12 Oct
2010
12 Oct
'10
10:02 a.m.
Hello,
I am currently going through an Inter-Op test with a new carrier and there
have been some issues. I have my Kamailio server integrated with many
tier-1 carriers, so I'm pretty sure I'm doing everything OK. The problem is
in the BYE coming from the ITSP, the URI isn't valid, but I am not sure
based on RFC3261 exactly why it isn't correct. If anyone can take a quick
glance, I would really appreciate it.
Thanks!
Via: SIP/2.0/UDP X.X.X.90;branch=z9hG4bK2aa.c3961912.0
Via: SIP/2.0/UDP
X.X.X.238:5060;received=X.X.X.238;branch=z9hG4bK10281a99;rport=5060
From: ""2026097878""
<sip:2026097878@mydomain.com<sip%3A2026097878@mydomain.com
Contact: <sip:2026097878@X.X.X.238
Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com
CSeq: 102 INVITE
User-Agent: G-Tel v1.0
Max-Forwards: 69
Date: Mon, 11 Oct 2010 17:50:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
P-Charge-Info: ""4019921313""
<sip:4019921313@connectfirst.com<sip%3A4019921313@connectfirst.com
Content-Type: application/sdp
Content-Length: 244
v=0
o=root 10137 10137 IN IP4 X.X.X.238
s=session
c=IN IP4 X.X.X.238
t=0 0
m=audio 11748 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
<<<<<<<<<<<<<<< IN
<<<<<<<<<<<<<<<<<<<<<<<<
<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
SIP/2.0 100 Trying
Via: SIP/2.0/UDP X.X.X.90;branch=z9hG4bK2aa.c3961912.0
Via: SIP/2.0/UDP
X.X.X.238:5060;received=X.X.X.238;branch=z9hG4bK10281a99;rport=5060
Record-Route: <sip:X.X.X.90;lr;ftag=as4c62e0b9
From: ""2026097878""
<sip:2026097878@mydomain.com<sip%3A2026097878@mydomain.com
Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com
CSeq: 102 INVITE
Contact: <sip:16782247008@67.150.160.10 <sip%3A16782247008(a)67.150.160.10> Content-Length: 0
<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
<<<<<<<<<<<<<<< IN
<<<<<<<<<<<<<<<<<<<<<<<<
<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP X.X.X.90;branch=z9hG4bK2aa.c3961912.0
Via: SIP/2.0/UDP
X.X.X.238:5060;received=X.X.X.238;branch=z9hG4bK10281a99;rport=5060
Record-Route: <sip:X.X.X.90;lr;ftag=as4c62e0b9
From: ""2026097878""
<sip:2026097878@mydomain.com<sip%3A2026097878@mydomain.com Content-Type: application/sdp
Content-Length: 109
v=0
o=ATSI-B1 0 0 IN IP4 67.150.160.10
s=sip call
c=IN IP4 67.150.160.10
t=0 0
m=audio 17308 RTP/AVP 0
<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
<<<<<<<<<<<<<<< IN
<<<<<<<<<<<<<<<<<<<<<<<<
<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP X.X.X.90;branch=z9hG4bK2aa.c3961912.0
Via: SIP/2.0/UDP
X.X.X.238:5060;received=X.X.X.238;branch=z9hG4bK10281a99;rport=5060
Record-Route: <sip:X.X.X.90;lr;ftag=as4c62e0b9
From: ""2026097878""
<sip:2026097878@mydomain.com<sip%3A2026097878@mydomain.com Content-Type: application/sdp
Content-Length: 109
v=0
o=ATSI-B1 0 0 IN IP4 67.150.160.10
s=sip call
c=IN IP4 67.150.160.10
t=0 0
m=audio 17308 RTP/AVP 0
From: ""2026097878""
<sip:2026097878@mydomain.com<sip%3A2026097878@mydomain.com
Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com
CSeq: 102 ACK
User-Agent: G-Tel v1.0
Max-Forwards: 70
Content-Length: 0
<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
<<<<<<<<<<<<<<< IN
<<<<<<<<<<<<<<<<<<<<<<<<
<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
BYE sip:X.X.X.90;lr;ftag=as4c62e0b9 SIP/2.0
Via: SIP/2.0/UDP 67.150.160.10;branch=z9hG4bKvNBHaK6ZByQmg
Max-Forwards: 70
From: <sip:+16782247008@X.X.X.90:5060>;tag=H3jDp3KZmHNja
To: ""2026097878""
<sip:2026097878@mydomain.com<sip%3A2026097878@mydomain.com
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>> OUT
>>>>>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
INVITE sip:16782247008@67.150.160.10:5060 SIP/2.0
Record-Route: <sip:X.X.X.90;lr;ftag=as4c62e0b9
;tag=as4c62e0b9
To: <sip:+16782247008@X.X.X.90:5060
;tag=as4c62e0b9
To: <sip:+16782247008@X.X.X.90:5060
;tag=as4c62e0b9
To: <sip:+16782247008@X.X.X.90:5060>;tag=H3jDp3KZmHNja
Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com
CSeq: 102 INVITE
Contact: <sip:16782247008@67.150.160.10 <sip%3A16782247008(a)67.150.160.10>
;tag=as4c62e0b9
To: <sip:+16782247008@X.X.X.90:5060>;tag=H3jDp3KZmHNja
Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com
CSeq: 102 INVITE
Contact: <sip:16782247008@67.150.160.10 <sip%3A16782247008(a)67.150.160.10>
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>> OUT
>>>>>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
ACK sip:16782247008@67.150.160.10 <sip%3A16782247008(a)67.150.160.10> SIP/2.0
Via: SIP/2.0/UDP X.X.X.238:5060;branch=z9hG4bK60b74898;rport
Route: <sip:X.X.X.90;lr;ftag=as4c62e0b9
;tag=as4c62e0b9
To: <sip:+16782247008@X.X.X.90:5060>;tag=H3jDp3KZmHNja
Contact: <sip:2026097878@X.X.X.238
;tag=as4c62e0b9
Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com
CSeq: 102 BYE
Content-Length: 0
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>> OUT
>>>>>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
SIP/2.0 404 Not here
Via: SIP/2.0/UDP 67.150.160.10;branch=z9hG4bKvNBHaK6ZByQmg
From: <sip:+16782247008@X.X.X.90:5060>;tag=H3jDp3KZmHNja
To: ""2026097878""
<sip:2026097878@mydomain.com<sip%3A2026097878@mydomain.com
;tag=as4c62e0b9
Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com
CSeq: 102 BYE
Server: Kamailio
Content-Length: 0
13 Oct
13 Oct
4:20 a.m.
New subject: Need some help with showing one of my ITSPs that they have a problem.
Hello,
the BYE is missing the Route header. Even if it looks like doing strict
routing instead of recommended loose routing, the BYE must have a Route
header with the contact address in INVITE or 200ok, depending whether
the BYE is sent by callee or caller.
Cheers,
Daniel
On 10/12/10 4:02 PM, Geoffrey Mina wrote:
Hello,
I am currently going through an Inter-Op test with a new carrier and
there have been some issues. I have my Kamailio server integrated
with many tier-1 carriers, so I'm pretty sure I'm doing everything
OK. The problem is in the BYE coming from the ITSP, the URI isn't
valid, but I am not sure based on RFC3261 exactly why it isn't
correct. If anyone can take a quick glance, I would really appreciate it.
Thanks!
--
Daniel-Constantin Mierla
http://www.asipto.com
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>> OUT
>>>>>>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
INVITE sip:16782247008@67.150.160.10:5060
<http://sip:16782247008@67.150.160.10:5060> SIP/2.0
Record-Route: <sip:X.X.X.90;lr;ftag=as4c62e0b9>
Via: SIP/2.0/UDP X.X.X.90;branch=z9hG4bK2aa.c3961912.0
Via: SIP/2.0/UDP
X.X.X.238:5060;received=X.X.X.238;branch=z9hG4bK10281a99;rport=5060
From: ""2026097878"" <sip:2026097878@mydomain.com
<mailto:sip%3A2026097878@mydomain.com>>;tag=as4c62e0b9
To: <sip:+16782247008@X.X.X.90:5060>
Contact: <sip:2026097878@X.X.X.238>
Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com
<mailto:4199b75502736b01448ce4bf04876d7e@mydomain.com>
CSeq: 102 INVITE
User-Agent: G-Tel v1.0
Max-Forwards: 69
Date: Mon, 11 Oct 2010 17:50:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
P-Charge-Info: ""4019921313"" <sip:4019921313@connectfirst.com
<mailto:sip%3A4019921313@connectfirst.com>>
Content-Type: application/sdp
Content-Length: 244
v=0
o=root 10137 10137 IN IP4 X.X.X.238
s=session
c=IN IP4 X.X.X.238
t=0 0
m=audio 11748 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
<<<<<<<<<<<<<<< IN
<<<<<<<<<<<<<<<<<<<<<<<<
<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
SIP/2.0 100 Trying
Via: SIP/2.0/UDP X.X.X.90;branch=z9hG4bK2aa.c3961912.0
Via: SIP/2.0/UDP
X.X.X.238:5060;received=X.X.X.238;branch=z9hG4bK10281a99;rport=5060
Record-Route: <sip:X.X.X.90;lr;ftag=as4c62e0b9>
From: ""2026097878"" <sip:2026097878@mydomain.com
<mailto:sip%3A2026097878@mydomain.com>>;tag=as4c62e0b9
To: <sip:+16782247008@X.X.X.90:5060>
Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com
<mailto:4199b75502736b01448ce4bf04876d7e@mydomain.com>
CSeq: 102 INVITE
Contact: <sip:16782247008@67.150.160.10
<mailto:sip%3A16782247008@67.150.160.10>>
Content-Length: 0
<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
<<<<<<<<<<<<<<< IN
<<<<<<<<<<<<<<<<<<<<<<<<
<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP X.X.X.90;branch=z9hG4bK2aa.c3961912.0
Via: SIP/2.0/UDP
X.X.X.238:5060;received=X.X.X.238;branch=z9hG4bK10281a99;rport=5060
Record-Route: <sip:X.X.X.90;lr;ftag=as4c62e0b9>
From: ""2026097878"" <sip:2026097878@mydomain.com
<mailto:sip%3A2026097878@mydomain.com>>;tag=as4c62e0b9
To: <sip:+16782247008@X.X.X.90:5060>;tag=H3jDp3KZmHNja
Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com
<mailto:4199b75502736b01448ce4bf04876d7e@mydomain.com>
CSeq: 102 INVITE
Contact: <sip:16782247008@67.150.160.10
<mailto:sip%3A16782247008@67.150.160.10>>
Content-Type: application/sdp
Content-Length: 109
v=0
o=ATSI-B1 0 0 IN IP4 67.150.160.10
s=sip call
c=IN IP4 67.150.160.10
t=0 0
m=audio 17308 RTP/AVP 0
<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
<<<<<<<<<<<<<<< IN
<<<<<<<<<<<<<<<<<<<<<<<<
<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP X.X.X.90;branch=z9hG4bK2aa.c3961912.0
Via: SIP/2.0/UDP
X.X.X.238:5060;received=X.X.X.238;branch=z9hG4bK10281a99;rport=5060
Record-Route: <sip:X.X.X.90;lr;ftag=as4c62e0b9>
From: ""2026097878"" <sip:2026097878@mydomain.com
<mailto:sip%3A2026097878@mydomain.com>>;tag=as4c62e0b9
To: <sip:+16782247008@X.X.X.90:5060>;tag=H3jDp3KZmHNja
Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com
<mailto:4199b75502736b01448ce4bf04876d7e@mydomain.com>
CSeq: 102 INVITE
Contact: <sip:16782247008@67.150.160.10
<mailto:sip%3A16782247008@67.150.160.10>>
Content-Type: application/sdp
Content-Length: 109
v=0
o=ATSI-B1 0 0 IN IP4 67.150.160.10
s=sip call
c=IN IP4 67.150.160.10
t=0 0
m=audio 17308 RTP/AVP 0
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>> OUT
>>>>>>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
ACK sip:16782247008@67.150.160.10
<mailto:sip%3A16782247008@67.150.160.10> SIP/2.0
Via: SIP/2.0/UDP X.X.X.238:5060;branch=z9hG4bK60b74898;rport
Route: <sip:X.X.X.90;lr;ftag=as4c62e0b9>
From: ""2026097878"" <sip:2026097878@mydomain.com
<mailto:sip%3A2026097878@mydomain.com>>;tag=as4c62e0b9
To: <sip:+16782247008@X.X.X.90:5060>;tag=H3jDp3KZmHNja
Contact: <sip:2026097878@X.X.X.238>
Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com
<mailto:4199b75502736b01448ce4bf04876d7e@mydomain.com>
CSeq: 102 ACK
User-Agent: G-Tel v1.0
Max-Forwards: 70
Content-Length: 0
<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
<<<<<<<<<<<<<<< IN
<<<<<<<<<<<<<<<<<<<<<<<<
<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
BYE sip:X.X.X.90;lr;ftag=as4c62e0b9 SIP/2.0
Via: SIP/2.0/UDP 67.150.160.10;branch=z9hG4bKvNBHaK6ZByQmg
Max-Forwards: 70
From: <sip:+16782247008@X.X.X.90:5060>;tag=H3jDp3KZmHNja
To: ""2026097878"" <sip:2026097878@mydomain.com
<mailto:sip%3A2026097878@mydomain.com>>;tag=as4c62e0b9
Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com
<mailto:4199b75502736b01448ce4bf04876d7e@mydomain.com>
CSeq: 102 BYE
Content-Length: 0
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>> OUT
>>>>>>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
SIP/2.0 404 Not here
Via: SIP/2.0/UDP 67.150.160.10;branch=z9hG4bKvNBHaK6ZByQmg
From: <sip:+16782247008@X.X.X.90:5060>;tag=H3jDp3KZmHNja
To: ""2026097878"" <sip:2026097878@mydomain.com
<mailto:sip%3A2026097878@mydomain.com>>;tag=as4c62e0b9
Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com
<mailto:4199b75502736b01448ce4bf04876d7e@mydomain.com>
CSeq: 102 BYE
Server: Kamailio
Content-Length: 0
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5242
days inactive
5243
days old
1 comments
2 participants
participants (2)
-
Daniel-Constantin Mierla -
Geoffrey Mina