Archives : September-2014
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I created a dummy dialplanwhere I ask the user to enter the age.[macro-age]exten => s,1,Background(my/age);;Play recorded message to enter age exten => s,n,WaitExten(10)exten => _XX,1,Set(AGE=${EXTEN});; this line is not executing, instead dialplan..
I am having the issue described in this question:http://lists.digium.com/pipermail/asterisk-users/2005-May/099075.htmlDoes anybody has an insight? I guess Asterisk is trying to match the combination IP:Port, but in H223 this changes call by call. Th..
I have a multihomed machine. How can I assign multiple IPs to and endpoint, not all of them, just two, for instance, out of many?Suppose the machine as 30 IPs, but my asterisk needs listen on two, and one single endpoint needs to be associated with th..
Den 2014-09-04 18:05, Stephen More skrev:You can try with Wait(2) to wait two seconds before you do Answer(). This will delay the response to the INVITE with two seconds.You dont want to wait to long thought, so maybe you can test your way to someth..
I get tons of these messages chan_sip.c:10088 process_sdp: Declining non-primary audio stream:audio 30660 RTP/AVP 4 101 13What does it mean and does it show a problem like one-way audio?Thanks for ..
Start from http://www.voip-info.org/or Asterisk : The Future of Telephony BookFrom: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Horace Miles Sent: Friday, September 05, 2014 12:19 AMTo: asterisk-users@lists.digium…
Try starting Asterisk with the -f option.It will NOT fork into the background so you will see all messages on startup (including any that might not end up in the log file).Search for DAHDI errors which will likely be there.Also, if you configure everyth..
Its my first post here, so Ill cut to the chaseI have 2 Asterisk servers and want to connect them using sip on one and pjsip on the other one. One is running at home and another at a VPS. The first one will be the client (with dynamic ip) and the ..
We cant do much with part of your debug. Youll want to post a pastebin link to your full SIP trace, and be sure that it includes at least VERBOSE messages turned up to 5.[1]Work on WebRTC support is on-going, so youll want to test in the very lat..
From the reading and testing I have done it doesnt look like SIP supports a username and password in the Dial string. I currently have the following mapping.priv => dundi-extens,0,SIP, dundi:pass@1.1.1.1/${NUMBER},nounsolicited,nocomunsolicit,nopartia..