Tag : sip
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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..
Ive had years of experience using ODBC for CDR, SIP, and extensions with Asterisk.One thing that has been problematic in the past is with documentation as far as database tables changing between versions (even within minor releases, though that was b..
I have trouble establishing a call between between two SIP phones. One sip phone is, with asterisk server, at home behind a firewall. The second sip phone is an iPhone with 3G wireless connection. When I call from the SIP device at home the SIP acco..
If I have Asterisk setup with local SIP phones configured but need to call a SIP phone which is not local but actually on another VLAN, what would the dialplan need to lo..
is there anyway to change Sip headers in local channels?if a user sets forward on their handset, calls coming in to the handset get diversion header added:Diversion: 2..
I am using the XML-browser and Call-Info header features for some SIP phones. SIPAddHeader(Call-Info: …) seems to work only in the outgoing direction. Does somebody know a way to send a Call-Info header to the originating SIP device by using only ..
All.Im running some tests with the latest Asterisk SVN-branch-12-r410493M compiled with fresh github pjsip and srtp 1.4.2 on an i386 CentOS machine (2.6.32-358.18.1.el6.i686). As a client Im using the sipMLP WebRTC javascript softphone running on Chr..
Ive got a small install with Asterisk 11. This box is connected to PSTN through a SIP trunk.I need to add a cellular phone as a remote agent of an existing queue.At the moment, this queue is configured according a ringall strategy and busy agent ..