Good day.
I have a problem when using android native sip client.
When dialplan used Progress (sending 183 Session Progress) after some
seconds android native sip client declines a call (the logs are at the
end of).
No ealry media be heard.
In same call using Ringing (180) is everything ok.
In same call using Progress and other SIP clients (any) is everything
ok. Early media exists.
Is it possible to make native SIP client works correctly?
Should I send 180 and 183 at the same time? In what sequence?
Standards and many discussions on the Internet claim that should be sent
just one of them.
Thanks.
Dmitriy Serov.
[2016-01-31 14:44:56] VERBOSE[1950] res_pjsip_logger.c: <---
Transmitting SIP response (874 bytes) to UDP:109.60.222.xxx:49912 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP
109.60.222.xxx:49912;rport=49912;received=109.60.222.xxx;branch=z9hG4bKf6c1211a627c45cf6e255cffc3e9c9203738
Call-ID: 46c41cb03be20add7f1b3e3c5423ba30@192.168.1.98
From: "16006" <sip:16...@talk37.ru>;tag=432079590
To: <sip:num...@talk37.ru>;tag=f1a0e09e-f94b-4aca-84c6-d3d9af678852
CSeq: 6847 INVITE
Server: ruVoIP.net PBX
Contact: <sip:85.142.148.xxx:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL,
UPDATE, PRACK, MESSAGE, REFER, REGISTER
Content-Type: application/sdp
Content-Length: 292
v=0
o=- 2541746601 2541746605 IN IP4 85.142.148.xxx
s=ruVoIP.net PBX
c=IN IP4 85.142.148.xxx
t=0 0
m=audio 25616 RTP/AVP 8 0 3 127
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:127 telephone-event/8000
a=fmtp:127 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
[2016-01-31 14:45:07] VERBOSE[1950] res_pjsip_logger.c: <---
Transmitting SIP response (527 bytes) to UDP:109.60.222.xxx:49912 --->
SIP/2.0 603 Decline
Via: SIP/2.0/UDP
109.60.222.xxx:49912;rport=49912;received=109.60.222.xxx;branch=z9hG4bKf6c1211a627c45cf6e255cffc3e9c9203738
Call-ID: 46c41cb03be20add7f1b3e3c5423ba30@192.168.1.98
From: "16006" <sip:16...@talk37.ru>;tag=432079590
To: <sip:num...@talk37.ru>;tag=f1a0e09e-f94b-4aca-84c6-d3d9af678852
CSeq: 6847 INVITE
Server: ruVoIP.net PBX
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL,
UPDATE, PRACK, MESSAGE, REFER, REGISTER
Reason: Q.850;cause=0
Content-Length: 0
--
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