IRC log for #asterisk on 20150330

00:00.50*** join/#asterisk fstd (~fstd@unaffiliated/fisted)
00:02.13muanangHi! Are you having echo issue with MG2 canceller using TDM800p?
00:02.43muanangI can hear myself when calling/receiving calls
00:03.07muanangbut normal when calling locally
00:04.35muanangon my test setup, im using tdm400, oslec has the same issue. So i used mg2 and that solved the echo problem.
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00:45.01ChannelZmuanang, fxotune
00:45.22muanangis that fxotune -i 4?
00:45.51muanangthe complete command?
00:46.01muanangi'll try that after office hours.
00:47.00ChannelZpossibly.  I use  fxotune -i -m 13 -l 1
00:47.28ChannelZDepends on how long your telco gives you before it busies out, etc.
00:47.59muanangi see.. i'll check fxotune command for reference..
00:48.04muanangthanks ChannelZ
00:48.08*** join/#asterisk aness (~aness@cm-84.215.80.229.getinternet.no)
00:49.23muanangim just noob using asterisk. your help is much appreciated. thanks!
00:50.05ChannelZDisable the echo cancellers, quit asterisk, restart the dahdi drivers, and run fxotune.  Then you can re-enable and restart everything.  If you've got /etc/init.d/dahdi that starts your DAHDI drivers, make sure it's running fxotune to load the config
00:50.17ChannelZ(it should say as much when starting)
00:51.03muanangnoted. just copied your message to my evernote. (y)
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01:56.15Kobazoh that's wiggity, hmm
01:56.39Kobazif you have a very low session timer duration, it leads to stuck channels in 1.8
01:56.49Kobazi should see if newer 1.8's have that fixed
01:57.05Kobazi could see it being used as a DoS
02:14.26mjordanKobaz: what version of 1.8?
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02:14.33mjordanthere's been about a million bug fixes for 1.8
02:14.40mjordanand a lot dealing with session timers.
02:14.55mjordanparticularly in the 1.8.15 to 1.8.20-ish range
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03:21.55oli-werk[2015-03-30 13:16:41] WARNING[24240][C-00000006] channel.c: Unable to find a codec translation path from (nothing) to (gsm)
03:21.56oli-werk[2015-03-30 13:16:41] WARNING[24240][C-00000006] file.c: Unable to open all-circuits-busy-now (format (nothing)): Function not implemented
03:22.17oli-werkgetting this error when dialling landlines... mobiles seem to work okay
03:22.33oli-werktried enabling and disabling different codecs, changing priorities
03:23.11oli-werkwhen i disabled all but one codec i get
03:23.12oli-werk13:35 <@oli> [2015-03-30 13:10:12] VERBOSE[23405][C-00000002] app_dial.c:     -- Called SIP/Exetel 0280090508/82920520
03:23.13oli-werk13:35 <@oli> [2015-03-30 13:10:12] VERBOSE[23334][C-00000002] chan_sip.c:     -- Got SIP response 482 "(Loop Detected)" back from 58.96.1.2:5060
03:23.43oli-werkthis also only occurs when using elastix dialerd (autodialler)
03:24.00oli-werkany ideas on how to debug / troubleshoot this
03:27.12oli-werk2015-03-30 14:26:39] NOTICE[25097][C-00000012]: chan_sip.c:29752 sip_request_call: Asked to get a channel of unsupported format (nothing) while capability is (gsm|ulaw|alaw|g726|g729)
03:27.12oli-werk[2015-03-30 14:26:39] WARNING[25097][C-00000012]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 58 - Bearer capability not available)
03:37.29oli-werkX-Asterisk-HangupCause: No user responding
03:37.30oli-werkX-Asterisk-HangupCauseCode: 18
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03:37.44[TK]D-Fender~pb
03:37.44infobotA "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude.
03:37.45[TK]D-Fender^^^
03:37.50[TK]D-FenderShow us the actual call.
03:38.19oli-werkthe full sip debug of the call?\
03:41.01[TK]D-FenderYou've Loop warnings and codec errors.  What do you think?
03:43.45oli-werkhttp://pastebin.com/HYzjYhHW
03:45.21[TK]D-Fender<PROTECTED>
03:47.22oli-werkwhoa that's a lot of info
03:47.29oli-werktaking a read through...
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03:50.52oli-werkhttp://pastebin.com/zgGkj1bQ  including verbose 10
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03:52.01[TK]D-Fenderg [s@macro-dialout-trunk:22] Dial("Local/0282920520@from-internal-0000000f;2", "SIP/Exetel 0280090508/0282920520,300,")
03:52.07[TK]D-FenderReliably Transmitting (NAT) to 58.96.1.2:5060:
03:52.13[TK]D-FenderNo real provider is behint NAT
03:52.22[TK]D-FenderContact: <sip:0280090508@10.1.13.38:5060>
03:52.36[TK]D-FenderAnd you are sending them your PRIVATE IP as the contact IP for media
03:52.42[TK]D-FenderYour entire NAT setup is wrong
03:52.43oli-werkhmm
03:53.14[TK]D-Fendera=rtpmap:18 G729/8000 <- you're also offer G.729.  Do you have licenses installed?
03:56.23[TK]D-Fender<PROTECTED>
03:57.53[TK]D-Fender<PROTECTED>
03:58.03[TK]D-FenderYou you have a FAILOVER that loops back into the dialplan.
03:58.09[TK]D-FenderYou have done a VERY bad Originate here...
03:58.44oli-werkthis is the first time i'm touching it :"(
03:58.56[TK]D-FenderVERY FINE MANUALS
03:58.59[TK]D-FenderRead 'em
03:59.40[TK]D-FenderFix your trunk, then re-examine how you are using your Originate
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06:08.36jzu_uhm, if I buy G729 license from Digium for one concurrent call am I able to buy more licenses later or do I need to know the exact amount?
06:08.58jzu_will I be issued new license or can I just install multiple licenses later?
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06:23.01jzu_G729 license is cheapo per concurrent call
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06:35.08jzu_"A G.729 key must be re-registered if any of the Ethernet devices in your  Asterisk server are changed, added, or removed. The unique G.729 license file which is located in your /var/lib/asterisk/licenses directory is tied to the MAC address of all the Ethernet devices installed in your system."
06:36.35jzu_hmm, interesting note
06:36.58jzu_I might swap phones somtimes - do I need to do something in that case? :O
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07:08.42aui_Hi again! Has anybody tried "early video" with a phone providing that preview functionality?
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07:18.59[Consultant]C-Asis there a way to set dnid ?
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08:00.04RadHi , i think i've asked this question before , how can i customize a sip response to my customer? one from our voip providers sends SIP response with a message , example : "SIP/2.0 100 trying -- your call is important to us"
08:00.26Radhow can i do that ? thank you.
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08:02.40wdoekes"your call is important to us", that's a SER router (kamailio, opensips, ..). you can customize everything there
08:03.01wdoekesasterisk doesn't allow you to customize the individual packets, but those sip routers do
08:03.20wdoekesif you want to do that, you'll either have to change a few strings in asterisk and recompile
08:03.31wdoekesor put a sip router in front of your asterisk to proxy the traffic
08:03.50wdoekesor you could not bother trying to change those messages -- my preferred option
08:04.00wdoekes@ Rad
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09:21.47Radwdoekes ok thank you. i will stick to your preferred option
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09:37.58imihaylovHello! Something happened on my asterisk box - the dial command is same => n, Dial(SIP/6001&SIP/6002&SIP/6003&SIP/6004&SIP/6005&SIP/6006&SIP/6010,30,M(call-macro)) and in call-macro asterisk sets in mysql table which operator picked up the phone. If no one pick up the phone in 30 seconds it should go to voice-mail but it began to ring again and again...
09:38.10imihaylovbefore it worked fine
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09:44.13stefan27a channel executes Set(myvar=) and then Read(myvar) and thus sets ${READSTATUS} to OK/ERROR/HANGUP/INTERRUPTED/SKIPPED/TIMEOUT is it the case that myvar can only be non-empty if READSTATUS is OK
09:44.36stefan27I don't really know what these statuses mean
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09:52.30dan_jHi. I'm trying to diagnose why Asterisk seems to take so long to start up. There seems to be something getting it stuck for about half a minute and that delay is causing me problems.
09:52.31dan_jhttp://pastebin.com/7Pg0yFUq
09:53.00dan_jFrom line 349
09:54.20dan_jAny idea why it would be getting stuck there?
09:59.41jzu_well well, I think I'll try some other ITSP besides Sonetel.. seems like I'm unable to get Caller ID to show and I'm having intermitted audio issues
09:59.46jzu_any suggestions?
09:59.59jzu_I need to be able to buy Finland DID
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10:02.52wdoekesstefan27: I'd expect myvar to be partial if TIMEOUT is hit. but you can code for safety and assume that only OK yields a complete myvar and !OK yields anything from (empty, complete, incomplete)
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10:41.51imihaylovCan somebody help with my loop issue mentioned before?
11:08.52wdoekesimihaylov: not if you don't help us first
11:09.01stefan27thanks wdoekes
11:09.07wdoekesyour didn't supply us with enough information to diagnose the problem
11:09.18imihaylovwhat do you need
11:09.27imihaylovi gave the dial command
11:09.33imihaylovand what happend
11:09.43imihaylovwhat should happend
11:10.36imihaylovafter the dial command I have same => n, ExecIf($[${DIALSTATUS} = NOANSWER]?Dial(SIP/ast_billing/${MACRO_EXTEN},30))
11:10.57imihaylovbut asterisk didn't come to this point at all
11:11.24wdoekes1. you've "told" us your dialplan instead of shown it. for all we know, it says Goto(previous-line) (ok, now you did)
11:11.45wdoekes2. you've told us what happened, but we'd rather see how asterisk sees it (core set verbose 10)
11:11.51imihaylov<PROTECTED>
11:12.07wdoekes~pastebin
11:12.08infobotA "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude.
11:13.23imihaylovok
11:13.27imihaylovjust a sec
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11:20.01cervajs2i'm trying configure static realtime. any ideas howto debug res_config_odbc? i need to know which select is failing ( [Mar 30 13:15:44] WARNING[9734]: res_config_odbc.c:922 config_odbc: SQL select error! )
11:20.40imihaylovhttp://pastebin.com/qWi2ygXF
11:20.46imihaylovhere is the output
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11:35.55sarvikHello. I am beginner in Asterisk. I have call forwarding issue.
11:35.55sarvikMore information is http://pastebin.com/Y4WNh0bc
11:35.55sarvikPlease let Me know, if Your need more information
11:50.20*** join/#asterisk cusco (~tralala@2001:41d0:1:6caf::cafe)
11:50.43cuscohey folks... so we have a few queues that membrs are Local channels...
11:50.57cuscoand Local channel dials to the sip peer with the A(audio) parameter
11:51.24cuscoso the callee can hear 'call type something' and be ready to greet
11:51.50cusconow, while the callee is hearing that audio, the call is not marked as answered
11:51.58cuscoany way to reverse the situation?
11:52.20cuscoso the call can be marked as answered as soon as the callee starts hearing that audio?
11:52.47cuscoit is a problem if the audio is longer than the timeout parameter of the queue
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12:01.27jzu_any of you using Multitel.net or Localphone.com as ITSP?
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12:28.31fpriorHi all! I need to troubleshhot this situation: a call come and Asterisk start to playback a file in a loop; for some reason caller's softphone crash (or peer loss network connection); call status stay in Up state and Asterisk repear infinite loops. I tried to use Hangup Handlers or check CHANNEL(state) but I cannot detect the event. Any idea ?
12:30.03WIMPyTwo ways: session-timers and rtptimeout.
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12:30.24WIMPyIf you don't use them, the situation cannot be detected.
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12:59.44fpriorWIMPy thanks, i was great. solved with rtptimeout in peer
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13:04.14anonymouz666hello
13:04.20anonymouz666everybody sleeping?
13:04.43fileI'm sure some people are
13:05.33anonymouz666hmmm don't sleep when the whole world is awake, it's dangerous
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13:15.55sarvikI'm not sleeping :-)
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13:22.48wdoekesimihaylov: calling macro's from macro's. not sure if that's a good idea. in any case, your failed Dial appears to exit the macro instead of continuing to the next prio. I would start by removing the M(call-macro) and see what changes
13:23.37wdoekesif possible, remove the use of macro's and start using Gosub's instead (also in the Dial option)
13:24.01wdoekesand, does your [macro-call-macro] context exist?
13:28.30imihaylovyes it exists
13:28.39imihaylovand its doing its purpose
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13:29.11imihaylovit worked well till some point
13:29.19imihaylovbut I don't know what changed
13:33.49anonymouz666hi wdoekes, are you using the new chan_pjsip?
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13:38.03wdoekesimihaylov: that's why you have version control (etckeeper?); to see what changed
13:38.41wdoekesanonymouz666: not really. I've set it up for someone, but didn't test more than a single call with it
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13:40.31imihaylovi have backups for the asterisk folder
13:41.05imihaylovi am pretty sure the thin about the Dial command that was changed is the number of sips
13:41.08imihaylovnothing else
13:41.36anonymouz666asterisk folder? are you using it on windows? :-)
13:42.27imihaylovno
13:42.39imihaylovdicrectory ..apf
13:42.59imihaylovso nothing changed in the behavior after removing the M(macro)
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13:43.13imihaylovit's looping all over
13:43.15imihaylovagain
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14:01.07wdoekesimihaylov: if you're so pretty sure, then reduce the number of destinations in the Dial string
14:01.32wdoekesyou're the one who's able to debug this by trial and error
14:02.08wdoekesperhaps one of your SIP peers sends a 302 back to your system, creating a loop
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14:03.29imihaylovhmmm
14:03.46imihaylovactually it worked
14:04.41imihaylovreducing the sip receivers
14:05.04imihaylovis there limit of dialing sips
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14:14.45wdoekesimihaylov: before asking more questions, find out exactly which SIP device and/or how many SIP devices changes things for the better/worse
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14:35.06imihaylovDial 6 sips - all logged in. some witgh dnd status - No M(call-macro) voice-mail works. Adding M(call-macro) and it began looping. Adding one sip that is not logged loop begin again. Before there were 7 sips that worked fine. Now one of them is not logged at all.
14:35.18imihaylovhere is the dialplan itself: http://pastebin.com/wgZwpTcY
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14:39.47[TK]D-FenderShow the CALL
14:53.07imihaylov<PROTECTED>
14:53.14imihaylovhere is the call
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15:08.57cuscohi
15:09.54cuscoif peer hangs up a channel while in a Gosub(), asterisk cli shows: "... Abnormal exit ..."
15:10.13[TK]D-Fenderimihaylov: I don't see a "loop".  I see a completely separate 2nd call
15:10.30[TK]D-Fender#4 1st call died completely
15:10.32cuscois there a variable set or so?
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15:11.05cuscoI'm using gosub for the first time, and I'm trying to playback a audio to a callee when he answers a queue()
15:13.23cuscobut if for some reason the callee hangs while playing that audio, I would like to route the caller back into queue
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15:26.50imihaylovyes there are 2 separate calls in asterisk but the caller (in this case me) makes one call and receives continues free dial tone
15:27.08imihaylovtill I cancel the call
15:31.54[TK]D-FenderLooks like didww does this.. not your system
15:32.02[TK]D-Fender"sip set debug on" <- look how the call actually ends
15:38.32imihaylovI will check it tomorrow , now it's pretty busy and there is ton of info from the debug
15:38.59imihaylovthanks for the tips.
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15:52.45jzu_hmmm
15:52.47jzu_<PROTECTED>
15:53.34jzu_http://pastebin.com/GHGvH7Qr
15:53.45jzu_trunk expect to get +358, eh?
15:53.48jzu_*expects
15:54.07jzu_https://www.localphone.com/help/voip/device_guides/softphone/asterisk - I'm tying to dial out through Localphone.com now
16:00.31[TK]D-Fenderhttps://www.localphone.com/help/voip <-----------
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16:13.38jzu_[TK]D-Fender: yes, fixed my dialplan to remove first 0 and add 358 for Finland country code
16:14.07[TK]D-FenderThat is NOT how they say you should be dialing
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16:15.56jzu_"If you are getting this error message it is because you are dialling our local access numbers from your VoIP client. When making a call from any Localphone enabled VoIP device or softphone you should dial the full international number as you would do normally without any + or 0011 prefix, just dial the number as for example 911234567890 for a number in India."
16:16.06jzu_what they say? :O
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16:17.25[Consultant]C-Ashi,  i am using asterisk 13 and was wondering if there is something wwrong with capturing session times on originatiion calls...   seems to work fine with direct calling,  but when i do a callback then a call  sessiontime is always 0
16:17.34[TK]D-Fender"How do I make a cheap international call using the Internet Phone? To make a call, enter the full international number including the country code (starting 00 or +) and hit call. Or, you can select a contact and double click to dial. You’ll find your contacts by clicking on the button with icons of people."
16:18.12jzu_exten => _0Z.,n,Macro(dialout-trunk,6,358${EXTEN:1},,off)
16:18.48jzu_[TK]D-Fender: :D
16:19.52[TK]D-FenderBRB
16:20.32jzu_wonder how I should make the dialplan so I could call real international calls if needed...
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16:24.38jzu_so yes I ditched Sonetel in favor of Localphone and now I'm happy.
16:24.51jzu_no matter what I did Sonetel didn't want to send out CID
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17:51.34Mango45Is there a way to add a non-standard SIP header to a 200 OK that Asterisk sends in response to an INVITE?
18:14.20[Consultant]C-Asi have a problem where after asterisk orriginates a call,   and the callee dials a number,   ANSWEREDTIME   always = 0   (asterisk 13)
18:16.36[Consultant]C-Asis there a setting to change this or fix it?
18:25.51[TK]D-FenderShow us
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18:57.58[Consultant]C-As<PROTECTED>
18:59.01[Consultant]C-Ashere is a complete dump of the bad  ANSWEREDTIME =0  example  http://pastebin.com/C3CEhhH1
19:14.25[Consultant]C-Asmy dialing out is using asterisk not a2billing in my test,   but to get a2billing to work, asterisk needs to give me the correct ANSWEREDTIME,    then ofc a2billing cant bill the call
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19:27.31[Consultant]C-Asit also shows BridgeID= (Not bridged)      I think ANSWEREDTIME  starts after bridge,    but if asterisk connects the 2 people why is the bridge missing on callback calls
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20:30.09nickse!bool
20:30.10nickse!book
20:30.15nickse~book
20:30.15infobotAsterisk: 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
20:31.13nickse$46.99 for a book :O
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20:32.36philfryhas anyone ran into issues specifically with goto meeting phone numbers?
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20:54.50zekoZekohey everyone, i'm taking over a raspbx system that checks caller ID and does something if the CID matches but i'm having trouble routing calls to it from another asterisk (raspbx uses a Sipura box on a PSTN line currently). I put the incoming traffic from the "real" PBX in the same context as PSTN calls get into, but it still passes through the configured DIDs. What should I look into? The freepbx config on
20:54.52zekoZekoraspbx is incomprehensible to me :(
20:57.41[Consultant]C-Asmy issue doesnt happen with asterisk 11 it seems
20:59.02[TK]D-FenderzekoZeko, Your description got a bit circular.  And you should keep FreePBX questions in their channel #freepbx
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20:59.56zekoZeko[TK]D-Fender: i'm actually trying to get rid of freepbx, this application is so simple freepbx just moots it so it's (as i've said) incomprehensible and I get lost in the dialplan :)
21:03.29[TK]D-FenderFix your description first of what it should be doing, and how it isn't
21:03.56zekoZekowill do and pastebin it somewhere.
21:07.02philfrySIP or IAX2? Is there a difference in reliability?
21:07.19[TK]D-Fendernot really
21:07.43[TK]D-FenderThat isn't a point between them
21:10.53philfryI guess from a network perspective IAX2 is easier to get through a firewall. Has anyone used Digium SIP Trunking before? I tried it out today and I had some issues but its hard to tell the issue. I was going to do a tcpdump to see if that helps at all. Just kinda frustrating i guess.
21:11.51[TK]D-FenderStart by getting rid of all the vague bits you just introduced there
21:12.06[TK]D-FenderAnd test with another provider
21:12.15[TK]D-FenderRemove YOURSELF as the weak link in the chain
21:12.53philfrycan you explain that a little bit more?
21:13.25robmalphilfry: [TK]D-Fender gets very angry when someone uses the word 'trunking'. I think he has some elephant problems.
21:13.25[TK]D-Fender"had some issues" doesn't offer anything for commentary
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21:13.53[TK]D-FenderBefore thinking that the experience was bad because of the vendor... go see if it's any better with another.
21:14.06[TK]D-FenderMaybe i'ts YOUR situation that's bad and it won't matter where you go
21:14.15philfryokay sorry. Phone calls were disconnected after a period of time. The first 2 were 15 minutes exactly and the last one was 50 minutes into the call.
21:14.22[TK]D-FenderAKA "do proper debugging and comparative testing"
21:14.41[TK]D-FenderAnd someone else's experine won't math your scenario because of your provider, and your equipment, etc
21:14.42philfryI've used different providers before and never had this issue. I think its very odd that it disconnects right at 15 minutes.
21:15.16[TK]D-Fender15 minutes sounds like an RTP timeout.  I'd be betting on an improper NAT setup on your side for starters
21:15.21[TK]D-FenderAnd start drilling the calls.
21:16.14philfryokay i'll investigate the firewall maybe there is an udp connection that is being blocked
21:16.15philfrythanks
21:18.29[TK]D-Fendercheck the start of your calls to verify that initial settings are right
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21:21.16philfrycan you explain what i should check for at the start of the call? I see it is passed to the correct outbound route and i don't see anything on the firewall being blocked.
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21:27.42[TK]D-FenderThat means digging the entire SIP setup of the call
21:27.50[TK]D-FenderFreePBX-speak really means nothing.
21:28.06[TK]D-FenderIt's all about what is being told to go where.
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21:33.46[Consultant]C-Aslol  @ robmal
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21:34.26robmal[Consultant]C-As: What? ;->
21:39.28zekoZekoi think I got closer to my problem... on calls asterisk -> raspbx DNIDDigits is not set, how do I set that in my dialplan? Set(CALLERID(dnid)=XXX) doesn't seem to do anything.
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21:48.29zekoZekoOK that was stupid... had to Dial(SIP/raspbx/${EXTEN}) instead of just SIP/raspbx. Seems I have a lot to learn still :)
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21:51.36j-fishany recommendations for a wirelss voip phone for a call center?
21:52.08[TK]D-FenderDECT handset + SIP receiver base.
21:52.18[TK]D-Fender~wifivoip
21:52.18infobot[~wifivoip] Wi-Fi (802.11a/b/g) VoIP phones on the market may possess any of the following drawbacks : poor battery life, limited range, lack of call features (many lack transfer, etc), poor or no NAT traversal, shoddy construction, etc. Because of this they are usually not recommended.  Some alternatives are : VoIP ATA + cordless, DECT w/ VoIP base, etc
21:52.45[TK]D-Fenderalso "for a call center" ... really doesn't mean anything at all.
21:55.27j-fishit is just that i heard , some bluethooth phones could have problems if too many are in the same room in close promixity
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21:58.26[TK]D-FenderAt no point did I mention Bluetooth
21:58.44[TK]D-FenderAnd what is a "bluetooth phone" exactly?
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22:03.31j-fishso which DECT+ sip receiver base brand/company you recommend
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22:13.18zekoZekoi have a few Siemens SIP/DECT phones and the users are quite happy... but they don't use them for much more that dialing and receiving calls. No LDAP directory support though...
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22:18.01j-fishzekoZeko: thank you
22:20.21pjensen00if I receive a "Resource not found" via a 'channels' ARI command where I'm trying to do an originate, am I correct in assuming that it means the Stasis app doesn't exist?
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