[Serusers] Force SER to send calls using TO header

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Mark Anthony C. Delfin

31 Jul 2006 31 Jul '06

1:39 a.m.

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(a)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.

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Mark Anthony C. Delfin

31 Jul 31 Jul

3:11 a.m.

New subject: [Users] Force SER to send calls using TO header

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(a)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.

Klaus Darilion

9:43 a.m.

New subject: [Users] Force SER to send calls using TO header

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(a)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(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

Mark Anthony C. Delfin

11:54 p.m.

New subject: [Users] Force SER to send calls using TO header

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(a)pernau.at> 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(a)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(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

Greg Fausak

1 Aug 1 Aug

2:29 a.m.

New subject: [Users] Force SER to send calls using TO header

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(a)gmail.com> wrote:

...

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(a)10.10.10.21

> <mailto:

317e120dd2385173-0094-44cc5343-282c(a)10.10.10.21>

> CSeq: 1154241348 INVITE > Contact: <sip:2589@10.10.10.21 <mailto:sip: 2589(a)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(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

-- Greg Fausak greg(a)thursday.com

Mark Anthony C. Delfin

2 Aug 2 Aug

6 a.m.

New subject: [Users] Force SER to send calls using TO header

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(a)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(a)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

> 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:

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(a)10.10.10.21

> <mailto:

317e120dd2385173-0094-44cc5343-282c(a)10.10.10.21> > > CSeq: 1154241348 INVITE > > Contact: <sip:2589@10.10.10.21 <mailto:sip: 2589(a)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(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

-- Greg Fausak greg(a)thursday.com

Klaus Darilion

7:48 a.m.

New subject: [Users] Force SER to send calls using TO header

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(a)gmail.com> wrote:

Hello Klaus, Thanks for the reply. Is it possible to rewrite the request URI based

from

> 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:

SIP/2.0

317e120dd2385173-0094-44cc5343-282c(a)10.10.10.21

> <mailto:

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(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users

_______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

-- Greg Fausak greg(a)thursday.com

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Greg Fausak
Klaus Darilion
Mark Anthony C. Delfin