[Kamailio-Users] problem with 200ok
BERGANZ François
francois at acropolistelecom.net
Mon Feb 23 15:49:52 CET 2009
Hello,
I have Asterisk1---SER---Asterisk2.
When I do INVITE from the left,
--the asterisk2 send 200ok to the SER
--the SER forward to the Asterisk1
--but the asterisk1 directly send the ACK to Asterisk2
Asterisk2 retransmit the 200ok… and error.
I think that it need that the ACK come from the SER and not directly from
the Asterisk1.
So, how can I detect a 200ok and reply a ACK with my SER?
Or, anyone have ever seen that problem?
Thank you
Next: my capture problem
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP IP_SER;branch=z9hG4bK815c.d8703d97.0;received=IP_SER
Via: SIP/2.0/UDP IP_PROVIDER_ASTERISK:5060;branch=z9hG4bK369cf0be;rport=5060
From: "francois berganz" <sip:170725014 at IP_PROVIDER_ASTERISK>;tag=as4a6f8fc6
To: <sip:URI_TEST at IP_SER>;tag=as272dc8d3
Call-ID: 5079515a2060a26155be2cbf6b73dc78 at IP_PROVIDER_ASTERISK
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:URI_TEST at IP_ASTERISK>
Content-Type: application/sdp
Content-Length: 287
v=0
o=root 1097653753 1097653753 IN IP4 IP_ASTERISK
s=Asterisk PBX 1.6.0.1
c=IN IP4 IP_ASTERISK
t=0 0
m=audio 16386 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
6§¢Iœÿ
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP IP_PROVIDER_ASTERISK:5060;branch=z9hG4bK369cf0be;rport=5060
From: "francois berganz" <sip:170725014 at IP_PROVIDER_ASTERISK>;tag=as4a6f8fc6
To: <sip:URI_TEST at IP_SER>;tag=as272dc8d3
Call-ID: 5079515a2060a26155be2cbf6b73dc78 at IP_PROVIDER_ASTERISK
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:URI_TEST at IP_ASTERISK>
Content-Type: application/sdp
Content-Length: 287
v=0
o=root 1097653753 1097653753 IN IP4 IP_ASTERISK
s=Asterisk PBX 1.6.0.1
c=IN IP4 IP_ASTERISK
t=0 0
m=audio 16386 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
SIP/2.0 200 OK
v: SIP/2.0/UDP IP_SER_TO_CLIENTS;branch=z9hG4bK6c33.6ae23971.0
v: SIP/2.0/UDP IP_ASTERISK:5060;branch=z9hG4bK598106c4;rport=5060
f: "francois berganz"<sip:170725014 at IP_ASTERISK>;tag=as1551f6d3
t: <sip:URI_TEST at IP_SER_TO_CLIENTS>;tag=c0a80101-41e5ef
i: 1593ec3c0a617037319349245be7c5ab at IP_ASTERISK
CSeq: 102 INVITE
Require: timer
x: 100;refresher=uac
Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,I
NFO
m: <sip:URI_TEST at 192.168.1.82:5060;user=phone>
Record-Route: <sip:IP_SER_TO_CLIENTS;lr=on;ftag=as1551f6d3>
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
c: application/sdp
l: 206
v=0
o=URI_TEST 4319348 4319348 IN IP4 192.168.1.82
s=-
c=IN IP4 192.168.1.82
t=0 0
m=audio 41000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
SIP/2.0 200 OK
v: SIP/2.0/UDP IP_ASTERISK:5060;branch=z9hG4bK598106c4;rport=5060
f: "francois berganz"<sip:170725014 at IP_ASTERISK>;tag=as1551f6d3
t: <sip:URI_TEST at IP_SER_TO_CLIENTS>;tag=c0a80101-41e5ef
i: 1593ec3c0a617037319349245be7c5ab at IP_ASTERISK
CSeq: 102 INVITE
Require: timer
x: 100;refresher=uac
Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,I
NFO
m: <sip:URI_TEST at IP_PHONE:5060;user=phone>
Record-Route: <sip:IP_SER_TO_CLIENTS;lr=on;ftag=as1551f6d3>
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
c: application/sdp
l: 206
v=0
o=URI_TEST 4319348 4319348 IN IP4 192.168.1.82
s=-
c=IN IP4 192.168.1.82
t=0 0
m=audio 41000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
ACK sip:URI_TEST at IP_PHONE:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP IP_ASTERISK:5060;branch=z9hG4bK0e97e7e8;rport
Route: <sip:IP_SER_TO_CLIENTS;lr=on;ftag=as1551f6d3>
Max-Forwards: 70
From: "francois berganz" <sip:170725014 at IP_ASTERISK>;tag=as1551f6d3
To: <sip:URI_TEST at IP_SER_TO_CLIENTS>;tag=c0a80101-41e5ef
Contact: <sip:170725014 at IP_ASTERISK>
Call-ID: 1593ec3c0a617037319349245be7c5ab at IP_ASTERISK
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0.1
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP_SER;branch=z9hG4bK815c.d8703d97.0;received=IP_SER
Via: SIP/2.0/UDP IP_PROVIDER_ASTERISK:5060;branch=z9hG4bK369cf0be;rport=5060
From: "francois berganz" <sip:170725014 at IP_PROVIDER_ASTERISK>;tag=as4a6f8fc6
To: <sip:URI_TEST at IP_SER>;tag=as272dc8d3
Call-ID: 5079515a2060a26155be2cbf6b73dc78 at IP_PROVIDER_ASTERISK
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:URI_TEST at IP_ASTERISK>
Content-Type: application/sdp
Content-Length: 287
v=0
o=root 1097653753 1097653754 IN IP4 IP_ASTERISK
s=Asterisk PBX 1.6.0.1
c=IN IP4 IP_ASTERISK
t=0 0
m=audio 16386 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP_PROVIDER_ASTERISK:5060;branch=z9hG4bK369cf0be;rport=5060
From: "francois berganz" <sip:170725014 at IP_PROVIDER_ASTERISK>;tag=as4a6f8fc6
To: <sip:URI_TEST at IP_SER>;tag=as272dc8d3
Call-ID: 5079515a2060a26155be2cbf6b73dc78 at IP_PROVIDER_ASTERISK
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:URI_TEST at IP_ASTERISK>
Content-Type: application/sdp
Content-Length: 287
v=0
o=root 1097653753 1097653754 IN IP4 IP_ASTERISK
s=Asterisk PBX 1.6.0.1
c=IN IP4 IP_ASTERISK
t=0 0
m=audio 16386 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
ACK sip:URI_TEST at IP_PHONE:5060;user=phone SIP/2.0
Record-Route: <sip:IP_SER_TO_CLIENTS;lr=on;ftag=as1551f6d3>
Via: SIP/2.0/UDP IP_SER_TO_CLIENTS;branch=z9hG4bK6c33.6ae23971.2
Via: SIP/2.0/UDP IP_ASTERISK:5060;branch=z9hG4bK0e97e7e8;rport=5060
Max-Forwards: 70
From: "francois berganz" <sip:170725014 at IP_ASTERISK>;tag=as1551f6d3
To: <sip:URI_TEST at IP_SER_TO_CLIENTS>;tag=c0a80101-41e5ef
Contact: <sip:170725014 at IP_ASTERISK>
Call-ID: 1593ec3c0a617037319349245be7c5ab at IP_ASTERISK
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0.1
Content-Length: 0
ACK sip:URI_TEST at IP_ASTERISK SIP/2.0
Via: SIP/2.0/UDP IP_PROVIDER_ASTERISK:5060;branch=z9hG4bK7a6d7cf4;rport
From: "francois berganz" <sip:170725014 at IP_PROVIDER_ASTERISK>;tag=as4a6f8fc6
To: <sip:URI_TEST at IP_SER>;tag=as272dc8d3
Contact: <sip:170725014 at IP_PROVIDER_ASTERISK>
Call-ID: 5079515a2060a26155be2cbf6b73dc78 at IP_PROVIDER_ASTERISK
CSeq: 102 ACK
User-Agent: MARS
Max-Forwards: 70
Content-Length: 0
Cordialement,
BERGANZ François
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
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