Hi All, I've recently stumbled upon this little hitch while using kamailio with topoh module that the Contact header do not contains the User part for the 302 Moved temporarily packet.
*My topology:* UserA<====\ ->Kamailio<===>FreeSwitch UserB<====/
The B party has set Call forwarding on their phone hence phone sends a 302 Moved Temporarily to Kamailio.
Via: SIP/2.0/UDP TOPOH.KAMAILIO.IP:5060;branch=z9hG4bK0767.4e4e57deca89da5c5f5ae97228325f85.0 Via: SIP/2.0/UDP TOPOH.KAMAILIO.IP;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.y7MGZLMGZwM.1LgRWIC9gIgx4fMGZAOBVAOBNfzuaVHRaYpB1LNSQLpx4uMx3Az6eL3RsBCxu-zRrUWSeOgjeBk.IVm4ds34aONc** From: "+4319714111" sip:+4319714111@FREESWITCH.IP.HERE;tag=yUr5UZ7eF794K To: sip:502@USER.B.IP.HERE:5060;tag=105223296 Call-ID: cd811276-4b4d-1234-66ae-005056867dbc CSeq: 87257355 INVITE Contact: sip:06606017597@FREESWITCH.IP.HERE:5060 User-Agent: Yealink SIP-T46G 28.80.0.70 Diversion: sip:502@USER.B.IP.HERE:5060;reason=unconditional Content-Length: 0
This is modified in Kamailio TOPOH and sent to FreeSwitch as following
SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 10.0.20.71;received=10.0.20.71;rport=5060;branch=z9hG4bKeBcy9HKK9cDKr From: "+4319714111" sip:+4319714111@FREESWITCH.IP.HERE;tag=yUr5UZ7eF794K To: sip:502@USER.B.IP.HERE:5060;tag=105223296 Call-ID: cd811276-4b4d-1234-66ae-005056867dbc CSeq: 87257355 INVITE Contact: <*sip:TOPOH.KAMAILIO.IP* ;line=sr-N6IAzBy6WBy6MxFwW.qwPSW5ohWINhaYNLu4g9W4OhWI3wKLgRsIpUg5kGs7g9P-W.y6My**> User-Agent: Yealink SIP-T46G 28.80.0.70 Diversion: sip:502@USER.B.IP.HERE:5060;reason=unconditional Content-Length: 0
Which results in a Call Originate from FreeSwitch with RURI as this:
INVITE *sip:TOPOH.KAMAILIO.IP*;line=sr-N6IAzBy6WBy6MxFwW.qwPSW5ohWINhaYNLu4g9W4OhWI3wKLgRsIpUg5kGs7g9P-W.y6My** SIP/2.0
I am thinking that in TOPOH Module some patch is required to atleast retain the $rU for 3XX replies , not sure if this will break some RFC or Kamailio stability etc !
Thanks, Sammy.
Yep, got same problem. Not sure it’s a bug, but seems to be topoh module is really lack some configuration.
2016-02-12 21:13 GMT+02:00 SamyGo govoiper@gmail.com:
Hi All, I've recently stumbled upon this little hitch while using kamailio with topoh module that the Contact header do not contains the User part for the 302 Moved temporarily packet.
*My topology:* UserA<====\ ->Kamailio<===>FreeSwitch UserB<====/
The B party has set Call forwarding on their phone hence phone sends a 302 Moved Temporarily to Kamailio.
Via: SIP/2.0/UDP TOPOH.KAMAILIO.IP:5060;branch=z9hG4bK0767.4e4e57deca89da5c5f5ae97228325f85.0 Via: SIP/2.0/UDP TOPOH.KAMAILIO.IP;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.y7MGZLMGZwM.1LgRWIC9gIgx4fMGZAOBVAOBNfzuaVHRaYpB1LNSQLpx4uMx3Az6eL3RsBCxu-zRrUWSeOgjeBk.IVm4ds34aONc** From: "+4319714111" sip:+4319714111@FREESWITCH.IP.HERE;tag=yUr5UZ7eF794K To: sip:502@USER.B.IP.HERE:5060;tag=105223296 Call-ID: cd811276-4b4d-1234-66ae-005056867dbc CSeq: 87257355 INVITE Contact: sip:06606017597@FREESWITCH.IP.HERE:5060 User-Agent: Yealink SIP-T46G 28.80.0.70 Diversion: sip:502@USER.B.IP.HERE:5060;reason=unconditional Content-Length: 0
This is modified in Kamailio TOPOH and sent to FreeSwitch as following
SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 10.0.20.71;received=10.0.20.71;rport=5060;branch=z9hG4bKeBcy9HKK9cDKr From: "+4319714111" sip:+4319714111@FREESWITCH.IP.HERE;tag=yUr5UZ7eF794K To: sip:502@USER.B.IP.HERE:5060;tag=105223296 Call-ID: cd811276-4b4d-1234-66ae-005056867dbc CSeq: 87257355 INVITE Contact: <*sip:TOPOH.KAMAILIO.IP* ;line=sr-N6IAzBy6WBy6MxFwW.qwPSW5ohWINhaYNLu4g9W4OhWI3wKLgRsIpUg5kGs7g9P-W.y6My**> User-Agent: Yealink SIP-T46G 28.80.0.70 Diversion: sip:502@USER.B.IP.HERE:5060;reason=unconditional Content-Length: 0
Which results in a Call Originate from FreeSwitch with RURI as this:
INVITE *sip:TOPOH.KAMAILIO.IP*;line=sr-N6IAzBy6WBy6MxFwW.qwPSW5ohWINhaYNLu4g9W4OhWI3wKLgRsIpUg5kGs7g9P-W.y6My** SIP/2.0
I am thinking that in TOPOH Module some patch is required to atleast retain the $rU for 3XX replies , not sure if this will break some RFC or Kamailio stability etc !
Thanks, Sammy.
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hello,
isn't the username then decoded from the value of
sip:TOPOH.KAMAILIO.IP;line=sr-N6IAzBy6WBy6MxFwW.qwPSW5ohWINhaYNLu4g9W4OhWI3wKLgRsIpUg5kGs7g9P-W.y6My**
In other words, when the follow up invite in Kamailio comes, can you print the value of $ru with xlog and see if it has username?
Cheers, Daniel
On 16/02/16 22:54, Igor Olhovskiy wrote:
Yep, got same problem. Not sure it’s a bug, but seems to be topoh module is really lack some configuration.
2016-02-12 21:13 GMT+02:00 SamyGo <govoiper@gmail.com mailto:govoiper@gmail.com>:
Hi All, I've recently stumbled upon this little hitch while using kamailio with topoh module that the Contact header do not contains the User part for the 302 Moved temporarily packet. _My topology:_ UserA<====\ ->Kamailio<===>FreeSwitch UserB<====/ The B party has set Call forwarding on their phone hence phone sends a 302 Moved Temporarily to Kamailio. Via: SIP/2.0/UDP TOPOH.KAMAILIO.IP:5060;branch=z9hG4bK0767.4e4e57deca89da5c5f5ae97228325f85.0 Via: SIP/2.0/UDP TOPOH.KAMAILIO.IP;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.y7MGZLMGZwM.1LgRWIC9gIgx4fMGZAOBVAOBNfzuaVHRaYpB1LNSQLpx4uMx3Az6eL3RsBCxu-zRrUWSeOgjeBk.IVm4ds34aONc** From: "+4319714111 <tel:%2B4319714111>" <sip:+4319714111 <tel:%2B4319714111>@FREESWITCH.IP.HERE>;tag=yUr5UZ7eF794K To: <sip:502@USER.B.IP.HERE:5060>;tag=105223296 Call-ID: cd811276-4b4d-1234-66ae-005056867dbc CSeq: 87257355 INVITE Contact: <sip:06606017597@FREESWITCH.IP.HERE:5060> User-Agent: Yealink SIP-T46G 28.80.0.70 Diversion: <sip:502@USER.B.IP.HERE:5060>;reason=unconditional Content-Length: 0 This is modified in Kamailio TOPOH and sent to FreeSwitch as following SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 10.0.20.71;received=10.0.20.71;rport=5060;branch=z9hG4bKeBcy9HKK9cDKr From: "+4319714111 <tel:%2B4319714111>" <sip:+4319714111 <tel:%2B4319714111>@FREESWITCH.IP.HERE>;tag=yUr5UZ7eF794K To: <sip:502@USER.B.IP.HERE:5060>;tag=105223296 Call-ID: cd811276-4b4d-1234-66ae-005056867dbc CSeq: 87257355 INVITE Contact: <*sip:TOPOH.KAMAILIO.IP*;line=sr-N6IAzBy6WBy6MxFwW.qwPSW5ohWINhaYNLu4g9W4OhWI3wKLgRsIpUg5kGs7g9P-W.y6My**> User-Agent: Yealink SIP-T46G 28.80.0.70 Diversion: <sip:502@USER.B.IP.HERE:5060>;reason=unconditional Content-Length: 0 Which results in a Call Originate from FreeSwitch with RURI as this: INVITE *sip:TOPOH.KAMAILIO.IP*;line=sr-N6IAzBy6WBy6MxFwW.qwPSW5ohWINhaYNLu4g9W4OhWI3wKLgRsIpUg5kGs7g9P-W.y6My** SIP/2.0 I am thinking that in TOPOH Module some patch is required to atleast retain the $rU for 3XX replies , not sure if this will break some RFC or Kamailio stability etc ! Thanks, Sammy. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Best regards, Igor
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Just a quick question, if possible. I can’t get $ru in reply, cause in reply_route it gives me $ru from original INVITE. In OpenSIPS I’m using $(<reply>hdr(contact)), but how to do this on Kamailio? Anyway, contact Header from reply was got from tcpdump capture on Kamailio server.
2016-02-17 10:06 GMT+02:00 Daniel-Constantin Mierla miconda@gmail.com:
Hello,
isn't the username then decoded from the value of
sip:TOPOH.KAMAILIO.IP;line=sr-N6IAzBy6WBy6MxFwW.qwPSW5ohWINhaYNLu4g9W4OhWI3wKLgRsIpUg5kGs7g9P-W.y6My**
In other words, when the follow up invite in Kamailio comes, can you print the value of $ru with xlog and see if it has username?
Cheers, Daniel
On 16/02/16 22:54, Igor Olhovskiy wrote:
Yep, got same problem. Not sure it’s a bug, but seems to be topoh module is really lack some configuration.
2016-02-12 21:13 GMT+02:00 SamyGo govoiper@gmail.com:
Hi All, I've recently stumbled upon this little hitch while using kamailio with topoh module that the Contact header do not contains the User part for the 302 Moved temporarily packet.
*My topology:* UserA<====\ ->Kamailio<===>FreeSwitch UserB<====/
The B party has set Call forwarding on their phone hence phone sends a 302 Moved Temporarily to Kamailio.
Via: SIP/2.0/UDP TOPOH.KAMAILIO.IP:5060;branch=z9hG4bK0767.4e4e57deca89da5c5f5ae97228325f85.0 Via: SIP/2.0/UDP TOPOH.KAMAILIO.IP;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.y7MGZLMGZwM.1LgRWIC9gIgx4fMGZAOBVAOBNfzuaVHRaYpB1LNSQLpx4uMx3Az6eL3RsBCxu-zRrUWSeOgjeBk.IVm4ds34aONc** From: "+4319714111" sip:+4319714111 @FREESWITCH.IP.HERE;tag=yUr5UZ7eF794K To: sip:502@USER.B.IP.HERE:5060;tag=105223296 Call-ID: cd811276-4b4d-1234-66ae-005056867dbc CSeq: 87257355 INVITE Contact: sip:06606017597@FREESWITCH.IP.HERE:5060 User-Agent: Yealink SIP-T46G 28.80.0.70 Diversion: sip:502@USER.B.IP.HERE:5060;reason=unconditional Content-Length: 0
This is modified in Kamailio TOPOH and sent to FreeSwitch as following
SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 10.0.20.71;received=10.0.20.71;rport=5060;branch=z9hG4bKeBcy9HKK9cDKr From: "+4319714111" sip:+4319714111 @FREESWITCH.IP.HERE;tag=yUr5UZ7eF794K To: sip:502@USER.B.IP.HERE:5060;tag=105223296 Call-ID: cd811276-4b4d-1234-66ae-005056867dbc CSeq: 87257355 INVITE Contact: <*sip:TOPOH.KAMAILIO.IP* ;line=sr-N6IAzBy6WBy6MxFwW.qwPSW5ohWINhaYNLu4g9W4OhWI3wKLgRsIpUg5kGs7g9P-W.y6My**> User-Agent: Yealink SIP-T46G 28.80.0.70 Diversion: sip:502@USER.B.IP.HERE:5060;reason=unconditional Content-Length: 0
Which results in a Call Originate from FreeSwitch with RURI as this:
INVITE *sip:TOPOH.KAMAILIO.IP*;line=sr-N6IAzBy6WBy6MxFwW.qwPSW5ohWINhaYNLu4g9W4OhWI3wKLgRsIpUg5kGs7g9P-W.y6My** SIP/2.0
I am thinking that in TOPOH Module some patch is required to atleast retain the $rU for 3XX replies , not sure if this will break some RFC or Kamailio stability etc !
Thanks, Sammy.
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Best regards, Igor
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hi Daniel,
There is nothing in $rU for this appearing in Kamailio. That is same according to the Packet capture there is not $rU in my INVITE coming from FS.
This is where in kamailio.cfg I'm getting a 484 Address Incomplete from Kamailio back to FS.
if (!is_method("REGISTER") && $rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; }
So Like I mentioned, if only the $rU part is maintained in the Contact header by TOPOH this would start working as I've experimented by disabling TOPOH and repeated the scenario and it worked.
Thanks for looking into this, Regards, Sammy
On Feb 17, 2016 08:31, "Igor Olhovskiy" igorolhovskiy@gmail.com wrote:
Just a quick question, if possible. I can’t get $ru in reply, cause in reply_route it gives me $ru from original INVITE. In OpenSIPS I’m using $(<reply>hdr(contact)), but how to do this on Kamailio? Anyway, contact Header from reply was got from tcpdump capture on Kamailio server.
2016-02-17 10:06 GMT+02:00 Daniel-Constantin Mierla miconda@gmail.com:
Hello,
isn't the username then decoded from the value of
sip:TOPOH.KAMAILIO.IP;line=sr-N6IAzBy6WBy6MxFwW.qwPSW5ohWINhaYNLu4g9W4OhWI3wKLgRsIpUg5kGs7g9P-W.y6My**
In other words, when the follow up invite in Kamailio comes, can you print the value of $ru with xlog and see if it has username?
Cheers, Daniel
On 16/02/16 22:54, Igor Olhovskiy wrote:
Yep, got same problem. Not sure it’s a bug, but seems to be topoh module is really lack some configuration.
2016-02-12 21:13 GMT+02:00 SamyGo govoiper@gmail.com:
Hi All, I've recently stumbled upon this little hitch while using kamailio with topoh module that the Contact header do not contains the User part for the 302 Moved temporarily packet.
*My topology:* UserA<====\ ->Kamailio<===>FreeSwitch UserB<====/
The B party has set Call forwarding on their phone hence phone sends a 302 Moved Temporarily to Kamailio.
Via: SIP/2.0/UDP TOPOH.KAMAILIO.IP:5060;branch=z9hG4bK0767.4e4e57deca89da5c5f5ae97228325f85.0 Via: SIP/2.0/UDP TOPOH.KAMAILIO.IP;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.y7MGZLMGZwM.1LgRWIC9gIgx4fMGZAOBVAOBNfzuaVHRaYpB1LNSQLpx4uMx3Az6eL3RsBCxu-zRrUWSeOgjeBk.IVm4ds34aONc** From: "+4319714111" sip:+4319714111 @FREESWITCH.IP.HERE;tag=yUr5UZ7eF794K To: sip:502@USER.B.IP.HERE:5060;tag=105223296 Call-ID: cd811276-4b4d-1234-66ae-005056867dbc CSeq: 87257355 INVITE Contact: sip:06606017597@FREESWITCH.IP.HERE:5060 User-Agent: Yealink SIP-T46G 28.80.0.70 Diversion: sip:502@USER.B.IP.HERE:5060;reason=unconditional Content-Length: 0
This is modified in Kamailio TOPOH and sent to FreeSwitch as following
SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 10.0.20.71;received=10.0.20.71;rport=5060;branch=z9hG4bKeBcy9HKK9cDKr From: "+4319714111" sip:+4319714111 @FREESWITCH.IP.HERE;tag=yUr5UZ7eF794K To: sip:502@USER.B.IP.HERE:5060;tag=105223296 Call-ID: cd811276-4b4d-1234-66ae-005056867dbc CSeq: 87257355 INVITE Contact: <*sip:TOPOH.KAMAILIO.IP* ;line=sr-N6IAzBy6WBy6MxFwW.qwPSW5ohWINhaYNLu4g9W4OhWI3wKLgRsIpUg5kGs7g9P-W.y6My**> User-Agent: Yealink SIP-T46G 28.80.0.70 Diversion: sip:502@USER.B.IP.HERE:5060;reason=unconditional Content-Length: 0
Which results in a Call Originate from FreeSwitch with RURI as this:
INVITE *sip:TOPOH.KAMAILIO.IP*;line=sr-N6IAzBy6WBy6MxFwW.qwPSW5ohWINhaYNLu4g9W4OhWI3wKLgRsIpUg5kGs7g9P-W.y6My** SIP/2.0
I am thinking that in TOPOH Module some patch is required to atleast retain the $rU for 3XX replies , not sure if this will break some RFC or Kamailio stability etc !
Thanks, Sammy.
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Best regards, Igor
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Best regards, Igor
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users