Tag : sip
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I just started with setting up a new asterisk system, that will operate on a sip trunk, but I wonder, how to transfer the calls to different extensions, because all calls appear as being send to the base number of the trunk.E.g. given the trunk ra..
I want to capture all SIP messages.I have about 30 hosts in about 6 colos.My first thought was dumpcap, but the output file name format bugs me.What do you use for long term SI..
Ive noticed that when I set a phone on DND (phone-side DND, meaning it rejects calls with a busy status, SIP 486 response code I believe) the queue keeps on trying the phone over and over again. This creates issues in terms of CDR entries – in a scena..
The channel motif and res_xmpp do not work. But there is one company that does make it work and charges $US 6 for a lifetime connection to your own free Google Voice number, from SIP. I wonder if anybody would be able to fix Asterisks libraries so peo..
if a PJSIP call fails, how can I capture SIP code, like 503,603 etc?in old SIP channel, we had ${HASH(SIP_CAUSE,)}but in PJSIP it has to be the outbound channel, which is gone when the control returns to the callin..
I have a host 192.168.1.3 that wants to run SIP on 5068 (long story). My host is 192.168.10.201. My host needs to stay on 5060 because of all the other devices I have connected.I tried putting portP68 in my SIP extension definition but that did not work..
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%22But this patch is for the SIP channel driver not PJSIP, right?Is it even possi..
Iam loocking for an programm to check the SIP port of an Asterisk asterisk.Ome time ago I have used #/usr/bin/sipsak but it seemed that it is not working anymore?Any ideas what I can use instead?best rega..
I finally secure SIP session between Asterisk server and a remote client. My questions is the following; do I need to open port 5061 UDP on my firewall or just port 5061 TCP for SIP sessions.? I am not interested in securing RTP only SIP sessions.Tha..
I have an Asterisk 13.8.2, which is supposed to be only a client to an encrypted SIP service. All local phones are connected via UDP.Since I cant use PJSIP (see my mailing list post from yesterday), Itried configuring chan_sip to work that way. My setti..