[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.

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20090223/49b3cdd0/attachment.htm>


More information about the sr-users mailing list