Hello Guys,
Just like to request assistance in trying to figure out how can I route the call from SER as seen on TO header. Below is the snippet of the sip log:
0(20457) DEBUG: get_hdr_field: <To> [34]; uri=[sip:8810844@24.90.219.179 :8700] 0(20457) DEBUG: to body [sip:8810844@24.90.219.179:8700 ] 0(20457) get_hdr_field: cseq <CSeq>: <1154241348> <INVITE> 0(20457) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(20457) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 0(20457) check_via_address(10.10.10.21, 10.10.10.21, 0) 0(20457) Sending: INVITE sip:8810844@10.10.10.86 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.86;branch=0 Via: SIP/2.0/UDP 10.10.10.21;branch=z9hG4bK13666f91365343 From: sip:2589@mandela;tag=cba-0094-44cc5343 To: sip:8810844@24.90.219.179:8700 Call-ID: 317e120dd2385173-0094-44cc5343-282c@10.10.10.21 CSeq: 1154241348 INVITE Contact: sip:2589@10.10.10.21 Date: Sun, 30 Jul 2006 06:35:47 GMT User-Agent: BRSIP v2.0.0.11 Max-Forwards: 16 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY Allow-Events: keep-alive, message-summary Supported: timer Session-Expires: 1800 Min-SE: 600 Expires: 300 Content-Type: application/sdp Content-Length: 220
v=0 o=BRSDP 177 177 IN IP4 10.10.10.21 s=BRSDP Session c=IN IP4 63.116.254.21 t=0 0 m=audio 15000 RTP/AVP 4 18 101 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15
I need SER to send the call based on the TO HEADER URI seen on get_hdr_field. This value changes depending on what another sip proxy is sending to the SER. The t_relay is not working as i like it to behave. Any help is greatly appreciated. Thanks in advance.
Hello Guys,
Just like to request assistance in trying to figure out how can I route the call from SER as seen on TO header. Below is the snippet of the sip log:
0(20457) DEBUG: get_hdr_field: <To> [34]; uri=[ sip:8810844@24.90.219.179 :8700] 0(20457) DEBUG: to body [sip:8810844@24.90.219.179:8700 ] 0(20457) get_hdr_field: cseq <CSeq>: <1154241348> <INVITE> 0(20457) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(20457) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 0(20457) check_via_address(10.10.10.21, 10.10.10.21, 0) 0(20457) Sending: INVITE sip:8810844@10.10.10.86 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.86;branch=0 Via: SIP/2.0/UDP 10.10.10.21 ;branch=z9hG4bK13666f91365343 From: sip:2589@mandela;tag=cba-0094-44cc5343 To: sip:8810844@24.90.219.179:8700 Call-ID: 317e120dd2385173-0094-44cc5343-282c@10.10.10.21 CSeq: 1154241348 INVITE Contact: sip:2589@10.10.10.21 Date: Sun, 30 Jul 2006 06:35:47 GMT User-Agent: BRSIP v2.0.0.11 Max-Forwards: 16 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY Allow-Events: keep-alive, message-summary Supported: timer Session-Expires: 1800 Min-SE: 600 Expires: 300 Content-Type: application/sdp Content-Length: 220
v=0 o=BRSDP 177 177 IN IP4 10.10.10.21 s=BRSDP Session c=IN IP4 10.10.10.21 http://63.116.254.21 t=0 0 m=audio 15000 RTP/AVP 4 18 101 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15
I need SER to send the call based on the TO HEADER URI seen on get_hdr_field. This value changes depending on what another sip proxy is sending to the SER. The t_relay is not working as i like it to behave. Any help is greatly appreciated. Thanks in advance.
Hi Mark!
First, routing based on the To header violates the RFC3261, and is usually an indicator that either your setup or your SIP hardware is non-optimal.
If you still want to do it, you can copy the URI in the To header in an AVP (check the list of available pseudo variables, there is a pseudo variable for the To URI I guess) with avp_write. Then push this AVP into the request URI using avp_pushto.
regards klaus
Mark Anthony C. Delfin wrote:
Hello Guys,
Just like to request assistance in trying to figure out how can I route the call from SER as seen on TO header. Below is the snippet of the sip log:
0(20457) DEBUG: get_hdr_field: <To> [34]; uri=[ sip:8810844@24.90.219.179:8700] 0(20457) DEBUG: to body [sip:8810844@24.90.219.179:8700 ] 0(20457) get_hdr_field: cseq <CSeq>: <1154241348> <INVITE> 0(20457) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(20457) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 0(20457) check_via_address(10.10.10.21 http://10.10.10.21, 10.10.10.21 http://10.10.10.21, 0) 0(20457) Sending: INVITE sip:8810844@10.10.10.86 mailto:sip:8810844@10.10.10.86 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.86 http://10.10.10.86;branch=0 Via: SIP/2.0/UDP 10.10.10.21 http://10.10.10.21 ;branch=z9hG4bK13666f91365343 From: sip:2589@mandela;tag=cba-0094-44cc5343 To: sip:8810844@24.90.219.179:8700 Call-ID: 317e120dd2385173-0094-44cc5343-282c@10.10.10.21 mailto:317e120dd2385173-0094-44cc5343-282c@10.10.10.21 CSeq: 1154241348 INVITE Contact: <sip:2589@10.10.10.21 mailto:sip:2589@10.10.10.21> Date: Sun, 30 Jul 2006 06:35:47 GMT User-Agent: BRSIP v2.0.0.11 Max-Forwards: 16 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY Allow-Events: keep-alive, message-summary Supported: timer Session-Expires: 1800 Min-SE: 600 Expires: 300 Content-Type: application/sdp Content-Length: 220
v=0 o=BRSDP 177 177 IN IP4 10.10.10.21 http://10.10.10.21 s=BRSDP Session c=IN IP4 10.10.10.21 http://63.116.254.21 t=0 0 m=audio 15000 RTP/AVP 4 18 101 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15
I need SER to send the call based on the TO HEADER URI seen on get_hdr_field. This value changes depending on what another sip proxy is sending to the SER. The t_relay is not working as i like it to behave. Any help is greatly appreciated. Thanks in advance.
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
Hello Klaus,
Thanks for the reply. Is it possible to rewrite the request URI based from the TO header for calls to be forwarded there?
On 7/31/06, Klaus Darilion klaus.mailinglists@pernau.at wrote:
Hi Mark!
First, routing based on the To header violates the RFC3261, and is usually an indicator that either your setup or your SIP hardware is non-optimal.
If you still want to do it, you can copy the URI in the To header in an AVP (check the list of available pseudo variables, there is a pseudo variable for the To URI I guess) with avp_write. Then push this AVP into the request URI using avp_pushto.
regards klaus
Mark Anthony C. Delfin wrote:
Hello Guys,
Just like to request assistance in trying to figure out how can I route the call from SER as seen on TO header. Below is the snippet of the sip
log:
0(20457) DEBUG: get_hdr_field: <To> [34]; uri=[ sip:8810844@24.90.219.179:8700] 0(20457) DEBUG: to body [sip:8810844@24.90.219.179:8700 ] 0(20457) get_hdr_field: cseq <CSeq>: <1154241348> <INVITE> 0(20457) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(20457) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 0(20457) check_via_address(10.10.10.21 http://10.10.10.21, 10.10.10.21 http://10.10.10.21, 0) 0(20457) Sending: INVITE sip:8810844@10.10.10.86 mailto:sip:8810844@10.10.10.86 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.86 http://10.10.10.86;branch=0 Via: SIP/2.0/UDP 10.10.10.21 http://10.10.10.21 ;branch=z9hG4bK13666f91365343 From: sip:2589@mandela;tag=cba-0094-44cc5343 To: sip:8810844@24.90.219.179:8700 Call-ID: 317e120dd2385173-0094-44cc5343-282c@10.10.10.21 mailto:317e120dd2385173-0094-44cc5343-282c@10.10.10.21 CSeq: 1154241348 INVITE Contact: <sip:2589@10.10.10.21 mailto:sip:2589@10.10.10.21> Date: Sun, 30 Jul 2006 06:35:47 GMT User-Agent: BRSIP v2.0.0.11 Max-Forwards: 16 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY Allow-Events: keep-alive, message-summary Supported: timer Session-Expires: 1800 Min-SE: 600 Expires: 300 Content-Type: application/sdp Content-Length: 220
v=0 o=BRSDP 177 177 IN IP4 10.10.10.21 http://10.10.10.21 s=BRSDP Session c=IN IP4 10.10.10.21 http://63.116.254.21 t=0 0 m=audio 15000 RTP/AVP 4 18 101 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15
I need SER to send the call based on the TO HEADER URI seen on get_hdr_field. This value changes depending on what another sip proxy is sending to the SER. The t_relay is not working as i like it to behave. Any help is greatly appreciated. Thanks in advance.
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
Mark,
I think that is what Klaus was suggesting.
First, copy the 'To' URI into an AVP (this holds the value) Then, copy the AVP to the RURI (request URI). It's a two step process.
I agree with Klaus, this is weird :-) But, entirely doable.
-g
On 7/31/06, Mark Anthony C. Delfin markanthonycdelfin@gmail.com wrote:
Hello Klaus,
Thanks for the reply. Is it possible to rewrite the request URI based from the TO header for calls to be forwarded there?
On 7/31/06, Klaus Darilion klaus.mailinglists@pernau.at wrote:
Hi Mark!
First, routing based on the To header violates the RFC3261, and is usually an indicator that either your setup or your SIP hardware is non-optimal.
If you still want to do it, you can copy the URI in the To header in an AVP (check the list of available pseudo variables, there is a pseudo variable for the To URI I guess) with avp_write. Then push this AVP into the request URI using avp_pushto.
regards klaus
Mark Anthony C. Delfin wrote:
Hello Guys,
Just like to request assistance in trying to figure out how can I route the call from SER as seen on TO header. Below is the snippet of the sip
log:
0(20457) DEBUG: get_hdr_field: <To> [34]; uri=[ sip:8810844@24.90.219.179:8700] 0(20457) DEBUG: to body [sip:8810844@24.90.219.179:8700 ] 0(20457) get_hdr_field: cseq <CSeq>: <1154241348> <INVITE> 0(20457) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(20457) DBG:maxfwd:process_maxfwd_header: value 70
decreased to 16
0(20457) check_via_address(10.10.10.21 http://10.10.10.21, 10.10.10.21 http://10.10.10.21, 0) 0(20457) Sending: INVITE sip:8810844@10.10.10.86 mailto:sip:8810844@10.10.10.86 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.86 <http://10.10.10.86 >;branch=0 Via: SIP/2.0/UDP 10.10.10.21 http://10.10.10.21 ;branch=z9hG4bK13666f91365343 From: sip:2589@mandela;tag=cba-0094-44cc5343 To: sip:8810844@24.90.219.179:8700 Call-ID:
317e120dd2385173-0094-44cc5343-282c@10.10.10.21
<mailto:
317e120dd2385173-0094-44cc5343-282c@10.10.10.21>
CSeq: 1154241348 INVITE Contact: <sip:2589@10.10.10.21 <mailto:sip: 2589@10.10.10.21>> Date: Sun, 30 Jul 2006 06:35:47 GMT User-Agent: BRSIP v2.0.0.11 Max-Forwards: 16 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY Allow-Events: keep-alive, message-summary Supported: timer Session-Expires: 1800 Min-SE: 600 Expires: 300 Content-Type: application/sdp Content-Length: 220
v=0 o=BRSDP 177 177 IN IP4 10.10.10.21 http://10.10.10.21 s=BRSDP Session c=IN IP4 10.10.10.21 <http://63.116.254.21 > t=0 0 m=audio 15000 RTP/AVP 4 18 101 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15
I need SER to send the call based on the TO HEADER URI seen on get_hdr_field. This value changes depending on what another sip proxy is sending to the SER. The t_relay is not working as i like it to behave. Any help is greatly appreciated. Thanks in advance.
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
Hello Again,
Is it possible to use AVP with DB integration? That is, storing the variable to a file on the server. Thanks.
On 7/31/06, Greg Fausak lgfausak@gmail.com wrote:
Mark,
I think that is what Klaus was suggesting.
First, copy the 'To' URI into an AVP (this holds the value) Then, copy the AVP to the RURI (request URI). It's a two step process.
I agree with Klaus, this is weird :-) But, entirely doable.
-g
On 7/31/06, Mark Anthony C. Delfin markanthonycdelfin@gmail.com wrote:
Hello Klaus,
Thanks for the reply. Is it possible to rewrite the request URI based
from
the TO header for calls to be forwarded there?
On 7/31/06, Klaus Darilion klaus.mailinglists@pernau.at wrote:
Hi Mark!
First, routing based on the To header violates the RFC3261, and is usually an indicator that either your setup or your SIP hardware is non-optimal.
If you still want to do it, you can copy the URI in the To header in
an
AVP (check the list of available pseudo variables, there is a pseudo variable for the To URI I guess) with avp_write. Then push this AVP
into
the request URI using avp_pushto.
regards klaus
Mark Anthony C. Delfin wrote:
Hello Guys,
Just like to request assistance in trying to figure out how can I
route
the call from SER as seen on TO header. Below is the snippet of the
sip
log:
0(20457) DEBUG: get_hdr_field: <To> [34]; uri=[ sip:8810844@24.90.219.179:8700] 0(20457) DEBUG: to body [sip:8810844@24.90.219.179:8700 ] 0(20457) get_hdr_field: cseq <CSeq>: <1154241348> <INVITE> 0(20457) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(20457) DBG:maxfwd:process_maxfwd_header: value 70
decreased to 16
0(20457) check_via_address(10.10.10.21 http://10.10.10.21, 10.10.10.21 http://10.10.10.21, 0) 0(20457) Sending: INVITE sip:8810844@10.10.10.86 mailto:sip:8810844@10.10.10.86
SIP/2.0
Via: SIP/2.0/UDP 10.10.10.86 <http://10.10.10.86 >;branch=0 Via: SIP/2.0/UDP 10.10.10.21 http://10.10.10.21 ;branch=z9hG4bK13666f91365343 From: sip:2589@mandela;tag=cba-0094-44cc5343 To: sip:8810844@24.90.219.179:8700 Call-ID:
317e120dd2385173-0094-44cc5343-282c@10.10.10.21
<mailto:
317e120dd2385173-0094-44cc5343-282c@10.10.10.21>
CSeq: 1154241348 INVITE Contact: <sip:2589@10.10.10.21 <mailto:sip: 2589@10.10.10.21>> Date: Sun, 30 Jul 2006 06:35:47 GMT User-Agent: BRSIP v2.0.0.11 Max-Forwards: 16 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY Allow-Events: keep-alive, message-summary Supported: timer Session-Expires: 1800 Min-SE: 600 Expires: 300 Content-Type: application/sdp Content-Length: 220
v=0 o=BRSDP 177 177 IN IP4 10.10.10.21 http://10.10.10.21 s=BRSDP Session c=IN IP4 10.10.10.21 <http://63.116.254.21 > t=0 0 m=audio 15000 RTP/AVP 4 18 101 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15
I need SER to send the call based on the TO HEADER URI seen on get_hdr_field. This value changes depending on what another sip
proxy is
sending to the SER. The t_relay is not working as i like it to
behave.
Any help is greatly appreciated. Thanks in advance.
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
-- Greg Fausak greg@thursday.com
On Wed, August 2, 2006 6:00, Mark Anthony C. Delfin said:
Hello Again,
Is it possible to use AVP with DB integration? That is, storing the variable to a file on the server. Thanks.
yes. http://www.openser.org/docs/modules/1.1.x/avpops.html
regards klaus
On 7/31/06, Greg Fausak lgfausak@gmail.com wrote:
Mark,
I think that is what Klaus was suggesting.
First, copy the 'To' URI into an AVP (this holds the value) Then, copy the AVP to the RURI (request URI). It's a two step process.
I agree with Klaus, this is weird :-) But, entirely doable.
-g
On 7/31/06, Mark Anthony C. Delfin markanthonycdelfin@gmail.com wrote:
Hello Klaus,
Thanks for the reply. Is it possible to rewrite the request URI based
from
the TO header for calls to be forwarded there?
On 7/31/06, Klaus Darilion klaus.mailinglists@pernau.at wrote:
Hi Mark!
First, routing based on the To header violates the RFC3261, and is usually an indicator that either your setup or your SIP hardware is non-optimal.
If you still want to do it, you can copy the URI in the To header in
an
AVP (check the list of available pseudo variables, there is a pseudo variable for the To URI I guess) with avp_write. Then push this AVP
into
the request URI using avp_pushto.
regards klaus
Mark Anthony C. Delfin wrote:
Hello Guys,
Just like to request assistance in trying to figure out how can I
route
the call from SER as seen on TO header. Below is the snippet of
the sip
log:
0(20457) DEBUG: get_hdr_field: <To> [34]; uri=[ sip:8810844@24.90.219.179:8700] 0(20457) DEBUG: to body [sip:8810844@24.90.219.179:8700 ] 0(20457) get_hdr_field: cseq <CSeq>: <1154241348> <INVITE> 0(20457) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(20457) DBG:maxfwd:process_maxfwd_header: value 70
decreased to 16
0(20457) check_via_address(10.10.10.21 http://10.10.10.21, 10.10.10.21 http://10.10.10.21, 0) 0(20457) Sending: INVITE sip:8810844@10.10.10.86 mailto:sip:8810844@10.10.10.86
SIP/2.0
Via: SIP/2.0/UDP 10.10.10.86 <http://10.10.10.86 >;branch=0 Via: SIP/2.0/UDP 10.10.10.21 http://10.10.10.21 ;branch=z9hG4bK13666f91365343 From: sip:2589@mandela;tag=cba-0094-44cc5343 To: sip:8810844@24.90.219.179:8700 Call-ID:
317e120dd2385173-0094-44cc5343-282c@10.10.10.21
<mailto:
317e120dd2385173-0094-44cc5343-282c@10.10.10.21>
CSeq: 1154241348 INVITE Contact: <sip:2589@10.10.10.21 <mailto:sip: 2589@10.10.10.21>> Date: Sun, 30 Jul 2006 06:35:47 GMT User-Agent: BRSIP v2.0.0.11 Max-Forwards: 16 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY Allow-Events: keep-alive, message-summary Supported: timer Session-Expires: 1800 Min-SE: 600 Expires: 300 Content-Type: application/sdp Content-Length: 220
v=0 o=BRSDP 177 177 IN IP4 10.10.10.21 http://10.10.10.21 s=BRSDP Session c=IN IP4 10.10.10.21 <http://63.116.254.21 > t=0 0 m=audio 15000 RTP/AVP 4 18 101 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15
I need SER to send the call based on the TO HEADER URI seen on get_hdr_field. This value changes depending on what another sip
proxy is
sending to the SER. The t_relay is not working as i like it to
behave.
Any help is greatly appreciated. Thanks in advance.
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
-- Greg Fausak greg@thursday.com
-
Greg Fausak -
Klaus Darilion -
Mark Anthony C. Delfin