| 00:26.11 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 00:26.11 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.4.0 (2013/05/17), 10.12.2 (2013/03/27), 1.8.22.0 (2013/05/17), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
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| 01:56.30 | raden | Is there a cheap way to make a PA speaker for Asterisk ? |
| 01:56.41 | raden | like hacka pap2t ? |
| 01:59.53 | ChannelZ | probably could |
| 02:00.41 | ChannelZ | I'm surprised someone hasn't made a little embedded SIP device with a giant speaker on it. Or maybe one exists and I've just never heard of it |
| 02:03.35 | WIMPy | Doesn't Snom do one? |
| 02:04.02 | WIMPy | Anyway, chan_alsa works pretty well for that. |
| 02:06.24 | ChannelZ | Huh. Snom PA1 |
| 02:06.47 | ChannelZ | See it on the YouTubes: http://www.youtube.com/watch?v=cupzRqd3X24 |
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| 03:28.53 | fulcan | anyone know a decent sms gateway? |
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| 03:34.43 | ChannelZ | some itsps offer it |
| 03:35.54 | ChannelZ | I know vitelity does |
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| 03:36.48 | igcewieling | Weezey: I mean Noop(HANGUPCAUSE is '${HANGUPCAUSE}') after the Dial. |
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| 03:51.37 | fulcan | ChannelZ vitality support asterisk sip sms? |
| 03:55.45 | ChannelZ | yeah |
| 03:56.01 | ChannelZ | I tried it once, I think I have it on my account in some capacity |
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| 04:00.21 | fulcan | ChannelZ and it's two way? |
| 04:02.13 | ChannelZ | from what I remember. It's XMPP based |
| 04:02.23 | ChannelZ | let me see if I can get it to work again |
| 04:04.29 | ChannelZ | it's "for person-to-person communication only", automation is frowned upon |
| 04:04.46 | fulcan | ChannelZ I am looking for sample sip.conf, I don't seem to see one. and chance you could show me what the provisioning looks like with them? |
| 04:05.22 | ChannelZ | oh great now my gtalk is all fucked up |
| 04:05.35 | ChannelZ | It's done via XMPP, not SIP |
| 04:09.16 | fulcan | ChannelZ I have a Python script that already knows how to looks for a new message in the asterisk /var/spool log and send via db. how difficult is it to morph to xmpp? |
| 04:11.10 | ChannelZ | well I suppose that depends |
| 04:11.24 | fulcan | I didn't know asterisk supported xmpp :) |
| 04:11.58 | ChannelZ | There might be some Python library which could do it directly.. otherwise it's a dialplan app |
| 04:14.19 | fulcan | for me, the hardest part is passing the first message, after that, I can code my why out of anything. Do you have a link to the asterisk xmpp provisioning for vitelity? |
| 04:16.35 | fulcan | It appears from for domestic us sms with just a phone number? (later it refers to a sms charge but it specifically states free above?) |
| 04:16.44 | ChannelZ | yes |
| 04:16.46 | fulcan | free |
| 04:16.58 | fulcan | very very interesting. |
| 04:17.47 | ChannelZ | http://pastebin.com/z8abdwww |
| 04:18.58 | fulcan | ChannelZ thank you very much! that's always the most difficult is figuring out that. |
| 04:19.11 | ChannelZ | then incoming messages will come into the dialplan and can be looked at with ${MESSAGE(from)} and ${MESSAGE(body)} |
| 04:19.37 | ChannelZ | and you can send with JabberSend(vitelity,phonenumber@sms,your message here) |
| 04:20.51 | fulcan | thank you my friend. you just saved me a few hours. |
| 04:21.25 | ChannelZ | they also have an email gateway |
| 04:21.41 | ChannelZ | might be easier |
| 04:22.06 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.210) |
| 04:22.29 | ChannelZ | and there's this: http://xmpppy.sourceforge.net/ |
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| 04:22.59 | fulcan | as long as asterisk does it's job of sending, recieving and stuffing it in /var/spool~ , I'm a happy camper. :) |
| 04:26.46 | fulcan | xmppy doesn't seem to have much support for receiving sms? |
| 04:28.11 | fulcan | looks like asterisk is still the best solution... xmppy is very interesting though. |
| 04:29.28 | *** join/#asterisk peetaur2 (~peter@x2f0d027.dyn.telefonica.de) |
| 04:30.46 | ChannelZ | well presumably it can with all those bot scripts, but I imagine you have to have a persistently running app that sits there listening |
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| 04:32.46 | fulcan | ChannelZ yup yup, always on the go. that's why having a queue already built, makes life a lot easier. Just plug in asterisk and you're done. Did it with anveo once already, but just to test. they are expensive. |
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| 04:34.14 | fulcan | you can do all kinds of neat tricks with an automated 'reply' sms message going into a db. |
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| 04:35.31 | ChannelZ | don't get busted |
| 04:36.27 | fulcan | ChannelZ I looked at their tos, they don't mention scripting and having replies is a lot less shady. :) |
| 04:37.03 | fulcan | just throttle to be safe :) |
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| 06:56.57 | firtina | hi all |
| 06:58.54 | firtina | i created a file agents_test.conf and queues_test.conf, should i include these files inside default queues.conf file? |
| 06:59.41 | firtina | or should i first include agents_test.conf inside queues_test.conf. Then include queues_test inside default queues.conf? |
| 07:01.06 | firtina | slaps ChanServ around a bit with a large trout |
| 07:11.15 | Changos | trout :o~~~ |
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| 07:16.13 | *** mode/#asterisk [+o blitzrage] by ChanServ |
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| 07:25.19 | pietro | hello |
| 07:27.23 | pietro | Can Asterisk sends generate RTCP reports ? |
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| 08:20.04 | *** join/#asterisk hrolf (~hrolf@unaffiliated/hrolf) |
| 08:21.02 | hrolf | Can we do SIP Transfer from inside an AGI (FastAGI to be specific) ? |
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| 08:37.05 | ChannelZ | with EXEC calling the Transfer app I'd assume so |
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| 09:03.19 | catphish | when callerid(pres) is set to hidden, does asterisk still send the callerid on to sip peers? |
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| 09:38.59 | kaldemar | catphish: sure. as far as i know "hidden" is not a valid value for presentation. |
| 09:39.21 | catphish | kaldemar: i mean CALLERID(num-pres)=prohib |
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| 09:40.25 | catphish | the issue i'm having is that after setting that, and sending the call to a second asterisk instance, the second instance doesn't return a value for CALLERID(num) |
| 09:40.38 | catphish | should i be using a different function to read the "hidden" callerid? |
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| 09:43.55 | kaldemar | catphish: it does not send the caller id. |
| 09:44.29 | catphish | interesting |
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| 09:46.33 | catphish | ah, i think it relates to the sendrpid and trustrpid options |
| 09:47.07 | kaldemar | catphish: it is sent in the Remote-Party-ID header if you have sendrpid=yes in sip.conf. |
| 09:47.16 | catphish | that will work for me i think |
| 09:47.33 | catphish | how would i then read it? just manually from the sip headers? |
| 09:47.58 | kaldemar | use func SIP_HEADER for that |
| 09:48.56 | *** join/#asterisk mceier (~mceier@89-69-201-93.dynamic.chello.pl) |
| 09:51.16 | catphish | kaldemar: thanks, Remote-Party-ID is present so that should work fine :) |
| 09:51.41 | kaldemar | you're welcome. |
| 09:52.24 | mceier | hi, is it possible to setup asterisk to use voice modem over serial port and AT command set ? was it implemented by chan_modem which I think is now obsolete ? |
| 09:53.40 | kaldemar | mceier: save yourself from a lot of trouble and use something else. |
| 09:54.31 | mceier | you mean not asterisk, or not voice modem ? |
| 09:56.45 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 09:59.09 | mceier | if I should use something else, is it something else than asterisk or something else than voice modem ? |
| 09:59.26 | kaldemar | mceier: modem |
| 09:59.40 | mceier | ok ;) |
| 09:59.42 | mceier | thanks |
| 10:00.17 | kaldemar | are you looking to connect an asterisk box to a POTS line or just looking to have any PSTN connectivity? |
| 10:01.53 | mceier | I want to receive faxes, and voice calls and forward them to some SIP phones |
| 10:02.08 | mceier | I have POTS line, and voice modem |
| 10:04.27 | mceier | I thought that asterisk would allow me to do that, but it seems it doesn't support voice modems |
| 10:04.44 | mceier | on serial port |
| 10:05.33 | mceier | so instead I should get FXO card, right ? |
| 10:05.40 | mceier | or FXS |
| 10:05.52 | kaldemar | if you insist on using the POTS line, get some hardware with an FXO port. either a card in your box or a gateway. you might have better luck with faxed with the former. |
| 10:06.00 | kaldemar | FXO is for lines, FXS is for phones. |
| 10:06.14 | mceier | ok ;) |
| 10:06.14 | kaldemar | you might want to do some reading on the basics in this: |
| 10:06.17 | kaldemar | ~book |
| 10:06.18 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 10:07.02 | mceier | thanks :) |
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| 10:11.28 | *** join/#asterisk firtina (~kamanato@5.46.7.145) |
| 10:11.39 | firtina | hello |
| 10:11.55 | firtina | could someone help me? |
| 10:14.49 | catphish | apparently not |
| 10:15.37 | *** join/#asterisk italorossi (~italoross@187.61.168.117) |
| 10:15.39 | *** join/#asterisk firtina (~kamanato@5.46.7.145) |
| 10:15.52 | firtina | how can we log an agent into a queue? |
| 10:26.35 | kaldemar | firtina: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ACD_id289508.html#ACD_id253000 |
| 10:26.48 | kaldemar | firtina: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-ACD.html |
| 10:28.39 | firtina | thanks, iam looking now |
| 10:35.07 | *** join/#asterisk hrolf (~hrolf@unaffiliated/hrolf) |
| 10:35.20 | hrolf | ChannelZ: Just checked, nope it doesn't transfers. |
| 10:35.46 | hrolf | ChannelZ: I did Exec Transfer(SIP/786) from FastAGI |
| 10:36.06 | hrolf | ChannelZ: The command appeared in Asterisk console, but as soon as I returned from the FastAGI, the call dropped. |
| 10:38.18 | firtina | kaldemar: when i want to add a channel to queue, i got SIP/6489@10.10.0.2 (dynamic) (Invalid) has taken no calls yet |
| 10:38.21 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.40) |
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| 10:42.29 | Tuju | is there some documentation about those cisco java based phones and their crappy 5060-src-port handling? |
| 10:43.03 | hrolf | When I do Transfer to a SIP peer from FastAGI, then the transfer doesn't happen, can anyone tell why? |
| 10:43.15 | hrolf | As soon as I return from the FastAGI, then the call is dropped. |
| 10:53.24 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 10:53.53 | Ice_Strike | Where can I buy second hand polycom 430? No luck on ebay :/ I need about 10. |
| 10:54.58 | catphish | ebay seems the obvious answer :( otherwise rather depends where you are |
| 10:56.38 | Ice_Strike | Any good phone similar to polycom 430? |
| 10:56.46 | Ice_Strike | same company ofcourse |
| 10:56.50 | Ice_Strike | well affforable phone. |
| 10:57.11 | hrolf | Any ideas? |
| 10:57.15 | catphish | eww that phone is incredibly ugly |
| 10:57.23 | catphish | any particular reason you want them? |
| 10:57.50 | Ice_Strike | Well we always have that model at work. |
| 10:57.57 | Ice_Strike | Which do you recommend? |
| 10:58.37 | catphish | well linksys (cisco) are very popular, we use yealink because they're cheap and work well |
| 10:58.59 | firtina | can we specify a strategy in agents.conf? |
| 10:59.23 | Ice_Strike | catphish what model? |
| 10:59.52 | catphish | cisco SPA or yealink, any model depending how much you want to spend |
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| 11:31.56 | hrolf | I call from a SIP peer, then do Answer(), afterwards, Playback() a file and do Transfer(SIP/786), but the call drops. Why? |
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| 11:35.27 | firtina | how can we add member to agents.conf without CLI commands? |
| 11:39.17 | hrolf | I call from a SIP peer, then do Answer(), afterwards, Playback() a file and do Transfer(SIP/786), but the call drops. Why? |
| 11:40.19 | file | <PROTECTED> |
| 11:40.23 | file | what are you trying to achieve? |
| 11:42.13 | *** join/#asterisk zapphir (~zapphir@46.195.17.247) |
| 11:42.58 | zapphir | Hi, anyone with experience in WebRTC and Asterisk? |
| 11:43.18 | file | maybe. |
| 11:43.42 | firtina | hrolf why don't you Dial(SIP/786) after playback? |
| 11:44.40 | *** join/#asterisk Thesulac (~Thesulac@82.94.204.46) |
| 11:45.40 | zapphir | @file: I am using Asterisk and Chrome with SIPML5 to make audio calls. It works great. But I need a way to detect if the browser is closed, or the tab is closed or the websocket goes down. Sometimes Asterisk catches this and executes a Hangup, but most of the time the call continues for the person making the call to chrome. |
| 11:45.50 | hrolf | firtina: Why is Transfer not working? |
| 11:47.07 | *** join/#asterisk bulkorok (~chatzilla@85.183.61.47) |
| 11:48.09 | zapphir | @file: Sometimes I see "Websocket disconnected" in asterisk console, but it is not consistent. Would SIP presence a way to detect when chrome closes down? |
| 11:48.33 | file | well there's rtptimeout so you know if the RTP stream stops, and session timers but I doubt sipml5 implements it |
| 11:48.49 | file | hrolf, my comment about Transfer was directed to you as was the question |
| 11:48.55 | jhlavacek | Hello, my Asterisk forwards SIP requests with Max-Forwards values 1 or 0, although the RFC says it should be rejected with Too Many Hops. Please, is there any special configuration to do? Event with debug enabled version all I have in the debug logs is Invite parsing, nothing about Max-Forwards (Asterisk 11, chan_sip.c:26593) |
| 11:50.26 | hrolf | file: What I want to do is, that receive a call, run the IVR (FastAGI) and then do Transfer. The reason I'm doing Transfer is that I want my FastAGI connection to be released. |
| 11:51.00 | file | Transfer doesn't do what you think it does/want it to |
| 11:51.07 | hrolf | file: When I do Dial, then the Dial application doesn't return until either the agent or caller disconnects. |
| 11:51.26 | hrolf | file: which blocks my FastAGI connection. |
| 11:51.40 | hrolf | file: So what should I do so that? |
| 11:52.01 | file | I haven't used FastAGI |
| 11:52.51 | hrolf | file: Can you tell why a simple: Answer() -> Playback(..) -> Transfer(..) doens't work? |
| 11:52.59 | *** join/#asterisk TSM (~the_softw@46-65-201-69.zone16.bethere.co.uk) |
| 11:53.13 | file | it does work, it just doesn't do what you think it does and you are using it incorrectly |
| 11:53.29 | file | it tells the remote device (like SIP phone) to contact the provided target directly, not going through Asterisk |
| 11:54.20 | file | since SIP/786 is not a properly formatted SIP URI it doesn't do anything |
| 11:55.05 | hrolf | file: then how should I write "SIP/786"? SIP/786 is on the same asterisk server. |
| 11:55.22 | file | you can't |
| 11:55.43 | file | that would be a REFER back into itself and chan_sip would likely get confused |
| 11:56.14 | file | what you want to do is exit FastAGI and execute additional dialplan logic, as I have not used FastAGI I can't tell you how to do that |
| 11:57.02 | firtina | may it be related to a context issue ? i.e SIP/786@context |
| 11:57.14 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:94e6:367c:6e95:9610) |
| 11:57.48 | file | jhlavacek, Asterisk doesn't forward SIP requests REALLY... each leg is independent so much SIP specific stuff doesn't get carried over... what you can probably do though is write some dialplan logic which enforces that if you so wish by using SIP_HEADER to get the value and rejecting using Hangup() |
| 11:58.01 | file | I think that's what Mark did at a SIPit... |
| 11:59.40 | firtina | i want to add a dynamic queue member? how can i add? And is agent.conf outdated? |
| 12:00.44 | jhlavacek | file: right, asterisk acts like a B2BUA and I have a dialplan for Max-Forward propagation, but I hesitate to open a bug for the problem of processing requests with Max-Forwards=1 or 0, even a B2BUA should reject I think |
| 12:01.24 | jhlavacek | there's even a code checking it in chan_sip |
| 12:02.02 | file | there's some for message handling cause it forwards headers on |
| 12:02.22 | file | you can file an issue if you wish, I just wouldn't expect it to get touched anytime soon |
| 12:04.05 | file | firtina, *agents.conf* configures chan_agent, I don't think an "agent.conf" has ever existed and as for adding... do you mean to a queue in app_queue? |
| 12:05.15 | hrolf | file: Okay, let's say I exit FastAGI, should then I use Transfer or Dial? |
| 12:05.21 | file | Dial |
| 12:06.18 | firtina | actually i did'nt understand how to use agents.conf and how to relate it to queues.conf. I define an agent in agents.conf like this, agent => 3101,3101,operator |
| 12:06.56 | hrolf | file: and if the person I'm Dialing to is on another PBX and my asterisk server has received the call from that PBX, then Dial will still use one channel of mine right? |
| 12:06.58 | zapphir | @file: rtptimeout worked like a charm with webrtc! thanks! |
| 12:06.58 | firtina | but queues.conf side is dark |
| 12:07.13 | file | hrolf, yes the call will go through Asterisk |
| 12:07.42 | jhlavacek | file: thanks, ok, I'll use my dialplan and I'll wait for the pjsip |
| 12:07.45 | file | unless you are in a position to use direct media |
| 12:07.56 | file | then media would go directly while signaling continues through Asterisk |
| 12:09.10 | firtina | file: how can i use password that i use when i create an agent. agent => 3101,passw,operator |
| 12:10.10 | file | chan_agent is a combination application and channel driver which allows agents to log in and wait to be called, the process of logging in requires a password - that is what it is for |
| 12:10.26 | firtina | then what is the relationship beetween queues.conf and agents.conf? |
| 12:10.31 | file | app_queue can have Agents as queue members |
| 12:10.43 | firtina | should i create a queue and then pass this agents to this queue as member? |
| 12:11.02 | file | the difference being that instead of an agent being off the phone and having to ring the phone a caller to the queue is immediately connected to an agent waiting on the phone |
| 12:11.04 | file | you can |
| 12:11.21 | *** join/#asterisk afournier (~admin@mx1.wisp-e.com) |
| 12:11.30 | file | it's up to you to figure out how you want to deploy |
| 12:11.34 | file | and now I go to grab a bus |
| 12:12.24 | firtina | file thanks a alot |
| 12:13.57 | *** join/#asterisk davlefouAMD (~david@197.15.65.76) |
| 12:14.07 | *** join/#asterisk riddlebox (~james@75-132-224-120.dhcp.stls.mo.charter.com) |
| 12:14.45 | *** join/#asterisk pouyou (~samy@LRouen-151-71-123-158.w193-253.abo.wanadoo.fr) |
| 12:14.47 | hrolf | file: If I want not to use Asterisk, then I'll have to Transfer right? |
| 12:15.48 | file | Yes but you can only transfer to an explicit SIP URI |
| 12:15.52 | file | Not a dial string |
| 12:16.26 | file | And what the remote side does depends on them |
| 12:17.49 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:19.46 | hrolf | file: We are doing SIP trunking (Asterisk and a PBX), receive call from that SIP trunk, we run our IVR (FastAGI) and then now want to transfer to an agent (SIP) on that PBX. Would I have to do something like Transfer(SIP/myTrunk/target) ? |
| 12:20.20 | hrolf | file: the person to transfer to is on the PBX and we receive the call from the PBX using SIP trunking |
| 12:20.46 | *** join/#asterisk ghost75 (~trechber@dslb-188-105-016-112.pools.arcor-ip.net) |
| 12:21.04 | hrolf | file: how would then my SIP URI look like? Let's assume, if I want to transfer to SIP peer "100" on the PBX?? |
| 12:22.05 | *** join/#asterisk firtina (~kamanato@5.46.7.145) |
| 12:22.30 | *** join/#asterisk chaz68 (~ChuckMast@wsip-24-234-137-89.lv.lv.cox.net) |
| 12:26.06 | file | You can't think in dial strings or peers. |
| 12:26.36 | file | sip:100@host would transfer to user 100 |
| 12:26.45 | riddlebox | do you guys sell/install switchvox at all? |
| 12:27.35 | file | If the other side is Asterisk it would treat that as a phone blind transferring to extension 100, which if its context has access to would then execute dial plan logic |
| 12:28.06 | file | And the call would no longer be going through the box that executed Transfer, at all |
| 12:28.50 | *** join/#asterisk vlad_starkov (~vlad_star@89.207.91.72) |
| 12:29.00 | riddlebox | I have a side job that switchvox would be perfect for but I do not think they will want to pay for the licenses every year...are the license needed for upgrades,and tech support? Or do you need them to continue using the product? |
| 12:29.49 | file | Salesy questions! Scary. |
| 12:30.31 | riddlebox | yeah..the systems I have put in so far have always just been a few phones for insurance companies(for a friend) |
| 12:31.28 | file | If no one here knows I'd give sales a shout tomorrow |
| 12:32.23 | riddlebox | @file I have been told that the main company I work for is discussing using Asterisk...28k employees and about 13k in one area |
| 12:34.13 | *** join/#asterisk threesome (~threesome@customer-79-127-150-148.net.angelnet.cz) |
| 12:34.36 | file | I'm sure I could make an AT&T joke with that... More Asterisk, more places! |
| 12:36.43 | riddlebox | yeah I am trying to fathom how we could achieve it..we have alot of analog stations..which would be gateways of course...but can you do multiple asterisk servers and administer them from one main server? I know we would need openser unless something better came out |
| 12:37.18 | riddlebox | I have been working on Avaya/Nortel/Cisco exclusively for the last 2 years and have not been keeping up with Asterisk development |
| 12:37.39 | file | That's more of a layer on top of Asterisk |
| 12:38.36 | file | (Management like that) |
| 12:39.00 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.85) |
| 12:43.03 | *** join/#asterisk firtina (~kamanato@5.46.7.145) |
| 12:43.11 | riddlebox | I hope our communications driving force looks into it more, since I am the only one on our Voice Services team that has any Asterisk experience I am sure I would be one of the first to digium training... |
| 12:44.45 | file | Would you be DCAP-itated? |
| 12:45.25 | riddlebox | haha I hope so |
| 12:46.05 | riddlebox | speaking of that my home asterisk server is down..need to get it going again |
| 12:46.42 | *** join/#asterisk ghost75 (~trechber@dslb-188-105-016-112.pools.arcor-ip.net) |
| 12:46.51 | firtina | file: if i paste my configuration text here, could you look? |
| 12:47.00 | riddlebox | maybe I will throw asterisknow on it..is that still going strong? |
| 12:47.06 | file | I've been using trunk whilst working on 12, its been stable for me... Quite happy |
| 12:47.19 | file | AsteriskNOW is still around yup |
| 12:47.39 | file | I am on my cellphone. My screen real estate is not large. |
| 12:48.23 | riddlebox | I told my wife if I could get a job with Digium we would be moving to Alabama..seems like a cool company to work for |
| 12:48.36 | riddlebox | maybe get a ride on Mark's jet... |
| 12:49.37 | pouyou | hi, is there anyone familiar with app_jack ? |
| 12:49.40 | file | Check our page for openings, I know we have a few |
| 12:50.01 | riddlebox | the digium site is a bit slow for me right now not sure whats going on |
| 12:50.20 | riddlebox | file: do you guys get to run linux on your laptop/desktops? |
| 12:51.23 | file | We can run what we want |
| 12:51.41 | file | I run Windows, some run OSX, others Linux |
| 12:51.55 | riddlebox | nice...everything I work on is based on linux and I am forced to run windows 7 with no permissions to do anything |
| 12:53.30 | firtina | http://pastebin.com/hRNFVMRa could you please look at my config files, where am i wrong? |
| 12:55.03 | file | What is the problem? |
| 12:56.21 | firtina | i get app_queue.c:5915 queue_exec: Unable to join queue 'my_queue' |
| 12:57.23 | [TK]D-Fender | firtina: "queue show" <- PASTEBIN |
| 12:57.32 | Greenlight | Do you have "my_queue" configured in queues.conf ? |
| 12:57.49 | file | And is an agent available to take the call? |
| 12:57.50 | firtina | yes |
| 12:58.11 | firtina | agents are available |
| 12:58.18 | *** join/#asterisk Rumbles (~Rumbles@mail.solutiontelecom.co.uk) |
| 12:58.28 | firtina | but when i look queue status NO members No Callers |
| 12:58.44 | firtina | within CLI: queue show |
| 12:58.56 | *** part/#asterisk fischli (~fischli@data.fischer-ing.de) |
| 12:59.29 | [TK]D-Fender | firtina: I see no code to LOG IN your memebers |
| 12:59.48 | [TK]D-Fender | firtina: "joinempty=no" <--- no members = no joining the queue |
| 12:59.56 | [TK]D-Fender | firtina: They have to be there. |
| 13:00.04 | firtina | ok i am adding |
| 13:01.25 | firtina | now i hear music but nothing happens |
| 13:01.48 | firtina | i could not add agent's to queue i think |
| 13:02.27 | pouyou | trying again :P : anyone familiar with app_julius and/or app_jack ? |
| 13:02.55 | firtina | when i write on CLI this => add queue member SIP/3100 to agents it works correctly |
| 13:03.32 | firtina | problem is probably adding this in extensions.conf or queues.conf |
| 13:04.57 | riddlebox | pouyou let me look at it, what is your question? |
| 13:05.04 | [TK]D-Fender | chan_agent is dead.... |
| 13:05.30 | firtina | so what do you suggest? |
| 13:05.51 | file | I would suggest doing some more reading firtina... You don't understand the queueing and agents and how it all fits together |
| 13:05.54 | [TK]D-Fender | using SIP... or local channels, or whatever. |
| 13:06.04 | riddlebox | pouyou ohh app_jack brings jack audio into Asterisk ok..soo whats your question? |
| 13:06.07 | file | And chan_agent still serves a purpose |
| 13:06.21 | pouyou | basically I'm building a processing queue which intends to do the following : calling a number, getting connected, pushing raw sound to julius to get some speech to text, which then gets interpreted and pushing back an answer to the channel |
| 13:06.24 | [TK]D-Fender | If you want them to log in, use AddQueueMember / RemoveQueueMember, etc |
| 13:06.42 | firtina | in extensions.conf right? |
| 13:06.44 | *** join/#asterisk Draecos (~Draecos@124-169-40-193.dyn.iinet.net.au) |
| 13:06.53 | [TK]D-Fender | firtina: Yes |
| 13:06.56 | pouyou | thing is, I do not seem to be able to make jack work or missing something obviously |
| 13:07.16 | firtina | [TK]D-Fender im trying now |
| 13:07.19 | riddlebox | pouyou what do you see in verbose mode? |
| 13:08.03 | riddlebox | pouyou if this ends up calling me and tries to get some kind of loan I will not be happy... ;) |
| 13:08.25 | firtina | file i am already reading asterisk definite guide and searching on internet but there is no accurate information |
| 13:08.29 | pouyou | don't worry, not trying to make you sign up for anything :P |
| 13:08.59 | file | The definitive guide is accurate |
| 13:09.17 | riddlebox | got a robo dial from an asterisk box the other day and the person on the other end said I had signed up to get a loan online.. |
| 13:10.08 | riddlebox | pouyou, do you see any error messages when the stream is trying to play? I have never needed to mess with jack so that part I am not familiar with |
| 13:10.19 | pouyou | just a sec, trying to get things working again |
| 13:10.33 | [TK]D-Fender | [09:08]firtinafile i am already reading asterisk definite guide and searching on internet but there is no accurate information <- that book is fairly accurate as is the Asterisk official WIKI. |
| 13:10.54 | file | What it may not tell you is how to built a call center |
| 13:11.57 | file | There is nothing quite like getting robocalled my something you helped code |
| 13:12.04 | file | Or by |
| 13:13.27 | pouyou | got to debug a few things, it seems i'm not able to register anymore |
| 13:13.28 | *** join/#asterisk sgimeno (~sgimeno@163.117.206.10) |
| 13:13.41 | pouyou | bare with me, definitely sorry, it was working a while ago |
| 13:16.20 | *** join/#asterisk vlad_starkov (~vlad_star@89.207.91.72) |
| 13:16.42 | riddlebox | file: I know right... |
| 13:17.55 | pouyou | so |
| 13:18.03 | pouyou | i end up having this : NOTICE[3165]: app_jack.c:187 log_jack_status: Client Open Status: Failure, Server Failed |
| 13:18.03 | pouyou | <PROTECTED> |
| 13:18.18 | pouyou | so i guessed the jack server needed to be set up first |
| 13:19.15 | pouyou | launching qjackctl |
| 13:19.26 | pouyou | things seem to be ok on jack side |
| 13:20.06 | pouyou | and end up with this now : app_jack.c:187 log_jack_status: Client Open Status: Failure, Server Error |
| 13:20.44 | pouyou | i end up having this in my jack log though |
| 13:20.49 | pouyou | (which I just noticed) |
| 13:20.51 | pouyou | Cannot write socket fd = 11 err = Broken pipe |
| 13:20.52 | pouyou | Cannot read socket fd = 11 err = Broken pipe |
| 13:20.52 | pouyou | Could not read result type = 7 |
| 13:21.18 | pouyou | so there might be my issue :P |
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| 13:32.58 | hrolf | file: I tried this. |
| 13:33.23 | hrolf | file: Set up two Asterisk servers. Called server A to server B using SIP trunk. |
| 13:33.35 | hrolf | file: Playback on server B. |
| 13:33.52 | hrolf | file: Then Transfer(sip:100@serverA IP address) |
| 13:33.59 | hrolf | file: Didn't work. |
| 13:39.13 | riddlebox | man I see alot of cudatel commercials on the net lately |
| 13:40.07 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 13:41.36 | *** join/#asterisk vlad_starkov (~vlad_star@89.207.91.72) |
| 13:46.37 | file | What happened? |
| 13:46.59 | file | Did a REFER get sent? What does the logging on both sides say? |
| 13:48.32 | *** join/#asterisk micols (~t@shell1.rlogin.dk) |
| 13:51.35 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 13:53.16 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 13:55.53 | hrolf | file: Transfer(SIP/any extension on the Server A) worked. |
| 13:56.44 | hrolf | file: Now, although it is working, I'm confused. Can you please explain why this is working? |
| 13:56.51 | hrolf | file: I'm using Asterisk 1.6. |
| 13:57.02 | file | Oh I don't know that old code |
| 13:57.24 | file | Not off the top of my head |
| 13:58.45 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 13:59.00 | file | It might be stripping out the SIP/ and constructing a SIP URI using the IP address within the dialog and the user you specified |
| 13:59.20 | hrolf | file: You mean, this shouldn't work on newer versions of Asterisk? |
| 13:59.48 | file | Unknown. I'd have to try or read the code. |
| 14:00.52 | file | And as I am presently on my cellphone both are unlikely to occur. |
| 14:02.12 | hrolf | file: Okay, thanks. No problem. We'll see this sometime later. |
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| 15:19.18 | g41n | hi all |
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| 15:51.50 | jmls | afternoon all |
| 15:52.17 | jmls | has anyone used / played with the sangoma lyra amd system ? |
| 16:05.27 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
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| 16:48.42 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 17:05.57 | *** join/#asterisk firtina (~kamanato@5.46.11.114) |
| 17:06.02 | firtina | hi |
| 17:08.34 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
| 17:09.59 | firtina | exten => s,1,Answer() |
| 17:10.00 | firtina | <PROTECTED> |
| 17:10.00 | firtina | <PROTECTED> |
| 17:10.00 | firtina | <PROTECTED> |
| 17:10.00 | firtina | <PROTECTED> |
| 17:10.00 | firtina | <PROTECTED> |
| 17:10.00 | firtina | <PROTECTED> |
| 17:10.44 | firtina | could you please tell me why this operates wrong ? |
| 17:10.55 | *** join/#asterisk Zopieux (~Zopieux@jerrycraft.tk) |
| 17:11.02 | Zopieux | hello there |
| 17:11.07 | firtina | hi |
| 17:11.07 | [TK]D-Fender | firtina: PASTEBIN <- |
| 17:11.12 | [TK]D-Fender | firtina: Do not flood in here... |
| 17:11.25 | [TK]D-Fender | show us the code... AND the call. |
| 17:11.27 | firtina | oke, next time i will |
| 17:11.39 | [TK]D-Fender | firtina: For all we know what you showed us here isn't even being called. |
| 17:11.44 | Zopieux | i don't understand why my asterisk logs do not contain lines of when it's receiving calls (from SIP) |
| 17:12.07 | [TK]D-Fender | [13:09]firtinasame => n, GotoIf($[Caller="SIP/3301"]?allow,deny) <- wrong use of Caller variable |
| 17:12.15 | [TK]D-Fender | firtina: You are not referncing its value |
| 17:12.27 | Zopieux | there is a flood of "call was rejected because no rule blabla" but no trace of accepted incoming calls |
| 17:12.58 | [TK]D-Fender | Zopieux: Because that is reporting ERRORS. No news is good news |
| 17:12.58 | Zopieux | is there a way to log these without setting log level to something too disk-space heavy? |
| 17:13.08 | firtina | you mean i should change Caller to -> ${Caller} |
| 17:13.14 | [TK]D-Fender | And no.. if you want full details it is going to get very heavy |
| 17:13.41 | [TK]D-Fender | FiRFor starters,.. next every character in a test like = has to match. this means quotes are LITERAL |
| 17:13.45 | [TK]D-Fender | firtina: ^ |
| 17:13.52 | [TK]D-Fender | firtina: And you have to have them on BOTH sides |
| 17:14.36 | firtina | thanks i am trying, and i will inform you if it is succeeded or not |
| 17:21.45 | firtina | GotoIf( $[ ${Caller} = ${"SIP/3101"} ]?allow,deny ) -> wrong // normally there is no space between them, i put for readability |
| 17:22.08 | [TK]D-Fender | ${"SIP/3101"} <- this is NOT a variable. |
| 17:22.19 | [TK]D-Fender | ~book |
| 17:22.19 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 17:22.21 | [TK]D-Fender | ^^ |
| 17:23.17 | firtina | i can't find a better teacher than you :D when i ask a question, you answer immediately thanks |
| 17:23.26 | [TK]D-Fender | ${SOMEVAR} will return the VALUE it holds. This trying does not have quotes IN it. |
| 17:23.42 | [TK]D-Fender | typically* |
| 17:24.05 | [TK]D-Fender | $["${var}" = "123"] |
| 17:24.31 | [TK]D-Fender | ${var} gets replaced by the valuew it holds. then each side has "" around it |
| 17:24.57 | [TK]D-Fender | because : 123 = "123" does not qork |
| 17:25.02 | [TK]D-Fender | work* |
| 17:25.57 | firtina | hehe it works, thank you very much |
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| 18:31.27 | floren | hi everyone |
| 18:32.02 | floren | i was wondering if anyone on the channel is using the cisco 99x1 with asterisk and voip.ms |
| 18:32.26 | floren | i created a set of centos6 x86_64 rpm's with garreth's patch |
| 18:32.50 | floren | i just have a bit of trouble setting the outbound calls with voip.ms, inbound calls work properly |
| 18:35.33 | [TK]D-Fender | We have no idea what patch you're referring to or how that impacts anything. As for your call failing... you should probably be showing us that call debug from * CLI/. that means SIP DEBUG included. |
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| 18:56.05 | Cuzner | Hey, if I do a "restart when convinient" how do I query for this status? |
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| 19:00.38 | floren | [TK]D-Fender: sorry i was away |
| 19:01.05 | floren | take a look at this please, placing a call from Cisco phone to normal telephone land line: http://codepad.org/odSuwybo |
| 19:01.20 | floren | placing a call from telephone land line to Cisco phone (call dropped): http://codepad.org/0C71bfp2 |
| 19:01.48 | floren | the phone patch is made by a asterisk developer: https://issues.asterisk.org/jira/browse/ASTERISK-13145 |
| 19:01.49 | LieutPants | [ASTERISK-13145] [Status: Open] [patch] Presence subscription on Cisco SIP phone needs special Cisco-styled XML - https://issues.asterisk.org/jira/browse/ASTERISK-13145 |
| 19:02.25 | floren | i have no idea why the call is dropped |
| 19:02.41 | floren | as the packets arrive but they get killed |
| 19:03.22 | [TK]D-Fender | floren: INVITE sip:15142723444@montreal.voip.ms SIP/2.0 Contact: <sip:145602@192.168.1.8:5060> |
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| 19:03.29 | [TK]D-Fender | floren: Your NAT settings are all wrong. |
| 19:03.42 | floren | ya that is my issue |
| 19:03.44 | [TK]D-Fender | well.. not ALL perhaps.. but THIS part, yes |
| 19:03.51 | [TK]D-Fender | You didn't not properly inform * of your WAN IP. |
| 19:03.53 | floren | i have no idea how to solve this [TK]D-Fender |
| 19:04.02 | floren | i only used asterisk for 3 days :D |
| 19:04.17 | floren | so far the easiest way was to build the centos rpm's :) |
| 19:04.40 | floren | i meant the easiest "thing", im on the learning curve |
| 19:05.08 | floren | i read a lot of documentation but still i cannot figure it |
| 19:05.27 | [TK]D-Fender | externhost, localnet, directmedia <--- set them appropriately in [general] |
| 19:05.49 | floren | i see i have none of those set in general :) |
| 19:07.54 | floren | my sip.conf now: http://codepad.org/lc245YtX |
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| 19:08.52 | floren | i'mgoing to read the documentation on externhost, localnet, directmedia |
| 19:09.06 | floren | thanks [TK]D-Fender much apreciated |
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| 19:10.05 | [TK]D-Fender | floren: your phone shoudl not be based on your peer for voip.ms either... |
| 19:10.17 | floren | can i use externip instead of externhost? i have a static ip |
| 19:10.34 | [TK]D-Fender | floren: IIRC externhost can be an IP... |
| 19:10.38 | [TK]D-Fender | and is the current field to use |
| 19:10.53 | floren | [TK]D-Fender: what do you mean by: my phone should not be based on voip.ms |
| 19:11.08 | floren | i apologize for the lack of knowledge :) |
| 19:11.12 | [TK]D-Fender | floren: the template you're using qwith 101 |
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| 19:11.44 | floren | well for now i just want to get it working, do you recommend a diff approach? |
| 19:11.54 | floren | i know the basic demo works |
| 19:12.10 | floren | but ultimatelly i wanted to use asterisk patched so i can use the cisco phone |
| 19:12.41 | floren | it is the only way those cisco phone models become registered, they are very picky |
| 19:13.04 | floren | configuring everything directly into phone simply does not work |
| 19:13.27 | [TK]D-Fender | You do not need to patch asterisk and that one is starting 2008.... |
| 19:14.09 | floren | well the patch is related to certain cieco features that are missing in asterisk, like call parking etc |
| 19:14.34 | floren | i could turn it off easy with use_callmanager = no |
| 19:14.36 | [TK]D-Fender | * has call parking |
| 19:14.42 | [TK]D-Fender | Since at least a decade |
| 19:14.42 | floren | i see |
| 19:14.45 | floren | heh |
| 19:14.52 | [TK]D-Fender | Stop looking for bad news. |
| 19:14.59 | floren | i unserstand |
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| 19:16.04 | floren | so i will follow your suggestion and start with the 3 settings you mentioned |
| 19:17.10 | floren | in my case, exterhost will be my nat address, i.e 70.51.13.39, localnet 192.168.1.0/255.255.255.0 and directmedia... i'm reading on it now |
| 19:18.24 | floren | i see canreinvite is now directmedia, does it still uses the same values? |
| 19:19.43 | [TK]D-Fender | not entirely. |
| 19:20.19 | floren | in my case i guess i need to set the directmedia = outgoing? |
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| 19:38.03 | igcewieling | The Book may be helpful. |
| 19:38.04 | igcewieling | ~book |
| 19:38.05 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 19:39.46 | floren_ | [TK]D-Fender: http://codepad.org/yT2omC3U |
| 19:39.57 | floren_ | it looks better imo, after implementing your settings |
| 19:40.33 | floren_ | i still get dropped calls :( |
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| 20:06.06 | [TK]D-Fender | floren: I see you answering the calll... and deliberately hanging up as your next step... |
| 20:06.17 | [TK]D-Fender | <PROTECTED> |
| 20:06.36 | [TK]D-Fender | and [101] should not be your peer to voip.ms" |
| 20:06.47 | [TK]D-Fender | you should give it a proper name and undo your use of templates. |
| 20:07.20 | [TK]D-Fender | ITSp peers should be the least like anything else.... best to do these things explicitly |
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| 20:27.25 | [TK]D-Fender | checkout time, later all |
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| 20:50.29 | littlepichurris | Does any body here know a spanish channel about asterisk ? |
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| 20:52.14 | s-hell | hello everyone |
| 20:52.58 | s-hell | just a short questions: I'm running asterisk 1.8 behind a firewall |
| 20:53.42 | s-hell | with nat=yes everything works fine but with nat=force_rport,comedia i can't connect to my sipgatetrunk |
| 20:54.53 | s-hell | anyone knows somethin about this? |
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| 21:03.55 | Jinxed- | So, I have some sip devices which auto answer when called however they have no physical buttons so they can't actually hang up a call etc. I was wondering if it would be possible to set up a dial peer which would create a conference or something in which it dials a series of these devices (which will all auto answer) and then hang up the call effectively leaving the users of those devices in a conference. |
| 21:04.11 | Jinxed- | Then the user could dial a different number to kick everyone out of the conference, or destroy it in some way |
| 21:05.26 | WIMPy | A case for originate in its various forms. |
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| 21:12.27 | Jinxed- | WIMPy: do you have any links by chance on originate |
| 21:12.43 | Jinxed- | a quick search of "the definative guide" didn't bring up anything too useful |
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| 21:14.40 | WIMPy | core showapplication originate |
| 21:14.45 | WIMPy | +" " |
| 21:15.16 | s-hell | no one knows anything to my nat problem? |
| 21:16.35 | WIMPy | s-hell: The was something about th new settings not having the same effect. Don't know if that's still the case. Try to look for it of http://issues.asterisk.org/jira/ |
| 21:16.57 | s-hell | WIMPy: Ok, thanks |
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| 21:18.39 | s-hell | WIMPy: Ha, that was easy: https://issues.asterisk.org/jira/browse/ASTERISK-20674 |
| 21:18.40 | LieutPants | [ASTERISK-20674] [Status: Closed] nat=force_rport,comedia does not behave the same as nat=yes - https://issues.asterisk.org/jira/browse/ASTERISK-20674 |
| 21:20.09 | Jinxed- | WIMPy: not super familar but what I can see, I should be able to setup a meetme conference number, use the Answer Application, wait a second then trasnfer the call to the conference |
| 21:20.26 | Jinxed- | meetme conferences appear to have admin capabilities (still looking into) |
| 21:21.33 | WIMPy | You can actually directly originate to the conference. |
| 21:23.04 | [TK]D-Fender | Jinxed-: if they enter an admin pin... then they are an admin. If you tell the app they are an admin... then they are an admin |
| 21:23.11 | WIMPy | If you have at least Asterisk 10 you shoudl look at ConfBridge. |
| 21:23.13 | [TK]D-Fender | And originate's instructions are rather clear... |
| 21:23.56 | Jinxed- | WIMPy: it would be max 4 maybe 5 users |
| 21:24.08 | Jinxed- | [TK]D-Fender: These devices have no buttons, so they can't enter a pin |
| 21:24.32 | [TK]D-Fender | Jinxed-: then you passed a bad parameter |
| 21:26.10 | Jinxed- | [TK]D-Fender: I haven't tried anything yet, I'm just researching |
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| 21:30.46 | Jinxed- | hmm looks like kicking people out will be the easiest part |
| 21:30.46 | Jinxed- | MeetMeAdmin(conference,command[,participant]) |
| 21:31.01 | Jinxed- | MeetMeAdmin(1001,K) |
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| 21:37.36 | Jinxed- | [TK]D-Fender: not finding much about originate beyond https://wiki.asterisk.org/wiki/display/AST/Application_Originate |
| 21:38.07 | [TK]D-Fender | Well taht is the dialplan app... It is BLOCKING BTW |
| 21:38.15 | [TK]D-Fender | So what is the question about it? |
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| 21:42.48 | Jinxed- | nvm too tired to figure this out tonight |
| 21:42.50 | Jinxed- | heading to bed |
| 21:42.56 | Jinxed- | thanks for the info on meet me |
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