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@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@XXX.XXX.XXX.XX6". 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@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@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@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
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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...
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@kurtkrenn.com:
-----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
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Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
-----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@kurtkrenn.com:
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@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
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@kurtkrenn.com:
-----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@kurtkrenn.com:
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@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@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
-----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@kurtkrenn.com:
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@kurtkrenn.com:
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@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
_______________________________________________ Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
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@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@XXX.XXX.XXX.XX6". 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@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@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@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@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
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@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@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@XXX.XXX.XXX.XX6". 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@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@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@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@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
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@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@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@XXX.XXX.XXX.XX6". 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@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@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@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
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