[Sems] DSM and b2bua without media

Saúl Ibarra saghul at gmail.com
Thu Jun 11 23:53:06 CEST 2009


Hi Stefan,

Thank you very much for your quick response :) I tried again with
r1434 but I'm getting some extrange error. By looking at SIP captures,
I see the INVITE is generated withoud SDP...

#
U +5.653972 192.168.1.112:5060 -> 192.168.1.115:5080
INVITE sip:1234 at sipproxy.saghul.lan:5080 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK55596cc3;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 192.168.1.112>;tag=as51b25bdf
To: <sip:1234 at sipproxy.saghul.lan:5080>
Contact: <sip:asterisk at 192.168.1.112>
Call-ID: 1ec73c6937032f0f273a28092c697e5f at 192.168.1.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.9
Date: Thu, 11 Jun 2009 22:24:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 208

v=0
o=root 1124002214 1124002214 IN IP4 192.168.1.112
s=Asterisk PBX 1.6.0.9
c=IN IP4 192.168.1.112
t=0 0
m=audio 8464 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

#
U +0.018650 192.168.1.115:5080 -> 192.168.1.112:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK55596cc3;rport
From: "asterisk" <sip:asterisk at 192.168.1.112>;tag=as51b25bdf
To: <sip:1234 at sipproxy.saghul.lan:5080>;tag=361D04EE-4A31842D000ED85A-B770FB90
Call-ID: 1ec73c6937032f0f273a28092c697e5f at 192.168.1.112
CSeq: 102 INVITE
Server: Sip Express Media Server (1.1.0-dev-r1434M (i386/linux))
Contact: <sip:1234 at 192.168.1.115:5080>
Content-Length: 0


#
U +0.000656 192.168.1.115:5080 -> 192.168.1.112:5060
SIP/2.0 488 could not find compatible payload
Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK55596cc3;rport
From: "asterisk" <sip:asterisk at 192.168.1.112>;tag=as51b25bdf
To: <sip:1234 at sipproxy.saghul.lan:5080>;tag=67D6E165-4A31842D000EDBFF-B75E8B90
Call-ID: 1ec73c6937032f0f273a28092c697e5f at 192.168.1.112
CSeq: 102 INVITE
Server: Sip Express Media Server (1.1.0-dev-r1434M (i386/linux))
Content-Length: 0


#
U +0.082672 192.168.1.115:5080 -> 213.192.59.75:5060
INVITE sip:conference at iptel.org SIP/2.0
Via: SIP/2.0/UDP 192.168.1.115:5080;branch=z9hG4bKUWp6haKu
From: "asterisk"
<sip:asterisk at 192.168.1.112>;tag=1FF17244-4A31842D000EDA14-B75E8B90
To: sip:conference at iptel.org
CSeq: 10 INVITE
Call-ID: 6DC74BE5-4A31842D000EDA19-B75E8B90 at 192.168.1.115
Contact: <sip:sems at 192.168.1.115:5080>
Content-Length: 0


#
U +0.081238 213.192.59.75:5060 -> 192.168.1.115:5080
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP
192.168.1.115:5080;branch=z9hG4bKUWp6haKu;rport=5080;received=85.84.127.147
From: "asterisk"
<sip:asterisk at 192.168.1.112>;tag=1FF17244-4A31842D000EDA14-B75E8B90
To: sip:conference at iptel.org
CSeq: 10 INVITE
Call-ID: 6DC74BE5-4A31842D000EDA19-B75E8B90 at 192.168.1.115
Server: Sip EXpress router (2.1.0-dev23-make (i386/linux))
Content-Length: 0
Warning: 392 213.192.59.75:5060 "Noisy feedback tells:  pid=2613
req_src_ip=85.84.127.147 req_src_port=5080
in_uri=sip:conference at iptel.org
out_uri=sip:conference at 213.192.59.78:5074 via_cnt==1"


I'm also a little confused about how the dsm is executed... can you
explain me with more detail when on invite and when on session start
are lanuched please?

Thanks in advance!


Regards,


PD: Find attached the dsm file. I made changes according to your
suggestions, buy I may be missing something... Thanks again! :)

-- 
Saúl -- "Nunca subestimes el ancho de banda de un camión lleno de disketes."
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http://www.saghul.net/
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