similar to: Bad Magic Internal Error

The Asterisk Development Team would like to announce the release of Asterisk 16.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.0.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

This is a great news, thank you. I have open the issue, https://issues.asterisk.org/jira/browse/ASTERISK-26177 and added the relevant files, let me know if you need more info. Leandro 2016-07-06 21:46 GMT+02:00 Joshua Colp <jcolp at digium.com>: > Leandro Dardini wrote: > >> Hello, >> I'd like to know if anyone of you is finding my same problems using any >>

On Fri, Apr 1, 2016 at 3:22 PM, George Joseph <george.joseph at fairview5.com> wrote: > > > On Fri, Apr 1, 2016 at 3:15 PM, Harley Peters <harley at thepetersclan.com> > wrote: > >> On 04/01/2016 04:06 PM, Joshua Colp wrote: >> >>> Harley Peters wrote: >>> >>>> I get the following error when trying to update date the database

06.10.2015 1:22, Joshua Colp ?????: > On 15-10-05 05:58 PM, Dmitriy Serov wrote: >> 05.10.2015 23:24, Joshua Colp ?????: >>> On 15-10-05 05:22 PM, Dmitriy Serov wrote: >>>> Hello. Do I understand correctly that the current implementation >>>> res_pjsip does not support ZRTP? >>>>

Marcelo Terres wrote: > I noticed another different behaviour. > > In older versions, when I call rasterisk, I receive some informations > about it. Fox example: > > [root at pbx2 ~]# rasterisk > Asterisk 11.22.0, Copyright (C) 1999 - 2013 Digium, Inc. and others. > Created by Mark Spencer<markster at digium.com> > Asterisk comes with ABSOLUTELY NO WARRANTY; type

On Tue, Feb 14, 2017, at 09:57 AM, Marcelo Terres wrote: > Same problem with me. > > I downloaded the file in 2 different places and had the same error... An issue was filed for tracking this[1] and it will be resolved later today. [1] https://issues.asterisk.org/jira/browse/ASTERISK-26791 -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL

Hi, I am using OpenSuse 12.2 64bit OS which uses Kernel 3.3.x version and downloaded Asterisk 1.8 current version, after installing Asterisk, while starting Asterisk it hangs at the stage of loading modules.conf, I checked the forum https://issues.asterisk.org/jira/browse/ASTERISK-19245 but still after updating through yast also i am facing the issue. Have anybody faced this type of issue with

Outoing calls I make using Motif Google Voice Calls continue ringing even after the other end picks up. I have to restart Asterisk to resolve the issue. I don't see any errors. It's not recognizing that the other party picked up the phone and restarting Asterisk fixes it only for a day. -- Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336

Hello, we just installed the latest 13 and 16 version of asterisk and face problem on incoming calls: they are ended like in Asterisk 16 [2020-02-04 19:19:48] ERROR[3768][C-00000001]: stasis_bridges.c:199 bridge_topics_init: Bridge id initialization required [2020-02-04 19:19:48] WARNING[3768][C-00000001]: bridge.c:809 bridge_base_init: Bridge da3bd3d1-cdea-4a05-8b3d-0ded8c561c5f: Could not

Hi, how do I make a bug report? I filled in the form to make a report and https://issues.asterisk.org/jira/issues/?filter=-2 still shows no issues reported by me. If someone knows how to get asterisk to re-register when using pjsip after the registration shows as Rejected, like after the internet connection to the VOIP provider goes away (and comes back), please let me know. This bug makes

Hi first post, so hope I'm not violating protocol. Been using Asterisk as home phone and hobby use for nearly 10 years. I love this project. Anyway, would someone mind verifying my pjsip.conf ? This seems to work well for 14.3.1 but I get no rtp into my natted Linphone when I upgrade to 14.4.1. Other than that the phone registers properly on 14.4.1. I can provide a pjsip log as well,

05.10.2015 23:24, Joshua Colp ?????: > On 15-10-05 05:22 PM, Dmitriy Serov wrote: >> Hello. Do I understand correctly that the current implementation >> res_pjsip does not support ZRTP? >> http://lists.digium.com/pipermail/asterisk-dev/2013-December/064401.html > > ZRTP is not supported in Asterisk itself. > >> Nothing has changed since 2013? P.S. I greatly

Hello, On a newly built Asterisk 13.13.1 system, I can't make HEP work with chan_sip (though I could make it work with PJSIP on another instance). Looking either at [1] or hep.conf, I can't see anything meaning HEP requires PJSIP. Before diging deeper, may I simply ask if HEP requires PJSIP or not ? What about a line mentioning the answer in above documents (to keep other from wondering

On 05/12/2017 at 08:49 PM, Joshua Colp wrote: > On Fri, May 12, 2017, at 02:46 PM, Michael Maier wrote: > > <snip> > >> >> If I'm doing exactly the same call originated with another extension, >> there can't be seen these frequent changes. But the strange thing is, >> that in both cases the part between extension and asterisk doesn't show

Dne 11.8.2015 v 12:18 Joshua Colp napsal(a): > Vinicius Fontes wrote: >> I'm having the same issue! The difference in my case is Asterisk server >> has a public IPv4 and the browser is behind a single NAT. >> >> I'm forwarding my configuration below (which I posted previously on >> asterisk-users). >> >> How can we debug ICE negotiation? >

examples of "interesting" information like ICE result and howto make "minimal" configuration of pjproject.conf i.e. for  debugging app_queue.so core set debug 5 app_queue.so for debugging RTP core set debug 10 rtp_engine core set debug 10 res_rtp_asterisk rtp set debug on logger.conf rtp => debug,verbose(5) so i mean in

Yes, this means the remote end was not sending any audio stream. But it shouldn't. According to [1] before remote end send OK or ACK there is one way SDP, no any audio stream. PJSIP channel (option rtp_timeout) does not take this one. Isn't it a mistake? What could be workarounds? 19.12.2016 11:33, Jean Aunis ?????: > > This means the remote end was not sending any audio stream,

Hi everyone, I have just upgraded to Asterisk 13.7.0-rc2 and noticed that when I type "pjsip show endpoints" at the CLI, I get "No Objects Found". However, if I request information on a specific endpoint, (for example: "pjsip show endpoint 101") then I get all of the information for that endpoint as expected. This seems to have started as soon as I upgraded to

Hi, I want to pass a part of a SIP INVITE multipart body. I found a quite old patch here: https://issues.asterisk.org/jira/browse/ASTERISK-14510?jql=text%20~%20%22body%20part%22 But this patch is for the SIP channel driver not PJSIP, right? Is it even possible without a patch? What do I have to put in the dialplan then? Thanks in advance, Simon

Hello, Does anyone know whether chan_sip in Asterisk supports DTMF in clock rates other than 8000? I looked for telephone-event/16000 in the changelog and in Jira but no luck. Any help would be appreciated. -- Best regards, Vlasis Chatzistavrou.