| 00:02.11 | LoneShadow | does "exten => s,1,NoOp(${EXTEN}) |
| 00:02.15 | LoneShadow | work as a default rule ? |
| 00:02.23 | marl | ok, sorry, found the inital problem, now have the call disconected if i dont hit the '#' key, but both ends apear to be able to hear each other while waiting for the '#' and theres no prompts of any kind, anyone point me to a way to silence the lines, and allow an anouncment to the called number? |
| 00:07.39 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:10.08 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:11.23 | Shaun2222 | when sending a reload to asterisk does it kill any calls or sip/iax connections? |
| 00:18.42 | Bullseye_Network | Shaun2222: No it SHOULDNT kill any calls |
| 00:20.04 | LoneShadow | is extension "s" supposed to be a default rule for undefined extensions ? |
| 00:20.59 | *** join/#asterisk Nitrus^ (n=ntisog@72-34-76-86.skyriver.net) |
| 00:21.22 | Nitrus^ | how do i prevent callers from hearing me dialing the feature tones for transfer and disconnect? |
| 00:23.50 | hads | LoneShadow: s is the extension that is used in the situation that there is no dialled number e.g a call from a POTS line. |
| 00:25.41 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 00:28.25 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 00:30.13 | LoneShadow | hads: I am trying to define a default rule for all undefined extensions rules, so that I can forward them out thru my pstn gateway |
| 00:30.34 | hads | _X. |
| 00:31.13 | hads | Put it in a context and include that context in your other context(s) |
| 00:31.18 | Nitrus^ | can someone recommend a voip provider they've experience with? |
| 00:31.35 | Nitrus^ | sorry to poll, im just sort of lost with all these damn providers |
| 00:31.53 | LoneShadow | can you give me an example where exention s might be useful ? |
| 00:32.30 | hads | erm.. I just did [12:23:49] <hads> LoneShadow: s is the extension that is used in the situation that there is no dialled number e.g a call from a POTS line. |
| 00:33.32 | *** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com) |
| 00:33.46 | LoneShadow | hmm, a call from POTS line ? like from an sip user ? |
| 00:34.19 | hads | Like if you have an analog telephone line, and it rings. |
| 00:34.46 | LoneShadow | I have a spa3k, so have a voip gw and pstn gw |
| 00:35.00 | LoneShadow | so are you talking about incoming calls from pstn gw ? |
| 00:35.51 | *** join/#asterisk MacWeenie (n=chatzill@cpe-74-65-228-134.nyc.res.rr.com) |
| 00:36.11 | LoneShadow | Nitrus^: I use sunrocket in US, I was able to get thier $199/2yrs deal, also was able to get the admin passwd for the sunrocket gizmo |
| 00:36.30 | LoneShadow | some folks seem to like viatalk |
| 00:37.10 | knarfly | I just revamped my dialplan and get this message when I connect to my sip phone. The connection works so I would like to know if the message needs attention. |
| 00:37.13 | knarfly | Aug 7 20:34:28 NOTICE[77135]: app_dial.c:1040 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 00:37.44 | LoneShadow | do "sip show peers" and "sip show registry" |
| 00:37.47 | *** join/#asterisk Samoied (n=Samoied@201.21.216.149) |
| 00:37.52 | LoneShadow | see if the ip addresses are right |
| 00:39.05 | knarfly | I have a sip registry. it's ok. there are also six sip peers in this sip.conf |
| 00:39.11 | knarfly | they all show up |
| 00:39.36 | knarfly | like I said the call works, this is a new DID setup I'm working |
| 00:41.50 | Nitrus^ | loneshadow: are you using for business or personal/ |
| 00:42.04 | LoneShadow | personal |
| 00:42.46 | Nitrus^ | ohh, i was trying to see if there is anyone out there who does multiple lines for cheap. i dont like vonage because i dont want to use their voip -> ptsn equipment |
| 00:43.07 | LoneShadow | you trying to use this for biz ? |
| 00:43.33 | Nitrus^ | yeah |
| 00:43.54 | LoneShadow | dont know who is best for that :) |
| 00:44.05 | Nitrus^ | yeah ill keep hunting |
| 00:44.08 | Nitrus^ | thanks though |
| 00:44.13 | LoneShadow | hads: thanks for the _X. rule |
| 00:44.18 | knarfly | Nitrus^: checkout myvoice.splitinfinity.com |
| 00:44.59 | knarfly | Nitrus^: just call the tech support number..tell 'em user #49 sent you |
| 00:46.36 | Nitrus^ | aight ill give it a shot |
| 00:50.54 | *** join/#asterisk tempest1 (n=asf@adsl-230-226-139.chs.bellsouth.net) |
| 00:51.22 | Nitrus^ | knarfly: do they support IAX? |
| 00:51.34 | Nitrus^ | or are they non-asterisk based |
| 00:53.15 | *** join/#asterisk hoytbowUE (i=webspinn@203.11.107.242) |
| 00:54.25 | hoytbowUE | Hi I was wondering if anybdy could help with a prob I am having compiling the latest SVN asterisk... |
| 00:55.19 | hoytbowUE | the error I get is 'No rule to make target `res_adsi.so', needed by `_all' |
| 00:56.40 | xheliox | hoytbowUE: Did you do make menuselect and customize anything? |
| 00:56.40 | *** join/#asterisk anthm (n=anthm@000-445-492.area4.spcsdns.net) |
| 00:56.41 | *** mode/#asterisk [+o anthm] by ChanServ |
| 00:57.22 | hoytbowUE | no... after I got this error I went into menuselect and turned off adsi, then it gave me the same error about res_agi.so |
| 00:57.50 | xheliox | res_agi or adsi? |
| 00:58.50 | *** join/#asterisk supjigatr (n=syslod@152.53.17.26) |
| 00:59.46 | hoytbowUE | originall res_adsi... then after that error I modified with menuselect turning off adsi, make clean, then make and then compile passed thru till error No rule to make target `res_agi.so', needed by `_all' |
| 01:01.59 | hoytbowUE | What I am actally trying to do is compile a trunk version with support for H264 so I can get the new grandstream phones going |
| 01:03.30 | *** join/#asterisk tempest1 (n=asf@adsl-144-35-104.chs.bellsouth.net) |
| 01:04.00 | hoytbowUE | I have got the version from asterisk.org 1.2.10 compiled an working finr but doesnt have support for H264 codec |
| 01:04.56 | LoneShadow | anyone using externhost and externrefresh ? |
| 01:12.05 | knarfly | Nitrus^: No SIP |
| 01:13.06 | knarfly | Nitrus^: they are * friendly...their techs are good with * setup |
| 01:13.13 | ManxPower | hoytbowUE, does the 1.2.x tarball cause the same error? |
| 01:13.29 | ManxPower | oh, nevermind. |
| 01:13.39 | hoytbowUE | ManxPower: no it doesnt |
| 01:13.55 | ManxPower | hoytbowUE, well SVN is the developement version of Asterisk, so it WILL be broken in various ways at various times. |
| 01:14.26 | hoytbowUE | any ideas about incorporating H264 into 1.2.x? |
| 01:14.40 | ManxPower | nope. |
| 01:14.51 | hoytbowUE | :0 |
| 01:14.55 | hoytbowUE | :) |
| 01:15.16 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 01:18.44 | *** join/#asterisk lin00bies (n=jsharryp@210.213.198.138) |
| 01:19.00 | *** join/#asterisk Flauto (n=HP_Owner@adsl-75-3-169-144.dsl.chcgil.sbcglobal.net) |
| 01:19.38 | *** join/#asterisk dec_ (n=tom@ppp165-186.lns3.adl4.internode.on.net) |
| 01:20.09 | dec_ | hi all, how can i tell the asterisk CLI application where to look for the asterisk.conf file? (so I can connect with asterisk -cr) |
| 01:21.12 | dec_ | oh -C :) |
| 01:21.16 | Flauto | hi all |
| 01:24.12 | *** part/#asterisk dec_ (n=tom@ppp165-186.lns3.adl4.internode.on.net) |
| 01:26.18 | *** join/#asterisk TeePOG (n=TeePOG@dsl-145-143-233.telkomadsl.co.za) |
| 01:26.40 | Flauto | hi all |
| 01:27.54 | *** join/#asterisk trbldwine (i=troubled@71.194.161.170) |
| 01:28.56 | *** join/#asterisk trbldwine (i=troubled@71.194.161.170) |
| 01:28.58 | *** join/#asterisk MnDBnDr (n=mndbndr@cpe-24-93-167-181.neo.res.rr.com) |
| 01:35.06 | *** join/#asterisk MnDBnDr (n=polx@cpe-24-93-167-181.neo.res.rr.com) |
| 01:37.21 | *** join/#asterisk tempest1 (n=asf@adsl-144-35-104.chs.bellsouth.net) |
| 01:38.57 | *** join/#asterisk Mike (n=mike@201.112.50.158) |
| 01:39.24 | Mike | hi guys, anyone using trunk mode in high traffic? |
| 01:40.07 | *** join/#asterisk MnDBnDr (n=mndbndr@cpe-24-93-167-181.neo.res.rr.com) |
| 01:40.18 | hoytbowUE | Mike: I havent found it to make much difference for iax |
| 01:40.31 | Mike | hoytbowUE, what do you mean? |
| 01:40.46 | Mike | hoytbowUE, on the wiki they seem to say there is a BIG difference |
| 01:41.03 | Mike | hoytbowUE, 100 calls in 1mb link using g729 and trunk mode |
| 01:41.48 | *** join/#asterisk rene1 (n=rene1@gea-gye-internet.telconet.net) |
| 01:41.55 | hoytbowUE | maybe I am not hi traffic :) only have 1000 users... have a decent link to the net tho but found that I hade more artifacts with it turned on than off |
| 01:42.11 | rene1 | hello, i am using astmanproxy and i wonder if it can accept both http and standard i/o from clients. |
| 01:42.38 | rene1 | it is working very well for me in standard i/o but http just keep showing events i didnt tell it to |
| 01:42.43 | Mike | hoytbowUE, it doesnt seem to work for me, i just added the trunk=yes on the context of my sending server i have on both digium cards and seems not to work |
| 01:42.49 | Mike | hoytbowUE, anything im missing? |
| 01:45.13 | hoytbowUE | hmmm... been a while.. just lemme check something brb |
| 01:46.44 | hoytbowUE | mike... is your hardware timer working fine? |
| 01:46.55 | Mike | hoytbowUE, how do i know that? |
| 01:48.28 | *** join/#asterisk LoneShadow (n=duh@59.92.133.150) |
| 01:49.14 | hoytbowUE | what modules are loaded for your zaptel interfaces |
| 01:50.01 | LoneShadow | is there a way to make client to re-register with the right ip address(dydns hostname) when server changes its ip ? |
| 01:50.10 | LoneShadow | between 2 asterisk box |
| 01:50.20 | hoytbowUE | mike:what modules are loaded for your zaptel interfaces? |
| 01:52.33 | hoytbowUE | mike: need an rtc module ie genrtc (or mebbe ztdummy)... trunking requires hardware timing to work |
| 01:53.06 | Mike | yes |
| 01:53.08 | LoneShadow | hmm ssh asterisk reload might work |
| 01:53.08 | Mike | i have rtc |
| 01:53.52 | Mike | hoytbowUE, zaptel rtc wct4xxp 119264 60 |
| 01:53.52 | Mike | zaptel 188228 203 wct4xxp |
| 01:54.27 | hoytbowUE | Trunking should work.. is it set in the recieving boxes context too? (I cant remember if it needs to be, if not then it should be fine) |
| 01:54.31 | LoneShadow | can anyone suggest anything better than "ssh root@remote-asterisk-client asterisk -rx 'iax2 reload'" ? |
| 01:55.58 | *** join/#asterisk trbldwine (i=troubled@71.194.161.170) |
| 01:56.01 | quid2478 | Anybody here run some serious call volumes through *? |
| 01:56.10 | Qwell | quid2478: 3 calls per week |
| 01:56.22 | Mike | hoytbowUE, only on the reciving boxes is set on the context of the master |
| 01:56.26 | quid2478 | dang... talk about workingyoure CPU into overdrive |
| 01:56.38 | Mike | hoytbowUE, the master is sending IAX2/user:pass/exten... |
| 01:56.57 | *** join/#asterisk kareena (n=k@adsl196-66-208-217-196.adsl196-15.iam.net.ma) |
| 01:57.43 | hoytbowUE | quid2478:1000+ day... not that high bet is getting a few thru |
| 01:57.57 | hoytbowUE | bet=but |
| 02:01.07 | hoytbowUE | mike... looks right to me, just needs trunk=yes in context and timer to work |
| 02:06.18 | *** join/#asterisk angom_h (n=angom@red-corp-200.38.15.140.telnor.net) |
| 02:15.07 | *** join/#asterisk IOscanner (n=IOscanne@c-67-164-154-209.hsd1.tx.comcast.net) |
| 02:18.45 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 02:21.06 | *** join/#asterisk apardo (n=apardo@87.217.147.151) |
| 02:23.51 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-179-72.lsanca.fios.verizon.net) |
| 02:24.30 | StewLG | Are the cause codes here the same as I'd see coming back from iax2 debug output?: http://tinyurl.com/jwlyk |
| 02:25.22 | *** join/#asterisk angom_h (n=angom@red-corp-200.79.146.88.telnor.net) |
| 02:30.29 | *** join/#asterisk Klitz (n=Klitz@24.96.201.72) |
| 02:32.47 | ManxPower | StewLG, those would be HANGUPCAUSE |
| 02:33.05 | ManxPower | They are ISDN PRI (Q.931?) codes. IAX uses them too. |
| 02:33.46 | StewLG | Ok, so, if I'm seeing a 21 (and I am), it's a AST_CAUSE_CALL_REJECTED. |
| 02:35.02 | *** join/#asterisk Un1x (i=Sean@CPE00d0b73f7bbf-CM00159a075322.cpe.net.cable.rogers.com) |
| 02:37.10 | ManxPower | IT should be. |
| 02:37.17 | ManxPower | but that should also show up on the console. |
| 02:37.54 | StewLG | Only shows me the reason (the code) with iax2 debugging on. |
| 02:39.35 | *** join/#asterisk atapi2 (n=virgill4@c-69-180-119-156.hsd1.fl.comcast.net) |
| 02:39.40 | atapi2 | hi all |
| 02:39.49 | atapi2 | anyone use lumenvox with asterisk? |
| 02:40.03 | Qwell | atapi2: file :D |
| 02:40.14 | file | uh oh? |
| 02:40.24 | file | what'cha wanna know? |
| 02:40.27 | atapi2 | speech to text |
| 02:40.47 | atapi2 | i need to record mailboxes |
| 02:40.52 | atapi2 | and convert them to text |
| 02:41.03 | file | that's not what the lumenvox SRE is for |
| 02:41.28 | file | it recognizes words you give to it essentially |
| 02:41.44 | file | so you write a grammar of words you want to listen for, and when it recognizes them... you get them back |
| 02:42.05 | atapi2 | i need to convert about 1000 messages to text |
| 02:42.29 | JT | hire a transcriber? |
| 02:42.59 | atapi2 | already do |
| 02:43.05 | atapi2 | its too slow |
| 02:43.50 | atapi2 | i just need the name from the message |
| 02:43.53 | JT | i don't think there's much voice recognition software that comes close to the level you need |
| 02:44.06 | Mike | hoytbowUE, i get this Ending trunk processing with 1 peers and 14 call chunks processed |
| 02:44.44 | atapi2 | i dont need alot of accuracy |
| 02:48.07 | ManxPower | atapi2, This is what you will experience: http://www.userfriendly.org/static/ |
| 02:49.15 | atapi2 | lol |
| 02:50.53 | *** join/#asterisk slinabery (n=slinaber@smooth.worldcycling.com) |
| 02:51.47 | slinabery | this is an irc noob with a question about the TE100P card. |
| 02:52.26 | slinabery | is this text going to all or just echoing on my irc client? |
| 02:52.40 | JT | just echoing |
| 02:54.02 | file | yeah, only you are seeing it |
| 02:54.24 | slinabery | That's what I thought. OK, I have a TE110P connected to a T1 with 20 channels for voice. When I dial into my asterisk pbx, * is getting only the first two of three dtmf digits. signalling is e&m wink; I don't know what parameters to adjust. |
| 02:54.42 | ManxPower | slinabery, where are you dialing in from? |
| 02:55.08 | ManxPower | cell phone, pbx, your grandmother's phone, which was last replaced in 1963 |
| 02:55.14 | slinabery | just calling in with my cell. mcleodusa is the provider of the T1, and we have 20 DIDs set up with them. So I'm just calling any of those. |
| 02:55.17 | hmmhesays | and two hours later I feel as though I want to hurl my guitar thorugh a window |
| 02:55.21 | hmmhesays | *through even |
| 02:55.52 | ManxPower | slinabery, I assume the call gets into Asterisk just fine, then you have DTMF issues? |
| 02:55.58 | Klitz | did mcleod test the T1 with a T-Berd before leaving? |
| 02:55.59 | hmmhesays | do the zaptel drivers support overlap dial? |
| 02:56.09 | ManxPower | hmmhesays, only on PRI |
| 02:56.35 | hmmhesays | cas can be a bitch |
| 02:56.47 | *** part/#asterisk kareena (n=k@adsl196-66-208-217-196.adsl196-15.iam.net.ma) |
| 02:57.06 | hmmhesays | where is [TK]D-Fender when I need some encouragement damnit |
| 02:57.09 | slinabery | I dial in on DID 535; * tries to send the call to extension 53. Receives the DTMF '5' after it's sent the call to extn 53. |
| 02:57.13 | ManxPower | slinabery, make it fast, I'm not getting any younger, grasshopper. |
| 02:57.27 | Klitz | I am so caloused to the telcos that I never believe them any more... I just assume it's their screw-up |
| 02:57.41 | ManxPower | slinabery, OK. This could be cause by several things. |
| 02:57.45 | hmmhesays | that is a good stance to take |
| 02:58.13 | ManxPower | slinabery, I would suspect your wink timing. |
| 02:58.22 | slinabery | the T1 is normally connected to a Merlin Magix which I'm trying to replace. So the T is functional with that other device. This isn't a new setup, in terms of the T. |
| 02:58.29 | ManxPower | slinabery, want me to check my E&M zaptel settings for you? |
| 02:58.51 | *** join/#asterisk javar (n=javar@200.118.174.253) |
| 02:59.01 | slinabery | shure, that would be great WP. |
| 02:59.28 | ManxPower | slinabery, Try these options BEFORE the channel => lines. |
| 02:59.30 | ManxPower | wink=270 |
| 02:59.30 | ManxPower | rxwink=270 |
| 02:59.39 | ManxPower | I don't think it will help, but see. |
| 02:59.46 | slinabery | okie I'll try it thanks |
| 03:00.18 | ManxPower | slinabery, you didn't do something stupid like set immediate=yes, did you? |
| 03:00.39 | slinabery | WP: no. |
| 03:00.52 | JT | WP? |
| 03:00.53 | ManxPower | slinabery, What is your signalling= set to? |
| 03:01.06 | ManxPower | JT, WanxPower. He types with a lisp. |
| 03:01.19 | JT | i see |
| 03:01.30 | Klitz | lmao - lisp |
| 03:02.19 | Klitz | don't know much about zap T1's, but could there be an 'inter-digit' timeout causing the problem? |
| 03:02.20 | slinabery | WanxPower signalling=em_w Just used e&m wink b/c that's what was in the configuration on the Avaya device. |
| 03:02.34 | ManxPower | slinabery, a logical thing to do. |
| 03:03.05 | ManxPower | Wanted to make sure you were not using plain E&M. Most DIDs T-1s that are non-PRI that I've seen are E&M Wink |
| 03:03.07 | *** part/#asterisk javar (n=javar@200.118.174.253) |
| 03:04.20 | ManxPower | slinabery, Try increasing or decreasing your rxgain= Asterisk may be missing digits. Oh, and set relaxdtmf=no |
| 03:04.27 | *** join/#asterisk javar (n=javar@200.118.174.253) |
| 03:04.50 | slinabery | WanxPower, ok will try that. first suggestion did not fix (wink & rxwink setting) |
| 03:05.02 | ManxPower | slinabery, As I said I didn't think it would. 8-) |
| 03:05.25 | slinabery | WanxPower s'ok I didn't think it would either |
| 03:05.37 | slinabery | WanxPower just because I'm a pessimist |
| 03:05.38 | ManxPower | slinabery, It's ACTING like Asterisk is not giving the telco enough time to send all the digits, but I've never actually seen that happen. |
| 03:06.03 | slinabery | WanxPower yes that's what it seems like to me also |
| 03:06.24 | ManxPower | And I don't know what option it would be to change that timeout. start= maybe? |
| 03:06.58 | slinabery | WanxPower but I have not been able to google up anyone else having a similar prob. I have not seen any place where you can set it. |
| 03:07.10 | ManxPower | slinabery, not many people use E&M. |
| 03:07.22 | ManxPower | If you have a choice, use PRI. |
| 03:07.37 | *** join/#asterisk Qwell_ (n=north@unaffiliated/qwell) |
| 03:07.37 | *** mode/#asterisk [+o Qwell_] by ChanServ |
| 03:08.07 | ManxPower | I only have 1 current E&M Wink T-1 in use, all the others are now PRI. |
| 03:08.30 | slinabery | WanxPower ah. I kinda have to work with what I've got; I'm trying to tweak this during off-hours, but the * config I'm working up is nowhere near production ready. So I've got to be able to plug back into the Avaya when I leave. |
| 03:08.45 | ManxPower | slinabery, *nod* |
| 03:09.47 | ManxPower | The location that that I manage that has not moved to PRI is totally undocumented as to where the channels are going, so we are leaving well enough alone. |
| 03:10.32 | slinabery | WanxPower did I mention that when I tail -f /var/log/asterisk/messages, it shows all the digits coming through eventually? So if I dial 535, * tries to send the call to extn 53, but the last '5' shows up. Nothing seems to be missing; it's just like it's picking up too soon. |
| 03:10.55 | ManxPower | slinabery, yup. |
| 03:11.18 | ManxPower | slinabery, if you don't fix it tonight, post a message to the -users mailing list. |
| 03:12.13 | ManxPower | slinabery, how many seconds (based on the timestamps in the logs) between the first digit and the last digit? |
| 03:12.54 | slinabery | WanxPower, The timing is within 1-2 seconds; don't know how (or if I can) to set log time resolution to milliseconds. |
| 03:13.09 | ManxPower | slinabery, try start=3000 |
| 03:13.18 | slinabery | for three digits, it's usually in the same second; sometimes over two secs. |
| 03:13.26 | ManxPower | slinabery, do you see the winks in the logs |
| 03:14.18 | slinabery | WanxPower don't know what I would be looking for. do I need to enable that level of debugging in the zaptel driver when it's module is loaded? |
| 03:14.34 | slinabery | its, that is, not it's (pet peeve) |
| 03:14.43 | ManxPower | slinabery, start asterisk as "asterisk -cvvvddd" |
| 03:15.03 | ManxPower | or "set debug 99" in the CLI |
| 03:16.34 | slinabery | WanxPower no I don't see the winks, but again don't know how they'd appear in the log. |
| 03:17.51 | ManxPower | slinabery, neither do I. |
| 03:18.18 | ManxPower | slinabery, are you SURE the existing PBX is set for E&M Wink and not just E&M? |
| 03:18.36 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 03:19.43 | slinabery | WanxPower No. I'll try plain e&m. |
| 03:20.53 | ManxPower | slinabery, again, I doubt it will fix the problem. |
| 03:21.12 | ManxPower | remember when you change signaling you have to stop and start asterisk or do a unload chan_zap.so and a load chan_zap.so |
| 03:21.19 | slinabery | yep |
| 03:22.36 | *** join/#asterisk lvit (n=kvit@203.209.56.65) |
| 03:24.54 | slinabery | WanxPower, no, it seems to be em_w alright. Do these config lines from the Avaya mean anything to you: |
| 03:25.00 | slinabery | A TRUNK 805 Slot/Port : 5/ 1 Toll |
| 03:25.08 | slinabery | A Direction: 2 Way E&M Signal: Type1S Dialtone : Remote |
| 03:25.11 | slinabery | A InType : Wink InMode : TouchTone AnsSupvr : 300 ms |
| 03:25.15 | slinabery | A OutType : Wink OutMode : TouchTone Disconnect: 300 ms |
| 03:25.16 | slinabery | end |
| 03:25.29 | *** join/#asterisk Dico_ (n=niko@60.51.217.61) |
| 03:25.59 | ManxPower | try setting wink and rxwink to 300 |
| 03:26.27 | ManxPower | let me look at a few things while you do that |
| 03:27.03 | *** join/#asterisk nwhit_home (n=chatzill@traffic.whittrio.com) |
| 03:27.09 | slinabery | WanxPower, same deal with wink=rxwink=300. |
| 03:27.46 | nwhit_home | i am having an issue with call parking that just came up with recent asterisk releases |
| 03:28.25 | ManxPower | make sure you have callprogress=no and usecallerid=no at well |
| 03:28.42 | nwhit_home | i am using snom hardphones and i use a moderated transfer to park the calls |
| 03:28.49 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 03:28.59 | shmaltz | hi everyone |
| 03:29.32 | nwhit_home | this has worked until recently.. now the system notifies the user of the park extension, but after transfering it, the system hangs up on the caller and says that he was tired of waiting |
| 03:29.33 | shmaltz | ~sleeping |
| 03:29.35 | jbot | extra, extra, read all about it, sleeping is the magical thing geeks have forgotten how to do |
| 03:29.47 | nwhit_home | if i transfer really quickly, there is no problem |
| 03:29.50 | shmaltz | ~sleep |
| 03:29.52 | jbot | i heard sleep is overrated, and a poor substitute for caffeine. |
| 03:30.12 | shmaltz | ~sex |
| 03:30.14 | jbot | updatedb; locate; talk; date; cd; strip; look; touch; finger; unzip; uptime; gawk; head; apt-get install condom; mount; fsck; gasp; more; yes; yes; yes; more; umount; apt-get remove --purge condom; make clean; sleep, or super extractor, http://sf.net/projects/sex/ |
| 03:30.55 | slinabery | WanxPower, I'm bailing. Muchas gracias; I'll put the details on the users list. Cheers, Steve. |
| 03:31.24 | shmaltz | ~time |
| 03:31.26 | jbot | You are educated stupid and therefore too dumb to understand nature's perfect time cube! (2006.08.08 3:31:26 GMT) |
| 03:31.46 | shmaltz | ~seen shmaltz |
| 03:31.51 | jbot | shmaltz is currently on #asterisk (3m 2s). Has said a total of 6 messages. Is idling for 5s, last said: '~seen shmaltz'. |
| 03:31.59 | shmaltz | ~pregnant |
| 03:32.00 | jbot | Yes, shmaltz, and it's your child. |
| 03:33.08 | *** part/#asterisk slinabery (n=slinaber@smooth.worldcycling.com) |
| 03:33.12 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 03:33.12 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:41.38 | *** join/#asterisk EvilDeshi (n=SkunK@oxford-bb-occam9-ws-25.dsl.maqs.net) |
| 03:43.39 | *** join/#asterisk salaud (n=salaud@72.11.116.122) |
| 03:46.58 | Mike | ok guys, im getting trunking working now on one side on the reciving server any ideas? |
| 03:51.22 | *** join/#asterisk jart (n=user@ool-4356f82f.dyn.optonline.net) |
| 03:58.59 | *** join/#asterisk froguz (n=froguz@224-138-222-201.adsl.terra.cl) |
| 03:59.07 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 03:59.07 | *** mode/#asterisk [+o denon] by ChanServ |
| 04:00.20 | froguz | hi russellb, i'm testing the trunk. how cool is menuselect! |
| 04:00.41 | russellb | thanks :) |
| 04:01.02 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 04:01.13 | froguz | does it make the old "make configure" automatically? |
| 04:01.33 | russellb | i'm not sure what you mean |
| 04:02.15 | froguz | "make configure" build the init scripts |
| 04:02.26 | russellb | build the configure script, you mean? |
| 04:02.46 | froguz | i made my contribution adding debian/ubuntu setup in that |
| 04:02.56 | russellb | ohhh, that, "make config" |
| 04:03.05 | russellb | no, you still run "make config" to do that |
| 04:03.10 | froguz | the scripts to make asterisk run at startup |
| 04:03.22 | hoytbowUE | russelb: did you get an error on compile like this No rule to make target `res_adsi.so', needed by `_all' |
| 04:03.34 | froguz | no. i'm not talking about make config |
| 04:04.00 | russellb | hoytbowUE: i've never seen that ... i would try make distclean and try again |
| 04:04.31 | *** join/#asterisk logicwrath (n=dfgaeefa@c-68-60-121-112.hsd1.mi.comcast.net) |
| 04:05.14 | froguz | aps.. "make config" you were right (again) |
| 04:05.37 | russellb | ;) |
| 04:06.01 | hoytbowUE | russellb: tried that thx... still same prob |
| 04:06.27 | froguz | cool, so i'm not pushed to always run * at startup... maybe a "configure init scripts" option would be nice |
| 04:06.53 | froguz | i mean, in menuselect |
| 04:08.39 | *** join/#asterisk sugardave (n=not@cpe-66-68-165-207.austin.res.rr.com) |
| 04:09.32 | *** join/#asterisk salaud (n=salaud@72.11.116.122) |
| 04:09.56 | sugardave | anybody using voicepulse connect? I can only get a SIP and IAX2 peer going on connect02 and no incoming, was wondering if anyone else is experiencing similar |
| 04:13.58 | *** join/#asterisk oadaeh (n=jason@wsip-24-234-160-51.lv.lv.cox.net) |
| 04:19.31 | ionix | I use vitelity, they are good but a bitch about international calls |
| 04:19.39 | ionix | they correctly handle inband DMTF |
| 04:19.41 | ionix | DTMF |
| 04:21.22 | sugardave | I just signed up a vitelity account tonight, was hoping to test a toll-free number, so now I wait until they can get me one (if they can) |
| 04:21.35 | *** join/#asterisk twisted[mobile] (n=twisted@pdpc/supporter/active/twisted) |
| 04:21.35 | *** mode/#asterisk [+o twisted[mobile]] by ChanServ |
| 04:21.44 | twisted[mobile] | whats up peeps |
| 04:21.57 | russellb | i wanna be vonage, you can be helping me? |
| 04:22.01 | sugardave | breakin' crap, learnin' stuff |
| 04:22.09 | file | russellb: for free??? |
| 04:22.18 | russellb | asterisk is free right? |
| 04:22.20 | twisted[mobile] | russellb, so you want to go bankrupt within 5 years and have a wide, yet unhappy customer base? |
| 04:22.21 | russellb | you can be helping me? |
| 04:22.28 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:22.45 | file | eep thunder |
| 04:22.48 | twisted[mobile] | OMG |
| 04:22.52 | russellb | file: lame |
| 04:22.53 | twisted[mobile] | look out for teh monsoon! |
| 04:22.59 | CunningPike | russellb: You'll need SVN trunk and an SPA-3000 to connect your customers to the PSTN |
| 04:23.06 | twisted[mobile] | hahaha |
| 04:23.16 | twisted[mobile] | CunningPike, that's it? no wires? |
| 04:23.23 | file | don't forget power over wireless |
| 04:23.27 | twisted[mobile] | oooh |
| 04:23.27 | russellb | CunningPike: can the SPA-3000 support up to 50,000 calls? |
| 04:23.27 | CunningPike | :D |
| 04:23.29 | twisted[mobile] | PoWIFI |
| 04:23.33 | *** join/#asterisk aaaazz (n=aaaazz@pool-70-20-24-133.bstnma.fios.verizon.net) |
| 04:23.40 | CunningPike | russellb: You'll have to overclock it, but yes |
| 04:23.48 | russellb | oh, ok, sounds easy |
| 04:23.53 | file | will it catch on fire? |
| 04:24.03 | file | can I pay more for it to catch on fire??? |
| 04:24.09 | russellb | file: sure |
| 04:24.12 | russellb | i offer a fire account |
| 04:24.19 | file | yay |
| 04:24.33 | *** part/#asterisk salaud (n=salaud@72.11.116.122) |
| 04:24.51 | russellb | mmm |
| 04:24.54 | aaaazz | anybody use the codec negotiation patch? (http://bugs.digium.com/view.php?id=4825) |
| 04:25.07 | russellb | i'm going to play the merge random stuff into the trunk game ... |
| 04:25.15 | twisted[mobile] | is that like the negotiation that occurs between the UN and Iran? |
| 04:25.36 | aaaazz | hehe |
| 04:25.43 | aaaazz | no its for negotiationg codecs between endpoints.. |
| 04:25.47 | twisted[mobile] | and Iran.... Iran's so far away.. |
| 04:25.58 | file | I reckin Iran away |
| 04:26.02 | Qwell | twisted[mobile]: "Please?" "No." "Please?!" "NO!" "Can we at least have our ball back?" "...no" |
| 04:26.08 | aaaazz | haha |
| 04:26.10 | twisted[mobile] | Qwell, hahaha |
| 04:26.25 | twisted[mobile] | "And get off my lawn!" |
| 04:26.31 | Qwell | heh |
| 04:26.37 | russellb | #asterisk-politics |
| 04:26.37 | russellb | kthx |
| 04:27.01 | file | russellb: noooooo |
| 04:27.04 | aaaazz | i want to try it but im not sure if trunk is stable enough |
| 04:27.04 | twisted[mobile] | asterisk is not a democracy... it's a theocracy |
| 04:27.19 | twisted[mobile] | depends on what kind of trunk it is |
| 04:27.20 | Qwell | who is Theo, and why does he control it? |
| 04:27.22 | file | twisted[mobile]: I thought it was a PBX... huh, shows what I know! |
| 04:27.23 | aaaazz | actually im not even sure if its in trunk |
| 04:27.23 | twisted[mobile] | wooden? aluminum? |
| 04:27.45 | EvilDeshi | how do i set up asterisk so that when someone dials up an extension it goes to voicemail if there is no answer |
| 04:27.59 | Qwell | EvilDeshi: look at the stdexten macro |
| 04:28.00 | aaaazz | EvilDeshi, see stdexten in extensions.conf |
| 04:28.08 | EvilDeshi | ok thanks |
| 04:28.23 | twisted[mobile] | or just put it up on the wall and pray to it |
| 04:28.39 | russellb | ~lart twisted[mobile] |
| 04:28.45 | twisted[mobile] | oooh |
| 04:28.46 | sugardave | ionix: is it normal IAX config with vitelity to use 'type=friend'? |
| 04:28.48 | twisted[mobile] | hot steamy AOL action |
| 04:28.52 | russellb | haven't used that one in a while |
| 04:28.54 | file | hawt |
| 04:29.13 | russellb | ~lart file |
| 04:29.20 | file | try? HA |
| 04:29.21 | twisted[mobile] | ahhaha |
| 04:29.22 | twisted[mobile] | TRIES |
| 04:29.26 | twisted[mobile] | ~lart russellb |
| 04:29.38 | russellb | ~frag twisted[mobile] |
| 04:29.42 | jbot | ACTION readies the nuke launcher and fires some rounds at twisted[mobile] |
| 04:29.43 | twisted[mobile] | it should say moran |
| 04:29.54 | twisted[mobile] | prepare for fire-ass! |
| 04:30.16 | aaaazz | ahh here it is http://svn.digium.com/svn/asterisk/team/oej/codecnegotiation |
| 04:32.20 | twisted[mobile] | HANGUP!? |
| 04:32.26 | twisted[mobile] | someone is using dialup?! |
| 04:33.53 | file | ATH++ |
| 04:34.12 | jart | hi file :) |
| 04:34.27 | twisted[mobile] | jart |
| 04:34.27 | file | hola |
| 04:34.29 | twisted[mobile] | as in lawn jart? |
| 04:34.39 | CunningPike | ~lart jart |
| 04:34.43 | CunningPike | Heh heh |
| 04:35.08 | jart | jart = Justin Alexander Roberts Tunney |
| 04:35.16 | twisted[mobile] | and your ssn? |
| 04:35.20 | twisted[mobile] | and DOB? |
| 04:35.37 | twisted[mobile] | j/k, don't post those |
| 04:35.38 | jart | 313-37-666, 12/18/84 |
| 04:35.43 | twisted[mobile] | hahah |
| 04:36.18 | *** join/#asterisk junbug (i=junya@c-67-191-62-53.hsd1.fl.comcast.net) |
| 04:40.11 | ionix | Hi, I am using DEADAGI to process a script but when a user hangup while I am playing a file, asterisk kills the script |
| 04:40.34 | ionix | I would like to keep the script alive as I am doing the hangup check via the script and I want the script to cleans itself up before dieing |
| 04:41.12 | twisted[mobile] | deadagi needs to be called ON hangup then |
| 04:41.14 | twisted[mobile] | using the h extension |
| 04:42.05 | ionix | hmm problem is that I am using a deadagi on an outgoing script |
| 04:43.10 | twisted[mobile] | i got an idea |
| 04:43.13 | twisted[mobile] | set a variable in the AGI on start |
| 04:43.19 | twisted[mobile] | say... AGICOMPLETE=0 |
| 04:43.28 | twisted[mobile] | if the AGI completes, set AGICOMPLETE=1 |
| 04:43.42 | twisted[mobile] | if it's hung up on, run another agi from the h extension IF AGICOMPLETE=0 |
| 04:43.45 | twisted[mobile] | that script could clean up |
| 04:44.16 | ionix | I see however how can I run a script on a h extension if I am calling the script from the outgoing call? |
| 04:44.37 | twisted[mobile] | because when the caller hangs up, it will still go to the h extension |
| 04:44.39 | twisted[mobile] | er |
| 04:44.40 | twisted[mobile] | callee |
| 04:44.49 | twisted[mobile] | if the h extension exists |
| 04:45.03 | twisted[mobile] | unless you're calling AGI directly |
| 04:45.25 | *** join/#asterisk linlin (n=linlin@c-67-184-230-25.hsd1.il.comcast.net) |
| 04:45.29 | twisted[mobile] | if you are, i'd modify that behaviour to call DIAL(Local/someroutine@somecontext/n) |
| 04:45.37 | twisted[mobile] | and that someroutine would run the agi for you |
| 04:45.45 | twisted[mobile] | that way you can be sure it'll fall out to the h extension |
| 04:45.56 | ionix | hmm, not a bad idea, I can call an extension on extensions.conf instead of the script directly and have it's it's agi called from the extension then add h to do the cleanup |
| 04:46.02 | twisted[mobile] | yep |
| 04:46.04 | ionix | let me try, this might just work |
| 04:46.07 | twisted[mobile] | ;) |
| 04:48.11 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
| 04:49.19 | *** join/#asterisk budmang (n=budman@12.206.134.162) |
| 04:49.44 | twisted[mobile] | see topic |
| 04:50.34 | *** join/#asterisk Corydon76-home (i=orange@pdpc/supporter/sustaining/Corydon76-home) |
| 04:50.34 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 04:52.29 | *** join/#asterisk javar (n=javar@200.118.174.253) |
| 04:54.48 | *** part/#asterisk javar (n=javar@200.118.174.253) |
| 04:54.54 | ionix | haha it works |
| 04:54.55 | ionix | thx dude |
| 04:55.52 | twisted[mobile] | np :) |
| 04:55.56 | *** join/#asterisk Floodbar (n=Flood@ip68-12-150-145.ok.ok.cox.net) |
| 04:56.24 | twisted[mobile] | if you're gonna make $ from it feel free to donate back to me :P |
| 04:56.31 | twisted[mobile] | ~twisted |
| 04:56.32 | jbot | somebody said twisted was twisted@indigent-networks.com, but you can paypal him at toastido@toastido.net |
| 04:56.48 | twisted[mobile] | </shameless plug> |
| 04:57.02 | Floodbar | what are you making $ from |
| 04:57.34 | Corydon76-home | Oooo, and what can I buy? |
| 04:57.34 | Floodbar | haha |
| 04:57.34 | twisted[mobile] | lol Corydon |
| 04:58.26 | twisted[mobile] | i'm broke so i'm shamelessly asking for donations in return for advice/help |
| 04:58.41 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 04:58.41 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 04:58.45 | Floodbar | so is this a development platform or can you ask general questions |
| 04:58.55 | Floodbar | I hear ya |
| 04:58.58 | twisted[mobile] | gen |
| 04:59.03 | Floodbar | cool |
| 04:59.20 | Floodbar | do you work for digium |
| 04:59.27 | twisted[mobile] | nope |
| 04:59.28 | Corydon76-home | russellb_: would you believe that twisted is selling himself on the channel? ;-) |
| 04:59.37 | ionix | hehe, I won't make money, in fact, it will cost me carrier fees and I won't make money |
| 04:59.45 | ionix | so I can give you part of the debt then :) |
| 04:59.53 | russellb_ | Corydon76-home: that's crazy talk |
| 05:00.10 | twisted[mobile] | ionix, hahah.. i'll pass :P |
| 05:00.17 | Floodbar | has anyone ever used ldapget or is there a better way to query ldap |
| 05:00.54 | twisted[mobile] | eep |
| 05:02.21 | russellb | meep meep |
| 05:02.27 | Floodbar | wow is ldap a conversation killer |
| 05:02.58 | russellb | Floodbar: yep, THANKS |
| 05:03.22 | Floodbar | sorry dude |
| 05:03.30 | russellb | just kidding :) |
| 05:03.50 | russellb | i have never used ldap |
| 05:03.52 | Floodbar | ahh its been something I have been working on for a while but its just not working |
| 05:03.56 | russellb | i just ... write code ... or something |
| 05:04.16 | Floodbar | I have tried the perl::Ldap, ldapget and another ldap deal |
| 05:04.24 | Floodbar | cool |
| 05:04.44 | Floodbar | I figure out how to use the code you write |
| 05:04.54 | russellb | i like my end of the deal |
| 05:04.57 | Floodbar | or try too |
| 05:05.03 | Floodbar | haha |
| 05:05.11 | russellb | if it doesn't work, oh well |
| 05:05.18 | Floodbar | true |
| 05:05.37 | russellb | just kidding, i actually work pretty hard at trying to make things not suck |
| 05:05.47 | Floodbar | I think asterisk works well |
| 05:05.54 | russellb | we have a passionate group of developers ... |
| 05:06.04 | Floodbar | I work for a school system and use it for our call center |
| 05:06.15 | file | yay passionate |
| 05:06.21 | Floodbar | pissed off the cisco guys |
| 05:06.29 | russellb | Floodbar: haha, awesome! |
| 05:06.30 | file | we all like eachother and enjoy what we do! |
| 05:06.47 | Floodbar | are most of the ops programmers |
| 05:06.53 | *** join/#asterisk topping (n=topping@207.47.6.192.static.nextweb.net) |
| 05:06.55 | russellb | file: i ... don't like you |
| 05:07.00 | file | russellb: zomg |
| 05:07.09 | denon | yeah . .come to think of it, I'm not real fond of file either |
| 05:07.15 | file | russellb: you're going into... A BUSH! |
| 05:07.17 | russellb | lol |
| 05:07.22 | Floodbar | haha |
| 05:07.30 | file | yeah nobody likes me |
| 05:07.35 | Floodbar | file just speaks up for a moment and gets hammered |
| 05:07.37 | russellb | file: been there, done tht |
| 05:07.39 | denon | once the muffins run out .. |
| 05:07.47 | denon | file just gets tossed aside like a dish towel |
| 05:07.59 | file | or a Gentoo install. |
| 05:08.14 | russellb | ooh, distro trolling |
| 05:08.24 | twisted[mobile] | OMFG |
| 05:08.26 | denon | redhat ownz you! |
| 05:08.28 | twisted[mobile] | GENTOO > * |
| 05:08.36 | file | I shouldn't troll on Gentoo... |
| 05:08.47 | file | CentOS, now that I can troll on |
| 05:08.51 | russellb | but you already did |
| 05:08.53 | twisted[mobile] | ~lart file |
| 05:09.18 | russellb | :/ |
| 05:09.20 | twisted[mobile] | hjahaha |
| 05:09.26 | file | jbot: file? |
| 05:09.27 | jbot | well, file is Joshua Colp, jcolp@digium.com, and is the best looking software developer at Digium |
| 05:09.29 | twisted[mobile] | jbot doesnt like u |
| 05:09.36 | russellb | haha, lame |
| 05:09.37 | twisted[mobile] | rofl |
| 05:09.42 | russellb | ~russellb |
| 05:09.43 | jbot | well, russellb is aka drumkilla |
| 05:10.04 | russellb | jbot: russellb is also much cooler than file |
| 05:10.08 | jbot | okay, russellb |
| 05:10.10 | twisted[mobile] | back to idle i go |
| 05:10.14 | Floodbar | drumkilla is the one who packages the files up? correct? |
| 05:10.14 | file | :( |
| 05:10.32 | file | a script does that, he just signs them |
| 05:10.34 | Floodbar | or sets the stable releases or something like that |
| 05:10.36 | Floodbar | haha |
| 05:10.37 | russellb | Floodbar: as of lately, it has been me and Kevin Fleming |
| 05:10.44 | Floodbar | well cool |
| 05:10.52 | file | and technically the script is running on the computer... so the computer does it |
| 05:10.52 | Floodbar | you guys are doing a hell of a job |
| 05:10.57 | russellb | jbot: russellb is Russell Bryant <russell@digium.com>, aka drumkilla |
| 05:10.58 | jbot | ...but russellb is already something else... |
| 05:11.01 | russellb | Floodbar: thanks :) |
| 05:11.07 | russellb | jbot: forget russellb |
| 05:11.29 | twisted[mobile] | hahaha |
| 05:11.31 | russellb | jbot: no, russellb is Russell Bryant <russell@digium.com>, aka drumkilla |
| 05:11.32 | jbot | russellb: okay |
| 05:11.43 | Floodbar | so far I have been able to get the phone system to do all kinds of things that the cisco cannot do easily or not at all |
| 05:11.43 | russellb | stupid bot |
| 05:11.47 | twisted[mobile] | jbot: russellb is also 1337 |
| 05:11.49 | jbot | okay, twisted[mobile] |
| 05:11.54 | file | Floodbar: exxxxxcellent |
| 05:11.54 | Corydon76-home | jbot: corydon76? |
| 05:11.56 | twisted[mobile] | ~russelb |
| 05:11.58 | twisted[mobile] | ~russellb |
| 05:12.01 | jbot | methinks russellb is Russell Bryant <russell@digium.com>, aka drumkilla, or 1337 |
| 05:12.11 | russellb | ~drumkilla |
| 05:12.13 | jbot | drumkilla is, like, Russell Bryant, the Asterisk release branch maintainer <russelb@clemson.edu> |
| 05:12.23 | russellb | it used to sat "ph33r him" |
| 05:12.25 | Floodbar | today I got queuemetrics 1.2 installed with chanspy working |
| 05:12.39 | CunningPike | ~cunningpike |
| 05:12.42 | jbot | somebody said cunningpike was not easily offended |
| 05:12.42 | twisted[mobile] | jbot, drumkilla is also someone who should be ph33r3d |
| 05:12.44 | jbot | okay, twisted[mobile] |
| 05:12.48 | Floodbar | its been done before but I was excited when I got it working |
| 05:12.52 | russellb | Floodbar: well, we stay up late many nights to make it possible :-p |
| 05:12.57 | CunningPike | Weird |
| 05:13.06 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 05:13.07 | Floodbar | I stayed up many late nights learning it |
| 05:13.13 | russellb | awesome :) |
| 05:13.20 | Floodbar | got dcap certified last year |
| 05:13.28 | file | dCAPitated? |
| 05:13.30 | russellb | i'm not dcap certified yet :( |
| 05:13.35 | twisted[mobile] | me either |
| 05:13.36 | Floodbar | hahah |
| 05:14.09 | bugz | why do asterisk -rx 'sip show peers' commands truncating output? |
| 05:14.09 | Floodbar | you all probably way smarter than that |
| 05:14.10 | russellb | file: think they'd let us go take the class sometime? |
| 05:14.13 | twisted[mobile] | if i take the dCap, i want a baseball cap with the letter 'd' on it |
| 05:14.16 | bugz | er |
| 05:14.21 | file | russellb: us? ha |
| 05:14.22 | bugz | why dont they show all the peers |
| 05:14.27 | bugz | iax2 show peers |
| 05:14.28 | russellb | bugz: what version |
| 05:14.34 | Floodbar | I like the wooden cert |
| 05:14.38 | bugz | 1.2.9.1-1.2.10 |
| 05:14.44 | bugz | even some older ones |
| 05:14.50 | bugz | some cases, it shows nothing at all |
| 05:14.54 | russellb | 1.2.10 should work, i think |
| 05:14.59 | Corydon76-home | jbot: Corydon76 is the master of asstricks |
| 05:15.00 | jbot | Corydon76-home: okay |
| 05:15.01 | russellb | in any case, should be fixed in the latest code |
| 05:15.05 | russellb | maybe just grab the 1.2 branch |
| 05:15.23 | Corydon76-home | jbot: Corydon76 is also an Asterisk developer |
| 05:15.25 | jbot | okay, Corydon76-home |
| 05:15.28 | bugz | hmm |
| 05:15.30 | bugz | ok thanks |
| 05:15.30 | Floodbar | I have a queue question if you don't mind taking a crack at it |
| 05:15.50 | russellb | jbot: Corydon76 is also a stalker of file |
| 05:15.51 | jbot | russellb: okay |
| 05:16.01 | Corydon76-home | ~corydon76 |
| 05:16.02 | jbot | corydon76 is, like, the master of asstricks, or an Asterisk developer, or a stalker of file |
| 05:16.06 | file | Your question will be answered in approximately ... 20 years ... |
| 05:16.08 | bugz | russellb: do you know much about the wcte24xxp cards? |
| 05:16.11 | twisted[mobile] | lsof file |
| 05:16.17 | bugz | i bought 3 and they are a little problematic... |
| 05:16.27 | Floodbar | when I have no device or agent logged into the queue there is no music on hold for the first caller that is waiting, however the second caller does |
| 05:16.29 | russellb | bugz: not really, you should contact technical support - support@digium.com |
| 05:16.40 | bugz | well they are already rma'ed... |
| 05:16.51 | bugz | bad batch of silicon i guess. |
| 05:16.54 | Floodbar | when the first caller hangs up the music on hold stops for the second person as now they are the first person in the queue |
| 05:17.00 | file | WCTE24XXP? the TDM2400? |
| 05:17.01 | twisted[mobile] | whoa |
| 05:17.08 | twisted[mobile] | Floodbar, sounds like a bug? |
| 05:17.08 | bugz | yes |
| 05:17.17 | russellb | Floodbar: that's odd. sounds like a bug ... |
| 05:17.23 | bugz | file: i put 2 in a box, with a quad port pri card |
| 05:17.26 | twisted[mobile] | russellb, GMTA biatch |
| 05:17.29 | Floodbar | if someone logs into the queue music on hold is restored for the first person |
| 05:17.43 | russellb | yeah, that's a bug, i guess |
| 05:17.43 | bugz | the channels on the 24 port started going out one by one |
| 05:17.47 | Floodbar | it happens on my 1.2.5 production and on my 1.2.10 development |
| 05:17.52 | bugz | eventually i had to comment out a whole card |
| 05:17.54 | bugz | ;P |
| 05:18.07 | Floodbar | whats weird is that queue anouncements still work |
| 05:18.17 | russellb | Floodbar: feel free to open a report on bugs.digium.com |
| 05:18.20 | file | I believe 4 were tested in a box, but just the TDM2400s... didn't throw a PRI one in... |
| 05:18.28 | russellb | and file will fix it |
| 05:18.35 | Floodbar | haha |
| 05:18.36 | file | what am I fixing now? |
| 05:18.38 | Floodbar | cool |
| 05:18.49 | bugz | asterisk |
| 05:18.50 | russellb | file: queue bug |
| 05:18.58 | Floodbar | its using a pri |
| 05:19.10 | russellb | music on hold sillyness |
| 05:19.13 | russellb | that's what |
| 05:19.20 | file | ah, just be sure to provide all the info and settings so I can lab it up and see... then I can try to fix it |
| 05:19.31 | russellb | yay |
| 05:19.31 | Floodbar | I had someone else test it with analog ports but I don't know if they replicated what I am experiencing |
| 05:19.38 | russellb | i think i'm going to break the trunk before i go to sleep |
| 05:19.41 | Floodbar | however they did not experience the problem |
| 05:19.42 | file | erm, I don't have any zaptel stuff |
| 05:19.58 | Floodbar | so put it in bug.digium.com |
| 05:20.03 | russellb | the problem shouldn't be channel specific ... |
| 05:20.07 | russellb | but who knows |
| 05:20.12 | Floodbar | I wouldn't think so |
| 05:20.26 | Floodbar | on my development I use a ztdummy driver |
| 05:20.42 | file | but uh, just provide the settings and info so I can lab it up exactly without too much trouble... I dislike having to figure out exactly how people have things |
| 05:20.56 | Floodbar | okay |
| 05:21.08 | file | and all your base are belong to me |
| 05:21.11 | Floodbar | on the bugs page is there a place to leave my email addy |
| 05:21.12 | Floodbar | hah |
| 05:21.22 | file | you'll need to sign up for an account |
| 05:21.25 | russellb | Floodbar: yeah, you have to create an account |
| 05:21.29 | Floodbar | that way if you have questions I would be willing to give you as much info as needed |
| 05:21.37 | Floodbar | cool |
| 05:21.46 | russellb | it should email you any time someone makes a note on your bug report |
| 05:21.47 | Floodbar | I will do that sometime tomorrow |
| 05:21.51 | Floodbar | cool |
| 05:22.10 | russellb | i wish my internet connection didn't suck SO MUCH |
| 05:22.20 | file | russellb: that's why you need to move to... |
| 05:22.24 | file | disney land! |
| 05:22.35 | Floodbar | brb |
| 05:23.01 | russellb | file: but but but |
| 05:23.07 | russellb | i'm moving in less than 2 weeks :( |
| 05:23.15 | file | ah, yes |
| 05:23.16 | file | I forgot |
| 05:23.38 | denon | russellb: use cdma and ssh to digium |
| 05:23.41 | russellb | i hate moving |
| 05:23.50 | denon | no need for a fast local connection |
| 05:23.51 | file | ssh? that's for silly people |
| 05:24.05 | denon | file: oh, sorry -- you use PCAnywhere, forgot .. |
| 05:24.15 | file | ! |
| 05:24.38 | file | Remote Desktop, mmmkthx |
| 05:24.53 | denon | oh? I thought you were all anti windows |
| 05:25.37 | file | geez, it's amazing how much stuff is packed onto an X400M module |
| 05:26.00 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 05:26.00 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 05:27.06 | *** join/#asterisk P-NuT (n=P-NuT@CPE-60-227-93-75.nsw.bigpond.net.au) |
| 05:27.07 | file | russellb: ... hi |
| 05:28.12 | russellb | hi. |
| 05:30.12 | Floodbar | I am a big fan of remote access any way you can get into it |
| 05:30.55 | Floodbar | russell do you live out in the boonies |
| 05:31.45 | russellb | nope |
| 05:31.58 | Floodbar | just have crappy internet? |
| 05:31.59 | russellb | not at all |
| 05:32.06 | russellb | it has sucked lately ... |
| 05:32.09 | russellb | it's BellSouth |
| 05:33.43 | Floodbar | who would know anything about ldap queries and when would they usually be on |
| 05:35.52 | EvilDeshi | is there a way when during an active conversation to tranfer the caller to a different extension? |
| 05:37.10 | Floodbar | evildeshi what kind of phone |
| 05:37.37 | EvilDeshi | xlite |
| 05:37.44 | EvilDeshi | is the phone im using |
| 05:37.47 | EvilDeshi | soft phone |
| 05:37.56 | Floodbar | there is a transfer button in the interface |
| 05:37.59 | EvilDeshi | or does all the transfering require the zt kernel mod |
| 05:38.13 | EvilDeshi | i have the free version |
| 05:38.18 | Floodbar | I do too |
| 05:38.28 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 05:38.31 | EvilDeshi | it says i need teh pro version to use xfer |
| 05:38.46 | Floodbar | one sec |
| 05:39.03 | Floodbar | I believe its * or # to initiate the transfer |
| 05:39.14 | Floodbar | I know the # completes it |
| 05:39.46 | Floodbar | so it would be something like *XXXX# |
| 05:40.13 | Floodbar | give it a whirl |
| 05:40.19 | EvilDeshi | im trying it out |
| 05:40.23 | Floodbar | hah |
| 05:40.34 | bugz | anyone seen this? http://demo.scopserv.com |
| 05:40.44 | bugz | looks like a good start. wish it was open source. |
| 05:41.34 | bugz | damn.. i have to create something |
| 05:41.40 | bugz | i dont know where to start |
| 05:41.48 | bugz | id like to use perl |
| 05:41.58 | bugz | just to be mean >=] |
| 05:42.03 | EvilDeshi | no Floodbar its not working that way |
| 05:42.07 | hoytbowUE | So... Can anyone tell me how to get the H264 Codec into 1.2.10 ?? |
| 05:42.19 | hoytbowUE | plz |
| 05:42.51 | P-NuT | hey guys, how do you set incoming caller ID for say 02 1234 5678 to be "John Smith" |
| 05:42.52 | Floodbar | one sec evil I will check it out |
| 05:43.04 | Floodbar | however I will be disconnected for a moment |
| 05:46.30 | hads | P-NuT: You can use the astdb to store your names for callerid and then do a lookup on that when calls come in. |
| 05:48.43 | *** join/#asterisk adorah (n=Administ@84.94.208.77.cable.012.net.il) |
| 05:51.55 | *** join/#asterisk Floodbar (n=Flood@ip68-12-150-145.ok.ok.cox.net) |
| 05:53.39 | Floodbar | evildeshi are you still there |
| 05:54.45 | EvilDeshi | yup |
| 05:55.16 | Floodbar | my vpn is not working for whatever reason to test it but from a google search you do it by pressing pound then it will ask you for an extension then you press pound to send the call |
| 05:55.43 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 05:56.18 | hads | If you are talking about using Asterisk to transfer it's setup in features.conf |
| 05:56.54 | bugz | hey dlynes_laptop |
| 05:57.01 | dlynes_laptop | good morning, bugz |
| 05:58.25 | Floodbar | well good night guys work comes early in the morning |
| 05:58.25 | bugz | these guys are insane: http://www.churchofwifi.org/default.asp?PageLink=Project_Display.asp?PID=84 |
| 05:58.38 | bugz | defcon really broke the mold this year |
| 05:58.42 | bugz | 2 guys at MIT built a safe cracking robot |
| 06:01.41 | *** join/#asterisk parag_ast (n=root@dxb-b18766.alshamil.net.ae) |
| 06:02.07 | parag_ast | can anybody see http://pastebin.ca/121827 and let me know the exect reason why Zap/4-1 didn't hangup... |
| 06:02.19 | *** join/#asterisk daysmen3 (n=primus@host86-141-242-43.range86-141.btcentralplus.com) |
| 06:02.45 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 06:03.15 | Shaun2222 | when sending a reload to asterisk does it kill any calls or sip/iax connections? |
| 06:03.25 | parag_ast | No shaun2222 |
| 06:03.26 | bugz | Shaun2222: no |
| 06:03.42 | Shaun2222 | any of you guys use mysql rather than flat files? |
| 06:03.48 | parag_ast | bugz:, can u see http://pastebin.ca/121827 and let me know the exect reason why Zap/4-1 didn't hangup... |
| 06:03.55 | bugz | parag_ast: looking |
| 06:03.59 | parag_ast | thanks |
| 06:06.41 | bugz | Aug 7 11:21:54 WARNING[15594] chan_zap.c: Ring/Off-hook in strange state 6 on channel 4 |
| 06:06.47 | bugz | this looks interesting |
| 06:06.53 | parag_ast | thats what |
| 06:07.13 | parag_ast | when i do show channels then i get this |
| 06:07.30 | parag_ast | asterisk1*CLI> show channels |
| 06:07.30 | parag_ast | Channel Location State Application(Data) |
| 06:07.30 | parag_ast | SIP/17-083b9d60 (None) Up Bridged Call(Zap/4-1) |
| 06:07.30 | parag_ast | Zap/4-1 s@macro-dial:10 Up Dial(SIP/10&SIP/11&SIP/13&SIP/2 active channels |
| 06:07.30 | parag_ast | 1 active call |
| 06:07.49 | parag_ast | actually its not a active call |
| 06:07.58 | parag_ast | its been already hanged up |
| 06:10.52 | hads | Is that AMP cruft in the dial plan? |
| 06:11.48 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 06:13.25 | dlynes_laptop | hads: he's using A@H |
| 06:13.48 | hads | Ah. |
| 06:13.50 | parag_ast | dlynes_laptop: http://pastebin.ca/121829 |
| 06:13.57 | bugz | looks more like Asterisk @ Neptune |
| 06:14.04 | dlynes_laptop | bugz: huh? |
| 06:14.11 | bugz | to me anyway heh... |
| 06:14.33 | parag_ast | Any comments on these strange problem |
| 06:14.40 | parag_ast | its day by day stuff |
| 06:14.48 | hads | Probably disconnect supervision. |
| 06:14.55 | hads | or lack of |
| 06:15.00 | parag_ast | daily one of my channel is getting engaged due to this |
| 06:15.29 | hads | parag_ast: You'll get more help in #freepbx |
| 06:15.32 | *** join/#asterisk kumamoto (n=kumamoto@24-178-2-212.dhcp.ftwo.tx.charter.com) |
| 06:15.56 | parag_ast | I tried , seems nobody knows |
| 06:22.40 | Maroder | parag_ast try with congestion |
| 06:23.14 | parag_ast | ?? |
| 06:23.46 | Maroder | show me your extension.conf where is extension from zap |
| 06:23.58 | parag_ast | okk i m uploading |
| 06:25.45 | EvilDeshi | i cant seem to get automon and attented transfering to work with features.conf |
| 06:28.30 | *** join/#asterisk Mw3 (i=mw3@national.t-error.hu) |
| 06:29.03 | parag_ast | Maroder, http://pastebin.ca/121850 my extention.conf.. |
| 06:30.35 | Maroder | wow second to turno on my lap and switch to X :) |
| 06:32.42 | Shaun2222 | Aug 7 22:28:40 WARNING[8090]: chan_iax2.c:9676 load_module: Unable to open IAX timing interface: No such file or directory |
| 06:32.50 | Shaun2222 | what am i missing? |
| 06:32.56 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 06:35.43 | *** join/#asterisk SaTLaN32 (n=satlan32@212.150.142.211) |
| 06:35.50 | SaTLaN32 | hi all |
| 06:36.08 | SaTLaN32 | need some help with Rhino R4FXO card installtion |
| 06:36.24 | SaTLaN32 | [root@asterisk1 sbin]# ztcfg |
| 06:36.24 | SaTLaN32 | Notice: Configuration file is /etc/zaptel.conf |
| 06:36.24 | SaTLaN32 | line 0: Unable to open master device '/dev/zap/ctl' |
| 06:36.24 | SaTLaN32 | 1 error(s) detected |
| 06:36.24 | SaTLaN32 | [root@asterisk1 sbin]# |
| 06:39.57 | Maroder | parag_ast try with congestion replace hagup |
| 06:39.58 | Maroder | ? |
| 06:40.23 | x86 | SaTLaN32: ls -la /dev/zap/ctl |
| 06:40.45 | SaTLaN32 | ls: /dev/zap/ctl: No such file or directory |
| 06:40.59 | x86 | there you go :) |
| 06:40.59 | parag_ast | congestion gets hanged |
| 06:41.07 | x86 | that's why it can't open it, it doesnt exist :) |
| 06:41.15 | x86 | SaTLaN32: lsmod | grep zaptel |
| 06:41.26 | SaTLaN32 | ok |
| 06:41.39 | SaTLaN32 | did that |
| 06:41.46 | x86 | anything? |
| 06:41.48 | bugz | parag_ast: what does this machine do? |
| 06:41.52 | SaTLaN32 | nop same |
| 06:41.59 | *** join/#asterisk Assid (i=assid@203.115.83.214) |
| 06:42.01 | bugz | parag_ast: what kind of operation i mean? if you dont mind me asking |
| 06:42.04 | x86 | SaTLaN32: did it show any output was my question :) |
| 06:42.27 | Mike | someone can help me with trunking iax2 channels? |
| 06:42.29 | bugz | its an advanced extensions layout with good examples of syntax semantics and heavy usage of macros and agi |
| 06:42.33 | SaTLaN32 | for the lsmod command? |
| 06:42.45 | EvilDeshi | in this howto im reading abotu call transfering says i need to add T or t to the Dial command to be able to allow that caller transfering abilities what and or how do i add the T or t option? |
| 06:42.48 | bugz | but the question is... wtf is it for? ;) |
| 06:43.21 | x86 | SaTLaN32: right |
| 06:43.38 | SaTLaN32 | [root@asterisk1 /]# lsmod | grep zaptel |
| 06:43.38 | SaTLaN32 | [root@asterisk1 /]# |
| 06:43.44 | SaTLaN32 | no output |
| 06:43.57 | x86 | then your zaptel driver is not loaded into the kernel |
| 06:43.59 | x86 | modprobe zaptel |
| 06:44.12 | SaTLaN32 | FATAL: Module zaptel not found. |
| 06:44.15 | SaTLaN32 | mmm |
| 06:44.25 | x86 | you need to compile the drivers then ;) |
| 06:44.57 | SaTLaN32 | i have restarted the server after installing the card, and changing the configuration files as they write in the support book, and then i did make && make install |
| 06:45.04 | SaTLaN32 | everything went well. |
| 06:45.28 | x86 | make && make install for what? |
| 06:45.33 | hads | heh |
| 06:45.34 | x86 | asterisk? zaptel? |
| 06:45.48 | SaTLaN32 | after changing the makefile for zaptel |
| 06:46.01 | SaTLaN32 | i added the r4fxo card. |
| 06:46.07 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 06:46.33 | SaTLaN32 | MODULES:=zaptel r4fxo tor2 torisa wcusb wcfxo wctdm wctdm24xxp \ |
| 06:46.52 | x86 | it didnt install properly then |
| 06:47.03 | *** join/#asterisk NoNeo (n=Artur@193.24.24.10) |
| 06:47.03 | SaTLaN32 | mm what should i try.? |
| 06:47.04 | x86 | ls -la /usr/src/linux |
| 06:47.20 | SaTLaN32 | lrwxrwxrwx 1 root root 33 Apr 30 15:44 /usr/src/linux -> /usr/src/kernels/2.6.9-34.EL-i686 |
| 06:47.26 | x86 | uname -a |
| 06:47.33 | hads | find /lib/modules/`uname -r` -name zap* |
| 06:47.46 | SaTLaN32 | Linux asterisk1.local 2.6.9-34.0.2.EL #1 Fri Jul 7 19:24:57 CDT 2006 i686 i686 i386 GNU/Linux |
| 06:48.03 | SaTLaN32 | for find /lib/modules/`uname -r` -name zap* i have no output |
| 06:48.19 | x86 | looks like you are running a different kernel version than what you compiled the module for |
| 06:48.39 | SaTLaN32 | how come? |
| 06:49.04 | x86 | your kernel tree is for 2.6.9-34.EL, you're running 2.6.9-34.0.2.EL |
| 06:49.11 | SaTLaN32 | mmm |
| 06:49.33 | x86 | find /lib/modules -name '*zap*' |
| 06:49.46 | x86 | you may be able to get away with simply moving them to the right location |
| 06:49.54 | SaTLaN32 | /lib/modules/2.6.9-34.EL/extra/zaptel.ko |
| 06:50.02 | SaTLaN32 | how? |
| 06:50.57 | x86 | mv /lib/modules/2.6.9-34.EL/extra/zaptel.ko /lib/modules/`uname -r`/extra/ |
| 06:51.28 | SaTLaN32 | [root@asterisk1 /]# mv /lib/modules/2.6.9-34.EL/extra/zaptel.ko /lib/modules/`uname -r`/extra/ |
| 06:51.28 | SaTLaN32 | mv: cannot move `/lib/modules/2.6.9-34.EL/extra/zaptel.ko' to `/lib/modules/2.6.9-34.0.2.EL/extra/zaptel.ko': No such file or directory |
| 06:51.39 | x86 | ls /lib/modules |
| 06:51.49 | x86 | do you see 2.6.9-34.0.2.EL listed? |
| 06:51.53 | SaTLaN32 | 2.6.9-34.0.2.EL 2.6.9-34.0.2.ELsmp 2.6.9-34.EL 2.6.9-34.ELsmp kabi-4.0-0 |
| 06:52.25 | x86 | mkdir /lib/modules/`uname -r`/extra |
| 06:52.28 | x86 | mv /lib/modules/2.6.9-34.EL/extra/zaptel.ko /lib/modules/`uname -r`/extra/ |
| 06:52.34 | x86 | depmod -ae |
| 06:52.37 | x86 | modprobe zaptel |
| 06:52.41 | Shaun2222 | SaTLaN32: yum install kernel-devel |
| 06:52.51 | *** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 06:53.09 | FuriousGeorge | hey all, anyone in here using asterlink |
| 06:53.11 | SaTLaN32 | [root@asterisk1 /]# depmod -ae |
| 06:53.11 | SaTLaN32 | [root@asterisk1 /]# modprobe zaptel |
| 06:53.12 | SaTLaN32 | [root@asterisk1 /]# |
| 06:53.20 | SaTLaN32 | this is better? |
| 06:53.23 | x86 | indeed |
| 06:53.26 | x86 | lsmod | grep zaptel |
| 06:53.30 | x86 | see it now? |
| 06:53.37 | SaTLaN32 | [root@asterisk1 /]# lsmod | grep zaptel |
| 06:53.38 | SaTLaN32 | zaptel 206852 0 |
| 06:53.38 | SaTLaN32 | crc_ccitt 2113 1 zaptel |
| 06:53.38 | SaTLaN32 | [root@asterisk1 /]# |
| 06:53.39 | x86 | ls -la /dev/zap* |
| 06:53.55 | SaTLaN32 | ok... i see ctl now |
| 06:54.01 | SaTLaN32 | drwxr-xr-x 2 root root 120 Aug 8 09:53 . |
| 06:54.01 | SaTLaN32 | drwxr-xr-x 9 root root 5240 Aug 8 09:53 .. |
| 06:54.01 | SaTLaN32 | crw-rw---- 1 asterisk asterisk 196, 254 Aug 8 09:53 channel |
| 06:54.01 | SaTLaN32 | crw-rw---- 1 asterisk asterisk 196, 0 Aug 8 09:53 ctl |
| 06:54.02 | SaTLaN32 | crw-rw---- 1 asterisk asterisk 196, 255 Aug 8 09:53 pseudo |
| 06:54.02 | SaTLaN32 | crw-rw---- 1 asterisk asterisk 196, 253 Aug 8 09:53 timer |
| 06:54.05 | x86 | whoa |
| 06:54.08 | x86 | too much paste |
| 06:54.09 | SaTLaN32 | :):) |
| 06:54.13 | SaTLaN32 | lol |
| 06:54.15 | x86 | so now you should be good |
| 06:54.24 | SaTLaN32 | now ztcfg? |
| 06:54.27 | x86 | yup |
| 06:54.38 | SaTLaN32 | ZT_SPANCONFIG failed on span 1: No such device or address (6) |
| 06:54.53 | x86 | that's beyond me ;) |
| 06:54.58 | SaTLaN32 | ha ha |
| 06:55.03 | SaTLaN32 | ok thanks.. |
| 06:55.12 | x86 | yep |
| 06:55.12 | SaTLaN32 | great help man |
| 06:55.16 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 06:55.17 | *** join/#asterisk af_ (n=af@ip-192-212.sn2.eutelia.it) |
| 06:56.13 | FuriousGeorge | SaTLaN32: i think thats a udev problem |
| 06:56.37 | FuriousGeorge | when you uncompress zaptel check readme.UDEV file (or something like that) |
| 06:57.20 | SaTLaN32 | hi |
| 06:57.43 | x86 | well, hello ;) |
| 07:01.16 | RoyK | . |
| 07:01.20 | x86 | ! |
| 07:01.50 | RoyK | ¡ |
| 07:02.13 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 07:03.17 | x86 | ? |
| 07:04.26 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 07:04.58 | FuriousGeorge | so i was thinking of switching from iax to sip for my voip minutes then i realized this would totally scre up every aspect of my dialplan, due to the stupid syntax difference between sip and iax2 dialing |
| 07:06.22 | *** join/#asterisk shaZwaz (n=chatzill@203.81.222.122) |
| 07:06.38 | shaZwaz | hi |
| 07:06.57 | shaZwaz | can anyone help me with Adit Channel Bank no ring issue |
| 07:07.27 | Shaun2222 | FuriousGeorge: whats so diffrent? |
| 07:07.30 | FuriousGeorge | exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@asterlink) is about the stupidest least intuitive syntax anyone could have come up with |
| 07:08.09 | FuriousGeorge | Shaun2222: vs dial(iax2/username@asterlink/${EXTEN}) |
| 07:08.36 | FuriousGeorge | the latter i can actually use a (gasp) variable for my provider |
| 07:08.56 | x86 | FuriousGeorge: you could use a variable in either |
| 07:08.57 | Shaun2222 | i see |
| 07:08.59 | FuriousGeorge | with the sip syntax i cant b/c what happens when i want to use ${EXTEN:1} |
| 07:09.14 | x86 | nothing different than normal behaviour? |
| 07:09.57 | x86 | Dial(SIP/${EXTEN}@${PROVIDER}) |
| 07:10.22 | x86 | Dial(IAX2/${USER}@${PROVIDER}/${EXTEN}) |
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| 07:10.29 | FuriousGeorge | x86: so i just define SIPOUT=sip/${EXTEN}@asterlink, then i can have VOIPOUT1=sip/${EXTEN:1}@asterlink |
| 07:10.45 | x86 | you said you wanted the provider in a variable |
| 07:10.47 | FuriousGeorge | x86: going back to what i was saying about needing to rewrite all my macros |
| 07:10.49 | x86 | not the extension too |
| 07:11.09 | FuriousGeorge | what i want is the exten out of my provider variable |
| 07:11.15 | x86 | it makes more sense to keep them separate anyway, you know |
| 07:11.21 | FuriousGeorge | i do now |
| 07:12.41 | FuriousGeorge | it was so simple before. VOIPOUT and POTSOUT were my vars and i could switch providers or use macro's for either tech... now i cant even do that |
| 07:13.04 | x86 | i dont see why not |
| 07:13.37 | FuriousGeorge | so i'd have to have a macro_pots_outbound and macro_voip_outbound, b/c arg1 and arg2 are the tech and extension parts of the dialplan |
| 07:14.56 | x86 | budmang: i'll do it for profit, but not trade |
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| 07:15.13 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 07:16.37 | FuriousGeorge | I would almost just assume stick with IAX2 but according to my new provider: We have recently become aware of a problem in the chan_iax2 implementation |
| 07:16.37 | FuriousGeorge | of IAX2 that may affect customers. This problem leads to degrading audio |
| 07:16.37 | FuriousGeorge | quality. |
| 07:17.14 | x86 | FuriousGeorge: only if you run multiple channels at the same time |
| 07:17.59 | x86 | FuriousGeorge: IAX2 is single-threaded, unlike SIP which can handle many many many concurrent threads |
| 07:18.20 | russellb | that is not correct at all |
| 07:18.41 | FuriousGeorge | the part about the multiple channels or about degrading sound quality? |
| 07:18.45 | FuriousGeorge | russellb: ? |
| 07:18.59 | x86 | russellb: oh? |
| 07:19.07 | russellb | well, the last statement about threading ...\ |
| 07:19.31 | x86 | russellb: URL? |
| 07:19.43 | x86 | russellb: as of 1.2.9 it was still single threaded anyway |
| 07:19.46 | russellb | so, in 1.2, both chan_sip and chan_iax2 use a single thread to handle incoming packets |
| 07:19.49 | x86 | maybe 1.2.10 fixed it |
| 07:20.07 | russellb | however, in chan_iax2, that includes media as well as the signalling |
| 07:20.30 | FuriousGeorge | russellb: what do you think about my provider recommending i avoid iax2 for sip? |
| 07:20.34 | russellb | but in the trunk, chan_iax2 uses a thread pool which can dynamically grow and shrink based on load |
| 07:21.08 | russellb | FuriousGeorge: well, i think it depends on your setup |
| 07:21.11 | Assid | actually i use iax2 for connecting to providers and sip for the clients |
| 07:21.22 | Assid | not to mention the additional nat advantage |
| 07:21.25 | russellb | if you're loading the crap out of your server, then sip will provide better performance in 1.2 |
| 07:22.09 | FuriousGeorge | Assid: thats what i used to do, but now that all our phones are sip im wondering if its unneccessary load to use two techs |
| 07:22.37 | FuriousGeorge | russellb: you think if im not loading the crap out of my server ill be ok with iax2 |
| 07:22.40 | Assid | well.. bridging takes care of it anwyasy |
| 07:22.56 | russellb | FuriousGeorge: absolutely |
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| 07:23.13 | *** part/#asterisk fourcheeze-away (n=rich@82.153.23.79) |
| 07:23.27 | Assid | i always though iax2 would be better since the bandwith you save on the headers is cut down tremendously |
| 07:23.34 | FuriousGeorge | Assid: i thought bridging iax2 and sip was more resource intensive than sip and sip or iax2 and iax2 |
| 07:23.34 | x86 | FuriousGeorge: assuming you keep up with the security vulnerabilities coming out left and right nowadays with IAX2 |
| 07:23.50 | FuriousGeorge | sigj |
| 07:23.54 | FuriousGeorge | *sigh |
| 07:23.56 | russellb | x86: pfft ... whatever, SIP has most of the same things |
| 07:24.09 | mitcheloc | anyone up for a moon lit stroll on the beach? |
| 07:24.10 | hads|home | x86: I don't know if you'd call it left and right. |
| 07:24.28 | x86 | russellb: i dont get flooded with asterisk SIP stack vulnerability notices from security focus like i do with IAX2 ;) |
| 07:24.31 | mitcheloc | russellb? hads? FG? |
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| 07:24.35 | Assid | russellb: so your saying if your planning to load the crap out of it.. just use sip - sip? |
| 07:24.40 | x86 | hads|home: slight exaggeration, but still ;) |
| 07:24.44 | Assid | how would it make a difference? |
| 07:24.47 | FuriousGeorge | mitcheloc: that's this weekend |
| 07:24.52 | hads|home | mitcheloc: Yeah, I like strolling :) |
| 07:25.16 | mitcheloc | yay :) |
| 07:25.26 | hads|home | x86: slight? What's it been? 1, 2? |
| 07:25.27 | russellb | Assid: well, multiple reasons, i guess. If it's the same channel type, it can transfer the media path away from asteirsk |
| 07:25.35 | x86 | hads|home: i counted 3 |
| 07:25.40 | russellb | but chan_sip in 1.2 can probably handle a higher call volume as well |
| 07:25.56 | x86 | IAX2 has lower overhead than SIP though |
| 07:26.06 | x86 | bandwidth-wise at leas |
| 07:26.07 | Assid | hrmm.. that means i gotta find sip based providers instead.. hrmmm.... |
| 07:26.09 | x86 | least* |
| 07:26.10 | russellb | indeed |
| 07:26.19 | x86 | Assid: you looking? :) |
| 07:26.23 | FuriousGeorge | russellb: can you guestimate the relative load cost of bridging sip and iax2 vs sip and sip? |
| 07:26.26 | x86 | Assid: http://www.shellshark.net/voip/ |
| 07:26.35 | mitcheloc | russellb: not interested? *sad face* |
| 07:26.35 | Assid | guess that leaves out voipjet |
| 07:26.44 | russellb | mitcheloc: sure! |
| 07:26.52 | mitcheloc | yay! :) |
| 07:26.53 | russellb | Assid: i wouldn't say don't use IAX2 |
| 07:27.22 | russellb | i'm talking about systems with hundreds of calls ... |
| 07:27.28 | Assid | well.. if it lightens the load.. why would i wanna use resources unnecessarily |
| 07:27.28 | FuriousGeorge | Assid: especially if you already set up your dialplan for it |
| 07:27.51 | russellb | Assid: because IAX2 makes your life easier? :) |
| 07:28.30 | Assid | well. if the box doesnt have any nat or anything.. wouldnt it pretty much fall in the same boat? |
| 07:28.48 | Assid | FuriousGeorge: pretty easy to modify my dialplan.. i use macros.. |
| 07:29.25 | russellb | i wouldn't go trying to fix something that isn't broken ... |
| 07:29.42 | FuriousGeorge | Assid: yeah thats what i thought. i was able to treat pots and voip the same way in those macros too since the first arg i would pass was the tech and the next the extension |
| 07:30.16 | FuriousGeorge | then i was thinking about switching to sip and as i just described to x86, all my macros would be useless |
| 07:30.30 | FuriousGeorge | since i need args for user, exten, and provider now |
| 07:30.35 | FuriousGeorge | vs just two before |
| 07:30.43 | Assid | well |
| 07:30.54 | Assid | i would normally create a third macro.. which would be the main macro |
| 07:31.02 | FuriousGeorge | not to mention i need to set them up differently for zap and voip |
| 07:31.07 | Assid | and from there.. generate the third one if needewd for zap |
| 07:31.51 | Assid | x86: your stuff? |
| 07:31.53 | x86 | third one is needed for IAX |
| 07:31.56 | x86 | Assid: yeah |
| 07:32.03 | x86 | Assid: interested? |
| 07:32.54 | Assid | only thing would be the special.. but i think a few people are giving 2 channels for a little bit around the same price |
| 07:33.09 | x86 | mine is two channels |
| 07:33.18 | x86 | residential |
| 07:33.32 | x86 | let me rephrase... all residentials are 2 channels |
| 07:34.41 | *** part/#asterisk mitcheloc (n=mitchelo@205.134.225.18) |
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| 07:45.09 | hads|home | mmmm. Moscow Mule |
| 07:47.42 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 07:48.20 | shaZwaz | help, I get no ring on my extensions |
| 07:58.06 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 08:01.01 | shaZwaz | ping |
| 08:02.41 | *** join/#asterisk roguebug (n=johnw2@dip-107-030.bras.dsl.breisnet.com) |
| 08:02.42 | roguebug | hi |
| 08:07.39 | *** part/#asterisk parag_ast (n=root@dxb-b18766.alshamil.net.ae) |
| 08:07.44 | *** join/#asterisk zachakit (n=Zachakit@cpe-24-174-39-126.houston.res.rr.com) |
| 08:07.56 | zachakit | hello |
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| 08:08.38 | zachakit | is there anybody that's an expert on asterisk? |
| 08:09.07 | roguebug | when i use the |
| 08:09.15 | roguebug | oops typo |
| 08:09.32 | roguebug | zachakit: i heard mark spencer is an expert on asterisk |
| 08:09.42 | mitcheloc | roguebug: haha... |
| 08:10.05 | zachakit | hmmm...I'm having a rather tricky problem |
| 08:10.09 | mitcheloc | zachakit: it's best to just ask what you need and if someone knows they will jump in |
| 08:10.13 | kaldemar | zachakit: just ask your question, you'll be more likely to get an answer. |
| 08:10.30 | mitcheloc | shaZwaz: nobody answered because the question you asked isn't detailed enough, it's extremely broad |
| 08:10.37 | zachakit | okay....well I'm constant getting a no such host error. This happend all of a sudden when I tried to make an outbound call... |
| 08:10.40 | zachakit | I'm using BV. |
| 08:10.59 | mitcheloc | zachakit: is your dns set up correctly? |
| 08:11.05 | zachakit | The server has been working non stop for a year.... |
| 08:11.20 | zachakit | the DNS has been fine |
| 08:11.55 | zachakit | I can email the configs...if needed. or they can be viewed at a message board...that nobody cares to respond to, lol |
| 08:12.28 | shaZwaz | mitcheloc: I have an Adit CB setup with 16x8, I can see * ringing the phone but on the phone I get no ring, but I can dialout from the phone |
| 08:12.40 | zachakit | http://forums.digium.com/viewtopic.php?t=8699 |
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| 08:17.25 | hwt | is OpenSER ready for production, or do you guys still use regular SER? |
| 08:21.57 | roguebug | hwt: what advantages does OpenSER have over SER? |
| 08:22.00 | zachakit | I need to some help resolving a no route to destination error. |
| 08:22.02 | roguebug | (and no, currently, i'm not using any *SER) |
| 08:23.18 | roguebug | zachakit: your problem has nothing to do with asterisk, it's a network config problem |
| 08:23.29 | *** join/#asterisk LoneShadow (n=duh@59.92.201.192) |
| 08:24.00 | zachakit | what makes you say that?, nothing has changed...the asterisk server rebooted...and all of a sudden quit working |
| 08:24.18 | zachakit | it registers just fine |
| 08:24.27 | zachakit | it just won't make any outbond calls |
| 08:24.29 | *** part/#asterisk budmang (n=budman@12.206.134.162) |
| 08:25.12 | LoneShadow | did you try verbose level 4 or 5 ? |
| 08:26.10 | LoneShadow | usually if you do "set verbose 5" and try making a call, should show what the problem is |
| 08:26.17 | LoneShadow | if required you can use "sip debug" |
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| 08:30.41 | zedkatuf | <PROTECTED> |
| 08:31.13 | LoneShadow | zedkatuf: I am running one of my asterisk box on openwrt right now |
| 08:31.18 | hwt | roguebug: that's what i'm wondering about. |
| 08:31.24 | hwt | roguebug: can't find any real comparisons. |
| 08:32.13 | LoneShadow | you dont need to worry about any NAT issues for your phone |
| 08:32.32 | LoneShadow | you just need to open sip port on your wan for asterisk to get incoming |
| 08:33.58 | zedkatuf | Do I keep the binaddr=0.0.0.0 in sip.conf? |
| 08:34.06 | LoneShadow | zedkatuf: add these rules in your /etc/firewall.user |
| 08:34.07 | LoneShadow | iptables -t nat -A prerouting_rule -i $WAN -p udp --dport 4569 -j ACCEPT |
| 08:34.08 | LoneShadow | iptables -A input_rule -i $WAN -p udp --dport 4569 -j ACCEPT |
| 08:34.13 | LoneShadow | and 2 more for ports 5060 |
| 08:34.18 | *** part/#asterisk kumamoto (n=kumamoto@24-178-2-212.dhcp.ftwo.tx.charter.com) |
| 08:34.21 | LoneShadow | 4569 is for IAX ports if you need them |
| 08:34.31 | zedkatuf | I'm running sip.... |
| 08:34.44 | zedkatuf | what would be the iptables comands for that? |
| 08:34.44 | LoneShadow | just use those 2 lines but port 5060 |
| 08:34.48 | hwt | zedkatuf: then open ports for SIP and RTP. |
| 08:34.50 | zedkatuf | (sorry iptables newbie) |
| 08:34.51 | LoneShadow | same thing as I pasted |
| 08:34.59 | hwt | zedkatuf: the rtp-ports used is specified in rtp.conf |
| 08:35.18 | hwt | zedkatuf: (5060/UDP, and possibly also TCP) |
| 08:36.13 | LoneShadow | yea forgot about the RTP ports |
| 08:36.35 | LoneShadow | use the same 2 lines, but --dport 10000:20000 or what ever the range is in your rtp.conf |
| 08:36.40 | zedkatuf | there's a nice-sh web front end to the firewall stuff on openwrt...shall I just ignore that & copy the firewall.up into the writeable filesystem? |
| 08:36.42 | *** join/#asterisk nailbags (n=nailbags@c220-237-123-137.randw1.nsw.optusnet.com.au) |
| 08:36.53 | zedkatuf | (it's a softlink to the rom atm..I'm running squashfs) |
| 08:37.09 | *** join/#asterisk fnordus (n=dnall@s142-179-111-243.bc.hsia.telus.net) |
| 08:37.13 | LoneShadow | you can delete it and copy the it from the /rom/etc/ folder |
| 08:37.28 | LoneShadow | if you have RC5, you can use thier web interface as well |
| 08:37.32 | *** join/#asterisk fnordus (n=dnall@s142-179-111-243.bc.hsia.telus.net) |
| 08:37.39 | LoneShadow | I think it picks it up from /etc/default/firewall |
| 08:38.04 | zedkatuf | yes, that's what I meant...ok.....yes, I'm using RC5 but it seems a bit fussy about how to add rules so I may as well just use the vi method :) |
| 08:38.31 | LoneShadow | I am used to firewall.user :D |
| 08:38.39 | zedkatuf | LoneShadow: bindaddr=0.0.0.0 in sip.conf? |
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| 08:39.12 | diablopico | hello |
| 08:39.21 | LoneShadow | yea |
| 08:39.26 | LoneShadow | I have 0s as well |
| 08:40.54 | zedkatuf | ok, sorry to be thick, but what flags to I put on the iptables command to reload the settings? |
| 08:41.04 | zedkatuf | (ie reload the firewall scripts) |
| 08:41.17 | LoneShadow | just run /etc/init.d/S45firewall |
| 08:41.22 | zedkatuf | k |
| 08:41.45 | sangee | i have two asterisk server on public and one private, i register private ip asterisk with public ip asterisk, i have may extension registered with private astersk when i make call it should go to public asterik, how to write the dialpeer? |
| 08:42.49 | LoneShadow | sangee: are both linux machines ? |
| 08:43.04 | sangee | yes |
| 08:43.19 | LoneShadow | if you have ztdummy loaded then create a iax trunk between those 2 boxes |
| 08:43.41 | sangee | how to do that? |
| 08:43.56 | LoneShadow | so both these boxes are from different domain ? |
| 08:44.02 | sangee | yes |
| 08:44.08 | diablopico | ?? i can call from my sip phone to pst phone , and call has audio ,, when i call from pstn to sip extension , it rings , and hangs up ... what should i be looking at ? |
| 08:44.42 | diablopico | pst = pstn |
| 08:44.43 | LoneShadow | diablopico: go to asterisk console, and enable verbose level to 5 |
| 08:45.11 | zedkatuf | LoneShadow is this OK for the iptables rules: |
| 08:45.13 | LoneShadow | sangee: what do you mean by private ip on one of the server |
| 08:45.14 | zedkatuf | http://pastebin.ca/121951 |
| 08:45.28 | diablopico | how do i set that at console , i onw how to on startup , but not after it is running |
| 08:45.28 | LoneShadow | does your isp give you a nat'ed ip address ? |
| 08:45.42 | LoneShadow | "set verbose 5" I think |
| 08:46.05 | sangee | private ip mean 10.0.0.2 |
| 08:46.50 | LoneShadow | zedkatuf: it looks good, while running S45firewall did you get any errors ? |
| 08:46.54 | zedkatuf | no |
| 08:47.08 | LoneShadow | sangee: is that your router setting the ip, or your isp ? |
| 08:47.10 | zedkatuf | gonna test 4 inbound now.... |
| 08:47.26 | sangee | my ISP |
| 08:47.27 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com) |
| 08:48.02 | diablopico | ok ,, i get an internal server error from mysipprovider |
| 08:48.32 | sangee | my basic question is how to work astersk to asterisk (like sipura box to asterisk, register sipura box then we can make call to asterisk)? |
| 08:49.54 | LoneShadow | sangee: check out iax trunk and iax register commands voip-info.org |
| 08:50.16 | LoneShadow | since you have private ip on one of the asterisk box, not really sure if its gonna work |
| 08:50.35 | diablopico | -- Got SIP response 500 "Internal Server Error" back from MySipProvider ... does this mean the provider is at fault ? |
| 08:50.48 | LoneShadow | you will probably be able to make outbound calls, but probably not get incoming |
| 08:50.48 | zedkatuf | hmmmm...LoneShadow: I was simplifying when I said that WAN side was internet ....it's not really, but the firewall I'm using upstream (ie between my internet connection & the openwrt box) is fowarding the relevant ports to the wan side of the openwrt box (I'm using smnoothwall for that firewall, so I didn't have to learn how to use iptables as it's done via web confg :)...but anyway, inbound connections are not getting thr |
| 08:50.48 | zedkatuf | at least nothing comes through on the *CLI of the openwrt |
| 08:52.26 | LoneShadow | hmm, havnt used smoothwall, is that the default web firewall page ? |
| 08:53.37 | LoneShadow | diablopico: when you mean pstn phone, do you mean voip service provider ? |
| 08:53.47 | zedkatuf | well, no, I'm using a smoothie mod which gives better control over port forwarding...& my setup of it has successfully forwarded ports to my asterisk@home box, so I know that the setup works fine for that computer......all I've done is changed the IP's to now point to the WAN side of the openwrt box |
| 08:54.30 | LoneShadow | oh you need to either change the aah's ports or comment them out |
| 08:54.52 | diablopico | LoneShadow when i dial to PSTN, i mena i dial thru * to the sip provider , and my landline rings |
| 08:54.59 | LoneShadow | "iptables -L" should show which ports are being opened/forwarded for which ip |
| 08:55.38 | LoneShadow | I dont think its a good idea to have same port opened on wan and forwarded to your aah box |
| 08:55.41 | zedkatuf | LoneShadow: all I've done is repointed the IP address to teh openwrt box instead of hte aah box |
| 08:55.48 | LoneShadow | aah ok |
| 08:55.57 | roguebug | (yeah, somewhere in those 20 pages of iptables output it must be) |
| 08:56.03 | LoneShadow | heh |
| 08:56.10 | roguebug | (just kiddin) |
| 08:56.16 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 08:58.15 | LoneShadow | zedkatuf: your openwrt is your nat/router right ? |
| 08:59.01 | LoneShadow | diablopico: you need to do sip register to your voip provider to receive the call, type "sip show registry" |
| 08:59.44 | zedkatuf | well, I'm more using it as an asterisk box with wifi capabilities...my real nat/router is upstream..a separate smoothwall box |
| 08:59.56 | zedkatuf | BINGO!!! |
| 08:59.59 | diablopico | Host Username Refresh State ,,,,,,, no data |
| 09:00.00 | zedkatuf | got it working |
| 09:00.42 | zedkatuf | ha |
| 09:00.45 | zedkatuf | sopeke to soon |
| 09:00.55 | zedkatuf | It dials in but no voice yet |
| 09:01.04 | zedkatuf | ahh |
| 09:01.20 | zedkatuf | prob need to modify the extensions.conf bit for incoming |
| 09:01.46 | LoneShadow | so you dont really nat on your openwrt box |
| 09:01.49 | LoneShadow | need* |
| 09:02.06 | zedkatuf | well, I am using it as it happens as I have my home network behind it :) |
| 09:02.12 | LoneShadow | oh lol |
| 09:02.19 | zedkatuf | VBG |
| 09:02.26 | diablopico | LoneShadow: thing is ,, when i dial from the PSTN phone , i see the call come into the asterisk box ,,, it rings the ip phone , and when i answer the ip phone , it hangs up immediatley |
| 09:02.32 | LoneShadow | whats this upstream box of yours ? dsl nat modem router ? |
| 09:02.51 | zedkatuf | zat a Q for diablo.. or zedkatuf? |
| 09:03.08 | zedkatuf | zat=is that |
| 09:03.17 | LoneShadow | was for you |
| 09:03.26 | zedkatuf | cable modem |
| 09:03.32 | zedkatuf | (above smoothwall box) |
| 09:03.36 | zedkatuf | (well not really) |
| 09:03.52 | zedkatuf | (I've got another router sandwiched inbetween the smoothy box & the cable modem) |
| 09:04.04 | zedkatuf | (I'm just a bit paranoid about security) |
| 09:04.05 | zedkatuf | :) |
| 09:04.45 | LoneShadow | diablopico: can you paste your console prints into pastebin.ca for the incoming call, change any personal data like passwds/ip addy |
| 09:05.14 | jeffjohnson | where I get a docu for the dial application? |
| 09:05.19 | LoneShadow | zedkatuf: a bit ? :P |
| 09:05.42 | zedkatuf | well, I've been hacked a coouple of times so I don't want them getting in anymore |
| 09:08.03 | zedkatuf | LoneShadow: inbound dailling works now (I'll tell u why it wasn't working b4 in a bit)..but when I answer the phone, I just get the engaged tone..is this because the incoming section in my extensions.conf isn't correct?: |
| 09:08.06 | zedkatuf | http://pastebin.ca/121957 |
| 09:08.36 | zedkatuf | basically, I seem to be answering the phone & then hanging up |
| 09:08.40 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 09:08.47 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 09:08.48 | zedkatuf | (I'm VERY new to Asterisk !) |
| 09:09.07 | zedkatuf | Is my incoming context wrong? |
| 09:10.43 | LoneShadow | change it to SIP/peer_name |
| 09:11.01 | *** join/#asterisk sangee (n=rkuru@82.194.57.206) |
| 09:11.05 | zedkatuf | ok, so it should read SIP/9118535 ? |
| 09:11.08 | LoneShadow | err sip user I guess |
| 09:11.20 | LoneShadow | whats the name in your sip.conf ? |
| 09:11.26 | zedkatuf | 9118535 |
| 09:11.27 | LoneShadow | [9118535] ? |
| 09:11.38 | LoneShadow | type sip show peers |
| 09:11.44 | zedkatuf | no, AUDIOLINE |
| 09:11.55 | diablopico | LoneShadow: posted |
| 09:12.11 | LoneShadow | diablopico: give me the link :D |
| 09:12.51 | diablopico | http://pastebin.ca/121962 |
| 09:13.00 | zedkatuf | sip show peers: http://pastebin.ca/121963 |
| 09:13.07 | zedkatuf | (busy day for LoneShadow!!!!!!!!!) |
| 09:13.15 | diablopico | when your good |
| 09:13.45 | sangee | how do i register one asterisk to another asterisk box? |
| 09:14.00 | LoneShadow | I am just giving back, got lot of help from past few days, got my box today morning :P |
| 09:14.05 | LoneShadow | my box up* |
| 09:14.17 | zedkatuf | k, thanks LoneShadow |
| 09:14.27 | zedkatuf | I'll try & do the same at some point... |
| 09:15.37 | diablopico | sangee: voip-info.org and read up on IAX |
| 09:15.44 | LoneShadow | diablopico: you need to have something showing up in your sip show registry |
| 09:16.24 | Assid | if a sip peer is on 729 and an iax2 outbound is on 729 .. shouldnt it attempt a native bridge |
| 09:16.53 | sangee | i put this on my sip.conf "register => 6000:6000@206.192.114.91/6000" |
| 09:17.14 | LoneShadow | diablopico: 67.133.234.125:5060 40840840840@s 75 Registered |
| 09:17.39 | LoneShadow | this is what I get when I do sip show registry, it shows that I am registerd with my sip provider to receive incoming calls |
| 09:17.59 | sangee | i have lot of extension on client asterisk, when extension make call i should send to server asterisk how to do that? |
| 09:18.09 | LoneShadow | looks like you somehow seem to receive calls still, but your voip provider is not liking something, probably codecs or rtp ports |
| 09:18.39 | zedkatuf | LoneShadow: if I don't specify a -p option in the iptables rules, will it forward both udp & tcp traffic by default? |
| 09:19.02 | zedkatuf | or do I do -p udp -p tcp |
| 09:19.26 | diablopico | zedkatuf -p all |
| 09:19.28 | *** join/#asterisk ghenry (n=ghenry@82-69-192-46.dsl.in-addr.zen.co.uk) |
| 09:19.34 | LoneShadow | zedkatuf: dump sip show users |
| 09:19.37 | zedkatuf | tnx diablopico |
| 09:20.50 | diablopico | LoneShadow: the sip provider is not set up for registry , just point to them ,, and they point to me for the DID numbers we have purchased |
| 09:21.42 | *** join/#asterisk tempest1 (n=asf@adsl-230-226-170.chs.bellsouth.net) |
| 09:21.44 | diablopico | LoneShadow: that is how i get the call from them ... |
| 09:22.07 | LoneShadow | zedkatuf: why do you need to forward tcp ? |
| 09:22.18 | zedkatuf | http://pastebin.ca/121967 |
| 09:22.40 | zedkatuf | Loneshadow: possibly don't..just copying what worked for my A'h box |
| 09:22.42 | zedkatuf | a@h |
| 09:22.51 | LoneShadow | diablopico: so they seem to have your ip address somehow, try sip debug when you receive calls, and try to read thru lots of messages and see where it is failing |
| 09:22.56 | zedkatuf | the pastebin is the sip show users |
| 09:23.10 | LoneShadow | sip debug before receiving the call, and once it fails, you can type "sip no debug" |
| 09:23.28 | zedkatuf | so maybe I should change the 9118535 to AUDIOLINE ?? |
| 09:24.09 | zedkatuf | or add a [9118535] extension in sip.conf |
| 09:24.13 | diablopico | LoneShadow: I read all the debug before i logged on here , and well , i can post it if you want |
| 09:25.20 | LoneShadow | hmm, maybe ask your voip provider whats wrong :D |
| 09:25.24 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 09:25.24 | *** mode/#asterisk [+o russellb] by ChanServ |
| 09:25.39 | LoneShadow | voxilla.com and voip-info.org might have sample settings for your vsp |
| 09:25.48 | LoneShadow | russellb can probably help you :P |
| 09:26.05 | *** join/#asterisk Yanie (n=Dave@82-69-4-3.dsl.in-addr.zen.co.uk) |
| 09:27.04 | diablopico | LoneShadow: I am pretty sure it is on the providers end now ,,,, they are new to SIP i think, so now i will get some sleep and call them in the morning.thanks for all your help |
| 09:30.35 | LoneShadow | np |
| 09:31.07 | LoneShadow | zedkatuf: what is audioline ? |
| 09:31.49 | diablopico | LoneShadow: I have another question. I have an asterisk box in another country and it uses Unicall. I have it configured properly to dial the correct output , and it places calls fine. the problem is that ( after the call is answered on the far end, there is a 20 second delay before voice converstion can take place.) once voice starts , everything is fine , i can talk with them , they talk with me ,,like normal |
| 09:32.32 | diablopico | LoneShadow: any idea what can cause the delay ? |
| 09:34.28 | zedkatuf | audioline is the extension I've called my internal phone |
| 09:34.41 | zedkatuf | LoneShadoe: audioline is the extension I've called my internal phone |
| 09:34.45 | zedkatuf | :) |
| 09:34.46 | diablopico | LoneShadow: more to that : it was working normal , then i had a dumb moment , i recompiled openh323 and pwlib, and now i have a delay. but they had the server offline for a week before i was able to log in ,, when i logged in finallyu , i was not able to get h323 to work. thats why i rebuilt the h323 |
| 09:34.47 | zedkatuf | e/w |
| 09:36.50 | LoneShadow | diablopico: not sure, I usually get a delay of like a second at the begining, but 20 is ouch |
| 09:37.13 | diablopico | thanks LoneShadow |
| 09:37.25 | LoneShadow | I am not sure if QOS might help |
| 09:37.34 | diablopico | can anyone answer that question ? |
| 09:38.17 | LoneShadow | zedkatuf: try your dial string at SIP/AUDIOLINE |
| 09:38.32 | diablopico | LoneShadow: they claim it isnt the network.. and i did rebuild openh323 and pwlib, so it might be on me |
| 09:38.36 | LoneShadow | instea of sip/9118535 |
| 09:39.06 | LoneShadow | I should probably back my configs before I mess things up :P |
| 09:39.10 | zedkatuf | u mean type: exten => SIP/AUDIOLINE |
| 09:39.16 | LoneShadow | backup* |
| 09:39.58 | LoneShadow | yea exten your_number => Dial(SIP/AUDIOLINE,30,rt) |
| 09:40.02 | LoneShadow | or whatever your settings are |
| 09:40.49 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 09:42.09 | diablopico | ===> can anyone answer my question about 20 second delays |
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| 09:44.20 | *** join/#asterisk victorz (i=victor@chac.rockone.fr) |
| 09:46.14 | jeffjohnson | I've try to forward isdn calls to a SIP gateway. Calls from an VoIP telephone over the sipgateway works. Asterisk also recognized my isdn calls... But from my isdn telephone I always get an busy sound... My extensions.conf looks like: "exten => _X.,2,Dial(SIP/${EXTEN}@sipgate,30,trg)" Anybody an idea? |
| 09:48.48 | *** join/#asterisk Arno[Slack] (i=100@master.infinityperl.org) |
| 09:59.47 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 10:08.53 | *** join/#asterisk xxoxx (n=xxoxxx@tor/regular/xxoxx) |
| 10:10.07 | zedkatuf | LoneShadow: inbound calling works-ish....but when I answer the phone, it imedialtely cuts off & gives the engaged tone (beep, beep, beep..not the fast busy tone) |
| 10:10.59 | LoneShadow | what do you see in the asterisk console, any paritcular errors ? |
| 10:11.11 | zedkatuf | sip debug info chucks out info so fast I can hardly have time to examine it |
| 10:11.20 | zedkatuf | how do I lessen the output somewhat? |
| 10:11.42 | *** join/#asterisk AsteriskAlbania (n=info@217.24.244.130) |
| 10:12.16 | zedkatuf | sip no debug |
| 10:12.22 | zedkatuf | ok, I can see error messages now |
| 10:12.26 | zedkatuf | I'll pastebin |
| 10:13.50 | zedkatuf | http://pastebin.ca/121990 |
| 10:14.00 | AsteriskAlbania | I need to call the asterisk box, I have a E1 connected to a DIGUM T410P and asterisk compalin that there is no context on the extension file , please give an URL with sample configuration |
| 10:14.09 | zedkatuf | ..it seems that I'm confused between extension names & extension numbers |
| 10:14.16 | zedkatuf | AFAICT |
| 10:16.58 | AsteriskAlbania | <PROTECTED> |
| 10:17.16 | AsteriskAlbania | I am dialing from 400074 to the E1 400995 |
| 10:17.24 | AsteriskAlbania | any help please |
| 10:18.26 | LoneShadow | zedkatuf: you can do sip debug for a particular ip or peer |
| 10:18.29 | LoneShadow | press tab |
| 10:19.34 | LoneShadow | zedkatuf: fix your codecs |
| 10:19.48 | LoneShadow | make sure you have disallow=all |
| 10:19.51 | LoneShadow | allow=ulaw |
| 10:19.55 | LoneShadow | allow=alaw |
| 10:19.57 | *** join/#asterisk ivanfm (n=ivanfm@201.52.129.236) |
| 10:19.58 | LoneShadow | and allow=gsm |
| 10:20.48 | AsteriskAlbania | <PROTECTED> |
| 10:20.50 | AsteriskAlbania | I am dialing from 400074 to the E1 400995 |
| 10:21.43 | creativx | well |
| 10:21.52 | creativx | try to paste it one more time AsteriskAlbania |
| 10:21.57 | creativx | im not sure those at the bottom of the channel list read it |
| 10:23.20 | *** join/#asterisk jpduyx (n=jpduyx@adsl-228-22.dsl.uva.nl) |
| 10:24.24 | *** join/#asterisk juice (i=1000@mo-71-50-31-176.dhcp.embarqhsd.net) |
| 10:24.42 | zedkatuf | LoneShadow: still got codec errors..should my local phone be on a different port to 5060 (eg 5062)? |
| 10:27.08 | zedkatuf | LoneShadow: here are the latest errors: http://pastebin.ca/121997 |
| 10:29.12 | *** join/#asterisk jailbreaker (n=teodory@mail.jetfinanceintl.com) |
| 10:30.31 | zedkatuf | ...looks like it's having problems doing codec conversions |
| 10:31.29 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 10:33.34 | *** join/#asterisk Delvar (n=Delvar@host-83-146-53-46.bulldogdsl.com) |
| 10:34.10 | Delvar | hi, anyone know about g729 in SVN trunk? i get a segfault with teh error: missing mod_data for codec_g729a.so |
| 10:34.11 | creativx | AsteriskAlbania: but have you checked the context default in your extensions.conf? is there a 400995 |
| 10:34.17 | diablopico | is there and expert in the house on openh323 and pwlib ? i have a delay problem 20 seconds before normal conversation starts.... |
| 10:35.16 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 10:35.24 | LoneShadow | diablopico: also ask in #freepbx |
| 10:35.29 | zedkatuf | LoneShadow: any ideas what's going wrong? |
| 10:35.37 | zedkatuf | looks like I'm 99% there now |
| 10:35.44 | zedkatuf | it's just falling over at the last hurdle :) |
| 10:36.10 | zedkatuf | actually, I can see what the problem is.just don't know what the solution is yet :) |
| 10:37.17 | zedkatuf | here's my show modules: |
| 10:37.22 | zedkatuf | http://pastebin.ca/122007 |
| 10:37.38 | LoneShadow | looks like your audiophone does only one codec for some reason |
| 10:37.58 | zedkatuf | well, if I stick to that maybe that'll sove it? |
| 10:38.47 | LoneShadow | what exactly is your audiophone ? |
| 10:38.56 | zedkatuf | BINGO |
| 10:39.16 | zedkatuf | allow=ulaw only |
| 10:39.22 | zedkatuf | It's an ATA adapter |
| 10:39.27 | zedkatuf | connected to a DECT phone |
| 10:39.28 | LoneShadow | aah ok |
| 10:39.30 | delmar | anyone here got an iTalk line with slingshot in NZ? |
| 10:39.32 | zedkatuf | OK, working now |
| 10:39.38 | LoneShadow | cool :D |
| 10:39.48 | zedkatuf | I'll email you a bottle of champagne :) :) :) |
| 10:39.56 | LoneShadow | ooo :D |
| 10:40.01 | zedkatuf | (don;t take mne too seriously) |
| 10:40.02 | LoneShadow | ok can ya make a free US call ? |
| 10:40.04 | zedkatuf | :D |
| 10:40.14 | zedkatuf | well, it's only 1.5 cents/minute |
| 10:40.22 | zedkatuf | so go ahead with ur number if u like :) |
| 10:40.25 | LoneShadow | never mind then :D |
| 10:40.26 | zedkatuf | ahhh |
| 10:40.44 | zedkatuf | except I haven't enabled international diallig yet in extensions.conf |
| 10:40.54 | zedkatuf | hey, 1.5 cents/minute is fine |
| 10:40.57 | LoneShadow | actually you can call my fwd # for free |
| 10:44.39 | creativx | hmm.. anyone here use swissvoice ip10s ? |
| 10:44.42 | puppet | 626163@fwd.pulver.com if anyone wants to try my Fwd, not sure it works |
| 10:44.53 | creativx | whatever I do to it it wont lower the ringing volume |
| 10:44.58 | creativx | it scares the crap out of me each time it rings |
| 10:45.15 | LoneShadow | puppet wanna try my fwd first ? :D |
| 10:45.15 | AsteriskAlbania | help needed for dialin , anyone can help ? |
| 10:45.25 | puppet | loneshadow: in my bedatm P |
| 10:57.13 | AsteriskAlbania | help needed for dialin , anyone can help ? |
| 11:04.20 | zedkatuf | Quick Q: can somone post a simple way to get more than one extension dialling at the same time for an incoming call? |
| 11:04.51 | zedkatuf | exten => 9118535,2,Dial(${EXTENSION1,EXTENSION3 },40,tr) |
| 11:04.55 | zedkatuf | would that work? |
| 11:06.17 | Mw3 | Dial(SIP/1&SIP/2) |
| 11:07.03 | delmar | Mw3, nice one dude.. single digit SIP devices... lol |
| 11:07.14 | delmar | Mw3, i dont think anyone here has done anything quite that dumb. |
| 11:07.26 | Mw3 | its just an example for dialing multiple extensions ... |
| 11:07.39 | Mw3 | i dont use single digit extensions |
| 11:07.46 | delmar | Mw3, ffs I hope not |
| 11:09.00 | delmar | zedkatuf, kinda.. depends if that is the second line in the context or not |
| 11:09.52 | delmar | zedkatuf, but generallt .. what u have after the Dial(etc etc... is about right. |
| 11:10.05 | delmar | generally* |
| 11:10.13 | delmar | bed time for me. l8r |
| 11:10.56 | *** part/#asterisk jpduyx (n=jpduyx@adsl-228-22.dsl.uva.nl) |
| 11:16.28 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
| 11:17.41 | jeffjohnson | If i want to forward isdn calls over SIP, I must dial the number after i have a connection to asterisk? Or can I forward all calls with a specified number from my tk to asterisk, an asterisk becomes the number from the isdn telephone? |
| 11:18.37 | zedkatuf | thanks Mw3 & delmar |
| 11:19.13 | zedkatuf | & a BIG thanks to LoneShadow|zzz |
| 11:19.18 | delmar | zedkatuf, np man. we take donations in the form of .. nekid pics of wife doing nauti thing. haha |
| 11:19.36 | zedkatuf | well, aint got any of those & it's not naughty |
| 11:19.48 | zedkatuf | just private :) |
| 11:20.13 | *** join/#asterisk [pyro] (i=pyro@tor/regular/bracketed-pyro) |
| 11:20.48 | zedkatuf | I'm just glad not to have to be running a 1.4ghz pc on 24/7 now (running it all on openwrt on a linksys wrt router instead now yipee) |
| 11:22.44 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 11:30.04 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 11:31.35 | AsteriskAlbania | need help with callback in asterisk , any ideas? |
| 11:31.39 | uwe | hello, im trying to configure meetme.conf, i set a conference on 8101 , but i keep getting this is an invalid conference number ... am i missing something? i just added it to meetme.conf |
| 11:32.29 | uwe | im using asterisk 1.0.10 ... still :( |
| 11:33.39 | uwe | i do the changes the i do reload |
| 11:34.58 | jeffjohnson | how can i view all isdn calls that asterisk receive? |
| 11:36.22 | jeffjohnson | or better, all numbers that was called from the isdn phone... misdn debug only shows me the from numbers |
| 11:37.01 | *** join/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
| 11:38.48 | Sonderblade | how does asterisk tell a phone what name or number to display when an extension receives a call? |
| 11:41.48 | charles___ | Sonderblade: sip messaging dude |
| 11:42.09 | charles___ | Sonderblade: it sends an invite with the name |
| 11:42.20 | charles___ | Sonderblade: you can configure on your voip phone |
| 11:42.28 | Sonderblade | charles___: what character encoding does the invite have? |
| 11:42.39 | *** join/#asterisk SwK (n=Silik0nJ@ip-207-145-9-58.atl.megapath.net) |
| 11:45.58 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 11:47.13 | charles___ | Sonderblade: nothing special |
| 11:47.18 | charles___ | Sonderblade: ascii stuff |
| 11:47.35 | charles___ | Sonderblade: you want to put unusual characters on the phone screen ? |
| 11:47.39 | charles___ | Sonderblade: it may not support |
| 11:49.51 | *** part/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
| 11:50.44 | Sonderblade | charles___: thats crack |
| 11:51.48 | Yanie | hmmm, is there an asterisk misconfiguration reason that results in it recognising DTMF signals from mobiles, but not from landlines? Calling out via VOIP. |
| 11:51.51 | AsteriskAlbania | any one has used callback.agi ? |
| 11:59.09 | *** join/#asterisk [TK]D-Fender (n=Administ@toronto-HSE-ppp4122655.sympatico.ca) |
| 11:59.32 | queuetue | Does anyone here use Voicepulse? Have they gone offline? |
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| 12:26.47 | jeffjohnson | what does the congestion app in the dialplan do? |
| 12:27.51 | *** join/#asterisk af_ (n=af@ip-192-212.sn2.eutelia.it) |
| 12:27.55 | benjk | it plays congestion tones |
| 12:28.17 | [TK]D-Fender | It gives you a virus :) |
| 12:29.08 | benjk | its better to use Hangup() though |
| 12:29.38 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
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| 12:31.59 | *** join/#asterisk NoRemorse (n=bah@203-214-13-168.dyn.iinet.net.au) |
| 12:32.02 | NoRemorse | hi all |
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| 12:34.56 | jeffjohnson | benjk: congestion means? :) a dictionary couldnt help me ./ |
| 12:35.07 | jeffjohnson | ben_d: busy sound? |
| 12:35.14 | jeffjohnson | ben_d: sry :) |
| 12:35.16 | *** join/#asterisk mut (n=animenod@65.111.222.120) |
| 12:39.14 | Sonderblade | anyone know of any sip phones that can display utf8 characters? |
| 12:39.31 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 12:39.55 | my007ms | Cisco one ;)O |
| 12:40.12 | NoRemorse | congestion usually refers to the call not being placed due to intermidiate route congestion rtaher than the b party being off hook |
| 12:41.14 | benjk | jeffjohnson, congestion ~~ traffic jam |
| 12:41.41 | jeffjohnson | benjk: ah ok, thx |
| 12:47.56 | *** part/#asterisk NoNeo (n=Artur@193.24.24.10) |
| 12:50.19 | [TK]D-Fender | benjk: Better to use Congestion(5), THEN Hangup |
| 12:50.49 | [TK]D-Fender | Sonderblade: Polycom IP501 and 601 IIRC. Lower models don't support Asian chars |
| 12:51.38 | benjk | not according to kapejod, he's documented quite a few cases where congestion has very nasty side effects on PRI and BRI circuits |
| 12:52.00 | *** join/#asterisk Gregabyte (n=greg@gateway.digium.com) |
| 12:53.43 | Sonderblade | [TK]D-Fender: does any support latin1 instead? |
| 12:56.30 | [TK]D-Fender | Sonderblade: Check the admin guide |
| 12:57.00 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.199) |
| 12:57.52 | Sonderblade | [TK]D-Fender: which admin guide? |
| 12:58.23 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 12:58.52 | [TK]D-Fender | Sonderblade: PJust go to www.polycom.com and check out their support for it. |
| 13:00.28 | Sonderblade | nope |
| 13:00.36 | Sonderblade | stupid phones |
| 13:02.05 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 13:02.05 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 13:04.31 | zedkatuf | Ha anyone got voicemail-to-email-attachments working with asterisk & openwrt? |
| 13:05.47 | NoRemorse | asterisk keeps core dumping on my openwrt and filling flash lol |
| 13:06.53 | *** join/#asterisk arcy (n=arcanum@ppp175-96.adsl.forthnet.gr) |
| 13:07.06 | *** join/#asterisk Mercestes (n=Merceste@216.54.143.2) |
| 13:13.26 | hmmhesays | ~seen anthm |
| 13:13.28 | jbot | anthm <n=anthmct@CPE-69-76-83-52.wi.res.rr.com> was last seen on IRC in channel #asterisk, 23h 33m 42s ago, saying: 'for?'. |
| 13:13.45 | *** join/#asterisk ph|ber (n=phiber@slackwaresupport.com) |
| 13:14.03 | ph|ber | hey..anyone have a prob with voicemail.. |
| 13:14.08 | ph|ber | Aug 8 09:12:16 WARNING[16196] app.c: Failed to lock path '/var/spool/asterisk/voicemail/default/0978/INBOX': File exists |
| 13:14.57 | hmmhesays | anyone ever try and create a dynamic dialplan natively, creating and reloading for every call |
| 13:15.11 | *** join/#asterisk falco_toadfoot (i=papir@210.84-48-86.nextgentel.com) |
| 13:15.29 | falco_toadfoot | is there any way to get h264 support in asterisk 1.0.7? |
| 13:15.47 | NoRemorse | hmmhesays have you seen realtime asterisk? |
| 13:16.15 | falco_toadfoot | yes, but must wait for next release of debian stable |
| 13:16.25 | hmmhesays | NoRemorse: yes I've seen realtime asterisk |
| 13:16.39 | NoRemorse | ok. |
| 13:16.48 | hmmhesays | haha I've been around here a long time |
| 13:16.52 | falco_toadfoot | oh, that was a nick, sorry. |
| 13:17.07 | NoRemorse | and you want to reload extensions.conf for every call? |
| 13:17.11 | [TK]D-Fender | 1.0.7. lol.... Debian packagin is so hopelessly out of date its kinda ridiculous.... |
| 13:17.27 | hmmhesays | or add and destroy a new context |
| 13:17.37 | hmmhesays | I have some 1.0.7 boxes out there |
| 13:17.43 | hmmhesays | running just fine |
| 13:17.51 | NoRemorse | well not sure if I am on the right track ehre but wouldnt you use rt for that? |
| 13:17.54 | falco_toadfoot | hmmhesays: without h264 I presume? |
| 13:18.04 | hmmhesays | falco_toadfoot: never used it |
| 13:18.09 | falco_toadfoot | I see |
| 13:18.11 | hmmhesays | NoRemorse: negative |
| 13:18.37 | hmmhesays | not for what i'm looking to accomplish it realtime is useless |
| 13:18.48 | falco_toadfoot | [TK]D-Fender: that may be true, still, fewer bugs |
| 13:18.54 | *** join/#asterisk dioedu (n=dioedu@200.207.150.85) |
| 13:18.56 | NoRemorse | and you would rather edit/create extensions.conf and reload each call, rather than write to a mysql table.. why? |
| 13:19.28 | *** join/#asterisk javar (n=javar@200.118.174.253) |
| 13:20.34 | *** join/#asterisk evol-emil (n=emile@landi.oddi.is) |
| 13:20.44 | ph|ber | anyone???/ |
| 13:21.10 | Pj_ | that would be me |
| 13:21.17 | [TK]D-Fender | falco_toadfoot: : Dunno... 1.08 came out for a reason, and several past it. |
| 13:21.54 | [TK]D-Fender | falco_toadfoot: 1.0.7 wasn't that stable.... plus the huge advances and security fixes. * from FTP release... the only way to fly. |
| 13:22.52 | falco_toadfoot | [TK]D-Fender: stable as a rock here.. newer versions doesn't neccesarily mean bug-fixes |
| 13:23.03 | falco_toadfoot | don't* |
| 13:23.33 | *** join/#asterisk tempest1 (n=asf@adsl-153-43-78.chs.bellsouth.net) |
| 13:24.03 | [TK]D-Fender | falco_toadfoot: Changelogs seem to tell of untold numbes for bug-fixes.... |
| 13:24.26 | queuetue | Is G.729 worth paying for? Is it an improvement in bandwidth at the expense of call quality? An improvement in quality over ulaw? |
| 13:24.33 | ph|ber | yes |
| 13:24.44 | [TK]D-Fender | falco_toadfoot: And theres the question of deprecation of config sytax. the longer you put it off the worse your final conversion is going to be |
| 13:24.55 | [TK]D-Fender | queuetue: Depends on your need. |
| 13:25.20 | [TK]D-Fender | queuetue: And G.729 is a compressed codec and of noticably lower quality than ULAW |
| 13:25.20 | hmmhesays | NoRemorse: I could sit and explain it, but it is complicated |
| 13:25.24 | ph|ber | how many phones do you have? |
| 13:25.31 | hmmhesays | bah chase bank sucks |
| 13:25.32 | queuetue | [TK]D-Fender: I do not have any bandwidth issues atm... I would like to improve call quality, though. |
| 13:25.41 | ph|ber | if you have 2 phones.. |
| 13:25.42 | ph|ber | no |
| 13:25.44 | ph|ber | if you have 20 |
| 13:25.44 | ph|ber | yes |
| 13:25.49 | hmmhesays | $15.00 late fee for being 30 minutes past the cut off time for payments that day |
| 13:25.53 | [TK]D-Fender | queuetue: then do NOTHING as far as codec is concerned. ULAW is as high as you'll get. |
| 13:26.03 | queuetue | [TK]D-Fender: Thank you. |
| 13:26.18 | [TK]D-Fender | queuetue: Look at your phones and PSTN connectivity next |
| 13:26.25 | [TK]D-Fender | queuetue: What do you use currently? |
| 13:27.15 | queuetue | [TK]D-Fender: PSTN sounds great - I was considering it for use over IAX channel to VoicePulse (most calls are routed over voip.). |
| 13:27.33 | falco_toadfoot | [TK]D-Fender: bugs.debian.org only reports 1 bug for asterisk 1.0.7 |
| 13:27.35 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 13:27.35 | *** mode/#asterisk [+o russellb] by ChanServ |
| 13:27.35 | ManxPower | hmmhesays, That's nothing. I used to have an account with Chase. Closed it because there are no branches anywhere near where I now live. The person that does my deposits for my business accounts had an old deposit slip and they made a deposit into my CLOSED Chase account. The account had been closed for 2 months. |
| 13:27.54 | ManxPower | The bank took the deposit, cleared the check, then took 6 weeks to return my money |
| 13:28.10 | queuetue | [TK]D-Fender: Thanks for helping me not waste 40 bucks. ;) |
| 13:28.55 | falco_toadfoot | [TK]D-Fender: configs aren't a problem.. we've always got voip-info.org with nice "only asterisk 1.2"-tags |
| 13:30.22 | [TK]D-Fender | falco_toadfoot: Yeah I suppose voip-info's outdatedness actually plays in your favour sometimes :) |
| 13:30.27 | ManxPower | Um, voip-info.org is full of wrong information. Use it only as a last resort. |
| 13:30.27 | hmmhesays | ManxPower: nice |
| 13:30.35 | hmmhesays | it is a wiki |
| 13:30.53 | [TK]D-Fender | falco_toadfoot: however we are about to end up in * 1.4. Think about that for a bit... |
| 13:31.00 | hmmhesays | is there way to add a context into the asterisk dialplan without reloading extensions.conf? |
| 13:31.31 | *** join/#asterisk trbldwine (i=troubled@adam.ur.northwestern.edu) |
| 13:31.49 | [TK]D-Fender | hmmhesays: Not really. Even realtime forces you to SWITCH the contexts out |
| 13:32.08 | hmmhesays | anthm *suggested* it was possible in a convo I was following in here |
| 13:32.09 | [TK]D-Fender | hmmhesays: A real downer in my books and one easily fixed. |
| 13:32.38 | [TK]D-Fender | hmmhesays: Anything is possible... you have the source don't you? :) Not the scale I'd appreciate having to adjust things on mind you |
| 13:33.03 | queuetue | I think I'll find something else to buy from digium. I'd like to show some support. :) |
| 13:33.13 | DarKnesS_WolF | the musiconhold works perfectly insde the same lan " sip " but i call from outside using a mobile the music in hold is very bad .. |
| 13:33.23 | ph|ber | this is the * business edition.. btw. |
| 13:33.27 | queuetue | Are The Voice recordings re-distributable? |
| 13:33.32 | hmmhesays | I think it was more of a *yes* you can do this without major modifications |
| 13:33.33 | ph|ber | anyone have a problem with voicemail lock files? |
| 13:34.06 | DarKnesS_WolF | how can i debug this ? |
| 13:34.31 | Mike | hello guys, how can i know if the timer on my asterisk box is working properly? |
| 13:35.03 | hmmhesays | exten => _X.1,Meetme |
| 13:36.09 | *** join/#asterisk tzafrir (n=tzafrir@local.xorcom.com) |
| 13:36.37 | Mike | hmmhesays, good point |
| 13:36.50 | queuetue | How do I "add a 1" to 10-digit calls? |
| 13:37.13 | hmmhesays | exten => _XXXXXXXXXX,1,Dial(TECH/1${EXTEN}) |
| 13:37.24 | hmmhesays | Mike, yeah i know |
| 13:37.26 | queuetue | Ah, ok. |
| 13:37.32 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 13:38.07 | DarKnesS_WolF | music on hold sound very cuted when it played between the SIP and zaptel channel :( " call from outside " how can i debug this ? |
| 13:38.41 | hmmhesays | anyone use lcdial in production? |
| 13:38.42 | falco_toadfoot | [TK]D-Fender: 1.2.10 is in debian _unstable_ think about that for a bit ;) |
| 13:39.14 | [TK]D-Fender | falco_toadfoot: Debian unstable is like everyone elses STABLE :) |
| 13:40.18 | falco_toadfoot | [TK]D-Fender: still, not ready for debian release |
| 13:40.30 | scorn | using a TE205P, if I 'cat /dev/zap/1 >> test.bin' and open test.bin with a hex editor, should I be able to directly see the data I sent in on E1 channel 1? Or will the data have to be processed on the RX side before it can be correlated to the transmitted data? |
| 13:40.36 | DarKnesS_WolF | [TK]D-Fender: don't kick my ass :P i need to debug a music on hold problem .. when ical lfrom cell phone the music played very poor played for 1 sec then d.c and so on . any idea? |
| 13:41.17 | Mike | hmmhesays, do you know something of trunking on iax? for bw saving? i been having trouble because it will only trunk me one way and i just check the times and they work on my 2 servers |
| 13:41.27 | ManxPower | DarKnesS_WolF, The problem COULD be simply that compressed codecs don't send music very well. |
| 13:41.42 | DarKnesS_WolF | ManxPower: what should i use ? |
| 13:41.48 | ManxPower | DarKnesS_WolF, What other types of pbones have you tested this with? |
| 13:41.51 | DarKnesS_WolF | i think zaptel uses slin ? |
| 13:42.02 | DarKnesS_WolF | ManxPower: inside the lan all are fine " ulaw " |
| 13:42.10 | DarKnesS_WolF | from outside using mobile it's really bad . |
| 13:42.24 | DarKnesS_WolF | i think the calles coming to zaptel using slin codecs |
| 13:42.25 | DarKnesS_WolF | ? |
| 13:42.30 | ManxPower | DarKnesS_WolF, And how are you calling in? direct to a zaptel port, into a SIP or IAX telco? |
| 13:42.39 | falco_toadfoot | [TK]D-Fender: you're right about documentation though..it's not exactly asterisk's strong point |
| 13:42.56 | DarKnesS_WolF | ManxPower: yes direct..it's analog 3rd world country :P egypt |
| 13:43.11 | ManxPower | DarKnesS_WolF, what make/model of Digium card? |
| 13:44.02 | DarKnesS_WolF | ManxPower: wctdm module card |
| 13:44.08 | DarKnesS_WolF | dont' remember TDM400P |
| 13:44.09 | DarKnesS_WolF | as i think |
| 13:44.13 | hmmhesays | Mike: no idea |
| 13:44.25 | DarKnesS_WolF | <PROTECTED> |
| 13:44.32 | ManxPower | DarKnesS_WolF, Does the music on hold sound good if you call from an ANALOG line? |
| 13:45.02 | ManxPower | Is Music on Hold the ONLY thing that sounds bad when you call in from your cell phone? |
| 13:45.26 | *** join/#asterisk Defraz (n=t0tal@001-717-226.area1.spcsdns.net) |
| 13:46.01 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 13:46.28 | *** join/#asterisk jaike (n=a@203.115.188.120) |
| 13:48.03 | *** part/#asterisk arcy (n=arcanum@ppp175-96.adsl.forthnet.gr) |
| 13:51.30 | DarKnesS_WolF | ManxPower: didn't try from fixed phone :-s |
| 13:51.37 | DarKnesS_WolF | ManxPower: yes it is |
| 13:51.41 | DarKnesS_WolF | the only thing that sonds bad |
| 13:51.52 | ManxPower | try calling from a fixed phone |
| 13:52.35 | jaike | just wanna make sure before i purchase additional modules, can one use any assortment of S110M or X100M on a TDM400P? |
| 13:52.58 | DarKnesS_WolF | ManxPower: hum |
| 13:52.59 | DarKnesS_WolF | will do it |
| 13:53.09 | ManxPower | jaike, on older cards you should not use FXO module on port 1 |
| 13:53.24 | my007ms | hello mr DarKnesS_WolF ;) |
| 13:53.38 | *** join/#asterisk devel (n=devel@wiggum.digitalcoven.com) |
| 13:54.43 | *** join/#asterisk trbldwine (i=troubled@adam.ur.northwestern.edu) |
| 13:54.46 | jaike | ManxPower: ok thanks! the card came with an FXS and FXO on ports 1 and 4. planning to buy 2 more FXOs |
| 13:55.07 | DarKnesS_WolF | ManxPower: the same |
| 13:55.10 | DarKnesS_WolF | hello my007ms :) |
| 13:55.27 | ManxPower | DarKnesS_WolF, put the output of "cat /proc/interrupts" on pastebin.ca |
| 13:55.46 | DarKnesS_WolF | haveing a problem with music on hold :P and another one my greetings is very short i want to put a 5 sec so the user can enter the ext. number he want to reach |
| 13:55.54 | my007ms | DarKnesS_WolF, what is your probelm there |
| 13:56.56 | DarKnesS_WolF | my007ms: the welcome messge too short so i want to do somehting like waitbackground(5) so the user has 5 sec to dial the ext. number but i don't know what the application name |
| 13:57.10 | DarKnesS_WolF | my007ms: then the music on hold from any fixed line is very bad u can't hear it |
| 13:57.40 | my007ms | DarKnesS_WolF, use wait |
| 13:57.48 | DarKnesS_WolF | ManxPower: http://pastebin.ca/122156 |
| 13:57.53 | DarKnesS_WolF | my007ms: wait will do the trick ? |
| 13:57.59 | DarKnesS_WolF | will give them time to enter the ext. number? |
| 13:58.11 | ManxPower | # |
| 13:58.11 | ManxPower | <PROTECTED> |
| 13:58.23 | my007ms | DarKnesS_WolF, did u use mpg123 |
| 13:58.28 | ManxPower | that means your card is sharing IRQ with the USB. That won't work very well. |
| 13:58.40 | jaike | Darkness_Wolf: i recorded a 15 second silent.wav file |
| 13:58.48 | my007ms | ManxPower, he not use that usb as i see |
| 13:59.01 | jaike | play it on the background after the greeting |
| 13:59.07 | ManxPower | my007ms, doesn't have to be USED, just loaded. |
| 13:59.25 | DarKnesS_WolF | ManxPower: ahh :-s no not mpg123 i'm using native |
| 13:59.41 | DarKnesS_WolF | ManxPower: so i have to disable the USB ? i'll never use it anyway |
| 14:00.04 | my007ms | DarKnesS_WolF, any how if u dont need this usb disp in motherbord |
| 14:00.14 | ManxPower | DarKnesS_WolF, disable it in the BIOS |
| 14:00.26 | DarKnesS_WolF | yep think will do so :-) but will need to take the machin out :P |
| 14:00.27 | DarKnesS_WolF | damn |
| 14:00.28 | DarKnesS_WolF | hehe |
| 14:02.02 | DarKnesS_WolF | will do so |
| 14:07.25 | *** join/#asterisk aaaazz (n=aaaazz@pool-70-20-24-133.bstnma.fios.verizon.net) |
| 14:10.25 | DarKnesS_WolF | disabled and still loaded i'll remove them manual |
| 14:16.42 | queuetue | How would I trim digits off of the beginning of ${EXTEN}? |
| 14:17.21 | Nugget | exactly in the manner described in the documentation or demonstrated in the example dialplan. ;) |
| 14:17.49 | queuetue | Nugget: I don't think I've ever seen an example dialplan. Is it online somewhere? |
| 14:18.21 | [TK]D-Fender | queuetue: Download and read THE BOOK |
| 14:18.22 | [TK]D-Fender | ~book |
| 14:18.24 | jbot | book is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 14:18.29 | Nugget | it's online and in the asterisk tarball and installed on your system. |
| 14:18.54 | queuetue | Ok, didn't know you'd all be so confounded by the question. |
| 14:19.57 | Nugget | I'm not confounded by the question at all. I'm just amazed that you'd be stumped by it since it's "page 1" stuff. |
| 14:20.08 | *** part/#asterisk falco_toadfoot (i=papir@210.84-48-86.nextgentel.com) |
| 14:20.12 | Nugget | You literally have to have read zero documentation to not know it |
| 14:20.22 | [TK]D-Fender | queuetue: and of course our fine WIKI pages... http://www.voip-info.org/wiki-Asterisk+variables |
| 14:20.44 | Nugget | if we answer this question it's like adopting you. |
| 14:20.58 | [TK]D-Fender | Nugget: I thought it'd be more like "outer sleeve" ;) |
| 14:21.47 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:22.10 | queuetue | Look, I know you guys like to be important and everything, but ... next time, just help or don't. It's not tough to ignore a question you don't feel like answering, and it's not a crime to ask the guys who know before you hit the docs. |
| 14:22.45 | queuetue | It's sometimes wierd to deal with this channel because so many of you are telephony guys, not OSS guys. |
| 14:22.51 | Nugget | we are helping. |
| 14:22.52 | Juggie | ~seen theplot |
| 14:23.09 | jbot | theplot <i=ThePlot@202.164.38.210> was last seen on IRC in channel #asterisk, 1d 20h 24m 18s ago, saying: 'I did set in the address field to match the username too'. |
| 14:23.13 | Nugget | encouraging you to make use of the resources available online is a far more valuable approach FOR YOU than just answering basic questions |
| 14:23.29 | *** join/#asterisk TrixV0x (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
| 14:23.38 | Nugget | oh, yeah, like "RTFM" is a telco invention and not an OSS invention. :P |
| 14:24.02 | queuetue | Nugget: Surprisingly, no, it isn't. Doing both would be, which is what you'd get is you asked me a question in a domain that I know more than you in. |
| 14:24.13 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 14:24.44 | Nugget | my experience is the opposite of what you're suggesting. |
| 14:26.02 | DarKnesS_WolF | ManxPower: didn't work :s but i'll try it tomorrow |
| 14:26.05 | DarKnesS_WolF | see u guys |
| 14:26.10 | queuetue | Nugget: That's odd. Your experience may be biased by your desire to ridicule people. |
| 14:26.19 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:26.19 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:26.24 | Nugget | I don't see where I've ridiculed you at all. |
| 14:26.34 | *** part/#asterisk Molotov (n=joe@unaffiliated/wiby) |
| 14:26.47 | queuetue | Oh... You were serious about the "adopting" comment? That's so cute. :) |
| 14:26.53 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 14:26.56 | Nugget | oh, ok, fair point. :) |
| 14:27.08 | Nugget | not ridicule per se, but that was definitely snide |
| 14:27.20 | queuetue | Yes, I suppose so. |
| 14:28.08 | DrkShdw | queuetue: I know right now you have a specific problem. the answer of "read the docs" may not be *what you want to hear* but it *is the best advice* because reading the docs won't just help you this time. It'll help you *every* time. Additionally, it teaches you how to find answers to problems you may have in the future, and possibly even the insight/knowledge to fix the problems without having to seek help or docs. |
| 14:28.13 | Nugget | and to be fair, it did come after your retaliatory implication that we didn't know the answer ourselves. |
| 14:28.56 | [TK]D-Fender | queuetue: Ok, we're all cool here.... Still you raelly do need to download the book and just go over the dialplan basics chapet. Its an easy read and good to keep handy. |
| 14:29.28 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-6.rcmdva.fios.verizon.net) |
| 14:29.30 | [TK]D-Fender | Nugget: I'd nix "snide" and raise you "slightly poking" :) |
| 14:29.44 | [TK]D-Fender | shapter* |
| 14:29.50 | [TK]D-Fender | CHAPTER!!!!!!! |
| 14:29.53 | queuetue | DrkShdw: Yes, and saying "Substrings use the python-like syntax of string:start:offset - look up asterisk variables in the wiki, under substrings" would have been even better. |
| 14:29.54 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 14:29.56 | [TK]D-Fender | eek, can't type today |
| 14:30.05 | scorn | using a TE205P, if I 'cat /dev/zap/1 >> test.bin' and open test.bin with a hex editor, should I be able to directly see the data I sent in on E1 channel 1? Or will the data have to be processed on the RX side before it can be correlated to the transmitted data? |
| 14:30.38 | *** join/#asterisk kfudge (n=kzymxa@24-217-137-21.dhcp.stls.mo.charter.com) |
| 14:31.36 | DrkShdw | queuetue: I don't disagree.. but you also have to understand, Nugget isn't being paid for his time here. It's really up to the individual offering the help, as to how far in depth he wants to go. It's extra work for someone to go lookup the URL, to give it to you, when you can just as easily google it as he can :) |
| 14:32.03 | DrkShdw | and in that last comment, replace "Nugget" with "anyone in a free support medium" |
| 14:32.27 | kfudge | I am having the start then instant stop of on hold music. has anyone figured out how to fix it? |
| 14:35.15 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 14:35.25 | *** join/#asterisk smackus (n=ckwall@63.149.122.93) |
| 14:35.29 | [TK]D-Fender | s/Nugget/"anyone in a free support medium"/ |
| 14:35.34 | [TK]D-Fender | :O |
| 14:35.39 | queuetue | DrkShdw: I put in an average of 20 hours a week giving free support in various online venues, most of them here on freenode. And the golden rules wrt to "helping" should be "Tell someone if you are guessing. If you're not going to help, then don't try. RTFM does nothing but alienate, but every obvious answer should be cited. Don't ridicule the temporarily ignorant. Don't engage trolls." It would have been even easier for Nugget t |
| 14:35.39 | queuetue | o not say anything at all. |
| 14:36.13 | tzanger | "temporarily ignorant" I like that :-) |
| 14:36.52 | [TK]D-Fender | queuetue: Ok we all get each other's points here and we have infact pointed you to 2 very direct sources of answers both for the question asked and probably the next 5 related to it. I think we can drop this now :) |
| 14:37.07 | *** join/#asterisk Piston` (n=no@ip72-192-233-71.dc.dc.cox.net) |
| 14:37.11 | DrkShdw | queuetue: We'll have to agree to disagree. IMO it's a slap in the face for someone in a free support medium to have to regurgitate info that is already widely available and documented online. |
| 14:37.20 | coppice | what about the permanently ignorant, and the plain old lazy? |
| 14:37.28 | [TK]D-Fender | tzanger: has a nice ring doesn't it? Sort of a self-fulfilling title no? Because it assumes the answer is impending :) |
| 14:37.30 | smackus | ok, looking for a recomendation: I was told by you guys not to use agentcallbacklogin because it is buggy and potentially system crashing... so i have been using agentlogin. works great. but my complaint with that is that it is only good for inbound only agents. (if i understand how to use it) is there a way besides agentcallbacklogin that will allow me to place an outbound call while still logged into the queue? I would want the agents to have to actuall |
| 14:37.41 | queuetue | coppice: Show me a human being that isn't lazy, and I'll show you a mistaken perception. |
| 14:37.51 | DrkShdw | coppice: it soon becomes obvious when somoene is permanently ignorant, and they typically get no responses ;) |
| 14:38.31 | coppice | queuetue: how quaint :-) |
| 14:38.32 | Piston` | I'm a little bit new to the home VOIP party. I got an old SPA-3000 from a friend. I am a US Marine stationed overseas and I want to be able to talk to my family at home. I know I need to buy a state-side phone number to place VOIP calls. Where can I buy one? |
| 14:38.47 | key2 | When someone is in a queue, is it possible to hang up on him without answering the call ? |
| 14:38.52 | jeffjohnson | the marines suck .) |
| 14:39.04 | [TK]D-Fender | key2: You mean like an agent rejecting the call? |
| 14:39.10 | key2 | [TK]D-Fender: yeah |
| 14:39.24 | key2 | [TK]D-Fender: and the client not being in the queue anymore |
| 14:39.35 | DrkShdw | Piston`: I've had good luck with teliax. but I doubt you'd want an itsp in the states, if you are overseas. and right off hand, I don't know any overseas itsp's with US numbers available |
| 14:39.39 | key2 | [TK]D-Fender: how would you do that ? |
| 14:39.43 | [TK]D-Fender | key2: when you say "hang up on" do you mean literally disconnect him, or just make your phone stop ringing? |
| 14:39.57 | key2 | [TK]D-Fender: disconnect him |
| 14:40.05 | Piston` | what's an itsp? |
| 14:40.10 | [TK]D-Fender | key2: And continue dialplan or just plain drop the call? |
| 14:40.10 | Piston` | :| |
| 14:40.14 | key2 | [TK]D-Fender: since if he stays in the queue, it would still ring |
| 14:40.33 | key2 | [TK]D-Fender: well either, since if it keeps on being in the dialplan I can Hangup() him |
| 14:40.35 | [TK]D-Fender | Piston`: Internet Telephony Sevice Provider. (Vonage, Broadvoice, etc) |
| 14:40.43 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 14:40.44 | SpaceBass | Piston`, you have tons of options.... telasip, broadvoice, vonage, sun rocket |
| 14:40.47 | Piston` | I thought it was something like that |
| 14:40.49 | DrkShdw | itsp = internet telephony service provider.. online phone company.. voip provider |
| 14:40.50 | key2 | [TK]D-Fender: have an idea ? |
| 14:41.02 | Piston` | ok I'll look at those. Thanks :) |
| 14:41.03 | [TK]D-Fender | key2: You'd have to actually pick up and transfer the call.... no other way really.... |
| 14:41.28 | SpaceBass | i'd agree with DrkShdw |
| 14:41.32 | [TK]D-Fender | I agree... I was just using them as a known sample. |
| 14:41.44 | key2 | [TK]D-Fender: unless I transfer him to an extension him |
| 14:41.46 | [TK]D-Fender | For the sake of obvious parallel |
| 14:42.04 | [TK]D-Fender | key2: that'd be the same as the agent picking up and hanging up. |
| 14:42.09 | Piston` | thanks for your help |
| 14:42.14 | SpaceBass | Piston`, depending on who you want to call....you might get a lot of value with setting up an Asterisk server in the states.... |
| 14:42.21 | [TK]D-Fender | key2: Which doesn't offer you anything and you'd know that you were hung up on. |
| 14:42.22 | key2 | [TK]D-Fender: from AMI ? |
| 14:42.28 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 14:42.33 | *** join/#asterisk BugKham (i=CKGLOB@61.47.112.140) |
| 14:42.39 | SpaceBass | let friends/family buy VoIP phones or something and everyone can connect directly....no need for a VoIP provider |
| 14:42.48 | [TK]D-Fender | key2: not sure if/how you'd be able to pull a queue call out like that... |
| 14:43.02 | key2 | [TK]D-Fender: yeah that's what I was thinkin |
| 14:43.31 | BugKham | hi, why is the "ACTION: COMMAND , command: reload" not working in 1.2.10? |
| 14:43.36 | BugKham | any one knows? |
| 14:43.51 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
| 14:43.52 | key2 | BugKham: cuz u probably don't have the right |
| 14:43.57 | key2 | BugKham: talkin about AMI right ? |
| 14:44.07 | BugKham | key2: yes |
| 14:44.13 | hmmhesays | what the crap, you can't use priority n in realtime? |
| 14:44.17 | hmmhesays | awful |
| 14:45.07 | key2 | hmmhesays: what u talking about |
| 14:45.21 | *** join/#asterisk javar (n=javar@200.118.174.253) |
| 14:45.31 | BugKham | key2: it said "Response: Success, Message: Authentication accepted" |
| 14:45.52 | key2 | ??? |
| 14:46.01 | key2 | BugKham: the machine is accessible from the outside ? |
| 14:46.03 | hmmhesays | key2 talking about realtime extensions |
| 14:46.08 | key2 | ohh |
| 14:46.18 | BugKham | key2: I'm executing this locally |
| 14:47.31 | Maroder | hi |
| 14:47.32 | BugKham | key2: the error line says "Response: Error, Message: Missing action in request" |
| 14:47.44 | Maroder | i have a following problem |
| 14:47.46 | Maroder | Aug 8 16:05:03 WARNING[7646]: codec_gsm.c:165 gsmtolin_framein: Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from oh323_read (10)? |
| 14:48.04 | Maroder | and 100 linex with this messages |
| 14:48.08 | Maroder | lines* |
| 14:48.14 | BugKham | key2: user right souldn't be a problem |
| 14:48.39 | BugKham | key2: I saw from the wiki that this is for 1.0.9 |
| 14:49.29 | BugKham | key2: don't know if there's a change for 1.2 |
| 14:50.20 | key2 | BugKham: have a look in manager.conf |
| 14:50.57 | *** join/#asterisk firstknight (n=firstkni@202.155.150.55) |
| 14:51.47 | *** join/#asterisk pengyong (n=lala@218.93.71.191) |
| 14:51.51 | sb_mx | BugKham, i think it has to do with the Action line |
| 14:52.06 | sb_mx | BugKham, it seems its not registering the request |
| 14:53.41 | BugKham | key2: permit=0.0.0.0/0.0.0.0 |
| 14:54.25 | BugKham | sb_mx: anyway to make the command registered? |
| 14:54.38 | Maroder | something about my question ? |
| 14:55.23 | sb_mx | BugKham, register wasnt the word i was looking for. I meant that the request is not getting to the manager. where are you sending this from? |
| 14:55.53 | BugKham | sb_mx: unix console |
| 14:56.35 | BugKham | sb_mx: but executing it from the browser gives the same result |
| 14:57.19 | sb_mx | BugKham, under that user.. what does the lines read and write have? |
| 14:57.33 | hmmhesays | extensions => mysql,ast_extensions is that proper syntax for mysql with database ast_extensions? |
| 14:58.22 | BugKham | sb_mx: system,call,log,verbose,command,agent,user for both |
| 14:58.22 | firstknight | Hi, I just installed asterisk ( have read until 4th chapter of asterisk book ) using ubuntu package manager. How do I register two sip user so they can call each other ( pc to pc using ekiga )? This is my configuration: http://www.pastecode.com/2794 . But I seem unable to register with ekiga... any idea? |
| 14:59.11 | ManxPower | What is "ekiga" |
| 14:59.31 | BugKham | sb_mx: before the error line I had "Response: Follows", "Privilege: Command" and then "--END COMMAND--" |
| 14:59.32 | hmmhesays | linux soft phone |
| 14:59.38 | ManxPower | I don't think the East Kentucky Independent Grocers Association has a softphone. |
| 14:59.39 | firstknight | ManxPower, www.ekiga.org ( another sip phone software ) |
| 14:59.45 | hmmhesays | and h.323 |
| 15:00.00 | ManxPower | firstknight, you would need to check the docs for your softphone to find out how to tell it to register with a SIP server. |
| 15:00.00 | sb_mx | BugKham, you said you got a Authenticated message right? |
| 15:00.13 | BugKham | sb_mx: seems it didn't wait for my "command: reload" |
| 15:00.22 | *** join/#asterisk gniretar_work (n=mark@aa2.linuxbox.com) |
| 15:00.23 | ManxPower | hmmhesays, ekiga is an *H.323* softphone? |
| 15:00.24 | BugKham | sb_mx: yes |
| 15:00.24 | gniretar_work | hi all |
| 15:00.38 | hmmhesays | h.323 and sip |
| 15:00.47 | gniretar_work | hey, about how many transcoded GSM calls can go through a 3Ghz asterisk box? |
| 15:00.50 | firstknight | ManxPower, any idea how to test without soft phone? Maybe something like ping if we want to check internet connection.... |
| 15:01.01 | ManxPower | firstknight, no. |
| 15:01.35 | hmmhesays | so realtime load table_name column_name field_value should return something if my realtime is setup properly right? |
| 15:01.57 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 15:02.02 | smackus | so what exactly is the problem with agentcallbacklogin? |
| 15:02.31 | firstknight | is it possible use client and asterisk as sip server in one computer....? my softphone in server complains udp port 5060 is in use, cannot initialize |
| 15:03.50 | [TK]D-Fender | firstknight: : set it to 5061 and make your entry for it match |
| 15:03.56 | key2 | http://88.191.12.21/3g/ |
| 15:04.28 | ManxPower | firstknight, not usually |
| 15:04.30 | StewLG | key2: I can't help, but I love the diagrams. |
| 15:04.49 | ManxPower | If you know what you are doing, you might be able to. |
| 15:05.38 | key2 | StewLG: which one :) |
| 15:06.28 | firstknight | [TK]D-Fender, when you say "set it..." it refers to server or client? entry for what? |
| 15:07.27 | *** join/#asterisk cytrak (n=kvirc@adelphi.geofocus.com) |
| 15:07.41 | cytrak | hey anyone here using RTA on asterisk ? |
| 15:08.00 | ManxPower | The Regional Transit Authority? |
| 15:08.13 | cytrak | ARA I mena |
| 15:08.16 | cytrak | mean |
| 15:08.21 | coppice | Really Terrible Acronym? |
| 15:08.26 | ManxPower | You mean, Realtime |
| 15:08.29 | cytrak | yes |
| 15:08.40 | ManxPower | firstknight, put the softphone on a different machine. |
| 15:09.31 | cytrak | I'm setting up a Xen server for testing that on the latest asterisk and I'm gonna use all the files that I already have configured on my production box |
| 15:09.35 | hmmhesays | ok this is driving me nuts now |
| 15:09.49 | BugKham | sb_mx: thanks |
| 15:10.05 | sb_mx | BugKham, yw |
| 15:10.10 | firstknight | ManxPower, okay, that means to test asterisk as sip server, I have to use three computer.... |
| 15:10.20 | ManxPower | cytrak, make it work in Xen without ARA first. |
| 15:10.28 | ManxPower | firstknight, no, you need 2 computers |
| 15:10.38 | ManxPower | Asterisk Server Computer and the SIP Client Computer |
| 15:10.39 | cytrak | Once I enable ARA to use realtime for my extensions.conf will it read through my extensions.conf file and load it into mysql |
| 15:10.45 | cytrak | ManxPower: it already does |
| 15:11.03 | phearless | WTF is proxy1_address in the Cisco VOIP phone config ? I don't want any proxy ... |
| 15:11.25 | ManxPower | phearless, proxy address would the address of your asterisk server |
| 15:11.35 | phearless | okay thanks :-) |
| 15:12.05 | phearless | and no thanks to cisco for their shitty-doc |
| 15:13.00 | [TK]D-Fender | Polycom > ALL |
| 15:13.04 | [TK]D-Fender | (for now) |
| 15:13.46 | *** join/#asterisk devel (n=devel@wiggum.digitalcoven.com) |
| 15:13.59 | *** join/#asterisk anthm (n=anthm@000-402-822.area4.spcsdns.net) |
| 15:13.59 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:15.11 | *** join/#asterisk sip_me (n=ask@80.179.11.31.static.012.net.il) |
| 15:15.52 | sip_me | Hi, When calling via Asterisk from one SIP client to the other, I hear no ringback. ?? |
| 15:16.12 | sumasuma | hi anyone please help me in configuring my sipura adapter |
| 15:16.18 | sumasuma | SIPURA 3000 |
| 15:16.29 | sumasuma | it never registers with my asterisk |
| 15:16.35 | sumasuma | while xlite works fine |
| 15:16.48 | hmmhesays | [TK]D-Fender: you using realtime anywhere? |
| 15:16.51 | sip_me | It seems that the Asterisk does create RTP connection to the SIP Client, but I think it does not send ring back tones |
| 15:17.06 | sb_mx | sip_me, are you sending the 'r' option when calling Dial? |
| 15:17.23 | sip_me | sb_mx, r option? |
| 15:17.26 | cytrak | has anyone tried voiceone ? |
| 15:17.28 | hmmhesays | ring ring ring ring |
| 15:17.31 | *** part/#asterisk trbldwine (i=troubled@adam.ur.northwestern.edu) |
| 15:17.48 | sb_mx | sip_me, if i recall correctly the 'r' option inside the DialString tells asterisk to generate ring-tones for the calling party |
| 15:18.16 | sip_me | OK. |
| 15:18.41 | sip_me | Where is the DialString configured? |
| 15:20.22 | ManxPower | sb_mx, WRONG!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! |
| 15:20.32 | sevard | How does one obtain g729 for educational/network testing before buying a license? |
| 15:20.35 | [TK]D-Fender | hmmhesays: nope |
| 15:20.40 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
| 15:20.47 | sumasuma | ManxPower: can you please help in configuring the sipura |
| 15:20.50 | kpettit | can you do allow=gsm,ulaw,alaw.... or does there have to be a spereate line for each codec? |
| 15:20.55 | sip_me | sb_mx, is this the string defined in extensions.conf? |
| 15:21.05 | ManxPower | Asterisk will generate a ringing sound to the caller by default. The "r" option tells Asterisk to send a ringing sound to the caller WHEN IT SHOULD BE SENDING SOME OTHER TONE OR MESSAGE |
| 15:21.11 | Corydon-w | sevard: licenses are cumulative... you can have a system with multiple license packs, no problem |
| 15:21.12 | [TK]D-Fender | kpettit: seperate lines only |
| 15:21.20 | kpettit | [TK]D-Fender, thanks |
| 15:21.36 | ManxPower | If Asterisk does not generate a ringing tone for you, "r" will not normally fix it. |
| 15:21.47 | sevard | Corydon-w: I understand that. How do I obtain a binary (or source) of said codec to try out? |
| 15:21.51 | kpettit | [TK]D-Fender, and if it's in the general context it will apply to all the sip peers correct? |
| 15:21.54 | *** join/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 15:21.59 | sb_mx | ManxPower, really? wow, i thought it wasnt by default. thanks for the info |
| 15:21.59 | Corydon-w | sevard: you buy them |
| 15:22.02 | qseek | hello all |
| 15:22.37 | sevard | Corydon-w: iirc on voip-info it says you can use the codec for educational uses for free as long as you don't incorporate it into a comercial service/product |
| 15:22.44 | kpettit | [TK]D-Fender, I'm seeing some peers crap out slin errors but I don't have slin specificed as a allowed codec, so I'm tring to force it out to see if that gets rid of the problem |
| 15:22.47 | Corydon-w | For instance, you can buy a single license to try it out, then buy 22 additional licenses for when your system goes into production |
| 15:23.01 | StewLG | Manx: What other tone or messae would it be supposed to be sending, if not ring, when doing a dial()? |
| 15:23.15 | Corydon-w | sevard: that's a different codebase |
| 15:23.21 | sevard | Corydon-w: and you purchase these from Sipro Labs i take it, do you know the price? |
| 15:23.32 | sevard | Corydon-w: now that does make things confusing ;) tell me more |
| 15:23.33 | Corydon-w | sevard: no, you buy from Digium |
| 15:23.33 | qseek | does anyone have a clue why if I have incoming call on say X100P and the caller disconnects the phone on the inside still keeps ringing and the call goes to voicemail..anyway to remedy this |
| 15:23.37 | sip_me | ManxPower, can you think of a reason why would Asterisk not send a ringtone to a sip client? |
| 15:23.39 | ManxPower | StewLG, busy sound, congestion tone, "all circuits are busy" message, "the cellular caller you are trying to reach is not in the service area" |
| 15:23.39 | sb_mx | ManxPower, heh. re-read the wiki. now i got it. thanks again |
| 15:23.39 | kpettit | <PROTECTED> |
| 15:23.55 | ManxPower | sip_me, do you have a /etc/asterisk/indications.conf ? |
| 15:24.00 | Corydon-w | sevard: G.729 is a specification, for which there are multiple implementations |
| 15:24.01 | kpettit | I'll get a few hundered of those really fast then it'll kill the call. Any ideas? |
| 15:24.05 | ManxPower | kpettit, "show codecs" |
| 15:24.10 | *** join/#asterisk remiss (i=bofh@191.80-203-38.nextgentel.com) |
| 15:24.20 | sip_me | ManxPower, if so, it is the default one. I never touched such a file after installation |
| 15:24.22 | kpettit | ManxPower, I've got all the codecs there |
| 15:24.29 | Corydon-w | sevard: $10/license |
| 15:24.32 | sevard | Corydon-w: but the royalities all go back to the orignating patent holders, no? |
| 15:24.39 | Corydon-w | Correct |
| 15:24.42 | ManxPower | kpettit, what are codecs 4 and 256? |
| 15:24.48 | kpettit | I'm just using this with Polycom phones which can use gsm,ulaw and alaw |
| 15:24.55 | sevard | So digium would be essentialy a reseller |
| 15:25.07 | hmmhesays | realtime is giving me hell, I have extconfig.conf extensions => mysql,ast_extensions,extensions_table ast_extensions is the database and extensions_table is the table name and in extensions.conf I have switch => Realtime/echo@extensions_table echo being the context in extensions_table |
| 15:25.18 | Corydon-w | sevard: Digium is selling licenses for a particular implementation that works with Asterisk |
| 15:25.26 | sevard | I see. |
| 15:25.33 | kpettit | ManxPower, G.711 u-law and G.729A |
| 15:25.43 | ManxPower | kpettit, do you have a G729 license? |
| 15:25.55 | kpettit | no, it should be using g729 though |
| 15:26.02 | coppice | sevard: it seems if you have a G.729 code base you are clear to use, sipro will now sell you one off licences. they used to only sell volume ones |
| 15:26.08 | kpettit | these are all local connections. Outside calls go through a PRI |
| 15:26.15 | ManxPower | Why? One leg is ulaw, one leg is G729 |
| 15:26.19 | qseek | does anyone know why the zaptel channel won't hangup on caller disconnect |
| 15:26.28 | ManxPower | then disable g729 in sip.conf |
| 15:26.41 | kpettit | ManxPower, that tripping me out. in sip.conf general I put a disallow=all and specified the codecs I want to use |
| 15:26.45 | ManxPower | qseek, Asterisk is not detecting the hangup. |
| 15:26.55 | ManxPower | kpettit, put your sip.conf on pastebin.ca |
| 15:26.58 | kpettit | ManxPower, if it's in general it should apply to all the peers correct |
| 15:26.59 | sevard | coppice: I see that the 'one off license' is a license to use the coded at an unlimited rate |
| 15:27.02 | qseek | is there a way to fix this is in the zapata.conf |
| 15:27.05 | kpettit | it's massive. have 200 phones |
| 15:27.18 | kpettit | let me put the genreal and first few peers though... |
| 15:27.21 | ManxPower | qseek, how does your telco indicate far end disconnect. |
| 15:27.25 | lehel | if i want to change the ASTRUNDIR, i need to recompile asterisk? |
| 15:27.35 | qseek | it is giving me hell as even though the caller disconnects the user gets a long empty message in his voicemail |
| 15:27.37 | sevard | Excellent. it's evident that Digium makes this quite easy. |
| 15:27.45 | qseek | i am not sure how to find that out Manx |
| 15:27.46 | *** join/#asterisk ronn (n=zakforev@87.112.9.177.bbplus.ptn-ag1.dyn.plus.net) |
| 15:27.48 | coppice | sevard: I have no idea what that means. what is an unlimited rate? G.729 turbo? :-) |
| 15:27.55 | ManxPower | qseek, what country are you in? |
| 15:28.01 | qseek | U.S |
| 15:28.13 | ManxPower | qseek, are you connected directly to the telco> |
| 15:28.18 | sumasuma | anyone here to help me to configure my sipura 300 plz |
| 15:28.38 | sumasuma | *sipura 3000 |
| 15:28.46 | sumasuma | i want to use fxo from asterisk |
| 15:28.47 | qseek | through a normal telephone line |
| 15:29.06 | Corydon-w | qseek: who is your telco? |
| 15:29.13 | qseek | verizon |
| 15:29.26 | ManxPower | qseek, then I don't know why it's not working. USA Telcos indicate far end disconnect by dropping battery on the line for some small amount of time (500ms?). Asterisk should detect that just fine. |
| 15:29.43 | qseek | do i need to set some parameter in the zapata.conf |
| 15:29.48 | ManxPower | qseek, no |
| 15:29.58 | kpettit | ManxPower, ughh hI found it. Some individual peers had it |
| 15:30.03 | Corydon-w | qseek: fxs_ks or fxs_ls ? |
| 15:30.07 | *** join/#asterisk mog (n=mogorman@gateway.digium.com) |
| 15:30.07 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:30.07 | ManxPower | kpettit, There you go. |
| 15:30.08 | sip_me | Can anyone suggest a freeware linux or win2K Sip-Phone that I test my asterisk with (known to work well) |
| 15:30.13 | Juggie | ManxPower, i dont think every line automatically provides disconnect supervision. |
| 15:30.23 | Juggie | sometimes you have to ask for it dont you? |
| 15:30.27 | Corydon-w | Juggie: most lines in the US do |
| 15:30.28 | sumasuma | sip_me: xlite ! |
| 15:30.38 | Corydon-w | Juggie: only backwoods telcos don't |
| 15:30.47 | qseek | manxpower: i have noticed that it will do that at certain stages in the call |
| 15:30.50 | sip_me | sumasuma, that's a win2k clinet right? |
| 15:30.51 | queuetue | Has anyone ever heard local radio stations on a zap POTS line? Is likely it a local grounding problem, a zaptel issue, or a telco problem? (I'll isolate line and check with analog equipment later, just not on site.) |
| 15:30.51 | Juggie | also known as rural? :) |
| 15:30.59 | qseek | like if I let it ring two or three times it will disconnect |
| 15:31.14 | qseek | however when it goes to voicemail and I hang up it does not hang up immediately |
| 15:31.14 | ManxPower | sip_me, once you have a /etc/asterisk/indications.conf then come back |
| 15:31.41 | ManxPower | queuetue, it's common. Radio shack should have filters for that |
| 15:31.43 | sip_me | ManxPower, I have one - the default one :) |
| 15:31.49 | queuetue | It's only on the PBX side of outgoing calls... |
| 15:32.02 | qseek | does it matter if it is a DSL line? |
| 15:32.15 | queuetue | ManxPower: Oh, ok. They are rj45 filters? |
| 15:32.18 | ManxPower | queuetue, no |
| 15:32.22 | ManxPower | ..er.. qseek no |
| 15:32.27 | *** join/#asterisk Ankhone (i=Ankhone@208.206.35.250) |
| 15:32.29 | ManxPower | queuetue, rj-11 filters |
| 15:32.37 | ManxPower | or call your telco and complain |
| 15:32.43 | qseek | k |
| 15:32.43 | ManxPower | or call the radio station and complain. |
| 15:32.45 | qseek | i will do that |
| 15:32.48 | queuetue | Heh. Yes, rj-11, my mistake. :) |
| 15:32.54 | qseek | thank you |
| 15:33.00 | queuetue | ManxPower: The radio station is in spanish, so I wouldn't get far there. :) |
| 15:33.11 | *** part/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 15:33.13 | ManxPower | Radion stations MAY be required to install filters on request. |
| 15:33.15 | remiss | bah... |
| 15:33.36 | *** join/#asterisk ebody (n=eBody@207.71.51.162) |
| 15:33.39 | ManxPower | I know that radio stations are required to install them on TVs on request |
| 15:33.51 | hmmhesays | <PROTECTED> |
| 15:33.51 | hmmhesays | <PROTECTED> |
| 15:33.55 | ebody | what would be a reason that my SIP call keep breaking up? |
| 15:34.06 | ronn | hi guys... any one knows if it's possible to match an incoming SIP call by username and IP address? i,e using both parameters? |
| 15:34.11 | hmmhesays | is that what switch should look like for realtime? |
| 15:34.34 | ManxPower | ronn, see permit/deny in sip.conf.sample |
| 15:34.48 | ManxPower | ebody, network issues |
| 15:34.54 | ronn | manx. i 'm using realtime. is there a field for that ? |
| 15:35.02 | ebody | ManxPower, not a config issue at all? |
| 15:35.23 | ManxPower | ebody, none that I can think of |
| 15:35.36 | hmmhesays | ahah ronn maybe you can help me out |
| 15:35.46 | *** join/#asterisk iscrow (n=ivani@208.48.9.6) |
| 15:36.07 | ronn | hmmhesays. what was yuor q? |
| 15:36.23 | hmmhesays | just tinkering with realtime right now and it is not working |
| 15:36.44 | ronn | hmmhesays: what error are u seeing ? |
| 15:36.47 | hmmhesays | i have extconfig.conf extensions => mysql,dbname,tablename |
| 15:37.14 | hmmhesays | and in extensions.conf switch => Realtime/context@table_name |
| 15:37.18 | hmmhesays | is that right? |
| 15:38.00 | ronn | that's the correct syntax. what's the name of your table ? |
| 15:38.12 | smackus | is anyone out here using the command AgentCallbackLogin? |
| 15:38.16 | hmmhesays | switch => Realtime/echo@extensions_table |
| 15:38.24 | hmmhesays | extensions_table is the name of the table |
| 15:38.40 | hmmhesays | and echo is a context I have in the context field of extensions_table |
| 15:39.09 | *** join/#asterisk W9SH (n=W9SH@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 15:40.25 | hmmhesays | nothing in the debugs, it is like it is not even trying to look up |
| 15:40.37 | iscrow | Does any of you know how I can force an zap FXO port to be permanently offhook. I need to take it out of service. |
| 15:40.45 | ronn | have u tried this? [echo] switch => Realtime/@extensions_table |
| 15:41.20 | ronn | i.e you create an entry in extensions.conf for [echo] |
| 15:42.01 | hmmhesays | that is the same as including it in the switch line |
| 15:42.03 | hmmhesays | and no good |
| 15:42.26 | ronn | what error do u see in the console? when u try to make a call ? |
| 15:42.46 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:43.32 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 15:43.53 | hmmhesays | Looking for 1234 in echo SIP/2.0 404 Not Found |
| 15:44.11 | hmmhesays | nothing no debug messages, which I think is odd |
| 15:44.36 | *** join/#asterisk _alex_mx_ (n=_alex_mx@200.78.229.18) |
| 15:46.51 | sumasuma | anyo ne please help me with my sipura 3000 box to configure with my asterisk ? |
| 15:48.49 | hmmhesays | it is freaking odd |
| 15:50.39 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 15:51.14 | *** join/#asterisk queuetue (n=scott@toronto-HSE-ppp4122670.sympatico.ca) |
| 15:51.47 | *** join/#asterisk acctor (n=heh@user-514d3b5b.l1.c2.dsl.pol.co.uk) |
| 15:52.52 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:52.53 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:53.32 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 15:54.23 | toerkeium | guys, if my hard/soft phones have licence for g729 and my SIP provider works with g729, I won't need a g729 licence because it just passthrough righ ? |
| 15:54.35 | Qwell | toerkeium: in most cases, yes |
| 15:54.55 | sumasuma | qwell: you have used sipura 3000 for fxo ? |
| 15:54.59 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 15:55.03 | toerkeium | you mean yes I am right? or I am wrong ? |
| 15:55.17 | Qwell | toerkeium: That was a big ambiguous, wasn't it? Yes, you're correct. |
| 15:55.31 | Qwell | but not all cases... |
| 15:55.55 | Qwell | s/big/bit/ |
| 15:55.56 | toerkeium | what would be one case where I would need a g729 licence? when I try to call another SIP with uses another codec? |
| 15:56.18 | Qwell | that, or when you try to use things like voicemail (there are ways to make voicemail work with g729) |
| 15:56.29 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 15:57.13 | toerkeium | Qwell, what if I allow=gsm after g279 ? woulnd't it pick up gsm for voicemail as the second alternative? |
| 15:57.24 | Qwell | toerkeium: in many circumstances, yes |
| 15:57.51 | toerkeium | ok so, I better have a couple of g729 just in case right? |
| 15:58.05 | Qwell | not necessarily... |
| 15:58.38 | toerkeium | so, what should I take care of ? |
| 15:58.41 | file | Qwell: you're being very vague... |
| 15:58.46 | Qwell | file: yeah...bbl :P |
| 15:58.48 | toerkeium | heh |
| 15:58.57 | toerkeium | I didn't wanted to tell him that :P |
| 15:59.10 | smackus | also, i am looking for a way to do skill based routing. is this possible with vanilla asterisk? i can only find the addon stuff. |
| 15:59.23 | smackus | skill based agents |
| 15:59.26 | Qwell | toerkeium: You just have to consider all of the scenarios where transcoding might be needed |
| 15:59.35 | Qwell | and understand when transcoding happens |
| 15:59.44 | Qwell | when/why |
| 16:00.32 | Qwell | bbl, work |
| 16:00.37 | toerkeium | I understand why it happens, it's just when one end doesn't support g729.. not sure about all scenarios |
| 16:00.40 | toerkeium | thanks Qwell |
| 16:00.55 | file | that's in a calling scenario... but it could also happen if you're doing music on hold |
| 16:01.14 | cytrak | this is strange for some reason meetme module was not compiled |
| 16:01.20 | ronn | hmmhesays: Looking for 1234 in echo SIP/2.0 404 Not Found -- do u have the 1234 extension in echo inside your table? |
| 16:01.29 | toerkeium | oh.. didn't take that in count |
| 16:02.02 | dlynes_laptop | cytrak: maybe you didn't have zaptel compiled and installed? |
| 16:02.05 | toerkeium | I guess I have to read more about it |
| 16:02.11 | toerkeium | thanks nice people :) |
| 16:02.27 | EvilDeshi | aanyone know if it is at all possible to use meetme without the zt_dummy kernel mod? |
| 16:02.30 | file | I can't let you go! why can't you stay? give it one more try! |
| 16:02.46 | file | EvilDeshi: you must have some zaptel timing source (ztdummy/actual hardware) to use meetme |
| 16:02.49 | dlynes_laptop | EvilDeshi: yes..you can use a different zaptel driver |
| 16:03.00 | EvilDeshi | the problem is i have a VPS for asterisk |
| 16:03.01 | cytrak | oh that's why .... I don't have a timing source (digium card installed) |
| 16:03.06 | EvilDeshi | and i cant install kernel mods |
| 16:03.12 | EvilDeshi | i dont have kernel level access |
| 16:03.14 | dlynes_laptop | EvilDeshi: then you're out of luck |
| 16:03.19 | EvilDeshi | thats what i thought |
| 16:03.23 | toerkeium | you don't talking to me right file ? |
| 16:03.31 | EvilDeshi | damit oh well |
| 16:03.40 | file | I'm talking to all those who will listen ;) |
| 16:03.47 | toerkeium | heh |
| 16:03.48 | EvilDeshi | what about call transfer that will work right? |
| 16:03.52 | dlynes_laptop | EvilDeshi: so don't be so cheap then...use a real server |
| 16:03.58 | EvilDeshi | im not cheap |
| 16:04.03 | EvilDeshi | its 100 bux a month |
| 16:04.10 | dlynes_laptop | HOLY CRAP |
| 16:04.13 | EvilDeshi | just the way midphase is set up |
| 16:04.13 | file | uh |
| 16:04.18 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 16:04.20 | dlynes_laptop | $100 a month, and you don't even get a real server? |
| 16:04.32 | EvilDeshi | well the cost is mostly the BW |
| 16:04.33 | EvilDeshi | 155mbit |
| 16:04.42 | dlynes_laptop | ah |
| 16:04.52 | toerkeium | lies, lies |
| 16:04.55 | EvilDeshi | i run a small radio station and we use asterisk for our shows |
| 16:04.56 | toerkeium | I hate lies |
| 16:05.12 | dlynes_laptop | why do you need so much bandwidth then? |
| 16:05.17 | EvilDeshi | streaming |
| 16:05.22 | file | pr0n. |
| 16:05.35 | toerkeium | what do you use for streaming EvilDeshi ? |
| 16:05.37 | file | or streaming, either or |
| 16:05.40 | hmmhesays | ok do realtime extensions no support pattern matching? |
| 16:05.44 | dlynes_laptop | damn....that's gonna have a nice impact on asterisk :) |
| 16:06.04 | file | hmmhesays: they should |
| 16:06.08 | EvilDeshi | toerkeium i use icecast shoutcast and sam3 broadcaster |
| 16:06.15 | *** join/#asterisk firejon (n=jellis@206-169-49-105.static.twtelecom.net) |
| 16:06.25 | hmmhesays | in _XXX and _XXX. format? |
| 16:06.27 | EvilDeshi | thats just to stream the fm signal |
| 16:06.36 | EvilDeshi | anyuways that is not important right now |
| 16:06.53 | toerkeium | icecast, good choice :) |
| 16:07.03 | hmmhesays | looks like it was case sensitive |
| 16:07.08 | hmmhesays | my bad |
| 16:07.58 | file | silly hmmhesays |
| 16:08.28 | smackus | does anyone here use queuing? |
| 16:08.40 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
| 16:09.27 | hmmhesays | yeah file realtime has been giving me hell all morning |
| 16:09.38 | toerkeium | file: speaking again about g729, if I allow all codecs and set g729 as the first one, I should not have any troubles right? as the allow list has priority.. (?) |
| 16:09.47 | EvilDeshi | yeah |
| 16:09.53 | hmmhesays | switch => Realtime/<context> doesn't work at all |
| 16:09.54 | EvilDeshi | if you list it as the first one it will sue that first |
| 16:09.58 | file | depends on direction too |
| 16:10.00 | EvilDeshi | use* |
| 16:10.16 | file | if Asterisk is calling a SIP phone, the SIP phone has control |
| 16:11.10 | toerkeium | hmmm |
| 16:11.55 | hmmhesays | anything I can get to work is adding switch => Realtime |
| 16:12.20 | file | hmmhesays: that sentence doesn't quite make sense to me |
| 16:14.08 | hmmhesays | oh file i'm playing with realtime, when i include a switch statement ex: switch => Realtime/context@table_name it does not work |
| 16:14.23 | file | ah, better |
| 16:15.31 | hmmhesays | but if I just have [context] switch => Realtime it works, because I have defined my table_name in extconfig.conf |
| 16:16.53 | hmmhesays | but according to the docs you should be able to use the first switch statement |
| 16:17.13 | sumasuma | what is SIP:ICMP Error ? |
| 16:18.07 | dlynes_laptop | sumasuma: probably nothing is set up to listen for incoming connections on UDP 5060, or there was a timeout while connecting to UDP 5060, or something similar |
| 16:18.30 | sumasuma | when i try with xlite it is working fine |
| 16:18.35 | smackus | am i just being ignored? someone reply to me just to let me know my posts are hitting the room. i have not had a single response to anything today. |
| 16:18.41 | sumasuma | i got that error only in sipura 3000 |
| 16:19.01 | dlynes_laptop | sumasuma: ICMP is typically used for tracing a route, or for checking ports, and that kinda thing |
| 16:19.08 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.71.71) |
| 16:19.08 | sumasuma | yes |
| 16:19.13 | dlynes_laptop | sumasuma: i.e. for response status |
| 16:19.16 | eKo1 | smackus: There are many people here that use queueing. |
| 16:19.22 | smackus | ahhh |
| 16:19.25 | smackus | i am heard |
| 16:19.26 | eKo1 | Not I though. |
| 16:19.38 | eKo1 | I don't have the need for it yet. |
| 16:19.40 | dlynes_laptop | smackus: and i've used sipura 3000's, but not queueing |
| 16:19.45 | smackus | just looked like maybe i was not even getting my posts out there. |
| 16:19.49 | dlynes_laptop | smackus: but that's the first i've seen anything about sipura 3000 today |
| 16:20.15 | smackus | 'twas not me that talked about the sipura 3000 |
| 16:20.19 | smackus | only queuing. |
| 16:20.31 | smackus | i am looking for some way to do skills based agent routing. |
| 16:20.43 | *** join/#asterisk Assid (i=assid@203.115.83.214) |
| 16:20.49 | dlynes_laptop | ah...yeah..i seen that...no idea what it is |
| 16:20.52 | ManxPower | ICMP is also sent when you are trying to connect to a port that nothing is listening on, if you try sending a packet that is prohibited by a firewall, and many other things. It is also used for MTU discovery (which is why, when you block all ICMP, some web sites don't load any more) |
| 16:20.57 | toerkeium | by default * doesn't come with the g729 for passtrough right? |
| 16:20.57 | smackus | google sends me to many other providers products, but i cannot find anything within asterisk |
| 16:21.10 | dlynes_laptop | ManxPower: that's what I said..port status :) |
| 16:21.20 | sumasuma | ManxPower: yes, but when i ping from a dos window it is working fine with ping |
| 16:21.34 | ManxPower | toerkeium, Asterisk bas PASSTHRU without a license, but remember than if Asterisk has to touch the audio then it's not passthru anymore. |
| 16:21.41 | dlynes_laptop | sumasuma: ping doens't test port status, and ping doesn't always use ICMP |
| 16:21.48 | dlynes_laptop | sumasuma: ping can also use UDP |
| 16:21.55 | Assid | hrm.. voicepulse seems to be facing issues today |
| 16:22.10 | toerkeium | ManxPower: but it comes buil-in with * ? |
| 16:22.13 | ManxPower | standard ping sends an ICMP echo request and gets back an IDMP Echo reply |
| 16:22.20 | ManxPower | toerkeium, yes. |
| 16:22.39 | toerkeium | thanks |
| 16:23.13 | ManxPower | just remember than many options to Dial will prevent passthru, you can passthru if you have to transcode (one leg not using G729 like a call to the PSTN), voicemail won't work, etc. |
| 16:23.33 | ManxPower | ...you CAN'T passthru if you transcode... |
| 16:25.19 | ManxPower | (DirecWay satellite internet proxy does not work with many audio streaming sites) |
| 16:25.56 | Nugget | I've been looking at wildblue.com for my parents. They're on starband.net right now and it's pretty awful. |
| 16:26.00 | hmmhesays | you need to vpn to a good proxy |
| 16:26.26 | ManxPower | hmmhesays, because of the latency VPNs only get about %10 of the rated speed. |
| 16:27.16 | *** join/#asterisk javar (n=javar@200.118.174.253) |
| 16:27.29 | scorn | using a TE205P, if I 'cat /dev/zap/1 >> test.bin' and open test.bin with a hex editor, should I be able to directly see the data I sent in on E1 channel 1? Or will the data have to be processed on the RX side before it can be correlated to the transmitted data? |
| 16:27.45 | hmmhesays | ManxPower: ever use openvpn? |
| 16:27.50 | ManxPower | when you have 3000ms RTTs and 2200ms of jitter lots of stuff doesn't work very well. |
| 16:27.52 | hmmhesays | I beg to differ |
| 16:28.25 | ManxPower | Heck, a simple GRE tunnel is unusable. |
| 16:28.30 | hmmhesays | I've used openvpn over satellite connections, nothing as bad as 3000ms, you have a shitty provider if you are getting those times |
| 16:28.52 | ManxPower | hmmhesays, with no traffic it's 800ms - 1200ms |
| 16:28.57 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 16:28.57 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 16:29.35 | ManxPower | When I finally have the budget for a T-1 I'll throw a party at the campground and invite everyone. |
| 16:29.55 | *** join/#asterisk BeBrA (n=abc@host56-241.pool8248.interbusiness.it) |
| 16:29.58 | BeBrA | hello! |
| 16:30.35 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 16:30.48 | BeBrA | can I configure asterisk to add default international prefix if dialed number starts with a 0 and dial as is if start with double 0? |
| 16:31.29 | *** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net) |
| 16:33.16 | vader-- | is there a way to merge two gsm files into one? |
| 16:33.18 | *** join/#asterisk xnon (i=xnon@200.82.223.113) |
| 16:33.26 | *** join/#asterisk quid2478 (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 16:33.34 | benjk | soxmix |
| 16:34.30 | vader-- | soxmix? |
| 16:34.39 | benjk | yep |
| 16:34.52 | benjk | man sox will tell ya |
| 16:40.50 | cytrak | even when you have * set to use realtime , does it still read the iax.conf file ? |
| 16:41.03 | cytrak | my iax.conf file has some include statements |
| 16:41.18 | jarrod | is there a script that will roll thru an extensions.conf and build a sql config |
| 16:43.40 | *** join/#asterisk Zalbag (n=tesmerjf@adsl-156-148-145.mia.bellsouth.net) |
| 16:43.46 | *** part/#asterisk smackus (n=ckwall@63.149.122.93) |
| 16:43.47 | *** join/#asterisk I-MOD (i=opticron@c-71-207-206-191.hsd1.al.comcast.net) |
| 16:47.05 | *** part/#asterisk firejon (n=jellis@206-169-49-105.static.twtelecom.net) |
| 16:50.04 | *** join/#asterisk Bullseye_Network (n=info@216.143.192.69) |
| 16:53.54 | toerkeium | is there any way to detect an answer machine than when someone pickups the phone? or for the caller party is the same? |
| 16:54.14 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 16:54.34 | pif | hi, anyone has polycom 1.6.7 firmware handy? |
| 16:54.53 | tzanger | pif: contact your reseller; they have it |
| 16:55.03 | pif | it's a hassle |
| 16:55.34 | tzanger | howso |
| 16:55.37 | tzanger | who's the reseller? |
| 16:55.41 | pif | zycko |
| 16:56.51 | hmmhesays | ok this thing is pissing me off |
| 16:56.59 | hmmhesays | where does gcc look for includes by default? |
| 16:57.24 | *** join/#asterisk jmls (n=jmls@62.49.235.130) |
| 16:58.22 | eKo1 | /usr/include ? |
| 16:58.34 | eKo1 | and your current dir. |
| 16:58.46 | eKo1 | and a whole bunch of other stuff depending on your setup |
| 17:03.21 | *** join/#asterisk momelod (n=momelod@bas5-toronto12-1168029679.dsl.bell.ca) |
| 17:05.11 | momelod | hey peeps, something strange is going on with my extensions.conf.. when im in a call, and if i hit the dtmf keys, then i start hearing wierd echos, one person hit some keys and heard a voicemail message in the background |
| 17:06.58 | *** join/#asterisk ACiDV (n=joel@c66.110.128-170.clta.globetrotter.net) |
| 17:07.55 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:07.55 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 17:08.56 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 17:08.59 | ACiDV | I use Queue 'rrmemory' and Dynamic Agents on Asterisk 1.2.10... Someone can explain me why some agent doesnt receive calls when the status is 'Not in use' ? Full 'SHOW QUEUES' and configs @ http://pastebin.ca/122379 |
| 17:09.08 | TripleFFFF | hey.. trunk not compiling lately ? |
| 17:09.09 | TripleFFFF | [2]: *** virtual memory exhausted. Stop. |
| 17:09.09 | TripleFFFF | gmake[2]: Leaving directory `/usr/src/asterisk/codecs/ilbc' |
| 17:09.09 | TripleFFFF | gmake[1]: *** [ilbc/libilbc.a] Error 2 |
| 17:09.09 | TripleFFFF | gmake[1]: Leaving directory `/usr/src/asterisk/codecs' |
| 17:09.52 | dlynes_laptop | cool beans |
| 17:10.00 | TripleFFFF | hmm |
| 17:10.01 | dlynes_laptop | TripleFFFF: how much physical memory you have? |
| 17:10.02 | TripleFFFF | any idea ? |
| 17:10.11 | TripleFFFF | Mem: 2073860k total, 914404k used, 1159456k free, 53604k buffers |
| 17:10.12 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 17:10.15 | dlynes_laptop | dayam |
| 17:10.17 | quid2478 | Has anybody pushed some serious traffic through *? I've read the wiki page on dimensioning... but looking for someone who has done it first hand |
| 17:10.29 | dlynes_laptop | TripleFFFF: i would report it on #asterisk-dev, personally |
| 17:10.34 | TripleFFFF | its freebsd |
| 17:10.37 | dlynes_laptop | TripleFFFF: let russellb know about it |
| 17:10.40 | dlynes_laptop | ah |
| 17:11.10 | dlynes_laptop | TripleFFFF: you're compiling trunk on freebsd? |
| 17:11.10 | dlynes_laptop | TripleFFFF: are you using gmake? |
| 17:11.11 | TripleFFFF | yes |
| 17:11.25 | dlynes_laptop | TripleFFFF: when did you grab trunk? |
| 17:11.30 | TripleFFFF | and mem was Mem: 131M Active, 1146M Inact, 165M Wired, 2884K Cache, 112M Buf, 556M Free |
| 17:11.34 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 17:11.34 | *** mode/#asterisk [+o mog] by ChanServ |
| 17:11.35 | TripleFFFF | more likely |
| 17:11.39 | *** join/#asterisk apardo (n=apardo@87.217.147.151) |
| 17:11.42 | TripleFFFF | just one sec ago i svn trunked |
| 17:11.51 | dlynes_laptop | gimme a sec...I'll try it under linux |
| 17:12.00 | dlynes_laptop | then you can rule out whether it's a freebsd problem or not |
| 17:12.24 | TripleFFFF | k |
| 17:12.38 | TripleFFFF | [13:12] file: upgrade GNU make |
| 17:13.29 | dlynes_laptop | TripleFFFF: which verfsion of gnu make are you using? |
| 17:13.45 | TripleFFFF | gmake -v |
| 17:13.45 | TripleFFFF | GNU Make 3.80 |
| 17:13.47 | file | need 3.81 |
| 17:13.52 | TripleFFFF | ah |
| 17:14.03 | dlynes_laptop | ah...i'm running 3.80 also :) |
| 17:14.20 | dlynes_laptop | is it a bug in 3.81, file? |
| 17:14.42 | file | sort of, the Makefiles do stuff that makes other versions sad |
| 17:15.02 | dlynes_laptop | like recursive macros? |
| 17:15.20 | file | recursive RECURSIVE recursive macros, 3 is better then 1 |
| 17:15.30 | Qwell[] | recursively recursive? |
| 17:15.39 | mut | missed one |
| 17:15.43 | TripleFFFF | hmm |
| 17:15.52 | TripleFFFF | any idea on how to upgrate gmake on fresbd |
| 17:15.52 | TripleFFFF | lol |
| 17:15.59 | TripleFFFF | tried portupgrade .. no luck |
| 17:16.10 | TripleFFFF | bah |
| 17:16.12 | TripleFFFF | thank |
| 17:17.59 | *** join/#asterisk SwK (n=Silik0nJ@ip-207-145-9-58.atl.megapath.net) |
| 17:18.24 | *** join/#asterisk rogier (n=rogier@83-67-dsl.ipact.nl) |
| 17:19.07 | *** join/#asterisk hunterotd (n=hunter@68-184-2-160.dhcp.mmll.ar.charter.com) |
| 17:19.54 | *** join/#asterisk c4t3l (n=c4t3l@69.15.174.114) |
| 17:20.36 | *** join/#asterisk SwK (n=Silik0nJ@ip-207-145-9-58.atl.megapath.net) |
| 17:21.24 | dlynes_laptop | wow...talk about fast |
| 17:21.30 | c4t3l | does anyone know of a speaker device that can act as a one-way pager |
| 17:21.43 | dlynes_laptop | virtual memory exhausted in a matter of a fraction of a second |
| 17:22.28 | c4t3l | i'm trying to use a Valcom pager system to no avail |
| 17:22.31 | dlynes_laptop | even does it on make clean |
| 17:22.38 | dlynes_laptop | now that's pretty lame |
| 17:22.45 | mut | c4t3l: dunno |
| 17:23.06 | dlynes_laptop | and it's a bs error, too |
| 17:23.19 | dlynes_laptop | virtual memory wasn't even touched, i don't think |
| 17:24.53 | *** part/#asterisk hunterotd (n=hunter@68-184-2-160.dhcp.mmll.ar.charter.com) |
| 17:24.58 | ronn | hi guys... any one knows if it's possible to match an incoming SIP call by username and IP address? i,e using both parameters? |
| 17:25.00 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 17:25.14 | mut | we're finally going to redo our company website!!!! |
| 17:26.02 | *** join/#asterisk saftsack (n=oliver@p54A7DF8E.dip.t-dialin.net) |
| 17:26.59 | eKo1 | I've been in that process for 4 months now. |
| 17:27.17 | *** join/#asterisk SwK (n=Silik0nJ@ip-207-145-9-58.atl.megapath.net) |
| 17:27.19 | mut | well that means i can redo it in php |
| 17:27.22 | mut | right now it's asp |
| 17:27.29 | mut | and the whole thing is a revamp |
| 17:27.37 | mut | so W00T! |
| 17:28.34 | *** join/#asterisk queuetue (n=scott@toronto-HSE-ppp4122670.sympatico.ca) |
| 17:29.44 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 17:31.16 | dlynes_laptop | file: looks like that problem only occurs with inefficiently written makefiles :) |
| 17:31.29 | dlynes_laptop | file: http://savannah.gnu.org/bugs/index.php?func=detailitem&item_id=14527 |
| 17:31.51 | file | don't look at me, I don't touch the Makefiles :P |
| 17:31.52 | *** join/#asterisk Eonz (n=Icarus@irc.americatelnet.com.pe) |
| 17:32.18 | file | patches welcome though! |
| 17:32.21 | file | as long as they don't break things |
| 17:32.30 | Juggie | makefiles scare me |
| 17:33.21 | Qwell[] | Juggie: better than scarring you |
| 17:33.42 | dlynes_laptop | file: i don't know enough about gnu makefiles to fix it |
| 17:33.54 | dlynes_laptop | file: only makefiles I know well are watcom makefiles and ant build scripts |
| 17:34.08 | Qwell[] | let's convert it to watcom then |
| 17:34.14 | dlynes_laptop | OK!!! |
| 17:34.17 | dlynes_laptop | not |
| 17:34.23 | *** part/#asterisk StewLG (i=user@216-99-218-126.dsl.aracnet.com) |
| 17:34.25 | dlynes_laptop | gnu make is a much better make |
| 17:34.30 | Juggie | Qwell, i guess :) |
| 17:34.31 | dlynes_laptop | I just don't know how to work with it |
| 17:34.41 | Juggie | watcom 32bit extender FTW |
| 17:35.01 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 17:35.01 | *** mode/#asterisk [+o anthm] by ChanServ |
| 17:35.11 | Juggie | RUSSELL! |
| 17:35.32 | *** join/#asterisk firstknight (n=firstkni@202.155.150.55) |
| 17:35.58 | Juggie | sigh |
| 17:36.01 | Juggie | i hate this databasde |
| 17:36.04 | Juggie | *database |
| 17:36.21 | Juggie | who the hell @ bell designed this thing, i have a CDRTABLE w/ 130million records |
| 17:37.18 | eKo1 | what is wrong with that? |
| 17:37.29 | Juggie | i' |
| 17:37.34 | Juggie | i'd rather it monthly |
| 17:37.43 | Juggie | but it also isnt indexed |
| 17:37.54 | Juggie | and its MSSQL |
| 17:37.59 | Juggie | its actually 155million |
| 17:37.59 | Qwell[] | no indexes? ugh |
| 17:38.06 | Qwell[] | still? :P |
| 17:38.08 | eKo1 | Create an index then. |
| 17:38.12 | Juggie | haha |
| 17:38.18 | eKo1 | and take the week off. |
| 17:38.23 | Juggie | turn an index on on a table w/ 155 million records? :) |
| 17:38.32 | Juggie | mssql would loose its mind. |
| 17:38.39 | Juggie | Qwell, we are upgrading to a new software release soon |
| 17:38.46 | Juggie | so i hope that one includes it |
| 17:40.01 | firstknight | Hi, I try to make asterisk as sip server in local lan. computer a ( with softphone; ekiga ) can call computer b ( with softphone; ekiga ) but computer b can not call computer a ( security check failed ). And webcam support is failed. Why? This is my configuration: http://www.pastecode.com/2799 |
| 17:40.39 | *** part/#asterisk a1fa (n=a1fa@207.210.210.202) |
| 17:41.16 | Juggie | i'm worknig on a report right now for calling patterns |
| 17:41.32 | Juggie | eg, a user calls a 1-800 number first, if they get busy what do they call next |
| 17:41.41 | Juggie | but its a pain because querys take like 4minutes... |
| 17:41.53 | Juggie | maybe i should dump a section of data into another table to work with. |
| 17:42.37 | *** join/#asterisk tempest1 (n=asf@adsl-153-35-194.chs.bellsouth.net) |
| 17:43.44 | *** join/#asterisk networkjedi (n=networkj@f0c216.gpcom.net) |
| 17:44.15 | *** join/#asterisk vpanayotov (n=vdp@213.91.154.185) |
| 17:46.59 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:47.00 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 17:51.42 | remiss | when using music on hold, the music disappears unless it is going sound the other way, any suggestions on what to search for? |
| 17:51.58 | hmmhesays | does gcc look in your working directory first for includes or /usr/include |
| 17:52.42 | Juggie | disable silence supression on your sip phone |
| 17:52.46 | remiss | and only seems to be a problem on sip |
| 17:53.52 | eKo1 | hmmhesays: I think it looks in your dir. first |
| 17:54.11 | *** part/#asterisk ph|ber (n=phiber@slackwaresupport.com) |
| 17:54.26 | iscrow | Do you guys know of a way to permanently put a zap fxo channel in offhook state. I need to take it out of service. |
| 17:54.26 | ManxPower | remiss, what SIP device? |
| 17:54.31 | *** join/#asterisk ComputerWarm (n=donc@209.29.158.47) |
| 17:54.35 | ManxPower | remiss, Asterisk will not send audio unless it's getting audio |
| 17:54.38 | *** join/#asterisk lvit (n=kvit@ppp-124.120.147.70.revip2.asianet.co.th) |
| 17:54.43 | ManxPower | remiss, Quite simple, really. |
| 17:54.45 | remiss | ManxPower: it's sip to pstn |
| 17:54.54 | ManxPower | remiss, correct. |
| 17:54.59 | *** part/#asterisk jaike (n=a@203.115.188.120) |
| 17:55.07 | ComputerWarm | hello is there anyone i can get india,Pak,BD,DR,Philippines dids from? |
| 17:55.13 | ManxPower | remiss, now WHAT sip device are you using? |
| 17:55.23 | ManxPower | ComputerWarm, ask on asterisk-biz |
| 17:55.27 | *** join/#asterisk Assid (i=assid@203.115.83.213) |
| 17:55.34 | ComputerWarm | oh ok thanks |
| 17:55.38 | remiss | ManxPower: i'm not quite sure i understand you :-/ |
| 17:55.45 | firstknight | How to enable video support in sip server? What configuration should I put in sip.conf beside videosupport=yes? |
| 17:55.54 | ComputerWarm | no one there |
| 17:55.56 | ManxPower | remiss, SIP device. Cisco, Polycom, X-Lite softphone, SIPura? |
| 17:56.11 | ManxPower | ComputerWarm, the asterisk-biz MAILING list. |
| 17:56.20 | ComputerWarm | oh already did that |
| 17:56.38 | remiss | ManxPower: it's a handytone 486, but it's not involved |
| 17:57.10 | ManxPower | then exactly what is involved. |
| 17:57.26 | remiss | my cellphone speaking to asterisk via sip |
| 17:57.26 | ManxPower | and what Digium card are you using to interface to the PSTN? |
| 17:57.53 | ManxPower | remiss, Cell phone -> PSTN -> Some odd SIP telco -> Asterisk |
| 17:57.56 | remiss | sorry.. it's sip connection terminated to pstn at the providers end.. |
| 17:57.58 | ManxPower | Is that the diagram |
| 17:58.17 | ManxPower | then your provider must disable CNG, VAD, and silence supression. |
| 17:58.22 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 17:58.27 | remiss | thath's what asterisk is saying too :( |
| 17:59.10 | remiss | Aug 8 19:54:21 NOTICE[25089]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 64.34.224.24 |
| 17:59.15 | remiss | or am i mistaken? |
| 17:59.30 | ManxPower | That is correct. |
| 17:59.47 | *** join/#asterisk blaylock (n=seth@snap.helixsystems.com) |
| 17:59.53 | ManxPower | Asterisk does not support RFC3389 and will not work correctly with devices that use it. |
| 18:01.01 | remiss | nothing i can do with sip.conf at all to make it better? |
| 18:01.19 | E-bola | change provider |
| 18:01.19 | remiss | it doesn't seem to be a problem when it's not music though.. |
| 18:01.22 | E-bola | should be easy enough |
| 18:01.37 | ManxPower | remiss, there is NOTHING you can do. |
| 18:01.42 | ManxPower | the PROVIDER must do it. |
| 18:01.45 | remiss | okay.. thanks.. |
| 18:01.51 | *** join/#asterisk tzafrir (n=tzafrir@local.xorcom.com) |
| 18:02.35 | ManxPower | remiss, when the call is set up between Asterisk and the provider, Asterisk will inform the provider that it does not support silence supression. The provider is ignoring that and using it anywat. |
| 18:02.39 | ManxPower | anyway too |
| 18:02.56 | remiss | aha.. practical.. |
| 18:03.21 | remiss | i'll give 'em a ring tomorrow, but i highly doubt they will care :p |
| 18:03.40 | ManxPower | Rather rude actually. It's like someone coming over to dinner. They tell you they are alergic to seafood and you serve shrimp anyway. |
| 18:03.51 | remiss | hehe |
| 18:06.21 | Zalbag | If I want to setup asterisk to use a PRI do build in the order zaptel/libpri/asterisk or libpri/zaptel/asterisk? |
| 18:06.47 | SpaceBass | anyone got a recommendation on an IAX or VoIP provider for my home office? need to be able to bill my company directly, not my credit card |
| 18:06.51 | SpaceBass | leaning towards telasip |
| 18:06.57 | *** join/#asterisk atapi2 (n=virgill4@c-69-180-119-156.hsd1.fl.comcast.net) |
| 18:07.43 | iscrow | Zalbag, libpri/zaptel/asterisk |
| 18:08.06 | Zalbag | SpaceBass: I don't know about billing your company directly but we use TelIAX, they've been ok but they're service as of late has been flakey |
| 18:08.18 | SpaceBass | flaky? |
| 18:08.26 | iscrow | That's what I do at least and it works. |
| 18:08.32 | Zalbag | SpaceBass, I've heard good things about voicepulse's connect service as well |
| 18:08.58 | Zalbag | Yeah like randomly during the day for 15 seconds to a couple minutes registration will fail and we can't receive or make phone calls |
| 18:08.59 | iscrow | I used voicepulse. They are pretty good. |
| 18:09.19 | SpaceBass | i use broadvoice now...and I think there is a lag |
| 18:09.30 | SpaceBass | people keep saying I'm cutting them off, but I think I'm talking when they aren't |
| 18:09.44 | Zalbag | Don't use BroadVoice, they're horrible...I've had nothing but problems witht hem |
| 18:09.46 | Zalbag | with them* |
| 18:10.10 | SpaceBass | I've used them for like 1 1/2 years....they are ok most of the time |
| 18:10.14 | ComputerWarm | SpaceBass: i heard AirStar was doing iax termination. never tried them yet though |
| 18:10.16 | SpaceBass | but they cannot bill my company |
| 18:10.20 | Zalbag | ah |
| 18:11.00 | Assid | voicepulse is decent.. |
| 18:11.12 | Zalbag | I tested BV out for a couple weeks and had nothing but constant voice quality issues (lag, drop packets, not being able to call, horrible sound, etc) |
| 18:11.23 | Zalbag | but it's true that YMMV |
| 18:11.23 | Assid | they are currently having issues tho |
| 18:11.39 | Assid | ymmv ? |
| 18:11.45 | Zalbag | your mileage may vary |
| 18:11.58 | quid2478 | Has anybody pushed some serious traffic through *? I've read the wiki page on dimensioning... but looking for someone who has done it first hand |
| 18:12.29 | Assid | quid2478: alot of people here run call centers.. so im guessing they must have pushed some traffic |
| 18:12.31 | E-bola | the wiki page seems hardcore exagerated if u ask me |
| 18:12.34 | E-bola | atleast with a sip setup |
| 18:12.52 | *** join/#asterisk Flauto (n=HP_Owner@adsl-75-3-169-144.dsl.chcgil.sbcglobal.net) |
| 18:13.14 | Zalbag | Is the TE110P fairly easy to work with? I've read voip-info.org and the install guide on digium's website but I am still a little skeptic |
| 18:13.22 | *** join/#asterisk anonymouz666 (n=anonymou@h1e2.compuland.com.br) |
| 18:13.23 | SpaceBass | Zalbag, I've heard mostly problems about BV...I think I've been lucky |
| 18:13.33 | SpaceBass | and thats why I want to migrate away...before I have any real issues |
| 18:13.38 | Zalbag | Ah |
| 18:13.50 | quid2478 | Ebola: Yeah, that's the problem... the data is all over the chart it seems |
| 18:14.39 | Zalbag | Honestly I don't think we'll ever use another internet based provider again. There's too much that could go wrong and generally it's really hard to get decent support |
| 18:15.34 | Zalbag | Yes a local telco could have problems as well but generally copper is a heck of a lot more reliable than internet |
| 18:16.10 | *** join/#asterisk rephorm (n=rephorm@cpe-24-27-8-18.austin.res.rr.com) |
| 18:16.37 | ManxPower | Zalbag, My clients use the local Telco (usually PRI). If we run out of trunks, we might send the call out via IP to a provider. |
| 18:17.28 | Zalbag | ManxPower, yeah I would keep it around for backup but relying on them as a main "trunk" is not a good idea :( |
| 18:17.49 | Zalbag | hence why I am here asking about PRI's ;) |
| 18:17.56 | Zalbag | PRIs* |
| 18:18.57 | ManxPower | Zalbag, I can never remember which order to build libpri so I build them in this order: libpri; zaptel; libpri; asterisk |
| 18:19.14 | ManxPower | or zaptel; libpri; zaptel; asterisk |
| 18:19.30 | Zalbag | hmmm |
| 18:19.30 | Zalbag | k |
| 18:19.37 | Zalbag | thanks :) |
| 18:19.37 | rephorm | ManxPower: zap, pri then asterisk |
| 18:19.46 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 18:19.46 | *** mode/#asterisk [+o anthm] by ChanServ |
| 18:20.33 | wunderkin | yups |
| 18:21.40 | ManxPower | Zalbag, as long as you don't have motherboard / irq issues, it's pretty easy to work with. |
| 18:22.21 | *** part/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 18:22.37 | Assid | okay if i get something like attempting native bridge |
| 18:22.55 | Assid | and sip show channels shows me 2 different channels using 2 different codecs |
| 18:23.00 | Assid | how is it a native bridge |
| 18:23.10 | *** join/#asterisk tempest1 (n=asf@adsl-144-59-43.chs.bellsouth.net) |
| 18:23.13 | Zalbag | ManxPower, is there any specific motherboard I should be looking for? |
| 18:23.54 | rephorm | is anyone here using realtime in a production environment? are there any cons to doing this? (i've been away from asterisk since pre-1.2) |
| 18:24.19 | ManxPower | Zalbag, search the mailing list archives for "HDLC" errors to find what ones you should stay away from. |
| 18:24.21 | rephorm | also, anyone know of any issues between te110p's and dell poweredge 1950's? |
| 18:24.46 | ManxPower | rephorm, your search of the mailing lists did not turn anything up? |
| 18:25.25 | scorn | using a TE205P, if I 'cat /dev/zap/1 >> test.bin' and open test.bin with a hex editor, should I be able to directly see the data I sent in on E1 channel 1? Or will the data have to be processed on the RX side before it can be correlated to the transmitted data? |
| 18:25.26 | rephorm | ManxPower: not on the 1950 |
| 18:26.04 | Assid | err.. anyone know how come im getting this? it shows attempting native bridge.. but still sip show channels shows me 2 diferent codecs |
| 18:26.15 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 18:26.15 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 18:26.18 | Zalbag | is there a better way to searching em besides ctrl+f |
| 18:26.20 | Zalbag | ? |
| 18:26.42 | ManxPower | ~mailinglist |
| 18:26.44 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 18:27.21 | Zalbag | <-- nub cake |
| 18:27.29 | ManxPower | ~docs |
| 18:27.30 | jbot | i guess docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 18:27.38 | ManxPower | ~book |
| 18:27.40 | jbot | rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 18:28.23 | TripleFFFF | ~* |
| 18:28.24 | jbot | ACTION sings with $who |
| 18:28.29 | rephorm | ManxPower: looks like someone else asked about the 1950 a few days ago. no responses (and google hadn't picked it up yet) |
| 18:28.46 | *** join/#asterisk junbug (i=junya@c-67-191-62-53.hsd1.fl.comcast.net) |
| 18:29.07 | *** join/#asterisk caloi (n=caloi@nat-66-218-1-166.usadatanet.com) |
| 18:29.14 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 18:30.38 | ManxPower | Results 1 - 10 of about 20 from lists.digium.com for fucked. (0.54 seconds) |
| 18:31.38 | mog | ? |
| 18:33.34 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.71.71) |
| 18:37.59 | rephorm | ManxPower: ? |
| 18:39.08 | Zalbag | Is echo usually really bad on a PRI line? Is it something I should consider when building out a new server possibly having to get a beefer processor handle echo cancelation? |
| 18:39.16 | quid2478 | ahh the 1950's... |
| 18:39.21 | quid2478 | those were the days... |
| 18:39.30 | quid2478 | or so I've been told, asn't around then |
| 18:39.40 | rephorm | quid2478: issues? |
| 18:39.47 | ManxPower | Zalbag, go with hardware EC. |
| 18:40.14 | ManxPower | Either an external EC (you can get Tellabs cheap on eBay, but they can be hard to get up and running). |
| 18:40.33 | ManxPower | I don't know if Digium's EC works well or not. Tellabs is what the TELCOS use to do EC |
| 18:40.39 | quid2478 | rephorm: isues in the 1950's were wage parity for women and the cold war. :) |
| 18:41.03 | ManxPower | the mid 60s to the mid 70s would have been more interesting. |
| 18:41.33 | rephorm | quid2478: hehe. :) |
| 18:41.48 | quid2478 | I was born in '72... wish I could have went to a real live "disco" ala Saturday Night Fever. would have been awesome putting on lots of polyester before a night on the town |
| 18:48.11 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 18:51.10 | *** join/#asterisk mtaht4 (n=m@adsl-75-10-213-145.dsl.pltn13.sbcglobal.net) |
| 18:52.56 | Zalbag | Has anyone used Sangoma's cards before? I just read a thing that says they don't have IRQ problems? |
| 18:53.44 | Corydon-w | LOL |
| 18:53.58 | tzanger | Zalbag: they work well, I've never had issues with digium or sangoma |
| 18:54.07 | Corydon-w | The only type of card that doesn't have IRQ problems is a card that doesn't claim an IRQ |
| 18:54.08 | tzanger | quid2478: erm... |
| 18:54.31 | tzanger | I haven't run across a digium card with IRQ issues in a LONG time now, I think they cleaned up alot of that |
| 18:54.39 | Zalbag | ah |
| 18:54.42 | tzanger | their earlier cards were VERY problematic in that regard |
| 18:54.46 | E-bola | mm arent IRQ issues a user problem? |
| 18:54.56 | Corydon-w | Generally the IRQ problems are caused by the motherboard not being quite up to spec |
| 18:55.18 | tzanger | Corydon-w: that doesn't explain how I could yank a digium card and put a sangoma in the same slot and have no issues |
| 18:55.33 | tzanger | generally the IRQ problems were due to poor drivers |
| 18:55.48 | dlynes_laptop | Zalbag: sangoma's claim to fame isn't lack of irq problems; it's their ability to be able to share irq's |
| 18:56.01 | tzanger | and NMI issues and other goodies due to the specification allowing for two things to both claim they're right when they're not. :-) |
| 18:56.26 | dlynes_laptop | Zalbag: iow, their cards will work just fine in a machine with only one pci slot, where that irq is shared with a network controller |
| 18:56.37 | Corydon-w | Well, that's the issue with rfc2833 dtmf |
| 18:56.48 | tzanger | dlynes_laptop: I had a TDM430P in a system that shared that IRQ with one of the disk controllers and the network card... the box was also a network server and stored movies that I streamed up to the living room (diskless) PC |
| 18:56.58 | tzanger | I had *NO* IRQ issues at all with that particular card or system |
| 18:57.03 | dlynes_laptop | tzanger: cool |
| 18:57.07 | Zalbag | hmm |
| 18:57.32 | dlynes_laptop | tzanger: i had one in a 3U rack mount system that didn't share irqs, or anything and it was still a huge headache |
| 18:57.49 | Zalbag | so really the only difference between the 2 is just a name? |
| 18:57.49 | dlynes_laptop | tzafrir: I ended up pulling it out, replaced it with a sangoma, and zero issues |
| 18:58.00 | *** join/#asterisk Tili (n=tili@202.133.67.111) |
| 18:58.01 | tzanger | Zalbag: no, there are differences |
| 18:58.09 | tzanger | Zalbag: but from terms of what they do, no not really |
| 18:58.15 | russellb_ | Zalbag: well and that Digium is the company that pays a team of developers to work on Asterisk all day |
| 18:58.18 | tzanger | Zalbag: heh, some systems are nasty yes |
| 18:58.21 | dlynes_laptop | tzanger: ummm...not quite true |
| 18:58.22 | russellb_ | but you know, whatever |
| 18:58.24 | tzanger | russellb_: exactly |
| 18:58.27 | Tili | how can I limit access to guest account to specific IPs |
| 18:58.32 | Tili | in sip.conf |
| 18:58.35 | dlynes_laptop | tzanger: there is one thing sangoma will do that digium won't for the pri cards |
| 18:58.59 | tzanger | dlynes_laptop: for all intents and purposes the cards are more or less identical. They use the same Xilinx Spartan II and PCI access blocks |
| 18:58.59 | dlynes_laptop | tzanger: you can use the sangoma cards to fractionalize the pri into both voice and network |
| 18:59.14 | Corydon-w | dlynes_laptop: you can do that with Digium cards, too |
| 18:59.17 | russellb_ | dlynes_laptop: you can do that with digium cards! |
| 18:59.18 | dlynes_laptop | Corydon-w: oh? |
| 18:59.23 | Corydon-w | Yeah. |
| 18:59.32 | russellb_ | i hate when people speak out of ignorance and spread bogus rumors |
| 18:59.36 | dlynes_laptop | russellb_: ok, so why does nobody ever talk about that with digium cards? |
| 18:59.49 | dlynes_laptop | russellb_: they always tell peeps on here to buy adtran 850's instead |
| 18:59.52 | Flauto | hi dlynes |
| 18:59.57 | dlynes_laptop | heya FlatFoot |
| 19:00.00 | dlynes_laptop | erm Flauto |
| 19:00.03 | russellb_ | i don't know, people do it all the time |
| 19:00.05 | Flauto | hehe |
| 19:00.05 | Corydon-w | We talk about it all the time with people who need hybrid voice and data |
| 19:00.05 | russellb_ | ask anyone in support |
| 19:00.08 | Flauto | how are you doing |
| 19:00.15 | russellb_ | ugh. |
| 19:00.18 | dlynes_laptop | russellb_: ok, cool...thanks for the heads up |
| 19:00.36 | Corydon-w | Of course, some customers would rather have multiple T1's... |
| 19:00.39 | dlynes_laptop | russellb_: do you need special drivers to take advantage of it? |
| 19:00.49 | Corydon-w | dlynes_laptop: nope, default zaptel drivers |
| 19:01.00 | dlynes_laptop | Corydon-w: so how do you get the digium card to show up as a network card, then? |
| 19:01.01 | *** join/#asterisk perdue (n=perdue@66.118.69.58) |
| 19:01.03 | *** part/#asterisk ComputerWarm (n=donc@209.29.158.47) |
| 19:01.20 | Corydon-w | dlynes_laptop: nethdlc in zaptel.conf, use sethdlc |
| 19:01.29 | dlynes_laptop | ah...so that's what those utilities are for |
| 19:01.49 | dlynes_laptop | i thought it was for some kinda low level tweaking or something |
| 19:02.03 | Corydon-w | There's several different modes, depending upon how the provider sets up the data |
| 19:02.42 | Corydon-w | frame relay is how we do it here |
| 19:02.42 | Flauto | dlynes, i have been playing with trixbox these days. it is a pain in the ass but it does provide a few interesting functions |
| 19:03.00 | dlynes_laptop | Flauto: but nothing that vanilla asterisk can't provide as well |
| 19:03.09 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 19:03.09 | *** mode/#asterisk [+o anthm] by ChanServ |
| 19:03.17 | dlynes_laptop | Flauto: the only difference is that trixbox includes a bunch of macros that vanilla asterisk doesn't |
| 19:03.32 | Flauto | right |
| 19:03.36 | dlynes_laptop | Flauto: but you can't take those out of trixbox and retool them a bit and drop them into vanilla asterisk, too |
| 19:03.50 | *** join/#asterisk twisla (i=twisla@lutin.jard.in) |
| 19:03.59 | dlynes_laptop | s/can't/can/ |
| 19:04.29 | Flauto | i have a littel problem now with wakeup call |
| 19:04.54 | dlynes_laptop | i've been hanging out in #freepbx once in a while lately to see how vanilla and trixbox differ |
| 19:05.08 | Flauto | it does not play music on hold any longer after i upgrade the asterisk to 1.2.10 |
| 19:05.22 | tzanger | dlynes_laptop: you can do that with digium too |
| 19:05.25 | dlynes_laptop | Flauto: no moh at all? |
| 19:05.43 | dlynes_laptop | tzanger: can do what? the same thing Corydon-w and russellb have already said? |
| 19:05.55 | tzanger | dlynes_laptop: heh maybe, I lost track :-) |
| 19:05.58 | Flauto | ii use moh-native |
| 19:06.02 | *** join/#asterisk adorah (n=Administ@87.68.173.125.cable.012.net.il) |
| 19:06.04 | Flauto | and i did not have any problem |
| 19:06.20 | dlynes_laptop | yeah...moh native works fine on 1.2.10 |
| 19:06.35 | Flauto | it does |
| 19:06.45 | Flauto | but when it goes into wakeup |
| 19:06.45 | dlynes_laptop | Flauto: when you upgraded to 1.2.10 did you also recompile and reinstall format_mp3.so? |
| 19:07.09 | *** join/#asterisk eBody (n=ehernand@207.71.51.162) |
| 19:07.10 | Flauto | no |
| 19:07.13 | Flauto | how would i do it |
| 19:07.16 | dlynes_laptop | there's your mistake, probably |
| 19:07.20 | Flauto | woul dyou show me |
| 19:07.23 | dlynes_laptop | you need to reinstall that as well |
| 19:07.25 | eBody | my SIP calls are all choppy, but why would my music on hold be choppy and broken up also? |
| 19:07.36 | dlynes_laptop | Flauto: it's in asterisk-addons-1.2.3 |
| 19:07.48 | Flauto | i did recompile that |
| 19:07.57 | Flauto | i mean addons |
| 19:08.08 | Flauto | hold on, i will be back in a few minutes |
| 19:08.12 | dlynes_laptop | Flauto: just download that, then cd /usr/local/src/asterisk-addons-1.2.3/format_mp3 && make clean && make && make install |
| 19:08.14 | Flauto | don't write anything |
| 19:08.22 | dlynes_laptop | or wherever you untarred it |
| 19:08.44 | *** join/#asterisk Mercestes (n=Merceste@216.54.143.2) |
| 19:10.29 | sevard | dlynes_laptop: do you use a ticketing system? |
| 19:10.40 | dlynes_laptop | sevard: you mean a trouble reporting system? |
| 19:10.47 | sevard | sure |
| 19:10.53 | dlynes_laptop | sevard: bugs.digium.com? |
| 19:11.00 | sevard | heh |
| 19:11.03 | sevard | i mean in house |
| 19:11.08 | dlynes_laptop | bugzilla? |
| 19:11.16 | sevard | I was thinking of bugzilla |
| 19:11.19 | sevard | but it's a piece of shit. |
| 19:11.39 | dlynes_laptop | There's probably about 40 or 50 ticketing systems out there |
| 19:11.41 | brookshire | everything out there is a piece |
| 19:12.08 | sevard | dlynes_laptop: there are, that's why I was asking if you used one inhouse. |
| 19:12.31 | dlynes_laptop | not atm |
| 19:12.31 | brookshire | rt is good :) |
| 19:12.36 | dlynes_laptop | I haven't had time to set one up again |
| 19:12.37 | brookshire | it's all email based though |
| 19:12.43 | dlynes_laptop | I used to have one setup, however |
| 19:13.10 | sevard | I'm looking at staying away from an all email based system |
| 19:13.15 | sevard | like ticketsmith is |
| 19:13.25 | dlynes_laptop | the one we were using was Java-based |
| 19:13.29 | sevard | BATTS looks alright, but it's full cli |
| 19:13.34 | sevard | dlynes_laptop: ewie. |
| 19:13.39 | TripleFFFF | git a qyucjue |
| 19:13.42 | sevard | a perl+mysql ticket system would be tite |
| 19:13.43 | TripleFFFF | got a quickie |
| 19:13.44 | brookshire | email systems just work |
| 19:13.57 | brookshire | but yes.. somewhat unorganized |
| 19:13.57 | TripleFFFF | if i prune realtime peer user1 |
| 19:14.04 | dlynes_laptop | sevard: we were a Java shop, so it only made sense to use a Java-based solution |
| 19:14.06 | TripleFFFF | it seems it still doesnt prune the memory |
| 19:14.11 | TripleFFFF | and i CANT just reload sip |
| 19:14.17 | TripleFFFF | sine that will reload all |
| 19:14.25 | TripleFFFF | anyway to say reload sip peer XXX ? |
| 19:14.29 | TripleFFFF | and WTF cant we |
| 19:14.30 | TripleFFFF | l;) |
| 19:14.45 | dlynes_laptop | sevard: have you tried Mantis? |
| 19:14.52 | brookshire | realtime ? |
| 19:14.54 | sevard | dlynes_laptop: I haven't even heard of it, oss? |
| 19:15.01 | TripleFFFF | yes |
| 19:15.09 | TripleFFFF | i changed a context for uiser1 |
| 19:15.11 | dlynes_laptop | sevard: www.mantisbt.org |
| 19:15.20 | dlynes_laptop | sevard: it's what bugs.digium.com runs on |
| 19:15.30 | TripleFFFF | and now when i dfial it wont actualy do it.. altough.. it does show new exten in the sip show peer user1 |
| 19:15.33 | TripleFFFF | it doesnt use it |
| 19:15.42 | dlynes_laptop | sevard: php/mysql |
| 19:15.54 | brookshire | triple: why don't you just do a reload every 15 minutes? or does it cause performance issues? |
| 19:16.03 | TripleFFFF | not good |
| 19:16.06 | TripleFFFF | reload lol |
| 19:16.14 | sevard | dlynes_laptop: but would it work well for tracking customer problems? |
| 19:16.18 | TripleFFFF | i got 90+ calls simult on that box |
| 19:16.43 | sevard | whoa bugs.digium.com is a bit complicated haha |
| 19:16.46 | dlynes_laptop | sevard: would you be using it inhouse, or having the customer updating it? |
| 19:17.04 | TripleFFFF | the xtens dont even push to the asterisk.. i mean wven sip debug on i ont see 404 or something.. |
| 19:17.14 | dlynes_laptop | sevard: yeah, but mantis is quite powerful |
| 19:17.24 | dlynes_laptop | sevard: really great search/filtering abilities |
| 19:17.25 | brookshire | RT! :) |
| 19:18.05 | TripleFFFF | what does prune do exaclty ? |
| 19:18.08 | TripleFFFF | ~define prune |
| 19:18.15 | jbot | Use ~dict for definitions. |
| 19:18.15 | TripleFFFF | ~prune |
| 19:18.19 | TripleFFFF | ~dict prune |
| 19:18.29 | sevard | dlynes_laptop: probably mostly inhouse |
| 19:18.31 | TripleFFFF | ~dict prune 2 |
| 19:18.38 | sevard | dlynes_laptop: but maybe a forum for a customer somewhere |
| 19:18.54 | TripleFFFF | ~dict prune |
| 19:19.16 | TripleFFFF | ~dict |
| 19:19.17 | dlynes_laptop | sevard: try http://sf.net/projects/ezt |
| 19:19.23 | TripleFFFF | hmm |
| 19:19.39 | TripleFFFF | ~dict 2 prune |
| 19:19.48 | dlynes_laptop | sevard: it might be alpha, but because it's so simplistic, I doubt it's got many bugs, if any |
| 19:19.52 | TripleFFFF | ~dict 7 prune |
| 19:20.02 | sevard | dlynes_laptop: looks neat |
| 19:20.05 | TripleFFFF | well i wont go trough whole list |
| 19:20.07 | TripleFFFF | ~dict 3prune |
| 19:20.13 | TripleFFFF | ~dict 3 prune |
| 19:20.22 | TripleFFFF | ~dict 4 prune |
| 19:20.25 | Flauto | Aug 8 14:20:06 WARNING[26162]: chan_sip.c:2561 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 64/64) |
| 19:20.31 | Flauto | dy |
| 19:20.35 | Flauto | this is what i got |
| 19:20.37 | dlynes_laptop | FlatFoot: codec mismatch |
| 19:20.42 | dlynes_laptop | stupid autocomplete |
| 19:20.44 | TripleFFFF | man ... this sucks lol.. |
| 19:20.47 | dlynes_laptop | Flauto: fix your name :) |
| 19:21.40 | Flauto | dlynes, what do i need to do |
| 19:21.51 | dlynes_laptop | TripleFFFF: you already got it with ~dict 3 prune |
| 19:22.07 | TripleFFFF | ; to clear of useless material; |
| 19:22.12 | TripleFFFF | but it doesnt do it |
| 19:22.26 | dlynes_laptop | TripleFFFF: what are you trying to do? write an irc script? |
| 19:22.34 | TripleFFFF | no |
| 19:22.56 | TripleFFFF | i got a sip user realtime i changed the context and * wont use new one.. altouh it does shoow it in sip show peer user1 |
| 19:23.03 | TripleFFFF | i see new context but tis not using it |
| 19:23.26 | dlynes_laptop | when you type sip show peers |
| 19:23.33 | dlynes_laptop | it doens't show up either? |
| 19:24.24 | TripleFFFF | yes it doews |
| 19:24.30 | TripleFFFF | and sip show peer user1 |
| 19:24.33 | TripleFFFF | show right context |
| 19:24.37 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 19:24.37 | *** mode/#asterisk [+o anthm] by ChanServ |
| 19:25.56 | TripleFFFF | found it |
| 19:25.58 | TripleFFFF | nice crap |
| 19:26.25 | TripleFFFF | sip prune realtime USERNAME |
| 19:26.31 | TripleFFFF | so it prunes both USER and PEER |
| 19:26.42 | TripleFFFF | if you put sip prune realtime peer USERNAME |
| 19:26.45 | TripleFFFF | then it wont do crap |
| 19:27.20 | TripleFFFF | hmm unless peer is all the peer ionfo ( ip , port ) etc and user is the local settings (codec, context ) etc |
| 19:27.21 | TripleFFFF | ???? |
| 19:27.27 | TripleFFFF | wich should be ex,plaiuned somehwere.. |
| 19:29.27 | Flauto | dlynes, what do i need to do to fix this problem |
| 19:29.42 | Flauto | even though, i have no problem with music on hold itself |
| 19:29.51 | Flauto | codec mismatch |
| 19:30.28 | Flauto | i always have ulaw as the first choice on all my extensions |
| 19:30.45 | Flauto | should i try to use only ulaw? |
| 19:31.39 | *** join/#asterisk jwsh (n=jwsh@c-65-96-209-153.hsd1.ma.comcast.net) |
| 19:32.41 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 19:33.03 | jwsh | anyone in here got any experience with the polycom 501s ? |
| 19:33.20 | *** join/#asterisk networkjedi (n=networkj@12.36.212.60) |
| 19:33.21 | Zalbag | I do |
| 19:33.30 | Zalbag | I am not expert though :( |
| 19:33.36 | jwsh | well, I'm trying to upgrade the firmware and it refuses to play nice |
| 19:33.46 | Zalbag | Well what have you done so far? |
| 19:34.45 | jwsh | Well, I dropped in the new bootrom.ld and sip.ld files and rebooted, it goes into a reboot loop.. I'm pretty sure I've tracked the problem down to the fact that it's not downloading the whole file |
| 19:35.05 | jwsh | it truncates the download at 21900 bytes every time |
| 19:35.56 | *** join/#asterisk BeBrA (n=abc@host56-241.pool8248.interbusiness.it) |
| 19:36.39 | BeBrA | Received message to <sip:123@100> from 100 <sip:100@100>;tag=6336, dropped it... |
| 19:36.45 | jwsh | but the file is actually 3125621 bytes long |
| 19:36.50 | BeBrA | how can I handle SMS on asterisk? |
| 19:37.04 | Zalbag | Your sip.ld file? |
| 19:37.18 | jwsh | bootrom.ld, actually |
| 19:37.24 | Zalbag | ah |
| 19:38.06 | Zalbag | I don't have the bootrom.ld file |
| 19:38.07 | Zalbag | :( |
| 19:38.11 | Dorphalsig | . |
| 19:38.29 | jwsh | I'm pretty sure it does the same thing with the sip.ld file too |
| 19:39.13 | ManxPower | no. bootrom.ld is the bootloader (same for all protocols) and sip.ld is the sip firmware. |
| 19:39.23 | Zalbag | right |
| 19:39.42 | *** join/#asterisk [hC] (n=hardcore@190.10.9.191) |
| 19:39.49 | Zalbag | hmm... |
| 19:39.53 | TripleFFFF | btw anyone got xten registry file ? i mean anyway to get xten to conf via a reg file ? |
| 19:40.04 | Zalbag | jwsh, did you setup a TFTP server with read/write? |
| 19:40.06 | ManxPower | older phones do not like version 3.x bootroms |
| 19:40.25 | x86 | if someone were to recommend a Cisco IP phone for use with Asterisk, which one would they recommend? |
| 19:40.51 | x86 | I'm looking at the 7970, which I don't think can be flashed to run a SIP stack, but Asterisk's SCCP support is decent, right? |
| 19:41.01 | TripleFFFF | 7960 |
| 19:41.10 | x86 | i wanted the color screen though |
| 19:41.15 | Zalbag | it's a shot in the dark but maybe it starts to pull the bootrom and at the same time write to the server...maybe if you didn't set the ability to write? |
| 19:41.25 | Zalbag | of course I probably sound like an idiot right nwo |
| 19:41.27 | Zalbag | now* |
| 19:41.55 | jwsh | yeah, it can write it's log files |
| 19:41.58 | Zalbag | ok |
| 19:42.09 | Zalbag | hmm...dunno then. I haven't had a problem |
| 19:42.23 | jwsh | if I move the bootrom.ld away, it will try to dl the sip.ld |
| 19:42.32 | jwsh | but it also quits on me |
| 19:42.36 | jwsh | but at a different place |
| 19:44.00 | jwsh | 4,863,260 bytes |
| 19:44.58 | diablopico | i am using openh323 on asterisk 1.0.7 when i make a call it works , problem is a 20 second delay before convesation can start. once started , it is normal 2 way conversation , .....can anyone help ? |
| 19:45.22 | jwsh | also, whenever the phone boots, it says "could not contact boot server using existing configuration" without ever trying to actually contact anything |
| 19:45.37 | jwsh | it doesn't try to download until after it does the "loading application...." bit |
| 19:46.41 | x86 | TripleFFFF: ? |
| 19:46.44 | Zalbag | jwsh, you've double checked your IP config on the phone right? |
| 19:47.23 | Zalbag | try pinging the phone |
| 19:47.30 | Zalbag | make sure it's actually reachable from the network |
| 19:47.39 | jwsh | Zalbag: it works fine, it downlads new configs, uploads log files |
| 19:47.43 | cytrak | anyone has setup voiceone ? |
| 19:47.45 | Zalbag | oh wait |
| 19:47.46 | jwsh | I can make calls and everything |
| 19:47.48 | Zalbag | duh :P |
| 19:47.57 | jwsh | it just won't upgrade the firmware |
| 19:47.57 | cytrak | I'm trying to set this up but i get nothing but problems |
| 19:48.07 | Zalbag | Hmmmm |
| 19:48.47 | *** join/#asterisk apardo (n=apardo@87.217.147.151) |
| 19:49.44 | TripleFFFF | man |
| 19:49.53 | Zalbag | What's the current version of your bootrom/sip and what are you trying to upgrade to? |
| 19:50.36 | jwsh | I'm trying to upgrade to the latest greatest (previously I had tried to upgrade to an older version and had similar results) |
| 19:50.42 | x86 | TripleFFFF: no answer still? |
| 19:50.44 | *** join/#asterisk cian (n=cian@cian.ws) |
| 19:50.47 | jwsh | it is currently running 1.6.2.0041 |
| 19:51.11 | TripleFFFF | in pn phone |
| 19:52.10 | jwsh | (that's the sip version - the boot rom is presently 2.6.1.0003) |
| 19:53.17 | jwsh | ManxPower: when you say "older phones do not like version 3.x bootroms" what exactly do you mean by "older phones" |
| 19:53.33 | ManxPower | jwsh, non x01 phones |
| 19:53.45 | Zalbag | jwsh said he has the 501 though? |
| 19:54.14 | quid2478 | anyone used SIPP (load testing) with *? |
| 19:54.27 | eKo1 | Yes. |
| 19:54.29 | ManxPower | Zalbag, I dunno, I can't read everythig on the channel |
| 19:55.46 | jwsh | ManxPower/Zalbag - yeah it's a 501 |
| 19:55.50 | *** join/#asterisk dasenjo_ (n=dasenjo@63.245.86.247) |
| 19:55.54 | TripleFFFF | voipsupply assholel around ? |
| 19:56.12 | TripleFFFF | x86 what was it ? |
| 19:56.15 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 19:56.15 | *** mode/#asterisk [+o anthm] by ChanServ |
| 19:56.20 | TripleFFFF | man you guys STAY away from voipsupply |
| 19:56.40 | quid2478 | ekor1: I've just been trying to play around with it... * receives all the invites and everything but does not seem to setup any RTP stream |
| 19:56.48 | TripleFFFF | they sent an ata to my client in canada.. and next morning they said.. sorry we sent rush.. now they billing me 260$ SHIPPING... no fuckign wa y im paying these assholes |
| 19:56.51 | Zalbag | jwsh, honestly I don't know what it's not working for you. I just checked the versions on my phone and I have the same bootrom as you but a new SIP version |
| 19:56.56 | TripleFFFF | they cant even use fedex right |
| 19:56.57 | quid2478 | FFFF: Oh yeah... thre shipping to Canada is ridiculous |
| 19:56.59 | Zalbag | s/what/why |
| 19:57.04 | TripleFFFF | anywher |
| 19:57.07 | x86 | TripleFFFF: i wanted the color screen of the 7970 or 7971 |
| 19:57.12 | TripleFFFF | got a shipment in usa same thing |
| 19:57.13 | TripleFFFF | 260$ |
| 19:57.17 | TripleFFFF | is this a scam ? |
| 19:57.21 | TripleFFFF | im reporting them |
| 19:57.23 | TripleFFFF | anyhow |
| 19:57.32 | TripleFFFF | x86 no idea .. |
| 19:57.45 | quid2478 | And they try and appear to look like they are in Canad (vopistore.ca) I think, onylthe fine print says they are shipping from the USA |
| 19:57.54 | TripleFFFF | not the point |
| 19:57.59 | TripleFFFF | he shipped and charge 45$ |
| 19:58.02 | jwsh | Zalbag: well, thanks for trying anyway.. I bought this phone from voipsupply.com actually.. they've been 0 help in getting it working |
| 19:58.05 | TripleFFFF | but he did error and shipped overnight |
| 19:58.12 | TripleFFFF | and SAID HIMSELF sorry i made an error |
| 19:58.22 | TripleFFFF | NOW im getting 2 bills of 260$ USD for his ERRORS |
| 19:58.27 | jwsh | so I guess they have two unhappy customers here |
| 19:58.32 | TripleFFFF | 540$ for 2 ata's |
| 19:58.40 | TripleFFFF | that fraud |
| 19:58.41 | quid2478 | TripleFFFF: Depending on what kind oF ATA or phones you need, consider Voxilla... although NCIX.com kills anybody on their prices for Linksys phones |
| 19:58.44 | TripleFFFF | and im reporting them |
| 19:58.54 | TripleFFFF | ncix is my new provider |
| 19:58.56 | TripleFFFF | btw |
| 19:59.08 | TripleFFFF | now they invoced my customers directly for 260$ each lol |
| 19:59.17 | quid2478 | FFFF: It's too bad NCIX doesn't sell Sangoma |
| 19:59.25 | TripleFFFF | i told them not to pay .. but im considering suing them .. since im loosing these 2 customers .. |
| 19:59.36 | TripleFFFF | them=vs |
| 19:59.42 | TripleFFFF | anyhow |
| 19:59.49 | TripleFFFF | they wont last long |
| 20:00.20 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 20:01.58 | x86 | TripleFFFF: there is a Cisco ATA for about $480 for one |
| 20:02.01 | *** join/#asterisk kisoku (n=msf@dns1.countersiege.com) |
| 20:02.10 | x86 | TripleFFFF: 188 I think is the model |
| 20:02.15 | x86 | just saw it on eBay |
| 20:02.40 | kisoku | mayeb someone here can help me figure this out |
| 20:02.59 | kisoku | asterisk machine as sip client behind a natr router running siproxd |
| 20:03.52 | kisoku | when I put outboundproxy=natrouterip in my sip.conf entry for my sip provider calls from outside work |
| 20:04.01 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 20:04.01 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 20:04.09 | kisoku | when I remove it outbound calls work |
| 20:04.17 | kisoku | how can I get both working ? |
| 20:06.47 | TripleFFFF | ?? |
| 20:06.51 | TripleFFFF | [16:02] x86: TripleFFFF: there is a Cisco ATA for about $480 for one |
| 20:08.30 | TripleFFFF | ncix has the Sipura SPA-2100 Analog Telephone VoIP Adapter |
| 20:08.32 | TripleFFFF | onlyu |
| 20:08.57 | TripleFFFF | and they do sangoma |
| 20:08.57 | TripleFFFF | SG108D |
| 20:08.57 | TripleFFFF | <PROTECTED> |
| 20:08.57 | TripleFFFF | Special Order, Usually 1-2 weeks. $4,494.29 |
| 20:11.31 | x86 | wow $5,000 for a single PCI card |
| 20:11.37 | x86 | that's totally ouch ;) |
| 20:11.45 | x86 | probably worth it, but still ouch ;) |
| 20:11.59 | mocker | A PCI card that can handle 192 phone lines! :) |
| 20:12.57 | file | phone lines? that is soooooo 20th century |
| 20:13.49 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-59.dslextreme.com) |
| 20:14.55 | hmmhesays | heh |
| 20:15.08 | hmmhesays | that makes me want to hurl |
| 20:15.11 | hmmhesays | 192 phone lines |
| 20:15.19 | Strom_C | 192 analog lines? |
| 20:15.25 | hmmhesays | anthm: you around? |
| 20:15.34 | *** join/#asterisk [koss] (i=koss@adsl-75-36-15-21.dsl.bcvloh.sbcglobal.net) |
| 20:15.37 | anthm | yes? |
| 20:16.17 | *** join/#asterisk fiber0pti (n=John@207.114.199.107) |
| 20:16.34 | Un1x | anyone have a DID from DIDWW.com? |
| 20:16.44 | fiber0pti | can RealTime use a MySQL database on another server? |
| 20:17.35 | Nugget | access the remote server over the internet and you can have the holy trifecta of unreliability! |
| 20:17.38 | Un1x | i have this in my extensions.conf |
| 20:17.38 | Un1x | [from-didww] |
| 20:17.38 | Un1x | exten => s,1,Dial(Zap/1,15) |
| 20:17.42 | Un1x | but it wont ring on my phone :( |
| 20:18.00 | ManxPower | Un1x, Why do you think calls will hit exten => s ? |
| 20:18.10 | *** join/#asterisk m4rkl4r (n=Administ@c-24-129-119-53.hsd1.fl.comcast.net) |
| 20:18.22 | Un1x | Manxpower then can you tell me how to change it please.. |
| 20:18.24 | ManxPower | calls only hit extension s when asterisk does not know what the destination umber is. |
| 20:18.29 | quid2478 | fiber0opi: I've only played with mySQL and * a bit, but I would assume so |
| 20:18.31 | ManxPower | Un1x, what is your DID? |
| 20:18.41 | [koss] | any ideas where i can find a local asterisk "expert" to help configure a PBX ? |
| 20:18.43 | ManxPower | Heck, first what is the ERROR MESSAGE ON THE CONSOLE? |
| 20:18.50 | quid2478 | koss: check the wiki |
| 20:18.54 | ManxPower | [koss], define "local" |
| 20:18.55 | quid2478 | ~wiki |
| 20:19.00 | quid2478 | hmm maybe not |
| 20:19.01 | Un1x | 01192422008213 |
| 20:19.03 | m4rkl4r | question: can call bridging be set up so that if the original caller hangs up, everyone else that has been bridged in stays on the line? |
| 20:19.17 | Un1x | ManxPower: 01192422008213 |
| 20:19.21 | [koss] | ManxPower: local to cleveland, ohio |
| 20:19.21 | ManxPower | Un1x, what errors do you get on the console when a call comes in? |
| 20:19.46 | ManxPower | m4rkl4r, that's call conferencing on zaptel |
| 20:20.09 | Un1x | ManxPower: it destroys the call it comes in into the server fine, but when it supposed to send the call to specifc extension or something it doesn't do that.. anyway to fix it basiclty the extensions.conf |
| 20:20.12 | *** join/#asterisk charles___ (n=charles@fw.invosat.com) |
| 20:20.15 | charles___ | hey Guys |
| 20:20.24 | charles___ | something wrong with my X chat |
| 20:20.25 | charles___ | but anyway |
| 20:20.35 | charles___ | anyone got Cisco with SIP dial peers ? |
| 20:20.36 | Un1x | ManxPower: this is what my current stuff looks like http://www.didww.com/support/index.php?_m=knowledgebase&_a=viewarticle&kbarticleid=3&nav=0,1 |
| 20:20.41 | Un1x | i added the sip.conf part |
| 20:20.42 | m4rkl4r | we are using sip, though |
| 20:20.50 | Un1x | but it tells me something completely different tho.... |
| 20:20.50 | m4rkl4r | ManxPower: we are using sip, though |
| 20:20.52 | Un1x | for extensionc.conf |
| 20:21.07 | ManxPower | Un1x, you are not going to give me the console output are you? |
| 20:21.13 | charles___ | I'm thinking about if there is a way to a cisco send a call to a Gateway and send user and password ? |
| 20:21.14 | ManxPower | m4rkl4r, that would depend on the SIP device. |
| 20:21.33 | hmmhesays | haha i suppose I should watch the chat when I ask if someone is around |
| 20:21.41 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 20:21.42 | anthm | ya |
| 20:22.06 | hmmhesays | I think it was you who was talking about reloading only pieces of the asterisk dialplan on a per call basis |
| 20:22.38 | hmmhesays | no realtime, just a dynamic dp that is generated on a per call basis |
| 20:22.39 | ManxPower | Hey, I'd be happy if the people I'm trying to help actually listened to me. Yeah, I know that's way too much to ask. |
| 20:22.52 | ManxPower | Un1x, if I have to ask a 3rd time you will be put on my /ignore list. |
| 20:22.59 | anthm | ya that's what we do |
| 20:23.13 | charles___ | hey guys |
| 20:23.23 | charles___ | Anyone got Cisco's with password on the dial peers ? |
| 20:23.32 | hmmhesays | care to share how you accomplish said feat? |
| 20:23.59 | Un1x | ManxPower: http://pastebin.ca/122624 |
| 20:24.09 | Un1x | i was getting the console output it's like 3 pages to copy and paste :P |
| 20:24.38 | Flauto | Aug 8 15:24:07 WARNING[26485]: chan_sip.c:2561 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 64/64) |
| 20:24.42 | ManxPower | Un1x, turn off sip debug. try another call then paste the console output. |
| 20:24.42 | charles___ | I mean specify auth on the peer |
| 20:24.46 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 20:25.02 | charles___ | Trying to connect a Cisco to an Asterisk with password auth |
| 20:25.09 | charles___ | not using regular peers |
| 20:25.13 | ManxPower | Flauto, you are trying to use G729 but do not have a G729 license |
| 20:25.18 | Un1x | ManxPower: if i turn off sip debug how wil i get the console output? |
| 20:25.25 | ManxPower | charles___, I don't think you can do that. |
| 20:25.44 | Flauto | manxpower, what should i do |
| 20:25.48 | charles___ | ManxPower: that's what I'm trying to figure out. need to have that |
| 20:25.54 | ManxPower | Un1x, you will get all the non sip info, which is what I want right now. |
| 20:26.01 | Un1x | oh okay |
| 20:26.07 | ManxPower | charles___, Cisco routers do not support as far as I know. |
| 20:26.21 | ManxPower | Flauto, disallow=g729 in sip.conf |
| 20:26.38 | Un1x | ManxPower: if i turn off debuging nothing happens heres a look |
| 20:26.38 | Un1x | Canucks*CLI> sip no debug |
| 20:26.39 | Un1x | SIP Debugging Disabled |
| 20:26.39 | Un1x | Canucks*CLI> |
| 20:26.51 | ManxPower | Un1x, so you do not get a "rejecting call" message |
| 20:26.57 | Un1x | no i do not... |
| 20:27.04 | ManxPower | you do not get a "cannot find extension X in context y" message? |
| 20:27.16 | *** join/#asterisk Assid (i=assid@203.115.83.213) |
| 20:27.21 | ManxPower | Un1x, put the out put of "sip show peers" and "sip show registry" on pastebin.ca |
| 20:27.22 | Un1x | nopes nothing i got into console doing `asterisk -r' but no msgs at all.. |
| 20:27.27 | Un1x | ok |
| 20:28.00 | *** join/#asterisk ESCulapio__ (n=ESCulapi@200.88.44.66) |
| 20:28.12 | ManxPower | Un1x, try "asterisk -rvvvddd" |
| 20:28.13 | hmmhesays | if not I won't bug you about it again |
| 20:28.51 | diablopico | can anyone help with an openh323 problem ? |
| 20:29.00 | ManxPower | Un1x, most of the time you do not need to wade thru 5 billion lines of SIP debug output to solve this stuff. |
| 20:29.50 | *** join/#asterisk mike800 (n=Michael@wsip-70-183-59-19.oc.oc.cox.net) |
| 20:30.05 | Un1x | manxpower pastebin wopnt let me paste domain names :// |
| 20:30.32 | Un1x | manxpower; mind if i paste you the 4 lines? |
| 20:30.38 | Un1x | there very small 4 lines |
| 20:31.02 | ManxPower | If you don't get kickbanned then you know it's OK. |
| 20:31.03 | TripleFFFF | in default (domain 192.168.1.7) |
| 20:31.09 | Un1x | ok |
| 20:31.13 | TripleFFFF | hmm why is this trying to find in a domain ? |
| 20:31.13 | Un1x | Canucks*CLI> sip show peers |
| 20:31.13 | Un1x | Name/username Host Dyn Nat ACL Port Status |
| 20:31.14 | Un1x | didww-in 212.150.36.116 5060 Unmonitored |
| 20:31.14 | Un1x | splitinfinity/10085 38.96.4.15 5060 Unmonitored |
| 20:31.14 | Un1x | 2 sip peers [2 online , 0 offline] |
| 20:31.14 | TripleFFFF | lol |
| 20:31.14 | Un1x | Canucks*CLI> |
| 20:31.38 | ManxPower | Un1x, and "sip show registry"? |
| 20:31.49 | Un1x | Canucks*CLI> sip show registry |
| 20:31.50 | Un1x | Host Username Refresh State |
| 20:31.50 | Un1x | voipgw1.splitinfinity.net:5060 10085 105 Registered |
| 20:31.50 | Un1x | Canucks*CLI> |
| 20:32.18 | ManxPower | Un1x, try it again, but with connecting to the console as "asterisk -rvvvddd" |
| 20:32.24 | ManxPower | no sip debug, try a call |
| 20:32.46 | mike800 | anyone wanna help me setup mysql cdr? |
| 20:33.01 | Un1x | Manxpower just did doesn't show anything... |
| 20:33.04 | ManxPower | mike800, not even if you wanted to pay me. |
| 20:33.14 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 20:33.24 | mike800 | hahaha |
| 20:33.36 | mitcheloc | ManxPower: at least he isn't using aah and he is trying to do it from scratch |
| 20:33.38 | Dr-Linux | aww |
| 20:33.48 | ManxPower | mitcheloc, OH! Can't help him with that. |
| 20:33.52 | Dr-Linux | what was the joke, please repeat again, i wanna laugh :S |
| 20:33.55 | mike800 | whoa- mitcheloc, you're the one who made Snap |
| 20:34.11 | *** join/#asterisk adker (n=adker@74-33-205-58.br1.glv.ny.frontiernet.net) |
| 20:34.14 | mike800 | i just installed your software and its amazing :-) |
| 20:34.15 | mitcheloc | heh, yes |
| 20:34.17 | diablopico | mike800 have you looked on info-voip.org there is full documentation on setting up mysql cdrs |
| 20:34.19 | mitcheloc | thanks :) |
| 20:34.20 | ManxPower | Un1x, do you have a /etc/asterisk/logger.conf file? |
| 20:34.36 | Un1x | yes |
| 20:34.44 | mike800 | diablopico, I have already read it, its just my first time, so I wanna make sure I do it right |
| 20:35.16 | ManxPower | Un1x, rename it to /etc/asterisk/logger.conf.wont.mess.things.up.anymore.will.you |
| 20:35.33 | ManxPower | then in the asterisk console type "logger restart" |
| 20:35.43 | ManxPower | or "logger reload" and try another call to your DID |
| 20:36.02 | diablopico | mike800: what questions do you have , i use it successfully at this time with a* realtime |
| 20:36.04 | Corydon-w | "logger rotate"? |
| 20:36.15 | Un1x | lol |
| 20:36.30 | ManxPower | Corydon-w, honestly I don't care if the logs are rotated or not 8-) |
| 20:36.53 | Un1x | Canucks*CLI> logger reload |
| 20:36.53 | Un1x | Asterisk Event Logger restarted |
| 20:36.53 | Un1x | Asterisk Queue Logger restarted |
| 20:36.53 | Un1x | Canucks*CLI> |
| 20:36.56 | *** join/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
| 20:37.08 | Un1x | Manxpower; done... |
| 20:37.10 | ManxPower | Un1x, still no output? |
| 20:37.36 | TripleFFFF | Looking for 17601231234 in default (domain 192.168.1.7) |
| 20:37.36 | TripleFFFF | Reliably Transmitting (NAT) to 192.168.1.79:5060: |
| 20:37.36 | TripleFFFF | SIP/2.0 404 Not Found |
| 20:37.50 | Un1x | Manxpower one sec let me make a call :D |
| 20:37.51 | TripleFFFF | ok so.. why isnt it finding the extension .... |
| 20:38.16 | Dr-Linux | TripleFFFF: it's finding the extension in "default" context |
| 20:38.25 | TripleFFFF | well |
| 20:38.32 | TripleFFFF | why the 404 ? |
| 20:38.34 | Dr-Linux | TripleFFFF: make sure that extension exists in "default" conetxt |
| 20:38.52 | TripleFFFF | exten => _1NXXNXXXXXX |
| 20:38.52 | TripleFFFF | exten => _XXXXXXXXXX |
| 20:38.52 | TripleFFFF | exten => _X., |
| 20:38.53 | Dr-Linux | TripleFFFF: bcoz it didn't find the extension in "default" context |
| 20:38.55 | TripleFFFF | exten => s, |
| 20:39.00 | TripleFFFF | exten => 17601231234 |
| 20:39.05 | TripleFFFF | got al those to match agains |
| 20:39.06 | TripleFFFF | so... |
| 20:39.16 | TripleFFFF | unless im missing something |
| 20:39.19 | ManxPower | TripleFFFF, there should be a NON sip deb ug message |
| 20:39.21 | TripleFFFF | it should match on of them. |
| 20:39.36 | TripleFFFF | i got sip debug on |
| 20:39.40 | Dr-Linux | TripleFFFF: maybe you are seding some prifix digits ? and not striping them? |
| 20:39.48 | ManxPower | I don't know what everyone's fetish for sip debug is. 90% of sip problems can be solved without ever looking at a sip debug |
| 20:39.51 | TripleFFFF | _1NXXNXXXXXX |
| 20:39.56 | TripleFFFF | should match that one then .. its not |
| 20:40.23 | Flauto | manpower, the problem is that when use wakeup, wakeup calls my extension and put musiconehold, then, it is showing |
| 20:40.23 | Flauto | Aug 8 15:31:06 WARNING[26554]: chan_sip.c:2561 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 64/64) |
| 20:40.37 | Flauto | when i just dial musiconhold, it is working |
| 20:40.44 | ManxPower | Dou do realize that _1NXXNXXXXXX will never match 17601231234 right? |
| 20:40.54 | Flauto | someone was telling me that it is codec mis match |
| 20:41.00 | TripleFFFF | ah BUT.. exten => 17601231234 |
| 20:41.01 | TripleFFFF | will |
| 20:41.01 | ManxPower | FlatFoot, did you put disallow=g729 in sip.conf? |
| 20:41.05 | Flauto | but the question is how can i fix it |
| 20:41.06 | TripleFFFF | and still doesnt |
| 20:41.15 | ManxPower | TripleFFFF, I do not believe it is looking in context default. |
| 20:41.19 | Flauto | i tried, and it really did not do anything |
| 20:41.26 | Flauto | i got the same thing |
| 20:41.39 | Dr-Linux | TripleFFFF: it's lookign in "default" context, i'm sure |
| 20:41.58 | Dr-Linux | look here: |
| 20:41.59 | Dr-Linux | Looking for 17601231234 in default (domain 192.168.1.7) |
| 20:42.21 | TripleFFFF | yes |
| 20:42.28 | TripleFFFF | but not matching for some ODD REASON |
| 20:42.39 | TripleFFFF | latst trunk |
| 20:42.46 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:42.48 | ManxPower | Dr-Linux, no, that is a SIP debug message saying it's looking for 17601231234 in the default SIP domain of "192.168.1.7" |
| 20:42.51 | Dr-Linux | TripleFFFF: i have same setup as you are trying .. so odd to me |
| 20:43.47 | ManxPower | that's why turning off sip debug is SO important, you can become confused about some of the messages. |
| 20:43.47 | *** join/#asterisk DrkShdw (n=DrkShdw@unaffiliated/drkshdw) |
| 20:43.58 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 20:43.58 | *** mode/#asterisk [+o mog] by ChanServ |
| 20:44.01 | ManxPower | TripleFFFF, do you get any console messages with sip debug turned off |
| 20:44.06 | Dr-Linux | ManxPower: but sir, whenever i configure my sip trunk, and send a call, and i get the same message as he is getting, and i see the context and place my extension over there |
| 20:44.09 | TripleFFFF | nope |
| 20:44.41 | ManxPower | TripleFFFF, do you have a /etc/asterisk/logger.conf? |
| 20:44.48 | Dr-Linux | ManxPower: only difference is that, i use "public" context and he is using "default" |
| 20:44.57 | TripleFFFF | yes |
| 20:45.36 | ManxPower | TripleFFFF, do this: put in context=INVALID in sip.conf [general] then put the correct context= line in the [section] for the incoming device. |
| 20:45.41 | ManxPower | do a reload, see what happens. |
| 20:45.45 | Dr-Linux | TripleFFFF: your both boxes are on same subnet? |
| 20:45.55 | TripleFFFF | tyes |
| 20:46.15 | *** join/#asterisk wunderkin (n=wunderki@216-19-202-6.getnet.net) |
| 20:46.19 | Un1x | ManxPower: nothing still |
| 20:46.19 | ManxPower | Dr-Linux, so your messages say Looking for 17601231234 in public (domain 192.168.1.7) or something similar |
| 20:46.21 | Dr-Linux | i see |
| 20:46.37 | Dr-Linux | ManxPower: yes sir |
| 20:46.51 | Dr-Linux | correct |
| 20:47.23 | Un1x | ManxPower: nothing still |
| 20:47.51 | Dr-Linux | ManxPower: but i suspect he is sending something, that already exists here . i mean "callerid" |
| 20:47.57 | ManxPower | Un1x, put your sip.conf up afain |
| 20:48.12 | *** join/#asterisk _GiGi_ (i=gigi@disc.more.pl) |
| 20:48.24 | Un1x | it's same as the one i showed u.. |
| 20:48.27 | Dr-Linux | TripleFFFF: what's your incoming box? |
| 20:48.37 | ManxPower | Un1x, I did not save the URL |
| 20:48.39 | Un1x | [didww-in] |
| 20:48.39 | Un1x | host=212.150.36.116 |
| 20:48.39 | Un1x | dtmfmode=inband |
| 20:48.39 | Un1x | dtmf=inband |
| 20:48.39 | Un1x | type=peer |
| 20:48.39 | Un1x | context=from-didww |
| 20:48.41 | Un1x | insecure=very |
| 20:48.43 | Un1x | nat=never |
| 20:48.45 | Un1x | allow=all |
| 20:49.25 | ManxPower | Un1x, perhaps you don't realize that I need the [general] section too |
| 20:49.44 | Dr-Linux | ManxPower: i have 5 servers connected in the same way, also one SIP provider, but i'm still talking here, bcoz i wanna learn. ofcos you are right |
| 20:51.44 | Un1x | ManxPower: http://pastebin.ca/122675 |
| 20:52.46 | Rowter | <PROTECTED> |
| 20:53.37 | TripleFFFF | did that no change |
| 20:53.39 | Un1x | ManxPower:? |
| 20:54.34 | TripleFFFF | AH! |
| 20:54.35 | TripleFFFF | in INVALID (domain 192.168.1.7) |
| 20:54.40 | TripleFFFF | its looking in invalud |
| 20:54.46 | TripleFFFF | how can i auth by ip ? |
| 20:54.51 | TripleFFFF | [IP] |
| 20:54.51 | TripleFFFF | host=IP ? |
| 20:55.16 | *** join/#asterisk JohnJacob (n=JohnJaco@pool-71-246-132-82.aubnin.fios.verizon.net) |
| 20:56.44 | ManxPower | TripleFFFF, no, host= is for OUTGOING |
| 20:56.53 | ManxPower | search the wiki for allow/deny stuff. |
| 20:57.05 | *** join/#asterisk Bobcat_1966 (n=chatzill@cpe-069-132-139-254.carolina.res.rr.com) |
| 20:57.11 | ManxPower | NOW you see why I ALWAYS put context=INVALID in general, then specifiy the context for each device in sip.conf |
| 20:57.23 | ManxPower | Unfortunatly sip.conf.sample HAS NO EXAMPLE OF PERMIT/DENY |
| 20:57.59 | ManxPower | Un1x, I can't help you further, but I'll bet you a copy of my [macro-std-exten] that the didww config example they have online is WRONG. |
| 20:58.32 | Dr-Linux | TripleFFFF: do like: |
| 20:58.34 | Dr-Linux | permit=IP.IP.IP.IP/255.255.255.255 |
| 20:58.34 | Dr-Linux | deny=0.0.0.0/0.0.0.0 |
| 20:59.04 | Dr-Linux | host=ip.ip.ip.ip |
| 20:59.11 | Dr-Linux | insecure=very |
| 20:59.33 | Dr-Linux | context=your-context |
| 20:59.44 | Dr-Linux | type=friend |
| 20:59.51 | eKo1 | Are the permit/deny options valid for IAX and H.323 peers? |
| 21:00.16 | Dr-Linux | at the top put something like [serverA] |
| 21:00.17 | TripleFFFF | then... |
| 21:01.10 | TripleFFFF | no userpas |
| 21:01.12 | Dr-Linux | TripleFFFF: no need for user/pass |
| 21:01.12 | Un1x | yes probably |
| 21:01.12 | Un1x | there gay |
| 21:01.12 | Dr-Linux | TripleFFFF: it's secure with IP address |
| 21:01.12 | Un1x | damn i cant beleive there is 285 people in here and not one has used didww before.. |
| 21:01.16 | eKo1 | Looks like it only works for SIP and IAX. |
| 21:01.28 | [TK]D-Fender | Un1x: DIDWW? |
| 21:01.28 | Dr-Linux | eKo1: i do this with SIP trunk |
| 21:01.29 | eKo1 | What is didww? |
| 21:01.57 | Un1x | ManxPower: http://www.didww.com/support/index.php?_m=knowledgebase&_a=viewarticle&kbarticleid=3&nav=0,1 |
| 21:01.58 | eKo1 | Dr-Linux: I just checked the code, only chan_sip and chan_iax2 mention permit/deny |
| 21:02.00 | TripleFFFF | <PROTECTED> |
| 21:02.01 | Un1x | didww.com |
| 21:02.03 | TripleFFFF | now its looking in htere |
| 21:02.30 | TripleFFFF | ooking for 17608607327 in default (domain 192.168.1.7) |
| 21:02.30 | TripleFFFF | Reliably Transmitting (NAT) to 192.168.1.79:5060: |
| 21:02.30 | TripleFFFF | SIP/2.0 404 Not Found |
| 21:02.42 | Dr-Linux | TripleFFFF: where? |
| 21:03.01 | *** join/#asterisk EvilDeshi (n=SkunK@oxford-bb-occam9-ws-25.dsl.maqs.net) |
| 21:04.06 | ManxPower | Un1x, I believe that page is wrong. |
| 21:04.07 | [TK]D-Fender | Un1x: What so special about those guys? Seem like jsut another ITSP to me... |
| 21:04.20 | ManxPower | TripleFFFF, set the context in sip.conf to something other than default and put your exten lines in that context. |
| 21:04.20 | *** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 21:04.40 | Dr-Linux | TripleFFFF: well put in default context something like >> exten => 17608607327,1,Goto(context,extension,priority) |
| 21:05.02 | TripleFFFF | i? |
| 21:05.14 | EvilDeshi | whats the Dial syntax if i want to give the caller the ability to do transfers |
| 21:05.32 | Dr-Linux | ManxPower: please try as ManxPower said |
| 21:05.33 | Strom_C | balls |
| 21:05.33 | ManxPower | TripleFFFF, are you dialing from 1 asterisk server to another asterisk server, both of which you manage? |
| 21:05.55 | TripleFFFF | exten => 17608607327,1,NOOP("GOT ON 1760") |
| 21:05.55 | TripleFFFF | exten => 17608607327,2,Set(CALLERID(name)=FAX) |
| 21:05.55 | TripleFFFF | exten => 17608607327,3,Set(CALLERID(number)=1231231234) |
| 21:05.55 | TripleFFFF | exten => 17608607327,4,DIAL(SIP/${EXTEN}@testfax,30,tr) |
| 21:05.57 | TripleFFFF | well |
| 21:06.00 | ManxPower | EvilDeshi, Smart people configure that in the devuce. |
| 21:06.02 | TripleFFFF | no its from my a[pp |
| 21:06.07 | ManxPower | NEVER USE "r" |
| 21:06.26 | [TK]D-Fender | TripleFFFF: I'm betting its in the wrong context.... |
| 21:06.43 | TripleFFFF | i dont see the noop |
| 21:06.50 | TripleFFFF | wrong contet ? |
| 21:06.57 | hmmhesays | show me how you want it to be |
| 21:06.58 | TripleFFFF | [default] |
| 21:07.00 | TripleFFFF | from extensions |
| 21:07.02 | hmmhesays | tell me baby cause I need to know |
| 21:07.10 | ManxPower | [TK]D-Fender, I think the message is confusing him. |
| 21:07.16 | TripleFFFF | [192.168.1.79] |
| 21:07.16 | TripleFFFF | type=friend |
| 21:07.16 | TripleFFFF | host=192.168.1.79 |
| 21:07.16 | TripleFFFF | context=default |
| 21:07.19 | Dr-Linux | TripleFFFF: wrong context, or your callerID exists in both servers |
| 21:07.24 | TripleFFFF | permit=192.168.1.79/255.255.255.255 |
| 21:07.24 | TripleFFFF | deny=0.0.0.0/0.0.0.0 |
| 21:07.25 | ManxPower | TripleFFFF, move it to another context. |
| 21:07.36 | EvilDeshi | i think its great that i come here and get insulted when i ask how to do somethign what the fuck does i tmatter to you jackass how i let callers xfer |
| 21:07.48 | hmmhesays | LOL |
| 21:07.53 | diablopico | anyone here able to help with openh323 |
| 21:07.55 | hmmhesays | dude settle down, this is IRC |
| 21:08.09 | TripleFFFF | oh |
| 21:08.14 | EvilDeshi | thats the umpteents time i have asked a legitimate question in here and get insulted |
| 21:08.14 | Strom_C | I can lie in haiku! |
| 21:08.17 | EvilDeshi | hard to calm down |
| 21:08.29 | Dr-Linux | trelane_: i think you should use "nat=yes" |
| 21:08.29 | eKo1 | What did you ask? |
| 21:08.30 | ManxPower | Evanrude, "show application dial" but you'll regret it later. |
| 21:08.36 | Dr-Linux | hehe |
| 21:08.38 | EvilDeshi | whats the Dial syntax if i want to give the caller the ability to do transfers |
| 21:08.43 | ManxPower | hence my suggesting you use the transfer feature of the device you are using |
| 21:08.44 | EvilDeshi | and i get insulted for that |
| 21:08.54 | EvilDeshi | what if there is no device being used |
| 21:09.02 | EvilDeshi | what if its all software and doesnt offer xfer ability |
| 21:09.06 | Strom_C | EvilDeshi: "show application dial" at the asterisk CLI |
| 21:09.08 | EvilDeshi | ever think of that perhaps? |
| 21:09.08 | eKo1 | EvilDeshi: I remember using that but I don't remember. |
| 21:09.13 | EvilDeshi | thanks Strom_C' |
| 21:09.16 | eKo1 | check the dial command |
| 21:09.18 | *** join/#asterisk [hC] (n=hardcore@201.198.6.194) |
| 21:09.20 | EvilDeshi | you were actually teh one i was looking for |
| 21:09.21 | TripleFFFF | other contex tworked |
| 21:09.23 | TripleFFFF | no ide a why |
| 21:09.41 | ManxPower | because it's pretty easy, when using the Dial options, to accidently allow people to call into your PBX then allow then to dial out and have you pay for the call |
| 21:09.50 | EvilDeshi | Strom_C do you know a new guy at digium "tony" he just started there maybe a month ago |
| 21:10.05 | file | Tony? |
| 21:10.34 | Dr-Linux | TripleFFFF: good to know, actually i never use "default" context |
| 21:10.39 | Strom_C | anyway, here's my attempt at lying in haiku: Look, the sky is green / Quarters fall up from the floor / And land on the clouds |
| 21:10.47 | Strom_C | EvilDeshi: no |
| 21:10.54 | file | Strom_C: ooh that was good |
| 21:11.19 | EvilDeshi | Strom_C and there is absolutly no way to use meetme without the zaptel kernel mod right? |
| 21:11.25 | diablopico | EvilDeshi: the dial command would be exten=><the extension>,<the priority>,Dial(<extension>,30,tr) |
| 21:11.47 | Strom_C | DON'T USE THE r |
| 21:11.53 | ManxPower | Here's one I stole: Coffee is so cool; Coffee, Coffee, Coffee, Yay!; I drink five big cups. |
| 21:11.55 | EvilDeshi | diablopico is it the t that adds transfer |
| 21:12.01 | diablopico | yes |
| 21:12.01 | Strom_C | EvilDeshi: correct |
| 21:12.08 | Strom_C | EvilDeshi: seriously dude, read the docs |
| 21:12.11 | EvilDeshi | thanks guys big help |
| 21:12.14 | Dr-Linux | ManxPower: fake callerID can be sent on both? i.e. PRI and analog ? |
| 21:12.15 | Strom_C | EvilDeshi: show application dial |
| 21:12.16 | ManxPower | T or t both allow transfer |
| 21:12.21 | EvilDeshi | Strom_C i have been reading |
| 21:12.24 | Strom_C | EvilDeshi: show application dial |
| 21:12.37 | EvilDeshi | im using voip-info.org wiki |
| 21:12.42 | Strom_C | EvilDeshi: show application dial |
| 21:12.48 | EvilDeshi | yes guy |
| 21:12.50 | ManxPower | EvilDeshi, The Wiki is frequently wrong. |
| 21:12.52 | Strom_C | I feel like I'm repeating myself |
| 21:13.06 | diablopico | google "astreisk command dila" |
| 21:13.08 | ManxPower | Use the DIGIUM docs, found in /path/to/src/asterisk/docs/ and "show applications" in the CLI |
| 21:13.12 | diablopico | err dial |
| 21:13.15 | EvilDeshi | Strom_C i think you should go shopping or something and lay off the crack |
| 21:13.19 | EvilDeshi | ;) |
| 21:13.30 | ManxPower | Strom_C, after the 3rd time I have to repeat, I usually put them on /ignore |
| 21:13.38 | Strom_C | heh |
| 21:14.12 | diablopico | can anyone help with a audio delay problem in openh323 ? |
| 21:14.23 | ManxPower | TripleFFFF, mayne you had a SPACE infront of the [default] |
| 21:14.41 | EvilDeshi | there is one thing i cant find in the docs and thats using mp3player |
| 21:14.50 | EvilDeshi | after about 15 minutes or so it just drops the caller |
| 21:14.56 | Strom_C | Here's a better attempt at lying in haiku: I'm sixty feet tall / I have nine million dollars / And ninety-nine phones |
| 21:14.57 | EvilDeshi | says poll timeout |
| 21:15.23 | ManxPower | EvilDeshi, "show application mp3player" the Asterisk CLI. |
| 21:15.44 | TripleFFFF | lol in the docs maybe...sample i mean that WOULD BE LAME\ |
| 21:15.44 | EvilDeshi | Strom_C we are setting our asterisk to play episodes of the radio shows we have in archives |
| 21:15.53 | EvilDeshi | ok i will check that one again |
| 21:16.11 | *** join/#asterisk JohnJacob (n=JohnJaco@pool-71-246-132-82.aubnin.fios.verizon.net) |
| 21:16.31 | ManxPower | the docs for each application are "show application X" in the Asterisk CLI, where X is the application name. There is no other official source of docs for those appliations. |
| 21:16.56 | EvilDeshi | the show application stuff is not telling me anything i dont already know |
| 21:17.06 | *** join/#asterisk techman97 (n=me@70-98-31-249.dr01.rsmt.mn.frontiernet.net) |
| 21:17.10 | EvilDeshi | it is workign but only for 15 minutes |
| 21:17.18 | hmmhesays | heh wow pear makes db connections incredibly easy |
| 21:17.37 | ManxPower | Evanrude, That would be some OTHER issue. |
| 21:17.59 | EvilDeshi | hence why i came here |
| 21:18.02 | techman97 | hey all - I have a place that is looking to put Asterisk in for an predictive AOD and regular ACD call center - any good leads / ideas on a module or flavor of Asterisk that could do this easily? |
| 21:18.22 | Strom_C | tell them to take their predictive dialer and burn in Hell |
| 21:18.25 | techman97 | ideally, I'd need a relatively easy interface (easy for a technical guy) to load in numbers |
| 21:18.28 | ManxPower | EvilDeshi, No other types of calls get disconnected after 15 mins? |
| 21:18.30 | techman97 | Strom_C: rofl |
| 21:18.32 | eKo1 | AOD and ACD? |
| 21:18.34 | techman97 | yaeh |
| 21:18.36 | techman97 | yeah |
| 21:18.37 | EvilDeshi | ManxPower nope just the mp3player |
| 21:18.43 | eKo1 | ~AOD |
| 21:18.51 | techman97 | ? |
| 21:19.03 | ManxPower | techman97, not a lot of people will help with predictive dialers |
| 21:19.21 | techman97 | I'm not looking for help - I just need to know if Asterisk has a module that can do it |
| 21:19.23 | techman97 | :S |
| 21:19.31 | Strom_C | that qualifies as 'help |
| 21:19.33 | Strom_C | ' |
| 21:19.34 | techman97 | =P |
| 21:19.39 | ManxPower | techman97, for predictive dialers, no. |
| 21:19.40 | Juggie | asterisk can be a predictive dialer |
| 21:19.45 | Juggie | but it has no 'module' |
| 21:19.54 | Juggie | however, you might find a 3rd party one if you look. |
| 21:20.14 | ManxPower | Juggie, you are helping again! |
| 21:20.19 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 21:20.25 | techman97 | that's kinda what I figured - not a lot of fans of a predictive dialer...and that I'd have to scrub around for something that could do it |
| 21:20.26 | generalhan | whats up all !? |
| 21:20.32 | Juggie | i'm not quite over the help i gave yesterday |
| 21:20.50 | techman97 | keep faith that it's not an ad agency - it's going to be used to communicate voice messages to a large employee base via telephone |
| 21:21.01 | Juggie | i can say with pretty much 99% accuracy that there is a * predictive dialer. |
| 21:21.05 | Strom_C | techman97: so for that you use voicemail |
| 21:21.08 | Juggie | that doesnt mean yuo wont pay for it, but it exists. |
| 21:21.19 | generalhan | i need some help ... i use a ChanisAvail before dialing an extension and i have one single extension that always is seen as unavailable ... any ideas how to troubleshoot this issue? |
| 21:21.24 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 21:21.34 | vader-- | well they ported our main numbers over today |
| 21:21.34 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 21:21.34 | *** mode/#asterisk [+o mog] by ChanServ |
| 21:21.40 | vader-- | our new phone system is finally live |
| 21:21.46 | Juggie | http://www.voip-info.org/wiki/view/Predictive+dialer |
| 21:21.51 | vader-- | yay now i have to go to the old phone room and move some wires for the fax machines |
| 21:21.56 | vader-- | and analog phones we kept |
| 21:22.00 | Bullseye_Network | techman97: like this? www.ephonealert.com |
| 21:22.20 | eKo1 | * is a pbx, not a predictive dialer |
| 21:22.25 | Bullseye_Network | The site is not finished yet tho |
| 21:23.30 | ManxPower | techman97, Um, you don't need a predictive dialer to have Asterisk call a lit of numbers, say "Your're fired!" and hang up. |
| 21:23.58 | EvilDeshi | qould this be right then for transfering ability exten => 1000,1,Macro(stdexten,SIP/1000),30,t |
| 21:24.39 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 21:24.41 | Strom_C | EvilDeshi: in an alternate universe, yes |
| 21:24.45 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 21:25.02 | EvilDeshi | i tried it with just a dial statement as well and that didnt let transfers occur |
| 21:25.09 | EvilDeshi | exten => 1000,1,Dial(SIP/1000),30,t |
| 21:25.10 | generalhan | in a deranged universe you mean |
| 21:25.21 | Strom_C | EvilDeshi: thats because your syntax is wrong |
| 21:25.28 | EvilDeshi | oh my |
| 21:25.57 | generalhan | exten => 1000,1,Dial(SIP/1000,30,Tt) |
| 21:26.05 | EvilDeshi | thanks |
| 21:26.06 | generalhan | EvilDeshi: like that |
| 21:26.07 | *** join/#asterisk AJaymn (i=AJaymn@70.59.126.198) |
| 21:26.20 | Strom_C | yes, then people calling in from the outside can transfer too! |
| 21:26.34 | EvilDeshi | thats the idea |
| 21:26.37 | Strom_C | no, that's not a security hole at ALL! /me sitcks head in sand and hums "brazil" |
| 21:26.50 | generalhan | lol i LOVE it Strom_C !! |
| 21:27.21 | generalhan | i call in from my cell to my personal phone directory on * and transfer it from there so that the CID shows up as work |
| 21:27.29 | *** join/#asterisk adorah (n=Administ@87.68.173.125.cable.012.net.il) |
| 21:27.35 | generalhan | like i want people knowing my cell number ! lol |
| 21:27.57 | Strom_C | generalhan: uh, that's what caller ID presentation restriction was invented for |
| 21:28.04 | generalhan | nah |
| 21:28.20 | diablopico | any ideas where to get help with * running openh323 ?????/ |
| 21:28.26 | generalhan | people that i call that are work related wont answer their phone if they dont know the number ... thats why i did this to begin with |
| 21:28.27 | EvilDeshi | its still not transfering |
| 21:28.44 | Strom_C | EvilDeshi: how are you trying to accomplish the transfer? |
| 21:29.12 | EvilDeshi | *1 and # |
| 21:29.32 | ManxPower | Cool! EvilDeshi's server is insecure! What's the number for his IVR again? |
| 21:29.51 | Strom_C | EvilDeshi: then it's a signaling issue |
| 21:30.06 | generalhan | HAHA |
| 21:30.28 | generalhan | how would i go about figuring out why an extension ALWAYS thinks it's busy ??? |
| 21:30.37 | *** join/#asterisk tamp4x (n=tampon@vonmail.vonworldwide.com) |
| 21:30.53 | tamp4x | where are the names stored for editting of people in the directory |
| 21:30.54 | tamp4x | ? |
| 21:30.59 | ManxPower | generalhan, try calling it. Use Noop to check the value of DIALSTATUS and HANGUPCAUSE |
| 21:31.16 | tamp4x | i thought it reads voicemail.conf, but i editted it, and it still uses the old format, wtf |
| 21:31.19 | generalhan | ManxPower: will try right now |
| 21:31.22 | ManxPower | tamp4x, /etc/asterisk/voicemail.conf |
| 21:31.45 | tamp4x | manxpower, i reversed the names, but it still does it by last name |
| 21:31.45 | ManxPower | tamp4x, "old format"? |
| 21:32.10 | ManxPower | tamp4x, did you do a "show application directory" and looked carefully at the "f" option? |
| 21:32.12 | tamp4x | so westine jason i moved to jason westine and still not reading by first name |
| 21:32.17 | Dovid | maxexpirey=1000 ; Max length of incoming registration we allow |
| 21:32.17 | Dovid | defaultexpirey=60 |
| 21:32.20 | Dovid | to low |
| 21:32.30 | Dovid | oops. didnt mean to post here. |
| 21:32.51 | tamp4x | ahh |
| 21:32.54 | tamp4x | nice |
| 21:33.07 | Dovid | for maxexpiret and defaultexpirey each number is a second ? |
| 21:33.13 | ManxPower | tamp4x, "show application" is your friend. Love it, hold it, buy it flowers. |
| 21:34.00 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 21:34.36 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 21:35.22 | tamp4x | the f option didnt works |
| 21:35.33 | tamp4x | Directory(vmailcontext|f) |
| 21:35.53 | Strom_C | did you reload app_voicemail.so? |
| 21:36.32 | tamp4x | nope |
| 21:36.34 | tamp4x | err |
| 21:36.37 | tamp4x | doesnt work still |
| 21:37.28 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 21:37.53 | ManxPower | tamp4x, Directory(vmailcontext,,f) |
| 21:37.57 | generalhan | ManxPower: i tried it ... now it goes straight to fast busy ??? just from adding a Noop ??? http://generalhan.pastebin.ca/122739 |
| 21:38.03 | ManxPower | if you don't use dial contexts you can't leave it off |
| 21:38.17 | *** join/#asterisk Derkommissar (n=Alberto@209.168.221.130) |
| 21:38.35 | shmaltz | anybody here know Perl data access scripting? |
| 21:38.46 | ManxPower | generalhan, the noop has to be AFTER the dial. |
| 21:38.52 | ManxPower | remove the chanisavail, that's not working |
| 21:39.12 | shmaltz | ManxPower, what do you mean chanisavail is not working? |
| 21:39.14 | generalhan | ManxPower: but thats the thing ... it IS working for all of my other 39 extensions ... just this ONE isnt |
| 21:39.34 | ManxPower | generalhan, and that is why I'm trying to help you diagnose it. |
| 21:39.40 | generalhan | hmmm |
| 21:39.42 | puppet | how safe is it to run asterisk on a PRIMERGY RX100S3 PD930 Dual Core 3.0GHz/2x1MB L2/1GB/CD-RW\DVD/PCI-X Risercard together with apache + mysql? |
| 21:39.54 | puppet | will asterisk handle that? |
| 21:39.57 | generalhan | well let me make another context then and copy this in it for this bad extension only |
| 21:40.01 | shmaltz | puppet, it wont explode on you |
| 21:40.03 | ManxPower | shmaltz, I suspect the phone is either not registered or the phone is set to do-not-disturb or something like that |
| 21:40.10 | shmaltz | oic |
| 21:40.21 | puppet | shmaltz: well will it handle it "ok" or not? :) |
| 21:40.26 | shmaltz | puppet, I run asterisk on a K62 400 mhz |
| 21:40.40 | puppet | shmaltz: well not with apache + mysql and customers using it right? |
| 21:40.42 | Derkommissar | im having a problem |
| 21:40.47 | Derkommissar | reciving a call trough zap |
| 21:40.52 | Derkommissar | get a wierd error |
| 21:40.58 | shmaltz | puppet, well for that I use a dual piii |
| 21:41.00 | Derkommissar | can someone check this out |
| 21:41.01 | Derkommissar | http://pastebin.ca/122741 |
| 21:41.10 | puppet | shmaltz: ok |
| 21:41.37 | Derkommissar | Aug 8 17:33:09 WARNING[9580]: chan_zap.c:4758 zt_write: Cannot handle frames in 256 format |
| 21:41.51 | shmaltz | Derkommissar, what type of device? |
| 21:41.58 | syzygyBSD | no error, just a warning... |
| 21:42.02 | ManxPower | Derkommissar, looks like the REAL error is above that. |
| 21:42.04 | Derkommissar | its a Digium t1. to a sip device |
| 21:42.18 | ManxPower | Derkommissar, you are allowing G729 and have not purchased the codecs |
| 21:42.37 | Derkommissar | i can make outgoing calls |
| 21:42.46 | [TK]D-Fender | Derkommissar: thats a G729 license issue |
| 21:42.46 | ManxPower | Derkommissar, not with G729 you can't. |
| 21:42.51 | Derkommissar | yes |
| 21:42.54 | Derkommissar | with g729 |
| 21:43.10 | Derkommissar | what is the command to show it...... show translations |
| 21:43.12 | Derkommissar | hang on |
| 21:43.26 | *** join/#asterisk [hC] (n=hardcore@190.10.12.97) |
| 21:43.26 | ManxPower | Derkommissar, I won't argue with you. If you do not have a G729 license, and you try to make a PSTN call out a Zap port, it will fail. |
| 21:43.40 | Derkommissar | <PROTECTED> |
| 21:44.25 | ManxPower | Derkommissar, then you have either bought the licenses or are using the illegal codec. |
| 21:44.36 | ManxPower | Derkommissar, what version of Asterisk? |
| 21:44.41 | Derkommissar | 1.2 |
| 21:44.45 | tamp4x | thanks manx |
| 21:44.58 | ManxPower | tamp4x, it worked? |
| 21:44.59 | generalhan | ManxPower: you are right about the chanisavail ... i can call that ext. once that is gone ... but the DIALSTATUS and HANGUPCAUSE dont show at all ... once it gets to the Dial() it stops on the CLI ... doesnt show my Noops at all |
| 21:45.05 | Derkommissar | its not illegal codec |
| 21:45.17 | Derkommissar | and i can recive calls if i dial directicly to the equipment |
| 21:45.26 | ManxPower | generalhan, pastebin that part of extensions.conf and the cli output. |
| 21:45.29 | Derkommissar | this only happens when it goes trough the queue |
| 21:45.35 | generalhan | ManxPower: rgre |
| 21:45.38 | *** part/#asterisk Nitrus^ (n=ntisog@72-34-76-86.skyriver.net) |
| 21:45.49 | *** join/#asterisk pfn (n=pfnguyen@netblock-66-245-252-239.dslextreme.com) |
| 21:47.34 | generalhan | ManxPower: http://generalhan.pastebin.ca/122745 |
| 21:48.58 | ManxPower | generalhan, so you can call it just fine? |
| 21:49.07 | generalhan | with this set up yes |
| 21:49.21 | ManxPower | generalhan, I have no idea how to help you. |
| 21:49.45 | generalhan | the issue is that i dont have the chanisavail check anymore ... so THAT is why it worked ... when the chanisavail goes back in it ALWAYS sends me to VM cause it says the line is busy |
| 21:50.20 | ManxPower | generalhan, if you can call it then it should work |
| 21:50.44 | Derkommissar | Manx Any Ideas |
| 21:51.01 | Derkommissar | it can recive the call if i send it to the device directicly |
| 21:51.12 | Derkommissar | but if it goes trhough a queue that happenns |
| 21:51.20 | *** join/#asterisk Z_God (n=Z_God@jabber.xs4all.nl) |
| 21:51.28 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 21:52.03 | Z_God | I'm trying to use asterisk with trixbox, but it just dies when I run it |
| 21:52.36 | Z_God | it says it started ok though !? |
| 21:52.52 | mog | do asterisk -vvvc from command line |
| 21:52.59 | mog | pastebin result |
| 21:53.02 | generalhan | ManxPower: im soooo confused as to why it is ONLY this one extension ... this is very interesting |
| 21:53.28 | Z_God | it dies after it loads the modem driver |
| 21:53.30 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 21:53.47 | Z_God | which I indeed just added |
| 21:54.08 | Z_God | I do not get any error at all though |
| 21:54.45 | ManxPower | Z_God, look at the /topic |
| 21:54.45 | *** join/#asterisk oadaeh (n=jason@las-static-208.57.199.83.mpowercom.net) |
| 21:55.09 | Z_God | hmm, I can't see the topic in this client unfortunately :S |
| 21:55.25 | Z_God | problems irc transport bug, I'll check with another client |
| 21:55.29 | Z_God | probably* |
| 21:55.40 | *** join/#asterisk rene1 (n=rene1@gea-gye-internet.telconet.net) |
| 21:55.42 | generalhan | here |
| 21:55.44 | generalhan | ill help |
| 21:55.45 | generalhan | * Topic is 'Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.10 and Zaptel 1.2.7 released! (July 14, 2006) -=- FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx' |
| 21:55.49 | rene1 | hey |
| 21:55.56 | Z_God | thanks :) |
| 21:55.59 | generalhan | lol np |
| 21:56.01 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
| 21:56.20 | rene1 | hey Qwell |
| 21:56.58 | rene1 | i have a question for fellow astmanproxy users |
| 21:58.12 | rene1 | i need both http and standard i/o i dont think it is currently possible. the question is.. can i run a second astmanproxy with a different asterisk manager username/pw and in a different listening port? |
| 21:58.54 | eKo1 | you mean the asterisk manager using another port? |
| 21:58.58 | rene1 | or probably daisy chain two astman proxys together with the first one using standard i.o and the second one using http |
| 21:59.01 | rene1 | no no |
| 21:59.10 | rene1 | asterisk in default port |
| 21:59.23 | rene1 | two users astman1 and astman2 |
| 21:59.36 | rene1 | each representing two different instances of astmanproxy |
| 21:59.50 | rene1 | running each one on its own port (1234, 1235) |
| 22:00.14 | rene1 | using standard and http i/o respectively |
| 22:00.19 | eKo1 | I'm sure you could. |
| 22:01.17 | TripleFFFF | how does one hadd a app in trunk ? |
| 22:01.24 | TripleFFFF | seems old way was depreciated for new module list thing |
| 22:01.30 | TripleFFFF | include ../menuselect.makeopts |
| 22:01.30 | TripleFFFF | etc |
| 22:02.11 | *** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
| 22:02.14 | eKo1 | What app are you trying to add? |
| 22:02.20 | TripleFFFF | spandsp |
| 22:02.24 | TripleFFFF | app_rxfaxc etc |
| 22:02.28 | TripleFFFF | to trunk |
| 22:02.29 | trelane | I'm setting up a Digium te110p. when dialing out I get the following: http://pastebin.ca/index.php. Has anyone seen this, and what have I missed? |
| 22:02.41 | rene1 | eKo1: because there is no way to tell a single astmanproxy to read and serve both http and standard right? |
| 22:03.06 | TripleFFFF | and OBVIOUSELY |
| 22:03.07 | TripleFFFF | http://soft-switch.org/downloads/spandsp/spandsp-0.0.2pre26/asterisk-1.2.x/apps_Makefile.patch |
| 22:03.10 | TripleFFFF | wont add |
| 22:03.41 | *** join/#asterisk Bobcat_1966 (n=chatzill@cpe-069-132-139-254.carolina.res.rr.com) |
| 22:04.12 | *** join/#asterisk Ajaymn (i=AJaymn@70.59.126.198) |
| 22:04.28 | nortex | trelane, You might try that pastebin again :) |
| 22:04.43 | trelane | http://pastebin.ca/122764+ |
| 22:04.44 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 22:04.44 | trelane | http://pastebin.ca/122764 |
| 22:04.50 | trelane | third time for the win! |
| 22:05.29 | *** part/#asterisk [TK]D-Fender (n=Administ@toronto-HSE-ppp4122655.sympatico.ca) |
| 22:05.42 | *** join/#asterisk frenzy (n=alithawe@196.46.104.113) |
| 22:06.18 | trelane | do I have to SetCallerID before dialing out on a PRI? |
| 22:06.42 | frenzy | hi can somone please help me build a profile for zonedata.c in zaptel for a specific country |
| 22:07.01 | *** part/#asterisk rogier (n=rogier@83-67-dsl.ipact.nl) |
| 22:07.08 | eKo1 | trelane: you don't have to |
| 22:07.20 | eKo1 | frenzy: what country? |
| 22:07.43 | frenzy | TZ |
| 22:08.23 | frenzy | I do see it at http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf |
| 22:08.34 | eKo1 | I ask for a country name and I get a two letter country code... |
| 22:08.39 | trelane | eKo1, can you determine what's causing that error? inbound calling works outbound doesn't |
| 22:08.53 | frenzy | eKo1: sorry, Tanzania |
| 22:10.28 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 22:10.28 | *** mode/#asterisk [+o mog] by ChanServ |
| 22:10.34 | TripleFFFF | ~centosbug |
| 22:10.35 | jbot | centosbug is probably a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 22:10.47 | rene1 | ~ecfo |
| 22:10.48 | jbot | Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as "screeching", "feedback", "static", or other useless terms. If users report "static" on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. |
| 22:12.41 | *** join/#asterisk LoneShadow (n=duh@59.92.160.36) |
| 22:12.44 | TripleFFFF | does fax work on anysolution ? or not |
| 22:13.05 | *** join/#asterisk adorah (n=Administ@87.68.173.125.cable.012.net.il) |
| 22:13.50 | eKo1 | what does 'on any solution?' mean? |
| 22:14.07 | eKo1 | trelane: sorry, haven't had a chance to look. |
| 22:14.20 | TripleFFFF | any solution to get fax working.. for fax to email |
| 22:14.21 | eKo1 | frenzy: I have no clue how to do that. |
| 22:14.42 | TripleFFFF | getting a weird In file included from app_rxfax.c:28: |
| 22:14.42 | TripleFFFF | ../include/asterisk/lock.h: In function `ast_mutex_init': error |
| 22:15.06 | eKo1 | TripleFFFF: fax is not guarenteed to work |
| 22:15.25 | TripleFFFF | nothign is gaanteed to wrok with astrisjk.. i know |
| 22:15.28 | TripleFFFF | still |
| 22:17.19 | eKo1 | ask hmmhesays, I think he/she was commenting about his/her setup a couple of days ago |
| 22:17.58 | TripleFFFF | ok then here is better formulation |
| 22:18.05 | TripleFFFF | whats more stable version of asterisk ;)\ |
| 22:18.10 | TripleFFFF | 1.2.5 ? |
| 22:18.11 | frenzy | anyone know how to build the cadience profile in zonedata.c? |
| 22:18.23 | eKo1 | TripleFFFF: 1.2.10 |
| 22:18.26 | TripleFFFF | ahahaha |
| 22:18.27 | TripleFFFF | no |
| 22:18.31 | TripleFFFF | surely not 1.2.10 |
| 22:18.40 | eKo1 | That is the latest stable so why not? |
| 22:19.04 | TripleFFFF | coz.. check bugs.. youll see that theres lots of crash types in these latest crap then bore |
| 22:19.08 | TripleFFFF | before |
| 22:19.16 | TripleFFFF | anyhow im talking one that works with app_Rxfax |
| 22:20.07 | TripleFFFF | like the business edition is what version as core |
| 22:20.15 | TripleFFFF | thats SHOULD give a hint |
| 22:20.19 | *** join/#asterisk smackus (n=ckwall@63.149.122.93) |
| 22:20.29 | eKo1 | The what now? |
| 22:20.32 | *** join/#asterisk Bobcat_1966 (n=chatzill@cpe-069-132-139-254.carolina.res.rr.com) |
| 22:21.35 | houdini | what does Business Edition sell for? |
| 22:21.39 | TripleFFFF | 999 |
| 22:21.42 | TripleFFFF | per license |
| 22:22.16 | houdini | Youch.. so if you wanted to run 100 calls... that's $100K? |
| 22:22.22 | trelane | I'm setting up a Digium te110p. when dialing out I get the following: .http://pastebin.ca/122764 Has anyone seen this (relevant google searches come up empty), and what have I missed? |
| 22:22.23 | russellb | no. |
| 22:22.23 | ManxPower | TripleFFFF, make sure your rxfax/txfax stuff is for the version of Asterisj you are running. |
| 22:22.37 | russellb | licensing for incrasing call volume is different, and negotiated directly with Digium. |
| 22:22.44 | houdini | ahh okay |
| 22:23.05 | russellb | and you get more than 1 call per server ........ |
| 22:23.11 | russellb | your math is quite a bit off :-p |
| 22:23.27 | ManxPower | trelane, cause code 28. Look it up. Google for PRI ISDN CAUSE CODE |
| 22:23.29 | TripleFFFF | hehe there are none |
| 22:23.41 | ManxPower | make sure you use one that uses DECIMAL codes, and not hex |
| 22:23.41 | houdini | I was thinking like G729 pricing... |
| 22:23.44 | TripleFFFF | houdini licence=1 box |
| 22:23.47 | TripleFFFF | not one call |
| 22:24.04 | russellb | ManxPower: or you can look in channel.c. It has a table of causes and descriptions |
| 22:24.19 | russellb | search for AST_CAUSE |
| 22:24.30 | russellb | and the associated numbers are in include/asterisk/channel.h ... |
| 22:24.35 | *** join/#asterisk techie (n=gus@voipops.net) |
| 22:24.52 | ManxPower | russellb, fine, tell everyone the easy way. |
| 22:24.55 | [hC] | is there anything like queuemetrics for providing agent stats on asterisk for free? |
| 22:25.48 | russellb | ManxPower: sorry :X |
| 22:27.30 | houdini | russellb: is there any hard published data on hardware/concurrent calls on *? The Wiki has information, but it's all over the board. |
| 22:27.58 | russellb | no |
| 22:28.07 | rene1 | [hC]: you can query the AMI or the cli by yourself and there is even a queue_stats log in /var/log/asterisk |
| 22:28.25 | russellb | for BE there is, it is heavily stress tested and so forth ... but that's all that i know of |
| 22:29.04 | *** part/#asterisk smackus (n=ckwall@63.149.122.93) |
| 22:29.36 | houdini | russell: Where can I find that data on BE? |
| 22:30.09 | russellb | houdini: look around digium.com, i'm not sure exactly where it is located |
| 22:31.05 | Un1x | god dam it |
| 22:31.06 | Un1x | Here is a call trace: |
| 22:31.06 | Un1x | Formating dialstring: SIP/72.61.82.119/422008213 |
| 22:31.06 | Un1x | <PROTECTED> |
| 22:31.06 | Un1x | <PROTECTED> |
| 22:31.10 | Un1x | i get a 404 |
| 22:31.28 | TripleFFFF | SIP/${EXTEN@HOST |
| 22:31.32 | TripleFFFF | not inverse |
| 22:31.36 | TripleFFFF | is a ISO way |
| 22:32.12 | ManxPower | Un1x, if you specify by IP address it bypasses sip.conf |
| 22:32.31 | Un1x | manzpower: how do i do that? |
| 22:32.48 | Un1x | *manxpower |
| 22:32.52 | *** part/#asterisk Z_God (n=Z_God@jabber.xs4all.nl) |
| 22:32.55 | ManxPower | Dial(SIP/12345@sipconfentry) |
| 22:33.00 | ManxPower | where 12345 is the number to dial |
| 22:34.50 | Un1x | ok |
| 22:35.01 | Un1x | so then id' dail to my SIp provider and from them to me correct? |
| 22:35.08 | *** join/#asterisk topping (n=topping@ppp-67-124-89-235.dsl.pltn13.pacbell.net) |
| 22:37.51 | *** join/#asterisk cp5 (n=cp5@72.54.121.98) |
| 22:38.04 | cp5 | hi |
| 22:39.01 | Strom_C | lo |
| 22:40.11 | cp5 | anyone happen to know which piece of code is responsible for joining channels together? for example, if a call goes into a queue and the member is Local/something@somewhere, if i start perform a Record(), i see a few seconds later the monitor stops but the bridge is still open meaning that some bridge in the middle must have been dropped and the two ends connected directly |
| 22:43.06 | eKo1 | cehck app_record and app_queue |
| 22:43.10 | eKo1 | *check |
| 22:44.34 | cp5 | looks like ast_channel_free is responsible, just have to find where it's happening |
| 22:44.51 | cp5 | just is very strange that it happens any time between 1 - 20 seconds after the call has been bridged and the call keeps going |
| 22:50.07 | Un1x | wq00t |
| 22:50.08 | Un1x | i fixed it |
| 22:50.09 | Un1x | :D |
| 22:51.50 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 22:53.26 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 22:53.26 | *** mode/#asterisk [+o russellb] by ChanServ |
| 22:54.31 | Un1x | w00t woreked man |
| 22:54.41 | Un1x | hehe soo happy now :P |
| 22:54.59 | *** join/#asterisk tempest1 (n=asf@adsl-144-58-221.chs.bellsouth.net) |
| 22:56.59 | *** join/#asterisk Penggu (n=yeah@220-245-58-70.static.tpgi.com.au) |
| 22:57.01 | Penggu | hi all |
| 22:57.42 | Penggu | just wondering - when you have a LAN with IP phones all over the place, does all the traffic between the phones go through the asterisk server, or do they run peer-2-peer? |
| 22:57.51 | Penggu | LAN with 1 subnet |
| 22:57.57 | Un1x | i think it's through server |
| 22:58.01 | Penggu | hmm |
| 22:58.06 | Un1x | not sure tho |
| 22:58.33 | Penggu | couldnt help thinking to myself that it seems a bit of a waste, esp when, for eg, the extensions are talking in the same lingo |
| 22:58.38 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 22:58.38 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 22:59.39 | hads | Penggu: If you are talking SIP phones then the RTP can go direct between phones. It depends on the canreinvite option in sip.conf |
| 22:59.51 | Penggu | ah k |
| 23:01.45 | hads | and a few other things. |
| 23:03.12 | *** join/#asterisk pjchilds (i=pjchilds@pdpc/supporter/student/pjchilds) |
| 23:03.13 | *** join/#asterisk pdt (n=ptinsley@209.12.249.243) |
| 23:03.35 | Penggu | i was thinking to setup a test environment for our workplace's pbx before we either upgrade it or replace it |
| 23:03.38 | pdt | hello |
| 23:04.18 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 23:04.28 | Penggu | the only thing appealing really is the scalability in terms of wiring things up and maybe the savings in terms of equipment. |
| 23:04.57 | Penggu | but overhaul is costly nevertheless... |
| 23:05.00 | Penggu | anyway, thx ppl |
| 23:05.29 | Penggu | unless everyone uses soft phones.. = get headseats = might as well buy handsets |
| 23:05.36 | pdt | anybody in here using a soundcard for music on hold, I can use the dsp as a |
| 23:05.44 | pdt | whoops |
| 23:07.11 | pdt | i meant to finish that though ;) I can use the dsp as a phone by using dial and hangup on the console but I cannot seem to get moh working |
| 23:08.09 | pdt | arecord segfaults I have tried sox with this command: sox -t ossdsp /dev/dsp -q -c 1 -r 8000 -t raw - and still I get moh started and stopped right after it |
| 23:09.44 | *** join/#asterisk aydiosmio (n=aydiosmi@65.213.70.43) |
| 23:10.18 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 23:11.14 | *** join/#asterisk EvilDeshi (n=SkunK@oxford-bb-occam9-ws-25.dsl.maqs.net) |
| 23:11.25 | *** join/#asterisk [hC] (n=hardcore@190.10.12.97) |
| 23:12.09 | EvilDeshi | is there a place other then FWD that can provide the same services at no charge? |
| 23:12.33 | Strom_C | cheapskate |
| 23:12.56 | EvilDeshi | if you saw where I lived you would understand |
| 23:13.13 | EvilDeshi | im just having trouble with FWD accounts registering |
| 23:13.16 | EvilDeshi | some do some dont |
| 23:13.21 | Strom_C | you and everyone else |
| 23:13.42 | aydiosmio | anyone know how many concurrent calls asterisk can handle bewteen ISDN channels only? |
| 23:13.53 | aydiosmio | on a decent pc |
| 23:14.02 | Strom_C | aydiosmio: PRI? |
| 23:14.02 | aydiosmio | is the CPU even a consideration? |
| 23:14.05 | aydiosmio | yeah |
| 23:14.12 | Strom_C | aydiosmio: how many PRIs do you have coming into the box? |
| 23:14.13 | Corydon-w | 191 |
| 23:14.36 | aydiosmio | Strom_C: I'll be roting 4 T1s to 4 channel banks |
| 23:14.41 | aydiosmio | routing |
| 23:14.45 | aydiosmio | and expanding later |
| 23:14.49 | Strom_C | wait |
| 23:14.58 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.239.50.Dial1.SanJose1.Level3.net) |
| 23:14.59 | Strom_C | ISDN and channel banks in the same sentence? |
| 23:15.02 | Corydon-w | You could do that with a 500MHz PC |
| 23:15.34 | eKo1 | aydiosmio: A lot of people come in here and ask the same type of question. The answer is always: it depends |
| 23:15.46 | Corydon-w | Routing between a PRI and a channel bank takes nearly zero CPU |
| 23:15.57 | aydiosmio | yeah that's what I thought |
| 23:16.00 | aydiosmio | thanks |
| 23:16.12 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 23:16.14 | *** join/#asterisk dasenjo_ (n=dasenjo@63.245.86.247) |
| 23:16.29 | Corydon-w | What takes all the CPU is doing things like transcoding and audio mixing |
| 23:16.35 | aydiosmio | I wouldn't presume to ask that question if I were routing to VoIP channels |
| 23:17.32 | diablopico | does anyone know what can cause a 20 second delay before 2 wy conversation can take place ... using * 1.0.7 and correct openh323/pwlib |
| 23:17.53 | aydiosmio | diablopico: wasn't the answer the caller id? |
| 23:18.47 | *** join/#asterisk rene1 (n=rene1@gea-gye-internet.telconet.net) |
| 23:18.55 | diablopico | aydiosmio: i never saw an answer , bi i must say caller id is all good ... any other ideas ? |
| 23:19.08 | ManxPower | Strom_C, apparently some channel banks CAN talk ISDN. |
| 23:19.17 | rene1 | is the softphone + sub 30 analog headset CONSIDERED HARMFUL for production call centers? |
| 23:19.18 | ManxPower | I can't imagine why anyone would want to..... |
| 23:19.34 | ManxPower | rene1, softphones cause cancer. |
| 23:19.54 | Strom_C | ManxPower: odd |
| 23:19.58 | Corydon-w | Yes, but the frequent usage of PRI and channel banks is: the existing plant is analog phones, and your calls are coming in across PRI |
| 23:20.28 | russellb_ | IMAP support for voicemail storage is now in trunk ... go test it people |
| 23:20.31 | Corydon-w | We did a few of those installs a few years ago, but now, almost everybody wants a SIP phone |
| 23:20.38 | Corydon-w | russellb_: woot |
| 23:20.50 | russellb_ | heh, i didn't do any of the work |
| 23:20.54 | russellb_ | i just want people to test it :D |
| 23:21.18 | rene1 | rusellb_ that is a sweet feature... |
| 23:22.38 | Un1x | hmm |
| 23:22.47 | rene1 | russelb: does that means one needs an special VM IMAP account or can asterisk use an existing IMAP account for VM storage |
| 23:22.55 | Un1x | is there a way i can have Asterisk use a different internet connection |
| 23:22.58 | Un1x | for different calls? |
| 23:23.22 | aydiosmio | hehe |
| 23:23.28 | russellb | rene1: not sure ... |
| 23:23.31 | aydiosmio | oh boy |
| 23:23.35 | russellb | exisiting i think |
| 23:23.39 | twisla | tagging them with iptables and then route it to different routes |
| 23:23.40 | russellb | haven't tried it |
| 23:23.41 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 23:23.41 | *** mode/#asterisk [+o mog] by ChanServ |
| 23:23.44 | rene1 | ahh |
| 23:23.54 | rene1 | twisla: sweet |
| 23:23.54 | twisla | but never tried it too |
| 23:24.01 | rene1 | it should work |
| 23:24.06 | aydiosmio | I agree |
| 23:24.08 | twisla | it should, yep |
| 23:24.09 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 23:24.25 | hads | Is there a bug# for the imap thing? |
| 23:24.25 | rene1 | aydiosmio: is a fun nickname |
| 23:24.32 | aydiosmio | why thank you |
| 23:24.54 | rene1 | it made me smile |
| 23:24.56 | aydiosmio | diablopico: /msg only plz |
| 23:25.41 | aydiosmio | freePBX should publish a feature code card |
| 23:25.53 | Un1x | hmm wonder if there is a AGI script that will use different inet connections for different calls |
| 23:26.11 | twisla | Un1x: just write it :) |
| 23:26.13 | *** join/#asterisk Alex (n=hauntedu@gentoo/user/Alex) |
| 23:26.23 | rene1 | Un1x you need to set routing first tho |
| 23:26.24 | Un1x | i dont know how to code perl still learning :( |
| 23:26.42 | rene1 | learn something else Ruby i would suggest |
| 23:26.50 | rene1 | it will be easier |
| 23:26.57 | Alex | php's okay, too =P |
| 23:27.28 | aydiosmio | Un1x: the problem is that if you're sending the calls to the same VoIP provider, there's no way to differentiate the traffic |
| 23:27.36 | rene1 | i just made a suggestion |
| 23:27.51 | Un1x | no i will be ussing 2 different voip providers |
| 23:27.57 | Un1x | but see my conenctions limit me to 100Kb upload max |
| 23:27.57 | aydiosmio | oh then that's easy |
| 23:28.07 | Un1x | so i was wanting to use 2 different inet connections |
| 23:28.12 | Un1x | i have 2 nics in my server ;D |
| 23:28.18 | aydiosmio | just change your routing table to change which network is on each interface |
| 23:28.18 | twisla | Un1x: just filter it at iptables level with dst ip :) |
| 23:29.01 | twisla | it's more a routing issue than an asterisk issue in this case |
| 23:29.06 | rene1 | yep |
| 23:29.17 | rene1 | twisla could it be done only with route or do you need iptables |
| 23:29.35 | rene1 | i have done some routing when setting up VPNs but cant remember |
| 23:30.31 | twisla | rene1: I think it could be dealed with some static routes, but filtering packets with iptables seems more powerfull and flexible to me |
| 23:31.17 | twisla | anyway, you always finish to mix up routes, filtering and traffic shapping in cases like that :) |
| 23:32.02 | twisla | the "basic static solution" never satisfies |
| 23:32.18 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 23:32.20 | rene1 | sure |
| 23:32.36 | rene1 | it is just that i havent crossed that networking chasm yet |
| 23:32.56 | twisla | time has come to do it :) |
| 23:35.08 | *** join/#asterisk adorah (n=Administ@87.68.173.125.cable.012.net.il) |
| 23:35.13 | Un1x | hmm time to setup voicemail :D |
| 23:35.20 | [hC] | anyone here often tune FXO chnnels (Specifically on sangoma a200's?) |
| 23:35.46 | Strom_C | I tune FXO channels on digium hardware |
| 23:38.21 | Strom_C | what are you trying to tune, exactly? |
| 23:41.00 | *** part/#asterisk frenzy (n=alithawe@196.46.104.113) |
| 23:41.15 | [hC] | gain levels i suppose |
| 23:41.36 | Strom_C | do you know the number for your telco's local milliwatt test tone? |
| 23:41.38 | rene1 | zapata [tx|rx]gain no good? |
| 23:41.51 | rene1 | ~milliwatt |
| 23:41.56 | twisla | 'night |
| 23:42.01 | rene1 | see ya |
| 23:42.19 | [hC] | so far ive been doing a milliwatt test tone and trying to get them as close to 14484 as possible, ... testing rx is hard though, trying to loop it |
| 23:42.19 | [hC] | but i worry that im doing it wrong and screwing up peoples lines. |
| 23:42.35 | Strom_C | what do you mean "testing rx is hard" |
| 23:42.40 | Strom_C | testing rx is ass-easy |
| 23:42.53 | Strom_C | you tune the rxgain and then you call something and see if it sounds OK |
| 23:43.34 | Strom_C | then you start with txgain at the same level and reduce it until your near-end echo goes away |
| 23:46.49 | *** join/#asterisk logicwrath (n=dfgaeefa@c-68-60-121-112.hsd1.mi.comcast.net) |
| 23:48.31 | Strom_C | you're welcome |
| 23:51.49 | *** join/#asterisk sangee (n=rkuru@217.17.253.30) |
| 23:52.34 | sangee | anyone know how to how to connect "two asterisk server"? |
| 23:52.41 | EyeCue | iax2?! |
| 23:52.45 | sangee | yes |
| 23:53.35 | Un1x | anyone |
| 23:53.37 | Un1x | know any tools |
| 23:53.40 | Un1x | for SIP CALL trace |
| 23:54.17 | rene1 | sip generation |
| 23:54.40 | rene1 | or pure sip traces |
| 23:55.04 | rene1 | sip debug on shows most of what you want |
| 23:55.13 | rene1 | ethereal can help too |
| 23:56.41 | Un1x | sip generation is a prog? |
| 23:59.46 | sangee | how to connect two asterisk server? |
| 23:59.53 | *** join/#asterisk [hC] (n=hardcore@190.10.12.97) |