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@IP_PROVIDER_ASTERISK>;tag=as4a6f8fc6
To:
<sip:URI_TEST@IP_SER>;tag=as272dc8d3
Call-ID:
5079515a2060a26155be2cbf6b73dc78@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@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@IP_PROVIDER_ASTERISK>;tag=as4a6f8fc6
To:
<sip:URI_TEST@IP_SER>;tag=as272dc8d3
Call-ID:
5079515a2060a26155be2cbf6b73dc78@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@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@IP_ASTERISK>;tag=as1551f6d3
t:
<sip:URI_TEST@IP_SER_TO_CLIENTS>;tag=c0a80101-41e5ef
i:
1593ec3c0a617037319349245be7c5ab@IP_ASTERISK
CSeq:
102 INVITE
Require:
timer
x:
100;refresher=uac
Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
m:
<sip:URI_TEST@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@IP_ASTERISK>;tag=as1551f6d3
t:
<sip:URI_TEST@IP_SER_TO_CLIENTS>;tag=c0a80101-41e5ef
i:
1593ec3c0a617037319349245be7c5ab@IP_ASTERISK
CSeq:
102 INVITE
Require:
timer
x:
100;refresher=uac
Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
m:
<sip:URI_TEST@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@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@IP_ASTERISK>;tag=as1551f6d3
To:
<sip:URI_TEST@IP_SER_TO_CLIENTS>;tag=c0a80101-41e5ef
Contact:
<sip:170725014@IP_ASTERISK>
Call-ID:
1593ec3c0a617037319349245be7c5ab@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@IP_PROVIDER_ASTERISK>;tag=as4a6f8fc6
To:
<sip:URI_TEST@IP_SER>;tag=as272dc8d3
Call-ID:
5079515a2060a26155be2cbf6b73dc78@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@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@IP_PROVIDER_ASTERISK>;tag=as4a6f8fc6
To:
<sip:URI_TEST@IP_SER>;tag=as272dc8d3
Call-ID:
5079515a2060a26155be2cbf6b73dc78@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@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@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@IP_ASTERISK>;tag=as1551f6d3
To:
<sip:URI_TEST@IP_SER_TO_CLIENTS>;tag=c0a80101-41e5ef
Contact:
<sip:170725014@IP_ASTERISK>
Call-ID:
1593ec3c0a617037319349245be7c5ab@IP_ASTERISK
CSeq:
102 ACK
User-Agent:
Asterisk PBX 1.6.0.1
Content-Length:
0
ACK
sip:URI_TEST@IP_ASTERISK SIP/2.0
Via:
SIP/2.0/UDP IP_PROVIDER_ASTERISK:5060;branch=z9hG4bK7a6d7cf4;rport
From:
"francois berganz"
<sip:170725014@IP_PROVIDER_ASTERISK>;tag=as4a6f8fc6
To:
<sip:URI_TEST@IP_SER>;tag=as272dc8d3
Contact:
<sip:170725014@IP_PROVIDER_ASTERISK>
Call-ID:
5079515a2060a26155be2cbf6b73dc78@IP_PROVIDER_ASTERISK
CSeq:
102 ACK
User-Agent: MARS
Max-Forwards: 70
Content-Length:
0
Cordialement,
BERGANZ François
P Pensez
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