similar to: Asterisk 11.9.0 crash and restart

I am currently running two different versions of Asterisk 11.0.1 11.2.1 I have noticed the bug occur on both servers. The issue is that when I try to dial a phone number sometimes the call will never go out. I will check the Asterisk server with NGREP and see that the SIP messages are making it to Asterisk but Asterisk isn't responding. I do the following command "netstat -nap |grep

The Asterisk Development Team has announced the release of Asterisk 13.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.8.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: New

Hi, after upgrade from 11.8.1 to 11.9.0 on our test server, and from 1.8.26.1 to 1.8.27 on production one, some CLI commands like "sip reload" or "iax2 reload" does nothing. We opened bug 23683 but it was immediately closed by Matt Jordan, telling that he can't reproduce it. But we can. Example: - switching back to 11.8.1 respectively 1.8.26.1 does the job working

Hopefully someone can point me in the correct direction. I had a 1.4x system die on me yesterday, while I was prepping a new machine to replace it. Took the machine on site yesterday and spent the day and part of the evening getting things working. This morning, I finished up converting my dial plan, knowing there'd be calls of things that I missed. While testing, I've noted that all

The Asterisk Development Team has announced the release of Asterisk 11.9.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.9.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

The Asterisk Development Team has announced the release of Asterisk 11.9.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.9.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

The Asterisk Development Team has announced the release of Asterisk 13.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.8.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: New

I am trying to find where the authoritative sql definitions for Asterisk Realtime Architecture ARA are located. I have found many locations but each and everyone seems to be different. http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html http://www.open-voip.org/index.php?title=Asterisk_Full_RealTime_example Files included with the distribution:

Hi, Have anyone tried using SIPML5 to connect to Asterisk over wss? I'm having the error as shown below Connecting to 'wss://54.xxx.xxx.xxx:8080/ws <wss://54.254.228.251:8080/ws>' SIPml-api.js?svn=224:1 ==stack event = starting SIPml-api.js?svn=224:1 __tsip_transport_ws_onerror SIPml-api.js?svn=224:1 __tsip_transport_ws_onclose SIPml-api.js?svn=224:1 ==stack event =

Hi ! My name is Gerald and I am working with WEBRTC and JsSIP.<div><br></div><div>I configure my Asterisk 11.7.0 to work wit WEBRTC.</div><div><br></div><div>Using a JsSIP (http://tryit.jssip.net/), the SIP extension can connect at the Asterisk, but when we try to make a call they send a 488 response and finish

Asterisk-11.9.0, Fedora 20: res_calendar_caldav.so => (Asterisk CalDAV Calendar Integration) [Apr 27 10:49:13] ERROR[4255]: res_calendar_ews.c:911 load_module: Exchange Web Service calendar module require neon >= 0.29.1, but neon 0.30.0: Library build, IPv6, Expat 2.1.0, zlib 1.2.8, GNU TLS 3.1.13. is installed. Is this a bug, or do I need to downgrade to 0.29? sean

Hi All, Currently i am integrating webRTC demo. I have issue using firefox,someone suggest me to enable DTLS for webRTC working in firefox using Asterisk. I am using Asterisk 11.9.0. https://groups.google.com/forum/#!searchin/doubango/bhavik/doubango/Mv9u0YkNb90/55VElJ1TdY8J Can any one tell me how to enable DTLS ? -- Thanks, Bhavik Patel -------------- next part -------------- An HTML

Hello friends, I have been experienced suddenly stops for my Asterisk server, I do not why is it happening. Asterisk's debug messages only tell me I have lacked g729 codec for translation to one peer minutes before the crashes occur [2014-05-27 09:48:30] WARNING[15384][C-0000017c] channel.c: Unable to find a codec translation path from (ulaw) to (g729) [2014-05-27 09:48:30]

On 11.9.0: > -- Accepting AUTHENTICATED call from 111.xxx.yyy.zzz: > -- > requested format = speex, > -- > requested prefs = (), > -- > actual format = ulaw, > -- > host prefs = (silk16|ulaw|gsm|g722), > -- > priority = mine > -- Executing [8447 at voip-in:1] Dial("IAX2/n4-5734",

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Is it possible to use the instant messaging feature of Polycom phones in Asterisk? At the moment I'm seeing this in the SIP messaging when I try to send one from a Polycom 450. <--- SIP read from UDP:<CENSORED POLYCOM IP>:5060 ---> INVITE sip:0100@<CENSORED>:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP <CENSORED POLYCOM

Hi, I spent the past week experimenting with webrtc + asterisk 11.9.0-rc1 + opus/vb8 codec patch. This is interesting technology and I try to find out how to connect all the moving parts. Firefox: Neither sipml5 or jssip works with calls to asterisk, audio/video doesn't matter. WARNING[977][C-00000005] chan_sip.c: Rejecting secure audio stream without encryption details: audio 35684

The Asterisk Development Team has announced the release of Certified Asterisk 13.8-cert1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/certified-asterisk The release of Certified Asterisk 13.8-cert1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the

Hi All, I am trying to configure webRTC phone example for SIPml5 and i found this info from https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support . I have asterisk 11.9.0 installed and downloaded source of SIPml5 from http://code.google.com/p/sipml5/source/checkout I copied sample code into web root directory and example loaded successfully and also able to register 2 extensions. I

Hi, I'm trying to get OpenSSH to work on Solaris 10 wich Sun C 5.8 compiler (SUNWspro 11). I've compiled OpenSSL 0.9.8a without problem and OpenSSH 4.3p2 as well. [user at compilationserver ~/openssh-4.3p2] ./ssh -V OpenSSH_4.3p2, OpenSSL 0.9.8a 11 Oct 2005 My problem is that I cannot connect to anything. When I try I always get an error [user at compilationserver ~/openssh-4.3p2]

Hello i'm experimenting a bit with asterisk to see if i can get it to work they way i want it to. i'm no asterisk expert and i've run into a bit of a problem that i can't figure out what is wrong. what i'm trying to do is to use my mobile phones built in sip client (a samsung s4 phone) to connect to asterisk. this is directly over wifi to the asterisk box so there is no