| 00:00.28 | voltagex | nope |
| 00:00.34 | voltagex | crontab decided to erase them |
| 00:00.41 | litage | JT: yeah i know what you mean. however, i need to fix this, and i unfortunately can't leave the alternative router in place |
| 00:00.41 | voltagex | instead of doing what it was supposed to |
| 00:00.42 | voltagex | :/ |
| 00:00.43 | voltagex | ffs |
| 00:00.45 | voltagex | ARRGHH |
| 00:01.01 | JT | litage: you might need to port forward |
| 00:01.28 | litage | JT: port forwarding's already configured on the [crappy] router, and i've re-checked the settings 10 times |
| 00:01.29 | Fieldy | how would an SIP user who is not part of my server call me, sip:ext@domain ? do i need to config anything special to allow these incoming calls? |
| 00:02.32 | JT | allowguest=yes iirc |
| 00:04.08 | Fieldy | hrm. it seems allowguest can be set to yes (default), no, or osb. anyone know what osp is? |
| 00:04.53 | Fieldy | s/osb/osp |
| 00:05.29 | JonathanW | i'm guesing i got burried? |
| 00:07.21 | voltagex | yes |
| 00:07.57 | JonathanW | http://channels.debian.net/paste/5850 |
| 00:08.29 | JonathanW | thats with out the bind and srv... added in the username= and thats verbose 10 |
| 00:09.10 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:09.51 | JT | JonathanW: what does the output of 'sip show peers' and 'sip show registry' say? |
| 00:10.10 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 00:14.52 | voltagex | do I have to do something special to make externhost=voltagex.homeip.com work? |
| 00:16.16 | litage | JT: in the [crappy] router, i've set the asterisk box as the DMZ host. however, qualify still says the sip server is unreachable. any other suggestions? |
| 00:18.09 | *** join/#asterisk mmartinn (n=martin@adsl-065-005-200-225.sip.gnv.bellsouth.net) |
| 00:20.50 | *** join/#asterisk Qwell (n=north@pdpc/sponsor/digium/Qwell) |
| 00:20.50 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 00:23.22 | *** join/#asterisk voiper1 (n=luke@ozvoip.dsl.onthenet.net) |
| 00:24.11 | *** join/#asterisk hansin321 (n=hansin32@c-67-174-180-72.hsd1.co.comcast.net) |
| 00:24.35 | JonathanW | JT http://channels.debian.net/paste/5851 |
| 00:25.31 | JonathanW | i did a doubble check to make sure all firewalls ect are not running right now |
| 00:27.24 | voiper1 | I have run a patlooptest on a te110p card and received the following results. does anyone know what it means? http://channels.debian.net/paste/5853 |
| 00:32.10 | *** join/#asterisk Innatech (n=daf@netblock-72-25-97-119.dslextreme.com) |
| 00:34.21 | mmartinn | Hey folks; what does "chan_sip.c:1227 retrans_pkt: Maximum retries exceeded on transmission" with an IP address mean? |
| 00:34.27 | mmartinn | Network failure? |
| 00:37.24 | Fieldy | mmartinn: i'm not sure, but it sure sounds like it |
| 00:37.34 | voltagex | how can I have asterisk update my externip because externhost isn't working :/ |
| 00:37.46 | Qwell | voltagex: got externrefresh set? |
| 00:39.33 | flenders | hey, to setup a sangoma a101, is this the best way? http://wiki.sangoma.com/wanpipe-asterisk-configure |
| 00:39.35 | mmartinn | Does anyone know what " Spawn extension (macro-outgoing, s-CHANUNAVAIL, 4) exited non-zero on 'SIP/2076-1728bae0' in macro 'outgoing'" means; does that mean the macro exited with that or was entered with those values? |
| 00:40.23 | voltagex | Qwell: have now, is there a way I can check what * is seeing as public IP? |
| 00:40.46 | *** join/#asterisk ZX81 (n=ZX81@60-234-238-188.bitstream.orcon.net.nz) |
| 00:41.27 | JonathanW | JT? |
| 00:41.44 | voltagex | JonathanW: I can help, state your problem |
| 00:41.56 | Ng | flenders: the sangoma wiki instructions seem to be pretty good, I just used them on an a102 on ubuntu edgy and they covered almost everything apart from little local config variations in asterisk (hardly their job ;) |
| 00:42.53 | JonathanW | well |
| 00:42.57 | mmartinn | Is there any way a NOANSWER would be the DIALSTATUS if I told Dial() not to timeout? |
| 00:43.00 | JonathanW | http://channels.debian.net/paste/5851 |
| 00:43.09 | voltagex | Quell... |
| 00:43.11 | JonathanW | i gues that dont apply tho thats what jt asked to see |
| 00:43.44 | voltagex | well JonathanW, there's nothing there for FreeWorldDialup, so it can't connect to it |
| 00:43.45 | *** join/#asterisk jkieffer (n=jkieffer@cpe-071-076-130-099.triad.res.rr.com) |
| 00:43.58 | JonathanW | i'm using no fwd |
| 00:44.12 | jkieffer | Can anyone help with chan_cellphone? |
| 00:44.17 | JonathanW | its all pvt sip in and out thrue this server no exturnal conection |
| 00:44.38 | JonathanW | to fwd needed i thought |
| 00:50.04 | jkieffer | Anybody out there? |
| 00:50.16 | *** part/#asterisk ZX81 (n=ZX81@60-234-238-188.bitstream.orcon.net.nz) |
| 00:50.21 | *** join/#asterisk ZX81 (n=ZX81@60-234-238-188.bitstream.orcon.net.nz) |
| 00:50.56 | jkieffer | Anyone that could help me with a chan_cellphone problem? |
| 00:51.27 | JonathanW | all the sip phones connect to the asterisk server, if there all set as extenctions it should route to them with out the use of fwd i would think |
| 00:53.50 | mmartinn | Is there any way a NOANSWER would be the DIALSTATUS if I told Dial() not to timeout? |
| 00:54.35 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 00:58.54 | tclark | hry anyone using this hpec $10/channel echo can now my ivr cant be heard when i dial in anyobe been there done that & fixed it ? |
| 01:03.56 | *** join/#asterisk NeonLevel (i=NeonLeve@200.52.142.184) |
| 01:04.40 | [TK]D-Fender | mmartinn, not that I can think of. |
| 01:05.36 | NeonLevel | good day, how do i match tos_audio=ef with tc ? is is something like tc filter.... match ip tos ___ 0xff flowid ..... what shoud i put there to match it? |
| 01:06.07 | Mercestes | [TK]D-Fender, Did you get that? |
| 01:07.25 | NeonLevel | i'm in the need of giving QoS priority to tos_audio, how can i match if tos_audio=ef please? |
| 01:11.40 | ltdwk | is ef hex ? |
| 01:11.50 | ltdwk | if so then just match tos 0xef |
| 01:11.52 | NeonLevel | dont know |
| 01:12.07 | [TK]D-Fender | Mercestes, Get what? |
| 01:12.24 | *** part/#asterisk ZX81 (n=ZX81@60-234-238-188.bitstream.orcon.net.nz) |
| 01:12.31 | NeonLevel | i dont think so, because sip.conf suggest tos_video=af41, how is that hex? |
| 01:12.58 | NeonLevel | or tos_sip=cs3, hex does not include s, right? |
| 01:15.23 | Mercestes | [TK]D-Fender, Whatever neonlevel is saying. |
| 01:15.58 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 01:16.01 | [TK]D-Fender | Mercestes, Not a sub-dialect of gibberish I'm familiar with, sorry... |
| 01:16.17 | NeonLevel | sorry to bother :-S |
| 01:16.55 | NeonLevel | is just that im in a very congested network, and had to do some priority in traffic so VoIP could sound OK |
| 01:16.58 | NeonLevel | that's all |
| 01:17.47 | JT | NeonLevel: is english your first language? |
| 01:18.16 | NeonLevel | no sorry |
| 01:18.22 | JT | ah ok |
| 01:18.23 | NeonLevel | español es mi lengua |
| 01:18.44 | NeonLevel | pero creo que no todo mundo habla español y menos aqui |
| 01:18.48 | dj-fu | #asterisk-es |
| 01:18.50 | NeonLevel | :-) |
| 01:19.04 | NeonLevel | ah ok ese canal no lo conocia |
| 01:19.17 | *** join/#asterisk bblack (n=bblack@38-118-35-20.rapiddsl.net) |
| 01:19.17 | NeonLevel | pero igual y pense que los genios estarian aqui! :-P |
| 01:19.29 | NeonLevel | alguien mas habla español aqui? |
| 01:19.39 | flenders | NeonLevel: I can understand it |
| 01:20.07 | NeonLevel | hola flenders, de donde eres? |
| 01:20.26 | flenders | NeonLevel: I'm brazilian, but been living in australia for about 5 years |
| 01:20.31 | bblack | hi guys. anybody have any ideea how can I stop asterisk 1.2 to accept to put calls on hold for SIP users? I guess in 1.4 you can do calllimit=1 and limitonpeers=yes, but what about SIP friends in 1.2? |
| 01:21.11 | NeonLevel | que bueno es eso flenders, encontraste trabajo en australia? |
| 01:21.15 | ManxPower | bblack: the best way is to configure your phone not to accept more than 1 call |
| 01:21.21 | NeonLevel | bblack: stop now |
| 01:21.34 | flenders | NeonLevel: yeah... it wasn't that hard |
| 01:21.44 | bblack | NeonLevel: and I thought rm -rf / would do it :p |
| 01:21.49 | NeonLevel | sorry bblack, bad answer |
| 01:21.59 | NeonLevel | :p |
| 01:22.28 | bblack | ManxPower: you can do that, but is there any way to do it from the Asterisk side? |
| 01:22.48 | ManxPower | bblack: yes, but it can be very complicated, using SetGroup, Groupcount, etc |
| 01:22.50 | NeonLevel | flenders, y la vida es mejor en australia?, puedo encontrar trabajo yo tambien alla? |
| 01:23.22 | flenders | NeonLevel: anyone can... |
| 01:23.30 | flenders | NeonLevel: and yeah, life here is amazing |
| 01:23.41 | flenders | NeonLevel: night life in spain is better though |
| 01:23.45 | bblack | ManxPower: that's actually a good ideea, thanks |
| 01:24.03 | flenders | NeonLevel: been to madrid a few times |
| 01:24.59 | NeonLevel | flenders: thanks! |
| 01:25.14 | *** join/#asterisk bkuhn (n=bkuhn@fsf/member/bkuhn/bkuhn) |
| 01:29.30 | *** join/#asterisk bkuhn (n=bkuhn@fsf/member/bkuhn/bkuhn) |
| 01:31.45 | *** join/#asterisk KuJaX (n=one@customtrading.dsl.xmission.com) |
| 01:33.21 | KuJaX | silly question, but asterisk isn't running. I get "Unable to connect to remote asterisk". Asterisk is currently stopped, any suggestions? |
| 01:34.00 | bblack | KuJaX: tail /var/log/asterisk/full |
| 01:34.12 | bblack | KuJaX: and see what's the last messages |
| 01:34.42 | KuJaX | chan_zap.c: Unable to open channel 1: No such device or address. loader.c: loading module chan_zap.so failed |
| 01:34.55 | KuJaX | bblack - Weird, because the PCI card for the zapel channel hasn't been changed or altered in anyway. |
| 01:35.06 | litage | if you use netcat to connect to a SIP server, what can you send to receive a response...any response? |
| 01:35.33 | KuJaX | Yet we don't even use that card at all anyways, can I somehow get rid of zapel |
| 01:35.51 | bblack | litage: it might be hard to construct a correct SIP message by hand. you can try to use sipsak for debugging |
| 01:36.57 | bblack | KuJaX: edit /etc/asterisk/modules.conf and add "noload => chan_zap.so" |
| 01:36.59 | litage | bblack: i've used sipsak successfully; when run from my asterisk box, it says it can register with the SIP server. but using the exact same credentials, asterisk says this on its CLI: chan_sip.c:5484 sip_reg_timeout: -- Registration for '8880001014@sip.voxpak.com' timed out, trying again (Attempt #99) |
| 01:37.59 | litage | bblack: i don't understand why sipsak can register, but asterisk can't |
| 01:38.12 | JonathanW | pastebin link? |
| 01:38.23 | bblack | litage: it can be a domain problem, for example you're supposed to register as 8880001014@voxpak.com and you're registering as ...@sip.voxpak.com and voxpack cares |
| 01:38.26 | litage | JonathanW: rafb.net/paste |
| 01:38.42 | JT | pastebin.ca |
| 01:38.49 | KuJaX | Okay bblack, I did that. How do i reload modules or do I have to reboot the machine? |
| 01:39.11 | bblack | KuJaX: you just try to start asterisk again |
| 01:39.21 | KuJaX | any specific command you would recommend? |
| 01:39.23 | JonathanW | JT i get this when somone trys to dial my extenction http://channels.debian.net/paste/5854 |
| 01:40.01 | JT | sounds like a firewall or nat issue |
| 01:40.01 | bblack | KuJaX: depends on the installation/linux distribution. on a correct installed redhat flavour, "service asterisk start" would do |
| 01:40.23 | KuJaX | wouldn't start. Did tail /var/log/asterisk/full and the last line says loader.c: loading module chan_zap.so failed! |
| 01:40.56 | KuJaX | Okay I added it in the module context, not global... |
| 01:41.22 | bblack | KuJaX: in [modules] in modules.conf |
| 01:41.42 | KuJaX | scary thing is that within [global] i have chan_zap.so=yes |
| 01:42.27 | JonathanW | http://channels.debian.net/paste/5856 thats with sip debuging enabled |
| 01:42.28 | bblack | KuJaX: put a ";" in front |
| 01:43.17 | litage | bblack: ``sipsak -v -U -a 'xxxxx' -s sip:8880001014@sip.voxpak.com'' works, and this is what's in sip.conf, which doesn't work: register => 8880001014:xxxxx@sip.voxpak.com/8880001014 |
| 01:43.47 | bblack | litage: can you do a sip trace and put it on a pastebin somewhere? |
| 01:44.23 | litage | bblack: can you be more specific please? by "sip trace", do you mean a packet capture while sipsak or asterisk is trying to register, or ...? |
| 01:44.59 | bblack | litage: "sip debug" in asterisk and a copy of the register dialog |
| 01:45.04 | mmartinn | Is there any way a NOANSWER would be the DIALSTATUS if I told Dial() not to timeout? |
| 01:46.06 | [TK]D-Fender | mmartinn, And why would you ever LEAVE Dial if you never time out? |
| 01:46.37 | mmartinn | [TK]D-Fender: I don't know; I have Dial() with no timeout and I get DIALSTATUS=NOANSWER after like 5 rings |
| 01:46.46 | mmartinn | Not always, but occaisonally |
| 01:47.09 | *** join/#asterisk bkuhn (n=bkuhn@fsf/member/bkuhn/bkuhn) |
| 01:47.27 | ManxPower | mmartinn: What is the value of HANGUPCAUSE? |
| 01:47.27 | TokyoJimu | Can I set an environment variable (like TZ) in extensions.conf? I need to set the timezone differently for certain callers. |
| 01:47.51 | ManxPower | TokyoJimu: Why? |
| 01:48.07 | bblack | mmartinn: do a grep after NOANSWER in app_dial.c; there is at least one other scenario, when the destination reject the call because of the privacy issues; I got the same result when the destination would return a timeout |
| 01:48.16 | ManxPower | voicemail is the only place callers would hear the time in normal situations and you can set different timezones in voicemail.cinf |
| 01:48.22 | TokyoJimu | Callers to a certain program are in a different timezone than the machine, so I need their timestamps to be correct. |
| 01:48.33 | mmartinn | ManxPower: I don't currently output that; is it displayed in "Spawn extension... non-zero" ? Otherwise I'd have to add that to NoOp |
| 01:48.44 | ManxPower | DIALSTATUS is VERY generic. HANGUPCAUSE is much more specific. |
| 01:48.47 | TokyoJimu | But we're not using the voicemail module. This is a custom app. |
| 01:49.03 | TokyoJimu | I see that I can read EnvVars but I'm not sure about setting them. |
| 01:49.06 | ManxPower | TokyoJimu: then you would have to do the timezone stuff in your app. |
| 01:49.30 | ManxPower | asterisk does all it's stuff in UTC |
| 01:49.44 | litage | bblack: http://rafb.net/p/23EN8l67.html |
| 01:49.52 | TokyoJimu | ManxPower: I was faraid of that. Thanks. |
| 01:49.55 | bblack | TokyoJimu: in 1.4 check ENV function |
| 01:50.04 | ManxPower | TokyoJimu: do your users at least have voicemailboxes? |
| 01:50.22 | ManxPower | bblack: that will not help him |
| 01:50.29 | TokyoJimu | Yes. They have mailboxes but it's all in our app. |
| 01:50.56 | ManxPower | TokyoJimu: do the user does not have entries in voicemail.conf? |
| 01:51.06 | ManxPower | Are they SIP callers or PSTN callers? |
| 01:51.25 | TokyoJimu | ManxPower: Correct. No relation to voicemail.conf. |
| 01:51.58 | JonathanW | JT http://channels.debian.net/paste/5856 thats with sip debuging enabled do you see somthing from that ouput |
| 01:52.20 | bblack | ManxPower, TokyoJimu: it depends what he wants to do with TZ. if he's looking for changing the response to date() in a PHP AGI, it will work, if he wants to change for example the timezone of the CDRs, it will not |
| 01:53.24 | JT | JonathanW: why are the softphones on private IPs? |
| 01:54.26 | ManxPower | bblack: he wants to change the time zone depending on what person is calling |
| 01:54.28 | bblack | litage: does sip.voxpack.com respond to your asterisk register message? |
| 01:54.31 | JonathanW | what do you mean |
| 01:54.32 | TokyoJimu | ManxPower: it's a dating system. +1 415 428 3000 if you're interested. |
| 01:54.34 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:54.37 | JonathanW | both softphones are on wan ip's |
| 01:54.44 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:54.47 | JT | JonathanW: no, they are RFC1918 PRIVATE IPs |
| 01:54.52 | JT | they are NOT PUBLIC IPs |
| 01:54.52 | litage | bblack: not as far as i can tell. is there a specific way to determine whether or not it does? |
| 01:55.14 | bblack | litage: it should show up in the console |
| 01:55.20 | JT | i have no idea what this WAN you have is, but it uses private IPs |
| 01:55.25 | JonathanW | dns 24.10.32.18 |
| 01:55.34 | litage | bblack: i didn't see a response displayed on asterisk's console |
| 01:55.41 | JT | 92.168.1.102 |
| 01:55.48 | JT | sorry 192.168.1.102 |
| 01:56.01 | JonathanW | oh crap that one is behind a router |
| 01:56.24 | JonathanW | the computer that one is on is on a router not direct to internet |
| 01:56.29 | ManxPower | JT: Wow! That is just 1 number off of MY IP!!!! |
| 01:56.48 | JonathanW | take stab in dark it needs to find a wan ip only |
| 01:57.04 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 01:57.04 | JonathanW | what ports should be forwarded in the router? |
| 01:57.37 | ManxPower | JonathanW: That totally depends on how your network is. |
| 01:57.50 | JT | JonathanW: please stop saying wan ip |
| 01:58.02 | bblack | litage: so you see asterisk retransmiting the register (at least a few times) and no response back? |
| 01:58.09 | ManxPower | Are only SIP clients on a private IP, or is Asterisk on the private IP? |
| 01:58.16 | JT | JonathanW: asterisk can be setup to work through a lot of nat routers |
| 01:58.24 | JT | port forwarding for the client is a last result |
| 01:58.42 | JT | resort |
| 01:58.43 | JT | even |
| 01:58.46 | litage | bblack: correct. i just enabled "sip debug" for longer, disabled it, and carefully looked through each message. plenty of registration requests were sent to sip.voxpak.com , but no responses came back |
| 01:58.56 | litage | bblack: it's very odd [to me |
| 01:59.09 | litage | bblack: it's very odd [to me] that sipsak can register |
| 01:59.23 | bblack | litage: and sipsak register from the same machine? |
| 01:59.28 | litage | bblack: yes |
| 01:59.42 | litage | bblack: i run sipsak from the asterisk box |
| 02:00.25 | bblack | litage: you can try to change the useragent in sip.conf to something else; some providers are alergic to asterisk :) |
| 02:00.47 | *** join/#asterisk Dovid (n=Dovid@l192-117-114-1.broadband.actcom.net.il) |
| 02:01.57 | litage | bblack: i have another asterisk box registered with the same provider. the only difference is that it has a public IP address, whereas the problem asterisk box has a private IP address |
| 02:02.09 | ManxPower | JT: I guess JonathanW didn't want our help |
| 02:02.48 | JT | heh, maybe not |
| 02:03.26 | bblack | litage: can you configure the nat in asterisk? nat=yes, externip, etc...? |
| 02:04.35 | litage | bblack: i have "nat=yes" enabled, but not externip as the public ip address is dynamic |
| 02:05.07 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
| 02:05.14 | bblack | litage: did you try externhost? |
| 02:05.21 | litage | bblack: not yet |
| 02:05.44 | bblack | litage: some providers do not work with clients behind nat |
| 02:06.11 | litage | bblack: i have IP phones on my local private subnet registered with the same provider though |
| 02:06.56 | Dovid | litage: asterisk works a bit diffrent |
| 02:07.30 | ManxPower | litage: You, of course did the localnet, extenip, and port forwarding as well as changing rtp.conf, right? |
| 02:07.31 | bblack | litage: do they use STUN? |
| 02:07.42 | ManxPower | What IS this fetish with STUN anyway? |
| 02:07.48 | Dovid | lol |
| 02:07.54 | JT | i wouldn't mind seeing STUN eliminated |
| 02:08.06 | Dovid | ~STUN |
| 02:08.17 | jbot | somebody said stun was that feeling you get when you realise your SIP call actually got through!. Simple Traversal of UDP over NATs |
| 02:08.17 | JT | its only purpose is to cater to RUBBISH SIP SERVERS |
| 02:08.17 | ManxPower | JT: replace it with KILL 8-) |
| 02:08.23 | JT | that are too dumb to read packet headers |
| 02:08.24 | bblack | STUN can be the explanation why the phones work |
| 02:08.38 | Dovid | haha. i luv the bit |
| 02:08.43 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 02:08.56 | ManxPower | bblack: Most providers do all the similar SIP NAT hacks, just like Asterisk does. |
| 02:09.15 | *** join/#asterisk pigpen2 (n=pigpen@fw.seamans.cc) |
| 02:09.16 | litage | ManxPower: localnet is set as "localnet=192.168.1.0/255.255.255.0" and port forwarding on the router is configured. externip is not configured because the public/wan ip address is dynamic |
| 02:09.23 | bblack | ManxPower: I agree with you regarding stun, but I know providers that doesn't |
| 02:09.29 | litage | bblack: no, the provider doesn't use stun |
| 02:09.41 | JT | jbot: stun is also a client side method to cater to crappy sip servers |
| 02:09.43 | jbot | okay, JT |
| 02:09.43 | bblack | litage: not the provider, your phones that actually register |
| 02:09.49 | JT | litage: no, clients use stun |
| 02:09.50 | ManxPower | litage: set externip and see what happens |
| 02:09.51 | JT | litage: not servers |
| 02:10.01 | litage | bblack: the phones don't use stun |
| 02:10.14 | JT | sipsak? |
| 02:10.49 | pigpen2 | Evening all...I |
| 02:11.18 | pigpen2 | I need to see if I can trim down the xfer time on my main IVR. |
| 02:11.41 | ManxPower | pigpen2 define "xfer time" |
| 02:11.44 | pigpen2 | ie: "press 1 for directions" takes 4 - 5 seconds to start the directions wav file. |
| 02:12.03 | ManxPower | pigpen2: then you have another extension that can start with 1 |
| 02:12.08 | ManxPower | that is the ONLY reason this would happen |
| 02:12.24 | pigpen2 | hmm.... |
| 02:12.34 | JT | timeout dialling |
| 02:12.47 | pigpen2 | did you tell me this the other night....if so...sorry, I think I get it now. |
| 02:12.51 | pigpen2 | I will check. |
| 02:12.51 | ManxPower | JT: not if he is already in the IVR |
| 02:13.07 | pigpen2 | ie: anything that starts with a "1" will wait for the digit timeout. |
| 02:13.36 | JT | ManxPower: it's still timeout dialling |
| 02:13.48 | ManxPower | pigpen2: otherwise how would asterisk know if you are dialing 1 or dialing 1234? |
| 02:13.51 | JT | it's in the dialplan, that's what matters |
| 02:13.56 | pigpen2 | I will paste my intro into pastebin just in case. |
| 02:13.58 | ManxPower | if another extension could match 1234 |
| 02:14.37 | pigpen2 | http://www.pastebin.ca/413007 |
| 02:14.39 | litage | ManxPower: i set "externip=121.50.193.61" in the "[general]" section of sip.conf and ran "sip reload". a few seconds later, asterisk said it couldn't register: chan_sip.c:5484 sip_reg_timeout: -- Registration for '8880001014@sip.voxpak.com' timed out, trying again (Attempt #5) |
| 02:15.20 | ManxPower | litage: unload chan_sip.sp and then load chan_sip.sol |
| 02:15.23 | ManxPower | s/l// |
| 02:16.07 | pigpen2 | ManxPower, Would I need to look into other contexts, or just the ones that are included? |
| 02:16.46 | ManxPower | pigpen2: only the current context and any included contexts |
| 02:16.47 | pigpen2 | ah..yes. |
| 02:17.01 | pigpen2 | I only have 100 extensions beginning with 1XXX |
| 02:17.03 | pigpen2 | :) |
| 02:17.08 | ManxPower | there you go. |
| 02:17.14 | pigpen2 | haha.. |
| 02:17.17 | JT | damnit |
| 02:17.21 | JT | use CONTEXTS |
| 02:17.24 | pigpen2 | Once again, and many more to come...Thanks. |
| 02:17.24 | JT | hi :) |
| 02:17.28 | litage | ManxPower: unloading chan_sip.so failed: http://rafb.net/p/eFLebf15.html |
| 02:17.36 | pigpen2 | You to JT. |
| 02:17.43 | ManxPower | litage: then stop and start asterisk. |
| 02:17.49 | bblack | litage: set nat=true, externip and localnet |
| 02:18.02 | JT | pigpen2: IVRs should have their own contexts |
| 02:18.09 | JT | that would prevent timeout dialling |
| 02:18.16 | JT | and potential breaches in dialplan security |
| 02:18.25 | JT | or unintended feature access |
| 02:18.26 | pigpen2 | Yeah..I was thinking that I don't need to include my main context. |
| 02:18.28 | ManxPower | litage: chances are a pending register or current calll is preventing sip from unloading |
| 02:19.27 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 02:20.08 | litage | ManxPower, bblack: even after setting "nat=true", "externip=121.50.193.61" and "localnet=192.168.1.0/255.255.255.0" and then stopping and starting asterisk, asterisk still can't register |
| 02:20.13 | aptura | whats the cause of a missing makefile when compiling zaptel |
| 02:20.43 | aptura | make file in kernal-headers |
| 02:20.43 | litage | aptura: you sure you downloaded it or didn't delete it? |
| 02:20.54 | aptura | ohh i have downloaded it the other day |
| 02:22.24 | *** join/#asterisk theblue (n=theblue@unaffiliated/theblue) |
| 02:22.27 | theblue | Hi all. |
| 02:22.36 | bblack | liptage: you can try a packet capture of what asterisk is sending and what sipsak is sending to see what are the differences and try to see why the server doesn't respond to asterisk's request |
| 02:22.59 | theblue | Is AGI usable through Python? |
| 02:23.01 | theblue | And if so, how? |
| 02:23.17 | bblack | litage: and double check that the IP in the asterisk message is ok |
| 02:23.27 | litage | bblack: k, i'll give that a shot. back in a bit :) |
| 02:25.02 | ManxPower | litage: you of course allow 5060 thru the nat router |
| 02:25.06 | litage | ManxPower: yes, otherwise sipsak wouldn't be able to registe |
| 02:25.17 | *** join/#asterisk aaronr (n=arussell@87.127.234.100) |
| 02:26.31 | JT | theblue: AGI is usable in ANY language that can communicate using STDIN and STDOUT |
| 02:26.36 | theblue | Ooo! |
| 02:26.40 | theblue | JT: Hi, by the way. |
| 02:26.43 | JT | hi |
| 02:26.55 | theblue | That could be VERY interesting... |
| 02:27.01 | JT | the book has a quick into to it |
| 02:27.03 | JT | ~thebook |
| 02:27.05 | jbot | well, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 02:27.13 | theblue | I've seen that. |
| 02:27.21 | theblue | Though I only read the first few chapters, I'll tak ea look. |
| 02:27.29 | JT | ok |
| 02:31.47 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 02:34.52 | *** join/#asterisk VoltageX (i=voltagex@creep.bur.st) |
| 02:35.06 | VoltageX | hi, any reason that externhost doesn't work? |
| 02:35.37 | VoltageX | still reporting my private IP to remote sip server |
| 02:42.47 | tengulre | VoltageX: anybody here sleeping now! |
| 02:44.00 | Dovid | hehe |
| 02:46.55 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 02:46.55 | *** mode/#asterisk [+o mog] by ChanServ |
| 02:49.30 | *** join/#asterisk Fieldy (i=RgZt4D7n@gentoo/contributor/Fieldy) |
| 02:52.06 | mmartinn | We had 80 people in our call center on Asterisk tonight |
| 02:52.13 | mmartinn | There was much drama and rejoicing |
| 02:52.15 | russellb | and? |
| 02:52.22 | russellb | it worked? :) |
| 02:52.25 | russellb | (i hope_ |
| 02:52.32 | mmartinn | We started one room at a time over the last few month |
| 02:52.33 | mmartinn | s |
| 02:52.36 | mmartinn | It mostly works |
| 02:52.44 | mmartinn | There's some weird hangups |
| 02:53.10 | mmartinn | But since you're present, thank you Russell for helping make it work :) |
| 02:53.42 | mmartinn | We definitely owe a lot to the devs and contributors, present or not =) |
| 02:55.21 | mmartinn | I've gotta say... I feel like the default config we started with is overcomplicated. Is there a historical or practical reason that some of them, such as the dialplan, have things like 10 trunks for outgoing calls? |
| 02:57.35 | russellb | the config sample extensions.conf neesd a rewrite |
| 02:57.56 | russellb | it probably started simple at one point, and got hacked up to death over the years |
| 02:58.05 | russellb | but you're welcome, I'm glad it is helping you out :) |
| 02:58.24 | mmartinn | We're trying to convince the university to go from Cisco to Asterisk |
| 02:58.25 | mmartinn | =) |
| 02:58.32 | russellb | cool deal |
| 02:58.34 | russellb | others have done it |
| 02:58.50 | mmartinn | We point to those examples, but they aren't "as big as us" |
| 02:59.03 | russellb | how big is that? |
| 02:59.12 | *** join/#asterisk Juggie (i=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 02:59.14 | mmartinn | They keep trying to use Asterisk to argue down the Cisco prices, and we keep trying to say that it would work because it works, not because it makes Cisco cheaper |
| 02:59.19 | ManxPower | I found that I had to write my dialplan from scratch |
| 02:59.21 | mmartinn | russellb: University of Florida |
| 02:59.40 | mmartinn | russellb: So like... however many offices support 45k students... I'm not sure how many are Cisco-based |
| 02:59.46 | mmartinn | but it is the "blessed solution" |
| 02:59.50 | russellb | 45k is a lot :) |
| 03:00.30 | JT | mmartinn: the dialplan is an example of the shear bredth of options |
| 03:00.47 | russellb | i know of installs that big, though ... |
| 03:00.54 | JT | mmartinn: what you're meant to do is have that as .samples or in another directory and write your dialplan from scratch |
| 03:01.14 | JT | 45k students... but how many extensions? |
| 03:01.16 | mmartinn | JT: I think we are definitely going to move in that direction |
| 03:01.26 | mmartinn | I don't think there are anywhere near that many phones |
| 03:01.39 | mmartinn | But staff offices are basically encouraged to switch as they upgrade |
| 03:02.05 | JT | 50 analogue phones is enough headache ;) |
| 03:02.06 | ManxPower | russellb: Digium needs to get more larger customers to admit they use Asterisk. |
| 03:02.22 | russellb | working on it ... |
| 03:02.29 | bkruse_home | that would be great......... |
| 03:02.34 | bkruse_home | :P |
| 03:02.35 | *** join/#asterisk k-man__ (n=jason@ppp244-232.static.internode.on.net) |
| 03:02.36 | mmartinn | Digium was very helpful for us at first; we have a support contract now |
| 03:02.44 | russellb | yay |
| 03:02.47 | JunK-Y | ManxPower: whats more larger customer btw? |
| 03:02.49 | bkruse_home | woot! |
| 03:03.12 | russellb | mmartinn: that lets me eat :-D |
| 03:03.16 | ManxPower | JunK-Y: people clients have heard of. |
| 03:03.27 | mmartinn | russellb: 8-D |
| 03:03.34 | bkruse_home | russellb: i still go hungry :[ |
| 03:03.41 | ManxPower | The ONLY larger installation I could find was Sam Houston Univ |
| 03:03.47 | bkruse_home | buy BE! |
| 03:03.49 | bkruse_home | :D |
| 03:03.51 | JunK-Y | ManxPower: i know a customers which 16 t1s using * |
| 03:03.53 | russellb | bkruse_home: because you're addicted to your job |
| 03:04.02 | ManxPower | JunK-Y: And their name is? |
| 03:04.04 | mmartinn | It seems that we actually get more hangups in the call center with Asterisk, due to the PRI lines -- we switched from E&M wink analog-ness |
| 03:04.05 | russellb | University of Pennsylvania uses it for voicemail |
| 03:04.09 | bkruse_home | so true |
| 03:04.13 | bkruse_home | we both are, its pretty awesome :] |
| 03:04.14 | JunK-Y | bkruse_home: you werent suppose to study? :) |
| 03:04.18 | russellb | they funded the IMAP storage work |
| 03:04.21 | JunK-Y | ManxPower: its private. |
| 03:04.27 | ManxPower | JunK-Y: Exactly |
| 03:04.42 | russellb | Colorado State uses Asterisk to some extent |
| 03:04.50 | bkruse_home | thanks file :D |
| 03:04.51 | ManxPower | if they won't admit to using asterisk it doesn not help me. |
| 03:04.57 | bkruse_home | ill getcha back for it, promise :] |
| 03:05.02 | russellb | ManxPower: those are ones that are public |
| 03:05.08 | mmartinn | We convinced UF telecom to let us conduct a permanent "test" =P |
| 03:05.31 | mmartinn | Being a call center, our needs were unique enough that they were happy to not support us |
| 03:05.42 | ManxPower | russellb: I think my clients would want more info than "uses Asterisk to some extent" |
| 03:05.49 | mmartinn | I couldn't believe how much they pay for Cisco =/ |
| 03:06.00 | ManxPower | they want press releases 8-) |
| 03:06.01 | bkruse_home | mmartinn: i think your looking for more custom developement then support |
| 03:06.27 | mmartinn | bkruse_home: Actually, we're honestly trying to just get it dialing |
| 03:06.35 | JT | cisco is just another proprietary phone system, that happens to use a bit of ethernet and IP |
| 03:06.54 | bkruse_home | mmartinn: ill do support for you :P |
| 03:07.15 | mmartinn | bkruse_home: We went from E&M wink -> PRI and that solved like 85% of our problems |
| 03:07.22 | bkruse_home | lol nice |
| 03:07.29 | bkruse_home | E&M *shudders* |
| 03:07.33 | bkruse_home | what equipment do you have? |
| 03:07.38 | ManxPower | mmartinn: I assume you checked your sync source for your cards? |
| 03:07.45 | mmartinn | Our old system was Nortel |
| 03:07.59 | mmartinn | ManxPower: With E&M, the timing was okay until you really loaded it down |
| 03:08.19 | ManxPower | mmartinn: I meant with the PRI |
| 03:08.33 | mmartinn | ManxPower: With PRI we don't have any more dropped calls period :) |
| 03:09.06 | mmartinn | ManxPower: It's calls where Dial() comes back very quickly even when there's no timeout -- I think I'm suspecting it's the other end though. |
| 03:09.36 | ManxPower | mmartinn: HANGUPCAUSE will tell you more. |
| 03:09.54 | mmartinn | The call center closed like 5 minutes before I added that to the dialplan :) |
| 03:10.01 | mmartinn | Tomorrow I will know lots more |
| 03:10.23 | mmartinn | I actually think the hardest part of this migration is managing expections about how the phone system should behave |
| 03:10.29 | ManxPower | mmartinn: I have written MANY lines of macros to deal with calls rejected by the telco one min and then accept it the the next try, so we retry the dial anytime a call fails except for a few HANGUPCAUSES |
| 03:11.00 | mmartinn | ManxPower: what's weird is that it doesn't fail right away, it fails after like... 4 rings |
| 03:11.22 | ManxPower | mmartinn: Do you use priindication=outofband? |
| 03:11.37 | ManxPower | or just not have one set at all |
| 03:11.38 | mmartinn | Tonight was our highest load yet, and one of the PRI D-channels disappeared a few times; I think I blame the CO |
| 03:12.02 | mmartinn | ManxPower: Ours isn't set at all (priindication) |
| 03:12.18 | ManxPower | mmartinn: what hangupcause are you getting on failed calls? |
| 03:12.40 | mmartinn | ManxPower: I just added that to the verbose output; I'll know once they start dialing tomorrow at 9am |
| 03:12.47 | mmartinn | I may be back :P |
| 03:13.09 | mmartinn | The D-Channel dropping today was definitely scary though |
| 03:13.28 | mmartinn | They had a bit of trouble bringing one of them up when they installed one of the circuits |
| 03:13.33 | bkruse_home | i bet ;X |
| 03:13.42 | bkruse_home | </3 nortel |
| 03:13.54 | mmartinn | Weirdly, the Nortel system <3'd the E&M wink |
| 03:14.32 | JT | it has probably undergone more testing and development than asterisk, for E&M |
| 03:14.40 | bkruse_home | oh, i knwo |
| 03:14.47 | mmartinn | Digium basically said the same thing -- PRI is more tested with Asterisk |
| 03:14.49 | bkruse_home | but i just dont link e&m |
| 03:14.54 | bkruse_home | pri is the awesome! |
| 03:15.00 | bkruse_home | and it def is |
| 03:15.49 | russellb | well if you have the option, you get a lot more cool stuff with a PRI :) |
| 03:15.59 | russellb | and you also get a warm feeling that you're not using lame analog signalling |
| 03:16.05 | bkruse_home | agreed |
| 03:16.29 | bkruse_home | you got a warm feeling when you do ztcfg and see all those lights go from red to green and scrolling good deeds are said on your asterisk console |
| 03:16.39 | JT | i don't think you can even get E&M trunks in Australia |
| 03:16.48 | bkruse_home | JT: doubt it |
| 03:16.59 | *** join/#asterisk squish102 (n=squish10@cpe-024-074-100-250.carolina.res.rr.com) |
| 03:17.06 | JT | doubt that you can, or can't? ;) |
| 03:18.18 | *** join/#asterisk mmartinn (n=martin@adsl-065-005-200-225.sip.gnv.bellsouth.net) |
| 03:18.21 | mmartinn | That was unfortunate... |
| 03:18.25 | mmartinn | I died ;( |
| 03:18.48 | ManxPower | <monty python>I doubt that you CAN and CANNOT use E&M in Aussieland!</monty python> |
| 03:18.50 | mmartinn | I was going to ask... is there any reason a D-Channel would go down after working a lot, that would be Asterisk's fault? |
| 03:19.04 | *** part/#asterisk squish102 (n=squish10@cpe-024-074-100-250.carolina.res.rr.com) |
| 03:19.13 | ManxPower | mmartinn: any time I've seen a D-channel go down it was the telco fault |
| 03:19.26 | mmartinn | ManxPower: I imagine they log these things? |
| 03:19.35 | mmartinn | ManxPower: So I can go complain :) ? |
| 03:19.41 | ManxPower | mmartinn: I doubt it |
| 03:20.02 | ManxPower | watchy our logs, then beat them to death with the aformentioned logs |
| 03:20.18 | mmartinn | Our PRI is long distance only, so testing it means grabbing 24 x $0.06/minute calls |
| 03:20.25 | mmartinn | To loop it up |
| 03:20.47 | JT | ManxPower: argh "aussieland" |
| 03:20.48 | mmartinn | Does your PRI show active channels in zttool? |
| 03:21.00 | mmartinn | err ManxPower: Does your PRI show active channels in zttool? |
| 03:21.10 | ManxPower | mmartinn: I strongly doubt it. |
| 03:21.14 | JT | mmartinn: D channels can go down also if your zap timing is bad or it is misconfigure |
| 03:21.18 | JT | misconfigured |
| 03:21.37 | mmartinn | JT: but could that really be transient? wouldn't that be like every call would die? |
| 03:21.55 | JT | D channels can flap |
| 03:22.13 | mmartinn | This happened once over heavy dialing for 4-5 hours |
| 03:22.31 | mmartinn | And it was the first D-channel, the one we use for timing |
| 03:22.37 | JT | you should check your zap timing accuracy, especially under load |
| 03:22.44 | JT | also ensure there is no irq sharing |
| 03:22.45 | ManxPower | Tell the people in your call center to stop generating so many calls. |
| 03:23.08 | mmartinn | JT: definitely no IRQ sharing already; how else do you test accuracy? won't zttest hurt the calls? |
| 03:23.26 | JT | zttest |
| 03:23.33 | mmartinn | Also, do you guys specify multiple PRIs for timing + backups, or just specify one? |
| 03:23.45 | JT | if you see any result under 99.97%, bad news |
| 03:24.10 | mmartinn | JT: with our test hardware, we saw it, but this new box we can't get to go below 99.9 |
| 03:24.24 | mmartinn | Won't that hurt/drop calls under heavy load, though? |
| 03:24.31 | JT | 99.9 is not specific enough |
| 03:24.33 | JT | it shouldn't |
| 03:24.41 | mmartinn | "Best: 100.000000 -- Worst: 99.987793 -- Average: 99.994451" |
| 03:25.03 | JT | watch the screen for any single result under 99.97% |
| 03:25.52 | mmartinn | 50 passes still perfect -- there's no load now though |
| 03:26.09 | JT | yeah |
| 03:26.18 | mmartinn | Is it better or worse to list multiple timing sources as backups? |
| 03:26.23 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:26.26 | JT | better |
| 03:26.36 | JT | usually |
| 03:26.59 | ManxPower | I figure that if more than 1 T-1 dies you have alot more serious problems than a no sync |
| 03:28.13 | mmartinn | So like: span=1,1,0,esf,b8zs / span=2,2,0,esf,b8zs / etc |
| 03:28.29 | JT | yes |
| 03:28.31 | mmartinn | Right now ours is span=1,1,0,esf,b8zs / span=2,0,0,esf,b8zs / etc |
| 03:32.05 | mmartinn | Any other recommendations? Do we like the zaptel drivers from 1.4 better than 1.2? |
| 03:32.21 | mmartinn | Or Asterisk/chan_zap better in 1.4 than 1.2, for that matter? |
| 03:32.45 | russellb | 1.4 ! |
| 03:33.14 | JT | ast 1.2 is more tested for production |
| 03:33.17 | russellb | but the community in general still mostly uses 1.2, i think |
| 03:33.24 | mmartinn | russellb: Can I gripe about the syntax of the AMI in 1.4's new commands? /ducks |
| 03:33.32 | russellb | people don't get as eager to update their phone systems as compared to some other apps, heh |
| 03:33.50 | russellb | mmartinn: sure, but I'll just agree and say that the whole AMI is gross. |
| 03:34.13 | mmartinn | russellb: I've been contributing to the Asterisk-Java project, and I'm the lucky person who's got the ticket assigned for 1.4 compatibility =/ |
| 03:34.30 | russellb | oh. :/ |
| 03:34.40 | russellb | Well it shouldn't be ... that much worse than 1.2 |
| 03:34.46 | russellb | if any at all |
| 03:34.52 | mmartinn | The Get/Update config stuff is weird |
| 03:35.04 | russellb | Ah, yes, it's interesting .. |
| 03:35.22 | *** join/#asterisk voltagex (n=voltagex@124-254-89-30-dsl.ispone.net.au) |
| 03:35.23 | russellb | then you'll just love the GetConfigJSON I put in trunk |
| 03:35.24 | theblue | ? |
| 03:35.26 | mmartinn | I don't like that a "key: value" has a key that is actually a value/argument for the command |
| 03:35.28 | theblue | voltagex: What are you doing here? |
| 03:35.38 | voltagex | what would cause externhost not to update, even with externrefresh set? |
| 03:35.46 | theblue | Oh. |
| 03:36.07 | mmartinn | russellb: JSON? Won't that only be for the web-exposed side? |
| 03:36.27 | russellb | yeah, it wouldn't be very useful otherwise :) |
| 03:36.32 | russellb | just cracking on weird formatting |
| 03:36.35 | mmartinn | hehe |
| 03:36.58 | mmartinn | I suppose we could consider moving to the web-service one, but that would require more work than just staying with the regular interface |
| 03:37.02 | mmartinn | =) |
| 03:37.10 | russellb | well, no, it wouldn't help you |
| 03:37.16 | russellb | the web thing is just manager over http ... |
| 03:37.28 | mmartinn | Right :) |
| 03:37.33 | russellb | you'll fight the same battles |
| 03:37.36 | mmartinn | I'm just saying... we could switch, but we'd be in the same spot |
| 03:37.37 | voltagex | ....... |
| 03:37.37 | *** join/#asterisk Sproket45 (n=sproket4@CPE001217214db0-CM00159a3c1146.cpe.net.cable.rogers.com) |
| 03:37.37 | mmartinn | =) |
| 03:37.42 | mmartinn | Plus, I get the feeling the AMI is just begging for a redesign |
| 03:37.47 | russellb | it is ... |
| 03:37.53 | mmartinn | Like it's time will come soon and it will get some major changes |
| 03:37.57 | russellb | it's a lot of code, though |
| 03:38.11 | mmartinn | I had to grep the source to figure out what changed in 1.4 :( |
| 03:38.12 | voltagex | can I *force* externhost to update? |
| 03:38.20 | mmartinn | I think it might be nice to make it all in the same spot |
| 03:38.28 | mmartinn | So at least it isn't handled everywhere through Asterisk |
| 03:38.38 | [TK]D-Fender | voltagex, Are you perhaps caching DNS somewhere? |
| 03:38.40 | mmartinn | So you could point to one spot which "is" the AMI |
| 03:38.48 | russellb | mmartinn: you could have just diff'ed the builtin in documenation for 'show manager command whatever' |
| 03:38.51 | voltagex | [TK]D-Fender: no |
| 03:39.03 | mmartinn | russellb: We'd have problems with accuracy in the past |
| 03:39.13 | mmartinn | Between the help and the actual sprintf |
| 03:39.17 | russellb | beautiful |
| 03:39.30 | voltagex | [TK]D-Fender: it's not updating *at all*, asterisk is still providing the private IP :/ |
| 03:39.34 | [TK]D-Fender | voltagex, pastbin the [general] section of your sip.conf |
| 03:39.42 | russellb | mmartinn: so you must think we're the best programmers ever :-p |
| 03:39.46 | [TK]D-Fender | ~pb |
| 03:39.57 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 03:40.15 | mmartinn | russellb: Actually, I managed just fine with most of it. The problem I'm actually stuck on (I could code around the command syntax) is the version-on-connect thing |
| 03:40.40 | russellb | trying to figure out which one you're connected to? |
| 03:40.45 | mmartinn | russellb: Where after a reload Asterisk answers the manager connections before it's done loading, and so is missing core commands |
| 03:40.57 | voltagex | [TK]D-Fender: one moment |
| 03:41.40 | mmartinn | russellb: We've been thrown for a loop for another way to detect version without relying on showing file versions |
| 03:42.08 | russellb | mmartinn: well what if we just add a GetVersion manager action? |
| 03:42.25 | Sproket45 | hi all, question about the -I option (announce user join/leave without review). I am running Asterisk 1.2.13 and it doens't seem to do anything, but the -i option works fine. ideas? thx |
| 03:42.38 | Sproket45 | sorry, i refer to the -I option in MEETME |
| 03:42.43 | mmartinn | russellb: Will GetVersion work when Asterisk reloads and the manager accepts connections before "core <..." commands work/are loaded? |
| 03:43.07 | mmartinn | russellb: We'd be happy to use a command if we could get a guarantee it would exist before the manager accepts our connection |
| 03:43.19 | mmartinn | s/a command/any command/ |
| 03:43.37 | voltagex | [TK]D-Fender: http://channels.debian.net/paste/paste |
| 03:43.47 | russellb | mmartinn: well, you tell me what is better. I could do that and ensure it is there, or I can just make it so connections aren't accepted until after initializing the rest of asterisk |
| 03:44.08 | russellb | well, i don't know ... it's tough |
| 03:44.22 | russellb | when you initialize the rest of asteirsk, it wants to fire events off when it starts ... |
| 03:44.23 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 03:44.24 | mmartinn | russellb: I understand that with the current way of things, it's hard |
| 03:44.35 | mmartinn | russellb: I looked through the source and realized that |
| 03:44.42 | mmartinn | russellb: with the load order requirements |
| 03:44.52 | mmartinn | russellb: And wanting the manager to be available to other modules |
| 03:44.56 | russellb | yeah, ok. Well if the GetVersion command is in manager.c, it should be fine |
| 03:45.00 | ManxPower | mmartinn: just add a sleep(120) or whatever at the beginning of the manasger module 8-) |
| 03:45.30 | voltagex | wtf |
| 03:45.36 | voltagex | it didn't paste the number |
| 03:45.43 | [TK]D-Fender | .... |
| 03:45.45 | mmartinn | ManxPower: Our library will have to sleep 120 seconds every time it connects? =/ |
| 03:45.47 | voltagex | http://channels.debian.net/paste/5858 |
| 03:45.58 | voltagex | xD |
| 03:46.10 | Corydon76-home | mmartinn: uh, what core command doesn't exist prior to the manager running? |
| 03:46.14 | mmartinn | russellb: I am not really looking for an immediate change or something -- I was mostly just curious to see if I was missing something in how it was handled |
| 03:46.17 | ManxPower | mmartinn: I was thinking of having manager sleep for a while before it accepts connections after a load/reload |
| 03:46.27 | ManxPower | seems like a rage condition waiting to happen, but it might help |
| 03:46.38 | russellb | mmartinn: no, not that I know of. I guess there isn't a really easy way |
| 03:46.40 | ManxPower | rage == race |
| 03:46.41 | Corydon76-home | Pretty much anything with a prefix of "core" should be in the core and thus immediately available |
| 03:46.52 | russellb | the web server reports the asterisk version, but you can't count on that running |
| 03:47.09 | russellb | Corydon76-home: depends on if they get registered before ast_manager_init() in manager.c |
| 03:47.10 | [TK]D-Fender | voltagex, your register should appear AFTER. Also you are missing "nat=yes", "canreinvite=no", "localnet=192.168.0.0/255.255.0.0" <- sample , and a few other "sanity" options. |
| 03:47.22 | mmartinn | Corydon76-home: If you connect immediately after a reload, AMI says that "show version file pbx.c" is a bad command |
| 03:47.48 | Corydon76-home | mmartinn: you mean after a restart |
| 03:47.48 | [TK]D-Fender | voltagex, the localnet sample I gave you is good for most class-c private subnets |
| 03:47.58 | mmartinn | Corydon76-home: correct |
| 03:48.07 | mmartinn | Corydon76-home: if you ask it to reload |
| 03:48.19 | voltagex | [TK]D-Fender: I accidentally wiped my config this morning |
| 03:48.21 | *** join/#asterisk e-milioX (n=emilio@adsl-8-70-169.mia.bellsouth.net) |
| 03:48.27 | Corydon76-home | mmartinn: a reload and a restart are two distinct events |
| 03:48.32 | [TK]D-Fender | voltagex, That'll teach you to BACKUP |
| 03:48.58 | mmartinn | Corydon76-home: I suppose I am referring to "restart now" |
| 03:49.04 | [TK]D-Fender | voltagex, How much more of you brain did it take with it? :) These are things you should still know to correct... |
| 03:49.06 | Corydon76-home | A reload should never make CLI commands go away |
| 03:49.25 | voltagex | [TK]D-Fender: I'm still learning...the hard way |
| 03:49.43 | [TK]D-Fender | voltagex, .... |
| 03:49.46 | [TK]D-Fender | ~osmosis |
| 03:49.54 | jbot | [osmosis] the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
| 03:49.54 | e-milioX | Hello guys |
| 03:49.54 | voltagex | ~alcohol |
| 03:49.56 | jbot | alcohol is probably the answer to most of lifes problems.., or a good excuse to alias rm='rm -i', or a cause of "rm -rf . / "'s, or the cause of, and solution to, all of lifes problems, or ask me about ambrosia |
| 03:50.00 | mmartinn | I ended up adding some sleep time in the thread of Asterisk-Java if the command isn't found, and then after 5 times, it assumes Asterisk version "very low" |
| 03:50.03 | Corydon76-home | mmartinn: why are you doing a 'restart now' in your normal course of events? |
| 03:50.27 | Sweeper | testing system? |
| 03:50.30 | Corydon76-home | That should be an exceptional condition |
| 03:50.33 | mmartinn | Corydon76-home: I'm not, but it IS a race condition and once it happens, you have no way of determining the version upon reconnect |
| 03:50.56 | [TK]D-Fender | voltagex, Good... now apply my suggestions, reload SIP, test and report back |
| 03:51.10 | mmartinn | Corydon76-home: We have a library that runs via the AMI, and we can't just ask it to reconnect a couple times to make sure it got the version right if Asterisk somehow restarted |
| 03:51.13 | Corydon76-home | mmartinn: so if you get an error, sleep and retry |
| 03:51.33 | mmartinn | Corydon76-home: That is what we are doing, but I was under the impression that early-early versions of Asterisk really just don't have the command |
| 03:51.35 | file | it's show version files pbx.c btw |
| 03:51.45 | russellb | mmartinn: that's true |
| 03:52.18 | russellb | mmartinn: but they have "show version" as a CLI command, can you run CLI commands over the manager in 1.2? |
| 03:52.25 | Corydon76-home | 'show uptime' has always existed, though |
| 03:53.43 | mmartinn | russellb: I believe so -- will the version number be as accurate as the svn revision number for differences in behavior? |
| 03:53.50 | theblue | ManxPower: I'm stealing that idea, you know. |
| 03:54.01 | ManxPower | theblue: Someone needs to. |
| 03:54.05 | theblue | ManxPower: Done. |
| 03:54.35 | russellb | mmartinn: yes, but only if it comes from an svn checkout |
| 03:54.42 | russellb | mmartinn: otherwise it will have the real version number |
| 03:54.49 | bkw__ | you have been able to run cli commands over the manager for some time |
| 03:54.49 | mmartinn | Corydon76-home: show uptime wouldn't tell us the version; and it would be a kludge if we used it to see if asterisk was finished loading -- some person somewhere might have a really big dialplan or something :P |
| 03:54.50 | bkw__ | before 1.2 |
| 03:54.52 | russellb | mmartinn: in any case, you get a nique ID |
| 03:55.14 | russellb | bkw__: ah, I couldn't remember |
| 03:55.15 | e-milioX | years ago i bought some single analog line port. Dont remeber exact model. The question is can they be used as a timing source ?? |
| 03:55.16 | mmartinn | russellb: I will look into that; that's a good tip then, if "show version" is loaded before AMI opens connections |
| 03:55.34 | voltagex | [TK]D-Fender: I get <1sec of audio now, no errors from console though |
| 03:55.39 | voltagex | better than no audio |
| 03:55.58 | [TK]D-Fender | voltagex, What ports have you forwarded to *? |
| 03:56.02 | e-milioX | years ago i bought some DIGIUM single por 1 analog line card i mean |
| 03:56.06 | [TK]D-Fender | voltagex, And is it picking up your IP? |
| 03:56.16 | voltagex | [TK]D-Fender: it's in the dmz |
| 03:56.26 | russellb | mmartinn: nope, show version will have the same race condition |
| 03:56.43 | mmartinn | russellb: okay, then I'll just hold on to it :) |
| 03:56.45 | russellb | but it looks like an easy fix |
| 03:57.08 | russellb | if you look in asterisk.c, move the call to load_pbx() to before the call to init_manager() |
| 03:57.13 | voltagex | [TK]D-Fender: yes, it's picking up my public ip |
| 03:57.27 | *** join/#asterisk asteriskguy (n=learnast@cpe-75-80-111-113.socal.res.rr.com) |
| 03:57.33 | mmartinn | russellb: When I looked at it, it made me think that load_pbx expected init_manager to have been done |
| 03:57.41 | [TK]D-Fender | voltagex, pastebin your new [general] section, and any sip peers involved in your test (minus passwords) and the CLI output at verbose 10 and SIP debug enabled. |
| 03:57.59 | Corydon76-home | "show version" is an immutable commands. It exists immediately upon start. |
| 03:58.21 | russellb | mmartinn: Well, init_manager doesn't have to done before attempting to register manager actions, but it probably does before creating events |
| 03:58.40 | mmartinn | russellb: Hmmm... okay, I thought before both |
| 03:58.40 | russellb | it is? |
| 03:58.50 | russellb | i thought file said it was in pbx.c |
| 03:59.02 | Corydon76-home | Ditto for "show uptime". Neither "show version" nor "show uptime" will ever be unavailable, no matter how quickly you connect to the manager port after restart |
| 03:59.04 | file | russellb: ha no |
| 03:59.20 | file | I said what the correct CLI command was for getting the revision number of a specific file |
| 03:59.24 | russellb | ok, so this whole conversation is moot |
| 03:59.31 | russellb | bah |
| 03:59.43 | mmartinn | it seems so... |
| 03:59.43 | file | now if you try to get the version of a file that isn't loaded yet... then no soup for you |
| 03:59.52 | Corydon76-home | russellb: they both exist in cli.c, in the builtins struct |
| 04:00.00 | russellb | cool |
| 04:00.16 | voltagex | [TK]D-Fender: sip.conf here http://channels.debian.net/paste/5861 |
| 04:00.31 | russellb | ok, so using show version for 1.2, and core show version 1.4 and beyond should always work |
| 04:00.33 | mmartinn | okay, so I suppose Asterisk-Java will be using that; the other dev gave me the impression the "show version" wasn't granular enough for certain command syntax changes |
| 04:00.50 | russellb | that will always be unique |
| 04:01.06 | mmartinn | I'll explore more though; you've given me plenty to run a test or two with |
| 04:01.19 | [TK]D-Fender | voltagex, You did not do the "nat=yes" ,"canreinvite=no" that I told you to, nor did you more the register line BELOW all of the [general] settings..... |
| 04:01.31 | voltagex | ? |
| 04:01.37 | [TK]D-Fender | voltagex, Correct, apply, test and repastebin on failure |
| 04:01.41 | voltagex | canreinvite is there |
| 04:01.43 | mmartinn | russellb, Corydon76-home -- thanks :) |
| 04:01.49 | russellb | you're welcome |
| 04:01.59 | [TK]D-Fender | voltagex, under [general] <------- |
| 04:02.13 | voltagex | oh crap. wrote allowreinvite |
| 04:02.14 | mmartinn | I'm off to retire so the call center people can chase me around with problems all morning! :) |
| 04:02.33 | mmartinn | Even though I'm just moonlighting as a sysadmin... I'm actually a developer for them. |
| 04:02.37 | mmartinn | =/ |
| 04:02.49 | [TK]D-Fender | voltagex, Location COUNTS. And "nat=yes" <- |
| 04:02.55 | [TK]D-Fender | voltagex, Pay attention please. |
| 04:03.25 | voltagex | [TK]D-Fender: register doesn't go in general? |
| 04:03.34 | [TK]D-Fender | voltagex, BELOW it. |
| 04:03.49 | voltagex | it is. |
| 04:03.55 | mmartinn | Corydon76-home: FYI, I just ran a little test, and got "No such command 'show version' (type 'help' for help)" after a restart in 1.2 |
| 04:04.25 | mmartinn | Corydon76-home: err 1.4 |
| 04:04.33 | mmartinn | Corydon76-home: I'm on the test box which is 1.4, I forgot ;) |
| 04:04.39 | [TK]D-Fender | voltagex, I mean below the settings in there. after all the "parm=value" stuff and before your first real peer/friend/usr entry |
| 04:04.47 | file | deprecation commands... |
| 04:05.05 | voltagex | ok |
| 04:05.47 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 04:05.48 | Corydon76-home | mmartinn: that's correct, "show version" is not immutable in 1.4 |
| 04:06.03 | mmartinn | Corydon76-home: Oh... I thought that's what you were saying above... |
| 04:06.07 | voltagex | ok, recv audio now |
| 04:06.16 | Corydon76-home | I was looking in 1.2 |
| 04:06.36 | Corydon76-home | It was removed in 1.4, because the command was renamed |
| 04:06.36 | mmartinn | Corydon76-home: oh, okay |
| 04:06.39 | voltagex | and transmit too, thanks |
| 04:06.52 | voltagex | going to make a backup of the conf now |
| 04:07.09 | Corydon76-home | So waiting after reconnect is going to be the best way forward |
| 04:07.33 | mmartinn | Corydon76-home: okay, that's where I'm at too then. nevermind :) |
| 04:07.47 | Corydon76-home | and don't worry about large dialplans... the core is always loaded before any modules |
| 04:08.30 | mmartinn | err okay. Well, thanks again; nite! |
| 04:08.39 | mmartinn | And thanks everyone else :) |
| 04:09.01 | [TK]D-Fender | voltagex, Good to hear |
| 04:09.42 | e-milioX | Anybody knows if the old single analog port cards fxo or fxs from digium can be used as a timing source ? |
| 04:10.28 | e-milioX | I dont have the box in front of me so i cant test but need to have it working tomorrow |
| 04:11.06 | asteriskguy | does anyone know if there's a script to help change the config files for the polycom phones? |
| 04:13.32 | JunK-Y | asteriskguy: yes, and its called the polycom web interface :) |
| 04:15.20 | asteriskguy | lol....I meant bulk provisioning |
| 04:15.43 | Corydon76-home | Write your own |
| 04:16.00 | [TK]D-Fender | asteriskguy, VI :) |
| 04:16.08 | asteriskguy | :( I do not know how to code in anything but Basic |
| 04:16.15 | asteriskguy | and that was 8 years ago |
| 04:16.19 | Corydon76-home | Learn. |
| 04:16.38 | [TK]D-Fender | PHP si easy stuff..... |
| 04:16.40 | Corydon76-home | It's really not all that difficult |
| 04:16.41 | JunK-Y | [TK]D-Fender: vim! |
| 04:16.50 | asteriskguy | Enventually yes, but we need to push this out in a few days |
| 04:16.59 | [TK]D-Fender | JunK-Y, mistar fan-c-pants! |
| 04:17.05 | Corydon76-home | Have some coffee |
| 04:17.13 | dj-fu | mm, coffee > * |
| 04:17.23 | asteriskguy | vim, nano, vi, emacs.....I'm looking to see if anyone has written a script before that could edit their .cfg files |
| 04:17.28 | [TK]D-Fender | asteriskguy, load chan_rality.so |
| 04:17.38 | [TK]D-Fender | chan_reality.so* |
| 04:18.00 | JunK-Y | asteriskguy: i doubt so, since ya asked to modify config files, and this is really general |
| 04:18.05 | Corydon76-home | I've written a script before that does that. However, it can only be gotten by buying a system from the company for which I work |
| 04:18.14 | [TK]D-Fender | asteriskguy, A quick SED or few to search&replce keyed field, and scripting is quick from there. |
| 04:18.58 | asteriskguy | SED? |
| 04:18.58 | [TK]D-Fender | asteriskguy, "man sed" |
| 04:18.58 | Corydon76-home | sed(1) - Stream editor |
| 04:19.04 | asteriskguy | hmm....ok...definitely need to look into that |
| 04:19.05 | [TK]D-Fender | asteriskguy, Time to learn the basic GNU tools out there... |
| 04:19.40 | JunK-Y | :1,$s/foo/bar/g |
| 04:20.00 | asteriskguy | I'm sorry, I didn't mention, I'm a noob when it comes to linux |
| 04:20.13 | [TK]D-Fender | regex / perl / bash / most other l33t Linux-y stuffs > me |
| 04:20.22 | CunningPike | asteriskguy: http://www.grymoire.com/Unix/Sed.html |
| 04:20.41 | [TK]D-Fender | asteriskguy, time to see how fast you can learn to swim, the water's rushing in ;) |
| 04:21.05 | [TK]D-Fender | asteriskguy, Is this a 1-shot config you've got to do, or a continual interface? How many phones? |
| 04:21.18 | asteriskguy | yeah, I'm trying, and it's hard to learn linux and learn to work with asterisk at the same time |
| 04:21.26 | asteriskguy | about 200 phones for the company |
| 04:21.28 | [TK]D-Fender | asteriskguy, And you really didn't need tomention, it kinda stood out. |
| 04:21.54 | e-milioX | nobody loves e-milioX |
| 04:22.23 | asteriskguy | I'm about 3 weeks into Linux and same with * |
| 04:22.41 | *** join/#asterisk annonimous (n=annonimo@189.136.65.124) |
| 04:22.50 | asteriskguy | Oh, and thanks CunningPike and TKD-Fender, I'm reading up on SED right now |
| 04:22.58 | annonimous | hello |
| 04:24.02 | Corydon76-home | e-milioX: Digium has never sold a single port FXS card |
| 04:24.30 | Corydon76-home | e-milioX: but yes, the X100P (FXO) can be used as a timing source |
| 04:25.09 | e-milioX | Corydon76-home: Thanks. I can rest now knwing that meetme will work tomorrow |
| 04:25.11 | asteriskguy | another question, if I buy a ATA device, can I connect that to a fax machine and direct multiple DIDs to that same fax machine |
| 04:25.12 | asteriskguy | ? |
| 04:25.37 | asteriskguy | Theoretically it should work right? Since it puts out an analog signal |
| 04:25.50 | Corydon76-home | ATA devices do not work reliably for fax |
| 04:26.13 | e-milioX | Corydon76-home: I have 1 port fx0 and 4 port fxs |
| 04:26.19 | *** join/#asterisk bawb2 (n=bawb2@ip51051.estcmp.ku.edu) |
| 04:26.50 | e-milioX | which would be "better" for timming ( lots of meetme for vicidial) |
| 04:27.11 | Corydon76-home | Neither is better |
| 04:27.12 | asteriskguy | oh, that's bad news for me then. What can I do to fix this. We do not have an extra slot for another card for FXS |
| 04:27.23 | Corydon76-home | They both produce 1000Hz timing signal |
| 04:28.10 | e-milioX | Corydon76-home: help very much apreciated. thanks again. |
| 04:28.39 | Corydon76-home | What might be better, in some circumstances is a T1 card, connected to a telco. |
| 04:29.02 | Corydon76-home | Then the 1000Hz signal is coordinated with other telcos |
| 04:29.35 | Corydon76-home | with an analog card, the timing is simply generated at the hardware level, directly on the card itself |
| 04:29.44 | e-milioX | Corydon76-home: "Much" better? will it make a diff for 60 phones with 60 conferences dialing out thorugh IAX ? |
| 04:29.57 | Corydon76-home | Nope |
| 04:30.22 | Corydon76-home | It's simply a matter of whether you need a single source or a coordinated source |
| 04:31.10 | e-milioX | I would say that for all practical purposes on a pure iax/sip conf (no telco) single is enough right ? |
| 04:31.47 | Corydon76-home | If you're not using a telco, then you do not need coordinated timing |
| 04:32.33 | JT | it's not super critical |
| 04:32.57 | e-milioX | I saw that there was a bug for ztdummy with IAX, I just wasn't able to make it work |
| 04:33.00 | JT | T and E series bearers are plesiochronous communications technologies |
| 04:33.18 | Sweeper | oo, big words |
| 04:33.32 | JT | they don't require the whole network to be synchronised, unlike OC/STM carriers |
| 04:34.52 | e-milioX | very nice word i learned today: plesiochronous |
| 04:35.08 | JT | yeah it's a cool word :) |
| 04:36.35 | e-milioX | good nigh everybody.. thanks for the transmission of knowledge... karma will reward you |
| 04:37.25 | *** part/#asterisk e-milioX (n=emilio@adsl-8-70-169.mia.bellsouth.net) |
| 04:37.54 | [TK]D-Fender | Everybody : My karma ran over your dogma :D |
| 04:39.46 | adamatomical | is there a way to register with an sip host before a call? |
| 04:40.10 | adamatomical | i figured out the error is 202 |
| 04:40.20 | JunK-Y | adamatomical: yes, see sip.conf |
| 04:40.20 | JT | use the register directive |
| 04:40.45 | adamatomical | i did use the register directive |
| 04:42.59 | *** join/#asterisk Landrocker (n=Anton@rimmfpc.clients.fishnet.net.nz) |
| 04:43.34 | [TK]D-Fender | adamatomical, pastebin the call with verbose 10, and SIP debug enabled. |
| 04:43.40 | [TK]D-Fender | ~pb |
| 04:43.53 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 04:43.54 | sevard | replace free with stolen and you're golden |
| 04:45.10 | flenders | JT: what model is that avaya phone you got? |
| 04:46.44 | JT | 4621SW iirc |
| 04:48.14 | *** join/#asterisk bird_of_Luck (n=melifaro@secured.by.ipfw.ru) |
| 04:48.26 | *** join/#asterisk dseeb_ (n=dcb@CPE-58-169-110-130.vic.bigpond.net.au) |
| 04:48.57 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 04:49.02 | adamatomical | http://pastebin.ca/413170 |
| 04:49.43 | flenders | JT: is it sip? |
| 04:51.35 | adamatomical | Host Username Refresh State |
| 04:51.36 | adamatomical | sip.inphonex.com:5060 6481000 105 Registered |
| 04:54.58 | [TK]D-Fender | adamatomical, hrm. So you connect, you issue the wait and the call dies during the wait? |
| 04:55.15 | adamatomical | I've recorded the message, because RTP audio is being sent |
| 04:55.19 | adamatomical | it says error 205 |
| 04:55.46 | JT | flenders: yes |
| 04:55.52 | adamatomical | http://www.inphonex.com/knowledgebase/error-notifications-answers.php?action=view&kb_id=6&tag_id= |
| 04:56.00 | flenders | JT: how much you paid for that again? |
| 04:56.10 | JT | you can get avaya phones that are not sip also |
| 04:56.14 | JT | $130 i think ;) |
| 04:56.27 | flenders | ebay? |
| 04:56.31 | JT | yeah |
| 04:56.34 | flenders | ebay .au? |
| 04:56.37 | JT | yes |
| 04:56.41 | flenders | nice |
| 04:56.46 | JT | it doesn't have DC in |
| 04:56.48 | JT | only does PoE |
| 04:57.03 | flenders | so you haven't plugged it in yet? |
| 04:57.30 | JT | yeah |
| 04:58.01 | flenders | just get one of those little netgear poe switches so you can test it |
| 04:58.10 | [TK]D-Fender | adamatomical, These guys seem psycho-picky |
| 04:58.34 | adamatomical | life is a struggle |
| 04:59.35 | adamatomical | i'm starting to get really pissed though |
| 04:59.35 | sevard | Brian702: those aren't girls. |
| 05:00.45 | JT | sevard: who are you talking to? |
| 05:00.50 | sevard | CHRIST |
| 05:00.56 | JT | jesus |
| 05:01.07 | sevard | tap dancers. |
| 05:02.06 | *** join/#asterisk lokkju_wrk_ (n=lokkju@unaffiliated/lokkju) |
| 05:03.52 | [TK]D-Fender | sevard, the voices are escaping again.... |
| 05:04.02 | sevard | DOOOMMED |
| 05:04.06 | sevard | The sky is blue |
| 05:04.09 | sevard | and all the leaves are green |
| 05:04.17 | sevard | My heart feel just like a BAKED POTATO |
| 05:04.36 | *** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au) |
| 05:05.31 | voltagex | on a phone without xfer cabability, is there a way to transfer a call? |
| 05:08.07 | [TK]D-Fender | sevard, Don't forget the sour-cream, onions, cheddar cheese , and bacon :) |
| 05:08.28 | sevard | [TK]D-Fender: you're a failure for not knowing what I'm singing |
| 05:08.57 | voltagex | isn't transfer #1 ? |
| 05:09.06 | voltagex | doesn't seem to work though |
| 05:10.32 | [TK]D-Fender | voltagex, "show application dial" |
| 05:11.30 | *** join/#asterisk voiper1 (n=luke@ozvoip.dsl.onthenet.net) |
| 05:11.34 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 05:11.34 | *** mode/#asterisk [+o denon] by ChanServ |
| 05:11.34 | JT | flenders: netgear :( |
| 05:11.41 | [TK]D-Fender | sevard, From what I've seen of the rest, I'm not missing much :) |
| 05:11.54 | sevard | [TK]D-Fender: do you like matt stone and trey parker? |
| 05:12.35 | [TK]D-Fender | sevard, You mean the illustrious makers of Baseketball? BLEH. Down with South Park! |
| 05:12.48 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 05:13.56 | [TK]D-Fender | sevard, uninspiring vulgar humour... |
| 05:14.15 | sevard | [TK]D-Fender: cannible the musical is one of the greatest movies ever made |
| 05:14.17 | [TK]D-Fender | sevard, Good for a few laughs if it passes a screen in front of you, but nothing to linger on. |
| 05:17.51 | *** join/#asterisk iratsu (n=iratsu@modemcable204.153-70-69.mc.videotron.ca) |
| 05:23.14 | flenders | JT: cheap |
| 05:23.48 | *** part/#asterisk adamatomical (n=adamatom@24-116-67-252.cpe.cableone.net) |
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| 05:24.59 | JT | and nasty ;) |
| 05:25.37 | flenders | I wouldn't use on production... but for testing... why not? |
| 05:26.22 | JT | let me see just how cheap it is ;) |
| 05:27.02 | *** join/#asterisk yidiyuehan (n=yidiyueh@58.185.253.70) |
| 05:27.33 | yidiyuehan | hello,guys, any one can tell me what is the functions of dnsmgr.conf? |
| 05:28.35 | yidiyuehan | i enable it then my dundi works fine, but the IAX trunk with Voicepulse broke, if i disable it the Voicepulse trunk works agan, but the dundi with Dynamic DNS fails. |
| 05:31.26 | JT | flenders: any idea on price? |
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| 05:46.31 | flenders | JT: yeah, around 150 for a little one |
| 05:46.51 | flenders | JT: 8 ports (4 poe) |
| 05:47.32 | *** part/#asterisk voiper1 (n=luke@ozvoip.dsl.onthenet.net) |
| 05:49.04 | JT | hrm non on ebay |
| 05:49.06 | JT | none |
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| 05:53.36 | zeeesh | hi |
| 05:55.55 | flenders | clearnet.com.au has it for 180 |
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| 05:57.03 | flenders | sorry, their website says 203, but jerry sent 180 on an e-mail a couple of days ago |
| 05:59.50 | JT | flenders: ah ok |
| 06:00.01 | JT | looking at some cisco options |
| 06:00.12 | flenders | and those linksys 8 port poe for about 350 |
| 06:00.19 | flenders | but you hate cisco |
| 06:00.20 | JT | ouch |
| 06:00.32 | JT | well as a rule, especially for voip stuff |
| 06:00.39 | JT | but for switches they can be acceptable |
| 06:01.26 | flenders | I like their stuff |
| 06:01.42 | flenders | and I think their phones look niiiiiice |
| 06:01.43 | JT | their voip stuff and firewalls are rubbish |
| 06:01.45 | flenders | smoooooth |
| 06:01.55 | flenders | firewalls are shit |
| 06:02.06 | flenders | pix of shit |
| 06:02.14 | JT | they look like a lot of recent made in china trendy crap, crappy fake silver plastic |
| 06:02.28 | JT | silver plastic looks so lame |
| 06:04.21 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:04.29 | flenders | I like it |
| 06:04.40 | flenders | it matches my dell desktop |
| 06:04.48 | flenders | and my dell monitor stand |
| 06:04.54 | flenders | and my headset |
| 06:05.05 | flenders | oh god! my desk is all matching |
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| 06:09.37 | JT | most of the dell stuff these days is black... |
| 06:10.01 | *** join/#asterisk hijacked (i=XSp4@cerebus.clandestineresearch.com) |
| 06:10.07 | flenders | I think most of is silver |
| 06:11.09 | flenders | all server face plates are silver |
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| 06:11.11 | flenders | the new dimension Cxx desktops are silver |
| 06:11.11 | flenders | inspiron laptops are silver |
| 06:13.33 | flenders | see? silver is the way! |
| 06:13.34 | JT | their monitors are black |
| 06:13.34 | JT | most desktops are |
| 06:13.34 | flenders | the monitor pedestal is silver |
| 06:13.34 | JT | servers are, except for the cheaparse shit covers they're putting on the front now |
| 06:13.34 | flenders | hahahha |
| 06:13.34 | JT | i'm not sure if i'd buy a new dell server anymore |
| 06:13.35 | JT | too much plastic |
| 06:13.35 | bjweeks | I'm running 1.2.17 on an embedded system and asterisk is failing to make asterisk.ctl when it starts up. It runs file with -c and when I start it in background mode it is still running just without the .ctl file. what could be wrong? |
| 06:13.35 | flenders | yeah, we don't have the covers |
| 06:13.35 | JT | among other things |
| 06:13.35 | JT | those covers were metal on the 2600 series |
| 06:13.35 | JT | 2800 they went plastic, that just pop off, don't even hinge |
| 06:14.03 | flenders | we have a few 2850s |
| 06:14.04 | flenders | they're pretty decent boxes |
| 06:14.19 | flenders | all our servers at rackspace are 2850s as well |
| 06:15.07 | flenders | what server would you buy in the same range? |
| 06:15.12 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 06:17.37 | JT | depends who for |
| 06:17.50 | JT | if brand names were the only option: ibm or hp |
| 06:17.59 | JT | i really like ibm but they're not that cheap |
| 06:18.15 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 06:18.22 | JT | if it's shoestring and warranty isn't greatest concern, something supermicro based |
| 06:19.11 | denon | psha |
| 06:19.14 | denon | I like supermicro |
| 06:19.21 | JT | supermicro is really innovative atm |
| 06:19.28 | denon | atm? they have been for years |
| 06:19.33 | JT | well |
| 06:19.42 | denon | did you know that Intel hires them to do their R&D boards on new chipsets? |
| 06:19.50 | denon | because they do better boards than their own designers :) |
| 06:20.04 | JT | they have a dual system dual xeon machine with 2 HDs a system (4 total) in 1RU now |
| 06:20.08 | denon | they use supermicro boards to test new chipset and cpu stuff |
| 06:20.10 | JT | heh |
| 06:20.41 | denon | density isnt real important to me - we have plenty of space in the noc |
| 06:20.52 | denon | I'm more concerned with stability and performance |
| 06:21.16 | JT | well i'm looking at datacentre applications |
| 06:21.24 | Tene | supermicro boards are great. |
| 06:22.00 | denon | JT: well, yeah, it only matters if you dont run the noc yourself :) |
| 06:22.00 | denon | ie: if you've gotta pay someone per U |
| 06:22.12 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 06:22.44 | JT | sure |
| 06:22.44 | JT | you mean datacentre not NOC |
| 06:23.00 | JT | imho a noc is just where you have screens that tell you your dc is screwed |
| 06:23.03 | JT | or working |
| 06:23.16 | denon | ours tends to be a merge |
| 06:23.36 | JT | yeah, it can be |
| 06:23.49 | JT | anyway, for a lot of applications these days, if they're Internet based |
| 06:23.50 | denon | but yeah, technically ops is kinda separated from the floor |
| 06:23.56 | JT | it's crazy to run your own datacentre |
| 06:24.21 | JT | you can't make the same economies of scale as a good datacente unless you're huge |
| 06:24.21 | denon | to each his own |
| 06:24.34 | denon | the upside, is you get to enjoy redundant power/etc :) |
| 06:24.37 | JT | and even if you're huge, you usually buy a bit of third part DC space anyway |
| 06:24.51 | JT | redundant power, cooling, fire suppression, connectivity |
| 06:25.02 | JT | connectivity is a big one |
| 06:25.21 | denon | anywho, tis late here |
| 06:25.35 | denon | looks like it's about time to knock off in your neck of the woods too |
| 06:25.55 | JT | heh, sooner for some than others ;) |
| 06:26.00 | denon | mhmm |
| 06:26.10 | denon | g'nite |
| 06:27.47 | JT | night |
| 06:31.02 | bjweeks | is any kernel option required to use 'asterisk -r' to connect to the running process? |
| 06:31.48 | JT | no... |
| 06:32.41 | bjweeks | bah, that was my last grasp for straws, I have no idea why this isn't working |
| 06:33.04 | JT | is asterisk running? |
| 06:33.04 | *** join/#asterisk cnbrk (n=cnbrk@35.11.193.34) |
| 06:33.06 | cnbrk | hi |
| 06:33.17 | bjweeks | JT: yes, confirmed with ps |
| 06:33.26 | cnbrk | i have a damn problem and i can't solve it. i used to work with asterisk with my old server |
| 06:33.34 | JT | are you trying to reattach as a suitably privedleged user? |
| 06:33.35 | cnbrk | and i ported the config files and other stuff to my new server |
| 06:33.48 | cnbrk | but now when i call somebody, for the first time i hear the sound. then no sound at all |
| 06:33.52 | bjweeks | JT: asterisk running as root, I'm running as root |
| 06:33.55 | cnbrk | what can be the problem please help |
| 06:34.06 | JT | bjweeks: unusual |
| 06:34.21 | JT | maybe asterisk is hung |
| 06:34.31 | bjweeks | JT: it is a MIPS system so that could be it |
| 06:34.40 | JT | try killing it and doing asterisk -cvvvvv |
| 06:35.28 | bjweeks | starts up and I can run commands |
| 06:35.35 | cnbrk | can you guys help me, i'll appreciate and give 25$ |
| 06:35.43 | bjweeks | asterisk is never making the asterisk.ctl file |
| 06:35.49 | bjweeks | cnbrk: did you check your logs? |
| 06:35.59 | cnbrk | i am connected to cli with debug |
| 06:36.06 | cnbrk | nothing seems to be wrong |
| 06:36.15 | cnbrk | but i don't think asterisk operates the codec normally |
| 06:37.34 | bjweeks | JT: do you have any idea what asterisk.ctl should contain? or it a device that gets connected to? |
| 06:38.14 | JT | not sure from memory |
| 06:38.15 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 06:38.15 | *** mode/#asterisk [+o denon] by ChanServ |
| 06:39.03 | bjweeks | JT: at this point I'm thinking of filing a bug, but not sure I have enough info to fix it |
| 06:39.17 | JT | bjweeks: did you try what i suggested? |
| 06:39.28 | bjweeks | JT: yes, it runs file |
| 06:39.50 | bjweeks | oh, sorry forgot to say something, though I did |
| 06:40.01 | JT | it runs file ;) |
| 06:40.13 | bjweeks | fine* |
| 06:40.21 | bjweeks | but I wish my box ran one of the devs |
| 06:40.29 | cnbrk | anyone who solves the problem can ask for everything |
| 06:44.24 | JT | cnbrk: no-one will be able to solve the problem without additional info |
| 06:45.22 | cnbrk | so |
| 06:45.30 | cnbrk | now i tried disabling gsm |
| 06:45.39 | cnbrk | and forced asterisk to use ulaw only |
| 06:46.05 | cnbrk | and when i called the other phone, when i answer it, it automatically hungup and on cli it said everyone is busy |
| 06:51.00 | JT | lol@evaluation copy |
| 06:51.48 | bjweeks | wow |
| 06:52.47 | JT | ? |
| 06:53.35 | bjweeks | that is funny |
| 06:53.48 | JT | yeah i've seen it before |
| 06:54.02 | JT | penalty for using shit non-free software ;) |
| 06:54.09 | JT | all good irc clients are FOSS |
| 06:54.23 | bjweeks | I'm just using the KDE one |
| 06:54.34 | JT | irssi :) |
| 06:55.18 | bjweeks | irssi is alright, if I'm in a shell mode |
| 06:55.35 | JT | i run irssi in screen 24/7 |
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| 07:06.47 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 07:06.49 | Chris-NB | hi |
| 07:07.02 | Chris-NB | is it possible to prohibit the callerid for a sip call? |
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| 07:17.29 | flenders | does anyone here run fax over sip? |
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| 07:19.33 | angryuser | can someone help me, my B410P is working bad, my pmtp ports go down after a first call passed by the port(any) if i put the port up manually, it works till the next call, i have all lates packages |
| 07:19.43 | angryuser | (stable) |
| 07:28.27 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
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| 07:31.18 | JT | how many ports, are they all TE mode? |
| 07:33.11 | menfin | hi guys ! |
| 07:38.48 | angryuser | JT: 4 ports and they are in te mode, i have tryed both te_ptp and te_ptmp, te difference is that in te_ptp ports goes up when i receive calls, in te_ptmp, they go down after i call out |
| 07:39.11 | JT | yeah that's one of the differences |
| 07:39.45 | angryuser | JT:it is a known problem? |
| 07:41.42 | angryuser | JT:i will set up a bug repors, hopefully i turned debug on when tested |
| 07:42.18 | tzafrir_laptop | angryuser, france should probably be ptmp |
| 07:43.23 | angryuser | tzafrir_laptop: yes, te_ptp was for testing purposes |
| 07:44.19 | angryuser | tzafrir_laptop: in te_ptmp ports go down after the first call passed, sometimes tey go down by themself.... |
| 07:44.39 | angryuser | dead end |
| 07:44.53 | tzafrir_laptop | so if you just call in and disconnect you can call ot immediately after that, right? |
| 07:45.48 | JT | angryuser: call digium tech support |
| 07:45.48 | JT | since they love people using misdn so much |
| 07:46.40 | angryuser | JT: hehe.... i am not alone with my problem, i have searche over the forums of asterisk-france, some people has same issue |
| 07:47.43 | voltagex | do I still need to recompile asterisk to set "payload type 96"? |
| 07:48.17 | JT | angryuser: normally i'd suggest to try bristuff, but i have no idea if it's compatible with the B410P |
| 07:49.21 | angryuser | JT: il search then |
| 07:50.44 | angryuser | bristuff are still on 1.2 :< |
| 07:52.24 | angryuser | is chan_capi compatible with 1.4.2? |
| 07:53.26 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:54.49 | JT | most people still use 1.2 in production |
| 07:54.54 | JT | 1.4 isn't as stable |
| 07:56.18 | angryuser | JT: if i will change the version, i will be forced to rewrite a big part of my dialplan(db functions) |
| 07:56.38 | JT | i dunno |
| 07:58.25 | angryuser | i am a bit lost, 2 chan_capi's here http://www.melware.org/ChanCapi (version 1.0) and here http://www.junghanns.net/downloads/chan_capi.0.3.5.tar.gz |
| 07:58.38 | angryuser | what is the diff? |
| 07:59.58 | JT | junghanns stopped maintaining chan_capi and moved it all into bristuff |
| 08:00.07 | JT | someone else kept doing chan_capi |
| 08:00.07 | voltagex | anyone here using mynetfone? |
| 08:00.12 | JT | as far as i understand it |
| 08:01.12 | angryuser | JT: so i "in theory" can intragrate capi only? |
| 08:01.45 | JT | umm it's more a driver issue |
| 08:01.54 | JT | i know digium have released a driver for misdn |
| 08:02.33 | JT | not sure if the multiport bri drivers for chan_capi/bristuff (qozap) work on the digium card |
| 08:02.45 | JT | i don't know what the digium card's chipset is |
| 08:03.36 | angryuser | JT: in any case it is writted on the firs site in hardware copatibility list, that all hardware supported by misdn, works with chan_capi 1.0 |
| 08:03.44 | angryuser | *written |
| 08:05.06 | angryuser | but nothing about asterisk compatible version |
| 08:06.40 | menfin | bulle, I want to say thanks for your help, all is right now, with ekiga and kphone on linux, and with other win32 client |
| 08:08.08 | JT | angryuser: well even so, it would probably still have errors |
| 08:08.16 | JT | it's a card driver issue i think |
| 08:08.24 | JT | not a channel driver issue |
| 08:09.15 | *** join/#asterisk shay (n=shay@unaffiliated/shay) |
| 08:09.16 | shay | yo |
| 08:10.50 | angryuser | JT: well i do remember it worked with some version 1.2.XX of asterisk, anyway i will submit the bug, and will try to move to 1.2 |
| 08:11.01 | JT | ok |
| 08:13.26 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 08:14.12 | voltagex | what does http://pastebin.ca/413264 mean? |
| 08:14.33 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:15.39 | JT | it means it doesn't like what you sent it |
| 08:16.05 | voltagex | ffs I'm starting to hate aussie voip providers |
| 08:16.38 | voltagex | mynetfone ask for pedantic=no to be set, that has to do with non-standard sip stuff right? |
| 08:18.42 | voltagex | and now it randomly starts working |
| 08:19.58 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 08:20.01 | voltagex | aha, it requires callerid to be sent |
| 08:20.13 | voltagex | but MNF are detecting the wrong callerid :/ |
| 08:20.33 | JT | lame |
| 08:20.47 | JT | they could just set a default if they don't like what you send... |
| 08:22.54 | voltagex | they have the caller id with my username |
| 08:23.00 | voltagex | wondering whether that pisses me off enough to not use them... |
| 08:23.12 | voltagex | coz I'm gonna get a victorian DID |
| 08:23.21 | voltagex | from someone, just don't know who |
| 08:23.25 | voltagex | anyway, gtg |
| 08:27.19 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 08:28.05 | *** join/#asterisk Ahrimanes (n=ma@81.7.159.2) |
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| 08:41.39 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 08:42.01 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 08:43.43 | tzafrir_laptop | to answer something that was asked here before I parted: the Digium cards use the same HFC-4s chipset used by the Junghanns cards. But the current qozap drivers won't work with the Digium cards |
| 08:44.13 | JT | what's the reason for that? |
| 08:44.17 | JT | and have you tested it? |
| 08:44.32 | tzafrir_laptop | JT, I never had the incentive to bother |
| 08:44.47 | angryuser | hehe concurrence |
| 08:44.47 | tzafrir_laptop | And I don't have that Digium card |
| 08:45.03 | JT | indeed |
| 08:45.04 | *** join/#asterisk menfin (n=cray@LNeuilly-152-21-127-49.w193-253.abo.wanadoo.fr) |
| 08:45.13 | JT | tzafrir_laptop: what makes you sure it won't work? |
| 08:45.52 | tzafrir_laptop | IIRC I saw a report of someone testing for it. The basic thing is that the hardware scan of qozap won't find it |
| 08:46.06 | tzafrir_laptop | and yeah: qozap includes its own stupid hardware scan |
| 08:46.38 | *** join/#asterisk friedrich| (n=friedric@e177240202.adsl.alicedsl.de) |
| 08:46.41 | angryuser | what do i need to do, if i want a proper uninstall of Asterisk 1.4.2 zaptel, libpri, misdn? |
| 08:47.32 | tzafrir_laptop | unlike, say, vzaphfc that does advertize the list of PCI IDS it supports |
| 08:48.04 | e-ddie | does anyone know if it's possible to make hints to notify on DND state or some variable in the astdb? |
| 08:48.42 | tzafrir_laptop | please take a look at http://voip-info.org/wiki/view/Asterisk+config+zapata.conf#CountrySpecificDialplans and let me know if it is stupid or not |
| 08:49.16 | tzafrir_laptop | I just don't understand exactly what the meaning the "dialplan" is in chan_zap |
| 08:51.12 | JT | yeah dialplans doesn't seem quite right |
| 08:51.16 | JT | in that context |
| 08:51.29 | JT | also i've never needed to set any of those options for Australia |
| 08:52.26 | Ahrimanes | e-ddie: look at func_devstate |
| 08:53.40 | e-ddie | Ahrimanes: i read about thatone, but it's only in 1.6, right? |
| 08:54.02 | Ahrimanes | e-ddie: i had russellb do it for me for 1.4 ;) |
| 08:54.09 | e-ddie | oh |
| 08:54.16 | *** join/#asterisk fbffff (n=fbffff@rrcs-24-153-196-185.sw.biz.rr.com) |
| 08:54.47 | tzafrir_laptop | JT, I needed to add "dialplan=dynamic" and the nationalprefix line, I believe. I haven't tested an international number |
| 08:55.00 | e-ddie | do you know if there's a way to make it check the astdb for some extension too? |
| 08:55.03 | Ahrimanes | http://www.asterisk.org/node/48325 <- "I popped into IRC and talking to someone there gave me the idea to write a new module..." <- i'm someone |
| 08:55.47 | Ahrimanes | e-ddie: it's a dialplan function, execute something in your dialplan to lookup astdb and use func_devstate |
| 08:56.38 | tzafrir_laptop | JT, so generally from your experince there is no need to manually edit the nationalprefix, internationalprefix, etc. ? |
| 08:56.48 | *** join/#asterisk gr1ncheux (n=devine@unaffiliated/gr1ncheux) |
| 08:56.53 | e-ddie | Ahrimanes: ok. thx |
| 08:56.57 | voltagex | Ahrimanes: so basically dialling a number to set "Away" mode? |
| 08:57.08 | *** join/#asterisk Fony_Vaio (n=Fony_Vai@81-178-58-216.dsl.pipex.com) |
| 08:57.34 | Ahrimanes | voltagex: could be done.. but so much more can be done.. this is meant for those flashy lamps on snom's |
| 08:58.17 | voltagex | Ahrimanes: ok, I just want to be able to set something so my phone doesn't ring in the middle of the night |
| 08:58.21 | Fony_Vaio | is this a dev only chan? |
| 08:58.48 | voltagex | Fony_Vaio: no |
| 09:00.10 | Ahrimanes | voltagex: most phones have a DND button |
| 09:00.31 | voltagex | $20 deskphone doesn't :P |
| 09:01.28 | Fony_Vaio | voltagex, ok cool. I work for a company who has way too many phone lines (six and counting) ringing in one retail store. It needs to be centralised so that all the phones which are not in-call ring whenever a call is recieved on any of the six lines, so that people aren't running around trying to find a specific phone. Is this possible with asterisk? Can someone point me to a guide? Any hardware recommendations for this particular proj |
| 09:01.28 | Fony_Vaio | ect? |
| 09:01.32 | Ahrimanes | voltagex: http://www.voip-info.org/wiki/index.php?page=PBX+Do+Not+Disturb |
| 09:01.58 | Ahrimanes | Fony_Vaio: that's a queue |
| 09:02.29 | voltagex | Ahrimanes: get error as soon as I dial *78 |
| 09:02.41 | Fony_Vaio | Ahrimanes, a queue defines how calls are routed to handsets? |
| 09:04.43 | Ahrimanes | Fony_Vaio: it can be used for that, yes |
| 09:05.25 | Ahrimanes | voltagex: ah sorry, those were hardcoded in chan_zap it seems |
| 09:05.32 | Ahrimanes | voltagex: it's really simple to do though |
| 09:05.41 | Fony_Vaio | Ahrimanes, excellent. Do i just need lots of speakerphone modems or is there someother (relatively cheap) hardware that i can use for all those pots ports and back out to the handsets? |
| 09:07.48 | Ahrimanes | Fony_Vaio: define relatively cheap? |
| 09:08.09 | *** join/#asterisk zepmantra (i=dsadsads@125.212.110.115) |
| 09:08.26 | Ahrimanes | voltagex: http://lists.digium.com/pipermail/asterisk-users/2003-July/009696.html <- there's some same dialplan code to do it |
| 09:08.42 | Fony_Vaio | Ahrimanes, well, not hundreds of GBP? |
| 09:11.18 | Ahrimanes | Fony_Vaio: how many phones? |
| 09:12.10 | Ahrimanes | Fony_Vaio: http://www.digium.com/en/wheretobuy/digiumdirect/products.php?category_id=18 <- a card like this should scale somewhat well for your setup and allow pots lines and phones to be connected. not the cheapest though |
| 09:12.29 | Fony_Vaio | Ahrimanes, 3 or 4 max. |
| 09:13.47 | Fony_Vaio | Ahrimanes, ouch.. £450 plus shipping to UK. |
| 09:16.32 | Ahrimanes | Fony_Vaio: yeah.. i'd go for cheap sip phones and maybe a http://www.digium.com/en/wheretobuy/digiumdirect/products.php?category_id=37 for pots |
| 09:18.33 | Fony_Vaio | Ahrimanes, that's a little more like it |
| 09:19.11 | Ahrimanes | Fony_Vaio: i guess, if you can find some sip phones for about $30 a piece it could be kept cheap |
| 09:19.30 | Ahrimanes | Fony_Vaio: at some time, pots should probably be exchanged for a T1/E1 if you get enough lines |
| 09:19.37 | Fony_Vaio | Ahrimanes, yah, the most expensive bit seems to be these pots ports |
| 09:19.54 | *** join/#asterisk SoftIce (n=phil@vc-196-207-45-253.3g.vodacom.co.za) |
| 09:20.57 | SoftIce | Asterisk SIP Invite Message Remote Denial of Service, Asterisk SIP Channel Driver Reponse Code Zero Remote Denial of Service, Grandstream Budgetone 200 Phone SIP INVITE Remote Denial of Service, 7940/7960 Phone SIP Invite Remote Denial of Service |
| 09:21.06 | SoftIce | wow, thats quite a few SIP vuln's for this week |
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| 09:23.26 | *** join/#asterisk dj-fu (n=ajc@unaffiliated/dj-fu) |
| 09:24.13 | *** join/#asterisk jeremy_g (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se) |
| 09:24.17 | jeremy_g | aright boys! |
| 09:24.22 | jeremy_g | wake up |
| 09:24.27 | jeremy_g | this is it |
| 09:25.44 | SoftIce | hmm, I don't seem to see a patch on the asterisk site for this new sip DoS vuln ? |
| 09:26.49 | Ahrimanes | SoftIce: when did it come out? |
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| 09:45.10 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
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| 09:53.28 | JT | tzafrir_laptop: correct |
| 09:53.55 | *** join/#asterisk tmjb (n=tmjb@195.252.125.18) |
| 09:54.45 | tzafrir | JT: that's PRI or BRI? Though I can't believe it should be different |
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| 09:57.09 | tmjb | is this smart to create exten => _0.,1,Dial(Zap/1-8/${EXTEN:1}&misdn/1/${EXTEN:1}) ? basicly i want to use ISDN and Analog lines and 0 for out. tnx |
| 09:58.14 | tzafrir | bah. I really miss the "talk" page in mediawiki |
| 09:58.40 | tzafrir | tmjb, you wanted: |
| 10:01.18 | tzafrir | you need it to ring all phones together? |
| 10:02.33 | jeremy_g | i need snmp monitoring on 1.2 |
| 10:02.35 | jeremy_g | any suggestions |
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| 10:04.09 | awk | jeremy_g what type of monitoring what do you want to do with the trap messages? |
| 10:05.12 | JT | tzafrir: bri for sure |
| 10:05.15 | JT | pri probably the same |
| 10:05.37 | awk | JT: what is happening with bristuff and this sip vuln |
| 10:05.40 | jeremy_g | awk:whatever is available. i am in no position to demand specific moinitoring |
| 10:05.44 | giasai68 | hello if i need to get a dial from a fxs if the number is 7 digits i can do exten=> _XXXXXXX., dial....... if i don't know thw number of digits how can i do start call? e.g. i i write exten=>_X. after 1 digit call start i need to do start the call after the caller digit some numbers on his telephone |
| 10:05.56 | awk | I know they using and early asterisk core |
| 10:06.01 | awk | have they fixed the sip vulns? |
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| 10:09.47 | zeeesh | hi |
| 10:11.14 | tmjb | tzafrir,no i want dial out of asterisk analog or ISDN bri which line is free example when i call from my SIP phone 0555-555 If all zap are busy user ISDN |
| 10:16.17 | jeremy_g | awk:tell me about snmp in 1.2? |
| 10:16.25 | jeremy_g | i have to add it to a gui |
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| 10:21.08 | geoaxis | has any one been able to configure and run Asterisk with SNMP support and OpenNMS |
| 10:22.18 | giasai68 | hello if i need to get a dial from a fxs if the number is 7 digits i can do exten=> _XXXXXXX., dial....... if i don't know thw number of digits how can i do start call? e.g. i i write exten=>_X. after 1 digit call start i need to do start the call after the caller digit some numbers on his telephone |
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| 10:25.55 | tzafrir | tmjb, use Zap/g0/NUMBER , and put all the zaptel channels in g0 |
| 10:25.57 | tmjb | when I call my Analog line first i here 2 two ring signal rings than my sip phone start ring any does this must work this way ? |
| 10:26.30 | tmjb | tzafrir,nice groups are great tnx |
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| 10:41.09 | shermster | does anyone know the syntax for installing the asterisk.XXX-patch.gz file to an existing base installation? |
| 10:42.20 | *** join/#asterisk Meaty (n=meaty3@office.abi.ca) |
| 10:43.31 | tzafrir | gzip <asterisk.XXX-patch.gz | patch -p1 --dry-run |
| 10:43.39 | tzafrir | see if this gives you an error ... |
| 10:43.52 | tzafrir | gunzip <asterisk.XXX-patch.gz | patch -p1 --dry-run |
| 10:44.01 | tzafrir | of course |
| 10:49.29 | Dr-Linux | tzafrir: |
| 10:49.31 | Dr-Linux | what's this: |
| 10:49.32 | Dr-Linux | LHR-PBX*CLI> reload chan_iax2.so |
| 10:49.32 | Dr-Linux | The previous reload command didn't finish yet |
| 10:49.43 | Dr-Linux | .... |
| 10:49.47 | Dr-Linux | why i'm getting this |
| 10:50.07 | Dr-Linux | even i don't have any probem with iax trunk |
| 10:50.49 | tzafrir | I understand what it technically means. I don't know what caused it |
| 10:51.15 | tzafrir | It seems that the module is still reloading its configuration |
| 10:51.28 | tzafrir | do you see iax users/peers? |
| 10:52.11 | Dr-Linux | since i'm using iax only for trunk between 2 asterisk servers and i am able to call both ways |
| 10:52.11 | shermster | tzafrir - I get a prompt asking me which file to patch? |
| 10:52.25 | Dr-Linux | that's strange |
| 10:52.48 | tzafrir | shermster, so it's a worng -p value. try -p0 instead |
| 10:52.57 | tzafrir | Dr-Linux, yeah |
| 10:53.10 | tzafrir | Dr-Linux, anything busy with 100% cpu? |
| 10:53.46 | shermster | tzafrir - that worked - thanks! |
| 10:53.49 | Dr-Linux | tzafrir: nope everything looks normal |
| 10:53.50 | Dr-Linux | Executing Dial("SIP/4092-3396", "IAX2/box2/8881|30|r") in new stack |
| 10:53.50 | Dr-Linux | <PROTECTED> |
| 10:53.56 | Dr-Linux | worked |
| 10:54.12 | Dr-Linux | tzafrir: also i had no problem with last related 2 months ago |
| 10:55.12 | tzafrir | Dr-Linux, and another reload attempt will still give that warning? |
| 10:56.11 | Dr-Linux | yes |
| 10:56.26 | Dr-Linux | tzafrir: aww, i didn't something else |
| 10:56.29 | Dr-Linux | and i'm okey now |
| 10:56.50 | Dr-Linux | what i did is, "unload chan_iax2.so" worked |
| 10:57.17 | Dr-Linux | then i did "reload chan_iax.so" same warning |
| 10:57.32 | Dr-Linux | then i did "load chan_iax.so" worked! |
| 10:58.36 | Dr-Linux | tzafrir: you can see here: http://phpfi.com/220870 |
| 10:59.16 | Dr-Linux | tzafrir: but "reload chan_iax.so" still does'nt work for me |
| 11:00.12 | *** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net) |
| 11:01.01 | tzafrir | strange |
| 11:01.32 | tzafrir | I'm not really familiar with that. But I really wonder why you got that error when the module was not really loaded |
| 11:03.01 | Dr-Linux | tzafrir: well, now my changes are done, and eveything is working just fine with new changes, but still error while reload :P |
| 11:03.43 | Dr-Linux | any way tzanger i'd like to ask a important before implementation |
| 11:06.04 | Dr-Linux | tzafrir: i wanna use different bandwith host for voice traffic, like sip/iax, so what voice ports should i put on different bandwidth? 5060 & 10 to 20k range and iax port? |
| 11:06.15 | Dr-Linux | tzafrir: or i need something else as well? |
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| 11:16.11 | giasai68 | <PROTECTED> |
| 11:17.23 | nasls_lsa | is there any timeout of input or something .. ? |
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| 11:53.42 | Jynger | hi |
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| 11:58.37 | Jynger | wich is the better choice 1.2.17 or the svn1.2 branch ? |
| 11:59.14 | MrWup | hey guys |
| 11:59.22 | MrWup | im having some problems with my digium 4fxo card |
| 11:59.29 | MrWup | i got my sip phones call connected up to asterisk |
| 11:59.42 | MrWup | and the quality of voice is very good between sip phones connected to my POE switch |
| 12:00.38 | MrWup | problem is... when one of the sip phones dials out via the digium card to an external line |
| 12:00.42 | MrWup | the voice quality is no good |
| 12:00.47 | MrWup | quite a bit of crackling on the line |
| 12:00.52 | MrWup | and also its rather quiet |
| 12:01.26 | Jynger | something to tune in zaptel config ? |
| 12:01.47 | MrWup | and when someone on the SIP phone speaks into the mouthpiece the voice from the external line coming through the SIP earpiece cuts out while you're talking into the SIP phone |
| 12:02.08 | MrWup | yeah i assume its zaptel config |
| 12:02.15 | MrWup | as the sip phones work great between each other |
| 12:02.17 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
| 12:02.23 | MrWup | but i got no clue really about what to do to make it right |
| 12:02.25 | MrWup | im an asterisk noob |
| 12:03.02 | Jynger | not sure, but could be something to do with echo canceling and vad, try to play with those ? |
| 12:03.09 | MrWup | hm ok |
| 12:03.33 | Jynger | www.voip-info.org has pretty long quides about zaptel optimisations |
| 12:03.41 | Jynger | detailed |
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| 12:28.27 | modcuts | what would be the recommend format of mp3 for music on hold to get the best sound quality? |
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| 12:30.01 | Corynick | Im a bit of a noob here and I just have a few questions, the first being: Is it possible to use Asterisk VOIP for internal building calls, and still use POSTN for external calls?? |
| 12:33.15 | JT | Corynick: yes |
| 12:33.15 | JT | and i think you mean PSTN |
| 12:33.15 | JT | or POTS |
| 12:34.08 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 12:34.32 | Corynick | hahhha thanks :-p |
| 12:34.46 | Corynick | i knew it was soemthing like that |
| 12:35.19 | Corynick | but now i am looking at the system requirements and I need to find a system to suit my needs |
| 12:35.25 | *** join/#asterisk ppyy (i=ppyy@58.216.29.156) |
| 12:35.25 | JT | POTS = Plain Old Telephone System = analogue lines |
| 12:35.27 | *** join/#asterisk angryuser (n=Miranda@i03v-213-44-169-43.d4.club-internet.fr) |
| 12:35.54 | JT | PSTN = Public Switched Telephone Network = phone network as a whole, can be accessed via digital methods too |
| 12:36.16 | JT | right |
| 12:36.24 | Corynick | ok |
| 12:36.26 | Makenshi | I suggest PANS should be private access network switch |
| 12:36.39 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 12:36.46 | angryuser | i have some problems compiling asterisk 1.2.17 http://www.pastebin.ca/413424 zaptel 1.2.12.xpp.r3332 |
| 12:36.56 | *** join/#asterisk p0g0 (n=pogo@madwifi/support/p0g0) |
| 12:37.06 | Corynick | i will have about 5 lines and 20 sets |
| 12:37.17 | *** join/#asterisk heart (n=zippetto@lugbari/people/heart) |
| 12:37.36 | angryuser | [codec_zap.o] Error 1 |
| 12:42.59 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 12:46.19 | *** join/#asterisk Crescendo_ (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 12:48.07 | puzzled | hi |
| 12:54.58 | *** join/#asterisk friedrich| (n=friedric@e177249002.adsl.alicedsl.de) |
| 12:56.45 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 13:00.25 | *** join/#asterisk kovger (n=kovger@217.27.219.154) |
| 13:03.31 | angryuser | if i dont compile codec_zap.so, is it imprtant module? |
| 13:04.00 | JunK-Y | only if ya want to use a specific card for pstn. |
| 13:04.02 | *** join/#asterisk phillipk (n=pkey@216.248.143.77) |
| 13:04.21 | angryuser | i have fxo's |
| 13:04.50 | JunK-Y | then ya need it. |
| 13:05.00 | JunK-Y | u need zaptel to use it. |
| 13:05.14 | angryuser | it is not comiling on my machine.....1.2.17...16..15... same error |
| 13:06.02 | JunK-Y | theres so much doc on that problem, just take a look at voip-info.org |
| 13:07.27 | *** join/#asterisk jeanmmi (n=besnard@front.tekkno.net) |
| 13:08.21 | angryuser | search codec_zap.so 1 result on voip-info.org |
| 13:09.15 | *** join/#asterisk friedrich| (n=friedric@e177249002.adsl.alicedsl.de) |
| 13:09.33 | *** join/#asterisk deeperror (n=deeperro@mail.banctel.com) |
| 13:10.11 | Katty | who knew - arby's has breakfast stuff |
| 13:12.32 | *** join/#asterisk friedrich| (n=friedric@e177249002.adsl.alicedsl.de) |
| 13:13.43 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 13:14.10 | *** join/#asterisk deeperror (n=deeperro@mail.banctel.com) |
| 13:14.33 | *** join/#asterisk kremiq (n=kremo@213.215.118.28) |
| 13:15.02 | *** part/#asterisk hexa (n=hexa@modemcable189.156-200-24.mc.videotron.ca) |
| 13:15.53 | *** join/#asterisk joeTSUNAMI (n=joeTSUNA@72.243.118.59) |
| 13:16.40 | gfraysse | <PROTECTED> |
| 13:17.08 | kremiq | is here anybody who use xorcom + astribank? |
| 13:17.13 | angryuser | yes |
| 13:17.30 | kremiq | angryuser: and can me help with small problem? |
| 13:17.45 | angryuser | decribe your problem |
| 13:19.10 | kremiq | angryuser: when I do software reboot from rapid-menu or simple reboot from console, xorcom stop and no reboot, I must poweroff it. When I don't have connected astribanks is no problem |
| 13:19.43 | LeddyHM | anyone know if sunrocket is better about supporting asterisk users? |
| 13:19.48 | kremiq | angryuser: And often Asterisk not detect connected astribanks in usb and no start :( |
| 13:20.05 | LeddyHM | I see a 199/2yr |
| 13:20.21 | angryuser | do you load firmware by youself? external script? |
| 13:20.21 | LeddyHM | might be worth investing if it is going to work, reliably |
| 13:20.51 | angryuser | ah sorry you are using rapid |
| 13:21.00 | kremiq | angryuser: firmware for astribank? I keep default |
| 13:21.21 | angryuser | i have no idea how rapid works, i load it manually |
| 13:21.41 | kremiq | angryuser: okey thanks for your time |
| 13:23.30 | angryuser | but the basic procedure is simple, /etc/hotplug/usb/xpp_fxloader load then echo 0 0 >/proc/xpp/sync |
| 13:25.20 | angryuser | kremiq: hint ask tzafrir ;) |
| 13:25.49 | kremiq | angryuser: yes I chating with him now in query :) |
| 13:26.27 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 13:27.41 | SoftIce | Asterisk 1.2.16 is affected with the sip vuln too |
| 13:27.54 | SoftIce | only 1.2.17 and 1.4.2 are not |
| 13:28.34 | angryuser | SoftIce: what this vuln can affect? |
| 13:28.48 | SoftIce | sip dos vuln |
| 13:28.57 | angryuser | SoftIce: crash? |
| 13:30.38 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-185-4.buckeyecom.net) |
| 13:30.45 | SoftIce | Asterisk is exposed to a remote denial of service issue while handling a single, malformed "SIP INVITE" request containing two headers. When the first header is valid and the second header is not valid, the |
| 13:30.55 | SoftIce | vulnerability is triggered |
| 13:31.58 | SoftIce | well what it can do is prevent people from acessing the phone system |
| 13:32.14 | SoftIce | so in a sense yes, a crash. |
| 13:33.01 | mosty | does anyone know how to use chanspy? what do all the beeps mean? |
| 13:33.34 | mosty | and is there any way too get debugging info from chanspy? |
| 13:33.34 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:33.46 | angryuser | mosty: it's when computer start ;)))) honestly no idea |
| 13:34.52 | *** join/#asterisk SlamX (n=Sad@AOrleans-157-1-45-214.w90-8.abo.wanadoo.fr) |
| 13:34.57 | SlamX | Hi |
| 13:35.01 | c4t3l | hello |
| 13:35.19 | SlamX | can some one help me with a chan_cellphone probleme ? |
| 13:36.42 | SlamX | i have my bluetooth working and i have connected my K75i as well. the probleme is whene i do 'cell show devices' !! it shows me nothin |
| 13:37.42 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@ppp-70-255-140-128.dsl.hstntx.swbell.net) |
| 13:37.48 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 13:37.48 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:38.07 | *** join/#asterisk af_ (n=getsmart@ip-156-32.sn2.eutelia.it) |
| 13:38.07 | SlamX | can someone helps me out plz ? |
| 13:39.10 | SlamX | mog are u alive :p ? |
| 13:41.51 | SlamX | i have my bluetooth working and i have connected my K75i as well. the probleme is whene i do 'cell show devices' !! it shows me nothin |
| 13:45.00 | SlamX | :\ |
| 13:46.42 | *** join/#asterisk phillipk (n=pkey@216.248.143.77) |
| 13:47.06 | SoftIce | hmm how g729 work in this sense that i know people that just copy it from box to box |
| 13:47.11 | *** join/#asterisk elmerbug (n=don@dsl017-061-162.sfo4.dsl.speakeasy.net) |
| 13:47.24 | SoftIce | couldnt anyone just do that? |
| 13:47.39 | SoftIce | how does it know when to get billed, etc |
| 13:47.50 | elmerbug | Howdy! I am having an issue with Endpoint Manager |
| 13:50.49 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 13:52.12 | Mercestes | ~seen c4t3l |
| 13:52.32 | jbot | c4t3l is currently on #asterisk (7h 41m 22s). Has said a total of 1 messages. Is idling for 17m 31s, last said: 'hello'. |
| 13:52.32 | Mercestes | ~last c4t3l |
| 13:52.36 | Mercestes | ... |
| 13:52.46 | Mercestes | :) |
| 13:52.51 | Mercestes | ~botsnack |
| 13:52.51 | jbot | Mercestes: aw, gee |
| 13:52.58 | Mercestes | Morning, c4t3l |
| 13:53.03 | SlamX | Mercestes can u help me ? |
| 13:53.15 | Mercestes | SlamX: Absolutely. |
| 13:53.20 | SlamX | with a chan_cellphone probleme |
| 13:53.24 | SlamX | i have my bluetooth working and i have connected my K75i as well. the probleme is whene i do 'cell show devices' !! it shows me nothin |
| 13:53.43 | SlamX | do u know why ? |
| 13:53.43 | Qwell | SlamX: put your phone into discoverable mode |
| 13:53.52 | SlamX | yes it is |
| 13:53.54 | Qwell | or pairing mode, or whatever |
| 13:54.07 | mosty | is there any way to break out of ChanSpy without hanging up? |
| 13:55.03 | Mercestes | mosty: I don't think so, you' d have to break the stream somehow, which is technically a hangup. Maybe you can MoH yourself or just put yourself wherever youw ant to be. |
| 13:55.59 | mosty | mercestes: i will submit a feature request then |
| 13:56.04 | Mercestes | SlamX: What version of.....is this even asterisk your using? |
| 13:56.35 | Mercestes | mosty: Maybe you should start with what end result you want to happen. |
| 13:56.36 | SlamX | i'm using svn |
| 13:56.50 | Qwell | Mercestes: 8919 :D |
| 13:57.05 | Mercestes | SlamX: Congratulaions. I heard it's a great repository. What are you using from SVN? |
| 13:57.16 | *** join/#asterisk joeTSUNAMI (n=joeTSUNA@72.243.118.59) |
| 13:57.20 | Mercestes | Qwell: omg, Check out #telapthy it really exists |
| 13:57.27 | Mercestes | Qwell: #telepathy even |
| 13:57.30 | Qwell | I'm already there |
| 13:57.34 | Mercestes | rofl |
| 13:58.13 | Mercestes | I so just randomly made that up one day. Talk about a good name. |
| 13:58.16 | SlamX | svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk-trunk |
| 13:58.58 | *** join/#asterisk bkuhn (n=bkuhn@fsf/member/bkuhn/bkuhn) |
| 13:59.10 | Mercestes | SlamX: Okies. I personally have never used chan_cellphone. Are you following a howto somewhere? |
| 14:00.18 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 14:01.06 | SlamX | yes |
| 14:01.07 | SlamX | http://www.saghul.net/blog/2007/02/02/howto-chan_cellphone-en-asterisk-14-trunk/ |
| 14:01.22 | SoftIce | somebody tell me how g729 payment works |
| 14:01.23 | SoftIce | ? |
| 14:03.45 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 14:03.58 | mosty | softice: you pay digium, they email you a licence key |
| 14:04.50 | Mercestes | you call up digium. |
| 14:05.06 | SoftIce | mosty: ok but how does it prevent X ammount of calls? |
| 14:05.28 | Mercestes | SoftIce: It's part of the software. If you attempt more calls than licenses it segfaults asterisk |
| 14:05.47 | SoftIce | ok but its not machine specific |
| 14:05.54 | mosty | no, it doesnt segfault asterisk |
| 14:05.56 | SoftIce | you can just copy the codec to any asterisk modules directory |
| 14:06.36 | mosty | it just refuses to use more than the allottet number of g729 licences, and falls back to another codec |
| 14:06.38 | *** join/#asterisk Vec (n=Vec@dsl-243-115-51.telkomadsl.co.za) |
| 14:07.23 | mosty | softice: the codec checks against the machines hardware, and needs to be re-registered if the hardware has changed |
| 14:08.39 | *** join/#asterisk ooor4 (i=blah@c-76-108-176-180.hsd1.fl.comcast.net) |
| 14:08.42 | Mercestes | SlamX: Are you using debian? |
| 14:09.34 | ooor4 | erm |
| 14:09.49 | *** join/#asterisk cfh (i=proxyuse@host197-137.pool8290.interbusiness.it) |
| 14:09.53 | Mercestes | ooor4, Digium support for that. |
| 14:10.02 | cfh | hi all |
| 14:10.03 | Vec | I want to see how many active channels there are on an IAX trunk from the dialplan, any ideas how I could do this, have looked at the function group_count but does not look right ? |
| 14:10.22 | Mercestes | Vec: Iax2 show channels maybe? |
| 14:11.24 | cfh | i m using chan_misnd with beronet isdn BRI card , is possibile monitoring the state of the isdn line with special button on a sip tel (ex line isdn busy = led on; line isdn free = led off ) ? |
| 14:12.23 | *** join/#asterisk Doodluv (n=jbaskew@24.214.206.158) |
| 14:12.35 | mosty | vec: i'd like to know the answer too. the only way i know at the moment would be to use an agi script which uses the console (or even better the manager interface) to find out, before setting a channel var with the result |
| 14:13.24 | *** join/#asterisk Greek-Boy (n=grb@196.46.105.176) |
| 14:13.55 | Mercestes | cfh: That sounds an awful lot like a turn key system. |
| 14:15.06 | Mercestes | cfh: What I think your referring to is "buddy watch" or "presence monitoring" or "hints" or other similar terms and I don't think it works with chan_isdn |
| 14:15.08 | *** join/#asterisk echo- (n=echo@64.184.118.232) |
| 14:15.39 | cfh | Mercestes: hints with chan_misnd ? |
| 14:16.13 | Mercestes | cfh: Right, I don't think that would work but I'm pretty sure that would be what you were looking for if it did work. |
| 14:16.32 | cfh | Mercestes: ok |
| 14:16.55 | cfh | Mercestes : with digium card is possibile ? |
| 14:17.06 | Mercestes | ... |
| 14:18.01 | Mercestes | cfh: Yes, it only works with digum cards. Be sure to ask for the "PXL" series, not the "P" series, and make sure to specifically ask for the cpe-hint capability. |
| 14:19.47 | Mercestes | cfh: An dyou'll need the svn trunk of asterisk before it will even work. Its' new stuff so it's really expensive and the 1st teir sales guys will probably have never even heard of it yet. Just ask to be escalated until you find a sales engineer whose familiar with the technology |
| 14:19.53 | *** join/#asterisk quidpro (n=quid@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 14:20.17 | tzafrir_laptop | chan_misdn should work with most misdn hardware, right? (e.g: hfc, multi hfc, fritz-avm) |
| 14:20.26 | Mercestes | cfh: I'm lying btw. Once again, I don't think you can hint chan_misdn. |
| 14:20.33 | *** part/#asterisk Doodluv (n=jbaskew@24.214.206.158) |
| 14:20.53 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:20.53 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:20.54 | Mercestes | I think hinting is a sip thing and the sip device has to support it/watch for it. |
| 14:21.11 | Mercestes | FoP or manager.api might provide a workaround for you. |
| 14:21.17 | Mercestes | or a google of asterisk turn key |
| 14:21.21 | cfh | Mercestes : ok , but where can i find the PXL series from digium ? do you have a link ? |
| 14:23.12 | Mercestes | ... |
| 14:23.25 | Mercestes | cfh: You have to call digium directly. Are you sure your getting *everything* I'm saying?? |
| 14:23.55 | Greek-Boy | anyone know where I can get voip phones in dubai? Polycom would be good... |
| 14:24.23 | Mercestes | Greek-Boy, from the internet? Most online shopping carts can do currency conversions for you. |
| 14:24.49 | aydiosmio | the troulbe is finding people who ship to Dubai |
| 14:25.15 | Mercestes | everything has a price. For enough virgins I'll hand deliver polycoms to Dubai. |
| 14:25.26 | cfh | Mercestes: first i will contact the beronet teem |
| 14:25.34 | cfh | team* |
| 14:25.43 | Mercestes | cfh: Okies. Good luck..they migh tnot have their hands on the tech yet. |
| 14:26.03 | Mercestes | cfh: Search the wiki for the howto. really good documentation there no how to get it up and working. |
| 14:26.46 | cfh | Mercestes: ok thank you |
| 14:26.47 | cfh | :) |
| 14:28.11 | Mercestes | cfh: There is no mISDN hinting..... |
| 14:30.44 | cfh | Mercestes: why ? |
| 14:30.53 | Mercestes | cfh: It's a sip thing. |
| 14:31.14 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 14:31.20 | Mercestes | cfh: I personally have never heard of/come across ISDN hinting. |
| 14:32.10 | *** join/#asterisk _DAW (n=chatzill@adsl-156-109-78.msy.bellsouth.net) |
| 14:32.25 | Mercestes | cfh: Again, you can attempt it through FoP or manager.api but, I don't think ISDN gives hint events |
| 14:32.41 | cfh | Mercestes:ok |
| 14:32.48 | Mercestes | cfh: You can also do some googling on "asterisk turn key" or "asterisk line system" I believe and come up with an older how to on how to emulate a turn key |
| 14:33.06 | Greek-Boy | lol |
| 14:33.09 | Greek-Boy | vpi |
| 14:33.17 | _DAW | Any recommendations on not being able to break dialtone from channel bank to asterisk T1? |
| 14:33.26 | Greek-Boy | voipsupply.com looks good but they only deliver to USA/CANADA |
| 14:34.33 | tzafrir_laptop | Greek-Boy, is that phone for usage in-house? because generally voip is blocked |
| 14:34.52 | Mercestes | cfh: I don't see anything for isdn presence, pri presence, line presence pri hints or line hints under google. |
| 14:35.29 | Mercestes | _DAW: Bad DTMF detection. Call an Answer(). canreinvite=yes, dtmfmode=auto |
| 14:36.49 | cfh | Mercestes : but many peoples talk me about this features ? :( |
| 14:36.50 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:37.51 | Mercestes | cfh: That is from the old turnkey PBX systems in which each button was tied to a single direct connect line. Asterisk doesn't do that, lines are dynamic and assigned on command, they are not static like the older systems. |
| 14:38.09 | cfh | Mercestes: i know |
| 14:38.29 | cfh | Mercestes: but many peoples doesnt understand :( |
| 14:38.34 | Mercestes | cfh: Therefore, you cannot meaningfully assign state supervision to a line on a button on the phone because that button would always be lit, because it could be *any* line and tha twont' change unless we revert to older technology and staticy assign lines to buttons. |
| 14:38.43 | Qwell[] | Mercestes: SLA :p |
| 14:39.27 | Mercestes | http://www.voip-info.org/wiki/index.php?page=Asterisk%20SLA |
| 14:39.35 | Mercestes | Thanks, Qwell. That's the page I was googling for. |
| 14:39.42 | Mercestes | "key system" not turn key. Awesome. |
| 14:39.53 | Mercestes | Qwell[]: that requise the PXL digum cards, right? >.> |
| 14:39.58 | Qwell[] | pxl? |
| 14:40.04 | Mercestes | oh, you missed it. |
| 14:40.08 | Qwell[] | and no, you can use SLA trunks |
| 14:40.11 | Qwell[] | erm |
| 14:40.13 | Qwell[] | SIP trunks |
| 14:40.17 | Qwell[] | "trunks"... |
| 14:40.25 | Mercestes | Morning buddy. |
| 14:40.30 | Qwell[] | indeed |
| 14:40.35 | Qwell[] | I assume I missed a bit of sarcasm |
| 14:41.16 | Mercestes | Qwell[]: to summarize, Merc: "ISDN hints do not work." CFH: "only works on digium cards?" Merc: "... yes. The PXL series. *demand* to be escalated until you find an engineer familiar with it." |
| 14:41.39 | Qwell[] | that's just mean |
| 14:41.42 | Mercestes | it is. |
| 14:41.50 | *** join/#asterisk vooduhal (n=vooduhal@tc-proxy2.catt.com) |
| 14:42.00 | Mercestes | <PROTECTED> |
| 14:42.06 | Mercestes | I think I got him set fairly straight now tho. |
| 14:42.24 | Mercestes | I had him sold on the "XL" cards tho. |
| 14:42.24 | Qwell[] | I don't see why isdn hints wouldn't work... |
| 14:42.29 | vooduhal | Strange question: Is: exten => ,1,Hangup() a valid extension line. It's the only solution to a strange problem we are having, but I don't know how. |
| 14:42.36 | Mercestes | vooduhal, no |
| 14:43.00 | Mercestes | vooduhal, I don't know how either. :P |
| 14:43.16 | vooduhal | I didn't think so. Glad I'm not the only one. |
| 14:43.35 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.218) |
| 14:43.37 | Mercestes | vooduhal, your mising your match . |
| 14:44.06 | vooduhal | I know. It was to solve a problem where we were getting a call from '' to ''. |
| 14:44.12 | Mercestes | Qwell[]: I've never tried any form of presence on chan_misdn so I dunno. It might. Would be a neat discovery |
| 14:44.18 | vooduhal | The _. match didn't actually work. |
| 14:44.29 | *** join/#asterisk Connor- (n=billy@198-144-174-5.knx.tn.nxs.net) |
| 14:44.34 | Mercestes | vooduhal, you don't want to use _. |
| 14:44.42 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 14:44.46 | Mercestes | vooduhal, specially ona hangup() |
| 14:44.50 | *** join/#asterisk eltech (i=G00Ds@69.124.148.163) |
| 14:45.09 | vooduhal | And what's even stranger, the DID is being delivered via DTMF. |
| 14:45.11 | vooduhal | This is on a PRI. |
| 14:45.57 | hermuli | i got a basic question. how to record a bunch of numbers from caller punching them on they keys of their phone to a variable and when the caller punches # jump to the next line of the extension? |
| 14:46.06 | hermuli | i hope my question was clear enough |
| 14:46.09 | Mercestes | hermuli, getDTMF I think. |
| 14:46.22 | hermuli | thanks, i'll see to that |
| 14:49.41 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 14:50.04 | Mercestes | hermuli, readdtmf |
| 14:50.24 | *** join/#asterisk sivana (n=sivana@gromit.mixdown.ca) |
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| 14:58.30 | *** join/#asterisk Fieldy (i=PP4Q2U6g@gentoo/contributor/Fieldy) |
| 14:59.26 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 15:02.29 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 15:04.00 | *** join/#asterisk kannan (i=kannan@58.68.25.66) |
| 15:04.40 | syzygyBSD | how many phones do you all have on your desk? |
| 15:04.53 | Qwell[] | right this second? 5.. |
| 15:05.00 | hedge77 | 5 + cells & softphones |
| 15:05.01 | shido6 | 3 |
| 15:05.12 | syzygyBSD | I only have 3 but it isn't enough, I am going to have to grab anohter |
| 15:05.23 | *** join/#asterisk P4C0 (n=ash@200.124.22.34) |
| 15:05.31 | shido6 | Qtek 9100/Tmobile Mda , Linksys 942 , and a uniden cordless connected to a pap2na |
| 15:05.37 | hedge77 | how can I determine why dynamic queue members aren't reloaded properly after a restart? |
| 15:05.58 | Qwell[] | Who wants to take bets on who has the oldest and most proprietary phones on their desk? :P |
| 15:06.09 | Qwell[] | crap |
| 15:06.20 | mihinomenest | I've got a comdial executech. |
| 15:06.42 | Qwell[] | I have a Rolmphone :D |
| 15:06.43 | syzygyBSD | hedge77: what version |
| 15:06.47 | hedge77 | 1.4.1 |
| 15:06.59 | P4C0 | 'Unable to find a codec translation path from g729 to slin' --> what's slin??? I'm using alaw and g729 only... |
| 15:07.06 | coppice | romn is a name I haven't heard for a while :-\ |
| 15:07.12 | syzygyBSD | hmm, outta luck there.. I know on 1.2.* but I haven't looked on 1.4 |
| 15:07.16 | coppice | s/romn/rolm |
| 15:08.20 | Mercestes | I have an AT&T original CS2500 |
| 15:08.32 | Mercestes | original AT&T brand phone. |
| 15:08.37 | Mercestes | :) |
| 15:08.46 | syzygyBSD | oh.. I need to get better hold music.... bannan phone is getting on my nerves |
| 15:08.51 | Mercestes | with redial and mut ebuttons |
| 15:08.54 | hedge77 | the members are listed after a restart but all marked invalid and it's driving me nuts |
| 15:09.01 | Mercestes | syzygyBSD, ROFLMAO!!!!!! |
| 15:09.17 | Mercestes | ring ring ring ring ring ring ring ring ring banana phone!!!!! |
| 15:09.28 | syzygyBSD | I wasn't thinking I was going to have to test a 10 minute hold |
| 15:12.11 | mihinomenest | http://www.telbroker.com/telephones/6614bk.jpg - $152. |
| 15:12.53 | *** join/#asterisk menfin (n=cray@LNeuilly-152-21-127-49.w193-253.abo.wanadoo.fr) |
| 15:13.00 | *** join/#asterisk p0g0 (n=pogo@madwifi/support/p0g0) |
| 15:13.12 | *** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse) |
| 15:13.13 | tzafrir_laptop | anybody noticed asterisk 1.4.2 getting 100% CPU for no good reason? |
| 15:13.19 | tzafrir_laptop | It remains functional |
| 15:14.02 | anonymouz666 | tzafrir_laptop very very strange |
| 15:14.43 | tzafrir_laptop | For some reason I don't have the time to invastegate it well when it happens. but it has already happened to me twice today. Once it happened I managed to get an strace of the specific thread, and it was in a loop trying to reply to "jabber test" to an asterisk -r |
| 15:14.59 | tzafrir_laptop | it kept getting an error at the write to the socket |
| 15:15.26 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:15.40 | pigpen | tzafrir_laptop, not happening here...but I will be deploying it to a larger installation later this week (200 sip phones) |
| 15:15.48 | Ryushin | Can someone familiar with PRI's and the USA help me with some questions that Qwest is asking me for before provisioning our PRI. |
| 15:16.14 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@ppp-70-255-140-128.dsl.hstntx.swbell.net) |
| 15:16.36 | pigpen | Ryushin, ask and ye shall be answered. |
| 15:16.40 | Ryushin | They are asking USD0 or DMTF. |
| 15:16.46 | Mercestes | DTMF |
| 15:16.54 | pigpen | Mercestes, too quick. |
| 15:16.59 | Ryushin | Yea, sorry, DTMF. |
| 15:17.06 | Ryushin | Okay, let me mark that. |
| 15:17.11 | Mercestes | lol |
| 15:17.11 | sivana | does anyone know if Calling Presentation is passed between * boxes |
| 15:17.20 | Mercestes | Sorry, I thought it was like Jeaopardy |
| 15:17.32 | Ryushin | I assume National as the protocol. |
| 15:17.43 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 15:18.16 | Mercestes | Ryushin: national/ni2 is fine |
| 15:18.17 | Ryushin | From their email: YIELD ON GLARE: CUSTOMER OR QWEST |
| 15:18.29 | Ryushin | Never heard of that before. |
| 15:18.31 | *** join/#asterisk De_Mon (i=de_mon@fl-71-55-184-242.dhcp.embarqhsd.net) |
| 15:18.34 | Mercestes | I just use a sunvisor. |
| 15:18.51 | Ryushin | Oh, what a whit. LOL. |
| 15:19.00 | pigpen | Good polarized sunglasses works for me. |
| 15:19.07 | Mercestes | lol. I would say Qwest. :) Or you coudl google it |
| 15:19.09 | pigpen | I would make it Customer on Glare |
| 15:19.20 | Ryushin | What is glare? |
| 15:19.21 | Mercestes | go with pigpen's answer. |
| 15:19.25 | Mercestes | I've never heard of it before. |
| 15:19.47 | pigpen | Glare is where a call is coming in on channel 1 and an outgoing call on channel 1 |
| 15:19.56 | pigpen | ie: how to avoid this. |
| 15:20.13 | zotz | polarize the channel? |
| 15:20.16 | De_Mon | I have a Aastra 480i, it works fine if I mute the base unit, but when dialing in to chanSpy about 10seconds into the call it gets real jittery and eventually becomes impossible to understand. |
| 15:20.20 | pigpen | the easiest way is to have incoming calls come into 1, outgoing at 23 |
| 15:20.28 | pigpen | ie: outgoing as G0 not g0 |
| 15:20.41 | pigpen | Gx = High to low |
| 15:20.48 | pigpen | g0 = low to high |
| 15:20.53 | De_Mon | I turned off silence supression and it seems to help a little, but it still goes wonkers in less than a minuite. |
| 15:21.00 | De_Mon | minute |
| 15:21.04 | Ryushin | Okay, that's how I do it anyway in the asterisk config. |
| 15:21.19 | Ryushin | Okay, next question: HUNTING: FWD, REV, UCD, GUCD, BGUCD |
| 15:21.26 | pigpen | your dialout, G0 instead of G1 |
| 15:21.35 | Ryushin | I'm sure that's for my hunting groups. |
| 15:21.46 | *** join/#asterisk wunderkin (n=kev@65.39.92.95) |
| 15:22.02 | pigpen | well, on a pri, all your did's are assigned to the entire thing, so I normally ditch hunting... |
| 15:22.14 | pigpen | but if you need it for some reason, I would do FWD. |
| 15:22.28 | pigpen | ie: one did can be rolled through many channels. |
| 15:22.55 | Ryushin | Well, I should need hunting. I can process the calls as long as it comes in the pri and send it from there. |
| 15:23.14 | Ryushin | I mean, I don't need hunting. |
| 15:23.19 | Ryushin | should not need hunting. |
| 15:23.22 | e-milio | Hello Everybody |
| 15:23.28 | pigpen | yeah..some people have issues with getting over hunting.... |
| 15:23.45 | *** join/#asterisk menfin (n=cray@LNeuilly-152-21-127-49.w193-253.abo.wanadoo.fr) |
| 15:23.48 | e-milio | I am (still) gettuing app_meetme.c:1546 conf_run: Unable to write frame to channel: Success |
| 15:24.09 | e-milio | I installed a x100p to get timer prob solved |
| 15:24.25 | Ryushin | Next question: CALLING NAME DELIVERY (inbound only) yes or no |
| 15:24.30 | Ryushin | Is this for caller id? |
| 15:25.06 | e-milio | It only happens when conference starts |
| 15:25.15 | pigpen | Ryushin, I would check with them.... |
| 15:25.22 | pigpen | kinda goofy wording. |
| 15:25.24 | Ryushin | pigpen: Okay. |
| 15:26.06 | Ryushin | Thanks, I appreciate it. |
| 15:26.23 | pigpen | Ryushin, also, make sure on your dialplan that you have about .5 sec as priority 1 in your incoming context to allow CID info to completly come in...otherwise, you may not get the name, only the number. |
| 15:26.27 | fetcher | e-milio: ztdummy by itself wouldn't work? |
| 15:27.04 | e-milio | fetcher: no, i tried but it had the same prob, icheck doc and said it was not enoguh and there was a bug also |
| 15:27.07 | fetcher | e-milio: I had similar problems with MeetMe & ztdummy here... had written it off as some hardware dependent quirk |
| 15:28.01 | hedge77 | Is there a way to automatically do a reload a couple of seconds after a restart? |
| 15:28.06 | e-milio | fetcher:were you able to make it work? |
| 15:28.38 | fetcher | e-milio: no, but I never tried very hard... we just wanted the conferencing for internal & experimental use, nothing serious... |
| 15:28.50 | fetcher | Good to know the X100P might be a workaround option, though |
| 15:28.57 | e-milio | yesterday i asked around and the solution was toinstall a hw timer but it continues |
| 15:29.11 | e-milio | fetcher: as of now it has NOT been a solution |
| 15:29.13 | fetcher | even if it does hammer the box with 1000 interrupts/second (probably not much real oad...) |
| 15:29.34 | *** join/#asterisk fbcit (n=cnighswo@nc-71-0-121-24.sta.embarqhsd.net) |
| 15:29.40 | fetcher | e-milio: Oh... hmm. |
| 15:29.56 | shido6 | errf? |
| 15:30.07 | shido6 | hedge77 yes |
| 15:30.23 | pigpen | Ryushin, fyi, some telco's don't like it when asterisk resets the "not in use" channels every 60 min...you may want to put in the zapata.conf: resetinterval=never |
| 15:30.37 | hedge77 | shido6: is there somewhere I might aquire this arcane knowledge? |
| 15:30.45 | pigpen | When this happens, many times it will fill up their trace, missing the pri errors. |
| 15:31.07 | shido6 | write a script that connects to the asterisk manager interface and reloads OR... just write a script that uses asterisk -rx 'reload' or extensions reload or whatever reload you need specifically |
| 15:31.14 | e-milio | anybody have an idea why app_meetme.c:1546 conf_run: Unable to write frame to channel: Success |
| 15:31.28 | e-milio | even with a HW timer ? |
| 15:31.36 | shido6 | Unable to pass gas: Success. heh |
| 15:31.52 | shido6 | what HW timer? |
| 15:31.57 | e-milio | x100p |
| 15:32.07 | shido6 | do u STILL have ztdummy in use? |
| 15:32.11 | e-milio | nope |
| 15:32.27 | shido6 | not a dell xeon box is it? |
| 15:33.04 | e-milio | no, dell p4 |
| 15:33.30 | fetcher | Does app_conference still not work with 1.2.xx ? |
| 15:33.33 | shido6 | x100p have its on irq? |
| 15:33.47 | e-milio | dont think so |
| 15:33.57 | fetcher | (alternative to MeetMe that doesn't need a timer, but apparently 1.0.x only) |
| 15:33.58 | e-milio | is it needed to be that way ? |
| 15:34.23 | fetcher | e-milio: yes, X100P's don't like to share interrupts |
| 15:34.39 | e-milio | will try to setup bios. be back... |
| 15:34.51 | fetcher | e-milio: the one time I tried it, there was huge jitter on the attached POTS line, and eventually the box locked up |
| 15:35.22 | fetcher | e-milio: you may have to physically move the card to another slot |
| 15:36.23 | pigpen | Also note the X100P is not a real advanced card....I moved to a TDM400 and my issues went away |
| 15:36.52 | pigpen | in both P3, P4 and 64bit Xeon's |
| 15:37.38 | *** join/#asterisk ToyMan (n=Stuart@74-32-55-210.dsl1.mdl.ny.frontiernet.net) |
| 15:37.38 | shido6 | can u use a TDM400P with no modules for a HW timer for conferencing? |
| 15:37.40 | pigpen | I am running TDM400, TDM2400, single/dual/quad pri's in 64bit xeon boxes. |
| 15:37.53 | shido6 | $74.95 |
| 15:37.58 | pigpen | hmm.... |
| 15:38.05 | shido6 | I never tried |
| 15:38.06 | pigpen | why not ztdummy? |
| 15:38.16 | *** join/#asterisk friedrich| (n=friedric@e177249002.adsl.alicedsl.de) |
| 15:38.20 | shido6 | cuz he tried ztdummy and got no luv |
| 15:38.31 | pigpen | you need luv. |
| 15:38.52 | pigpen | if you have issues with ztdummy, you will have issues with most zap issue. |
| 15:39.11 | pigpen | ztdummy is pretty basic. |
| 15:39.24 | shido6 | try giving the x100p its own irq in the bios and try conferencing again. |
| 15:39.32 | fetcher | pigpen: under 2.6, ztdummy uses the motherboard RTC as a timing source, right? |
| 15:39.34 | pigpen | but, I think that the hw timer comes from the chipset..not the module. |
| 15:39.49 | pigpen | think so. |
| 15:39.56 | aydiosmio | oh really? the hw timer comes from the hw? |
| 15:40.00 | aydiosmio | strange. |
| 15:40.05 | pigpen | haha. |
| 15:40.46 | shido6 | http://www.voip-info.org/wiki/index.php?page=Asterisk+timer+ztdummy |
| 15:40.57 | fetcher | aydiosmio: right, either /dev/rtc, or a a UHCI USB host controller (OHCI won't work for some reason) |
| 15:41.00 | pigpen | I always have at least a TDM400 in my boxes, mostly pri's...so I don't screw much ztdummy. |
| 15:41.03 | sivana | does anyone know if Calling Presentation is passed between * boxes |
| 15:41.15 | fetcher | and of course all my boxes are OHCI |
| 15:41.46 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool4-153.mtco.com) |
| 15:42.14 | pigpen | sivana, I would think so. |
| 15:43.08 | mosty | is ztdummy going to be scrapped for posix timers? |
| 15:43.33 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 15:43.42 | mosty | sivana: it can be. it can also be disabled |
| 15:44.05 | fbcit | e-milio: I had to move my X100P through several slots to find one that it liked. Works fine now. Except for the echo which I have not had time to tune. |
| 15:44.17 | e-milio | mmmm |
| 15:44.37 | syzygyBSD | if someone connects with no callerid, how can I trace back who they are? |
| 15:44.38 | pigpen | the X100p can be fun to tune. |
| 15:44.54 | e-milio | I also have a 4port fxs card ? |
| 15:45.03 | Makenshi | syzygyBSD, ask the fbi, i hear they're pretty good at getting call logs |
| 15:45.04 | fbcit | pigpen: So I understand from the reading on the wiki.... |
| 15:45.08 | e-milio | Is there any reason to try the onther one ? |
| 15:45.23 | e-milio | I mean the other card instead of the x100p? |
| 15:45.26 | shido6 | u need a court order if you arent a part of the club |
| 15:45.48 | syzygyBSD | Makenshi: I'm not lazy, I will do it myself if I know how, |
| 15:46.05 | e-milio | quit |
| 15:46.20 | mosty | syzygyBSD: if there is no callerid, you cant |
| 15:46.22 | syzygyBSD | it isn't like they are harrassing, they just call into my automated system and stay for hours in a loop that would drive anyone crazy |
| 15:46.30 | *** join/#asterisk friedrich| (n=friedric@e177249002.adsl.alicedsl.de) |
| 15:46.34 | fetcher | syzygyBSD: you'd need help from the telco providing your inbound service |
| 15:46.51 | syzygyBSD | damn, I knew we should have gotten on the ss7 network |
| 15:47.05 | mosty | syzygyBSD: are you sure that it's not a bug in your dialplan? invalid timeout or something? |
| 15:47.21 | syzygyBSD | yes |
| 15:47.29 | fetcher | Although, strangely enough one of my PRIs shows callerID even for "blocked" (*67) calls. Misconfiguration on the switch? :) |
| 15:48.10 | syzygyBSD | fetcher: hmm, going to have to try to get my telco to do that, we are a clec and they should... |
| 15:50.04 | *** join/#asterisk kovger (n=kovger@217.27.219.154) |
| 15:53.51 | hedge77 | derp |
| 15:56.52 | fetcher | Will a Polycom 301 run the same sip.ld software image (SIP 2.01) used by a 501/601? |
| 15:57.18 | e-milio | I run out of slots and nothing |
| 15:57.43 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 15:58.17 | e-milio | but the card shows ok in zttool |
| 15:58.54 | e-milio | maybe it is because i have to nics and a agp video |
| 15:59.59 | *** join/#asterisk _-zar-_ (n=zar@125-24-129-193.adsl.totbb.net) |
| 16:00.01 | Mercestes | syzygyBSD, You should always get number identification, syzygyBSD, regardless of CID. |
| 16:00.13 | *** part/#asterisk _-zar-_ (n=zar@125-24-129-193.adsl.totbb.net) |
| 16:00.23 | syzygyBSD | that is what I was thinking, which variable is that |
| 16:00.42 | Mercestes | syzygyBSD, ${EXTEN} |
| 16:00.58 | Mercestes | syzygyBSD, wait, that's what they dialed. Should be wh atyou see in the CLI or in the CDR |
| 16:00.59 | syzygyBSD | lol.. not called number, of course I have that |
| 16:01.16 | syzygyBSD | ya, I will look for it |
| 16:01.18 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 16:07.06 | *** join/#asterisk vlt (n=dm@p54b33719.dip0.t-ipconnect.de) |
| 16:07.11 | aydiosmio | does this channel have the bot that parses s// syntax? |
| 16:07.24 | Qwell[] | no, that's invalid syntax |
| 16:07.34 | Mercestes | This channel does not have a bot that parses s// syntax. |
| 16:07.38 | Mercestes | s/does not/does/ |
| 16:07.50 | Qwell[] | s^s//^s///^ |
| 16:07.51 | aydiosmio | thanks. |
| 16:08.12 | *** join/#asterisk gr1ncheux_ (n=devine@unaffiliated/gr1ncheux) |
| 16:08.31 | *** join/#asterisk Fieldy (i=Y2L9lymz@gentoo/contributor/Fieldy) |
| 16:08.44 | fetcher | "Unknown RTP codec 96 received" <-- is that T.38 ? |
| 16:08.54 | aydiosmio | show codecs? |
| 16:09.53 | aydiosmio | If you are getting messages such as the one below, chances are big you are trying to connect to a Cirpack/v4.39a (gw_sip) sip gateway or to a Mediatrix 1124 gateway |
| 16:09.57 | fetcher | 96 isn't on Asterisk's list. That shows up on calls from a FAX machine (via an ATA), though so I'm suspecting the ATA tries to shift to T.38 |
| 16:10.23 | Mercestes | fetcher: Likely something like that, aye |
| 16:12.54 | *** join/#asterisk MrWup (i=Neil@i-83-67-202-134.freedom2surf.net) |
| 16:13.03 | MrWup | hey guys im completely at sea |
| 16:13.08 | MrWup | i had just set up asterisk |
| 16:13.10 | Qwell[] | completely? |
| 16:13.19 | MrWup | got a few things working, a few SIP handstes conected, calling each other all fine |
| 16:13.31 | MrWup | got an outside line working ok, accepting calls from outside and playing monmkey noises |
| 16:13.36 | MrWup | got the sip handsets to dial out too |
| 16:13.43 | MrWup | but i was having a problem with crackle and echo |
| 16:13.48 | MrWup | so i closed the asterisk server |
| 16:13.52 | MrWup | ran fxotune -i 4 |
| 16:13.57 | MrWup | then fxotune -s |
| 16:14.03 | MrWup | and it "tuned" the channels |
| 16:14.15 | MrWup | and now when i start asterisk and try to dial a zap channel it just gives an error |
| 16:14.40 | *** join/#asterisk phillipk (n=pkey@216.248.143.77) [NETSPLIT VICTIM] |
| 16:15.33 | *** join/#asterisk friedrich| (n=friedric@e177249002.adsl.alicedsl.de) [NETSPLIT VICTIM] |
| 16:15.33 | *** join/#asterisk Juggie (i=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) [NETSPLIT VICTIM] |
| 16:15.34 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) [NETSPLIT VICTIM] |
| 16:15.34 | Vec | Net split :0 |
| 16:15.34 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) [NETSPLIT VICTIM] |
| 16:15.34 | *** join/#asterisk wunderkin (n=kev@dslstat-ppp-95.fastq.com) [NETSPLIT VICTIM] |
| 16:15.34 | *** join/#asterisk aaronr (n=arussell@87.127.234.100) [NETSPLIT VICTIM] |
| 16:15.34 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) [NETSPLIT VICTIM] |
| 16:15.34 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) [NETSPLIT VICTIM] |
| 16:15.34 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
| 16:15.56 | *** join/#asterisk SlamX (n=Sad@AOrleans-157-1-45-214.w90-8.abo.wanadoo.fr) |
| 16:16.02 | SlamX | hi |
| 16:16.03 | Mercestes | I wonder if that happens *every* time he tries to dial a zap |
| 16:16.05 | *** join/#asterisk MrWup (i=Neil@i-83-67-202-134.freedom2surf.net) [NETSPLIT VICTIM] |
| 16:16.24 | SlamX | some one working here on chan_cellphone ? |
| 16:16.32 | Qwell[] | Mercestes: what, a netsplit? :p |
| 16:16.36 | MrWup | but i was having a problem with crackle and echo so i closed the asterisk server ran fxotune -i 4 then fxotune -s and it "tuned" the channels |
| 16:16.43 | MrWup | and now when i start asterisk and try to dial a zap channel it just gives an error |
| 16:16.44 | Makenshi | SlamX, just use an fct |
| 16:16.48 | MrWup | cannot create channel of type zap |
| 16:16.53 | wunderkin | wow it can all fit on one line... almost |
| 16:17.03 | MrWup | and when i do core show channels there are no channels listed |
| 16:17.10 | SlamX | Makenshi are u working on it ? |
| 16:17.30 | Mercestes | Qwell[]: Exactly |
| 16:18.28 | MrWup | im really in a bind here, been searching google and working on it for hous |
| 16:18.34 | MrWup | no end in sight |
| 16:20.00 | aydiosmio | perhaps suicide is finally the answer to one of the questions asked here |
| 16:20.27 | mosty | mrwup: what does ztcfg -vvvv tell you? |
| 16:20.36 | *** part/#asterisk SlamX (n=Sad@AOrleans-157-1-45-214.w90-8.abo.wanadoo.fr) |
| 16:20.40 | Makenshi | SlamX, no i am not working on it because it is pointless |
| 16:20.48 | MrWup | mosty where do i run that from? |
| 16:21.06 | mosty | from your linux shell |
| 16:21.24 | aydiosmio | from your windows cmd prompt |
| 16:21.40 | aydiosmio | sssh, let's see if he does it. |
| 16:23.20 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:23.36 | *** join/#asterisk fbcit (n=cnighswo@nc-71-0-121-24.sta.embarqhsd.net) [NETSPLIT VICTIM] |
| 16:24.26 | MrWup | it says 4 channels configured |
| 16:24.28 | MrWup | no erros |
| 16:24.34 | MrWup | and lists the channels |
| 16:24.38 | MrWup | fxs kewlstarts |
| 16:24.47 | MrWup | (they are fxo sockets) so this is as it should be |
| 16:25.00 | fbcit | Qwell[]: I'm still looking for the association of the chan_skinny.c code and the button layout on the 30VIP. |
| 16:25.10 | Qwell[] | fbcit: most of the buttons do not work |
| 16:25.26 | mosty | mrwuo: and what does "zap show channels" show in the asterisk console? |
| 16:25.51 | wunderkin | they just sit and look pretty |
| 16:26.00 | Qwell[] | wunderkin: nah, it's a pretty ugly phone |
| 16:26.04 | MrWup | 0 configured channels |
| 16:26.07 | MrWup | 0 active channels |
| 16:26.08 | fbcit | Qwell[]: I assume there is no way around it? No code mods that could be done? |
| 16:26.16 | Qwell[] | fbcit: You could fix the buttons, sure |
| 16:26.17 | MrWup | no such thing as zap show channels btw |
| 16:26.22 | MrWup | on mine only core show channels exists |
| 16:26.26 | MrWup | but shows 0 channels |
| 16:26.42 | Mercestes | MrWup: what is the ls -l on zapata.conf |
| 16:26.45 | wunderkin | with a knife |
| 16:27.10 | MrWup | ls -l? what is that? |
| 16:28.12 | Mercestes | MrWup: It destroys your root partition, of course. Same as all other unknown linux commands. |
| 16:28.19 | Mercestes | MrWup: except in this case...it actually lists in "long" format as man ls would tell you and tells me who owns zapata.conf and what it's read/write permissions are. |
| 16:28.23 | fbcit | Qwell[]: I have looked at the code, but fail to see the association between the row,column config in the code and on the phone itself. |
| 16:28.32 | wunderkin | -l: LetLooseSomeExploitToAllowmercestesWithALittleMToHaxorYourBox |
| 16:28.48 | Mercestes | MrWup: my next question would be, what does ps aux |grep -i asterisk says, which also destroys your root parition, until either man or google prove differently. |
| 16:28.51 | Qwell[] | fbcit: 1.4? |
| 16:29.07 | fbcit | Qwell[]: Yes. |
| 16:29.26 | *** join/#asterisk Cybertoy (n=cybertoy@dsl254-123-112.nyc1.dsl.speakeasy.net) [NETSPLIT VICTIM] |
| 16:29.27 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 16:31.50 | fbcit | Qwell[]: I have seven of these phones that are nib. I hate to scrap them and dump money into new ones. I'd rather attempt changing the code. |
| 16:34.05 | MrWup | rw r r |
| 16:34.06 | MrWup | root owns it |
| 16:34.09 | MrWup | and im root |
| 16:34.54 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 16:35.54 | mosty | MrWup: check that the contents of that file are correct |
| 16:37.57 | fbcit | Qwell[]: I'm off to lunch. I'd like to understand how the code relates to the phone itself. Any successful mods I make I'll be glad to contribute to the driver. |
| 16:38.07 | *** join/#asterisk MarkWD (n=MarkWD@rrcs-67-78-88-186.sw.biz.rr.com) |
| 16:38.09 | *** join/#asterisk apardo (n=apardo@87.217.144.110) |
| 16:38.50 | *** join/#asterisk Opperior (n=chatzill@ns2.lannetwork.com) |
| 16:42.49 | *** join/#asterisk corruptor (n=corrupto@styx.mcn.ru) |
| 16:44.09 | anonymouz666 | how do I configure * to handle SIP subscriptions ? |
| 16:44.29 | mosty | google asterisk sip hints |
| 16:45.27 | *** join/#asterisk shinux__ (n=shinux@80.89.187.51) |
| 16:45.47 | anonymouz666 | Mar 28 13:32:43 ERROR[2003]: chan_sip.c:11076 handle_request_subscribe: Got SUBSCRIBE for extension *8@access-level-5 from 192.168.10.75, but there is no hint for that extension |
| 16:46.44 | mosty | anonymouz666: create a hint for that extension |
| 16:48.25 | *** join/#asterisk mmartinn (n=martin@adsl-065-005-200-225.sip.gnv.bellsouth.net) |
| 16:50.00 | *** part/#asterisk jffmriii (n=jffmriii@c-24-1-191-136.hsd1.il.comcast.net) |
| 16:50.01 | anonymouz666 | the damn phone is sending subscribe for every called number, is it right? |
| 16:50.05 | *** join/#asterisk hphinc (n=michael@c-24-126-234-252.hsd1.ga.comcast.net) |
| 16:50.07 | *** join/#asterisk Jave27 (n=jason@cpe-24-33-243-193.twmi.res.rr.com) |
| 16:50.39 | hphinc | does anyone know if you can transfer calls from within the CLI? |
| 16:51.01 | mmartinn | Is there any reason ChanSpy won't go to the extensions I ask it to spy on? I know people are on those extensions because when I scroll through with *, I find them. |
| 16:51.06 | hphinc | or...more specifically, what's the syntax to do it. |
| 16:51.08 | hphinc | ? |
| 16:52.12 | mosty | hphinc: you should probably use the manager interface for that i'd think |
| 16:54.13 | *** part/#asterisk cfh (i=proxyuse@host197-137.pool8290.interbusiness.it) |
| 16:54.26 | *** join/#asterisk SamBr (n=sam@kmg2.kpm-net.com) |
| 16:54.49 | *** join/#asterisk AdamCrowe (n=newbie@host-87-74-27-138.bulldogdsl.com) |
| 16:54.55 | AdamCrowe | In PTMP mode with B410P, I can make outbound calls but cannot receive any calls at all. All I get is a message from mISDN: GOT IGNORE SETUP |
| 16:54.57 | AdamCrowe | GOT IGNORE SETUP |
| 16:54.58 | AdamCrowe | CC_RELEASE_COMPLETE|CONFIRM [TE] |
| 16:54.58 | AdamCrowe | release_chan: Ch not found! |
| 16:54.58 | AdamCrowe | CC_RELEASE_COMPLETE|CONFIRM [TE] |
| 16:55.00 | AdamCrowe | release_chan: Ch not found! |
| 16:55.12 | AdamCrowe | Anybody know how to fix this??? |
| 16:57.05 | AdamCrowe | And also, every once in a while, I get a message saying: "l2 D-channel frame wrong EA0/EA1" |
| 16:58.07 | SamBr | Having Trouble with IVR and accepting digits. Asterisk 1.2.12.1 and TrixBox. Everything works correctly except that you have to wait until the message is done playing before you can dial an extention or if using the directory you have to wait until the instructions are done before you try to dial the person's last name. I looked around at the background() command but that is already set. If someone can point me in the correction direction I would apprec |
| 16:58.45 | Gido-E | SamBr dtmf mode? |
| 16:58.53 | SamBr | dtmf inband |
| 16:59.11 | Gido-E | dtmfmode=rfc2833 |
| 16:59.17 | Gido-E | did you trie that? |
| 16:59.28 | Gido-E | try |
| 16:59.29 | SamBr | Yes I tried rfc2833 |
| 16:59.47 | echo- | what kind of phones, SamBr? |
| 16:59.49 | SamBr | The digits work but only after the message is done playing |
| 17:00.21 | SamBr | Phones are Polycom 501. But this problem is with incoming callers. The Polycom phones are able to navigate the Menu tree with no issues. |
| 17:00.34 | Mercestes | SamBr: did you even read the topic? |
| 17:00.49 | SamBr | read the topic? |
| 17:01.08 | Mercestes | novel concept, I know |
| 17:01.32 | SamBr | read the topic from where? I just spent the last 4 hours doing searches with google. |
| 17:01.34 | hphinc | mosty: I was trying to do it from the CLI when I want to steal a call from another extension. |
| 17:01.35 | Mercestes | that part at the top that says something about trixbox support being in #freepbx. |
| 17:01.54 | Mercestes | and. you need to call an Answer() first before trying to accept DTMF btw. |
| 17:01.59 | hphinc | ocassionally we have someone in our office that doesn't transfer a call, but puts them on hold instead. |
| 17:02.06 | mosty | hphinc: why from the cli? |
| 17:02.17 | hphinc | so I want to type transfer that call here in the CLI and have it work. |
| 17:02.20 | Mercestes | othrewise you cannot recieve DTMF only play audio out. |
| 17:02.34 | hphinc | because I keep the CLI open all day long on one of my montors, and it would be conveneient. |
| 17:02.38 | mosty | hphinc: i think the manager interface is designed for that sort of thing |
| 17:02.38 | Mercestes | SamBr: So you wanna Wait(1) Answer() Wait(1) Background(sound) |
| 17:03.04 | JunK-Y | instead of answer() and wait(1), ya can pass that 1 to the answer |
| 17:03.10 | JunK-Y | answer(time to wait) |
| 17:03.19 | *** join/#asterisk gbodemantv (n=gbodeman@corpex.pivotal.televerde.com) |
| 17:03.19 | Mercestes | lies! |
| 17:03.25 | gbodemantv | hello all |
| 17:03.28 | echo- | is there a channel that is Grandstream specific? I'm having an issue with GXP2000's and 802.1q. |
| 17:03.35 | gbodemantv | is anyone having trouble compiling 1.2.17 |
| 17:03.41 | hphinc | so the AMI is going to be a better resource to accomplish that task than the CLI? |
| 17:03.42 | gbodemantv | will not make |
| 17:03.46 | JunK-Y | answer(1000) if you prefer ;) |
| 17:03.46 | Mercestes | echo-: Try #masochism |
| 17:04.03 | gbodemantv | get codec_zap.c:681: error: dereferencing pointer to incomplete type |
| 17:04.16 | JunK-Y | hphinc: particulary for web stuff. |
| 17:04.16 | mosty | hphinc: yes, i think various manager clients allow you to do that sort of thing easily |
| 17:04.26 | gbodemantv | tmake[1]: *** [codec_zap.o] Error 1 |
| 17:04.33 | mmartinn | Is there any reason ChanSpy won't go to the extensions I ask it to spy on? I know people are on those extensions because when I scroll through with *, I find them. |
| 17:04.47 | gbodemantv | finally make[1]: Leaving directory `/usr/src/asterisk-1.2.17/codecs' |
| 17:04.47 | gbodemantv | make: *** [subdirs] Error 1 |
| 17:04.55 | gbodemantv | any ideas? |
| 17:05.02 | mosty | mmartinn: how are you specifying the cannel? |
| 17:05.07 | mosty | channel, even |
| 17:05.18 | SamBr | Mercestes: ext-local is an include in the [ivr-10] file of the extentions_additonal.conf and ext-local is the fiel that just list all of the extentions |
| 17:05.22 | mmartinn | mosty: exten => 321,n,ChanSpy(SIP/20) |
| 17:05.35 | mmartinn | mosty: We want them, if they dial 24, to go to SIP/2024 |
| 17:06.17 | mosty | mmartinn: does it work if you specify the entire sip extension in the chanspy command? |
| 17:06.21 | *** join/#asterisk kyoorius (n=kyoorius@ool-44c5a05a.dyn.optonline.net) |
| 17:06.27 | mmartinn | mosty: Yes |
| 17:06.40 | mmartinn | mosty: I tried, with success: exten => _321XX,n,ChanSpy(SIP/20${EXTEN:3}) |
| 17:07.02 | mmartinn | mosty: I also tried just exten => 321,n,ChanSpy(SIP/2024) |
| 17:07.06 | mmartinn | mosty: with success |
| 17:07.27 | mmartinn | mosty: And using the asterisk to scroll through channels can find SIP/2024 just fine |
| 17:07.32 | *** join/#asterisk tkowal (n=nospamto@74.93.82.14) |
| 17:07.50 | mosty | then why not use some dialplan logic to read the last 2 digits and build the working ChanSpy call? |
| 17:07.56 | *** join/#asterisk SamBr (n=sam@kmg2.kpm-net.com) |
| 17:08.14 | mmartinn | mosty: We are. I just wondered why the ChanSpy(SIP/20) wasn't working -- it used to |
| 17:08.49 | kyoorius | hi, i'm having some issues with latency. |
| 17:09.02 | mmartinn | mosty: Is that DTMF issue or a dialplan one? |
| 17:09.11 | mmartinn | s/that/that a/ |
| 17:09.32 | kyoorius | just ran some tests here. I'm using a bluetooth headset->x-lite softphone->asterisk->x100p fxo |
| 17:09.43 | mosty | mmartinn: sounds like a chanspy issue to me |
| 17:10.06 | mmartinn | mosty: k, we'll settle for dialplan solution then :) I don't want to get into chanspy :) |
| 17:10.13 | mosty | mmartinn: btw do you know of a way to exit from chanspy without hanging up? |
| 17:10.31 | gbodemantv | anyone else having troubles with compiling asterisk 1.2.17 in fedora 5 |
| 17:10.34 | mmartinn | mosty: hmmm... I don't, but ZapBarge might have a way |
| 17:11.01 | kyoorius | i made an outside call, picked up a physical extension of the landline, made a popping sound, and recorded the orginating sound and also the landline at the same time. 130ms delay |
| 17:11.17 | kyoorius | gbodemantv. any reason why you're compiling instead of using yum? |
| 17:11.20 | heison | Asterisk 1.4.2 crashes when it invokes rxfax (with spandsp-0.0.3pre28 / 20061206 / 20070327), i have enabled debug on console via logger.conf: [Mar 28 12:59:25] DEBUG[2871]: chan_zap.c:4790 zt_read: DTMF digit: f on Zap/5-1 |
| 17:11.21 | heison | <PROTECTED> |
| 17:11.21 | heison | [Mar 28 12:59:25] DEBUG[2871]: pbx.c:1795 pbx_extension_helper: Launching 'RxFAX' |
| 17:11.21 | heison | <PROTECTED> |
| 17:11.21 | heison | [Mar 28 12:59:25] DEBUG[2871]: channel.c:2845 set_format: Set channel Zap/5-1 to read format slin |
| 17:11.22 | heison | [Mar 28 12:59:25] DEBUG[2871]: channel.c:2845 set_format: Set channel Zap/5-1 to write format slin |
| 17:11.23 | mosty | mmartinn: i need it for sip extensions :/ oh well |
| 17:11.24 | heison | asterisk*CLI> |
| 17:11.27 | heison | Disconnected from Asterisk server |
| 17:11.32 | heison | Executing last minute cleanups |
| 17:11.34 | mmartinn | mosty: =/ |
| 17:11.56 | mmartinn | mosty: Before this version of Asterisk, we couldn't get chanspy to work at all |
| 17:15.35 | *** join/#asterisk Slingky (n=na@modemcable199.182-200-24.mc.videotron.ca) |
| 17:16.43 | gbodemantv | kyoorius - its what I have laways done, honestly did not know I could yum |
| 17:17.30 | kyoorius | yum install asterisk |
| 17:17.37 | kyoorius | thats all there is to it! |
| 17:18.15 | kyoorius | caveat: it doesnt come with chan_zap.so |
| 17:18.36 | kyoorius | in the end, I did have to compile asterisk to get that and then copy it to /usr/lib/asterisk/modules/ |
| 17:18.40 | Strom_C | caveat: it's probably an old version |
| 17:18.45 | kyoorius | but if you arent using zaptel it doesnt matter. |
| 17:19.01 | gbodemantv | I am |
| 17:22.04 | JunK-Y | heison: report it to the author with backtraces. |
| 17:23.02 | anonymouz666 | anyone that use FOP know if it is possible to monitor a channel (chanspy)? |
| 17:23.08 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 17:24.26 | Slingky | hi guys! could someone point me to a fax solution for asterisk (fax to email) and viceversa |
| 17:24.31 | *** join/#asterisk w9sh (n=w9sh@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 17:24.31 | Slingky | like this: http://nerdvittles.com/index.php?p=88 |
| 17:25.21 | SamBr | Dial by Extention does not work while recorded message for the menu is being played. After the message finished the dial by extention does work. I looked at the full log and it looks like the IVR is trying to process the extention as a menu option so when dialing 234 it shows as no menu for 2 then no menu for 3 etc dtmfmode=inband |
| 17:25.36 | mosty | slinkgy: what connection to the phone system are you planning on using? |
| 17:26.13 | Slingky | mosty, i use acanac, but right now i use primus with a voip adapter and standard fax |
| 17:26.13 | *** join/#asterisk CrashHD (n=crashhd@c-67-166-155-233.hsd1.ca.comcast.net) |
| 17:26.38 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 17:27.22 | mosty | but what pstn connection are you faxing over? a modem? |
| 17:27.22 | *** join/#asterisk delight85 (n=delight@ip215.9.1411M-CUD12K-01.ish.de) |
| 17:27.22 | Slingky | mosty, is asterfax the best solution ? |
| 17:27.27 | mosty | hylafax is my preferred fax sever |
| 17:29.17 | delight85 | sorry for interruption but i have some serious problmens getting my cisco 7960 to work with asterisk...does anyone have some time? |
| 17:29.45 | Slingky | mosty, is it compatible with asterisk ? i mean i want something free and integrated if possible |
| 17:30.41 | mosty | slingky: no, it's a seperate server |
| 17:33.11 | gbodemantv | zaptel would not compile either |
| 17:33.21 | fbcit | Is anyone using an icecast stream for moh? |
| 17:33.31 | kyoorius | hbodemantv: yum install zaptel |
| 17:33.41 | gbodemantv | had to modify a xbus-core.c file to reflect kernel |
| 17:33.48 | *** join/#asterisk Fieldy (i=hspS2ojb@gentoo/contributor/Fieldy) |
| 17:34.01 | kyoorius | gbode: point your yum.conf to http://dl.atrpms.net/fc5-i386/atrpms/stable |
| 17:36.07 | gbodemantv | yum does not work for m |
| 17:36.07 | gbodemantv | me |
| 17:36.07 | kyoorius | do you have it installed? |
| 17:36.12 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@ppp-70-255-140-128.dsl.hstntx.swbell.net) |
| 17:36.15 | kyoorius | i used to build everything from scratch, but managing upgrades and all become unwieldly after a while |
| 17:36.31 | kyoorius | rpm and yum really take the headache out of upgrades and removing stuff. |
| 17:37.02 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 17:38.22 | delight85 | does anyone has an idea why i get an chan_sip.so:undefined symbol: ast_pickup_ext & terminating asterisk, when trying to make a call? |
| 17:38.26 | gbodemantv | if I try to yum is saus I do not have anything to update |
| 17:38.38 | gbodemantv | I am trying to go from 1.2.13 to 17 |
| 17:38.50 | gbodemantv | i typed yum install asterisk |
| 17:39.29 | kyoorius | is your yum.conf configured? |
| 17:39.58 | gbodemantv | i can yum other things |
| 17:40.00 | kyoorius | http://atrpms.net/install.html |
| 17:40.04 | *** join/#asterisk antotrace (n=antotrac@host211-151-dynamic.3-87-r.retail.telecomitalia.it) |
| 17:41.00 | Strom_C | yum may only have 1.2.13 as the latest version |
| 17:41.35 | kyoorius | that repository installed asterisk-1.4.2-36.fc5.at for me yesterday. |
| 17:42.06 | gbodemantv | I don't want to go to 1.4 yet |
| 17:42.12 | gbodemantv | that would be a problem |
| 17:42.14 | syzygyBSD | 81 missed calls... I don't think they were important, they would have left a voicemail right? |
| 17:42.34 | kyoorius | do a "yum search asterisk" and save the output. i think i saw asterisk 1.2 in there. |
| 17:43.47 | gbodemantv | only find libpri for 1.4 |
| 17:46.32 | kyoorius | configure your yum.conf as described in the url above. |
| 17:47.46 | *** part/#asterisk BruXo (n=celio@201.17.146.166) |
| 17:48.39 | echo- | Can anyone tell me what the following lines in my asterisk logs mean: |
| 17:48.43 | echo- | Mar 28 13:47:52 DEBUG[3694] chan_sip.c: Stopping retransmission on '03d28bdd7f84914a33a03b5929561046@10.1.1.226' of Request 102: Match Found |
| 17:49.12 | echo- | repeated about 20 times once a minute |
| 17:49.19 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 17:50.16 | gbodemantv | changing yum.conf adds 1.4 only |
| 17:53.38 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-021f1d796161c25a) |
| 17:53.41 | echo- | that's on 1.2.14, btw. |
| 17:54.21 | AdamCrowe | In PTMP mode with B410P, I can make outbound calls but cannot receive any calls at all. All I get is a message from mISDN: GOT IGNORE SETUP |
| 17:54.22 | AdamCrowe | GOT IGNORE SETUP |
| 17:54.22 | AdamCrowe | CC_RELEASE_COMPLETE|CONFIRM [TE] |
| 17:54.22 | AdamCrowe | release_chan: Ch not found! |
| 17:54.23 | AdamCrowe | CC_RELEASE_COMPLETE|CONFIRM [TE] |
| 17:54.25 | AdamCrowe | release_chan: Ch not found! |
| 17:54.27 | AdamCrowe | Anybody know how to fix this??? |
| 17:59.50 | AdamCrowe | FIXED!!!! HAHAHAH!!! |
| 18:00.09 | Strom_C | what was the solution |
| 18:00.23 | AdamCrowe | It's all misdn.conf problems... |
| 18:01.07 | AdamCrowe | I am now experiencing something weird... An incoming call is registered both on line one and two on my Grandstream 2000... Any ideas? |
| 18:03.11 | Mercestes | AdamCrowe, Is line one and Line 2 the same username?? |
| 18:03.26 | AdamCrowe | Line two is disabled... :/ |
| 18:04.03 | AdamCrowe | It only acts as another virtual line for the same user... To make more than one call in parallel... |
| 18:04.03 | *** join/#asterisk _osiris (i=ob30ut6r@bsd02.evansengineering.net) |
| 18:05.04 | Mercestes | That wuold be why it rings both lines |
| 18:05.41 | AdamCrowe | But why not for the other two line that the Grandstream has? (it has four virtual lines) |
| 18:05.51 | antotrace | i've got a question for the channel |
| 18:05.56 | antotrace | as new feature |
| 18:06.07 | antotrace | can i explain the scenario ? |
| 18:07.23 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-155-97-204.red.bezeqint.net) |
| 18:09.12 | *** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru) |
| 18:09.22 | kyoorius | has anyone here experimented or measured the latency on their systems? |
| 18:09.42 | kyoorius | I'm getting 130ms between my headset and the landline. |
| 18:09.46 | *** join/#asterisk boort (n=a@IGLD-84-228-220-247.inter.net.il) |
| 18:09.52 | kyoorius | which i think is quite a lot. |
| 18:11.22 | boort | could someone point me to some instructions on how to set up two pstn lines (connected via spa3102) so that asterisk uses whichever line is currently free? |
| 18:15.06 | *** join/#asterisk dahunter3 (n=dahunter@static-71-189-121-177.lsanca.dsl-w.verizon.net) |
| 18:15.08 | *** join/#asterisk Braxus (n=braxus@66.147.214.164) |
| 18:15.19 | Mercestes | boort: Put them in a group |
| 18:15.25 | Mercestes | boort: or just use two dial statements |
| 18:15.35 | *** join/#asterisk wubba (n=kmurrey@adsl-76-211-157-158.dsl.akrnoh.sbcglobal.net) |
| 18:15.37 | dahunter3 | Is there a good link to how I can debug a red alarm on a TE110P? |
| 18:16.39 | *** join/#asterisk tuan_modulis (n=chatzill@3-82-252-216-static.enter-net.com) |
| 18:16.52 | wubba | Is Asterisk Now using the newest Asterisk version? |
| 18:17.07 | aydiosmio | would be sad if it weren't, wouldn' tit? |
| 18:17.12 | *** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com) |
| 18:17.21 | wubba | It would be - so I take it that it is. |
| 18:18.45 | *** join/#asterisk Dovid (n=Dovid@89-138-9-217.bb.netvision.net.il) |
| 18:18.51 | EmleyMoor | Has anyone got dring detection and caller ID working together from BT into Asterisk yet? Also, how can I use a BT-style caller display box on my FXS ports? |
| 18:26.44 | heison | JunK-Y: you there/ |
| 18:27.51 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 18:28.10 | JunK-Y | busy |
| 18:32.25 | EmleyMoor | Can someone help me understand something about the "reverse hold" example on the wiki? I want to know what the caller ID stuff achieves |
| 18:38.25 | *** join/#asterisk itaro (n=dmiller@firewall.edm1.com) |
| 18:39.13 | *** part/#asterisk itaro (n=dmiller@firewall.edm1.com) |
| 18:39.30 | *** join/#asterisk Rhab (n=Rhab@p3EE0CF8B.dip.t-dialin.net) |
| 18:39.53 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:39.54 | Rhab | hi |
| 18:40.35 | Rhab | i compiled and installed asterisk 1.2.17 |
| 18:40.54 | Rhab | and when i type asterisk it says" illegal instruction" |
| 18:41.00 | Rhab | whys that? |
| 18:41.18 | Rhab | btw i got a debian server install with kernel 2.6.8 |
| 18:41.39 | Rhab | !help |
| 18:41.40 | *** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse) |
| 18:41.55 | _ys | Rhab http://bugs.digium.com is best place for this question |
| 18:42.39 | Rhab | hm k |
| 18:43.01 | JunK-Y | u using the deb package or you compiled everything? |
| 18:43.25 | Rhab | i compiled it manually since the stable package was too old for my needs |
| 18:43.48 | JunK-Y | no errors at all during compilation? |
| 18:44.12 | Rhab | well i dont think so |
| 18:44.22 | Rhab | make install and make samples also worked |
| 18:44.29 | tuan_modulis | what versions of asterisk is asterisk now compatible with? |
| 18:44.34 | Rhab | the config files are there |
| 18:44.47 | JunK-Y | and asterisk -c is saying illegal "instruction", nothing more? |
| 18:44.47 | *** join/#asterisk yogurt2ungue (n=yogurt2u@150-52-231-201.fibertel.com.ar) |
| 18:45.04 | Rhab | yes nothing more |
| 18:45.12 | Rhab | i did a strace |
| 18:45.22 | Rhab | but the output doesnt help me |
| 18:45.31 | Rhab | cause i dont understand it |
| 18:45.32 | JunK-Y | never saw that. |
| 18:45.39 | JunK-Y | debian etch 32 bits? |
| 18:45.56 | Rhab | no its stable |
| 18:46.19 | Rhab | 2.6.8.-3-386 |
| 18:46.27 | *** part/#asterisk kyoorius (n=kyoorius@ool-44c5a05a.dyn.optonline.net) |
| 18:46.41 | JunK-Y | try a make distclean && ./configure && make && make install |
| 18:46.51 | JunK-Y | if its still an issue, submit a bug with ur strace. |
| 18:47.26 | Rhab | do i need to ./configure in1.2 ? |
| 18:47.36 | Rhab | i didnt do that |
| 18:47.54 | JunK-Y | ha, 1.2! |
| 18:48.02 | JunK-Y | then make && make install |
| 18:48.10 | JunK-Y | 1.4 has ./configure |
| 18:48.18 | Rhab | thats what i did |
| 18:48.23 | Rhab | make and make install |
| 18:48.41 | JunK-Y | submit a bug. |
| 18:48.50 | _ys | Rhab no ./configure script in 1.2, simple try 'gmake clean && gmake && gmake install' |
| 18:49.07 | JunK-Y | report infos regarding ur compilers. |
| 18:49.27 | wubba | Is there software out there to integrate with MS Outlook? |
| 18:49.30 | JunK-Y | _ys: hes on debian, not freebsd. |
| 18:49.39 | Rhab | hm i installed kernel-package and did a apt-get install gcc |
| 18:49.56 | Rhab | make wasnt availive at first |
| 18:50.12 | Rhab | available |
| 18:51.04 | JunK-Y | you're using deb package make right? |
| 18:51.36 | _ys | JunK-Y it is not important |
| 18:51.53 | Rhab | yes JunK-Y |
| 18:52.21 | JunK-Y | just submit a bug. |
| 18:52.46 | _ys | Rhab that CPU on You box? |
| 18:52.59 | _ys | s/on/in/ |
| 18:53.13 | Rhab | ^^ |
| 18:53.20 | EmleyMoor | How do I pick up the caller ID of the transferring party rather than the originally calling party, in the dialplan? |
| 18:53.22 | Rhab | its a VIA 533 MHz |
| 18:54.00 | Rhab | is there a comand to find it out more detailed? |
| 18:54.15 | JunK-Y | cat /proc/cpuinfo |
| 18:54.30 | _ys | .. and uname -a |
| 18:55.03 | Rhab | k |
| 18:55.16 | JunK-Y | this is probably due to ur cpu. |
| 18:55.24 | Rhab | im compiling again |
| 18:55.31 | JunK-Y | ive never touched a via cpu, so cant help. |
| 18:56.03 | Rhab | and it says many times "march=i686" but i have 386 headres |
| 18:56.11 | Rhab | mbe this is a prob |
| 18:56.29 | *** join/#asterisk bkuhn (n=bkuhn@fsf/member/bkuhn/bkuhn) |
| 18:58.50 | JunK-Y | Rhab: which via is it exactly? |
| 18:59.08 | Mercestes | Rhab, likely...what distro? |
| 18:59.20 | JunK-Y | hes on debian. |
| 18:59.24 | Rhab | ja |
| 18:59.26 | Mercestes | ah |
| 18:59.36 | JunK-Y | which model of cpu exactly? |
| 18:59.43 | Rhab | ill try to find out more about my cpu when the compiling is finished |
| 19:00.14 | JT | a via 533 is not a 386 |
| 19:00.15 | Rhab | i installed with debian-31r5-i386-netinst.iso |
| 19:00.36 | Rhab | JT_: really? so what is irt? |
| 19:00.47 | JT | it's a C3 isn't it? |
| 19:00.49 | JT | i586 arch iirc |
| 19:00.59 | JT | you can compile a kernel with specific optimisations for it |
| 19:01.08 | _ys | Rhab :) try ctrl-Z and cat /proc/cpuinfo |
| 19:01.13 | _ys | fg |
| 19:01.19 | Rhab | ja |
| 19:02.19 | Rhab | processor: 0 |
| 19:02.31 | _ys | 0? |
| 19:02.34 | Rhab | vendor_id: centaurhauls |
| 19:02.44 | JunK-Y | pastebin all this |
| 19:02.44 | Rhab | cpe family :6 |
| 19:02.45 | JunK-Y | ~pb |
| 19:02.47 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 19:02.57 | Rhab | model name: VIA samuel 2 |
| 19:03.06 | JT | _ys: what's unusual about that? |
| 19:03.27 | Rhab | sry |
| 19:03.37 | Rhab | w8 a mom |
| 19:03.39 | _ys | Rhab vendor_id? |
| 19:03.53 | Rhab | CentaurHauls |
| 19:04.10 | JT | _ys: why do you need to repeat back everything Rhab says? |
| 19:04.12 | Rhab | that means Ce? |
| 19:04.19 | Rhab | 3 |
| 19:05.57 | Rhab | i dont get this pastebin stuff |
| 19:06.00 | _ys | JT No, but strange information from cpuinfo. |
| 19:06.20 | Rhab | mbe i just look in the motherboard manual |
| 19:06.22 | JT | _ys: cpu 0 is not strange information |
| 19:06.43 | _ys | JT: "vendor_id" |
| 19:07.10 | JunK-Y | _ys: you should do a local cat /proc/cpuinfo |
| 19:07.13 | Rhab | ok here it is |
| 19:07.19 | JT | so? it's not intel or amd, we've established that |
| 19:07.31 | Rhab | CPU: VIA C3 EBGA 533 |
| 19:07.43 | Rhab | so 386 is awfully wrong? |
| 19:07.53 | _ys | [ys@rlk ~]$cat /proc/cpuinfo |
| 19:07.55 | _ys | processor : 1 |
| 19:08.03 | _ys | vendor_id : GenuineIntel |
| 19:08.05 | _ys | hm |
| 19:08.08 | CunningPike | If asterisk starts saying "Can't destroy call <nnnnn>" over and over, is that bad? |
| 19:08.12 | JT | _ys: so? |
| 19:08.25 | JT | _ys: seriously, i don't see where you're heading |
| 19:09.04 | _ys | JT: "processor : 1" mb old vers |
| 19:09.13 | JT | _ys: how many processors do you have, do you have hyperthreading, multiple cores? |
| 19:09.28 | _ys | JT: yes |
| 19:09.29 | JT | linux has always started at proc 0 |
| 19:09.30 | JunK-Y | it should start at 0, for ur first one |
| 19:09.54 | *** join/#asterisk sysreq (n=sysreq@41-198-0-72-ppp.3menatwork.com) |
| 19:10.04 | JT | _ys: the answer to the question should be numerical, not "yes" :) |
| 19:10.28 | Rhab | [21:09] Rhab: hey when i isntalled debian with 386 iso and i have a via c3 cp - this should be the main problem, shouldnt it? |
| 19:10.40 | JT | Rhab: doubt it |
| 19:10.53 | JT | i386 is just not optimised |
| 19:11.02 | _ys | JT: 2 (4) :) |
| 19:11.20 | JT | _ys: so it'd go 0, 1, 2, 3 |
| 19:11.51 | _ys | JT: yes, You right |
| 19:12.16 | _ys | but cpu 0 is out of scroll buffer :( |
| 19:12.37 | JunK-Y | cat /proc/cpuinfo |more |
| 19:12.49 | JT | |less |
| 19:13.05 | Rhab | mom |
| 19:13.12 | JunK-Y | more! |
| 19:13.13 | JunK-Y | :) |
| 19:13.21 | JunK-Y | more > less |
| 19:13.22 | JunK-Y | :) |
| 19:13.30 | JT | JunK-Y: stuck in dos ways are we? |
| 19:13.54 | JunK-Y | in dos or DoS ?! |
| 19:14.06 | Rhab | how can i copy the output of that cat ? |
| 19:14.22 | JT | with a mouse |
| 19:14.30 | *** join/#asterisk beek (n=klinebl@64.9.22.203) |
| 19:14.32 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 19:14.38 | JunK-Y | cat /proc/cpuinfo > rhab.txt |
| 19:14.43 | JunK-Y | then open that file. |
| 19:14.47 | JT | dont copy it to here though |
| 19:15.10 | JT | i don't see how that will help him copy the text ;) |
| 19:15.17 | JT | seems a bit unnecessary |
| 19:15.27 | Rhab | ?? |
| 19:15.59 | JunK-Y | JT: i will let ya help him, ive some jobs to get done. |
| 19:16.04 | JT | Rhab: select the text with a mouse... you'll probably need to be remotely logged into your system using ssh |
| 19:16.30 | Rhab | yaya im working on it w8 a sec |
| 19:16.54 | syzygyBSD | Rhab: how do you want to copy it? |
| 19:17.41 | _ys | Rhab google say me, that this cpu is like P1 - p3 , try "march=i386" |
| 19:18.15 | *** join/#asterisk `p4r14h`work (n=josh@24-119-48-78.cpe.cableone.net) |
| 19:18.15 | JT | _ys: like i've said before a few times, i586 is mosst optimal, but i386 should work fine |
| 19:18.33 | beek | Is this an appropriate channel to ask a question about Asterisk::AGI? |
| 19:18.48 | _ys | JT, not sure ... |
| 19:18.48 | Rhab | i did march=i386 |
| 19:18.52 | Rhab | what is it for? |
| 19:18.53 | JT | i am sure |
| 19:18.56 | JT | it's a VIA C3 |
| 19:19.07 | Rhab | y |
| 19:19.14 | JT | why do you even need to play with march= settings anyway? |
| 19:19.24 | Rhab | ppl from debian channel told me i386 is the right for my cpu |
| 19:19.45 | JT | it will work, yes, it won't take advantage of all optimisations though |
| 19:19.48 | JunK-Y | Rhab: just use 1.4 and ./configure ! |
| 19:20.02 | Rhab | yes ill try it |
| 19:20.05 | JT | Rhab: why are you trying to set march? |
| 19:20.18 | Rhab | [21:17] _ys: Rhab google say me, that this cpu is like P1 - p3 , try "march=i386" |
| 19:20.43 | JT | it should work without you manually setting march |
| 19:20.52 | Rhab | how can i get rid of the installed but not working asterisk 1.2? |
| 19:21.54 | JunK-Y | just install 1.4 |
| 19:23.02 | JT | meh, avoid 1.4 unless you need it :P |
| 19:25.56 | Rhab | how find out the ip of my server? |
| 19:26.07 | Rhab | its configured with dhcp |
| 19:27.15 | _ys | Rhab ifconfig ? |
| 19:27.23 | Rhab | ija :) |
| 19:29.37 | *** join/#asterisk syzygyBSD_ (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 19:30.36 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 19:36.30 | *** join/#asterisk harleya (n=harleya@207.108.166.2) |
| 19:36.58 | *** join/#asterisk zoa (n=zoa@pirus.securax.be) |
| 19:37.03 | zoa | yo yo |
| 19:37.05 | Rhab | why is damn ssh not working? |
| 19:38.29 | Rhab | im moving in circles for 2 hours |
| 19:38.33 | Rhab | tt |
| 19:38.58 | *** join/#asterisk mrbnet (n=sureal@corpmail1.mrbnetworks.com) |
| 19:39.07 | syzygyBSD | is it on? |
| 19:39.21 | syzygyBSD | plugged in? |
| 19:39.44 | Rhab | "connection closed by remote host" |
| 19:39.56 | Rhab | i think i forgot something very simple |
| 19:40.41 | Gido-E | ssh -v |
| 19:42.10 | zoa | anybody using idefisk in here ? |
| 19:42.58 | inv_arp[work] | Rhab: just tryin to simply login? |
| 19:43.45 | Rhab | i use telnet now |
| 19:43.56 | Rhab | i installed the package |
| 19:44.04 | Rhab | (on server) |
| 19:44.32 | Rhab | is there any telnet deamon i need to run |
| 19:45.01 | syzygyBSD | telnet? |
| 19:45.03 | syzygyBSD | really? |
| 19:45.08 | Rhab | ja |
| 19:45.47 | syzygyBSD | I can't tell you in good conscious to run `/etc/init.d/telnet start` |
| 19:46.21 | Rhab | i give a damn for encryption atm |
| 19:47.06 | *** join/#asterisk Strom_M (n=strom@70.141.71.195) |
| 19:47.14 | Rhab | its on my server and im behind a router |
| 19:47.22 | *** join/#asterisk PupenoR (n=pupeno@2002:c87b:b75b:0:240:f4ff:fe6b:7650) |
| 19:47.34 | Rhab | on my server is nothing interesting at all atm |
| 19:48.42 | Rhab | puuuh |
| 19:48.53 | Rhab | that telnet at init.d doesnt exist |
| 19:49.04 | JunK-Y | rhab: just apt-get install ssh? |
| 19:49.31 | Rhab | ja |
| 19:50.04 | Rhab | it asks me to remove my running kernel image |
| 19:50.18 | *** join/#asterisk PupenoR (n=pupeno@2002:c87b:b75b:0:240:f4ff:fe6b:7650) |
| 19:50.23 | Rhab | there must be something seriously wrong with my system |
| 19:50.35 | Rhab | is totaly fucked up |
| 19:50.38 | JunK-Y | wtf ssh is related with kernel? |
| 19:50.40 | JunK-Y | hehe |
| 19:50.50 | Rhab | however i just instlled it yesterda< |
| 19:50.56 | friedrich| | maybe you just don't have any clue at all. |
| 19:51.01 | Rhab | <PROTECTED> |
| 19:51.09 | Rhab | lol |
| 19:51.52 | Rhab | I rather use the master switch and go to bed |
| 19:51.55 | Rhab | ... tt |
| 19:52.02 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 19:52.03 | Rhab | thanks for help |
| 19:52.03 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:52.14 | blitzrage | For anyone interested: http://digg.com/gadgets/Bounty_offered_for_Asterisk_PBX_on_Apple_TV |
| 19:52.16 | Rhab | but im screwed atm |
| 19:52.53 | Qwell[] | blitzrage: 2 words... |
| 19:52.57 | Qwell[] | why? :P |
| 19:53.05 | blitzrage | just because :) |
| 19:53.19 | Rhab | this mine lovely debian installation sucks horse cocks |
| 19:53.20 | blitzrage | although $299 for that kind of hardware is pretty decent |
| 19:53.21 | Qwell[] | sokol bounty? or did he just post it? |
| 19:53.24 | inv_arp[work] | Rhab: maybe system resources comsumed.. |
| 19:53.32 | inv_arp[work] | too max that is.. |
| 19:53.36 | Rhab | whatever |
| 19:53.50 | Rhab | tomorow its erased |
| 19:54.01 | gbodemantv | anyone having trouble compiling 1.2.17 |
| 19:54.06 | gbodemantv | I cannot get it to work |
| 19:54.07 | Rhab | and ill try the testing branch |
| 19:54.14 | gbodemantv | even the svn |
| 19:54.15 | blitzrage | Qwell[]: looks to be a sokol bounty |
| 19:55.06 | Rhab | bb tnx 4 help |
| 19:55.09 | *** part/#asterisk Rhab (n=Rhab@p3EE0CF8B.dip.t-dialin.net) |
| 19:55.20 | zoa | yeah |
| 19:55.21 | zoa | wtf |
| 19:55.28 | syzygyBSD | why are polycom speaker phones so much better than the competition? |
| 19:55.36 | JunK-Y | <--- SIP read from 10.2.94.245:5060 ---> |
| 19:55.36 | JunK-Y | SIP/2.0 200 OK |
| 19:55.36 | JunK-Y | Via: SIP/2.0/UDP 10.2.117.179:5060;branch=z9hG4bK7f6217ec;received=10.2.117.179;rport=5060 |
| 19:55.37 | zoa | because they have better marketing |
| 19:55.40 | zoa | and they look ugly |
| 19:55.40 | zoa | :) |
| 19:55.49 | JunK-Y | what may force the rtp passing thru the other-end localnet? |
| 19:56.04 | syzygyBSD | well, I have 3 different phones on my desk, polycom is by far the best |
| 19:56.13 | blitzrage | syzygyBSD: they seem to work, have good Asterisk support, and seem to do most things fine for the price point |
| 19:56.35 | zoa | Qwell: http://www.asteriskguru.com/idefisk/ you might want to upgrade |
| 19:56.41 | zoa | it should fix all known issues |
| 19:56.43 | Qwell[] | k |
| 19:58.40 | syzygyBSD | is asterisk mostly written in C or C++? |
| 19:58.45 | Qwell[] | c |
| 19:59.04 | zoa | its mostly written in VI |
| 19:59.12 | blitzrage | and emacs |
| 19:59.20 | Qwell[] | nano all the way |
| 19:59.21 | *** join/#asterisk bkruse (i=root@nat/digium/x-fb296bd652a18c9c) |
| 19:59.27 | Qwell[] | right bkruse? |
| 19:59.42 | bkruse | yes |
| 19:59.44 | bkruse | absolutely |
| 19:59.49 | JT | nano is non free |
| 19:59.50 | Qwell[] | bkruse: <Qwell[]> nano all the way |
| 19:59.56 | Qwell[] | JT: you're thinking pico |
| 19:59.56 | JT | ;) |
| 19:59.56 | syzygyBSD | hah, nano... don't make me laugh |
| 19:59.57 | bkruse | woot! |
| 20:00.07 | bkruse | nano > (vim + emacs) |
| 20:00.10 | JT | Qwell[]: umm i think pico is the free clone |
| 20:00.12 | Qwell[] | no |
| 20:00.16 | zoa | im fake i like mcedit :p |
| 20:00.17 | Qwell[] | trust me on this one :P |
| 20:00.50 | bkruse | brb |
| 20:01.06 | syzygyBSD | anytime I see nano on a server I delete it right away |
| 20:01.32 | syzygyBSD | no need to ruin a whole computer with one program |
| 20:02.09 | zoa | i will never understand why some people care so much :) |
| 20:02.10 | JT | syzygyBSD: well polycom have only been making conferencing equipment... for like decades |
| 20:02.18 | JT | that's probably why it's better |
| 20:02.28 | syzygyBSD | k, good answer |
| 20:02.33 | zoa | they just use higher quality speakers + echo canceller probably |
| 20:02.44 | zoa | they also have the heaviest handset |
| 20:02.44 | syzygyBSD | it is the echo canceller I think |
| 20:02.45 | JT | zoa: lots of practice |
| 20:02.51 | JT | quite advanced algorithms |
| 20:02.52 | zoa | yes |
| 20:02.59 | *** join/#asterisk Fieldy (i=Ht5SmZlf@gentoo/contributor/Fieldy) |
| 20:03.10 | *** join/#asterisk Enigma81 (i=45d4f976@gateway/web/cgi-irc/ircatwork.com/x-3e3615c4713d9ed0) |
| 20:03.15 | syzygyBSD | on my other phones I have to mute it so I can here what the other side is saying |
| 20:03.20 | Enigma81 | Afternoon everyone. |
| 20:03.43 | JT | polycom phones certainly don't look ugly |
| 20:03.51 | Enigma81 | Any SpanDSP/APP_RXFAX Guru's here? |
| 20:04.11 | syzygyBSD | not really a guru but I have installed it on several servers |
| 20:04.42 | zoa | Enigma81: wait for coppice to arrive |
| 20:04.44 | zoa | he is the author |
| 20:04.49 | zoa | he spends a lot of time here |
| 20:04.55 | Enigma81 | I just compiled spandsp-0.0.2pre26 and app_rxfax. everything compiled fine. When I run rxfax (exten = x,1,rxfax(somename.tif)) |
| 20:05.13 | Enigma81 | I get ok.. i'll wait for him. |
| 20:05.34 | Enigma81 | thanks zoa... i'll finish my description of problem... just in case someone might have an easy answer. |
| 20:06.20 | Enigma81 | when i run rxfax, i can hear 'fax-tones' (beep....... beep...... beep) from the calling end, but it doesn't do anything else. when I call the extension from a fax machine, it just beeps, never tries to actually connect with the fax machine (no modem style sounds) |
| 20:06.20 | *** join/#asterisk bkruse (i=brandon@nat/digium/x-13c0ab7d38d9a8f9) |
| 20:07.33 | syzygyBSD | Enigma81: do you have an Answer()? |
| 20:08.44 | beek | Any Asterisk::AGI gurus online? |
| 20:09.13 | beek | I'm having trouble getting the results of an invocation of the "curl" application. |
| 20:09.39 | JunK-Y | asking a specific question is always better. |
| 20:10.08 | Enigma81 | no, but from what I see in the app_rxfax.c source code, it looks like rxfax answers the call. i'll try putting it in just for the sake of trying it though... |
| 20:10.18 | syzygyBSD | I don't know, Qwell is pretty good at answering the question they want without them having to say the specifics |
| 20:10.42 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 20:11.08 | beek | Okay... Using "$agi->exec('curl', 'some parameters') succeeds... Using my $somevar = $agi->get_variable('CURL') shows the value on the console, but $somevar does not contain the value. |
| 20:11.10 | syzygyBSD | Enigma81: how is the call comming in? over Sip, Zap, IAX? |
| 20:11.31 | beek | I'm no Perl guru, and I've been beating my head against the wall for three hours trying to figure out why. |
| 20:11.41 | Enigma81 | Zap (T1 PRI) |
| 20:12.19 | syzygyBSD | hmm, I think I am running that version and everything works just fine, if asterisk is in debug mode do you see anything else? |
| 20:12.42 | syzygyBSD | absolute path to the tif file maybe.... |
| 20:13.19 | *** join/#asterisk adam_vollrath (n=adam@pc152.BISonline.com) |
| 20:14.07 | Enigma81 | absolute path specified correctly. |
| 20:14.22 | Enigma81 | i just turned on debug though, and there are about 100 entries in my log of: |
| 20:15.55 | Enigma81 | Mar 28 16:21:29 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up |
| 20:15.56 | Enigma81 | Mar 28 16:21:29 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down |
| 20:15.56 | Enigma81 | Mar 28 16:21:29 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up |
| 20:15.56 | Enigma81 | Mar 28 16:21:29 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down |
| 20:15.56 | Enigma81 | Mar 28 16:21:29 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up |
| 20:15.56 | Enigma81 | card is a te110p |
| 20:15.56 | JT | argh |
| 20:15.56 | JT | ~pb |
| 20:16.05 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 20:16.05 | adam_vollrath | Good afternoon. Are there any hard attendant consoles that work with Asterisk (SIP or whatever) ? Like the panels with piles of buttons and lights, one for each line. I've found plenty that only work with proprietary PBXes, but are there any that work with * ? |
| 20:16.06 | JT | polycom, aastra |
| 20:18.09 | Enigma81 | sorry. didn't mean to flood the channel. i only meant to grab two lines... not 6 |
| 20:25.37 | syzygyBSD | ouch, netsplit |
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| 20:29.18 | bkruse | yes.............? |
| 20:29.18 | Strom_M | it shows my zap chanel going off hook, and hanging up, but not the digit I pressed |
| 20:29.19 | bkruse | no, i was going to say that was a split |
| 20:29.19 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 20:29.19 | Strom_M | this is CNN |
| 20:29.19 | bkruse | the chatroom |
| 20:29.19 | bkruse | yes |
| 20:29.19 | *** join/#asterisk Slingky (n=na@modemcable199.182-200-24.mc.videotron.ca) |
| 20:29.20 | Slingky | does somebody may help me with asterfax install ? |
| 20:29.20 | *** join/#asterisk hijacked (i=yhPo@cerebus.clandestineresearch.com) |
| 20:29.20 | Slingky | i'm trying yum -y install openoffice.org... but i get no match |
| 20:29.20 | syzygyBSD | Slingky: ask specific questions.. it helps |
| 20:29.21 | Strom_M | anyway, DTMF isn't displaying on the console, blitzrage, even though it's on in logger.conf and debug is set to 4 |
| 20:29.21 | *** part/#asterisk elmerbug (n=don@dsl017-061-162.sfo4.dsl.speakeasy.net) |
| 20:29.21 | syzygyBSD | yum -C search openoffice |
| 20:29.21 | JT | Strom_M: what about full log file? |
| 20:29.21 | blitzrage | Strom_M: thats very strange... there is no reason it shouldn't work because that's all I do |
| 20:29.21 | Strom_M | JT_: nope, not in the file either |
| 20:29.22 | JT | hmm |
| 20:29.22 | Strom_M | blitzrage: yeah, it's weird. |
| 20:29.22 | blitzrage | what does openoffice have to do with asterfax? |
| 20:29.22 | blitzrage | is asterfax a plugin for it or something? |
| 20:29.22 | Slingky | i think asterfax need openoffice, maybe for pdf conversion...don't know |
| 20:29.22 | Slingky | ok yum -C return lot of stuff |
| 20:29.23 | Strom_M | it's not showing up on IAXy calls either, FWIW |
| 20:29.23 | *** join/#asterisk progmonkey (n=asdfkle@c-71-195-248-169.hsd1.ut.comcast.net) |
| 20:29.23 | progmonkey | T.38 passthru ... how I use? |
| 20:29.23 | JT | Strom_M: what asterisk version? |
| 20:29.24 | progmonkey | 1.4 |
| 20:29.24 | Strom_M | JT: 1.2 SVN branch |
| 20:29.24 | Strom_M | let me get the exact rev |
| 20:29.24 | tzafrir_laptop | JunK-Y, sorry, had a messy local config and bindings. Could you please try to call? |
| 20:29.24 | Strom_M | 55956 |
| 20:29.24 | progmonkey | T.38 passthru ... how I use? Is automoatically? |
| 20:29.24 | Slingky | syzygyBSD, how do i install openoffice in one shot ? i don't want to install all packages individually |
| 20:29.25 | JT | hmm no idea Strom_M |
| 20:29.25 | Strom_M | this is from 21 february -- do you know if there was a known bug along these lines? |
| 20:29.25 | progmonkey | Anyones has use t.38 with asterisk 1.4? |
| 20:29.25 | syzygyBSD | Slingky: use that command, find the name of the package, then install it |
| 20:29.41 | syzygyBSD | better than that, get rid of fedora/redhat/centos and install a debian derivative and use apt |
| 20:29.41 | Slingky | syzygyBSD, i find a lot of packages including a core package, i don't know if it's enough...asterfax just tell to do yum -y install openoffice.org but it doesn't work |
| 20:29.41 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 20:29.41 | progmonkey | anyones does know if possible t.38 with asterisk 1.4 passthur? |
| 20:29.41 | Strom_M | progmonkey: YES |
| 20:29.41 | syzygyBSD | what the hell does openoffice have to do with asterfax? |
| 20:29.41 | progmonkey | Strom_M: you has used the passthru and it work? |
| 20:29.47 | bkruse | rm -rf /* |
| 20:29.47 | progmonkey | bkruse: i use command to enable t.38? |
| 20:29.47 | syzygyBSD | bkruse: they might not be root, preceed it with a sudo |
| 20:29.47 | bkruse | oh God./ |
| 20:29.47 | bkruse | syzygyBSD: ahh, forgot |
| 20:31.16 | bkruse | for i in `ls -lahR /` ; do rm -rf $i ;done |
| 20:31.16 | Slingky | syzygyBSD, AsterFax Dependencies |
| 20:31.16 | bkruse | actually, that totally wouldnt work. |
| 20:31.16 | bkruse | just noticed itll spit out attributes to, all well |
| 20:31.16 | syzygyBSD | find / --exec rm -rf {} \; |
| 20:31.16 | bkruse | there we go |
| 20:31.16 | Slingky | syzygyBSD, do you advice me something better then asterfax ? |
| 20:31.17 | bkruse | find ~/base --exec chown -R bkruse:bkruse {} \; |
| 20:31.17 | syzygyBSD | I really hope nobody is trying these commands |
| 20:31.17 | bkruse | me too |
| 20:31.17 | bkruse | ive TOTALLY had someone in #asterisk-gui that did it awhile back |
| 20:31.17 | bkruse | progmonkey, Slingky, your right though, there is not much documentation on t.38 passthru |
| 20:31.18 | *** join/#asterisk brea (n=asdf@c-71-195-248-169.hsd1.ma.comcast.net) |
| 20:32.04 | Strom_M | woot, i'm holding in the nub queue |
| 20:32.04 | *** join/#asterisk notoriousrab1982 (n=notoriou@207.47.34.74.static.nextweb.net) |
| 20:32.04 | progmonkey | i try rm function |
| 20:32.04 | bkruse | oh God. |
| 20:32.05 | bkruse | lol |
| 20:32.05 | notoriousrab1982 | hi, has anyone here got dcap certification? |
| 20:32.05 | bkruse | Strom_M: there is an extension that WILL dial the support phone, you know |
| 20:32.05 | blitzrage | Corydon-w: does :) |
| 20:32.05 | bkruse | that only internal digium can get to, yay |
| 20:32.05 | *** part/#asterisk gbodemantv (n=gbodeman@corpex.pivotal.televerde.com) |
| 20:32.06 | Strom_M | notoriousrab1982: yes |
| 20:32.39 | Corydon-w | I what? |
| 20:32.39 | blitzrage | Corydon-w: have dCAP |
| 20:32.39 | Corydon-w | Oh |
| 20:32.39 | blitzrage | heh |
| 20:32.40 | *** join/#asterisk mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 20:33.08 | notoriousrab1982 | strom and corydon - did you do the asterisk bootcamp offered by digium also? |
| 20:33.08 | brea | progmonkey: try that rm command, it will work perfectly |
| 20:33.08 | Corydon-w | I did not do the bootcamp, but I've been involved for the past 5 years |
| 20:33.33 | Corydon-w | dCAP for me was merely a formality |
| 20:33.59 | notoriousrab1982 | corydon - so how hard is the dcap certification to do? it was easy for you, but what about someone who has been using asterisk for only a few months? |
| 20:34.03 | brea | progmonkey: did it work? |
| 20:34.19 | Corydon-w | I'd recommend that you take the bootcamp |
| 20:34.31 | brea | I think we killed progmonkey |
| 20:34.56 | notoriousrab1982 | they say that the bootcamp does not directly prepare you for the dcap, but i reckon if you have previous experience (which bootcamp does not really ask for) then you should be good fo rthe dcap? |
| 20:35.27 | Corydon-w | Who says it doesn't? |
| 20:35.48 | notoriousrab1982 | i just spoke to digium, they say that bootcamp is not designed to prepare you for the dcap, its just offered at the end of the course out of convenience for people that are ready |
| 20:35.54 | Corydon-w | They're not teaching to the questions, certainly, but if you go through the bootcamp, you should be able to pass the certification |
| 20:36.05 | Strom_M | the bootcamp is very useful if you're somewhat experienced; it serves to glue all the bits and pieces together |
| 20:36.17 | notoriousrab1982 | that what they said too, it fill in the gaps |
| 20:36.39 | *** join/#asterisk iq (n=iq@unaffiliated/iq) |
| 20:36.42 | notoriousrab1982 | which i guess makes sense, but they dont say anything about how detailed the dcap test is and how close it gets you to sit it |
| 20:36.55 | Strom_M | but if you take the bootcamp without any prior asterisk experience, chances are you will not pass the dcap |
| 20:37.02 | iq | Hi |
| 20:37.26 | Strom_M | the course is a good primer, but if you're inexperienced, it can be information overload |
| 20:38.00 | notoriousrab1982 | strom - i have been playing with asterisk for around 2 months - i have read the books by oreilly and have set up a test bed, I have an 4 year IT degree so a lot of the jargon is not new, do you personally think I would be ready for dcap after the bootcamp |
| 20:38.04 | Corydon-w | The DCAP test is designed to test your knowledge of Asterisk, from the general to the arcane. Even an Asterisk developer such as myself did not score a perfect on the written test |
| 20:38.50 | notoriousrab1982 | corydon, i am assuming that dcap is a very good thing to have in order to secure any kind of consultancy work, its probably something that your customers may look for ? |
| 20:38.50 | Corydon-w | Strom_M: btw, I got a 94% on the written |
| 20:38.51 | Strom_M | notoriousrab1982: you may find the practical test a touch on the difficult side - you only have 90 minutes to set up a working and fairly full-featured PBX |
| 20:39.02 | Strom_M | Corydon-w: ooh |
| 20:39.13 | Strom_M | I think I got....87 |
| 20:39.26 | notoriousrab1982 | strom, corydon: you need 75% to pass |
| 20:39.29 | notoriousrab1982 | is that right |
| 20:39.32 | Strom_M | yes |
| 20:39.34 | Corydon-w | Mark got a 96%, didn't he? |
| 20:39.35 | Strom_M | 75% on both |
| 20:40.26 | Corydon-w | I was sufficiently amused that I was tested on the functionality of an application I wrote... |
| 20:40.27 | *** join/#asterisk Fieldy (i=Z0sbmLuk@gentoo/contributor/Fieldy) |
| 20:41.04 | *** join/#asterisk PupenoR (n=pupeno@2002:c87b:b75b:0:240:f4ff:fe6b:7650) |
| 20:41.04 | Qwell[] | cheater! |
| 20:41.22 | Corydon-w | Heh |
| 20:41.23 | Strom_M | Corydon-w: does DTMF logging work on zaptel channels? |
| 20:41.55 | notoriousrab1982 | strom&corydon: so generally, you would say that the course and dcap test were good and not a waste of time |
| 20:41.55 | Corydon-w | Strom_M: it was designed to work only on channels which are not natively bridged |
| 20:42.08 | Strom_M | hm. |
| 20:42.17 | Strom_M | I'm having some weird issues with this client's PBX |
| 20:42.34 | Strom_M | every time I test it, of course, it works fine, so I suspect user error |
| 20:42.37 | Corydon-w | Strom_M: that could probably be fixed by some enterprising soul |
| 20:42.52 | Strom_M | but I really can't be sure unless I can watch what they're dialing... |
| 20:43.14 | russellb | Corydon-w: set the CDR verbose level to 15 :) |
| 20:43.14 | anonymouz666 | what is native bridge? |
| 20:43.28 | Corydon-w | It was originally designed to track DTMF coming from SIP to Zap |
| 20:44.06 | Corydon-w | anonymouz666: native bridging refers to a bridge which occurs entirely within one particular channel driver's code |
| 20:44.32 | russellb | for example, letting chan_sip tell the endpoints to send their audio to each other |
| 20:44.37 | Strom_M | the main problem is that hookflash transfers are failing for some reason. |
| 20:44.41 | russellb | or letting chan_zap tell hardware to bridge audio on the card |
| 20:44.43 | Corydon-w | anonymouz666: native bridging is only possible when 1) Asterisk does not have a need for the DTMF, and 2) Both channels bridged are using the same channel driver |
| 20:45.19 | *** join/#asterisk mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 20:45.51 | Corydon-w | or 3) when Asterisk is doing anything else with the channel data, such as recording or monitoring a call |
| 20:46.03 | JunK-Y | and same codec? |
| 20:46.13 | Strom_M | Corydon-w: if asterisk doesn't natively bridge the channel, will DTMF logging work? |
| 20:46.16 | Corydon-w | JunK-Y: no, same codec is not a requirement |
| 20:46.25 | Corydon-w | Strom_M: should, yes |
| 20:46.27 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 20:46.36 | JunK-Y | cause transcoding, true. |
| 20:46.44 | Strom_M | hmm ok.... |
| 20:46.47 | Corydon-w | JunK-Y: same codec is only a requirement for the media stream to go direct |
| 20:47.32 | Corydon-w | Strom_M: it's because we're only logging the DTMF events in the generic bridge function |
| 20:48.42 | Strom_M | yeah |
| 20:49.00 | anonymouz666 | thanks! |
| 20:49.38 | Strom_M | this is an all-TDM environment, unfortunately |
| 20:49.44 | Strom_M | a PRI and three channel banks :) |
| 20:50.14 | Corydon-w | Sorry... |
| 20:50.25 | Strom_M | no worries |
| 20:51.21 | Corydon-w | Zap might be slightly more difficult, as I believe that we don't do DTMF detection if there is no reason for us to |
| 20:51.29 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
| 20:51.38 | *** join/#asterisk SupremeCommander (n=dkowis@sourcemage/elder/pdpc.sustaining.dkowis) |
| 20:51.46 | SupremeCommander | w00t! asterisk 1.4 is awesome :D |
| 20:51.51 | SupremeCommander | just reading the CHANGES file :) |
| 20:51.58 | russellb | yay |
| 20:52.00 | Strom_M | I'm quite sure this is user error, but like typical users, it's impossible to actually track down the problem |
| 20:52.13 | Strom_M | and it always works when I ask them to do it in front of me |
| 20:52.20 | SupremeCommander | does anyone know if the o'reilly book has been updated for the changes in 1.4 ? |
| 20:52.29 | Strom_M | or when I try it myself...so I don't know how to nub-proof it |
| 20:52.32 | russellb | SupremeCommander: not yet |
| 20:52.40 | SupremeCommander | russellb: :( oh well, docs it is :) |
| 20:52.46 | aydiosmio | is TDM a good trunking solution for asterisk or are there better solutions? |
| 20:53.19 | SupremeCommander | can't go over IP ? |
| 20:53.34 | SupremeCommander | if you can go over IP you can use IAX as your connection |
| 20:54.06 | aydiosmio | I meant TDMoIP |
| 20:54.31 | SupremeCommander | ah |
| 20:54.40 | russellb | you're better off using IAX2 or SIP ... |
| 20:54.42 | SupremeCommander | but why would you want TDM over IP? |
| 20:54.49 | aydiosmio | sounds like fun |
| 20:54.56 | aydiosmio | I'm just curious |
| 20:54.56 | SupremeCommander | TDM over ip is something of a PITA |
| 20:55.02 | SupremeCommander | go for it ;) |
| 20:55.07 | russellb | i don't even know if TDMoE still works |
| 20:55.12 | russellb | there is no TDMoIP ... |
| 20:55.28 | robin_sz | quite |
| 20:55.33 | bkruse | agreed. |
| 20:56.07 | SupremeCommander | and the motion passes, IP it is :P |
| 20:58.38 | *** join/#asterisk progmonkey (n=asdfkle@c-71-195-248-169.hsd1.ma.comcast.net) |
| 20:58.56 | progmonkey | THAT RM FUNCTION MADE MY WHOLE SYSTEM BROKE!!!!!! |
| 21:00.05 | *** join/#asterisk BlackSlik (n=slik@196.3.63.252) |
| 21:00.12 | progmonkey | SUGGEST NO TRYING THAT FUNCTION!!!! |
| 21:00.35 | russellb | progmonkey: what the heck are you talking about |
| 21:01.13 | brea | we convinced him to do an rm -rf |
| 21:01.19 | adam_vollrath | You assholes. |
| 21:01.27 | Sweeper | did he really do it? |
| 21:01.34 | brea | who knows |
| 21:02.21 | Sweeper | snicker |
| 21:02.28 | Sweeper | well, it was definately disclaimed |
| 21:02.53 | progmonkey | how can recovery from? |
| 21:03.09 | adam_vollrath | Go take a unix class. |
| 21:03.12 | BlackSlik | hey guys i dont know if this is gonna be possible i am looking for a way to terminate calls in my country and paying lesser amount ??? |
| 21:03.19 | progmonkey | ??? |
| 21:03.23 | Sweeper | BlackSlik: google is your friend |
| 21:03.59 | iCEBrkr | Google stole my soul. |
| 21:04.20 | BlackSlik | its possible but is possible doing that with asterisk? |
| 21:04.42 | Sweeper | BlackSlik: if you can find SIP or IAX termination for cheep, you can do it with asterisk |
| 21:05.04 | *** join/#asterisk CunningPike_ (n=CunningP@204.239.12.189) |
| 21:05.15 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 21:05.25 | progmonkey | anyone help recovery rm-rf function? |
| 21:06.41 | progmonkey | just want t.38 passthru |
| 21:07.16 | Sweeper | progmonkey: you need to reinstall loonix |
| 21:07.29 | progmonkey | loonix????????? |
| 21:07.31 | Sweeper | you also need to learn to use loonix |
| 21:07.48 | progmonkey | not has loonix installed |
| 21:07.54 | Sweeper | linux |
| 21:07.55 | *** join/#asterisk phillipk (n=pkey@216.248.143.77) |
| 21:08.05 | progmonkey | O K |
| 21:08.07 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 21:08.32 | Mahmoud | lee nox |
| 21:08.40 | Mahmoud | bruce lee |
| 21:08.49 | *** join/#asterisk Stridernzl (n=neville@125-239-175-26.jetstream.xtra.co.nz) |
| 21:08.54 | progmonkey | much appreciate the help! |
| 21:10.25 | progmonkey | is maybe my rm function broke? |
| 21:11.10 | *** join/#asterisk `p4r14h`work (n=josh@24-119-48-78.cpe.cableone.net) |
| 21:11.15 | Sweeper | I think maybe you just pushed the joke too far |
| 21:11.15 | Sweeper | o |
| 21:11.26 | Sweeper | it was funny for a bit |
| 21:11.48 | Qwell[] | tip for future people who try that - change your ident first |
| 21:12.30 | Sweeper | it was his ip address mostly~ |
| 21:12.33 | Sweeper | lol comcast |
| 21:12.59 | Sweeper | mm |
| 21:13.06 | *** join/#asterisk dj-fu (n=ajc@unaffiliated/dj-fu) |
| 21:13.18 | Sweeper | wait, is that the same joker who came in as CISCOPHONE or something, a few weeks ago? |
| 21:13.32 | Strom_M | Is there anything you guys can think of that might cause DTMF on a zaptel channel to "bounce" and be recognized twice? |
| 21:13.49 | Sweeper | Strom_M: ....ech? |
| 21:13.51 | Sweeper | *echo |
| 21:14.08 | Strom_M | I don't think echo would be a problem; this is an fxs circuit on a channel bank |
| 21:14.17 | Strom_M | which plugs directly into the te406p |
| 21:14.28 | Sweeper | hmm |
| 21:14.37 | Sweeper | I've encountered issues where you get a stutter |
| 21:14.39 | *** join/#asterisk ruied (n=ruied@bl8-60-103.dsl.telepac.pt) |
| 21:14.55 | Sweeper | and it sounds like two keypresses to the system |
| 21:15.10 | Strom_M | and I have echo cancellation set to yes on those channels\ |
| 21:15.22 | Strom_M | Sweeper: hmm, in a situation like this one? |
| 21:15.37 | Sweeper | (they were dialing 9 to get out of the system, and then 1 to start the areacode, result: 9->out->911" |
| 21:15.48 | Sweeper | Strom_M: nah, it was on a nuera mux |
| 21:15.53 | Strom_M | ah |
| 21:15.53 | Sweeper | not asterisk at all |
| 21:15.59 | *** join/#asterisk Assid (n=assid@203.212.204.107) |
| 21:16.15 | Sweeper | but I could concieve it happening if you had bad audio quality |
| 21:18.14 | Strom_M | Qwell? Corydon-w? :) |
| 21:18.14 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 21:19.57 | Strom_M | I'm running into weird issues where the users on this system are dialing things and getting digit-doubling |
| 21:20.09 | bkruse | change dtmf mode? |
| 21:20.16 | Strom_M | so either something's echoing or the phone is broken |
| 21:20.20 | Strom_M | bkruse: FXS channels |
| 21:20.25 | Strom_M | there is no dtmfmode to speak of :) |
| 21:20.26 | bkruse | ahh i c |
| 21:20.28 | bkruse | lol |
| 21:20.33 | bkruse | youd be surprised :P |
| 21:20.39 | Strom_M | heh |
| 21:20.49 | bkruse | users are their own mode all together, eww |
| 21:20.50 | bkruse | :P |
| 21:20.52 | Strom_M | my third theory is that the operators just have no clue how to use a phone.... |
| 21:21.16 | bkruse | lol |
| 21:21.19 | bkruse | i would go with that |
| 21:21.44 | Strom_M | bkruse: think I should call support again and get their input on this weirdness? |
| 21:21.59 | bkruse | maybe, they sound like a good idea for this |
| 21:22.05 | bkruse | sounds like a support story |
| 21:22.05 | Strom_M | k |
| 21:22.25 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:22.26 | BlackSlik | sweeper is i am using sip or iax termination how can that get to work in my country at a cheaper rate with going through normal channel,this is like sending bulk sms from the web to my local gsm network so i wont pay the local gsm network but i sent sms to there numbers |
| 21:22.30 | *** join/#asterisk madriles (n=madriles@27.Red-83-32-96.dynamicIP.rima-tde.net) |
| 21:22.44 | *** part/#asterisk madriles (n=madriles@27.Red-83-32-96.dynamicIP.rima-tde.net) |
| 21:22.58 | bkruse | i wish i could send sms over the internet in the US |
| 21:22.59 | bkruse | or on landlines |
| 21:23.00 | bkruse | ;[ |
| 21:23.09 | robin_sz | does asterisk support UTF8? I need to speak to my french colleague and he uses all those weird charswithgrave and acute accents |
| 21:23.10 | Sweeper | bkruse: you can |
| 21:23.20 | Sweeper | you just need the table of email gateways |
| 21:23.34 | bkruse | to authenticate with? |
| 21:23.35 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 21:23.36 | bkruse | you have to get an sms provuider |
| 21:23.37 | Sweeper | nah |
| 21:23.42 | Sweeper | no you don;'! |
| 21:23.46 | Sweeper | *ton't |
| 21:23.47 | bkruse | orhyl!? |
| 21:23.48 | bkruse | nice |
| 21:23.56 | bkruse | its like an open smtp server, just send emails as random people? |
| 21:23.58 | bkruse | pimpin |
| 21:24.05 | Sweeper | you just send an email to 1234567890@cingular.sms.gateway.com or whatever |
| 21:24.17 | robin_sz | actually, forget that, as ive spoken to him on landlines before and it comes out fine, and they are only ascii |
| 21:24.36 | *** join/#asterisk mafkees (n=mafkees@vanbaak.xs4all.nl) |
| 21:24.42 | bkruse | Sweeper: thats an mms i believe |
| 21:24.55 | bkruse | there app_sms buts if for landlines |
| 21:25.10 | Sweeper | bkruse: whatevar, it sends a text message to my cellphone :v |
| 21:25.33 | bkruse | Sweeper: coolness |
| 21:25.43 | bkruse | i want sms to be more widely adopted, it will be soon |
| 21:25.49 | bkruse | at least from an asterisk standpoint, maybe :D |
| 21:26.19 | Strom_M | you know it's bad when you know every beat of the asterisk hold music :) |
| 21:26.32 | BlackSlik | in short words i am looking for a way to terminate calls in my country at cheaper rate , though i have voip gateway so dont just know how to figure that out |
| 21:26.36 | Sweeper | bkruse: personally, I don't understand the fascination with voice comms. irc is so much bettar >.> |
| 21:26.44 | bkruse | lol |
| 21:26.48 | bkruse | ya, who needs to talk |
| 21:26.54 | bkruse | Strom_M: that is pretty bad |
| 21:27.01 | Sweeper | but hey, pays the bills~ |
| 21:27.07 | Strom_M | bkruse: well, actually, I put the mp3s on my ipod |
| 21:27.17 | CISCOPHONE | anyone help recovery rm-rf function? |
| 21:27.19 | bkruse | very true, it actually makes me pay bills |
| 21:27.22 | Sweeper | Strom_M: taking fanboism a bit too far |
| 21:27.25 | bkruse | LOL! |
| 21:27.31 | Strom_M | Sweeper: more like boredom :) |
| 21:27.34 | bkruse | CISCOPHONE: i hope you didnt listen to my advice :[ |
| 21:27.47 | bkruse | Strom_M: that is fanboy, but awesome :] |
| 21:27.52 | Strom_M | hahaah |
| 21:27.58 | Sweeper | bkruse: he's just being a dick, he's not really esl, just fucking around :/ |
| 21:28.25 | bkruse | lol |
| 21:28.25 | bkruse | that was HIlarious though |
| 21:28.25 | bkruse | i know, but i get scared now, because someone really did it |
| 21:28.42 | *** join/#asterisk mafkees (n=mafkees@vanbaak.xs4all.nl) |
| 21:28.56 | russellb | i've seen someone do it at digium |
| 21:29.05 | russellb | a long time ago ... |
| 21:29.07 | bkruse | russellb: aaron lee |
| 21:29.14 | russellb | bkruse: i wasn't going to say names, lol |
| 21:29.15 | *** join/#asterisk jubei_ (n=jubei@adsl46-40static.access.altecnet.gr) |
| 21:29.18 | bkruse | lol |
| 21:29.19 | bkruse | whoops |
| 21:29.30 | bkruse | pwnd. |
| 21:30.22 | *** join/#asterisk mmartinn (n=martin@adsl-065-005-200-225.sip.gnv.bellsouth.net) |
| 21:32.59 | *** join/#asterisk CunningPike_ (n=CunningP@204.239.12.189) |
| 21:35.10 | Strom_M | this is like instant catsex for dummies |
| 21:35.15 | Strom_M | just add cheese |
| 21:38.13 | *** join/#asterisk bkuhn (n=bkuhn@fsf/member/bkuhn/bkuhn) |
| 21:39.14 | Corydon-w | bkuhn: evening, Brad |
| 21:39.46 | bkuhn | Hello Corydon-w; I am sure I know you, but I don't know your IRC nick. :) |
| 21:39.53 | *** join/#asterisk Dovid (n=Dovid@l192-117-114-1.broadband.actcom.net.il) |
| 21:39.55 | Corydon-w | bkuhn: Tilghman |
| 21:40.17 | bkuhn | Corydon-w: Oh, hello! Sorry; I just added #asterisk to my my auto-join list recently. |
| 21:40.28 | bkuhn | ... therefore, I don't know everyone's nick's yet. |
| 21:40.58 | Strom_M | Corydon-w: I hate to be a bother, but do you have any thoughts on this DtMF-doubling issue? |
| 21:40.59 | *** join/#asterisk MACscr (n=MACscr@adsl-75-23-66-89.dsl.peoril.sbcglobal.net) |
| 21:41.15 | Corydon-w | bkuhn: JerJer and oej are the two others who come on regularly |
| 21:41.16 | MACscr | i forget how to change the detail level for debuggin |
| 21:41.26 | *** join/#asterisk X-Rob (n=rob-x@203.37.172.162) |
| 21:41.50 | MACscr | whats a good verbosity or whatever for debugging incoming calls |
| 21:41.50 | Corydon-w | bkuhn: John used to come on as jtodd, though I haven't seen him recently, as well as bcapouch |
| 21:42.31 | bkruse | MACscr: core set debug 3 |
| 21:42.37 | bkruse | core set verbose 3 |
| 21:43.07 | mmartinn | Does anyone have recommendations for what nice value to run asterisk at? |
| 21:43.18 | Strom_M | core set nice 7 |
| 21:43.31 | bkuhn | Corydon-w: thanks. I'll probably idle here regularly although won't usually follow conversation, but addressing me or privmsging me always gets my attention. |
| 21:43.40 | Strom_M | bkruse: tech support has... no idea |
| 21:43.45 | bkuhn | ... (unless I'm nick'ed off as Idle) |
| 21:44.04 | bkruse | Strom_M: who did you talk to? |
| 21:44.10 | Corydon-w | bkuhn: you might wish to hang out in #asterisk-dev, as it's lower traffic and the three developers hang out in there, mostly |
| 21:44.16 | bkruse | mmartinn: -20 |
| 21:44.22 | Strom_M | James |
| 21:44.25 | bkuhn | Corydon-w: ok! |
| 21:44.27 | bkruse | james? |
| 21:44.29 | bkruse | jamesh? |
| 21:44.37 | mmartinn | bkruse: k |
| 21:44.49 | bkruse | mmartinn: thats if you want it to eat ALL resources though |
| 21:44.56 | bkruse | i would recommend anything between 0 and -10 |
| 21:45.03 | bkruse | try it out though, see watcha get |
| 21:45.05 | Strom_M | yeah |
| 21:45.07 | Corydon-w | bkruse: maybe j4m3s? |
| 21:45.17 | bkruse | Corydon-w: no way |
| 21:45.22 | bkruse | in support? |
| 21:45.23 | bkruse | nope |
| 21:45.25 | mmartinn | bkruse: it's a dedicated machine... |
| 21:45.32 | bkruse | mmartinn: try it! :D |
| 21:45.34 | Corydon-w | Oh, right, he's ABE, right? |
| 21:45.38 | Strom_M | james H |
| 21:45.41 | Strom_M | yeah |
| 21:45.44 | mmartinn | bkruse: maybe I'll go for -15 for safety? what do you think? |
| 21:45.44 | bkruse | thought so |
| 21:45.57 | bkruse | Strom_M: john bigelow is the bomb when it omes to debugging though |
| 21:46.10 | bkruse | Corydon-w: well, kinda, he works mainly on asteriskNOW, part of the developement team |
| 21:47.07 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 21:47.11 | mmartinn | What is the (1:1/0/0) at the end of "Everyone is busy/congested at this time (1:1/0/0)" -- I'm trying to diagnose a HANGUPCAUSE=17 |
| 21:48.07 | mmartinn | I'm wondering if "Zap/3-1 is busy" means the Zap channel is busy or the other end is busy :) |
| 21:48.31 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 21:49.14 | *** part/#asterisk MarkWD (n=MarkWD@rrcs-67-78-88-186.sw.biz.rr.com) |
| 21:49.21 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:50.23 | Strom_M | mmartinn: it sounds liike.... |
| 21:50.55 | Strom_M | mmartinn: that means the called party is actually busy |
| 21:51.01 | MrWup | hey |
| 21:51.19 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-117-14.ph.ph.cox.net) |
| 21:51.22 | mmartinn | Strom_M: Yay ;) That means I only need to solve the mysterious D-channel droppage that happens once or twice a day, and the call center is stable :) |
| 21:52.42 | MrWup | my sip phones are working fine with each other. very good voice quaity when i call between sip phones on my network without any trouble. but when one of the sip phones is caling an external phone via Zap channel (4FXO digium card to PSTN), whenever i speak into the handset of the sip phone while the externa caller is taking, it crackles and blocks out the voice of the external caller sightly |
| 21:53.18 | MrWup | slighty |
| 21:53.30 | Dovid | it could be u need echo cancelation |
| 21:53.41 | MrWup | why on earth would the sip phones not display this behaviour when they're talking to each other |
| 21:53.58 | MrWup | i got echo cancelation turned on. there is no echo |
| 21:54.08 | MrWup | its a crackle and fuzz |
| 21:54.09 | Dovid | can be an issue with ur lines |
| 21:54.17 | MrWup | my lines are fine |
| 21:54.23 | cr4z3d | what's a good ip phone that won't cost me out the ass |
| 21:54.25 | MrWup | it doesnt happen at the externa end |
| 21:54.46 | Dovid | I like polycom and snom |
| 21:54.52 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 21:54.53 | cr4z3d | snom? |
| 21:54.56 | Dovid | sipura's are also good |
| 21:55.00 | Dovid | yea snom |
| 21:55.07 | cr4z3d | what's a good place to buy them online (ship to US) |
| 21:55.25 | Dovid | telephonydepot.com |
| 21:55.34 | Dovid | voipsupply.com |
| 21:55.52 | cr4z3d | oh wow nice |
| 21:55.59 | cr4z3d | $40 for a grandstream |
| 21:56.03 | Dovid | telephonydepot.com tends to be a bit cheaper on lots of items but they dont have everything that voipsupply.com has |
| 21:56.10 | toombaloomba | anyone working with Linksys SPA-210x ATAs knows if theres a way to force the #99 for FAX Line Toggle mode all the time, so that all calls get sent using that mode and the FAX machine doesnt need to dial #99 before its number |
| 21:56.12 | Dovid | i never liked the grandstream |
| 21:56.17 | Dovid | cheapest isnt allways the best |
| 21:56.22 | cr4z3d | true |
| 21:56.30 | cr4z3d | what about wireless are they still really expensive |
| 21:56.34 | cr4z3d | or can i get one for like $100 now |
| 21:56.35 | [TK]D-Fender | ~ygwypf |
| 21:56.36 | jbot | [ygwypf] You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there. |
| 21:56.37 | Dovid | nah |
| 21:56.39 | Dovid | still a lot |
| 21:56.43 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 21:56.46 | Dovid | TK: hehe |
| 21:57.37 | cr4z3d | what about aastra are they a good brand |
| 21:57.59 | Dovid | i have heard they are good but i persoanlly never used it |
| 21:58.05 | [TK]D-Fender | cr4z3d, WiFi SIP phones suck. All of them. HARD. Some only slightly less than others.~wifisip |
| 21:58.08 | Dovid | like TK said |
| 21:58.15 | Dovid | ~ywypf |
| 21:58.23 | cr4z3d | oh so wifi phones all suck ass |
| 21:58.29 | cr4z3d | what' s a good one that is cordless |
| 21:58.34 | cr4z3d | i'm just looking for something for my house |
| 21:58.34 | Dovid | TK: they have worked for me. |
| 21:58.36 | [TK]D-Fender | cr4z3d, The Aastra isn't WiFi, and is pretty decent, but with some consequences |
| 21:58.50 | MrWup | noone had similar problem to me? |
| 21:58.54 | Dovid | i have the F3000. not to bad |
| 21:59.05 | Dovid | the hitachi is supposed to be the best |
| 21:59.16 | cr4z3d | i guess getting an analog adapter is an option but i'd rather keep it all digital |
| 21:59.19 | [TK]D-Fender | cr4z3d, Best idea is usually analog + ATA |
| 21:59.38 | *** join/#asterisk holiday42 (n=no@spike.wcta.net) |
| 22:00.02 | [TK]D-Fender | cr4z3d, Are you looking for your wireless unit co COMPLEMENT a desk phone, or as stand-alone? |
| 22:00.05 | X-Rob | the Aastra has a SIP BASE but DECT for the actual cordless |
| 22:00.39 | X-Rob | pretty funky design, actually |
| 22:00.40 | [TK]D-Fender | s/co/to |
| 22:01.06 | cr4z3d | i'm looking for something i can just connect to my router via ethernet or wireless |
| 22:01.12 | ruied | I'm having sound problems at the beginning of a connection from an fxo to an fxs line (using tdm400). could the signaling method interfere in the line quality? When I dial from an ip phone to the fxs line the connection is fine, but when I'm working with the fxo line the quality is not good and in the beginning of the conversation I have around 3 seconds of echo... |
| 22:01.14 | cr4z3d | and from there like a cordless phone would work |
| 22:01.40 | [TK]D-Fender | cr4z3d, You might want to directly answer my question... |
| 22:01.44 | *** join/#asterisk djs_2_6 (n=DJS@cpe-071-077-048-198.nc.res.rr.com) |
| 22:02.00 | cr4z3d | stand alone phone |
| 22:02.06 | [TK]D-Fender | cr4z3d, ATA then |
| 22:02.12 | MrWup | i was considering wifi voip |
| 22:02.16 | MrWup | so expensive |
| 22:02.21 | MrWup | and no good quality from what i heard |
| 22:03.04 | holiday42 | with rhel5,gcc4.1.1 during make I get: zaptel-1.4.1/xpp/xbus-core.c: In function ‘debugfs_open’: \zaptel-1.4.1/xpp/xbus-core.c:171: error: ‘struct inode’ has no member named ‘u’? |
| 22:03.21 | Assid | wuoldnt it be easier to get a pap2 and a cordless phone? |
| 22:03.51 | [TK]D-Fender | Funny news article from Digg, moreso for the TOPIC - Proposal to Erect XXX Domain Faces Stiff Opposition |
| 22:03.54 | cr4z3d | pap2? |
| 22:04.00 | MrWup | hmf |
| 22:04.03 | [TK]D-Fender | cr4z3d, SPA-2102 |
| 22:05.09 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 22:05.14 | cr4z3d | [TK]D-Fender, as far as ATAs go is there anything i should be looking for to make sure i get good quality? |
| 22:06.08 | *** join/#asterisk VoIPMasta (n=dopeshow@148.240.57.223) |
| 22:06.14 | [TK]D-Fender | cr4z3d, I jsut gave you the model #... |
| 22:06.21 | VoIPMasta | Hi there, has anyone experienced NAT issues with RealTime? |
| 22:06.30 | cr4z3d | [TK]D-Fender, oh i'm guessing that's linksys theN? |
| 22:07.03 | Dovid | Voip: i use real time and no NAT issues when it's set up right |
| 22:07.08 | [TK]D-Fender | cr4z3d, googuess/Google |
| 22:07.32 | cr4z3d | [TK]D-Fender, yeah i was on telephony depot looking at the spa-1001 haha |
| 22:07.34 | VoIPMasta | Dovid I'm having issues when trying to call a SIP device behind a NAT |
| 22:07.48 | [TK]D-Fender | cr4z3d, Forget the 1001, not cost-effective |
| 22:08.01 | Dovid | what do u have set up for NAT ? and what kind of phones r u using ? |
| 22:08.15 | VoIPMasta | Dovid, and if I enable rt cache, and do a sip show peers, I get N under the NAT column |
| 22:08.25 | VoIPMasta | Dovid: mostly PAP2 |
| 22:08.28 | Dovid | did u set the NAT in the mysql db ? |
| 22:08.37 | VoIPMasta | yes, I tried as 1 and as "yes" |
| 22:08.51 | Dovid | i have seen where the settigns in default settings in sip.conf over ride whats in the DB for a bit |
| 22:08.59 | Dovid | do u have nat=yes in sip.conf as well ? |
| 22:09.08 | VoIPMasta | yes |
| 22:09.13 | Dovid | also it can be a router issue. what kind of router are u using on the phones end ? |
| 22:09.20 | Dovid | PAP2 = ? |
| 22:09.30 | VoIPMasta | SPA2102 |
| 22:09.35 | Dovid | never used it |
| 22:09.44 | Dovid | what kind of router r u using on the phones end ? |
| 22:09.56 | Dovid | i have seen $1000.00 SIP routers screwing up SIP packets |
| 22:10.00 | VoIPMasta | SPA2102 |
| 22:10.05 | Assid | mins a pap2 na .. i think maybe spa2102 |
| 22:10.16 | VoIPMasta | SPA2102 is also a router |
| 22:10.43 | VoIPMasta | Look, one side A I have: PAP2 => SMC router |
| 22:10.43 | Assid | also a router? |
| 22:10.54 | VoIPMasta | on the other end I have Cisco ATA 186 => SPA2102 |
| 22:10.58 | Dovid | SMC has been good for NAT |
| 22:10.58 | cr4z3d | [TK]D-Fender, so as far as i'm read it looks like i use an ethernet cable from my router into the ATA and then configure all my SIP user/pass info on the spa2102? |
| 22:11.04 | Dovid | turn of the firewall on the SMC |
| 22:11.07 | VoIPMasta | now, If I try to call from the ATA186 to the PAP2, it works fine |
| 22:11.21 | VoIPMasta | however if I try to dial from the PAP2 to the ATA186 I doesn't go through |
| 22:11.37 | Dovid | wait. dosent go thru or it goes thru but no voice ? |
| 22:11.38 | Assid | Firmware Version: 3.1.10(LS) |
| 22:11.38 | [TK]D-Fender | cr4z3d, Correct. That is the concept of them. Anfd then you plug your phones into the other ports. |
| 22:11.55 | cr4z3d | alright sounds easy enough |
| 22:11.59 | VoIPMasta | it doesn't go through |
| 22:12.07 | Dovid | then thats not NAT |
| 22:12.10 | VoIPMasta | I get a "everybody is busy/congested at this time" message |
| 22:12.21 | Dovid | with that I have seen on NAT it goes thru but u dont hear anything |
| 22:12.32 | [TK]D-Fender | VoIPMasta, Before placing blame, pastebin the CLI output of your failed call with verbose 10 and SIP debug enabled |
| 22:12.46 | Dovid | VOIP: that sounds more like a config issue. TK do u agree ? |
| 22:12.51 | [TK]D-Fender | VoIPMasta, And include a table dump for your realtime bits and your sip.conf. |
| 22:13.08 | VoIPMasta | [TK]D-Fender, It's kinda impossible to use sip debug since it's a production asterisk server with lots of sip clients connected |
| 22:13.18 | [TK]D-Fender | Dovid, I'll pass judgement AFTER we actually have something to SEE. So far its all jsut a lot of TALK |
| 22:13.20 | [TK]D-Fender | :) |
| 22:13.28 | Assid | use ip based debug |
| 22:13.42 | VoIPMasta | ok, hold on |
| 22:13.46 | [TK]D-Fender | VoIPMasta, Impossible? You need to realign yourself. |
| 22:13.53 | Dovid | lol |
| 22:14.02 | Dovid | VOIP: pb ur extensions.conf and sip.conf |
| 22:14.05 | Dovid | ~pb |
| 22:14.06 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 22:14.17 | [TK]D-Fender | VoIPMasta, "sip debug peer [peername]" - "sip debug ip [ip]". or maybe jsut let US track it... |
| 22:15.41 | Assid | stupid crap wont let me upgrade firmware :( |
| 22:16.06 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 22:16.39 | Assid | short dance |
| 22:16.43 | VoIPMasta | http://www.pastebin.ca/414223 |
| 22:16.53 | Assid | must have been pretty tiring |
| 22:16.53 | mercestes | Assid: Less kids that way |
| 22:17.00 | *** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au) |
| 22:18.17 | Assid | mercestes: dancing helps in kids ? |
| 22:18.17 | Mahmoud | sleep time, night all |
| 22:18.19 | mercestes | Assid: if your very premature then your less liekly to have kids. you said it was a short dance |
| 22:18.19 | Assid | hehe |
| 22:19.23 | VoIPMasta | [TK]D-Fender, Did you see the SIP debug output? |
| 22:19.23 | Assid | okay i need milk |
| 22:19.23 | [TK]D-Fender | VoIPMasta, and your sip.conf please... |
| 22:20.38 | *** join/#asterisk Vec (n=Vec@dsl-243-115-51.telkomadsl.co.za) |
| 22:21.04 | VoIPMasta | http://www.pastebin.ca/414228 |
| 22:21.07 | VoIPMasta | sip.conf at the bottom |
| 22:21.17 | Assid | [TK]D-Fender: if the pap2-na only says its a pap2-na .. how do i iknow if its a spa-2102 |
| 22:21.21 | Assid | or a simple pap2na |
| 22:21.29 | [TK]D-Fender | Assid, it ISN'T |
| 22:21.37 | [TK]D-Fender | totally different models |
| 22:21.48 | Assid | is there any difference |
| 22:21.48 | VoIPMasta | Assid, SPA2102 has 2 ethernet ports |
| 22:21.48 | mvanbaak | zzzzzzzzzzzzzzzzzzzz time |
| 22:21.48 | Assid | oh |
| 22:21.48 | [TK]D-Fender | do not get the PAP2 |
| 22:21.48 | VoIPMasta | Assid, 1 wan, 1 lan |
| 22:21.49 | mvanbaak | latero |
| 22:21.55 | Assid | already have it |
| 22:22.01 | Vec | I am trying to get asterisk to dial my mobile but only put the call through if I push a key (eg#) if I do not it must continue with the dial plan. I tried to put a c after the trunk Zap/g1c as I read somewhere that the c will only connect the call if a # is pressed but this did not work, any ideas ? |
| 22:22.22 | mercestes | Vec: match your cell# plus # |
| 22:22.27 | [TK]D-Fender | VoIPMasta, Your * is also behind NAT, ccorrect? And what is the remote phone? |
| 22:22.40 | VoIPMasta | no, * is not behind NAT |
| 22:22.41 | mercestes | Vec: and then match your cell # without th # and playback(tt-monkeys) hangup() |
| 22:22.42 | cr4z3d | what's a good phone brand for analog phones w/ base stations |
| 22:22.57 | mercestes | cr4z3d, AT&T |
| 22:23.02 | VoIPMasta | the 5200001 friend/peer in the sip debug is a remote PAP2 |
| 22:23.04 | mvanbaak | cr4z3d: siemens |
| 22:23.07 | VoIPMasta | sorry ATA186 |
| 22:23.07 | Vec | mercestes : sorry not getting u ? |
| 22:23.29 | cr4z3d | what about like panasonic |
| 22:23.31 | mercestes | cr4z3d, Hitachi. :D |
| 22:23.42 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 22:23.44 | mercestes | Vec: DIalplan it to require a # |
| 22:23.47 | mvanbaak | or get the aastra with dect handsets |
| 22:23.48 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-124425.home.otenet.gr) |
| 22:23.52 | mvanbaak | they are sweet |
| 22:24.01 | Assid | [TK]D-Fender: besides the 2 lan ports.. any difference in audio quality etc? |
| 22:24.14 | *** join/#asterisk jks (n=njks@193.189.92.56) |
| 22:24.21 | mvanbaak | anywayz, time to please the lady |
| 22:24.22 | mvanbaak | bye |
| 22:24.26 | [TK]D-Fender | Assid, better performance, SPU on the unit, T.38 I believe as well. |
| 22:24.39 | jks | anyone know a channel for people using siemens gigaset? (or does anyone know what the new 02509 version includes?) |
| 22:24.43 | Assid | mvanbaak: does this mean your leaving her alone for a few hours? |
| 22:24.44 | Vec | mercestes : not sure how to dial(mymobile) then wait for key Background(please push # to accept call), The background audio will be played to the person dialing rather then my cell phone which is being dialed ? |
| 22:25.04 | mvanbaak | Assid: ;) |
| 22:25.06 | Assid | spu ? |
| 22:25.41 | mercestes | Vec: Then you ${CELLNUMBER}1,Background(Please press # to dial) and you pattern match exten => #,1,Dial(Zap/G1/${CELLNUMBER}) |
| 22:25.55 | BlackSlik | any got any telebanking solution that is basically webbased |
| 22:25.58 | cr4z3d | does asterisk handle FoIP or is there some other software i should look into for that? i'm just trying to get faxes intercepted and sent to my email |
| 22:26.03 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-124425.home.otenet.gr) |
| 22:26.43 | [TK]D-Fender | cr4z3d, w/ * 1.4 we are supposed to have T.38 pass-through, so this might work for you witha device like mentioned. |
| 22:27.22 | Assid | [TK]D-Fender: i cant connect a regular fax to this pap2-na ? |
| 22:27.27 | cr4z3d | oh alright i was just wondering if it was possible to just do it all through a computer and not actually have a fax machine |
| 22:27.34 | [TK]D-Fender | Assid, Go check its manual |
| 22:27.48 | [TK]D-Fender | cr4z3d, Not currently over VoIP. |
| 22:28.18 | *** join/#asterisk wunderkin (n=kev@dslstat-ppp-95.fastq.com) |
| 22:28.40 | VoIPMasta | [TK]D-Fender, any ideas? |
| 22:30.31 | [TK]D-Fender | VoIPMasta, that is a call TO the device correct? |
| 22:30.56 | VoIPMasta | yes |
| 22:31.07 | [TK]D-Fender | VoIPMasta, Does inbound work? |
| 22:31.14 | VoIPMasta | From Asterisk on a public IP to a Cisco ATA 186 behind a NAT |
| 22:31.26 | VoIPMasta | inbound... do you mean from the ATA to Asterisk? |
| 22:31.43 | [TK]D-Fender | VoIPMasta, Correct. From this device. TO it doesn, do calls work FROM it? |
| 22:31.50 | MrWup | i am using my SIP phone to call an external PSTN line via 4FXO card, but when i hang up with the SIP phone, the external phone remainds connected |
| 22:31.56 | MrWup | any idea why? |
| 22:32.01 | VoIPMasta | If I call ATA => Asterisk it works, ATA => Asterisk => PSTN works |
| 22:32.04 | MrWup | asterisk console seems to register the hangup |
| 22:32.15 | MrWup | but yet the FXO card doesnt actually cut off the caller |
| 22:32.16 | VoIPMasta | However Asterisk => ATA doesn't and PSTN => Asterisk => ATA doesn't |
| 22:32.22 | [TK]D-Fender | VoIPMasta, pastebin "sip show peer [peer]" |
| 22:33.05 | VoIPMasta | http://www.pastebin.ca/414241 |
| 22:33.07 | VoIPMasta | At the bottom |
| 22:33.47 | [TK]D-Fender | VoIPMasta, you must have "qualify=yes" and "canreinvite=no". Both are fatal. |
| 22:33.53 | [TK]D-Fender | (lack thereof) |
| 22:34.29 | VoIPMasta | ok they're seeded from the db |
| 22:34.45 | VoIPMasta | I have canreinvite=yes in the db |
| 22:34.57 | VoIPMasta | and I have qualify=999 in the db |
| 22:35.12 | [TK]D-Fender | canreinvite = NO <------ |
| 22:36.51 | [TK]D-Fender | VoIPMasta, And extend your qualify to 2000. Upon reload and reregistration of the remote device, do sip show peers and pastebin. Once OK'd then I'll ask you to make another call |
| 22:37.17 | VoIPMasta | [TK]D-Fender, the database schema does only allow for 3 digits |
| 22:37.25 | *** join/#asterisk jubei_ (n=Stormtro@147.27.47.114) |
| 22:37.52 | *** join/#asterisk ilan (i=ilan@69.60.110.251) |
| 22:38.20 | [TK]D-Fender | VoIPMasta, set to YES then |
| 22:38.36 | MrWup | =[ |
| 22:39.05 | JT | bah 6 months experience required |
| 22:39.13 | JT | looking into getting a cabling licence |
| 22:39.16 | JT | stupid laws |
| 22:40.46 | VoIPMasta | [TK]D-Fender, done |
| 22:41.52 | *** join/#asterisk garreel (n=garreel@host143-34-dynamic.17-87-r.retail.telecomitalia.it) |
| 22:42.26 | *** join/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 22:42.35 | [TK]D-Fender | VoIPMasta, reset the remote device |
| 22:43.01 | VoIPMasta | Done |
| 22:43.07 | VoIPMasta | The calls are going through perfectly :) |
| 22:44.31 | [TK]D-Fender | VoIPMasta, Let it site for 3 minutes without activity (intentional), and then try calling it again. I'd like to make sure the UDP window is kept open |
| 22:45.06 | VoIPMasta | ok |
| 22:45.18 | VoIPMasta | btw, do you think that it will have a negative impact if I disable rtcache? |
| 22:46.14 | *** join/#asterisk Fieldy (i=KvO3GLyL@gentoo/contributor/Fieldy) |
| 22:46.30 | [TK]D-Fender | VoIPMasta, I'm not really familiar with RT personally.... I highly suggest you leave it on in some capacity unless you aare comfortable enough with the load and uptime on it. That evaluation is in your hands. |
| 22:46.57 | *** join/#asterisk stridernzl (n=neville@125-237-110-53.jetstream.xtra.co.nz) |
| 22:47.10 | VoIPMasta | personally I don't like RT either, but I'm testing a web-gui that allows instant signup/activation of new users |
| 22:47.25 | VoIPMasta | once it's finished it will be GPLed :) |
| 22:48.09 | wunderkin | [TK]D-Fender, hey dude whats up? i put on your config + some of them reset to defaults and we have had one reboot since monday so i already know it is still bad, there was a thread on users on 3/21 & 3/23 about one guy having the same problem.. would you mind me tooing on there for me or are you not on the list? going to scream at voipsupply tomorrow to swap the other phones, trying to get some ammo together :D |
| 22:49.57 | Vec | How would I go about joining to seperate call legs, would I just put them in a meetme room, or is there an easier way ? |
| 22:50.46 | [TK]D-Fender | VoIPMasta, Test time |
| 22:51.13 | VoIPMasta | There was a test 1 minute ago (unintentional) |
| 22:51.21 | VoIPMasta | someone called my DID |
| 22:51.28 | VoIPMasta | and it's forwarded to my ATA186 |
| 22:51.29 | VoIPMasta | :) |
| 22:52.49 | MrWup | my sip phones are working fine with each other. very good voice quaity when i call between sip phones on my network without any trouble. but when one of the sip phones is caling an external phone via Zap channel (4FXO digium card to PSTN), whenever i speak into the handset of the sip phone while the externa caller is taking, it crackles and blocks out the voice of the external caller slightly |
| 22:53.27 | [TK]D-Fender | VoIPMasta, nothing says Thank-You quite like Paypal ;) |
| 22:53.29 | MrWup | anyone had this problem? know of a way to fix it? why it only happens when calling an external line? |
| 22:53.40 | MrWup | cause between local sips it is 100% ok in all ways |
| 22:54.25 | VoIPMasta | Name it |
| 22:55.17 | MrWup | name it? |
| 22:55.49 | VoIPMasta | [TK]D-Fender, name it :) |
| 22:59.37 | *** join/#asterisk kimo_sabe (i=nick@206.169.90.29) |
| 23:03.14 | jks | anyone knows if VAD / Comfort Noise is supported in the new asterisk versions? |
| 23:03.58 | russellb | jks: for the most part, yeah |
| 23:04.06 | russellb | if you enable internaltiming=yes in asterisk.conf |
| 23:04.12 | russellb | which requires zaptel ... |
| 23:04.28 | jks | I'm not using zaptel at all |
| 23:04.35 | russellb | load ztdummy |
| 23:04.39 | jks | I was just trying to use a G.729 annex B VoIP phone |
| 23:04.51 | jks | and I got an error from Asterisk saying comfort noise wasn't supported, please turn it off on the client |
| 23:04.53 | JT | russellb: what do you mean by "supported" |
| 23:04.59 | JT | does new version == 1.4? |
| 23:05.03 | russellb | yeah |
| 23:05.17 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 23:05.17 | *** mode/#asterisk [+o mog] by ChanServ |
| 23:05.22 | jks | russellb, so if I upgrade to 1.4.x that message would go away and it would work? |
| 23:05.23 | russellb | but ... we don't claim fully supported still |
| 23:05.33 | JT | russellb: what does "supported" mean? |
| 23:05.57 | russellb | i think the warning is still there |
| 23:06.09 | jks | okay... well, at least it's progressing :-) |
| 23:06.18 | jks | would be nice to have, but I'll probably just leave it disabled for now then |
| 23:06.20 | [TK]D-Fender | russellb, what kind of setups do not function (specific known failures)? |
| 23:06.44 | JT | i'm trying to find out just what supported means ;) |
| 23:06.50 | russellb | i don't even remember ... call recording might not work .. |
| 23:06.50 | *** join/#asterisk kgx (n=karuna@60.234.196.160) |
| 23:06.51 | *** join/#asterisk dseeb_ (n=dcb@58.169.110.130) |
| 23:07.02 | russellb | pretend i didn't say it :-p |
| 23:07.08 | JT | call recording works in 1.2 with silence supression |
| 23:07.27 | [TK]D-Fender | russellb, That some don't... or that any WILL? ;) |
| 23:07.45 | JT | imho, for it to be considered "fully supported" it MUST produce CNG noises when receving RTP with CNG |
| 23:07.55 | JT | and it must be able to send VAD |
| 23:08.04 | JT | receiving RTP with VAD i meant |
| 23:08.07 | russellb | yeah, so it does neither |
| 23:08.14 | JT | no comfort noises sounds awful |
| 23:08.16 | russellb | well, it can receive RTP with VAD ... |
| 23:08.19 | JT | well it doesn't sound |
| 23:08.26 | russellb | but doesn't send CNG i don't think |
| 23:08.31 | russellb | i don't feel like looking right now |
| 23:08.33 | russellb | working on something else |
| 23:08.34 | russellb | :) |
| 23:09.08 | JT | i know 1.2 doesn't do CNG, it's really hard to tell if a call is still working or if it has dropped out |
| 23:09.11 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 23:09.18 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 23:09.46 | [TK]D-Fender | russellb, whats the feedback on SLA looking like? |
| 23:10.13 | russellb | [TK]D-Fender: i haven't really heard people say it doesn't work, other than a few bug reports i already fixed |
| 23:10.20 | russellb | just reports of people trying to figure out how to set it up ... |
| 23:10.24 | russellb | So, I'm assuming it is working fine |
| 23:10.35 | russellb | and I am on to working on other things until I hear otherwise |
| 23:11.08 | [TK]D-Fender | russellb, Where those fixes PRE, or POST 1.4.2 release? |
| 23:11.31 | russellb | hm, I think one may have been post-1.4.2 |
| 23:11.50 | kimo_sabe | hmm, there's no IRC support for switchvox is there? |
| 23:12.00 | [TK]D-Fender | russellb, I guess I might still do well to begin my move to 1.2 here...... |
| 23:12.08 | [TK]D-Fender | 1.4* |
| 23:12.23 | russellb | [TK]D-Fender: but if you end up trying it sometime, please let me know how it goes |
| 23:12.35 | russellb | I still haven't heard someone say "I'm using it and it works for me" |
| 23:12.43 | russellb | but generally, you don't hear that .... you only hear from people having problems |
| 23:12.48 | russellb | the happy people keep to themselves |
| 23:13.34 | russellb | right now I'm really excited about this distributed generic event system I am building ... |
| 23:13.34 | [TK]D-Fender | russellb, I will test w Polycom IP 501 & 301 for you and report back specifically. |
| 23:13.41 | russellb | I should have distributed MWI pretty soon |
| 23:14.53 | [TK]D-Fender | russellb, HEY... how about adding generic CPIP support to the channel driver system and beginning work on the SIP specific end of the implentation? there is already a largely functional patch out there screaming to be templated for inclusion... |
| 23:15.02 | [TK]D-Fender | CPID* |
| 23:15.13 | mmartinn | Distributed generic event system? <perks up> |
| 23:15.15 | russellb | but that is exactly it, it is SIP specific |
| 23:15.30 | russellb | we *have* to solve it in a protocol independent way |
| 23:15.42 | russellb | but, it'll get done eventually |
| 23:15.52 | russellb | in fact, I have to have that to be able to have fully functional CallerID support in SLA |
| 23:16.15 | russellb | it's ... complicated |
| 23:16.15 | [TK]D-Fender | russellb, Yes, that specific implementation, but I'm sure a common abstract can be dummied up for it and the SIP part bolted on for the same net end effect.... |
| 23:16.39 | russellb | yeah, maybe so ... |
| 23:16.52 | russellb | but it's going to take a pretty significant amount of work to get it to where it can be merged |
| 23:16.58 | JT | cpid? |
| 23:16.59 | [TK]D-Fender | russellb, Well.. jsut "food for thought" from the list of "highly demanded by real world users" :) |
| 23:17.08 | russellb | [TK]D-Fender: I appreciate it :) |
| 23:17.16 | russellb | I would say it's on my list of things we need by 1.6 ... |
| 23:17.26 | [TK]D-Fender | russellb, I'll do my part in tested SLA for you with Polycom / Aastra |
| 23:17.28 | Strom_M | russellb: how familiar are you with zaptel and channel banks? |
| 23:17.32 | Strom_M | [TK]D-Fender, also |
| 23:17.41 | russellb | Strom_M: I know nothing |
| 23:17.49 | [TK]D-Fender | Strom_M, No doubt far less than russellb, but ask away |
| 23:17.51 | russellb | [TK]D-Fender: that would rock, whenever you have time |
| 23:17.54 | Strom_M | I'm having a weeeeird issue where DTMF is occasionally bouncing |
| 23:18.01 | [TK]D-Fender | Strom_M, Ok, maybe I know SLIGHTLY more than him then ;) |
| 23:18.01 | Strom_M | here's the scenario |
| 23:18.09 | russellb | [TK]D-Fender: in fact, if you test with polycom and use real config files to set it up, I'd like to add them to the documentation |
| 23:18.13 | mmartinn | 1.6? /me's head explodes |
| 23:18.26 | russellb | mmartinn: it's my job to think ahead :) |
| 23:18.28 | [TK]D-Fender | russellb, I'll MAKE time. glad to hear of your prgress on all this great stuff. |
| 23:18.29 | Qwell[] | 1.8! |
| 23:18.45 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 23:18.48 | Qwell[] | chan_skinny will probably support the redial softkey in 1.8 |
| 23:18.52 | Strom_M | this system has a TE406P with a PRI on span 1 and Adtran TA624 channel banks on spans 2-4; each channel bank has analog telephones hanging off of it |
| 23:18.58 | russellb | after distributed MWI, I will make distributed devicestate notification using the same framework |
| 23:19.04 | [TK]D-Fender | russellb, I ONLY provision, and from scratch. Somehow my rep hasn't entirely preceeded me with you ;) |
| 23:19.18 | russellb | and thennn, well ... I have a lot i want to do, hehe |
| 23:19.37 | russellb | [TK]D-Fender: ha, sorry ... |
| 23:19.45 | Strom_M | caller calls in, gets the IVR, dials operator. Operator does a hookflash transfer and dials extension 24. Occasionally, Asterisk thinks extension 22 is being dialed. |
| 23:20.00 | [TK]D-Fender | russellb, Oh yeah! Do we have a decicestate dialplan toggleable device yet in 1.4? |
| 23:20.17 | Strom_M | I can't figure out /what/ is causing this |
| 23:20.20 | russellb | [TK]D-Fender: ooh, ooh, I wrote that! |
| 23:20.21 | garreel | Strom_M: did you tryed relaxdtmf settings? |
| 23:20.22 | [TK]D-Fender | Strom_M, "relaxdtmf=yes" |
| 23:20.25 | [TK]D-Fender | :D |
| 23:20.38 | russellb | [TK]D-Fender: see my "Custom Device State" blog post on *.org |
| 23:20.38 | garreel | ;-) |
| 23:20.48 | [TK]D-Fender | garreel, Only reason you beat me is by not answering him FORMALLY ;) |
| 23:20.52 | Strom_M | what exactly will relaxdtmf do? wait for longer intervals between dialed digits? |
| 23:21.01 | garreel | I had same some weeks ago |
| 23:21.03 | garreel | ;-) |
| 23:21.05 | [TK]D-Fender | russellb, :O |
| 23:21.23 | Strom_M | garreel: you had a similar problem? |
| 23:21.26 | russellb | it's in trunk, but you can use it in 1.4 ... |
| 23:21.28 | *** join/#asterisk Erik2300 (n=ewilhelm@67-132-248-66.dia.static.qwest.net) |
| 23:21.28 | garreel | yes |
| 23:21.31 | Strom_M | ok |
| 23:21.35 | garreel | some weeks ago |
| 23:21.48 | russellb | [TK]D-Fender: I am on fire now that I get to do this full-time :-D |
| 23:21.51 | garreel | but only when i dialed on my pri box from an analog phone |
| 23:21.52 | Strom_M | was this on stations connected to asterisk? |
| 23:21.55 | Strom_M | ok |
| 23:22.22 | [TK]D-Fender | russellb, That was a good read.... |
| 23:22.24 | garreel | changet to relaxdtmf to yes and the problem has gone |
| 23:22.32 | [TK]D-Fender | ~russellb |
| 23:22.45 | jbot | hmm... russellb is Russell Bryant <russell@digium.com>, aka drumkilla, or 1337 |
| 23:22.49 | [TK]D-Fender | russellb, Ok... you just f'n ROCK.... |
| 23:22.52 | Strom_M | garreel: was the analog phone on a channel bank? |
| 23:23.19 | garreel | no... the analog phone was my home's phone |
| 23:23.21 | russellb | [TK]D-Fender: thanks :-D |
| 23:23.33 | Strom_M | hm |
| 23:23.56 | mmartinn | Does " Primary D-Channel on span 1 up " happen always after a down, or it can it happen when it was already up? |
| 23:24.04 | [TK]D-Fender | russellb, that IS scheduled for merge in the next 1.4 release, eright? |
| 23:24.17 | russellb | [TK]D-Fender: no, we don't add new features in release branches. |
| 23:24.23 | [TK]D-Fender | mmartinn, You should only see it if it was down prior |
| 23:24.24 | garreel | but noticed some duplicated digits on dtmf recognition |
| 23:24.33 | russellb | [TK]D-Fender: but, cp trunk/funcs/func_devstate.c 1.4/funcs |
| 23:24.41 | russellb | and it will work (until we make some API breakage in the future) |
| 23:25.02 | garreel | Strom_M: that seems your case also |
| 23:25.12 | [TK]D-Fender | russellb, *wah*... and a makefile patch as well I presume? |
| 23:25.22 | Strom_M | garreel: so this was only on calls where you had TEL -> PSTN -> PRI -> Asterisk ? |
| 23:25.32 | russellb | [TK]D-Fender: nope, the 1.4/trunk build system has a lot of magic ... it will get automatically compiled and installed if you drop it in |
| 23:25.44 | mmartinn | [TK]D-Fender: How would I rule out problems with the D-Channel on my end? |
| 23:25.46 | garreel | Strom_M: I don't have channel banks |
| 23:25.55 | Strom_M | right |
| 23:25.55 | [TK]D-Fender | russellb, Ok, well small blessing in that... will save this chat in reference |
| 23:26.02 | russellb | sweet |
| 23:26.21 | Strom_M | i'm just trying to confirm, garreel, that this was all TDM and not VoIP at any point in the connection |
| 23:26.53 | garreel | Strom_M: for me that happened only on TEL -> PSTN -> PRI -> * |
| 23:26.56 | Strom_M | ok |
| 23:27.23 | *** join/#asterisk Andretii (i=dupont@adsl-75-22-238-229.dsl.chcgil.sbcglobal.net) |
| 23:27.26 | garreel | Strom_M: no problems when I tried with CELL -> PSTN -> PRI -> * |
| 23:27.49 | garreel | or with SIP -> * -> PRI -> PSTN -> TEL |
| 23:28.23 | Andretii | Hello everyone, can anyone tell me the command to restart SAP thabks |
| 23:28.26 | garreel | "relaxdtmf=yes" solved this issue completely |
| 23:33.39 | [TK]D-Fender | Andretii, Sorry... I'm stuck using PeopleSoft.... |
| 23:34.37 | mmartinn | What is PRI/ISDN cause code 127, i.e. "This cause indicates that an interworking call (usually a call to 5W56 service) has ended."? What does that mean? |
| 23:35.27 | Andretii | i meant ZAP |
| 23:35.40 | Andretii | i need to know the command to restart the process |
| 23:37.30 | [TK]D-Fender | what process? |
| 23:37.42 | Erik2300 | can anyone help with voicemail IMAP configuration? |
| 23:38.30 | Andretii | [TK]D-Fender my incoming calls are not working and somebody told me to restart zap |
| 23:39.17 | Andretii | also my asterisks' system menu dont show anything |
| 23:41.00 | *** join/#asterisk Dr-Linux|home (n=asfdf@DSL-202-59-73-131.nexlinx.net.pk) |
| 23:41.37 | Dr-Linux|home | anybody tried LumenVox speech recognition with AGI? or perl? |
| 23:41.38 | [TK]D-Fender | Andretii, "reload chan_zap.so" |
| 23:44.08 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 23:44.20 | Andretii | [TK]D-Fender shows command not found |
| 23:45.06 | [TK]D-Fender | Andretii, check the .4 syntax for it... its changed. |
| 23:45.16 | Dr-Linux|home | Andretii: try it on cosol like CLI> reload chan_zap.so |
| 23:45.29 | [TK]D-Fender | russellb, Btw... your sample screams for me to reinstall my X-10 setup I sued to have running on* as it is ;) |
| 23:45.34 | [TK]D-Fender | used* |
| 23:45.44 | Dovid | or: asterisk -rx 'reload chan_zap.so' |
| 23:45.49 | dj-fu | Is there a security issue in 1.4 at the moment? |
| 23:46.05 | Dovid | with 1.4 yes not 1.4.2 |
| 23:46.16 | russellb | [TK]D-Fender: heck yeah. If I had that, I'd do it |
| 23:46.24 | dj-fu | Dovid, just 1.4.0? |
| 23:46.25 | russellb | [TK]D-Fender: but I'd be happy to know that someone else was able to :) |
| 23:46.45 | Dovid | 1.40 and 1.4.1 |
| 23:46.46 | mmartinn | Is there any way to get pri debug to have a timestamp? |
| 23:46.49 | Dovid | as well as some of 1.2.X |
| 23:47.34 | [TK]D-Fender | russellb, unfortunately I only run a CM11A and 1-way modules, so they can't report back, but you headed in ALL the right directions with this. I'm jsut sad to hear that it isn't to be included in MAINLINE. that means a LONG period of patching manually afor this |
| 23:49.21 | MrWup | i use my cellphone to dial into asterisk and dial a sip handset. when i hang up on the sip handset, asterisk says hungp up on the zap channel, but my cellphone still remains connected for almost a minute afterwards |
| 23:49.25 | MrWup | any idea why? |
| 23:49.43 | russellb | [TK]D-Fender: yeah, but for the sake of stability, you should appreciate our resistance to unneccessarily changing code in releases :) |
| 23:50.11 | Erik2300 | anyone have experience with getting asterisk 1.4.2 IMAP working? |
| 23:50.30 | Erik2300 | I have it installed and it gets an error |
| 23:50.46 | Erik2300 | when trying to connect using the UW c-client |
| 23:50.58 | [TK]D-Fender | russellb, wel.. this IS merely dependant on the core functionality while not being PART of it. This is a mixed issue I guess :) Make just a change in revision parttern.... |
| 23:51.02 | Erik2300 | to my IMAP server |
| 23:51.16 | [TK]D-Fender | russellb, But for as long as this is repeatable, I'll happily live with it |
| 23:51.26 | MrWup | anyone? |
| 23:52.44 | [TK]D-Fender | MrWup, try a call, pastebin it at verbose 10. upon declaration of hangup do a "show channels", then "zap show channels" and tell us where in there you regain access to your Zap again |
| 23:53.52 | MrWup | ok will try it at the office tomorrow |
| 23:53.58 | MrWup | dont have access from here |
| 23:54.02 | mmartinn | Is there any way to get pri debug to have a timestamp? |
| 23:55.29 | Erik2300 | anyone with IMAP and asterisk |
| 23:56.20 | [TK]D-Fender | MrWup, Don't ask us to help you and then tell us we CAn'T :) |
| 23:59.04 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
