Re: [SR-Users] TLS Support
Thanks a lot :-) -Ursprüngliche Nachricht- Von: Klaus Darilion [mailto:klaus.mailingli...@pernau.at] Gesendet: Donnerstag, 10. Februar 2011 19:19 An: Bernhard Suttner Cc: sr-users@lists.sip-router.org Betreff: Re: [SR-Users] TLS Support Bernhard Suttner wrote: Hi, how could I compile the TLS module and only the TLS module - not the other sources. cd modules/tls make regards klaus BTW: The documentation within the http://www.kamailio.org/docs/tls.html is not up-to-date. Thanks in advance. Best regards, Bernhard ___ 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 ___ 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
[SR-Users] NAT, TLS and location table
Hi, I am using TLS and recognize the following problem: The TLS connection are build up successfully but the natping (natping_interval = 10) does not send small dummy packets to the phones. The phones are behind a firewall with NAT. Registered phones with NAT but UDP do work correctly. They are getting the natping every 10 seconds. After 120 seconds (should be the tcp_connection_timeout) kamailio does send a FIN to the TLS phone to close the TLS connection. Should I increase the tcp_connection_timeout to a value bigger than the registration timeout? I thought I do not need that, because of the natping_interval. Is it maybe better to use a SIP-Options Ping instead of the small dummy packets? I would prefer the dummy packets because they are much smaller. BTW: Kamailio 3.1.1 and snom360 with latest v8 firmware. Thanks in advance! Best regards, Bernhard ___ 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
Re: [SR-Users] NAT, TLS and location table
Hi, thanks for you response. Do you think that kamailio does send sip-options-ping within TCP/TLS (instead of sending the dummy packets)? Best regards, Bernhard -Ursprüngliche Nachricht- Von: Klaus Darilion [mailto:klaus.mailingli...@pernau.at] Gesendet: Freitag, 11. Februar 2011 12:30 An: Bernhard Suttner Cc: sr-users@lists.sip-router.org Betreff: Re: [SR-Users] NAT, TLS and location table Am 11.02.2011 12:15, schrieb Bernhard Suttner: Hi, I am using TLS and recognize the following problem: The TLS connection are build up successfully but the natping (natping_interval = 10) does not send small dummy packets to the phones. The phones are behind a firewall with NAT. Registered phones with NAT but UDP do work correctly. They are getting the natping every 10 seconds. After 120 seconds (should be the tcp_connection_timeout) kamailio does send a FIN to the TLS phone to close the TLS connection. IIRC the keep-alive code in nathelper module sends CRLF only on UDP. IMO it would be nice if it sends it also on TCP/TLS connections, at least as a config option. Of course the code should also take care of not setting up a new TCP connection if the old one is gone. I once have seen a client which was confused by the CRLF and then closes the TCP connection, so there might be other problems as well. Of course the proper solution (IETF view) is that the clients sends keep-alive (SIP outbound RFC). Should I increase the tcp_connection_timeout to a value bigger than the registration timeout? I thought I do not need that, because of the natping_interval. Is it maybe better to use a SIP-Options Ping instead of the small dummy packets? I would prefer the dummy packets because they are much smaller. http://www.kamailio.org/dokuwiki/doku.php/install:1.5.x-to-3.0.0#tcp_connection_lifetime http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:3.1.x#tcp_keepalive and http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:3.1.x#set_forward_no_connect (to be used after lookup()) might be interesting too. regards klaus ___ 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
[SR-Users] Change transport=tls within Contact Header
Hi, is there a function to change the transport field within the Contact Header? t_relay does relay it to the UDP port of the PBX but the transport within the Contact Header is still set to TLS. Later, the PBX does try to establish a TLS connection. The idea is now, that kamailio does set the transport of the Contact Header to UDP. Another possible way would to be, that I use the re.subst method but maybe there is a special function for that. Thanks in advance. Best regards, Bernhard ___ 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
[SR-Users] TLS Support
Hi, how could I compile the TLS module and only the TLS module - not the other sources. BTW: The documentation within the http://www.kamailio.org/docs/tls.html is not up-to-date. Thanks in advance. Best regards, Bernhard ___ 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
Re: [SR-Users] Redundancy between 2 Kamailio servers
Would be possible.
Have a look at DRBD!
Von: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] Im Auftrag von Danny Dias
Gesendet: Mittwoch, 2. Februar 2011 11:20
An: kamailio
Betreff: Re: [SR-Users] Redundancy between 2 Kamailio servers
Hello,
I was thinking that if i store the contacts that are registered into my proxy
in a database, like this:
.
.
# - usrloc params -
modparam(usrloc, db_mode, 3)
.
.
if (is_method(REGISTER))
{
if (!save(location))
sl_reply_error();
exit;
}
.
.
This way i would store the contacts in table location, and then i could make
a backup of the DB and restore on the other server...is that ok? or i'm missing
something?
Thanks in advance for your help!
2011/1/27 Iñaki Baz Castillo i...@aliax.net
2011/1/27 Danny Dias ing.diasda...@gmail.com:
Two Kamailios in a HeartBeat cluster which manages the kamailio
service along with a virtual IP in which kamailios are supposed to
listen. Just one kamailio is running (HA manages them).
So, the heartbeat cluster shall manage that both are ok and also check that
the virtual ip and the kamailio service in the primary server is OKif
something fails it will activate the virtual IP address and the kamailio
process in the other server? so this heartbeat cluster is installed in both
kamailio servers?
which HA software do you recommend?
As I said at the top of my previous mail: HeartBeat (as it is the only
I'm used to).
Regsitration can be done in a shared database with db_mode=3 (or 2) so
no locations are lost when HA stops the running instance of kamailio
(or the server is down) and starts kamailio in the other cluster node.
So, the database of the kamailios should be dedicated and externalised
server?
Could be, or not. It doesn't matter too much.
--
Iñaki Baz Castillo
i...@aliax.net
___
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
[SR-Users] t_local_replied() ?
Hi, I am using the dispatcher module and want to check within the failure_route if the 408 was internally generated from kamailio or it was received from the dispatcher gateway. There was previously a function called t_local_replied() in the TM-module but I could not find this function in the current documentation. Was it removed? Is there a alternative to check if the 408 was local generated or if it was received from the peer (= from the dispatcher gateway)? Thanks in advance! Best regards, Bernhard ___ 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
Re: [SR-Users] t_local_replied() ?
Hi, just to be sure: - If the gateway does send back a 100 Trying and then a 408 is detected within failure_route the method t_branch_replied does return false (means: gateway is up and running) - dont go to next gateway (dispatcher) - If the gateway does not respond and a 408 is detected within failure_route (= 408 was generated from kamailio) t_branch_replied does return true (means: gateway is down) - go to the next gateway (dispatcher) Is that correct or am I wrong? Best regards, Bernhard - Original Message - From: Daniel-Constantin Mierla [mailto:mico...@gmail.com] To: Bernhard Suttner [mailto:bernhard.sutt...@winet.ch] Cc: sr-users@lists.sip-router.org Sent: Tue, 01 Feb 2011 13:30:54 +0100 Subject: Re: [SR-Users] t_local_replied() ? Hello, On 2/1/11 12:08 PM, Bernhard Suttner wrote: Hi, I am using the dispatcher module and want to check within the failure_route if the 408 was internally generated from kamailio or it was received from the dispatcher gateway. There was previously a function called t_local_replied() in the TM-module but I could not find this function in the current documentation. Was it removed? Is there a alternative to check if the 408 was local generated or if it was received from the peer (= from the dispatcher gateway)? see the example of: http://kamailio.org/docs/modules/stable/modules/tm.html#t_branch_replied Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com ___ 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
Re: [SR-Users] t_local_replied() ?
argh. I just mixed up the true/false return values in the eMail but you still got it :-) Thanks a lot! - Original Message - From: Daniel-Constantin Mierla [mailto:mico...@gmail.com] To: Bernhard Suttner [mailto:bernhard.sutt...@winet.ch] Cc: sr-users@lists.sip-router.org Sent: Tue, 01 Feb 2011 20:31:17 +0100 Subject: Re: [SR-Users] t_local_replied() ? Hello, not sure if you ask about the options, or you tried them and don't give you the needed feature, since there are some improper true/false return codes in your email. t_branch_replied() will return false if the 408 is generated locally. Cheers, Daniel On 2/1/11 8:25 PM, Bernhard Suttner wrote: Hi, just to be sure: - If the gateway does send back a 100 Trying and then a 408 is detected within failure_route the method t_branch_replied does return false (means: gateway is up and running) - dont go to next gateway (dispatcher) - If the gateway does not respond and a 408 is detected within failure_route (= 408 was generated from kamailio) t_branch_replied does return true (means: gateway is down) - go to the next gateway (dispatcher) Is that correct or am I wrong? Best regards, Bernhard - Original Message - From: Daniel-Constantin Mierla [mailto:mico...@gmail.com] To: Bernhard Suttner [mailto:bernhard.sutt...@winet.ch] Cc: sr-users@lists.sip-router.org Sent: Tue, 01 Feb 2011 13:30:54 +0100 Subject: Re: [SR-Users] t_local_replied() ? Hello, On 2/1/11 12:08 PM, Bernhard Suttner wrote: Hi, I am using the dispatcher module and want to check within the failure_route if the 408 was internally generated from kamailio or it was received from the dispatcher gateway. There was previously a function called t_local_replied() in the TM-module but I could not find this function in the current documentation. Was it removed? Is there a alternative to check if the 408 was local generated or if it was received from the peer (= from the dispatcher gateway)? see the example of: http://kamailio.org/docs/modules/stable/modules/tm.html#t_branch_replied Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com ___ 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 -- Daniel-Constantin Mierla http://www.asipto.com ___ 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
Re: [SR-Users] [sr-dev] planning release of v3.1.2
Hi, whats about: http://sip-router.org/tracker/index.php?do=detailstask_id=108 I think that some users are trying to register to an asterisk system. Or does somebody have a solution for that? A new release would be nice!!! Thanks in advance. Best regards, Bernhard - Original Message - From: Daniel-Constantin Mierla [mailto:mico...@gmail.com] To: kamailio [mailto:sr-users@lists.sip-router.org], sr-dev [mailto:sr-...@lists.sip-router.org] Sent: Fri, 28 Jan 2011 20:15:50 +0100 Subject: [sr-dev] planning release of v3.1.2 Hello, I think it is time to release v3.1.2, first date that comes in my mind is next Thursday if everyone feels it is enough time to take care of backporting any fix he/she did and it is not yet there. That will provide us a fresh release for the FOSDEM event. If not, then maybe the other week, Tuesday, so the participants at the Kamailio Devel training in Barcelona can practice on it. Soon after we should plan also a release for previous stable, branch 3.0. Anyone having other options? Thanks, Daniel -- Daniel-Constantin Mierla http://www.asipto.com ___ sr-dev mailing list sr-...@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev ___ 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
[SR-Users] onreply_route and failure_route
Hi,
is it possible to call a route within onreply_ and failure_route? I mean,
within the main route { ... } its possible to call the route[NAT]. Is this also
possible for this special routes? If this is possible, is it also allowed to
call a route which will normally called from the main route { ... }
Thanks in advance.
Best regards,
Bernhard
___
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
Re: [SR-Users] onreply_route and failure_route
Hi,
that was _really_ fast. Thanks. :-)
One more question: Could I somehow test with the route[NAT] for example, if the
request was from the main route { ... } or from the onreply_route[] ?
Best regards,
Bernhard
- Original Message -
From: Alex Balashov [mailto:abalas...@evaristesys.com]
To: Bernhard Suttner [mailto:bernhard.sutt...@winet.ch]
Cc: sr-users@lists.sip-router.org [mailto:sr-users@lists.sip-router.org]
Sent: Sat, 29 Jan 2011 21:37:34 +0100
Subject: Re: [SR-Users] onreply_route and failure_route
Yes.
--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/
On Jan 29, 2011, at 3:35 PM, Bernhard Suttner bernhard.sutt...@winet.ch
wrote:
Hi,
is it possible to call a route within onreply_ and failure_route? I
mean, within the main route { ... } its possible to call the route
[NAT]. Is this also possible for this special routes? If this is
possible, is it also allowed to call a route which will normally
called from the main route { ... }
Thanks in advance.
Best regards,
Bernhard
___
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
___
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
[SR-Users] xlogl and pseudo variables
Hi, I am using xlogl for debugging and logging the most important actions within the kamailio script. Does it take much resources if I use the pseudo variables with the xlogl statements? Example: xlogl(L_NOTICE, Route $rm to destination! F=$fu T=$tu IP=$si \n); I´m planning to do this in all the xlogl statements to be able to debug / see whats going on on the system. Thanks in advance. Best regard, Bernhard ___ 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
Re: [SR-Users] Redundancy between 2 Kamailio servers
Try to set this option to bind to an non existing ip: net.ipv4.ip_nonlocal_bind = 1 does work for SSH and thefore should also work with kamailio -Ursprüngliche Nachricht- Von: sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] Im Auftrag von Klaus Darilion Gesendet: Donnerstag, 27. Januar 2011 15:18 An: Alex Hermann Cc: sr-users@lists.sip-router.org Betreff: Re: [SR-Users] Redundancy between 2 Kamailio servers Am 27.01.2011 13:05, schrieb Alex Hermann: Yes - the problem with SIP based replication is that both proxies must be running. This is a problem as Kamailio binds to the virtual IP at start up - thus adding the virtual IP address to the backup server does not make Backup-Kamailio listening to the new IP address - you would have to restart the backup Kamailio. Just bind kamailio to the HA IP on both servers and do REGISTER replication between the two (on SIP level). Then if the IP migrates to the other server, it will take over the rgistrar function with no loss of records. No restart needed. Is it possible to bind Kamailio to an IP address which is not active? (e.g. start Kamailio on the backup server) klaus ___ 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 ___ 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
Re: [SR-Users] fix_nated_sdp()
Hi, thanks for your answer. The question was about, if the fix_nated_sdp() (in the example below) could maybe break something. I am sure, that the fix_nated_sdp() would work in some cases but I am not sure, if the function could break something: if (method==INVITE has_sdp() nat_uac_test(8)) fix_nated_sdp(3) and within the onreply route: if (status=~(180|183|200) has_sdp() nat_uac_test(8)) fix_nated_sdp(3) Or is there a better solution to fix these damend NAT scenarios? Best regards, Bernhard -Ursprüngliche Nachricht- Von: kaiserbo...@googlemail.com [mailto:kaiserbo...@googlemail.com] Im Auftrag von Carsten Bock Gesendet: Dienstag, 25. Januar 2011 18:56 An: Bernhard Suttner Cc: sr-users@lists.sip-router.org Betreff: Re: [SR-Users] fix_nated_sdp() Hi Bernhard, have you checked the SDP which is going to and from the Freeswitch-Server? Is it modified? If yes: You may want to check your NAT-Firewall. A working (but not nice) solution might be to relay the RTP through an proxy in this case... If No: Verify the result of nat_uac_test: Why is it returning false? Carsten 2011/1/25 Bernhard Suttner bernhard.sutt...@winet.ch: Hi, someone has an idea? Thanks in advance! Best regards, Bernhard -Ursprüngliche Nachricht- Von: sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] Im Auftrag von Bernhard Suttner Gesendet: Montag, 24. Januar 2011 20:38 An: sr-users@lists.sip-router.org Betreff: [SR-Users] fix_nated_sdp() Hi, I have a question about the fix_nated_sdp() function. Different devices (phones, pbxes, gateways) are using Kamailio as SBC. The SBC does t_relay() the message to a pool of FreeSWITCH Servers. The scenario does work quite well but: if a call is initiated from one of the freeswitch server, pass it through kamailio to the PBX and the PBX does have a forwarding back to kamailio and in the end to freeswitch I have no audio because the PBX is behind a NAT and uses private ip addresses in all the SDP of 200 OK and 183 Session Progress. Therefore FreeSWITCH could not do the (very cool) RTP Auto-Adjustment (check if IP in incoming packet is different to the IP in the SDP) to detect the real IP address. Therefore I have to use fix_nated_sdp(). I am not sure, what this function could break therefore I want to be very sure and use it only, if really necessary. Test which User-Agent or Server the peer (in this example the PBX) has, is not really possible because some devices does not send this header. What do you think about the follwoing pseudo code: if (method==INVITE has_sdp() nat_uac_test(8)) fix_nated_sdp(3) and within the onreply route: if (status=~(180|183|200) has_sdp() nat_uac_test(8)) fix_nated_sdp(3) Somebody has a better idea to handle this? What could go wrong? Thanks in advance. Best regards, Bernhard Suttner ___ 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 ___ 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 -- Carsten Bock Schomburgstr. 80 22767 Hamburg Germany Mobile +49 179 2021244 Home +49 40 34927217 Büro (Verl) +49 5246 801427 Fax +49 40 34927218 mailto:cars...@bock.info ___ 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
Re: [SR-Users] fix_nated_sdp()
Hi, thanks for your response. I think I understand the scenario very good - but im not so familiar with the fix_nated_sdp() functionality. Currently the SDP will be modified with the source IP of the message. This does work good but currently I only do that for some special devices (matching the User-Agent / Server string). I want to do that fix_nated_sdp() in all cases and not only for some special devices. The FreeSWITCH RTP-Auto Adjustment does work in nearly every case but in one (currently known) scenario it does not. Maybe I should just give it a try to do fix_nated_sdp() in combination with nat_uac_test (RFC1918 private address in SDP). Thanks a lot for this discussion. Best Regards, Bernhard - Original Message - From: Ovidiu Sas [mailto:o...@voipembedded.com] To: Bernhard Suttner [mailto:bernhard.sutt...@winet.ch] Cc: Carsten Bock [mailto:li...@bock.info], sr-users@lists.sip-router.org Sent: Tue, 25 Jan 2011 20:38:14 +0100 Subject: Re: [SR-Users] fix_nated_sdp() You need to experiment with nathelper module and while doing that, you need to inspect the SIP messages in order to understand how it works. Only when you fully understand your scenarios, you will be able to properly configure, debg and setup kamailio. Best thing to do here is to use rtpproxy and relay media if needed. Another option would be to let the SDP untouched and let freeswitch do it's magic (if it can). Not sure if this will work in all scenarios. Calling fix_nated_sdp() on the SBC will definitely not work (you are just messing with the IP on the SDP and you don't know if the real RTP port on the NATed side will match the advertised port in your fixed SDP). Regards, Ovidiu Sas On Tue, Jan 25, 2011 at 1:23 PM, Bernhard Suttner bernhard.sutt...@winet.ch wrote: Hi, thanks for your answer. The question was about, if the fix_nated_sdp() (in the example below) could maybe break something. I am sure, that the fix_nated_sdp() would work in some cases but I am not sure, if the function could break something: if (method==INVITE has_sdp() nat_uac_test(8)) fix_nated_sdp(3) and within the onreply route: if (status=~(180|183|200) has_sdp() nat_uac_test(8)) fix_nated_sdp(3) Or is there a better solution to fix these damend NAT scenarios? Best regards, Bernhard -Ursprüngliche Nachricht- Von: kaiserbo...@googlemail.com [mailto:kaiserbo...@googlemail.com] Im Auftrag von Carsten Bock Gesendet: Dienstag, 25. Januar 2011 18:56 An: Bernhard Suttner Cc: sr-users@lists.sip-router.org Betreff: Re: [SR-Users] fix_nated_sdp() Hi Bernhard, have you checked the SDP which is going to and from the Freeswitch-Server? Is it modified? If yes: You may want to check your NAT-Firewall. A working (but not nice) solution might be to relay the RTP through an proxy in this case... If No: Verify the result of nat_uac_test: Why is it returning false? Carsten 2011/1/25 Bernhard Suttner bernhard.sutt...@winet.ch: Hi, someone has an idea? Thanks in advance! Best regards, Bernhard -Ursprüngliche Nachricht- Von: sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] Im Auftrag von Bernhard Suttner Gesendet: Montag, 24. Januar 2011 20:38 An: sr-users@lists.sip-router.org Betreff: [SR-Users] fix_nated_sdp() Hi, I have a question about the fix_nated_sdp() function. Different devices (phones, pbxes, gateways) are using Kamailio as SBC. The SBC does t_relay() the message to a pool of FreeSWITCH Servers. The scenario does work quite well but: if a call is initiated from one of the freeswitch server, pass it through kamailio to the PBX and the PBX does have a forwarding back to kamailio and in the end to freeswitch I have no audio because the PBX is behind a NAT and uses private ip addresses in all the SDP of 200 OK and 183 Session Progress. Therefore FreeSWITCH could not do the (very cool) RTP Auto-Adjustment (check if IP in incoming packet is different to the IP in the SDP) to detect the real IP address. Therefore I have to use fix_nated_sdp(). I am not sure, what this function could break therefore I want to be very sure and use it only, if really necessary. Test which User-Agent or Server the peer (in this example the PBX) has, is not really possible because some devices does not send this header. What do you think about the follwoing pseudo code: if (method==INVITE has_sdp() nat_uac_test(8)) fix_nated_sdp(3) and within the onreply route: if (status=~(180|183|200) has_sdp() nat_uac_test(8)) fix_nated_sdp(3) Somebody has a better idea to handle this? What could go wrong? Thanks in advance. Best regards, Bernhard Suttner ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip
Re: [SR-Users] $ua in on_reply route not set?
Hi, found the problem. The device does sometimes use User-Agent and sometimes Server. Is it better to use $hdr() or the Search() function? I use this to check for the User-Agent and then to do a fix_nated_sdp() (in route[] and onreply[]) because I am not really sure, if the fix_nated_sdp() could break something. Or should kamailio break nothing here? Sometimes the User-Agent/Server is missing in Session-Progress 183. Therefore a global fix_nated_sdp() would be nice to have. Best regards, Bernhard - Original Message - From: Daniel-Constantin Mierla [mailto:mico...@gmail.com] To: Bernhard Suttner [mailto:bernhard.sutt...@winet.ch] Cc: sr-users@lists.sip-router.org Sent: Mon, 24 Jan 2011 17:09:12 +0100 Subject: Re: [SR-Users] $ua in on_reply route not set? Hello, On 1/24/11 5:03 PM, Bernhard Suttner wrote: Hi, could it be, that the $ua pseudo variable is not set within in a onreply route? (Version 3.1). no, should be set, there was no change in this regard for quite long time. Can you sent the sip reply plus log with debug=3? What is the best alternative for that? Search()? The alternative is $hdr(User-Agent) which is practically returning the same value as $ua, using generic header search mechanism. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com ___ 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
[SR-Users] fix_nated_sdp()
Hi, I have a question about the fix_nated_sdp() function. Different devices (phones, pbxes, gateways) are using Kamailio as SBC. The SBC does t_relay() the message to a pool of FreeSWITCH Servers. The scenario does work quite well but: if a call is initiated from one of the freeswitch server, pass it through kamailio to the PBX and the PBX does have a forwarding back to kamailio and in the end to freeswitch I have no audio because the PBX is behind a NAT and uses private ip addresses in all the SDP of 200 OK and 183 Session Progress. Therefore FreeSWITCH could not do the (very cool) RTP Auto-Adjustment (check if IP in incoming packet is different to the IP in the SDP) to detect the real IP address. Therefore I have to use fix_nated_sdp(). I am not sure, what this function could break therefore I want to be very sure and use it only, if really necessary. Test which User-Agent or Server the peer (in this example the PBX) has, is not really possible because some devices does not send this header. What do you think about the follwoing pseudo code: if (method==INVITE has_sdp() nat_uac_test(8)) fix_nated_sdp(3) and within the onreply route: if (status=~(180|183|200) has_sdp() nat_uac_test(8)) fix_nated_sdp(3) Somebody has a better idea to handle this? What could go wrong? Thanks in advance. Best regards, Bernhard Suttner ___ 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
Re: [SR-Users] Dispatcher Setting
Hi, thanks for your response. It's like that how I handle it currently but some 500 Internal server error from freeswitch will result, that the message is send to all the other freeswitch server, too. They will of course respond in the same way than the first does. All the freeswitch server should have the same version and dialplan logic. Best regards, Bernhard -Ursprüngliche Nachricht- Von: kaiserbo...@googlemail.com [mailto:kaiserbo...@googlemail.com] Im Auftrag von Carsten Bock Gesendet: Donnerstag, 13. Januar 2011 11:39 An: Bernhard Suttner Cc: sr-users@lists.sip-router.org Betreff: Re: [SR-Users] Dispatcher Setting Hi Bernhard, that is simply up to you. But for my production setups so far, i used to catch 408, 5xx and 6xx. - 408 of course, it is a timeout (others were always valid responses in my case) - 5xx is, according to the RFC, a server failure, so that particular server cannot process the request. - 6xx is, according to the RFC, a global failure, a definite answer about the destination. Actually, if you take the RFC seriously, you should not disable based upon a 6xx response, since if it is implemented correctly, no-one can answer the question, not only the choosen destination. However, many network components do not take this too seriously, so that was a valid failed answer for a system Kind regards, Carsten 2011/1/13 Bernhard Suttner bernhard.sutt...@winet.ch: Hi, I am using the Dispatcher module to load-balance the calls to several freeswitch servers. On which responses within the failure_route should be a fallback to another server (dst_next_dst())? I am sure, that on a 408 response there should be a fallback but whats about other 4XX responses and 5XX or 6XX? Thanks in advance. Best regards, Bernhard Suttner ___ 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 -- Carsten Bock Schomburgstr. 80 22767 Hamburg Germany Mobile +49 179 2021244 Home +49 40 34927217 Büro (Verl) +49 5246 801427 Fax +49 40 34927218 mailto:cars...@bock.info ___ 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
[SR-Users] IP Address
Hi, I have a very short question: Kamailio does listen to many IP Addresses and does forward messages to the peer with t_relay. Which IP address will kamailio use for the outgoing traffic as source-ip, if there are many IP addresses on the interface? Is it possible to specify this somehow? Thanks in advance. Best regards, Bernhard ___ 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
[SR-Users] Dispatcher Setting
Hi, I am using the Dispatcher module to load-balance the calls to several freeswitch servers. On which responses within the failure_route should be a fallback to another server (dst_next_dst())? I am sure, that on a 408 response there should be a fallback but whats about other 4XX responses and 5XX or 6XX? Thanks in advance. Best regards, Bernhard Suttner ___ 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
Re: [SR-Users] Change the SIP-To-User
Hi, ah, thanks. Would be good to have a uac_replace_to. Sounds like a 30min task to someone familiar with kamailio programming :-) BR, Bernhard - Original Message - From: Klaus Darilion [mailto:klaus.mailingli...@pernau.at] To: Bernhard Suttner [mailto:bernhard.sutt...@winet.ch] Cc: sr-users@lists.sip-router.org Sent: Fri, 07 Jan 2011 21:59:55 +0100 Subject: Re: [SR-Users] Change the SIP-To-User Bernhard Suttner wrote: Hi, thanks to all responses. I wonder if there is a function to modify the To/From Header stateful. I mean, change the TO header in the INVITE to A and then the 200 OK (for example) reverse the change back to the previous value to B? but the magic $tU is amazing... If you use uac_replace_from(), the changes will stateful and fixed for every in-dialog requests and responses. But there is no similar feature for To header manipulation. regards klaus ___ 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
Re: [SR-Users] Replace number in pseudo variable
Hi, thanks for your fast response. $ou is the original request uri but I tried with $rU and it still does not work. Is it not possible to use a pseudo variable with re.subst ? Best regards, Bernhard -Ursprüngliche Nachricht- Von: Juha Heinanen [mailto:j...@tutpro.com] Gesendet: Montag, 3. Januar 2011 16:04 An: Bernhard Suttner Cc: sr-users@lists.sip-router.org Betreff: [SR-Users] Replace number in pseudo variable Bernhard Suttner writes: I try to replace the user-name of the Request URI and save that into a AVP later. userpart of request uri is $rU. se pvar doc. -- juha ___ 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
[SR-Users] Change the SIP-To-User
Hi, what is the best way to replace the userpart of the SIP-To URI with the content of a pseudo variable? Thanks for every hint. Best regards, Bernhard ___ 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
[SR-Users] Store Source IP in DB
Hi, I am using Kamailio with usrloc, nathelper and register module (and some other). All the data will be stored within a MySQL database. The contact address will be stored within the database. Is it somehow possible to store the SOURCE-IP of a incoming REGISTER from a phone to Kamailio somewhere in the database? It's just for debugging purposes. Best regards, Bernhard ___ 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
Re: [SR-Users] Store Source IP in DB
Hi, thanks for these replys. I already saw the nathelper module but I am not sure, which method will store the received value. Does the nat_uac_test save the received value? Best regards, Bernhard - Original Message - From: Klaus Darilion [mailto:klaus.mailingli...@pernau.at] To: Bernhard Suttner [mailto:bernhard.sutt...@winet.ch] Cc: sr-users@lists.sip-router.org Sent: Tue, 07 Dec 2010 18:59:36 +0100 Subject: Re: [SR-Users] Store Source IP in DB Am 07.12.2010 15:20, schrieb Bernhard Suttner: Hi, I am using Kamailio with usrloc, nathelper and register module (and some other). All the data will be stored within a MySQL database. The contact address will be stored within the database. Is it somehow possible to store the SOURCE-IP of a incoming REGISTER from a phone to Kamailio somewhere in the database? It's just for debugging purposes. If you are doing NAT traversal, e.g. fix_nated_register(), you will have the source IP address of the REGISTER in the location table in the received column. regards Klaus ___ 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
[SR-Users] DoS on Kamailio
Hi, is there a Kamailio module to prevent DoS Attacks like SIP Register Scanners? I know, that there is a way to implement that with Fail2Ban and xlog but it would be nice to have a pure Kamailio solution for that? Any idea is appreciated. Best regards, Bernhard Suttner ___ 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
Re: [SR-Users] loop through variables
Hi,
yeah, that looks like a good solution. But I assume that there should be the
similar way to use the shm or pseudo variables as a array. Maybe someone else
does know more about this way?
Best regards,
Bernhard
-Ursprüngliche Nachricht-
Von: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] Im Auftrag von Alex Balashov
Gesendet: Dienstag, 2. November 2010 14:41
An: sr-users@lists.sip-router.org
Betreff: Re: [SR-Users] loop through variables
On 11/02/2010 09:35 AM, Sergey Okhapkin wrote:
htable module should do the trick to you.
On Tuesday 02 November 2010, Bernhard Suttner wrote:
Hi,
I want something like that:
# define some router to use later in the script (global section)
Mod_param(router(1), 10.10.10.1)
mod_param(router(2), 10.10.10.2)
mod_param(router(3), 10.10.10.3)
Mod_param(router_count, 3)
within the different routes I want to loop through the routers like that:
$var(i)
while($(var(i) $var(router_count)) {
if ($var(router($var(i)) == src_ip) {
.. do something
}
}
Another option is to push the constants into AVP arrays at startup (I
assume there is some sort of init route at this point, otherwise a
module init event-route):
$(avp(s:routers)[0]) = 10.10.10.1;
$(avp(s:routers)[1]) = 10.10.10.2;
$(avp(s:routers)[2]) = 10.10.10.3;
You can then iterate through them, using is_avp_set() to check if
something exists at the current index/array subscript:
$var(i) = 0;
while(is_avp_set($(avp(s:routers)[$var(i)]))) {
xlog(L_INFO, Router is: $(avp(s:routers)[$var(i)])\n;
$var(i) = $var(i) + 1;
}
--
Alex Balashov - Principal
Evariste Systems LLC
1170 Peachtree Street
12th Floor, Suite 1200
Atlanta, GA 30309
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/
___
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
___
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
Re: [SR-Users] loop through variables
Hi, thanks @ all. I will try out the AVP method. Best regards, Bernhard -Ursprüngliche Nachricht- Von: Daniel-Constantin Mierla [mailto:mico...@gmail.com] Gesendet: Dienstag, 2. November 2010 17:16 An: Alex Balashov Cc: Bernhard Suttner; sr-users@lists.sip-router.org Betreff: Re: [SR-Users] loop through variables On 11/2/10 5:13 PM, Alex Balashov wrote: On 11/02/2010 12:01 PM, Daniel-Constantin Mierla wrote: What Alex provided for assignment is not working exactly that way, because an assignment to an AVP always adds at the first position like a stack (the index is not relevant for avp in the left side of assignment unless it is '*' and assigned value is null - meaning delete all avps with that name). That is true. I forgot about that. That still means he could insert the IPs in backward order, though, doesn't it? yes, it inserts them in backward order. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com ___ 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
Re: [SR-Users] loop through variables
Ok. Thx. And where could I add the global variables (the configured routers)? I would like to have them in the global section (like the mod_param). -Ursprüngliche Nachricht- Von: Daniel-Constantin Mierla [mailto:mico...@gmail.com] Gesendet: Dienstag, 2. November 2010 17:57 An: Bernhard Suttner Cc: sr-users@lists.sip-router.org Betreff: Re: [SR-Users] loop through variables On 11/2/10 5:52 PM, Bernhard Suttner wrote: Hi, thanks @ all. I will try out the AVP method. beware that normally the avps are persistent per transaction. Meaning that when the transaction processing (or the message in stateless mode) is done, avps are removed automatically. There are global avps coming from ser 2.0 (therefore you have to use version 3.0+) where the name has to contain 'g.' prefix - $avp(g.foo) Cheers, Daniel Best regards, Bernhard -Ursprüngliche Nachricht- Von: Daniel-Constantin Mierla [mailto:mico...@gmail.com] Gesendet: Dienstag, 2. November 2010 17:16 An: Alex Balashov Cc: Bernhard Suttner; sr-users@lists.sip-router.org Betreff: Re: [SR-Users] loop through variables On 11/2/10 5:13 PM, Alex Balashov wrote: On 11/02/2010 12:01 PM, Daniel-Constantin Mierla wrote: What Alex provided for assignment is not working exactly that way, because an assignment to an AVP always adds at the first position like a stack (the index is not relevant for avp in the left side of assignment unless it is '*' and assigned value is null - meaning delete all avps with that name). That is true. I forgot about that. That still means he could insert the IPs in backward order, though, doesn't it? yes, it inserts them in backward order. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com ___ 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
