search for: moh_class

Hello, I have a problem with the following: When I dial a PSTN number from a UAC, the call is made through a SIP Trunk (which has a connection to the PSTN) in Asterisk. The PSTN Gateway returns a 180 Ringing WITH SDP, but Asterisk forwards the 100 Ringing WITHOUT SDP: As you can see below, the SIP message from 10.254.254.1 (the PSTN Gateway) has SDP, while * (with 192.168.0.173) removes the SDP

Hi there, I need some help with a 'custom' feature. I have following feature defined in features.conf : [applicationmap] opnemencallee => #3,self/callee,Monitor,wav,/var/samba/profiles/jonaskl/recording,m In my dialplan : [from-HostAst] exten => s,1,Set(__DYNAMIC_FEATURES=opnemencallee) exten => s,n,Dial(SIP/grandstream,30) I want the callee to be able to press #3 to be able

...; Attended transfer ;parkcall => #72 ; Park call (one step parking) ; Set(__DYNAMIC_FEATURES=myfeature1#myfeature2#myfeature3) ;<FeatureName> => <DTMF_sequence>,<ActivateOn>[/<ActivatedBy>],<Application>[,<AppArguments>[, MOH_Class]] testfeature => #9,peer,Playback,tt-monkeys ;Allow both the caller and callee to play pauseMonitor => #1,self/callee,Pausemonitor ;Allow the callee to pause monitoring unpauseMonitor => #3,self/callee,UnPauseMonitor ;Allow the callee to unpause monitoring -----------...

...SIP message from 10.254.254.1 (the PSTN > Gateway) has SDP, while * (with 192.168.0.173) removes the SDP content. > > How can this be solved? > Well, I am not that expert but AFAIK your PSTN gateway should send a 183 (Session progress) than a simple 180. Do you use Dial(SIP/blah|30|m(moh_class)) to start early media? Regards, Hauke ------------------------------ _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users End of Asterisk-Users Digest, Vol 14, Issue 161 ***********...