24 Feb
2005
24 Feb
'05
5:44 a.m.
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.
.
24 Feb
24 Feb
8:52 a.m.
New subject: [Serusers] HELP: SER-0.9 Record Route Problem Between SER andAsterisk
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.
.
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
9:08 a.m.
New subject: [Serusers] HELP: SER-0.9 Record Route Problem Between SER andAsterisk
It looks like a ser related problem. A ser config would help to
troubleshoot.
Richard
-----Original Message-----
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On
Behalf Of Java Rockx
Sent: Wednesday, February 23, 2005 6:44 PM
To: serusers(a)lists.iptel.org
Subject: [Serusers] HELP: SER-0.9 Record Route Problem Between SER
andAsterisk
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=103
3.
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=103
3.
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.
.
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
1:10 p.m.
New subject: [Serusers] HELP: SER-0.9 Record Route Problem Between SER andAsterisk
Here is my ser.cfg which I've removed all the non-BYE related route blocks.
Many Thanks!
Paul
listen=24.11.12.24
listen=10.255.255.1
mhomed=1
# loadmodule stuff goes here
# modparam stuff goes here
route {
# ------------------------------------------------------------------------
# Sanity Check Section
# ------------------------------------------------------------------------
# ------------------------------------------------------------------------
# Record Route Section
# ------------------------------------------------------------------------
if (method!="REGISTER") {
record_route();
};
if (loose_route()) {
if (method=="BYE") {
setflag(1); # enable accounting for BYE messages. We do not
# enable accounting for Record-Routed INVITE messages
# because these are re-INVITEs and we already captured
# the original INVITE during the call setup
};
# I'm using mediaproxy - so tear down the RTP stuff
if (method=="BYE" || method=="CANCEL")) {
end_media_session();
};
if (method=="INVITE") {
# re-INVITE messages must be handled differently because
# our transparent RTP proxy needs to test the caller and
# callee for NATed IP addresses otherwise we could lose
# audio on one or both directions
route(8);
} else {
route(1);
};
break;
};
# Only handle message destined for out served domains. Other messages
are just relayed
if (!uri==myself) {
route(1);
break;
};
# ------------------------------------------------------------------------
# Message Handler Logic
# ------------------------------------------------------------------------
if (method=="BYE") {
# NOTE: all BYE messages should be record-routed so they should never
# hit this code block since they should be handled in the
# loose_route section above, so this is just a safety net
end_media_session();
route(1);
} else if (method=="CANCEL") {
# I'm not sure if any CANCELs messages would ever hit this block
# but it's here for good measure :-)
end_media_session();
route(1);
} else if (method=="INVITE") {
setflag(1); # enable accounting for *original* INVITE messages
route(6);
} else if (method=="NOTIFY") {
route(2);
} else if (method=="OPTIONS") {
route(3);
} else if (method=="REFER") {
route(6);
} else if (method=="REGISTER") {
route(4);
} else if (method=="SUBSCRIBE") {
route(5);
} else {
# all other messages come here for default handling
route(1);
};
}
route[1] {
# ------------------------------------------------------------------------
# default message handler
# ------------------------------------------------------------------------
# A note on when we need to call lookup("location")
#
# This was posted to serusers on 02/23/2005 by Daniel-Constantin Mierla
#
# lookup("location") has to be used for any request that has the domain
# part of R-URI pointing to your SIP server, should be delivered to an
# end-user and does not have to follow any Route header -- it does not
# matter the type of method. Could be said that only REGISTERs are just
# for servers, the others are either mixed (e.g, OPTIONS) or only for
# end-users.
if (!search(^"Route:") && !search(^"Record-Route:")) {
lookup("location");
};
if (!t_relay()) {
sl_reply_error();
};
}
route[2] {
# ------------------------------------------------------------------------
# NOTIFY Message Handler
# ------------------------------------------------------------------------
}
route[3] {
# ------------------------------------------------------------------------
# OPTIONS Message Handler
# ------------------------------------------------------------------------
}
route[4] {
# ------------------------------------------------------------------------
# REGISTER Message Handler
# ------------------------------------------------------------------------
}
route[5] {
# ------------------------------------------------------------------------
# SUBSCRIBE Message Handler
# ------------------------------------------------------------------------
}
route[6] {
# ------------------------------------------------------------------------
# INVITE Message Handler
# ------------------------------------------------------------------------
}
route[7] {
# voicemail route
rewritehostport("10.255.255.2:5060");
append_branch();
if (!isflagset(15)) {
use_media_proxy();
};
t_on_reply("1");
if (!t_relay()) {
end_media_session();
sl_reply_error();
};
}
route[8] {
# ------------------------------------------------------------------------
# re-INVITE Message Handler
#
# This route is a stripped down version of route[6]. Here we only
# lookup('location') in order to get the NAT flag from the location
# table because we need to know wheather or not to enable RTP proxying
}
onreply_route[1] {
}
failure_route[1] {
}
On Wed, 23 Feb 2005 22:08:12 -1000, Richard <richard(a)o-matrix.org> wrote:
It looks like a ser related problem. A ser config
would help to
troubleshoot.
Richard
-----Original Message-----
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On
Behalf Of Java Rockx
Sent: Wednesday, February 23, 2005 6:44 PM
To: serusers(a)lists.iptel.org
Subject: [Serusers] HELP: SER-0.9 Record Route Problem Between SER
andAsterisk
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=103
3.
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=103
3.
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.
.
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
25 Feb
25 Feb
5:59 a.m.
New subject: [Serusers] HELP: SER-0.9 Record Route Problem Between SER andAsterisk
Greg,
I really do need to buy you a pint because that was the problem. I'll
test further tomorrow, but Asterisk really seemed to be unhappy with
that ;lr> attribute in the record-route headers.
I've asked our 3rd party pstn gateway provider to modify their Sonus
box (if they can) to write the lr tag correctly. But for now I have no
choice but to kludge my ser.cfg and search() for it in every message.
:-(
Regards,
Paul
On Thu, 24 Feb 2005 07:10:53 -0500, Java Rockx <javarockx(a)gmail.com> wrote:
Here is my ser.cfg which I've removed all the
non-BYE related route blocks.
Many Thanks!
Paul
listen=24.11.12.24
listen=10.255.255.1
mhomed=1
# loadmodule stuff goes here
# modparam stuff goes here
route {
# ------------------------------------------------------------------------
# Sanity Check Section
# ------------------------------------------------------------------------
# ------------------------------------------------------------------------
# Record Route Section
# ------------------------------------------------------------------------
if (method!="REGISTER") {
record_route();
};
if (loose_route()) {
if (method=="BYE") {
setflag(1); # enable accounting for BYE messages. We do not
# enable accounting for Record-Routed INVITE
messages
# because these are re-INVITEs and we already
captured
# the original INVITE during the call setup
};
# I'm using mediaproxy - so tear down the RTP stuff
if (method=="BYE" || method=="CANCEL")) {
end_media_session();
};
if (method=="INVITE") {
# re-INVITE messages must be handled differently because
# our transparent RTP proxy needs to test the caller and
# callee for NATed IP addresses otherwise we could lose
# audio on one or both directions
route(8);
} else {
route(1);
};
break;
};
# Only handle message destined for out served domains. Other messages
are just relayed
if (!uri==myself) {
route(1);
break;
};
# ------------------------------------------------------------------------
# Message Handler Logic
# ------------------------------------------------------------------------
if (method=="BYE") {
# NOTE: all BYE messages should be record-routed so they should never
# hit this code block since they should be handled in the
# loose_route section above, so this is just a safety net
end_media_session();
route(1);
} else if (method=="CANCEL") {
# I'm not sure if any CANCELs messages would ever hit this block
# but it's here for good measure :-)
end_media_session();
route(1);
} else if (method=="INVITE") {
setflag(1); # enable accounting for *original* INVITE messages
route(6);
} else if (method=="NOTIFY") {
route(2);
} else if (method=="OPTIONS") {
route(3);
} else if (method=="REFER") {
route(6);
} else if (method=="REGISTER") {
route(4);
} else if (method=="SUBSCRIBE") {
route(5);
} else {
# all other messages come here for default handling
route(1);
};
}
route[1] {
# ------------------------------------------------------------------------
# default message handler
# ------------------------------------------------------------------------
# A note on when we need to call lookup("location")
#
# This was posted to serusers on 02/23/2005 by Daniel-Constantin Mierla
#
# lookup("location") has to be used for any request that has the
domain
# part of R-URI pointing to your SIP server, should be delivered to an
# end-user and does not have to follow any Route header -- it does not
# matter the type of method. Could be said that only REGISTERs are just
# for servers, the others are either mixed (e.g, OPTIONS) or only for
# end-users.
if (!search(^"Route:") && !search(^"Record-Route:"))
{
lookup("location");
};
if (!t_relay()) {
sl_reply_error();
};
}
route[2] {
# ------------------------------------------------------------------------
# NOTIFY Message Handler
# ------------------------------------------------------------------------
}
route[3] {
# ------------------------------------------------------------------------
# OPTIONS Message Handler
# ------------------------------------------------------------------------
}
route[4] {
# ------------------------------------------------------------------------
# REGISTER Message Handler
# ------------------------------------------------------------------------
}
route[5] {
# ------------------------------------------------------------------------
# SUBSCRIBE Message Handler
# ------------------------------------------------------------------------
}
route[6] {
# ------------------------------------------------------------------------
# INVITE Message Handler
# ------------------------------------------------------------------------
}
route[7] {
# voicemail route
rewritehostport("10.255.255.2:5060");
append_branch();
if (!isflagset(15)) {
use_media_proxy();
};
t_on_reply("1");
if (!t_relay()) {
end_media_session();
sl_reply_error();
};
}
route[8] {
# ------------------------------------------------------------------------
# re-INVITE Message Handler
#
# This route is a stripped down version of route[6]. Here we only
# lookup('location') in order to get the NAT flag from the location
# table because we need to know wheather or not to enable RTP proxying
}
onreply_route[1] {
}
failure_route[1] {
}
On Wed, 23 Feb 2005 22:08:12 -1000, Richard <richard(a)o-matrix.org> wrote:
It looks like a ser related problem. A ser config
would help to
troubleshoot.
Richard
-----Original Message-----
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On
Behalf Of Java Rockx
Sent: Wednesday, February 23, 2005 6:44 PM
To: serusers(a)lists.iptel.org
Subject: [Serusers] HELP: SER-0.9 Record Route Problem Between SER
andAsterisk
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=103
3.
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=103
3.
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.
.
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
9:08 a.m.
New subject: [Serusers] HELP: SER-0.9 Record Route Problem Between SER andAsterisk
Route field should follow the reverse record-route field of last incoming
packet. In this case, BYE should use the record-route of ACK to construct
its route field. But I don't think it is the case with *.
It should be something like (with tag),
<sip:10.255.255.1;lr=on>, <sip:24.11.12.24;lr=on>,
<sip:68.86.100.20:5060;lr=on>
however it uses,
<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>.
Richard
-----Original Message-----
From: Java Rockx [mailto:javarockx@gmail.com]
Sent: Thursday, February 24, 2005 2:11 AM
To: Richard
Cc: serusers(a)lists.iptel.org
Subject: Re: [Serusers] HELP: SER-0.9 Record Route Problem Between SER
andAsterisk
Here is my ser.cfg which I've removed all the non-BYE related route
blocks.
Many Thanks!
Paul
listen=24.11.12.24
listen=10.255.255.1
mhomed=1
# loadmodule stuff goes here
# modparam stuff goes here
route {
# ------------------------------------------------------------------
------
# Sanity Check Section
# ------------------------------------------------------------------
------
# ------------------------------------------------------------------
------
# Record Route Section
# ------------------------------------------------------------------
------
if (method!="REGISTER") {
record_route();
};
if (loose_route()) {
if (method=="BYE") {
setflag(1); # enable accounting for BYE
messages. We do
not
# enable accounting for
Record-Routed INVITE
messages
# because these are re-INVITEs and
we
already captured
# the original INVITE during the
call setup
};
# I'm using mediaproxy - so tear down the RTP stuff
if (method=="BYE" || method=="CANCEL")) {
end_media_session();
};
if (method=="INVITE") {
# re-INVITE messages must be handled differently
because
# our transparent RTP proxy needs to test the
caller
and
# callee for NATed IP addresses otherwise we could
lose
# audio on one or both directions
route(8);
} else {
route(1);
};
break;
};
# Only handle message destined for out served domains. Other
messages
are just relayed
if (!uri==myself) {
route(1);
break;
};
# ------------------------------------------------------------------
------
# Message Handler Logic
# ------------------------------------------------------------------
------
if (method=="BYE") {
# NOTE: all BYE messages should be record-routed so they
should never
# hit this code block since they should be handled in
the
# loose_route section above, so this is just a
safety
net
end_media_session();
route(1);
} else if (method=="CANCEL") {
# I'm not sure if any CANCELs messages would ever hit this
block
# but it's here for good measure :-)
end_media_session();
route(1);
} else if (method=="INVITE") {
setflag(1); # enable accounting for *original* INVITE
messages
route(6);
} else if (method=="NOTIFY") {
route(2);
} else if (method=="OPTIONS") {
route(3);
} else if (method=="REFER") {
route(6);
} else if (method=="REGISTER") {
route(4);
} else if (method=="SUBSCRIBE") {
route(5);
} else {
# all other messages come here for default handling
route(1);
};
}
route[1] {
# ----------------------------------------------------------------
--------
# default message handler
# ----------------------------------------------------------------
--------
# A note on when we need to call lookup("location")
#
# This was posted to serusers on 02/23/2005 by Daniel-Constantin
Mierla
#
# lookup("location") has to be used for any request that has the
domain
# part of R-URI pointing to your SIP server, should be delivered to
an
# end-user and does not have to follow any Route header -- it does
not
# matter the type of method. Could be said that only REGISTERs are
just
# for servers, the others are either mixed (e.g, OPTIONS) or only
for
# end-users.
if (!search(^"Route:") && !search(^"Record-Route:")) {
lookup("location");
};
if (!t_relay()) {
sl_reply_error();
};
}
route[2] {
# ----------------------------------------------------------------
--------
# NOTIFY Message Handler
# ----------------------------------------------------------------
--------
}
route[3] {
# ------------------------------------------------------------------
------
# OPTIONS Message Handler
# ------------------------------------------------------------------
------
}
route[4] {
# ------------------------------------------------------------------
------
# REGISTER Message Handler
# ------------------------------------------------------------------
------
}
route[5] {
# ------------------------------------------------------------------
------
# SUBSCRIBE Message Handler
# ------------------------------------------------------------------
------
}
route[6] {
# ------------------------------------------------------------------
------
# INVITE Message Handler
# ------------------------------------------------------------------
------
}
route[7] {
# voicemail route
rewritehostport("10.255.255.2:5060");
append_branch();
if (!isflagset(15)) {
use_media_proxy();
};
t_on_reply("1");
if (!t_relay()) {
end_media_session();
sl_reply_error();
};
}
route[8] {
# ----------------------------------------------------------------
--------
# re-INVITE Message Handler
#
# This route is a stripped down version of route[6]. Here we only
# lookup('location') in order to get the NAT flag from the location
# table because we need to know wheather or not to enable RTP
proxying
}
onreply_route[1] {
}
failure_route[1] {
}
On Wed, 23 Feb 2005 22:08:12 -1000, Richard <richard(a)o-matrix.org> wrote:
10.255.255.1;branch=z9hG4bKb929.21080974.0;received=10.255.255.1;rport=103
10.255.255.1;branch=z9hG4bKb929.21080974.0;received=10.255.255.1;rport=103
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.
> > .
> >
> > _______________________________________________
> > Serusers mailing list
> > serusers(a)lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
>
>
It looks like a ser related problem. A ser config
would help to
troubleshoot.
Richard
> -----Original Message-----
> From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org]
On
> Behalf Of Java Rockx
> Sent: Wednesday, February 23, 2005 6:44 PM
> To: serusers(a)lists.iptel.org
> Subject: [Serusers] HELP: SER-0.9 Record Route Problem Between SER
> andAsterisk
>
> 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
> 3.
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
> 3.
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-
> 
1 Mar
1 Mar
5:51 p.m.
New subject: [Serusers] HELP: SER-0.9 Record Route Problem Between SER andAsterisk
Actually RFC3261 specifies that loose routers should use ;lr, not
;lr=on. ;lr=on is our extension because it helps to deal with broken
implementations. There were some implementations that stripped the ;lr
parameter because it contained no value, so we temporarily put =on
there.
The spec does not exactly forbid this, but ;lr is correct (and from time
to time someone complains about ;lr=on).
You can control behavior of SER using enable_full_lr parameter of rr
module.
modparam("rr", "enable_full_lr", 1) would make ser to insert ;lr=on
modparam("rr", "enable_full_lr", 0) would make ser to insert ;lr
Jan.
On 24-02 22:08, Richard wrote:
Route field should follow the reverse record-route
field of last incoming
packet. In this case, BYE should use the record-route of ACK to construct
its route field. But I don't think it is the case with *.
It should be something like (with tag),
<sip:10.255.255.1;lr=on>, <sip:24.11.12.24;lr=on>,
<sip:68.86.100.20:5060;lr=on>
however it uses,
<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>.
Richard
-----Original Message-----
From: Java Rockx [mailto:javarockx@gmail.com]
Sent: Thursday, February 24, 2005 2:11 AM
To: Richard
Cc: serusers(a)lists.iptel.org
Subject: Re: [Serusers] HELP: SER-0.9 Record Route Problem Between SER
andAsterisk
Here is my ser.cfg which I've removed all the non-BYE related route
blocks.
Many Thanks!
Paul
listen=24.11.12.24
listen=10.255.255.1
mhomed=1
# loadmodule stuff goes here
# modparam stuff goes here
route {
# ------------------------------------------------------------------
------
# Sanity Check Section
# ------------------------------------------------------------------
------
# ------------------------------------------------------------------
------
# Record Route Section
# ------------------------------------------------------------------
------
if (method!="REGISTER") {
record_route();
};
if (loose_route()) {
if (method=="BYE") {
setflag(1); # enable accounting for BYE
messages. We do
not
# enable accounting for
Record-Routed INVITE
messages
# because these are re-INVITEs and
we
already captured
# the original INVITE during the
call setup
};
# I'm using mediaproxy - so tear down the RTP stuff
if (method=="BYE" || method=="CANCEL")) {
end_media_session();
};
if (method=="INVITE") {
# re-INVITE messages must be handled differently
because
# our transparent RTP proxy needs to test the
caller
and
# callee for NATed IP addresses otherwise we
could
lose
# audio on one or both directions
route(8);
} else {
route(1);
};
break;
};
# Only handle message destined for out served domains. Other
messages
are just relayed
if (!uri==myself) {
route(1);
break;
};
# ------------------------------------------------------------------
------
# Message Handler Logic
# ------------------------------------------------------------------
------
if (method=="BYE") {
# NOTE: all BYE messages should be record-routed so they
should never
# hit this code block since they should be handled in
the
# loose_route section above, so this is just a
safety
net
end_media_session();
route(1);
} else if (method=="CANCEL") {
# I'm not sure if any CANCELs messages would ever hit this
block
# but it's here for good measure :-)
end_media_session();
route(1);
} else if (method=="INVITE") {
setflag(1); # enable accounting for *original* INVITE
messages
route(6);
} else if (method=="NOTIFY") {
route(2);
} else if (method=="OPTIONS") {
route(3);
} else if (method=="REFER") {
route(6);
} else if (method=="REGISTER") {
route(4);
} else if (method=="SUBSCRIBE") {
route(5);
} else {
# all other messages come here for default handling
route(1);
};
}
route[1] {
# ----------------------------------------------------------------
--------
# default message handler
# ----------------------------------------------------------------
--------
# A note on when we need to call lookup("location")
#
# This was posted to serusers on 02/23/2005 by Daniel-Constantin
Mierla
#
# lookup("location") has to be used for any request that has the
domain
# part of R-URI pointing to your SIP server, should be delivered to
an
# end-user and does not have to follow any Route header -- it does
not
# matter the type of method. Could be said that only REGISTERs are
just
# for servers, the others are either mixed (e.g, OPTIONS) or only
for
# end-users.
if (!search(^"Route:") && !search(^"Record-Route:")) {
lookup("location");
};
if (!t_relay()) {
sl_reply_error();
};
}
route[2] {
# ----------------------------------------------------------------
--------
# NOTIFY Message Handler
# ----------------------------------------------------------------
--------
}
route[3] {
# ------------------------------------------------------------------
------
# OPTIONS Message Handler
# ------------------------------------------------------------------
------
}
route[4] {
# ------------------------------------------------------------------
------
# REGISTER Message Handler
# ------------------------------------------------------------------
------
}
route[5] {
# ------------------------------------------------------------------
------
# SUBSCRIBE Message Handler
# ------------------------------------------------------------------
------
}
route[6] {
# ------------------------------------------------------------------
------
# INVITE Message Handler
# ------------------------------------------------------------------
------
}
route[7] {
# voicemail route
rewritehostport("10.255.255.2:5060");
append_branch();
if (!isflagset(15)) {
use_media_proxy();
};
t_on_reply("1");
if (!t_relay()) {
end_media_session();
sl_reply_error();
};
}
route[8] {
# ----------------------------------------------------------------
--------
# re-INVITE Message Handler
#
# This route is a stripped down version of route[6]. Here we only
# lookup('location') in order to get the NAT flag from the location
# table because we need to know wheather or not to enable RTP
proxying
}
onreply_route[1] {
}
failure_route[1] {
}
On Wed, 23 Feb 2005 22:08:12 -1000, Richard <richard(a)o-matrix.org> wrote:
10.255.255.1;branch=z9hG4bKb929.21080974.0;received=10.255.255.1;rport=103
10.255.255.1;branch=z9hG4bKb929.21080974.0;received=10.255.255.1;rport=103
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.
> > .
> >
> > _______________________________________________
> > Serusers mailing list
> > serusers(a)lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
>
>
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers It looks like a ser related problem. A ser config
would help to
troubleshoot.
Richard
> -----Original Message-----
> From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org]
On
> Behalf Of Java Rockx
> Sent: Wednesday, February 23, 2005 6:44 PM
> To: serusers(a)lists.iptel.org
> Subject: [Serusers] HELP: SER-0.9 Record Route Problem Between SER
> andAsterisk
>
> 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
> 3.
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
> 3.
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-
>
7005
days inactive
7010
days old
6 comments
4 participants
participants (4)
-
Greger V. Teigre -
Jan Janak -
Java Rockx -
Richard