| 00:00.50 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 00:02.13 | muanang | Hi! Are you having echo issue with MG2 canceller using TDM800p? |
| 00:02.43 | muanang | I can hear myself when calling/receiving calls |
| 00:03.07 | muanang | but normal when calling locally |
| 00:04.35 | muanang | on my test setup, im using tdm400, oslec has the same issue. So i used mg2 and that solved the echo problem. |
| 00:15.01 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 00:15.51 | *** join/#asterisk superscrat (~asanders@173-17-133-2.client.mchsi.com) |
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| 00:45.01 | ChannelZ | muanang, fxotune |
| 00:45.22 | muanang | is that fxotune -i 4? |
| 00:45.51 | muanang | the complete command? |
| 00:46.01 | muanang | i'll try that after office hours. |
| 00:47.00 | ChannelZ | possibly. I use fxotune -i -m 13 -l 1 |
| 00:47.28 | ChannelZ | Depends on how long your telco gives you before it busies out, etc. |
| 00:47.59 | muanang | i see.. i'll check fxotune command for reference.. |
| 00:48.04 | muanang | thanks ChannelZ |
| 00:48.08 | *** join/#asterisk aness (~aness@cm-84.215.80.229.getinternet.no) |
| 00:49.23 | muanang | im just noob using asterisk. your help is much appreciated. thanks! |
| 00:50.05 | ChannelZ | Disable 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 |
| 00:50.17 | ChannelZ | (it should say as much when starting) |
| 00:51.03 | muanang | noted. just copied your message to my evernote. (y) |
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| 01:56.15 | Kobaz | oh that's wiggity, hmm |
| 01:56.39 | Kobaz | if you have a very low session timer duration, it leads to stuck channels in 1.8 |
| 01:56.49 | Kobaz | i should see if newer 1.8's have that fixed |
| 01:57.05 | Kobaz | i could see it being used as a DoS |
| 02:14.26 | mjordan | Kobaz: what version of 1.8? |
| 02:14.28 | *** join/#asterisk cyford (~cyford@c-73-207-183-115.hsd1.ga.comcast.net) |
| 02:14.33 | mjordan | there's been about a million bug fixes for 1.8 |
| 02:14.40 | mjordan | and a lot dealing with session timers. |
| 02:14.55 | mjordan | particularly in the 1.8.15 to 1.8.20-ish range |
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| 03:21.55 | oli-werk | [2015-03-30 13:16:41] WARNING[24240][C-00000006] channel.c: Unable to find a codec translation path from (nothing) to (gsm) |
| 03:21.56 | oli-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 |
| 03:22.17 | oli-werk | getting this error when dialling landlines... mobiles seem to work okay |
| 03:22.33 | oli-werk | tried enabling and disabling different codecs, changing priorities |
| 03:23.11 | oli-werk | when i disabled all but one codec i get |
| 03:23.12 | oli-werk | 13:35 <@oli> [2015-03-30 13:10:12] VERBOSE[23405][C-00000002] app_dial.c: -- Called SIP/Exetel 0280090508/82920520 |
| 03:23.13 | oli-werk | 13:35 <@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 |
| 03:23.43 | oli-werk | this also only occurs when using elastix dialerd (autodialler) |
| 03:24.00 | oli-werk | any ideas on how to debug / troubleshoot this |
| 03:27.12 | oli-werk | 2015-03-30 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) |
| 03:27.12 | oli-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) |
| 03:37.29 | oli-werk | X-Asterisk-HangupCause: No user responding |
| 03:37.30 | oli-werk | X-Asterisk-HangupCauseCode: 18 |
| 03:37.34 | *** join/#asterisk bkruse (~Adium@24.42.207.11) |
| 03:37.44 | [TK]D-Fender | ~pb |
| 03:37.44 | infobot | A "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-Fender | Show us the actual call. |
| 03:38.19 | oli-werk | the full sip debug of the call?\ |
| 03:41.01 | [TK]D-Fender | You've Loop warnings and codec errors. What do you think? |
| 03:43.45 | oli-werk | http://pastebin.com/HYzjYhHW |
| 03:45.21 | [TK]D-Fender | <PROTECTED> |
| 03:47.22 | oli-werk | whoa that's a lot of info |
| 03:47.29 | oli-werk | taking a read through... |
| 03:49.16 | *** join/#asterisk king1337-2 (~king1337@S0106c8fb2641d848.vs.shawcable.net) |
| 03:50.52 | oli-werk | http://pastebin.com/zgGkj1bQ including verbose 10 |
| 03:51.28 | *** join/#asterisk D30 (~D30@58.71.19.178) |
| 03:52.01 | [TK]D-Fender | g [s@macro-dialout-trunk:22] Dial("Local/0282920520@from-internal-0000000f;2", "SIP/Exetel 0280090508/0282920520,300,") |
| 03:52.07 | [TK]D-Fender | Reliably Transmitting (NAT) to 58.96.1.2:5060: |
| 03:52.13 | [TK]D-Fender | No real provider is behint NAT |
| 03:52.22 | [TK]D-Fender | Contact: <sip:0280090508@10.1.13.38:5060> |
| 03:52.36 | [TK]D-Fender | And you are sending them your PRIVATE IP as the contact IP for media |
| 03:52.42 | [TK]D-Fender | Your entire NAT setup is wrong |
| 03:52.43 | oli-werk | hmm |
| 03:53.14 | [TK]D-Fender | a=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-Fender | You you have a FAILOVER that loops back into the dialplan. |
| 03:58.09 | [TK]D-Fender | You have done a VERY bad Originate here... |
| 03:58.44 | oli-werk | this is the first time i'm touching it :"( |
| 03:58.56 | [TK]D-Fender | VERY FINE MANUALS |
| 03:58.59 | [TK]D-Fender | Read 'em |
| 03:59.40 | [TK]D-Fender | Fix your trunk, then re-examine how you are using your Originate |
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| 06:08.36 | jzu_ | 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? |
| 06:08.58 | jzu_ | will I be issued new license or can I just install multiple licenses later? |
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| 06:23.01 | jzu_ | G729 license is cheapo per concurrent call |
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| 06:31.06 | *** join/#asterisk airjump (~Thunderbi@p5B0A3AB0.dip0.t-ipconnect.de) |
| 06:35.08 | jzu_ | "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." |
| 06:36.35 | jzu_ | hmm, interesting note |
| 06:36.58 | jzu_ | I might swap phones somtimes - do I need to do something in that case? :O |
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| 07:08.42 | aui_ | Hi again! Has anybody tried "early video" with a phone providing that preview functionality? |
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| 07:18.59 | [Consultant]C-As | is there a way to set dnid ? |
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| 07:59.19 | *** join/#asterisk Rad (~Rad@cpe-xd001622.cust.jaguar-network.net) |
| 08:00.04 | Rad | Hi , 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" |
| 08:00.26 | Rad | how can i do that ? thank you. |
| 08:02.30 | *** join/#asterisk rogerh (~rogerh@202.37.70.18) |
| 08:02.40 | wdoekes | "your call is important to us", that's a SER router (kamailio, opensips, ..). you can customize everything there |
| 08:03.01 | wdoekes | asterisk doesn't allow you to customize the individual packets, but those sip routers do |
| 08:03.20 | wdoekes | if you want to do that, you'll either have to change a few strings in asterisk and recompile |
| 08:03.31 | wdoekes | or put a sip router in front of your asterisk to proxy the traffic |
| 08:03.50 | wdoekes | or you could not bother trying to change those messages -- my preferred option |
| 08:04.00 | wdoekes | @ Rad |
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| 09:21.47 | Rad | wdoekes ok thank you. i will stick to your preferred option |
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| 09:34.35 | *** join/#asterisk imihaylov (~imihaylov@87.121.90.151) |
| 09:37.58 | imihaylov | Hello! 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... |
| 09:38.10 | imihaylov | before it worked fine |
| 09:42.35 | *** join/#asterisk stefan27 (~stefan27@212.247.4.149) |
| 09:43.20 | *** join/#asterisk ghostlines (~ghostline@82.201.93.50) |
| 09:44.13 | stefan27 | a channel executes Set(myvar=) and then Read(myvar) and thus sets ${READSTATUS} to OK/ERROR/HANGUP/INTERRUPTED/SKIPPED/TIMEOUT is it the case that myvar can only be non-empty if READSTATUS is OK |
| 09:44.36 | stefan27 | I don't really know what these statuses mean |
| 09:44.58 | *** join/#asterisk ghostlines (~ghostline@82.201.93.50) |
| 09:52.30 | dan_j | Hi. I'm trying to diagnose why Asterisk seems to take so long to start up. There seems to be something getting it stuck for about half a minute and that delay is causing me problems. |
| 09:52.31 | dan_j | http://pastebin.com/7Pg0yFUq |
| 09:53.00 | dan_j | From line 349 |
| 09:54.20 | dan_j | Any idea why it would be getting stuck there? |
| 09:59.41 | jzu_ | well well, I think I'll try some other ITSP besides Sonetel.. seems like I'm unable to get Caller ID to show and I'm having intermitted audio issues |
| 09:59.46 | jzu_ | any suggestions? |
| 09:59.59 | jzu_ | I need to be able to buy Finland DID |
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| 10:02.52 | wdoekes | stefan27: I'd expect myvar to be partial if TIMEOUT is hit. but you can code for safety and assume that only OK yields a complete myvar and !OK yields anything from (empty, complete, incomplete) |
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| 10:41.51 | imihaylov | Can somebody help with my loop issue mentioned before? |
| 11:08.52 | wdoekes | imihaylov: not if you don't help us first |
| 11:09.01 | stefan27 | thanks wdoekes |
| 11:09.07 | wdoekes | your didn't supply us with enough information to diagnose the problem |
| 11:09.18 | imihaylov | what do you need |
| 11:09.27 | imihaylov | i gave the dial command |
| 11:09.33 | imihaylov | and what happend |
| 11:09.43 | imihaylov | what should happend |
| 11:10.36 | imihaylov | after the dial command I have same => n, ExecIf($[${DIALSTATUS} = NOANSWER]?Dial(SIP/ast_billing/${MACRO_EXTEN},30)) |
| 11:10.57 | imihaylov | but asterisk didn't come to this point at all |
| 11:11.24 | wdoekes | 1. you've "told" us your dialplan instead of shown it. for all we know, it says Goto(previous-line) (ok, now you did) |
| 11:11.45 | wdoekes | 2. you've told us what happened, but we'd rather see how asterisk sees it (core set verbose 10) |
| 11:11.51 | imihaylov | <PROTECTED> |
| 11:12.07 | wdoekes | ~pastebin |
| 11:12.08 | infobot | A "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. |
| 11:13.23 | imihaylov | ok |
| 11:13.27 | imihaylov | just a sec |
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| 11:18.17 | *** join/#asterisk cervajs2 (~cervenka@178.148.broadband4.iol.cz) |
| 11:20.01 | cervajs2 | i'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! ) |
| 11:20.40 | imihaylov | http://pastebin.com/qWi2ygXF |
| 11:20.46 | imihaylov | here is the output |
| 11:23.33 | *** join/#asterisk MarkSX (~MarkSX@unaffiliated/marksx) |
| 11:35.55 | sarvik | Hello. I am beginner in Asterisk. I have call forwarding issue. |
| 11:35.55 | sarvik | More information is http://pastebin.com/Y4WNh0bc |
| 11:35.55 | sarvik | Please let Me know, if Your need more information |
| 11:50.20 | *** join/#asterisk cusco (~tralala@2001:41d0:1:6caf::cafe) |
| 11:50.43 | cusco | hey folks... so we have a few queues that membrs are Local channels... |
| 11:50.57 | cusco | and Local channel dials to the sip peer with the A(audio) parameter |
| 11:51.24 | cusco | so the callee can hear 'call type something' and be ready to greet |
| 11:51.50 | cusco | now, while the callee is hearing that audio, the call is not marked as answered |
| 11:51.58 | cusco | any way to reverse the situation? |
| 11:52.20 | cusco | so the call can be marked as answered as soon as the callee starts hearing that audio? |
| 11:52.47 | cusco | it is a problem if the audio is longer than the timeout parameter of the queue |
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| 12:01.27 | jzu_ | any of you using Multitel.net or Localphone.com as ITSP? |
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| 12:28.31 | fprior | Hi 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 ? |
| 12:30.03 | WIMPy | Two ways: session-timers and rtptimeout. |
| 12:30.22 | *** join/#asterisk bulkorok (~Benjamin_@85.183.61.47) |
| 12:30.24 | WIMPy | If you don't use them, the situation cannot be detected. |
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| 12:59.44 | fprior | WIMPy thanks, i was great. solved with rtptimeout in peer |
| 13:00.57 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-66-28.user.veloxzone.com.br) |
| 13:04.14 | anonymouz666 | hello |
| 13:04.20 | anonymouz666 | everybody sleeping? |
| 13:04.43 | file | I'm sure some people are |
| 13:05.33 | anonymouz666 | hmmm don't sleep when the whole world is awake, it's dangerous |
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| 13:15.55 | sarvik | I'm not sleeping :-) |
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| 13:22.48 | wdoekes | imihaylov: 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 |
| 13:23.37 | wdoekes | if possible, remove the use of macro's and start using Gosub's instead (also in the Dial option) |
| 13:24.01 | wdoekes | and, does your [macro-call-macro] context exist? |
| 13:28.30 | imihaylov | yes it exists |
| 13:28.39 | imihaylov | and its doing its purpose |
| 13:28.46 | *** join/#asterisk italorossi (~italoross@200.174.102.85) |
| 13:29.11 | imihaylov | it worked well till some point |
| 13:29.19 | imihaylov | but I don't know what changed |
| 13:33.49 | anonymouz666 | hi wdoekes, are you using the new chan_pjsip? |
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| 13:38.03 | wdoekes | imihaylov: that's why you have version control (etckeeper?); to see what changed |
| 13:38.41 | wdoekes | anonymouz666: not really. I've set it up for someone, but didn't test more than a single call with it |
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| 13:40.31 | imihaylov | i have backups for the asterisk folder |
| 13:41.05 | imihaylov | i am pretty sure the thin about the Dial command that was changed is the number of sips |
| 13:41.08 | imihaylov | nothing else |
| 13:41.36 | anonymouz666 | asterisk folder? are you using it on windows? :-) |
| 13:42.27 | imihaylov | no |
| 13:42.39 | imihaylov | dicrectory ..apf |
| 13:42.59 | imihaylov | so nothing changed in the behavior after removing the M(macro) |
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| 13:43.13 | imihaylov | it's looping all over |
| 13:43.15 | imihaylov | again |
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| 14:01.07 | wdoekes | imihaylov: if you're so pretty sure, then reduce the number of destinations in the Dial string |
| 14:01.32 | wdoekes | you're the one who's able to debug this by trial and error |
| 14:02.08 | wdoekes | perhaps one of your SIP peers sends a 302 back to your system, creating a loop |
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| 14:03.29 | imihaylov | hmmm |
| 14:03.46 | imihaylov | actually it worked |
| 14:04.41 | imihaylov | reducing the sip receivers |
| 14:05.04 | imihaylov | is there limit of dialing sips |
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| 14:14.45 | wdoekes | imihaylov: 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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| 14:35.06 | imihaylov | Dial 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. |
| 14:35.18 | imihaylov | here is the dialplan itself: http://pastebin.com/wgZwpTcY |
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| 14:39.47 | [TK]D-Fender | Show the CALL |
| 14:53.07 | imihaylov | <PROTECTED> |
| 14:53.14 | imihaylov | here is the call |
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| 15:08.57 | cusco | hi |
| 15:09.54 | cusco | if peer hangs up a channel while in a Gosub(), asterisk cli shows: "... Abnormal exit ..." |
| 15:10.13 | [TK]D-Fender | imihaylov: I don't see a "loop". I see a completely separate 2nd call |
| 15:10.30 | [TK]D-Fender | #4 1st call died completely |
| 15:10.32 | cusco | is there a variable set or so? |
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| 15:11.05 | cusco | I'm using gosub for the first time, and I'm trying to playback a audio to a callee when he answers a queue() |
| 15:13.23 | cusco | but 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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| 15:26.50 | imihaylov | yes there are 2 separate calls in asterisk but the caller (in this case me) makes one call and receives continues free dial tone |
| 15:27.08 | imihaylov | till I cancel the call |
| 15:31.54 | [TK]D-Fender | Looks like didww does this.. not your system |
| 15:32.02 | [TK]D-Fender | "sip set debug on" <- look how the call actually ends |
| 15:38.32 | imihaylov | I will check it tomorrow , now it's pretty busy and there is ton of info from the debug |
| 15:38.59 | imihaylov | thanks for the tips. |
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| 15:52.45 | jzu_ | hmmm |
| 15:52.47 | jzu_ | <PROTECTED> |
| 15:53.34 | jzu_ | http://pastebin.com/GHGvH7Qr |
| 15:53.45 | jzu_ | trunk expect to get +358, eh? |
| 15:53.48 | jzu_ | *expects |
| 15:54.07 | jzu_ | 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-Fender | https://www.localphone.com/help/voip <----------- |
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| 16:13.38 | jzu_ | [TK]D-Fender: yes, fixed my dialplan to remove first 0 and add 358 for Finland country code |
| 16:14.07 | [TK]D-Fender | That is NOT how they say you should be dialing |
| 16:15.21 | *** join/#asterisk [Consultant]C-As (~Pro-Aster@c-73-207-183-115.hsd1.ga.comcast.net) |
| 16:15.56 | jzu_ | "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." |
| 16:16.06 | jzu_ | what they say? :O |
| 16:16.22 | *** part/#asterisk caveat- (hoax@shell.bshellz.net) |
| 16:17.25 | [Consultant]C-As | hi, 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. Youâll find your contacts by clicking on the button with icons of people." |
| 16:18.12 | jzu_ | exten => _0Z.,n,Macro(dialout-trunk,6,358${EXTEN:1},,off) |
| 16:18.48 | jzu_ | [TK]D-Fender: :D |
| 16:19.52 | [TK]D-Fender | BRB |
| 16:20.32 | jzu_ | wonder how I should make the dialplan so I could call real international calls if needed... |
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| 16:24.38 | jzu_ | so yes I ditched Sonetel in favor of Localphone and now I'm happy. |
| 16:24.51 | jzu_ | no matter what I did Sonetel didn't want to send out CID |
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| 17:51.34 | Mango45 | Is 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-As | i 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-As | is there a setting to change this or fix it? |
| 18:25.51 | [TK]D-Fender | Show us |
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| 18:57.58 | [Consultant]C-As | <PROTECTED> |
| 18:59.01 | [Consultant]C-As | here is a complete dump of the bad ANSWEREDTIME =0 example http://pastebin.com/C3CEhhH1 |
| 19:14.25 | [Consultant]C-As | my 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-As | it 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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| 20:30.09 | nickse | !bool |
| 20:30.10 | nickse | !book |
| 20:30.15 | nickse | ~book |
| 20:30.15 | 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 |
| 20:31.13 | nickse | $46.99 for a book :O |
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| 20:32.36 | philfry | has anyone ran into issues specifically with goto meeting phone numbers? |
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| 20:54.50 | zekoZeko | hey 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 |
| 20:54.52 | zekoZeko | raspbx is incomprehensible to me :( |
| 20:57.41 | [Consultant]C-As | my issue doesnt happen with asterisk 11 it seems |
| 20:59.02 | [TK]D-Fender | zekoZeko, Your description got a bit circular. And you should keep FreePBX questions in their channel #freepbx |
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| 20:59.56 | zekoZeko | [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-Fender | Fix your description first of what it should be doing, and how it isn't |
| 21:03.56 | zekoZeko | will do and pastebin it somewhere. |
| 21:07.02 | philfry | SIP or IAX2? Is there a difference in reliability? |
| 21:07.19 | [TK]D-Fender | not really |
| 21:07.43 | [TK]D-Fender | That isn't a point between them |
| 21:10.53 | philfry | I 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-Fender | Start by getting rid of all the vague bits you just introduced there |
| 21:12.06 | [TK]D-Fender | And test with another provider |
| 21:12.15 | [TK]D-Fender | Remove YOURSELF as the weak link in the chain |
| 21:12.53 | philfry | can you explain that a little bit more? |
| 21:13.25 | robmal | philfry: [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-Fender | Before thinking that the experience was bad because of the vendor... go see if it's any better with another. |
| 21:14.06 | [TK]D-Fender | Maybe i'ts YOUR situation that's bad and it won't matter where you go |
| 21:14.15 | philfry | okay 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-Fender | AKA "do proper debugging and comparative testing" |
| 21:14.41 | [TK]D-Fender | And someone else's experine won't math your scenario because of your provider, and your equipment, etc |
| 21:14.42 | philfry | I'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-Fender | 15 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-Fender | And start drilling the calls. |
| 21:16.14 | philfry | okay i'll investigate the firewall maybe there is an udp connection that is being blocked |
| 21:16.15 | philfry | thanks |
| 21:18.29 | [TK]D-Fender | check the start of your calls to verify that initial settings are right |
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| 21:21.16 | philfry | can 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-Fender | That means digging the entire SIP setup of the call |
| 21:27.50 | [TK]D-Fender | FreePBX-speak really means nothing. |
| 21:28.06 | [TK]D-Fender | It's all about what is being told to go where. |
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| 21:33.46 | [Consultant]C-As | lol @ robmal |
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| 21:34.26 | robmal | [Consultant]C-As: What? ;-> |
| 21:39.28 | zekoZeko | i 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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| 21:48.29 | zekoZeko | OK 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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| 21:51.36 | j-fish | any recommendations for a wirelss voip phone for a call center? |
| 21:52.08 | [TK]D-Fender | DECT handset + SIP receiver base. |
| 21:52.18 | [TK]D-Fender | ~wifivoip |
| 21:52.18 | infobot | [~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-Fender | also "for a call center" ... really doesn't mean anything at all. |
| 21:55.27 | j-fish | it 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-Fender | At no point did I mention Bluetooth |
| 21:58.44 | [TK]D-Fender | And what is a "bluetooth phone" exactly? |
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| 22:03.31 | j-fish | so which DECT+ sip receiver base brand/company you recommend |
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| 22:13.18 | zekoZeko | i 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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| 22:18.01 | j-fish | zekoZeko: thank you |
| 22:20.21 | pjensen00 | if 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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