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| 01:19.54 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.19.0 (2018/01/11), Standard: 15.2.0 (2018/01/11); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 06:44.21 | cwhuang | file: about the incorrect answer and end time in cdr_mysql, it's a regression introduced by commit 1199927f. |
| 06:44.37 | cwhuang | file: yes, it's introduced several months ago |
| 06:44.56 | cwhuang | file: I fixed it by https://github.com/cwhuang/asterisk/commit/4e012662f2d982f090f76904f6a714581f9ac078 |
| 06:45.46 | cwhuang | It works as expected. If you think it's OK, I'll send a pull request. |
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| 08:29.32 | BeeBuu | hello,all. I need a help on how to using playSIP to make a call to a client and record it...please? |
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| 08:41.28 | BeeBuu | anyone help me ,thanks. |
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| 15:42.37 | Covale | hello. is anyone here familiar with configuring Asterisk while using a 4000 series Cisco router? I am asking because we just upgraded to the 4000 series routers from 2900 series and now calls going through CUBE are ignoring the audio packets where before they were going through just fine. |
| 15:43.28 | Covale | I was given a long explanation from the Cisco engineer that might better explain my situation. however, I do not 100% trust the Cisco engineer's explanation. |
| 15:44.21 | sekil | what is the explanation? |
| 15:45.18 | Covale | he said something about the SIP connection being set to "re-invite based transfer" where it should to be set to "refer based transfer" |
| 15:46.25 | Covale | it's pretty long. is it acceptable etiquette to just paste to the channel? |
| 15:46.48 | Covale | sorry, been out of touch on IRC for a while, so please forgive me ignorance. |
| 15:47.45 | sibiria | use pastebin.com |
| 15:48.33 | Covale | ahh, now I remember. I thought I remembered something about it not being cool to paste walls to the channel. thanks for the tip. back in a min |
| 15:50.12 | igcewieling | For the most part if you disable SIP ALG things will start working better. |
| 15:51.09 | Covale | https://pastebin.com/kdA72PLh |
| 15:51.44 | Covale | that's the emailed explanation by the Cisco engineer. I hope it made sense. |
| 15:53.08 | Covale | I heard something about ALG and it being the mother of all cockblockers... |
| 15:54.15 | Covale | I will have to look into that as well. but before I suggest this to the Cisco fellas, I want to make sure there is not some setting in Asterisk I can try first. |
| 15:55.00 | Covale | like a super secret "youwantyourcallstowork=yes" |
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| 16:06.37 | sekil | Covale: this is to stop rtpbleed |
| 16:07.09 | sekil | Covale: https://www.rtpbleed.com/ |
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| 16:18.10 | yeoj | is there another more generic sip/voip channel on freenode? i have a question that's sip/voip/foip related, but not specific to asterisk, but i'm not sure where to find a larger group of people with similar subject matter knowledge... |
| 16:22.46 | Covale | sekil: Thanks for the link, was an interesting read. however, what does the rtp bleed bug have to do with my problem? |
| 16:23.26 | sekil | SR 4000 router will check the source port and IP to consider the packets are valid. So, that is the reason, new CUBE is ignoring these packet because source port and ip are different( IP phoneâs) then the IVRâs |
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| 16:24.53 | Covale | so they are blocking my audio packets in a wild attempt to eliminate the rtp bleed bug? |
| 16:25.01 | Covale | the router, I mean |
| 16:26.13 | Covale | so I need to pass that link on to the network folks so they can stop blocking MY packets, is where you are going with that, right? |
| 16:27.14 | sekil | Covale: no |
| 16:27.20 | sekil | Covale: signalling is the issue |
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| 16:27.59 | sekil | Covale: Agent phone got instruction from CUCM to send media to CUBEâs IP. So, Phone started sending media to CUBEâs IP. |
| 16:29.59 | sekil | Covale: what he's telling you is that with CUCM sending REFER back with proper phone information..it should work.. |
| 16:30.54 | sekil | Covale: cisco CUCM usually sends INVITEs w/o SDP |
| 16:31.06 | Covale | ok, that sounds like what he was talking about. so how to I configure my Asterisk SIP connection to send REFER instead of INVITE? |
| 16:32.04 | sekil | Covale: where's Asterisk in this picture? |
| 16:32.13 | sekil | Covale: it says CUCM and CUBE |
| 16:32.30 | Covale | transferring calls to CUCM for zero-out |
| 16:32.53 | Covale | sorry, I been breathing this, forgot that wasn't obvious. |
| 16:33.04 | Covale | been too close to this for a while. you understand. |
| 16:33.13 | igcewieling | You can't disable SIP ALG on the routers? |
| 16:33.20 | file | I think the use of "transferring" is being incorrectly used |
| 16:34.17 | Covale | the Asterisk system takes a call, the user does whatever he wants to do, presses options, and when it comes time to reach an operator, they press zero which the Asterisk system sends the call to a CUCM system on the same internal network over a SIP connection. |
| 16:34.41 | file | yeah, that's a normal call not a transfer |
| 16:34.56 | Covale | I have someone looking into ALG as we speak, but it takes forever to hear back from network guys - ironically. |
| 16:35.15 | file | have you configured chan_sip (presumably) to not do a re-invite? |
| 16:35.17 | Covale | and you are probably right about the call/transfer thing. |
| 16:35.27 | sekil | so I'm completely lost now...not the flow I thought it's there |
| 16:35.45 | Covale | @file: no, how would I do that? |
| 16:35.48 | sekil | Covale: just don't do re-invites like file said.. |
| 16:35.52 | file | directmedia=no |
| 16:36.15 | Covale | thank you! I think that is the magic setting I needed. I will try this and let you guys know. |
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| 16:40.01 | Covale | so if I have a canreinvite=yes, that setting was renamed to directmedia at some point? so I should remove the canreinvite and replace that with directmedia=no? |
| 16:40.46 | file | yes. |
| 16:41.01 | Covale | perfect, thanks! |
| 16:41.04 | file | that will force media to be relayed through Asterisk, as apparently your new setup can't handle reinviting. |
| 16:41.25 | file | you also can't change to using a REFER because they serve two completely different purposes |
| 16:41.50 | file | a re-invite changes the media address information to have media flow directly, a REFER is used for blind or attended call transfers |
| 16:42.41 | sekil | file: yeah I didn't know about asterisk at all |
| 16:43.23 | sekil | file: Cisco usually does 3PCC INVITEs |
| 16:47.31 | Samot | Well.. |
| 16:47.36 | Samot | REFER is not an Asterisk thing. |
| 16:47.40 | Samot | It's a SIP thing. |
| 16:48.31 | file | I REFER you to the spec, ahahahahahaha |
| 16:48.35 | file | falls over |
| 16:48.48 | file | uses NAPTR and SRV to locate an alternative server for failover purposes |
| 16:48.50 | Samot | Dear god. |
| 16:49.51 | Samot | I run my own DNS to have full control of the NAPTR/SRV records for my sip domains. |
| 16:50.27 | Samot | And XMPP stuff. |
| 16:51.40 | sekil | Cisco chose to use 3PCC for its SIP to emulate SCCP |
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| 16:52.47 | file | Cisco: 10 SIP stacks are better than one. |
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| 18:58.39 | Worldexe | hmmm, do we have any way to call some macro/extension/sub periodically? i mean, once per minute or so - regardless of channel status as far as channel exists? |
| 18:58.49 | Worldexe | didnt find any myself ( |
| 18:59.54 | JohnWigley | worldexe: not internally to Asterisk, but you can call one externally through a call file dropped via a cron job, or through the other interfaces such as ARI |
| 19:00.32 | [TK]D-Fender | Worldexe, To do what exactly? |
| 19:01.58 | Worldexe | yeah, got it... |
| 19:02.58 | Covale | file: if I am changing "directmedia=no" should I also change the setting for "insecure="? I currently have it set to "insecure=port,invite" |
| 19:03.11 | file | they are unrelated. |
| 19:03.31 | Covale | ok, I suspected so, but wanted to check first. thanks! |
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| 19:04.44 | Worldexe | well, its complicated ) Im trying to find/mitigate very rare calls, that I can not handle properly, because hangup part of dialplan is not called for them, so they become 'trapped' in my statistics system |
| 19:06.51 | JohnWigley | worldexe: What does your scheduled extension do in this process? |
| 19:11.04 | igcewieling | Worldexe: hangups using the 'h' extension? |
| 19:11.28 | igcewieling | If so, switch to using hangup handlers, they are far more reliable than exten 'h' |
| 19:11.59 | igcewieling | Worldexe: you could also try setting an RTP timeout to drop calls which are dead. |
| 19:13.17 | Worldexe | nono, they are dropped properly, there are just no 'h' being called - so they become 'zombie' in my db table. |
| 19:13.31 | Worldexe | hmm, hangup handlers is an idea, thanks |
| 19:13.45 | Worldexe | i was using 'h' and never had any problems untils now ) |
| 19:14.19 | igcewieling | Worldexe: the problem with 'h' is you need one in every context a hangup might happen in. With hangup handlers that isn't an issue anymore. |
| 19:15.04 | Worldexe | yeah; i use Lua dialplan and add them automatically at the end of the script; so no missing handlers, im sure ) |
| 19:15.41 | Worldexe | i guess this happens when my Originate call being answered and immediately dropped by far side while being Queue()-ed |
| 19:16.08 | Worldexe | something like this |
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| 20:08.34 | sebastienthiry | Hello guys, I'm trying to configure asterisk to work with Csipsimple on my smartphone with secure connexion. My phone can register with secure sip but I can't pass a call, I get "process_sdp: Rejecting secure audio stream without encryption details: audio 4000 RTP/SAVP 8 101". |
| 20:08.58 | sebastienthiry | Here is my configuration |
| 20:08.59 | sebastienthiry | [413] |
| 20:09.00 | sebastienthiry | type=friend |
| 20:09.00 | sebastienthiry | host=dynamic |
| 20:09.00 | sebastienthiry | qualify=yes |
| 20:09.00 | sebastienthiry | dtmfmode=rfc2833 |
| 20:09.00 | sebastienthiry | callerid=Test TLS <413> |
| 20:09.00 | sebastienthiry | fullname = Test TLS |
| 20:09.01 | sebastienthiry | username = testtls |
| 20:09.01 | sebastienthiry | secret=testTLS |
| 20:09.02 | sebastienthiry | context = maison |
| 20:09.02 | sebastienthiry | transport=tls |
| 20:09.03 | sebastienthiry | encryption=yes |
| 20:09.03 | sebastienthiry | ignorecryptolifetime=yes |
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| 21:30.32 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.19.0 (2018/01/11), Standard: 15.2.0 (2018/01/11); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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