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| 00:11.07 | przerull | nm everyone I figured it out |
| 00:11.17 | przerull | I just installed asterisk 13 |
| 00:11.24 | przerull | thanks though! |
| 00:14.13 | WIMPy | eduzimrs: Make sure you have all features disabled and directmedia enabled except for your ITSPs. |
| 00:17.58 | eduzimrs | ok |
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| 00:42.40 | GreatSUN | hi all |
| 00:43.10 | GreatSUN | can someone tell me where I can find configurations for jabber/xmpp in asterisk other than jabber.conf? |
| 00:43.28 | GreatSUN | I always have WARN/ERR from res_jabber.c |
| 00:43.40 | GreatSUN | but I don't want jabber/xmpp connection anymore |
| 00:55.59 | WIMPy | noload the module. |
| 01:01.34 | GreatSUN | WIMPy: hmmm... how do I find the place where it is loaded? |
| 01:02.14 | WIMPy | modules.conf |
| 01:02.26 | GreatSUN | in modules.conf it is not loaded |
| 01:02.35 | GreatSUN | at least not the named one |
| 01:02.44 | GreatSUN | I don't find any jabber in there |
| 01:02.57 | WIMPy | So I guess you have aoutoload on. |
| 01:03.53 | GreatSUN | right |
| 01:04.11 | WIMPy | So back to answer 1: |
| 01:04.14 | WIMPy | noload the module. |
| 01:04.24 | GreatSUN | hehe right |
| 01:04.44 | GreatSUN | in my case though noload => res_jabber.so |
| 01:04.45 | WIMPy | Or turn off autoload and load all modules you want manually. |
| 01:04.46 | GreatSUN | right? |
| 01:04.59 | WIMPy | Whatever the name is. |
| 01:08.33 | GreatSUN | that worked nicely after restart |
| 01:08.39 | GreatSUN | thanks a lot |
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| 01:16.34 | tompaw | Hello |
| 01:17.21 | tompaw | Is there a way to call Statis from CLI? |
| 01:18.01 | tompaw | s/Statis/Stasis/ |
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| 04:41.56 | todwetsprock | hi ! id like to make my own.call file, in order to let my mobile-phone ring and send me hello-world. |
| 04:50.15 | todwetsprock | i think i need to put Channel:SIP/trunkname/mobilephonenumber .but it doesnt ring |
| 04:50.41 | todwetsprock | on a local extension, it rings, and has already played hello-world |
| 04:50.46 | todwetsprock | ideas ? |
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| 06:56.33 | refj | Where can I find SRPMs for asterisk 13.x? |
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| 07:01.18 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.5.0 (2015/08/07), 11.19.0 (2015/08/07), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 07:03.07 | refj | Found this: http://packages.asterisk.org/centos/6/asterisk-13/SRPMS/ |
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| 08:46.22 | carrar | moo |
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| 09:09.38 | RadJAckson | Hello , i've noticed lately when receiving up to 150 calls on my asterisk server , the server CPU Gets > 90% and ive noticed some zombie processes, |
| 09:09.41 | RadJAckson | http://hpics.li/48781bc |
| 09:09.59 | RadJAckson | i get a bunch of these when typing : ps -ef | grep "asterisk" |
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| 10:38.06 | babak | Hi , Is it possible to do from cli : originate DAHDI/g0/num1 application park |
| 10:38.36 | babak | and originate DAHDI/g0/num2 application pickup 701 |
| 10:38.47 | babak | to connect num1 to num2 ? |
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| 11:00.31 | pawiecki | Hello! I set timestamp in asterisk.conf to "yes" but it shows only with system messages. How to set it to show in every line? Code: http://pastebin.com/1UFynp0t |
| 11:12.06 | pawiecki | ok, I got it to work - just needed to restart asterisk :) |
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| 11:28.52 | pawiecki | exit |
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| 12:40.20 | [TK]D-Fender | babak: You COULD, but you'd better hope the first call answers first and gets parked otherwise the 2nd call will try to pickup something that doesn't exist. |
| 12:40.50 | [TK]D-Fender | babak: OBut it's far safer and more direct to just call the first then send them to dial the second one... |
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| 12:49.16 | Milos | just finished 1hr of slimming asterisk |
| 12:49.24 | Milos | I'm pretty sure I had this before but I want to get to the bottom of it |
| 12:49.26 | Milos | channel.c: Dropping incompatible voice frame on SIP/milos_desk-00000001 of format g722 since our native format has changed to (ulaw) |
| 12:49.38 | Milos | does this indicate an actual issue, and if so, what's the best way to solve it |
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| 13:41.58 | _omer | Is jitterbuffer enabled in asterisk 11 by default or do I have to enable it manually? |
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| 14:26.05 | babak | [TK]D-Fender: originate DAHDI/g0/num1 application Dial DAHDI/g0/num2 works ok |
| 14:26.25 | babak | originate DAHDI/g0/num1 application park |
| 14:26.25 | babak | originate DAHDI/g0/num2 extension 701@parkedcalls |
| 14:26.25 | babak | also works ok |
| 14:26.43 | [TK]D-Fender | Depends who answers first |
| 14:26.58 | babak | but I cannot find any way to run pickup application |
| 14:27.01 | [TK]D-Fender | if num2 answers first... game over |
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| 14:30.00 | babak | from cli |
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| 14:32.23 | i0x81 | Hey, i am trying to setup an extension which auto answers and plays a greeting, in extensions it starts as: exten => 1003,1,Answer() calling that extension from another softphones works, however the problem is that im getting an error: Subscriber absent when trying to forward calls to that extension |
| 14:33.17 | [TK]D-Fender | "forward" is a very vague term. And nothing about setting up a call has the word "subscriber" attached to it |
| 14:33.35 | [TK]D-Fender | You'll want to be clearer with your wording and more detailed for what you are actually doing |
| 14:38.00 | i0x81 | forward by doing Dial |
| 14:38.14 | i0x81 | let me attach a pastebin |
| 14:39.11 | [TK]D-Fender | Good idea. |
| 14:40.15 | i0x81 | http://pastebin.com/w7G08hsr |
| 14:40.54 | i0x81 | im guessing since im not using an actual softphone for that extension 1003, asterisk thinks that its offline |
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| 14:42.42 | [TK]D-Fender | You don't dial an "extension" in the dialplan like that |
| 14:42.46 | [TK]D-Fender | You are mixing ideas up |
| 14:43.14 | [TK]D-Fender | You would simply GOTO it. You are then just changing where you are in the dialplan. |
| 14:43.41 | [TK]D-Fender | exten => 18888684603,1,Goto(thecontext,1003,1) |
| 14:45.50 | i0x81 | ahhh that makes sense, thanks, that works |
| 14:48.30 | [TK]D-Fender | Quickly it becomes a better idea to put special shared dialplan into its own context and each differnt kind of place that could dial it by some means would Goto that common place. |
| 14:52.07 | i0x81 | for now, the 100x extensions are kept in the local context and the 1888 is kept in the [incoming] |
| 14:52.43 | [TK]D-Fender | Good start. |
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| 14:55.55 | [TK]D-Fender | This would be the more agnostic way of looking at ithttp://pastebin.com/q0E4JmJ9 |
| 15:00.39 | i0x81 | Agreed, thats much better |
| 15:00.41 | i0x81 | thanks |
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| 15:05.19 | i0x81 | [TK]D-Fender: do you know how to properly set timeout on the no input, im trying to have asterisk hangup after x amount of seconds if no dtmf input has been received: http://pastebin.com/x3vtJ2V6 |
| 15:05.44 | i0x81 | what i have does not seem to be working right past the auth step |
| 15:06.33 | [TK]D-Fender | Count your commas right.... |
| 15:06.58 | [TK]D-Fender | exten => 1002,n,Set(AUTH=${SHELL(/usr/bin/perl /root/asterisk_scripts/auth.pl '${AUTHCODE}')}) |
| 15:07.10 | [TK]D-Fender | Also a very bad idea to have * touch anything "root" |
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| 15:11.16 | i0x81 | [TK]D-Fender: commans look right, exten => 1002,n,Read(AUTHCODE,filename,maxdigits,option,attempts,10) |
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| 15:11.31 | i0x81 | according to this: http://www.voip-info.org/wiki/view/Asterisk+cmd+Read |
| 15:13.18 | [TK]D-Fender | never use that source |
| 15:13.28 | [TK]D-Fender | almost certainly dated crap. |
| 15:13.35 | i0x81 | what do you use ? |
| 15:13.41 | [TK]D-Fender | "core show application read" <- Always use CLI for syntax |
| 15:13.58 | [TK]D-Fender | And the official WIKI for any bigger breakdown. |
| 15:14.00 | [TK]D-Fender | ~asteriskwiki |
| 15:14.00 | infobot | well, asteriskwiki is http://wiki.asterisk.org |
| 15:14.44 | i0x81 | hmm, lookng at that, looks correct as well: Read(variable[,filename[&filename2[&...]][,maxdigits[,options[,attempts[,timeout]]]]]) |
| 15:14.54 | i0x81 | or were you referring to a different line ? |
| 15:16.33 | [TK]D-Fender | that does seem to be in the right position |
| 15:16.43 | [TK]D-Fender | check the instructions as to exactly what kind of timeout that is. |
| 15:17.33 | i0x81 | Reads a #-terminated string of digits a certain number of times from the user |
| 15:17.33 | i0x81 | in to the given <variable>. |
| 15:19.14 | i0x81 | basically the first read works as expected |
| 15:19.41 | i0x81 | however after getting to the second read, it only expires after 60 something seconds |
| 15:21.24 | i0x81 | 1 min 12 seconds |
| 15:21.32 | i0x81 | which makes no sense to me |
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| 15:36.07 | Synthase_ | Deploying an HA cluster for a private school, and am at the point to where I'm choosing between implementing PJSIP vs chan_sip. My priority is hardening at the moment, and thus I come to 'allowguest' and 'alwaysauthreject'. Having a cursory look, I don't see such analogs in PJSIP. Can anyone provide some insight? |
| 15:44.20 | i0x81 | [TK]D-Fender: turns out Read worked the way it was supposed to, the problem was that it was passing an empty variable after failing to receive the input, this was the fix: exten => 1002,n,GotoIf($["${CID}" = ""]?kick) |
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| 18:41.41 | jluc | hello |
| 18:42.13 | jluc | i'm looking for a stable SIPphone tool that enables to call, receive calls and manage contacts and messages |
| 18:42.30 | jluc | i'm using linphone but its very week for contacts and does not record messages |
| 18:42.36 | jluc | what would you advise ? |
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| 18:45.42 | salz212 | hi guys,Is it possible to use queues application through ARI leveraging stasis ? I want to control the state management of extensions and distribute them over multiple asterisk boxes. makes sense? |
| 18:46.02 | file | ARI is for building a queue application, it's not for using Queue() |
| 18:48.35 | salz212 | is there any queue app available based on ARI ? I saw playload-voice work on queues but it seem not ready, correct me if I am wrong? |
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| 19:00.07 | file | people have written some, but I do not know any off the top of my head that I can point to |
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| 19:04.27 | i0x81 | jluc: you are looking for a client for linux ? |
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| 20:56.57 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:07.03 | jluc | yes i0x81 |
| 21:15.41 | *** join/#asterisk mjordan (~mjordan@2600:1005:b142:eef1:b455:8e19:8db1:e002) |
| 21:15.41 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 21:36.48 | *** join/#asterisk theron (~theron@199.201.64.131) |
| 21:59.11 | cj | o/ |
| 21:59.27 | *** join/#asterisk Marquel (~Marquel@fuchsfanclub/allerdings/marquel) |
| 22:00.01 | cj | carrar: do we have a seattle ix channel that I could bother with a few questions regarding standing up an IX on Orcas? |
| 22:03.23 | [TK]D-Fender | ... IX? |
| 22:03.28 | [TK]D-Fender | Orcas? |
| 22:06.27 | cj | [TK]D-Fender: sorry. I'm slightly off topic. Looking at bringing up a Washington State VoIP exchange point. |
| 22:08.28 | cj | [TK]D-Fender: I ran in to carrar at a board meeting of the Seattle Internet Exchange (Seattle IX or SIX for short) a few years ago and have been paying his company money every month for voice service. I thought he (or others on the channel) might have some interest in and might benefit financially from helping to get such a thing off the ground. |
| 22:09.50 | *** join/#asterisk lbazan (~LoKoMurdo@fedora/LoKoMurdoK) |
| 22:12.45 | carrar | cj, just use the SIX |
| 22:12.51 | carrar | Thats what it's for |
| 22:12.57 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 22:13.29 | carrar | There is no irc channel for it, only a discussioning mailing list |
| 22:13.38 | carrar | (sp) |
| 22:14.03 | carrar | the SIX is great |
| 22:14.21 | carrar | man thats a huge explosion in China |
| 22:28.38 | *** join/#asterisk bmg505 (~leon@196-215-18-102.dynamic.isadsl.co.za) |
| 22:38.36 | *** join/#asterisk aness (~aness@cm-84.215.80.229.getinternet.no) |
| 22:39.25 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-cloepqcshyiojhvm) |
| 22:41.27 | *** join/#asterisk superscrat (~asanders@173-21-89-217.client.mchsi.com) |
| 22:57.54 | cj | okay, I guess I'll join the list. I just don't have an ASN, so I'm a bit shy. |
| 22:59.43 | carrar | How else are you gonna exchange traffic with other companies? |
| 23:00.10 | cj | yeah, I guess I should get that straightened first, eh? |
| 23:07.38 | *** join/#asterisk iulhk (~iulhk@119.63.142.28) |
| 23:09.41 | iulhk | asterisk callfile feature, which automaticall dial sip or any number, is it possible that when i sent file /var/spool/asterisk/outgoing its just initiate first my script and then from the script after some authentication i will make able to make call? |
| 23:17.19 | *** part/#asterisk kharwell (kharwell@nat/digium/x-zssjmckwmvyyhava) |
| 23:27.14 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 23:40.47 | lvlinux | lynx http://www.belkin.com/us/support-article?articleNum=10771 |
| 23:41.31 | lvlinux | wooops! wrong window! lol |
| 23:59.37 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |