Aterisk stops working suddenly
https://wiki.asterisk.org/wiki/display/ ... +BacktraceStatistics : Posted by marcosfig • on Sun Aug 23, 2015 9:52 pm • Replies 1 • Views 62
View ArticleAterisk stops working suddenly
During the execution of the dialplan, the arterisk for work giving error in the kernelIt does not happen all the time. It is a while and suddenly it stops working.Use the asterisk 11:19, 64 bits....
View ArticleThe Asterisk 13.4.0 'Failed to set remote offer/answer' sdp
Sip/userX-0000001 has reported that a specific device has redirected the call to somewhere else. Asterisk is forwarding that information to SIP/userY-0000001 so that it also knows who redirected the...
View ArticleThe Asterisk 13.4.0 'Failed to set remote offer/answer' sdp
Ok I have eventually found out the following in asterisk.conf :Asterisk was pointing to my keys directory inside /var/lib/asterisk instead of /etc/asterisk/.putting it back to that default...
View ArticleThe Asterisk 13.4.0 'Failed to set remote offer/answer' sdp
As a new-commer to the Asterisak 13.4.0 pbx,I have been working on a worst-so-far WEB RTC issue for too much longer than I'd actually expected; I am using WebRTComm for my sip client stack and chrome...
View ArticleThe Asterisk 13.4.0 'Failed to set remote offer/answer' sdp
Hi,Can someone be generous enough to help me with this please.It is urgent !!!My latest bits include activating and configuring pjsip but still.even with directmedia=yes will not let asterisk generate...
View Article*13.4-SIP- no early media even after 183 Session Progress
Hello,RTP does not flows-out after I get Progress on the channel, but as soon as the channel is answered RTP starts flowing-out.I have "Asterisk 13.4.0" and I am trying to playback a prompt without...
View ArticleMultiple trunk dialing?
The definition of trunk pretty much means that it can support simultaneous calls.One call at a time SIP services are intended for people with a simple SIP phone, not for use with PABXes.Examples of...
View ArticleMultiple trunk dialing?
Dial.http://forums.asterisk.org/line1Dial.http://forums.asterisk.org/line2CongestionIdeally check the hangup cause to avoid calling people who don't answer twice, etc. Congestion is only needed if you...
View ArticleMultiple trunk dialing?
We recently switched from Vonage to running our own Asterisk server with one trunk. This worked great after a learning curve for myself (I had to set it up), and we have multiple extensions and such...
View ArticleThe Asterisk 13.4.0 'Failed to set remote offer/answer' sdp
Sorry I don't understand.You mean you don't use WebRTC/browser-to-browser websocket call (via a pbx or sip proxy ) ? what do you mean by 'locked into a pre-ICE version'. Enlighten me please and where...
View ArticleThe Asterisk 13.4.0 'Failed to set remote offer/answer' sdp
I don't use WebRTC and we are locked into a pre-ICE version.Statistics : Posted by kader2gx • on Fri Aug 21, 2015 10:09 am • Replies 6 • Views 350
View ArticleThe Asterisk 13.4.0 'Failed to set remote offer/answer' sdp
Hi david,Thank you indeed for coming in !That msg is no longer coming up as the call can be picked up but with a second related issue as I explained above;endless ICE messages trying to connect to an...
View ArticleSIP Trunk - problem to connect
Hello! Thnxs for reading!I've an IPLAN virtual PBX, that allows me to connect via zoiper or gigaset, for instance (and it works!)Connection parameters are:Authentication Name: Número...
View ArticleSIP Options replaces replace timer
At a guess, by changing the peer (not the Asterisk description of the peer).Statistics : Posted by BurhanKhan • on Wed Aug 26, 2015 12:59 pm • Replies 1 • Views 51
View ArticleSIP Options replaces replace timer
HiI was facing issues in inbound dialing. I observed trunk details by using command'sip show peer 123'and observed SIP Options : replaces replace timer after restarting asterisk issue has resolved. and...
View ArticleHow to enable jitter buffer in PJSIP
There are no equivalent configuration options. Instead, you should use the JITTERBUFFER function.https://wiki.asterisk.org/wiki/display/ ... TTERBUFFERStatistics : Posted by fchin • on Tue Apr 21, 2015...
View ArticleHow to enable jitter buffer in PJSIP
Hi,There used to be a few variables in sip.conf for configuring jitter buffer, e.g. jbenable, jbmaxsize and etc. I have now switched to using pjsip. So I'm wondering where to find the equivalent...
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*13.4-SIP- no early media even after 183 Session Progress
edited the posts as guided, thank you for the education ... appreciated (Y)re-posting in support forumStatistics : Posted by usmanbaiga • on Mon Aug 24, 2015 3:31 pm • Replies 5 • Views 293
View Article*13.4-SIP- no early media even after 183 Session Progress
a is not media direction. It is a general attribute field.Please re-edit so that the log is in a code block; that results in a lot less scrolling.I wonder if this relates to the probation message, i.e....
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