| 00:01.01 | *** join/#asterisk kio (n=kio@ool-4577a5a5.dyn.optonline.net) |
| 00:01.04 | *** join/#asterisk eth0x (n=dcfletch@octane.lax.c2h2.net) |
| 00:02.52 | *** part/#asterisk Utah_Dave (n=boucha@0-1pool139-198.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 00:02.58 | kimosabe | is any one using the wrt54g for qos on thir lan for the sip devices |
| 00:03.26 | *** join/#asterisk edwin__ (n=edwin@252-131-222-203.rev.techex.net.au) |
| 00:05.20 | *** part/#asterisk eth0x (n=dcfletch@octane.lax.c2h2.net) |
| 00:05.25 | *** join/#asterisk eth0x (n=dcfletch@octane.lax.c2h2.net) |
| 00:05.36 | *** part/#asterisk eth0x (n=dcfletch@octane.lax.c2h2.net) |
| 00:05.55 | *** join/#asterisk sonic2wb (n=dcfletch@octane.lax.c2h2.net) |
| 00:06.01 | *** part/#asterisk sonic2wb (n=dcfletch@octane.lax.c2h2.net) |
| 00:06.59 | kink0 | Netrunz, ERROR[21165]: chan_zap.c:11273 setup_zap: Signalling must be specified before any channels are. |
| 00:07.02 | *** join/#asterisk eth0x (n=dcfletch@octane.lax.c2h2.net) |
| 00:07.08 | *** part/#asterisk eth0x (n=dcfletch@octane.lax.c2h2.net) |
| 00:07.29 | choongii | since I am calling to an ISDN card, I should get output on my screen if I tail -f /dev/thedevice and call the number, right? |
| 00:08.21 | robin_z | ummm |
| 00:08.36 | robin_z | using CAPI? |
| 00:09.36 | choongii | uh. no idea |
| 00:09.46 | choongii | oh wait, no, HFC |
| 00:10.01 | kink0 | Netrunz, after install bristuff, I am unable to load chan_zap and asterisk stops |
| 00:10.26 | robin_z | sory no clue on how to debug hfc .. whtever it is. |
| 00:10.28 | kink0 | even I set signalling=pri_net and so. |
| 00:11.04 | choongii | uh. I dont really know either. thanks anyway |
| 00:11.46 | robin_z | choongii: with CAPI in * you can get a lot from setting "capi debug" |
| 00:12.00 | robin_z | and turning verbosity up to 6 or more |
| 00:12.09 | robin_z | no clue about hfc |
| 00:12.22 | robin_z | oh, heres one ... |
| 00:12.28 | robin_z | sending faxes ... |
| 00:12.34 | Dr-Linux | i have changed sip port to port=5089 but when i do "sip show setting" it still shows sip port 5060 what could be happend? |
| 00:12.50 | robin_z | I have multiple ISDN channels on a Eicon card, used by * for incoming calls |
| 00:13.05 | dudes | Dr-Linux - perhaps the peer is using 5060 |
| 00:13.12 | robin_z | can I use, say, hylafax, to send stuff, using CAPI .. |
| 00:13.22 | robin_z | and not screw up * while doing it? |
| 00:13.32 | choongii | thanks robin_z |
| 00:13.35 | robin_z | or do I have to dedicate a channel fromthe group? |
| 00:13.49 | Dr-Linux | dudes: yeah you are right, |
| 00:13.53 | *** join/#asterisk hans (n=fugalh@dhcp25.cs.nmsu.edu) |
| 00:14.14 | *** join/#asterisk Wishbringer (n=ripper@gtcyber.ist.utl.pt) |
| 00:14.24 | Wishbringer | good evening all |
| 00:14.30 | Wishbringer | can I get some help please? :) |
| 00:15.22 | Wishbringer | I've got an TE110p and an asterisk config, and I am not able to dial out (over the line) from a SIP telephone! |
| 00:15.41 | Wishbringer | do u guys have any clues?? i've been over 14hours and I could use some help! :( |
| 00:16.16 | robin_z | Wishbringer: and the phone registers OK? |
| 00:16.19 | robin_z | ir gets auth |
| 00:17.01 | Wishbringer | robin_z: here's what I've been able to make it work: I can dial from a public telefone (regular number) to a SIP telephone (in the intranet) and dial between extensions |
| 00:17.39 | robin_z | right .. so the SIP phone is able to dial other extensions, but not outbound to the PSTN right? |
| 00:18.14 | Wishbringer | robin_z: yes , I can't dial out! I just wanted to make the SIP phone call back, through the line, to a regular phone! |
| 00:18.17 | Dr-Linux | can i assign any individaul port to my softphone user other than 5060 ? |
| 00:18.24 | cyburdine | anyone ever see this error? "Got 404 Not found on SIP register to service" |
| 00:18.45 | robin_z | Wishbringer: err, that last thing .. I cant make any clue from what you mean |
| 00:18.47 | cyburdine | just trying to connect my asterisk box up to my provider |
| 00:18.56 | robin_z | oh I CANT dial out .. OK |
| 00:18.58 | cyburdine | and am almost there... ( i think) |
| 00:19.30 | robin_z | Wishbringer: so ... what does * say when you try? in the console ... |
| 00:19.32 | Wishbringer | robin_z: that's it :) dialing in (from the telephone company) works fine. Dialing out, to the telephone company, doesn't work. |
| 00:19.35 | Wishbringer | ok |
| 00:19.37 | Wishbringer | gimme a sec |
| 00:19.43 | robin_z | pastebin ... |
| 00:19.58 | Wishbringer | (I am using BitchX :P) |
| 00:20.28 | cyburdine | I can connect to my provider via xlite |
| 00:20.31 | kink0 | is bristuff ok for Digium TE405 cards ??? Now, after installed bristuff, all crashes, unable to load chan_zap |
| 00:20.34 | Wishbringer | Jan 26 00:20:20 DEBUG[5841] chan_sip.c: Allocating new SIP dialog for 3315e3a3118690c5@192.168.70.17 - INVITE (With RTP) |
| 00:20.34 | Wishbringer | Jan 26 00:20:20 DEBUG[5841] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE |
| 00:20.34 | Wishbringer | Jan 26 00:20:20 DEBUG[5841] chan_sip.c: * SIP extension value: 1 for call 3315e3a3118690c5@192.168.70.17 |
| 00:20.34 | Wishbringer | Jan 26 00:20:20 DEBUG[5841] chan_sip.c: Setting NAT on RTP to 0 |
| 00:20.34 | Wishbringer | Jan 26 00:20:20 DEBUG[5841] chan_sip.c: = Found Their Call ID: 3315e3a3118690c5@192.168.70.17 Their Tag 0cba08ff84328267 Our tag: as56b6f050 |
| 00:20.36 | Wishbringer | Jan 26 00:20:20 DEBUG[5841] chan_sip.c: **** Received ACK (6) - Command in SIP ACK |
| 00:20.39 | Wishbringer | Jan 26 00:20:20 DEBUG[5841] chan_sip.c: Stopping retransmission on '3315e3a3118690c5@192.168.70.17' of Response 63067: Match Found |
| 00:20.40 | cyburdine | but can't get the asterisk box to onnect |
| 00:20.41 | Wishbringer | Jan 26 00:20:20 DEBUG[5841] chan_sip.c: = Found Their Call ID: 3315e3a3118690c5@192.168.70.17 Their Tag 0cba08ff84328267 Our tag: as56b6f050 |
| 00:20.44 | Wishbringer | Jan 26 00:20:20 DEBUG[5841] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE |
| 00:20.46 | Wishbringer | Jan 26 00:20:20 DEBUG[5841] chan_sip.c: Setting NAT on RTP to 0 |
| 00:20.48 | Wishbringer | Jan 26 00:20:20 DEBUG[5841] chan_sip.c: Checking SIP call limits for device ricardo.pascoal |
| 00:20.50 | Wishbringer | Jan 26 00:20:20 DEBUG[5841] chan_sip.c: Updating call counter for incoming call |
| 00:20.54 | Wishbringer | Jan 26 00:20:20 DEBUG[5841] chan_sip.c: = Found Their Call ID: 3315e3a3118690c5@192.168.70.17 Their Tag 0cba08ff84328267 Our tag: as56b6f050 |
| 00:20.56 | robin_z | stop it ! |
| 00:20.57 | Wishbringer | Jan 26 00:20:20 DEBUG[5841] chan_sip.c: **** Received ACK (6) - Command in SIP ACK |
| 00:20.59 | Wishbringer | Jan 26 00:20:20 DEBUG[5841] chan_sip.c: Stopping retransmission on '3315e3a3118690c5@192.168.70.17' of Response 63068: Match Found |
| 00:21.02 | Wishbringer | (I am sorry for those that got this!) |
| 00:21.02 | robin_z | arghhhh. |
| 00:21.03 | jsaunders | Heheh |
| 00:21.04 | Wishbringer | (this is the best I can do) |
| 00:21.06 | wunderkin | raaape |
| 00:21.10 | Wishbringer | sorry guys |
| 00:21.13 | Wishbringer | :( |
| 00:21.13 | jsaunders | My eyes! |
| 00:21.18 | *** join/#asterisk ke4qqq (n=chatzill@srv.fgp.com) |
| 00:21.26 | wunderkin | kink0, why are you using bristuff with a te405 |
| 00:22.05 | Wishbringer | robin_z: did u get the idea? :) |
| 00:22.06 | kink0 | wunderkin, because I have Cause: Unknown (100), class = Protocol Error (6) |
| 00:22.24 | De_Mon | I've got 1,Answer() n,MusicOnHold() n,Wait(10) n,Hangup and have been on hold over 1min |
| 00:22.34 | *** join/#asterisk c4r3x (n=carex@202.159.57.221) |
| 00:22.35 | kink0 | wunderkin, while I send calls from Asterisk (Digium te405) to 2N PRI E1 port |
| 00:22.47 | wunderkin | ? why arent you just using regular asterisk, zaptel, and libpri? |
| 00:23.08 | De_Mon | also getting big gaps in audio.. a behavior also seen in channelspy |
| 00:23.25 | robin_z | Wishbringer: I think you have debugging turned WAY high :) |
| 00:23.28 | kink0 | wunderkin, I was ussing just that, but I got that error and I was unable to send calls to my 2N gateway |
| 00:23.28 | cyburdine | I'll be someone's best friend if they'll help me get connected to my voip provider?!? ;) |
| 00:23.50 | De_Mon | cyburdine you arn't the kind of friend I want. Whos your provider? |
| 00:23.57 | brockj49464 | cyburdine: who is the voip? |
| 00:23.58 | cyburdine | hey... ;) |
| 00:24.01 | Wishbringer | robin_z: yes... that is a usual simptom of a desperate person! :> |
| 00:24.03 | cyburdine | fonixvoip.com |
| 00:24.10 | kink0 | wunderkin, I think I will need to reinstall again standard asterisk+zaptel+libpri and forget bristuff |
| 00:24.18 | cyburdine | I can connect via xlite |
| 00:24.46 | De_Mon | cyburdine are you trying to connect asterisk or xlite to your provider? |
| 00:24.58 | robin_z | Wishbringer: I think the real problem is lost somewhere in that mess. its probably a dialplan thing .. ratehr than a SIP issue |
| 00:25.00 | brockj49464 | cyburdine: what problems do you have in *? |
| 00:25.01 | cyburdine | I'm trying to connect asterisk to the provider |
| 00:25.19 | cyburdine | when * goes to register i get a 404 error |
| 00:25.32 | Wishbringer | robin_z: ok. now, then what should I do? |
| 00:25.42 | mishehu | anybody know why asterisk won't work with speex 1.1.x series? |
| 00:25.56 | cyburdine | Got 404 Not found on SIP register to service 8018675309@sd.fonixvoip.com, giving up |
| 00:26.05 | kink0 | wunderkin, do you think I will able to use PRI E1 without bristuff ? |
| 00:26.10 | justinu | mishehu: can you help me with my one way audio problem? :P |
| 00:26.40 | wunderkin | kink0, yeah... as long as its setup right |
| 00:26.42 | mishehu | justinu: yeah, all you need to do is use 2 threads instead of one... |
| 00:26.56 | Wishbringer | robin_z: can u show me some site with something like what I want? like a dial plan of some sort? I've been arround so many that I reaaly can't tell what each one is supposed to do. |
| 00:27.01 | *** join/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 00:27.15 | *** part/#asterisk jarg (n=jarg@200.56.225.61) |
| 00:27.34 | robin_z | http://www.dnsreport.com/tools/dnsreport.ch?domain=voipgate.com # idiots |
| 00:27.34 | kink0 | wunderkin, ahh ok , that sounds good, I was desesperate today with that error ( Cause: Unknown (100), class = Protocol Error (6) |
| 00:27.42 | brockj49464 | cyburdine: Do you manually edit * files or do you use something like AMP |
| 00:27.45 | robin_z | Wishbringer: like for dialing out? |
| 00:27.55 | cyburdine | manual edit... |
| 00:28.07 | Wishbringer | robin_z: yes, please!!!!! (that's what I need :D) |
| 00:28.15 | rpm | what does a _ mean infront of a telephone number in the dialplan? |
| 00:28.33 | brockj49464 | cyburdine: Can you pastebin your sip_additional if that is where you put the info |
| 00:28.36 | Wishbringer | robin_z: my conf's are a mess, I know it, but I do need some light at the end of the tunnel! |
| 00:28.42 | cyburdine | will do |
| 00:28.50 | robin_z | Wishbringer: just set the TRUNK variable to whatever your outgoing trunk is in * and it will work withthe standard extensions.conf |
| 00:29.13 | rob0 | Wishbringer: the light is a TRAIN! You are doomed!! |
| 00:29.18 | Wishbringer | robin_z: ok... now what is a TRUNK (in a nutshell plz!) |
| 00:29.43 | Wishbringer | rob0: either way, I am dead! so I might try and face it! :P |
| 00:30.19 | robin_z | Wishbringer: a trunk is an external line or group of lines |
| 00:30.59 | robin_z | exten => _0.,1,Dial(${TRUNK}/g1/${EXTEN}/B,30) |
| 00:31.13 | *** part/#asterisk Netrunz (n=FF@195.137.136.242) |
| 00:31.23 | robin_z | dial out on group 1 of the trunk for any extension begining 0 |
| 00:31.58 | Wishbringer | robin_z: you are going a bit too fast for me! |
| 00:32.15 | *** join/#asterisk hans (n=fugalh@128.123.232.3) |
| 00:32.51 | Wishbringer | robin_z: so {$TRUNK} can have a value like Zap/1? |
| 00:32.56 | robin_z | yes |
| 00:33.03 | robin_z | or just write that |
| 00:33.23 | fdask | g1 means group 1, right? |
| 00:33.36 | Wishbringer | robin_z: ok, so whats the B,30 Options goes for? |
| 00:33.55 | robin_z | oh, thats CAPI specific I guess |
| 00:34.02 | kink0 | well, I get again: < Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) |
| 00:34.02 | kink0 | < Ext: 1 Cause: Unknown (100), class = Protocol Error (6) |
| 00:34.03 | cyburdine | brockj49464: here's sip.conf and output from sip debug |
| 00:34.03 | robin_z | early B3 and 30 second timeout |
| 00:34.07 | cyburdine | http://pastebin.com/523228 |
| 00:34.34 | *** join/#asterisk yxa (n=diablo@58.185.90.110) |
| 00:34.51 | Wishbringer | robin_z: so this would seem right for you: exten => _0.,1,Dial(Zap/g1/${EXTEN}/B,30) |
| 00:35.23 | robin_z | drop the B, probably I would try .. |
| 00:35.43 | robin_z | exten => _0.,1,Dial(Zap/1/${EXTEN}/30) |
| 00:35.55 | robin_z | if I had only one channel, groups are then irrelevant |
| 00:36.17 | Wishbringer | I have a E1 |
| 00:36.18 | robin_z | but I know not what I type :) |
| 00:36.42 | brockj49464 | cyburdine: How about sd.fox for the host and fromdomain in out section? |
| 00:36.54 | Wishbringer | robin_z: and that's why I am asking a person that knows :D |
| 00:37.03 | yxa | guys my zap channels are not detecting hangups. they used to but they don't now... |
| 00:37.14 | robin_z | Wishbringer: really? who? |
| 00:37.31 | c4r3x | a |
| 00:37.49 | cyburdine | just updated it with working debug output from x-lite |
| 00:38.00 | Wishbringer | robin_z: you. I put it on extensions.conf under the [default] group. |
| 00:38.06 | kink0 | any idea about Cause : Unknown (100) Protocol Error (6) ? |
| 00:38.14 | Wishbringer | robin_z: and it still doesn't work :| |
| 00:38.24 | robin_z | Wishbringer: so, OK, with 30 channels a group is useful |
| 00:38.36 | robin_z | did you define the channels in the group in zap.conf |
| 00:38.39 | *** join/#asterisk Jonathan_ (n=chatzill@66-168-63-104.dhcp.mdsn.wi.charter.com) |
| 00:38.39 | cyburdine | brockj: are you saying I should make them both sd.fonix... ? |
| 00:38.44 | Wishbringer | erm |
| 00:38.48 | Wishbringer | *cof* |
| 00:38.49 | robin_z | eg group=1 inthe channels? |
| 00:38.56 | Wishbringer | er... |
| 00:39.10 | robin_z | this way you can make up different groups |
| 00:39.14 | harryvv | off topic, Are there alot of idiots in this world that send email with no topic or message in the body of the mail or is this some kind of script to verify my email exist? |
| 00:39.25 | robin_z | 1-10 in group 1, 11-15 group 2 |
| 00:39.38 | *** join/#asterisk carex (n=widiasto@202.159.57.221) |
| 00:39.39 | Wishbringer | zap.conf or zapata.conf? |
| 00:39.42 | robin_z | harryvv: your postulate is too specific. |
| 00:39.51 | harryvv | postulate? |
| 00:39.52 | robin_z | Wishbringer: yeah, probably. |
| 00:39.54 | Jonathan_ | After a recent install of *, SIP clients are unable to register. Turning on SIP Debuging shows that Asterisk is responding 404 Not found to the register request (if I'm reading it correctly) but the users are built in sip.conf. Does anyone have suggestions on how to fix this? |
| 00:40.15 | cyburdine | brockj: doesn't seem to make a difference |
| 00:40.24 | robin_z | harryvv: simply "Are there alot of idiots in this world" wold have been more than sufficient |
| 00:40.29 | robin_z | and yes, there are. |
| 00:40.52 | robin_z | there is no such thing as a stupid question |
| 00:41.01 | cyburdine | brockj: in the xlite debug info it seems to throw fonixvoip.com in for both to and from |
| 00:41.03 | robin_z | but there are plenty of inquisitive idiots ;))))) |
| 00:41.09 | Wishbringer | ok, i got |
| 00:41.14 | robin_z | kewl. |
| 00:41.29 | Wishbringer | group=1 |
| 00:41.29 | Wishbringer | signalling=pri_cpe |
| 00:41.29 | Wishbringer | context=pri1 |
| 00:41.30 | Wishbringer | channel => 1-15,17-31 |
| 00:41.40 | robin_z | Wishbringer: please remember, I know nothing :) |
| 00:41.46 | Wishbringer | under the [channels] |
| 00:41.48 | festr_ | Wishbringer: whats the problem? :) |
| 00:42.16 | harryvv | robin_z I definatly get my share of emails from people I DONT know that contains no message in the topic or the body of the email. |
| 00:42.16 | Wishbringer | festr_: hello. I can't dial out from a SIP phone to a regular telefone (through the line) though I can call in! |
| 00:42.26 | kink0 | Wishbringer, has you connected asterisk to E1 trunk ?? successfully ? |
| 00:42.29 | yxa | anyone help pls? my zap channels are not detecting hangups |
| 00:42.30 | robin_z | Wishbringer: and you did do a "estensions reload" after editing your files? |
| 00:42.36 | festr_ | Wishbringer: regular telephone through PRI? |
| 00:42.44 | robin_z | harryvv: no, they are not from people ... |
| 00:42.50 | Wishbringer | robin_z: I reloaded all asterisk |
| 00:43.07 | robin_z | harryvv: they are "probe" mails to see if your address accepts mail or just bounces ... |
| 00:43.24 | harryvv | robin_z I kinda thought that was the case. |
| 00:43.25 | Wishbringer | kink0: I can receive calls that I make from my mobile phone to the SIP phone! |
| 00:43.26 | *** join/#asterisk konfuzed (n=KonfuzeD@H135.C72.B0.tor.eicat.ca) |
| 00:43.33 | robin_z | harryvv: a common technique by spammers to detect possible working email addies |
| 00:43.45 | Wishbringer | festr_: I didn't understand ur question...:| |
| 00:44.06 | kink0 | Wishbringer, and can you call SIP phone -> asterisk -> E1 -> PSTN ? |
| 00:44.11 | harryvv | robin, I want to block all and allow only those into my inbox. firefox thunderbird does not offer this. what else. |
| 00:44.14 | robin_z | festr_: it looks like his dialplan is not routing SIP to pstn calls out over zap |
| 00:44.18 | festr_ | Wishbringer: i understand. in asterisk console, any message which could call whats wrong? |
| 00:44.24 | Wishbringer | kink0: no, I cannot. and that's the problem |
| 00:44.28 | *** join/#asterisk groogs (n=greg@d226-27-136.home.cgocable.net) |
| 00:44.35 | brockj49464 | cyburdine: Yes I think that they need to be the same. |
| 00:44.42 | Wishbringer | festr_: I can pastebin the errors I have... :> |
| 00:44.43 | *** join/#asterisk muzzz (n=chatzill@218.111.65.17) |
| 00:44.48 | festr_ | Wishbringer: do you see line Dial(SIP.... Zap/g1) |
| 00:44.49 | Wishbringer | (learned this from previous experiences!) |
| 00:44.49 | kink0 | Wishbringer, where have you the problem ? SIP -> PRI or PRI -> PSTN ? |
| 00:44.49 | festr_ | ? |
| 00:44.54 | festr_ | Wishbringer: paste it |
| 00:45.10 | robin_z | festr_: NOOOOOOOOO ... |
| 00:45.18 | festr_ | Wishbringer: to pastebian |
| 00:45.29 | brockj49464 | cyburdine: Sorry missed the other msg |
| 00:45.34 | festr_ | robin_z: not here off course :) |
| 00:46.42 | *** join/#asterisk tmbinc (i=XXX@dslb-082-083-085-056.pools.arcor-ip.net) |
| 00:46.50 | Wishbringer | here it goes: http://pastebin.com/523248 |
| 00:46.53 | *** join/#asterisk mistral (i=mistral@jstevenson.plus.com) |
| 00:46.55 | *** join/#asterisk Lord_Drachenblut (n=Lord@12-210-115-191.client.insightBB.com) |
| 00:47.37 | festr_ | Wishbringer: that's all? |
| 00:47.46 | robin_z | eech ... debugging still too high |
| 00:47.52 | Wishbringer | er yes |
| 00:47.54 | Wishbringer | :| |
| 00:48.04 | robin_z | festr_: I suspect we didnt see the bit we need in all that mess |
| 00:48.24 | robin_z | Wishbringer: try asterisk -cvvv |
| 00:48.25 | Wishbringer | robin_z: no, that's all that came out. |
| 00:48.31 | robin_z | not vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv |
| 00:48.49 | Wishbringer | kk |
| 00:49.07 | festr_ | Wishbringer: make sure, that /etc/asterisk/logger.conf is console=verbose,warning,error,notice,debug |
| 00:49.17 | festr_ | s/is/contain |
| 00:49.19 | festr_ | s/is/contain/ |
| 00:49.36 | funxion | b4 I try this can anyone tell me if Dial(Zap/g6/) will just go off hook on a chnnel assigned to that group and conect the call without spilling digits? |
| 00:49.53 | tmbinc | what's wrong when i get a SIP-client (fritzbox fon) gets "100 Trying", but no connection? we have 4 fritzboxes here through a VPN, and 3 of them work. only one doesn't work. on asterisk side i'm only getting a "Jan 26 01:44:02 WARNING[17127]: chan_sip.c:6393 register_verify: Failed to parse contact info", on the fritzbox side i get a ">>>udp Request: REGISTER sip:192.168.5.183", "<<<udp Status: 100 Trying" |
| 00:50.01 | robin_z | so .. where was I .. |
| 00:50.03 | Wishbringer | http://pastebin.com/523251 |
| 00:50.04 | robin_z | oh yes |
| 00:50.08 | Wishbringer | hwere ir is |
| 00:50.10 | Wishbringer | it* |
| 00:50.18 | robin_z | FAX .. using ReceiveFax is great .. |
| 00:50.28 | De_Mon | also getting big gaps in audio.. a behavior also seen in channelspy. Any ideas? cpu load is low, latency is low |
| 00:50.32 | Wishbringer | and I get a 404 error on the SIP Phone! :) |
| 00:50.35 | robin_z | lovely .. gets my faxes and emails them to the secreatry |
| 00:50.43 | festr_ | Wishbringer: paste here log, when you call from mobile to sip |
| 00:50.46 | De_Mon | normal calls work fine... |
| 00:50.47 | robin_z | now .. sending faxes .. |
| 00:50.50 | Wishbringer | ok |
| 00:51.07 | festr_ | robin_z: but iaxmodem with hylafax is more greater :) |
| 00:51.10 | robin_z | fax printer of some sort? |
| 00:51.21 | Wishbringer | damn |
| 00:51.28 | Wishbringer | it's not working any more... |
| 00:51.30 | robin_z | hmm |
| 00:51.30 | Wishbringer | wait a sec |
| 00:51.42 | exstatica | anyone know about the zaptel drivers? |
| 00:51.47 | festr_ | Wishbringer: pri show span 1 |
| 00:51.55 | festr_ | exstatica: it depends |
| 00:51.57 | exstatica | i keep getting an error |
| 00:51.57 | exstatica | Freshmaker failed register test |
| 00:51.57 | exstatica | wctdm: probe of 0000:00:04.0 failed with error -5 |
| 00:52.12 | *** join/#asterisk buzzyd (n=buzzyd@82-45-247-173.cable.ubr01.enfi.blueyonder.co.uk) |
| 00:52.19 | festr_ | exstatica: contact support@digium.com |
| 00:52.26 | Wishbringer | http://pastebin.com/523255 and I can no longer connect |
| 00:52.34 | Wishbringer | festr_: er ... what is that? :| |
| 00:52.40 | exstatica | so they can tell me to use a different motherboard? |
| 00:53.12 | festr_ | exstatica: nonsense |
| 00:53.34 | festr_ | Wishbringer: paste CLI> show dialplan pri1 |
| 00:53.45 | *** join/#asterisk hans (n=fugalh@falcon.fugal.net) |
| 00:53.53 | *** join/#asterisk pifiu-laptop (n=someone@c-65-34-166-146.hsd1.fl.comcast.net) |
| 00:54.23 | festr_ | Wishbringer: you should see 501 extensions, if not, extensions.conf is broken or pri1 does not contain 501 |
| 00:55.03 | festr_ | Wishbringer: also check in your sip.conf [youraccount] ... context = some_context and if this context exist |
| 00:55.40 | kpettit | anybody notice any new probelms with 1.2.3? I'm getting alittle gun shy now with new versions |
| 00:56.05 | Jonathan_ | I am unable to get any SIP clients to register to my recent * install. I have disabled iptables/ipchains to ensure they are not interfearing. The CLI is showing all devices unregistered. Any ideas on how I can begin to resolving this? |
| 00:56.21 | Wishbringer | festr_: ok, I got it to work again: |
| 00:56.27 | Wishbringer | http://pastebin.com/523260 |
| 00:56.30 | Wishbringer | (long lg!) |
| 00:56.33 | funxion | does anyone if I can dial to a ZAP channel without outpulsing digits? |
| 00:56.49 | Wishbringer | (extension 502 this time) |
| 00:57.33 | festr_ | Wishbringer: paste your sip.conf |
| 00:57.40 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:57.58 | Aughey | funxion: you can, but asterisk chokes up a warning when you do. |
| 00:58.01 | festr_ | Wishbringer: and extensions.conf |
| 00:58.08 | funxion | hmm |
| 00:58.20 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 00:58.22 | funxion | I guess thats not soo bad |
| 00:58.22 | Wishbringer | http://pastebin.com/523263 |
| 00:58.28 | rpm | my home phone is broken, i ugraded to 1.2.2-netsec and whenever someone calls now my asterisk system freezes. |
| 00:58.29 | Wishbringer | (and don't mind the passwords :P) |
| 00:58.35 | rpm | <PROTECTED> |
| 00:58.36 | Dr-Linux | anyone used anyother port for sip other than 5060 ? |
| 00:58.39 | rpm | the load went up to 9.9 |
| 00:58.49 | funxion | Aughey is there any downside to it other than a warning? |
| 00:58.50 | festr_ | Wishbringer: :) |
| 00:58.58 | rpm | i think i have found a bug |
| 00:59.06 | Wishbringer | they will be changed, of course :) |
| 00:59.10 | funxion | Dr-Linux I have port 80 |
| 00:59.14 | festr_ | Wishbringer: and paste your extensions.conf |
| 00:59.15 | Wishbringer | when the system is full and funcional :D |
| 00:59.24 | Wishbringer | festr_: u sure u want me to do that? |
| 00:59.35 | Dr-Linux | great |
| 00:59.43 | Dr-Linux | funxion: can i /msg you ? |
| 00:59.46 | Aughey | funxion: not that I know of. I used it to just get a raw dialtone |
| 00:59.48 | festr_ | Wishbringer: yes |
| 00:59.50 | funxion | sure |
| 00:59.52 | Wishbringer | festr_: why don't we start with the zapata.conf?? |
| 00:59.57 | Dr-Linux | ty |
| 01:00.05 | Wishbringer | festr_: ok... u know what u'r talking about :) |
| 01:00.09 | festr_ | Wishbringer: cause you cannot dial sip->zapata |
| 01:00.31 | festr_ | Wishbringer: paste extensins.conf and lets see :) |
| 01:00.36 | Wishbringer | festr_: yes, Sir! :) |
| 01:01.01 | festr_ | Wishbringer: hurry up, my battery is almost gone (10minutes) |
| 01:01.02 | FuriousGeorge | can anyone tell me why asterisk sometimes just makes up the cid when its unknown? i got one server where its always "maria", i got another where it always says "asterisk" |
| 01:01.04 | festr_ | ;] |
| 01:01.23 | De_Mon | I called musiconhold and the dialplan wolnt move to the next command!(wait(10) then hangup) heeelp |
| 01:01.47 | festr_ | De_Mon: musiconhold will pplay music forever |
| 01:01.59 | festr_ | De_Mon: try waitmusiconhold or something like that |
| 01:01.59 | Wishbringer | festr_: http://pastebin.com/523268 |
| 01:02.07 | festr_ | De_Mon: (show applications) |
| 01:02.16 | Wishbringer | (oh please!!!!) |
| 01:02.30 | harryvv | FuriousGeorge thats a odd one. |
| 01:02.40 | festr_ | Wishbringer: so there is the problem |
| 01:02.46 | festr_ | Wishbringer: your sip is in context tutorial |
| 01:02.52 | Zodiacal | anyone know if its normal for a ring group to take longer to ring than dialing into a direct ext.? from outside lines.. |
| 01:02.59 | festr_ | Wishbringer: and there is only 501 exten |
| 01:03.01 | Wishbringer | festr_: I do need to clean up the stuff |
| 01:03.10 | rpm | anyone know if g729 works or not with 1.2.2-netsec? |
| 01:03.20 | De_Mon | festr_ I want musiconhold instead of rininging.. waitmusiconhold doesn't appear do do that either |
| 01:03.26 | FuriousGeorge | harryvv: depends on your perspective, to me, at this point, it would be weird to see an asterisk server display "Unknown" |
| 01:03.27 | srt | -cl |
| 01:03.33 | *** join/#asterisk msw_ (n=msw@rdu-nat.rpath.com) |
| 01:03.34 | FuriousGeorge | ive never seen one do it |
| 01:03.47 | festr_ | Wishbringer: so you have to change context of your sip.conf accounts to some context, which should look like this: exten => _X.,1,Dial(Zap/g1) |
| 01:03.49 | Wishbringer | festr_: yes... and?? only one 501 for one phone....I don't understand! |
| 01:03.56 | *** join/#asterisk ke4qqq (n=chatzill@srv.fgp.com) |
| 01:04.20 | Wishbringer | festr_: ok, I am messed up! please take it slow! |
| 01:04.29 | festr_ | Wishbringer: context in sip.conf mean, that calls from sip is placed to that context (yours [tutorial]) and there is only 501 |
| 01:04.41 | *** join/#asterisk coolhp (n=Toshiba@modemcable189.136-81-70.mc.videotron.ca) |
| 01:04.48 | coolhp | Good evening all 1 |
| 01:05.03 | Wishbringer | festr_: so I must put _X.,1,Dial(Zap/g1) in the toturial context? |
| 01:05.06 | festr_ | De_Mon: use dial(.....,m) m option (CLI> show application dial) |
| 01:05.11 | festr_ | Wishbringer: yes |
| 01:05.40 | festr_ | Wishbringer: exactly |
| 01:05.51 | coolhp | I was wondering if anyone has been experiencing the following problem lately : I dont know when it started but when using MeetMe, the usual beep sound when someone enters or leave the conference are distorted. They sould aweful... Any ideas ? |
| 01:06.06 | Wishbringer | festr_: now whats the difference between Zap/g1 and Zap/1? |
| 01:06.30 | festr_ | Wishbringer: for security i recomend you that you do setcallerid in sip.conf or use it in dialplan before go to Zap/g1 (exten _X.,1,setcallerid...) |
| 01:06.42 | De_Mon | festr_ just found it. I feel silly now |
| 01:06.54 | festr_ | Wishbringer: what about reading documentation? |
| 01:07.14 | Wishbringer | festr_: hum.. can u please be more precise? and about the documentation, I've been around it for the las 10 hours :/ |
| 01:07.32 | festr_ | Wishbringer: go sleep and start tomorrow |
| 01:07.43 | festr_ | Wishbringer: getting into asterisk is not about day :") |
| 01:07.52 | Wishbringer | festr_: and I don't my reading it. This is my desperate move! Oh I can't go sleep without a call geting out of this LAN :D |
| 01:08.12 | festr_ | Wishbringer: Zap/g1 will use first ascendant channel |
| 01:08.22 | festr_ | Wishbringer: Zap/1 will always use first channel |
| 01:08.26 | Wishbringer | nice :) |
| 01:08.34 | festr_ | Wishbringer: Zap/g1 first "FREE" channel |
| 01:08.42 | Wishbringer | that's nice:) why on hell don't YOU make some decent documentation? :) |
| 01:08.54 | *** join/#asterisk bawks (n=0x3e44d@71-213-20-208.slkc.qwest.net) |
| 01:08.58 | festr_ | Wishbringer: there is a lot of documentation |
| 01:09.00 | Wishbringer | festr_: and another thing, I put _0X.,1,Dial(Zap/g1) |
| 01:09.01 | festr_ | Wishbringer: and howtos |
| 01:09.17 | Wishbringer | festr_: for every number starting with 0, correct? |
| 01:09.18 | *** join/#asterisk MrBelvedr (n=tt@ip70-187-237-193.dc.dc.cox.net) |
| 01:09.24 | festr_ | Wishbringer: try to find some on voip-info.org. there is almost everything about asterisk |
| 01:09.33 | Wishbringer | festr_: quantity is different from quality, as you might know :) |
| 01:09.42 | bawks | Does anyone know how to listen to voicemail with Asterisk@home? |
| 01:09.47 | FuriousGeorge | am i the only one whos "unknown" caller ids on incoming calls show up as one of your internal extensions? |
| 01:09.50 | festr_ | Wishbringer: uhmmm. _X.,1,Dial(Zap/g1/${EXTEN}) is the right one |
| 01:10.03 | festr_ | Wishbringer: dont fuck me man :) |
| 01:10.26 | festr_ | Wishbringer: voip-info.org is very good resource |
| 01:10.35 | Wishbringer | festr_: oh no, don't worry about that :D I don't fuck on first dates :D |
| 01:10.37 | festr_ | Wishbringer: my batteries in my NTB is gone |
| 01:10.43 | festr_ | Wishbringer: ok :) |
| 01:11.01 | festr_ | Wishbringer: have a nice night i have to shut down ntb |
| 01:11.05 | *** join/#asterisk Iam8up|lappy (n=dontemai@user-0cdv2kk.cable.mindspring.com) |
| 01:11.10 | Wishbringer | festr_: THANKS |
| 01:11.20 | Wishbringer | festr_: a lot :) (I hope this works!) |
| 01:11.24 | FuriousGeorge | let me ask this way: what does asterisk do when no cid info is sent? |
| 01:11.45 | Iam8up|lappy | erm..i'm trying to have asterisk run as non-root |
| 01:12.04 | FuriousGeorge | what does it do in terms of choosing what to display as a cid |
| 01:12.06 | festr_ | Wishbringer: try to search on voip-info.org :) bye |
| 01:12.12 | *** join/#asterisk DynaGuy (n=dynaguy@S01060011954e8668.vc.shawcable.net) |
| 01:12.17 | Iam8up|lappy | is this how to do it? (i got this from the amp INSTALL doc) |
| 01:12.19 | Wishbringer | festr_: oh god.... lemme have your children, IT WORKS!!! |
| 01:12.19 | Iam8up|lappy | <PROTECTED> |
| 01:12.19 | Iam8up|lappy | <PROTECTED> |
| 01:12.19 | Iam8up|lappy | <PROTECTED> |
| 01:12.48 | Wishbringer | festr_: I do wish I knew what that line does! |
| 01:12.55 | Wishbringer | festr_: never the less |
| 01:12.58 | Wishbringer | it WORKS! :D |
| 01:13.00 | Wishbringer | wow |
| 01:13.03 | pifiu-laptop | so what does * do when no cid info is available? |
| 01:13.29 | Iam8up|lappy | pifiu-laptop - 'unknown caller' |
| 01:13.38 | FuriousGeorge | pifiu-laptop: if you ask me it picks an internal extension and it picks that cid |
| 01:13.49 | *** join/#asterisk Q-At-Home (n=Queue@S0106000c41bb87af.ed.shawcable.net) |
| 01:14.46 | FuriousGeorge | Iam8up|lappy: can you think of any reason mine would pick, say, extension "Maria" 104@10.0.0.10 |
| 01:15.25 | Iam8up|lappy | FuriousGeorge - uhm..information in the configuration? |
| 01:15.25 | Q-At-Home | hiya all long time no see |
| 01:15.35 | Q-At-Home | where have ${CALLERIDNUM} ${CALLERIDNAME} gone? (if anywhere) |
| 01:15.45 | Wishbringer | ok |
| 01:15.46 | *** join/#asterisk brookshire[home] (n=matt@pcp01541028pcs.huntsv01.al.comcast.net) |
| 01:15.48 | Wishbringer | I am off to sleep |
| 01:15.49 | Wishbringer | :D |
| 01:15.58 | Wishbringer | tks all |
| 01:15.59 | Wishbringer | :D |
| 01:16.01 | Wishbringer | [[[]]]s |
| 01:17.09 | *** join/#asterisk litecode (n=andrewb@12-217-30-205.client.mchsi.com) |
| 01:17.31 | litecode | i'm staring at the documentation, and i can't figure out the difference between "RealTime" and "RealTime Static" can somebody enlighten me? |
| 01:18.25 | *** join/#asterisk vmwarez (n=jjones@64.212.45.45) |
| 01:19.12 | *** join/#asterisk pablasso (n=pablasso@dsl-201-133-179-254.prod-infinitum.com.mx) |
| 01:20.29 | drumkilla | litecode: 'Realtime' refers to different parts of Asterisk that have been set up with the ability to pull configuration out of a database as it is needed |
| 01:20.45 | kink0 | I still getting Ext: 1 Cause: Unknown (100), class = Protocol Error (6) |
| 01:20.48 | litecode | drumkilla, so RealTime Static means pulling "whole" configuration files out? |
| 01:20.49 | kink0 | any idea ? |
| 01:20.49 | drumkilla | litecode: 'Realtime Static' refers to the ability to map *any* asterisk configuration file to a database |
| 01:20.51 | bawks | What would cause two identical phones connected to the same asterisk box not to send audio to each other? They dial each other just fine. |
| 01:21.10 | litecode | bingo. that's what i watn to do, i don't want an extensions.conf, i want it completely in the database. |
| 01:21.11 | drumkilla | litecode: but still requires a reload, and the data is put in the db in the same kind of way that it is put into the .conf file |
| 01:21.40 | FuriousGeorge | Iam8up|lappy: ironically, i realized at least in one server's case it was pulling the cid i named from the previous fxs' configs |
| 01:21.48 | litecode | drumkilla, I can deal with the reload. |
| 01:21.52 | litecode | drumkilla, thanks |
| 01:22.01 | drumkilla | you are welcome |
| 01:22.24 | Iam8up|lappy | FuriousGeorge - how is that ironic..? |
| 01:22.39 | drumkilla | any conf file can be put in a db using 'realtime static', but only certain parts of asterisk support 'realtime' config ... |
| 01:22.46 | drumkilla | it's ... confusing |
| 01:23.01 | drumkilla | and a terrible name for the whole thing, heh |
| 01:23.05 | FuriousGeorge | Iam8up|lappy: becuase i asked how it can be happening and you answered "your configs" and you were right, the irony comes in because... forget it |
| 01:23.21 | Iam8up|lappy | rofl =P |
| 01:23.27 | Zodiacal | is it normal for asterisk to wait until the second ring before acctualy ringing on my sip phone? with a pots line and a tdm400p and fxo module... |
| 01:23.48 | drumkilla | Zodiacal: yes, for callerid ... |
| 01:24.02 | Zodiacal | drumkilla i don't have it and i disabled it in my zapa config |
| 01:24.14 | kink0 | any one have connected a digium TExxx E1 to the PSTN or gateway ? or so ? |
| 01:24.34 | Zodiacal | drumkilla any options you know of off hand that could speed this up? |
| 01:24.39 | coolhp | Has anyone ever had a problem with the enter.h and leave.h files in app_meetme ? For the past couple of weeks and even having uped to the latest source, these tones sound awefully distorted when entering and leaving the conference. |
| 01:26.31 | rpm | [2006-01-25 18:23:11] WARNING[13881]: chan_zap.c:10829 setup_zap: Ignoring signalling |
| 01:26.35 | bawks | What would cause two identical phones connected to the same asterisk box not to send audio to each other? They dial each other just fine. Echo test works fine. |
| 01:26.43 | rpm | does asterisk not like my fxo_ks signalling no more ? :) |
| 01:28.52 | brookshire[home] | bawks: are they sip? |
| 01:28.54 | *** join/#asterisk _blop (i=blop@openbeer.be) |
| 01:29.04 | bawks | brookshire[home]: yup |
| 01:29.17 | *** part/#asterisk vmwarez (n=jjones@64.212.45.45) |
| 01:29.23 | brookshire[home] | http://www.asterisk.org/node/55 |
| 01:29.59 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 01:30.05 | bawks | eeek ok |
| 01:30.12 | Ariel_ | hello folks |
| 01:30.19 | brookshire[home] | hi |
| 01:30.34 | *** part/#asterisk DynaGuy (n=dynaguy@S01060011954e8668.vc.shawcable.net) |
| 01:30.53 | brookshire[home] | bawks: 1.2.3 was released to fix this issue |
| 01:31.35 | bawks | ugh... and asterisk@home uses 1.2.2 |
| 01:32.02 | *** join/#asterisk webman (n=adamg@52.87.233.220.exetel.com.au) |
| 01:32.05 | yxa | anyone help pls? my zap channels are not detecting hangups |
| 01:32.11 | tehdely | ass turd dicks |
| 01:32.51 | FuriousGeorge | you guys think that relaxeddtmf will fix my non-working #-transfers |
| 01:32.58 | FuriousGeorge | on zap channels |
| 01:33.30 | Ariel_ | bawks, so upgrade the asterisk |
| 01:34.11 | Ariel_ | looks like nufone got hit hard over this issue. They were down most of today. |
| 01:34.14 | *** join/#asterisk Techie-Micheal (n=Techie-M@phpbb/support/techie-micheal) |
| 01:34.35 | Techie-Micheal | pablasso: Something I can help you with? |
| 01:34.36 | FuriousGeorge | yxa: you using kew start signalling? |
| 01:34.44 | yxa | FuriousGeorge yeah |
| 01:34.49 | *** join/#asterisk Grok_ (n=blah@c-71-196-75-163.hsd1.fl.comcast.net) |
| 01:34.55 | FuriousGeorge | Ariel_: they sent me an email, said it was "multiple harware failures" |
| 01:35.21 | FuriousGeorge | yxa: then im not sure |
| 01:35.32 | Ariel_ | FuriousGeorge, yes all there servers needed to be updated |
| 01:35.49 | yxa | FuriousGeorge i called my incumbent regarding disconnect supervision. they were totally clueless |
| 01:35.51 | Ariel_ | yxa, what pstn you have? |
| 01:36.01 | yxa | i'm in asia. Singapore. |
| 01:36.08 | Ariel_ | OK hummm |
| 01:36.35 | justinu | cool |
| 01:36.36 | Ariel_ | I think if you find coppice when he is around he might now some settings for that area. |
| 01:36.47 | justinu | yeah, coppice is a good resource |
| 01:36.55 | *** part/#asterisk Techie-Micheal (n=Techie-M@phpbb/support/techie-micheal) |
| 01:36.58 | Ariel_ | if it's an old system they might use ground start |
| 01:37.17 | Q-At-Home | out of curiosity, what was 1.2.3 set to fix, it was running forever until this morning was this some kind of date bug? |
| 01:37.24 | yxa | Ariel_ i tried. didn't work |
| 01:37.36 | Ariel_ | Q-At-Home, sound |
| 01:37.53 | Q-At-Home | ok, but why did it work yesterday, with no changes break overnight? |
| 01:38.01 | Err | Q-At-Home: yes, it was related to the date - a date value was signed, and rolled over to negative |
| 01:38.05 | Q-At-Home | ahhhh |
| 01:38.06 | Q-At-Home | thanks |
| 01:38.18 | De_Mon | I've got several sip phones configured as [phone1] [phone2] should I have to add phone1,Dial(SIP/phone1) to extensions.conf for them to be callable? |
| 01:39.10 | Q-At-Home | heh cool |
| 01:39.10 | Ariel_ | what extension do you want phone1 to have like exten => 100,1,Dial(Sip/phone1,20) |
| 01:39.16 | Q-At-Home | thanks Ariel_ |
| 01:39.52 | Dr-Linux | if i change sip port to anything else in sip.conf should i also need to change it in rtp.conf ? |
| 01:39.54 | brockj49464 | So is aah2.3 bad due to the date bug in * 1.2.2 |
| 01:40.17 | Ariel_ | Dr-Linux, no |
| 01:40.35 | Ariel_ | port 5060 is for registration only rtp uses different ports |
| 01:41.05 | shmaltz | is EM64T supported in 2.4 kernel? |
| 01:41.15 | Q-At-Home | I'm trying to find a cheap wifi sip/iax phone :) |
| 01:41.35 | Ariel_ | Q-At-Home, it's just sip and it's working after the firmware upgrade to 3.8 fine. |
| 01:41.42 | fdask | anyone here use an openline4 card? |
| 01:41.55 | bawks | brockj49464: it's bad for me |
| 01:42.11 | Q-At-Home | hrmm |
| 01:42.29 | FuriousGeorge | anyone know if #-transfers work on sip? |
| 01:42.33 | Q-At-Home | I should pick one up for "cool" factor |
| 01:42.38 | FuriousGeorge | or is it just for zap |
| 01:42.44 | brockj49464 | aah2.3 released 1/24/2006. Does not sound (haha) good for aah2.3 then Bawks. |
| 01:42.58 | Ariel_ | wow yesterday there was a release of 2.3 for aah so I guess it's needing the patch |
| 01:43.18 | bawks | brockj49464: I'm using that version now, and it's borked. |
| 01:43.36 | Q-At-Home | FuriousGeorge: yes they do |
| 01:43.37 | Ariel_ | FuriousGeorge, yes if you use the Tt at the dial rule |
| 01:44.01 | brockj49464 | bawks: Is it just ext->pstn or ext->ext also |
| 01:44.08 | webman | ChangeLog for 1.2.3 says "channel.c: Issue #6439" but bugs.digium.com says "Issue 6439 not found." anyone know the correct bug number ?? |
| 01:44.35 | FuriousGeorge | so relaxdtmf wont solve this mystery |
| 01:44.35 | Grubs | http://bugs.digium.com/view.php?id=6349 |
| 01:44.51 | Ariel_ | FuriousGeorge, not really |
| 01:45.43 | bawks | brockj49464: only tested sip->sip |
| 01:45.46 | FuriousGeorge | Ariel_: its exten => s,3,dial(zap/g1,,t,T) |
| 01:45.49 | FuriousGeorge | right? |
| 01:45.52 | harryvv | Ariel_ :) Talked to dale slagar about his enclosed atx boxes and he said thay can accomidate up to 7 pci slots. |
| 01:46.09 | *** join/#asterisk j_vianna (n=joaovian@207.190.206.161) |
| 01:46.17 | harryvv | Sorry It is Dale Sliger |
| 01:46.18 | *** join/#asterisk katakefalos (i=katakefa@194.214.77.65.in-addr.arpa.ethernext.com) |
| 01:46.37 | j_vianna | Any more pathes for asterisk today ? |
| 01:46.40 | Ariel_ | harryvv, nice |
| 01:47.25 | fdask | i'm looking to pick up a new fxo card for an asterisk setup |
| 01:47.30 | fdask | looking at the Digium TDM04B Quad FXO |
| 01:47.47 | fdask | having 0 luck with these voicetronix cards |
| 01:47.53 | fdask | so i figure digium might be better |
| 01:47.56 | fdask | anyone used this card? |
| 01:48.02 | webman | anyone know if bug 6349 affected any older versions of asterisk (ie, prior to 1.2.2) ?? |
| 01:49.19 | FuriousGeorge | solved that mystery, if i specify a timeout i dont have to escape the comma |
| 01:49.30 | Aughey | fdask: I was referred to Sangoma cards yesterday. Bought $810 worth of them |
| 01:49.38 | fdask | Sangoma? |
| 01:49.42 | *** join/#asterisk bkw__ (n=brian@70.103.248.130) |
| 01:49.44 | Q-At-Home | Ariel_: if you dont mind my asking, what did you get your F1000 for |
| 01:49.49 | fdask | i'm only after fxo port cards |
| 01:49.55 | fdask | not up with t1 or anything yet |
| 01:50.05 | fdask | just looking for something relatively painless to setup |
| 01:50.06 | Aughey | fdask: http://www.sangoma.com/datasheets/p_a200-specs |
| 01:50.08 | j_vianna | fdask: I have a friend in Brazil using voicetronix card successfuly... |
| 01:50.25 | fdask | j_vianna: i've got a pair of openline4 cards here, having a hell of a time bridging calls |
| 01:50.42 | fdask | like using a dial() in my dialplan, things just bottom out |
| 01:51.26 | Q-At-Home | I've got a TDM04B and it works fine. |
| 01:51.31 | Jonathan_ | Does anyone know why I would be able to receive volume (such as from * vm) but not transmit sound on a SIP phone? |
| 01:51.45 | katakefalos | hello! to dial a number 123456 via a sip trunk you do exten => s,2,Dial(sip/123456@trunkname,60,r) what do u do when you want to dial it via a specific zap channel lets say zap/1? |
| 01:51.47 | Q-At-Home | Jonathan_: see topic |
| 01:52.49 | Q-At-Home | sounds like the bug |
| 01:53.15 | brookshire[home] | Jonathan_: 1.2.2 ? |
| 01:53.23 | Jonathan_ | Ok, I'll update to 1.2.3 quick and see if that resolves. |
| 01:53.24 | fdask | Q-At-Home: can you bridge ports together? |
| 01:53.46 | *** part/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca) |
| 01:53.47 | Q-At-Home | fdask: you mean like take one in, then send one out? |
| 01:53.49 | fdask | ya |
| 01:53.59 | Q-At-Home | I've got 2 boards, 4 ports each, one fxo one fxs |
| 01:54.00 | fdask | i call in, dialplan phones out, connects the two so they can talk |
| 01:54.01 | Q-At-Home | so yes |
| 01:54.03 | [av]bani | fdask: skip fxo entirely, go ITSP :) |
| 01:54.08 | fdask | itsp? |
| 01:54.15 | [av]bani | internet telephony service provider |
| 01:54.31 | fdask | we've got a multitech gateway here too, was thinking of just running my pbx into it, and patching it into asterisk with sip |
| 01:54.36 | [av]bani | avoid POTS if you can, its a PITA |
| 01:54.38 | Q-At-Home | fdask: yes that works fine |
| 01:54.45 | katakefalos | anyon knows whats the script to dial via zap ? |
| 01:54.50 | *** join/#asterisk jebba (n=jebba@ip-216-17-203-198.rev.frii.com) |
| 01:54.53 | fdask | Q-At-Home: does it use the zap channel? |
| 01:54.57 | Q-At-Home | yes |
| 01:55.06 | Q-At-Home | 2 of them |
| 01:55.14 | jebba | FYI, i upgraded to 1.2.2 a couple days ago and got lots of "I can't hear you" problems. I downgraded and the problems are gone. Anyone else seeing thign? |
| 01:55.14 | fdask | levels ok? |
| 01:55.19 | Q-At-Home | sounds fine |
| 01:55.29 | brookshire[home] | jebba: see topic |
| 01:55.30 | fdask | hrm |
| 01:55.36 | jebba | heh |
| 01:55.38 | jebba | ;) |
| 01:56.33 | [av]bani | fdask: spa3000 prolly sux the least for cheapy FXO |
| 01:56.42 | *** join/#asterisk ptiggerdine_ (n=ptiggerd@203-206-59-72.dyn.iinet.net.au) |
| 01:56.48 | [av]bani | fiddling with pci cards is a real pain |
| 01:56.50 | *** part/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 01:57.09 | sivana | jebba: go to 1.2.3 |
| 01:57.26 | rpm | i really need some help with my asterisk server, i can recieve and make phone calls.. but no audio is passed through |
| 01:57.34 | *** join/#asterisk troy (n=central@CPE00907f17e478-CM014300013422.cpe.net.cable.rogers.com) |
| 01:57.35 | fdask | spa3000? |
| 01:57.40 | fdask | better than a Digium TDM04B? |
| 01:57.41 | [av]bani | rpm: upgrade to 1.2.3 |
| 01:57.43 | Lord_Drachenblut | anyone have any suggestions for a wifi sip phone |
| 01:57.47 | Lord_Drachenblut | or wifi iax phone |
| 01:57.55 | rpm | [av]bani, i did upgrade to 1.2.3 and zaptel 1.2.2 |
| 01:57.57 | [av]bani | fdask: from what i can tell, yes |
| 01:58.01 | fdask | hrm |
| 01:58.04 | troyb1 | hey slePP |
| 01:58.07 | [av]bani | only marginally though |
| 01:58.07 | fdask | who makes that spa3000? |
| 01:58.15 | ptiggerdine_ | linksys/cisco |
| 01:58.29 | ptiggerdine_ | well l/c bought out sipura |
| 01:58.48 | [av]bani | http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-34702223616.htm |
| 01:59.00 | katakefalos | Lord_Drachenblut> i have tried zyxel and UT i dont like them |
| 02:00.06 | Lord_Drachenblut | katakefalos, so any suggestions of one that is decent |
| 02:00.28 | katakefalos | did not found one yet |
| 02:00.44 | robin_z | meep? |
| 02:00.46 | katakefalos | they are all like alpha stage |
| 02:01.19 | robin_z | Lord_Drachenblut: i have a Zyxel wifi phone too. it sucks. |
| 02:01.47 | *** join/#asterisk Chuji (i=Chuji@pcp09180602pcs.nash01.tn.comcast.net) |
| 02:01.54 | robin_z | Lord_Drachenblut: failsl to work, fails to upgrade, fails to even pick up charge from cradle ... |
| 02:01.57 | fdask | Due to changes in Linksys/Sipura Policies, Atacomm may only sell the SPA-3000 devices to Linksys Approved Service Providers |
| 02:02.03 | Lord_Drachenblut | what about terrasip |
| 02:02.57 | Zodiacal | how come my cisco 7960 doesn't answer when i pickup the handset, it asks me to press the answer softkey? internal calls are answered by lifting the handset just fine, but external calls coming in want me to press Answer.. |
| 02:03.22 | robin_z | Zodiacal: different incoming channels? |
| 02:03.46 | Zodiacal | robin_z yeah i think i just got it workin.. i had two channels on it.. ok thanks! |
| 02:04.15 | robin_z | ok bedtime |
| 02:04.19 | *** part/#asterisk robin_z (n=robin@adsl.redpoint.org.uk) |
| 02:04.40 | *** join/#asterisk flynux (i=nc1m3c1@pingou.in) |
| 02:06.01 | rpm | anyone know what would cause no audio to work? im using gsm, g729, ulaw and alaw codecs.. when i restart asterisk it screams at me about: chan_zap.c:10829 setup_zap: Ignoring signalling. I am using fxo_ks signalling for my phones which is a FXS PSTN line.. |
| 02:06.32 | fugitivo | rpm: what version? |
| 02:06.48 | rpm | version 1.2.3, zaptel 1.2.2 |
| 02:07.37 | fugitivo | you should use fxs_ks signalling |
| 02:07.46 | fugitivo | not fxo_ks |
| 02:08.10 | fugitivo | i suppose you mean pstn line and not phones |
| 02:08.26 | rpm | no, phone.. i think i meant to say FXO actually.. |
| 02:08.39 | rpm | i have a phone plugged into the card |
| 02:08.46 | fugitivo | what card? |
| 02:08.56 | Err | if you're using a POTS line, compression methods shouldn't matter (since they're analog, not digital) |
| 02:09.02 | rpm | tdm400p |
| 02:09.13 | fugitivo | with fxo modules or fxs modules? |
| 02:09.29 | *** join/#asterisk snewpy (n=markl@210.84.44.179) |
| 02:09.51 | brockj49464 | netrunz: Here it is with a sip debug http://pastebin.com/523149 IS this the problem I have? http://bugs.digium.com/view.php?id=5177 |
| 02:10.16 | rpm | fugitivo, Module 3: Installed -- AUTO FXS/DPO |
| 02:10.29 | fugitivo | only one module? |
| 02:10.32 | rpm | yes |
| 02:10.35 | rpm | just one phone |
| 02:10.43 | fugitivo | what did you say about a PSTN line? |
| 02:11.08 | rpm | its a analog phone.. sorry, getting terms confused.. its been a long day at work |
| 02:11.12 | *** join/#asterisk NoRemorse (n=bah@202.161.68.6) |
| 02:11.14 | NoRemorse | hi all |
| 02:11.16 | fugitivo | oh, ok |
| 02:11.22 | fugitivo | it should work that way |
| 02:11.33 | fugitivo | what do you have in zaptel.conf? |
| 02:11.40 | fugitivo | fxoks=3 ? |
| 02:11.48 | NoRemorse | is it possible to send calls to vonage's network from an asterisk server without signing up for their $15 usd a month account? |
| 02:12.00 | *** join/#asterisk kiswanto (n=kiswanto@222.124.24.61) |
| 02:12.04 | rpm | fxoks=4 |
| 02:12.06 | rpm | loadzone=us |
| 02:12.07 | fugitivo | NoRemorse: yes, call the pstn number |
| 02:12.08 | rpm | defaultzone=us |
| 02:12.16 | fugitivo | rpm: why 4? module is 3 |
| 02:12.26 | shmaltz | NoRemorese, yes, call any of the published numbers that ring into the Vonage network :P |
| 02:12.28 | fugitivo | channel 3 |
| 02:12.41 | NoRemorse | fugitivo: problem is my server is in Australia and I'd be paying IDD trunk rates for that |
| 02:12.51 | kiswanto | hi, how to connection between asterisk and siemens PABX |
| 02:12.52 | rpm | module 0 - channel 1, module 1 - channel 2 ... module 3 - channel 4 |
| 02:13.05 | fugitivo | hmm |
| 02:13.14 | *** join/#asterisk LoRez_ (i=lorez@freenode/staff/lorez) |
| 02:13.15 | NoRemorse | vonage need a 1800 freecall entry intot heir network |
| 02:13.43 | fugitivo | rpm: no, it's not like that |
| 02:13.48 | NoRemorse | on another topic does 1700 iaxtel network still work? |
| 02:13.58 | jebba | yep. 1.2.3 fixed my 1.2.2 probs. :) |
| 02:14.19 | fugitivo | rpm: wait, i'll check one box with a tdm400 |
| 02:14.25 | rpm | ok |
| 02:14.27 | *** join/#asterisk Jun (n=chatzill@pool-138-89-62-149.nwrk.east.verizon.net) |
| 02:14.40 | rpm | fugitivo, it was working with asterisk 1.2.0 |
| 02:15.51 | fugitivo | rpm: you're right, module 3 is channel 4 |
| 02:17.10 | fugitivo | rpm: can you pastebin your zapata.conf? |
| 02:17.59 | rpm | http://pastebin.ca/523365 |
| 02:18.05 | fugitivo | rpm: did you just made a reload or unload and load the zaptel modules? |
| 02:18.22 | rpm | i mean pastebin.com |
| 02:18.34 | rpm | no, i have stopped and restarted asterisk multiple times |
| 02:18.45 | fugitivo | rpm: unload and load zaptel |
| 02:18.50 | fugitivo | i mean, the modules |
| 02:18.54 | rpm | yeah i did that too |
| 02:19.01 | fugitivo | stop asterisk, unload and load modules, start asterisk |
| 02:19.17 | fugitivo | hell, you need to delete commented lines :) |
| 02:19.48 | rpm | yeah, my configs are pretty messy like that |
| 02:20.14 | fugitivo | meanwhile, could you do that? |
| 02:20.21 | rpm | yewah |
| 02:20.23 | fugitivo | stop asterisk, unload zaptel, load zaptel, start asterisk |
| 02:20.28 | rpm | yup just did that |
| 02:20.48 | fugitivo | where is the signalling configured in that pastebin? |
| 02:20.55 | rpm | at the very bottom |
| 02:21.04 | rpm | lemme clean these files up |
| 02:21.54 | *** join/#asterisk Lurr (n=pr0ph3t@adsl-223-183-169.mia.bellsouth.net) |
| 02:22.02 | *** part/#asterisk Lurr (n=pr0ph3t@adsl-223-183-169.mia.bellsouth.net) |
| 02:22.40 | bch | does anyone here run asterisk non-root? |
| 02:23.03 | fugitivo | sure |
| 02:23.12 | fugitivo | who run asterisk as root??? |
| 02:23.45 | bch | hah |
| 02:24.04 | bch | good point |
| 02:24.38 | bch | well, I have been, going to switch to running it non-root |
| 02:24.42 | bch | and upgrading from 1.0.9 |
| 02:24.57 | *** join/#asterisk tomben (n=tomben@208.51.87.172) |
| 02:25.02 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
| 02:25.28 | bch | it doesn't seem like the setup for running it as nonroot is too difficult, but anything I should look out for |
| 02:25.29 | bch | ? |
| 02:25.46 | fugitivo | the wiki has a good article about it |
| 02:25.52 | fugitivo | just follow the steps |
| 02:26.15 | bch | cool, I checked that out, looks pretty straightforward |
| 02:26.24 | bch | what distro do you prefer for production boxes? |
| 02:26.28 | brookshire[home] | bch: it's pretty simple really |
| 02:26.30 | fugitivo | none |
| 02:26.31 | fugitivo | lfs |
| 02:27.02 | brookshire[home] | i run it as a non-root user on my powerbook |
| 02:27.10 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 02:27.22 | rpm | its weird, i should be able to hear myself over my speakers when i call my home phone and am running this SIP phone |
| 02:27.22 | bch | nice |
| 02:27.44 | rpm | as far as i know nothing as changed in my configs |
| 02:27.49 | bch | fugitivo: don't know if I'm that advanced |
| 02:28.06 | Err | the only non-root issue would be interface boards, I suspect |
| 02:28.08 | fugitivo | rpm: that's only a warning, not an error, it should work |
| 02:28.25 | Err | (the non-root user has to have permission to use whatever interface devices it's configured for) |
| 02:28.31 | fugitivo | rpm: if you clean up the config files maybe we'll find the problem |
| 02:29.08 | bch | Err: it's for an IVR over SIP |
| 02:29.10 | fugitivo | bch: just follow the steps and be sure to setup udev with user permissions |
| 02:29.19 | fugitivo | bch: errr, for zaptel device nodes |
| 02:29.19 | rpm | fugitivo, yeah im working on that.. gotta love comments |
| 02:29.46 | Lord_Drachenblut | anyone ever have luck getting a 3com voip phone working with asterisk |
| 02:30.02 | fugitivo | bch: asterisk will not start if it can't log, so be sure that the user will have access to /var/log/asterisk |
| 02:30.03 | Err | bch: if all of your links are IP, I doubt you'll run into any issues |
| 02:30.12 | hans | there's also ToS issues with non-root, but if you have control over a router or firewall that can set that you're ok |
| 02:30.18 | hans | or if you don't need QoS |
| 02:30.35 | Err | actually, ToS isn't the issue - it's if you try to set the IP precedence bits, which you shouldn't :-) |
| 02:30.37 | *** join/#asterisk techie (i=gus@antibala.com) |
| 02:30.58 | hans | Err: oh, you can set ToS without root? |
| 02:31.09 | hans | I was under the impression you couldn't |
| 02:31.10 | brockj49464 | * gods: Here it is with a sip debug http://pastebin.com/523149 IS this the problem I have? http://bugs.digium.com/view.php?id=5177 |
| 02:31.12 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 02:31.12 | *** mode/#asterisk [+o denon] by ChanServ |
| 02:31.13 | Err | everything but your local router will ignore those, anyway, so it doesn't matter (actually, they'll ignore the entire field) |
| 02:31.24 | bch | well, I'm not too worried about QoS |
| 02:31.41 | bch | most calls will be from mobile phones coming in, so they'll think their call cutout |
| 02:32.09 | Err | the only "QoS" settings that matter with IP involve throwing more money at your ISP |
| 02:32.19 | Err | even then, that only guarantees bandwidth on *your* end |
| 02:32.38 | bch | hah, true |
| 02:32.53 | hans | a cynical view that's probably mostly true. :) |
| 02:33.16 | hans | but a lot of devices out there support qos by default, so it's worth the small effort it takes to set it, I think |
| 02:33.18 | Err | the core router folks claim that they're overprovisioned, and they don't ever drop packets except on route failures |
| 02:33.49 | Err | that leaves only the edges, which almost always means your last-mile link; do some reasonable QoS on your outbound router, and that's all you can hope for |
| 02:34.13 | bch | yeah, that's good enough for me |
| 02:36.42 | Err | speaking of... |
| 02:37.16 | rpm | fugitivo, i have posted the uncommented to config /etc/zaptel.conf to http://pastebin.com/523390 |
| 02:38.42 | NoRemorse | is there an asterisk peering organisation? |
| 02:39.08 | brookshire[home] | dundi? |
| 02:39.13 | NoRemorse | along the lines of earning credits for delivering calls to your local network for other members |
| 02:39.18 | NoRemorse | dundi thats it thanks |
| 02:40.15 | NoRemorse | ah looks a little hard core for what I was thinking of. |
| 02:40.46 | Atacomm | any thoughts on sangoma's analog cards? they seem to have way too many peices...... |
| 02:41.12 | rpm | fubster |
| 02:41.33 | *** join/#asterisk hans (n=fugalh@falcon.fugal.net) |
| 02:41.40 | rpm | fugitivo, http://pastebin.com/523393 contains my configurations, i don't know if they are related to this.. but its a bit more information. |
| 02:45.37 | *** join/#asterisk Math[laptop] (n=Math_@70.81.4.148) |
| 02:49.11 | *** join/#asterisk drumkilla (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 02:49.11 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 02:49.52 | *** join/#asterisk ta^3 (n=tacvbo@dsl-200-95-8-31.prod-infinitum.com.mx) |
| 02:49.57 | drumkilla | 415 people here, awesome :) |
| 02:50.07 | Math[laptop] | :) |
| 02:50.24 | ta^3 | I'm reading an Asterisk ChangeLog, so I have an Issue # and a r####. How to find the corresponding bug in http://bugs.digium.com/ using that reference? (Just for know more about a fix) |
| 02:50.24 | *** join/#asterisk annonimous (n=annonimo@dsl-201-129-251-47.prod-infinitum.com.mx) |
| 02:50.27 | ta^3 | ? |
| 02:50.29 | annonimous | hello |
| 02:50.57 | drumkilla | ta^3: the issue # is the mantis issue number |
| 02:51.05 | drumkilla | you can type that into mantis to jump to that issue |
| 02:51.08 | Math[laptop] | there's more people here than in the small irc networks everyone on undernet's #coder-com users were trying to start |
| 02:51.29 | drumkilla | ta^3: top right, by the "Jump" button |
| 02:51.37 | justinu | drumkilla: do you know ke4qqq? |
| 02:51.46 | annonimous | hello good ppl i have troubles im using zaptel-1.2.2 and im looking for the line that i need to modify to get my clone card to init but i cant found it its bad my version of zaptel or what do i need to do? |
| 02:51.54 | drumkilla | justinu: i don't think so ... |
| 02:51.57 | katakefalos | 415 people but noone can tell what the script is to dial a number via a zap channel |
| 02:51.59 | ta^3 | APPLICATION ERROR #1100 ; Issue 6439 not found. |
| 02:52.04 | justinu | his name is david |
| 02:52.10 | drumkilla | ta^3: try 6349 |
| 02:52.13 | justinu | in south carolina |
| 02:52.17 | drumkilla | I probably made a typo ... it was realllllllly early |
| 02:52.18 | ptiggerdine_ | drumkilla, have you seen any whitepapes on security and asterisk. |
| 02:52.32 | Math[laptop] | or click http://bugs.digium.com/view.php?id=6349 |
| 02:52.35 | drumkilla | ptiggerdine_: well, there is a DUNDi whitepaper |
| 02:52.46 | ptiggerdine_ | ok thanks |
| 02:52.54 | drumkilla | on http://www.dundi.com |
| 02:52.58 | ta^3 | I just want to read more about the 'timebomb' bug, but the reference in the asterisk's 1.2.3changelog doesnt exist in mantis. |
| 02:53.02 | drumkilla | that has some security related info it it ... :) |
| 02:53.05 | ta^3 | drumkilla: :-| |
| 02:53.09 | Dr-Linux | drumkilla |
| 02:53.09 | drumkilla | ta^3: ok, then I made a typo in the issue number, sorry |
| 02:53.18 | katakefalos | i would like to repeat my question: --to dial a number 123456 via a sip trunk you do exten => s,2,Dial(sip/123456@trunkname,60,r) what do u do when you want to dial it via a specific zap channel |
| 02:53.20 | drumkilla | it's 6349 |
| 02:53.31 | *** join/#asterisk Feral_Kid (n=Feral@red-corp-200.56.96.178.telnor.net) |
| 02:53.38 | brookshire[home] | ta: the link is right in oej's blog on asterisk.org |
| 02:53.49 | brookshire[home] | it's like the second story |
| 02:53.52 | brookshire[home] | on the front page |
| 02:53.54 | Dr-Linux | i changed the SIP port in sip.conf to 8080, but its not being changed |
| 02:53.58 | Jonathan_ | (has upgraded to 1.2.3) Is anyone aware why music on hold using mp3 files would be silent with the occasional click/crackle? |
| 02:54.00 | De_Mon | katakefalos no you Dial(sip/User:password@truck/${EXTEN},60,r) |
| 02:54.05 | Dr-Linux | when i do "sip show setting" it shows same 5060 |
| 02:54.12 | drumkilla | katakefalos: look in configs/extensions.conf.sample for documentation on the Zap dialing options |
| 02:54.13 | Math[laptop] | De_Mon: er his was is ok... |
| 02:54.18 | De_Mon | Jonathan_ 1.2.23? |
| 02:54.24 | Jonathan_ | correct |
| 02:54.46 | Dr-Linux | http://voxilla.com/PNphpBB2-viewtopic-t-7176.html << these guys have ssame problem like me .. |
| 02:54.56 | De_Mon | Math[laptop] that syntax didn't work with 2 of the 3 providers I've used |
| 02:54.57 | ta^3 | drumkilla: thanks, great, yes its a typo in the ChangeLog. :) |
| 02:55.16 | Math[laptop] | De_Mon: then you didnt configure your peer entry properly |
| 02:55.35 | katakefalos | De_Mon> i mean in like in extensions.conf the: do exten => s,2,Dial(sip/123456@trunkname,60,r) works but i would like to dial via a zap channel |
| 02:55.38 | *** join/#asterisk budman2 (i=budman@12-216-205-236.client.mchsi.com) |
| 02:56.10 | Feral_Kid | Help... After adding a couple of trunks, I suddenly find that I cannot hear anything outgoing or incoming... Although the console says that it operation just fine. Any suggestions? |
| 02:56.20 | De_Mon | Math[laptop] I'll look at them again in that case, good to know. |
| 02:56.44 | drumkilla | ta^3: I could fix that ... buuuuut ... |
| 02:56.46 | Jonathan_ | Any ideas on the no music on hold music anyone? |
| 02:56.51 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 02:56.52 | drumkilla | ta^3: I'll fix it, heh |
| 02:56.52 | Dr-Linux | anybody have any idea ? |
| 02:57.25 | *** join/#asterisk retentiveboy (n=retentiv@h73.90.40.69.ip.alltel.net) |
| 02:57.30 | ta^3 | drumkilla: :D yea!, minor-minor-minor bugfixes. (to the chagelog) xD |
| 02:58.42 | retentiveboy | Wonder if jer is having a few beers after a day like today :) |
| 02:58.44 | drumkilla | ta^3: ha, i fixed the ChangeLog on the ftp |
| 02:58.47 | drumkilla | but not in the tarball |
| 03:01.57 | *** part/#asterisk gongoputch (n=gongoput@pcp01486721pcs.limstn01.de.comcast.net) |
| 03:02.39 | *** join/#asterisk greskdo (n=lesgsod@Toronto-HSE-ppp3746159.sympatico.ca) |
| 03:02.45 | brookshire[home] | we need # |
| 03:02.46 | brookshire[home] | :) |
| 03:03.00 | brookshire[home] | # a few beers |
| 03:03.02 | greskdo | hey, is a mailbox setting valid in an iax context? |
| 03:03.38 | greskdo | I hope I am not too rude to ask the question |
| 03:05.49 | *** join/#asterisk jef_ (i=fischer@p54846C55.dip.t-dialin.net) |
| 03:05.54 | Jonathan_ | greskdo: Are you connecting it to an FXO ATA or ? |
| 03:06.11 | greskdo | I am trying to get the "VM" button from idefisk to do something |
| 03:06.23 | greskdo | and I suppose I need to tell idefisk what mailbox to access |
| 03:06.41 | greskdo | I do not know if this is a setting of idefisk (I don't think) or a setting of the iax channel |
| 03:06.52 | Jonathan_ | the mailbox "button" should be a speedial somewhere in the settings. |
| 03:07.05 | Jonathan_ | the only thing the channel can send is a MWI of some type |
| 03:07.20 | *** join/#asterisk iq (n=iq@71-214-2-97.omah.qwest.net) |
| 03:07.38 | greskdo | and how do I make it send that mwi? |
| 03:08.06 | Jonathan_ | a message waiting indicator (MWI) should get sent down the channel if it has a mailbox associated with it |
| 03:08.28 | greskdo | I see, although it does not seem to work for me (yet, at the very least) |
| 03:08.41 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 03:09.17 | greskdo | see, the table of cmmand on http://www.voip-info.org/wiki/index.php?page=Asterisk%20config%20iax.conf |
| 03:09.38 | greskdo | says that the mailbox setting is not a peer setting nor a user setting |
| 03:10.50 | Jonathan_ | hmm |
| 03:10.54 | X-Rob | I would have to be a user setting |
| 03:10.55 | X-Rob | it |
| 03:11.03 | X-Rob | otherwise there's no way for MWI to work |
| 03:11.32 | greskdo | that's what I'd think, but it does not seem to work, although my settings are a bit weird (one mailbox for all users) |
| 03:11.36 | Jonathan_ | i would think you would just need to add the mailbox=1000@context under the channel definition |
| 03:11.53 | Jonathan_ | that's 1000@vmcontext |
| 03:12.00 | greskdo | I did so |
| 03:12.43 | X-Rob | uh, it's a peer setting. |
| 03:12.55 | greskdo | well my context is a friend |
| 03:13.01 | Dr-Linux | == Parsing '/etc/asterisk/manager.conf': Found |
| 03:13.02 | Dr-Linux | Jan 26 08:14:23 WARNING[3179]: manager.c:1747 init_manager: Unable to bind socke |
| 03:13.02 | Dr-Linux | t: Address already in use |
| 03:13.06 | Dr-Linux | why is this? |
| 03:13.13 | greskdo | which, if I understand, should be both a peer and a friend |
| 03:13.41 | X-Rob | Dr-Linux, it means something is already listening on the manager port. |
| 03:13.44 | *** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 03:13.48 | *** join/#asterisk Lord_Drachenblut (n=Lord@12-210-115-191.client.insightBB.com) |
| 03:13.52 | Jonathan_ | greskdo: that's my understanding as well |
| 03:14.09 | X-Rob | or, your bindaddr=.... is incorrect |
| 03:15.15 | X-Rob | greskdo, try 'iax2 show peer peername' |
| 03:16.22 | *** part/#asterisk kio (n=kio@ool-4577a5a5.dyn.optonline.net) |
| 03:16.41 | Dr-Linux | X-Rob: i changed the sip default port 5060 to 8080 in sip.conf but it still shows 5060 when i do "sip show settings" |
| 03:16.49 | Dr-Linux | any idea? |
| 03:16.53 | X-Rob | Dr-Linux, that's the manager port. |
| 03:17.05 | X-Rob | notice how it says 'manager.c' not 'chan_sip.c' |
| 03:17.12 | X-Rob | oops |
| 03:17.17 | greskdo | X-rob, it does show the mailbox 100@default |
| 03:17.18 | X-Rob | sorry, brain dyslexia. |
| 03:17.41 | Dr-Linux | X-Rob: its okey that problem is fixed, i changed the manager port, and i don't need manager stuff |
| 03:17.47 | Dr-Linux | asterisk is running |
| 03:17.48 | *** join/#asterisk Underhand (n=gavan@frog.coolfactor.org) |
| 03:17.49 | X-Rob | Dr-Linux, no idea irt changing sip ports. Never played with it. |
| 03:17.57 | *** join/#asterisk iq (n=iq@71-214-2-97.omah.qwest.net) |
| 03:18.04 | X-Rob | greskdo, so, what's the problem then? (I missed the start of the convo) |
| 03:18.05 | greskdo | I restarted idefisk |
| 03:18.11 | greskdo | and it's doing something funnier now |
| 03:18.20 | greskdo | it tells me I have 255 new and 255 old messages |
| 03:18.26 | greskdo | I'm pretty popular |
| 03:18.38 | greskdo | among myself and my other personalities... |
| 03:19.18 | *** join/#asterisk gandhijee (i=user@ip72-192-218-111.dc.dc.cox.net) |
| 03:19.19 | greskdo | X-Rob, I'm trying to get idefisk's voicemail button to "light" up when I have messages on my mailbox |
| 03:19.35 | Underhand | folks, i've set up an asterisk server, but can't get the IAX test in the demo to work properly. could somebody spend a few minutes with me trying to debug this, or to explain what I've done wrong? |
| 03:19.40 | Jonathan_ | greskdo: do we know if this is actually a feature of idefisk? |
| 03:19.44 | gandhijee | hey has stuff in asterisk changed alot between 1.0 and 1.2 to the point i can use the old conf files? |
| 03:20.11 | X-Rob | gandhijee, depends. A lot of stuff in 1.0.9 isn't in 1.2.3 |
| 03:20.27 | X-Rob | *agree* Jonathan_ |
| 03:20.37 | greskdo | Jonathan_, maybe it is not a feature, although the button is definitely there |
| 03:20.49 | Jonathan_ | *downloads idefisk* |
| 03:20.49 | Dr-Linux | X-Rob: here is some solution but, i don't understand >> http://voxilla.com/PNphpBB2-viewtopic-t-7176.html |
| 03:20.54 | greskdo | it seems like I did the right thing, so it might just be a problem with idefisk |
| 03:20.56 | gandhijee | ok |
| 03:21.11 | gandhijee | that might explain why i can't dial |
| 03:21.57 | X-Rob | Underhand, you need to give us more information. What youv'e tried, and what's not working |
| 03:23.09 | Underhand | i'm using kphone to connect to asterisk, i've configured in sip.conf the bare minimum i need to connect kphone to asterisk. everything else is the default configuration |
| 03:23.19 | Underhand | i have no hardware interfaces yet |
| 03:23.22 | Jonathan_ | greskdo: how do you tell idefisk what server you want it to register against? |
| 03:23.39 | Underhand | i dial the server, to get into the demo. i dial 500 to connect to digium's test server |
| 03:23.46 | *** join/#asterisk festr_ (n=festr@ns.regnet.cz) |
| 03:23.47 | greskdo | right-click on the interface and go to the account menu |
| 03:23.57 | Underhand | and i see packets going to and fro, but no audio reaches kphone. |
| 03:24.08 | *** part/#asterisk retentiveboy (n=retentiv@h73.90.40.69.ip.alltel.net) |
| 03:24.32 | Jonathan_ | ok |
| 03:24.45 | X-Rob | Underhand, possibly a firewall issue, but unlikely, as IAX is reasonably firewall tolerant. |
| 03:24.57 | X-Rob | can you get audio out of your local server? |
| 03:25.00 | Underhand | x-rob: during the test period, my firewall hasn't blocked any packets. |
| 03:25.28 | Underhand | yes, everything works fine with the demo until i try to connect out. the voice menus work, dtmf works, the echo test works |
| 03:25.58 | greskdo | Underhand, I don't understand, you started by saying you are testing IAX and you modified sip.conf? |
| 03:26.35 | Underhand | greskdo, i'm using SIP to communicate between kphone and asterisk. The asterisk demo uses IAX to communicate between asterisk and digium's test server. |
| 03:26.38 | *** join/#asterisk Iam8up|lappy (n=dontemai@user-0cdv2kk.cable.mindspring.com) |
| 03:26.46 | greskdo | oh, sorry, I missed this part |
| 03:27.30 | Underhand | to expand a little - once asterisk has dialled out to digium, and digium has answered, there's a bidirectional packet flow between asterisk and digium |
| 03:27.36 | X-Rob | ok. digium works. |
| 03:27.40 | Underhand | and there's a bidirectional packet flow between asterisk and kphone. |
| 03:27.49 | Underhand | but no audio. |
| 03:28.06 | Underhand | right up until the call is transfered to digium, including for one ring of ringtone, audio works. |
| 03:28.09 | brookshire[home] | asterisk version? |
| 03:28.36 | X-Rob | I'm guessing it's a codec issue |
| 03:28.39 | Underhand | 1.2.2 |
| 03:28.42 | brookshire[home] | ttsia (the topic says it all) |
| 03:28.48 | X-Rob | Bingo |
| 03:29.15 | brookshire[home] | bringing also |
| 03:29.21 | Underhand | i looked at the topic, looked at the bug report and changelog, which seemed to imply it was only an issue with sip <-> zap |
| 03:29.36 | Underhand | so, sorry if i misread that |
| 03:29.59 | *** join/#asterisk bizvn (n=nghiatha@203.210.207.217) |
| 03:30.01 | brookshire[home] | Underhand: usually upgrading asterisk fixes problems :D |
| 03:30.07 | X-Rob | Underhand, usually if you come in here and you're not running (latest release) or SVN Trunk people will tell you to upgrade. |
| 03:30.40 | *** join/#asterisk ronaldl79 (n=chatzill@c-24-8-54-203.hsd1.co.comcast.net) |
| 03:32.48 | Underhand | is there a known workaround to that issue that i can play with tonight, until i get the latest version compiled up? |
| 03:33.07 | X-Rob | dude, it takes 5 minutes |
| 03:33.35 | Underhand | not on this box it doesn't. |
| 03:33.38 | ManxPower | ~docs |
| 03:33.40 | jbot | somebody said docs was probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 03:33.41 | ManxPower | ~mailinglist |
| 03:33.42 | jbot | i heard mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives/search.php, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.html |
| 03:33.47 | wunderkin | set the date back? |
| 03:34.01 | *** join/#asterisk SibrPhrek (i=SibrPhre@user-12lccke.cable.mindspring.com) |
| 03:34.11 | ronaldl79 | Everyone recovering since 1.2.3 was dropped? |
| 03:34.24 | SibrPhrek | how do i know what version i'm running? |
| 03:34.31 | SibrPhrek | running asterisks on OS X server |
| 03:34.33 | ptiggerdine_ | rpm -qi asterisk |
| 03:34.39 | ptiggerdine_ | lol |
| 03:34.39 | ronaldl79 | How's that going, Sibr? |
| 03:34.47 | ptiggerdine_ | asterisk -V |
| 03:34.50 | ptiggerdine_ | i think |
| 03:34.51 | SibrPhrek | ronaldl79: works - but i'm having issues with audio |
| 03:35.07 | ronaldl79 | Are you running on a G5? |
| 03:35.13 | SibrPhrek | x-serve |
| 03:35.23 | SibrPhrek | i am running 1.2.2 |
| 03:35.26 | SibrPhrek | what's new in the 1.2.3 ? |
| 03:35.37 | Grubs | a fix for your audio problem |
| 03:35.39 | ronaldl79 | Well, the 'time bomb' fix is the major one....and some others. |
| 03:35.54 | SibrPhrek | Grubs: seriously? |
| 03:36.07 | ronaldl79 | Sibr -- He's not kidding. |
| 03:36.08 | Grubs | http://www.asterisk.org/node/55 |
| 03:36.10 | SibrPhrek | whoa |
| 03:36.11 | SibrPhrek | haha |
| 03:36.12 | SibrPhrek | ok |
| 03:36.13 | SibrPhrek | lemme try this |
| 03:36.28 | drumkilla | just so you guys know, my phone rang at 5 AM about that bug :-p |
| 03:36.32 | Underhand | may i suggest /topic Asterisk 1.2.3 Release (If you are running 1.2.2, this will fix your audio) |
| 03:36.33 | ronaldl79 | I was up late into the night, and tried making an outbound call...nothing...I was about to flame BroadVoice.... |
| 03:36.46 | Aughey | what's the diff between asterisk-1.2.3.tar.gz and asterisk-1.2.3-netsec.tar.gz on the ftp server? |
| 03:36.46 | SibrPhrek | are there special rules in terms of upgrading? |
| 03:36.46 | ronaldl79 | lol @ drumkilla |
| 03:36.50 | SibrPhrek | i'm new to this whole thing |
| 03:37.03 | ronaldl79 | drumkilla -- You're not K Fleming, are you? I'm not sure who's who. |
| 03:37.13 | drumkilla | no, Kevin Fleming is kpfleming on IRC |
| 03:37.14 | ronaldl79 | Welcome, Sibr. |
| 03:37.16 | drumkilla | I am Russell Bryant |
| 03:37.20 | wunderkin | drumkilla, and there wasn't any audio? |
| 03:37.25 | ronaldl79 | Oh, right....sorry, my brain is fried. |
| 03:37.28 | drumkilla | wunderkin: lol |
| 03:37.30 | wunderkin | HEH |
| 03:37.50 | drumkilla | I'd switch to "russell" on IRC probably, except someone already took it :( |
| 03:38.05 | ptiggerdine_ | lol@ wunderkin |
| 03:38.16 | Feral_Kid | I have changed codecs on the sipura and I still can't hear anything... |
| 03:38.26 | SibrPhrek | ronaldl79: i found easy installers for OS X, as well as extention adder GUI |
| 03:38.26 | Rowter | anyone knows if I can run agis on macros it seems not to work mmh |
| 03:38.28 | SibrPhrek | looks nice |
| 03:38.29 | ronaldl79 | Has anyone tried Monitor since 1.2.3 dropped? |
| 03:38.55 | wunderkin | ronaldl79, um no not yet but maybe i should do my testing, are you the one that just asked about that on the mailing list? |
| 03:39.07 | ronaldl79 | Yes, wunderkin. |
| 03:39.23 | ronaldl79 | The audio isn't sync'd at all, and I didn't have these problems until I upgraded to 1.2.3. |
| 03:39.38 | *** join/#asterisk aogail (n=aogail@c-67-168-255-131.hsd1.or.comcast.net) |
| 03:40.07 | reza | yay |
| 03:40.11 | SibrPhrek | my audio issue is that i can't hear the other person and they can't hear me |
| 03:40.19 | reza | shit works. wasted HOURS because of that dumb timebomb bug |
| 03:40.20 | SibrPhrek | so forget sync'ing - i have a lack of audio |
| 03:40.37 | reza | seems NuFone had the broken code and I thought it was me |
| 03:40.53 | reza | ok, now that asterisk works, what are some fun things to do with it? |
| 03:41.00 | ronaldl79 | Sibr -- Upgrade to 1.2.3, or downgrade to 1.2.1 -- I am doing that right now to see if it resolves my issue with Monitor |
| 03:41.02 | SibrPhrek | reza - make phone calls! |
| 03:41.09 | reza | that works |
| 03:41.09 | JonR800 | SibrPhrek: upgrade to 1.2.3 or.. yes what ronaldl79 |
| 03:41.09 | aogail | Howdy all. |
| 03:41.19 | reza | it works fine, though the ringing is choppy. audio is good after the ringing |
| 03:41.20 | SibrPhrek | ronaldl79: i'm copying the 1.2.3.gz to my server right now |
| 03:41.41 | reza | is 1.2.3 just 1.2.2 with the timebomb patch? |
| 03:41.42 | Grubs | Is this is an OK way to handle a fallback from one trunk to another? http://pastebin.com/523449 |
| 03:41.59 | JonR800 | reza: no, read the changelog. there are other changes. |
| 03:43.02 | greskdo | I have a question: my phone calls to a sip or iax softphone are very choppy over a land line, but sound fine over a cell phone. any idea why this would happen? |
| 03:43.05 | reza | i got the post-patch svn from last night; wonder if it's worth upgrading again |
| 03:43.08 | SibrPhrek | ronaldl79: are there special upgrade instructions - or is this just ./configure make make install |
| 03:43.40 | reza | i dont think there is a configure, just 'make install' |
| 03:43.44 | ronaldl79 | Sibr -- That's the only way I do it. :) |
| 03:43.58 | ronaldl79 | Right, 'make install.' |
| 03:43.59 | SibrPhrek | thanks |
| 03:44.04 | Underhand | ok, setting the date back gives me audio. thanks guys, upgrading now. sorry for the noise. :) |
| 03:44.43 | reza | i wonder who checked in the broken code in the first place? and how much in damage it caused |
| 03:45.09 | Grubs | reza - dont go there ! :p |
| 03:45.11 | SibrPhrek | hrm |
| 03:45.13 | SibrPhrek | i do make |
| 03:45.20 | SibrPhrek | i get no target specified and no makefile found |
| 03:45.32 | aogail | I'm running the latest AAH installation and am getting no sound when calling between extensions or to the PSTN using an IAX trunk. During the calls I get tons of "Dropping voice to exceptionally long queue" messages in the log. The few relevant google results haven't helped fix this. Does anyone have any suggestions? |
| 03:45.33 | SibrPhrek | which is BS cuz that Makefile file is there |
| 03:46.02 | reza | make -f <path to makefile> |
| 03:46.06 | SibrPhrek | oh |
| 03:46.06 | SibrPhrek | der |
| 03:46.10 | SibrPhrek | i wasn't in the directory |
| 03:46.11 | SibrPhrek | duh! |
| 03:46.12 | reza | make all -f <path to makefile> |
| 03:46.16 | Grubs | aogail: A@H needs to be upgraded to Asterisk 1.2.3. 1.2.2 has a critical audio bug. |
| 03:46.20 | reza | make install -f <path to makefile> |
| 03:46.22 | SibrPhrek | i'm dumb |
| 03:46.38 | ronaldl79 | For anyone that's running 1.2.3 .... could you do a test call with Monitor applied??? |
| 03:46.40 | Grubs | for now set your date back to before Jan 25 and wait for the updated A@H |
| 03:46.50 | SibrPhrek | what's Monitor? |
| 03:47.05 | Grubs | (or compile 1.2.3 into your box) |
| 03:47.50 | *** join/#asterisk Eight (n=blake@12-227-169-99.client.mchsi.com) |
| 03:47.56 | greskdo | ok, I'll try another question |
| 03:48.08 | greskdo | how can I know which code is in use during a call? |
| 03:48.10 | Grubs | ronaldl79: I'm trying it now |
| 03:48.11 | greskdo | code=codec |
| 03:48.15 | aogail | Grubs: Thanks for the info. I wondered about that update, but didn't see the message referenced in the changelog. Do you know if AAH uses a vanilla installation of *? i.e. would downloading and installing normally break it? |
| 03:49.33 | *** join/#asterisk bmg505 (n=leon@dsl-146-31-151.telkomadsl.co.za) |
| 03:49.38 | Dr-Linux | X-Rob: great have changed the sip port, but it was some odd expereince |
| 03:50.03 | inv_Arp | it is poss to place a call via cmd line right? |
| 03:50.25 | ronaldl79 | Wonderful, Grubs |
| 03:50.44 | ronaldl79 | It would probably help to test for a few minutes ... and see if the flat wav remains sync'd... |
| 03:51.16 | SibrPhrek | nice |
| 03:51.17 | SibrPhrek | it worked! |
| 03:51.21 | SibrPhrek | 1.2.3 is GREAT! |
| 03:51.48 | ronaldl79 | Nice, Sibr. |
| 03:51.49 | Grubs | ronaldl79: Monitor worked perfectly |
| 03:51.57 | ronaldl79 | How long did you test, Grubs? |
| 03:51.58 | SibrPhrek | grubs - what's Monitor? |
| 03:52.11 | ronaldl79 | Sibr -- Monitor allows you to recall calls |
| 03:52.18 | ronaldl79 | record |
| 03:52.19 | Grubs | only 30 seconds or so - a few sentances |
| 03:52.22 | ronaldl79 | I am so fried |
| 03:52.27 | SibrPhrek | is it part of the cli |
| 03:52.32 | SibrPhrek | or is part of the sip phone |
| 03:52.38 | ronaldl79 | Grubs -- try a few minutes ... call a buddy ... ask them random questions .. |
| 03:52.40 | Grubs | Sibr: http://www.voip-info.org/wiki-Asterisk+cmd+Monitor |
| 03:53.08 | ravenpi | Query: anyone know why faxes might be received "offset", such that the .tif file has (say) the right 3/4 of the fax shifted left, and then the remainder "pasted" on to the right side? |
| 03:53.16 | Grubs | ok... I'm calling my own soft phone through a providor on the other side of the country |
| 03:54.06 | SibrPhrek | my phone system still sounds a little choppy - i would like it clearer - is that because of the sip trunk i'm connected to? |
| 03:54.22 | wunderkin | ronaldl79, works ok for me too recording in ulaw, but only recorded for about 25 sec |
| 03:54.28 | *** join/#asterisk Prival (i=user65@Kitchener-HSE-ppp3571800.sympatico.ca) |
| 03:54.42 | Prival | Hi, is there a way to force a SIP unregistration? |
| 03:54.44 | ronaldl79 | So, not sync issues, wunder? |
| 03:54.49 | ronaldl79 | no |
| 03:55.08 | wunderkin | ronaldl79, you are using option m to mix the files together? i'm not |
| 03:55.10 | Grubs | ronaldl79: : Monitor is currently recording AC/DCs "You shook me all night long" :) |
| 03:55.16 | *** join/#asterisk srt (n=nobody@18.120.9.213.dsl.getacom.de) |
| 03:55.38 | Prival | ok... Then what you do when you want ot change the extension used in the registration let's say... |
| 03:56.14 | *** join/#asterisk hd420_ (n=hdiwan@c-69-181-3-188.hsd1.ca.comcast.net) |
| 03:57.07 | ronaldl79 | rofl |
| 03:57.12 | *** join/#asterisk MikeJ__ (n=vircuser@70.103.248.130) |
| 03:57.34 | ronaldl79 | Yes, wunder, though I've considered trying a separate recording... |
| 03:58.25 | *** join/#asterisk Qwell[laptop] (n=Qwell[]@unaffiliated/qwell) |
| 04:01.05 | SibrPhrek | how come when i do "safe_asterisk" - i get Cannot find your TTY (9) |
| 04:01.40 | Grubs | ronaldl79: : seems good. Made a couple of 4.2MB wav files - both seem OK |
| 04:01.54 | Q-At-Home | safe_asterisk spawns the console on a tty in this case, 9, try changing the file to like... 4 |
| 04:02.10 | Grubs | duration 4.3 minutes |
| 04:02.12 | SibrPhrek | Q-At-Home: how do i do that? |
| 04:02.19 | SibrPhrek | i'm using FreeBSD on OS X |
| 04:02.20 | ronaldl79 | Hmmm, alright, Grubs....thanks for testing.... |
| 04:02.26 | Q-At-Home | edit safe_asterisk and change the line TTY=2 |
| 04:02.30 | Q-At-Home | 2 is mine |
| 04:02.38 | Grubs | np |
| 04:02.41 | ronaldl79 | Well, I guess I won't downgrade afterall... |
| 04:02.49 | wunderkin | ok for me too |
| 04:02.57 | SibrPhrek | edit safe_asterisk |
| 04:02.59 | SibrPhrek | uhm |
| 04:02.59 | SibrPhrek | ok? |
| 04:03.18 | Q-At-Home | pico `which safe_asterisk` |
| 04:03.21 | Grubs | of course recoding a phone call without permission is illegal here ;) |
| 04:03.26 | a1fa | sup :p |
| 04:03.29 | Grubs | recording |
| 04:03.33 | SibrPhrek | Q-At-Home: from the /etc/asterisk folder? |
| 04:03.34 | Q-At-Home | ` is a backtick |
| 04:03.45 | Q-At-Home | no, wherever its installed |
| 04:03.49 | Q-At-Home | which safe_asterisk |
| 04:03.52 | Q-At-Home | will tell you where it is |
| 04:04.09 | SibrPhrek | ah |
| 04:04.10 | SibrPhrek | thanks |
| 04:04.39 | SibrPhrek | ok i tried 4 and 2, and still can't find TTY |
| 04:05.11 | SibrPhrek | do i have to set it to /dev/ttyp1 ? |
| 04:05.15 | greskdo | I'll try my question again, if you don't mind |
| 04:05.22 | greskdo | how can I know which codec is used during a call? |
| 04:05.34 | a1fa | sip status |
| 04:05.36 | a1fa | :P |
| 04:05.37 | *** join/#asterisk pengyong (n=lala@222.188.134.60) |
| 04:05.46 | a1fa | greskdo : sip? |
| 04:05.50 | greskdo | doesn't matter |
| 04:05.52 | greskdo | in general |
| 04:05.59 | greskdo | I want to debug some sound issues |
| 04:06.09 | greskdo | and I am wondering if it's not a problem with a particular codec |
| 04:06.13 | greskdo | when the sound is choppy |
| 04:06.33 | greskdo | in general it's a PSTN->Zap->(SIP or IAX) kind of problem |
| 04:06.36 | Grubs | in general - connect to asterisk with the -vvvvvvr switch and watch the console. it tells you the codec |
| 04:06.42 | greskdo | ok |
| 04:06.56 | greskdo | so there is no "show" something command that could tell me this |
| 04:06.59 | greskdo | quite weird |
| 04:07.21 | a1fa | hm |
| 04:07.30 | *** join/#asterisk brookshire[home] (n=matt@pcp09462771pcs.huntsv01.al.comcast.net) |
| 04:07.42 | SibrPhrek | OMG |
| 04:07.46 | SibrPhrek | i changed the TTY to 1 |
| 04:07.48 | Underhand | well you can do sip show channels |
| 04:07.50 | SibrPhrek | and i still get Cannot find your TTY |
| 04:08.31 | ronaldl79 | IAXTel ... is it down? |
| 04:08.39 | SibrPhrek | Q-At-Home: any suggestions? |
| 04:09.15 | greskdo | Underhand, thanks, I must have been really tired last not for not having seen that |
| 04:09.45 | *** join/#asterisk dextro (n=dextro@cpe-70-116-14-87.austin.res.rr.com) |
| 04:10.39 | Grubs | SibrPhrek - just comment out the TTY command alltogether and connect to asterisk using "asterisk -r" |
| 04:10.56 | SibrPhrek | ok |
| 04:10.59 | *** join/#asterisk r0d3nt (i=r0d3nt@tinfoilhat.net) |
| 04:11.28 | SibrPhrek | oh no |
| 04:11.30 | SibrPhrek | it's in a loop |
| 04:11.32 | SibrPhrek | help! |
| 04:11.46 | ptiggerdine_ | killall asterisk. |
| 04:12.34 | ptiggerdine_ | OSX doesn't have tty's in the traditional sence as unix/linux has. |
| 04:12.39 | *** join/#asterisk bofh42 (n=bofh42@p54828725.dip0.t-ipconnect.de) |
| 04:12.46 | SibrPhrek | yeah i see that |
| 04:13.14 | ptiggerdine_ | have you managed to stop it form looping. |
| 04:13.33 | ptiggerdine_ | mmm I need to reboot I think. |
| 04:13.36 | Supaplex | from looping from looping from looping *hic* |
| 04:13.44 | Grubs | pretty please - Is this is an OK way to handle dialing a second trunk if the first fails? http://pastebin.com/523449 |
| 04:18.26 | SibrPhrek | if you type safe_asterisk |
| 04:18.28 | SibrPhrek | what should happen |
| 04:18.51 | SibrPhrek | TTY=a # TTY (if you want one) for Asterisk to run on |
| 04:18.56 | SibrPhrek | if [ "$TTY" != "" ]; then |
| 04:18.56 | SibrPhrek | <PROTECTED> |
| 04:18.56 | SibrPhrek | <PROTECTED> |
| 04:19.04 | SibrPhrek | i added the p there |
| 04:19.13 | SibrPhrek | and now when i type safe_asterisk - i don't get an error |
| 04:19.16 | SibrPhrek | i just don't get anything |
| 04:21.10 | *** join/#asterisk fibergod (n=scourtne@pcp0011071153pcs.reston01.va.comcast.net) |
| 04:22.08 | *** join/#asterisk Muckl (n=yo@p54BEDBAA.dip.t-dialin.net) |
| 04:22.58 | *** join/#asterisk pimpwell (n=pimpwell@ool-44c768ec.dyn.optonline.net) |
| 04:25.39 | *** join/#asterisk ipso (n=ipso@d207-81-249-35.bchsia.telus.net) |
| 04:25.51 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
| 04:26.01 | Feral_Kid | Can anyone point me to where I can solve my no audio issue... The consoles shows that I am dialing out, and also recieving, but I cannot hear a things.. |
| 04:26.06 | SibrPhrek | Feral_Kid: |
| 04:26.08 | SibrPhrek | look @ topic |
| 04:26.12 | SibrPhrek | you have to get 1.2.3 |
| 04:26.16 | SibrPhrek | i had the same problem |
| 04:26.19 | wisdom | Feral_Kid check firewall settings |
| 04:26.42 | SibrPhrek | wisdom: how do you activate music on hold? |
| 04:26.53 | brookshire[bar] | TTSIA! |
| 04:27.02 | wisdom | SibrPhrek read the documentation |
| 04:27.38 | Feral_Kid | SibrPhrek: Actually, it was running just fine prior to me adding some trunks... |
| 04:28.18 | Grubs | coincidence perhaps |
| 04:28.36 | SibrPhrek | timebomb? |
| 04:28.40 | Feral_Kid | wisdom: No change to the firewall, and as I said earlier, prior to adding some trucks, it was working fine... |
| 04:28.46 | Grubs | my audio died when I was editing conf files also - cost me 4 hours or so in debugging |
| 04:28.58 | litage | what's the difference res_mysql and cdr_mysql? |
| 04:29.27 | Feral_Kid | I think that I did something wonky with the way the codecs are being handled... |
| 04:29.58 | Feral_Kid | I seem to remember this problem before... |
| 04:30.09 | greskdo | Feral_Kid, listen to what SibrPhrek is saying, try upgrading to 1.2.3 |
| 04:30.17 | greskdo | if you are running 1.2.2 |
| 04:30.59 | SibrPhrek | hrm |
| 04:31.05 | SibrPhrek | mp3 support you need mpg123 |
| 04:31.09 | SibrPhrek | i wonder if that will work on OS X |
| 04:32.59 | ronaldl79 | You can also use format_mp3 |
| 04:33.33 | Nugget | I'm holding out for format_aac and format_m4p |
| 04:33.43 | ronaldl79 | :) |
| 04:33.53 | SibrPhrek | yeah mpg123 don't like OS X |
| 04:34.03 | SibrPhrek | ronaldl79: format_mp3? |
| 04:34.11 | ronaldl79 | Native MP3 playback, Sibr. |
| 04:34.17 | SibrPhrek | work in OS X? |
| 04:34.25 | ronaldl79 | Not sure, I'm a PC guy. |
| 04:34.50 | bsdfreak | heh |
| 04:35.01 | ronaldl79 | Or, how about, PC X guy? :) |
| 04:35.39 | ronaldl79 | Anyone using the zaprtc from BRI for timing? |
| 04:35.57 | bsdfreak | anyone familiar with the sipura 3000 |
| 04:36.16 | SibrPhrek | ronaldl79: i'm just gonna tkae the addon package and sounds package from asterisk's home page |
| 04:36.17 | budman2 | anyone have asterlink and asterisk? |
| 04:36.20 | SibrPhrek | hopefully somethign in there will work |
| 04:36.38 | bsdfreak | budman: i do |
| 04:36.48 | budman2 | can you help me :-) |
| 04:36.56 | *** join/#asterisk ptiggerdine_ (n=ptiggerd@203-206-59-72.dyn.iinet.net.au) |
| 04:37.00 | gandhijee | hey can someone give me a hand? |
| 04:37.06 | *** join/#asterisk johnnyb (n=jonathan@207.155.33.225) |
| 04:37.09 | gandhijee | i have a broadvoice account and fwd |
| 04:37.16 | gandhijee | i can recieve calls |
| 04:37.24 | gandhijee | from both but i can't make calls |
| 04:37.34 | gandhijee | i am using my old confs from 1.0.9 |
| 04:38.04 | gandhijee | and i am runnin 1.2.1 |
| 04:38.19 | ronaldl79 | Gand -- Any issues with BV recently? |
| 04:38.30 | gandhijee | not that i know of. |
| 04:38.39 | ronaldl79 | No dropped audio or anything, gand? |
| 04:38.45 | gandhijee | like i said i can't make outbound calls on either broadvoice or fwd |
| 04:38.51 | gandhijee | but i can get the calls |
| 04:38.59 | budman2 | bsdfreak can you help me? |
| 04:39.07 | bsdfreak | with? |
| 04:39.08 | *** join/#asterisk asteriskmonkey (n=phil@67.71.177.202) |
| 04:39.11 | asteriskmonkey | hey |
| 04:39.19 | SibrPhrek | ronaldl79: now that i did the make and make install of the asterisk addons - how do i know format_mp3 is working? |
| 04:39.24 | asteriskmonkey | im having some issue getting spandsp to load |
| 04:39.30 | budman2 | conntecting asterlink to my asterisk and playing the monkeys |
| 04:39.47 | budman2 | ive got it registered via iax2 |
| 04:39.49 | gandhijee | one sec, lemme paste bin my confs |
| 04:40.02 | bsdfreak | heh |
| 04:40.03 | asteriskmonkey | ive tried multiple things and all i get is the silly eroor Unable to load module is there a more extensive why reason |
| 04:40.05 | bsdfreak | you should probably use sip |
| 04:40.43 | budman2 | would not connect with sip |
| 04:40.48 | budman2 | they say i have a firewall |
| 04:40.54 | bsdfreak | heh |
| 04:40.55 | budman2 | with my server i have had 2 years |
| 04:41.01 | bsdfreak | heh |
| 04:41.12 | bsdfreak | that's not good |
| 04:41.22 | *** join/#asterisk zu (n=raz@28-pool1.ras14.floca.alerondial.net) |
| 04:41.25 | zu | hy all |
| 04:41.33 | zu | hy all |
| 04:41.57 | budman2 | can you switch me to sip in their systems? |
| 04:41.59 | budman2 | i can try that again |
| 04:42.12 | bsdfreak | no, i don't work there. you can do that in their control panel. |
| 04:42.51 | Grubs | asteriskmonkey: what version spandsp and asterisk |
| 04:43.29 | FuriousGeorge | so i got a bunch of includes => for time of day in my inbound context, and now i call and it complains of no s extension. why isnt it including based on the time of day and going to the right s extension |
| 04:44.08 | asteriskmonkey | Grubs: version spandsp-0.0.2pre23 |
| 04:44.14 | gandhijee | http://pastebin.ca/38629 |
| 04:44.23 | gandhijee | http://pastebin.ca/38630 |
| 04:44.31 | gandhijee | http://pastebin.ca/38631 |
| 04:44.47 | gandhijee | extensions |
| 04:45.03 | budman2 | ok should take about 5 minutes |
| 04:45.07 | budman2 | sip is way better? |
| 04:45.24 | Grubs | SibrPhrek: I couldnt get a peep out of format_mp3. If you run "make mpg123" from the asterisk source directory then it will download and isntall the right version of mpg123 |
| 04:45.34 | bsdfreak | heh |
| 04:45.37 | bsdfreak | sip is better with asterlink |
| 04:45.49 | SibrPhrek | Grubs: mpg123 won't run on OS X |
| 04:45.52 | SibrPhrek | i got format_mp3 in |
| 04:45.57 | SibrPhrek | i just don't know what to do now |
| 04:46.02 | SibrPhrek | like how do i "activate" it |
| 04:46.08 | SibrPhrek | or any of the sound files |
| 04:46.14 | SibrPhrek | i'm such a n00b |
| 04:46.17 | SibrPhrek | i'll admit it |
| 04:46.24 | Grubs | exten => 6601,1,WaitMusicOnHold(30) |
| 04:46.25 | asteriskmonkey | Grubs: anyway of getting more info other than module wont load :P |
| 04:46.38 | Grubs | then dial 6601 and you should get 30 seconds of MOH. |
| 04:46.55 | SibrPhrek | does that go in macr-extenstions of extensions.conf? |
| 04:47.12 | budman2 | sip |
| 04:47.15 | budman2 | unregistered |
| 04:47.38 | *** join/#asterisk r0d3nt|m (i=r0d3nt@tinfoilhat.net) |
| 04:47.55 | Grubs | ummm...... depends on your extensions.conf file. In mine its under [default] |
| 04:48.09 | Grubs | as I dial in on the default context |
| 04:48.27 | SibrPhrek | ok hang on |
| 04:48.28 | SibrPhrek | lemme try it |
| 04:49.01 | Grubs | find some other extension that you know works (e.g. 600 for echo test?) locate that and put it under that. |
| 04:49.14 | Grubs | reload |
| 04:49.21 | asteriskmonkey | grubs : spandsp-0.0.2pre23 and asterisk 1.2 |
| 04:49.28 | SibrPhrek | reload from CLI? |
| 04:49.35 | Grubs | yes |
| 04:49.43 | Feral_Kid | Ok, just taking a look at my console... Could someone take a look at http://pastebin.com/523525 and see if that is the reason I am having audio problems? The 201 extention is a SIpura 2000 |
| 04:50.02 | Feral_Kid | Extension even... :) |
| 04:50.08 | gandhijee | http://pastebin.ca/38629, http://pastebin.ca/38631, http://pastebin.ca/38632 |
| 04:50.10 | Grubs | asteriskmonkey - I just set up 1.2.3 with spandsp 0.0.2pre23 and it worked ok... sorry - no idea |
| 04:50.11 | budman2 | bsdfreak |
| 04:50.15 | gandhijee | can someone help me |
| 04:50.20 | budman2 | which conf do you use the control panel or email |
| 04:50.20 | gandhijee | i can't make out going calls |
| 04:50.24 | budman2 | they are different |
| 04:50.39 | gandhijee | those are the extentsions.conf sip.conf and iax.conf |
| 04:51.12 | asteriskmonkey | Grubs: do i have to delete all my old files? i went from 1.0.9 a while ago and never got it workign again :( |
| 04:51.22 | SibrPhrek | Grubs: i get call failed 404 not found |
| 04:51.40 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 04:51.45 | SibrPhrek | fuck |
| 04:51.47 | SibrPhrek | need to reboot |
| 04:51.52 | asteriskmonkey | load app_rxfax.so |
| 04:51.52 | asteriskmonkey | Unable to load module app_rxfax.so |
| 04:51.59 | Grubs | asteriskmonkey - I started from scratch with a clean hard drive after I could never get faxing under 1.0.9 working |
| 04:52.04 | X-Rob | so look at /var/log/asterisk/full |
| 04:52.14 | Grubs | with the clean setup its working brilliantly |
| 04:52.16 | X-Rob | my money says that you haven't installed spandsp, or if you have, you haven't run ldconfig |
| 04:52.30 | Feral_Kid | I am wondering if this just a sipura issue... |
| 04:52.38 | SibrPhrek | hrm |
| 04:52.41 | SibrPhrek | i'm getting static now |
| 04:52.43 | SibrPhrek | werid |
| 04:52.47 | SibrPhrek | has to be my comp |
| 04:52.53 | asteriskmonkey | X-Rob: i have run ldconfig :( |
| 04:52.58 | SibrPhrek | god damnit i can't reboot until this CD is done burning |
| 04:53.15 | asteriskmonkey | i wish it would tell me more other thatn unable to load module |
| 04:53.21 | SibrPhrek | ? |
| 04:53.22 | X-Rob | asteriskmonkey, is the path to libspandsp in /etc/ld.so.conf? |
| 04:53.29 | asteriskmonkey | yes |
| 04:53.30 | gandhijee | yo can i get some help |
| 04:53.34 | X-Rob | asteriskmonkey, it does. look at /var/log/asterisk/full |
| 04:53.37 | gandhijee | PLEASE |
| 04:54.43 | asteriskmonkey | ah ha .. i found something |
| 04:54.46 | Grubs | gandhijee - what version asterisk |
| 04:54.48 | asteriskmonkey | but i dont know what it means |
| 04:54.51 | gandhijee | 1.2.1 |
| 04:54.54 | asteriskmonkey | Jan 25 09:19:12 WARNING[27885] loader.c: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: t30_completion_code_to_str |
| 04:54.54 | asteriskmonkey | Jan 25 09:19:12 WARNING[27885] loader.c: Loading module app_rxfax.so failed! |
| 04:55.08 | gandhijee | the files are re-edits from 1.0.9 |
| 04:55.17 | gandhijee | but i don't see what the heck i did wrong |
| 04:55.33 | Grubs | what version of asterisk are you running now |
| 04:55.36 | gandhijee | i don't see the diffs that the UPGRADE.txt file tells me about |
| 04:55.37 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@84.205.154.241) |
| 04:55.41 | gandhijee | 1.2.1 |
| 04:56.02 | PoWeRKiLL | hi |
| 04:56.03 | PoWeRKiLL | gandhijee what is your problem ? |
| 04:56.13 | gandhijee | <PROTECTED> |
| 04:56.17 | gandhijee | at all |
| 04:56.29 | gandhijee | i know the FWD logic works |
| 04:56.40 | PoWeRKiLL | What happened in the console ? |
| 04:56.41 | gandhijee | well i know it works with 1.0.9 |
| 04:57.17 | gandhijee | one sec |
| 04:57.17 | X-Rob | gandhijee, dude. Give up. I'm, right now, trying to get AMP to work with 1.2 |
| 04:57.24 | X-Rob | uh 1.2.3 even |
| 04:57.28 | X-Rob | but you're running 1.2.1 |
| 04:57.32 | gandhijee | no, because i had it running 1.0.9 |
| 04:57.37 | Grubs | asteriskmonkey: I got the same error. I went back to the tiff and spandsp sources - run "make uninstall" and then re-did the configre with "./configure --prefix=/usr" to match my asterisk install. - reisntalled tiff-3.8.0 and spandsp and its all gravey. |
| 04:57.39 | gandhijee | it just means i am missing something |
| 04:58.32 | budman2 | bsd it never registers with their servers |
| 04:58.38 | gandhijee | Jan 25 20:07:28 WARNING[1933]: chan_iax2.c:6970 socket_read: Call rejected by 192.246.69.186: No such context/extension |
| 04:58.38 | gandhijee | <PROTECTED> |
| 04:58.38 | gandhijee | <PROTECTED> |
| 04:58.39 | gandhijee | <PROTECTED> |
| 04:58.48 | Grubs | note - I'm a unix n00b and just decided its easier if all the parts fitted the same directory structure - someone wiht more skill would prolly do differently! |
| 04:58.51 | gandhijee | does someone have a FWD number i can dial? |
| 04:59.01 | asteriskmonkey | Grubs: thanks ill try some of that |
| 04:59.04 | gandhijee | one sec |
| 04:59.14 | gandhijee | nm |
| 04:59.26 | PoWeRKiLL | gandhijee : it(s simple you just don't set the context or the extension to fall in when dialing |
| 04:59.44 | Grubs | both tiff and spandsp needed to be reconfigured and reinstalled - then recompile and reinstall asterisk (no configure needed) |
| 04:59.45 | gandhijee | ?? |
| 04:59.51 | gandhijee | what do you mean powerkill |
| 05:00.02 | gandhijee | http://pastebin.ca/38629, http://pastebin.ca/38631, http://pastebin.ca/38632 |
| 05:00.03 | PoWeRKiLL | you get No such context/extension |
| 05:00.08 | gandhijee | those are my confs |
| 05:00.32 | asteriskmonkey | where do i find my tiff stuff |
| 05:00.44 | Grubs | did you isntall tiff? |
| 05:00.54 | Grubs | libtiff? |
| 05:00.57 | gandhijee | can you give me an example powerkill |
| 05:00.58 | gandhijee | ? |
| 05:01.20 | asteriskmonkey | yes i dont have src for that :( |
| 05:01.29 | Grubs | how did you install? |
| 05:01.47 | asteriskmonkey | rpm |
| 05:02.29 | *** join/#asterisk SibrPhrek (i=SibrPhre@user-12lccke.cable.mindspring.com) |
| 05:02.46 | SibrPhrek | ah |
| 05:02.48 | SibrPhrek | better |
| 05:03.13 | Grubs | I uninstalled the default debian libtiff and then grabbed the latest from "ftp://ftp.remotesensing.org/libtiff/tiff-3.8.0.tar.gz" |
| 05:03.23 | gandhijee | PoWeRKiLL: i don't understand what you are saying |
| 05:03.25 | *** join/#asterisk EriSan (n=erisan@81-174-23-205.f5.ngi.it) |
| 05:03.32 | PoWeRKiLL | It's meachecking your conf file |
| 05:03.39 | gandhijee | ok |
| 05:04.01 | gandhijee | a bunch is commented out |
| 05:04.10 | gandhijee | cuz this one doesn't have a TDM card |
| 05:05.05 | SibrPhrek | Grubs: 6601 i get 404 not found |
| 05:05.18 | Grubs | bugger |
| 05:05.28 | *** join/#asterisk Lord_Drachenblut (n=Lord@12-210-115-191.client.insightBB.com) |
| 05:05.37 | Grubs | is your musiconhold.conf and features.conf set up? |
| 05:05.47 | PoWeRKiLL | gandhijee all your exten in dialables are like this exten=_1. instead of this exten=>_1. |
| 05:05.56 | gandhijee | does that make a diff now??? |
| 05:05.58 | PoWeRKiLL | Your forgot the > |
| 05:06.21 | gandhijee | i know it didn't used to |
| 05:06.26 | gandhijee | like i said these file worked in 1.0.9 |
| 05:06.40 | SibrPhrek | Grubs - i haven't touched them...they are the base configs |
| 05:06.40 | *** join/#asterisk jeffik (n=jeffik@CPE0050babf4cd5-CM014350000760.cpe.net.cable.rogers.com) |
| 05:07.38 | Grubs | my mistake |
| 05:08.10 | SibrPhrek | Grubs - i'm soooo confused :/ |
| 05:08.57 | Grubs | ok - in musiconhold.conf under [default] si the path to the mp3s correct? |
| 05:09.12 | Grubs | "/var/lib/asterisk/mohmp3" |
| 05:09.18 | *** join/#asterisk jeffik (n=jeffik@CPE0050babf4cd5-CM014350000760.cpe.net.cable.rogers.com) |
| 05:09.22 | SibrPhrek | yes |
| 05:09.25 | gandhijee | powerkill |
| 05:09.43 | Grubs | and if you go there do you have mp3 files? |
| 05:09.50 | SibrPhrek | yes |
| 05:09.51 | gandhijee | i can't even dial us numbers |
| 05:09.56 | gandhijee | that is a broadvoice conf file too |
| 05:10.54 | Grubs | If you change "exten => 6601,1,Playback(demo-echotest)" then save, reload, and try again do you hear the echo test prompt? |
| 05:11.07 | PoWeRKiLL | gandhijee can you add in local_phones exten => 123,1,Playback(demo-echotest) |
| 05:11.23 | PoWeRKiLL | then extension reload and dial 123 |
| 05:11.25 | Grubs | (I mean change "WaitMusicOnHold(30)" to something else like "Playback(demo-echotest)") |
| 05:11.30 | gandhijee | ok |
| 05:11.40 | Grubs | just to test you are hitting the extension OK |
| 05:11.47 | gandhijee | i can dial my other sips from |
| 05:11.48 | gandhijee | phone |
| 05:12.10 | *** join/#asterisk tecnico_ (n=tecnico@user-24-236-120-2.knology.net) |
| 05:13.06 | SibrPhrek | grubs |
| 05:13.10 | SibrPhrek | i still get called failed 404 |
| 05:13.33 | Grubs | ok - so the problem has nothing to do with music on hold |
| 05:13.50 | Grubs | are there any test extensions you CAN dial successfully? |
| 05:14.02 | SibrPhrek | yeah voice amil |
| 05:14.03 | SibrPhrek | that works fine |
| 05:14.14 | Grubs | what do you dial |
| 05:14.19 | Grubs | 1234? |
| 05:14.25 | SibrPhrek | 250 |
| 05:14.55 | Grubs | Can you locate "exten => 250" in your extensions.conf? |
| 05:15.28 | gandhijee | PoWeRKiLL: i can call the echo demo |
| 05:16.00 | gandhijee | <PROTECTED> |
| 05:16.01 | gandhijee | <PROTECTED> |
| 05:16.01 | gandhijee | <PROTECTED> |
| 05:16.01 | gandhijee | <PROTECTED> |
| 05:16.01 | gandhijee | Jan 25 20:24:43 NOTICE[2062]: chan_iax2.c:7198 socket_read: Rejected connect attempt from 10.0.0.1, request '8046900359@local_phones' does not exist |
| 05:16.25 | gandhijee | is what i get when i try to call my cell via broadvoice |
| 05:16.29 | PoWeRKiLL | 8046900359@local_phones don't exist |
| 05:16.41 | SibrPhrek | yeah i see it |
| 05:16.42 | gandhijee | yeah i know its my cell phone |
| 05:16.43 | SibrPhrek | grub |
| 05:16.46 | SibrPhrek | i see that line |
| 05:16.58 | gandhijee | so how do i get it to dial out to the SIP's line? |
| 05:17.04 | gandhijee | i am using the broadvoice config files |
| 05:17.27 | PoWeRKiLL | if you dial 8 you are going to dial_fwd |
| 05:17.51 | Grubs | SibrPhrek: place the line to call your music on hold just below the 250 voicemail one - then save and reload |
| 05:18.08 | asteriskmonkey | how do you remove an rpm and tell it to ignore dependencies |
| 05:18.11 | Grubs | then at least we know you are in the right context |
| 05:18.17 | *** join/#asterisk Nivex (i=kjotte@user-0c8hq5r.cable.mindspring.com) |
| 05:18.38 | PoWeRKiLL | gandhijee what country are you in ? |
| 05:18.42 | SibrPhrek | when i do the reload i get a couple errors and notices |
| 05:18.53 | De_Mon | PoWeRKiLL I thought it was *8 |
| 05:19.08 | gandhijee | US |
| 05:19.14 | De_Mon | SibrPhrek you sould read them... |
| 05:19.31 | PoWeRKiLL | when you dial 804XXXX you have to dial 1804 |
| 05:19.32 | SibrPhrek | i'll get to them |
| 05:19.35 | gandhijee | PoWeRKiLL: i can't call my FWD number either though |
| 05:19.44 | PoWeRKiLL | try to dial 1804XXXXXXX |
| 05:19.57 | Grubs | asteriskmonkey: rpm -e xxxxx --nodeps |
| 05:19.59 | gandhijee | one sec i found something else |
| 05:20.08 | gandhijee | i think i might have it now |
| 05:20.09 | SibrPhrek | Grubs - when i call the extention |
| 05:20.12 | SibrPhrek | it goes thru |
| 05:20.14 | SibrPhrek | i don't hear anything |
| 05:20.16 | SibrPhrek | but this comes across |
| 05:20.18 | *** join/#asterisk Jenna (n=cherryRe@209.8.233.248) |
| 05:20.21 | SibrPhrek | Jan 26 00:22:46 WARNING[832]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x1805e00', 10 retries! |
| 05:20.22 | SibrPhrek | Jan 26 00:23:06 NOTICE[832]: res_musiconhold.c:507 monmp3thread: Request to schedule in the past?!?! |
| 05:20.31 | *** part/#asterisk Jenna (n=cherryRe@209.8.233.248) |
| 05:20.47 | gandhijee | nm that doesn't work either |
| 05:21.06 | De_Mon | SibrPhrek those are ok |
| 05:21.09 | *** join/#asterisk xygoat`` (n=xygoat@c-71-194-90-204.hsd1.il.comcast.net) |
| 05:21.11 | PoWeRKiLL | what are you getting ? |
| 05:21.16 | Grubs | closer then :) |
| 05:21.25 | gandhijee | samething |
| 05:21.40 | gandhijee | <number>@local_phones not avail |
| 05:21.48 | SibrPhrek | the call timer isn't moving either |
| 05:21.49 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 05:22.01 | xygoat`` | any of you guys ever debugged a phpagi script? |
| 05:22.05 | SibrPhrek | Jan 26 00:24:43 NOTICE[1153]: res_musiconhold.c:507 monmp3thread: Request to schedule in the past?!?! |
| 05:22.05 | SibrPhrek | Jan 26 00:24:52 NOTICE[1153]: rtp.c:330 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client |
| 05:22.19 | asteriskmonkey | Grubs: you can just update rpms too i fougn out :) rpm -Uvh |
| 05:22.20 | gandhijee | does anyone have broadvoice working with 1.2.x? |
| 05:22.29 | redder86 | xygoat``: yes |
| 05:22.30 | gandhijee | if so can they post thier conf file |
| 05:22.33 | gandhijee | plz |
| 05:22.34 | SibrPhrek | ugh it's late |
| 05:22.46 | SibrPhrek | Grubs - open for suggestions |
| 05:22.48 | Grubs | SibrPhrek: -= you have ztdummy or a digium card? |
| 05:22.51 | PoWeRKiLL | add in you local-phones exten => _1804X.,1,Dial(SIP/${EXTEN}@sip.broadvoice.com,30) |
| 05:22.59 | SibrPhrek | Grubs: - no idea |
| 05:23.28 | SibrPhrek | this OS X box is connected to a sip trunk connected to a coppercom connected to broadview.....i only have control of the OS X box |
| 05:24.09 | konfuzed | SibrPhrek: are you running asterisk on that osx box |
| 05:24.15 | SibrPhrek | yup :) |
| 05:24.19 | De_Mon | one more degree and you should be connected to kevin bacon |
| 05:24.32 | SibrPhrek | De_Mon har har |
| 05:25.08 | konfuzed | then its just like every other base install of asterisk and its your config on the osx box that determines ztdummy is in use or a real physical card |
| 05:25.33 | gandhijee | PoWeRKiLL: so now i get this |
| 05:25.35 | gandhijee | <PROTECTED> |
| 05:25.35 | gandhijee | <PROTECTED> |
| 05:25.35 | gandhijee | Jan 25 20:34:05 NOTICE[2151]: chan_sip.c:9516 handle_response_invite: Failed to authenticate on INVITE to '"Anish Patel" <sip:7038320122@sip.broadvoice.com>;tag=as01c3168e' |
| 05:25.35 | gandhijee | <PROTECTED> |
| 05:25.35 | gandhijee | <PROTECTED> |
| 05:25.37 | gandhijee | <PROTECTED> |
| 05:25.58 | *** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 05:26.01 | bsdfreak | heh |
| 05:26.11 | xygoat`` | redder86: is there an easy way to do it - or just brute it? i am getting a constant return 0 in the cli |
| 05:26.13 | asteriskmonkey | mm i have libtiff install and i get this when i try and config spandsp now.. Can't build without libtiff |
| 05:26.22 | *** join/#asterisk pifiu-laptop (n=someone@c-65-34-166-146.hsd1.fl.comcast.net) |
| 05:26.25 | Grubs | SibrPhrek - I'd probbaly abandon format_mp3 and try rawplayer - instructions here: http://www.asteriskguru.com/tutorials/request_schedule_past.html |
| 05:26.30 | *** join/#asterisk dasuberdavid (n=dasuberd@gateway.digium.com) |
| 05:26.33 | konfuzed | SibrPhrek: only you can determine if you asterisk on osx is using ztdummy or not |
| 05:27.11 | Grubs | asteriskmonkey: thats what I got - which is sy I removed the rpm and isntalled tiff-3.8.0 from source with the ./configure --prefix=/usr |
| 05:27.47 | PoWeRKiLL | gandhijee now it's better |
| 05:28.02 | asteriskmonkey | i tired to isntall tiff from source its broken for me when i complile it |
| 05:28.06 | asteriskmonkey | im using centos |
| 05:28.08 | gandhijee | ?? |
| 05:28.12 | gandhijee | what do you mean better? |
| 05:28.24 | gandhijee | is broadvoice fucked up? |
| 05:28.34 | PoWeRKiLL | gandhijee now the problem is in your sip.conf but I don't know if you configure right the parameter of broadvoice please check on the wiki |
| 05:28.42 | gandhijee | ok |
| 05:28.46 | Grubs | :( |
| 05:28.52 | gandhijee | but then why can't i dail any FWD numbers? |
| 05:29.27 | *** join/#asterisk twisted[mobile] (n=twisted@24.96.140.230) |
| 05:29.39 | PoWeRKiLL | Dial a FWD number and paste the error |
| 05:30.00 | SibrPhrek | Grubs |
| 05:30.06 | SibrPhrek | normal install of raw player doesn't work on OS X |
| 05:30.09 | SibrPhrek | will play with it tommorrow |
| 05:30.11 | SibrPhrek | need sleep |
| 05:30.43 | gandhijee | http://pastebin.ca/38633 |
| 05:30.53 | Grubs | sleep well |
| 05:31.21 | *** join/#asterisk tecnico (n=tecnico@user-24-236-120-2.knology.net) |
| 05:32.09 | pobre | q |
| 05:33.33 | Grubs | asteriskmonkey - this is how I installed spandsp. http://pastebin.com/523554 If you cant get tiff installed perhaps thats telling you thats a problem you need to solve? |
| 05:33.55 | *** join/#asterisk r0d3nt|m (i=r0d3nt@tinfoilhat.net) |
| 05:36.14 | PoWeRKiLL | gandhijee add exten => 8855555,1,Dial(IAX2/${FWDNUMBER}:${FWDPASSWORD}@iax2.fwdnet.net/${EXTEN:1},60,r) |
| 05:36.36 | PoWeRKiLL | btw why did you remove the first 3 number ? |
| 05:36.59 | gandhijee | first 3 number? |
| 05:37.10 | gandhijee | i thought exten:1 got the first number!? |
| 05:37.57 | *** join/#asterisk tecnico_ (n=tecnico@user-24-236-120-2.knology.net) |
| 05:38.13 | gandhijee | ok so i fixed the FWD problems |
| 05:38.14 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@84.205.154.241) |
| 05:38.45 | gandhijee | now i just need a lil help with BV... anyone know where i can go? |
| 05:40.08 | justinu | grubs: where in the world are you from? |
| 05:41.07 | *** part/#asterisk Grubs (n=Miranda@c211-28-119-169.eburwd3.vic.optusnet.com.au) |
| 05:45.09 | asteriskmonkey | ah well cant stay up anymore thanks for help gusy ttyl |
| 05:45.43 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@84.205.154.241) |
| 05:46.38 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@84.205.154.241) |
| 05:49.26 | *** join/#asterisk jeffgus (n=jeffgus@alpha.zimage.com) |
| 05:52.00 | SkramX | When was asterisk first released? |
| 05:52.08 | gandhijee | like in 2001 |
| 05:52.20 | gandhijee | i remember mark talking about it at phreaknic |
| 05:52.40 | *** join/#asterisk xtr (i=01928375@S0106000c41ed11e1.vf.shawcable.net) |
| 05:52.52 | SkramX | Coo Coo |
| 05:52.57 | SkramX | anyone from gNumber around? |
| 05:53.45 | *** join/#asterisk santiago (n=santiago@208.195.215.222) |
| 05:54.01 | SkramX | I was talking to ctoole a couple weeks ago. |
| 05:56.35 | *** join/#asterisk Dibbler_ (n=Dibbler@snaddy.plus.com) |
| 05:57.07 | gandhijee | sweet, i finally got the shit working |
| 05:57.43 | ronaldl79 | Gotta love it when you forget stuff....like my forgetting how I installed ztdummy...and got it working on an OHCI USB controller...I sure as hell don't remember using zaprtc!!! :P |
| 06:07.06 | ronaldl79 | Anyone around? |
| 06:08.01 | ronaldl79 | Here's food for thought: Any time I modprobe ztdummy, the audio is severely affected....lots of choppy audio....but I can hear enough to know that I'm able to join a conference..hmmm. |
| 06:08.01 | ptiggerdine_ | kinda |
| 06:08.12 | FuriousGeorge | anyone use fop? |
| 06:08.16 | FuriousGeorge | or apache? |
| 06:20.32 | *** join/#asterisk argos73 (i=1000@jason.argos.org) |
| 06:21.59 | argos73 | upgraded to 1.2.2 (from 1.0.9), copied all the config files from orig to 122... sip<>zap calls dial, but audio rcvd on the phone is muted almost immediately... any ideas? |
| 06:22.36 | reza | ok |
| 06:22.53 | reza | argos - see the topic |
| 06:23.02 | reza | furiousgeorge - yes, apache |
| 06:23.15 | reza | 1.2.2 has a serious bug (no audio) upgrade to 1.2.3 |
| 06:23.22 | argos73 | reza - duh... :) tnx |
| 06:23.22 | *** join/#asterisk zzxxcc (n=zzxxcc@221.232.10.225) |
| 06:23.42 | reza | np |
| 06:31.00 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 06:32.12 | argos73 | reza: that fixed it |
| 06:34.15 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 06:35.28 | reza | argos - go figure ;) |
| 06:35.59 | reza | anyone here have any experience w/ a fax -> FXS -> * -> FXO -> phone line? |
| 06:36.02 | reza | and visa-versa? |
| 06:38.06 | argos73 | reza: somewhat... |
| 06:38.23 | argos73 | running t1 - channel bank - * - channel bank - faxmodem |
| 06:41.11 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 06:44.04 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
| 06:44.40 | *** join/#asterisk g0mb0 (n=test@202.179.0.164) |
| 06:46.59 | *** join/#asterisk festr_ (n=festr@ns.regnet.cz) |
| 06:47.04 | *** join/#asterisk dokhench (n=dochench@adsl-065-080-180-134.sip.bna.bellsouth.net) |
| 06:48.50 | *** join/#asterisk Mark_Halverson (n=mhlvrs@67-139-119-152.dsl1.pco.ca.frontiernet.net) |
| 06:50.50 | Mark_Halverson | ok here's a weird one - i can dial on my Polycom SIP phone.. and the call goes through - when I try any windows IAX client - it falls.. * is coming back saying: |
| 06:50.55 | Mark_Halverson | Jan 25 22:50:20 NOTICE[24678]: chan_iax2.c:6769 socket_read: Rejected connect attempt from 67.139.119.152, who was trying to reach '18006396111@' |
| 06:51.05 | Mark_Halverson | where the hell is it getting the @ from????? |
| 06:54.25 | litecode | Mark_Halverson, maybe you need a default context for that client? (shot int eh dark) |
| 06:54.42 | argos73 | just what I was thinking... |
| 06:55.51 | litecode | argos73, i was scerwing with the same problem you had last night. |
| 06:55.57 | litecode | argos, i'm glad to see your post |
| 06:56.00 | litecode | heh |
| 06:56.08 | litecode | (upgrading now) |
| 06:56.14 | argos73 | he |
| 06:56.16 | argos73 | hehe |
| 06:57.52 | argos73 | fortunately, the bug didn't affect the zap<>zap calls that are the main job for this ast box |
| 06:59.37 | litecode | my problem was sip-to-sip |
| 06:59.44 | litecode | i screwed with my firewall for hours. |
| 06:59.59 | litecode | then i realized.. i get sound for half a second. |
| 07:00.04 | litecode | because i was cursing into the phone. |
| 07:00.19 | litecode | who says getting angry never solved anything. |
| 07:00.37 | argos73 | sounds familiar... :) |
| 07:02.23 | litecode | it tripped me up for a whole day, and this is going to be going online as a pbx host :P |
| 07:02.36 | litecode | can't wait to be using RealTimeS |
| 07:02.59 | argos73 | funny thing (sorta) is that I'm using asterisk to convince the company prez that it's time to switch out the phone system... oops... |
| 07:03.13 | litecode | hahahahahahahaha |
| 07:03.36 | litecode | open source = irony |
| 07:04.20 | mog_home | ? |
| 07:04.21 | argos73 | tomorrow I can march in and say "the problem's fixed... if Microsloth wrote this, we'd still be waiting for them to admit there even was a problem.." |
| 07:04.44 | litecode | yah. |
| 07:07.15 | *** join/#asterisk watchy (n=watchy@h124.184.255.206.cable.cmdn.cablelynx.com) |
| 07:07.49 | watchy | is it possible to put a call on hold and beable to see the call on hold from a 7960? |
| 07:09.04 | argos73 | watchy: you mean putting the call on hold from that 7960, or a different phone? |
| 07:09.23 | watchy | say someone calls i put them on hold with my 7960 |
| 07:09.35 | watchy | how do i make it show on all my 7960s theres a call on hold |
| 07:10.48 | argos73 | not sure you can, at least in the traditional sense. |
| 07:11.00 | watchy | hrm |
| 07:11.08 | watchy | well you can do it on other phone systems cant you? |
| 07:11.22 | argos73 | recall something on voip-info that makes me think it's doable in a different way |
| 07:11.34 | watchy | hrm |
| 07:11.47 | watchy | so your not gonna beable to tell what lines are in use etc |
| 07:12.08 | argos73 | depends on the phone system... |
| 07:12.23 | X-Rob | watchy, you want a key system, not a pabx. |
| 07:12.28 | X-Rob | asterisk is a key system. |
| 07:12.37 | *** part/#asterisk kiswanto (n=kiswanto@222.124.24.61) |
| 07:12.39 | X-Rob | Snom's 4S can emulate a key system, though. But you need Snom phones. |
| 07:12.39 | watchy | pabx? |
| 07:12.52 | watchy | so i want a key system |
| 07:12.54 | X-Rob | uh. |
| 07:12.57 | X-Rob | asterisk is _not_ a key system |
| 07:13.00 | watchy | oh |
| 07:13.07 | watchy | yea you just ocnfused me baddly |
| 07:13.12 | argos73 | english is goofy that way. |
| 07:13.14 | X-Rob | sorry |
| 07:13.15 | X-Rob | 8) |
| 07:13.24 | watchy | so if someones on hold |
| 07:13.27 | watchy | are they just fucked? |
| 07:13.40 | rabelais | watchy: there are ways to do that, it's handled through the call manager api, it just so happens that the cisco phones don't tie into asterisk's call manager api to check the status (obviously...) |
| 07:13.41 | litecode | yah, rob.. +10 on content, -1 on delivery :) |
| 07:14.00 | watchy | rabelais: do any phones? |
| 07:14.19 | rabelais | watchy: maybe some softphones....but no hardware phones that I know of |
| 07:14.20 | X-Rob | watchy, you can use parking. |
| 07:14.24 | X-Rob | but you can't get lights for it |
| 07:14.28 | X-Rob | (not yet, anyway) |
| 07:14.53 | watchy | hrm |
| 07:14.57 | watchy | so hows it work |
| 07:15.01 | watchy | you put them on hold and? |
| 07:15.10 | X-Rob | you park them |
| 07:15.16 | X-Rob | ~parking |
| 07:15.17 | jbot | app_valetparking is probably http://www.bkw.org/archives/000291.html |
| 07:15.25 | X-Rob | hrm. no that's not it |
| 07:15.27 | argos73 | transfer them to a special park xtn... after a certain amount of time, they bounce back to you |
| 07:15.38 | watchy | oh i see |
| 07:15.40 | litecode | "parking in rear" |
| 07:15.43 | litecode | sorry. couldn't resist. |
| 07:15.44 | X-Rob | :q! |
| 07:15.48 | watchy | so it puts them on an extension |
| 07:15.50 | watchy | you dial it |
| 07:15.53 | argos73 | dial a special "pick up this parking slot" number from any phone to retrieve the call |
| 07:15.55 | watchy | from another fone |
| 07:15.56 | reza | anyone play with the gxp-2000 grandstream hardphone |
| 07:16.04 | watchy | and you get them back |
| 07:16.14 | argos73 | basically |
| 07:16.33 | watchy | hrm |
| 07:16.37 | litecode | watchy: example: hit # dial 700, asterisk says "CAll parked at ....." it then hangs up on you. You go to anoter phone, hit that number and walla. |
| 07:16.46 | watchy | yea |
| 07:16.51 | watchy | you hit # 701 or whatever |
| 07:16.57 | watchy | and the call is picked back up |
| 07:17.05 | watchy | i mean just 701 ignore the # |
| 07:17.09 | litecode | minus the # |
| 07:17.09 | argos73 | yup |
| 07:17.10 | litage | after running ``asterisk -c'', how do you quit asterisk but leave it running? |
| 07:17.13 | X-Rob | reza, whats your GXP question? |
| 07:17.13 | litecode | yah |
| 07:17.14 | watchy | hrm |
| 07:17.22 | reza | is it a good phone? |
| 07:17.26 | X-Rob | Hurm. |
| 07:17.26 | reza | does it play well w/ * |
| 07:17.38 | wasim | litage: you don't |
| 07:17.41 | X-Rob | Short answer, yes. Long answer 'Not right now, but it's getting much better' |
| 07:17.41 | watchy | it kinda sucks you cant tell if someones on the phone or not |
| 07:17.47 | X-Rob | it plays with * very well |
| 07:18.00 | X-Rob | just the firmware reeks of arse. |
| 07:18.04 | litage | wasim: really? how come? |
| 07:18.07 | reza | i'm thinking of doing 3 normal phones plugged into fxs's (tdm400) and one ip phone for the receptionist |
| 07:18.12 | watchy | you think ventually youll beable to tell if people are on the phone or not? |
| 07:18.20 | wasim | litage: becuase thats console mode, you can CTRL-Z and then bg |
| 07:18.26 | X-Rob | reza, if you've got the bucks, go the Snom 360s |
| 07:18.27 | X-Rob | they look horn |
| 07:18.37 | reza | how much? |
| 07:18.38 | X-Rob | they do BLF (so you can see if a phone is in use) |
| 07:18.45 | X-Rob | FIIK, are you in my country? |
| 07:18.47 | wasim | litage: but either run it in screen OR use safe_asterisk and then attach with -rx |
| 07:19.00 | reza | ultimatly ease of use is important as receptionist = clueless |
| 07:19.05 | argos73 | watchy: i imagine it'll be worked in eventually |
| 07:19.08 | reza | and it should support 3 or 4 lines |
| 07:19.15 | litecode | reza, isn't that understood? |
| 07:19.27 | X-Rob | watchy, BLF (eg, if someone's on the phone or not) works with the GXP2000's, SNom and (apparently) Polycom phones. |
| 07:19.36 | X-Rob | call pickup works on the GXP and Snom. |
| 07:19.43 | watchy | whats BLF? |
| 07:19.55 | X-Rob | (eg, extension 200 is ringing, the light flashes, you push it and you're answering that call) |
| 07:19.57 | argos73 | busy lamp field |
| 07:20.02 | watchy | oh |
| 07:20.05 | wasim | its like MILF but for telephony |
| 07:20.12 | watchy | haha |
| 07:20.13 | argos73 | ooo... bad |
| 07:20.14 | litage | wasim: i'll look into safe_asterisk |
| 07:20.17 | X-Rob | IPPILF? |
| 07:20.21 | watchy | is a polycom 501 any good? |
| 07:20.57 | *** join/#asterisk mattwj2005 (n=Matt@dialup-4.254.84.69.Dial1.Chicago1.Level3.net) |
| 07:21.25 | reza | x-rbo - $175 for snom 320 |
| 07:21.27 | *** join/#asterisk yxa (n=diablo@58.185.90.98) |
| 07:21.28 | argos73 | the 7960s are capable of doing what you want, just not with asterisk right now |
| 07:21.59 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
| 07:22.09 | reza | looks the same, what's the diff w/ snom 320 and 360? |
| 07:22.28 | reza | $200 for 360 @ some atacomm.com... hmm |
| 07:24.10 | [av]bani | 360 does xml, has bitmap lcd |
| 07:24.25 | [av]bani | http://bani.anime.net/phonez/ |
| 07:24.32 | reza | why do i need xml and what's the bitmap for? pr0n on the display? |
| 07:24.40 | [av]bani | yes exactly |
| 07:24.47 | [av]bani | pr0n only |
| 07:26.17 | wasim | what square nipples you have, my dear ... |
| 07:26.21 | *** join/#asterisk mattwj2005 (n=Matt@dialup-4.254.84.69.Dial1.Chicago1.Level3.net) |
| 07:27.11 | reza | so xml for configuration? |
| 07:27.18 | [av]bani | no |
| 07:27.38 | [av]bani | http://snom.com/wiki/index.php/Xmlobjects |
| 07:27.41 | yxa | might zap channels not detecting hangups possibly be caused by faulty lines? |
| 07:27.56 | reza | i was just at that page |
| 07:27.59 | reza | and can't figure out a thing from it |
| 07:28.10 | wasim | yxa: yes |
| 07:28.18 | [av]bani | you can write your own stuff to display on teh screen |
| 07:28.23 | argos73 | reza: i use xml with my ciscos to display all sorts of info from our production equipment on the phones... client information, status of various pieces of equipment, etc.. |
| 07:28.54 | reza | that sounds cool, though i cant possibly see why the receptionist would need to know anything via her phone |
| 07:28.55 | reza | hmm |
| 07:29.04 | [av]bani | salespeople |
| 07:29.06 | [av]bani | tech support |
| 07:29.06 | reza | however, i wouldnt mind one of those phones |
| 07:29.06 | [av]bani | etc |
| 07:29.14 | yxa | wasim coz i really don't know what to try anymore. its not getting me anywhere |
| 07:29.41 | wasim | its just a matter of getting your indications.conf right and play with zapata.conf |
| 07:29.42 | reza | http://www.snom.com/minibrowser/snombox.bmp <- what's that supposed to be? |
| 07:29.45 | argos73 | also have our client phone directory (maintained in a sql database) available on the phones |
| 07:29.59 | [av]bani | pr0n i think |
| 07:30.03 | [av]bani | like you said |
| 07:30.07 | reza | why dont they use html? why xml? |
| 07:30.15 | yxa | wasim indications are right. |
| 07:30.26 | watchy | whats the point of a 7960g having more then 1lines |
| 07:31.05 | reza | other than the display/xml, is there any other reason to go snom over gxp-2000? The price isn't that big of a deal |
| 07:31.36 | argos73 | watchy: I have a few 7960s with "normal" calls coming in on one line button, and calls to our "employees only" number coming on another... |
| 07:31.58 | [av]bani | reza: http://www.o2m8.com/modules.php?name=News&file=article&sid=25 |
| 07:32.02 | [av]bani | you can do stuff like that |
| 07:32.30 | [av]bani | if you dont care, then shrug.... dont buy a 360 |
| 07:33.19 | reza | hmm |
| 07:33.34 | reza | again, that's cool for me, but i dont think it would be usefull for them |
| 07:33.53 | [av]bani | what does 'them' need? |
| 07:33.57 | reza | what's better, astra or snom |
| 07:34.07 | [av]bani | better depends on your parameters |
| 07:34.14 | reza | 3-4 lines, able to transfer line to other extension, make outgoing calls... about it |
| 07:34.25 | [av]bani | then a $80 chinese POS will do |
| 07:34.37 | reza | aka the 2000 |
| 07:34.49 | wasim | even $40 POS |
| 07:34.50 | [av]bani | no, crappier |
| 07:34.50 | reza | down to $79 it seems |
| 07:35.00 | wasim | which are pretty good |
| 07:35.04 | wasim | PA168 |
| 07:35.05 | [av]bani | lemme find a really crappy one with only those functions and nothing else |
| 07:35.07 | [av]bani | not even a screen |
| 07:35.10 | [av]bani | or speakerphone |
| 07:35.19 | [av]bani | im sure i can find a $10 phone :) |
| 07:35.25 | wasim | especially since it has IAX |
| 07:35.32 | wasim | [av]bani: in 5 million qty you can |
| 07:35.43 | argos73 | heh - my dlink 140 speakerphone's so prone to feedback, it might as well not exist... |
| 07:35.55 | wasim | upfront payment, delivery over 3 years |
| 07:36.09 | hmodes | hrmm |
| 07:36.17 | [av]bani | http://www.voipsupply.com/index.php?cPath=95_105 |
| 07:36.22 | hmodes | does anyone know if a moderated meetme does any unnecessary mixing? |
| 07:36.30 | reza | speakerphone isn't bad - and screen for caller id isn't bad. |
| 07:36.43 | hmodes | e.g. would an $x channel moderated meetme use significantly more cpu then an $x channel MoH? |
| 07:36.50 | reza | $80 is totally fine -- though i might swap out some of the analog phones with ip phones if you find some good cheap ones |
| 07:37.16 | [av]bani | http://www.voipsupply.com/product_info.php?products_id=39 |
| 07:37.29 | [av]bani | ther eyou go, will do transfer, do sip but only just |
| 07:37.33 | [av]bani | make outgoing calls |
| 07:37.50 | [av]bani | construction is just above cardboard box quality |
| 07:38.08 | wasim | atcom or yntx |
| 07:38.16 | reza | for $58, i would rather pay an extra $20 for the 2000 |
| 07:38.16 | wasim | atcom preferably |
| 07:38.20 | [av]bani | hell, lets try ebay |
| 07:39.19 | [av]bani | theres some real garbage there :) |
| 07:39.31 | watchy | i have to put in a phone system at an office |
| 07:39.33 | [av]bani | http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-30988135680.htm |
| 07:39.35 | [av]bani | $49!! |
| 07:39.38 | watchy | should i do polycoms or ciscos |
| 07:39.51 | wasim | snoms |
| 07:40.05 | reza | http://www.youtube.com/watch?v=Gi2CfuqcUGE |
| 07:40.05 | watchy | why snoms |
| 07:40.11 | reza | messed up |
| 07:40.43 | dokhench | i like the sipura 941. have a polycom 501 and a spa941. spa941 is a bit less expensive and doesn't have the ethernet bridge. |
| 07:41.04 | watchy | i need ethernet bridges |
| 07:41.14 | *** join/#asterisk Farrukh (n=linux@202.5.145.58) |
| 07:41.35 | [av]bani | http://cgi.ebay.com/IP-TALK-203-VOIP-IP-PHONE-ETHERNET-no-PC-needed_W0QQitemZ5857794145QQcategoryZ61841QQrdZ1QQcmdZViewItem |
| 07:41.39 | [av]bani | yay PA168 |
| 07:41.44 | [av]bani | real garbage territory :) |
| 07:42.28 | dokhench | what ya expect for 15 bucks tho. |
| 07:42.44 | dokhench | has a grandstream look to it. |
| 07:43.00 | ManxPower | I just send a rather nasty message to our helpdesk bitching because the Asterisk system with the analog cards is not being rebooted once a week. |
| 07:43.00 | *** join/#asterisk mattwj2006 (n=Matt@dialup-4.159.80.238.Dial1.Chicago1.Level3.net) |
| 07:43.03 | [av]bani | it has a less than grandstream look, if thats possible :) |
| 07:43.09 | dokhench | lol |
| 07:43.41 | [av]bani | http://cgi.ebay.com/New-IP-Phone-SIP-VOIP-Asterisk-compare-to-Grandstream_W0QQitemZ5858281783QQcategoryZ61840QQrdZ1QQcmdZViewItem |
| 07:43.45 | [av]bani | another pa168 |
| 07:44.00 | dokhench | crontab. ;) |
| 07:44.33 | bsdfreak | heh |
| 07:44.42 | bsdfreak | anyone familiar with the sipura 3k? |
| 07:44.45 | *** join/#asterisk felipex (n=dsfdsf@85-18-250-142.ip.fastwebnet.it) |
| 07:44.59 | ManxPower | dokhench, The MIS manager did not want to do that because if the reboot fails in the middle of the night we are screwed. |
| 07:45.06 | dokhench | ah |
| 07:46.11 | watchy | asterisk with analog cards needs to be rebooted once a week? |
| 07:46.17 | bsdfreak | no |
| 07:46.24 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
| 07:46.30 | wasim | only stable ones do |
| 07:47.04 | *** join/#asterisk kiwnix (n=egarcia@82.158.158.77) |
| 07:48.51 | *** join/#asterisk EriSan (n=erisan@151.8.109.117) |
| 07:49.07 | mattwj2006 | I have actually been quite impressed....I can send gsm and ilbc through my dial-up connection...and the sound isn't that bad :) |
| 07:49.22 | mattwj2006 | especially ilbc |
| 07:49.32 | *** join/#asterisk razu (n=razu@tln-kontor.norby.ee) |
| 07:49.37 | *** join/#asterisk Psykick (n=anon@remote.phoenixone.co.nz) |
| 07:49.39 | Psykick | hey guys |
| 07:50.17 | Psykick | I'm becoming more and more disappointed in asterisk the longer I use it |
| 07:50.27 | mattwj2006 | why is that? |
| 07:50.58 | Psykick | I've got 25 people trying to make calls concurrently and everyone is complaining |
| 07:51.12 | reza | get a bigger box/bandwidth |
| 07:51.12 | Psykick | people are complaining that no one can hear them |
| 07:51.18 | Psykick | or it's breaking up |
| 07:51.28 | denon | get a real network |
| 07:51.29 | denon | :) |
| 07:52.05 | Psykick | we've got a dual xeon 3ghz server running asterisk and we're using gigabit switches with QoS |
| 07:52.10 | mattwj2006 | you might be able to change codecs.....but you should lose quality to the sound.....I am assuming your using ulaw or alaw |
| 07:52.14 | Psykick | it's got 4 GB RAM in it as well |
| 07:52.31 | Psykick | tonight we tried out the G729 codec and still the same issues |
| 07:52.38 | *** join/#asterisk greendisease (n=jack@fedora/greendisease) |
| 07:52.40 | yxa | dsp.o(.text+0x809): In function `ast_dsp_process': |
| 07:52.40 | yxa | /usr/src/asterisk/dsp.c:963: undefined reference to `ast_dsp_busydetect' |
| 07:52.40 | yxa | collect2: ld returned 1 exit status |
| 07:52.40 | yxa | make: *** [asterisk] Error 1 |
| 07:52.47 | reza | what happens when one person uses it only? |
| 07:52.52 | Psykick | it's fine |
| 07:52.54 | [av]bani | g729 is expensive |
| 07:52.55 | Psykick | no problems |
| 07:52.58 | [av]bani | cpu wise |
| 07:52.59 | Psykick | tell me about it |
| 07:53.10 | yxa | i only get that when i specify -DBUSYDETECT_COMPARE_TONE_AND_SILENCE |
| 07:53.13 | Psykick | it ain't that bad for a dual xeon |
| 07:53.19 | *** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv) |
| 07:53.23 | yxa | am i missing any dep? |
| 07:53.24 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-127.claranet.co.uk) |
| 07:53.36 | reza | when do you start having problems (how many users)? |
| 07:53.39 | Psykick | the servers barely even breaking a sweat |
| 07:53.50 | Psykick | probably around 16 |
| 07:53.54 | [av]bani | Psykick: asterisk is at fault because your ethernet backbone cant handle it? |
| 07:53.58 | denon | grandstream phones? |
| 07:54.05 | Psykick | softphones |
| 07:54.07 | denon | linksys switches? |
| 07:54.12 | denon | oh, softphones heheh |
| 07:54.12 | Psykick | nope SMC |
| 07:54.32 | watchy | smc is kinda low end aint it |
| 07:54.36 | reza | hmm |
| 07:54.36 | Psykick | we've been using IAX extensions so far (recommended by the guy that installed it) |
| 07:54.42 | reza | mythtv is supposed to support sip |
| 07:54.43 | reza | hmph |
| 07:54.46 | denon | tigerswitches arent bad |
| 07:54.47 | Psykick | watchy: 550 each |
| 07:54.52 | *** join/#asterisk rajiv|work (n=rajiv@gentoo/developer/rajiv) |
| 07:54.55 | Psykick | oops |
| 07:54.56 | denon | their consumer switches arent the greatest, though |
| 07:55.01 | Psykick | I meant 2000 each |
| 07:55.13 | watchy | for $2000 each you shoulda got cisco |
| 07:55.26 | Psykick | cisco ain't that much better either |
| 07:55.28 | [av]bani | yeah, that will get you real nice cisco switch |
| 07:55.31 | denon | google uses SMC switches |
| 07:55.32 | Psykick | we've tried them already |
| 07:55.40 | denon | tigerswitches |
| 07:55.58 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 07:56.00 | denon | to handle terrabits of traffic |
| 07:56.01 | Psykick | tiger switches are what we have at the moment |
| 07:56.14 | argos73 | ethernet card up to par? it's amazing the difference between a 3com or intel and a netgear.. |
| 07:56.15 | [av]bani | yeah well, google runs a lot f hardware end users dont... |
| 07:56.28 | [av]bani | a friend of mine makes some stuff google uses... |
| 07:56.43 | Psykick | we're using CNet gigabit network cards |
| 07:56.54 | Psykick | that's not on all of them though |
| 07:56.58 | denon | Psykick: I wouldnt brag about that ... :) |
| 07:57.01 | Psykick | on some we're using Dlink as well |
| 07:57.05 | watchy | dewd |
| 07:57.08 | watchy | get a fucking intel |
| 07:57.10 | watchy | or 3com nic |
| 07:57.12 | denon | [av]bani: actually, kind of on the contrary .. google bigtable stuff excels on consumer hardware |
| 07:57.16 | Psykick | and on others 3com |
| 07:57.22 | watchy | whats the server have |
| 07:57.34 | Psykick | funny thing is the cnet nics work better than the 3com ones |
| 07:57.39 | [av]bani | denon: end users dont use the same hardware google does... |
| 07:57.40 | denon | Psykick: throw up a sniffer and watch what's happening .. also, I assume you dont have any interrupt conflicts? |
| 07:57.54 | Psykick | nope no interrupt conflicts |
| 07:57.57 | [av]bani | end users typically dont buy custom rackmount systems with custom built management systems |
| 07:58.15 | denon | Psykick: are you positive on that? |
| 07:58.16 | Psykick | server has intel gigabit nics |
| 07:58.25 | Psykick | denon: absolutely |
| 07:58.25 | *** part/#asterisk mattwj2006 (n=Matt@dialup-4.159.80.238.Dial1.Chicago1.Level3.net) |
| 07:58.47 | denon | Psykick: make sure the server and switch are AUTO-NEGing .. not staticly set to full or such |
| 07:58.49 | argos73 | and the smc is a true switch, not a hub, right? (rare these days, but anything's possible) |
| 07:58.53 | denon | SMC switches have a known issue there |
| 07:58.57 | Psykick | I've spent 200+ hours trying to fix this |
| 07:59.06 | *** join/#asterisk nico__ (n=nico@34.101.94.80.dslz1.libello.mc) |
| 07:59.11 | Psykick | denon: they are auto sensing |
| 07:59.16 | denon | argos73: I know the switches he's running, they're decent .. |
| 07:59.27 | [av]bani | Psykick: and you've determined it's the fault of asterisk |
| 07:59.29 | argos73 | k |
| 07:59.38 | denon | in fact, we run a lot of sip traffic on smc tigerswitches |
| 08:00.00 | Psykick | [av]bani: not entirely but can you think of anything that I haven't thought of or tried |
| 08:00.17 | denon | Psykick: im guessing you have somethin screwy in your kernel or such .. anything else running on that box? you better not have X installed... |
| 08:00.18 | [av]bani | Psykick: dunno, dont know your network is engineered |
| 08:00.18 | reza | psykick |
| 08:00.19 | reza | yeah |
| 08:00.22 | reza | do this |
| 08:00.26 | argos73 | i'd try shutting down asterisk and running some big FTPs (both directions) from the server to some other higher-end machine - see what kind of throughput you get |
| 08:00.44 | Psykick | argos73: done that |
| 08:00.58 | reza | get like 1 connection going, figure out the bandwidth, then flood the network with 15x that, then 20x that - see if you see any problems |
| 08:01.09 | [av]bani | Psykick: if you have phones doing reinvite, then asterisk is removed from the equation totally, since its no longer in the audio path |
| 08:01.12 | reza | with non-sip/aix traffic |
| 08:01.24 | denon | Psykick: no X right? |
| 08:01.35 | Psykick | I've got QoS enabled and set for highest priority to UDP 4569 |
| 08:01.42 | denon | Psykick: disable QoS |
| 08:02.00 | [av]bani | heh yeah, when you have a ton of devices all demanding qos, some switches dont like that |
| 08:02.03 | Psykick | denon: UDP 4569 is the only rule in the switches |
| 08:02.26 | [av]bani | what codec are oyu using (not g729) |
| 08:02.31 | denon | Psykick: we've determined its not HELPING .. so now we assume it could be hurting :) |
| 08:02.32 | [av]bani | and what phones |
| 08:03.06 | Psykick | [av]bani: the only other codecs we use is g726 |
| 08:03.31 | [av]bani | why not g711? |
| 08:03.34 | Psykick | but the softphone majority of the clients use doesn't support it |
| 08:03.43 | denon | Psykick: you never answered .. no X, right? |
| 08:04.06 | Psykick | denon: no X windows |
| 08:04.32 | Psykick | [av]bani: we were using g711 but the traffic was too much |
| 08:04.32 | denon | Psykick: you're saying your client doesnt support ulaw? dont think Ive ever seen one that didnt |
| 08:04.48 | Psykick | denon: doesn't support g726 |
| 08:04.58 | [av]bani | Psykick: traffic was too much? on what, 100mb ethernet? |
| 08:05.00 | denon | 20 users of ulaw traffic was too much for gig-e? |
| 08:05.06 | [av]bani | that sounds very very wrong |
| 08:05.09 | denon | nod |
| 08:05.11 | Psykick | exactly |
| 08:05.17 | denon | fire up a sniffer |
| 08:05.17 | [av]bani | its not asterisk at all, almost certainly |
| 08:05.22 | denon | you've got somethin else weird there |
| 08:05.24 | [av]bani | your backbone is jacked |
| 08:05.41 | Psykick | didn't know you could jack a backbone |
| 08:05.42 | Psykick | lol |
| 08:05.47 | wasim | 20 users on sip/ulaw is about 1.3 mbps |
| 08:05.53 | Psykick | can I jack yours |
| 08:05.56 | [av]bani | Psykick: use ttcp, get udp going between a bunch of nodes |
| 08:06.05 | *** join/#asterisk Feral_Kid (n=Feral@red-corp-200.56.96.178.telnor.net) |
| 08:06.05 | [av]bani | wasim: exactly. its nothing |
| 08:06.20 | wasim | and 1000 pps |
| 08:06.27 | *** join/#asterisk Joeymn (i=AJay@63.231.252.9) |
| 08:06.31 | Feral_Kid | Has anyone ever setup a trunk for iconnecthere? |
| 08:06.31 | Joeymn | IS there an easy way to make a bootable CD Backup of a newly configured Asterisk system for restoring at a later time? |
| 08:06.58 | denon | Feral_Kid: yes, hit the wiki |
| 08:07.00 | wasim | Joeymn: tar /etc/asterisk |
| 08:07.00 | Psykick | this is the confusing part |
| 08:07.06 | denon | think there's a detailed instruction there |
| 08:07.18 | Feral_Kid | Which wiki? |
| 08:07.20 | [av]bani | 1000pps is nothing |
| 08:07.21 | Psykick | I know that g711 should be fine but it just doesn't make sense |
| 08:07.23 | wasim | !docs |
| 08:07.25 | wasim | ~docs |
| 08:07.26 | jbot | from memory, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 08:07.30 | *** join/#asterisk konfuzed (n=KonfuzeD@H135.C72.B0.tor.eicat.ca) |
| 08:07.36 | wasim | 3rd one down ... that wiki |
| 08:07.39 | [av]bani | Psykick: something is very very bad with your ethernet backbone, look at that first |
| 08:07.45 | denon | Feral_Kid: http://asterisk.paperwork.com |
| 08:07.52 | Joeymn | wasim: i want to make a bootable recovery disk for the whole HD |
| 08:08.00 | [av]bani | or you have a crappy router in there somewhere like a low end cisco which cant handle the pps |
| 08:08.16 | wasim | Joeymn: thats got nothing to do with asterisk, readup on how to do that for linux |
| 08:09.09 | Psykick | I'm going to do what [av]bani suggested |
| 08:09.11 | [av]bani | Psykick: either that or your switch isnt negotiating 1000mb FD properly |
| 08:09.30 | [av]bani | you could try putting some random 10/100 nic in your asterisk pc |
| 08:09.46 | [av]bani | 100mb FD is way more than needed for even g711 and 20 clients |
| 08:10.05 | denon | or 10 half for that matter :) |
| 08:10.12 | Psykick | ok just ran that test |
| 08:10.21 | Feral_Kid | Ok, now that I have that out of my system... Has anyone successfully got iconnecthere working... Incidentally, I thought that I was getting some different information, but I have tried the one listed at voip-info (didn't work), the forums from asterisk@home (didn't)... |
| 08:10.59 | Feral_Kid | Ok, now that I have that out of my system... Has anyone successfully got iconnecthere working... Incidentally, I thought that I was getting some different information, but I have tried the one listed at voip-info (didn't work), the forums from asterisk@home (didn't)... |
| 08:11.25 | Psykick | [av]bani: what stat's do you want |
| 08:12.11 | [av]bani | Psykick: what test did you do? |
| 08:12.55 | *** join/#asterisk jhiver_ (n=jhiver@AStDenis-105-1-4-4.w193-253.abo.wanadoo.fr) |
| 08:13.00 | jhiver_ | hiya all |
| 08:13.02 | Psykick | did the simple pcattcp -u -t |
| 08:13.15 | *** join/#asterisk Bambr (n=Bambr@213-35-235-26-dsl.end.estpak.ee) |
| 08:13.16 | Psykick | and the pcattcp -u -r |
| 08:13.23 | Psykick | also did the same for tcp |
| 08:13.29 | [av]bani | do it between a couple of nodes at the same time |
| 08:13.53 | [av]bani | look at ifconfig stats for eth0 |
| 08:13.56 | Psykick | so you think I should try it on say 10+ clients? |
| 08:14.09 | [av]bani | on your asterisk box |
| 08:14.25 | [av]bani | whats it say for errors, dropped, overruns, frame, carrier and collisions |
| 08:15.10 | Psykick | <PROTECTED> |
| 08:15.10 | Psykick | <PROTECTED> |
| 08:15.26 | [av]bani | collisions? |
| 08:15.38 | Psykick | <PROTECTED> |
| 08:15.39 | Psykick | <PROTECTED> |
| 08:15.39 | Psykick | <PROTECTED> |
| 08:15.40 | jhiver | hey anybody is any good with SER in there? I have a problem with getting ser-acc-db-module to work properly |
| 08:15.43 | *** join/#asterisk burton (i=mimx@w201.ljudmila.org) |
| 08:15.44 | [av]bani | good |
| 08:15.57 | [av]bani | so its FD, and no cable problems to the switch at least |
| 08:16.08 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 08:16.22 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:16.28 | Psykick | you know .... now that you guys mention it |
| 08:16.32 | burton | hello, anyone has some info about any wifi iax phone? |
| 08:16.33 | [av]bani | try -u -t between asterisk and a node and see what speed you get |
| 08:16.49 | [av]bani | it should get max easily |
| 08:17.31 | Psykick | [av]bani: is there a linux executable somewhere for pcattp |
| 08:17.54 | [av]bani | ttcp is included in most distros |
| 08:18.04 | Psykick | ok lemme just try that |
| 08:18.40 | [av]bani | might also want to look at your switch vendors support site for any errata related to qos :) |
| 08:19.09 | Psykick | will do |
| 08:19.42 | [av]bani | as someone mentioned before, try turning qos off on your switches totally.. since it obviously isnt helping, it cant hurt either :) |
| 08:20.14 | denon | heh |
| 08:20.19 | Psykick | tarue |
| 08:20.39 | Psykick | hmm ... just seems to be sitting there doing nada |
| 08:20.58 | [av]bani | it runs for a while then finishes, it doesnt give a progress meter or anything |
| 08:21.12 | Psykick | been running for quite awhile so far |
| 08:21.19 | Psykick | think I'll go have a cigi and come back |
| 08:21.22 | Psykick | bbiab |
| 08:21.27 | [av]bani | ttcp -s -r on the receiving end |
| 08:21.38 | [av]bani | ttcp -s -t (receiver ip address) on the sending end |
| 08:22.01 | [av]bani | for 100mb ether it should finish pretty quickly |
| 08:22.22 | Psykick | yeah the receiving end got a shitload of garbage |
| 08:23.06 | *** join/#asterisk Libila (n=vye@ip68-8-174-154.sd.sd.cox.net) |
| 08:23.28 | Libila | I have a WRT54GS router, does that support SIP? |
| 08:23.32 | Psykick | ttcp-t: buflen=8192, nbuf=2048, align=16384/0, port=5001 udp -> 192.168.17.22 |
| 08:23.32 | Psykick | ttcp-t: socket |
| 08:23.32 | Psykick | ttcp-t: 16777216 bytes in 0.16 real seconds = 100630.79 KB/sec +++ |
| 08:23.32 | Psykick | ttcp-t: 2054 I/O calls, msec/call = 0.08, calls/sec = 12615.70 |
| 08:23.32 | Psykick | ttcp-t: 0.0user 0.0sys 0:00real 6% 0i+0d 0maxrss 1+2pf 0+0csw |
| 08:23.46 | [av]bani | thats plenty fast |
| 08:23.52 | burton | hello, anyone has some info about any wifi iax phone? |
| 08:24.01 | [av]bani | well |
| 08:24.11 | [av]bani | on the receiving end do ttcp -s -r > /dev/null |
| 08:24.12 | [av]bani | heh |
| 08:24.15 | Psykick | burton: don't know of any IAX wifi phones |
| 08:24.18 | Psykick | only sip based |
| 08:24.33 | [av]bani | and on the sending end do ttcp -s -t -n 262144 |
| 08:24.45 | [av]bani | makes it run a bit longer |
| 08:25.52 | Psykick | ok well while that's running I'll go have a cigi |
| 08:27.34 | *** join/#asterisk lorinc (n=ang@caracas-0534.adsl.interware.hu) |
| 08:30.32 | reza | damn, fucking directv pooped out and my tivo recorded an hour of an error message rather than lost |
| 08:30.34 | burton | psykick: tnx, but sip based is working thru nat only with SIP-proxy? am i right? |
| 08:30.40 | reza | what's a good bittorrent site for tv shows? |
| 08:32.28 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 08:34.13 | Psykick | reza: this is probably not the best channel to ask |
| 08:34.34 | Psykick | [av]bani: it's still running |
| 08:34.43 | xygoat`` | hi all.. |
| 08:35.23 | xygoat`` | is there a way to instantiate a call from the shell? |
| 08:35.46 | xygoat`` | i want to set up a cron job to call my cell and tell me the load on my server |
| 08:36.28 | argos73 | Psykick: should have been done by now, if i've been following this correctly... activity lights on the switch? |
| 08:36.33 | Psykick | xygoat``: you could create a php script that can do that using phpagi |
| 08:36.43 | Psykick | I'll just go check |
| 08:37.23 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 08:37.39 | *** join/#asterisk veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 08:38.11 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 08:38.16 | Psykick | [av]bani: traffic is still passing back and forth |
| 08:38.20 | argos73 | k |
| 08:39.15 | argos73 | (lights should be solidly "on", not flickering... assuming they're "on" when traffic is active) |
| 08:40.04 | xygoat`` | i have done some work with phpagi - but i am not really sure how to execute it and make it call.. i mean.. i can handle an incoming call like nobodies business - just outgoing.. |
| 08:40.21 | Psykick | just use Originate |
| 08:40.55 | Psykick | setup an extension that the server will use to place a call |
| 08:40.55 | xygoat`` | i see my problem. |
| 08:40.57 | xygoat`` | my bad |
| 08:41.13 | xygoat`` | i wasn't connecting to asterisk throught the asterisk mananager interface |
| 08:41.14 | xygoat`` | hah |
| 08:42.22 | argos73 | i need sleep.... |
| 08:42.57 | argos73 | Psykick: don't give up on ast - it's almost definitely a network issue... good luck |
| 08:42.58 | argos73 | cya |
| 08:43.07 | Psykick | cya |
| 08:44.06 | *** join/#asterisk dasuberdavid (n=dasuberd@gateway.digium.com) |
| 08:44.37 | *** join/#asterisk felipex (n=dsfdsf@85-18-250-142.ip.fastwebnet.it) |
| 08:46.40 | [av]bani | psykick: ^C the sender |
| 08:46.48 | [av]bani | the remote should report stats |
| 08:47.15 | Psykick | ttcp-r: buflen=8192, nbuf=262144, align=16384/0, port=5001 tcp |
| 08:47.15 | Psykick | ttcp-r: socket |
| 08:47.15 | Psykick | ttcp-r: accept from 192.168.17.252 |
| 08:47.15 | Psykick | ttcp-r: 766442608 bytes in 67.72 seconds = 11052.88 KB/sec +++ |
| 08:47.15 | Psykick | ttcp-r: 164780 I/O calls, msec/call = 0.42, calls/sec = 2433.33 |
| 08:47.32 | Psykick | I started it again |
| 08:47.44 | [av]bani | thats pretty much max from 100mb ethernet |
| 08:47.57 | [av]bani | so you can max it out at least between 2 peers |
| 08:48.01 | Psykick | I changed switches |
| 08:48.07 | *** join/#asterisk NirS (n=nirs@84.94.9.117.cable.012.net.il) |
| 08:48.22 | Psykick | was at least 10 times that on the other switches |
| 08:48.27 | [av]bani | 100630.79 is pretty damned good for gbe |
| 08:49.07 | Psykick | so I guess it's not a throughput issue |
| 08:49.13 | [av]bani | not between 2 peers |
| 08:49.23 | [av]bani | throw a bunch of simultaneous peers up |
| 08:50.04 | [av]bani | hmm.. ttcp really needs qos options |
| 08:52.33 | *** join/#asterisk tzafrir_laptop (n=tzafrir@82.166.242.248) |
| 08:52.36 | Psykick | ttcp on the server only accepts a connection from one of the peers |
| 08:53.12 | [av]bani | dont you have other linux servers |
| 08:53.30 | Psykick | of course I do |
| 08:53.55 | [av]bani | get a couple of em going, independent of your asterisk server |
| 08:53.58 | [av]bani | so the switches get bashed |
| 08:54.04 | [av]bani | then run some calls thru asterisk |
| 08:54.10 | Libila | anyone know how to reset the password on a Budge Tone 100? |
| 08:54.19 | Libila | doing a reset doesn't do it |
| 08:56.04 | Psykick | ttcp-r: buflen=8192, nbuf=262144, align=16384/0, port=5001 tcp |
| 08:56.04 | Psykick | ttcp-r: socket |
| 08:56.04 | Psykick | ttcp-r: accept from 192.168.17.251 |
| 08:56.04 | Psykick | ttcp-r: 801275584 bytes in 68.13 real seconds = 11484.84 KB/sec +++ |
| 08:56.04 | Psykick | ttcp-r: 528949 I/O calls, msec/call = 0.13, calls/sec = 7763.49 |
| 08:56.04 | Psykick | ttcp-r: 0.3user 2.2sys 1:08real 3% 0i+0d 0maxrss 0+2pf 0+0csw |
| 08:56.32 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 08:56.59 | [av]bani | -n 262144 to keep em going |
| 08:57.04 | [av]bani | while you make asterisk calls |
| 08:57.49 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 08:57.49 | Psykick | calls to extensions on the network ok? |
| 08:57.59 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 08:58.48 | [av]bani | yeah, you just want to keep the switches busy |
| 08:59.07 | [av]bani | is it extension-extension that gets problems? |
| 08:59.14 | [av]bani | do you have upstrea provider or something |
| 08:59.25 | Psykick | it's outbound calling |
| 08:59.29 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 08:59.54 | [av]bani | outbound calling to where? |
| 09:01.24 | Psykick | to places around the country |
| 09:01.31 | [av]bani | what do you use for outbound |
| 09:01.32 | Psykick | these are the stats and the calls were fine |
| 09:01.33 | Psykick | ttcp-r: buflen=8192, nbuf=262144, align=16384/0, port=5001 tcp |
| 09:01.33 | Psykick | ttcp-r: socket |
| 09:01.33 | Psykick | ttcp-r: accept from 192.168.17.251 |
| 09:01.33 | Psykick | ttcp-r: 2147483648 bytes in 182.60 real seconds = 11484.81 KB/sec +++ |
| 09:01.33 | Psykick | ttcp-r: 1422476 I/O calls, msec/call = 0.13, calls/sec = 7790.03 |
| 09:01.35 | Psykick | ttcp-r: 0.8user 5.7sys 3:02real 3% 0i+0d 0maxrss 0+2pf 0+0csw |
| 09:01.52 | Psykick | what do you mean what do we use for outbound |
| 09:01.58 | Psykick | we use softphones |
| 09:02.17 | [av]bani | what device/service do you use to send calls out to the PSTN |
| 09:02.32 | [av]bani | external FXO? PRI? vonage? |
| 09:02.39 | Psykick | we've got an te210p card that has a e1 pri |
| 09:03.00 | [av]bani | so its bridged calls sip->pri that give you stuttering |
| 09:03.09 | Psykick | we don't use sip |
| 09:03.11 | Psykick | we use IAX |
| 09:03.16 | [av]bani | well, bridged calls |
| 09:03.19 | Psykick | but yes |
| 09:03.23 | [av]bani | k |
| 09:03.38 | Psykick | throughput issue on our pri? |
| 09:03.45 | [av]bani | shouldnt be |
| 09:03.48 | Psykick | taking wild stabs in the dark here |
| 09:03.57 | [av]bani | breakdown at 25 channels? |
| 09:04.15 | Psykick | we only have 20 channels at the moment |
| 09:04.35 | Psykick | great ... now I feel like an idiot |
| 09:04.46 | [av]bani | eheheh? |
| 09:04.47 | Psykick | perhaps we should get more channels? |
| 09:05.22 | [av]bani | well, you should get busies if you try to establish more outgoing than you have channels |
| 09:05.24 | *** join/#asterisk dasuberdavid (n=dasuberd@gateway.digium.com) |
| 09:05.29 | [av]bani | unless you misconfigured the card... |
| 09:05.48 | [av]bani | or screwed up the dialplan |
| 09:06.13 | Psykick | that's the thing I didn't configure the card |
| 09:06.22 | Psykick | we had someone else do the configuration |
| 09:06.33 | *** join/#asterisk [bofh42] (n=bofh42@p5482A504.dip0.t-ipconnect.de) |
| 09:06.34 | Psykick | and by the sounds of things he's f**ked it |
| 09:06.38 | RoyK | what card is this? |
| 09:06.44 | Psykick | TE210p |
| 09:06.44 | [av]bani | <Psykick> we've got an te210p card that has a e1 pri |
| 09:06.48 | *** join/#asterisk dkoum33 (n=dk@adsl16-210static.access.acn.gr) |
| 09:06.57 | [av]bani | well, the dialplan might be f00ked too |
| 09:07.05 | RoyK | Psykick: there aren't too many ways to configure that card |
| 09:07.10 | [av]bani | ive had dialplans go into endless loops when i screwed them up |
| 09:07.11 | Psykick | that's always a possibilty |
| 09:07.29 | RoyK | Psykick: where're you located? |
| 09:07.29 | [av]bani | its possible you dont handle channels unavailable properly |
| 09:07.33 | Psykick | can you guys recommend anything to create custom dialplans |
| 09:07.38 | Psykick | RoyK: new zealand |
| 09:07.42 | RoyK | ok |
| 09:07.52 | RoyK | i know nothing about what telcos do down there :) |
| 09:07.54 | X-Rob | Ooh |
| 09:07.57 | X-Rob | That's me. |
| 09:08.02 | Psykick | [av]bani: anythings possible |
| 09:08.05 | X-Rob | Psykick, what's your problem? |
| 09:08.06 | [av]bani | Psykick: if its a te210p, its a digium, poke them for support |
| 09:08.10 | Psykick | X-Rob: where are you? |
| 09:08.14 | X-Rob | Queensland |
| 09:08.19 | [av]bani | ask them to look and see if it looks fooked for 20 channels |
| 09:08.28 | [av]bani | if they say its ok, then its probably your dialplan |
| 09:08.35 | Psykick | X-Rob: have the telepermitted the digium cards there yet? |
| 09:08.43 | X-Rob | yeah |
| 09:08.47 | Psykick | kewl |
| 09:08.55 | Psykick | shouldn't be too long before they've done the same here |
| 09:08.59 | X-Rob | beginning of last year for the tdm's, the pri ones have been done for years. |
| 09:09.07 | Psykick | only one telco has telepermitted them |
| 09:09.46 | Psykick | I'm not sure if we can .... the same guy that did the config supplied us the digium card |
| 09:09.55 | Psykick | and the server it's running on |
| 09:10.02 | [av]bani | well, poke him then |
| 09:10.03 | X-Rob | So what's the problem? |
| 09:10.04 | dkoum33 | hi all, just a quick question: is there any way to select codecs on the call prefix? |
| 09:10.10 | diLLec | I've got a problem on Asterisk 1.2.2 with a PRi and some cellphone numbers. Can anyone help ? |
| 09:10.14 | [av]bani | witha bayonet if needed |
| 09:10.22 | Psykick | X-Rob: I work in a call center that has at the moment 25 seats |
| 09:10.27 | [av]bani | diLLec: upgrade to 1.2.3 |
| 09:10.35 | diLLec | no - thats not the problem |
| 09:10.48 | diLLec | fixed it yesterday by adding the patch |
| 09:10.49 | Psykick | X-Rob: progressively things get worse and worse as more people make calls |
| 09:10.59 | diLLec | the problem exists only on some cellphones |
| 09:11.00 | X-Rob | what's 'worse and worse'? |
| 09:11.29 | X-Rob | diLLec, I suggest installing windows. |
| 09:11.34 | [av]bani | Psykick: make sure you dont have any NMI in /proc/interrupts |
| 09:11.36 | Psykick | X-Rob: people can't hear our staff ... our staff can't hear people ... staggered speech ... etc etc |
| 09:11.40 | X-Rob | (or, you know, you could help us out with more information of what the problem si) |
| 09:11.50 | X-Rob | Psykick, that's an overloaded machine or network |
| 09:11.56 | [av]bani | make sure dmesg doesnt say anything unusual |
| 09:12.03 | Psykick | nope no NMI |
| 09:13.08 | [av]bani | Psykick: do you have any problems at say 10 sessions? |
| 09:13.25 | Psykick | yes and no |
| 09:13.37 | Psykick | intermittent speech |
| 09:13.44 | Libila | I'm trying to restore the settings on my budge tone 100, says to go the the -- reset -- menu then enter in the mac address, I put in the mac right but then it says it will restart automatically I just sat here for a few minutes till the numbers disapeared. anyone have a hint? |
| 09:13.57 | [av]bani | Psykick: but internal calls are fine? |
| 09:14.08 | Psykick | yup |
| 09:14.16 | Psykick | err ... this is the first time I've noticed this |
| 09:14.20 | [av]bani | it sounds like a pri misconfig |
| 09:14.31 | Psykick | output from dmesg > Registered tone zone 0 (United States / North America) |
| 09:14.44 | [av]bani | heheh... is that correct for nz? |
| 09:14.50 | Psykick | shouldn't that say europe |
| 09:15.33 | Psykick | and I've just noticed ... looks like an interrupt conflict |
| 09:15.40 | Psykick | 2G: Got interrupt, status = 0000000c, GIS = 0080 |
| 09:15.40 | Psykick | 2G: Got interrupt, status = 0000000b, GIS = 0081 |
| 09:15.54 | Psykick | keeps looping the same status message |
| 09:15.59 | [av]bani | { 10, "nz", "New Zealand", { 400, 200, 400, 2000 } |
| 09:16.05 | [av]bani | looks like theres a new zealand tone zone... |
| 09:16.31 | [av]bani | cat /proc/interrupts |
| 09:16.44 | [av]bani | see what irq your card is on and what its sharing with |
| 09:16.52 | Psykick | <PROTECTED> |
| 09:16.53 | Psykick | <PROTECTED> |
| 09:16.53 | Psykick | <PROTECTED> |
| 09:16.53 | Psykick | <PROTECTED> |
| 09:16.53 | Psykick | <PROTECTED> |
| 09:16.53 | Psykick | <PROTECTED> |
| 09:16.55 | Psykick | <PROTECTED> |
| 09:16.57 | Psykick | <PROTECTED> |
| 09:16.59 | Psykick | <PROTECTED> |
| 09:17.01 | Psykick | NMI: 0 0 0 0 |
| 09:17.03 | Psykick | LOC: 354449058 354449299 354449062 354449310 |
| 09:17.05 | Psykick | ERR: 0 |
| 09:17.07 | Psykick | MIS: 0 |
| 09:17.09 | Psykick | sorry peeps |
| 09:17.11 | [av]bani | looks ok |
| 09:17.14 | Psykick | [av]bani: where can I change that NZ tone |
| 09:17.23 | [av]bani | dual HT xeon? |
| 09:17.51 | [av]bani | <PROTECTED> |
| 09:18.56 | Psykick | don't know why but the idiot that has installed the box had commented them out |
| 09:20.11 | Psykick | guys I gotta go |
| 09:20.24 | Psykick | I'm just leaving work now .... thanks for all your help so far |
| 09:20.35 | Psykick | it's 10:20 pm and I've been here since 8:00am |
| 09:20.45 | *** join/#asterisk Andrew_C (n=Andrew@58.6.118.227) |
| 09:20.45 | [av]bani | http://www.digium.com/index.php?menu=compatibility |
| 09:21.24 | Psykick | be back tomorrow to get more info |
| 09:21.29 | Psykick | thanks heaps |
| 09:21.33 | Psykick | it's much appreciated |
| 09:21.36 | Psykick | byeeeeee |
| 09:21.48 | Andrew_C | hey all |
| 09:22.33 | Andrew_C | any chance of some help with an asterisk problem? |
| 09:23.47 | *** join/#asterisk nurfe (n=rgff@h24-207-70-68.dlt.dccnet.com) |
| 09:24.25 | JunK-Y | Andrew_C: asking a question will be a good start. |
| 09:25.10 | Andrew_C | ok, trying to make a call between X-Lite softphone and a SPA-2000 ATA. Connection goes through but no audio either way |
| 09:25.31 | Andrew_C | most of the google results i've found are about NAT problems, but there's no NAT here |
| 09:26.31 | X-Rob | there's no firewall at all between the three devices? |
| 09:26.50 | Andrew_C | none at all |
| 09:26.57 | JunK-Y | Andrew_C: which * version? |
| 09:27.15 | X-Rob | I should be asking that first with 'no audio' problems, shouldn't I. Duh. |
| 09:27.18 | Andrew_C | server is at another site via a VPN, but all ports are open |
| 09:28.18 | Andrew_C | good q, hang on |
| 09:28.27 | JunK-Y | Andrew_C: we had issue yesterday with no audio, so ur answer coul has an impact. |
| 09:28.52 | Andrew_C | 1.2.2 |
| 09:28.56 | JunK-Y | thats it |
| 09:28.59 | JunK-Y | upgrade to 1.2.4 |
| 09:29.02 | JunK-Y | 1.2.3 |
| 09:29.03 | JunK-Y | :) |
| 09:29.08 | Andrew_C | you're kidding me |
| 09:29.17 | Andrew_C | damnit should've looked |
| 09:29.18 | JunK-Y | no, thats the bug from yesterday |
| 09:29.19 | Andrew_C | is there a bug? |
| 09:29.19 | JunK-Y | :) |
| 09:29.21 | JunK-Y | yes |
| 09:29.24 | JunK-Y | let me see the id |
| 09:29.28 | JunK-Y | 6319 iirc |
| 09:30.02 | oej | There are tons of good source code in mantis |
| 09:30.06 | Andrew_C | so 1.2.3 is the latest? |
| 09:30.12 | JunK-Y | Andrew_C: yes |
| 09:30.39 | oej | 1.2.3 is the greatest. |
| 09:30.45 | oej | 1.2.2 contains a serious bug |
| 09:31.03 | JunK-Y | oej: deja vu :) |
| 09:31.11 | Andrew_C | i should've read the topic when i joined hey |
| 09:31.13 | Andrew_C | i'll shut up now |
| 09:31.20 | oej | :-) |
| 09:31.33 | oej | JunK-Y: Up late in SF? |
| 09:31.40 | JunK-Y | 1:32 now |
| 09:31.56 | JunK-Y | whoch is like 4:32 in my real timezone. |
| 09:32.31 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
| 09:32.47 | *** join/#asterisk jozsab1 (n=jozsab1@86.125.91.54) |
| 09:33.32 | jozsab1 | How do i make/configure my server to suport 183 message (to send it) ? |
| 09:33.32 | oej | Question for Europeans: Where would you like Astricon Europe to take place? |
| 09:33.32 | JunK-Y | hey areski ! |
| 09:33.32 | oej | jozsab1: It does support it natively |
| 09:33.50 | areski | JunK-Y, Hey man |
| 09:33.53 | oej | jozsab1: If you want to generate one, run playback with the noanswer option before answering |
| 09:33.57 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 09:34.01 | jozsab1 | but level3 testcases show that i do not send them |
| 09:34.12 | areski | my god, I start to work that u still dont sleep :) |
| 09:34.27 | JunK-Y | areski: heheeh |
| 09:34.28 | oej | jozsab1: Test with playback and noanswer and you will see that you do send them over SIP |
| 09:34.29 | jozsab1 | thanks oej, i will try that |
| 09:35.08 | oej | If we get early audio in on a PRI or E1, we will send out a 183 with SDP by default |
| 09:35.10 | fenlander | oej: the UK :-) |
| 09:35.29 | oej | fenlander: It's so expensive for everyone outside of the Pound stirling area |
| 09:35.50 | zoa | eastern europe or the netherlands |
| 09:36.08 | zoa | so either 1) very cheap or 2) very central in europe |
| 09:36.11 | oej | zoa: I think Dave and I ruled out netherlands |
| 09:36.12 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 09:36.18 | zoa | brussels then :) |
| 09:36.29 | zoa | or somewhere in france |
| 09:36.38 | zoa | uk is across the water, makes it more difficult |
| 09:36.46 | oej | I would love to run it in Nice, but haven't found any cheap conference centers there |
| 09:36.51 | zoa | germany or luxemburg would also be ok |
| 09:37.00 | oej | What about Stockholm? |
| 09:37.14 | oej | It is summer then, so no snow and a lot of sunshine in the evenings |
| 09:37.24 | zoa | to us (belgians) the countries there sound very far away |
| 09:37.33 | zoa | and people will not come by car |
| 09:38.04 | oej | Come on, it's not that far away. And there's a bridge between denmark and Sweden, so you can drive |
| 09:38.12 | oej | We are connected to the mainland! |
| 09:38.13 | areski | I ve a great flat in Barcelona, if this can help we can do Astricon in my place |
| 09:38.26 | oej | That would be a huge flat! |
| 09:38.27 | areski | good thing, I wont have to travel |
| 09:38.30 | areski | :) |
| 09:38.30 | zoa | it sounds really far away |
| 09:38.49 | JunK-Y | lets do the party at zoa's place! :) |
| 09:38.52 | oej | zoa: I need to force you up here to discover all the beautiful girls, and then you will not object any more :-) |
| 09:39.01 | zoa | olle, its not for me, i would come anyway |
| 09:39.07 | zoa | im just thinking about how others would feel |
| 09:39.14 | oej | Did not check brussels last year, don't know about pricing levels in Belgium |
| 09:39.45 | oej | I hear you |
| 09:39.53 | oej | Any more suggestions on a location for Astricon Europe? |
| 09:39.56 | areski | you can probably get easily contact with ULB university and perhaps get a conference room there |
| 09:40.19 | areski | I mean for Brussel |
| 09:40.22 | [av]bani | oej: bulgaria |
| 09:40.24 | JunK-Y | time for bed now, see ya. |
| 09:40.30 | zoa | [av]bani are you from bulgaria ? |
| 09:40.31 | oej | 118 comments on the jitter buffer patch, that must be some sort of record |
| 09:40.36 | zoa | hehe :) |
| 09:40.36 | oej | Night, Junky |
| 09:40.43 | zoa | i think its ready to go in btw |
| 09:40.53 | [av]bani | zoa: no :) |
| 09:40.53 | zoa | not a single crash or problem so far with this latest patch |
| 09:40.55 | oej | Any comments from Mark/Kevin |
| 09:40.58 | zoa | nopez |
| 09:40.59 | areski | JunK-Y, please sleep more than 2 hours this time |
| 09:41.06 | oej | I heard that royk is running it in production, right |
| 09:41.08 | zoa | they dont seem interested |
| 09:41.13 | zoa | no, i dont think he is |
| 09:41.18 | zoa | he has a special setup |
| 09:41.27 | oej | I think they are overloaded, they have very little time to go through patches |
| 09:41.29 | JunK-Y | areski: actually theres few remaining |
| 09:41.38 | oej | It's not that they are not interested, just too much to do |
| 09:41.49 | oej | But there's no other committer that can handle that patch |
| 09:41.59 | JunK-Y | on my way back, i<ll have like 5 hours from sf to chicago that will be |
| 09:42.11 | oej | Let's stir up the community a bit |
| 09:42.17 | oej | I'll mail about it. |
| 09:42.18 | areski | :) |
| 09:42.26 | [av]bani | oej: monaco! |
| 09:42.29 | JunK-Y | cool. |
| 09:42.30 | oej | zoa: Can you update the description on it |
| 09:42.42 | oej | Monaco? Not cheap. |
| 09:42.52 | zoa | monaco would be extremely expensive |
| 09:42.56 | oej | zoa: Very user friendly instructions on how to patch |
| 09:42.59 | [av]bani | oh it has to be cheap? |
| 09:43.03 | oej | which files can I remove from the bug report |
| 09:43.14 | zoa | i will remove the unneeded files |
| 09:43.18 | oej | We need a decent price for the hotel nights and the conference |
| 09:43.24 | oej | zoa: thanks |
| 09:43.37 | oej | zoa: Don't forget to update the description... ;-) |
| 09:43.45 | [av]bani | some place in eastern europe then, ex communist country |
| 09:44.20 | zoa | eastern europe would be cheap, but flights are not so easy |
| 09:44.24 | dant | nottingham, UK |
| 09:44.33 | [av]bani | oh now it has to be easy too |
| 09:44.34 | oej | zoa: And we need a patch for svn trunk |
| 09:44.35 | [av]bani | so many demands |
| 09:44.42 | dant | that's easy for me :) |
| 09:44.50 | zoa | somebody made that already i think |
| 09:44.56 | fenlander | lets go visit skype in talin |
| 09:44.58 | *** join/#asterisk mmmToop (n=chatzill@196.31.11.194) |
| 09:45.14 | [av]bani | :() |
| 09:45.17 | zoa | yeah, do it in front of the skype office :) |
| 09:45.29 | zoa | we can all show our ass there |
| 09:45.32 | zoa | then take a nice picture |
| 09:45.36 | zoa | it would be in the news for sure |
| 09:45.38 | zoa | :) |
| 09:46.04 | areski | I am IN :D |
| 09:46.45 | fenlander | Tallinn is a great place to visit anyway |
| 09:46.53 | oej | zoa: I think we need more generic names in the coud thank "stevek" or "scx" :-) |
| 09:47.04 | oej | remove_stevek is an interesting name |
| 09:48.11 | oej | zoa: And kudos for all comments. We need to convert them to doxygen |
| 09:48.16 | puzzled | morning all |
| 09:48.35 | mmmToop | morning... |
| 09:49.49 | zoa | you went through all that code ? :) |
| 09:50.07 | oej | zoa: Looking at it now. Needs to update so I can patch my trunk |
| 09:50.27 | puzzled | oej: as usual I suggest Amsterdam for Astricon Europe :) |
| 09:50.49 | oej | puzzled: Noted |
| 09:50.53 | puzzled | areski: but Barcelona is nice too since the airfare is only about €100 |
| 09:50.59 | puzzled | and it has a beach |
| 09:51.01 | oej | no more spain |
| 09:51.14 | puzzled | The Hague has a beach too |
| 09:51.18 | oej | Lulea has a beach too, up in northern Sweden |
| 09:51.19 | JunK-Y | amsterdam, i might go there :) |
| 09:51.36 | puzzled | tickets are cheap since it's a European hub and everything flies to it |
| 09:51.54 | puzzled | flies? is that english? |
| 09:52.41 | Andrew_C | close enough |
| 09:53.00 | *** join/#asterisk lahaine (n=lahaine@8.69.119-80.rev.gaoland.net) |
| 09:53.11 | lahaine | hi ppl |
| 09:53.25 | oej | zoa: Ahh, doxygen coded comments in abstract_jb.h - extra points to slav!!! |
| 09:53.32 | puzzled | oej: if you are serious about investigating .nl as a location then I can put you in contact with a couple of people who might be able to help you |
| 09:53.46 | oej | puzzled: please. |
| 09:53.56 | zoa | oej, it has some strange things like putting a jitter buffer in chan_zap, it makes sense, but takes some talks with slav to understand why |
| 09:54.09 | oej | At this point, I am looking into everything other than spain, just because we did spain last year |
| 09:54.18 | areski | puzzled, you can even get cheaper flight inside europe |
| 09:54.28 | zoa | i could also bring you in contact with some people there probably |
| 09:54.42 | oej | zoa: Well, if the incoming channel sends jittery sound through to the zap outbound, it makes sense, right |
| 09:54.45 | *** join/#asterisk s (n=moreece@196.46.142.23) |
| 09:54.46 | puzzled | ok. though belgium is nice too. I like brussels and antwerp and brugge would be really nice |
| 09:54.52 | oej | I don't think it is for incoming audio on zap |
| 09:54.57 | zoa | indeed its not there |
| 09:54.59 | oej | But channels may pass jitter through the pbx |
| 09:54.59 | s | hello |
| 09:55.08 | puzzled | areski: yup :) |
| 09:55.21 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 09:55.23 | oej | Belgium is worth investigating. |
| 09:55.25 | moreece | just like to thank everyone for helping get my asterisk box going |
| 09:55.49 | Andrew_C | hey if i do a make install with this new version is it going to overwrite my configs? |
| 09:56.02 | moreece | I do have another question though regarding weather I should use SIP or IAX |
| 09:56.06 | puzzled | zoa: are you talking about a new version of the sip jb? |
| 09:56.14 | moreece | current SIP softphones are working |
| 09:56.22 | JunK-Y | Andrew_C: no, not a make install |
| 09:56.31 | Andrew_C | ta |
| 09:56.37 | oej | Seems to be a generic jitterbuffer, even though it's implemented in chan_sip now |
| 09:56.40 | moreece | we are opening a remote branch up in a couple weeks and I want to implement my current VOIP solution over to them |
| 09:56.41 | oej | Right, zoa? |
| 09:56.59 | moreece | is there any disadvantages to using sIP over IAX? |
| 09:57.03 | oej | moreece: If you ask me, I'll go for SIP :-) |
| 09:57.08 | moreece | thanks |
| 09:57.12 | oej | moreece: But that's a religious issue |
| 09:57.21 | moreece | ah alright |
| 09:57.28 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 09:57.41 | moreece | I understand that because IAX uses 1 port for communication its easier when it comes to NAT and firewalls |
| 09:58.02 | moreece | but that is find I'm going to get the boss to buy a couple SIP IP phones for testing |
| 09:58.07 | oej | moreece: With some technical details added to the argumentation. there are things you can only do with one of the protcols. If you have bandwidth problems, IAX trunking might help. If you have security concerns, IAX encryption might help |
| 09:58.07 | puzzled | what was the bug# of the sip jb again? |
| 09:58.12 | zoa | 3854 i think |
| 09:58.16 | puzzled | thanks |
| 09:58.20 | zoa | its generic, can be easily added |
| 09:58.32 | zoa | we have some patch to put it on iax2 too, but that one needs more work |
| 09:58.42 | oej | zoa: Tell me when you have an updated patch for trunk, and I'll start making noice again and test it |
| 09:58.50 | *** join/#asterisk chat_jokey (n=chat_jok@221-128-139-8.exatt.net) |
| 09:58.54 | zoa | k |
| 09:59.06 | zoa | will start on it next week |
| 09:59.24 | chat_jokey | hi there |
| 09:59.26 | moreece | thanks oej! but unforuntally I havent got IAX going yet, I did see an IAX error when start the console earlier. But for the moment we'll stick with the SIP phones I think |
| 09:59.27 | areski | anybody have been to the FOSDEM or project to go ? |
| 09:59.45 | zoa | areski im going |
| 09:59.54 | puzzled | zoa: do you need the jan 26 version of the patch tested on 1.2.3? |
| 09:59.58 | chat_jokey | anyone has any pointers .. if its possible to create a channel driver for Skype in asterisk ? |
| 10:00.10 | areski | zoa, hoo great, we r going to see any voip topics ? |
| 10:00.14 | puzzled | chat_jokey: reverse engineer the protocol first |
| 10:00.22 | zoa | yes |
| 10:00.25 | zoa | SER and asterisk |
| 10:00.26 | oej | moreece: Stick to SIP - that's the way to the future :-) |
| 10:00.41 | oej | Where's FOSDEM? |
| 10:00.45 | oej | And what? |
| 10:00.47 | zoa | belgium |
| 10:00.49 | chat_jokey | puzzled: yeah i am working on the same .. is it legal ? :) well i dont want to be flamed here though |
| 10:00.50 | areski | in brussel |
| 10:00.53 | puzzled | zoa: ok, I'll add it to today's 1.2.3 build |
| 10:00.54 | zoa | mark is speaking there |
| 10:00.57 | areski | fosdem.org |
| 10:01.05 | areski | ho really |
| 10:01.09 | zoa | chat_jokey: it would be legal |
| 10:01.11 | *** join/#asterisk guyee (n=izomtrik@nextra.nudli.equitas.hu) |
| 10:01.12 | zoa | i looked into it |
| 10:01.18 | *** part/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 10:01.18 | zoa | you will need an uber reverse engineer for it |
| 10:01.24 | viperdude | hi all, on a IAX debug for a incoming call I get 'IAX Subclass: INVAL', any ideas? |
| 10:01.30 | JunK-Y | Jean-Marc Valin too, cool |
| 10:01.37 | moreece | last question oej: Jan 26 11:56:43 WARNING[20253]: res_musiconhold.c:833 moh_register: Unable to open pseudo channel for timing... Sound may be choppy. |
| 10:01.45 | moreece | hmmm any ideas? |
| 10:01.51 | zoa | areski |
| 10:01.52 | chat_jokey | zoa: you think it would be a very interesting feature to add to asterisk ? |
| 10:01.53 | zoa | speak to me |
| 10:01.55 | zoa | in private |
| 10:01.59 | oej | moreece: You need a zaptel timer, check the wiki for information |
| 10:01.59 | zoa | chat_jokey: yes |
| 10:02.09 | zoa | i looked at doing it, but it will take a lot of money to do |
| 10:02.19 | puzzled | zoa: which one do I use for 1.2.3? ast_jb-1.2.0.patch3 or ast_jb-svn.patch? |
| 10:02.22 | moreece | even if I'm not using any hardware and simply using an IP |
| 10:02.25 | zoa | the svn |
| 10:02.29 | puzzled | ok thanks |
| 10:02.29 | oej | zoa: and to protect yourself from Ebay/Skype lawyers -t hat requires money |
| 10:02.33 | zoa | nah |
| 10:02.41 | zoa | i wouldnt mind about that too much |
| 10:02.53 | oej | moreece: Try different musiconhold ways, I am sure you can get it to work without the timer |
| 10:02.54 | zoa | release it from russia |
| 10:03.12 | chat_jokey | well oej: if that would be the case, then asterisk would have been sued long time back by some of the big telcos :) |
| 10:03.15 | oej | moreece: But if you need meetme conferencing, you will end up needing a timer, either hardware or ztdummy software timing |
| 10:03.21 | chat_jokey | oej: i feel thats the power of open source .. |
| 10:03.44 | oej | chat_jokey: Well, we don't implement licensed and protected stuff, we are very careful of that |
| 10:03.58 | [av]bani | isnt enhanced rtc accurate enough these days to not need ztdummy? |
| 10:04.04 | oej | chat_jokey: That's why the G.729 software codec is *not* open source, but available as commercial software with a license |
| 10:04.14 | zoa | reverse engineering is legal for compatibility things |
| 10:04.26 | oej | The power of open source is not to break copyrights and licenses |
| 10:04.28 | moreece | yes I've been doing the reading on it, also if u change the musiconhold mp3format to the native asterisk one instead of mpg123 |
| 10:04.54 | oej | zoa: I guess that depends on under which law you work. The skinny driver is a bit on the edge, if you want to find something risky in Asterisk |
| 10:04.54 | chat_jokey | oej: what we would need to do is reverse engineer and write a channel driver, so that people can use asterisk as their client, rather then their buggy client :P |
| 10:05.03 | moreece | I read that with the newer kernels asterisk using Processor for timer? |
| 10:05.18 | [av]bani | seems to me under 2.6 that ztdummy could go away entirely |
| 10:05.18 | oej | chat_jokey: it's not a new idea, it's been up for discussion on the mailing lists thousands of times |
| 10:05.37 | oej | moreece: Yes, we use the linux timer in ztdummy |
| 10:05.43 | zoa | chat_jokey: skype is probably one of the best protected and obfuscated codes |
| 10:06.01 | moreece | ah thanks guys appreciate it |
| 10:06.19 | [av]bani | oej -> what about ditching ztdummy entirely and using HPET |
| 10:06.31 | oej | zoa: Fosdem fits nicely into my trip to paris. I might go there. Would be fun. |
| 10:06.55 | hackeron | hey, I'm trying to share a calendar, but the "Select a user to add" droplist is empty. Any ideas why that is? |
| 10:06.58 | oej | Please submit code so that we can use timing directly in asterisk, without having to load kernel drivers. I would like that. |
| 10:07.09 | [av]bani | yes, i would too. |
| 10:07.15 | oej | hackeron: That is a good question, since Asterisk doesn't support calendars at all. |
| 10:07.28 | *** join/#asterisk thosa (n=thosa@p54879774.dip0.t-ipconnect.de) |
| 10:07.33 | hackeron | oej: oh, crap, wrong window, lol |
| 10:08.11 | oej | :-) hackeron |
| 10:08.29 | hackeron | that was a question for #horde :) |
| 10:08.33 | skeffling | hello, I'm geting lots of these warnings on nearly every call: Jan 26 09:59:56 WARNING[4242]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0xb6f05880', 10 retries! (zap 1.2.2 asterisk 1.2.3) calls seem to work OK though. |
| 10:10.12 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 10:14.26 | RoyK | wasim: ping |
| 10:14.31 | RoyK | ~seen wasim |
| 10:14.34 | jbot | wasim is currently on #asterisk (1d 20h 4m 56s). Has said a total of 42 messages. Is idling for 2h 6m 18s, last said: 'Joeymn: thats got nothing to do with asterisk, readup on how to do that for linux'. |
| 10:15.55 | chat_jokey | RoyK: any know list of commands for bot ? |
| 10:16.05 | RoyK | chat_jokey: msg it help |
| 10:17.31 | chat_jokey | ~help |
| 10:17.38 | *** join/#asterisk VJ (n=vijay@203.122.28.98) |
| 10:17.50 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 10:17.55 | RoyK | /msg jbot help |
| 10:17.56 | RoyK | might work |
| 10:18.16 | RoyK | ~openpbx |
| 10:18.18 | jbot | from memory, openpbx is an asterisk fork without asterisk's limitations of using other GPLed code. see http://openpbx.org/ for more info, or join #openpbx |
| 10:19.18 | chat_jokey | another project coming up : freeswitch .. i am trying to understand ... |
| 10:19.23 | chat_jokey | ~freeswitch |
| 10:19.49 | chat_jokey | what is freeswitch? |
| 10:19.57 | chat_jokey | phew, bot went crazy |
| 10:21.42 | MrChimpy | great. forking openpbx really looks helpful |
| 10:21.54 | RoyK | chat_jokey: i beleive freeswitch is getting built in quite another architecture |
| 10:22.31 | chat_jokey | MrChimpy: i might be a little lame.. could ya throw more lite on openpbx ? |
| 10:22.56 | MrChimpy | and freeswitch has a horrible looking website, so must be terrible :) |
| 10:23.32 | chat_jokey | agrees with MrChimpy reg freeswitch.. i dont understand it as well |
| 10:23.47 | RoyK | chat_jokey: opbx has what asterisk has, only in addition to that, it has stuff like chan_exosip2 and woomera, something that can, of course, be patched into asterisk, but will never be part of the distro since asterisk is dual licenced |
| 10:24.16 | MrChimpy | asterisk is the most popular and best documented, so that's what i'll use. |
| 10:24.51 | RoyK | http://tinypic.com/ml4lea.jpg |
| 10:25.02 | RoyK | MrChimpy: best documented :) |
| 10:25.07 | RoyK | rotfl |
| 10:25.14 | MrChimpy | royk: well, there's a book :) |
| 10:25.20 | RoyK | yeah |
| 10:25.31 | RoyK | opbx is pretty close to asterisk, though |
| 10:25.38 | *** join/#asterisk acehunky (n=chat_jok@221-128-138-85.exatt.net) |
| 10:25.42 | *** join/#asterisk skeffling (n=Andrew_H@andrew.1ec.aaisp.net.uk) |
| 10:25.57 | MrChimpy | i found the asterisk book pretty good, but didn't cover any E1 card stuff which would have been handy. |
| 10:26.05 | MrChimpy | would've only taken a few pages |
| 10:26.11 | Andrew_C | yes! audio is now working |
| 10:26.13 | Andrew_C | thanks guys |
| 10:26.31 | MrChimpy | what's up with the asterisk licence? I should check these things before we start developing :) |
| 10:26.48 | acehunky | is there any good howto on integrating openSer + asterisk (yeah i went through the one posted on their wiki page ) |
| 10:27.01 | RoyK | MrChimpy: you need to offer your soul to digium to get any code into the main tree |
| 10:28.02 | MrChimpy | if opbx is a fork how has that changed? |
| 10:28.13 | MrChimpy | surely their licence still applies |
| 10:28.48 | RoyK | they have stuffed it full with nice gpl stuff |
| 10:28.51 | acehunky | ~openser + asterisk |
| 10:28.53 | RoyK | check it out |
| 10:29.00 | RoyK | not openser, though |
| 10:29.21 | RoyK | ser is completely different |
| 10:29.33 | MrChimpy | Digium, Inc. (formerly Linux Support Services) holds copyright |
| 10:29.33 | MrChimpy | and/or sufficient licenses to all components of the Asterisk |
| 10:29.33 | MrChimpy | package, and therefore can grant, at its sole discretion, the ability |
| 10:29.33 | MrChimpy | for companies, individuals, or organizations to create proprietary or |
| 10:29.33 | MrChimpy | Open Source (even if not GPL) modules which may be dynamically linked at |
| 10:29.34 | MrChimpy | runtime with the portions of Asterisk which fall under our |
| 10:29.36 | MrChimpy | copyright/license umbrella, or are distributed under more flexible |
| 10:29.38 | MrChimpy | licenses than GPL. |
| 10:29.40 | MrChimpy | that totally reeks. |
| 10:30.31 | RoyK | http://tinypic.com/ml4lea.jpg |
| 10:30.48 | MrChimpy | hum. then the following paragraph seems to contradict what that implies. |
| 10:33.09 | acehunky | does it sound more like MPL ? |
| 10:33.23 | acehunky | all your code belongs to us kinds ? |
| 10:33.23 | MrChimpy | sounds like a load of BS to me |
| 10:33.53 | MrChimpy | look at LICENSE inside your asterisk tree |
| 10:34.22 | *** join/#asterisk bigjb (n=bigjb@195.60.10.113) |
| 10:35.15 | MrChimpy | seems like if you develop stuff with AGI/AMI they don't care and you can do what you like |
| 10:35.24 | bigjb | can anyone tell me why i get a "pbx_extension_helper: Cannot find extension context 'default'" |
| 10:35.40 | zoa | MrChimpy: that is correct |
| 10:35.41 | zoa | they dont |
| 10:35.42 | *** join/#asterisk Hit3k (n=Snake@128.125.dsl.mel.iprimus.net.au) |
| 10:36.04 | MrChimpy | loadable extensions however have to be developed at the discretion of digium |
| 10:36.04 | *** part/#asterisk Hit3k (n=Snake@128.125.dsl.mel.iprimus.net.au) |
| 10:36.45 | acehunky | so all loadable module is property of digium ? |
| 10:36.49 | MrChimpy | makes no sense, given I can happily change the source tree itself |
| 10:37.05 | MrChimpy | ace: scrollback... i pasted that bit |
| 10:37.49 | MrChimpy | i find it very odd when people GPL then attempt to put their own bits on top |
| 10:38.14 | acehunky | well GPL has a great affinity towards businesses .. |
| 10:38.48 | acehunky | businesses always like monopoly ... and GPL says dare you do that :) |
| 10:40.25 | MrChimpy | in fact I seriously doubt they can claim copyright on extensions that other people write |
| 10:41.11 | DarkFlibble | The digium license only matters if you want to use asterisk in a commercial product and *not* release the source... otherwise just use the GPL and relase the source with it |
| 10:41.11 | MrChimpy | as they're dynamically linked and if there's no digium code in there they have no claim of ownership at all. |
| 10:41.17 | acehunky | anyways i was here to find out if anyone has links to some good howto .. on integrating OpenSER + Asterisk .. |
| 10:41.47 | MrChimpy | i'll be using asterisk commercially but not in a product. GPL suits us for that. |
| 10:41.52 | DarkFlibble | acehunky, there is one info in the wiki... |
| 10:42.03 | MrChimpy | as we won't release anything we don't have to let our code go |
| 10:42.16 | acehunky | DarkFlibble: is it the OpenSER + Asterisk Real time one ? |
| 10:42.18 | MrChimpy | but if there's anything actually useful to the community we can release that at will. |
| 10:42.28 | DarkFlibble | acehunky, I don't remember |
| 10:42.34 | *** join/#asterisk asr__ (i=asr@pimpbox.latency.net) |
| 10:42.50 | acehunky | http://openser.org/dokuwiki/doku.php?id=integrating_openser_and_asterisk_realtime |
| 10:42.55 | acehunky | any other ? |
| 10:51.01 | DarkFlibble | I meant the voip-info wiki |
| 10:56.52 | *** join/#asterisk bzbw (n=wlwzhang@ip-33-107-134-202.rev.dyxnet.com) |
| 10:57.22 | bzbw | anyone knows if I can use busy tone to close the zap channel for my x100p? |
| 10:58.32 | *** join/#asterisk Dabba (n=d@ipv6.mfnx.ip6net.net) |
| 11:01.00 | Dabba | Hi all could someone in the know please confirm if this syntax is correct as im unsure |
| 11:01.03 | Dabba | http://pastebin.ca/raw/38645 |
| 11:01.46 | acehunky | DarkFlibble: wat were ur search string ? |
| 11:02.02 | wasim | bzbw: yes |
| 11:02.16 | RoyK | wasim: hi |
| 11:02.17 | *** join/#asterisk BlueDevi1 (n=BlueDevi@scooby.csn.tu-chemnitz.de) |
| 11:02.58 | wasim | etowi RoyK |
| 11:04.17 | wasim | err ... goddag |
| 11:04.56 | *** join/#asterisk kiwnix (n=egarcia@77.red-82-158-158.user.auna.net) |
| 11:05.07 | DarkFlibble | acehunky, I'm looking now |
| 11:05.39 | SERGEUS|W | can somebody give me an advice about cards with 1 bri port? what is the best buy for today? :) |
| 11:05.41 | DarkFlibble | but I think it was linked fromt eh asterisk at large section |
| 11:05.42 | Dabba | ast_expr2.fl:176 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected TOK_GT, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input: |
| 11:06.19 | acehunky | umm okk ..@ DarkFlibble |
| 11:06.54 | *** part/#asterisk zzxxcc (n=zzxxcc@221.232.10.225) |
| 11:13.27 | *** join/#asterisk basta (n=basta@213-156-52-98.fastres.net) |
| 11:13.51 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 11:14.20 | *** join/#asterisk cpm (n=Chip@border0.avitecture.net) |
| 11:15.05 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
| 11:15.47 | myiagy | hey, is rtpproxy the only way to implement symmetric rtp ? |
| 11:15.55 | basta | could it be that if I have ser and asterisk on the same machine (* with chan_sip disabled) asterisk will have problems sending audio ? |
| 11:16.50 | *** join/#asterisk fulgas (n=fulgas@209.8.233.243) |
| 11:17.10 | *** join/#asterisk frenzy (n=frenzy@196.45.144.40) |
| 11:17.24 | frenzy | hello |
| 11:17.27 | DarkFlibble | basta, not really... although you will have to make sure either ser or asterisk isn't using the default sip port |
| 11:17.42 | frenzy | does any one know of a GSM modem that works with Asterisk? |
| 11:18.35 | DarkFlibble | frenzy, do a search for cell socket... I have heard good things of them... but not anything lately... |
| 11:18.43 | DarkFlibble | the mailling list should help you |
| 11:18.58 | bzbw | wasim: sorry I was off for few minutes, how do I set up busy detection for my X100p? |
| 11:19.10 | wasim | bzbw: you read up on teh wiki |
| 11:20.40 | basta | when they call from that machine I can't hear them but they can hear me, seems the rtp stream never starts |
| 11:21.25 | basta | a identical machine without ser works .... |
| 11:23.14 | guyee | I upgraded from 1.0.9 to the latest trunk, and ${CALLERIDNUM} is now empty. It's been the user id of the SIP account, and now I can only use the ${CALLERID(num)} which is the configured caller id of the calling account. And I need the user id. Any ideas? |
| 11:23.44 | DarkFlibble | guyee, its now CALLERID(num) its listed under functions on the wiki |
| 11:23.55 | DarkFlibble | err... CALLERID(number) |
| 11:26.31 | guyee | DarkFibble: almost. CALLERID(number) gives the "callerid" property of the SIP peer. What I need is the account name (the something between brackets) |
| 11:26.34 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 11:26.36 | *** join/#asterisk basta (n=basta@213-156-52-98.fastres.net) |
| 11:26.47 | *** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 11:26.49 | *** join/#asterisk franck (n=franck@tikiwiki/franck) |
| 11:27.17 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 11:27.26 | fulgas | morning |
| 11:27.38 | DarkFlibble | hey |
| 11:27.59 | guyee | DarkFibble: callerid is necessary to call out through our voip provider, so I cannot change that. So CALLERIDNUM and CALLERID(num) are _not_ equivalent. |
| 11:29.18 | DarkFlibble | Set(CALLERID(number)=mynumberhere) i think is the new method... not got time to double check it... |
| 11:29.35 | DarkFlibble | but it should be fairly easy to find |
| 11:30.54 | guyee | DarkFibble: Argh. but then I must rewrite the whole dialplan. Until now it was quite easy to do anything, because I had access to both the userid and the callerid. |
| 11:32.06 | DarkFlibble | if you live on the cutting edge expect cuts |
| 11:32.20 | DarkFlibble | :P |
| 11:32.39 | guyee | DarkFibble: Yeah. I have to agree... :/ |
| 11:32.42 | *** join/#asterisk ckruetze (n=ckruetze@131.8.dsl3.ip.foni.net) |
| 11:32.47 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 11:32.52 | ckruetze | hi |
| 11:34.05 | *** join/#asterisk arkanis (n=test@80-219-10-141.dclient.hispeed.ch) |
| 11:39.02 | *** join/#asterisk Inkubot (n=inkubot@quirquincho.manquehue.cl) |
| 11:39.07 | Inkubot | hi |
| 11:39.18 | Inkubot | i've got a little proble with a x101p card |
| 11:39.33 | Inkubot | when i make or receive a call, there's no audio |
| 11:40.53 | DarkFlibble | Inkubot, was it a clone card....? |
| 11:40.59 | darkskiez | Asterisk 1.2.2 ? |
| 11:41.18 | Inkubot | 1.2.3 |
| 11:41.22 | DarkFlibble | darkskiez, 1.2.2 has a bug... |
| 11:41.34 | Inkubot | DarkFlibble: a clone card |
| 11:41.41 | *** join/#asterisk kore (i=kore@mindwipe.org) |
| 11:41.41 | darkskiez | Dar |
| 11:41.56 | darkskiez | DarkFlibble: yes, just wondering if inku was running it :) |
| 11:42.17 | DarkFlibble | kk.. |
| 11:42.57 | Inkubot | the call reach the Asterisk, but nothing happens |
| 11:43.01 | Inkubot | no audio at all |
| 11:43.06 | DarkFlibble | many of the clone cards aren't 100% compatible... |
| 11:43.48 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 11:45.32 | Inkubot | apz |
| 11:50.17 | VJ | for audio problem consider this as well http://www.asterisk.org/blog/73 |
| 11:50.40 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 11:50.47 | VJ | Inkubot: a bug has been found in 1.2.2 |
| 11:51.25 | Dabba | can anyone confirm if app MYsql works in 1.2.3 |
| 11:51.33 | Dabba | it didnt in 1.2.2 |
| 11:51.40 | Dabba | but does in 1.2.0 |
| 11:51.42 | frenzy | I'm looking for a gsm modem that works with asterisk. used for receiving SMS |
| 11:52.05 | VJ | Mysql works in 1.2.2 as well |
| 11:52.14 | Dabba | hmm |
| 11:52.18 | Dabba | it segs on me |
| 11:52.24 | Dabba | but fine in1.20 |
| 11:52.36 | VJ | might be some conf problem |
| 11:52.46 | VJ | but i am using mysql in 1.2.2 and no problem at all |
| 11:52.55 | Dabba | you think, wierd |
| 11:53.05 | VJ | which version of mysql are you using? |
| 11:53.12 | DarkFlibble | frenzy, look for GSM terminal |
| 11:53.34 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 11:53.42 | Dabba | VJ > 4.0.25-standard |
| 11:53.49 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 11:54.10 | VJ | this should be ok |
| 11:54.16 | VJ | whats the problem you are facing |
| 11:54.46 | VJ | which lunux are you using? |
| 11:54.50 | VJ | *linux |
| 11:54.53 | *** join/#asterisk SeicherlBoB (n=noyb@dsl-80-16.utaonline.at) |
| 11:54.56 | *** join/#asterisk S3RG3US|W (n=SERGEUS@ippe-245.ippe.ru) |
| 11:55.10 | Inkubot | VJ: i known, i'm using 1.2.3 |
| 11:55.26 | Dabba | VJ > http://pastebin.ca/raw/38647 |
| 11:55.40 | Dabba | it segs but it doesnt in 1.2.0 |
| 11:57.30 | *** join/#asterisk Delvar (n=irc@host-83-146-53-34.bulldogdsl.com) |
| 11:59.22 | Dabba | VJ > i just noticed your other question RH9 smp w/2cpu |
| 11:59.31 | *** join/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com) |
| 11:59.47 | Dabba | anyway, i guess it is ok for you so must be fine in 1.2.3 |
| 12:00.05 | *** join/#asterisk stefan2 (n=someone@193.138.167.4) |
| 12:00.06 | stefan2 | hi |
| 12:00.38 | *** join/#asterisk mut (n=animenod@65.111.201.79) |
| 12:00.40 | warthawg | especially now that nufone is back in operation |
| 12:00.51 | stefan2 | I managed to get a fresh asterisk 1.2.3 with chan_capi running. it even reports my controller correct when starting with "asterisk -vvvvgc" |
| 12:01.02 | stefan2 | but it immediately exits without giving a console |
| 12:01.14 | stefan2 | does somebody have a hint? |
| 12:01.32 | stefan2 | nothing in the logs - except "1138275664|NONE|NONE|NONE|QUEUESTART|" in queue.log |
| 12:02.10 | franck | What is nonce is sip protocol? |
| 12:02.11 | zigman | Dabba try to recompile asterisk-addons |
| 12:04.13 | *** join/#asterisk enric (n=enric@62.81.151.129) |
| 12:06.55 | *** join/#asterisk r0d3nt (i=r0d3nt@tinfoilhat.net) |
| 12:07.29 | warthawg | what happened to 1.2.2? |
| 12:08.15 | mut | fookin comcast |
| 12:08.35 | *** join/#asterisk tecnico (n=tecnico@user-24-236-120-2.knology.net) |
| 12:08.45 | warthawg | i mean, what is the timebomb? |
| 12:08.53 | mut | audio dies |
| 12:09.02 | warthawg | Ahrimanes, thanks |
| 12:11.59 | *** part/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 12:13.53 | *** join/#asterisk SGM (n=stoyan@home.marinov.us) |
| 12:16.58 | warthawg | was that really a timebomb, deliberately coded? |
| 12:17.05 | DarkFlibble | no |
| 12:17.14 | DarkFlibble | its was an overflow in a signed variable |
| 12:17.20 | warthawg | thanks |
| 12:17.56 | DarkFlibble | franck, a nonce is used for authentication... |
| 12:19.57 | Dabba | zigman I tried that, I guess I will just stick with 1.2.0 it works and it works well |
| 12:21.25 | franck | My sip phone is sending lot of nonce authentications, is it normal? |
| 12:22.26 | *** join/#asterisk Jun_Wang (n=chatzill@pool-138-89-62-149.nwrk.east.verizon.net) |
| 12:22.51 | Dabba | anyone familiar with variables and GotoIf check this over please |
| 12:22.54 | Dabba | http://pastebin.ca/raw/38645 |
| 12:26.26 | *** join/#asterisk heliosj (n=jeff@pdpc/supporter/active/xheliox) |
| 12:26.46 | *** join/#asterisk Abbas (i=Abbas@203.81.226.100) |
| 12:28.56 | *** join/#asterisk trererre (n=sergio@joltid-gw.joltid.org) |
| 12:33.03 | cpm | sourceforge really gets on my nerves sometimes. |
| 12:33.24 | warthawg | how so, cpm |
| 12:33.41 | stefan2 | cpm: was unavailable for me as well, an hour or so ago |
| 12:33.58 | Err | heh, it's pretty much garbage, but you do get what you pay for :-) |
| 12:35.30 | cpm | warthawg, It's just that I remember when projects had a website, with links to the source, and mirrors, and howtos written in the howto style, and a link to the mailling list archives, and that was that. Sourceforge is painful to navigate in comparison. That and when it is down, it's down. No hopping to a mirror |
| 12:35.46 | fugitivo | cpm: I agree |
| 12:35.51 | *** join/#asterisk bmg505 (n=leon@dsl-146-46-56.telkomadsl.co.za) |
| 12:36.14 | fugitivo | but most projects have a website, you just need to click on the link "Homepage" |
| 12:36.22 | warthawg | cpm, thanks, i'll pass that up the food chain |
| 12:36.45 | cpm | Please know, I'm just whining for the sake of whining. |
| 12:36.51 | warthawg | heh |
| 12:36.55 | cpm | or wenging if you will |
| 12:36.58 | warthawg | i have a blackbelt in whining |
| 12:37.01 | cpm | heh |
| 12:38.03 | cpm | thanks for letting me whine |
| 12:39.03 | warthawg | we're getting reports that sourceforge is blocked in china |
| 12:39.03 | Err | actually, you can view mirrors almost always, for downloads |
| 12:39.10 | Err | I don't know if web pages are mirrored or not |
| 12:39.11 | warthawg | dunno why they would bother |
| 12:39.42 | cpm | Yeah, but when sourceforge is down, it's difficult to find the mirrors, unless you intuitively know where they might be. |
| 12:40.26 | Err | it's the kind of thing you should bookmark, since sourceforge goes down like a $3 whore |
| 12:40.36 | cpm | heh, shame shame! |
| 12:40.51 | fugitivo | blah, you don't need sourceforge, just emerge search the project |
| 12:41.09 | *** join/#asterisk crich1999 (n=crich@p54BFA504.dip0.t-ipconnect.de) |
| 12:41.34 | tzafrir_laptop | also, add links you like to http://swik.net/ |
| 12:42.00 | tzafrir_laptop | http://swik.net/asterisk |
| 12:42.13 | fugitivo | tzafrir_laptop: that's new for me, great |
| 12:42.15 | Err | I love how SF's cvs goes down for hours at a time (anoncvs) any time gaim has any sort of announcement |
| 12:43.03 | tzafrir_laptop | well, they finally have a SVN. |
| 12:44.05 | Err | kind of - it's not globally available yet |
| 12:47.39 | trererre | hi, i have a major problem in meetme, this is the scenario: a client connects to meetme using sip and ilbc codec, then some other clients connect also with SIP and whatever codec, if everyone leaves the conference leaving only a client(using ilbc) in the conference, the CPU of the machine goes up 10% and never recovers. If someone enters the conference again, the CPU goes down again. If this scenario happens in 10 conferences for example, the CPU |
| 12:48.55 | *** join/#asterisk tecnico (n=tecnico@user-24-236-120-2.knology.net) |
| 12:49.08 | *** join/#asterisk Jun_Wang (n=chatzill@pool-138-89-62-149.nwrk.east.verizon.net) |
| 12:50.55 | *** join/#asterisk kink0 (n=kinko@pluton.interec.com) |
| 12:50.58 | kink0 | hello |
| 12:51.19 | kink0 | I still geting: < Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) |
| 12:51.20 | kink0 | < Ext: 1 Cause: Unknown (100), class = Protocol Error (6) ] |
| 12:51.24 | *** part/#asterisk frenzy (n=frenzy@196.45.144.40) |
| 12:51.29 | *** join/#asterisk DannyF (n=dannyf@dialup-82-96-30-92.rixtele.com) |
| 12:51.36 | kink0 | while send calls from Digium TE405 to 2N Stargate PRI port |
| 12:51.46 | kink0 | any idea about what is wrong ? |
| 12:52.54 | trererre | please, can someone help me out? |
| 12:55.06 | mut | weird |
| 12:55.12 | mut | this user timed out while i was on the phone |
| 12:55.24 | mut | their qualify didn't return in time |
| 12:55.31 | stefan2 | does anybody have an idea why my asterisk with chan_capi starts up fine, but immediately exists after init? I don't get a console, though started with "asterisk -vvvvgc" |
| 12:55.31 | mut | but the call is still going |
| 12:56.52 | *** join/#asterisk Grubs (n=Miranda@c211-28-119-169.eburwd3.vic.optusnet.com.au) |
| 12:58.54 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 12:59.17 | *** part/#asterisk dwildes (n=dwildes@209.164.237.195) |
| 13:00.26 | trererre | hi, i have a major problem in meetme, this is the scenario: a client connects to meetme using sip and ilbc codec, then some other clients connect also with SIP and whatever codec, if everyone leaves the conference leaving only a client(using ilbc) in the conference, the CPU of the machine goes up 10% and never recovers. If someone enters the conference again, the CPU goes down again. If this scenario happens in 10 conferences for example, the CPU |
| 13:01.27 | JimmyGulp | trererre, you might want to try popping that onto 2 lines, we get cut off after "happens in 10 conferences for example, the CPU" |
| 13:01.44 | warthawg | you got the U? |
| 13:02.04 | *** join/#asterisk enemy^x (n=null@85.196.70.98) |
| 13:02.06 | DarkFlibble | no U here |
| 13:02.13 | *** join/#asterisk pengyong (n=lala@222.185.18.133) |
| 13:03.01 | bennyben | mmm.. someone motivated to help me about Flash() and how to reproduce the 'R' key of my phone ? |
| 13:03.32 | *** join/#asterisk jono (n=jono@mail.openadvantage.org) |
| 13:03.36 | RoyK | if you don't have a R key, just hit the fork |
| 13:03.38 | jono | hi all |
| 13:03.42 | RoyK | R == flash == short break |
| 13:03.43 | jono | which plugin is needed in ARI when you play a voicemail ? |
| 13:03.51 | RoyK | wtf is ari? |
| 13:04.01 | jono | asterisk recording interface |
| 13:04.18 | dokhench | depends on the format recorded in. |
| 13:04.48 | jono | its in wav format |
| 13:04.55 | bennyben | RoyK: hit the fork ? I tried to put a Flash() in my dialplan, it does a short break, but not generates exactly the same signal than my phone |
| 13:05.26 | jono | dogtanian, no, in the ARI web interface, there is a Play link to play a voicemail |
| 13:06.05 | dokhench | jono: what is this ARI web interface you a talking about? got a http:// ? |
| 13:06.14 | *** join/#asterisk imb (n=root@194.149.96.251.adsl.nextra.cz) |
| 13:06.35 | trererre | JimmyGulp, the CPU goes down again. If this scenario happens in 10 conferences for example, the CPU goes 100 %. Can someone help me out? I know this is a bug, but i cant find where is coming from, it is not directly dependent in meetme since version 1.2 with app_meetme 1.0.9 also has the bug, but if u use 1.0.9 * and app_meetme it works fine |
| 13:06.49 | fugitivo | dokhench: http://www.littlejohnconsulting.com/ari |
| 13:06.57 | bennyben | I tried to play with flash_time in my zapata.conf too, and if it's just finding the good parameters, how can I find it ? |
| 13:08.04 | dokhench | fugitivo, thx. |
| 13:08.08 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 13:08.09 | Grubs | bennyben - search for "hook-flash" in vpb.conf for an example (perhaps). |
| 13:08.32 | bennyben | okay |
| 13:08.50 | trererre | <PROTECTED> |
| 13:08.57 | trererre | If this scenario happens in 10 conferences for example, the CPU goes 100 %. Can someone help me out? I know this is a bug, but i cant find where is coming from, it is not directly dependent in meetme since version 1.2 with app_meetme 1.0.9 also has the bug, but if u use 1.0.9 * and app_meetme it works fine |
| 13:09.16 | trererre | can someone help me identifying this bug? |
| 13:09.27 | *** join/#asterisk burtonez (i=mimx@w201.ljudmila.org) |
| 13:09.57 | bennyben | Grubs: thanks, I'll try it soon |
| 13:10.03 | Grubs | bennyben - flash timings are set in zapata.conf |
| 13:10.26 | *** join/#asterisk Grok_ (n=blah@c-71-196-75-163.hsd1.fl.comcast.net) |
| 13:10.31 | Lathos42 | Good Morning everyone |
| 13:10.40 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 13:12.26 | *** join/#asterisk Modcuts (n=sam@proporta.gotadsl.co.uk) |
| 13:13.30 | *** join/#asterisk apardo (n=apardo@87.218.44.151) |
| 13:13.35 | bennyben | Grubs: I know, ever played with it, but either it does nothing, either it hangs up, hard to find exactly the good timings, or a good doc about these parameters |
| 13:14.02 | Grubs | what country are you in? |
| 13:14.24 | mut | anyone use format_mp3 for MoH? |
| 13:14.36 | Grubs | some refernce docs are linked to in indications.conf |
| 13:15.10 | *** join/#asterisk Chuji_ (i=Chuji@pcp09180602pcs.nash01.tn.comcast.net) |
| 13:15.11 | kink0 | Ext: 1 Cause: Unknown (100), class = Protocol Error (6) ] what to see ? what to do to fix this error ? |
| 13:15.22 | Grubs | (e.g. AU recall is only 100ms when Asterisk Flash defaults to 750ms) |
| 13:15.33 | bennyben | france, under an adsl box that does pabx itself |
| 13:15.47 | bennyben | mmmmmm ok :) |
| 13:16.26 | bennyben | (an hook-flash doesn't resolve my problem) |
| 13:19.44 | *** join/#asterisk linville (n=linville@nat-pool-rdu.redhat.com) |
| 13:20.00 | *** join/#asterisk BladeRunner05 (n=feelme@adsl-80-217.37-151.net24.it) |
| 13:20.39 | bennyben | Grubs: ok, thx a lot, I will try to take a look at this, perhaps the problem is my adsl box that would like special parameters |
| 13:21.01 | *** part/#asterisk lahaine (n=lahaine@8.69.119-80.rev.gaoland.net) |
| 13:23.24 | *** join/#asterisk jozsab1 (n=jozsab1@86.125.91.54) |
| 13:24.33 | *** join/#asterisk Stephnie (i=Stephnie@u15157627.onlinehome-server.com) |
| 13:24.39 | jozsab1 | Is there a way to send instead of "m=audio 12252 RTP/AVP 0 101" only "m=audio 12252 RTP/AVP 0" on sip protocol ? (i think this means disabling RFC 8233) |
| 13:24.54 | mut | whgat exactly does res_config_mysql.c:551 parse_config: MySQL RealTime: No database socket found, using '/tmp/mysql.sock' as default. |
| 13:24.57 | mut | mean? |
| 13:25.04 | mut | cause res_mysql.conf |
| 13:25.19 | mut | i use the exact same stuff on command line.. mysql -u -h -p |
| 13:25.25 | mut | and it works fine |
| 13:25.44 | SGM | this means you havent specified the path to the mysql socket |
| 13:25.47 | Aughey | mut: does /tmp/mysql.sock exist? |
| 13:25.57 | mut | it's not connecting to localhost |
| 13:26.02 | SGM | or the one you specified wasn't found |
| 13:26.04 | mut | i specified poty |
| 13:26.08 | mut | port |
| 13:26.27 | Stephnie | hi I got an error when I do "make install" in ASTCC source ..... http://pastebin.ca/38652 |
| 13:26.29 | SGM | mut: try using 127.0.0.1 instead of localhost |
| 13:26.42 | mut | i'm not using localhost.. thats the point |
| 13:26.45 | mut | it's a remote db |
| 13:26.50 | SGM | hmm |
| 13:27.15 | NirS | steph, what distribution are you using ? |
| 13:28.00 | Stephnie | RH 9 .... |
| 13:28.04 | NirS | ok |
| 13:28.06 | NirS | hold on |
| 13:28.12 | mut | ya, hmm is what i thought too |
| 13:28.19 | NirS | you need to install the perl-cgi rpm according to the error message |
| 13:28.26 | Stephnie | okey. |
| 13:28.44 | BladeRunner05 | with * 1.2.3 I have to install ztdummy or any zap interface. the question is are installed by default or I have to do it manually ? |
| 13:28.48 | jozsab1 | Is there a way to send instead of "m=audio 12252 RTP/AVP 0 101" only "m=audio 12252 RTP/AVP 0" on sip protocol ? (i think this means disabling RFC 8233) |
| 13:28.49 | NirS | let me check which RPM it's exactly |
| 13:29.14 | SGM | mut: paste your res_mysql.conf |
| 13:29.32 | Stephnie | NirS: I think this one perl-CGI-2.81-88.i386.rpm |
| 13:29.38 | gaupe | CGI.pm is included in the perl packages, at least for fedora/redhats of this milennium |
| 13:29.43 | NirS | yes |
| 13:29.52 | Stephnie | NirS: ok going to install |
| 13:29.55 | gaupe | it's beyond me anyone would like to run RH9 today |
| 13:29.57 | NirS | gaupe, RedHat 9 isn't from this millenium |
| 13:30.04 | gaupe | I know |
| 13:30.23 | NirS | gaupe, it's beyond me why people would like to use RH or Fedora, they both suck |
| 13:30.31 | mut | http://pastebin.ca/38653 |
| 13:30.46 | *** join/#asterisk secure75 (n=mic@host-82-135-62-14.customer.m-online.net) |
| 13:31.00 | gaupe | NirS: sounds like a professional opinion ;) |
| 13:31.08 | cpm | It all sucks to some degree or another, its the sucking one is used to. |
| 13:31.18 | NirS | ROTFL |
| 13:31.20 | gaupe | RoyK: debian? |
| 13:31.32 | RoyK | :) |
| 13:31.35 | imb | debian riks |
| 13:31.40 | imb | roks even |
| 13:31.40 | RoyK | no way |
| 13:31.42 | NirS | actually, I managed to come quite friendly with Mandrake |
| 13:31.45 | NirS | and CentOS |
| 13:31.49 | RoyK | I use debian on all sorts of boxes |
| 13:31.54 | *** join/#asterisk KeX (n=KeX@argos.campus-sbg.at) |
| 13:31.56 | KeX | hi |
| 13:31.56 | warthawg | i'm running asterisk from the ubuntu package |
| 13:32.00 | gaupe | NirS: well, centos is redhat really :) |
| 13:32.01 | imb | Redhat sucks like a hoover |
| 13:32.08 | cpm | heh |
| 13:32.10 | NirS | i didn't say I like CentOS |
| 13:32.10 | mut | .. |
| 13:32.14 | KeX | how can i get an variable from an shell script (agi) back into asterisk ? |
| 13:32.14 | RoyK | centos and fedora is redhattish |
| 13:32.16 | NirS | I just got friendly with it |
| 13:32.18 | warthawg | but i ran it first on openwrt |
| 13:32.25 | RoyK | KeX: SET VARIABLE |
| 13:32.28 | NirS | Mandriva kicks ass, no matter what you say |
| 13:32.35 | cuco | NirS, i am having problems surfing into http://asterisk.org.il |
| 13:32.36 | SGM | mut: and you can connect to the remote server using mysql cli? |
| 13:32.38 | imb | asterisk -rx "$command" ?? |
| 13:32.40 | RoyK | NO WAY, NirS |
| 13:32.43 | mut | yes |
| 13:32.50 | NirS | why roy ? |
| 13:32.56 | RoyK | because it's redhattish? |
| 13:34.03 | NirS | not any more |
| 13:34.03 | *** join/#asterisk mohsin (n=mohsin@202.176.229.209) |
| 13:34.03 | NirS | the only relation between Mandriva and RH these days is the fact they both use RPM |
| 13:34.04 | NirS | anything else is pure imagination |
| 13:34.04 | KeX | RoyK: i tried this: echo "SET(new_str=$new_str)"; than return $new_str; and echo $new_str; but nothin worked : / |
| 13:34.04 | mohsin | hi |
| 13:34.04 | Stephnie | NirS : please check it now http://pastebin.ca/38655 |
| 13:34.04 | RoyK | kend: set? |
| 13:34.04 | NirS | I've been using Mandriva/Mandrake for the past 8 years now, never disappointed me |
| 13:34.04 | RoyK | kend: you need to do an AGI call SET VARIABLE |
| 13:34.08 | RoyK | NirS: rpm is enough to hate it :) |
| 13:34.10 | jozsab1 | Is there a way to send instead of "m=audio 12252 RTP/AVP 0 101" only "m=audio 12252 RTP/AVP 0" on sip protocol ? (i think this means disabling RFC 8233) |
| 13:34.10 | KeX | RoyK, k, i'll try |
| 13:34.17 | KeX | RoyK: and echo or return? |
| 13:34.40 | SGM | mut: this seems to be just a warning |
| 13:34.43 | RoyK | KeX: http://www.voip-info.org/wiki-Asterisk+AGI |
| 13:34.52 | SGM | that no socket was specified |
| 13:34.52 | Stephnie | NirS: do I need this now ?? perl-DBD-Pg-1.21-2.i386.rpm ??? |
| 13:34.53 | RoyK | ~rtfw |
| 13:34.54 | jbot | hmm... rtfw is http://www.voip-info.org, the only place to get any real answers about * and it's many, many uses. |
| 13:34.55 | NirS | you are missing the perl-dbd-mysql module |
| 13:34.57 | SGM | in the general section |
| 13:34.59 | BladeRunner05 | How can I configure ztdummy ????? |
| 13:35.01 | SGM | of the config |
| 13:35.05 | RoyK | BladeRunner05: you don't |
| 13:35.16 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 13:35.19 | BladeRunner05 | royk: is yet installed in 1.2.3 ? |
| 13:35.21 | mut | hm |
| 13:35.27 | NirS | steph, perl-dbd-pg is for Postgres |
| 13:35.36 | RoyK | BladeRunner05: just make install with zaptel on 2.6 and it installs |
| 13:35.43 | SGM | res_config_mysql.c line 550 |
| 13:35.43 | RoyK | remember to modprobe ztdummy before starting asterisk |
| 13:35.43 | SGM | :) |
| 13:36.02 | RoyK | BladeRunner05: it comes with zaptel, not asterisk |
| 13:36.06 | BladeRunner05 | RoyK: ok I try.... |
| 13:36.13 | jozsab1 | anybody knows a softphone wich has conferrence feture and it's free or trial ? |
| 13:36.16 | cpm | the reason mandriva roks, is plf |
| 13:36.18 | SGM | mut: try adding "dbsock=" to the config |
| 13:36.45 | Stephnie | NirS: oh.....I got it....plz check it http://pastebin.ca/38656 |
| 13:36.48 | SGM | gotta go... bye guys |
| 13:36.54 | jozsab1 | bye sgm |
| 13:39.02 | cpm | g'night! |
| 13:39.15 | stefan2 | hey - i finally got it to work :-) |
| 13:39.32 | stefan2 | all I needed was create a logger.conf to also output my any warnings |
| 13:39.56 | *** join/#asterisk saftsack (n=oliver@p54A7CDE6.dip.t-dialin.net) |
| 13:40.25 | Stephnie | NirS: what do I do next? should I get this file from another box??? |
| 13:41.13 | Aughey | jozsab1: have you tried sjphone? |
| 13:41.26 | saftsack | _Sam--, hi are you here? |
| 13:41.53 | NirS | hold on |
| 13:41.56 | NirS | on the phone |
| 13:42.10 | Stephnie | ok |
| 13:42.19 | Stephnie | take your time :) |
| 13:45.15 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:46.52 | *** join/#asterisk Iam8up|lappy (n=dontemai@cpe-71-65-112-38.woh.res.rr.com) |
| 13:47.00 | mut | i also get this warning |
| 13:47.01 | mut | Jan 26 08:45:30 WARNING[31117]: pbx.c:3742 ast_merge_contexts_and_delete: Requested contexts didn't get merged |
| 13:47.05 | mut | whats it mean? |
| 13:47.47 | *** join/#asterisk hans (n=fugalh@falcon.fugal.net) |
| 13:47.49 | *** join/#asterisk msw (n=msw@rdu-nat.rpath.com) |
| 13:48.28 | NirS | hold on |
| 13:48.36 | NirS | let me check the pastbin |
| 13:49.07 | NirS | ok, this means you need to install the mysql-devel package |
| 13:49.38 | Stephnie | NirS: ok let me install that |
| 13:51.31 | BladeRunner05 | with linux kernel 2.6.12 I get a lot of errors compiling zaptel-1.2.2 see http://pastebin.com/523900 |
| 13:53.01 | *** join/#asterisk Defraz (i=t0tal@72.165.56.43) |
| 13:53.02 | tzafrir_laptop | BladeRunner05, are those the first errors? |
| 13:53.50 | tzafrir_laptop | BladeRunner05, Isn't there some message about a missing header higher up? |
| 13:54.19 | Stephnie | NirS: http://pastebin.ca/38657 |
| 13:54.57 | BladeRunner05 | I check |
| 13:55.09 | Stephnie | NirS: do I need this ?? MySQL-shared-3.22.32-1.i386.rpm ??? or ....? |
| 13:55.29 | fdask | might wanna try and find a newer version |
| 13:56.11 | tzafrir_laptop | BladeRunner05, BTW: ztdummy needs no extra config after loading the module |
| 13:56.14 | BladeRunner05 | tzafir: I don't see errors like that, consider that I have yet compiled asterisk without errors.... could be missing headers for zap... ? |
| 13:56.54 | tzafrir_laptop | BladeRunner05, when you build zaptel, zaptel.h is in the same directory. Kind of hard to miss |
| 13:57.25 | BladeRunner05 | yes zaptel.h is in the zaptel compiling direct... |
| 13:57.28 | tzafrir_laptop | Though those fields come from zaptel |
| 13:58.43 | BladeRunner05 | what can I check ? |
| 13:58.59 | *** join/#asterisk SGM (n=stoyan@213.91.216.130) |
| 13:59.45 | NirS | well |
| 13:59.55 | NirS | I'm not that familiar with the RH9 structure |
| 14:00.01 | NirS | try installing all of them |
| 14:00.01 | *** join/#asterisk SibRw0rk (n=DaPhrek@66.234.235.84) |
| 14:00.02 | tzafrir_laptop | BladeRunner05, sorry, not sure |
| 14:00.10 | Katty | arrow |
| 14:00.13 | BladeRunner05 | tzafir: I see this upper Makefile:485: .config: No such file or directory |
| 14:00.21 | tzafrir_laptop | RH9??? upgrade upgrade upgrade |
| 14:00.30 | NirS | hey tzafrir |
| 14:00.33 | tzafrir_laptop | hi |
| 14:00.37 | NirS | wassup ? |
| 14:00.42 | NirS | you coming to OSDC in Feb? |
| 14:00.52 | Stephnie | NirS : this file has created alot of troubles for me ...yesterday I was unable to install MYSQL due ti this file and today I have reinstalled RH9 but stucked at ASTCC installation... |
| 14:00.56 | Katty | OSDC? |
| 14:01.03 | cuco | NirS, i am having problems surfing into http://asterisk.org.il |
| 14:01.14 | NirS | it's www.asterisk.org.il |
| 14:01.17 | Katty | tzafrir_laptop: maybe cause it's free. |
| 14:01.24 | Katty | tzafrir_laptop: (= |
| 14:01.27 | NirS | Katyt: Open Source Developers Conference in Israel |
| 14:01.30 | cuco | NirS, i know... that is the problem ;-) |
| 14:01.30 | NirS | wait a minute |
| 14:01.33 | Katty | NirS: way out /there/ |
| 14:01.40 | BladeRunner05 | tzafir: check this http://pastebin.com/523913 |
| 14:01.44 | NirS | how many people here are from israel ? |
| 14:01.50 | saftsack | debian:~# faxmodem /dev/ttyS0 |
| 14:01.50 | saftsack | FIFO: open: No such device or address |
| 14:01.51 | saftsack | Oo |
| 14:01.52 | iDunno | Katty: why not? |
| 14:01.54 | tzafrir_laptop | (when they can use Centos, Fedora, another Distro, or pay for RHEL) |
| 14:02.01 | saftsack | /dev/ttyS0 is existing |
| 14:02.09 | Katty | iDunno: do i really need to list off all the reasons? |
| 14:02.29 | NirS | steph, you got me there |
| 14:02.30 | iDunno | Katty: nah - it's alright, as long as there are reasons :) |
| 14:02.34 | NirS | suggestion: dump redhat9 |
| 14:02.42 | tzafrir_laptop | NirS, any reason not to make asterisk.org.il an alias to wwww.asterisk.org.il ? |
| 14:02.49 | Katty | iDunno: k |
| 14:03.14 | *** join/#asterisk coppice (n=chatzill@223.143.17.210.dyn.pacific.net.hk) |
| 14:03.37 | cuco | iDunno, come... it's raining and stuf... it's so much fun here... |
| 14:03.37 | tzafrir_laptop | saftsack, /dev/tty |
| 14:03.37 | NirS | hold on |
| 14:04.08 | tzafrir_laptop | saftsack, /dev/ttyS0 "exists": is it from udev/devfs or just a static device file? |
| 14:04.08 | iDunno | ENOTENOUGHCASH |
| 14:04.16 | iDunno | and it's alll the way over to the right a bit. |
| 14:04.29 | saftsack | i think it is from devfs |
| 14:05.19 | tzafrir_laptop | saftsack, so it is a serial port. What did you expect to find there, a modem? internal card or a serial one? |
| 14:05.26 | NirS | done |
| 14:05.27 | saftsack | a modem |
| 14:05.31 | NirS | asterisk.org.il is also available now |
| 14:05.40 | saftsack | with cu i can connect on this modem |
| 14:05.41 | jozsab1 | <Aughey> : thanks, that is what i was looking for, it has conference and evrything |
| 14:05.49 | tzafrir_laptop | yey |
| 14:06.00 | BladeRunner05 | tzafir: have u seen the pastebin error msg... its seems 'cause it don't find header file at the position if looks for... |
| 14:06.20 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 14:06.37 | tzafrir_laptop | BladeRunner05, what pastebin file? |
| 14:06.49 | *** part/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com) |
| 14:06.51 | BladeRunner05 | tzafir: this one http://pastebin.com/523913 |
| 14:06.56 | *** join/#asterisk sack (n=sack@55.Red-83-55-220.dynamicIP.rima-tde.net) |
| 14:07.40 | *** join/#asterisk vixtor (n=victor@193.226.128.142) |
| 14:08.13 | vixtor | hello all |
| 14:08.37 | vixtor | i have a question for you.. if an agent is paused with PauseQueueMember, how can i retrieve its status, to know that it is paused? |
| 14:10.51 | *** join/#asterisk dsfr (n=dsfr@gateway.digium.com) |
| 14:12.30 | *** join/#asterisk techy (n=tecky@66.9.96.115) |
| 14:14.21 | BladeRunner05 | tzafir: have u seen ? |
| 14:15.05 | *** join/#asterisk ruza (n=ruza@holly.cervenytrpaslik.cz) |
| 14:15.16 | tzafrir_laptop | BladeRunner05, hmmm, I believe that the zaptel 1.2.2 tarball is missing a .version file |
| 14:15.29 | tzafrir_laptop | I had to add it to the Debian package |
| 14:15.36 | tzafrir_laptop | Anybody else encontered this? |
| 14:16.02 | fugitivo | kill the 1.2.2 release! |
| 14:16.02 | tzafrir_laptop | I forgot to file a bug |
| 14:16.03 | fugitivo | burn it! |
| 14:16.11 | fugitivo | :) |
| 14:16.28 | tzafrir_laptop | try: |
| 14:16.51 | tzafrir_laptop | make cleanecho 1.2.2 >.version |
| 14:16.54 | coppice | I just squash bugs. filing them seems pointlessly cruel :-) |
| 14:16.55 | tzafrir_laptop | make clean |
| 14:17.12 | tzafrir_laptop | echo 1.2.2 >.version |
| 14:17.12 | tzafrir_laptop | make |
| 14:17.16 | BladeRunner05 | tzafir: it say: make: *** No rule to make target `cleanecho'. Stop. |
| 14:17.34 | tzafrir_laptop | coppice, I tend to file [patch] bugs |
| 14:17.48 | tzafrir_laptop | yes, ignore that one |
| 14:17.57 | BladeRunner05 | now run make ? |
| 14:18.22 | tzafrir_laptop | BladeRunner05, I just hope that this is a related problem, but it should not hurt |
| 14:18.32 | tzafrir_laptop | so try running 'make', yes |
| 14:18.44 | BladeRunner05 | a lot of errors too.... |
| 14:19.11 | BladeRunner05 | and stop with make: *** [linux26] Error 2 |
| 14:19.20 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:20.07 | fugitivo | what package contains zlib.h in debian? |
| 14:20.28 | fugitivo | ok, found it |
| 14:21.00 | BladeRunner05 | fugitivo: is for me ? |
| 14:21.05 | fugitivo | no |
| 14:21.09 | BladeRunner05 | kk |
| 14:21.49 | jozsab1 | Is there a way to send instead of "m=audio 12252 RTP/AVP 0 101" only "m=audio 12252 RTP/AVP 0" on sip protocol ? (i think this means disabling RFC 8233) |
| 14:22.42 | trererre | hi everyone, i have a major problem in meetme, this is the scenario: a client connects to meetme using sip and ilbc codec, then some other clients connect also with SIP and whatever codec, if everyone leaves the conference leaving only a client(using ilbc) in the conference, the CPU of the machine goes up 10% and never recovers. |
| 14:22.50 | trererre | If someone enters the conference again, the CPU goes down again. If this scenario happens in 10 conferences for example, the CPU goes 100 %. Can someone help me out? I know this is a bug, but i cant find where is coming from, it is not directly dependent in meetme since version 1.2 with app_meetme 1.0.9 also has the bug, but if u use 1.0.9 * and app_meetme it works fine |
| 14:23.01 | *** join/#asterisk }btorch{ (n=kvirc@208.63.19.172) |
| 14:23.01 | trererre | can someone help me out |
| 14:23.23 | }btorch{ | is it possible to have #include in the voicemail.conf file ? |
| 14:23.31 | }btorch{ | whats up trererre ? |
| 14:24.37 | cuco | NirS, just seen the change 10x! |
| 14:24.46 | trererre | }btorch{, are u familiar with meetme? |
| 14:24.59 | }btorch{ | no haven't tried that yet |
| 14:25.05 | }btorch{ | sorry |
| 14:25.10 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 14:25.11 | }btorch{ | my go over that today though |
| 14:25.21 | *** join/#asterisk SteveGibbs (n=fds@bspark2dsl.claranet.co.uk) |
| 14:25.22 | SteveGibbs | Does anyone have experience of configuring a PervTech ASP-X box from scratch? |
| 14:25.29 | *** join/#asterisk nico__ (n=nico@34.101.94.80.dslz1.libello.mc) |
| 14:25.35 | _Sam-- | pervtech...nice name! |
| 14:25.45 | SteveGibbs | Hehe, have u heard of them? |
| 14:25.55 | _Sam-- | nope, but i am by no means any expert |
| 14:26.17 | SteveGibbs | I like their Creep range of servers |
| 14:26.25 | }btorch{ | no one knows if I can add #incluedes in any conf file ? |
| 14:26.49 | _Sam-- | i think you can in any conf file |
| 14:26.51 | _Sam-- | try it |
| 14:28.21 | *** join/#asterisk kink0 (n=k@62.37.205.161) |
| 14:29.17 | kink0 | I need urgent help with this error: Ext: 1 Cause: Unknown (100), class = Protocol Error (6) |
| 14:29.30 | hans | except I read you don't want to in voicemail.conf |
| 14:29.34 | kink0 | while I route a call from Asterisk E1 to 2N Stargate E1 port |
| 14:30.10 | *** join/#asterisk kiwnix (n=egarcia@77.red-82-158-158.user.auna.net) |
| 14:30.24 | kink0 | hola kiwnix |
| 14:30.53 | kiwnix | hola :) |
| 14:31.04 | kink0 | kiwnix, tu usas E1 ? |
| 14:31.31 | kiwnix | no, lo siento |
| 14:31.36 | *** join/#asterisk KeX (n=KeX@argos.campus-sbg.at) |
| 14:31.51 | kink0 | ok |
| 14:33.39 | fugitivo | kink0: estas usando la señalización correcta? |
| 14:35.26 | Katty | woah. |
| 14:35.30 | Katty | microsoft release source code. |
| 14:35.41 | Katty | this is pretty nutty. |
| 14:35.47 | fugitivo | yeah, they're tired of being stolen :) |
| 14:36.38 | RoyK | hm |
| 14:36.49 | RoyK | are cdr data available to the h extension? |
| 14:37.12 | coppice | Katty: not nutty. deceptively worded |
| 14:37.17 | SteveGibbs | I want to render a link_to_remote from my PervTech Stalker-Range controller, but I'm not really sure how to do this. do I have to create a partial just for the link_to_remote and render the partial? OR do I have to render_text the html output (created by hand) |
| 14:37.56 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 14:38.22 | *** join/#asterisk madounet (n=mad_net@juv34-2-82-226-155-19.fbx.proxad.net) |
| 14:38.23 | fugitivo | Katty: if they don't release source code, they lose money, a lot |
| 14:38.31 | Katty | so i read. 2 million a day. |
| 14:38.56 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 14:39.33 | Katty | but at least we found another planet! |
| 14:40.17 | Katty | -220C (= |
| 14:40.35 | iDunno | hmm. toasty warm ;) |
| 14:40.53 | RoyK | 52K is nice |
| 14:40.57 | iDunno | 100C or so above 0K |
| 14:41.12 | iDunno | oh, is it only 272? dammit. |
| 14:41.14 | SteveGibbs | ! |
| 14:41.35 | Katty | iDunno: yes, but it's /how/ they found it. |
| 14:41.40 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
| 14:42.11 | RoyK | 273.16 iirc |
| 14:42.13 | asteriskmonkey | can anyone help me with this? i have libtiff installed and spandsp says its not when i go to compile it . |
| 14:42.20 | RoyK | so 52.84, then ;P |
| 14:42.26 | *** part/#asterisk ckruetze (n=ckruetze@131.8.dsl3.ip.foni.net) |
| 14:42.38 | RoyK | asteriskmonkey: bug coppice about it :) |
| 14:43.08 | *** join/#asterisk Szolke (n=Szolke@217.116.36.22) |
| 14:43.24 | Szolke | Hi all. |
| 14:43.47 | jalsot | Hi szolke |
| 14:44.08 | fugitivo | debian installs by default postgresql 7.4, god, that's old |
| 14:44.14 | coppice | asteriskmonkey: are you using RPMs or similar? do you have the devel package installed? |
| 14:44.20 | asteriskmonkey | rpms |
| 14:44.23 | asteriskmonkey | yes |
| 14:44.27 | asteriskmonkey | both |
| 14:44.41 | *** join/#asterisk Delvar2 (n=irc@host-83-146-53-34.bulldogdsl.com) |
| 14:44.45 | SteveGibbs | Anyone want to watch me shitting on my webcam? |
| 14:44.51 | coppice | then it must be an illusion that it is not found :-) |
| 14:45.04 | asteriskmonkey | oh no wait |
| 14:45.09 | cpm | Naw, thanks though |
| 14:45.14 | asteriskmonkey | my libbtifdev has gone |
| 14:45.29 | asteriskmonkey | i only have libtiff-3.6.1-8 installed :P |
| 14:45.41 | asteriskmonkey | anyone know where to get the dev rpm for centos from? |
| 14:45.55 | SteveGibbs | . ÷. ].×Ð\@§¶¯þ·µß¿|¥.¸¯.©....Don't .+ |
| 14:45.55 | SteveGibbs | ~Ç%^ª#Ø<Îæ¡®´ø¢»$*Þ.°./"...Drink ..)³ |
| 14:45.57 | SteveGibbs | .`±¤¦~£Ä¬½}ÿå"«Ö_&(X:*..& Draw. ._*@^; |
| 14:46.04 | Katty | ... |
| 14:46.10 | *** join/#asterisk j0n (n=jellis@206-169-48-226.gen.twtelecom.net) |
| 14:46.11 | fugitivo | ok, ban him! |
| 14:46.21 | sivana | heh |
| 14:46.28 | coppice | looks like a UTF-8 problem |
| 14:46.36 | SteveGibbs | i need your attention guys |
| 14:46.50 | Katty | why? are you an attention-whore? |
| 14:46.59 | SteveGibbs | yes i am :) |
| 14:47.39 | j0n | is it possible to save the return value of an application to a variable in ael? |
| 14:47.53 | imb | pwd |
| 14:48.03 | Szolke | we wouldlike to use IAX2 trunk with gsm or speex codec on our servers. BUT during the call the callee allways hears stranger voices (like a guitar voice) |
| 14:48.29 | fugitivo | Szolke: ok, you need to call your local priest |
| 14:48.34 | sivana | heh |
| 14:48.42 | jono | what does all circuits busy mean? |
| 14:48.44 | SteveGibbs | i'll try a better one |
| 14:48.52 | SteveGibbs | .........//\\.......... |
| 14:48.54 | SteveGibbs | ......../// \\......... |
| 14:48.55 | SteveGibbs | .......|||..||........ |
| 14:48.57 | SteveGibbs | .......|||__||........ |
| 14:48.59 | SteveGibbs | Wohoo!A|||...|||A...... |
| 14:48.59 | Katty | jono: that usually means all outgoing lines at the provider are busy |
| 14:49.00 | SteveGibbs | Rowdie||||/°\||||...... |
| 14:49.01 | SteveGibbs | is hav|||||_|||||...... |
| 14:49.03 | SteveGibbs | ing a ||||Ž°Ž||||...... |
| 14:49.04 | SteveGibbs | Birth-|||||"|||||...... |
| 14:49.07 | SteveGibbs | day!!!||·/^^\·||...... |
| 14:49.07 | SteveGibbs | Time 4||.|| ||.||...... |
| 14:49.09 | SteveGibbs | Rocket||//| |\\||...... |
| 14:49.10 | SteveGibbs | Take-æ///| |\\\æ...... |
| 14:49.11 | fugitivo | jono: using a trunk, that means that the trunk is in use or busy |
| 14:49.12 | SteveGibbs | Off!/____/| |\____\cyfo |
| 14:49.13 | SteveGibbs | ·.°¿æ|..| || |..|æ¿°.· |
| 14:49.14 | Iam8up|lappy | can anyone tell me why amportal doesn't show up? these errors are shown in apache's error_log over and over... http://pastebin.com/523967 |
| 14:49.15 | SteveGibbs | .æ@((æ§@%VAVAV%@§æ))@æ. |
| 14:49.16 | SteveGibbs | ~ (@(@??)(?V?)(??@)@).~ |
| 14:49.18 | SteveGibbs | Space Travel |
| 14:49.20 | SteveGibbs | Rocks! |
| 14:49.21 | fugitivo | ~amp |
| 14:49.22 | jbot | it has been said that amp is NOT supported here! people using it should join #amportal |
| 14:49.27 | fugitivo | SteveGibbs: stop that |
| 14:49.31 | Szolke | fugitivo: thank you :D |
| 14:49.37 | Iam8up|lappy | ok, thanks |
| 14:49.43 | SteveGibbs | ok |
| 14:49.48 | Iam8up|lappy | * Added *!*n=fds@bspark2dsl.claranet.co.uk to ignore list |
| 14:50.27 | asteriskmonkey | oh now this sucks donkey dong.. anyone know how to fix this |
| 14:50.29 | asteriskmonkey | libtiff-devel needs libtiff = 3.5.7, this is not available. |
| 14:50.49 | asteriskmonkey | thats from a yum install libtiff-devel |
| 14:52.05 | Stephnie | Nirs: can you check this problem now ?? while installing ASTCC ? http://pastebin.ca/38666 |
| 14:53.31 | VJ | Hi guys |
| 14:53.51 | VJ | if i want to use g79 codec on asterisk |
| 14:54.05 | fugitivo | cool |
| 14:54.05 | VJ | should i specify allow=g729 |
| 14:54.09 | fugitivo | yes |
| 14:54.10 | Stephnie | yep |
| 14:54.20 | VJ | will this be sufficient, or something else is aso required? |
| 14:54.21 | fugitivo | but you need a license |
| 14:54.28 | VJ | ok |
| 14:54.29 | fugitivo | you need to install a license |
| 14:54.38 | VJ | how much will that cost |
| 14:54.49 | fugitivo | $10 |
| 14:54.56 | fugitivo | www.digium.com |
| 14:55.04 | VJ | any other codec which is freely available but not so heavy like alaw or ulaw |
| 14:55.07 | Stephnie | anyone please check http://pastebin.ca/38666 |
| 14:55.11 | VJ | which i am using currently |
| 14:55.12 | fugitivo | VJ: gsm |
| 14:55.39 | VJ | thats not supproted by AUDIOCODE gateway which i am using |
| 14:56.14 | fugitivo | well, what codecs does your gateway support? |
| 14:56.44 | j0n | does anyone know if it's possible to get the return value of an application in ael? |
| 14:56.49 | *** join/#asterisk _blop (n=blop@213-193-176-70.adsl.easynet.be) |
| 14:57.39 | VJ | ulaw, alaw, g729, g7231, netcode, g726, transparent |
| 14:58.07 | fugitivo | wel |
| 14:58.08 | VJ | couple of sub verions of g7231, netcode & g726 |
| 14:58.17 | fugitivo | go with g729, buy the license |
| 14:58.50 | *** join/#asterisk maggit (n=maggit@customer-200-36-59-130.uninet.net.mx) |
| 14:58.51 | mut | can anyone explain this? |
| 14:58.51 | mut | Jan 26 09:57:31 NOTICE[22621]: chan_sip.c:9689 handle_response_peerpoke: Peer '9896851082' is now REACHABLE! (31ms / 3000ms) |
| 14:58.58 | mut | Jan 26 09:58:07 NOTICE[22621]: chan_sip.c:11328 sip_poke_noanswer: Peer '9896851082' is now UNREACHABLE! Last qualify: 31 |
| 14:59.08 | mut | it's there |
| 14:59.20 | mut | but it drops all the time |
| 15:00.23 | *** join/#asterisk KeX (n=KeX@argos.campus-sbg.at) |
| 15:01.14 | Iam8up|lappy | are there other web based configs like AMP? |
| 15:02.20 | VJ | ok, this g729 needs to be procured from digium? |
| 15:04.18 | asteriskmonkey | damn it, cand seem to get libtiff dev installed becuase it cant get libtiff although libtiff is installed and im trying to get lib tiff out using rpm -r libtiff and i get error: Failed dependencies: |
| 15:04.18 | asteriskmonkey | <PROTECTED> |
| 15:04.43 | fugitivo | that's why i hate redhat based distros... |
| 15:04.46 | asteriskmonkey | sorry rpm -e |
| 15:04.54 | asteriskmonkey | yes gimme BSD ones anybad |
| 15:05.03 | Szolke | we wouldlike to use IAX2 trunk with gsm or speex codec on our servers. BUT during the call the callee allways hears stranger voices (like a guitar voice) CAn you help me? |
| 15:05.37 | Hmmhesays | guitar voice? |
| 15:06.01 | coppice | he's calling eric clapton |
| 15:06.07 | fugitivo | your asterisk boxes need an exorcism, call your local priest |
| 15:06.28 | *** join/#asterisk alx_ (n=alx@203.156.48.73) |
| 15:06.35 | asteriskmonkey | coppice: it says i need a different verison of libtiff and i cant seem to get the current one out |
| 15:06.50 | coppice | isn't life tragic |
| 15:06.53 | asteriskmonkey | well with rpm -e that is, do you know of another way? |
| 15:07.06 | fugitivo | asteriskmonkey: be a man, use sources, not packages |
| 15:07.11 | fugitivo | asteriskmonkey: life will be easier |
| 15:07.13 | asteriskmonkey | life is opensource |
| 15:07.32 | Hmmhesays | life is fucked up |
| 15:07.38 | asteriskmonkey | hey now :P point me to sources and im all over it |
| 15:07.40 | Hmmhesays | but then again i'm still drunk |
| 15:07.49 | fugitivo | google |
| 15:07.53 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 15:08.20 | *** join/#asterisk maggit (n=maggit@customer-200-36-59-130.uninet.net.mx) |
| 15:08.28 | asteriskmonkey | libtiff.org |
| 15:08.33 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 15:08.41 | alx_ | hello, i tested a sip provider last week, and it was working ok with asterisk... suddently, it stop working with asterisk but still works with software/ip phone.. what can i assume here? |
| 15:09.03 | fugitivo | alx_: what provider? |
| 15:09.13 | alx_ | the asterisk is running on a fixed ip |
| 15:09.27 | alx_ | fugitivo: local provider.. |
| 15:09.35 | *** join/#asterisk hypnox (n=dan@cornelyn.force9.co.uk) |
| 15:09.47 | kink0 | I need urgent help with this error: Ext: 1 Cause: Unknown (100), class = Protocol Error (6) |
| 15:09.48 | fugitivo | local provider? |
| 15:10.01 | alx_ | local sip provider in my country. |
| 15:10.04 | fugitivo | kink0: when do you get that error? |
| 15:10.25 | fugitivo | alx_: ask your local sip provider what changed |
| 15:10.31 | kink0 | fugitivo, when I send a call from Asterisk PRI Digium card to 2N Stargate PRI E1 port |
| 15:10.45 | alx_ | what im wondering is,.... can the provider be blocking my IP for accessing their server? |
| 15:10.46 | fugitivo | kink0: do you have the right signalling? |
| 15:10.55 | kink0 | I get that error while debuging PRI. |
| 15:10.56 | hypnox | quick question: has anyone ever encountered a "long beep" from voicemail? Everything is normal except the final beep tone is a continious tone. It doesn't appear on the recording. |
| 15:10.59 | fugitivo | alx_: what error do you get? |
| 15:11.06 | alx_ | i sent about 5 simulately calls with it ,.. the provider might not like that .. |
| 15:11.10 | fugitivo | kink0: are you sure you have to use PRI? |
| 15:11.15 | alx_ | i get 507 something ,.. |
| 15:11.16 | alx_ | 2sec |
| 15:11.19 | alx_ | i will check |
| 15:11.20 | kink0 | fugitivo, not sure at all, but calls from 2N PRI E1 to Asterisk console goes ok |
| 15:11.55 | kink0 | fugitivo, yes, I have a 2N PRI at one end, and a Digium TE405 at the other end |
| 15:11.57 | fugitivo | is that a channel bank or.. ? |
| 15:11.59 | alx_ | SIP/2.0 503 Service unavailable |
| 15:11.59 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
| 15:12.13 | fugitivo | alx_: sip show registry |
| 15:12.16 | fugitivo | alx_: is it registered? |
| 15:12.20 | alx_ | jepp |
| 15:12.26 | alx_ | registered without problem |
| 15:12.30 | kink0 | fugitivo, http://pastebin.com/523003 |
| 15:12.33 | fugitivo | mmm |
| 15:12.45 | kink0 | this is the debug at both sides, 2N and Asterisk |
| 15:12.45 | kpettit | so has anybody noticed any new problems with 1.2.3? |
| 15:12.47 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 15:13.26 | alx_ | i will give them a call as you told me,. . lets see what they have to say .. |
| 15:14.35 | asteriskmonkey | oh slutty day... the make has errors |
| 15:15.19 | *** join/#asterisk nico__ (n=nico@34.101.94.80.dslz1.libello.mc) |
| 15:17.15 | *** join/#asterisk nico__ (n=nico@34.101.94.80.dslz1.libello.mc) |
| 15:18.01 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:18.01 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:18.47 | asteriskmonkey | i get a g++ error but i have gcc installed |
| 15:19.22 | kink0 | fugitivo, I am ussing this: signalling=pri_net |
| 15:19.22 | kink0 | switchtype=euroisdn |
| 15:19.22 | kink0 | <PROTECTED> |
| 15:19.38 | *** join/#asterisk Bambr (n=Bambr@213-35-235-26-dsl.end.estpak.ee) |
| 15:19.40 | *** join/#asterisk mmmToop (n=chatzill@dsl-146-122-237.telkomadsl.co.za) |
| 15:19.46 | kink0 | and span=1,1,0,ccs,hdb3,crc4 at zaptel.conf |
| 15:20.02 | *** part/#asterisk Grubs (n=Miranda@c211-28-119-169.eburwd3.vic.optusnet.com.au) |
| 15:20.26 | kink0 | asteriskmonkey, /usr/bin/g++ -> g++-gcc-3.3.4 |
| 15:20.56 | *** join/#asterisk natix (n=Bambr@213-35-235-26-dsl.end.estpak.ee) |
| 15:21.37 | natix | Hello. I was wondering if the configuration for a IAX terminal would be exactally the same as the configuration for a SIP terminal |
| 15:21.49 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 15:21.52 | kink0 | where must I change "asterisk" string by a numerical string like a callerid , at CGN-asterisk CDN-657897123 SEC ? |
| 15:22.24 | kink0 | that is the caller at PRI session |
| 15:23.09 | *** join/#asterisk thieumS (n=darkmind@bea75-1-82-234-122-35.fbx.proxad.net) |
| 15:23.41 | ManxPower | kink0, Caller*ID of "asterisk" will only be set if Asterisk did not receive any caller id informtion. |
| 15:24.23 | ManxPower | on a PRI you might want to do a Wait(1) at the beginngin of any extensions that are dialed from the outside to give Asterisk time to receive the PRI message with the caller id info in it. |
| 15:24.29 | kink0 | ManxPower, I originate the call from one asterisk box to other asterisk connected to the E1 |
| 15:24.56 | ManxPower | kink0, then the lack of the Wait(1) might be the problem. |
| 15:25.22 | kink0 | ManxPower, hmmmm ... let me try it, that is an easy to fix then. |
| 15:25.47 | ManxPower | The Wait(1) must be before anything that might answer the line. |
| 15:26.40 | *** join/#asterisk bkw__ (n=brian@70.103.248.130) |
| 15:28.00 | kink0 | ManxPower, yes, I set it as priority 1, and Dial Zap with priority 2 |
| 15:28.07 | fugitivo | natix: not exactly the same |
| 15:28.21 | kink0 | that is the only two lines I am ussing in my extensions now, but appears did not fix |
| 15:28.52 | ManxPower | kink0, no, you need it on the asterisk that RECEIVES the call |
| 15:30.05 | *** join/#asterisk AlexCTI (n=alex@weston-69.65.90.105.myacc.net) |
| 15:30.37 | *** part/#asterisk AlexCTI (n=alex@weston-69.65.90.105.myacc.net) |
| 15:30.41 | *** join/#asterisk AlexCTI (n=alex@weston-69.65.90.105.myacc.net) |
| 15:31.18 | kink0 | ManxPower, yes, I set in the second asterisk, where the call is getting and send to the E1 |
| 15:31.25 | *** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
| 15:31.25 | kink0 | ManxPower, http://pastebin.com/524034 |
| 15:31.36 | kink0 | that is what I got, even with a Wait(1) |
| 15:31.54 | natix | Ok let me be a little more specific. I don't have access to the asterisk server files but would the [general] context be the same in the sip.conf file as in the iax.conf file? Like if I was to just replace the iax.conf general context with the general context that I have in my sip.conf file would it work or are the specific settings for the iax.conf file? |
| 15:32.03 | kink0 | the mistake is I am able to log in the 2N and originate a call from the 2N PRI to the Asterisk PRI |
| 15:33.04 | fugitivo | kink0: did you talk with 2N? |
| 15:33.12 | fugitivo | i'm sure they'll give you support |
| 15:33.14 | ManxPower | kink0, So you have Asterisk -> Asterisk -> 2N -> Carrier? |
| 15:33.36 | ManxPower | OR do you have Asterisk -> 2N -> Carrier |
| 15:33.45 | ManxPower | Or do you have 2N -> Asterisk -> Carrier ? |
| 15:33.46 | kink0 | fugitivo, yes, I have been spoken this morning, they changed 2N PRI from TE/M to TE/S |
| 15:34.08 | kink0 | ManxPower, yes, that is the scenary, but first Asterisk , may be a Xlite or so instead |
| 15:34.15 | fugitivo | i think they work with asterisk, so I'm sure they know how to fix it |
| 15:34.18 | kink0 | any SIP originating |
| 15:34.23 | ManxPower | kink0, diagram the correct one. |
| 15:34.46 | kink0 | fugitivo, the told me this morning to go to Asterisk/Digium support |
| 15:34.56 | fugitivo | heh |
| 15:35.20 | kink0 | ManxPower, [Any SIP,like Asterisk, X-lite] --> ASTERISK --> 2N Stargate GSM gateway |
| 15:35.32 | kink0 | that is the exact scenary |
| 15:35.38 | ManxPower | kink0, in the sip.conf what is the callerid= line? |
| 15:35.42 | fugitivo | so, is it a PRI or not? |
| 15:35.43 | AlexCTI | Hi, everyone. I have a little issue I'm using my asterisk as a Gateway to send calls trough VoIP from my current PBX, I have 4 PRI connected between the PBX and Asterisk BOX, so if I dial manual, the calls goes true, but if i use the dialer, the call is generate but never completed, I think so this is a delay problem, somebody knows how cani fix it? |
| 15:36.00 | *** join/#asterisk SPoon_TSX (n=Administ@24.87.103.250) |
| 15:36.15 | cpm | heh |
| 15:36.23 | SPoon_TSX | hello everyone. I got a quick questions. |
| 15:36.36 | fugitivo | AlexCTI: what dialer? |
| 15:36.46 | kink0 | ManxPower, I have not any callerid= at my sip.conf, all are commneted out. |
| 15:36.57 | AlexCTI | I'm using Telesynergy PBX |
| 15:37.00 | ManxPower | kink0, that would be a problem then. |
| 15:37.06 | SPoon_TSX | I have Asterisk 1.2.2 installed but even when I use my IP Phone call another IP Phone, phone ring but no sound at all. What would be the problem? |
| 15:37.07 | fugitivo | AlexCTI: i don't know what is that |
| 15:37.19 | kink0 | ManxPower, ok, that is easy.. i put some callerid now and try |
| 15:37.23 | fugitivo | SPoon_TSX: read topic |
| 15:37.35 | cpm | topic |
| 15:37.41 | ManxPower | in the sip.conf entry for the X-line device put callerid=their name <thenumber> Where the number is without theleading 0 or 00 |
| 15:37.46 | fugitivo | SPoon_TSX: and read www.asterisk.org |
| 15:38.11 | *** join/#asterisk XeratuL (n=afsdf@208.195.215.222) |
| 15:38.29 | SPoon_TSX | DMAN! |
| 15:38.30 | AlexCTI | Fugitivo: I thinks so there is no te issuen on the PBX, because if I put the PRI direct to it the dialer works fine |
| 15:38.34 | *** join/#asterisk mmmToop__ (n=chatzill@dsl-146-122-237.telkomadsl.co.za) |
| 15:39.00 | *** join/#asterisk santiago (n=santiago@208.195.215.222) |
| 15:39.15 | *** join/#asterisk SplasPood (i=jwb@206.252.198.100) |
| 15:39.48 | kink0 | ManxPower, the same, I got the Error 6 even if I not send any call |
| 15:40.05 | kink0 | that is when I have restarted Asterisk with pri intense debug on |
| 15:40.05 | fugitivo | AlexCTI: and how does that dialer work? |
| 15:40.58 | kink0 | ManxPower, as 2N docs, appears to be something like Invalid Information Element Content |
| 15:41.00 | AlexCTI | simple, the dialer start to generate call to asterisk box and it to my VoIP carrier to complete the call |
| 15:41.27 | SPoon_TSX | fugitivo: Thanks for your hints, when I do make, do I need to stopo everything including my zaptel? |
| 15:41.27 | AlexCTI | the calls comes through zap to VoIp |
| 15:41.28 | fugitivo | AlexCTI: how does the dialer generate the call? |
| 15:41.47 | fugitivo | SPoon_TSX: what do you mean? compiling asterisk? |
| 15:42.26 | AlexCTI | Figitivo: You load a BD with phones to dialer and it start to Dial |
| 15:42.27 | SPoon_TSX | fugitivo: I mean when I do "make install" do I need to unload the zaptel module first? |
| 15:42.40 | fugitivo | SPoon_TSX: no |
| 15:42.50 | fugitivo | AlexCTI: so you are a telemarketer |
| 15:42.58 | AlexCTI | yes |
| 15:43.18 | cpm | heh |
| 15:43.32 | fdask | the default vpb.conf that gets installed when you do a make samples with asterisk has an error in it |
| 15:43.34 | fugitivo | are you happy with that? |
| 15:43.38 | fdask | where would i report that? |
| 15:43.50 | synthetiq | spoontsx jdm apexi n1 eibach greddy y0! |
| 15:44.08 | *** join/#asterisk burton (i=mimx@w201.ljudmila.org) |
| 15:45.34 | AlexCTI | Fugitivo: If i'm happy? |
| 15:46.40 | *** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 15:48.12 | *** join/#asterisk copantl (n=copantl@205.240.200.88) |
| 15:49.49 | *** join/#asterisk folsson (n=filip@h147n1fls32o985.telia.com) |
| 15:49.56 | SPoon_TSX | fugitivo: May I give you a kiss!!!! Man you just safe my life! |
| 15:50.44 | *** join/#asterisk Jkx (n=jkx@84.4.64.27) |
| 15:50.51 | *** join/#asterisk jpablo (n=jpablo@200.94.130.194) |
| 15:50.52 | Jkx | Hi all |
| 15:50.54 | fugitivo | well, i'm far from you for a kiss, but paypal will give me the same effect |
| 15:51.07 | fugitivo | :) |
| 15:51.36 | coppice | paypal can give you herpes? :-\ |
| 15:52.00 | fugitivo | errr, not, only a smile |
| 15:52.58 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-89.rockynet.com) |
| 15:53.35 | *** join/#asterisk Skumling (n=skumling@fw.sg12.dk) |
| 15:54.04 | jpablo | Hi, im having some problems with my PRI, first, some times when i place a call on it and something start sendings audio from the other side (said a record saying the phone doesn't exits) i can still hear a ringing tone in the background. |
| 15:54.07 | jpablo | any ideas? |
| 15:54.51 | kink0 | ManxPower, do you see something strange here http://pastebin.com/524066 ? |
| 15:54.59 | kink0 | jpablo, do you have E1 ? |
| 15:56.30 | jpablo | kink0: yes |
| 15:57.17 | ManxPower | kink0, nothing looks wrong to me |
| 15:57.31 | *** join/#asterisk qazwsx (n=qazwsx@142097.fln.virtua.com.br) |
| 15:57.52 | *** join/#asterisk apardo (n=apardo@62.97.121.92) |
| 15:58.03 | Skumling | Hrm, I've messed around with asterisk for a couple of weeks now, and thought I was starting to control it... I've got some Zap-interfaces runnning on the "outside" and some Zap and SIP devices on my inside. Now I've added a external SIP-account, but I simply can't get incoming calls working :-/... |
| 15:58.25 | *** join/#asterisk Simon-_ (i=byte@proxima.arlott.org.uk) |
| 15:58.28 | jpablo | kink0: any sugestions ? |
| 15:58.45 | *** join/#asterisk baconbuttie (n=bob@213-162-112-249.knetso161.adsl.metronet.co.uk) |
| 15:59.23 | baconbuttie | 'lo |
| 15:59.31 | brad_mssw | Skumling: sounds like a nat issue ... make sure you have nat=yes for that sip setup both on the sip device and within sip.conf ... also make sure you have externip= set properly inside of sip.conf |
| 15:59.50 | brad_mssw | Skumling: if you still have problems, make your SIP client use a STUN server |
| 16:00.12 | brad_mssw | Skumling: so that it can determine it's own IP address and NAT type |
| 16:00.18 | jlewis | is it possible to hangup a zap channel from the CLI, or do I have to use the manager interface? |
| 16:00.29 | Skumling | brad_mssw: I thought so too, but now I've moved the box to a routed IP-address, and that doesn't help |
| 16:00.57 | Skumling | when I tcpdump, I see that my VoIP provider contacts me on port 5060, when I dial the number |
| 16:01.10 | Skumling | http://pastebin.com/524072 < that is all I get in my /etc/asterisk/full |
| 16:01.19 | Skumling | no output on the console at all |
| 16:01.25 | wunderkin | jlewis, soft hangup zap/... |
| 16:01.46 | kend | If I want to disable echo cancellation in a dialplan, how do I do that? |
| 16:01.47 | jlewis | excellent |
| 16:01.48 | brad_mssw | Skumling: what version of asterisk ? |
| 16:01.57 | *** part/#asterisk jono (n=jono@mail.openadvantage.org) |
| 16:02.12 | ManxPower | kend, I don't believe you can. Why do you want to do that? |
| 16:02.23 | kend | Because I've got an incoming fax. |
| 16:02.27 | Skumling | brad_mssw: 1.2.1-bristuffed |
| 16:03.08 | ManxPower | kend, echo cancel is automatically disabled when Asterisk detects the fax tone. |
| 16:03.16 | ManxPower | kend, well, on zap, of course. |
| 16:03.24 | kend | Yeah -- but my auto-detection has been working poorly. |
| 16:03.29 | brad_mssw | Skumling: k ... not sure what '/etc/asterisk/full' is, considering I don't have one ... if you bind to the running asterisk like asterisk -Rvvvd what do you get ? |
| 16:03.41 | kend | So I finally said, "Screw it. Faxes come in on these DID's. Send those to rxfax." |
| 16:03.51 | kend | But now the page format's all weird, probably because of timing/latency. |
| 16:03.58 | brad_mssw | Skumling: also, is there a NAT at all between your asterisk server and your sip client ? |
| 16:03.59 | Skumling | brad_mssw: in this testing phase, I'm always starting asterisk with -vvvvvvvvcg |
| 16:04.38 | Skumling | brad_mssw: yeah, but I'm not using the SIP clients right now - I'm testing via a ISDN phone on a Zap-interface |
| 16:05.02 | kink0 | jpablo: I am in a fight also with my E1 |
| 16:05.09 | brad_mssw | Skumling: oh, you're just going to a voip provider ... and you can't get incoming calls |
| 16:05.18 | *** join/#asterisk sherbang (n=sherbang@69.182.224.2) |
| 16:05.19 | Skumling | brad_mssw: when dialing in on the SIP-phonenumber, I get no output on the console at all... when dialing in to ISDN, I get lot's of information |
| 16:05.44 | Skumling | brad_mssw: that's right - my local SIP-phones are working fine |
| 16:05.44 | jpablo | kink0: mine works mostly fine, i just have a problem when i hear two dialtones |
| 16:05.50 | brad_mssw | Skumling: what does it do when you dial? just give you a busy signal? |
| 16:06.02 | brad_mssw | Skumling: you are registering with your sip provider, right? |
| 16:06.13 | brad_mssw | Skumling: when you do a 'sip show registry' it says, registered, right ? |
| 16:06.19 | Skumling | brad_mssw: hrm, here in Denmark we call it at "do-re-mi"... the three tones |
| 16:06.33 | ManxPower | kend, We use analog lines outside of Asterisk for faxex. |
| 16:06.48 | *** part/#asterisk madounet (n=mad_net@juv34-2-82-226-155-19.fbx.proxad.net) |
| 16:06.57 | Skumling | brad_mssw: I suppose so, since I can see in the log-file, that i registrers properly... 2 sec. |
| 16:07.06 | *** join/#asterisk ravenpi (n=chatzill@londonderry-cuda1-68-234-68-160.lndnnh.adelphia.net) |
| 16:07.13 | Skumling | brad_mssw: voip9.telsome.com:5060 36930822 105 Registered |
| 16:07.15 | Skumling | oh yes |
| 16:07.58 | Skumling | brad_mssw: when testing, I'm calling the number up from my cell |
| 16:08.15 | *** join/#asterisk ravenpi (n=chatzill@londonderry-cuda1-68-234-68-160.lndnnh.adelphia.net) |
| 16:08.15 | brad_mssw | Skumling: and you don't have a firewall blocking ports 5060,10000-20000 or a NAT between the 2 points, correct? |
| 16:08.31 | *** join/#asterisk brc_ (n=brian@pdpc/supporter/basic/brc) |
| 16:09.08 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfnj6.dialup.mindspring.com) |
| 16:09.18 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 16:09.18 | Skumling | brad_mssw: correct ;)... the box running asterisk has a global IP address directly on its eth0 interface - no filtering or NAT |
| 16:09.32 | Skumling | no iptables rules on the box itself neither |
| 16:09.35 | kend | ManxPower: Hmmm. I guess I could have a POTS line come in, with my DID's pointing to that. That makes sense. |
| 16:09.41 | SeicherlBoB | hi there! i can connect with idefisk to asteriks via vpn and make an outbound call. but when i try to reach an internal extension i only come to the voicebox. (Asterisk 1.10.010) |
| 16:10.19 | brad_mssw | Skumling: ok, if you do a sip debug ip voip2.telsome.com then try to call, do you get debugging on your console? |
| 16:10.23 | *** join/#asterisk Johnnie (n=jdlewis@dynamic-acs-24-154-53-16.zoominternet.net) |
| 16:10.31 | Skumling | brad_mssw: 2 sec... |
| 16:10.34 | thieumS | somebody's using spandsp and * 1.2.1 ? |
| 16:11.00 | brad_mssw | thieumS: 1.2.3, and spandsp-0.0.2-pre23 ... works fine |
| 16:11.03 | kend | thieumS: I'm using spandsp and 1.2.1... |
| 16:11.04 | coppice | thousands are using spandsp and 1.2.1 |
| 16:11.07 | Skumling | brad_mssw: oh yes... plenty of output |
| 16:11.22 | brad_mssw | Skumling: ok, pastebin that |
| 16:11.29 | kend | (Sadly, having some timing issues -- as noted above. ;-) |
| 16:11.30 | *** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfnj6.dialup.mindspring.com) |
| 16:11.32 | fugitivo | i do |
| 16:11.44 | thieumS | ok, which release for spandsp and 1.2.1 ? |
| 16:11.48 | Skumling | brad_mssw: gimme a sec, it filled my putty's buffer, I'll have to turn on some logging... |
| 16:12.30 | *** join/#asterisk tuxinator_linux (n=tuxinato@70-32-106-248.ontrca.adelphia.net) |
| 16:12.51 | fugitivo | i don't know |
| 16:12.59 | fugitivo | coppice: how do i check my spandsp version? |
| 16:13.28 | asteriskmonkey | kink0: i have gcc but no /use/bin/g++ |
| 16:13.43 | asteriskmonkey | i though g++ was a part of gcc |
| 16:13.57 | thieumS | 0.0.2(release) |
| 16:14.08 | Skumling | brad_mssw: http://pastebin.com/524092 |
| 16:14.13 | brad_mssw | asteriskmonkey: you can choose not to compile g++ when you compile gcc |
| 16:14.23 | SeicherlBoB | can anybody help me, plz! I can't make an internal call and dont know why. |
| 16:14.33 | brad_mssw | thieumS: there is no official 0.0.2 release, it's all pre-XX afaik |
| 16:14.53 | asteriskmonkey | ah : P doh... so i guess ill ahve to redownload it and go through the notes |
| 16:14.55 | kink0 | jpablo: porfa, pasame tus zaptel.conf y zapata.conf :) |
| 16:15.05 | *** join/#asterisk svenna_ (n=svenna@p548D1C37.dip0.t-ipconnect.de) |
| 16:16.25 | *** join/#asterisk _Thor (i=CS@user-vc8fl7l.biz.mindspring.com) |
| 16:17.25 | brad_mssw | Skumling: and that's the end of the output, I take it ? |
| 16:17.31 | jpablo | kink0: que tipo de e1 tienes ? |
| 16:17.36 | Skumling | brad_mssw: god damnit, seems I were just missing the "s" in the wmc-incoming context? |
| 16:17.40 | Skumling | brad_mssw: yes :) |
| 16:18.09 | kink0 | jpablo, lo que tengo es conectado una digium TE405 ( en modo E1 ) a un PRI E1 de un 2N |
| 16:18.26 | kink0 | del 2N puedo mandar al Asterisk, pero al reves, que es lo que se pretende, ni forma |
| 16:18.41 | fugitivo | kink0: estas seguro que estas utilizando la senalizacion correcta? |
| 16:18.43 | kink0 | da un error de protocolo que el 2N dice que es un Elemento con datos no validos |
| 16:18.48 | brad_mssw | Skumling: yep, it's generating a SIP/2.0 404 Not Found |
| 16:18.51 | Skumling | brad_mssw: I added a exten => s,1,Dial(zap/g3/21,30) under wmc-incoming, and now my ISDN phone here on my desk is dailing... |
| 16:18.54 | kink0 | fugitivo, no, ya no estoy seguro de nada |
| 16:19.04 | fugitivo | kink0: has probado con fxo, fxs, etc? |
| 16:19.18 | fugitivo | kink0: si trabaja igual que un channel bank, deberias usar esa senalizacion |
| 16:19.45 | kink0 | fugitivo, no del todo, no las he probado todas, salvo fxsks, que era lo mas probable, en caso de ver al otro como un banco |
| 16:19.46 | brad_mssw | Skumling: yeah, you shouldn't need an s=> extension unless you're using a proxy ... typically for coming into a PBX system, you don't use a proxy ... you really want sip with DNIS support |
| 16:19.59 | fugitivo | kink0: y con fxsks no funciono? |
| 16:20.11 | Skumling | brad_mssw: I just didn't think that could be the problem, as when I earlier have had similar problems, it has outputtet something about "missing 's'" right on the console without need to turn debuggning on |
| 16:20.13 | brad_mssw | Skumling: so that you get the phone number that was dialled, and in your context, it would require an extension matching your phone number |
| 16:20.17 | kink0 | fugitivo, daba otro problema, espera, te digo cual era, lo pongo otra vez con fxsks |
| 16:20.35 | *** join/#asterisk ibob63 (n=hp@bb-87-82-6-79.ukonline.co.uk) |
| 16:21.13 | ibob63 | hi guys,iis there a php/mysql opensource asterisk gui? |
| 16:21.19 | Skumling | brad_mssw: I had a "exten => 36930822,1,Playback(what-are-you-wearing)" test-entry in my dialplan... I thought that the number was passed by to me |
| 16:21.33 | brad_mssw | Skumling: yes, exactly, it _should_ have been passed to you |
| 16:21.34 | Skumling | brad_mssw: doesn't my VoIP provider act as a proxy? |
| 16:21.50 | brad_mssw | Skumling: a lot of times they have additional settings for your account |
| 16:21.55 | *** join/#asterisk tecnico_ (n=tecnico@user-24-236-120-2.knology.net) |
| 16:21.58 | brad_mssw | Skumling: or a different address you should register to |
| 16:22.19 | Skumling | brad_mssw: ah okay... well well, I'm learning all the time :) |
| 16:22.23 | brad_mssw | Skumling: those dumb sip clients don't want the exten/phone number passed to them, but a PBX you do |
| 16:22.27 | Skumling | brad_mssw: thanks very much for your help :) |
| 16:23.05 | SeicherlBoB | can anybody help me, plz! I can't make an internal call and dont know why. I get forwarded to the mailbox automatically. |
| 16:23.18 | Skumling | (now I just need to figure out, why it chooses the wrong context... I've got 2 SIP-accounts at the same provider, but it chooses the wrong one) |
| 16:25.26 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-81-201.tvcablenet.be) |
| 16:28.17 | kink0 | fugitivo, no, asi da muchos errores continuamente, si lo pongo como fxsks |
| 16:29.05 | Inkubot | español! |
| 16:29.09 | Inkubot | ee que bien |
| 16:30.54 | *** join/#asterisk HamYaI (i=HamYai@125.24.8.239) |
| 16:30.57 | fugitivo | kink0: que errores? |
| 16:31.18 | *** join/#asterisk DruidFyr (n=druidfyr@rrcs-24-73-79-30.se.biz.rr.com) |
| 16:31.38 | *** join/#asterisk mkrufky (n=mk@68.160.103.77) |
| 16:31.38 | kink0 | fugitivo, tu tienes tu E1 como fxsks ? o en Argentina son T1 ? |
| 16:32.09 | kink0 | jpablo, me pasas la parte de config de tu E1 ? para ver que estoy yo haciendo mal, porque no me saca ni una sola llamada. |
| 16:32.09 | fugitivo | E1, pero no son PRI, sino R2 |
| 16:32.40 | kink0 | fugitivo, el caso es que si lo pongo como PRI, me permite enviar llamadas desde el 2N al Asterisk |
| 16:33.03 | kink0 | pero al reves no va, desde el ASterisk -> 2N es donde falla, porque dice algo de un elemento con datos no valido |
| 16:33.12 | fugitivo | kink0: probaste con el otro PRI? |
| 16:33.22 | fugitivo | uno es CPE, el otro network side |
| 16:33.38 | kink0 | si, si lo pongo en cpe se queja de que el otro extremo tambien quiere ser cpe |
| 16:33.57 | kink0 | asi que si el otro es cpe, este lo tengo que poner en modo net, claro |
| 16:34.12 | fugitivo | pri_net? |
| 16:34.16 | kink0 | si |
| 16:34.24 | fugitivo | y no hay nada en el 2N que pueda cambiarse con respecto a eso? |
| 16:34.27 | kink0 | asi lo tengo, es lo mas proximo a funcionar, si no fuera por ese error en esa trama |
| 16:34.57 | kink0 | no... lo que permite cambiar se ponerlo como TE o NT, master o slave, y el TEI , nada mas. |
| 16:36.09 | SeicherlBoB | I cant make internal calls. I always get forwarded to the voicebox. what am i doin wrong? |
| 16:36.45 | *** part/#asterisk qazwsx (n=qazwsx@142097.fln.virtua.com.br) |
| 16:36.46 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 16:36.59 | wunderkin | heh, here we go again |
| 16:37.24 | *** join/#asterisk cron (n=cron@blk-224-156-15.eastlink.ca) |
| 16:38.22 | cron | anyone with a gxp-2000 try out the new 1.0.2.3fw yet? |
| 16:38.46 | *** join/#asterisk batphone (n=will@69.15.174.114) |
| 16:38.54 | *** part/#asterisk batphone (n=will@69.15.174.114) |
| 16:39.38 | _Sam-- | cron: never heard of that firmware for the gxp2000...do you have a url? |
| 16:39.53 | _Sam-- | the latest beta , at least of when i checked last nite, is 1.0.13 |
| 16:39.53 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 16:40.06 | *** join/#asterisk fa_back (i=faceoff@devel.acdbddh.eu.org) |
| 16:40.37 | fugitivo | kink0: si lo pones como slave y pones el asterisk como pri_cpe? |
| 16:41.10 | cron | _Sam--: http://www.grandstream.com/BETATEST/GXP2000/ |
| 16:41.16 | *** join/#asterisk ToeJam (n=ToeJam@jam.it.utah.edu) |
| 16:41.22 | cron | _Sam--: link came from http://www.voip-info.org/tiki-index.php?page=GXP-2000 |
| 16:41.34 | _Sam-- | i was on the tiki page last night... |
| 16:41.37 | *** join/#asterisk bkw__ (n=brian@70.103.248.130) |
| 16:41.37 | *** part/#asterisk ToeJam (n=ToeJam@jam.it.utah.edu) |
| 16:41.45 | _Sam-- | well what do you know, its from today.... |
| 16:42.30 | _Sam-- | why dont oyu install it and let me know how it works :) |
| 16:42.48 | cron | _Sam--: no mention of echo problems resolved in the release notes :( |
| 16:42.56 | _Sam-- | not true |
| 16:43.11 | _Sam-- | gxp2000.bin: |
| 16:43.12 | _Sam-- | . Added Acoustic Echo Cancellation (AEC) with Acoustic Gain Control (AGC) for |
| 16:43.16 | cron | _Sam--: give me a week or so when my other gxp comes back onsite |
| 16:44.22 | _Sam-- | i'll give it a go in a few minutes when im done lunch |
| 16:44.46 | *** part/#asterisk SplasPood (i=jwb@206.252.198.100) |
| 16:44.56 | Inkubot | kink0 o fugitivo ayer estuve instalando una x101p (clon) y fue muy raro, pero las llamadas llegaban al Asterisk, pero al contestar en un cliente (X-Lite) no existia audio, de hecho no existia audio para ninguna llamanda a traves de la linea analogica, se les ocurre que puede ser ? |
| 16:45.05 | Inkubot | ademas de la posibilidad de q la tarjeta este mala |
| 16:45.33 | _Sam-- | Inkubot: nat / firewall |
| 16:45.46 | cron | _Sam--: don't wreck a set just curious if it did help out, I cannot wait to get a remote users back onsite to test it out :) |
| 16:46.09 | _Sam-- | nah i know those phones pretty good...and how to fix them when they're dead. |
| 16:46.41 | *** join/#asterisk batphone (n=will@69.15.174.114) |
| 16:46.51 | _Sam-- | if you take an ethereal dump of the phone when they die...most of the time they are sending out an arp request. if you can respond to that arp request with the right ip, then you can fix dead gxp2000s |
| 16:47.15 | *** join/#asterisk frenzy (n=frenzy@196.45.144.41) |
| 16:49.34 | fugitivo | Inkubot: que version de asterisk estas usando? |
| 16:49.49 | Inkubot | 1.2.3 |
| 16:49.59 | Inkubot | quedo todo bien |
| 16:50.05 | Inkubot | a nivel de lan funciona perfecto |
| 16:50.14 | Inkubot | el problema fue la linea analogica |
| 16:50.26 | frenzy | any open source billing available for asterisk to do wholesale termination |
| 16:50.41 | Inkubot | de hecho, configure para que cuando una llamada llegara la mandara a un cliente X-Lite y la llamada llega, pero al contestar no hay audio |
| 16:50.49 | _Sam-- | inkubot : que occuridad cuando llama una extensione internale? (sorry for bad spanish/mexican/argentinian |
| 16:50.50 | Inkubot | en ninguna de las 2 partes |
| 16:51.11 | Inkubot | _Sam--: work just fine |
| 16:51.38 | Inkubot | _Sam--: a internal call works great |
| 16:51.43 | Inkubot | no problem at all |
| 16:51.59 | Inkubot | i think the proble maybe is the card |
| 16:52.07 | Inkubot | proble/problem |
| 16:52.20 | *** join/#asterisk gammacoder (n=chatzill@64-132-192-33.gen.twtelecom.net) |
| 16:52.25 | batphone | i have a problem where calls get dropped unexpectedly in a 30 seat location |
| 16:52.28 | batphone | really sporadic |
| 16:52.35 | *** join/#asterisk agreenberg (n=alex@64.201.13.170) |
| 16:52.36 | batphone | i can find no errors on the console |
| 16:52.40 | file | okay |
| 16:52.41 | batphone | just that they hang up |
| 16:52.42 | *** join/#asterisk jonathh (n=asd@host86-142-221-157.range86-142.btcentralplus.com) |
| 16:52.44 | *** part/#asterisk jonathh (n=asd@host86-142-221-157.range86-142.btcentralplus.com) |
| 16:52.55 | file | does anybody have a contact at Global Crossing who can help me with a routing issue I think they have with one of our IP assignments? |
| 16:52.58 | file | please say yes |
| 16:53.02 | batphone | ... the only packet loss i can find is across a switch near the edge |
| 16:53.23 | batphone | it loses maybe 3 out of 1000 packets in a flood |
| 16:53.32 | batphone | but i wouldnt think that is enough to kill a call |
| 16:54.06 | *** join/#asterisk cidez (n=sdgseg@modemcable159.198-81-70.mc.videotron.ca) |
| 16:54.29 | _Sam-- | Inkubot: ¿cuándo usted llama externo, los anillos externos del teléfono? |
| 16:54.55 | *** join/#asterisk SplasPood (i=jwb@206.252.198.100) |
| 16:55.15 | _Sam-- | sorry, bad translation |
| 16:55.37 | Inkubot | _Sam--: jajaja i think that my english is a little better than your Spanish, _Sam-- talk in english |
| 16:55.40 | Inkubot | kaka |
| 16:55.42 | Inkubot | jaja |
| 16:56.03 | _Sam-- | hah...when you call the external phone, the external phone rings , but just has no audio? |
| 16:56.09 | [TK]D-Fender | ~seen jbalcomb |
| 16:56.20 | jbot | jbalcomb is currently on #asterisk (2d 27m 55s). Has said a total of 57 messages. Is idling for 1d 19h 29m 21s, last said: 'by his mom'. |
| 16:56.36 | Inkubot | nops.. when i call to the external phone nothing happens |
| 16:56.48 | jbalcomb | [TK]D-Fender hello |
| 16:56.52 | Inkubot | but in the CLI i can see that the call is made |
| 16:56.58 | wunderkin | batphone: zap? |
| 16:57.11 | *** join/#asterisk GraGra (n=GraGra@p5499021B.dip0.t-ipconnect.de) |
| 16:58.33 | [TK]D-Fender | There you are :) |
| 16:59.46 | Inkubot | _Sam--: any idea ? |
| 17:00.31 | Skumling | argh, I don't get this. 2 different accounts at the same VoIP provider. Two different sections in sip.conf pointing to two different contextes - but asterisk chooses the same context for incoming calls via both of the SIP-links (and it's not the default-context set in the top of sip.conf)... any ideas? |
| 17:01.26 | *** part/#asterisk GraGra (n=GraGra@p5499021B.dip0.t-ipconnect.de) |
| 17:01.43 | *** join/#asterisk Lurr (n=pr0ph3t@host-63-69-20-3.compusource.net) |
| 17:01.47 | brc_ | *sigh* |
| 17:01.51 | *** part/#asterisk Lurr (n=pr0ph3t@host-63-69-20-3.compusource.net) |
| 17:02.05 | brc_ | so asterisk has magically decided to stop passing audio for phone to phone and phone to pstn calls |
| 17:02.14 | brc_ | works fine for playing audio files (vm) |
| 17:02.49 | *** join/#asterisk EksilAndyCap (n=aoy@pdpc/supporter/sustaining/AndyCap) |
| 17:03.23 | brc_ | seems asterisk isn't forwarding any rtp |
| 17:03.35 | cpm | brc_ what version? |
| 17:03.49 | brc_ | 1.2-r7831 |
| 17:03.53 | cpm | topic |
| 17:04.03 | brc_ | har har |
| 17:05.18 | brc_ | know off hand which of the critical bugs the fix is for? |
| 17:05.27 | _Sam-- | the one that is affecting you. |
| 17:05.47 | _Sam-- | the one that doesnt allow bridged calls to work. |
| 17:06.11 | brc_ | 6349? |
| 17:06.17 | brc_ | aite |
| 17:06.28 | *** part/#asterisk JimmyGulp (n=james@ns0.esagroup.co.uk) |
| 17:07.36 | brc_ | anybody familiar with the bug? Once it broke (with no changes) it's stayed broken even through restarts |
| 17:07.49 | file | it's to do with time |
| 17:07.53 | Katty | file: yes. |
| 17:08.14 | *** join/#asterisk troy|A (n=central@CPE00907f17e478-CM014300013422.cpe.net.cable.rogers.com) |
| 17:08.15 | Katty | file: actually, i don't |
| 17:08.19 | Katty | file: but you still love me, right? |
| 17:08.21 | brc_ | !muffin for file |
| 17:08.25 | file | Katty: of course! |
| 17:08.28 | Katty | file: k |
| 17:08.36 | file | I don't like Global Crossing though :( |
| 17:08.39 | twisted[asteria] | can you feel |
| 17:08.40 | twisted[asteria] | a little lov |
| 17:08.42 | twisted[asteria] | +e |
| 17:09.02 | zoa | I dont want to feel your love, put that thing away! |
| 17:09.10 | jpablo | hey people, i hear two dialtones in my pri, anyone has an idea ? |
| 17:09.22 | _Sam-- | why does your PRI have a dialtone? |
| 17:09.24 | *** join/#asterisk Idle (n=brian@S010600a024969312.ed.shawcable.net) |
| 17:09.35 | _Sam-- | i thought the dialtone comes from the phones |
| 17:09.49 | Idle | so, what does asterisk provide me that cisco would not? |
| 17:09.55 | jpablo | _Sam--, sorry, maybe i made the wrong question. When i dial to a phone trought my pri i hear two dialtones, both with a grandstreamd and a sipura ata |
| 17:09.58 | _Sam-- | Idle: open source? |
| 17:10.01 | coppice | file: global crossing means hours in a bloody plane. i don't like that either :- |
| 17:10.21 | jpablo | _Sam--, no dialtones, riging tones, the tuuuuu, tuuuu thing |
| 17:10.26 | rob0 | Eh, Cisco! Eh, Pancho!! |
| 17:10.32 | zoa | jpablo: because the phone generates one |
| 17:10.38 | zoa | + there is a ringing tone on the pri |
| 17:10.49 | Idle | _Sam--: :) of course, but I'm thinking, if cost isn't a factor, for actual services, what does asterisk have extra? aside from obviously better interoperability |
| 17:10.49 | jpablo | zoa, what's the best way to fix it ? |
| 17:10.51 | batphone | wunderkin: zap? |
| 17:11.50 | _Sam-- | Idle: ive never worked for/with anyone where cost wasnt a factor...so ive never actually used and cisco pbx type stuff |
| 17:11.58 | _Sam-- | and = any |
| 17:12.03 | *** join/#asterisk Guggemand (i=Guggeman@tester2.har-tabt.dk) |
| 17:12.05 | Idle | ah, :) |
| 17:12.30 | Katty | twisted[asteria]: wuverly. |
| 17:12.31 | *** join/#asterisk ascas (n=sdgseg@modemcable159.198-81-70.mc.videotron.ca) |
| 17:12.37 | coppice | well, people seem to loath call manager, so i guess doing better than cisco shouldn't be too hard |
| 17:12.40 | twisted[asteria] | Katty, ? |
| 17:12.50 | Idle | all I've heard from a friend is that Asterisk has more, or does it better... but I've never really looked into what asterisk does better |
| 17:13.23 | Idle | I wanna sell Asterisk as an alternative to Cisco to my boss :) |
| 17:13.39 | twisted[asteria] | coppice, they loathe it (spelling corrected... loath means to leave) because it's a beas |
| 17:13.41 | twisted[asteria] | beast |
| 17:13.43 | _Sam-- | get him a cisco phone and call it a day |
| 17:13.47 | _Sam-- | connect it to asterisk |
| 17:13.51 | *** join/#asterisk markit (n=konversa@host119-245.pool8172.interbusiness.it) |
| 17:13.58 | Idle | ;) |
| 17:14.01 | markit | hi :) anyone using visdn, by chance? |
| 17:14.05 | Katty | twisted[asteria]: you. |
| 17:14.22 | twisted[asteria] | Katty, thanks methinks |
| 17:14.25 | cron | Idle: I really like the scripting language and the pluggable sources you can add to it |
| 17:14.27 | Katty | twisted[asteria]: (= |
| 17:14.39 | Idle | well, we're replacing a 3com call manager on a customer site, and he wants to use the godawefully expensive Call Manager, not only do I want to save money, I want to add more features |
| 17:14.53 | wunderkin | batphone: yes what technology are you using to call out or maybe for the agents too? zap iax sip.. |
| 17:14.59 | batphone | sip |
| 17:15.00 | Idle | cron: but is there pluggable sources that outdo ciscos features? |
| 17:15.15 | batphone | its driving me nuts |
| 17:15.21 | batphone | every now and then my ssh locks up |
| 17:15.24 | _Sam-- | Idle: i dont know how many seats/phones yuo are talking about for your system...but i would venture that you could implement an asterisk based solution for probably 50% less cost than a cisco system using all cisco....i dont know too many bosses/business owners that want to pay double for the same crap |
| 17:15.28 | batphone | sometimes it resumes after a few seconds |
| 17:15.30 | batphone | sometimes not at all |
| 17:15.36 | Idle | I love modularity as much as the next guy, but its nothing if its not utilized |
| 17:15.44 | batphone | i have a pretty extensive iptables setup on there |
| 17:15.52 | batphone | but have been able to duplicate the packet loss without it applied |
| 17:16.02 | znoG | is it bad for 2 TDM cards to share the same irq? |
| 17:16.06 | Idle | _Sam--: I remember seeing the license for Cisco call manager, something like 150,000 for 15 licenses... |
| 17:16.20 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-2.rcmdva.fios.verizon.net) |
| 17:16.23 | SpaceBass | hey folks |
| 17:16.23 | Idle | I would venture Asterisk could do it for alot less then 50% |
| 17:16.32 | *** join/#asterisk opensa_ivan (n=istepani@201.250.7.164) |
| 17:16.32 | tuxinator_linux | znoG: yes |
| 17:16.39 | zoa | idle, give us a call :p |
| 17:16.43 | [TK]D-Fender | Idle : With * you can do just about anything. Scripting allows you to make a cal go anywhere you want it to, lookup info, time fo day or just about any other routing you can imagin. Running on Linux it also lets you access all sort of other things. Mine can make coffe for instance :) |
| 17:16.51 | batphone | znoG: its not as bad as having them on the same logical PCI bus as, say, eth0 or hda |
| 17:16.58 | wunderkin | batphone, sounds like a network or system problem |
| 17:16.59 | Katty | tuxinator_linux: erro. |
| 17:17.03 | Idle | [TK]D-Fender: haha, :D nice |
| 17:17.19 | znoG | batphone: whats what I thought. I have 2 tdms AND the network card sharing an IRQ which could be the cause of the echo problems I'm having |
| 17:17.26 | tuxinator_linux | Katty: Morning Katty |
| 17:17.34 | batphone | what kind of problem znoG? |
| 17:17.37 | batphone | blips and garbling? |
| 17:17.40 | Idle | [TK]D-Fender: but cant you do that on cisco as well? (aside from making coffee of course) |
| 17:17.45 | znoG | batphone: i'm just not sure how I'll be able to trick the system into giving each an individual irq |
| 17:17.52 | Idle | the phone routing that is |
| 17:17.59 | znoG | batphone: echo at the beginning of a call for 5 seconds (until echotraining kicks in) then its OK |
| 17:18.08 | batphone | znoG: try "noirqdebug" kernel option and disable APIC |
| 17:18.24 | batphone | znoG: you have to move the cards around alot too |
| 17:18.37 | znoG | yeah I'm thinking of just moving them around until it works |
| 17:18.44 | [TK]D-Fender | Idle : How much control do you have over a call? Can you make it go ANYWHERE, do anything? If you want to add "features" like ACD, does it cost extra? If you want to use a non-cisco device, is there licensing? |
| 17:18.44 | batphone | znoG: this works for me using a PDSMA motherboard from supermicro, it has 4 tdm405p's in it |
| 17:18.48 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 17:18.56 | batphone | znoG: it handls 150,000 calls a day dude |
| 17:19.00 | znoG | batphone: i'm just wondering if the fact they're sharing an IRQ with eth0 is the problem |
| 17:19.01 | [TK]D-Fender | Idle : Ask yourself what Cisco offers... that a far easier question. |
| 17:19.22 | znoG | batphone: i would have expected a crackly phone call if it was sharing IRQ the problem, but its just the terrible echo at the beginning of a call, then it's fine. |
| 17:19.25 | Idle | [TK]D-Fender: true, :) |
| 17:19.32 | batphone | znoG: ideally you dont want anything sharing irq's, and i can pretty much guarantee that you are gonna have problems until you sort that out |
| 17:19.57 | znoG | batphone: yea, i think i'll go play with the CMOS and move the cards around a bit |
| 17:20.04 | cron | Idle: Sorry work related call, our dba is using ruby-agi for stats collection etc, but can provide some really nice live interaction with calls etc that we currently don't use but the features that can open up/built on it is amazing |
| 17:20.10 | Idle | [TK]D-Fender: but cost not being a factor, since my boss seems to have no problem with dropping in a 150,000$ cisco call manager |
| 17:20.22 | [TK]D-Fender | Idle : * fits the nedds of largest portion of the PBX market at a fraction of the cost offering far greater control and the freedom to choose from a massive market of standardized commodity equipment at a fraction of the price. What could Cisco possibly offer to compete with that? |
| 17:20.28 | batphone | znoG: that sounds like a different problem all together |
| 17:20.49 | [TK]D-Fender | Idle : Control & Freedom <- Theres the selling point for you then. |
| 17:20.50 | _Sam-- | Idle: figure a way to pocket the difference. |
| 17:21.03 | Idle | [TK]D-Fender: according to the Cisco guy here, it provides everything... I dont buy that tho |
| 17:21.21 | _Sam-- | "I can implement it for 80,000, why dont you give me 20,000 and i will make sure it runs 24/7" :) |
| 17:21.36 | Idle | configureability, modularity, coffee, thats some good extras. ;) |
| 17:21.40 | [TK]D-Fender | idle have them show you their version of "everything", and then invent a sick-ass scenario for them to explain the implementation of. |
| 17:21.56 | *** join/#asterisk maggit (n=maggit@customer-200-36-59-130.uninet.net.mx) |
| 17:22.03 | [TK]D-Fender | their* |
| 17:22.08 | _Sam-- | you easily use a scripting language (like php) and sql with cisco's stuff? |
| 17:22.18 | Idle | yea, thats true, gotta see what he wants to implement, and enhance it with Asterisk specific examples |
| 17:22.19 | EksilAndyCap | [TK]D-Fender: You know what they sell;a brand name. :-P |
| 17:22.22 | znoG | batphone: possibly, buggered if I know what it could be though. I've tried playing with the rxgain and txgain to no avail |
| 17:22.27 | cron | ie calling a voice menu and turning a coffee machine on and off and reading back the temp stats ;) |
| 17:22.45 | Idle | _Sam--: nope, :D I would really love to do that tho, with asterisk that is |
| 17:22.56 | [TK]D-Fender | cron : I'm working on adding Sphinx for just that purpose :D |
| 17:23.06 | *** join/#asterisk roulduke_ (i=qi51dmtl@p508D3F0B.dip0.t-ipconnect.de) |
| 17:23.10 | cron | [TK]D-Fender: your my hero! |
| 17:23.10 | [TK]D-Fender | "Computer COFFEE!!" |
| 17:23.17 | Idle | cron: as in, have it talk it out? or show it back on Caller ID/screen? both? ;) |
| 17:23.23 | _Sam-- | your boss wont care as long as he has a swanky cisco phone on his desk |
| 17:23.31 | cron | idle: all of the above |
| 17:23.33 | cpm | http://www.bgcfreedom.com/~jparr/callmanager.jpg Cisco call Manager |
| 17:23.35 | Idle | nice |
| 17:23.55 | *** part/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 17:24.04 | cron | wtf would you throw out the phones :( |
| 17:24.18 | [TK]D-Fender | Idle : Which you're at it , ditch Cisco and go Polycom :D |
| 17:24.25 | cpm | Because you are done with them, and never want to see one of those damed boxes again. |
| 17:24.33 | [TK]D-Fender | While* Dammit... circuits crossing a lot today... |
| 17:24.44 | SeicherlBoB | I cant make internal calls. I always get forwarded to the voicebox. what am i doin wrong? |
| 17:24.48 | _Sam-- | Idle: do you really think a non-technical kind of boss will really understand fundamental technical differences between cisco and *? |
| 17:24.50 | Idle | [TK]D-Fender: polycom? |
| 17:25.01 | _Sam-- | i dont think he/she will care what you use as long as it WORKS, and he/she has a nice phone |
| 17:25.11 | Idle | _Sam--: hes technical. :) |
| 17:25.18 | [TK]D-Fender | SeicherlBoB : Some DETAILS maybe? Pastebin your extensions.conf and for the love of God if this is an AMP question go to #amportal. |
| 17:25.20 | [TK]D-Fender | ~pb |
| 17:25.21 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 17:25.25 | *** join/#asterisk dionerata (n=me@201.138.171.210) |
| 17:25.25 | EksilAndyCap | _Sam--: and as long as all the employees remains easily replaceable. |
| 17:25.32 | Idle | and he likes his Cisco Channel Partnerships |
| 17:25.39 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 17:25.39 | *** mode/#asterisk [+o denon] by ChanServ |
| 17:25.44 | [TK]D-Fender | Idle : Yes... Polycom phones are far cheaper for the same functionality... |
| 17:25.49 | SeicherlBoB | [TK]D-Fender: I use the online-tool. the configs seem too complex for me at the beginning |
| 17:26.10 | cron | cpm: but they ebay for silly ammounts ;) |
| 17:26.11 | [TK]D-Fender | Idle : If you've got a great deal on Cisco's, then by all means, but if you're paying like the rest of us, then think twice :) |
| 17:26.23 | cpm | good day for dumpster diving. |
| 17:26.27 | SeicherlBoB | [TK]D-Fender: I use idefisk as client and i can connect to the server and make outbound calls. |
| 17:26.28 | [TK]D-Fender | SeicherlBoB : So you're running A@H? |
| 17:26.29 | _Sam-- | if cost is no object, dont think twice, just once |
| 17:26.41 | cpm | I was kinda shocked at the pic myself. |
| 17:26.42 | cron | cpm: no doubt :) |
| 17:26.56 | SeicherlBoB | [TK]D-Fender: APM or how it called |
| 17:27.02 | cron | hell those are 7940/60's... could still sip them ;) |
| 17:27.07 | SeicherlBoB | [TK]D-Fender: AMP, sorry |
| 17:27.23 | [TK]D-Fender | SeicherlBoB : Then ask in #amportal , you won't get much help here I'm betting.... |
| 17:27.31 | dionerata | hey guys i downloaded asterisk and was so excited however i have configured and reconfigured and i cannot get it to do the things i want it to. I set incoming service and outgoing, and used Xlite as softphone. Does someone have a real easy setup that i could use, any suggestions would be appreciated |
| 17:27.36 | SeicherlBoB | [TK]D-Fender: thx! |
| 17:27.40 | EksilAndyCap | cpm: but did someone do it or photoshop/gimp it? |
| 17:28.10 | Idle | [TK]D-Fender: ah, phones. :D |
| 17:28.29 | *** join/#asterisk felixD (n=felix@local.epyron.de) |
| 17:28.46 | Idle | yea, I'm just thinking about the call manager now, but when I'm looking at phones I'll look at polycom |
| 17:28.52 | _Sam-- | Idle: name a single advantage of using the cisco call manager software vs. asterisk (not trying to be a smartass, just curious) |
| 17:29.04 | drumkilla | EksilAndyCap: looks 'photoshopped', doesn't it ... |
| 17:29.33 | _Sam-- | and that is the same question i would ask the other person who is pro-call manager |
| 17:29.45 | _Sam-- | it costs more, but what is the ADVANTAGE of that cost |
| 17:29.51 | *** join/#asterisk pointer (i=pointer@aj.catt.com) |
| 17:29.53 | fugitivo | _Sam--: one word |
| 17:29.56 | fugitivo | _Sam--: SUPPORT |
| 17:30.11 | _Sam-- | they seem pretty technically adedpt |
| 17:30.12 | Idle | _Sam--: he seems to think it provides better service, as in, calls are setup faster under load |
| 17:30.38 | pointer | anyone here using VMCOUNT() in their dialplan? |
| 17:30.58 | pointer | it seems to be returning 0 no matter how many messages the mailbox has |
| 17:31.17 | SeicherlBoB | [TK]D-Fender: can you recommend a tutorial on Asterisk? I've read through a couple but they all... suck. |
| 17:31.41 | EksilAndyCap | SeicherlBoB: the book? |
| 17:31.48 | jbalcomb | ~docs |
| 17:31.50 | jbot | well, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 17:31.57 | jbalcomb | there you are SeicherlBoB |
| 17:32.20 | [TK]D-Fender | SeicherlBoB : Try the TFOT book, and the WIKI |
| 17:32.25 | jbalcomb | SeicherlBoB the wiki has been my best resource so far |
| 17:32.26 | [TK]D-Fender | ~docs |
| 17:32.27 | jbot | somebody said docs was probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 17:32.31 | [TK]D-Fender | ~thebook |
| 17:32.32 | jbot | well, thebook is Asterisk: The Future of Telephony, released under the Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 17:33.10 | Exstatica | haha digium is in my system trying to figure out my zaptel issue |
| 17:33.18 | _Sam-- | i actually, personally, believe that asterisk would probably outperform call manager on the same hardware...i am by no means an expert on how asterisk is programmed, but i believe that the benefit of the open source has probably helped asterisks call perforamance |
| 17:33.29 | cron | _Sam--: bbiab, drop me a message to let me know how the beta fw works out |
| 17:33.41 | _Sam-- | will do, thanks for reminding me |
| 17:33.48 | pointer | Exstatica: echo? |
| 17:33.50 | *** join/#asterisk Mark_Halverson (n=mhlvrs@67-139-119-152.dsl1.pco.ca.frontiernet.net) |
| 17:34.47 | batphone | this is some great wallpaper |
| 17:34.47 | RoyK | humtitum |
| 17:35.02 | Idle | _Sam--: me too, but he needs quantitative data.... :P |
| 17:35.25 | _Sam-- | i think the events you are talking about...handling max clients under extreme load...is probably not the common scenario... |
| 17:35.33 | RoyK | is there a way to reset the CDR counter, as in billsec? i get a call, answer it, do som IVR and dial another one, and I want the billsec from the last call, not the initial one |
| 17:35.36 | *** part/#asterisk SibRw0rk (n=DaPhrek@66.234.235.84) |
| 17:35.37 | SeicherlBoB | wow... ok. thanks. i'll have a look on those |
| 17:35.38 | _Sam-- | and if it were, you could just get another box :) |
| 17:36.10 | _Sam-- | but if he arguing the same box could handle more calls, then i would point him to the asterisk wiki where people with like 386s are handling 10,000 sip clients (sorry exaggerated) |
| 17:36.27 | Idle | :D |
| 17:36.41 | _Sam-- | but there are some hard numbers there..."P4 1G ram..handled 1000 simultaneous connections"...or something along those lines...its there |
| 17:36.55 | jbalcomb | Exstatica thats nice. how do you get them to do that? |
| 17:37.05 | Idle | I wanna setup an asterisk box at home, I have an FXO/FXS Digium card comming, but I'm gonna need a new machine to handle it all... |
| 17:37.06 | *** join/#asterisk crich1999 (n=crich@p54BFB774.dip0.t-ipconnect.de) |
| 17:37.06 | Exstatica | yeah i'm saving the session just so i can see how they fix their issues |
| 17:37.10 | _Sam-- | Idle: http://www.voip-info.org/wiki/view/Asterisk+dimensioning |
| 17:37.15 | Exstatica | well they might have to RMA my card but they want to check first |
| 17:37.30 | *** join/#asterisk Stephnie (i=Stephnie@u15157627.onlinehome-server.com) |
| 17:38.44 | _Sam-- | Idle: if money is no object, and someone else wants to use the cisco, i dont understand your motivation to push asterisk |
| 17:39.10 | Idle | cause I'm gonna be the one setting it up (and I've never used either of them)' |
| 17:39.11 | zoa | me neither |
| 17:39.13 | _Sam-- | are you in charge of administering the system? |
| 17:39.15 | zoa | go for cisco then |
| 17:39.25 | zoa | nobody gets fired for buying cisco |
| 17:39.37 | Exstatica | buy juniper |
| 17:39.43 | Exstatica | haha |
| 17:39.53 | Exstatica | you know those new linksys phones are really nice |
| 17:39.59 | cpm | heh |
| 17:40.04 | bsdfreak | heh |
| 17:40.13 | Idle | _Sam--: will be |
| 17:40.15 | _Sam-- | Idle: if you've never used asterisk and are going to implement yourself, in a big production environment...that is a big task for a first timer. |
| 17:40.15 | cpm | for a small system, why not? |
| 17:40.22 | jbalcomb | Exstatica ah, im calling them today because we are having echo/jitter issues including dropped calls and a wierd clicking noise. |
| 17:40.27 | _Sam-- | you could save some hair loss and sleep time by not using asterisk, probably |
| 17:40.36 | cron | Idle: thats why there are courses that your dept/staff require before implementing such a system build it into the implementation/setup costs |
| 17:40.40 | Exstatica | jbalcomb: send an email to support, they answer pretty quickly |
| 17:41.32 | jbalcomb | Idle I would recommend discussing your production directly with [TK]D-Fender. I'm certain it would be worth your dollar to have him help. |
| 17:41.34 | _Sam-- | i have a really small production environment (20 desks) and i couldnt implement asterisk on my own the first time -- was easier to pay someone else (zoa) and learn from his work. |
| 17:41.46 | *** join/#asterisk Assid (n=assid@203.115.64.14) |
| 17:41.48 | jbalcomb | Exstatica excellent. doing so now actually. thank you and good luck on your issue. |
| 17:42.11 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 17:42.11 | *** join/#asterisk razu (n=razu@ip59.cab62.mus.starman.ee) |
| 17:42.17 | Assid | hey [TK]D-Fender: you around? |
| 17:42.20 | Assid | weird issue |
| 17:42.26 | bsdfreak | heh |
| 17:42.27 | bsdfreak | \o\ |
| 17:42.33 | *** join/#asterisk Katty_ (n=angela@64.82.232.54) |
| 17:42.46 | Ivan_Stepaniuk | Hi there, i have a simple IAX trunking betwen 2 * boxes, setting all codec configuration to ulaw, alaw, or g726 works ok, but setting all to g729 fails with: Jan 26 14:36:54 WARNING[17266]: channel.c:1914 ast_request: No translator path exists for channel type IAX2 (native 63744) to 64 ... any ideas? |
| 17:43.23 | [TK]D-Fender | Assid : here |
| 17:43.29 | Assid | on the poly501 .. i have reg.1.lineKeys="2" and i have reg.3.lineKeys="1" and am registering line 3 |
| 17:43.37 | Assid | but i dont see it connected |
| 17:43.40 | Idle | jbalcomb: yea, its small tho, not large |
| 17:43.53 | Assid | rebooted the phone |
| 17:43.53 | [TK]D-Fender | You're out of line keys silly! |
| 17:43.59 | Ivan_Stepaniuk | im using Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k on both sides |
| 17:44.00 | Exstatica | jbalcomb: thanks |
| 17:44.07 | Idle | 5 or 10 users to start, and growing to a half dozen 10-15 user locations over a couple years |
| 17:44.08 | Assid | isnt it a 3 line phone? |
| 17:44.21 | Stephnie | aa little bit confusion at ASTCC ..please check http://pastebin.ca/38687 ?? anybody |
| 17:44.22 | *** join/#asterisk mattwj2005 (n=Matt@dialup-4.159.110.70.Dial1.Chicago1.Level3.net) |
| 17:44.46 | Assid | i want first 2 lines on same registration.. and 3rd line to have its own tregistration |
| 17:46.07 | [TK]D-Fender | Assid : You need a line-key to access it from though. for 3 lines you could only give 1 line key per line. |
| 17:46.24 | Ivan_Stepaniuk | [TK]D-Fender: i thought only g723 was licenced |
| 17:46.35 | [TK]D-Fender | Assid : Otherwise you'd have no way to pick the 3rd registration. |
| 17:46.35 | brad_mssw | anyone know what voicepulse's IAX server is? need to do a traceroute before signing up ... |
| 17:46.48 | *** join/#asterisk twisted[mobile] (n=twisted@24.96.140.230) |
| 17:46.53 | [TK]D-Fender | Ivan_Stepaniuk : Nope, G.729.... read up on it... |
| 17:47.21 | Assid | so if i want lines 1 and 2 to be same registration.. and third line seperate.. what do i gotta do? |
| 17:47.32 | _Sam-- | iidle: you said the budget was 150,000 dollars....for 6 phones?? |
| 17:47.34 | _Sam-- | count me in |
| 17:47.51 | Assid | i thought linekeys means the number of lines to use for this registration |
| 17:48.09 | Hmmhesays | why is behringer so cheap |
| 17:48.24 | Idle | _Sam--: no, :P |
| 17:48.27 | Hmmhesays | geebus its like 1/4 of the price of peavey, qsc, crown |
| 17:48.35 | *** join/#asterisk QbY (n=QbY@adsl-068-209-210-253.sip.cha.bellsouth.net) |
| 17:48.49 | Idle | I said thats what I heard cisco call manager cost, and that I didnt believe it was _that_ much |
| 17:48.53 | Assid | [TK]D-Fender ? |
| 17:49.17 | [TK]D-Fender | Assid : If you use 3 line keys for your first reg, you have no way of accessing any other reg! |
| 17:49.30 | *** part/#asterisk thieumS (n=darkmind@bea75-1-82-234-122-35.fbx.proxad.net) |
| 17:49.48 | Assid | right.. thayts why my first reg is only set to 2 |
| 17:49.59 | QbY | I'm receiving calls from an area code that isn't supposed to use the 800# they are dialing. How can I catch that area code and play them a message to call another number? |
| 17:50.07 | Assid | oh wait |
| 17:50.11 | Assid | i thinkj i know |
| 17:50.19 | Assid | i gotta use. reg.2. instead |
| 17:50.21 | [TK]D-Fender | Assid : Handling multiple calls per key isn't really hard (not as nice mind you), but if you want 3 unique regs, they're going to need to take 1 key each |
| 17:50.26 | *** part/#asterisk jeffik (n=jeffik@CPE0050babf4cd5-CM014350000760.cpe.net.cable.rogers.com) |
| 17:50.40 | [TK]D-Fender | Assid : Well you're out of keys before getting to your 3rd reg! |
| 17:50.48 | *** join/#asterisk fa_back (n=faceoff@devel.acdbddh.eu.org) |
| 17:51.01 | Assid | yeah i got it |
| 17:51.09 | [TK]D-Fender | Assid : I think we need to start over : How many regs do you need? |
| 17:51.13 | Assid | i guess that sipura method was screwed |
| 17:51.28 | Assid | sipura i would have had to register as the 3rd reg |
| 17:51.34 | [TK]D-Fender | Sipura is definately a whacked setup. |
| 17:51.37 | Assid | i need 2 reg... first reg - 2 lines |
| 17:51.47 | Assid | 2nd reg .. 1 line |
| 17:51.52 | Assid | so i gotta reg.2. |
| 17:51.53 | [TK]D-Fender | Assid : then you need reg 1 w/ 2 keys, reg 2 w/ 1 key |
| 17:52.08 | Assid | yes |
| 17:52.19 | Assid | i was doing reg.3 .. cause i thought third line... |
| 17:52.27 | Stephnie | http://pastebin.ca/38687 ?? anybody |
| 17:53.25 | oogle | does digium provide any sort of bug notification service? ya know, so if timebomb happens again we can get one of those wonderful 3am wake-up calls |
| 17:53.42 | [TK]D-Fender | Assid : no, reg 3 is for the 3rd ACCOUNT, how you divvyup the line keys is up to you. |
| 17:53.52 | *** join/#asterisk lithi (n=interp3@toronto-HSE-ppp4045394.sympatico.ca) |
| 17:53.54 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@72-254-39-40.client.stsn.net) |
| 17:53.58 | [TK]D-Fender | oogle : you're sitting in it! |
| 17:54.15 | Assid | yeah |
| 17:54.24 | Assid | [TK]D-Fender : whats label for ? |
| 17:54.28 | oogle | [TK]D-Fender: I don't dream on freenode |
| 17:55.03 | [TK]D-Fender | Assid : You could lable your first reg as "Fred" while reg'ing with 8000 as a username. Itrs just text for the screen |
| 17:55.22 | lithi | How can I add +1 to my dial plan without making the user dial 1 first, something like ${EXTEN:+1} ? |
| 17:55.23 | [TK]D-Fender | oogle : There is Mantis (the bug tracker), and the WIKI. |
| 17:55.39 | Assid | aah |
| 17:55.44 | [TK]D-Fender | oogle : Keep in mind * is FREE and isn't required to send e-mails to everyone about bugs... |
| 17:55.47 | Assid | like "private line" or "PBX" |
| 17:55.52 | Assid | if you have them seperate |
| 17:55.56 | [TK]D-Fender | Assid : Something like that. |
| 17:55.59 | Assid | okay |
| 17:56.10 | [TK]D-Fender | Assid : I typically find few times where I need a phone to have 2 regs..... |
| 17:56.23 | Assid | yeah.. like this one |
| 17:56.25 | zoa | oogle: haha yes sure call them at 3 am if your server goes down |
| 17:56.29 | zoa | they will really like that |
| 17:56.55 | [TK]D-Fender | Assid : Why are you looking to use 2 regs on that phone? |
| 17:56.55 | oogle | i understand * is free, but when you're pumping thousands of dollars in to digium cards, it's nice to hear about bugs like timebomb from digium first |
| 17:57.26 | Assid | 2 different profiles for these guys |
| 17:57.39 | [TK]D-Fender | oogle : The cards work fine... the software you're USING with them isn't! get ABE and maybe they'll follow up more.. mind you they avoid the "bleeding edge anyways...) |
| 17:57.40 | oogle | even if it was a pay-for service |
| 17:58.38 | Stephnie | http://pastebin.ca/38687 ?? ASTCC HELP ??? |
| 17:58.45 | *** join/#asterisk j0n (n=jellis@206-169-48-226.gen.twtelecom.net) |
| 17:59.00 | SplasPood | lithi: 1${EXTEN} |
| 17:59.12 | lithi | SplasPood: Thanks :) |
| 17:59.29 | SplasPood | np |
| 17:59.53 | agreenberg | Has anyone done anything about the type size miss-matches in pbx_spool.c? I'm reviewing the code to see if I can track down some memory leaks. Look the usage of retrytime in the outgoing stuct. It's an int when the use is t_time. |
| 18:00.29 | _Sam-- | agreenberg: get a new ver |
| 18:00.31 | _Sam-- | check topic |
| 18:00.32 | Assid | hrmm.. can i have a macro .. where i can specify extension instead of s ? |
| 18:01.30 | agreenberg | Point me to the url if you have one please. |
| 18:01.48 | file | oogle: it wasn't a planned timebomb :P it was an honest coding mistake |
| 18:02.02 | rob0 | Did the big bad bug only bite 1.2.2? I have a 1.2.1 install which seems to be okay. |
| 18:02.18 | [TK]D-Fender | Assid : huh? |
| 18:02.26 | file | rob0: aye |
| 18:03.12 | file | I need to yell at them right now... |
| 18:03.17 | file | but I can't get to the right person |
| 18:03.33 | jbalcomb | tell them you are from NTT/Verio.. |
| 18:03.52 | file | packet goes in... but it doesn't come out... |
| 18:04.39 | jbalcomb | file tell them you're Jim Templin from the NOC in houston. :) |
| 18:04.40 | Assid | gimme a sec.. i dont think this it working yet |
| 18:04.51 | *** join/#asterisk maggit (n=maggit@customer-200-36-59-130.uninet.net.mx) |
| 18:05.14 | _Sam-- | cron: i think you made a good choice not installing this firmware. |
| 18:05.25 | _Sam-- | first phone im working with now is stuck in a reboot loop |
| 18:05.25 | cpm | file: what asn is at issue? |
| 18:05.46 | file | I don't even know anymore, my brain is shot from debugging this |
| 18:05.50 | fulgas | "Extension '' in context 'from-zap' from '' does not exist" |
| 18:05.52 | Assid | [TK]D-Fender: that didnt work either apparently |
| 18:05.53 | cpm | gimme an ip address |
| 18:05.56 | fulgas | anyone knows what this might be? |
| 18:06.11 | *** join/#asterisk justinu (n=justin@72.18.13.34) |
| 18:06.11 | SpaceBass | anyone else have a problem sending e-mails (VM or Fax) from their asterisk box to users on a mail server on their same LAN? |
| 18:06.16 | Assid | i see the final file downloaded.. but dont see it in sip show peers as registere |
| 18:06.17 | bsdfreak | yeah |
| 18:06.18 | bsdfreak | truly one |
| 18:06.33 | SpaceBass | my firewall doesnt allow internal boxes to connect to external address, so send mail cannot deliver my voice mails to my local MTA |
| 18:07.31 | *** join/#asterisk loick (n=loick@APuteaux-151-1-12-101.w82-120.abo.wanadoo.fr) |
| 18:08.41 | bsdfreak | heh |
| 18:08.45 | bsdfreak | can you tunnel the connection? |
| 18:08.53 | bsdfreak | use mail forwarding over port 587? |
| 18:08.55 | Err | set up a mail relay |
| 18:08.57 | bsdfreak | there's gotta be a way :p |
| 18:09.12 | Stephnie | any help for ASTCC ??? |
| 18:09.12 | Err | (send the mail the same way you're *supposed* to as a client...) |
| 18:09.14 | Stephnie | http://pastebin.ca/38687 ?? ASTCC HELP ??? |
| 18:10.53 | *** join/#asterisk file (n=joshnet@mctnnbsa24w-142167051174.pppoe-dynamic.nb.aliant.net) |
| 18:12.31 | kpettit | SpaceBass, Yes, I have that problem all the time |
| 18:12.44 | kpettit | You basically need to fake a MX record. Each MTA has there own way of doing it. |
| 18:13.12 | Nugget | how do you fake an MX record. Or, rather, what is a "fake" record? |
| 18:13.14 | kpettit | I use courier-mta and there is a esmtproutes file that's kind of like a /etc/hosts file for MX records |
| 18:13.34 | kpettit | it will use what you specify in there rather than the public MX record |
| 18:13.41 | Nugget | *nod* |
| 18:13.45 | _Sam-- | cron: using the new firmware...all is good so far. |
| 18:14.17 | *** join/#asterisk Defraz (i=t0tal@72.165.56.43) |
| 18:14.26 | kpettit | My asterisk box is the gateway/firewall. So it's IP address is the one show on the public MX record so I have to tell voicemail to email to use a internal nat address by using those fake mx records |
| 18:14.30 | *** join/#asterisk flynux (i=xy74d67@pingou.in) |
| 18:14.44 | pifiu | hey everyone |
| 18:15.31 | Assid | [TK]D-Fender: no man.. it hates me |
| 18:15.33 | Assid | dont wanna work |
| 18:15.41 | bsdfreak | heh |
| 18:15.49 | *** part/#asterisk pointer (i=pointer@aj.catt.com) |
| 18:16.04 | _Sam-- | damn teliax |
| 18:16.08 | _Sam-- | no incoming or outgoing phone server |
| 18:16.10 | _Sam-- | service. |
| 18:16.15 | |vinsik| | haha..how to use svn.digium.com ? |
| 18:16.35 | |vinsik| | says forbitten. |
| 18:18.15 | |vinsik| | nevermind, got it |
| 18:18.28 | *** join/#asterisk slan (n=lba@user-12lml5g.cable.mindspring.com) |
| 18:19.00 | *** join/#asterisk loick_ (n=loick@APuteaux-151-1-56-32.w82-120.abo.wanadoo.fr) |
| 18:19.10 | slan | Has anyone used Asterisk in Slackware? Any problems? How do you update since Slack doesn't use package managers like Yum? |
| 18:19.11 | *** join/#asterisk bkw__ (n=brian@72-254-39-40.client.stsn.net) |
| 18:19.34 | |vinsik| | slan: update what exacly? |
| 18:19.34 | _Sam-- | cron: i think the new gxp2000 firmware does work better. |
| 18:19.48 | _Sam-- | the first two people i called thought i was on a different phone than normal |
| 18:20.10 | SpaceBass | kpettit, thanks! |
| 18:20.21 | slan | vinsik: Meaning when it comes time to update Asterisk |
| 18:20.30 | kpettit | SpaceBass, np |
| 18:20.31 | |vinsik| | slan: oh.. |
| 18:20.45 | [TK]D-Fender | slan : Completely trouble free for me... |
| 18:20.53 | EksilAndyCap | slan: same as you installed it I guess. |
| 18:20.55 | |vinsik| | slan: just compile it and install |
| 18:21.01 | kpettit | I'm tryeing to find a good platform to embed * on |
| 18:21.20 | SpaceBass | kpettit, know of a way to spoof the MX with sendmail? |
| 18:21.28 | [TK]D-Fender | slan : I use "swaret" for the typical packages, and for Asterisk , I only use SOURCE downloads and I suggest the same for everyone. |
| 18:21.41 | kpettit | I've only done it with courier, sorry |
| 18:21.57 | _Sam-- | kpettit: i run some solid state asterisk boxes on compact flash cards (not quite embedded, but close) |
| 18:22.06 | slan | Let me ask a related q - How can I preserve all my conf files when I update Asterisk? |
| 18:22.08 | Assid | [TK]D-Fender: whats the default user/pass for poly501? any clue? |
| 18:22.10 | _Sam-- | and i have some special tricks so there is barely any writes to the card |
| 18:22.14 | [TK]D-Fender | Assid : Pastebin your phone config |
| 18:22.21 | kpettit | _Sam--, sweet. What hardware you using? I'll just be happy to get rid of the fans |
| 18:22.28 | [TK]D-Fender | Assid : Why would you need that? |
| 18:22.41 | [TK]D-Fender | Assid : Should be "Polycom", "456" |
| 18:22.54 | _Sam-- | the last 3 have been on mini-atx motherboards actually, with a fanless cpu cooler |
| 18:22.56 | kpettit | _Sam--, I want the OS on a flash card but I'll probally have a HD in there just for Voicemail and logs |
| 18:23.03 | _Sam-- | they do still have power supply fans |
| 18:23.12 | *** join/#asterisk mroth_imm[workin (n=mroth@63.65.26.220) |
| 18:23.22 | _Sam-- | i write voicemail to ram, and then to flash only once in a while. |
| 18:23.28 | kpettit | _Sam--, I'm thinking of something like this... http://www.mbx.com/oem/reference_platforms/RP-1009.cfm |
| 18:23.29 | mroth_imm[workin | anyone know if there is a users' group for SNOM VOIP phones? |
| 18:23.42 | _Sam-- | and then i backup to USB thumb drive using rsync every so often |
| 18:23.46 | kpettit | _Sam--, have you had any problems with the flash wearing out? |
| 18:23.57 | kpettit | that's awsome |
| 18:24.00 | Assid | okay thats weird |
| 18:24.01 | _Sam-- | kpettit: i barely write to it...like i said, i have some "special" tricks i use |
| 18:24.06 | _Sam-- | i dont run /var from the compact flash |
| 18:24.18 | _Sam-- | <PROTECTED> |
| 18:24.19 | Assid | my phone.cfg has numlinekeys at 2 not 3.. web inteface shows 3 |
| 18:24.26 | kpettit | How big is your OS on the flash card |
| 18:24.26 | _Sam-- | and then every so often, i rsync /var from memory back to the card |
| 18:24.47 | [TK]D-Fender | Assid : Did you forget to reboot the phone? |
| 18:24.48 | _Sam-- | i have a big flash card, because i have a nice interface for clients...my CF card has Xwindows on it. |
| 18:24.55 | kpettit | _Sam--, does it have very much memory to handle that? How many uses can you support? |
| 18:25.00 | *** join/#asterisk flynux (i=qv49ivc@cl-8.bru-01.be.sixxs.net) |
| 18:25.02 | |vinsik| | wha? .. when i svn to digium my server hangs.. :) |
| 18:25.13 | Assid | [TK]D-Fender: NO.. i rebooted. i saw the files being downloaded from ftp |
| 18:25.20 | kpettit | I'm hoping to do up to 20 users, and I also want to put a 4port fxs or fxo card on it |
| 18:25.23 | Assid | with sip notify |
| 18:25.29 | *** join/#asterisk loic1 (n=loick@APuteaux-151-1-56-32.w82-120.abo.wanadoo.fr) |
| 18:25.36 | _Sam-- | my machine isnt affected by the cf card in terms of capacity...it has the same capacity as any other asterisk box on the same hardware |
| 18:25.51 | [TK]D-Fender | slan : The only way you'd blow away your config is if you did "make samples". As log as you don't do that you're old configs will just stay there. I upgraded 2 servrs this weel. |
| 18:26.01 | *** join/#asterisk EriSan (n=erisan@81-174-23-205.f5.ngi.it) |
| 18:26.07 | kpettit | _Sam--, do you have a publick image of it? I'd love ot check it out |
| 18:26.08 | _Sam-- | kpettit: its a good project, and one you will enjoy im sure |
| 18:26.14 | [TK]D-Fender | Assid : Maybe something else was left in.... double checp sip.cfg and your phones CFG. |
| 18:26.16 | Assid | [TK]D-Fender: weird thing.. the new registration shows in the web interface |
| 18:26.21 | |vinsik| | load average: 16.05, 6.80, 2.96 <= w000t?! |
| 18:26.25 | _Sam-- | i could let ya have it, its for a 1G compact flash card (which are pretty cheap) |
| 18:26.29 | slan | D-Fender: Thanks |
| 18:26.42 | kpettit | _Sam--, these big boxes are driving me nuts. I really want something small I can screw onto a wall or something like that |
| 18:26.45 | _Sam-- | you would need to do some tweaking in terms of how x-windows starts for your environment |
| 18:26.55 | kpettit | _Sam--, cool thanks,. |
| 18:27.17 | _Sam-- | also, for me, it was easiest to work with IDE compact flash in terms of booting |
| 18:27.23 | _Sam-- | you may want to buy an IDE compact flash adapter |
| 18:27.28 | Assid | [TK]D-Fender: nah thats fine |
| 18:27.30 | _Sam-- | instead of booting from USB cf |
| 18:27.35 | kpettit | _Sam--, that's what I was thinking. It seemed the easiest way to develop |
| 18:27.41 | [TK]D-Fender | slan : np |
| 18:28.12 | _Sam-- | kpetit: i wont be able to give you the image of the cf card until i get home later.... |
| 18:28.26 | kpettit | no prolbme. I won't have hardware for another week or so. |
| 18:29.03 | _Sam-- | my image comes standard with mysql 5, cdr_mysql, apache, and plus some other tools like an mp3 player for when you're pulling your hair out |
| 18:29.41 | *** join/#asterisk Falle (i=falstaf@213.141.80.88) |
| 18:30.02 | Falle | Is anyone here trying the new GXP firmware? |
| 18:30.08 | _Sam-- | Falle: i am |
| 18:30.14 | _Sam-- | works nice after you get out of the reboot loop |
| 18:30.29 | [TK]D-Fender | Assid : If you want you can tar them up, email them over and I'll take a look for you? |
| 18:30.30 | Falle | _sam--: mine never stops the reboot loop |
| 18:30.37 | _Sam-- | Falle: move the firmware files |
| 18:30.43 | _Sam-- | so they dont exist where it is getting them |
| 18:30.46 | _Sam-- | then it stops |
| 18:30.52 | mut | there any way to change attenuation on a te405p? |
| 18:30.55 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 18:31.03 | Falle | can i do that even thouh its in the middle of the loop? |
| 18:31.09 | Falle | +g |
| 18:31.14 | _Sam-- | wait until it is done a loop, and then pull the plug. |
| 18:31.16 | Assid | sure.. gime a sec |
| 18:31.17 | _Sam-- | then move the files |
| 18:31.19 | _Sam-- | and turn it back on |
| 18:31.20 | Assid | l;emme try something |
| 18:31.53 | Falle | _sam--: Ok, I'll do that. sound like a bug anyhow :) |
| 18:32.01 | _Sam-- | i thought my phone was toast |
| 18:32.09 | Falle | same here :P |
| 18:32.12 | _Sam-- | seems like it works ok though |
| 18:32.22 | *** join/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com) |
| 18:32.41 | Falle | i really like the new features and GUI |
| 18:32.46 | warthawg | can you call skype users from a phone connected to asterisk? |
| 18:33.05 | _Sam-- | i havent seen the menu yet from the phone when push the round button |
| 18:33.21 | _Sam-- | but the main feature im hoping works well is the echo cancel thing |
| 18:33.22 | redder86 | warthawg: doubt it, unless there's a chan_skype out there |
| 18:33.30 | warthawg | thanks, ReD-MaN |
| 18:33.35 | warthawg | whooops, i mean redder86 |
| 18:33.47 | Falle | _sam--: no echo's at all.. i tried that inbetween loops |
| 18:34.42 | Falle | _sam--: its been in this loop for 35 minutes now, getting scared :-) |
| 18:34.49 | *** join/#asterisk festr__ (n=festr@ns.regnet.cz) |
| 18:35.12 | _Sam-- | it will be fine, i promise |
| 18:35.16 | *** join/#asterisk angom_w (n=angom@red-corp-200.38.16.10.telnor.net) |
| 18:35.26 | _Sam-- | <famous last words> |
| 18:35.52 | *** join/#asterisk Katty (n=angela@64.82.232.54) |
| 18:36.41 | Falle | :) |
| 18:38.39 | fulgas | i'm getting Extension '' in context 'from-zap' from '' does not exist. Rejecting call on channel 0/29, span 2 |
| 18:38.42 | fulgas | anyone knows why? |
| 18:38.46 | *** join/#asterisk TonyM_ (n=TonyM@adsl-solo-80-168-225-127.claranet.co.uk) |
| 18:38.49 | fulgas | * with a alcatel |
| 18:38.57 | fulgas | 405p |
| 18:39.18 | *** part/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com) |
| 18:40.12 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 18:41.25 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:41.36 | *** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 18:41.41 | [TK]D-Fender | fulgas : The extension its looking for doesn't exist. The error is self explanatory. |
| 18:42.44 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 18:42.53 | fulgas | ( [TK]D-Fender ): not quite |
| 18:43.02 | rayvd | Is there a (built-in) way to do the equivalent of a show voicemail users (see mailboxes and messages waiting) when using voicemail+realtime? |
| 18:43.02 | *** join/#asterisk Zodiacal- (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 18:43.26 | Feral_Kid | I someone could give me a hand with the an iconnect trunk, I would would be a very happy camper... According to the console, iconnect is registered, but I get congestion tone when I dial out... |
| 18:44.09 | fulgas | * moves in misterious ways |
| 18:44.10 | fulgas | eheh |
| 18:44.16 | *** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-163.sw.biz.rr.com) |
| 18:45.00 | *** join/#asterisk ruza (n=ruza@holly.cervenytrpaslik.cz) |
| 18:45.09 | ctooley | I'm trying to set up a soft client. Asterisk is running on the NAT network bridge and the soft client is behind it. I'm using a SIP service provider. I can't get it to stop attempting a native bridge for this client. |
| 18:46.18 | rayvd | you want asterisk to stay out of the media path? |
| 18:46.41 | *** join/#asterisk pointer (i=pointer@aj.catt.com) |
| 18:46.45 | *** join/#asterisk Curi (n=no@200.24.227.58) |
| 18:47.00 | ctooley | rayvd, no I specifically want asterisk to stay _IN_ the media path to straddle the NAT |
| 18:47.18 | pointer | anyone around with cvs write access that can save me from filing a really dumb bug report on app_hasnewvoicmail.c? |
| 18:47.21 | Curi | hello, is there a way to disable an extension for a while? |
| 18:47.48 | _Sam-- | if you had a phone that supported it, you could just put it on do not disturb |
| 18:48.28 | Curi | but i mean from asterisk, i'm trying to build a callshop so i want to be able to disable the extension so it won't be able to make any calls |
| 18:49.06 | _Sam-- | there alot of ways to do it. |
| 18:49.16 | Katty | wocka! |
| 18:49.17 | Curi | really? |
| 18:49.37 | Curi | could you help me? |
| 18:49.47 | _Sam-- | i could but i cant right now |
| 18:49.57 | _Sam-- | a place to look would be extensions.conf |
| 18:50.10 | _Sam-- | and maybe and/or sip.conf or iax.conf |
| 18:50.16 | *** join/#asterisk jabuka (n=edumatao@200.205.205.254) |
| 18:50.26 | jabuka | hello, somebody can help me? |
| 18:50.38 | Katty | i'm not a licensed therapist, sorry. |
| 18:50.43 | Katty | next! |
| 18:50.53 | jabuka | Call pickup works with IAX? |
| 18:50.55 | pointer | hasnewvoicemail/hasvoicemail/VMCOUNT() all will never report more than 1 voicemail in any given mailbox with the break in the while loop in hasvoicemail_internal() |
| 18:52.12 | Katty | ! |
| 18:52.16 | Katty | hey junky (= |
| 18:52.21 | JunK-Y | whats up? |
| 18:52.26 | Katty | my weight. |
| 18:52.28 | jabuka | HEELLLOOOOO |
| 18:52.29 | Katty | just had chinese! |
| 18:52.36 | Katty | jabuka: chill. |
| 18:52.45 | Katty | jabuka: patience is a virtue for a /reason/ |
| 18:52.55 | slan | ctooley, Hello Chris this is Larry Alkoff. Nice to see you here. |
| 18:52.58 | jabuka | katty, sei, sei |
| 18:53.03 | mroth_imm[workin | jabuka: HELLO TO YOU SIR |
| 18:53.05 | ctooley | :) Hi Larry |
| 18:53.06 | Katty | JunK-Y: what's up with you? |
| 18:53.16 | Katty | JunK-Y: i hear damin molested you for me. |
| 18:53.34 | cpm | jabuka, what's yer issue? |
| 18:53.37 | *** part/#asterisk Ivan_Stepaniuk (n=istepani@201.250.7.164) |
| 18:53.57 | slan | ctooley, My AAH is working better and better. Spent last night forwarding the proper ports and registering with dyndns. |
| 18:53.58 | jabuka | call pickup works with IAX? |
| 18:54.09 | ctooley | slan, I see. |
| 18:54.31 | ctooley | I'll be happy to chat later, just wanted to ask a quick question. My 1 o'clock meeting is here |
| 18:54.37 | slan | ctooley, From an old Isaac Asimov book ;-) |
| 18:54.41 | jbalcomb | iCEBrkr just found out my PRI is sharing IRQ 11 with four other devices!! woot. |
| 18:54.41 | *** join/#asterisk MikeJ__ (n=vircuser@70.103.248.130) |
| 18:54.58 | *** join/#asterisk rob314 (n=root@207.58.194.55) |
| 18:55.04 | iCEBrkr | jbalcomb: That totally rocks.. |
| 18:55.09 | iCEBrkr | jbalcomb: You got a Digium card in there? |
| 18:55.54 | *** join/#asterisk zzxxcc (n=zzxxcc@221.232.10.225) |
| 18:56.07 | slan | ctooley, Nice seeing ya. Bye |
| 18:56.56 | JunK-Y | Katty: yeah damin hugs me, dont worry :) |
| 18:57.12 | JunK-Y | bkw should have the pic. |
| 18:57.20 | oej | ~seen malverian |
| 18:57.32 | jbot | malverian <n=malveria@adsl-065-005-207-210.sip.gnv.bellsouth.net> was last seen on IRC in channel #asterisk, 35d 21h 23m 47s ago, saying: 'mog_work: Okay, later then :)'. |
| 18:57.32 | jabuka | cpm, did you see my message? |
| 18:58.10 | oej | ~seen malverian[work] |
| 18:58.12 | jbot | malverian[work] <n=pawalls@209.208.68.61> was last seen on IRC in channel #asterisk, 1d 18h 7m 54s ago, saying: 'Strange..'. |
| 18:58.43 | cpm | You mean the statement in the form of a question " call pickup works with IAX" ? that I saw. sounds like an empircal statement to me. |
| 18:59.25 | [TK]D-Fender | cpm : Yeah... long live the Empire! |
| 18:59.50 | file | I'm hungry, annoyed, and irritated |
| 18:59.52 | file | I'm going to get food. |
| 19:00.09 | cpm | Do that thing. |
| 19:01.53 | pointer | can someone try to do a checkout from the digium subversion tree for me, I keep getting a subversion error |
| 19:01.57 | pointer | please :) |
| 19:02.19 | file | bah all I got was a drink |
| 19:02.35 | asteriskmonkey | is there anyway of know what firmware revision a digium card is using |
| 19:02.39 | *** join/#asterisk Drew_____ (n=foo@zux221-065-169.adsl.green.ch) |
| 19:03.10 | pointer | bleh, I browsed the repo |
| 19:03.14 | pointer | looks like they fixed it |
| 19:04.12 | Drew_____ | what do i have to do to my extensions.conf/sip.conf to process incoming SIP calls that go directly to my IP rather than coming in through my SIP provider? |
| 19:06.22 | *** join/#asterisk dijit0 (n=dijit0@adsl-68-125-48-100.dsl.pltn13.pacbell.net) |
| 19:06.32 | *** join/#asterisk gambolputty (n=root@64.74.225.131) |
| 19:07.08 | dijit0 | can anyone help me with a callerid issue? i am using idefisk, but i want to set the callerid in iax.conf, not allow the user to set it... but it seems that idefisk sets it either way... |
| 19:07.29 | gambolputty | Hi. Can * do a select from mysql where two tables are in a one to many relationship and the result back is more than one row? |
| 19:07.50 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 19:08.28 | _Sam-- | if you have the caller id set in iax.conf i didnt think it would be overridden by the client |
| 19:08.33 | _Sam-- | zoa would know |
| 19:08.34 | _Sam-- | zoa |
| 19:08.37 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 19:08.47 | dijit0 | well, it didnt used to be, lol |
| 19:09.00 | _Sam-- | what * version? |
| 19:09.07 | dijit0 | 1.2.3 |
| 19:09.24 | dijit0 | i am not experienced with it too well though, so its probablysomething stupid |
| 19:09.40 | asteriskmonkey | anyone know how to find out what firmware a DIGIUM BOARD is using |
| 19:09.58 | file | asteriskmonkey: ask it nicely |
| 19:10.42 | Katty | file: GIVE ME A MUFFIN |
| 19:10.45 | Katty | file: pretty please. |
| 19:14.00 | dijit0 | so no one knows anything about the caller id issue? |
| 19:14.59 | *** join/#asterisk elg (n=fugalh@falcon.fugal.net) |
| 19:15.32 | mut | what'de cause a double ring on a line? |
| 19:15.42 | _Sam-- | who maintainst the GXP2000 tiki page on voip-info.org? it says "Come visit us sometime #Asterisk on irc.freenode.net... |
| 19:15.45 | mut | if i dial sip -> zap-pstn |
| 19:15.50 | _Sam-- | just wondering who takes care of that page |
| 19:15.52 | mut | sometimes it does a double ring |
| 19:15.55 | mut | sometimes it doesn't |
| 19:16.06 | *** join/#asterisk Delvar (n=irc@host-83-146-53-44.bulldogdsl.com) |
| 19:17.29 | *** join/#asterisk zkillz (i=koma@d001046.adsl.hansenet.de) |
| 19:17.31 | zkillz | moin |
| 19:17.52 | mut | and do changes in zapata.conf need a asterisk restart? |
| 19:18.00 | justinu | most seem to |
| 19:18.17 | *** join/#asterisk bkw__ (n=brian@70.103.248.130) |
| 19:18.38 | [av]bani | moinmoin |
| 19:18.45 | nroej | *yeeha* |
| 19:18.56 | nroej | tomorrow i will get my sirrix crad |
| 19:19.02 | nroej | s/crad/card |
| 19:20.03 | *** join/#asterisk eliel (n=eliel@200.123.183.89) |
| 19:20.19 | *** join/#asterisk muzzz_ (n=chatzill@218.111.66.117) |
| 19:21.02 | QbY | does asterisk play well with any particular scripting language ? i need it to check the system to do a database hit when a user takes a particular option--and if the database says "yes" then asterisk needs to do something else.. |
| 19:21.10 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@70.103.248.130) |
| 19:21.19 | tehdely | asterisk plays well with pretty much anything |
| 19:21.29 | tehdely | the AGI interface is about as basic as you can get; simple command-set, all communications done over pipes |
| 19:21.33 | QbY | any examples of what i'm talking about?? i need to know how to pass info back and forth |
| 19:22.01 | tehdely | http://www.voip-info.org/tiki-index.php?page=Asterisk+AGI |
| 19:22.21 | tehdely | asterisk is scripting-language agnostic. as long as your script can read on stdin and write on stdout, and understands the very simple command set, you can write it in anything |
| 19:22.53 | QbY | awesome |
| 19:22.58 | iCEBrkr | QbY: I'd look into the ODBC function stuff |
| 19:23.08 | iCEBrkr | AGI doesn't scale well and is actually pretty slwo. |
| 19:23.09 | iCEBrkr | slow |
| 19:23.46 | QbY | well here's the thing.. we use asterisk for our helpdesk.. and people need to know when there is an outage.. currently we have to call record the outage notification, then change the dial plan.. |
| 19:24.12 | iCEBrkr | It's high-volume then I'd use the new ODBC function stuff |
| 19:24.20 | QbY | i'd like to call in, make the recording and then * know that its that way.. when the condition is cleared we'd call back and reset it, and it'd play the normal "everything is good" recording |
| 19:24.39 | [TK]D-Fender | asteriskmonkey : IIRC Digium boards have HARDWARE revisions, not "firmware" they are not field upgradeable. |
| 19:24.46 | iCEBrkr | QbY: Oh, that's easier than needing any AGI |
| 19:24.57 | QbY | k? |
| 19:25.13 | iCEBrkr | QbY: You can use the native Asterisk DB() functions |
| 19:25.25 | [av]bani | [TK]D-Fender: digium boards dont really have firmware :) |
| 19:25.36 | [TK]D-Fender | QbY : You can make a purely diaplan solution for that extremely easily. |
| 19:25.54 | [TK]D-Fender | [av]bani : Funny... thats twhat I just said :) |
| 19:26.20 | [TK]D-Fender | QbY : Set a value with DB and then based on that play stuff back. |
| 19:26.58 | iCEBrkr | QbY: Yeah, you could make a simple IVR menu for recording the message and setting the status |
| 19:26.58 | Drew_____ | i have a * box that acts as a proxy for the voip phones in my lan to my provider - what do i have to do to accept incoming calls that dont come in through my sip provider but come directly to i.e. myhostname.dyndns.org:5060 ?? |
| 19:27.04 | *** join/#asterisk ToTo (n=ToTo@host42-157.pool874.interbusiness.it) |
| 19:28.00 | *** join/#asterisk Delvar2 (n=irc@host-83-146-53-34.bulldogdsl.com) |
| 19:28.11 | [TK]D-Fender | Drew_____ : Put a "context=" statement in [general] along with "allowguests=yes". |
| 19:28.19 | iCEBrkr | QbY: http://www.voip-info.org |
| 19:30.47 | sivana | does * look for .call files automatically? |
| 19:31.50 | QbY | now if i could only get my music on hold to work for parked calls |
| 19:34.37 | Drew_____ | d-fender - i but that in but the caller gets a 404 |
| 19:34.51 | Drew_____ | sipdebug shows some sip packets though |
| 19:34.53 | *** join/#asterisk rpm (n=russell@S010600111155e117.cg.shawcable.net) |
| 19:35.10 | Drew_____ | but * doesnt answer the call or output any console info |
| 19:35.29 | *** join/#asterisk Lord_Drachenblut (n=Lord@12-210-115-191.client.insightBB.com) |
| 19:38.23 | Drew_____ | any ideas? |
| 19:39.23 | *** join/#asterisk Skkip (n=Skipper@216.160.91.91) |
| 19:39.33 | *** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
| 19:39.40 | *** join/#asterisk hroarke (n=whois@216.190.144.90) |
| 19:40.08 | *** join/#asterisk jpablo (n=jpablo@200.94.130.194) |
| 19:40.27 | Zeeek | beer? |
| 19:40.40 | hroarke | yum |
| 19:40.45 | jpablo | hey people, i have soome problem passing calls to the console/dsp, when i use mpg123 the sound works fine, asterisk *seems* to work fine (console answer and everything) but i don't hear anything, any ideas? |
| 19:41.45 | *** join/#asterisk parsec_ (n=parsec@69.90.194.201) |
| 19:42.03 | hroarke | is mgcp support something that is in its infancy and will be developed further or is it being phased out? |
| 19:42.06 | Zeeek | thanks! |
| 19:43.24 | brettnem | hroarke: mgcp support just kinda sucks |
| 19:43.28 | [TK]D-Fender | Drew_____ : Pastebin your sip.conf and the context it points to in extensions.conf |
| 19:43.29 | Zeeek | anyone have any issues like a FXS module dying on a TDM400 |
| 19:43.30 | [TK]D-Fender | ~pb |
| 19:43.31 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 19:43.52 | brettnem | wow. I don't think I've said anything in here in like 3 weeks |
| 19:43.55 | brettnem | hi |
| 19:44.00 | Zeeek | lo |
| 19:44.39 | hroarke | brettnem: thanks. specifically is it the implementation in asterisk that is not too good or the protocol |
| 19:44.45 | *** join/#asterisk dsfr (n=dsfr@gateway.digium.com) |
| 19:44.57 | brettnem | the protocol is great for certain applications |
| 19:45.00 | brad_mssw | Zeeek: well, no, my FXS ports on my TDM400P have been fine, no issues ... though we're phasing them out in favor of SIP phones everywhere |
| 19:45.07 | brettnem | asterisk has a poor implementation of it. |
| 19:45.16 | Drew_____ | k one moment |
| 19:45.19 | brettnem | mgcp is a good protocol for dump endpoints |
| 19:45.26 | brettnem | sip is good for smart endpoints. |
| 19:45.34 | hroarke | brettnem: is more involved than i want to mess with to port the vocal stuff to a driver asterisk can use? |
| 19:45.53 | brettnem | personally, I prefer the endpoint to be dumb.. I don't like varibility at the far end.. but they both have their merits |
| 19:46.05 | *** join/#asterisk santiago (n=santiago@208.195.215.222) |
| 19:46.10 | Zeeek | brad_mssw I have had a very occasional thing in the past where severe noises appear in the dialtone of one of three fxs. But now it just happened twice in a few days. |
| 19:46.24 | Zeeek | reloading zaptel fixes it |
| 19:46.50 | brettnem | hroarke: if you like the vocal stuff, just use it aside from asterisk.. you don't really need to integrate it.. but I didn't think the vocal mgcp stuff was very good either. |
| 19:47.01 | brad_mssw | Zeeek: what zaptel version ? |
| 19:47.03 | Drew_____ | http://pastebin.ca/38700 <-sip.conf |
| 19:47.14 | Zeeek | brad_mssw 1.2 |
| 19:47.17 | *** part/#asterisk santiago (n=santiago@208.195.215.222) |
| 19:47.40 | hroarke | brettnem: never messed with vocal just saw the had a driver.. The product we are developing is an asterisk product just looking to get more into it than sip and IAX2 |
| 19:47.55 | brettnem | hroarke: <choke> |
| 19:48.06 | brad_mssw | Zeeek: hmm, any irq conflicts ? |
| 19:48.14 | brettnem | hroarke: really, don't you think it has enough in it? why do you want to put more in? |
| 19:48.24 | Zeeek | naw nothing but digium cards on the box |
| 19:48.29 | *** join/#asterisk Psykick (n=anon@203-167-215-33.dsl.clear.net.nz) |
| 19:48.34 | hroarke | brettnem: from what i have looked at nothing has an API as nice as asterisk and would really like to stick with it |
| 19:48.36 | Psykick | hi guys |
| 19:48.36 | Zeeek | been running fine for almost two years |
| 19:48.42 | Drew_____ | http://pastebin.ca/38701 <-relevant suff from extensions.conf |
| 19:48.44 | brettnem | I think a real MGCP interface is probably worth money |
| 19:49.07 | brettnem | hroarke: really? asterisk is a total mess I think.. each day I wake up and pray it works |
| 19:49.12 | VJ | Hi guys, can anyone tell me the concept of g729 codec |
| 19:49.17 | Psykick | does anyone know of a good HOWTO for installing linux on an RPM based distro besides CentOS |
| 19:49.19 | VJ | in case i need to test it |
| 19:49.22 | *** join/#asterisk Skkip (n=Skipper@216.160.91.91) |
| 19:49.26 | VJ | can i do it without purchasing it |
| 19:49.29 | hroarke | brettnem: it does do alot and i am not faulting asterisk jsut that someone we do business is an all MGCP network and they would really like our product to suppor tit |
| 19:49.29 | brettnem | VJ: low bandwidth with reasonably low CPU overhead |
| 19:49.30 | Psykick | sorry guys |
| 19:49.33 | Psykick | I meant Asterisk |
| 19:49.39 | brettnem | VJ: if you don't know why you'd need it, you proabably dont |
| 19:49.55 | Drew_____ | VJ: http://en.wikipedia.org/wiki/G729 |
| 19:50.11 | hroarke | brettnem: still new to it -- but i really like the way its set up so far |
| 19:50.30 | Drew_____ | d-fender - any ideas? |
| 19:50.46 | brettnem | hroarke: well, I'd like good mgcp support in *.. but it's not going to happen.. I've watched that driver for years and it's not going anywhere.. |
| 19:51.20 | Psykick | brettnem: that makes two of us that pray that it works |
| 19:51.36 | hroarke | brettnem: ah hehe thanks for the input bro :) |
| 19:51.36 | Psykick | brettnem: I believe most of our issue is that its not configured properly |
| 19:51.37 | [TK]D-Fender | Drew_____ : Ok, right now incoming@youriphere will give the menu |
| 19:51.43 | brettnem | Psykick: I'm sure we arn't the only ones.. :) |
| 19:52.06 | brettnem | hroarke: no problem |
| 19:52.19 | Psykick | brettnem: slowly been discovering that the issues I've been facing have really nothing to do with asterisk ... but more to how incorrectly it's configured |
| 19:52.23 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 19:52.33 | Drew_____ | no that gives the caller a 404 |
| 19:52.55 | [TK]D-Fender | Drew_____ : You need to have an exten named accordingly. You should have something like "exten => bernhard,1,Dial(SIP/bernhard,60)" to do like "berhard@yourip" |
| 19:53.13 | [TK]D-Fender | Drew_____ : pastebin the error |
| 19:53.17 | Psykick | Quick Question to everyone |
| 19:53.25 | Psykick | how many people prefer SIP over IAX2? |
| 19:53.44 | brettnem | Psykick: there is just so much varibility in the network |
| 19:53.55 | *** join/#asterisk SPoon_TSX (n=Administ@h24-83-96-211.sbm.shawcable.net) |
| 19:53.57 | Psykick | brettnem: that's the same thing I thought |
| 19:54.15 | Psykick | brettnem: after doing some tests etc I've discovered it's not the network |
| 19:54.20 | file | varibility, what a nice word... OF THE DAY! |
| 19:54.29 | QbY | SIP Proxies. I'm getting conflicting information--I was told to use a SIP Proxy to handle all the clients and * For Calling Features (Voicemail, etc). So does this mean the SIP Proxy (SER) would actually make the call or would asterisk? |
| 19:54.30 | brettnem | ugh.. hours on a sipura behind a linksys that was trashing the natted port mappings.. grrr!!! what a pain! |
| 19:54.40 | Drew_____ | there is no error or console output on the * box |
| 19:54.46 | brettnem | file! you are the person OF THE DAY! |
| 19:54.49 | Drew_____ | the error occurs on the phone of the caller |
| 19:54.51 | file | YAY!!! |
| 19:55.00 | Zeeek | Drew_____ what is the default context? |
| 19:55.03 | SPoon_TSX | Hello everyone, got a question. May I know how can I enable that 1 SIP extension can take mulitpules call at the same time? |
| 19:55.08 | Katty | file: yay! |
| 19:55.12 | [TK]D-Fender | Drew_____ : exactly what did they try to dial?> |
| 19:55.15 | Zeeek | Katty, hi |
| 19:55.25 | file | QbY: phone <-> SER <-> phone for normal calling, phone <-> SER <-> Asterisk if using Asterisk for PSTN or features |
| 19:55.27 | Katty | Zeeek: herro. |
| 19:55.35 | Drew_____ | zeeek defualt is [incoming] |
| 19:55.39 | [TK]D-Fender | SPoon_TSX : the multiple call capability is an option on theDEVICE, not * |
| 19:55.42 | Drew_____ | fender - my ip |
| 19:55.52 | brettnem | file: there is aparently quite a bit of variability in the word of the days |
| 19:55.53 | Zeeek | Drew_____ is there something in the incoming context to handle the number being called? |
| 19:55.58 | Drew_____ | sip:81.221.65.169:5060 |
| 19:55.59 | [TK]D-Fender | Drew_____ : Show me EXACTLY what they are attempting to dial. |
| 19:56.09 | file | brettnem: I bet |
| 19:56.14 | QbY | file. so if i had wholesale accounts with a provider, my sip proxy would connect the phone to that provider? or does that go through Asterisk? |
| 19:56.27 | [TK]D-Fender | Drew_____ :There is no "s" exten in that context...... thats why it fails. |
| 19:56.42 | file | QbY: if the provider is SIP you could go phone <-> SER <-> provider |
| 19:56.51 | file | QbY: but the billing solutions for SER aren't exactly plentiful or easy |
| 19:57.32 | SPoon_TSX | [TK]D-Fender: Oh, My IP Phone DO support mulitline, I have 4 line to use. But when i use one of the line, the other line got busy signal. I don't have such problem, just happen once I upgraded to 1.2.3 |
| 19:57.36 | QbY | file: ok.. that makes sense now.. i just read somewhere that the SER couldn't originate calls.. |
| 19:57.45 | file | QbY: it can't |
| 19:57.53 | file | QbY: it acts as proxy and routes SIP messages |
| 19:59.00 | [TK]D-Fender | SPoon_TSX : What model? |
| 19:59.04 | *** join/#asterisk RussCC (n=face@216.157.205.211) |
| 19:59.11 | *** join/#asterisk zock (n=zock@p54B18239.dip0.t-ipconnect.de) |
| 19:59.26 | SPoon_TSX | [TK]D-Fender: I got 2 Aastra 480i and PolyCOM 501i |
| 19:59.31 | Drew_____ | d-fender - i put in a s exten but that doesnt change anything |
| 20:00.01 | zock | Hi. |
| 20:00.45 | Zeeek | Drew_____ have you tried sip debug? |
| 20:00.53 | Zeeek | that'll show you what's what |
| 20:00.59 | [TK]D-Fender | Drew_____ : Pastebin the new versions of both |
| 20:01.14 | gammacoder | _Sam--: i've been contributing to that page |
| 20:01.17 | *** join/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net) |
| 20:01.24 | Drew_____ | http://pastebin.ca/38704 <- extensions |
| 20:01.32 | [TK]D-Fender | SPoon_TSX : Does your problem occur on both? |
| 20:01.33 | RussCC | hello, I am having a problem with my TDM13B I have my line pluged into the first FXO port but when a call is recived I dont get Caller ID to come across |
| 20:01.42 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-89.rockynet.com) |
| 20:01.48 | SPoon_TSX | [TK]D-Fender: yes. |
| 20:01.48 | RussCC | I do when I use my clone card though |
| 20:01.56 | [TK]D-Fender | SPoon_TSX : Hmmm, makes no sense |
| 20:02.16 | _Sam-- | gammacoder: the newest firmware has a reboot loop |
| 20:02.17 | Drew_____ | no change to sip.conf |
| 20:02.19 | _Sam-- | maybe you could put that in there |
| 20:02.32 | [TK]D-Fender | Drew_____ : pastebin it anyways |
| 20:02.35 | twisted[asteria] | think you've got enough underscores in your name there? |
| 20:02.43 | _Sam-- | the only way to use the newest firmware is to let the phone upgrade, then remove the files from the location the phone is getting them from |
| 20:02.46 | Drew_____ | http://pastebin.ca/38700 <- sip.conf |
| 20:02.47 | gammacoder | _Sam--: I just posted upgrade instructions |
| 20:02.51 | _Sam-- | otherwise, the phone will be in a reboot loop |
| 20:02.53 | SPoon_TSX | [TK]D-Fender: I remember that somewhere say I need to set sip allow something... any idea? |
| 20:03.20 | [TK]D-Fender | Drew_____ : Is your * behind NAT? |
| 20:03.23 | Drew_____ | yes |
| 20:03.33 | [TK]D-Fender | SPoon_TSX : Nothing you need to set |
| 20:03.35 | Drew_____ | i portforwarded port 5060 to the * box |
| 20:03.43 | _Sam-- | i dont see any upgrade instructions on that page |
| 20:03.53 | Drew_____ | so the caller should reach * when calling my IP |
| 20:04.06 | *** join/#asterisk Seldon1975 (n=someone@199.243.101.131) |
| 20:04.06 | gammacoder | _Sam--: my instructions cover that issue - I've done 15 phones successfully so far with the instructions I just posted to the page (within the last 10 mins - they are 1/2 way down) |
| 20:04.10 | [TK]D-Fender | Drew_____ : You are missing all of the required NAT settings for un-initiated SIP calls to work. |
| 20:04.26 | [TK]D-Fender | Drew_____ : You need to set either EXTERNHOST or EXTERNIP, and LOCALNET. |
| 20:04.39 | _Sam-- | those directions are fine, if your upgrade works |
| 20:04.49 | _Sam-- | but so far, ive done three phones where those instructions are no good |
| 20:04.56 | _Sam-- | and i helped someone else on here fix theirs |
| 20:05.01 | Drew_____ | so i need to know my friends IPs for them to call me? |
| 20:05.11 | Seldon1975 | can someone please tell me what this measn (without being sarcastic): |
| 20:05.11 | Seldon1975 | WARNING[2337]: file.c:583 ast_readaudio_callback: Failed to write frame |
| 20:05.58 | _Sam-- | so far ive done 3 of my own phones (out of 20)...and all three got stuck in a reboot loop |
| 20:06.02 | _Sam-- | upgrading from 0.13 |
| 20:06.13 | Zeeek | what phone? |
| 20:06.19 | _Sam-- | gxp2000 |
| 20:06.22 | Zeeek | ah. |
| 20:06.57 | _Sam-- | i called bill @ grandstream and told him, dont know how long til they fix it |
| 20:06.59 | gammacoder | i've done phones from 0.9, 0.12, and 0.13 - and yes - it is a pain, but I think my instructions covered any hiccup I've seen - feel free to modify them, if it will help others |
| 20:07.24 | [TK]D-Fender | Drew_____ : No, you don't need their IPS. you just need * to know its behind NAT and forge the IP details accordingly. Also you need to forward the normal set of ports to * (5060,10000-20000) All UDP |
| 20:07.54 | hroarke | can anyone reccomend 3 or 4 very well tested and recommended phones to use with asterisk |
| 20:08.11 | *** part/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 20:08.13 | hroarke | or even 1 or 2 |
| 20:08.42 | Zeeek | polycom ip series |
| 20:08.44 | _Sam-- | hroarke i think there are plenty...but mostly depends on your budget. |
| 20:08.59 | Zeeek | sipura SPA841-941 |
| 20:09.11 | hroarke | 200 and less |
| 20:09.11 | Zeeek | I've heard good things about snom |
| 20:09.17 | hroarke | ive heard snom too |
| 20:09.28 | [av]bani | hroarke: if you buy polycom 601 then [TK]D-Fender will give you 24/7 technical support on this channel |
| 20:09.28 | Zeeek | Sipura is less than 200 |
| 20:09.36 | Zeeek | hah |
| 20:09.54 | [TK]D-Fender | 50/50 even! |
| 20:09.57 | hroarke | polycom 601 rh? |
| 20:10.00 | hroarke | eh? |
| 20:10.05 | [TK]D-Fender | hehe |
| 20:10.07 | [av]bani | snom, polycom, cisco, all well tested with asterisk |
| 20:10.12 | [av]bani | recommended depends on your needs |
| 20:10.13 | Assid | [av]bani: thats kinda taking advantage of the poor guy |
| 20:10.22 | [av]bani | Assid: no, its revenge |
| 20:10.28 | Assid | but hell hes pretty good at it |
| 20:10.32 | hroarke | well im trying to avoid stuff like the cisco where i have to flash firmware to sip to make it work |
| 20:10.42 | hroarke | these are for customers and i just cant expcet them to do that ;p |
| 20:10.52 | shido6 | erf? |
| 20:10.56 | [av]bani | hroarke: hold on a sec and i'll make a page with some info for you |
| 20:10.57 | [TK]D-Fender | hroarke : Polycom and Cisco make the best phones to date, Polycom at a much better price point however. Sipura is iffy, Snom is an "interesting" option. |
| 20:11.00 | wisdom | hroarke what's what conslutants get paid for :) |
| 20:11.02 | [av]bani | <types really fast> |
| 20:11.06 | *** join/#asterisk intensedr (n=scolson@209.172.11.52) |
| 20:11.08 | [av]bani | ok hroarke here you go -> http://bani.anime.net/phonez/ |
| 20:11.17 | hroarke | thanks ! :) |
| 20:11.17 | Seldon1975 | is anyone here a member of Orkut |
| 20:11.32 | _Sam-- | and the grandstream gxp2000s really arent that bad if you are on a tighter budget :) |
| 20:11.34 | Zeeek | i got a haircut |
| 20:11.47 | EksilAndyCap | hroarke: aren't you going to deliver the phones already set up? |
| 20:11.48 | [av]bani | i used to be, till we got flooded with racists from south america |
| 20:11.50 | wisdom | Zeeek which one did they cut ? |
| 20:11.58 | Zeeek | all four! |
| 20:12.09 | znoG | guys, if I have an incoming call to a FXO module on the TDM400P, and I make it ring a FXS module, and there's no echo, but when the call comes in via FXO and I make it ring an ATA there IS echo, would it be safe to deduce that the echo is being produced by the ATA? |
| 20:12.32 | Zeeek | where is the ATA? |
| 20:12.39 | znoG | its on the network |
| 20:12.53 | Zeeek | at likek 5ms lag? |
| 20:12.57 | znoG | i guess i could cross over connect it to the box, to be certain thats not a prob |
| 20:13.01 | hroarke | EksilAndyCap: well if we are shipping 1,000's of systems that probably wont be too efficient |
| 20:13.02 | znoG | i'll check the lag, brb |
| 20:13.11 | gammacoder | _Sam--: I don't see the reboot loop you are referring to - and I haven't changed moved the bin files off my TFPT server - can you provide any more details |
| 20:13.13 | hroarke | EksilAndyCap: im not sure yet |
| 20:13.37 | znoG | Zeeek: yep about 1ms lag or so |
| 20:13.50 | Zeeek | who hears the echo? |
| 20:13.58 | hroarke | EksilAndyCap: but i still would hate to add flashing firmware on a phone to the process for every phone |
| 20:13.59 | _Sam-- | sure, using http to update the phone....the phone will download repeatedly the 2nd stage files over and over and over and just keep turning itself off then booting up and getting the same files...FALLE on this channel had his phone do it for 48 minutes before i told him how to solve it |
| 20:14.04 | znoG | Zeeek: i kept getting echo on inbound FXO calls so I thought I'd try and ring a FXS port to see what happens, and there's no echo. |
| 20:14.11 | gammacoder | ahh - I'm using TFTP |
| 20:14.12 | znoG | Zeeek: the ATA side hears echo |
| 20:14.14 | pointer | was the STAT function implemented in 1.2.0? |
| 20:14.16 | _Sam-- | and my phones experienced the same problems |
| 20:14.19 | Zeeek | hmmmm |
| 20:14.27 | _Sam-- | the only way to get it out of the reboot loop was to remove the files |
| 20:14.28 | pointer | the wiki says it's there...but the function isn't registered |
| 20:14.29 | _Sam-- | or mv them |
| 20:14.47 | Drew_____ | d-fender - ok i forwarded the RTP ports aswell as the SIP ports, i added externhost, externrefresh and localnet to sip.conf but the caller still gets a 404 - new sip.conf: http://pastebin.ca/38705 |
| 20:14.57 | EksilAndyCap | hroarke: yeah, should suffice to plug them in and make a test call, the provisioning system doing the rest. :-P |
| 20:14.57 | gammacoder | guess I got luck in my choice to use TFTP |
| 20:15.09 | _Sam-- | i have tftp, just easier for http |
| 20:15.14 | Zeeek | Drew_____ use sip debug to see if anything is getting there |
| 20:15.15 | _Sam-- | maybe i will try it for the others |
| 20:15.25 | Zeeek | or ethereal even |
| 20:15.35 | *** join/#asterisk OnuR (n=Ce_S_eT@62.220.217.78) |
| 20:16.01 | gammacoder | i'll test and add a warning about the HTTP reboot loop bug to my instructions - thx |
| 20:16.04 | Drew_____ | zeeek - there are sip packets coming from the caller |
| 20:16.15 | Drew_____ | so i guess its not the NAT |
| 20:16.17 | Zeeek | so you need to see what they say |
| 20:16.36 | [TK]D-Fender | Drew_____ : Sorry, "allowguest" should be singular, not plural. |
| 20:16.38 | Zeeek | register, invite... |
| 20:16.49 | [TK]D-Fender | You DID neeed the NAT stuff too though... |
| 20:16.50 | *** join/#asterisk jyukes (i=jyukes@0.sub-70-197-233.myvzw.com) |
| 20:17.06 | _Sam-- | no, thank you! |
| 20:17.25 | Seldon1975 | [av]bani: in Orkut? |
| 20:17.39 | [av]bani | Seldon1975: yep |
| 20:17.41 | Hmmhesays | i lost my damn cell phone |
| 20:17.43 | Hmmhesays | argh |
| 20:17.53 | twisted[asteria] | did you call it? |
| 20:17.54 | Drew_____ | what am i looking for in the sip debug? |
| 20:18.02 | Seldon1975 | [av]bani: so you're no longer a member at all? |
| 20:18.12 | cpm | Hmmhesays, Gimme the number, I'll call it :) |
| 20:18.13 | Hmmhesays | it ran out of batteries yesterday |
| 20:18.14 | [av]bani | Seldon1975: no, i just never logon anymore |
| 20:18.20 | Hmmhesays | thats the problem |
| 20:18.20 | [TK]D-Fender | I had my SMTP port blocked, and it had queued up all my VM's and I just got a huge batch at once :D |
| 20:18.25 | Zeeek | you should see a wordy description of what's going on |
| 20:18.27 | Drew_____ | there is somthing about "found no matching peer or user for '{ip of caller}:5060' " |
| 20:18.34 | cpm | that's pretty hateful |
| 20:18.37 | Hmmhesays | i didn't make it home for a couple nights in a row |
| 20:18.47 | [TK]D-Fender | Drew_____ : Just fix the "allowguest" line like I said, do a "reload" and should work. |
| 20:18.53 | Zeeek | sounds right |
| 20:19.00 | [av]bani | Seldon1975: racists from brazil floded all the groups i was in, i got so disguste with it that i stopped going |
| 20:19.21 | Zeeek | by the way, somoen called my phone directly the other day. |
| 20:19.28 | [TK]D-Fender | [av]bani : Damn Brazilians!!! All the same!!! ;) |
| 20:19.30 | Seldon1975 | [av]bani: how weird |
| 20:19.36 | Zeeek | there was a missed call from xyz@ipaddr |
| 20:19.43 | Drew_____ | dfender - the allowguest is in the config and is reloaded |
| 20:19.49 | Zeeek | I tried calling it back for grins but no go |
| 20:19.55 | Hmmhesays | i'm going ot have to sign a contract to get a new one too |
| 20:19.55 | Seldon1975 | [av]bani: were they racist against whites? |
| 20:20.02 | Zeeek | not thru asterisk, direct to the ip phone |
| 20:20.35 | EksilAndyCap | Seldon1975: blanco! hmm, what movie was that |
| 20:20.40 | brettnem | hey my mpg123 process is at 99% cpu.. can I kill it??? |
| 20:20.50 | Seldon1975 | heheh |
| 20:20.50 | brettnem | without breaking calls that are up? |
| 20:21.10 | Zeeek | brettnem you'd better befoore it leaves town with your wife! |
| 20:21.33 | Hmmhesays | i liked that phone tool |
| 20:21.35 | Hmmhesays | *too |
| 20:21.42 | Seldon1975 | can someone please tell me what this measn (without being sarcastic): |
| 20:21.45 | Seldon1975 | WARNING[2337]: file.c:583 ast_readaudio_callback: Failed to write frame |
| 20:22.00 | brettnem | seriously.. I'm getting a lot of static on my calls on that server.. it's averaging 40% idle but mpg123 is showing 99% in top |
| 20:22.39 | _Sam-- | gammacoder: keep me posted if you have the same troubles via http upgrade |
| 20:22.41 | Zeeek | brettnem there's some stuff on the mailing list about that |
| 20:22.44 | brettnem | any ideas anyone? |
| 20:23.10 | brettnem | Zeeek: most of them are about using the wrong version of mpg123 or something.. |
| 20:23.21 | wunderkin | brettnem: don't use mpg123 |
| 20:23.22 | gammacoder | _Sam--: just got through one - it rebooted 2x and seems to be working |
| 20:23.33 | Zeeek | I had a runnaway a while back that did that btw |
| 20:23.41 | wunderkin | brettnem: what version of asterisk? are all of the calls using the same codec? |
| 20:23.41 | brettnem | yeah, I know that.. but this system has been running for a while.. need to get it in a downtime to change it. |
| 20:23.46 | [av]bani | Seldon1975: against jews mainly |
| 20:23.48 | Zeeek | I never knew it until a friend logged in and told me! |
| 20:24.15 | Seldon1975 | [av]bani: how strange. Any chance can you send me an invite to join? |
| 20:24.29 | brettnem | so can I kill mpg123 without effecting in progress calls? or will asterisk barf? |
| 20:24.30 | Hmmhesays | sweet caroline |
| 20:24.37 | Hmmhesays | god i feel like a million bucks now that i'm not drunk |
| 20:24.43 | [av]bani | Seldon1975: if i can find my login stuff again maybe |
| 20:24.46 | Zeeek | brettnem you can kill it I'd say |
| 20:24.59 | Seldon1975 | [av]bani: thanx - I've /msgd you my email |
| 20:25.05 | gammacoder | _Sam--: http://pastebin.ca/38706 http logs look as I'd expect them - so I guess I can't reproduce it yet |
| 20:25.22 | _Sam-- | well, too bad falle isnt still here |
| 20:25.26 | _Sam-- | he could concur my story |
| 20:25.32 | Drew_____ | zeeek, dfender - i pasted the sip debug output of a attempted call to my ip adress: http://pastebin.ca/38707 |
| 20:25.33 | _Sam-- | and ive seen it on every phone ive flashed so far |
| 20:25.49 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 20:25.50 | znoG | Zeeek: strange! i disabled echotraining and the echo has stopped??? |
| 20:26.01 | *** join/#asterisk ckruetze (n=ckruetze@i577A4B56.versanet.de) |
| 20:26.16 | [TK]D-Fender | Drew_____ : Make a NAMED exten in the contet and have him test that |
| 20:26.24 | Zeeek | znoG good to know, thanks! |
| 20:26.27 | [av]bani | yay the phones arrived |
| 20:26.30 | [av]bani | polycom and snom |
| 20:26.33 | [av]bani | and gxp2000 :) |
| 20:26.49 | Drew_____ | i.e. call incoming@myip:5060 ? |
| 20:26.52 | Seldon1975 | [av]bani: 301s? |
| 20:26.52 | _Sam-- | [av]bani: go upgrade the gxp2000 :) |
| 20:26.59 | _Sam-- | and tell me if it gets stuck in the reboot loop via http |
| 20:27.00 | Seldon1975 | [av]bani: 501s? |
| 20:27.07 | [av]bani | we already have one, its running .13 |
| 20:27.10 | [av]bani | Seldon1975: 601s |
| 20:27.13 | znoG | can anyone explain the reason why echo might stop when echotraining is set to "no" ?? when set to "yes", it was *causing* echo?!?! |
| 20:27.14 | Seldon1975 | niice |
| 20:27.17 | [av]bani | will make fender happy |
| 20:27.18 | [TK]D-Fender | Drew_____ : sure |
| 20:27.33 | [av]bani | so when they go south, i can yell @ fender |
| 20:27.44 | gammacoder | _Sam--: will you try my instructions at http://www.voip-info.org/wiki/view/GXP-2000 and see if they work w/o issue for you (they are about 1/3 way down and tagged with "gammacoder") |
| 20:28.15 | [TK]D-Fender | [av]bani : You go south you cheap-assed pessimist! :D |
| 20:28.17 | gammacoder | then I'll add your warning about HTTP loops |
| 20:28.21 | Zeeek | znoG maybe if it's training it isn't good at it yet? :) |
| 20:28.32 | Seldon1975 | hahah |
| 20:28.54 | Zeeek | but what do I know, my wife is out with 30 guys tonight |
| 20:29.02 | brettnem | echotraining sucks |
| 20:29.05 | [av]bani | [TK]D-Fender: 601s arent cheap :P |
| 20:29.17 | brettnem | echo training will happen each time there is silence on the line.. over and over.. |
| 20:29.25 | brettnem | retraining each time there is silence |
| 20:29.39 | [TK]D-Fender | [av]bani : I meant YOU :) |
| 20:30.12 | [av]bani | [TK]D-Fender: i bought a snom! |
| 20:30.14 | [av]bani | 360 |
| 20:30.18 | Zeeek | what I find is regardless of the technology of the channel, incoming FXO calls have a long bridging delay (maybe half a sec) when you answer |
| 20:30.54 | _Sam-- | gammacoder: i will definitely try it your way ...have a few loose ends to wrap up first |
| 20:30.54 | [TK]D-Fender | [av]bani : Fine, now just try them out. I may help with provisioning them with you one night |
| 20:31.02 | gammacoder | thx |
| 20:31.14 | [av]bani | woo, a new gxp2000 beta |
| 20:31.17 | *** join/#asterisk niZon (n=ilt@S0106deadbeefbeef.wp.shawcable.net) |
| 20:31.18 | _Sam-- | i do think the firmware is better than .13 |
| 20:31.28 | _Sam-- | [av]bani: where you been, its been out for HOURS! |
| 20:31.30 | [av]bani | _Sam--: cant be any worse than stock firmware |
| 20:31.38 | [av]bani | _Sam--: HOURS!! @_@!!! |
| 20:31.39 | Hmmhesays | now i lost my debian stable cd |
| 20:31.45 | niZon | anyone know why two asterisk servers would just suddenly refuse to communicate? |
| 20:32.11 | Zeeek | sex change? |
| 20:32.43 | [av]bani | niZon: one of them found the other sleeping with a cisco callmanager? |
| 20:32.44 | brettnem | niZon: solar flares |
| 20:32.56 | [TK]D-Fender | niZon : read the topic |
| 20:33.11 | Zeeek | <[av]bani heheh |
| 20:33.12 | [av]bani | [TK]D-Fender: shhh thats a secret |
| 20:33.30 | _Sam-- | [av]bani: can you upgrade your GXP using http? |
| 20:33.38 | SwK[Work] | hey whats the alarm debounce thing you can set on the T1 cards? |
| 20:33.42 | _Sam-- | we (gammacoder and myself) are trying to see if you have a problem |
| 20:33.43 | SwK[Work] | wheres the docs on tht hidden |
| 20:33.55 | RussCC | hello, I am having a problem with my TDM13B I have my line pluged into the first FXO port but when a call is recived I dont get Caller ID to come across |
| 20:33.56 | Zeeek | I've always been a little shy with upgrades but this latest is scary! |
| 20:34.04 | RussCC | anyone have any idea |
| 20:34.14 | Drew_____ | zeek, fender - it works with incoming@myip - audio is oneway - but my friend hasnt forwarded his RTP ports so that should be fixable - is there no way of capuring any call? ie. without the extension? |
| 20:34.19 | Zeeek | RussCC what country? |
| 20:34.30 | RussCC | Zeeek USA |
| 20:34.44 | Zeeek | what's in zapata? |
| 20:34.57 | Seldon1975 | wow |
| 20:35.08 | Seldon1975 | i just told digium about my crosstalk issue and their reply was:\ |
| 20:35.10 | Seldon1975 | Sir, In wctdm2400.c try changing the line "wctdm_vpm_out(wc, x, 0x023, 0x7f);" to "wctdm_vpm_out(wc, x, 0x023, 0x3f);" |
| 20:35.23 | [TK]D-Fender | Drew_____ : I think with one of "i" or "_." |
| 20:35.25 | Zeeek | shit that was obvious! |
| 20:36.51 | [av]bani | _Sam--: i didnt have problems with .12 and .13 via http before |
| 20:36.53 | *** join/#asterisk areski (n=areski@201.Red-88-5-211.staticIP.rima-tde.net) |
| 20:36.59 | _Sam-- | ravenpi: me either |
| 20:37.02 | RussCC | Zeeek http://pastebin.ca/38709 |
| 20:37.02 | Seldon1975 | Zeeek: the change to wctdm2400.c |
| 20:37.04 | _Sam-- | er [av] : me iether |
| 20:37.09 | _Sam-- | not until this version |
| 20:37.14 | QbY | True or False: "Hold" doesn't exist in Asterisk -- Only Call Parking? |
| 20:37.26 | _Sam-- | myself, and falle...both had gxps get caught in a reboot loop when upgrading via http |
| 20:38.22 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 20:38.29 | _Sam-- | it was easily fixed (no worries there) and the phones upgraded fine |
| 20:38.34 | [TK]D-Fender | QbY : False. |
| 20:38.37 | _Sam-- | im just wondering if you will have the same experience |
| 20:38.43 | brettnem | QbY: how about "hold doesn't exist at all. it's simply a media redirection" |
| 20:38.51 | intensedr | anyone know if a VOIP Provider can go to NANPA and order NPA-NXX Is every state? |
| 20:38.52 | Zeeek | RussCC looks ok but I don't know AMP |
| 20:38.54 | *** join/#asterisk cyburdine (n=cyburdin@208.2.145.2) |
| 20:39.02 | QbY | media redirection eh? |
| 20:39.04 | Zeeek | Seldon1975 I was being sarcastic |
| 20:39.07 | [TK]D-Fender | brettnem : Don't go all metaphysical on him now! |
| 20:39.11 | brettnem | intensedr: if that voip provider is a LEC, sure |
| 20:39.18 | intensedr | what about a CLEC? |
| 20:39.22 | brettnem | [TK]D-Fender: heh ... :) |
| 20:39.25 | [TK]D-Fender | QbY : Yes these is such a thing as "hold" |
| 20:39.31 | RussCC | Zeeek its odd because my $30 clone card does it with out error |
| 20:39.32 | Zeeek | I have a change to a line in wcfxs.c to get our phones to ring |
| 20:39.37 | brettnem | intensedr: sure, a CLEC can do it.. I'm a clec.. I get my own NPANXX |
| 20:39.55 | intensedr | Im a CLEC Out of MI.. I can order NPA-NXX's in California? |
| 20:40.05 | QbY | [TK]D-Fender: I don't see it anywhere in my PAP2 -- I don't know which HookFlash code to use to activate it.. |
| 20:40.10 | brettnem | intensedr: come on now.. answer your own question |
| 20:40.12 | Zeeek | RussCC it is odd. I amnot using FXO modules at the moment tho |
| 20:40.25 | brettnem | intensedr: what happens when someone calls that calif NPANXX?? where does the call go? |
| 20:40.31 | [TK]D-Fender | QbY : It should be in there somewhere... just like all the SPA's |
| 20:40.41 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 20:40.41 | RussCC | Zeeek: thank you trying |
| 20:40.45 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt) |
| 20:40.48 | Zeeek | np |
| 20:41.06 | intensedr | to my switch |
| 20:41.42 | brettnem | intensedr: ok. good.. so you have T1s from your switch in MI to the ILEC switch in CA.. sure that works |
| 20:42.14 | brettnem | intensedr: I do the same thing.. I have a switch in Austin and I get NPANXX in San Antonio for example.. I do that in lots of places. |
| 20:42.40 | brettnem | intensedr: but you need connectivity to that lata to pick up the NPANXX.. to terminate locally as well |
| 20:43.10 | brettnem | ARGGGH |
| 20:43.12 | brettnem | I hate static |
| 20:43.15 | intensedr | ahh So going voip nation wide going with DID's from major CLECso or lecs would be the best route |
| 20:43.38 | brettnem | intensedr: I'm not sure how you put 1 and 1 together.. but you make a reasonably valid statment |
| 20:43.57 | brettnem | for other reasons.. |
| 20:44.02 | intensedr | Take Vonage for example |
| 20:44.11 | intensedr | They down own any of their NPA-NXX's |
| 20:44.14 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.9) |
| 20:44.17 | brettnem | yay vonage.. the golden example |
| 20:44.27 | intensedr | All the big boys dont |
| 20:44.40 | brettnem | yeah, like L3.. sure go on |
| 20:44.45 | [av]bani | wow, 601 takes ages to boot |
| 20:44.50 | Dr-Linux | hi guys, howdy? |
| 20:44.57 | brettnem | [av]bani: all the polycoms do.. sux |
| 20:45.01 | brettnem | nice phones to |
| 20:45.02 | brettnem | tho |
| 20:45.05 | Zeeek | all Polycoms take a while |
| 20:45.05 | intensedr | So its more cost effective to go the DID route than have Facilities in each state |
| 20:45.24 | justinu | probably depends on your volume |
| 20:45.30 | *** join/#asterisk rculp (n=rculp@66.173.240.20) |
| 20:45.37 | brettnem | intensedr: there are advantages of being a facilities based provider that are well beyond the scope of this group. :) |
| 20:45.40 | [TK]D-Fender | 90-120 sec boot... |
| 20:46.06 | [av]bani | whats the default user/pass for 601? |
| 20:46.08 | brettnem | but if you are intending to be a large carrier.. facilities based is the way to go.. Especially since UNE-P is gone. |
| 20:46.16 | brettnem | 456 |
| 20:46.19 | Zeeek | for info, the SPA-941 takes about 10 secs to boot including DHCP |
| 20:46.26 | [TK]D-Fender | [av]bani : "Polycom", "456" |
| 20:46.29 | Zeeek | Polycom 456 |
| 20:46.34 | Zeeek | FUCKING ECHO |
| 20:46.37 | Drew_____ | dfender - works - thanks for the help |
| 20:46.39 | brettnem | I win |
| 20:46.50 | [TK]D-Fender | Drew_____ : YWC |
| 20:46.51 | Zeeek | brettnem no, no pw |
| 20:46.56 | brettnem | Zeeek: gotta get to your buzzer quicker |
| 20:47.01 | Dr-Linux | can we connect to asterisk manager API from any other no-asterisk linux machine? |
| 20:47.03 | Drew_____ | thanks to zeeeks too |
| 20:47.03 | QbY | [TK]D-Fender: Could you peek at this and tell me where hold would be?? http://www.myfone.com.au/faq/LinksysPAP2/Linksys%20PAP2%20Configuration.htm |
| 20:47.04 | [TK]D-Fender | Zeeek : I think its on YOUR end ;) |
| 20:47.14 | Zeeek | HUH huh huh huh? |
| 20:47.18 | brettnem | Dr-Linux: it's just a tcp port.. |
| 20:47.28 | brettnem | Dr-Linux: you in Tx? |
| 20:47.31 | Zeeek | Drew_____ it works? Great! |
| 20:47.34 | EksilAndyCap | Dr-Linux: manager.conf has the allow settings iirc. |
| 20:48.06 | Dr-Linux | [TK]D-Fender: i have changed the SIP port |
| 20:48.23 | Dr-Linux | brettnem: i'm in Pakistan |
| 20:48.37 | *** join/#asterisk bprice20 (n=brandon@Dynamic-216.120.224.153.hrnoc.net) |
| 20:48.37 | brettnem | Dr-Linux: people say howdy in pakistan? nice.. :) |
| 20:48.39 | Hmmhesays | sixtel takes their sweet time activating numbers |
| 20:48.51 | brettnem | ooohh.. their sweeeett time.. |
| 20:49.05 | brettnem | Hmmhesays: but those are sweeeeeeet dids theys got |
| 20:49.10 | Hmmhesays | anyone knows how aah stores the numbers for call forwarding? |
| 20:49.22 | Hmmhesays | i think i shall borrow the script |
| 20:49.23 | brettnem | Hmmhesays: molecularly |
| 20:49.28 | intensedr | how is UNE-P gone? |
| 20:49.43 | Dr-Linux | brettnem: no, no one knows here whats "howdy" i use this for US guys though :P |
| 20:50.25 | brettnem | ahhh.. I see.. "what's a howdy" haha |
| 20:50.47 | brettnem | intensedr: here's some light reading for you http://hraunfoss.fcc.gov/edocs_public/attachmatch/FCC-01-361A1.pdf |
| 20:51.05 | Dr-Linux | brettnem: its not "how day" ? :P |
| 20:51.15 | Hmmhesays | ugh it uses dbput and dbget |
| 20:51.19 | Hmmhesays | that blows monkey goats |
| 20:51.45 | RussCC | Zeeek: one last thing when I do a ztcfg -vvv I get that all of my moduels are FXS but when I purchaed them it says they are FXO any ideas on that? |
| 20:51.48 | [TK]D-Fender | QbY : I think you do a hook-flash, then hangup |
| 20:51.59 | brettnem | Dr-Linux: it's more like "how dee" :) |
| 20:52.05 | QbY | k |
| 20:52.15 | Zeeek | RussCC doesn't it say they USE FXS signalling? |
| 20:52.40 | Zeeek | what is the exact line? |
| 20:52.52 | brettnem | intensedr: the incumbants like to squash non-facilities providers like bugs |
| 20:53.04 | Dr-Linux | brettnem: yeah, but meanings go towards like >> hows your day ? isn't it |
| 20:53.22 | [TK]D-Fender | QbY : unless I'm mistaken it will give you a new tone, and then "blip" the phone intermittantly to let you know you left them hanging. |
| 20:53.26 | QbY | [TK]D-Fender: Excellent.. Thank you.. |
| 20:53.30 | brettnem | Dr-Linux: yeah it's a nonsense word that just means hi in a friendly kinda way |
| 20:53.38 | [TK]D-Fender | QbY: test it fast though |
| 20:53.41 | Zeeek | how day may be true tho |
| 20:53.54 | [TK]D-Fender | Today is "Why day" :) |
| 20:53.56 | QbY | I did just test it.. I'm listening to hold music riight now.. |
| 20:54.01 | Zeeek | and when day? |
| 20:54.04 | brettnem | [TK]D-Fender: why is today why day? |
| 20:54.07 | Zeeek | that was yesterday |
| 20:54.12 | [TK]D-Fender | brettnem : EXACTLY! |
| 20:54.15 | Zeeek | Wenday |
| 20:54.19 | RussCC | Zeeek: Channel 01: FXO Kewlstart (Default) (Slaves:01) |
| 20:54.34 | brettnem | [TK]D-Fender: We're having earthquake weather here today |
| 20:54.37 | Zeeek | that doesn't look right |
| 20:54.37 | RussCC | Zeeek: Channel 02: FXS Kewlstart (Default) (Slaves:02) |
| 20:54.44 | RussCC | and so on |
| 20:54.49 | Zeeek | which is which? |
| 20:54.57 | brettnem | Zeeek: exactally |
| 20:55.06 | Zeeek | exatimenos |
| 20:55.07 | Dr-Linux | earthquake .. ufff |
| 20:55.21 | Zeeek | RussCC what is the first and second modules? |
| 20:55.23 | brettnem | yep.. looks like earthquakes outside |
| 20:55.33 | brettnem | I can smell it |
| 20:55.34 | RussCC | Zeeek: the FXO is what I thought and Digium says is a FXS module |
| 20:55.56 | justinu | that's normal |
| 20:55.58 | Zeeek | looks like 1 is a FXS and 2 is FXO |
| 20:56.00 | RussCC | Zeeek: and the FXS is what I thought and Digium says is a FXO module |
| 20:56.02 | [TK]D-Fender | brettnem : Los Angeles? |
| 20:56.02 | Dr-Linux | Paki peoples afraid now from this name "earthquake" |
| 20:56.07 | brettnem | Digium has made a mess of this FXO/FXS business.. They didn't have to go explain it and confuse everyone.. |
| 20:56.09 | Dr-Linux | word* |
| 20:56.13 | Zeeek | haha |
| 20:56.16 | brettnem | [TK]D-Fender: Texas |
| 20:56.26 | rculp | need help from someone real quick to help me point blame back at our clec for their mistake that they're not owning up to |
| 20:56.28 | justinu | Dr-Linux: what's up? |
| 20:56.28 | rculp | http://pastebin.ca/38715 |
| 20:56.36 | rculp | that's the output from pri show span 1 |
| 20:56.36 | brettnem | [TK]D-Fender: It probably won't happen.. but it smells like it. |
| 20:56.37 | Zeeek | RussCC FXO connects to the phone line |
| 20:56.40 | [TK]D-Fender | ok, gtg, later all |
| 20:56.46 | Zeeek | FXS gives dialtone to a phone |
| 20:56.49 | Dr-Linux | hey Justin |
| 20:56.53 | brettnem | rculp what's the prob? |
| 20:56.58 | RussCC | Zeeek: oh ok |
| 20:57.05 | Dr-Linux | justinu: you were much busy last day? hain na? |
| 20:57.11 | Dr-Linux | :) |
| 20:57.12 | Zeeek | and you can burn one out by connecting to the wrong side |
| 20:57.14 | justinu | rculp: your T1 appears down |
| 20:57.20 | rculp | right |
| 20:57.23 | justinu | Dr-Linux: : yeah, quite busy lately |
| 20:57.23 | rculp | it was working |
| 20:57.27 | brettnem | rculp: what makes you think this isn't your fault? |
| 20:57.34 | rculp | and then dropped in the middle of calls |
| 20:57.38 | brettnem | you got rats nibbling your wires |
| 20:57.47 | rculp | verizon tested the line |
| 20:57.54 | rculp | from co to our building |
| 20:57.56 | rculp | but our clec |
| 20:57.58 | rculp | didn't test |
| 20:58.00 | [av]bani | aggghhh |
| 20:58.01 | rculp | we've replaced |
| 20:58.05 | rculp | all physical lines |
| 20:58.06 | [av]bani | i change one little thing on the 601 and it has to reboot |
| 20:58.09 | rculp | from dmarc |
| 20:58.11 | brettnem | your clec will have verizon do the test.. that's normal |
| 20:58.13 | rculp | to wildcard |
| 20:58.15 | [av]bani | another long wait |
| 20:58.16 | Zeeek | That's Polycom av |
| 20:58.22 | rculp | to eliminate |
| 20:58.24 | rculp | issues |
| 20:58.25 | brettnem | did you test from the demarc to your PC? |
| 20:58.29 | Zeeek | the only reasonable way if by the XML files |
| 20:58.36 | [av]bani | :< |
| 20:58.44 | Zeeek | or infinite patience! |
| 20:58.50 | RussCC | Zeeek: FXO module allows the TDM400P card to terminate analog telephone lines (POTS) |
| 20:58.52 | rculp | well, looking for the best way to eliminate our gear |
| 20:58.57 | rculp | but |
| 20:59.01 | rculp | nothing has changed |
| 20:59.02 | Zeeek | Every new Polycom owner goes thru this phase grasshopper :) |
| 20:59.03 | Hmmhesays | ok my phone is in the fitness place, how the fark did it get there |
| 20:59.05 | Hmmhesays | i don't exercise |
| 20:59.12 | brettnem | small arms explosives show eliminate your gear |
| 20:59.23 | Zeeek | RussCC yup; FXO module connects to the phone line as I said |
| 20:59.42 | brettnem | s/show/should/ |
| 20:59.47 | brettnem | thanks jbot |
| 21:00.06 | brettnem | s/thanks/you suck/ |
| 21:00.07 | rculp | any ideas to help eliminate my end? |
| 21:00.16 | RussCC | Zeeek: right I have 3 FXO ports and one FXS ztcfg -v says the oposite |
| 21:00.23 | franck | Hi |
| 21:00.40 | justinu | RussCC: you need to use the opposite signalling, it's normal |
| 21:00.44 | Zeeek | RussCC no it's right |
| 21:00.50 | *** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 21:00.53 | franck | anyway to detect the difference with SIP, between the person is on the phone with the SIP phone is not registered? |
| 21:00.53 | brettnem | DON'T PANIC |
| 21:01.03 | Zeeek | chnnel 1 is FXS, 234 FXO |
| 21:01.14 | brettnem | franck: for what purpose? |
| 21:01.35 | jbalcomb | iCEBrkr yeah, Digium TE411P |
| 21:01.45 | rculp | anyone? |
| 21:01.56 | Dr-Linux | to interact with external program, AGI is good approach or API ? |
| 21:01.58 | justinu | rculp: use loopbacks |
| 21:01.59 | RussCC | Zeeek: correct but zapta.conf and ztcfg say that 1 is a FXO and 234 are FXS |
| 21:02.08 | franck | brettnem: when you call a sip phone, I would like to differentiate the phone is off hook or * cannot find the phone. "The person is on the phone" "The person is not available" |
| 21:02.12 | rculp | justinu: on the wildcard? |
| 21:02.20 | rculp | justinu: used one on the line over |
| 21:02.21 | jbalcomb | iCEBrkr Tonight we are taking out the Wildcard TE110P that is also in there. I don't know about IRQ stuff in the Linux though. |
| 21:02.24 | brettnem | franck: show application chanisavail |
| 21:02.24 | Zeeek | franck look in the db, there are variables |
| 21:02.31 | RussCC | Zeeek: which is kind of screwed up if you ask me |
| 21:02.32 | justinu | rculp: you could try, but phsyical plugs are often more reliable |
| 21:02.42 | Zeeek | RussCC no I think it's right |
| 21:02.48 | brettnem | franck: also read README.variables in the docs dir and the ENTIRE show application dial |
| 21:02.48 | justinu | russcc: it's normal. |
| 21:02.51 | franck | how do I do that with extensions.conf ? |
| 21:02.54 | Zeeek | I believe it's telling you the signalling |
| 21:02.59 | brettnem | franck: read |
| 21:03.04 | justinu | zeeek is correct. |
| 21:03.06 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 21:03.06 | franck | brettnem: ok |
| 21:03.09 | justinu | for FXO ports, you use FXS signalling |
| 21:03.09 | rculp | justinu: when it was plugged into just the smartcard with a loopback there's no alarm on the smart card. when the line is plugged into the wildcard there's no alarm on the smart card |
| 21:03.10 | Zeeek | franck database show |
| 21:03.31 | RussCC | franck: thank you |
| 21:03.38 | brettnem | Zeeek: I don't think that's what he wants.. |
| 21:03.40 | RussCC | Zeeek: thank you |
| 21:03.48 | [av]bani | ahah. and another 2 minutes after its "booted" before the web admin comes up |
| 21:03.57 | Zeeek | ok RussCC - can't figure what your problem is tho |
| 21:03.57 | cyburdine | can someone help me with commands in sip.conf? |
| 21:04.02 | Dr-Linux | justinu: whats difference between port= and bindport= ? :) |
| 21:04.11 | brettnem | one is right, and one is wrong? |
| 21:04.13 | justinu | Dr-Linux: one works, the other doesn't :P |
| 21:04.17 | rculp | justinu: determined that by changing wires and having one of my techs watch the smart card as I plugged and unplugged cables |
| 21:04.25 | *** join/#asterisk cjk (n=cjk@11.121.9.213.dsl.getacom.de) |
| 21:04.31 | justinu | rculp: smartjack loopback facing your CPE? |
| 21:04.41 | cyburdine | I'm looking at sip debug and wanting to change info and need to know what commands change the From: and Call-ID: fields |
| 21:04.51 | cjk | hi, is there a tool tos niff the network traffic and decod an rtp or iax stream into audio in realtime? |
| 21:04.54 | rculp | justinu: that's the only thing I have not done is plug it in to our asterisk box |
| 21:04.54 | justinu | fromdomain, fromuser |
| 21:04.56 | Zeeek | RussCC DON'T try switching th two tho, you'll burn one of them out |
| 21:05.01 | brettnem | cyburdine: don't change call-id.. you can't |
| 21:05.15 | cyburdine | hmm that doesn't seem to make a difference |
| 21:05.18 | justinu | rculp: if you can have the smartjack loop to your * box, and the circuit goes GREEN, you're all set. |
| 21:05.27 | justinu | rculp: don't expect the d-channel to come up to a loop tho |
| 21:05.30 | cyburdine | it seems to be taking anything that is in my register=> line |
| 21:05.31 | jebba | Any idea why MeetMe won't record? I have this: Set(MEETME_RECORDINGFILE=/tmp/recording_test) and on the console it says: "Starting recording of MeetMe Conference 101 into file /tmp/recording_test.wav" But no file appears there (same for /var/spool/asterisk/recordings/) |
| 21:05.33 | RussCC | Zeeek: Right thank you |
| 21:05.36 | cyburdine | and putting that in as the from |
| 21:05.47 | *** join/#asterisk rene- (i=rene@201.144.61.222) |
| 21:05.47 | [av]bani | man, my linux pc boots up faster than the 601 |
| 21:05.48 | *** join/#asterisk Husk_ (n=tcg@202.55.153.169) |
| 21:05.53 | brettnem | from is the callerid stuff |
| 21:05.53 | Zeeek | haha |
| 21:06.11 | rene- | has anybody seen jerjer? |
| 21:06.15 | Zeeek | my old windows laptop boots faster than the ip500! |
| 21:06.19 | justinu | actually, RPID is the callerid stuff in real SIP \ |
| 21:06.21 | brettnem | ~seen jerjer |
| 21:06.25 | jbot | jerjer <n=jj@pdpc/supporter/bronze/jerjer> was last seen on IRC in channel #debian, 7d 23h 35m 31s ago, saying: 'thanks again'. |
| 21:06.25 | justinu | :P |
| 21:06.28 | Zeeek | jerjer is busy fixing the netwrok |
| 21:06.32 | justinu | lol |
| 21:06.32 | brettnem | justinu: whew.. thanks.. |
| 21:06.38 | rene- | heh |
| 21:06.38 | justinu | brettnem: :) |
| 21:07.05 | rculp | justinu: looped back the card |
| 21:07.11 | rculp | justinu: and it went green |
| 21:07.16 | cyburdine | my register line has 5551212:password@something.proxyname.com |
| 21:07.22 | justinu | rculp: sounds like you're ok then. |
| 21:07.36 | brettnem | cyburdine: that line has nothing to do with any kind of cid.. |
| 21:07.36 | rculp | justinu: going to have a tech do it while I watch the actual output on the terminal to see what it shows |
| 21:07.50 | brettnem | ok, I have to go.. have fun all.. |
| 21:07.54 | justinu | l8 |
| 21:08.00 | rculp | justinu: figured I was, but just wanted to have proof :) |
| 21:08.04 | brettnem | nah.. it's still early.. |
| 21:08.11 | rculp | justinu: thanks for the suggestion of looping back the wildcard |
| 21:08.14 | justinu | np |
| 21:08.23 | *** part/#asterisk pointer (i=pointer@aj.catt.com) |
| 21:08.38 | cyburdine | my fromdomain is proxyname.com |
| 21:08.42 | *** join/#asterisk Agreenberg (n=agreenbe@64.201.13.50) |
| 21:09.04 | cyburdine | when I watch sip debug it keeps placing something.proxyname.com in the from field |
| 21:09.16 | cyburdine | with I think is part of my problem not being able to register |
| 21:09.21 | Dr-Linux | <justinu> Dr-Linux: one works, the other doesn't :P >> yeah, but its not mentioned in WIKI or any common docs :S |
| 21:09.33 | *** part/#asterisk franck (n=franck@tikiwiki/franck) |
| 21:10.13 | Dr-Linux | justinu: if you change the sip port you need to do some odd changes at client side as well :P |
| 21:11.14 | *** join/#asterisk JSabines (i=JSabines@201.138.136.71) |
| 21:12.14 | justinu | if you used DNS SRV you wouldn't have that problem :P |
| 21:12.50 | cyburdine | dns srv? |
| 21:13.17 | Dr-Linux | DNS ok but sRV ? |
| 21:14.00 | Dr-Linux | ooo ic |
| 21:14.11 | Dr-Linux | justinu: understand |
| 21:14.15 | rene- | it is a mechanism where dns server helps you get redundancy by defining secundary servers in case of failutre |
| 21:14.17 | justinu | cool |
| 21:14.28 | justinu | IP addresses and ports |
| 21:14.30 | rene- | the sip endpoint must support it |
| 21:14.41 | justinu | which most seem to do |
| 21:15.13 | *** join/#asterisk Sjeemz (i=sjeemz@ipv6.sjeemz.nl) |
| 21:15.50 | Zodiacal- | anyone know why my * pots line picks up on the first ring, but it waits until the second ring before my sip extention starts rining? |
| 21:15.57 | Sjeemz | wow, busy channel |
| 21:16.00 | Dr-Linux | but what you will do with softclients by default they send requesto to server on 5060 port, server has sip port 80 tho ? |
| 21:16.40 | Dr-Linux | it listens on ONE port |
| 21:16.41 | justinu | DNS SRV handles that. |
| 21:16.49 | justinu | it points the client to port 80, instead of 5060. |
| 21:17.10 | justinu | Zodiacal-: caller ID capture, more than likely. |
| 21:17.20 | Zodiacal- | justinu i turned it off tho.. |
| 21:17.27 | justinu | dunno then... |
| 21:17.29 | Zodiacal- | usecallerid=no |
| 21:17.33 | Zodiacal- | justinu no ideas? |
| 21:17.38 | Zodiacal- | its quite frustrating.. |
| 21:17.56 | Zodiacal- | justinu does yours transfer instantly? |
| 21:18.11 | justinu | sorry, i don't use analog lines |
| 21:18.16 | Zodiacal- | ic |
| 21:19.17 | Sjeemz | question: I've just upgraded from 1.0.9 to 1.2.3, but all incoming calls with blocked callerid get displayed on my cisco 7960 as '"Uknown" <Unknown>', this should not happen, because I use Set(CALLERID(name)=name) on all incoming calls. CIDName does seem to get overridden for all incoming calls that do supply a callerid |
| 21:19.24 | Dr-Linux | justinu: is this fine, if i redirect 80 port request to 5060 in router? |
| 21:19.50 | Sjeemz | oh, and it did work in asterisk 1.0.9 :) |
| 21:19.53 | Dr-Linux | or change 5060 port to 80 |
| 21:20.01 | Dr-Linux | is fine |
| 21:20.28 | *** join/#asterisk alexxgreenberg (n=Alex@64.201.13.50) |
| 21:21.03 | Hmmhesays | you still trying to get around the pakistani government? |
| 21:21.13 | Seldon1975 | can someone please tell me what this measn (without being sarcastic): |
| 21:21.16 | *** join/#asterisk folsson_ (n=filip@lund-meje-sr0-vl101-249.perspektivbredband.net) |
| 21:21.17 | Seldon1975 | WARNING[2337]: file.c:583 ast_readaudio_callback: Failed to write frame |
| 21:21.32 | Hmmhesays | exactly what it says |
| 21:21.45 | justinu | dr-linux: yeah, that's called PAT and you could do that. |
| 21:22.01 | Hmmhesays | Seldon1975: it tried but theirs no channel |
| 21:22.05 | *** join/#asterisk razu (n=razu@ip59.cab62.mus.starman.ee) |
| 21:22.05 | Hmmhesays | *theres even |
| 21:22.13 | Seldon1975 | Hmmhesays: yes, but does it indicate something is wrong with my setup? |
| 21:22.13 | Hmmhesays | me grammar it good ja |
| 21:22.22 | Hmmhesays | Seldon1975: that can safely be ignored |
| 21:22.25 | Dr-Linux | yeah, but pating is better than changing 5060 ? |
| 21:22.27 | Dr-Linux | right |
| 21:22.33 | Seldon1975 | Hmmhesays: ok thx |
| 21:22.51 | Hmmhesays | Dr-Linux: don't screw around, just use openvpn with linksys openvpn enabled routes |
| 21:22.56 | *** join/#asterisk _cleric_ (n=dacleric@p5482824A.dip0.t-ipconnect.de) |
| 21:22.59 | Hmmhesays | headache, gone. |
| 21:23.17 | Hmmhesays | you don't want to fark around with changing your call signaling ports |
| 21:23.35 | *** part/#asterisk alexxgreenberg (n=Alex@64.201.13.50) |
| 21:24.50 | Hmmhesays | for 60 bucks you can get a linksys that you can use openvnp on |
| 21:25.15 | Hmmhesays | no hassle, no headache, no changing your callsignaling ports |
| 21:25.52 | *** part/#asterisk rob314 (n=root@207.58.194.55) |
| 21:25.55 | Sjeemz | Hmmhesays: but only on certain models |
| 21:26.01 | rene- | Hmmhesays do you think openvpn is harder to setup than vtun? |
| 21:26.15 | Hmmhesays | vtun? |
| 21:26.32 | Hmmhesays | being openvpn is a breeze i can't imagine anything much easier |
| 21:27.09 | rene- | in debian you have to create your tun inferfaces with mknod |
| 21:27.14 | rene- | in openvpn |
| 21:27.25 | rene- | in red hat and that is where i have used vtun you dont |
| 21:27.50 | rene- | but im a noob with vtun and probably in redhat it is easy |
| 21:27.53 | Nivex | which version of Debian? I've not had that problem. |
| 21:27.57 | rene- | sarge |
| 21:28.07 | Hmmhesays | wrt54gs |
| 21:28.15 | rene- | openwrt? |
| 21:28.22 | Hmmhesays | aye |
| 21:28.47 | rene- | Nivex, so you just edit your openvpn.conf and your good to go? |
| 21:28.52 | Hmmhesays | we're going to be deploying those with voip phones in russia and china, avoid any goverment hassle there |
| 21:29.02 | Nivex | rene-: that's been my experience |
| 21:29.04 | Hmmhesays | you can't block what you can't see |
| 21:29.26 | jbalcomb | Anyone have feelings about Digiums recomendation that we should use IDE hard drives rather than SCSI or SATA? |
| 21:29.30 | rene- | not every phone supports encription, i think it is alsoeasier to route calls tru iax tru vpn connections |
| 21:29.54 | _Sam-- | i use SATA w/ linux software raid fine with my digium card and asterisk |
| 21:29.56 | Hmmhesays | the phone doesn't have to encrypt anything |
| 21:30.12 | *** join/#asterisk crich1999 (n=crich@port-212-202-0-102.dynamic.qsc.de) |
| 21:30.17 | _Sam-- | had no problems, drive related, --been running this way for over a year |
| 21:30.20 | *** join/#asterisk pointer (i=pointer@aj.catt.com) |
| 21:30.40 | rene- | Hmmhesays: do you have udp/tcp issues? can you run tunnels tru udp with openvpn? is it easy to go tru firewalls that way? |
| 21:30.54 | Hmmhesays | its cake |
| 21:30.57 | *** join/#asterisk Romik_ (n=romik_@1.fix.netvision.net.il) |
| 21:31.01 | Hmmhesays | no issues |
| 21:31.11 | *** join/#asterisk Qwellj2me (n=Qwell@unaffiliated/qwell) |
| 21:31.13 | *** join/#asterisk SERGEUS|W (n=SERGEUS@ippe-245.ippe.ru) |
| 21:31.16 | Hmmhesays | unless of course your network is crap |
| 21:31.20 | rene- | udp tunnels were tricker with vtun |
| 21:31.25 | *** join/#asterisk agreenberg (n=Alex@64.201.13.50) |
| 21:31.38 | rene- | probably firewall related |
| 21:31.39 | Hmmhesays | it takes about 7 seconds to setup in openvpn |
| 21:32.00 | rene- | i ve seen that is not hard to have a database backend for those tunnels |
| 21:32.10 | rene- | with openvpn |
| 21:33.21 | rene- | have you considered something like free swan? has interoperability with third parties using industry standard vpns been important? |
| 21:33.30 | Hmmhesays | why bother |
| 21:33.46 | Hmmhesays | openvpn is easy to configure and it does way more than I need it to |
| 21:33.59 | rene- | i know there is a gui that could be handy for remote workers |
| 21:34.05 | pointer | what is the status of the function STAT()? it doesn't appear to be in 1.2.3, but it was added to the wiki on Jan 13, 2006 |
| 21:34.15 | _Sam-- | gammacoder: you there? |
| 21:34.18 | *** part/#asterisk Jkx (n=jkx@84.4.64.27) |
| 21:34.19 | rene- | have you done much of that kind of vpn? |
| 21:34.35 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 21:34.47 | *** part/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 21:35.08 | gammacoder | _Sam--: yep |
| 21:35.43 | _Sam-- | gammacoder: i have 6 phone sales guys on the gxp2000s on the new firmware...they are complaining about the "missed calls"...because i use a "ringall" for incoming calls...so their phones all show like 30 missed calls |
| 21:35.55 | _Sam-- | and i cant seem to find a way to turn that off...do you know anything about what im talking about? |
| 21:36.05 | _Sam-- | i want to disable the call history in the phone i guess |
| 21:36.35 | gammacoder | yeah - annoying, I'm sure |
| 21:36.55 | rene- | does anybody know the freenode nick of Howard White from vcch? |
| 21:37.45 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 21:37.47 | gammacoder | i don't know of a solution - I'll keep my eyes out |
| 21:37.47 | Romik_ | hello, somebody can point me how agent can call transfer and continue to hear MOH and wait new call when current call transfered to other agent...? |
| 21:37.47 | _Sam-- | i will put in a feature request |
| 21:38.31 | gammacoder | _Sam--: do you have a good contact at Grandstream? i'd like to get alpha releases from them |
| 21:39.02 | _Sam-- | i do have a contact but he is really tight with releasing firmware |
| 21:39.08 | Hmmhesays | i hate tech support "suddenly it stopped working!" "oh yeah, what changed?" "nothing" "are you sure?" |
| 21:39.18 | _Sam-- | he was upset that 13 became a public type beta |
| 21:39.27 | _Sam-- | and said that he is trying to tighten things up |
| 21:39.40 | Hmmhesays | "oh yeah, this setting didn't change itself", " i meant besides that" |
| 21:39.46 | justinu | lol |
| 21:39.46 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 21:40.21 | gammacoder | gotcha |
| 21:40.24 | Hmmhesays | "that'll be $75 usd, plus a 10% moron tax please" |
| 21:40.44 | justinu | 100% moron tax |
| 21:40.56 | justinu | it's a standard fee, ma'am |
| 21:40.56 | _Sam-- | honestly, i dont think id WANT to use anything alpha from grandstream. |
| 21:41.04 | Hmmhesays | anyone in here using SER currently? |
| 21:41.09 | Hmmhesays | jutinu: no doubt |
| 21:41.16 | gammacoder | ha |
| 21:41.26 | harryvv | Anyone know of a DID reseller of BC canada 604 0r 778 phone numbers? |
| 21:41.29 | Hmmhesays | file[laptop]: question |
| 21:41.33 | file[laptop] | Hmmhesays: answer |
| 21:41.40 | justinu | openser here |
| 21:41.59 | Hmmhesays | i'm looking to do a simple radius auth and then forward, ser or openser |
| 21:42.14 | file[laptop] | auth_radis |
| 21:42.20 | file[laptop] | er |
| 21:42.22 | file[laptop] | auth_radius |
| 21:42.32 | Hmmhesays | auth_radius i know that much |
| 21:43.03 | Hmmhesays | does it matter if I use auth_radius in ser or openser? |
| 21:43.06 | *** join/#asterisk Johnnie (n=jdlewis@dynamic-acs-24-154-53-16.zoominternet.net) |
| 21:43.16 | *** part/#asterisk pointer (i=pointer@aj.catt.com) |
| 21:43.18 | file[laptop] | http://www.iptel.org/ser/doc/ser_radius/ser_radius.html |
| 21:43.21 | file[laptop] | probably work similar |
| 21:43.33 | Hmmhesays | yeah pretty much identical from what I glanced at |
| 21:43.58 | *** join/#asterisk tuxinator_linux (n=tuxinato@70-32-106-248.ontrca.adelphia.net) |
| 21:44.31 | Hmmhesays | ser is documented better |
| 21:47.30 | fugitivo | why developers assume that everybody use mysql? |
| 21:47.33 | fugitivo | they're so wrong |
| 21:48.39 | Nugget | I got an email the other day about flightaware from some PFY who wanted a job or something. "I see you're using Linux, PHP, and MySQL!" he said. |
| 21:48.42 | Nugget | Wrong on three counts. |
| 21:48.51 | sivana | fugitivo: what do you use |
| 21:48.53 | fugitivo | an opensource project that uses only mysql is useless |
| 21:49.03 | _Sam-- | Nugget: i think he meant to send that email to me |
| 21:49.06 | sivana | I use Access with * myself |
| 21:49.07 | fugitivo | sivana: postgresql |
| 21:49.33 | sivana | fugitivo: same here ;) |
| 21:49.43 | Nugget | We took the mail to heart, though, and promptly hired someone to redesign the site so it doesn't look like it was built with PHP, MySQL, and Linux. :) |
| 21:49.55 | fugitivo | Nugget: lol |
| 21:50.05 | fugitivo | Nugget: how did you do it? |
| 21:50.08 | fugitivo | Nugget: using jsp? |
| 21:50.14 | Nugget | PostgreSQL, Rivet, and FreeBSD. |
| 21:50.15 | _Sam-- | ajax |
| 21:50.16 | Hmmhesays | what I wouldn't give to get my hands on quintums firmware code |
| 21:50.19 | *** join/#asterisk mattwj2006 (n=Matt@dialup-4.159.110.252.Dial1.Chicago1.Level3.net) |
| 21:50.27 | fugitivo | ajax is not a language |
| 21:50.35 | wunderkin | what is rivet |
| 21:50.41 | sivana | ; pressed 9999 - voicemail |
| 21:50.41 | sivana | exten => 9999,1,Macro(dial-switch,IAX2/2020software@tor-sw,9999) |
| 21:50.45 | sivana | bah |
| 21:50.53 | sivana | damn right-click |
| 21:51.05 | _Sam-- | Nugget: gee what would give anyone the idea you use php? |
| 21:51.10 | *** join/#asterisk Qwell[laptop] (n=Qwell[]@unaffiliated/qwell) |
| 21:51.12 | *** join/#asterisk droops (n=droops@adsl-065-005-212-128.sip.jan.bellsouth.net) |
| 21:51.12 | _Sam-- | http://slacker.com/legal.php |
| 21:51.30 | _Sam-- | http://slacker.com/people.php |
| 21:51.41 | fugitivo | iuse.php |
| 21:51.42 | Nugget | I wasn't talking about slacker, you futz. |
| 21:51.52 | Nugget | it was about flightaware, as I said. |
| 21:51.58 | _Sam-- | sorry i came in late |
| 21:52.02 | _Sam-- | nice picture of you tho |
| 21:52.13 | fugitivo | Nugget: what is rivet? |
| 21:52.35 | *** join/#asterisk bkw__ (n=brian@70.103.248.130) |
| 21:52.41 | Nugget | http://tcl.apache.org/rivet/ |
| 21:52.49 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 21:53.02 | Libila | fugitivo: define:rivet in google. |
| 21:54.39 | *** join/#asterisk Vyeperman (n=Vye@ip68-8-174-154.sd.sd.cox.net) |
| 21:54.56 | *** join/#asterisk jaike (n=a@203.131.137.76) |
| 21:55.14 | _Sam-- | Nugget: do you supply the flightaware info to airports, or who uses it? |
| 21:55.22 | jaike | anyone using MixMonitor? how do you set it to record in wav49 format |
| 21:55.37 | Nugget | We supply it to a lot of people, and the ad revenue is not insignificant. |
| 21:55.55 | _Sam-- | im just trying to figure out who really needs that info |
| 21:56.13 | Nugget | pilots, their friends family and colleagues, and aviation enthusiasts. |
| 21:56.14 | _Sam-- | like if i wanted to check on my wife's flight...id go the airports webpage, or to the airlines webpage to check status |
| 21:56.22 | Nugget | not everyone flies commercially. |
| 21:56.23 | *** join/#asterisk harryk (n=me@195.245.80.178) |
| 21:56.30 | _Sam-- | trying to figure who needs the flightaware info -- its cool stuff. |
| 21:56.35 | *** join/#asterisk SibRw0rk (n=DaPhrek@66.234.235.84) |
| 21:56.54 | SibRw0rk | hey - what do you guys think the most stable distro is for asterisk to sit on? |
| 21:57.05 | Nugget | there's a huge amount of general aviation, charter, and corporate air traffic that benefits from the site. |
| 21:57.51 | pifiu | can someone point me in the right direction for a program to send emails to customers as a group? |
| 21:57.56 | pifiu | sorry a bit out of topic |
| 21:58.02 | *** join/#asterisk Defraz_ (i=t0tal@72.165.56.43) |
| 21:58.06 | Nugget | pifiu: use FSF's "mailman" app. |
| 21:58.16 | Nugget | http://www.gnu.org/software/mailman/ |
| 21:59.54 | *** join/#asterisk Mike_TK (n=Mike@p1-dialup-178.if.ukrtel.net) |
| 22:00.12 | *** join/#asterisk tartar (n=tartar@CPE0004e27b716e-CM014370001917.cpe.net.cable.rogers.com) |
| 22:00.13 | evilrabbi | pifiu http://mikeduggan.tripod.com/playingcards/hpm04.jpg |
| 22:00.19 | Nugget | SibRw0rk: asterisk doesn't care at all which linux you use. Heck, a lot of people use some other unix. |
| 22:00.25 | Nugget | so pick whatever distro makes you happy |
| 22:00.34 | SibRw0rk | Nugget: yeah i know - my boss seems to think gentoo is the most stable |
| 22:00.39 | SibRw0rk | tho i got it working find with my OS X server |
| 22:00.42 | _Sam-- | Nugget: i know you didnt ask for it, but you get it anyway....your page at flightaware would be more aesthetically pleasing if you moved the top header to the bottom |
| 22:01.07 | _Sam-- | the header that has the date/time, and those links on the right side |
| 22:01.38 | Mike_TK | I think it's bad when boss choosing what distro will be used ;) |
| 22:01.48 | SibRw0rk | haha |
| 22:02.16 | Nugget | yeah, we're completely redoing the site layout. the current one is ugly. |
| 22:02.48 | Lord_Drachenblut | is it possible for an iax client to talk to a sip client threw asterisk |
| 22:02.49 | evilrabbi | I would be using freebsd if i wasn't required to use AMP |
| 22:03.00 | jaike | fedora :) |
| 22:03.02 | evilrabbi | if you wanna use AMP use linux |
| 22:03.14 | _Sam-- | amp sux |
| 22:03.16 | mattwj2006 | asterisk@home :P |
| 22:03.18 | evilrabbi | The people who made AMP thought it was a great idea to hard code everything |
| 22:03.20 | mattwj2006 | lol |
| 22:03.21 | *** part/#asterisk tartar (n=tartar@CPE0004e27b716e-CM014370001917.cpe.net.cable.rogers.com) |
| 22:03.22 | Nugget | http://lnk.nu/slacker.com/q2 |
| 22:03.59 | _Sam-- | bah, thats not old, i should scan my slackware 1.0 |
| 22:04.15 | Nugget | it's the oldest *CD* I have. :) |
| 22:04.16 | *** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-162.sw.biz.rr.com) |
| 22:04.24 | jbalcomb | Woot!! New firmware update for GrandStream GXP-2000. Includes echo cancellation with gain control for speakerphone. |
| 22:04.34 | ctooley | Anyone here use Ekiga/Gnomemeeting with Asterisk? |
| 22:04.38 | _Sam-- | jbalcomb: works nice. |
| 22:04.44 | fugitivo | hell |
| 22:04.46 | Nugget | less interesting to scan a 3.5" floppy that's got "slack AP 4 of 18" written on it. |
| 22:04.49 | _Sam-- | careful on the reboot loop during upgrade...happened to myself and one other from here |
| 22:05.01 | fugitivo | i have to modify all database connections for that ARI interface |
| 22:05.12 | fugitivo | anyone has it working for postgresql? |
| 22:05.19 | jbalcomb | _Sam-- yeah, I'm kind of hoping they fixed that too. :/ |
| 22:05.29 | _Sam-- | they didnt |
| 22:05.32 | jbalcomb | _Sam-- seems not... |
| 22:05.40 | _Sam-- | falle had his phone rebooting over and over for 48 minutes |
| 22:05.47 | _Sam-- | until i told him to move the firmware files |
| 22:05.50 | _Sam-- | so it couldnt get them anymore |
| 22:05.56 | Romik_ | hello, somebody can point me how agent can call transfer and continue to hear MOH and wait new call when current call transfered to other agent...? |
| 22:06.02 | _Sam-- | on the other hand, gammacoder has had no problems with his |
| 22:06.16 | jbalcomb | _Sam-- Hrmm.. I didn't have that, just the lock up on firmware update |
| 22:06.16 | _Sam-- | the firmware seems to be worth it though. |
| 22:06.19 | justinu | nugget: how about a 5.25" HD floppy with "SCO Xenix/286" written on it? |
| 22:06.44 | Nugget | I used to work on that stuff. On altos servers as a base for running mas-90. :) |
| 22:07.10 | jbalcomb | _Sam-- I'm hoping so. I just found out my PRI card is sharing an IRQ with four other devices so I'm hoping to fix that tonight. |
| 22:07.31 | _Sam-- | should be an easy fix |
| 22:08.02 | jaike | anyone know how to set MixMonitor to record in wav49 format? |
| 22:08.08 | bkw__ | es |
| 22:08.09 | bkw__ | yes |
| 22:08.11 | bkw__ | use WAV |
| 22:08.13 | bkw__ | all caps |
| 22:08.16 | bkw__ | thats wav49 |
| 22:08.18 | bkw__ | or wav49 |
| 22:10.15 | Romik_ | bkw: could you please help with agent things problem? |
| 22:10.34 | *** join/#asterisk jeffik (n=Jeff@CPE0050babf4cd5-CM014350000760.cpe.net.cable.rogers.com) |
| 22:12.51 | De_Mon | I read somewhere about using meetme rooms for call queues instead of agents anyone have a link? |
| 22:13.05 | Qwell[laptop] | De_Mon, just use a Local/ channel |
| 22:13.09 | Qwell[laptop] | one that runs a meetme |
| 22:13.18 | *** join/#asterisk deb_user (n=Hypnotis@71-36-56-107.albq.qwest.net) |
| 22:13.27 | bkw__ | Qwell[laptop], where are you? and do you want another drink? |
| 22:13.40 | Qwell[laptop] | bkw__, I'm in the conf, and...you buying? :) |
| 22:13.50 | De_Mon | how would I manage the calls though? I don't remember all the details from, whatever I read |
| 22:13.54 | bkw__ | Qwell[laptop], but we know your limit now |
| 22:13.59 | *** join/#asterisk nettie (i=esivieri@85-18-54-38.ip.fastwebnet.it) |
| 22:14.02 | Qwell[laptop] | bkw__, no... |
| 22:14.10 | Qwell[laptop] | bkw__, You know what is PASSED my limit. |
| 22:14.16 | jaike | thanks bkq |
| 22:14.16 | Qwell[laptop] | the trick is finding the limit |
| 22:14.18 | jaike | thanks bkw |
| 22:14.18 | bkw__ | WAY passed |
| 22:14.22 | Qwell[laptop] | :P |
| 22:14.24 | Qwell[laptop] | ZOACON! |
| 22:14.27 | bkw__ | haha |
| 22:14.38 | Qwell[laptop] | anything this guy says about me...is ALL made up |
| 22:14.45 | bkw__ | NO |
| 22:14.52 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 22:14.53 | Qwell[laptop] | I didn't even go with them on Wednestuesday night, I swear |
| 22:14.53 | bkw__ | I have the astlinux guy to back me up |
| 22:14.59 | Qwell[laptop] | meh, damn fvwm |
| 22:15.05 | bkw__ | Kris will back me up |
| 22:15.12 | Qwell[laptop] | never happened |
| 22:15.16 | sivana | heh |
| 22:15.37 | *** part/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 22:15.52 | jbalcomb | how do you set IRQs for PnP devices in linux? |
| 22:16.07 | jpablo | hey people, what's the sintaxys for the alsa channel ? |
| 22:16.47 | cyburdine | in my initial registration to my voip provider it seems that fromdomain seems to not make any changes to my sip debug output... |
| 22:16.56 | cyburdine | am i missing something obvious... |
| 22:17.06 | cyburdine | i.e. fromdomain = bob.sam.com |
| 22:17.33 | justinu | does your register statement match the name in []'s in sip.conf? |
| 22:17.41 | cyburdine | From <sip:phonenumber@sam.com> |
| 22:18.07 | Qwell[laptop] | alright, I've got about an hour |
| 22:18.10 | Qwell[laptop] | bkw__, You guys in the lobby? |
| 22:18.18 | justinu | if your peer is in [hello] |
| 22:18.33 | cyburdine | no... just changed it to match |
| 22:18.38 | justinu | you want to register => blah@hello |
| 22:19.29 | cyburdine | oh man... I can't believe I overlooked that! |
| 22:19.34 | Cresl1n | hey, does anybody have chan_misdn working? |
| 22:19.41 | cyburdine | damn... sorry about that... and Thanks! |
| 22:20.24 | justinu | cyburdine: np, it's kinda not-so-obvious |
| 22:20.48 | cyburdine | yeah... but I should have double checked that.. |
| 22:20.51 | *** join/#asterisk Grubs (n=Miranda@c211-28-119-169.eburwd3.vic.optusnet.com.au) |
| 22:21.52 | *** join/#asterisk Powerkill (n=PoWeRKiL@84.205.154.241) |
| 22:23.49 | bkw__ | Qwell[laptop], next to registration desk |
| 22:23.56 | *** join/#asterisk sandra78 (n=sandra@200.106.40.54) |
| 22:23.59 | Qwell[laptop] | brt |
| 22:24.29 | sandra78 | hi does anyone knows where to get a sip softphone open source? i would like to put my logo skin etc |
| 22:25.02 | De_Mon | does chan_agent still misbehave in some cases? http://www.voip-info.org/wiki/view/Agents+without+agent+channel |
| 22:25.17 | cyburdine | justinu: I think I figured out why I was confused... the docs on voip-info.org seem to be really missleading... |
| 22:25.27 | justinu | yep, they are pretty nebulous sometimes |
| 22:26.23 | cyburdine | in the definition of host it says that each sip server needs a definition in a secion of it's own in SIP.conf (mysipprovider.com) |
| 22:26.33 | tdonahue | i read about a utility that can recreate an rtp stream from a pcap a while back, anyone know what it is? |
| 22:26.37 | cyburdine | yet in the example secion it's mysipprovider-out |
| 22:26.39 | *** join/#asterisk sherbang (n=sherbang@c-71-192-235-90.hsd1.ma.comcast.net) |
| 22:26.46 | cyburdine | hrm... |
| 22:27.00 | *** join/#asterisk CoKane (n=CoKane@85.134.137.104) |
| 22:27.06 | cyburdine | anyone have edit privs on that page? |
| 22:27.08 | justinu | tdonahue: ethereal should be able to do that. |
| 22:27.11 | *** join/#asterisk BladeRunner05 (n=feelme@adsl-172-213.38-151.net24.it) |
| 22:27.19 | CoKane | hi there |
| 22:27.23 | CoKane | anyone tell what is the button combination to hit when you want to start on demand recording in A@H |
| 22:27.47 | *** join/#asterisk adeel (n=adeel@HSE-Montreal-ppp333357.sympatico.ca) |
| 22:28.00 | adeel | hi |
| 22:28.01 | *** join/#asterisk Martincit0 (n=martin@litigaractivos1.att.net.co) |
| 22:28.02 | tdonahue | justinu: ethereal doesn't convert it back to audio, but it does do a fairly nice statistical analysis of it... i'm looking for the audio though |
| 22:28.07 | *** part/#asterisk loic1 (n=loick@APuteaux-151-1-56-32.w82-120.abo.wanadoo.fr) |
| 22:29.13 | adeel | can some one guide me on this ... I was using the system command to check if a file exists now it no longer works in ast 1.2. what can i do |
| 22:29.33 | CoKane | anyone |
| 22:30.03 | justinu | tdonahue: rtpscan |
| 22:30.41 | fugitivo | CoKane: /etc/asterisk/features.conf |
| 22:30.44 | Martincit0 | hello all |
| 22:30.46 | fugitivo | CoKane: search "automon" |
| 22:31.12 | adeel | System(...) doest not work any more to see the existance of a file, does any one know a solution |
| 22:31.23 | tdonahue | justinu: thats the one :-) thanks |
| 22:31.26 | justinu | np |
| 22:31.54 | CoKane | fugitivo: whats the button press for automon is it *1 |
| 22:32.13 | fugitivo | CoKane: did you look features.conf? |
| 22:32.21 | CoKane | yes |
| 22:32.25 | CoKane | and thats what it has |
| 22:32.29 | CoKane | *1 |
| 22:32.31 | fugitivo | then? why are you asking me again? |
| 22:32.47 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 22:33.54 | *** join/#asterisk Qwell[\0] (n=Qwell[]@70.103.248.130) |
| 22:34.00 | Feral_Kid | Still looking for a help on getting a iconnecthere trunk working... According to the console, it says that the trunk is registered, but when I dial out, I don't see any response on the console, and I get a congestion... |
| 22:34.41 | adeel | What is automon, is this an automation thing for asterisk |
| 22:34.45 | *** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at) |
| 22:34.50 | fugitivo | auto-monitor |
| 22:34.58 | adeel | ok |
| 22:35.06 | fugitivo | record ondemand |
| 22:35.14 | adeel | ok cool |
| 22:36.33 | adeel | does any one know how to see if a file exists or not in asterisk ... thx |
| 22:37.02 | *** join/#asterisk elephantMan (n=elephant@252.205.103-84.rev.gaoland.net) |
| 22:38.32 | *** join/#asterisk mzo (n=moz@ool-435193b3.dyn.optonline.net) |
| 22:38.44 | *** part/#asterisk mkrufky (n=mk@68.160.103.77) |
| 22:38.47 | mzo | does anyone have a faq for upgrading from 1.2.2 to 1.2.3 |
| 22:39.13 | *** join/#asterisk Charlie (n=Charlie@HSE-Montreal-ppp333357.sympatico.ca) |
| 22:39.23 | Charlie | hi guys |
| 22:40.02 | *** join/#asterisk stack_ (n=stack@63.239.190.202) |
| 22:40.55 | Charlie | can some one tell how do i change by text colour |
| 22:41.07 | stack_ | What I would like to do is have a Queue that, once it has reached the end of the queue, it drops out of the queue and goes to the next priority. I don't see any way to do this except for setting a timeout at the end of the Queue command. Is there another way to do this? |
| 22:41.32 | *** part/#asterisk mattwj2006 (n=Matt@dialup-4.159.110.252.Dial1.Chicago1.Level3.net) |
| 22:41.44 | Charlie | is the asterisk help line |
| 22:41.56 | Charlie | i have a small trouble |
| 22:42.31 | Charlie | is this port 6667 |
| 22:42.46 | tuxinator_linux | Charlie: You're in the right place |
| 22:43.00 | tuxinator_linux | Charlie: I don't think people understand your question. |
| 22:43.09 | Charlie | cool i finely reached |
| 22:43.32 | *** join/#asterisk Ferrari (n=Ferrari_@rrcs-24-123-225-162.central.biz.rr.com) |
| 22:43.36 | Charlie | ok, well i guess i have to rephrase |
| 22:43.41 | Ferrari | good day all |
| 22:43.56 | Ferrari | having a slight issue with a pri channel getting hung |
| 22:44.06 | Martincit0 | Frequently yellow alarms are triggered for all channels of my Digium TE110P, which is connected to the local PSTN. However it seems to be working fine. Any ideas on how to get rid of these warnings? |
| 22:44.12 | Ferrari | if i do a zap show channels channel 49 shows an inbound call |
| 22:44.23 | Ferrari | however if i do show channels there is nothing there |
| 22:44.33 | Ferrari | anyway to reset that channel |
| 22:45.08 | Hmmhesays | ok it didn't build my auth_radius module |
| 22:47.00 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 22:47.03 | Charlie | well, what i am tring to do is to see if a sound file exists so asterisk can play it, in order to do that i use the asterisk sound command like this system(/var/lib/asterisk/system/fexists.sh /mnt/d2/prompts/${file}.gsm) now this use to work but no longer works in asterisk 1.2+, so my question does any one know a solution to this or a workaround. Thanks |
| 22:47.22 | Seldon1975 | can someone please verify that this is the correct command to check out asterisk 1.2.3 source: |
| 22:47.23 | Seldon1975 | svn co http://svn.digium.com/svn/asterisk/tags/1.2.3 asterisk1.2.3 |
| 22:47.25 | *** part/#asterisk Romik_ (n=romik_@1.fix.netvision.net.il) |
| 22:47.52 | Martincit0 | Messages are: chan_zap.c:6468 handle_init_event: Detected alarm on channel 1: Yellow Alarm |
| 22:47.52 | Martincit0 | chan_zap.c:1442 zt_disable_ec: Unable to disable echo cancellation on channel 1 |
| 22:48.23 | justinu | yellow alarm means that the far end can't receive your TX signal properly |
| 22:48.43 | Martincit0 | Seldon: svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk |
| 22:48.45 | SkramX | How many calls could 80MB ram handle, do you think? |
| 22:49.07 | Seldon1975 | Martincit0: thank you |
| 22:49.10 | sevrdseot | I there such a thing as a WiMax VoIP hardphone? |
| 22:49.12 | SkramX | ? |
| 22:49.14 | sivana | anyone here good with macros? |
| 22:49.14 | mogorman | ram is not limiting factor SkramX it will be cpu |
| 22:49.16 | Charlie | 2 calls |
| 22:49.18 | _Sam-- | sevrdseot: yes |
| 22:49.24 | SkramX | mogorman: hrmm |
| 22:49.26 | Powerkill | someone is familiar with rtpproxy ? |
| 22:49.29 | Seldon1975 | Martincit0: that seems odd though; previously I've had to specify a version number |
| 22:49.34 | sevrdseot | _Sam--: Do you happen to know of any names off hand? |
| 22:49.41 | _Sam-- | utstarcomm f-1000 |
| 22:49.45 | mogorman | 2 calls wtf are you smoking Charlie |
| 22:49.51 | _Sam-- | i think cisco has a wifi phone |
| 22:49.56 | _Sam-- | i think zyxel has a wifi phone |
| 22:50.00 | Charlie | good quality so you can understand |
| 22:50.01 | sevrdseot | _Sam--: have you used this phone? |
| 22:50.11 | _Sam-- | i have 2 utstarcomm f1000s |
| 22:50.17 | Charlie | i had 320 ram and i was hoaving trouble with 10 calls |
| 22:50.33 | sevrdseot | _Sam--: And it's WiMax? |
| 22:50.35 | mzo | i have a zyxel wifi phone, it's crap, save your money. :P |
| 22:50.38 | Charlie | actually my processor was 500 mhz |
| 22:50.49 | _Sam-- | no, this is just wifi |
| 22:50.54 | mzo | is asterisk weird with SMP? |
| 22:51.05 | sevrdseot | right, my question was if anyone had heard of a WiMax hardphone |
| 22:51.33 | Charlie | try to abstain from lower quality phones |
| 22:51.50 | stack_ | I'd like my dialplan to leave a queue after everyone in the queue has been dialed once, is there a way to do this? |
| 22:51.52 | mzo | wts pos zyxel, pst. :P |
| 22:52.27 | _Sam-- | sevrdseot: sorry for the confusion..i dont know of any wimax phones...and i dont of anyone who uses wimax |
| 22:52.30 | _Sam-- | at least, not yet |
| 22:52.38 | cyburdine | justinu: thanks again for the help... it seems like it's progressing through the registration further than before |
| 22:52.49 | [av]bani | ugh, polycom webconfig is horrible |
| 22:52.57 | justinu | cyburdine: cool |
| 22:52.59 | [av]bani | grandstream config: 15 seconds polycom: 15 minutes |
| 22:53.01 | cyburdine | I am getting a bizarre Scheduling destruction of call notice though do I need to worry about that |
| 22:53.06 | _Sam-- | [av]bani: did you upgrade your grandstream yet? |
| 22:53.09 | [av]bani | yep |
| 22:53.10 | justinu | [av]bani: give up on the polyocm web config, and just setup the boot server |
| 22:53.16 | Charlie | gentek webconfig is ok |
| 22:53.16 | _Sam-- | no problems with the reboot loop? |
| 22:53.20 | Dr-Linux | question, asterisk what APP communicate on *SOCKET Base* with any external language i.e. java/db/etc ? |
| 22:53.21 | *** join/#asterisk SGM (n=stoyan@home.marinov.us) |
| 22:53.23 | EriSan | hi all, are the cisco ata 183 good devices ? |
| 22:53.25 | mzo | zyxel webconfig is awful, you have to reboot the phone after EACH setting change. |
| 22:53.27 | [av]bani | yeah, it keeps redownloading the same firmware |
| 22:53.33 | [av]bani | they obviously didnt add a proper check |
| 22:53.43 | _Sam-- | makes me feel better, i knew i wasnt that crazy. |
| 22:53.49 | _Sam-- | all you have to do is move the firmware files |
| 22:53.52 | [av]bani | nice features though |
| 22:53.53 | sevrdseot | _Sam--: I'm keeping my eyes peeled for a WiMax phone. I've been searching and I can't even find reports of a company thinking about attempting to make one. |
| 22:53.57 | Charlie | i want to know i using a scsi drive will improve asterisk perfomance |
| 22:54.00 | _Sam-- | so it cant get them |
| 22:54.07 | [av]bani | yeah, thats what i did |
| 22:54.26 | [av]bani | i think they want you to make a http provision to clear out the upgrade url on it |
| 22:54.29 | [av]bani | or something |
| 22:54.36 | *** join/#asterisk |omni| (n=rob@net98.limelyte.net) |
| 22:54.38 | _Sam-- | i think this new firmware may really be good |
| 22:54.42 | _Sam-- | heard 0 echo so far |
| 22:54.49 | _Sam-- | and people ive called said i sound better |
| 22:54.54 | mzo | which phone firmware? |
| 22:55.00 | _Sam-- | gxp2000 beta firmware |
| 22:55.05 | mzo | oh, a good phone :P |
| 22:55.22 | _Sam-- | cheap and good are noy synonomous in this case :) |
| 22:55.26 | _Sam-- | at least , not according to some. |
| 22:55.38 | cyburdine | justinu: do I need to worry about the "Scheduling destruction of call" message I get? |
| 22:55.40 | _Sam-- | its a cheap phone that works pretty good for a cheap phone |
| 22:55.46 | _Sam-- | and i think, it works even better now |
| 22:55.53 | mzo | I have two grandstream 100s and they are crap. :P I don't mind too much, now if I could just get my server to stop blowing up :P |
| 22:55.58 | justinu | cyburdine: no, it's normal. |
| 22:56.15 | cyburdine | cool... |
| 22:56.23 | sevrdseot | _Sam--: How much experience do you have with WiFi hardphones? Can you recommend a nice phone that would support WPA, 802.11g and SIP? |
| 22:56.24 | Charlie | how is it crap |
| 22:56.40 | justinu | sevrdseot: i don't think there is such a thing yet. |
| 22:56.46 | *** join/#asterisk franck (n=franck@tikiwiki/franck) |
| 22:57.00 | mzo | the grandstream? just cheezy, really. Setting ringtones is a nightmare, and lots of silly little things that i forgot about, because i set them up months ago. These are the old crappy ones. |
| 22:57.06 | sevrdseot | I've found phones that do SIP and 802.11g, but so far encryption on these phones are weak as hell. |
| 22:57.25 | mzo | the zyxel does WEP64 or WEP128 bit but at the cost of massive latency. |
| 22:57.30 | _Sam-- | serv: i only have experience with the utstarcom F1000 phone...and that is another cheap phone, that works pretty good for a cheap phone...im not sure about WPA (i use mac filtering and WEP) |
| 22:57.51 | *** join/#asterisk MikeJ__ (n=vircuser@70.103.248.130) |
| 22:58.09 | sevrdseot | Wireless security saddens me, WEP is easily broken and MAC address can be spoofed simply. |
| 22:58.40 | _Sam-- | im not that worried about someone listening to my wife's conversations....and my neighbors are too far to get on my wirelss network |
| 22:58.47 | sevrdseot | heh. |
| 22:58.58 | _Sam-- | and i dont they would know how to crack a WEP key even if i gave them a hint :) |
| 22:58.59 | mzo | i only get bad phone sex from 900 numbers, and now i can't even do that. Let's hope 1.2.3 fixes this! |
| 22:59.07 | cpm | well, there was pgp phone, but no one ever used it. |
| 22:59.29 | cpm | you could put the sip call through ssh tunnel |
| 22:59.43 | *** join/#asterisk johnnyb (n=jonathan@adsl-38-9-196.tulsaconnect.com) |
| 23:00.30 | johnnyb | Is anyone using Sipura SPA-841 or Zultys Zip2 phones? Can anyone comment how they compare to each other or to the Grandstream Budgetones? |
| 23:00.45 | *** join/#asterisk QuAd|Haudrauf (n=hau@port-212-202-185-60.dynamic.qsc.de) |
| 23:00.50 | sevrdseot | cpm: there was a PGP phone? That's awesome! How would one go about tunneling sip calls over ssh? Can you also do SSL certs? |
| 23:00.52 | johnnyb | Or the ArtDio phones as well. |
| 23:00.56 | [av]bani | _Sam--: for $80, gxp2000 is an awesome phone, considering the garbage in the same price class |
| 23:01.11 | [av]bani | i dont think anything else comes close for $80 |
| 23:01.12 | sherbang | The SPA-841 is much nicer then the Budgetones. |
| 23:01.18 | _Sam-- | [av]bani: ive taken that position for a while...and taken some licks on this channel for it |
| 23:01.25 | _Sam-- | thats why i was happy to hear when you got some other phones today |
| 23:01.28 | johnnyb | sherbang: what's the difference? |
| 23:01.32 | _Sam-- | because now you can really tell me how the gxp stacks up |
| 23:01.34 | [av]bani | _Sam--: the phone just works, which is nice. no frills |
| 23:01.40 | sherbang | but transferring calls is difficult on the SPA |
| 23:01.45 | mzo | Connected to Asterisk 1.2.3 currently running on asterisk1 (pid = 3045) <--YAY :) |
| 23:01.55 | [av]bani | _Sam--: and config is very smple, unlike polycom... |
| 23:02.02 | johnnyb | sherbang: what's difficult about it? |
| 23:02.10 | sherbang | too many button presses. |
| 23:02.12 | [av]bani | oh, we also found out... nat doesnt work for SHIT with polycom |
| 23:02.14 | [av]bani | period |
| 23:02.14 | QuAd|Haudrauf | it's a pitty that it's common to let products mature at the client's desk |
| 23:02.15 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 23:02.51 | johnnyb | sherbang: so what's better about it? |
| 23:03.21 | johnnyb | [av]bani: how does the gxp2000 do with speakerphone? |
| 23:03.31 | [av]bani | johnnyb: totally crap |
| 23:03.37 | [av]bani | no echo, but its too quiet |
| 23:03.45 | _Sam-- | it does intercom mode now? |
| 23:03.48 | [av]bani | it always did |
| 23:03.57 | sevrdseot | Finding a good phone is hard business. |
| 23:03.59 | _Sam-- | i thought with a workaround |
| 23:04.06 | _Sam-- | but now it does it from the sip header? |
| 23:04.11 | sherbang | johnnyb: well, the budgetone works like a 1-line phone with call waiting. The SPA has multiple line lights and such. Feels more like an office phone. |
| 23:04.11 | johnnyb | How does the Zip2 phone work? |
| 23:04.15 | [av]bani | well, i take that back |
| 23:04.23 | [av]bani | speakerphone isnt too bad now, on max volume |
| 23:04.41 | [av]bani | _Sam--: answer-after:0 in the sip header |
| 23:04.47 | [av]bani | forces the gxp to pick the extension up |
| 23:05.04 | _Sam-- | i also noticed some new phone-specific settings in the firmware related to that |
| 23:05.04 | johnnyb | Where can you get one for $80? |
| 23:05.14 | _Sam-- | johnnyb: voipstore.atacomm.com is like 84 |
| 23:05.19 | [av]bani | https://www.discountvoipoutlet.com/store/product.php?productid=2&cat=1&page=1 |
| 23:05.42 | sevrdseot | Ask the guy a question and he /quits |
| 23:05.45 | [av]bani | 'Grandstream GXP-2000 features market leading superb sound quality' hahahhahaha |
| 23:05.48 | [av]bani | that it does not |
| 23:05.52 | _Sam-- | thats the cheapest place ive seen so far |
| 23:05.53 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 23:05.53 | [av]bani | it is just average |
| 23:06.00 | _Sam-- | discountvoipoutlet, i mean |
| 23:06.22 | QuAd|Haudrauf | w00t, 80$, pitty export, i only could get one for 95 euro (germany) and that was the lowest price available :) |
| 23:06.32 | mzo | what's better than that if you have say $150 to spend? |
| 23:06.32 | jaike | does mixmonitor still use soxmix? |
| 23:06.34 | [av]bani | this new gxp firmware is pretty nice |
| 23:06.48 | [av]bani | all i could possibly ask for over this is http/sendtext |
| 23:06.49 | QuAd|Haudrauf | grandstream made a big leap forward with it today |
| 23:06.58 | *** part/#asterisk Skkip (n=Skipper@216.160.91.91) |
| 23:06.59 | *** join/#asterisk Husk (n=tcg@202.55.153.169) |
| 23:07.01 | jaike | polycoms |
| 23:07.08 | [av]bani | if gs would do that, it would be unbelievably awesome |
| 23:07.16 | [av]bani | but they added http provisioning, which is pretty cool |
| 23:07.22 | mzo | which polycom? |
| 23:07.25 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.252) |
| 23:07.28 | Aughey | I've hears the snom 330 is good |
| 23:07.37 | Aughey | er, 320 |
| 23:07.48 | jaike | 501s |
| 23:09.03 | _Sam-- | why would you want to spend more money, if you could spend 80 bucks and get a phone that does everything you need? |
| 23:09.22 | mzo | i want a pretty spiffy shiny phone. :P not cheap plastic |
| 23:09.23 | sevrdseot | _Sam--: I missed it, what phone are you talking about again? |
| 23:09.32 | _Sam-- | fair enough! |
| 23:09.34 | QuAd|Haudrauf | btw, has anyone managed to upload a ring3.bin to his gxp2000? mine is 23kb big, so it should work - but the syslog always says: ERROR GET ring3.bin .. 1.0.1.12 had the same problems |
| 23:09.37 | jaike | cisco! |
| 23:09.55 | tuxinator_linux | jaike: cisco what? |
| 23:10.15 | jaike | nah..too expensive |
| 23:10.26 | mzo | does anyone know what format i have to have the ringtones for the budgetones? i've looked online and never foud a clear pic of what they should be |
| 23:10.39 | QuAd|Haudrauf | the tv-series 24 uses cisco-phones in their headquarters :) |
| 23:10.41 | _Sam-- | mzo: alot of people here say the SPA 941 is a nice phone compared to the grandstream gxp2000 |
| 23:10.50 | tuxinator_linux | QuAd|Haudrauf: I noticed that also |
| 23:10.54 | mzo | how much are ciscos phones anyway? |
| 23:11.03 | tuxinator_linux | QuAd|Haudrauf: but is that a cisco ring you hear? |
| 23:11.10 | QuAd|Haudrauf | they're expensively aimed for corporate use.. puh :) |
| 23:11.13 | jaike | $300+ and up |
| 23:11.20 | mzo | worth it though? |
| 23:11.22 | QuAd|Haudrauf | tuxinator_linux, i have no idea. :) |
| 23:11.29 | mzo | i mean do they sound that clear and no echo and feel solid? |
| 23:11.36 | tuxinator_linux | QuAd|Haudrauf: I don't think it is |
| 23:11.45 | QuAd|Haudrauf | i got me the ringtone, so i could install it to the gxp2000, haven't managed it |
| 23:11.59 | [av]bani | man, the default ringtone on the polycom 601 is real wussy |
| 23:12.17 | [av]bani | bleedle bleedle bleedle, sounds like mr. rogers neighborhood |
| 23:12.29 | mzo | i hate the default ringtones on my budgetones, it annoys me |
| 23:12.34 | sivana | are variables created using Set() global or local? |
| 23:12.34 | QuAd|Haudrauf | mzo, they have some funny xml applications stuff, etc.. very special |
| 23:12.37 | [av]bani | the gxp ringtone is ANSWER ME GOD DAMMIT |
| 23:12.48 | *** join/#asterisk CaT[tm] (n=cat@nessie.weebeastie.net) |
| 23:13.00 | sevrdseot | i like classic ring tones, ones people are familar with |
| 23:13.13 | CaT[tm] | heylahey. is there a good debian asterisk repository available? |
| 23:13.23 | _Sam-- | debian asterisk repository? |
| 23:13.25 | justinu | i followed fenders lead and put the cisco 24 ring tone on my polycoms |
| 23:13.26 | _Sam-- | what would it containt? |
| 23:13.28 | QuAd|Haudrauf | and then, after you get the phone up, you say: "operator ..?" |
| 23:13.29 | _Sam-- | contain |
| 23:13.34 | sevrdseot | i kind of hate custom ring tones and all the shitastic ring tones new sip phones come out with |
| 23:13.55 | _Sam-- | CaT[tm]: if you want an apt source that has a good asterisk version, i can help....i have an apt source with 1.2.1...i think most stable sources still use 1.0.9 |
| 23:13.56 | mzo | where do you get a nice ringtone? im still trying to get a niice simple one. |
| 23:13.59 | tzanger | sevrdseot: you and me both |
| 23:14.00 | *** join/#asterisk junbug (n=junya@c-66-176-211-109.hsd1.fl.comcast.net) |
| 23:14.01 | jaike | the latest debian asterisk package is only 1.0.7 methinks |
| 23:14.06 | tzanger | I want a nice normal OFFICE ringer selection |
| 23:14.11 | _Sam-- | nah i have 1.2.1 debian package |
| 23:14.13 | mzo | i compiled from source. I feel so proud! :P |
| 23:14.14 | _Sam-- | need right source |
| 23:14.22 | tuxinator_linux | mzo, good for you |
| 23:14.32 | jaike | mzo: nothing like source |
| 23:14.34 | *** join/#asterisk sdgusler-M33 (i=WebChat@i.think.napoleon.dynamiteblows.com) |
| 23:14.35 | mzo | i've never compiled it before, i was expecting demons to appear and smite me :P |
| 23:14.35 | tzanger | not "fur elise", not the latest pop tune... and worst of all NOT some gay-ass impersonation of a bell ringer |
| 23:14.47 | sdgusler-M33 | what'de cause audio in one direction to breakup? |
| 23:14.59 | mzo | 'you are unworthy to compile this source, try again at level 60!' *death* |
| 23:15.00 | sdgusler-M33 | on my voip i can hear the person fine but they hear breakups really back |
| 23:15.01 | sdgusler-M33 | bad |
| 23:15.03 | QuAd|Haudrauf | ;} |
| 23:15.04 | _Sam-- | Package: asterisk |
| 23:15.04 | _Sam-- | Versions: |
| 23:15.04 | _Sam-- | 1:1.2.1.dfsg-3(/var/lib/apt/lists/ftp.de.debian.org_debian_dists_unstable_main_binary-i386_Packages) |
| 23:15.11 | QuAd|Haudrauf | 60? .. that's so wow'ish |
| 23:15.14 | CaT[tm] | sam: I'd prefer something I can use to keep uptodate with if possible. |
| 23:15.18 | sdgusler-M33 | all i did was switch over from using a cisco gateway for pri to a te405p card |
| 23:15.26 | mzo | well, yeah, if I played EQ I'd still be levelling before i ever get to install asterisk :P |
| 23:15.53 | QuAd|Haudrauf | epic asterisk 1.2.3 found |
| 23:15.58 | _Sam-- | Cazper: if you're that worried about keeping up to date why dont you compile it yourself |
| 23:16.03 | _Sam-- | and then just keep getting the latest svn |
| 23:16.15 | _Sam-- | er not Cazper, Cat |
| 23:16.33 | mzo | haha, it's PURPLE! |
| 23:16.50 | jaike | am using 1.2.1 now...still scared to go 1.2.3 |
| 23:16.56 | sdgusler-M33 | tzafrir_home? |
| 23:17.01 | QuAd|Haudrauf | https://www.discountvoipoutlet.com/store/product.php?productid=2&cat=1&page=1 <- that site thinks gxp2000 is fond of encryption? |
| 23:17.02 | mutilator | any ideas? |
| 23:17.07 | mzo | i upgraded from 1.21 to 1.2.3 no problems. :P |
| 23:17.21 | _Sam-- | i am on 1.2.1 but have found no reason to upgrade |
| 23:17.29 | QuAd|Haudrauf | AES based encryption |
| 23:17.33 | mzo | my asterisk blew up last night, so i had to upgrade. :P |
| 23:18.17 | Drew_____ | quad - maybe a future firmware release? |
| 23:18.21 | QuAd|Haudrauf | hm, does the phone need enum support to work or is it just the provider? |
| 23:18.30 | junbug | it is poss to dial in cmd line right? |
| 23:18.32 | sherbang | How does the gxp2000 compare to the Snom320? |
| 23:18.46 | Dr-Linux | what is API ? its manager interface, API stand for? :S |
| 23:18.47 | QuAd|Haudrauf | Drew, possible.. still it's a lie written there. :) |
| 23:19.03 | mzo | my dream is to make a 'fuckyou' phone list, and if the callerid of the caller matches it'll like hang up on them, or not pick up or go to a voicemail hell where there is noescape, but i never figured out how to do it |
| 23:19.04 | fdask | is there a way to change the audio codec used in asterisk? |
| 23:19.13 | fdask | or would that be something i should do in my card specific driver |
| 23:19.17 | _Sam-- | mzo: that should take about 20 minutes, max. |
| 23:19.20 | *** join/#asterisk kll (i=kll@insomnia.juniks.net) |
| 23:19.22 | _Sam-- | if you wanted to use an sql table to it |
| 23:19.22 | mzo | teach me! |
| 23:19.30 | mzo | or ithere a faq? |
| 23:19.31 | _Sam-- | if you wanted to do it in your extensions.conf it should take about 2 |
| 23:19.37 | mzo | i have so many numbers i wanted to block forever! |
| 23:19.49 | QuAd|Haudrauf | mzo, your ex-girlfriend? |
| 23:19.52 | mzo | 'the party you have aclled is not taking your calls ever again' |
| 23:19.54 | mzo | no, citibank |
| 23:19.57 | mzo | and the local PBA |
| 23:19.59 | QuAd|Haudrauf | that's similar |
| 23:20.04 | QuAd|Haudrauf | ;} |
| 23:20.33 | Seldon1975 | i just checked out revision 8756 from SVN - is that the best (1.2.3)? |
| 23:20.43 | Seldon1975 | of asterisk source |
| 23:20.46 | Charlie | Asterisk 1.2.3 Released (If you are running 1.2.2, this is a critical update) get it now. |
| 23:20.58 | drumkilla | mzo: look at the LookupBlacklist application |
| 23:21.09 | *** join/#asterisk Mr-packet (n=a@202-0-61-91.adsl.paradise.net.nz) |
| 23:21.11 | mzo | oooh ty |
| 23:21.16 | Drew_____ | lol @ citibank |
| 23:21.45 | Mr-packet | i'm looking for a steer in the right direction.. any suggestion on phone handsets to use with asterix? |
| 23:22.03 | Charlie | Asterisk 1.2.3 Released (If you are running 1.2.2, this is a critical update) get it now. |
| 23:22.06 | tuxinator_linux | Mr-packet: what is your application and budget? |
| 23:22.08 | mzo | that would work fine on regular phone numbers, too? :P |
| 23:22.37 | Drew_____ | quad - i just asked a friend who has some insight into the grandstream firmware betas - he says AES is very far fetched |
| 23:22.49 | Mr-packet | tux: small office enviroment about 20 phones, 6 external lines |
| 23:22.59 | Mr-packet | budget is moderate, but not tiny |
| 23:23.08 | mutilator | so anyone know what'de cause audio on the pstn side to be really choppy and not on the voip side? |
| 23:23.08 | Charlie | Does any one know how to check for a existance of a sound file before playing it |
| 23:23.09 | QuAd|Haudrauf | Drew, hehe :) thought so. |
| 23:23.21 | _Sam-- | Mr-packet: get some gxp2000s with the new firmware! |
| 23:23.25 | tuxinator_linux | Mr-packet: well, if it where me, I would go with polycom |
| 23:23.27 | _Sam-- | seems to be all the rage, at least, today. |
| 23:23.39 | Mr-packet | who makes the gxp2000? |
| 23:23.44 | _Sam-- | grandstream |
| 23:23.44 | QuAd|Haudrauf | snom ought to be nice phones, linux on them.. |
| 23:24.07 | QuAd|Haudrauf | (which doesn't mean much but ...) |
| 23:24.07 | mzo | weird my asterisk info thing doesn't work on the website anymore, at least it still answers phone. |
| 23:24.12 | *** join/#asterisk cpm (n=Chip@orthanc.avwashington.com) |
| 23:24.16 | Drew_____ | hack the phones! ;-) |
| 23:24.22 | *** join/#asterisk fiber0pti (n=John@invinine.com) |
| 23:24.27 | Mr-packet | do the cisco phones work with asterix? |
| 23:24.37 | QuAd|Haudrauf | Mr-packet, they work with obelix. |
| 23:24.39 | *** join/#asterisk adeel_ (n=Charlie@HSE-Montreal-ppp333357.sympatico.ca) |
| 23:24.39 | fiber0pti | Anyone use Asterisk@home? Any downfalls to configuring it yourself? |
| 23:24.43 | sherbang | Mr-packet: I'm in a similar environment, I've been using the snom phones lately. The Budgettone and the SPA-841 are too cheap, and the Polycom was a pain to configure. |
| 23:24.45 | junbug | it is poss to dial in cmd line right? |
| 23:25.09 | _Sam-- | ive setup 3 medium size offices with the gxp2000 phones and nobody has complained |
| 23:25.11 | QuAd|Haudrauf | Mr-packet, yes, they work with asterisk. :) |
| 23:25.15 | Mr-packet | i need it to work pretty well |
| 23:25.18 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 23:25.28 | _Sam-- | 1 has 15 phones, one has 13, and one has 11 |
| 23:25.29 | Mr-packet | so i'm not lookign to super cheap it |
| 23:25.36 | junbug | fiber0pti: u learn it better doing it yourself |
| 23:25.41 | jaike | does mixmonitor still use soxmix? or is it native |
| 23:25.44 | adeel_ | Does any one know how to check for a existance of a sound file before playing it in the dialplan |
| 23:25.53 | mzo | fiber0pti, you have to update to 1.2.3 :P |
| 23:25.56 | *** join/#asterisk lancey (i=Shady@support.net1.cc) |
| 23:25.59 | lancey | hi all |
| 23:26.00 | fiber0pti | heh.. |
| 23:26.11 | lancey | what's one supposed to replace ${TIMESTAMP} with in 1.2.2? |
| 23:26.13 | mzo | fiber0pti http://sourceforge.net/forum/forum.php?thread_id=1428398&forum_id=420324 :P |
| 23:26.15 | lancey | or 1.2.3, whatever :) |
| 23:26.18 | Mr-packet | we are about to move offices in about 3 months, and we have an old PABX that i can't be bothered moving, so it seems like a good time to do a VoIP solution |
| 23:26.21 | [TK]D-Fender | sherbang : The Polycom's do have a bit of a learning curve, but the accelleration grows as does your control over them. Makes for a very solid experience. |
| 23:26.22 | sherbang | _Sam--: No-one has complained? Seems someone complains about something no-matter what you install :) |
| 23:26.27 | Drew_____ | mr packet - according to nugget at http://slacker.com/~nugget/asterisk8.php they seam to work |
| 23:26.30 | Mr-packet | i just can't bring myself to paying cisco for call manager again |
| 23:26.42 | _Sam-- | only complaints have been network related, not phone related |
| 23:27.11 | adeel_ | I think there should be an asterisk hardware section this is too crazy |
| 23:27.13 | Mr-packet | and what do you guys use for connecting to ptsn? |
| 23:27.25 | QuAd|Haudrauf | cisco doesn't natively package a powersupply, they already think poe 802.3af is the standard in corporate environments |
| 23:27.28 | adeel_ | there are a million phones out there are we going to talk about each one |
| 23:27.40 | Mr-packet | Quad.. i'll be using a poe switch |
| 23:27.46 | Mr-packet | they are cheap enough.. |
| 23:27.48 | QuAd|Haudrauf | then it's alright |
| 23:27.49 | Drew_____ | packet - i use a SIP account at a local provider |
| 23:27.51 | adeel_ | just get some phones and try it out |
| 23:27.55 | adeel_ | you wont louse |
| 23:28.00 | [TK]D-Fender | QuAd|Haudrauf : Those of course that support 802.3af and not just their proprietary PoE |
| 23:28.05 | *** join/#asterisk silistra`busy (i=0range@213.137.48.212) |
| 23:28.07 | *** join/#asterisk lancey (i=Shady@support.net1.cc) |
| 23:28.07 | silistra`busy | wazaaaaaaaa |
| 23:28.08 | silistra`busy | :) |
| 23:28.13 | tzanger | ha |
| 23:28.14 | silistra`busy | lancey sux :) |
| 23:28.18 | tzanger | not my wallpaper |
| 23:28.22 | QuAd|Haudrauf | hm right, :) btw, has anyone tested if gxp2000 works with cisco-style poe too? |
| 23:28.24 | tzanger | I just spent the afternoon tearing it down |
| 23:28.25 | _Sam-- | adeel_: cant you do what you want from an AGI? |
| 23:28.35 | Drew_____ | nutin - watching the game, having a bud... |
| 23:28.36 | adeel_ | how Sam |
| 23:28.53 | _Sam-- | like use php or something? |
| 23:28.56 | adeel_ | is there a script that i can use |
| 23:29.10 | adeel_ | i was using System command but no longer works |
| 23:29.24 | _Sam-- | adeel_: check this url...im not sure exactly how it will help you, but i think it will |
| 23:29.25 | _Sam-- | http://free.oxymium.net/Asterisk/AGIAddOns/ |
| 23:29.36 | adeel_ | ok thanks |
| 23:30.05 | *** join/#asterisk insomnia (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 23:30.08 | silistra`busy | i think asterisk sux :) |
| 23:30.09 | adeel_ | i have been to this page but its too complicated for me cuz |
| 23:30.15 | mutilator | whats this mean |
| 23:30.15 | mutilator | Jan 26 18:26:48 WARNING[29464]: chan_zap.c:8360 pri_dchannel: Ring requested on channel 0/4 already in use on span 1. Hanging up owner. |
| 23:30.39 | *** join/#asterisk insomni_ (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 23:30.44 | silistra`busy | this means your in big trouble ;) |
| 23:30.53 | _Sam-- | adeel_: if you odnt understand AGI / php...you will probably need to find someone to help you i think...but i am usually wrong. |
| 23:30.54 | adeel_ | cuz i am a beginer |
| 23:30.56 | mutilator | uh right.. |
| 23:31.13 | adeel_ | ok... |
| 23:31.29 | _Sam-- | there are probably a dozen ways to do what you want, adeel. |
| 23:31.40 | adeel_ | these is one thing that i am still tring to find out |
| 23:31.47 | _Sam-- | but they all , in my opinion, involve some type scripting (php or shell or perl, etc) |
| 23:32.11 | *** join/#asterisk lunaphyte (n=lunaphyt@pool-71-115-128-96.gdrpmi.dsl-w.verizon.net) |
| 23:32.35 | adeel_ | i am running a script |
| 23:32.56 | _Sam-- | good, then change it to do what you need |
| 23:33.13 | adeel_ | this is what i got the info from http://www.voip-info.org/wiki/view/Asterisk+tips+fileexistance |
| 23:34.03 | adeel_ | but no longer works |
| 23:34.18 | silistra`busy | yeah |
| 23:34.55 | QuAd|Haudrauf | has anyone ever wondered when doing "less gxp2000.bin" why there's written: "This is free software; see the source for copying conditions."? :) |
| 23:35.11 | QuAd|Haudrauf | i wonder which tool they used to compile it to .bin |
| 23:35.14 | _Sam-- | never crossed my mind to 'less' a bin file |
| 23:35.24 | *** join/#asterisk pifiu-laptop (n=someone@adsl-068-213-231-041.sip.mia.bellsouth.net) |
| 23:35.30 | justinu | strings usually works better for that kidna stuff |
| 23:35.44 | _Sam-- | there had been requests to open source the gxp2000 firmware |
| 23:35.57 | QuAd|Haudrauf | ! |
| 23:36.07 | adeel_ | how to enable the global priority jumping option |
| 23:36.07 | QuAd|Haudrauf | if that happens i will be really happy |
| 23:36.12 | Seldon1975 | can someone please remind me of the commands to stop zaptel, asterisk and the wctdm driver; I'm about to upgrade to 1.2.3 |
| 23:36.29 | jaike | service zaptel stop |
| 23:36.30 | _Sam-- | i dont think it will, but i did read where a few people had suggested the gxp2000 would be a better phone if they would open source the firware |
| 23:36.33 | jaike | fedora? |
| 23:36.35 | jaike | :) |
| 23:36.39 | Dr-Linux | what is API ? its manager interface, API stand for? :S |
| 23:36.56 | [av]bani | _Sam--: minor annoyance, handset volume = speakerphone volume on gxp |
| 23:37.17 | [av]bani | forspeakerphone to work usefully i have to crank it all the way up, which means handset is a megaphone |
| 23:37.20 | adeel_ | how to enable the global priority jumping option does any on know |
| 23:37.24 | QuAd|Haudrauf | justinu, with strings you don't find that notice i just mentioned. |
| 23:37.33 | _Sam-- | i thought there are seperate volume settings in the new phone GUI |
| 23:37.33 | silistra`busy | basi ko burzo pi6ete :)) |
| 23:37.41 | [av]bani | _Sam--: i didnt see any |
| 23:37.58 | [av]bani | ring tone and ring volume |
| 23:38.01 | [av]bani | thats about it it seems |
| 23:38.17 | *** part/#asterisk silistra`busy (i=0range@213.137.48.212) |
| 23:38.21 | *** join/#asterisk pifiu-laptop (n=someone@adsl-068-213-231-041.sip.mia.bellsouth.net) |
| 23:38.28 | adeel_ | Does any one know where i can find uptodate documentaion on asterisk |
| 23:38.30 | _Sam-- | will have to remember that...i cant remember the last time i used a speakerphone for other than just dialing |
| 23:38.56 | Seldon1975 | jaike: asterisk @ home |
| 23:39.01 | Seldon1975 | jaike: ie: redhat |
| 23:39.05 | jaike | adeel: in this channel |
| 23:39.34 | QuAd|Haudrauf | the ringtone files ringX.bin have the same notice. they get created by the modified "sox" they have on their webbie |
| 23:39.38 | jaike | seldon: if you did make config when you compiled zaptel...just do service zaptel stop |
| 23:41.11 | QuAd|Haudrauf | oCypirhg tc( )9199 ,dApaitevD gitilaT cenhlogoei,sI cn -- could translate to copyright 1999 Adaptive Digital Technologies |
| 23:41.53 | QuAd|Haudrauf | aha, DSP ppl |
| 23:42.14 | QuAd|Haudrauf | Adaptive Digital develops DSP algorithms and solutions for telephony and video applications such as Voice and Video Over IP, conferencing, echo cancellation, and video security. |
| 23:42.16 | QuAd|Haudrauf | ;} |
| 23:42.27 | QuAd|Haudrauf | grandstream obviously use their tech |
| 23:42.34 | *** join/#asterisk eieiyo (n=eieiyo@68-119-68-89.dhcp.mtgm.al.charter.com) |
| 23:42.45 | _Sam-- | they are doing the echo canc |
| 23:43.01 | _Sam-- | NOTE: (Nov25/05) (i agree please OPEN SOURCE) i understand that you likely have segments when are propriatry to other companies (like the Adaptive Digital Echo cancellation) but even if you were to release source code with the propriatry parts removed it would be great as we woul still be able to write code to replace the missing parts and changes could still be rolled into the grandstream firmware if this was your wish... - SoloFlyer |
| 23:43.25 | QuAd|Haudrauf | :) |
| 23:43.58 | QuAd|Haudrauf | i wish the firmware was opensourced too |
| 23:44.04 | _Sam-- | me three |
| 23:44.08 | _Sam-- | that makes 5 of us so far |
| 23:44.13 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-43-124.cybersurf.com) |
| 23:44.27 | *** join/#asterisk DynaGuy (n=dynaguy@S01060011954e8668.vc.shawcable.net) |
| 23:44.31 | mutilator | anyone know what'de cause audio on the pstn side to breakup really bad but not on the voip(sip) side, using g711u and g729a (is even worse than 711u) |
| 23:45.19 | mutilator | box is a dual 3ghz xeon w/o anything running but asterisk |
| 23:45.49 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@84.205.154.241) |
| 23:50.04 | QuAd|Haudrauf | gonna add some stuff to the feature-request list of the wiki |
| 23:51.33 | *** join/#asterisk morale (i=russell@S010600111155e117.cg.shawcable.net) |
| 23:52.15 | eieiyo | whats the reason for doing make linux26? |
| 23:52.26 | eieiyo | 2.6 kernel? |
| 23:53.24 | jebba | eieiyo, ya |
| 23:53.26 | Nugget | the linux26 make tag is no longer necessary, I believe. |
| 23:53.31 | Nugget | it's vestigial |
| 23:53.36 | eieiyo | ok cool |
| 23:53.37 | fugitivo | eieiyo: you don't need it |
| 23:53.41 | eieiyo | thanks guys |
| 23:53.44 | jebba | Any idea how to play a file into a MeetMe conference? |
| 23:53.54 | jebba | Or which is the best way? :) |
| 23:54.47 | eieiyo | what about figuring out what package needs what for the makefile... somtimes you do make clean and then make... and then sometimes other packages need make clean then make then make install... etc.. can you examine the makefile and figure this out? or is there a webpage with this information on it? |
| 23:55.33 | Nugget | Welcome To Linux Country, eieiyo. |
| 23:55.34 | jebba | eieiyo, make clean ; make ; make install should work fine. |
| 23:55.49 | eieiyo | ok gotcha... |
| 23:55.56 | eieiyo | just making sure i am doing it right |
| 23:56.09 | eieiyo | its working, but i just wanted to make sure i was not missing out on a feature ;) |
| 23:56.42 | Seldon1975 | please tlel me the command to stop the wctdm driver |
| 23:56.51 | tzanger | rmmod wctdm zaptel |
| 23:56.54 | Seldon1975 | rmmod.. something |
| 23:56.57 | Seldon1975 | aha |
| 23:56.58 | Seldon1975 | thx |
| 23:57.20 | Seldon1975 | can you tell me how to list the modules? |
| 23:57.41 | Seldon1975 | modprobe.. something |
| 23:57.41 | *** join/#asterisk Coccyx (n=clint@typhoon.org) |
| 23:58.15 | *** join/#asterisk copantl (n=galel@63.245.93.138) |
| 23:58.33 | Nugget | lsmod |
| 23:58.36 | Seldon1975 | thx |
| 23:58.39 | Nugget | np |
| 23:59.06 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
