Open B-R-Bender opened 1 year ago
now I'm thinking I'm doing it wrong (obviously, as it's now working :-) )
as I wan't to be able to place a call to my phone number from asterisk using my sip id/credentials
so there in no such thing
sip:${requstedPhone}@${asteriskHost}
That code looks correct to me at first glance. Are you saying it is working now? If so please close the issue, it not please provide sip traces
Nope it's not working :( I'll get you a sip trace for sure meanwhile I've added some logging to my code, and getting that after I've got dialog in my hand and destroy handler is called
'CSeq: 24347 BYE\r\n' +
'Reason: Q.850;cause=44\r\n' +
Hello guys!
Can some, please, clarify/help with workflow on the following script:
Let's assume I have RTP stream on a local device, and what I'm trying to accomplish is to place a call from asterisk (I've got sip identity with credentials) using drachtio-srf and use that RTP stream as an audio for the initiated call.
I'm assuming I should you UAC from srf instance, and I'm building local SDP for it but the call seems to be dropping my test code (which is not working) is
thank you in advance!