2006/9/22, Daniel-Constantin Mierla <daniel@voice-system.ro >:

Do you get "using mediaproxy" message in the logs? If not, that the
search() matches, I cannot sot right now what is wrong with the
expression. But you can move t_on_reply("1") into if*method=="INVITE")
statement and replace the search condition with if (status =~
"(183)|(2[0-9][0-9])").

See:
http://voip-info.org/wiki/view/OpenSER+And+Mediaproxy

Cheers,
Daniel

On 09/21/06 21:46, Arne Van Theemsche wrote:
> 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 < http://e.evonet.be>>..Call-ID:
> 1064dc44-514a90c3-13c4-7a70a-1de331be-529@<ip
> <mailto: 1064dc44-514a90c3-13c4-7a70a-1de331be-529@%3Cip>
> client>..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
> < http://pgate.evonet.be>>;tag=514a90c3-13c4-7a70a-1de331c0-5e4f..To:
> "701"<sip:701@<sip domain>>..Call-ID:
> 1064dc44-514a90c3-13c4-7a70a-1de331be-529@<ip
> <mailto:1064dc44-514a90c3-13c4-7a70a-1de331be-529@ %3Cip> 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 <mailto:a-1de331be-529@ %3Cip> 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
> <mailto:1064dc44-514a90c3-13c4-7a70a-1de331be-529@ %3Cip>
> 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
> <mailto:1064dc44-514a90c3-13c4-7a70a-1de331be-529@ %3Cip>
> 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
> <mailto: 1064dc44-514a90c3-13c4-7a70a-1de331be-529@%3Cip>
> 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
> <mailto: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 reply
>     > gets 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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