5 Apr
2006
5 Apr
'06
2:09 a.m.
Hi All,
I am facing a problem when terminating calls to Voice Master.
The exact problem is :
Voice Master is configured for Pin prefix Authentication.
I add the gateway prefix in my openser.cfg using prefix("123#")
But the Voice Master is rejecting the calls with error 488 Not acceptable
here.
User dialing number and openser adding prefix to it.
But when I send the calls from the userend itself with prefix ie. User is
dialing 123#<number> then the calls are going fine. What comes to my mind is
that Voice Master is somehow reading the to_uri being sent. Can I somehow
add prefix to the to_uri as well ?
This is the ngrep trace of the rejected call :
U XXX.XXX.XXX.XX6:5060 -> XXX.XXX.XXX.XX2:5060
INVITE sip:10235#14085264000@XXX.XXX.XXX.XX2:5060 SIP/2.0.
Record-Route: <sip:XXX.XXX.XXX.XX6;ftag=4036363136;lr=on>.
Via: SIP/2.0/UDP XXX.XXX.XXX.XX6;branch=z9hG4bK5e7a.83627b27.0.
Via: SIP/2.0/UDP
61.16.188.4:5060;rport=5060;branch=z9hG4bKE6B692A1F9A1460484EF5F786E105BFE.
From: test <sip:test@XXX.XXX.XXX.XX6>;tag=4036363136.
To: <sip:14085264000@XXX.XXX.XXX.XX6>.
Contact: <sip:test@61.16.188.4:5060>.
Call-ID: 4D0342A8-6F0D-4C93-9218-6B423B9027EB(a)61.16.207.90.
CSeq: 38322 INVITE.
Authorization: Digest
username="test",realm="XXX.XXX.XXX.XX6",nonce="4433562d730464572c23207912e46
d50281d2136",response="59311eb34dd7ce8b3550064960af1508",uri="sip:1408526400
0(a)XXX.XXX.XXX.XX6"quot;.
Max-Forwards: 69.
Content-Type: application/sdp.
User-Agent: X-PRO build 1082.
Content-Length: 314.
.
v=0.
o=test 3846000 3846000 IN IP4 61.16.188.4.
s=X-PRO.
c=IN IP4 61.16.188.4.
t=0 0.
m=audio 8000 RTP/AVP 18 0 8 3 98 97 101.
a=rtpmap:0 pcmu/8000.
a=rtpmap:8 pcma/8000.
a=rtpmap:3 gsm/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:98 iLBC/8000.
a=rtpmap:97 speex/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
#
U XXX.XXX.XXX.XX2:5060 -> XXX.XXX.XXX.XX6:5060
SIP/2.0 100 Trying.
From: <sip:test@XXX.XXX.XXX.XX2:5060>;tag=4036363136.
To: <sip:14085264000@XXX.XXX.XXX.XX2:5060>;tag=669a9b6321e12fb3fd8a82d7.
CSeq: 38322 INVITE.
Call-ID: 4D0342A8-6F0D-4C93-9218-6B423B9027EB(a)61.16.207.90.
Via: SIP/2.0/UDP XXX.XXX.XXX.XX6:5060;branch=z9hG4bK5e7a.83627b27.0.
Via: SIP/2.0/UDP
61.16.188.4:5060;rport=5060;branch=z9hG4bKE6B692A1F9A1460484EF5F786E105BFE.
Content-Length: 0.
.
#
U XXX.XXX.XXX.XX2:5060 -> XXX.XXX.XXX.XX6:5060
SIP/2.0 488 Not Acceptable Here.
Record-Route: <sip:XXX.XXX.XXX.XX6;ftag=4036363136;lr=on>.
Via: SIP/2.0/UDP XXX.XXX.XXX.XX6;branch=z9hG4bK5e7a.83627b27.0.
Via: SIP/2.0/UDP
61.16.188.4:5060;rport=5060;branch=z9hG4bKE6B692A1F9A1460484EF5F786E105BFE.
From: "test" <sip:test@XXX.XXX.XXX.XX6>;tag=4036363136.
To: <sip:14085264000@XXX.XXX.XXX.XX6>.
Contact: <sip:test@61.16.188.4:5060>.
Call-ID: 4D0342A8-6F0D-4C93-9218-6B423B9027EB(a)61.16.207.90.
CSeq: 38322 INVITE.
Authorization: Digest username="test", realm="XXX.XXX.XXX.XX6",
nonce="4433562d730464572c23207912e46d50281d2136",
response="59311eb34dd7ce8b3550064960af1508",
uri="sip:14085264000@XXX.XXX.XXX.XX6".
Max-Forwards: 68.
Content-Type: application/sdp.
User-agent: X-PRO build 1082.
Content-Length: 314.
.
v=0.
o=test 3846000 3846000 IN IP4 61.16.188.4.
s=X-PRO.
c=IN IP4 61.16.188.4.
t=0 0.
m=audio 8000 RTP/AVP 18 0 8 3 98 97 101.
a=rtpmap:0 pcmu/8000.
a=rtpmap:8 pcma/8000.
a=rtpmap:3 gsm/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:98 iLBC/8000.
a=rtpmap:97 speex/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
#
U XXX.XXX.XXX.XX6:5060 -> XXX.XXX.XXX.XX2:5060
ACK sip:10235#14085264000@XXX.XXX.XXX.XX2:5060 SIP/2.0.
Via: SIP/2.0/UDP XXX.XXX.XXX.XX6;branch=z9hG4bK5e7a.83627b27.0.
From: test <sip:test@XXX.XXX.XXX.XX6>;tag=4036363136.
Call-ID: 4D0342A8-6F0D-4C93-9218-6B423B9027EB(a)61.16.207.90.
To: <sip:14085264000@XXX.XXX.XXX.XX6>.
CSeq: 38322 ACK.
User-Agent: OpenSer (1.0.1 (i386/linux)).
Content-Length: 0.
.
exit
5 received, 0 dropped
5 Apr
5 Apr
2:44 a.m.
New subject: [Users] Openser Warning
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi,
I'm getting this Warning from OpenSER (1.1.0-dev16-tls) but all calls
work fine. What does it mean?
Warning: 392 xxx.xxx.xxx.156:5060 "Noisy feedback tells: pid=30117
req_src_ip=xxx.xxx.xxx.143 req_src_port=1467
in_uri=sip:test@xxx.xxx.xxx.156 out_uri=sip:test@xxx.xxx.xxx.130:5061
via_cnt==1"
chris...
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.1 (GNU/Linux)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
iD8DBQFEM2dpR0exH8dhr/YRAnMTAJwPtqzaCmUFqzAA2PLxGi15rJnDQwCgzgpA
wceOSgL9NGmMmwjDG5Kjd3A=
=ICJ0
-----END PGP SIGNATURE-----
3:41 a.m.
New subject: [Users] Openser Warning
It's just a header added by openser to help debuging purpouses. You
can disable this feature adding at the top of your config file
sip_warning=0
and this header will not appear anymore in the processed SIP messages.
Samuel.
2006/4/5, Christoph Fürstaller <christoph.fuerstaller(a)kurtkrenn.com>om>:
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi,
I'm getting this Warning from OpenSER (1.1.0-dev16-tls) but all calls
work fine. What does it mean?
Warning: 392 xxx.xxx.xxx.156:5060 "Noisy feedback tells: pid=30117
req_src_ip=xxx.xxx.xxx.143 req_src_port=1467
in_uri=sip:test@xxx.xxx.xxx.156 out_uri=sip:test@xxx.xxx.xxx.130:5061
via_cnt==1"
chris...
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.1 (GNU/Linux)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
iD8DBQFEM2dpR0exH8dhr/YRAnMTAJwPtqzaCmUFqzAA2PLxGi15rJnDQwCgzgpA
wceOSgL9NGmMmwjDG5Kjd3A=
=ICJ0
-----END PGP SIGNATURE-----
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
3:45 a.m.
New subject: [Users] Openser Warning
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi samuel,
samuel wrote:
It's just a header added by openser to help
debuging purpouses. You
can disable this feature adding at the top of your config file
sip_warning=0
Ok, but what does the warning mean? Don't think it's a good idea to
clean warnings by suppress printing them?
chris...
and this header will not appear anymore in the
processed SIP messages.
Samuel.
2006/4/5, Christoph Fürstaller <christoph.fuerstaller(a)kurtkrenn.com>om>:
Hi,
I'm getting this Warning from OpenSER (1.1.0-dev16-tls) but all calls
work fine. What does it mean?
Warning: 392 xxx.xxx.xxx.156:5060 "Noisy feedback tells: pid=30117
req_src_ip=xxx.xxx.xxx.143 req_src_port=1467
in_uri=sip:test@xxx.xxx.xxx.156 out_uri=sip:test@xxx.xxx.xxx.130:5061
via_cnt==1"
chris...
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.1 (GNU/Linux)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
iD8DBQFEM3WLR0exH8dhr/YRAiPAAJ4t4GBdCpUnH4ZaNMeK8Ge/t5clKgCdEjHh
cSlD8gDGQNgOJ1AVCBX+j/w=
=l3a3
-----END PGP SIGNATURE-----
4:07 a.m.
New subject: [Users] Openser Warning
It's a header that openser adds in the messages it processes to insert
the incoming IP, incoming URI, ougoing IP, and outgoing URI. This
information can be used later from any capture and check where the SIP
message came from and where is it going.
It's not a warning in the sense of that somethign went wrong, it's
ONLY useful for debuging the path of the SIP message.
If you are not using it, you can safely remove it.
Samuel.
2006/4/5, Christoph Fürstaller <christoph.fuerstaller(a)kurtkrenn.com>om>:
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi samuel,
samuel wrote:
It's just a header added by openser to help
debuging purpouses. You
can disable this feature adding at the top of your config file
sip_warning=0
Ok, but what does the warning mean? Don't think it's a good idea to
clean warnings by suppress printing them?
chris...
and this header will not appear anymore in the
processed SIP messages.
Samuel.
2006/4/5, Christoph Fürstaller <christoph.fuerstaller(a)kurtkrenn.com>om>:
Hi,
I'm getting this Warning from OpenSER (1.1.0-dev16-tls) but all calls
work fine. What does it mean?
Warning: 392 xxx.xxx.xxx.156:5060 "Noisy feedback tells: pid=30117
req_src_ip=xxx.xxx.xxx.143 req_src_port=1467
in_uri=sip:test@xxx.xxx.xxx.156 out_uri=sip:test@xxx.xxx.xxx.130:5061
via_cnt==1"
chris...
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.1 (GNU/Linux)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
iD8DBQFEM3WLR0exH8dhr/YRAiPAAJ4t4GBdCpUnH4ZaNMeK8Ge/t5clKgCdEjHh
cSlD8gDGQNgOJ1AVCBX+j/w=
=l3a3
-----END PGP SIGNATURE-----
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
4:35 a.m.
New subject: [Users] Openser Warning
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
samuel wrote:
It's a header that openser adds in the messages it
processes to insert
the incoming IP, incoming URI, ougoing IP, and outgoing URI. This
information can be used later from any capture and check where the SIP
message came from and where is it going.
It's not a warning in the sense of that somethign went wrong, it's
ONLY useful for debuging the path of the SIP message.
If you are not using it, you can safely remove it.
Ah, ok. I understand. Thanks, I'll remove it : )
chris...
Samuel.
2006/4/5, Christoph Fürstaller <christoph.fuerstaller(a)kurtkrenn.com>om>:
Hi samuel,
samuel wrote:
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
It's just a header added by openser to help
debuging purpouses. You
can disable this feature adding at the top of your config file
sip_warning=0
Ok, but what does the warning mean? Don't think it's a good idea to
clean warnings by suppress printing them?
chris...
and this header will not appear anymore in the
processed SIP messages.
Samuel.
2006/4/5, Christoph Fürstaller
<christoph.fuerstaller(a)kurtkrenn.com>om>:
Hi,
I'm getting this Warning from OpenSER
(1.1.0-dev16-tls) but all calls
work fine. What does it mean?
Warning: 392 xxx.xxx.xxx.156:5060 "Noisy
feedback tells: pid=30117
req_src_ip=xxx.xxx.xxx.143 req_src_port=1467
in_uri=sip:test@xxx.xxx.xxx.156 out_uri=sip:test@xxx.xxx.xxx.130:5061
via_cnt==1"
chris...
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.1 (GNU/Linux)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
iD8DBQFEM4FJR0exH8dhr/YRAoN/AKCxp3BbBx4N9ZNXLgKuQCA2T0CvLwCgqG00
/pKVJ76I2VpN9v1Lb84Sc8Q=
=LFPA
-----END PGP SIGNATURE-----
4:56 a.m.
Hi,
my first guess is that this Voice Master looks into the TO uri and not
into RURI. When user dials directly, the TO uri and RURI are the same,
but when you perform the prefix on proxy, the TO is not affected, but
only RURI.
regards,
bogdan
Deepak Singhal wrote:
Hi All,
I am facing a problem when terminating calls to Voice Master.
The exact problem is :
Voice Master is configured for Pin prefix Authentication.
I add the gateway prefix in my openser.cfg using prefix("123#")
But the Voice Master is rejecting the calls with error 488 Not acceptable
here.
User dialing number and openser adding prefix to it.
But when I send the calls from the userend itself with prefix ie. User is
dialing 123#<number> then the calls are going fine. What comes to my mind is
that Voice Master is somehow reading the to_uri being sent. Can I somehow
add prefix to the to_uri as well ?
This is the ngrep trace of the rejected call :
U XXX.XXX.XXX.XX6:5060 -> XXX.XXX.XXX.XX2:5060
INVITE sip:10235#14085264000@XXX.XXX.XXX.XX2:5060 SIP/2.0.
Record-Route: <sip:XXX.XXX.XXX.XX6;ftag=4036363136;lr=on>.
Via: SIP/2.0/UDP XXX.XXX.XXX.XX6;branch=z9hG4bK5e7a.83627b27.0.
Via: SIP/2.0/UDP
61.16.188.4:5060;rport=5060;branch=z9hG4bKE6B692A1F9A1460484EF5F786E105BFE.
From: test <sip:test@XXX.XXX.XXX.XX6>;tag=4036363136.
To: <sip:14085264000@XXX.XXX.XXX.XX6>.
Contact: <sip:test@61.16.188.4:5060>.
Call-ID: 4D0342A8-6F0D-4C93-9218-6B423B9027EB(a)61.16.207.90.
CSeq: 38322 INVITE.
Authorization: Digest
username="test",realm="XXX.XXX.XXX.XX6",nonce="4433562d730464572c23207912e46
d50281d2136",response="59311eb34dd7ce8b3550064960af1508",uri="sip:1408526400
0(a)XXX.XXX.XXX.XX6"quot;.
Max-Forwards: 69.
Content-Type: application/sdp.
User-Agent: X-PRO build 1082.
Content-Length: 314.
.
v=0.
o=test 3846000 3846000 IN IP4 61.16.188.4.
s=X-PRO.
c=IN IP4 61.16.188.4.
t=0 0.
m=audio 8000 RTP/AVP 18 0 8 3 98 97 101.
a=rtpmap:0 pcmu/8000.
a=rtpmap:8 pcma/8000.
a=rtpmap:3 gsm/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:98 iLBC/8000.
a=rtpmap:97 speex/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
#
U XXX.XXX.XXX.XX2:5060 -> XXX.XXX.XXX.XX6:5060
SIP/2.0 100 Trying.
From: <sip:test@XXX.XXX.XXX.XX2:5060>;tag=4036363136.
To: <sip:14085264000@XXX.XXX.XXX.XX2:5060>;tag=669a9b6321e12fb3fd8a82d7.
CSeq: 38322 INVITE.
Call-ID: 4D0342A8-6F0D-4C93-9218-6B423B9027EB(a)61.16.207.90.
Via: SIP/2.0/UDP XXX.XXX.XXX.XX6:5060;branch=z9hG4bK5e7a.83627b27.0.
Via: SIP/2.0/UDP
61.16.188.4:5060;rport=5060;branch=z9hG4bKE6B692A1F9A1460484EF5F786E105BFE.
Content-Length: 0.
.
#
U XXX.XXX.XXX.XX2:5060 -> XXX.XXX.XXX.XX6:5060
SIP/2.0 488 Not Acceptable Here.
Record-Route: <sip:XXX.XXX.XXX.XX6;ftag=4036363136;lr=on>.
Via: SIP/2.0/UDP XXX.XXX.XXX.XX6;branch=z9hG4bK5e7a.83627b27.0.
Via: SIP/2.0/UDP
61.16.188.4:5060;rport=5060;branch=z9hG4bKE6B692A1F9A1460484EF5F786E105BFE.
From: "test" <sip:test@XXX.XXX.XXX.XX6>;tag=4036363136.
To: <sip:14085264000@XXX.XXX.XXX.XX6>.
Contact: <sip:test@61.16.188.4:5060>.
Call-ID: 4D0342A8-6F0D-4C93-9218-6B423B9027EB(a)61.16.207.90.
CSeq: 38322 INVITE.
Authorization: Digest username="test", realm="XXX.XXX.XXX.XX6",
nonce="4433562d730464572c23207912e46d50281d2136",
response="59311eb34dd7ce8b3550064960af1508",
uri="sip:14085264000@XXX.XXX.XXX.XX6".
Max-Forwards: 68.
Content-Type: application/sdp.
User-agent: X-PRO build 1082.
Content-Length: 314.
.
v=0.
o=test 3846000 3846000 IN IP4 61.16.188.4.
s=X-PRO.
c=IN IP4 61.16.188.4.
t=0 0.
m=audio 8000 RTP/AVP 18 0 8 3 98 97 101.
a=rtpmap:0 pcmu/8000.
a=rtpmap:8 pcma/8000.
a=rtpmap:3 gsm/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:98 iLBC/8000.
a=rtpmap:97 speex/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
#
U XXX.XXX.XXX.XX6:5060 -> XXX.XXX.XXX.XX2:5060
ACK sip:10235#14085264000@XXX.XXX.XXX.XX2:5060 SIP/2.0.
Via: SIP/2.0/UDP XXX.XXX.XXX.XX6;branch=z9hG4bK5e7a.83627b27.0.
From: test <sip:test@XXX.XXX.XXX.XX6>;tag=4036363136.
Call-ID: 4D0342A8-6F0D-4C93-9218-6B423B9027EB(a)61.16.207.90.
To: <sip:14085264000@XXX.XXX.XXX.XX6>.
CSeq: 38322 ACK.
User-Agent: OpenSer (1.0.1 (i386/linux)).
Content-Length: 0.
.
exit
5 received, 0 dropped
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
6:06 a.m.
Yes , I also suspected that .. But can I add prefix to to_uri as well ? I
tried finding a way to achieve it .but was unsuccessful.
Deepak Singhal
-----Original Message-----
From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro]
Sent: Wednesday, April 05, 2006 2:27 PM
To: Deepak Singhal
Cc: users(a)openser.org
Subject: Re: [Users] Openser and Voice Master
Hi,
my first guess is that this Voice Master looks into the TO uri and not into
RURI. When user dials directly, the TO uri and RURI are the same, but when
you perform the prefix on proxy, the TO is not affected, but only RURI.
regards,
bogdan
Deepak Singhal wrote:
Hi All,
I am facing a problem when terminating calls to Voice Master.
The exact problem is :
Voice Master is configured for Pin prefix Authentication.
I add the gateway prefix in my openser.cfg using prefix("123#") But the
Voice Master is rejecting the calls with error 488 Not acceptable here.
User dialing number and openser adding prefix to it.
But when I send the calls from the userend itself with prefix ie. User
is dialing 123#<number> then the calls are going fine. What comes to my
mind is that Voice Master is somehow reading the to_uri being sent. Can
I somehow add prefix to the to_uri as well ?
This is the ngrep trace of the rejected call :
U XXX.XXX.XXX.XX6:5060 -> XXX.XXX.XXX.XX2:5060 INVITE
sip:10235#14085264000@XXX.XXX.XXX.XX2:5060 SIP/2.0.
Record-Route: <sip:XXX.XXX.XXX.XX6;ftag=4036363136;lr=on>.
Via: SIP/2.0/UDP XXX.XXX.XXX.XX6;branch=z9hG4bK5e7a.83627b27.0.
Via: SIP/2.0/UDP
61.16.188.4:5060;rport=5060;branch=z9hG4bKE6B692A1F9A1460484EF5F786E105BFE.
From: test <sip:test@XXX.XXX.XXX.XX6>;tag=4036363136.
To: <sip:14085264000@XXX.XXX.XXX.XX6>.
Contact: <sip:test@61.16.188.4:5060>.
Call-ID: 4D0342A8-6F0D-4C93-9218-6B423B9027EB(a)61.16.207.90.
CSeq: 38322 INVITE.
Authorization: Digest
username="test",realm="XXX.XXX.XXX.XX6",nonce="4433562d730464572c232079
12e46
d50281d2136",response="59311eb34dd7ce8b3550064960af1508",uri="sip:14085
26400
0(a)XXX.XXX.XXX.XX6"quot;.
Max-Forwards: 69.
Content-Type: application/sdp.
User-Agent: X-PRO build 1082.
Content-Length: 314.
.
v=0.
o=test 3846000 3846000 IN IP4 61.16.188.4.
s=X-PRO.
c=IN IP4 61.16.188.4.
t=0 0.
m=audio 8000 RTP/AVP 18 0 8 3 98 97 101.
a=rtpmap:0 pcmu/8000.
a=rtpmap:8 pcma/8000.
a=rtpmap:3 gsm/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:98 iLBC/8000.
a=rtpmap:97 speex/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
#
U XXX.XXX.XXX.XX2:5060 -> XXX.XXX.XXX.XX6:5060 SIP/2.0 100 Trying.
From: <sip:test@XXX.XXX.XXX.XX2:5060>;tag=4036363136.
To: <sip:14085264000@XXX.XXX.XXX.XX2:5060>;tag=669a9b6321e12fb3fd8a82d7.
CSeq: 38322 INVITE.
Call-ID: 4D0342A8-6F0D-4C93-9218-6B423B9027EB(a)61.16.207.90.
Via: SIP/2.0/UDP XXX.XXX.XXX.XX6:5060;branch=z9hG4bK5e7a.83627b27.0.
Via: SIP/2.0/UDP
61.16.188.4:5060;rport=5060;branch=z9hG4bKE6B692A1F9A1460484EF5F786E105BFE.
Content-Length: 0.
.
#
U XXX.XXX.XXX.XX2:5060 -> XXX.XXX.XXX.XX6:5060 SIP/2.0 488 Not
Acceptable Here.
Record-Route: <sip:XXX.XXX.XXX.XX6;ftag=4036363136;lr=on>.
Via: SIP/2.0/UDP XXX.XXX.XXX.XX6;branch=z9hG4bK5e7a.83627b27.0.
Via: SIP/2.0/UDP
61.16.188.4:5060;rport=5060;branch=z9hG4bKE6B692A1F9A1460484EF5F786E105BFE.
From: "test" <sip:test@XXX.XXX.XXX.XX6>;tag=4036363136.
To: <sip:14085264000@XXX.XXX.XXX.XX6>.
Contact: <sip:test@61.16.188.4:5060>.
Call-ID: 4D0342A8-6F0D-4C93-9218-6B423B9027EB(a)61.16.207.90.
CSeq: 38322 INVITE.
Authorization: Digest username="test", realm="XXX.XXX.XXX.XX6",
nonce="4433562d730464572c23207912e46d50281d2136",
response="59311eb34dd7ce8b3550064960af1508",
uri="sip:14085264000@XXX.XXX.XXX.XX6".
Max-Forwards: 68.
Content-Type: application/sdp.
User-agent: X-PRO build 1082.
Content-Length: 314.
.
v=0.
o=test 3846000 3846000 IN IP4 61.16.188.4.
s=X-PRO.
c=IN IP4 61.16.188.4.
t=0 0.
m=audio 8000 RTP/AVP 18 0 8 3 98 97 101.
a=rtpmap:0 pcmu/8000.
a=rtpmap:8 pcma/8000.
a=rtpmap:3 gsm/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:98 iLBC/8000.
a=rtpmap:97 speex/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
#
U XXX.XXX.XXX.XX6:5060 -> XXX.XXX.XXX.XX2:5060 ACK
sip:10235#14085264000@XXX.XXX.XXX.XX2:5060 SIP/2.0.
Via: SIP/2.0/UDP XXX.XXX.XXX.XX6;branch=z9hG4bK5e7a.83627b27.0.
From: test <sip:test@XXX.XXX.XXX.XX6>;tag=4036363136.
Call-ID: 4D0342A8-6F0D-4C93-9218-6B423B9027EB(a)61.16.207.90.
To: <sip:14085264000@XXX.XXX.XXX.XX6>.
CSeq: 38322 ACK.
User-Agent: OpenSer (1.0.1 (i386/linux)).
Content-Length: 0.
.
exit
5 received, 0 dropped
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
2:05 p.m.
Hi,
no, you cannot change TO uri without breaking things. Is not advisable
to do so, no to mention that routing based on TO uri is obsolete for a
long time ago.
regards,
bogdan
Deepak Singhal wrote:
Yes , I also suspected that .. But can I add prefix to
to_uri as well ? I
tried finding a way to achieve it .but was unsuccessful.
Deepak Singhal
-----Original Message-----
From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro]
Sent: Wednesday, April 05, 2006 2:27 PM
To: Deepak Singhal
Cc: users(a)openser.org
Subject: Re: [Users] Openser and Voice Master
Hi,
my first guess is that this Voice Master looks into the TO uri and not into
RURI. When user dials directly, the TO uri and RURI are the same, but when
you perform the prefix on proxy, the TO is not affected, but only RURI.
regards,
bogdan
Deepak Singhal wrote:
Hi All,
I am facing a problem when terminating calls to Voice Master.
The exact problem is :
Voice Master is configured for Pin prefix Authentication.
I add the gateway prefix in my openser.cfg using prefix("123#") But the
Voice Master is rejecting the calls with error 488 Not acceptable here.
User dialing number and openser adding prefix to it.
But when I send the calls from the userend itself with prefix ie. User
is dialing 123#<number> then the calls are going fine. What comes to my
mind is that Voice Master is somehow reading the to_uri being sent. Can
I somehow add prefix to the to_uri as well ?
This is the ngrep trace of the rejected call :
U XXX.XXX.XXX.XX6:5060 -> XXX.XXX.XXX.XX2:5060 INVITE
sip:10235#14085264000@XXX.XXX.XXX.XX2:5060 SIP/2.0.
Record-Route: <sip:XXX.XXX.XXX.XX6;ftag=4036363136;lr=on>.
Via: SIP/2.0/UDP XXX.XXX.XXX.XX6;branch=z9hG4bK5e7a.83627b27.0.
Via: SIP/2.0/UDP
61.16.188.4:5060;rport=5060;branch=z9hG4bKE6B692A1F9A1460484EF5F786E105BFE.
From: test <sip:test@XXX.XXX.XXX.XX6>;tag=4036363136.
To: <sip:14085264000@XXX.XXX.XXX.XX6>.
Contact: <sip:test@61.16.188.4:5060>.
Call-ID: 4D0342A8-6F0D-4C93-9218-6B423B9027EB(a)61.16.207.90.
CSeq: 38322 INVITE.
Authorization: Digest
username="test",realm="XXX.XXX.XXX.XX6",nonce="4433562d730464572c232079
12e46
d50281d2136",response="59311eb34dd7ce8b3550064960af1508",uri="sip:14085
26400
0(a)XXX.XXX.XXX.XX6"quot;.
Max-Forwards: 69.
Content-Type: application/sdp.
User-Agent: X-PRO build 1082.
Content-Length: 314.
.
v=0.
o=test 3846000 3846000 IN IP4 61.16.188.4.
s=X-PRO.
c=IN IP4 61.16.188.4.
t=0 0.
m=audio 8000 RTP/AVP 18 0 8 3 98 97 101.
a=rtpmap:0 pcmu/8000.
a=rtpmap:8 pcma/8000.
a=rtpmap:3 gsm/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:98 iLBC/8000.
a=rtpmap:97 speex/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
#
U XXX.XXX.XXX.XX2:5060 -> XXX.XXX.XXX.XX6:5060 SIP/2.0 100 Trying.
From: <sip:test@XXX.XXX.XXX.XX2:5060>;tag=4036363136.
To: <sip:14085264000@XXX.XXX.XXX.XX2:5060>;tag=669a9b6321e12fb3fd8a82d7.
CSeq: 38322 INVITE.
Call-ID: 4D0342A8-6F0D-4C93-9218-6B423B9027EB(a)61.16.207.90.
Via: SIP/2.0/UDP XXX.XXX.XXX.XX6:5060;branch=z9hG4bK5e7a.83627b27.0.
Via: SIP/2.0/UDP
61.16.188.4:5060;rport=5060;branch=z9hG4bKE6B692A1F9A1460484EF5F786E105BFE.
Content-Length: 0.
.
#
U XXX.XXX.XXX.XX2:5060 -> XXX.XXX.XXX.XX6:5060 SIP/2.0 488 Not
Acceptable Here.
Record-Route: <sip:XXX.XXX.XXX.XX6;ftag=4036363136;lr=on>.
Via: SIP/2.0/UDP XXX.XXX.XXX.XX6;branch=z9hG4bK5e7a.83627b27.0.
Via: SIP/2.0/UDP
61.16.188.4:5060;rport=5060;branch=z9hG4bKE6B692A1F9A1460484EF5F786E105BFE.
From: "test" <sip:test@XXX.XXX.XXX.XX6>;tag=4036363136.
To: <sip:14085264000@XXX.XXX.XXX.XX6>.
Contact: <sip:test@61.16.188.4:5060>.
Call-ID: 4D0342A8-6F0D-4C93-9218-6B423B9027EB(a)61.16.207.90.
CSeq: 38322 INVITE.
Authorization: Digest username="test", realm="XXX.XXX.XXX.XX6",
nonce="4433562d730464572c23207912e46d50281d2136",
response="59311eb34dd7ce8b3550064960af1508",
uri="sip:14085264000@XXX.XXX.XXX.XX6".
Max-Forwards: 68.
Content-Type: application/sdp.
User-agent: X-PRO build 1082.
Content-Length: 314.
.
v=0.
o=test 3846000 3846000 IN IP4 61.16.188.4.
s=X-PRO.
c=IN IP4 61.16.188.4.
t=0 0.
m=audio 8000 RTP/AVP 18 0 8 3 98 97 101.
a=rtpmap:0 pcmu/8000.
a=rtpmap:8 pcma/8000.
a=rtpmap:3 gsm/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:98 iLBC/8000.
a=rtpmap:97 speex/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
#
U XXX.XXX.XXX.XX6:5060 -> XXX.XXX.XXX.XX2:5060 ACK
sip:10235#14085264000@XXX.XXX.XXX.XX2:5060 SIP/2.0.
Via: SIP/2.0/UDP XXX.XXX.XXX.XX6;branch=z9hG4bK5e7a.83627b27.0.
From: test <sip:test@XXX.XXX.XXX.XX6>;tag=4036363136.
Call-ID: 4D0342A8-6F0D-4C93-9218-6B423B9027EB(a)61.16.207.90.
To: <sip:14085264000@XXX.XXX.XXX.XX6>.
CSeq: 38322 ACK.
User-Agent: OpenSer (1.0.1 (i386/linux)).
Content-Length: 0.
.
exit
5 received, 0 dropped
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
6857
days inactive
6857
days old
8 comments
4 participants
participants (4)
-
Bogdan-Andrei Iancu -
Christoph Fürstaller -
Deepak Singhal -
samuel