12 Feb
2016
12 Feb
'16
9:13 p.m.
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.
16 Feb
16 Feb
11:54 p.m.
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(a)gmail.com>om>:
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(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Best regards,
Igor
17 Feb
17 Feb
10:06 a.m.
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(a)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(a)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(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.com
http://miconda.eu
3:30 p.m.
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(a)gmail.com>om>:
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(a)gmail.com>om>:
--
Best regards,
Igor
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(a)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(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
6:17 p.m.
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(a)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(a)gmail.com>om>:
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(a)gmail.com>om>:
--
Best regards,
Igor
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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(a)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(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
3231
days inactive
3236
days old
4 comments
3 participants
participants (3)
-
Daniel-Constantin Mierla -
Igor Olhovskiy -
SamyGo