similar to: Asterisk 10.0.0-rc3 Now Available

Hi All, I have configured WebRTC according to the install document. The clients register correctly. I'm use SIPjs. The clients are able to send messages to the server. The SIP debug shows the messages being received. However I'm stumped for directions on how to route the messages between the clients. Asterisk 11.11.0 Here is my client sip config: [1060] type=friend username=1060 ; The

Hi, I only run a home-based asterisk (v1.4.18), and have never patched it, so I'm a unfamiliar with what time frame to expect for patches being implimented. I just downloaded (April 14) svn asterisk V1.6.1 r188415, on a "play" machine and noticed that when I stop asterisk, the astcanary module does not exit - when I restart asterisk, a new copy of astcanary also starts. In browsing

Went from 1.6.0.6 to 1.6.0.8 and resulted in segmentation fault. Reverted to 1.6.0.6 and back to normal. ------------------ Linux asterisk.hulber.com 2.6.18-128.1.1.el5 #1 SMP Mon Jan 26 13:58:24 EST 2009 x86_64 x86_64 x86_64 GNU/Linux Apr 3 11:49:56 asterisk kernel: asterisk[3780]: segfault at 00002ce1ac0537a8 rip 0000003e980715a8 rsp 00007fff5bf00c30 error 4 Apr 3 11:50:00 asterisk

Recently safe_asterisk is failing to pick up ASTSBINDIR. I've never had this problem before and even when I move to back versions I have the issue. I did upgrade safe_asterisk and the init.d scripts a version or so ago but even when I try older ones I still have the problem. When I hard code the location things seem to work. The problem that occurs is: cat:

The Asterisk Development Team is pleased to announce the release of Asterisk 1.6.0.7. Asterisk 1.6.0.7 is available for immediate download at http://downloads.digium.com/pub/asterisk/ This release resolves an issue where IMAP voicemail message retrieval and Message Waiting Indication (MWI) would not work properly with the same mailbox name in multiple voicemail contexts. This release also fixes a

The Asterisk Development Team is pleased to announce the release of Asterisk 1.8.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.8.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in

The Asterisk Development Team is pleased to announce the release of Asterisk 1.8.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.8.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in

Hi iam not able to start asterisk give me following error any help STARTING ASTERISK /usr/sbin/safe_asterisk: line 42: 4633 Illegal instruction (core dumped) ${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS} >&/dev/${TTY} </dev/${TTY} Asterisk ended with exit status 132 Asterisk exited on signal 4. Automatically restarting Asterisk. /usr/sbin/safe_asterisk: line 42: 4637 Illegal

I downloaded asterisk-12.4.0 (asterisk-12-current.tar.gz, dated 10-JUL_2014) from http://downloads.asterisk.org/pub/telephony/asterisk/. The packaged configured OK. However, I'm getting a compile error: use of undeclared identifier 'pvt'. I've found similar reports for other versions (asterisk-11 and dongle; macports; a couple others), but nothing for 12.4. Any suggestions to

A week ago, I tried give realtime priority to asterisk proces using -p switch, asterisk was running inside astcanary, but yestarday asterisk probably starts eating all cpu and lock any access to computer, only ping was possible, so, anybody have experience, that ascanary process does really work to lower process priority in case of overloading? PJ

Thanks Paradise, this seems to have worked a treat!!! I commented out the: exten => 110,hint,SIP/110 lines which were in extensions_additional.conf for each sip extension I had. This seems to have stopped the crashes which were previously 3-5 times a day, now: System uptime: 1 day, 18 hours, 10 minutes, 3 seconds Interestingly it had the knock on effect of fixing another problem I had

Hello there! I'm working on some modifications on Asterisk to adapt it to our needs considering some particular demandings of the infraestructure we want to provide. Two of these modifications are: 1- A proprietary configuration driver that will communicate with a server that will be the source of information for the entire infraestructure; and, 2- A call control application that will be

I have just setup asterisk on a debian sarge box. I am running Asterisk 1.21 with AMP and chan_capi_cm 0.6.1 using a BT Speedway (AVM Fritz) ISDN card, connected to a BT ISDN2e line. Currently we have 6 extensions (SIP) configured all using CounterPath(Xten) eyebeam softphone. After many hours of Googling I have finally got it all setup and working. We can transfer calls internally and make and

Hi to everybody my system is be attack, but I dont know what this means [May 27 15:12:24] WARNING[26018] chan_skinny.c: Partial data received, waiting (76 bytes read of 786) [chan_skinny.c] skinny_session[0][C-00000000] skinny_session: WARNING[May 27 15:52:32] Asterisk 13.8.0 built by root @ asterisk on a x86_64 running Linux on 2016-04-04 19:02:51 UTC [May 27 15:52:32] NOTICE[2306] cdr.c: CDR

i m using asterisk 1.8.3 on a centos 5.5 computer when i try to change my name on the voicemail asterisk dump core here what i got on the console -- User ended message by pressing # -- <SIP/6672-00000002> Playing 'auth-thankyou.alaw' (language 'fr') -- <SIP/6672-00000002> Playing 'vm-review.alaw' (language 'fr') -- Saving message as

On 15-09-28 10:19 AM, Emil Ohlsson wrote: > (Still no not receiving the mail, revisited the settings.) > > OK, so SendText doesn't work with this scenario. But can MessageSend > handle this, and respond even when the transport protocol is TLS? Or > do I need to modify Asterisk to add this support? MessageSend has no concept of TLS, it gets passed to chan_sip which then sends

automon is not working for me with asterisk 1.4.22.1 in extension.conf [globals] DYNAMIC_FEATURES=>automon dial is with "w" feature.conf automon => *1 -- Executing [11 at internal:1] Playback("SIP/218-007556b0", "transfer") in new stack -- <SIP/218-007556b0> Playing 'transfer' (language 'en') -- Executing [11 at internal:2]

Hi list, Can someone please explain how to get one touch recording (automon) to work reliably? I'm using Asterisk 1.4.14 on a Debian etch system. My current configuration includes the following settings: In /etc/asterisk/sip.conf: [2000] ; Siemens Gigaset S675 IP wireless SIP phone. type=friend secret=1234 context=phones-j dtmfmode=rfc2833 qualify=yes

The Asterisk Development Team has announced the release of Asterisk 1.8.14.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.14.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

MessageSend is command for send message, however I don't know what the context for sending message. I create a pjsip with the context 'from-internal' then when i config the extension for context 'from-internal' it works but then the my call dialplan does not work. Because they both sms and call are coming to the same context 'from-internal', as I notice. I wonder how