Hi List,
We are using Kamailo 3.01 and testing topoh for starters:
Kamailio3 = IP of Kamailio Server and Asterisk=IP of Asterisk server
Observations:
U 2010/03/26 12:05:56.665583 Kamailio3:5060 -> Asterisk:5060 SIP/2.0 100 trying -- your call is important to us^@ Via: SIP/2.0/UDP Kamailio3:5060;branch=z9hG4bK08fffea4;rport=5060^@ From: "xxxxxxxxx" sip:xxxxx@asterisk;tag=as7ac63338^@ To: sip:Xxxxxxx@Kamailio3^@ Call-ID: 2f7a1c040f4920d5591695956ec8c42c@Kamailio3^@ CSeq: 103 INVITE^@ Server: kamailio (3.0.1 (i386/linux))^@ Content-Length: 0^@ Warning: 392 Kamailio:5060 "Noisy feedback tells: pid=23782 req_src_ip=Asterisk req_src_port=5060 in_uri=sip:xxxxxx@Kamailioout_uri=sip:Translated@ *UPSTREAMPROVIDERSIP* Provider via_cnt==1"^@
The noisy feedback contains everything that is hidden by topoh in the contact:
Contact: sip:10.1.1.10;line=sr-N6IAzBVAWxFAzx3lzxclMxylPxcsOBVlWGZ6WJZfWlq-W.y6My**^
Also:
U 2010/03/26 12:06:00.125828 Kamailio3:5060 -> Asterisk:5060 SIP/2.0 180 Ringing^@ From: "xxxxxxx"sip:xxxxxxxx@Kamailio3;tag=as7ac63338^@ To: <sip:xxxxxx@Kamailio3
;tag=fccd978-b342ea59-13c4-50022-82e974-27d2fa70-82e974^@
Call-ID: 2f7a1c040f4920d5591695956ec8c42c@Kamailio3^@ CSeq: 103 INVITE^@ Via: SIP/2.0/UDP Asterisk:5060;branch=z9hG4bK08fffea4;rport=5060^@ Record-Route: sip:10.1.1.10;line=sr-N6IAzBclOBF4WLZfWBF7M.VLz6fLHRQ7z6srpxusg9MXgSIqHReEgJZwgSKEgSjLMc**^@ Record-Route: sip:Kamailio3;lr=on;did=bff.60851b12^@ Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, INFO, REFER, NOTIFY^@ Contact: sip:10.1.1.10;line=sr-N6IAzBVAWxFAzx3lzxclMxylPxcsOBVlWGZ6WJZfWlq-W.y6My**^@ Content-Type: application/sdp^@ Content-Length: 228^@ ^@ v=0^@ *o=Upstream Switch Name 160624700 160624700 IN IP4 Upstream Providers IP^@* *s=Upstream Switch Name^@* * * The upstream info appears here also.
I am using engage_media_proxy and using topoh with defaul parameters.
Am I missing something ?
Thanks, Stephen.
nathelper modules has a certain flag to change o= line too: http://www.kamailio.org/docs/modules/1.5.x/nathelper#id2468157
Maybe mediaproxy has a similar feature?
Regarding s= line. I think you have to change it manually using textops module.
regards klaus
Am 26.03.2010 13:36, schrieb dotnetdub:
Hi List,
We are using Kamailo 3.01 and testing topoh for starters:
Kamailio3 = IP of Kamailio Server and Asterisk=IP of Asterisk server
Observations:
U 2010/03/26 12:05:56.665583 Kamailio3:5060 -> Asterisk:5060 SIP/2.0 100 trying -- your call is important to us^@ Via: SIP/2.0/UDP Kamailio3:5060;branch=z9hG4bK08fffea4;rport=5060^@ From: "xxxxxxxxx" sip:xxxxx@asterisk;tag=as7ac63338^@ To: sip:Xxxxxxx@Kamailio3^@ Call-ID: 2f7a1c040f4920d5591695956ec8c42c@Kamailio3^@ CSeq: 103 INVITE^@ Server: kamailio (3.0.1 (i386/linux))^@ Content-Length: 0^@ Warning: 392 Kamailio:5060 "Noisy feedback tells: pid=23782 req_src_ip=Asterisk req_src_port=5060 in_uri=sip:xxxxxx@Kamailio out_uri=sip:Translated@*UPSTREAMPROVIDERSIP* Provider via_cnt==1"^@
The noisy feedback contains everything that is hidden by topoh in the contact:
Contact: sip:10.1.1.10;line=sr-N6IAzBVAWxFAzx3lzxclMxylPxcsOBVlWGZ6WJZfWlq-W.y6My**^
Also:
U 2010/03/26 12:06:00.125828 Kamailio3:5060 -> Asterisk:5060 SIP/2.0 180 Ringing^@ From: "xxxxxxx"sip:xxxxxxxx@Kamailio3;tag=as7ac63338^@ To: sip:xxxxxx@Kamailio3;tag=fccd978-b342ea59-13c4-50022-82e974-27d2fa70-82e974^@ Call-ID: 2f7a1c040f4920d5591695956ec8c42c@Kamailio3^@ CSeq: 103 INVITE^@ Via: SIP/2.0/UDP Asterisk:5060;branch=z9hG4bK08fffea4;rport=5060^@ Record-Route: sip:10.1.1.10;line=sr-N6IAzBclOBF4WLZfWBF7M.VLz6fLHRQ7z6srpxusg9MXgSIqHReEgJZwgSKEgSjLMc**^@ Record-Route: sip:Kamailio3;lr=on;did=bff.60851b12^@ Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, INFO, REFER, NOTIFY^@ Contact: sip:10.1.1.10;line=sr-N6IAzBVAWxFAzx3lzxclMxylPxcsOBVlWGZ6WJZfWlq-W.y6My**^@ Content-Type: application/sdp^@ Content-Length: 228^@ ^@ v=0^@ *o=Upstream Switch Name 160624700 160624700 IN IP4 Upstream Providers IP^@* *s=Upstream Switch Name^@*
The upstream info appears here also.
I am using engage_media_proxy and using topoh with defaul parameters.
Am I missing something ?
Thanks, Stephen.
sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hi Klaus,
Thanks for reply.
Regarding the noisy feedback. Is it possible to just turn this off in Kamailio? I've looked but the only way I can see to do it is to comment it in source and recompile. It seems to be only transmitted certain times.
Would it not make sense for topoh module to mask all topology in the SIP messages?
Regards, Stephen
On 26 March 2010 12:53, Klaus Darilion klaus.mailinglists@pernau.at wrote:
nathelper modules has a certain flag to change o= line too: http://www.kamailio.org/docs/modules/1.5.x/nathelper#id2468157
Maybe mediaproxy has a similar feature?
Regarding s= line. I think you have to change it manually using textops module.
regards klaus
Am 26.03.2010 13:36, schrieb dotnetdub:
Hi List,
We are using Kamailo 3.01 and testing topoh for starters:
Kamailio3 = IP of Kamailio Server and Asterisk=IP of Asterisk server
Observations:
U 2010/03/26 12:05:56.665583 Kamailio3:5060 -> Asterisk:5060 SIP/2.0 100 trying -- your call is important to us^@ Via: SIP/2.0/UDP Kamailio3:5060;branch=z9hG4bK08fffea4;rport=5060^@ From: "xxxxxxxxx" sip:xxxxx@asterisk;tag=as7ac63338^@ To: sip:Xxxxxxx@Kamailio3^@ Call-ID: 2f7a1c040f4920d5591695956ec8c42c@Kamailio3^@ CSeq: 103 INVITE^@ Server: kamailio (3.0.1 (i386/linux))^@ Content-Length: 0^@ Warning: 392 Kamailio:5060 "Noisy feedback tells: pid=23782 req_src_ip=Asterisk req_src_port=5060 in_uri=sip:xxxxxx@Kamailio out_uri=sip:Translated@*UPSTREAMPROVIDERSIP* Provider via_cnt==1"^@
The noisy feedback contains everything that is hidden by topoh in the contact:
Contact:
sip:10.1.1.10;line=sr-N6IAzBVAWxFAzx3lzxclMxylPxcsOBVlWGZ6WJZfWlq-W.y6My**^
Also:
U 2010/03/26 12:06:00.125828 Kamailio3:5060 -> Asterisk:5060 SIP/2.0 180 Ringing^@ From: "xxxxxxx"sip:xxxxxxxx@Kamailio3;tag=as7ac63338^@ To: <sip:xxxxxx@Kamailio3
;tag=fccd978-b342ea59-13c4-50022-82e974-27d2fa70-82e974^@
Call-ID: 2f7a1c040f4920d5591695956ec8c42c@Kamailio3^@ CSeq: 103 INVITE^@ Via: SIP/2.0/UDP Asterisk:5060;branch=z9hG4bK08fffea4;rport=5060^@ Record-Route:
sip:10.1.1.10;line=sr-N6IAzBclOBF4WLZfWBF7M.VLz6fLHRQ7z6srpxusg9MXgSIqHReEgJZwgSKEgSjLMc**^@ Record-Route: sip:Kamailio3;lr=on;did=bff.60851b12^@ Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, INFO, REFER, NOTIFY^@ Contact:
sip:10.1.1.10;line=sr-N6IAzBVAWxFAzx3lzxclMxylPxcsOBVlWGZ6WJZfWlq-W.y6My**^@ Content-Type: application/sdp^@ Content-Length: 228^@ ^@ v=0^@ *o=Upstream Switch Name 160624700 160624700 IN IP4 Upstream Providers IP^@* *s=Upstream Switch Name^@*
The upstream info appears here also.
I am using engage_media_proxy and using topoh with defaul parameters.
Am I missing something ?
Thanks, Stephen.
sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
http://sip-router.org/wiki/cookbooks/core-cookbook/devel#sip_warning_noisy_f...
regards klaus
Am 26.03.2010 16:39, schrieb dotnetdub:
Hi Klaus,
Thanks for reply.
Regarding the noisy feedback. Is it possible to just turn this off in Kamailio? I've looked but the only way I can see to do it is to comment it in source and recompile. It seems to be only transmitted certain times.
Would it not make sense for topoh module to mask all topology in the SIP messages?
Regards, Stephen
On 26 March 2010 12:53, Klaus Darilion <klaus.mailinglists@pernau.at mailto:klaus.mailinglists@pernau.at> wrote:
nathelper modules has a certain flag to change o= line too: http://www.kamailio.org/docs/modules/1.5.x/nathelper#id2468157 Maybe mediaproxy has a similar feature? Regarding s= line. I think you have to change it manually using textops module. regards klaus Am 26.03.2010 13:36, schrieb dotnetdub: Hi List, We are using Kamailo 3.01 and testing topoh for starters: Kamailio3 = IP of Kamailio Server and Asterisk=IP of Asterisk server Observations: U 2010/03/26 12:05:56.665583 Kamailio3:5060 -> Asterisk:5060 SIP/2.0 100 trying -- your call is important to us^@ Via: SIP/2.0/UDP Kamailio3:5060;branch=z9hG4bK08fffea4;rport=5060^@ From: "xxxxxxxxx" <sip:xxxxx@asterisk>;tag=as7ac63338^@ To: <sip:Xxxxxxx@Kamailio3>^@ Call-ID: 2f7a1c040f4920d5591695956ec8c42c@Kamailio3^@ CSeq: 103 INVITE^@ Server: kamailio (3.0.1 (i386/linux))^@ Content-Length: 0^@ Warning: 392 Kamailio:5060 "Noisy feedback tells: pid=23782 req_src_ip=Asterisk req_src_port=5060 in_uri=sip:xxxxxx@Kamailio out_uri=sip:Translated@*UPSTREAMPROVIDERSIP* Provider via_cnt==1"^@ The noisy feedback contains everything that is hidden by topoh in the contact: Contact: <sip:10.1.1.10;line=sr-N6IAzBVAWxFAzx3lzxclMxylPxcsOBVlWGZ6WJZfWlq-W.y6My**>^ Also: U 2010/03/26 12:06:00.125828 Kamailio3:5060 -> Asterisk:5060 SIP/2.0 180 Ringing^@ From: "xxxxxxx"<sip:xxxxxxxx@Kamailio3>;tag=as7ac63338^@ To: <sip:xxxxxx@Kamailio3>;tag=fccd978-b342ea59-13c4-50022-82e974-27d2fa70-82e974^@ Call-ID: 2f7a1c040f4920d5591695956ec8c42c@Kamailio3^@ CSeq: 103 INVITE^@ Via: SIP/2.0/UDP Asterisk:5060;branch=z9hG4bK08fffea4;rport=5060^@ Record-Route: <sip:10.1.1.10;line=sr-N6IAzBclOBF4WLZfWBF7M.VLz6fLHRQ7z6srpxusg9MXgSIqHReEgJZwgSKEgSjLMc**>^@ Record-Route: <sip:Kamailio3;lr=on;did=bff.60851b12>^@ Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, INFO, REFER, NOTIFY^@ Contact: <sip:10.1.1.10;line=sr-N6IAzBVAWxFAzx3lzxclMxylPxcsOBVlWGZ6WJZfWlq-W.y6My**>^@ Content-Type: application/sdp^@ Content-Length: 228^@ ^@ v=0^@ *o=Upstream Switch Name 160624700 160624700 IN IP4 Upstream Providers IP^@* *s=Upstream Switch Name^@* * * The upstream info appears here also. I am using engage_media_proxy and using topoh with defaul parameters. Am I missing something ? Thanks, Stephen. _______________________________________________ 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
Hi Klaus.
Prefect, thanks. I used textops to replace the s= and o= lines also so now all fine. Maybe would be nice for topoh to do all this, point of the module is to secure your topology.
Thanks, Stephen.
On 26 March 2010 15:47, Klaus Darilion klaus.mailinglists@pernau.at wrote:
http://sip-router.org/wiki/cookbooks/core-cookbook/devel#sip_warning_noisy_f...
regards klaus
Am 26.03.2010 16:39, schrieb dotnetdub:
Hi Klaus,
Thanks for reply.
Regarding the noisy feedback. Is it possible to just turn this off in Kamailio? I've looked but the only way I can see to do it is to comment it in source and recompile. It seems to be only transmitted certain times.
Would it not make sense for topoh module to mask all topology in the SIP messages?
Regards, Stephen
On 26 March 2010 12:53, Klaus Darilion <klaus.mailinglists@pernau.at mailto:klaus.mailinglists@pernau.at> wrote:
nathelper modules has a certain flag to change o= line too: http://www.kamailio.org/docs/modules/1.5.x/nathelper#id2468157
Maybe mediaproxy has a similar feature?
Regarding s= line. I think you have to change it manually using textops module.
regards klaus
Am 26.03.2010 13:36, schrieb dotnetdub:
Hi List, We are using Kamailo 3.01 and testing topoh for starters: Kamailio3 = IP of Kamailio Server and Asterisk=IP of Asteriskserver
Observations: U 2010/03/26 12:05:56.665583 Kamailio3:5060 -> Asterisk:5060 SIP/2.0 100 trying -- your call is important to us^@ Via: SIP/2.0/UDP Kamailio3:5060;branch=z9hG4bK08fffea4;rport=5060^@ From: "xxxxxxxxx" <sip:xxxxx@asterisk>;tag=as7ac63338^@ To: <sip:Xxxxxxx@Kamailio3>^@ Call-ID: 2f7a1c040f4920d5591695956ec8c42c@Kamailio3^@ CSeq: 103 INVITE^@ Server: kamailio (3.0.1 (i386/linux))^@ Content-Length: 0^@ Warning: 392 Kamailio:5060 "Noisy feedback tells: pid=23782 req_src_ip=Asterisk req_src_port=5060 in_uri=sip:xxxxxx@Kamailio out_uri=sip:Translated@*UPSTREAMPROVIDERSIP* Providervia_cnt==1"^@
The noisy feedback contains everything that is hidden by topoh in the contact: Contact:sip:10.1.1.10;line=sr-N6IAzBVAWxFAzx3lzxclMxylPxcsOBVlWGZ6WJZfWlq-W.y6My**^
Also: U 2010/03/26 12:06:00.125828 Kamailio3:5060 -> Asterisk:5060 SIP/2.0 180 Ringing^@ From: "xxxxxxx"<sip:xxxxxxxx@Kamailio3>;tag=as7ac63338^@ To: <sip:xxxxxx@Kamailio3;tag=fccd978-b342ea59-13c4-50022-82e974-27d2fa70-82e974^@
Call-ID: 2f7a1c040f4920d5591695956ec8c42c@Kamailio3^@ CSeq: 103 INVITE^@ Via: SIP/2.0/UDP Asterisk:5060;branch=z9hG4bK08fffea4;rport=5060^@ Record-Route:sip:10.1.1.10;line=sr-N6IAzBclOBF4WLZfWBF7M.VLz6fLHRQ7z6srpxusg9MXgSIqHReEgJZwgSKEgSjLMc**^@ Record-Route: sip:Kamailio3;lr=on;did=bff.60851b12^@ Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, INFO, REFER, NOTIFY^@ Contact:
sip:10.1.1.10;line=sr-N6IAzBVAWxFAzx3lzxclMxylPxcsOBVlWGZ6WJZfWlq-W.y6My**^@ Content-Type: application/sdp^@ Content-Length: 228^@ ^@ v=0^@ *o=Upstream Switch Name 160624700 160624700 IN IP4 Upstream Providers IP^@* *s=Upstream Switch Name^@* * * The upstream info appears here also.
I am using engage_media_proxy and using topoh with defaul parameters. Am I missing something ? Thanks, Stephen. _______________________________________________ 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
On 03/26/2010 12:46 PM, dotnetdub wrote:
Prefect, thanks. I used textops to replace the s= and o= lines also so now all fine. Maybe would be nice for topoh to do all this, point of the module is to secure your topology.
I would say, in my interpretation, that the purpose of the module is more to obscure your topology. You still can't change the m= line without breaking audio, for one.
Obscure yes,
We are using mediaproxy so m= is not an issue here.
The two lines that were revealing:
o=Upstream Switch Name 160624700 160624700 IN IP4 Upstream Providers IP^@ s=Upstream Switch Name^@
textops has sorted these for me. Klaus pointed out that the RTPproxy module will look after the o= line. Mediaproxy doesn't have an option.
Regards Stephen.
On 26 March 2010 16:52, Alex Balashov abalashov@evaristesys.com wrote:
On 03/26/2010 12:46 PM, dotnetdub wrote:
Prefect, thanks. I used textops to replace the s= and o= lines also so
now all fine. Maybe would be nice for topoh to do all this, point of the module is to secure your topology.
I would say, in my interpretation, that the purpose of the module is more to obscure your topology. You still can't change the m= line without breaking audio, for one.
-- Alex Balashov - Principal Evariste Systems LLC
Tel : +1 678-954-0670 Direct : +1 678-954-0671 Web : http://www.evaristesys.com/
sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-
Alex Balashov -
dotnetdub -
Klaus Darilion