Re: [asterisk-users] Queue show : failed to extend from 240 to 327

On Fri, Sep 9, 2016 at 5:37 PM, Jonas Kellens wrote: > Hello > > when I type on the Asterisk CLi 'queue show', I first get a list of my > queues and then the following : > > > failed to extend from 240 to 327 > failed to extend from 240 to 334 > > > I could not find

[asterisk-users] Queue show : failed to extend from 240 to 327

Hello when I type on the Asterisk CLi 'queue show', I first get a list of my queues and then the following : failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 323 failed to extend from 240 to 327 failed to extend

[asterisk-users] Asterisk 13.11.2 Now Available

The Asterisk Development Team has announced the release of Asterisk 13.11.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.11.2 resolves an issue reported by the community and would have not been possible

[asterisk-users] How to get a list of DAHDI channels

Anyone know an efficient way to get a list of the DAHDI channels? Is there an AMI or ARI variable to get a list of all the channels? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)9116-91161 --

Re: [asterisk-users] Asterisk 13 PJSIP with Snom 710

thanks for the reply. if i config the extension in softphone it works fine. but with snom its not working Bet Regards, Madushan On Fri, Sep 9, 2016 at 10:31 PM, Madushan Geethanga wrote: > yes I have unchecked it. > > On Fri, Sep 9, 2016 at 10:27 PM, Administrator

Re: [asterisk-users] Asterisk 13 PJSIP with Snom 710

yes I have unchecked it. On Fri, Sep 9, 2016 at 10:27 PM, Administrator TOOTAI wrote: > Le 09/09/2016 à 18:32, Madushan Geethanga a écrit : > >> Hi, >> > > If you're not using RTP encryption did you uncheck the option in your RTP > TAB from identity ? > > >> This is the log.

Re: [asterisk-users] Asterisk 13 PJSIP with Snom 710

Le 09/09/2016 à 18:32, Madushan Geethanga a écrit : Hi, If you're not using RTP encryption did you uncheck the option in your RTP TAB from identity ? This is the log. ex dialling 0 from snom phone <--- Received SIP request (1230 bytes) from UDP:123.231.72.210:33878

Re: [asterisk-users] Asterisk 13 PJSIP with Snom 710

Hi, This is the log. ex dialling 0 from snom phone <--- Received SIP request (1230 bytes) from UDP:123.231.72.210:33878 ---> INVITE sip:0@54.206.59.252;user=phone SIP/2.0 Via: SIP/2.0/UDP 123.231.72.210:45835;branch=z9hG4bK-bskkkx1t5bas;rport From: "outburns00-nhvg5vjjn6-2001" <

[asterisk-users] Asterisk 13 PJSIP with Snom 710

Hi, I'm trying to setup snom 710 phone with asterisk 13 with PJSIP. inbound is working fine but i cannot dial out. i don't hear anything on the phone and asterisk CLI also does not show anything. my config is. please advice. [2001] type=endpoint context=out-local

[asterisk-users] (no subject)

Hi, Im trying to setup snom 710 phone with asterisk 13 with PJSIP. inbound is working fine but i cannot dial out. i don't hear anything on the phone and asterisk CLI also does not show anything. my config is. please advice. [2001] type=endpoint context=out-local

Re: [asterisk-users] Asterisk 13 and WebRTC

Hello, I mean a working configuration (SIP o PJSIP) without patches or code corrections. Thank you Regards El 09/09/2016 a las 03:47, marek cervenka escribió: using in production last asterisk 13 + pjsip bundled + pjsip patch for RTP/SAVPF (search pjsip conf) + sipml5 version from

Re: [asterisk-users] [SOLVED] Re: Feature Request: what about "core stop panic" ?

2016-09-09 13:24 GMT+02:00 Jacek Konieczny : > On 2016-09-09 11:34, Olivier wrote: > >> Adding an /etc/sysctl.d/foobar.conf file with the bellow content allowed >> me to at last produce core dump files (in /var/tmp directory), even if >> asterisk is run by asterisk user (and by

Re: [asterisk-users] 13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed

09.09.2016 15:18, Joshua Colp пишет: Dmitry Melekhov wrote: 09.09.2016 14:08, Dmitry Melekhov пишет: And, as I already said, there was no such messages while using asterisk 13.10. I'll open bug report. I was mistaken and it is indeed a bug. I've got a fix up but even without the fix it

Re: [asterisk-users] [SOLVED] Re: Feature Request: what about "core stop panic" ?

On 2016-09-09 11:34, Olivier wrote: Adding an /etc/sysctl.d/foobar.conf file with the bellow content allowed me to at last produce core dump files (in /var/tmp directory), even if asterisk is run by asterisk user (and by root). I choosed this /var/tmp directory to make sure core dumps are not

Re: [asterisk-users] 13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed

Dmitry Melekhov wrote: 09.09.2016 14:08, Dmitry Melekhov пишет: And, as I already said, there was no such messages while using asterisk 13.10. I'll open bug report. I was mistaken and it is indeed a bug. I've got a fix up but even without the fix it isn't blocking requests as you've

Re: [asterisk-users] 13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed

09.09.2016 14:13, Joshua Colp пишет: Dmitry Melekhov wrote: 09.09.2016 14:08, Dmitry Melekhov пишет: 09.09.2016 13:45, Joshua Colp пишет: Dmitry Melekhov wrote: Hello! Upgraded 13.10 to 13.11.1 today and now I see messages in log: [Sep 9 12:23:16] NOTICE[6064]

Re: [asterisk-users] 13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed

Dmitry Melekhov wrote: 09.09.2016 14:08, Dmitry Melekhov пишет: 09.09.2016 13:45, Joshua Colp пишет: Dmitry Melekhov wrote: Hello! Upgraded 13.10 to 13.11.1 today and now I see messages in log: [Sep 9 12:23:16] NOTICE[6064] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"3563"

Re: [asterisk-users] 13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed

09.09.2016 14:08, Dmitry Melekhov пишет: 09.09.2016 13:45, Joshua Colp пишет: Dmitry Melekhov wrote: Hello! Upgraded 13.10 to 13.11.1 today and now I see messages in log: [Sep 9 12:23:16] NOTICE[6064] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"3563" '

Re: [asterisk-users] 13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed

09.09.2016 13:45, Joshua Colp пишет: Dmitry Melekhov wrote: Hello! Upgraded 13.10 to 13.11.1 today and now I see messages in log: [Sep 9 12:23:16] NOTICE[6064] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"3563" ' failed for '192.168.32.116:5060' (callid:

Re: [asterisk-users] 13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed

Dmitry Melekhov wrote: Hello! Upgraded 13.10 to 13.11.1 today and now I see messages in log: [Sep 9 12:23:16] NOTICE[6064] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"3563" ' failed for '192.168.32.116:5060' (callid: 0_1409534529@192.168.32.116) - No

Re: [asterisk-users] [SOLVED] Re: Feature Request: what about "core stop panic" ?

Adding an /etc/sysctl.d/foobar.conf file with the bellow content allowed me to at last produce core dump files (in /var/tmp directory), even if asterisk is run by asterisk user (and by root). I choosed this /var/tmp directory to make sure core dumps are not erased after a reboot and because this

[asterisk-users] 13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed

Hello! Upgraded 13.10 to 13.11.1 today and now I see messages in log: [Sep 9 12:23:16] NOTICE[6064] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"3563" ' failed for '192.168.32.116:5060' (callid: 0_1409534529@192.168.32.116) - No matching endpoint found

Re: [asterisk-users] Asterisk 13 and WebRTC

using in production last asterisk 13 + pjsip bundled + pjsip patch for RTP/SAVPF (search pjsip conf) + sipml5 version from roginvs https://github.com/DoubangoTelecom/sipml5/pull/238 Dne 08/09/2016 v 23:36 Annus Fictus napsal(a): Hello list, before to lost my time, I'd like know if someone

Re: [asterisk-users] Asterisk 13 and WebRTC

Hi. It have big audio delay because using extenral ICE servers. Better to use kamailio/opensips + rpenigne infront 2016-09-09 0:36 GMT+03:00 Annus Fictus : > Hello list, > > before to lost my time, I'd like know if someone have a WebRTC working > configuration on Asterisk

[asterisk-users] Different cachertclasses setting for different Music on Hold

As you know, there is the following settings [general] cachertclasses=yes ; use 1 instance of moh class for all users who are using it, ; decrease consumable cpu cycles and memory ; disabled by default It allows to use a single instance of MOH for all