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| 07:29.24 | topro | Hi there, running Asterisk 13 on Debian (Devuan) on a german Telekom all-IP SIP line. I've had provider SIP failing about 1-4 times per year, but since most recent update of Asterisk packages (13.14.1~dfsg-2+deb9u4) I'm losing provider SIP line functionality multiple times a day (!!). When trying to establish a call then remote SIP will return 403-Forbidden. Anyone ever experienced something similar and could give me a hint on where to look at, please? |
| 07:31.11 | topro | Btw. only measure I figured out so far to get it working again is to restart asterisk |
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| 10:46.11 | Helenah1 | Hi |
| 10:46.46 | Helenah1 | I can't get a single Android SIP client to authenticate properly while desktop clients authenticate no problem. |
| 10:46.49 | Helenah1 | [Oct 21 11:44:36] NOTICE[10625]: res_pjsip/pjsip_distributor.c:649 log_failed_request: Request 'REGISTER' from '<sip:2003@10.1.1.254>' failed for '10.1.1.63:35096' (callid: wl_jZD9L-KqP0zvvSXW0YA..) - Failed to authenticate |
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| 12:59.25 | Helenah | I forgot to rejoin. |
| 13:00.32 | sibiria | there were no replies to your question while you were gone |
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| 14:14.17 | Helenah | sibiria: That's good to know, I may ask again in a bit. |
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| 15:03.57 | Helenah | 44/buffer 2 |
| 15:17.48 | igcewieling | Ouch: "I would just like to advise that your term fees to disconnect the 2 ckts early will roughly be around $40,000" |
| 15:19.11 | file | that's a few dollars |
| 15:19.29 | sibiria | sounds like a nasty contract for your soul |
| 15:19.50 | Samot | Those are some big circuits or a long term that is being cancelled almost as soon as it started or both. |
| 15:23.52 | igcewieling | I suspect a 3-year contract |
| 15:24.19 | igcewieling | We just pass on the disco fee from the carrier. |
| 15:25.19 | Samot | Then those are some costly circuits. |
| 15:27.43 | igcewieling | Yup. |
| 15:28.56 | igcewieling | They are not a VOIP client so I'm not really involved with them, I was a bit shocked by the early term fee. |
| 15:43.53 | *** join/#asterisk cusco (~tralala@neptune.tretas.eu) |
| 15:44.29 | cusco | hello, using mixmonitor what sets the quality of the recorded call? the formwat we're writting it in + codec being used? |
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| 15:45.11 | cusco | until now we're recording everything in gsm, it has been fine for ages, but there are certain calls where we would like to record with more quality, we mostly use alaw |
| 15:45.21 | cusco | so saving as a file.al will improve it? |
| 15:48.23 | adeel | anyone have experience with the WaitForNoise() application? if so, how sensitive is its detection? |
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| 16:16.12 | Helenah | Hi, every device can authenicate with my Asterisk system, but the one on my mobile phone. I have double checked the password, even changed the password. Still no luck. I then switched SIP clients and still no luck. [Oct 21 17:15:41] NOTICE[11421]: res_pjsip/pjsip_distributor.c:649 log_failed_request: Request 'REGISTER' from '<sip:2003@10.1.1.254>' failed for '10.1.1.63:54535' (callid: |
| 16:16.13 | Helenah | a8fb1c82-387932ab-41609dd7-a85bc09d) - Failed to authenticate |
| 16:17.01 | igcewieling | Good luck! |
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| 19:04.32 | NWATechSupport | ls |
| 19:04.36 | NWATechSupport | oh good lord. :P |
| 19:05.46 | NWATechSupport | I am trying to find to location of provisioning logs (over tftp or http) on the Sangoma7 distro. I'm not recognizing the logs in /var/log/*. Can someone point me in the right direction? |
| 19:06.48 | NWATechSupport | I've found the mail logs, asterisk logs, and httpd logs well enough. |
| 19:07.49 | lagzilla | Shouldn't the provisioning be in httpd? I don't have the applicance |
| 19:15.56 | NWATechSupport | That would make sense for provisioning over http. How about tftp? |
| 19:16.46 | NWATechSupport | eeeh. May have found my answer, myself. Looks like it needs to specifically be enabled, first. |
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| 19:17.21 | lagzilla | yeah and it will end up in /var/log/messages |
| 19:17.33 | NWATechSupport | Thank you |
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| 21:28.42 | adeel | i'm having trouble getting the WaitForNoise() application to work properly....from the description, i'd expect that if there's "talking" aka noise for the minimum duration, it should then move on to the next dialplan priority...however, i don't see that happening |
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| 22:08.04 | adeel | interesting...figured out the waitfornoise() application issue i was having....the app checks each RTP packet for any audio beyond the silence threshold, and needs it for the whole duration of the time threshold, so any pauses between words resets |
| 22:14.33 | Helenah | Hi, every device can authenicate with my Asterisk system, but the one on my mobile phone. I have double checked the password, even changed the password. Still no luck. I then switched SIP clients and still no luck. [Oct 21 17:15:41] NOTICE[11421]: res_pjsip/pjsip_distributor.c:649 log_failed_request: Request 'REGISTER' from '<sip:2003@10.1.1.254>' failed for '10.1.1.63:54535' (callid: |
| 22:14.35 | Helenah | a8fb1c82-387932ab-41609dd7-a85bc09d) - Failed to authenticate |
| 22:16.13 | Helenah | Hiya adeel :) |
| 22:22.24 | *** join/#asterisk Tagor (~Tagor@94-208-76-54.cable.dynamic.v4.ziggo.nl) |
| 22:23.04 | Tagor | how can I detect DTMF input while Dial() is ringing? |
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| 22:35.22 | Helenah | Ohai Tagor :) |
| 22:52.05 | Helenah | Apparently MixMonitor recordings are stored in /var/spool/asterisk/monitor, however they are not there nor do I have a monitor directory. |
| 22:54.28 | Helenah | Sorry about that, permissions issue, they are there. |
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