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| 00:16.40 | *** topic/#asterisk is Take the March 2019 Asterisk User Survey! https://goo.gl/forms/xL1VUHRsf95saly13 -- #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.27.1 (2019/7/11) 16.4.1 (2019/7/11), Security Only: 15.7.3 (2019/7/11); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
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| 05:12.05 | fourhundredtheca | I have couple of SIP provider accounts registered on my asterisk |
| 05:12.16 | fourhundredtheca | how can I see whether they are connected via TLS ? |
| 05:12.24 | fourhundredtheca | "sip show tcp" only shows me registered phone devices |
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| 12:38.07 | twanny796 | Error 399, what should I remedy? |
| 12:38.38 | Samot | That is absolutely useless. |
| 12:38.50 | Samot | It means nothing. Be more descriptive. |
| 12:39.30 | twanny796 | I have a PABX, /not asterisk/, and I am trying to connect a sip trunk with an ITSP. |
| 12:40.01 | Samot | So it doesn't even use Asterisk? |
| 12:40.09 | *** topic/#asterisk by file -> Take the March 2019 Asterisk User Survey! https://goo.gl/forms/xL1VUHRsf95saly13 -- #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.28.0 (2019/7/25) 16.5.0 (2019/7/25), Security Only: 15.7.3 (2019/7/11); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
| 12:40.37 | twanny796 | No, this time it's not asterisk, but it's sip |
| 12:41.02 | Samot | OK so Error 399 means nothing in SIP. |
| 12:41.41 | Samot | So you either need to look at your PBX docs or call the ITSP for help or both. |
| 12:41.46 | twanny796 | Samot: I'm very sorry, that was the length of the packet :( |
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| 12:44.40 | Samot | twanny796: I'm not sure what you expect us to solve here... |
| 12:45.17 | twanny796 | I'll tell you the story. |
| 12:46.39 | twanny796 | first I call a number 'Request: INVITE', then 'Status: 183 Session Progress, from itsp. Then 'Status 487 Request terminated, from itsp. |
| 12:47.02 | twanny796 | then Request: Ack from the pabx. |
| 12:47.17 | twanny796 | Can I find some error in the packets? |
| 12:48.10 | Samot | So why is the ITSP terminating the call? |
| 12:48.58 | twanny796 | They told me there's no PPI ??? |
| 12:49.05 | twanny796 | in the header. |
| 12:49.18 | Samot | I have no idea what that means. |
| 12:49.19 | twanny796 | This is a test account |
| 12:49.59 | Samot | What type of PBX is this? |
| 12:50.21 | twanny796 | OpenScape Business UNIY |
| 12:50.24 | twanny796 | UNIFY |
| 12:50.42 | Samot | Then you need to go seek help with them. |
| 12:50.51 | twanny796 | yep |
| 12:51.12 | Samot | So far this is looking like you don't have things configured properly. |
| 12:51.50 | twanny796 | yes I know, but I cannot see any reason in the packets |
| 12:52.19 | twanny796 | just request terminated |
| 12:57.18 | Samot | That's the reason. |
| 12:57.23 | Samot | They terminated the call. |
| 12:57.36 | Samot | There is no error because everything is correct. |
| 12:57.44 | Samot | Again, they said it was because of no PRI. |
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| 12:57.59 | Samot | I'm not sure what they means for them but they must be expecting a setting from you. |
| 12:59.08 | twanny796 | I asked the tech to send me the call log, but he didn't come back yet. Probabbly it would show something. |
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| 13:06.36 | Samot | OK so are you sending them all the right details? |
| 13:06.39 | Samot | All the right settings? |
| 13:06.53 | Samot | This wasn't a 4XX or a 6XX error. |
| 13:07.02 | Samot | Well a user error. |
| 13:07.17 | Samot | They sent back CANCEL. |
| 13:07.34 | Samot | Generally the called side doesn't CANCEL the call. The calling side does. |
| 13:07.37 | twanny796 | Can I upload a tcpdump somewhere? |
| 13:07.43 | Samot | pastebin |
| 13:13.25 | twanny796 | I wrote to the itsp, and send them the tcpdump. |
| 13:14.01 | twanny796 | On another note. In PJSIP how do I set up an itsp with an SRV record? |
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| 13:15.40 | file | you put their hostname in the SIP URI, and that's it |
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| 13:27.41 | twanny796 | ok, that worked. |
| 13:28.04 | twanny796 | have to go |
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| 13:58.44 | twanny796 | now I'm trying the test account on Asterisk and for an incoming call I get 'No matching endpoint found'. |
| 13:59.11 | twanny796 | I have another sip account with the same configuration and it's working fine. |
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| 15:35.26 | qakhan | Hi all, I have completed call pickup from queue. now here are few questions. If call is ringing on exten 3004 and exten 3005 pick up the call. Queue_log table shows the call was answered by 3004. |
| 15:35.37 | qakhan | Can we pickup a call in queue which is not ringing on exten and waiting in the queue? |
| 15:37.26 | [TK]D-Fender | I've answered this on at elast 2 separate days already |
| 15:37.35 | [TK]D-Fender | What is your problem with comprehending this? |
| 15:37.53 | [TK]D-Fender | egcweiling has also told you the same |
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| 15:38.16 | [TK]D-Fender | use AMI or ARI to hijack the call with a Redirect or Bridge |
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| 15:49.24 | Kobaz | SwK: |
| 15:49.25 | Kobaz | [2019-07-24 15:55:26] <SwK> I like some of the automation they added in so you can trigger things on the phone |
| 15:49.36 | Kobaz | what kind of automation? re: polycom vvx obi stuffs |
| 15:52.11 | Samot | It's not really automation more than it's an API. |
| 15:52.22 | Samot | So you can do things like make calls, hangup, etc. etc. |
| 15:53.06 | NWATechSupport | Is there a compelling reason to note use the built-in SMTP server? |
| 15:53.13 | NWATechSupport | not* |
| 15:53.25 | Samot | What built in SMTP server? |
| 15:53.50 | NWATechSupport | System Admin->Email Setup -- Options are to use built in or external SMTP server. |
| 15:53.58 | NWATechSupport | Is this a freepbx question instead of asterisk? |
| 15:54.32 | Samot | It means do you want to send mail locally from this machine straight to the Internet or do you want to relay through an SMTP server. |
| 15:54.54 | Samot | So it basically will configure the STMP server to either do one or the other. |
| 15:55.09 | Kobaz | Samot: and this is only on obi's ? |
| 15:55.28 | Samot | Kobaz: Not really many makers have added APIs. |
| 15:55.36 | [TK]D-Fender | Asterisk doesn't have a mail server |
| 15:56.06 | Samot | Kobaz: The REST API was introduced in UC 5.9.0 |
| 15:56.11 | Samot | It's not an Obi only thing. |
| 15:56.13 | [TK]D-Fender | You should understand what Asterisk actually is by now.... |
| 15:57.18 | Samot | And UC 5.9.x is the last UC release for the first gen VVX series. |
| 15:57.26 | Samot | So the 400/401, 500/501, etc. |
| 15:57.53 | NWATechSupport | Thank you, Samot. |
| 15:59.21 | Kobaz | right |
| 15:59.40 | Kobaz | yeah, i haven't researched anything in the new firmwares, that'll be useful for testing |
| 15:59.59 | Samot | Testing what? |
| 16:07.54 | Kobaz | my unit testing |
| 16:08.14 | Kobaz | i use a modified sipp for unit testing |
| 16:08.26 | Kobaz | it would be nicer to use a real polycom phone for certain things |
| 16:09.04 | Kobaz | i basically wrote a remote control for sipp, so i can make it place calls, hang up calls, blind transfer calls, and replaces transfer calls |
| 16:09.34 | Kobaz | the cool thing about sipp vs a real phone, i can spwan hundreds of them doing different tasks |
| 16:09.42 | Kobaz | and do load testing or whatever |
| 16:09.50 | Kobaz | try and replicate deadlocks, things like that |
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| 16:18.31 | Kobaz | i was using asterisk as a sip ua.. sending commands on ami to start and stop calls |
| 16:18.47 | Kobaz | but i couldn't figure out how to do replaces transfer on asterisk without really digging deep into chan_sip |
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| 17:09.59 | qakhan | [TK]D-Fender i queue pickup call has been completed. I use AMI originate to an agent who want to answer the call. once he/she anser the call then i use PickupChan() to pickup the call. |
| 17:12.21 | qakhan | in order to use Redirect or Bridge AMI actions. i have to first get the agent (who wants to anser the call) channel and then connect both channels. |
| 17:13.12 | qakhan | question is why Queue_log records that 3004 answered the call which 3005 picked up the call |
| 17:20.23 | [TK]D-Fender | Because that's who the queue called |
| 17:20.32 | [TK]D-Fender | if you cheat Asterisk doesn't care |
| 17:23.02 | qakhan | ok, understand. can we answer the call which is waiting in the queue and not ringing on any exten? |
| 17:23.39 | [TK]D-Fender | WE'VE JUST ANSWERED THIS OVER 5 TIMES NOW |
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| 18:48.34 | seanbright | anyone here familiar with rtpengine's release versions? |
| 18:49.12 | seanbright | there's a 6.5, 7.1, 7.2, 7.3, and 7.4 all released 6 days ago - do i just download the latest? |
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