24 Feb
2005
24 Feb
'05
9:52 a.m.
Paul,
I'm stretching my RFC understanding here, but I'll give my 2c:
I agree with you that a route header is missing in the Asterisk-generated
BYE. The question is why. I can see no problems with your record-routing
and I doubt that Asterisk does something wrong. The only anomaly I can see
is the 68.86.100.20 Record-Route:
Record-Route: <sip:68.86.100.20:5060;lr>.
The way I read the RFC, lr should be specified as lr=on if the proxy
supports loose routing. May it be that Asterisk interprets the proxy as a
strict router and thus (correctly) does not include a Route header? From
the RFC, section 19.1.1:
URI parameters: Parameters affecting a request constructed from
the URI. URI parameters are added after the hostport component and
are
separated by semi-colons.
URI parameters take the form:
parameter-name "=" parameter-value
Even though an arbitrary number of URI parameters may be
included in a URI, any given parameter-name MUST NOT appear
more than once. This extensible mechanism includes the transport,
maddr, ttl,
user, method and lr parameters.
g-)
Java Rockx wrote:
Hi All.
I'm using ser-0.9
Can anyone take a quick look at this short SIP conversation and tell
me if they think the problem is with my ser.cfg or a bug in Asterisk
1.0.2.
We use a 3rd party for PSTN gateway functionality. This 3rd party uses
a Sonus box behind a SIP proxy. Our SER proxy talks directly to their
SIP proxy as needed to complete PSTN calls.
The problem is that when a PSTN caller dials a SIP phone and gets sent
to voice mail (Asterisk) because of a no answer or busy condition,
Asterisk hangs up after the caller leaves a message. When Asterisk
hangs up, the BYE from Asterisk is sent to SER, however, SER
incorrectly forwards the BYE directly to their Sonus gateway, rather
than the their SIP proxy. This causes our PSTN gateway provider to
have "open" billing records in their system.
If you look at the BYE message from Asterisk to SER you can see that
route headers are missing (I think). The final BYE should have been
sent to 68.86.100.20, but it was sent to 68.86.100.30 instead.
I am record_route()ing all messages except for REGISTER and I have the
mhomed=1 parameter set.
Can anyone help me put the blame on either my ser.cfg or Asterisk?
Regards,
Paul
IP LEGEND
-----------
68.86.100.30 - 3rd Party Sonus PSTN Gateway
68.86.100.20 - 3rd Party SIP Proxy
24.11.12.24 - Sip Express Router (eth0)
10.255.255.1 - Sip Express Router (eth1)
10.255.255.2 - Asterisk PBX
NOTE: I have Asterisk connected to the SER server with a crossover
cable.
U 2005/02/23 22:24:18.848582 68.86.100.20:5060 -> 24.11.12.24:5060
INVITE sip:4075551212@24.11.12.24:5060 SIP/2.0.
Via: SIP/2.0/UDP 68.86.100.20:5060;branch=z9hG4bKed82c1ba766-c3014f40.
Via: SIP/2.0/UDP
68.86.100.30:5060;branch=7a874abcac87c7752a3d3c5c7ba10dc0.
To: 4075551212 <sip:4075551212@68.86.100.30:5060>.
From: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD.
Call-ID: 9028535-3318204258-749010(a)68.86.100.30.
CSeq: 1 INVITE.
Max-Forwards: 4.
Contact: sip:3211231234@68.86.100.30:5060.
Record-Route: <sip:68.86.100.20:5060;lr>.
Content-Type: application/sdp.
Content-Length: 312.
.
v=0.
o=NexTone-MSW 1234 187 IN IP4 66.236.245.98.
s=sip call.
c=IN IP4 66.236.245.98.
t=0 0.
m=audio 16814 RTP/AVP 18 0 4 8 101.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:0 PCMU/8000.
a=rtpmap:4 G723/8000.
a=fmtp:4 annexa=yes.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
#
U 2005/02/23 22:24:18.860022 24.11.12.24:5060 -> 68.86.100.20:5060
SIP/2.0 100 trying -- your call is important to us.
Via: SIP/2.0/UDP 68.86.100.20:5060;branch=z9hG4bKed82c1ba766-c3014f40.
Via: SIP/2.0/UDP
68.86.100.30:5060;branch=7a874abcac87c7752a3d3c5c7ba10dc0.
To: 4075551212 <sip:4075551212@68.86.100.30:5060>.
From: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD.
Call-ID: 9028535-3318204258-749010(a)68.86.100.30.
CSeq: 1 INVITE.
Content-Length: 0.
.
#
U 2005/02/23 22:24:18.860259 10.255.255.1:1033 -> 10.255.255.2:5060
INVITE sip:699@10.255.255.2:5060 SIP/2.0.
Record-Route: <sip:10.255.255.1;r2=on;ftag=27DECB5C-17BD;lr=on>.
Record-Route: <sip:24.11.12.24;r2=on;ftag=27DECB5C-17BD;lr=on>.
Via: SIP/2.0/UDP 10.255.255.1;branch=z9hG4bKb929.21080974.0.
Via: SIP/2.0/UDP 68.86.100.20:5060;branch=z9hG4bKed82c1ba766-c3014f40.
Via: SIP/2.0/UDP
68.86.100.30:5060;branch=7a874abcac87c7752a3d3c5c7ba10dc0.
To: 4075551212 <sip:4075551212@68.86.100.30:5060>.
From: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD.
Call-ID: 9028535-3318204258-749010(a)68.86.100.30.
CSeq: 1 INVITE.
Max-Forwards: 3.
Contact: sip:3211231234@68.86.100.30:5060.
Record-Route: <sip:68.86.100.20:5060;lr>.
Content-Type: application/sdp.
Content-Length: 312.
.
v=0.
o=NexTone-MSW 1234 187 IN IP4 66.236.245.98.
s=sip call.
c=IN IP4 24.11.12.24.
t=0 0.
m=audio 36574 RTP/AVP 18 0 4 8 101.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:0 PCMU/8000.
a=rtpmap:4 G723/8000.
a=fmtp:4 annexa=yes.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
#
U 2005/02/23 22:24:18.871131 10.255.255.2:5060 -> 10.255.255.1:1033
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP
10.255.255.1;branch=z9hG4bKb929.21080974.0;received=10.255.255.1;rport=1033.
Via: SIP/2.0/UDP 68.86.100.20:5060;branch=z9hG4bKed82c1ba766-c3014f40.
Via: SIP/2.0/UDP
68.86.100.30:5060;branch=7a874abcac87c7752a3d3c5c7ba10dc0.
From: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD.
To: 4075551212 <sip:4075551212@68.86.100.30:5060>;tag=as588114d9.
Call-ID: 9028535-3318204258-749010(a)68.86.100.30.
CSeq: 1 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER.
Contact: <sip:699@10.255.255.2>.
Content-Length: 0.
.
#
U 2005/02/23 22:24:18.879160 10.255.255.2:5060 -> 10.255.255.1:1033
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
10.255.255.1;branch=z9hG4bKb929.21080974.0;received=10.255.255.1;rport=1033.
Via: SIP/2.0/UDP 68.86.100.20:5060;branch=z9hG4bKed82c1ba766-c3014f40.
Via: SIP/2.0/UDP
68.86.100.30:5060;branch=7a874abcac87c7752a3d3c5c7ba10dc0.
Record-Route: <sip:10.255.255.1;r2=on;ftag=27DECB5C-17BD;lr=on>.
Record-Route: <sip:24.11.12.24;r2=on;ftag=27DECB5C-17BD;lr=on>.
Record-Route: <sip:68.86.100.20:5060;lr>.
From: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD.
To: 4075551212 <sip:4075551212@68.86.100.30:5060>;tag=as588114d9.
Call-ID: 9028535-3318204258-749010(a)68.86.100.30.
CSeq: 1 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER.
Contact: <sip:699@10.255.255.2>.
Content-Type: application/sdp.
Content-Length: 362.
.
v=0.
o=root 550 550 IN IP4 10.255.255.2.
s=session.
c=IN IP4 10.255.255.2.
t=0 0.
m=audio 17900 RTP/AVP 97 18 3 4 2 0 8 101.
a=rtpmap:97 iLBC/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:4 G723/8000.
a=rtpmap:2 G726-32/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
#
U 2005/02/23 22:24:18.883882 24.11.12.24:5060 -> 68.86.100.20:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 68.86.100.20:5060;branch=z9hG4bKed82c1ba766-c3014f40.
Via: SIP/2.0/UDP
68.86.100.30:5060;branch=7a874abcac87c7752a3d3c5c7ba10dc0.
Record-Route: <sip:10.255.255.1;r2=on;ftag=27DECB5C-17BD;lr=on>.
Record-Route: <sip:24.11.12.24;r2=on;ftag=27DECB5C-17BD;lr=on>.
Record-Route: <sip:68.86.100.20:5060;lr>.
From: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD.
To: 4075551212 <sip:4075551212@68.86.100.30:5060>;tag=as588114d9.
Call-ID: 9028535-3318204258-749010(a)68.86.100.30.
CSeq: 1 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER.
Contact: <sip:699@10.255.255.2>.
Content-Type: application/sdp.
Content-Length: 363.
.
v=0.
o=root 550 550 IN IP4 10.255.255.2.
s=session.
c=IN IP4 24.11.12.24.
t=0 0.
m=audio 36574 RTP/AVP 97 18 3 4 2 0 8 101.
a=rtpmap:97 iLBC/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:4 G723/8000.
a=rtpmap:2 G726-32/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
#
U 2005/02/23 22:24:19.097436 68.86.100.20:5060 -> 24.11.12.24:5060
ACK sip:699@10.255.255.2 SIP/2.0.
Via: SIP/2.0/UDP
68.86.100.20:5060;branch=ef5ce1f8a400adab242436d0564cb045.
Via: SIP/2.0/UDP
68.86.100.30:5060;branch=ef5ce1f8a400adab242436d0564cb045.
To: 4075551212 <sip:4075551212@68.86.100.30:5060>;tag=as588114d9.
From: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD.
Call-ID: 9028535-3318204258-749010(a)68.86.100.30.
CSeq: 1 ACK.
Max-Forwards: 4.
Contact: sip:3211231234@68.86.100.30:5060.
Record-Route: <sip:68.86.100.20:5060;lr>.
Route: <sip:24.11.12.24;r2=on;ftag=27DECB5C-17BD;lr=on>.
Route: <sip:10.255.255.1;r2=on;ftag=27DECB5C-17BD;lr=on>.
Content-Length: 0.
.
#
U 2005/02/23 22:24:19.098087 10.255.255.1:1033 -> 10.255.255.2:5060
ACK sip:699@10.255.255.2 SIP/2.0.
Record-Route: <sip:10.255.255.1;r2=on;ftag=27DECB5C-17BD;lr=on>.
Record-Route: <sip:24.11.12.24;r2=on;ftag=27DECB5C-17BD;lr=on>.
Via: SIP/2.0/UDP 10.255.255.1;branch=0.
Via: SIP/2.0/UDP
68.86.100.20:5060;branch=ef5ce1f8a400adab242436d0564cb045.
Via: SIP/2.0/UDP
68.86.100.30:5060;branch=ef5ce1f8a400adab242436d0564cb045.
To: 4075551212 <sip:4075551212@68.86.100.30:5060>;tag=as588114d9.
From: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD.
Call-ID: 9028535-3318204258-749010(a)68.86.100.30.
CSeq: 1 ACK.
Max-Forwards: 3.
Contact: sip:3211231234@68.86.100.30:5060.
Record-Route: <sip:68.86.100.20:5060;lr>.
Content-Length: 0.
.
###
U 2005/02/23 22:24:25.104860 10.255.255.2:5060 -> 10.255.255.1:1033
BYE sip:3211231234@68.86.100.30:5060 SIP/2.0.
Via: SIP/2.0/UDP 10.255.255.2:5060;branch=z9hG4bK2da77693;rport.
Route:
<sip:24.11.12.24;r2=on;ftag=27DECB5C-17BD;lr=on>,<sip:68.86.100.20:5060;lr>,<sip:3211231234@68.86.100.30:5060>.
From: 4075551212 <sip:4075551212@68.86.100.30:5060>;tag=as588114d9.
To: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD.
Contact: <sip:699@10.255.255.2>.
Call-ID: 9028535-3318204258-749010(a)68.86.100.30.
CSeq: 102 BYE.
User-Agent: Asterisk PBX.
Content-Length: 0.
.
#
U 2005/02/23 22:24:25.108961 24.11.12.24:5060 -> 68.86.100.30:5060
BYE sip:3211231234@68.86.100.30:5060 SIP/2.0.
Max-Forwards: 10.
Record-Route: <sip:24.11.12.24;r2=on;ftag=as588114d9;lr=on>.
Record-Route: <sip:10.255.255.1;r2=on;ftag=as588114d9;lr=on>.
Via: SIP/2.0/UDP 24.11.12.24;branch=z9hG4bK231b.07949ea5.0.
Via: SIP/2.0/UDP 10.255.255.2:5060;branch=z9hG4bK2da77693;rport=5060.
From: 4075551212 <sip:4075551212@68.86.100.30:5060>;tag=as588114d9.
To: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD.
Contact: <sip:699@10.255.255.2>.
Call-ID: 9028535-3318204258-749010(a)68.86.100.30.
CSeq: 102 BYE.
User-Agent: Asterisk PBX.
Content-Length: 0.
Route: <sip:3211231234@68.86.100.30:5060>.
.
#
U 2005/02/23 22:24:25.175832 68.86.100.30:5060 -> 24.11.12.24:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 24.11.12.24;branch=z9hG4bK231b.07949ea5.0.
Via: SIP/2.0/UDP 10.255.255.2:5060;branch=z9hG4bK2da77693;rport=5060.
From: 4075551212 <sip:4075551212@68.86.100.30:5060>;tag=as588114d9.
To: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD.
Call-ID: 9028535-3318204258-749010(a)68.86.100.30.
CSeq: 102 BYE.
Content-Length: 0.
.
#
U 2005/02/23 22:24:25.176182 10.255.255.1:1033 -> 10.255.255.2:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 10.255.255.2:5060;branch=z9hG4bK2da77693;rport=5060.
From: 4075551212 <sip:4075551212@68.86.100.30:5060>;tag=as588114d9.
To: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD.
Call-ID: 9028535-3318204258-749010(a)68.86.100.30.
CSeq: 102 BYE.
Content-Length: 0.
.
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