| 00:03.56 | navaismo | Hi there, anyone knows if IAX2 protocl still support the Register information Element(IE) in plaintext? |
| 00:18.20 | *** join/#asterisk pcangel (~yoink@www.vehiclestars.com) |
| 00:20.10 | pcangel | Does anybody know of publicly released/open source C++ class that would read messages in the format Asterisk's manager interface outputs them in? (A1: B1\nA2: B2\nA3: B3\n\n) I'm trying to save some time if I can |
| 00:21.18 | pcangel | I want the messages in a format I can pass through a function that checks the Event type to see if there's any hooks set up for it |
| 00:21.35 | NetAtWorld | got a question have a remote extension that calls the pbx but after a few seconds the pbx just disconnects it what do you think it could be ? |
| 00:21.42 | NetAtWorld | scheduling destruction of sip dialog (method: notify) |
| 00:27.43 | *** join/#asterisk iprouteth0 (~iprouteth@unaffiliated/iprouteth0) |
| 00:31.57 | *** join/#asterisk dtoronto (~dtoronto@173-14-233-70-Utah.hfc.comcastbusiness.net) |
| 00:33.48 | dtoronto | I'm trying to set up three asteriskNow servers with a centralized voicemail server. I would like to continue to use the FreePBX if on the three asterisk servers if possible, I would like to know if anyone has had experience with this |
| 00:40.54 | *** join/#asterisk bjhaid (~abejide@41.220.69.11) |
| 00:42.43 | carrar | stupid bug 0018344 |
| 00:42.47 | carrar | !@#$!@#$ |
| 00:43.50 | bjhaid | i am trying to connect to an ITSP but i cannot get to connect, i need to do a sip trace, please how can i get this done |
| 00:46.16 | p3nguin | dtoronto: There is an entire channel dedicated to FreePBX... and this isn't it. |
| 00:46.17 | carrar | sip set debug on |
| 00:46.45 | carrar | or |
| 00:46.46 | carrar | ngrep -q -d eth0 -W byline net IPADDRESSBLOCK/24 and udp port 5060 |
| 00:48.55 | *** join/#asterisk sourcode (~code@ppp-115-87-200-176.revip4.asianet.co.th) |
| 00:48.59 | *** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
| 00:50.14 | bjhaid | carrar: thanks |
| 01:02.24 | *** join/#asterisk JonnyD_work (~Jon@cpe-071-075-036-057.carolina.res.rr.com) |
| 01:04.03 | philippel_mac | anyone know off hand, on 1.8, is there a way to get the call pickup of a parked call to use the CONNECTEDLINE() ability to get the CID on the phone updated from the call that was parked? |
| 01:04.31 | philippel_mac | I've got the CONNECTEDLINE() stuff working, but that one seems like it's up to the ParkedCall() function to do it and no apparent options |
| 01:05.13 | bjhaid | carrar: i connect to the internet via a modem, you have any ideas of what i can do to bye pass NAT, i think the problem is from my end, due to the type of connection |
| 01:06.04 | *** join/#asterisk envisean (~envisean@unaffiliated/envisean) |
| 01:14.30 | carrar | bjhaid, I'd put your modem in bridge mode and let your Asterisk box be internet device |
| 01:18.01 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 01:34.41 | *** join/#asterisk chopp (~chopp@unaffiliated/chopp) |
| 01:39.35 | dtoronto | I am unable to send voicemail to a central asterisk voicemail server using following dialplan: exten => _s-BUSY,n,Dial(IAX2/To-Asterisk4/u${ARG1}) any help would be appreciated |
| 01:41.18 | dtoronto | current version of asterisk is 1.8.3 |
| 01:47.20 | *** join/#asterisk Kumbang (~kumbang@180.245.137.5) |
| 01:53.23 | *** part/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 02:11.41 | *** join/#asterisk shmaltz (~chatzilla@mail.dmaven.com) |
| 02:12.06 | shmaltz | anyone know what a tdma over IP gateway (if there is no cell coverage) is called? |
| 02:16.46 | drmessano | femtocell |
| 02:17.42 | shmaltz | thank you drmessano |
| 02:20.51 | *** join/#asterisk bjhaid (~abejide@41.220.69.11) |
| 02:25.45 | tonsofpcs | that's cell coverage though... |
| 02:26.00 | tonsofpcs | I've done IP over TDMA... I suppose TDMA over IP is just 'buffering' |
| 02:26.15 | tonsofpcs | (IP over TDMA is often done with T1 and ISDN circuits) |
| 02:27.26 | shmaltz | tonsofpcs, read the post again, thats what I wanted cell coverage |
| 02:28.26 | nestAr | i have contacts and hints on my IP650 and all the phones that aren't currently registered show unavailable.. however, is there a way to make a phone on DND show as unavailable on the ip650? |
| 02:29.53 | tonsofpcs | shmaltz: what you wanted and what you said are two different things entirely, apparently. |
| 02:30.19 | shmaltz | tonsofpcs, wow how so? read it again |
| 02:30.54 | tonsofpcs | shmaltz: I have. |
| 02:31.16 | shmaltz | nestAr, you would have to remat the dnd button on the polycom to an extension that: A. manually sets the blf to unavail and B. makes sure thru dialplan not to ring the phone |
| 02:31.32 | shmaltz | tonsofpcs, didn't I say if there is no cell coverage? |
| 02:32.08 | tonsofpcs | ... |
| 02:32.19 | tonsofpcs | goes back to messing with autotune |
| 02:32.39 | shmaltz | moves on |
| 02:35.51 | nestAr | just realized that the IP450 doesn't have a hard DND button. |
| 02:38.27 | Kobaz | anyone have any good info on writing polycom xml apps |
| 02:49.36 | nestAr | does the polycom im thing work under asterisk? |
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| 03:20.55 | *** join/#asterisk dzup2 (~alex@unaffiliated/dzup) |
| 03:21.02 | dzup2 | hello, i setup the intercom to console/dsp but when i dial *808# which dials my custom ext 8 ...makes a beep and then when i speak i dont heard my voice on the speaker, i test aplay some.wav file and sounds good, what a may doing wrong? thankyou |
| 03:33.32 | Kobaz | yeah |
| 03:37.34 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 04:03.31 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net) |
| 04:24.24 | dzup2 | never mind people i fix it, but ....hmmm |
| 04:25.34 | dzup2 | how can i intercom to dsp in "realtime" ...the way i have it is i record a msg then is playit over pc soundcard output, i wonder if there is a way to play the msg intercom in realtime |
| 04:28.52 | *** join/#asterisk c0dyhi11 (~c0dyhi11@208-47-100-77.dia.static.qwest.net) |
| 04:28.56 | c0dyhi11 | anyone here that can help with a voicemail email issue? |
| 04:30.40 | p3nguin | State the problem. |
| 04:31.12 | c0dyhi11 | It used to send emails outbound from the asterisk local user account. |
| 04:31.19 | c0dyhi11 | Now it sends from root. |
| 04:31.39 | c0dyhi11 | I can't find anywhere where to change that. |
| 04:32.24 | p3nguin | Run this command and show me what it says: ps u -C asterisk |
| 04:33.03 | c0dyhi11 | USER PID %CPU %MEM VSZ RSS TTY STAT START TIME COMMAND |
| 04:33.03 | c0dyhi11 | root 11548 1.3 1.0 38056 22204 ? Sl Feb08 164:00 /usr/sbin/asterisk -f -vvvg -c |
| 04:33.11 | c0dyhi11 | so asterisk is running as root? |
| 04:33.35 | p3nguin | That's correct. You need to fix that ASAP. |
| 04:34.01 | p3nguin | What distro are you running asterisk on? |
| 04:34.26 | c0dyhi11 | i just fixed it. |
| 04:34.28 | c0dyhi11 | it's asterisk now |
| 04:34.43 | c0dyhi11 | i did a: servie asterisk stop |
| 04:34.50 | c0dyhi11 | then an: amportal stop |
| 04:34.55 | c0dyhi11 | then an amportal start |
| 04:34.57 | c0dyhi11 | all fixed |
| 04:35.12 | beardy | wtf is amportal. |
| 04:35.15 | c0dyhi11 | stupid mistake on my end |
| 04:35.16 | p3nguin | Most of the time the init/rc script has a parameter to set the user and group to run asterisk. |
| 04:35.36 | c0dyhi11 | amportal is freepbx's special service |
| 04:39.09 | c0dyhi11 | Thanks a bunch for the help. |
| 04:40.11 | p3nguin | I'm glad you had something easy. My mind is wiped and I'm about to retire for the day. |
| 04:41.25 | c0dyhi11 | I google for like an hour and nothing popped up. I love IRC. |
| 04:44.37 | dzup2 | i google like 23hrs already and can not get my intercom to be play at my audio device, follow bunch of tutorias and offcourse i ask the channel, no luck either way, you are luck :) |
| 04:44.44 | dzup2 | have* |
| 04:45.13 | dzup2 | you are lucky lol |
| 04:45.59 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 04:46.59 | c0dyhi11 | haha. Yeah i think you are wiped out! :-) have a great nite |
| 05:03.12 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
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| 05:52.22 | *** join/#asterisk nfsnobody (~Whitey@ppp191-96.static.internode.on.net) |
| 05:53.01 | nfsnobody | hey all, I'm getting an error trying to use AGI with a PHP script - the error is ERROR[29115]: utils.c:1128 ast_carefulwrite: write() returned error: Broken pipe |
| 05:53.14 | nfsnobody | No matter what the content of the script it, this comes up - can anyone point me in the right direction? |
| 05:53.54 | nfsnobody | I'm running 1.6.12, compiled from source |
| 05:55.56 | *** join/#asterisk Defraz (~Defraz@96.18.85.158) |
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| 05:59.40 | kaldemar | nfsnobody: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg231722.html |
| 05:59.51 | kaldemar | read some of that thread |
| 06:01.40 | nfsnobody | kaldemar - so that would mean it's a scripting error? |
| 06:01.53 | nfsnobody | I've tried about 4 different example scripts, along with a simply blank script... |
| 06:04.25 | kaldemar | it's one possibility. |
| 06:04.54 | nfsnobody | thanks - I'll give it a try :) |
| 06:07.36 | kaldemar | it wouldn't be the first time there were erroneous examples floating around. |
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| 06:45.55 | *** join/#asterisk Babak__ (Babak.y@85.133.129.100) |
| 06:48.19 | Babak__ | hi can someone tell me what a CO port is? |
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| 06:54.08 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 06:57.05 | kaldemar | Babak__: where did you see it? it might be meaning an FXO interface or an ISDN interface in NT mode. |
| 06:57.47 | Babak__ | kaldemar: I was searching for a way to connect a kx td 1232 to sip |
| 07:06.29 | *** join/#asterisk asterisk-learner (~chatzilla@77.42.241.114) |
| 07:13.55 | asterisk-learner | hello, when compiling dahdi, I am not getting dahdi_dummy.so , although dahdi_dummy.c is present ... am i missing something ? |
| 07:17.54 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-138-161.ks.ks.cox.net) |
| 07:20.28 | kaldemar | asterisk-learner: the dummy functionality is built in the "dahdi" module. you won't get a separate dummy module in the newer versions. |
| 07:22.05 | asterisk-learner | kaldemar: I am using asterisk 1.4.36, even in that version i shouldn't get a dahdi_dummy.so ? Also when i do lsmod | grep dummy, i dont get any "dahdi" module.... |
| 07:22.34 | kaldemar | asterisk-learner: dahdi_dummy is a part of dahdi, not asterisk. |
| 07:22.57 | kaldemar | i was talking about dahdi versions. |
| 07:23.30 | kaldemar | and you won't see a dahdi_dummy module in lsmod because there is no such module, as stated earlier. |
| 07:23.38 | asterisk-learner | mmm |
| 07:24.00 | asterisk-learner | but in my asterisk application, I have this line failing : p->zaptel_chan = ast_request("dahdi", AST_FORMAT_ULAW, "pseudo", NULL); |
| 07:24.03 | kaldemar | do you have some kind of a dahdi related problem? |
| 07:24.08 | asterisk-learner | althoguh chan_dahdi.so is laoded in asterisk |
| 07:24.34 | asterisk-learner | dahdi + asterisk = yes |
| 07:25.43 | asterisk-learner | actually, in ast cli, if i run module show like dahdi, i get 3 modules, are thoses enough or am i missing more modules ? |
| 07:26.34 | kaldemar | how does your line fail? |
| 07:28.29 | asterisk-learner | ast_request returns NULL |
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| 07:29.46 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 07:29.49 | kaldemar | there are many reasons why it could return NULL. what is yours? |
| 07:30.46 | asterisk-learner | how can i know ? I am making this calls and it is returning NULL (when it shouldn't if evreything is set up correctly) |
| 07:30.48 | asterisk-learner | ... |
| 07:31.43 | kaldemar | modify ast_request to tell you |
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| 07:36.15 | asterisk-learner | ok let me try this and get back to you |
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| 07:37.18 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 07:38.10 | killown | its possible to asterisk control my voip authenticated through ata modem? |
| 07:41.13 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 07:42.26 | asterisk-learner | kaldemar: unfortunatly I am getting the reson : AST_CAUSE_NOTDEFINED, does this helps us in any way ? |
| 07:42.29 | asterisk-learner | :-# |
| 07:42.33 | asterisk-learner | :-$ |
| 07:45.17 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:47.22 | kaldemar | asterisk-learner: still 3 different possibilities for ast_request to return NULL. |
| 07:49.54 | asterisk-learner | kaldemar: ok i will add debuging info in ast_request() to see from where it is returning |
| 07:55.41 | asterisk-learner | kaldemar: it is returning NULL from this line : if (!(c = chan->tech->requester(type, capabilities | videoformat, data, cause))) |
| 07:55.47 | asterisk-learner | ? |
| 07:58.30 | *** join/#asterisk benngard (~mabe@213.88.138.230) |
| 08:04.28 | asterisk-learner | Does anyone have dahdi installed with asterisk 1.4.xxx ? |
| 08:05.01 | *** join/#asterisk manji (~manjiki@penicuik.physics.upatras.gr) |
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| 08:18.15 | asterisk-learner | kaldemar: any ideas why the tech->requester() call might fail even when chan_dahdi.so is loaded in asterisk ? |
| 08:20.30 | *** part/#asterisk aiguojun (~agj@61.141.20.110) |
| 08:22.03 | kaldemar | asterisk-learner: no. |
| 08:22.18 | asterisk-learner | ok |
| 08:22.25 | asterisk-learner | thx anw |
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| 08:30.13 | *** join/#asterisk aiguojun (~agj@61.141.20.110) |
| 08:30.24 | aiguojun | HI erverbody |
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| 08:31.00 | *** part/#asterisk aiguojun (~agj@61.141.20.110) |
| 08:32.56 | ChannelZ | HI Dr. Nick! |
| 08:44.29 | Russ | HI EVERYBODY |
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| 08:56.32 | _schmidts | good morning |
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| 09:05.08 | asterisk-learner | kaldemar: it turned out i didnt do a make samples when moving from asterisk 1.4.10 to 1.4.36 |
| 09:05.10 | asterisk-learner | and now it works |
| 09:05.12 | asterisk-learner | :-) |
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| 11:23.13 | mzb | I'd like to find a nice way of importing trunk definitions with a script ... is there an 'accepted method' ? |
| 11:24.26 | mzb | ie: I want to setup an ENUM trunk (and outbound routes to suit) by default |
| 11:25.27 | WIMPy | Maybe you should avoid the word "trunk". That might make it more understandable. |
| 11:26.34 | kaldemar | sounds like freepbx terminology. |
| 11:26.43 | mzb | ah ... oops ;) |
| 11:26.44 | kaldemar | mzb: is that what you're using? |
| 11:26.50 | mzb | wrong channel ... my mistake |
| 11:27.10 | mzb | I've been writing (yet another) debian installer script for Asterisk+FreePBX |
| 11:28.02 | mzb | sorry about that ... I'll slink off to #freepbx-dev ;) |
| 11:28.08 | WIMPy | ENUM trunk certainly looks even mor senseless than the damous SIP trunk. |
| 11:28.17 | mzb | hmm |
| 11:28.54 | WIMPy | Damn |
| 11:29.01 | WIMPy | ENUM trunk certainly looks even more senseless than the famous SIP trunk. |
| 11:29.26 | mzb | good for testing some aspects of a virgin system |
| 11:32.46 | mzb | (along with an "extension" that does direct sip dial) |
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| 11:44.40 | Weazel | hey guys, i'm looking for a way i can tell an ext to be able to dial only internally + feature codes + specific numbers |
| 11:51.42 | Tozz_ | Yes |
| 11:52.17 | kaldemar | Weazel: make a context that allows what you want. |
| 11:53.33 | Weazel | kaldemar: i'm still a noob with syntaxes |
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| 11:58.02 | kaldemar | Weazel: learn dialplan basics and you'll be able to do that. |
| 11:58.05 | kaldemar | ~newbook |
| 11:58.05 | infobot | Asterisk: The Definitive Guide. A preview of the upcoming release is available at http://ofps.oreilly.com/titles/9780596517342/. The book is being released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/ |
| 11:58.30 | Weazel | thanks |
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| 12:05.17 | phyber | What's the best way to secure Asterisk? I'm reading http://www.voip-info.org/wiki/view/Asterisk+IAX+authentication and it seems like mission impossible. |
| 12:05.28 | phyber | asterisk will do anything that it can to let a user connect |
| 12:05.50 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 12:06.12 | phyber | just randomly picking a user and letting something authenticate if it doesn't provider a user but happens to provide a password that matches something? that's horrible. (unless you can demonstrate some awesome use for it) |
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| 12:11.56 | kaldemar | phyber: use secrets or keys and ACL. |
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| 12:12.19 | oelewapperke | How can you detect which phone(s) issued SIP REDIRECTS in the dialplan ? |
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| 12:34.52 | SirSquidness | I've got chan_sccp_b set up (latest trunk from SVN) with Asterisk 1.6.2.16.1. I have four Cisco IP phones (three 7906G and a 7945). After powering the phones on, the first call to another phone will work and any calls after that will be silent. The asterisk log indicates the phones report sending 0 packets. |
| 12:35.06 | SirSquidness | Any suggestions, or even a "yes, it works for me, you're doing something wrong" would be swell :) |
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| 13:01.45 | tzafrir | phyber, "secure" could mean a host of different things. For starters, hire a guard to prevent physical access to the server |
| 13:02.35 | tzafrir | What sorts of threats do you have in mind? |
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| 13:50.28 | skrusty | afternoon |
| 13:50.50 | skrusty | is there a way to send custom manager events from an AGI? |
| 13:51.03 | skrusty | if so, what are peoples thoughts on how easy this would be to implement? |
| 13:51.07 | skrusty | if not even |
| 13:53.25 | WIMPy | Why don't you communicate directly? |
| 13:54.27 | skrusty | i have an app that listens to manager events, and i want to send some information to it from a running AGI |
| 13:54.41 | skrusty | so it gets handled in the same way any other * event does |
| 13:55.00 | skrusty | so, just wondered if such a function existed, and if not, if there was a reason why not :) |
| 13:55.52 | Kobaz | skrusty: UserEvent |
| 13:56.03 | skrusty | ah :) |
| 13:56.13 | skrusty | i thought there was something like that, but it's been a while :) |
| 13:56.16 | skrusty | thanks Kobaz |
| 13:56.30 | Kobaz | yeah, it rebroadcasts whatever you send it as an event |
| 13:56.43 | skrusty | excellent, just what i wanted, thanks! |
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| 14:17.03 | *** join/#asterisk donatas (~donatas@office.kis.lt) |
| 14:17.20 | donatas | hey. how can I do a call forward if one channel is busy? |
| 14:19.47 | Kobaz | it would be if a device is busy |
| 14:19.50 | Kobaz | you can use DEVICSTATE |
| 14:20.41 | donatas | okey, but how to forward? |
| 14:21.00 | *** join/#asterisk boedabelg (~kvirc@41.251.37.147) |
| 14:22.31 | Kobaz | Dial() |
| 14:22.38 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-qtrrtfmlwwitmxyp) |
| 14:22.42 | Kobaz | er DEVICESTATE.. missed the E |
| 14:24.37 | no1peanut | I would like to use background during a conversation - can that be done ? |
| 14:28.10 | donatas | If I want to forward blind, I can use only Dial? |
| 14:29.58 | Kobaz | there's also Transfer() |
| 14:30.38 | Kobaz | which is a request for remove transfer |
| 14:30.40 | Kobaz | remote |
| 14:30.48 | Kobaz | you'll probably want to use Dial() though |
| 14:31.43 | donatas | understand |
| 14:32.43 | *** join/#asterisk xheliox (jeff@pdpc/supporter/student/xheliox) |
| 14:33.04 | donatas | Transfer(SIP/33331/remote_number); ? |
| 14:33.06 | donatas | like that? |
| 14:33.19 | donatas | SIP/33331 is my peering site |
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| 14:36.53 | *** join/#asterisk Mhaddog__ (~Mhaddog@adsl-072-149-063-056.sip.bct.bellsouth.net) |
| 14:38.20 | matach | Hi , Can anyone explain me the difference between "port" and "bindport" in [general] section of sip.conf ? |
| 14:40.37 | matach | should I use 'bindport" just in case of using NAT ? |
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| 14:44.42 | malcolmd | Asterisk Weekly Developer Conference Call beginning in 15 minutes! https://wiki.asterisk.org/wiki/display/AST/Asterisk+Weekly+Developer+Conference+Call |
| 14:46.45 | *** join/#asterisk philfine (~philfine@a213-22-178-36.cpe.netcabo.pt) |
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| 14:51.13 | malcolmd | Asterisk Weekly Developer Conference Call beginning in 10 minutes! https://wiki.asterisk.org/wiki/display/AST/Asterisk+Weekly+Developer+Conference+Call |
| 14:51.18 | Kobaz | aww, i keep missing them |
| 14:51.19 | Kobaz | er |
| 14:51.26 | Kobaz | stupid customers |
| 14:52.25 | *** join/#asterisk matach (~matach@170-117.76-83.cust.bluewin.ch) |
| 14:53.27 | leifmadsen | Kobaz++ |
| 14:54.59 | *** join/#asterisk funpunx (daemon@173-15-183-209-BusName-Philadelphia.hfc.comcastbusiness.net) |
| 14:55.43 | Kobaz | heh |
| 14:55.48 | malcolmd | Asterisk Weekly Developer Conference Call beginning in 5 minutes! https://wiki.asterisk.org/wiki/display/AST/Asterisk+Weekly+Developer+Conference+Call |
| 14:58.07 | malcolmd | sip:100@devcon.asterisk.org skype:devcall.digium tel:18772747510 bridge: 26052 pin: 35806 |
| 15:00.13 | malcolmd | Asterisk Weekly Developer Conference Call beginning in 1 minute! https://wiki.asterisk.org/wiki/display/AST/Asterisk+Weekly+Developer+Conference+Call sip:100@devcon.asterisk.org skype:devcall.digium tel:18772747510 bridge: 26052 pin: 35806 |
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| 15:23.26 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 15:23.36 | Sertys | i need some advice on using srtp |
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| 15:24.34 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:24.43 | pabelanger | ~ask |
| 15:24.44 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 15:24.47 | pabelanger | Sertys: ^ |
| 15:25.24 | malcolmd | Sertys: couple of wiki pages here: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling |
| 15:25.28 | Sertys | i'm intending to use srtp with nokia voip clients, which might come out a bit cluterry |
| 15:25.54 | Sertys | and i've deployed my asterisk from src, but it's not svn |
| 15:26.19 | malcolmd | was the src version 1.8? |
| 15:26.27 | Sertys | it's 1.6 branch |
| 15:26.33 | Sertys | should i migrate to 1.8 already? |
| 15:26.38 | malcolmd | srtp support didn't make it in until 1.8 |
| 15:26.42 | malcolmd | 1.6 has tls, but no srtp |
| 15:26.47 | Sertys | does it have srtp support out of the box |
| 15:26.52 | malcolmd | 1.8 does, yes |
| 15:27.01 | malcolmd | upi |
| 15:27.02 | malcolmd | er... |
| 15:27.29 | malcolmd | you'll have to enable it, but that's done easily through configuration - you don't need anything magical |
| 15:28.03 | malcolmd | you will need libsrtp installed before you start building asterisk 1.8 though if you want to use srtp |
| 15:33.27 | ariel_ | anyone have a very simple *72 *73 setup for sip phones that they are willing to share? very simple no gui, no complicated dial plan |
| 15:34.01 | *** join/#asterisk cd101 (d07cd9d5@gateway/web/freenode/ip.208.124.217.213) |
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| 15:38.23 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 15:38.26 | Sertys | re |
| 15:40.53 | Sertys | i |
| 15:41.11 | Sertys | have already deployed vpns for the clients, so srtp is not a necessity for me |
| 15:41.20 | Sertys | but i'd like to test it otu |
| 15:41.29 | Sertys | that's why i was asking for someone with xp on the field |
| 15:42.19 | cd101 | Can anyone tell me how I would start to resolve a "Received Response: Forbidden" message? This occurs while trying to send a call out from an asterisk server in a VM to an asterisk server on the host computer. The VM is a custom build of asterisk, and the host is configured using freepbx |
| 15:44.04 | cd101 | The call is going out from the VM, and the host displays only: == Using SIP RTP TOS bits 186\n== Using SIP RTP CoS mark 5 |
| 15:45.11 | cd101 | sip show peers on both appears correct |
| 15:49.16 | Sertys | cd101: check out the insecure parameter |
| 15:49.52 | *** join/#asterisk philfine (~philfine@a213-22-178-36.cpe.netcabo.pt) |
| 15:51.47 | Kobaz | anyone have any good info on writing polycom xml apps |
| 15:52.45 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 15:55.42 | cd101 | insecure is set port,invite |
| 15:58.07 | *** part/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 15:58.09 | cd101 | is it a problem on the host or the vm? I would expect that the host is rejecting, but there is no output on the host... |
| 15:58.24 | cd101 | just the Using SIP ..... |
| 16:05.43 | cd101 | its listed as not working in Asterisk 1.6+ |
| 16:07.06 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 16:11.31 | Sertys | and u're dialin out how? |
| 16:11.41 | Sertys | with .call files or a client? |
| 16:12.07 | Sertys | what does ur extensions.conf look like? |
| 16:12.24 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 16:28.36 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 16:29.44 | *** join/#asterisk damencho (~damencho@2001:470:1f13:536:21f:5bff:fef5:d2ef) |
| 16:33.55 | *** join/#asterisk marl_scot (~matt.lowe@office.unk.com) |
| 16:36.50 | damencho | hi all, I have a question about asterisk 1.8.2.3 installed and used by not so many users on a server machine happens from time to time to hangs for 4-5 minutes, and I'm wondering what can be the problem? and how to debug it? |
| 16:37.34 | leifmadsen | check logs to see where it hangs (what was it doing, etc), file an issue and provide output of 'core show locks' and a backtrace from the running process |
| 16:37.46 | *** join/#asterisk awclin_ (~alinford@g0962184.demon.co.uk) |
| 16:37.59 | leifmadsen | ~debugging |
| 16:37.59 | infobot | if debugging is the process of removing bugs, then programming must be the process of putting them in. |
| 16:38.10 | leifmadsen | ~asteriskdebug |
| 16:38.23 | damencho | most of the users are connected with tcp, and sometimes after it hangs and start running normally it for example sends all the invites that were stuck for that time |
| 16:38.24 | WIMPy | ~collectdebug |
| 16:38.24 | infobot | it has been said that collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 16:38.30 | leifmadsen | perfect |
| 16:38.31 | leifmadsen | thanks! |
| 16:38.36 | damencho | there are no logs during that time |
| 16:38.39 | _Corey_ | I like the definition :) |
| 16:38.48 | WIMPy | :-) |
| 16:39.06 | leifmadsen | damencho: the other forms of debug information is still relative. The logging might include SIP traces, etc. |
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| 16:48.17 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 16:49.31 | *** join/#asterisk lordvadr (~something@jose-tc.ctc.biz) |
| 16:50.13 | lordvadr | Is there any mechanism to pop a call off of a ringing queue and answer it? |
| 16:50.24 | lordvadr | without being a member of the queue |
| 16:51.00 | kaii | pickup one of the ringing channels? |
| 16:51.11 | lordvadr | that doesn't work |
| 16:51.26 | kaii | what kind of pickup do you use? |
| 16:52.02 | lordvadr | I'll have to look, it's not a pbx I designed. We're trying to find a work around for this problem... brb |
| 16:52.41 | cd101 | Sertys: Sorry for the lack of response...I am dialing out from the VM using a physical phone. This is what I have for extensions.conf in the VM: |
| 16:52.45 | cd101 | exten => 2523,1,Macro(phone) exten => _2XXX,1,Log(NOTICE, Dialing out from ${CALLERID(all)} to ${EXTEN} through STEP trunk) exten => _2XXX,n,Dial(SIP/step/${EXTEN},60) |
| 16:54.47 | cd101 | the VM outputs the notice to the console, then displays the WARNING handle_response_invite: Received response: "Forbidden" from 2523 |
| 16:55.03 | Sertys | and why are u using SIP/step/${EXTEN} as an uri, is this how ur peer is registered |
| 16:56.19 | cd101 | first, because I am not knowledgable on the subject, and this is what the "how to" instructed....but yes, the peer is registered as step |
| 16:57.00 | Sertys | and where is the macro-phone section? |
| 16:57.49 | cd101 | should I paste it here? |
| 16:58.02 | cd101 | its about 8 lines |
| 16:58.41 | *** join/#asterisk zooz (~zooz@188-220-41-110.zone11.bethere.co.uk) |
| 16:58.43 | zooz | hi people |
| 16:58.58 | zooz | I am trying to register to my SIP VoIP provider |
| 16:59.39 | zooz | my asterisk server is able to register with it, but when I place a call to the number which i got assigned from my voip provider it goes to s:default dialplan |
| 16:59.45 | damencho | leifmadsen: thank you |
| 17:00.45 | *** join/#asterisk JonnyD_work (~Jon@12.222.63.34) |
| 17:00.55 | zooz | my login details from the voip provider looks like this: username:password@provider.tld, the phone number is different from the username |
| 17:01.02 | zooz | this is what confuses me |
| 17:01.09 | titter | Is there a reason MeetMe(blah,dicP,1234) would ignore the pin? |
| 17:01.46 | zooz | how do I describe my sip provider in sip.conf ? |
| 17:02.02 | zooz | register => username:password@provider.tld |
| 17:02.08 | zooz | this is pretty straight forward |
| 17:02.20 | zooz | but I am not sure what I need to put below in [ ] |
| 17:02.41 | zooz | anyone any ideas? |
| 17:03.20 | cd101 | Sertys: The entire extensions.conf file @ http://pastebin.com/xC0w0TpM |
| 17:06.01 | cd101 | VM console output: @ http://pastebin.com/DwMkN9KR |
| 17:06.24 | *** join/#asterisk mort_gib (~mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 17:12.34 | lordvadr | kaii: Pickup(${EXTEN}@context) |
| 17:13.54 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.234.213) |
| 17:17.33 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
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| 17:26.15 | titter | Found the answer to my own question - https://issues.asterisk.org/view.php?id=18070 ... this did work in 1.6.1.17, bummer! |
| 17:27.58 | *** join/#asterisk khlaed (~khlaed@ppp-119-10.pppoe.canl.vu) |
| 17:28.57 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-156-107.dsl.stlsmo.sbcglobal.net) |
| 17:29.56 | LemensTS | Anyone know what form has IVR information in it for SX-200? |
| 17:29.57 | Qwell | what? |
| 17:30.20 | LemensTS | Mitel SX-200. I know some of you work on those. Didn't see a MITEL room. |
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| 17:31.13 | drmessano-lt | Where's Anonymous when you need then |
| 17:31.15 | drmessano-lt | them |
| 17:31.34 | WIMPy | Is there a way to hand parameters with commas to Macro()? Dial accepts \, but Macro doesn't seem to do so. |
| 17:31.44 | marl_scot | hi there, i have an * 1.8 (SVN-trunk-r297536) with a BeroNet BN4SminiPCI ISDN2 4 port card (this is one of 2 identical machines, the first of which i have nno problems with). This is running lcr 20100601, every so often (seams to be about twice a day, and only during hte day) * stops responding to SIP calls and to anything from the LCR, i can still see calls coming in via lcradmin, but * doesnt seem to register that anything is coming in, at the s |
| 17:31.45 | marl_scot | ame time, my SIP hadnsets (YEalinks) stop being able to call each other, or make outbound calls :( |
| 17:31.46 | pabelanger | LemensTS: The SX200 is a PBX, with very basic voicemail menu tree support for IVRs |
| 17:32.05 | *** join/#asterisk Tozz_ (Tozz@hardwire.duocast.net) |
| 17:32.09 | pabelanger | Also does not support native IP |
| 17:32.28 | LemensTS | pabelanger: Oh it uses the voicemail forms for the IVR |
| 17:32.42 | LemensTS | We have a icp controller on it. |
| 17:32.54 | pabelanger | LemensTS: yes, like I said limited support |
| 17:33.03 | WIMPy | marl_scot: Do you have IAX in use? |
| 17:34.11 | marl_scot | i have a couple of IAX accounts setup, but am not using them just now |
| 17:34.44 | WIMPy | There is something up on mantis regarding IAX and lockups. |
| 17:35.13 | WIMPy | It hasn't happened to me for some versions but it seems to be an issue for others. |
| 17:35.49 | marl_scot | disable all iax maybe? |
| 17:36.53 | marl_scot | strange thing is when sip/lcr stop working i can still asterisk -r and run normal commands etc. only problem seems to be that the stop command doenst work, i have to kill the process with sugnal '9' |
| 17:37.56 | WIMPy | There is an issue with can_lcr. You can only 'stop now'. The other options make Asterisk hang. |
| 17:39.54 | tash | for some reason when a call isn't answered by a user, the retries fail over and over. In fact, I never see them in the asterisk console. The only thing different that I noticed about the .call file's mtime/timestamp is that it doesn't have 00 for 'seconds' ... any clues? |
| 17:39.54 | marl_scot | ok, will make sure thats the first one i try when it goes again, think i did try it before, but only after trying the init script stop |
| 17:40.25 | WIMPy | I usually find that using the tab key on *cli is a perfect indication of Asterisk's state. |
| 17:40.42 | WIMPy | Usually that locks up the shell if something is going on. |
| 17:42.26 | marl_scot | nope tab key shows and compleates commands with no problems |
| 17:43.03 | WIMPy | Oh, I've never had that working with anything else stuck. |
| 17:43.14 | *** join/#asterisk TimeRider (~steve@92.27.131.175) |
| 17:44.59 | marl_scot | lol, i ALWAYS have the problems no one else has had :( |
| 17:45.20 | WIMPy | Ha, I know that all to well. |
| 17:45.37 | WIMPy | Have you tried to originate a call from the console to some sip phone? |
| 17:45.56 | marl_scot | nope, how would i do that? |
| 17:46.06 | WIMPy | Do you see anything on *cli at all in that situation? |
| 17:46.18 | WIMPy | channel originate ... |
| 17:47.49 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 17:47.57 | marl_scot | dont seem to remember seeing anyting other than responses to my commands when i connect with 'asterisk -rv' i should probably try with -rvvvvv or something to get more data |
| 17:48.59 | WIMPy | tries to think harder. |
| 17:49.25 | WIMPy | I think I have had a situation like that. But I can't remember when or why. |
| 17:49.42 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 17:50.56 | anonymouz666 | ~newbook |
| 17:50.57 | infobot | Asterisk: The Definitive Guide. A preview of the upcoming release is available at http://ofps.oreilly.com/titles/9780596517342/. The book is being released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/ |
| 17:51.04 | anonymouz666 | Sorry I just miss the link |
| 17:53.06 | zooz | I got totally confused :-( |
| 17:53.22 | zooz | my asterisk registered with SIP gw |
| 17:53.44 | zooz | but for some reason it goes to default dialplan |
| 17:54.04 | zooz | even though I specified it to go to context=incominf |
| 17:54.13 | zooz | context=incoming* |
| 17:54.18 | hardwire | your context has a lisp. |
| 17:56.15 | *** join/#asterisk F0x24 (~Fox@AToulouse-553-1-241-103.w92-162.abo.wanadoo.fr) |
| 17:56.20 | F0x24 | Hello |
| 17:56.37 | zooz | hardwire, could you please have a look: http://p.defau.lt/?W_CMdv_E5_ZlUSOG44XG0w |
| 17:56.44 | zooz | that's my sip.conf |
| 17:57.00 | hardwire | I was just trying to be funny |
| 17:57.10 | hardwire | did it inadvertantly involve me in your problem? |
| 17:57.12 | hardwire | karma says maybe. |
| 17:57.48 | F0x24 | Please help me, i have no sound when i try to connect to my server from outside. |
| 17:58.06 | F0x24 | i'm connected to server, but no sound in calls |
| 17:58.10 | skrusty | quick question: does anyone know why in the sip header from a linksys phone, to asterisk, the from and to are both the server and not just to to? |
| 17:58.13 | zooz | F0x24, nat issues? |
| 17:58.30 | F0x24 | i guess not |
| 17:58.33 | F0x24 | i openned ports |
| 17:58.36 | zooz | F0x24, I will rephrase my answer: issues due to NAT? |
| 17:58.42 | hardwire | zooz: turn on sip debugging.. make a call in .. see if it relates to the correct user. |
| 17:59.01 | F0x24 | i think no |
| 17:59.15 | zooz | hardwire, did not know if that was possible, thanks :-) |
| 17:59.24 | hardwire | sip set debug on |
| 17:59.35 | hardwire | F0x24: you too.. sip set debug on. |
| 17:59.40 | F0x24 | ok |
| 17:59.47 | hardwire | check out specifically the RTP portion of the SIP/SDP |
| 17:59.51 | *** join/#asterisk dank14 (~dank14@78.32.115.1) |
| 17:59.54 | F0x24 | yes |
| 17:59.58 | hardwire | hopefully it will enlighten you into a facepalm |
| 18:00.17 | *** join/#asterisk lanning (~lanning@208.87.233.137) |
| 18:00.48 | dank14 | can i ask who would be interested in helping debug pri disconnects? |
| 18:00.53 | hardwire | zooz: turn off insecure first |
| 18:00.59 | zooz | hardwire, where does the username should be visible? |
| 18:01.09 | zooz | To: username@ ? |
| 18:01.39 | hardwire | bbl |
| 18:01.43 | hardwire | I need to do my own work |
| 18:01.45 | F0x24 | I openned : 5004-5082 |
| 18:01.46 | F0x24 | 10000-20000 |
| 18:01.46 | F0x24 | is it correct ? |
| 18:02.00 | marl_scot | to use channeloringinate, from the cli, what do i use for the 'tech/data' ? i know i need to use exten@context for the extension (am using SIP/LCR) |
| 18:02.01 | F0x24 | udp and tcp |
| 18:02.50 | WIMPy | channel originate SIP/somephone application sayunixtime |
| 18:05.01 | WIMPy | marl_scot: Both machines have the same software versions and same hardware? |
| 18:05.33 | marl_scot | ah, after running the command 'channel originate SIP/702 application sayunixtime' i can hit return, but if i try tab, the cli locks uintil the command times out (no one answering the extension just now) |
| 18:05.57 | WIMPy | That's normal. |
| 18:05.59 | marl_scot | WIMPy, yup, only differances has been in the extensions.conf files |
| 18:06.06 | WIMPy | But did it ring? |
| 18:06.13 | marl_scot | ah ok, thougt i might have hit on somehting :( |
| 18:06.24 | hardwire | F0x24: tcpdump is your beeyatch. |
| 18:06.55 | *** join/#asterisk dzup2 (~alex@unaffiliated/dzup) |
| 18:07.18 | WIMPy | marl_scot: Really the same versions and the same configuration? i.e. same modules loaded? |
| 18:07.26 | hardwire | people.. there are tools.. many tooooools! |
| 18:07.37 | dzup2 | if i want to send audio to my dsp (computer audio output jack) from CLI or from a extention to dsp, how? |
| 18:08.07 | leifmadsen | dzup2: use JACK |
| 18:08.13 | WIMPy | dzup2: Chan_alsa, chan_oss, chan_console or use a spftphone. |
| 18:08.17 | marl_scot | once the first box was up and running, i cloned the hard drive using 'dd' re-installed grub, and tested the machine in that office, then took machine down to second office, and changed the dialplans |
| 18:08.19 | leifmadsen | sorry, I read that wrong |
| 18:08.23 | leifmadsen | dzup2: what WIMPy said |
| 18:08.28 | dzup2 | WIMPy: console/JACK ? |
| 18:08.29 | *** join/#asterisk timahvo1 (~rogue@41.223.57.78) |
| 18:08.38 | leifmadsen | dzup2: no ignore the JACK part I misread |
| 18:08.51 | leifmadsen | dzup2: chan_alsa, or chan_oss is your best bet |
| 18:08.54 | dzup2 | /console/chan_console ? |
| 18:08.58 | *** join/#asterisk Tim_Toady (~moi@178.128.24.211.dsl.dyn.forthnet.gr) |
| 18:09.02 | dzup2 | ahh |
| 18:09.27 | dzup2 | if i create a custom ext then console/chan_alsa ? |
| 18:09.42 | WIMPy | marl_scot: Hmm. That's not that helpful. I still don't know when I hit sich a thin. |
| 18:10.02 | WIMPy | I think I had a version once whe chan_lcr locked up. |
| 18:10.10 | F0x24 | hardwire: you want me to run wireshark ? |
| 18:10.13 | WIMPy | But I can't remember when. |
| 18:10.55 | marl_scot | if chan_lcr locked up, would 'lcradmin state' still show when calls where coming in, and show there cli/ddi etc ? |
| 18:11.18 | WIMPy | Yes./ I thought you said, it did? |
| 18:12.18 | WIMPy | But possibly only for a very short time. |
| 18:12.28 | hardwire | F0x24: run whatever gets you knowledge. |
| 18:13.02 | leifmadsen | dzup2: I think it's just Dial(Console/DSP) |
| 18:13.09 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 18:13.34 | F0x24 | hardwire: In SIP stats i got SIP 401 Unauthorized and SIP 404 Not Found |
| 18:13.41 | marl_scot | have been unable to let it run in an error state longer than about 10 mins |
| 18:13.49 | leifmadsen | dzup2: although I've never done that (oddly enough in all these years) |
| 18:13.52 | F0x24 | but also SIP 200 OK |
| 18:14.14 | marl_scot | is it posible to reload chan_lcr without restarting asterisk? |
| 18:15.21 | WIMPy | marl_scot: No, that's the same bug that prevents core stop/restart when ... from working. |
| 18:15.40 | WIMPy | Or at least I think so. Iihaven't tried to look at that part. |
| 18:16.11 | WIMPy | But LCR writes its own log when you can see what cam in. |
| 18:16.13 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 18:16.23 | dank14 | does anyone know about this in pri debug T200 requested to stop when not started? |
| 18:16.46 | killown | I am trying set up my voip account http://bpaste.net/show/13757/ and I get 302 moved temporarily back from 200.147.35.32, do anyone know what means the 302 error? |
| 18:17.13 | F0x24 | hardwire : i think i'm forbidden to call from outside but why ? (in local it works) |
| 18:17.31 | hardwire | but why indeed |
| 18:17.31 | retentiveboy | Anybody know why res_jabber and chan_gtalk are not available via the RPMs in the YUM repository? |
| 18:17.33 | hardwire | I'm not your network |
| 18:17.43 | F0x24 | lol |
| 18:17.52 | hardwire | stare at the sip traces until your eyes bleed |
| 18:17.53 | Qwell | retentiveboy: there is no iksemel package right now |
| 18:17.59 | F0x24 | is there any option to write in sip.conf ? |
| 18:18.07 | F0x24 | except : nat=yes |
| 18:18.10 | hardwire | stare at the sip traces until your eyes bleed |
| 18:18.28 | killown | can anyone help me? |
| 18:18.42 | zooz | I see more people online now, I will ask my question again: I got asterisk to register to my SIP gw, but when I dial to the number I got assigned by my SIP gw the call goes to default context, but I have different context specified. Here is my sip.conf --> http://p.defau.lt/?W_CMdv_E5_ZlUSOG44XG0w . any ideas why this is happening? |
| 18:18.51 | retentiveboy | Qwell: it's available in EPEL |
| 18:19.03 | *** join/#asterisk matach (~matach@194-70.76-83.cust.bluewin.ch) |
| 18:19.07 | retentiveboy | guess that's not blessed yet |
| 18:19.09 | Qwell | retentiveboy: EPEL breaks things in the Digium repos |
| 18:19.41 | retentiveboy | Qwell: yeah, see that |
| 18:20.24 | *** part/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 18:21.30 | WIMPy | dank14: Just ignore it. |
| 18:22.05 | *** join/#asterisk BlackBishop (dexter@ipv6.d3xt3r01.tk) |
| 18:22.21 | dank14 | ta |
| 18:22.36 | *** join/#asterisk manji (~manjiki@adsl-142-168.adsl.ntua.gr) |
| 18:23.19 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 18:23.20 | dank14 | stuffed with these disconnects on te121 any suggestions |
| 18:23.33 | WIMPy | is out for a meeting. BBL. |
| 18:23.41 | BlackBishop | if I have something like this: |
| 18:23.41 | BlackBishop | exten => s,n,Set(testlocal="0000000000") |
| 18:23.41 | BlackBishop | exten => s,n,Dial(LOCAL/0000000000@robotel) |
| 18:23.48 | BlackBishop | shouldn't test local also be available in [robotel] ? |
| 18:24.05 | dzup2 | WIMPy, leifmadsen if i want to identify the JACK (microphone) device from console is console/JACK or is there a list of local devices i can use of * to redirect the input from a telefone to one of those (dsp, jack, console) am trying to get from extention voice to my audio dsp, thank you |
| 18:25.00 | leifmadsen | dzup2: I told you to ignore what I said about JCK |
| 18:25.02 | leifmadsen | JACK* |
| 18:25.11 | leifmadsen | if you want to use JACK then look at app_jack |
| 18:25.19 | leifmadsen | but JACK is probably not what you want |
| 18:25.32 | *** join/#asterisk dajhorn (~dajhorn@adsl-75-17-124-41.dsl.rcsntx.sbcglobal.net) |
| 18:25.41 | leifmadsen | BlackBishop: not if you don't make the channel variable become inherited |
| 18:25.56 | leifmadsen | exten => s,n,Set(_testlocal=0000000000) |
| 18:26.10 | dzup2 | leifmadsen: ok, ignoring jack |
| 18:26.36 | leifmadsen | BlackBishop: you're calling a new channel, so the variable you've set is not associated with that channel. Use single underscore for single channel inheritance or double underscore for infinite inheritance |
| 18:28.15 | BlackBishop | ow |
| 18:28.19 | BlackBishop | learned something new |
| 18:29.01 | leifmadsen | :) |
| 18:30.28 | leifmadsen | if you were looking at a webpage and saw a list of icons that were really simple (basically thing stick figures, single colour), what would represent a page for creating "administrators" |
| 18:30.52 | leifmadsen | actually, perhaps it'd make more sense for a page to chang permissions |
| 18:31.09 | Kobaz | stick figure above a group of stick figures on the bottom |
| 18:31.21 | leifmadsen | with a large head |
| 18:31.24 | Kobaz | yeah |
| 18:31.28 | Kobaz | and pointy hair |
| 18:31.32 | leifmadsen | and a beard |
| 18:32.07 | Kobaz | the little stick figures on the bottom should also be begging for mercy |
| 18:32.40 | leifmadsen | lol |
| 18:33.10 | *** join/#asterisk Diffen2 (~diffen2@c-2472e555.042-17-73746f11.cust.bredbandsbolaget.se) |
| 18:34.09 | *** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
| 18:34.13 | *** join/#asterisk bn-7bc (~bjarne-im@m83-177-58-150.cust.tele2.no) |
| 18:36.07 | Diffen2 | Evning. Is it possible to set an extension state via the asterisk cli? |
| 18:38.35 | p3nguin | I don't know if extensions have states at all. |
| 18:38.43 | p3nguin | I know devices have states. |
| 18:38.54 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 18:39.32 | Diffen2 | Hmm ok |
| 18:39.38 | leifmadsen | it's possible... but you'd have to set a custom device state, then execute it by dialing a local channel that executes dialplan from the console -- it's a bit of a haphazard way to do it, but it's certainly possible. |
| 18:39.53 | leifmadsen | there is no, "device set state <foo> for <bar>" |
| 18:40.11 | *** join/#asterisk cd101 (d07cd9d5@gateway/web/freenode/ip.208.124.217.213) |
| 18:40.51 | F0x24 | my client can't answer the SDP paquet, what does that mean please ? |
| 18:41.00 | Diffen2 | ok what I want to achieve is that using extension state for users that only uses cell phones and the find me function. so when a call is sent to their cell phone the extension should be set to busy |
| 18:41.07 | BlackBishop | leifmadsen: then I should be able to use {testlocal} (without any _ ) in [robotel] right ? |
| 18:41.34 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 18:42.06 | BlackBishop | I'm still asking because I did a exten => _XXXXXXXXXX,1,Verbose(testlocal=${testlocal}) in [robotel] .. and it says testlocal= nothing :| |
| 18:42.26 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-156-107.dsl.stlsmo.sbcglobal.net) |
| 18:43.37 | killown | I am trying set up my voip account http://bpaste.net/show/13757/ and I get 302 moved temporarily back from 200.147.35.32, do anyone know how fix it? |
| 18:44.36 | Qwell | there's nothing to "fix". 302 is a normal response, and the new address given will be tried |
| 18:44.45 | Qwell | if it's causing breakage, yell at your provider |
| 18:45.41 | p3nguin | diffen2: BUSY is a DIALSTATUS returned on a channel by a device. That has nothing to do with extensions. |
| 18:46.09 | pabelanger | or set promiscredir = yes and handle it within your dialplan |
| 18:46.29 | *** join/#asterisk dank14 (~dank14@87.127.172.193) |
| 18:46.43 | BlackBishop | damn variables |
| 18:47.09 | p3nguin | I still doubt that extensions have states. It's only been like eight minutes since the last time I told you this -- I doubt anyone had time to do the magic stuff and create code for it. |
| 18:47.12 | Diffen2 | p3nguin ok so in order to achieve this i need to have a device registered and that are called simultanious as the cell phone? |
| 18:47.45 | p3nguin | It is possible for BUSY to be returned on the channel used to dial the cell phone. |
| 18:48.05 | p3nguin | But the channel does need to be busy in order for it to happen. |
| 18:48.26 | Diffen2 | ok but i the call is picked up on the cell phone is it possible to set the device to busy even if its not busy? |
| 18:48.31 | p3nguin | Since the cell phone won't be a device "on asterisk," asterisk won't know its state. |
| 18:49.03 | Mhaddog__ | u have g729 support in your box? |
| 18:50.02 | dzup2 | leifmadsen: if i make a custom extention to point to Console/dsp <-- is that the correct way to call the dsp device of * ? |
| 18:51.49 | dank14 | any reasons in particular that conference and mobile calls get disconnected randomly? |
| 18:53.36 | *** join/#asterisk ubuspire (~ubuspire@41.190.129.207) |
| 18:54.17 | ubuspire | hi all, when using res_pgsql on 1.6.x must I create an extensions_conf entry for each sip peer in the database? |
| 18:54.53 | ubuspire | (i'm currently failing to get the DB peers to logon with the error matching peer not found) |
| 18:56.23 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 18:59.17 | dank14 | anyone help with WARNING[16670] pbx.c: Context 'app-blacklist-add-invalid' tries to include nonexistent context 'app-blacklist-add-invalid-custom' |
| 18:59.42 | Qwell | the error is pretty uambiguous |
| 18:59.56 | dank14 | not to me |
| 19:00.03 | pabelanger | dank14: *CLI> dialplan show app-blacklist-add-invalid-custom |
| 19:02.08 | dank14 | failed |
| 19:02.21 | justdave | ok, I'm having an interesting problem... I have an office in Paris, with a trunk to the main office over iax2. When we place station-to-station calls from paris to HQ or vice versa, they're nice and clear. If anyone from Paris dials into a Meetme conf in HQ, they're all choppy. |
| 19:02.39 | justdave | anyone in HQ is nice and clear in the conferences. |
| 19:02.49 | justdave | (and anyone from the Toronto office is clear, too) |
| 19:03.16 | pabelanger | dank14: that is your issue then |
| 19:03.22 | dank14 | ok |
| 19:04.19 | *** join/#asterisk jkroon (~jkroon@dsl-241-228-69.telkomadsl.co.za) |
| 19:06.18 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 19:08.13 | ubuspire | rephrases. |
| 19:08.58 | ubuspire | i keep getting http://pastie.org/1575775 for any sip peer that is in my database. clues on where to start looking? |
| 19:09.46 | pcangel | I have a macro called macro-dialout that does include => macro-dialout-custom, and I have to inject a MixMonitor line in to the Macro's s extension ..from macro-dialout-custom... is this possible? |
| 19:10.37 | titter | justdave: You can implement a jitter buffer by using chan_local ... iirc iax2 to meetme doesn't bridge two channels, which doesn't allow for any jb to be applied |
| 19:11.02 | pabelanger | ubuspire: *CLI> sip show peer 0202400204 |
| 19:11.55 | justdave | hmmm |
| 19:12.27 | justdave | how would that work, incoming iax2 goes into a fake exten that dials again using Local/ ? |
| 19:12.50 | justdave | so then it gets dejittered on that bridge before connecting to meetme? |
| 19:13.09 | Qwell | justdave: exactly |
| 19:13.15 | ubuspire | pabelanger: peer 0202400204 not found. i get that for any peer in the database (using res_pgsql) and not for any statically defined peers. |
| 19:13.23 | titter | justdave: Correct, you need to use option j |
| 19:13.25 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 19:13.28 | Qwell | ubuspire: then your odbc config isn't setup properly |
| 19:13.37 | pabelanger | ^^ |
| 19:15.30 | Kobaz | do cisco 7960's play nice with asterisk |
| 19:15.33 | BlackBishop | -- Executing [0760905294@users:2] GotoIf("SIP/500-00000000", ""5" = 0]?read:call") in new stack .. it should go to (call) right ? |
| 19:15.35 | Kobaz | it looks like you can load sip firmware on them |
| 19:15.38 | BlackBishop | since 5 != 0 ! |
| 19:15.55 | titter | justdave: Dial(Local/someexten@somecontext/nj) is what I do in my dp |
| 19:16.00 | Kobaz | BlackBishop: GotoIf $[5 == 0]... |
| 19:16.12 | _Corey_ | Kobaz: The 7960s work fine |
| 19:16.13 | titter | justdave: Had similar issues until I implemented that from remote iax users. |
| 19:16.20 | BlackBishop | the whole line is like this |
| 19:16.20 | BlackBishop | exten => _XXXXXXXXXX,n,Gotoif("${CALLERID(num):0:1}" = "0"]?read:call) |
| 19:16.37 | dzup2 | module load /usr/lib/asterisk/modules/chan_oss.so , Unable to load module /usr/lib/asterisk/modules/chan_oss.so , Command 'module load /usr/lib/asterisk/modules how load a module then?/chan_oss.so' failed. |
| 19:16.40 | p3nguin | |
| 19:16.40 | p3nguin | |
| 19:16.48 | Kobaz | BlackBishop: you have a syntax error |
| 19:16.58 | dzup2 | module load /usr/lib/asterisk/modules/chan_oss.so , Unable to load module /usr/lib/asterisk/modules/chan_oss.so , Command 'module load /usr/lib/asterisk/modules how load a module then?/chan_oss.so' failed. <--how to load a module than? |
| 19:17.13 | justdave | titter: found an example on voip-info even |
| 19:17.15 | Kobaz | BlackBishop: exten => _XXXXXXXXXX,n,Gotoif($["${CALLERID(num):0:1}" = "0"]?read:call) |
| 19:17.24 | BlackBishop | oh damn |
| 19:17.26 | BlackBishop | right ! |
| 19:17.28 | BlackBishop | fsck |
| 19:17.51 | justdave | http://www.voip-info.org/wiki/view/Asterisk+new+jitterbuffer#Asterisk14Jitterbuffers |
| 19:18.02 | kerframil | 19:10 < titter> justdave: You can implement a jitter buffer by using chan_local ... iirc iax2 to meetme doesn't bridge two channels, which doesn't allow for any jb to be applied |
| 19:18.10 | kerframil | oops, sorry |
| 19:19.14 | p3nguin | kobaz: Yes, Cisco 7960s work with Asterisk. You can use SCCP or SIP. I personally prefer SCCP and chan_sccp-b. |
| 19:19.58 | p3nguin | I used to use SIP, then reverted to SCCP after I found out that the Cisco SIP images aren't that wonderful. |
| 19:20.09 | kerframil | Kobaz: I also use chan_sccp (in my view, Federico Santulli's 'v2' trunk is the most reliable of the various branches). it's a bid fiddly to set up for the uninitiated though. |
| 19:20.32 | BlackBishop | Kobaz: thanks, that was it ! :) |
| 19:20.33 | Kobaz | p3nguin: my whole everything is based on sip, i really would prefer sip... is it buggy? |
| 19:20.40 | p3nguin | v3 is had more features. |
| 19:20.54 | p3nguin | is had? v3 has more features. |
| 19:21.00 | kerframil | p3nguin: yes it does. but production stability is my priority. |
| 19:21.11 | p3nguin | I haven't had any problems with v3. |
| 19:21.23 | ubuspire | Qwell, pabelanger ok. I'll take a look at it again. I've checked the logs of my DBMS (postgres) and the queries do get executed, (i've updated http://pastie.org/1575775 ). the entries do exist, but I'm not sure what's wrong/missing in the output. |
| 19:21.31 | Kobaz | so what's the biggest issue with sip on those phones? |
| 19:21.40 | p3nguin | lack of features |
| 19:21.49 | Kobaz | hmm... like what? |
| 19:22.25 | Qwell | ubuspire: theres no type |
| 19:22.38 | ubuspire | Qwell, doh. thx. let me fix that. |
| 19:23.05 | p3nguin | It's been long enough that I don't remember what SIP had and didn't have. I just know when I switched over to SCCP and chan_sccp-b, I was much more satisfied with my phone. |
| 19:23.06 | Qwell | also no context, so nothing will happen when you use it |
| 19:23.20 | Kobaz | heh |
| 19:23.43 | *** join/#asterisk philfine (~philfine@a213-22-45-215.cpe.netcabo.pt) |
| 19:24.04 | Kobaz | i think the only stuff I need is 1) making/receiving phone calls 2) SLA/BLF |
| 19:24.11 | *** join/#asterisk matach (~matach@194-70.76-83.cust.bluewin.ch) |
| 19:24.23 | killown | I am trying set up my voip account http://bpaste.net/show/13757/ and I got sip response 302 moved temporarily back from 200.147.35.32, do I have a configuration issue on my side that I missed? |
| 19:24.34 | p3nguin | I doubt SIP will give you BLF on Cisco. |
| 19:25.11 | _Corey_ | Yeah, negative for BLF on Cisco SIP |
| 19:25.23 | Kobaz | SLA? |
| 19:25.24 | _Corey_ | The new SPA stuff can do it though |
| 19:25.35 | _Corey_ | Negative on SLA |
| 19:25.45 | Kobaz | okay so sccp it is then i suppose |
| 19:25.51 | titter | lol |
| 19:26.29 | kerframil | Kobaz: also, I think g729 support is missing in some instances of the SIP firmware (7940/7960 at least) |
| 19:26.58 | Kobaz | k |
| 19:27.47 | _Corey_ | I haven't found that to be an issue... we've mostly eliminated the 79xx's in favor of Polycom here and with our customers |
| 19:27.51 | p3nguin | It was present when I used SIP on the 7900 series. |
| 19:28.12 | Kobaz | yeah |
| 19:28.12 | _Corey_ | Cisco just doesn't care to develop or support these phones with anything other than Call Manager etc |
| 19:28.28 | Kobaz | but this site has a whole bunch of criscos already |
| 19:28.30 | kerframil | p3nguin: Cisco seem to back it up: http://tinyurl.com/5rdroxo |
| 19:28.34 | *** join/#asterisk Tim_Toady (~moi@178.128.24.211.dsl.dyn.forthnet.gr) |
| 19:28.57 | kerframil | ah, that's annex B - never mind |
| 19:29.26 | p3nguin | yep |
| 19:29.40 | p3nguin | g.729a is right in the codec preference menu. |
| 19:29.46 | *** part/#asterisk phyber (phyber@irssi.co.uk) |
| 19:30.30 | _Corey_ | Can't knock them on reliability though, we've got some here that have been running 6+ years |
| 19:31.23 | ubuspire | Qwell, working now. thanks. think I need more coffee. |
| 19:31.35 | cd101 | Does anyone have any hints for resolving: handle_response_received: "Forbidden" on a SIP channel connecting 2 asterisk boxes on an internal network? |
| 19:32.23 | p3nguin | For one, I'd use IAX2 to interconnect two Asterisk boxes. |
| 19:33.08 | p3nguin | And two, I'll need a lot more informatioin than a simple "Forbidden" message to solve it. |
| 19:34.12 | cd101 | Ok, bare with me for details |
| 19:34.20 | cd101 | I have Asterisk 1.4 in a VM, and 1.6 on another server |
| 19:34.54 | cd101 | 1.4 server has a single physical polycom phone connected using ext 2523 |
| 19:35.15 | p3nguin | The extension used to reach the phone is irrelevant at this point. |
| 19:35.18 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v013-068.mobile.uci.edu) |
| 19:35.55 | cd101 | they are currently connected by using register => localname:password@ip/remotename |
| 19:36.36 | cd101 | I am trying to connect from he 1.4 server to other extensions on the 1.6 server |
| 19:36.52 | cd101 | 1.6 server has 2XXX extensions |
| 19:37.29 | cd101 | basically, at this point I have 1.4 showing it is connecting, and 1.6 saying only: Using SIP RTP TOS bits 184 |
| 19:37.47 | cd101 | at which point 1.4 says the forbidden message |
| 19:38.20 | Qwell | The error says more than "Forbidden". What does it say? |
| 19:39.14 | cd101 | Here is the console output from 1.4: http://pastebin.com/wSzKwrQY |
| 19:40.55 | *** part/#asterisk F0x24 (~Fox@AToulouse-553-1-241-103.w92-162.abo.wanadoo.fr) |
| 19:42.16 | cd101 | sip.conf on 1.6: http://pastebin.com/Gu94zwub |
| 19:43.14 | cd101 | extensions.conf on 1.6: http://pastebin.com/aQ2txXWy |
| 19:43.30 | Qwell | You aren't registering, are you? |
| 19:43.38 | cd101 | yes |
| 19:43.50 | cd101 | should I not be? |
| 19:43.53 | Qwell | I don't see that in the config. |
| 19:44.03 | Qwell | but no, you really shouldn't be using host=dynamic with that setup |
| 19:44.48 | cd101 | 1.6 sip.conf [general]: register => step:welcome@10.76.123.164/orlo |
| 19:45.18 | cd101 | 1.4 sip.conf [general]: register => orlo:welcome@10.76.123.36/step |
| 19:46.39 | *** join/#asterisk luckman212 (~irc@pool-173-77-253-145.nycmny.fios.verizon.net) |
| 19:47.12 | cd101 | it appears as though the dialplan for orlo_incoming never gets run? The first line is to log a notice |
| 19:47.40 | p3nguin | I would rather use Verbose() so I could see it on the CLI while testing. |
| 19:48.06 | p3nguin | Verbose(This message is printed on the Asterisk console) |
| 19:49.09 | *** join/#asterisk luckman212 (~irc@pool-173-77-253-145.nycmny.fios.verizon.net) |
| 19:54.17 | *** join/#asterisk awclin_ (~alinford@g0962184.demon.co.uk) |
| 19:55.03 | cd101 | I changed to Verbose, but there still is no output besides the: Using SIP RTP messages |
| 19:55.15 | Kobaz | Verbose(This message brought to you by the letter *) |
| 19:55.22 | cd101 | =) |
| 19:55.47 | p3nguin | Yeah, my idea of changing to Verbose() wasn't intended to fix the underlying problem. |
| 19:55.49 | Kobaz | what's your verbosity level? do like core set verbose 5 |
| 19:56.14 | cd101 | I understand that, but I was expecting some output to say something more than just the 2 Using SIP messages |
| 19:56.25 | p3nguin | or 3 or 4, since it won't increase any more by going up further. |
| 19:56.37 | cd101 | it was set to 4 |
| 19:56.42 | cd101 | now set to 5 |
| 19:57.04 | p3nguin | There will be no additional output by going from 4 to 5. |
| 19:57.33 | cd101 | I am just not sure how to proceed debugging...I am not even sure which side of the channel is having the problem |
| 19:58.08 | *** join/#asterisk Chris_1234 (~chatzilla@42-236.62-188.cust.bluewin.ch) |
| 19:58.49 | Chris_1234 | hi all, anyone here having asterisk 1.8.2.3 running? |
| 19:59.21 | _Corey_ | Chris_1234: Yeah, that's what I have |
| 19:59.49 | Kobaz | Chris_1234: i was, until i kept crashing it |
| 19:59.59 | kleszcz | ;) |
| 20:00.15 | Chris_1234 | is not crashing so far, but whatever I do, I can't get clients behind firewall working. nat properties changed since 1.6.2 |
| 20:01.00 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 20:01.35 | cd101 | p3nguin: is there any further information I could provide that may help? Or do you know how I can get more output in the 1.6 console? |
| 20:01.40 | p3nguin | That's weird. I set up nat on 1.8.2.2 the exact same way I set up nat on 1.4.38. |
| 20:03.05 | _Corey_ | I haven't changed anything |
| 20:03.38 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v013-068.mobile.uci.edu) |
| 20:03.44 | _Corey_ | We went from 1.4x to 1.6.2 and then 1.8.2.3 on this lab PBX and it's got some NATed phones working |
| 20:04.39 | cd101 | Here is a clip of the SIP debug: http://pastebin.com/zP8D4Pap |
| 20:07.26 | Chris_1234 | My sip.conf at http://pastebin.com/UEcqMZMp |
| 20:09.04 | p3nguin | chris_1234: Make sure you put in the peer definition "canreinvite=no". |
| 20:09.44 | nestAr | anyone familar with the SPA phones? |
| 20:10.32 | _Corey_ | nestAr: What's the question? |
| 20:10.58 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-156-107.dsl.stlsmo.sbcglobal.net) |
| 20:11.05 | nestAr | I've got a SPA525G with a single registration, but it shows the same registration on all 5 soft buttons |
| 20:11.21 | nestAr | even with those lines set disabled |
| 20:11.36 | nestAr | example: <Line_Enable_3_ ua="na">No</Line_Enable_3_> |
| 20:13.31 | *** join/#asterisk lvlolvlo (~lvlolvlo@unaffiliated/lvlolvlo) |
| 20:14.42 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 20:15.38 | _Corey_ | nestAr: What does <Extension_2_ look like? |
| 20:15.56 | _Corey_ | <Extension_3_ (etc) |
| 20:18.20 | nestAr | aahhh |
| 20:18.25 | nestAr | i see it.. |
| 20:18.37 | nestAr | they're all set to 1, i need to change to disabled. |
| 20:18.46 | nestAr | ill check that in a second.. thanks! |
| 20:19.11 | _Corey_ | nestAr: No problem... :) |
| 20:20.51 | *** join/#asterisk fofware (~fofware@wdctf.siup.gov.ar) |
| 20:23.20 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 20:23.24 | killown | what is the difference between context = from-pstn and context = from-trunk? |
| 20:24.07 | nestAr | the names, you just have to see where your lines are configured to |
| 20:24.48 | nestAr | like /etc/asterisk/dahdi-channels.conf needs to match your dialplan |
| 20:25.22 | p3nguin | killown: Contexts names can be arbitrary, but the set contexts do need to match to have proper call flow. |
| 20:25.23 | killown | nestAr there is a way to know what context a sip server does use? |
| 20:25.40 | p3nguin | killown: Check your sip.conf |
| 20:25.48 | nestAr | usually they will give you an example config for sip.conf |
| 20:25.54 | nestAr | and that will dictate the rest of it.. |
| 20:25.55 | _Corey_ | killown: You may want to check #FreePBX |
| 20:26.20 | p3nguin | killown: There is absolutely no reason for you to know what contexts are used on someone else's server. |
| 20:26.27 | killown | p3nguin, nestAr _Corey_ thank you |
| 20:26.49 | killown | I am trying set up my voip account http://bpaste.net/show/13757/ and I got sip response 302 moved temporarily back from 200.147.35.32, do I have a configuration issue on my side that I missed? |
| 20:33.06 | LemensTS | killown: can you pb the sip debug of the 302 error |
| 20:33.27 | LemensTS | http://lists.digium.com/pipermail/asterisk-users/2005-April/095216.html |
| 20:33.31 | *** join/#asterisk bjhaid (~abejide@41.220.69.8) |
| 20:39.56 | skrusty | evening |
| 20:40.51 | *** join/#asterisk UserReg_CL (~COB@pc-82-83-120-200.cm.vtr.net) |
| 20:41.19 | UserReg_CL | hi !!! install debian or centos for asterisk ? |
| 20:41.38 | JerJer | debian |
| 20:41.57 | UserReg_CL | thank JerJer |
| 20:43.07 | skrusty | debian |
| 20:43.30 | skrusty | anyone know why i would get SIP/2.0 401 Unauthorized, but nothing shown from chan_sip at the console? |
| 20:43.58 | skrusty | got about 10 phones that are just getitng 401 (even though they were working fine yesterday) |
| 20:45.28 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 20:45.43 | cd101 | p3nguin: I switched to IAX2 and it works |
| 20:46.02 | cd101 | I still wish I understood where the problem was with the SIP channel though... |
| 20:46.24 | cd101 | appreciate your assistance |
| 20:46.56 | nestAr | _Corey_: worked as one might have expected. Thanks. |
| 20:47.06 | _Corey_ | :) no prob |
| 20:47.14 | *** join/#asterisk FL1SK (~cseader@96.18.203.155) |
| 20:47.21 | FL1SK | Hi |
| 20:47.40 | FL1SK | anyone here answer my question about asterisk and cell phones |
| 20:47.57 | FL1SK | i have a cell phone i need to record phone conversations on |
| 20:48.20 | FL1SK | is there a way to route the converstation through an asterisk server for recording somehow |
| 20:48.31 | FL1SK | any thoughts or suggestions? |
| 20:49.23 | *** join/#asterisk devnull111 (~Miranda@p5B213B80.dip.t-dialin.net) |
| 20:49.39 | *** join/#asterisk pabelanger (~pabelange@50.22.5.41-static.reverse.softlayer.com) |
| 20:49.39 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 20:49.51 | Sertys | FL1SK: pfff |
| 20:50.01 | FL1SK | what Sertys |
| 20:50.07 | Sertys | it's a trick setup |
| 20:50.13 | Sertys | do u wanna do that stealth? |
| 20:50.16 | cd101 | Does anyone know if IAX2 has any limitations on the usable codecs? Can I implement a custom codec over IAX2 as I would have from a SIP channel? |
| 20:50.21 | FL1SK | yes stealth Sertys |
| 20:50.52 | FL1SK | Sertys: what do you mean its a trick setup? |
| 20:50.54 | Sertys | well, if u get the phone to dial the asterisk, u can do pretty much everything with the call |
| 20:51.13 | FL1SK | ok |
| 20:51.46 | FL1SK | Sertys: If they dial my cell phone # though how can it be forwarded or ring in both places so that it records |
| 20:52.20 | lvlolvlo | is there a way to force comfort noise (CNG) on Asterisk 1.6+ for a particular extension if the extension doesn't support it? |
| 20:52.53 | Sertys | well |
| 20:53.10 | Sertys | u get a DID and do a call forward on ur cell to that did |
| 20:53.17 | p3nguin | cd101: You can use the same codecs on IAX2 that you would use on SIP. The big advantage that IAX2 will give you is that it has a trunking mode to reduce bandwidth usage when you have high call volume. |
| 20:53.30 | devnull111 | hey i opened a topic in the asterisk forum http://forums.digium.com/viewtopic.php?f=1&t=77155&sid=60fbdec1d66ee6158969f7f53aee22a9 could somebody of you perhaps help me |
| 20:53.36 | FL1SK | whats a DID |
| 20:53.45 | Sertys | direct-inward dial |
| 20:53.46 | cd101 | thank you p3nguin. |
| 20:53.46 | FL1SK | :-) sorry for my ignorance Sertys |
| 20:53.58 | justdave | russellb: is that chan_local jitterbuffer patch yours? (It just says "russel" on the username of the svn account it's in) |
| 20:54.08 | Sertys | FL1SK: a number from a dialing plan |
| 20:54.10 | russellb | yes |
| 20:54.17 | justdave | if so, any chance of getting that updated for 1.4.39+ ? The current patch no longer successfully applies |
| 20:54.25 | russellb | oh, boo |
| 20:54.27 | russellb | justdave: one sec |
| 20:54.40 | Sertys | FL1SK: what's your phone? |
| 20:54.48 | Sertys | the cell that is |
| 20:54.52 | FL1SK | Verizon Moto DROID |
| 20:54.59 | *** join/#asterisk kfife (~Miranda@home.chicagoventure.com) |
| 20:55.17 | FL1SK | i do have a SIP phone too with Broadvoice |
| 20:55.19 | Sertys | u get a number from any voip provider |
| 20:55.24 | FL1SK | ok |
| 20:55.28 | FL1SK | got one |
| 20:55.40 | Sertys | u put call forwarding on ur phone to that number |
| 20:55.57 | Sertys | and install 3cxphone on ur android phone |
| 20:56.08 | FL1SK | 3cxphone |
| 20:56.09 | FL1SK | what is that |
| 20:56.14 | Sertys | a softphone |
| 20:56.16 | FL1SK | ok |
| 20:56.40 | FL1SK | so that it rings from the forwarded # |
| 20:56.46 | Sertys | u route the call through the asterisk, and then to your softphone |
| 20:56.59 | Sertys | and u have a stable voice recording setup |
| 20:57.05 | FL1SK | Cool |
| 20:57.12 | FL1SK | is this hard to setup |
| 20:57.22 | FL1SK | i need to do it all today |
| 20:57.30 | FL1SK | :-) |
| 20:57.32 | p3nguin | Is there a way to configure Polycom phones for auto-answer by Alert-info header without having a provisioning server to serve up the modified sip.cfg? Some way to configure it either in the web interface or the phone's display menu? |
| 20:57.47 | russellb | justdave: done |
| 20:57.59 | justdave | russellb: thanks |
| 20:58.01 | justdave | this is still trying to solve our audio problems with the Paris office, our Paris IT guy apparently talked to you about it at FOSDEM. :) |
| 20:58.08 | devnull111 | hey does somebody of you know what the error dahdi_pri_error: 1 !! Not yet handling pre-handle message type SEGMENT (0x60) means ? |
| 20:58.09 | justdave | (Mozilla) |
| 20:58.20 | *** part/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 20:58.51 | russellb | justdave: gotcha ... well hopefully it helps. |
| 20:59.04 | russellb | justdave: mozilla foundation office in paris? |
| 20:59.08 | justdave | russellb: yep |
| 20:59.17 | FL1SK | Sertys: is there a white paper on the setup of this? Is this hard to do? |
| 20:59.34 | justdave | getting a new ISP, too, apparently our existing network connection out of there just sucks, too |
| 20:59.35 | russellb | justdave: cool. well feel free to drop me an email if things are still messed up (or even if it works :-) ), and i'll throw out some ideas |
| 20:59.50 | Sertys | FL1SK: well, it might take from 10 minutes to 10 days |
| 20:59.56 | russellb | yeah, network quality is pretty important |
| 21:00.05 | FL1SK | 10 days |
| 21:00.07 | FL1SK | ouch |
| 21:00.18 | justdave | it's a business class DSL line, working on getting a 10gigE fiber put in |
| 21:00.19 | FL1SK | why do you say that Sertys, I'm good with linux already |
| 21:00.38 | Sertys | then go out setup your extensions |
| 21:01.07 | Sertys | it would take me an hour to setup and test |
| 21:01.11 | russellb | justdave: excellent. well like I said, drop me a line if I can help ... russell at russellbryant dot net |
| 21:01.15 | FL1SK | I'll give it a go |
| 21:01.20 | FL1SK | thanks Sertys |
| 21:01.57 | Sertys | just install a call recorder to your moto, FL1SK |
| 21:02.14 | FL1SK | Sertys: they don't work on the DROID |
| 21:02.22 | FL1SK | pisses me off |
| 21:02.39 | devnull111 | hey guys its my first time at irc can you read my messages ? |
| 21:02.40 | FL1SK | Motorola did something to it |
| 21:02.43 | justdave | russellb: does anything in asterisk import code from chan_local.c other than chan_local.so? |
| 21:02.53 | russellb | justdave: no |
| 21:03.10 | justdave | I'm assuming if not, it'd be safe to drop that recompiled module in and just module unload/load it to pick up the changes |
| 21:03.14 | justdave | instead of having to restart the whole thing |
| 21:03.19 | russellb | justdave: yes, in theory :-) |
| 21:03.20 | Sertys | FL1SK: |
| 21:03.20 | justdave | (lots of calls in progress) |
| 21:03.21 | Sertys | hahha |
| 21:03.24 | Sertys | u're right |
| 21:03.27 | justdave | none using Local at the moment |
| 21:03.32 | russellb | right, should be fine then |
| 21:03.48 | Sertys | i guess it's illegal to record call in the us |
| 21:03.53 | FL1SK | Sertys: I'd need to get an HTC |
| 21:04.01 | Sertys | and they ripped the part for incoming audio |
| 21:04.08 | FL1SK | Sertys: i'm only doing this because my lawyer told me too, and its not illegal in some states |
| 21:04.18 | Sertys | though 2.1+ works |
| 21:04.18 | marl_scot | devnull111, yup can see your messages |
| 21:04.29 | FL1SK | wait... 2.1+ works what |
| 21:04.53 | FL1SK | Sertys 2.1+ works |
| 21:04.58 | FL1SK | really |
| 21:05.23 | *** join/#asterisk zooz (~zooz@host86-164-219-51.range86-164.btcentralplus.com) |
| 21:06.28 | marl_scot | FL1SK, not sure about the USA, but in the UK, if a call is 'personal' and the recording is for 'personal use' then you dont normally have to notify the other party. (at leaste that was the way the data protection people explained it) |
| 21:06.59 | FL1SK | marl_scot: its that way in most states in the use, exept like 8 of them |
| 21:07.22 | marl_scot | lol, must be a nightmare :( |
| 21:07.46 | FL1SK | Sertys: where did you find that 2.1+ works? |
| 21:08.48 | Sertys | off the top of my head :) |
| 21:08.58 | Sertys | root the damn android phone :) |
| 21:08.58 | FL1SK | lol |
| 21:09.11 | FL1SK | rooting this phone is a pain in the Ass |
| 21:09.23 | Sertys | actually i was planning on getting one |
| 21:09.33 | justdave | of course by the time it finishes compiling we've got 8 channels open using Local/ :( |
| 21:10.33 | Sertys | http://www.youtube.com/watch?v=k4Hfs99vmHU |
| 21:10.51 | *** join/#asterisk devnull111 (~Miranda@p5B213B80.dip.t-dialin.net) |
| 21:11.20 | Sertys | http://theunlockr.com/2010/06/04/how-to-root-the-motorola-droid-android-2-1/ |
| 21:11.34 | justdave | meh. gonna have people pissed at me now. I replaced chan_local.so on disk and asterisk sigbussed |
| 21:11.34 | Sertys | (i'm not pasting anymore nasty android-related links) |
| 21:11.46 | FL1SK | thanks Sertys |
| 21:11.49 | thehar | whoopsies |
| 21:11.50 | justdave | oops :) |
| 21:11.59 | Sertys | wah |
| 21:14.51 | justdave | I suppose I should have renamed the old one first instead of overwriting it |
| 21:18.21 | justdave | ok, the Local /nj passthrough didn't help even with the patch :( |
| 21:18.42 | justdave | people from paris are all still choppy in meetme :( |
| 21:18.46 | tash | Hello, if a call goes out to a user and I use the WaitExten application ... and they don't press anything, my next step in the exten is to use System to copy the call file to a directory, then give it a timestamp of 30 minutes from now (touch -d "30 minutes"). Then I have a script that runs via cron to move that file to the outgoing directory, but the call never gets placed again. Can someone help me understand why? |
| 21:20.09 | tash | I don't even see anything in the asterisk console when the 30 minutes is up. I tried to set the permissions on the .call to 777 just in case it was permissions |
| 21:20.20 | *** join/#asterisk drmessano-lt (~nonya@pdpc/supporter/active/drmessano) |
| 21:24.35 | justdave | guess we wait on the new network connection |
| 21:34.46 | *** join/#asterisk clintc (~clintc@n128-227-2-246.xlate.ufl.edu) |
| 21:34.53 | *** part/#asterisk clintc (~clintc@n128-227-2-246.xlate.ufl.edu) |
| 21:37.15 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 21:37.15 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 21:38.09 | *** join/#asterisk awclin_ (~alinford@g0962184.demon.co.uk) |
| 21:44.29 | *** join/#asterisk philfine (~philfine@a213-22-45-215.cpe.netcabo.pt) |
| 21:50.23 | WIMPy | Can anyone tell me how to pass a parameter containing commas to Macro()? |
| 21:58.13 | *** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey) |
| 21:59.49 | Kobaz | is there any way to conference more than two people on a polycom? |
| 21:59.59 | Kobaz | even with a 650 (6 line keys) i can only have a three way |
| 22:00.33 | kfife | Kobaz: You need to pay $10 and buy the polycom productivity suite. |
| 22:00.40 | kfife | This will allow you to have a 4 way |
| 22:00.45 | Kobaz | haha |
| 22:00.48 | Kobaz | 4 way? |
| 22:00.52 | kfife | indeed. |
| 22:01.00 | kfife | you, and THREE other connecte parties |
| 22:01.01 | Kobaz | snom can conference like 8 calls for nothing extra |
| 22:01.03 | nestAr | A fourgy, if you will. |
| 22:01.04 | kfife | connected parties |
| 22:01.19 | Kobaz | 10 bucks per phone? |
| 22:01.20 | kfife | Go Snom! |
| 22:01.25 | WIMPy | Really? Never tried that. |
| 22:01.38 | kfife | Even if SNOM will allow it, my wilfe won't allow it. |
| 22:01.47 | kfife | She's strictly 2 way |
| 22:01.51 | kfife | Son of a... |
| 22:01.52 | nestAr | if the snom desk phones are anything like the snom m3 interface, i'll pass on that deal. |
| 22:01.56 | nestAr | kfife: LOL |
| 22:02.17 | Qwell | kfife: nobody said you had to conference with your wife |
| 22:02.19 | Qwell | just sayin |
| 22:02.22 | Kobaz | i used snom like forever ago |
| 22:02.25 | kfife | Qwell: LOL |
| 22:02.29 | Kobaz | the configuration process was horrible |
| 22:02.38 | kfife | Kobaz: that's why i like polycom. |
| 22:02.52 | Kobaz | is that 10 bucks for a license like, forever? |
| 22:02.54 | kfife | The provisioning is so flexible even if the learning curve is rather steep. |
| 22:02.58 | Kobaz | or is that per phone, or per site, or what |
| 22:03.19 | kfife | http://kfife.com/voip/ (old, but you get the idea) |
| 22:03.25 | WIMPy | I can't even do 3-way with a third connection active. |
| 22:03.38 | kfife | Kobaz: it's per endpoint |
| 22:03.58 | *** join/#asterisk drdru (~Adium@76.77.182.145) |
| 22:04.01 | Kobaz | ghetto |
| 22:04.11 | kfife | I have a vague recollection that it's included for free now. |
| 22:04.21 | Kobaz | efk is free now |
| 22:04.28 | Kobaz | productivity suite i don't think |
| 22:04.48 | drdru | I'm a asterisk newbie, looking to find a good sample SIP configuration - any tutorials would be greatly appreciated |
| 22:05.03 | drdru | so far, I can connect my sip phone, but calls are not working |
| 22:05.20 | nestAr | drdru: do you have a voip peer to pass calls through? |
| 22:05.27 | Kobaz | http://www.snom.com/en/products/ip-phones/snom-870/ 5-way conferencing (bridge on phone) |
| 22:05.34 | nestAr | do local calls (like to voicemail) work? |
| 22:05.48 | kfife | Kobaz: You may be right. I don't mind paying them the $10, but I fu¢ked around for a long time trying to figure out why the the ip7000 series phones wouldn't do a 4-way. |
| 22:05.54 | drdru | well, I don't want to do that part yet - right now I just want to connect multiple sip clients to my server, and make calls between them |
| 22:06.14 | drdru | you're asking if I have access to a SIP provider, right? |
| 22:06.44 | kfife | Kobaz: IOW $1000 worth of time only to realize that I wasn't doing anything wrong. |
| 22:06.51 | LemensTS | drdru: voip-info.org <-- |
| 22:07.18 | kfife | LATER that day, the ip6xx's provisionied smoove like buttah |
| 22:07.33 | zooz | I got a problem with my Asterisk configuration, I cannot get my external SIP provider to bet redirected to the right context, it always goes to [default]. Any ideas? |
| 22:07.34 | LemensTS | http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf |
| 22:07.58 | drdru | LemensTS: I've been pouring over that URL for the past 24 hours |
| 22:08.26 | Kobaz | kfife: yeah |
| 22:08.38 | LemensTS | Just know that it is not all up to date for 1.6/.18 |
| 22:08.38 | drdru | ok I remember what my question is - I'm getting invalid context |
| 22:09.32 | drdru | [Feb 17 21:14:30] NOTICE[2130]: chan_sip.c:20314 handle_request_invite: Call from 'jane' to extension 'john' rejected because extension not found in context 'from-sip'. |
| 22:09.51 | drdru | I looked at extensions.conf and couldn't find anything about setting the context |
| 22:10.00 | zooz | this is what my sip/extensions conf look like: http://p.defau.lt/?DkOeIyaZjkC9fsXLLhkClw |
| 22:10.14 | zooz | have been trying to solve this issue for the last day and a half |
| 22:10.34 | zooz | please spare me some help and professional knowledge |
| 22:10.37 | Kobaz | drdru: that's a pretty easy one... make a john extension in from-sip |
| 22:11.31 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 22:11.31 | kfife | Qwell: still laughing |
| 22:11.31 | LemensTS | drdru: in sip.conf, context "jane", it is pointing to from-sip. apparently it is not finding "john" in extensions.conf context from-sip |
| 22:11.33 | drdru | I don't have any context defined |
| 22:12.08 | *** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 22:12.43 | kfife | Qwell: Actually define 'Nobody'. IIRC she MAY have said somethign to that effect. |
| 22:13.13 | LemensTS | drdru: you better define that context in extensions.conf than |
| 22:13.21 | LemensTS | drdru: and add john to it |
| 22:13.29 | drdru | that's the problem I'm having |
| 22:13.36 | drdru | not sure how to create a context |
| 22:13.42 | LemensTS | [from-sip] |
| 22:13.45 | drdru | I'm reading extensions.conf |
| 22:13.48 | drdru | and it's just over my head |
| 22:13.58 | LemensTS | drdru: 1 sec |
| 22:14.26 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 22:15.10 | drdru | another question - does asterisk support SIP MESSAGE capability? |
| 22:15.13 | LemensTS | drdru: http://pastebin.com/bxWE7FBx |
| 22:15.20 | drdru | I want to be able to send a message between my sip clients |
| 22:15.49 | LemensTS | drdru: you mean sms? |
| 22:16.25 | drdru | yeah |
| 22:16.32 | LemensTS | drdru: http://www.voip-info.org/wiki/view/Asterisk+cmd+SendText |
| 22:17.13 | LemensTS | drdru: ive never needed to do it before, but I think that shoud point you in the right direction |
| 22:17.32 | drdru | I think I want this â http://www.voip-info.org/wiki/view/Asterisk+SIP+Messaging |
| 22:18.45 | LemensTS | drdru: i dont even know what that page is for, looks like it is just to tell about a patch...idk |
| 22:18.53 | LemensTS | not what you are wanting tho |
| 22:19.12 | drdru | I want to use the SIP MESSAGE protocol |
| 22:20.32 | *** join/#asterisk ketema (~ketema@kjhmacpro.ketema.net) |
| 22:24.02 | ketema | hello. I was experimenting with ChanIsAvail today and I keep getting back -16 for the AVAILSTATUS variable as well as I get ERROR[29381] utils.c broken pipe in the messages log file when I use the function. I am using it on a SIP device. Can anyone tell me what -16 means ? |
| 22:28.59 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-156-107.dsl.stlsmo.sbcglobal.net) |
| 22:32.09 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 22:32.26 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 22:33.05 | pcangel | I have a macro called macro-dialout that does include => macro-dialout-custom, and I have to inject a MixMonitor line in to the Macro's s extension ..from macro-dialout-custom... is this possible? |
| 22:33.35 | pcangel | the macro-dialout is in a file that my client's interface is rewriting, and I don't have access to their GUI tools |
| 22:33.49 | Qwell | ie; freepbx |
| 22:33.52 | Qwell | ask them |
| 22:33.52 | pcangel | correct |
| 22:33.55 | pcangel | ugh |
| 22:33.57 | pcangel | I hate freepbx |
| 22:34.11 | _Corey_ | lol |
| 22:34.22 | pcangel | It's still just asterisk files. Is there actually a way for this "macro-dialout-custom" to inject a line in to the macro? |
| 22:34.23 | ChannelZ | but it's free! |
| 22:34.34 | _Corey_ | you're in good company :) |
| 22:34.37 | pcangel | I don't want to test anything without knowledge of what it'll do |
| 22:35.56 | pcangel | I imagine that either the s line in macro-dialout-custom will not be ran at all, that it'll be prepended or appended to the macro-dialout, or that the two conflicting priority level 1s will start two separate chains, and only one will work (and if it's in the order I feel is logical, the code I write will run, and their code won't, which would disable all of their outbound calls) |
| 22:39.26 | envisean | does anyone have a good readup on silence suppression / vad & asterisk? |
| 22:43.02 | philfine | Hi everyone, what should be the rule to redirect calls that arrive from pstn line ? |
| 22:43.03 | pcangel | Undocumented results? |
| 22:43.35 | envisean | philfine: you'll want to check out your extensions.conf and add some inbound rules |
| 22:43.38 | pcangel | philfine: you'd need to be more specific- ie where do you want to redirect them to, how are they coming in |
| 22:43.40 | envisean | philfine: depends on what you want to do |
| 22:44.45 | philfine | I have a single analog line connecting to TDM400 FXS port. I want to me able any incoming call to match a rule. Can seem to be able to intersect any call in asterisk |
| 22:45.22 | frigidzephyr | philfine: exten => s,1,SomeApplication |
| 22:45.33 | frigidzephyr | that has to be in the context the call arrives in |
| 22:45.53 | frigidzephyr | and SomeApplication should be whatever you want to do, Answer, Dial, etc |
| 22:48.32 | zooz | I cannot get SIP channel to be directed to the specific context, it always goes to [default] one. here is my configuration: http://p.defau.lt/?DkOeIyaZjkC9fsXLLhkClw What am I doing wrong? |
| 22:48.33 | philfine | Is there any variable set to the callerid ? |
| 22:48.49 | citywok | CALLERID(NUM) (ALL) |
| 22:49.25 | *** join/#asterisk maxagaz (~maxagaz@114.241.174.227) |
| 22:49.38 | citywok | zooz ~pb the console output when the call comes in with verbosity on |
| 22:49.56 | zooz | sure |
| 22:49.57 | nestAr | philfine: exten => 2,2,Set(CALLERID(name)="WHSL: ${CALLERID(name)}") |
| 22:51.23 | *** join/#asterisk x1nux (~x1nux@unaffiliated/x1nux) |
| 22:51.26 | x1nux | hi |
| 22:51.55 | x1nux | i want to know, how close extension session en asterisk CLI> |
| 22:52.09 | x1nux | excuse my english .. ! |
| 22:52.41 | citywok | type exit |
| 22:53.46 | x1nux | example |
| 22:54.02 | x1nux | exit 2000, kick exten 2000 ? |
| 22:54.40 | citywok | oooh, you want to close a channel |
| 22:54.45 | zooz | citywok, here it is: http://p.defau.lt/?xdYpU_gulYSzFJ_Ogc4w_g |
| 22:55.09 | citywok | depending on the channel and your version of *, in 1.8 it's channel request hangup ${CHANNEL} |
| 22:55.26 | *** join/#asterisk ^rage^ (~ragus@82.208.57.182) |
| 22:59.54 | *** join/#asterisk jcongdon (~jcongdon@161.156.121.70.cfl.res.rr.com) |
| 23:00.31 | jcongdon | Hello all |
| 23:00.57 | jcongdon | Does anyone know where I can find a good resource for programming ADSI phones? |
| 23:01.08 | zooz | citywok, and that is what I get if I change host= to dynamic: http://pastebin.com/tu5uXUSv |
| 23:02.59 | citywok | zooz: read this for how type=friend/peer work http://fonality.com/trixbox/forums/trixbox-forums/trunks/sip-trunk-peer-details-type-peer-vs-type-friend |
| 23:05.04 | x1nux | sterisk 1.4.17 = i need to unregister Extension SIP, from CLI> Any idea ? |
| 23:05.10 | x1nux | asterisk 1.4.17 = i need to unregister Extension SIP, from CLI> Any idea ? |
| 23:05.53 | citywok | i think in 1.4 it was "soft hangup sip/whatever" |
| 23:06.06 | x1nux | testing |
| 23:06.51 | x1nux | asterisk*CLI> soft hangup SIP/200 |
| 23:06.51 | x1nux | SIP/2000 is not a known channel |
| 23:07.05 | zooz | citywok, the last pastebin I did was with type=peer option |
| 23:07.21 | tash | anyone had much luck with voicemail/answering machine detectin? |
| 23:07.23 | citywok | zooz use the config your provider suggests |
| 23:07.31 | tash | detection* |
| 23:07.36 | citywok | x1nux: you have to hang up the CHANNEL |
| 23:07.47 | kfife | Scope question: if my dialplan runs the application DIAL(), and the channel returns BUSY, my endpoint hears a busy signal without explicitly running the busy() application. |
| 23:08.27 | kfife | OTOH, if I Gosub to a dialing routine, the channel must look at DIALSTATUS and explictly run BUSY() to hear the busy tone? |
| 23:08.40 | x1nux | i just have extension logedIN i just want to Kick any extension .. |
| 23:08.46 | x1nux | i don have channels |
| 23:08.49 | *** join/#asterisk jcongdon (~jcongdon@161.156.121.70.cfl.res.rr.com) |
| 23:08.52 | x1nux | 0 active SIP channels |
| 23:10.08 | philfine | nestAr: Is that rule for setting caller id or to get callerid from incomming call from pstn ? |
| 23:10.13 | Besticles | is there a way to hangup channels thare at a particular extension? |
| 23:11.13 | kfife | channel request hangup |
| 23:11.40 | p3nguin | x1nux: Extensions do not "log in." Channels are created when activity between a device and asterisk occurs. |
| 23:14.00 | x1nux | so .. who i do .. to kick a extension or a channel .. |
| 23:14.00 | x1nux | ? |
| 23:14.31 | p3nguin | You cannot "kick" an extension. But if there is an active channel, you can ask it to hangup using the soft hangup command on the cli. |
| 23:15.06 | p3nguin | If you want to remove an extension, open up extensions.conf, delete or comment out the lines for that extension, save the file, then run "dialplan reload" from the cli. |
| 23:15.59 | p3nguin | Alternatively, you could use "dialplan remove extension" on the cli to remove an extension. Be careful with this, since you have no idea what you're doing. |
| 23:16.58 | x1nux | so what wappen whan 3 persona login with same extension from any place ? |
| 23:17.08 | x1nux | so what wappen whan 3 persons login with same extension from any place ? |
| 23:17.28 | Qwell | then 1 of them will get the call |
| 23:17.29 | p3nguin | Saying the same thing over and over won't make it a possible situation. |
| 23:17.47 | p3nguin | Like I already told you, extensions do not log in. |
| 23:18.11 | p3nguin | Extensions can allow agents to log in. Extensions can call three different phones. |
| 23:18.15 | citywok | x1nux: i suggest you learn asterisk |
| 23:18.35 | x1nux | jejej Ok ! |
| 23:18.46 | x1nux | thks for your help bye |
| 23:18.48 | *** part/#asterisk x1nux (~x1nux@unaffiliated/x1nux) |
| 23:18.59 | p3nguin | Yeah, hilarious to think someone would be willing to learn something before getting started! |
| 23:23.20 | jcongdon | does anyone know of a good resource for learning ADSI phone programming? |
| 23:24.30 | jcongdon | I was going to suggest agent queues to x1nux, if I am understanding his questions |
| 23:24.34 | jcongdon | too late |
| 23:28.03 | *** join/#asterisk nix8n82-phone (~AndChat@81.sub-174-253-179.myvzw.com) |
| 23:28.06 | jcongdon | I have another strange issue that someone may know. CallerID issue on DAHDI channels. SIP and voicemail always seem to work, but our handsets in the office don't seem to get callerid transferred someones. It's random, not every phone call. |
| 23:30.29 | *** join/#asterisk bjhaid (~abejide@41.220.69.10) |
| 23:34.46 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 23:37.10 | killown | I am trying to set up my voip account and I on debug - got <--- SIP rea<--- SIP read from UDP:200.147.35.32:5060 ---> SIP/2.0 405 Method Not Allowed from UDP:200.147.35.32:5060, do anyone know why i got this error? |
| 23:38.02 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-170.cablep.bezeqint.net) |
| 23:39.03 | lvlolvlo | could be a codec issue or a timer error |
| 23:39.18 | lvlolvlo | do you have the sip debug of the INVITE and the 405 that you received ? |
| 23:39.26 | lvlolvlo | please pastebin that |
| 23:40.44 | killown | lvlolvlo, http://bpaste.net/show/13773/ |
| 23:41.07 | killown | lvlolvlo, could be a ipv6 problem? I don't have enabled it |
| 23:42.04 | *** join/#asterisk matach (~matach@194-70.76-83.cust.bluewin.ch) |
| 23:42.06 | Qwell | once again... call your provider |
| 23:42.14 | lvlolvlo | yes agreed with Qwell |
| 23:42.30 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 23:42.34 | Qwell | That's at least the 6th time today I've told you that. |
| 23:42.41 | lvlolvlo | but before you do that, try changing the User Agent |
| 23:42.54 | lvlolvlo | it could just be that they are redirecting you and sending the 405 because you're using Asterisk |
| 23:43.14 | killown | Qwell, MANNNNNN I DID THAT "my provider does not offer support for third part apps" |
| 23:43.23 | Qwell | Then fire them. |
| 23:43.30 | p3nguin | Their system is not a third-party app. |
| 23:43.49 | p3nguin | If they won't support their services, get a new provider. |
| 23:43.51 | killown | Qwell, I don't know another good brazilian voip provider :( |
| 23:44.09 | Qwell | Clearly you don't know ANY good brazilian voip providers. |
| 23:44.12 | p3nguin | Another? You don't even know ONE, apparently. |
| 23:44.21 | lvlolvlo | try a non brasillian provider |
| 23:44.24 | p3nguin | high fives qwell |
| 23:44.30 | lvlolvlo | sometimes you'd be surprised what other itsps offer |
| 23:46.05 | _Corey_ | I've never seen a 302 redirect on a REGISTER before, that's really whacked |
| 23:47.05 | lvlolvlo | yeah, it's a really poorly implemented logic on the way this ITSP is having OpenSIPS deal with it on the proxy |
| 23:49.03 | _Corey_ | Nothing worse than trying to fix a carrier's f'd up network |
| 23:49.25 | p3nguin | It's never their problem. |
| 23:49.52 | *** join/#asterisk WWGD (~WWGD@c-24-22-54-155.hsd1.wa.comcast.net) |
| 23:49.56 | *** part/#asterisk WWGD (~WWGD@c-24-22-54-155.hsd1.wa.comcast.net) |
| 23:50.30 | _Corey_ | I've determined there's literally 2-3 people at a major national CLEC that actually understand their Broadsoft platform |
| 23:50.59 | p3nguin | I never knew there were so many people who were unable to be accountable. |
| 23:51.38 | _Corey_ | You should have heard the conversation I had with one whacko that told me to solve a customer's caller-id presentation problem by inserting a Diversion tag into the SIP header |
| 23:54.17 | _Corey_ | They all just need to hire that Jeopardy supercomputer |
| 23:56.41 | lvlolvlo | Diversion only? really? not even rpid? |
| 23:56.47 | lvlolvlo | or passerted? |
| 23:57.05 | _Corey_ | Yeah, so the issue was the customer wants to present whatever number they want |
| 23:57.23 | _Corey_ | They were only letting the BTN go out |
| 23:57.33 | _Corey_ | obviously so they could charge the call |
| 23:57.57 | lvlolvlo | damn |
| 23:58.05 | _Corey_ | They had so many screwy solutions... |
| 23:58.21 | _Corey_ | first it was 20 different sip registrations.... one for each possible number |
| 23:58.42 | _Corey_ | then when they realized it needed to be more open, they found the nutjob who proposed the Diversion header |
| 23:59.03 | khlaed | help: i want to use different port range (10000-20000). |
| 23:59.11 | lvlolvlo | rtp.conf |
| 23:59.20 | jcongdon | anyone here know ADSI? |
| 23:59.21 | _Corey_ | finally, the solution is adding a "P-Charge Identity" field |
| 23:59.28 | p3nguin | You want to use 10000 - 20000? |
| 23:59.42 | _Corey_ | as long as they can bill the call, they're fine with whatever's in the From header |
| 23:59.59 | lvlolvlo | interesting |