below is the transaction of the failed mediaproxy invite. I allready could tell that replies go through openser, but I don't see the reason why ser doesn't see them as replies (and use the mediaproxy function).
as you can see, the invite from <ip client> to <ip asterisk> (through <ip OPENSER>, which is also ip of mediaproxy) goes in one direction good (the ip in the SDP is changed from <ip client> to <ip openser>, but the return path en the OK (with it's SDP) is not changed
I did a tcpdump with a call between 2 clients, where the proxy works, and the only difference I see is that in the reply of asterisk, there is no rinstance field in the contact header
thanks arne
U <ip client>:5060 -> <ip OPENSER>:5060 INVITE sip:701@<sip domain>;transport=UDP SIP/2.0..From: "arne" <sip:1002@<sip domain>>;tag=514a90c3-13c4-7a70a-1de331c0-5e4f..To: "701"< sip:701@sipgat e.evonet.be>..Call-ID: 1064dc44-514a90c3-13c4-7a70a-1de331be-529@<ipclient>..CSeq: 1 INVITE..Via: SIP/2.0/UDP <ip client>:5060;rport;branch=z9hG4bK-7a70a-1d e331c2-69dc..Max-Forwards: 70..Supported: replaces,100rel,timer..Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, OPTIONS, INFO, PRACK..User-Agent: Swissvoice IP1 0 SP v1.0.1 (Build 3) 3.0.5.1..Allow-Events: talk, hold, conference..Contact: "arne" <sip:1002@<ip client>:5060;transport=UDP>..Session-Expires: 1800..Content- Type: application/sdp..Content-Length: 246....v=0..o=rtp/1 501514 501514 IN IP4 <ip client>..s=-..c=IN IP4 <ip client>..t=0 0..m=audio 50000 RTP/AVP 18 0 8.. a=fmtp:18 annexb=yes..a=ptime:40..a=SilenceSupp:on..a=rtpmap:18 g729/8000..a=rtpmap:0 pcmu/8000..a=rtpmap:8 pcma/8000..a=sendrecv.. #
U <ip OPENSER>:5060 -> <ip asterisk>:5060 INVITE sip:701@<sip domain>;transport=UDP SIP/2.0..Record-Route: <sip:<ip OPENSER>;lr=on;ftag=514a90c3-13c4-7a70a-1de331c0-5e4f>..From: "arne" < sip:1002@si pgate.evonet.be>;tag=514a90c3-13c4-7a70a-1de331c0-5e4f..To: "701"<sip:701@<sip domain>>..Call-ID: 1064dc44-514a90c3-13c4-7a70a-1de331be-529@<ip client>..C Seq: 1 INVITE..Via: SIP/2.0/UDP <ip OPENSER>;branch=0..Via: SIP/2.0/UDP <ip client>:5060;rport=5060;branch=z9hG4bK-7a70a-1de331c2-69dc..Max-Forwards: 69..Supp orted: replaces,100rel,timer..Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, OPTIONS, INFO, PRACK..User-Agent: Swissvoice IP10 SP v1.0.1 (Build 3) 3.0.5.1..Allo w-Events: talk, hold, conference..Contact: "arne" <sip:1002@<ip client>:5060;transport=UDP>..Session-Expires: 1800..Content-Type: application/sdp..Content-Leng th: 246....v=0..o=rtp/1 501514 501514 IN IP4 <ip client>..s=-..c=IN IP4 <ip OPENSER>..t=0 0..m=audio 60106 RTP/AVP 18 0 8..a=fmtp:18 annexb=yes..a=ptime:40..a =SilenceSupp:on..a=rtpmap:18 g729/8000..a=rtpmap:0 pcmu/8000..a=rtpmap:8 pcma/8000..a=sendrecv.. #
U <ip asterisk>:5060 -> <ip OPENSER>:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP <ip OPENSER>;branch=0;received=<ip OPENSER>..Via: SIP/2.0/UDP <ip client>:5060;rport=5060;branch=z9hG4bK-7a70a-1de331c2- 69dc..From: "arne" <sip:1002@<sip domain>>;tag=514a90c3-13c4-7a70a-1de331c0-5e4f..To: "701"<sip:701@<sip domain>>..Call-ID: 1064dc44-514a90c3-13c4-7a70 a-1de331be-529@<ip client>..CSeq: 1 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Contact: <sip:701@ <ip asterisk>>..Content-Length: 0.... #
U <ip OPENSER>:5060 -> <ip client>:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP <ip client>:5060;rport=5060;branch=z9hG4bK-7a70a-1de331c2-69dc..From: "arne" <sip:1002@<sip domain>>;tag=514a90c3-13c 4-7a70a-1de331c0-5e4f..To: "701"<sip:701@<sip domain>>..Call-ID: 1064dc44-514a90c3-13c4-7a70a-1de331be-529@<ip client>..CSeq: 1 INVITE..User-Agent: Asteri sk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Contact: <sip:701@<ip asterisk>>..Content-Length: 0.... #
U <ip asterisk>:5060 -> <ip OPENSER>:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP <ip OPENSER>;branch=0;received=<ip OPENSER>..Via: SIP/2.0/UDP <ip client>:5060;rport=5060;branch=z9hG4bK-7a70a-1de331c2-69dc ..Record-Route: <sip:<ip OPENSER>;lr=on;ftag=514a90c3-13c4-7a70a-1de331c0-5e4f>..From: "arne" <sip:1002@<sip domain>>;tag=514a90c3-13c4-7a70a-1de331c0-5e4f ..To: "701"<sip:701@<sip domain>>;tag=as60ebd3fc..Call-ID: 1064dc44-514a90c3-13c4-7a70a-1de331be-529@<ip client>..CSeq: 1 INVITE..User-Agent: Asterisk PBX ..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Contact: <sip:701@<ip asterisk>>..Content-Type: application/sdp..Content-Length: 188....v= 0..o=root 26276 26276 IN IP4 <ip asterisk>..s=session..c=IN IP4 <ip asterisk>..t=0 0..m=audio 13434 RTP/AVP 0 8..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a= silenceSupp:off - - - -.. #
U <ip OPENSER>:5060 -> <ip client>:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP <ip client>:5060;rport=5060;branch=z9hG4bK-7a70a-1de331c2-69dc..Record-Route: <sip:<ip OPENSER>;lr=on;ftag=514a90c3-13c4-7a70 a-1de331c0-5e4f>..From: "arne" <sip:1002@<sip domain>>;tag=514a90c3-13c4-7a70a-1de331c0-5e4f..To: "701"<sip:701@<sip domain>>;tag=as60ebd3fc..Call-ID: 1064dc44-514a90c3-13c4-7a70a-1de331be-529@<ip client>..CSeq: 1 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NO TIFY..Contact: <sip:701@<ip asterisk>>..Content-Type: application/sdp..Content-Length: 188....v=0..o=root 26276 26276 IN IP4 <ip asterisk>..s=session..c=IN IP4 <ip asterisk>..t=0 0..m=audio 13434 RTP/AVP 0 8..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=silenceSupp:off - - - -.. #
2006/9/21, Daniel-Constantin Mierla daniel@voice-system.ro:
Hello,
watch the network traffic with ngrep on your sip server. You can see the call flow which may help to identify the issue. You can paste it to the list and someone may give you hints.
Cheers, Daniel
On 09/21/06 12:28, Arne Van Theemsche wrote:
hi
my users subscribe with openser, en asterisk is used as connectivity to pstn
i am now installing a mediaproxy, for all users, so every call goes via a mediaproxy.
I'm doing this as follows (relevant statements only)
in route
#I installed the t_on_reply here to be sure that every replygets parsed, but normally in the INVITE section should be enough? t_on_reply("1");
if (method==INVITE) { use_media_proxy(); }onreply_route[1] { log(-3,"reply received"); if (!search("^Content-Length:[ ]*0")) { log(-3,"using mediaproxy"); use_media_proxy(); }; }
the weird is, for all local users, this works fine, but as soon as asterisk is involved, the reply doesn't get triggered (not seeing the "reply received" either, only when disconnecting the call). The call get's established fine, asterisk is sending media to the mediaproxy, but the SDP towards the calling phone is not modified (since the onreply isn't triggered)
am I missing something here?
thanks Arne
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