Archives : March-2015
I am trying to determine how the transfer button on the Snom-870 works with Asterisk.Is the ## special code employed as when it is entered through the handset or is the blind transfer through the phone function accomplished in a differen..
all,I have Asterisk 13 running and Im currently trying to get PJSIP working on TCP.My transport looks like this. My box is not behind NAT.[transport-tcp]type=transport protocol=tcp bind=0.0.0.0:5061My endpoint looks like this:[user1]type=endpoint transport=transport-..
All;I build a conference server using Asterisk 1.8 and the third party module app_konference.so. I would ask on their forum, but the forum seems to be pretty dead. The problem I am having is that when I have conferences that have a lot of members, ..
Im facing some problems with RTP during queue agent calls.Randomly during the call the agent cant hear the other side. This happens for two or three seconds and the the call continue without problems.The weird thing is that the recording for this c..
For the mailing list archive and for anyone else interested.A few years ago we needed to automatically run a second AGI if the first AGI failed i.e. a failsafe setup.Mainly because Im not a very good programmer. 8-|The code below is very similar to w..
Hello.I am using asterisk and chan_sip a lot of years. And newbie in chan_pjsip. Now i am transfering all from chan_sip to chan_pjsip. And have a lot of questions. First of…system: Asterisk 13.2 on slackware 14.1Errors on outgoing call:[2015-03..
Im having some problems with a vega sangoma, if a call comes into my ivr and hangs up, the call continues to ring and leaves hanging the channel, I have to restart Asterisk and everything works Okmy sangoma is a vega 50 , 4 FXO .I tried different t..
Stuck with TLS transport,I have 2 phones (both in local network for tests)one connected with up second with tlswhen I calling TLS to UPD everything is fine, but when UDP calls TLS I getting an error ERROR[44230]: pjsip:0 : tlsc0x7f143012 TLS connec..
Background: I dabbled with asterisk years ago, and more recently have more-or-less functioning IncrediblePBX systems for experimenting, but I want to understand more so Im now working with distro packages(Ubuntu) and hand edited configurations file..
Id like to dial two extensions (or external number) and ask for confirmation to accept the call.Dialing an extension, asking for confirmation and then dialing a second extension if the call has not been accepted is easy by using the dial option U(….