[SR-Users] Doubts with Registration
Hello,
I'm having problems with my kamailio, my phones are unable to register into
the Proxy. I have been checking with ngrep-sip and all i'm receiving is many
REGISTER from the phones but never getting some response from the Proxy
The Proxy has a public IP and the phones are behind NAT. The weir thing is
that my logs are not showing any output...and as you can see on the script,
i have a section to receive REGISTER requests
if (is_method("REGISTER")) {
if (!save("location")) {
sl_reply_error();
}
exit;
}
Thanks in advance for your help
kamailio.cfg
Description: Binary data
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Re: [SR-Users] Doubts with Registration
Hello...
2010/11/18 Daniel-Constantin Mierla :
> never write private emails. They will be simply discarded.
Sorry Daniel i did not realize that i was sending the email to you
>
> ngrep gets the traffic before the firewall. Do you have a firewall?
No i do not have a firewall. This is so weird!
>
> start kamailio with log_stderror=yes and debug=5. if you don't see any sip
> message processing, then sip traffic does not reach the application layer.
I do have log_stderror=yes and debug=5 and i do not see any message
and no any answer to my REGISTER, am i missing something in my script?
(in the initial message of this thread you can see the kamailio.cfg)
>
> Cheers,
> Daniel
>
> On 11/18/10 10:39 AM, Danny Dias wrote:
>>
>> Thanks for the answer Daniel,
>>
>> 2010/11/17 Daniel-Constantin Mierla
>>>
>>> Hello,
>>>
>>> On 11/17/10 6:58 PM, Danny Dias wrote:
>>>
>>> Hello,
>>> I'm having problems with my kamailio, my phones are unable to register
>>> into the Proxy. I have been checking with ngrep-sip and all i'm receiving is
>>> many REGISTER from the phones but never getting some response from the Proxy
>>> The Proxy has a public IP and the phones are behind NAT. The weir thing
>>> is that my logs are not showing any output...and as you can see on the
>>> script, i have a section to receive REGISTER requests
>>> if (is_method("REGISTER")) {
>>> if (!save("location")) {
>>> sl_reply_error();
>>> }
>>> exit;
>>> }
>>>
>>> If everything is fine, then no log message should be printed if debug
>>> parameter is low.
>>
>> But i should see in the Kamailio logs the answers, right? or i have to
>> configure this on the scripts?
>>
>> My debug is on "3"
>>
>>> But if you run in debug mode with a high value and nothing is printed,
>>> then the SIP traffic does not reach the>application. A reason for that might
>>> be a firewall, do you run one?
>>
>> The SIP traffic is arriving the machine (at least), i have checked
>> this with the ngrep on the server running kamailio:
>>
>> #
>> U 2010/11/18 10:26:03.331925 212.230.253.164:43686 -> 212.230.19.191:5060
>> REGISTER sip:212.230.19.191 SIP/2.0'
>> Via: SIP/2.0/UDP 10.134.16.164:43686;branch=z9hG4bK247264420;rport'
>> From:;tag=1251867918'
>> To:'
>> Call-ID: 2022066534-1646...@ba.bde.bg.bge'
>> CSeq: 4647 REGISTER'
>>
>> Contact:;reg-id=2;+sip.instance=""'
>> Max-Forwards: 70'
>> User-Agent: Grandstream GXV3140 1.0.1.1'
>> Supported: path'
>> Expires: 3600'
>> Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO,
>> REFER, UPDATE, MESSAGE'
>> Content-Length: 0'
>> '
>>
>> #
>> U 2010/11/18 10:26:05.339925 212.230.253.164:43686 -> 212.230.19.191:5060
>> REGISTER sip:212.230.19.191 SIP/2.0'
>> Via: SIP/2.0/UDP 10.134.16.164:43686;branch=z9hG4bK247264420;rport'
>> From:;tag=1251867918'
>> To:'
>> Call-ID: 2022066534-1646...@ba.bde.bg.bge'
>> CSeq: 4647 REGISTER'
>>
>> Contact:;reg-id=2;+sip.instance=""'
>> Max-Forwards: 70'
>> User-Agent: Grandstream GXV3140 1.0.1.1'
>> Supported: path'
>> Expires: 3600'
>> Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO,
>> REFER, UPDATE, MESSAGE'
>> Content-Length: 0'
>>
>> And i never gets a response from Kamailio, for example, a 200 OK, or a
>> 407 unauthorized...
>>
>> What can i do?
>>
>>> Cheers,
>>> Daniel
>>>
>>> --
>>> Daniel-Constantin Mierla
>>> http://www.asipto.com
>>
>> --
>> Ing. Danny Dias
>> www.DannTEL.net
>
> --
> Daniel-Constantin Mierla
> Kamailio (OpenSER) Advanced Trainings
> Nov 22-25, 2010, Berlin, Germany
> Jan 24-26, 2011, Irvine, CA, USA
> http://www.asipto.com
>
>
--
Ing. Danny Dias
www.DannTEL.net
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Re: [SR-Users] SIP Scanning Attacks Experiences
Thanks Daniel...Very nice article!!! 2010/11/18 Daniel-Constantin Mierla : > Hello, > > during the testing period of Kamailio 3.1.0, while running it at > voipuser.org, I had the chance to watch live and analyze a SIP scanning > attack. Yesterday I noticed another one by looking at Siremis 2.0 charts, > therefore I wrote an article with some hints about what you can use to > protect your SIP services within Kamailio configuration file. > > You can read it at: > * http://asipto.com/u/i > > Hope is going to be useful for many of you! > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > Kamailio (OpenSER) Advanced Trainings > Nov 22-25, 2010, Berlin, Germany > Jan 24-26, 2011, Irvine, CA, USA > 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 > -- Ing. Danny Dias www.DannTEL.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
Re: [SR-Users] Doubts with Registration
Hello Daniel, 2010/11/19 Daniel-Constantin Mierla : > > > On 11/18/10 3:56 PM, Danny Dias wrote: >> >> Hello... >> >> 2010/11/18 Daniel-Constantin Mierla: >>> >>> never write private emails. They will be simply discarded. >> >> Sorry Daniel i did not realize that i was sending the email to you >> >>> ngrep gets the traffic before the firewall. Do you have a firewall? >> >> No i do not have a firewall. This is so weird! >> >>> start kamailio with log_stderror=yes and debug=5. if you don't see any >>> sip >>> message processing, then sip traffic does not reach the application >>> layer. >> >> I do have log_stderror=yes and debug=5 and i do not see any message >> and no any answer to my REGISTER, am i missing something in my script? >> (in the initial message of this thread you can see the kamailio.cfg) > > How do you start kamailio? I start Kamailio with /etc/init.d/kamailio restart: r...@kamailio-danndp:~# /etc/init.d/kamailio restart Restarting Kamailio SIP server: kamailioListening on udp: 212.230.19.191 [212.230.19.191]:5060 Aliases: udp: Kamailio-DANNDP:5060 > > When debug=5 you get log messages before the config is executed, as soon as > something is received. So if you don't see anything, then kamailio does not > receive anything on the sockets it listens to. Attached you will see the tail -f of my kamailio.log while i'm restarting kamailio > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > Kamailio (OpenSER) Advanced Trainings > Nov 22-25, 2010, Berlin, Germany > Jan 24-26, 2011, Irvine, CA, USA > http://www.asipto.com > > -- Ing. Danny Dias www.DannTEL.net tail_kamailio_log Description: Binary data ___ 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] Doubts with Registration
Thanks Iñaki,
2010/11/19 Iñaki Baz Castillo :
> 2010/11/19 Danny Dias :
>> I start Kamailio with /etc/init.d/kamailio restart:
>>
>> r...@kamailio-danndp:~# /etc/init.d/kamailio restart
>> Restarting Kamailio SIP server: kamailioListening on
>> udp: 212.230.19.191 [212.230.19.191]:5060
>> Aliases:
>> udp: Kamailio-DANNDP:5060
>
> The daemonized mode of Kamailio 3.X has been improved so it would
> result in error exit status in case the main process fails to start
> (so by invoking init service you would receive an error and exit
> status != 0).
That's good to know.
>
> However in Kamailio 1.5.X this is not true, and the daemon mode exits
> with zero (success) without waiting for the main process to properly
> start, so you get "success" after invoking the init service but
> Kamailio fails to start (due to errors in modules configuration,
> listening IP's).
I decided to comment the following lines in my kamailio.cfg:
#loadmodule "auth.so"
#loadmodule "auth_db.so"
#loadmodule "alias_db.so"
#modparam("auth_db", "calculate_ha1", yes)
#modparam("auth_db", "password_column", "password")
And now, restarting kamailio works:
r...@kamailio-danndp:/home# /etc/init.d/kamailio restart
Restarting Kamailio SIP server: kamailioListening on
udp: 212.230.19.191 [212.230.19.191]:5060
Aliases:
udp: Kamailio-DANNDP:5060
r...@kamailio-danndp:/home# ps -ef | grep kamailio
root 20482 1 0 19:22 ?00:00:00 /usr/sbin/kamailio -P
/var/run/kamailio/kamailio.pid -m 64 -u root -g root
root 20483 20482 0 19:22 ?00:00:00 /usr/sbin/kamailio -P
/var/run/kamailio/kamailio.pid -m 64 -u root -g root
root 20484 20482 0 19:22 ?00:00:00 /usr/sbin/kamailio -P
/var/run/kamailio/kamailio.pid -m 64 -u root -g root
root 20485 20482 0 19:22 ?00:00:00 /usr/sbin/kamailio -P
/var/run/kamailio/kamailio.pid -m 64 -u root -g root
root 20486 20482 0 19:22 ?00:00:00 /usr/sbin/kamailio -P
/var/run/kamailio/kamailio.pid -m 64 -u root -g root
root 20487 20482 0 19:22 ?00:00:00 /usr/sbin/kamailio -P
/var/run/kamailio/kamailio.pid -m 64 -u root -g root
root 20488 20482 0 19:22 ?00:00:00 /usr/sbin/kamailio -P
/var/run/kamailio/kamailio.pid -m 64 -u root -g root
root 20489 20482 0 19:22 ?00:00:00 /usr/sbin/kamailio -P
/var/run/kamailio/kamailio.pid -m 64 -u root -g root
root 20490 20482 0 19:22 ?00:00:00 /usr/sbin/kamailio -P
/var/run/kamailio/kamailio.pid -m 64 -u root -g root
root 20491 20482 0 19:22 ?00:00:00 /usr/sbin/kamailio -P
/var/run/kamailio/kamailio.pid -m 64 -u root -g root
root 20492 20482 0 19:22 ?00:00:00 /usr/sbin/kamailio -P
/var/run/kamailio/kamailio.pid -m 64 -u root -g root
root 20493 20482 0 19:22 ?00:00:00 /usr/sbin/kamailio -P
/var/run/kamailio/kamailio.pid -m 64 -u root -g root
root 20495 20429 0 19:23 pts/000:00:00 grep kamailio
By the way, in my kamctlr i have the #STORE_PLAINTEXT_PW=1, that means
that it wont store the the passwords in the subscriber table...right¿?
Also, the REGISTER are having the same result:
interface: any
filter: (ip or ip6) and ( port 5060 )
match: 100
#
U 2010/11/19 19:25:14.239925 212.230.253.164:10091 -> 212.230.19.191:5060
REGISTER sip:212.230.19.191 SIP/2.0'
Via: SIP/2.0/UDP 10.134.16.164:10091;branch=z9hG4bK1590232060;rport'
From: ;tag=1251867918'
To: '
Call-ID: 2022066534-1646...@ba.bde.bg.bge'
CSeq: 6878 REGISTER'
Contact:
;reg-id=2;+sip.instance=""'
Authorization: Digest username="101", realm="212.230.19.191",
nonce="4ce5bb380047aa806944ef2f6754dbff9fb1757db9ae",
uri="sip:212.230.19.191", response="1f0f04efa3e0271c6937007dc9fd77e7",
algorithm=MD5'
Max-Forwards: 70'
User-Agent: Grandstream GXV3140 1.0.1.1'
Supported: path'
Expires: 3600'
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO,
REFER, UPDATE, MESSAGE'
Content-Length: 0'
'
#
U 2010/11/19 19:25:18.247925 212.230.253.164:10091 -> 212.230.19.191:5060
REGISTER sip:212.230.19.191 SIP/2.0'
Via: SIP/2.0/UDP 10.134.16.164:10091;branch=z9hG4bK1590232060;rport'
From: ;tag=1251867918'
To: '
Call-ID: 2022066534-1646...@ba.bde.bg.bge'
CSeq: 6878 REGISTER'
Contact:
;reg-id=2;+sip.instance=""'
Authorization: Digest username="101", realm="212.230.19.191",
nonce="4ce5bb380047aa806944ef2f6754dbff9fb1757db9ae",
uri="sip:212.230.19.191", response="1f0f04efa3e0271c6937007dc9fd77e7",
algorithm=MD5'
Max-Forwards: 70'
User-Agent: Grandstream GXV3140 1.0.1.1'
Supported
Re: [SR-Users] Doubts with Registration
Thanks Iñaki,
>
> 2010/11/19 Iñaki Baz Castillo :
>> 2010/11/19 Danny Dias :
>>> I start Kamailio with /etc/init.d/kamailio restart:
>>>
>>> r...@kamailio-danndp:~# /etc/init.d/kamailio restart
>>> Restarting Kamailio SIP server: kamailioListening on
>>> udp: 212.230.19.191 [212.230.19.191]:5060
>>> Aliases:
>>> udp: Kamailio-DANNDP:5060
>>
>> The daemonized mode of Kamailio 3.X has been improved so it would
>> result in error exit status in case the main process fails to start
>> (so by invoking init service you would receive an error and exit
>> status != 0).
>
That's good to know.
>
>>
>> However in Kamailio 1.5.X this is not true, and the daemon mode exits
>> with zero (success) without waiting for the main process to properly
>> start, so you get "success" after invoking the init service but
>> Kamailio fails to start (due to errors in modules configuration,
>> listening IP's).
>
I decided to comment the following lines in my kamailio.cfg:
#loadmodule "auth.so"
#loadmodule "auth_db.so"
#loadmodule "alias_db.so"
#modparam("auth_db", "calculate_ha1", yes)
#modparam("auth_db", "password_column", "password")
And now, restarting kamailio works:
r...@kamailio-danndp:/home# /etc/init.d/kamailio restart
Restarting Kamailio SIP server: kamailioListening on
udp: 212.230.19.191 [212.230.19.191]:5060
Aliases:
udp: Kamailio-DANNDP:5060
r...@kamailio-danndp:/home# ps -ef | grep kamailio
root 20482 1 0 19:22 ? 00:00:00 /usr/sbin/kamailio -P
/var/run/kamailio/kamailio.pid -m 64 -u root -g root
root 20483 20482 0 19:22 ? 00:00:00 /usr/sbin/kamailio -P
/var/run/kamailio/kamailio.pid -m 64 -u root -g root
root 20484 20482 0 19:22 ? 00:00:00 /usr/sbin/kamailio -P
/var/run/kamailio/kamailio.pid -m 64 -u root -g root
root 20485 20482 0 19:22 ? 00:00:00 /usr/sbin/kamailio -P
/var/run/kamailio/kamailio.pid -m 64 -u root -g root
root 20486 20482 0 19:22 ? 00:00:00 /usr/sbin/kamailio -P
/var/run/kamailio/kamailio.pid -m 64 -u root -g root
root 20487 20482 0 19:22 ? 00:00:00 /usr/sbin/kamailio -P
/var/run/kamailio/kamailio.pid -m 64 -u root -g root
root 20488 20482 0 19:22 ? 00:00:00 /usr/sbin/kamailio -P
/var/run/kamailio/kamailio.pid -m 64 -u root -g root
root 20489 20482 0 19:22 ? 00:00:00 /usr/sbin/kamailio -P
/var/run/kamailio/kamailio.pid -m 64 -u root -g root
root 20490 20482 0 19:22 ? 00:00:00 /usr/sbin/kamailio -P
/var/run/kamailio/kamailio.pid -m 64 -u root -g root
root 20491 20482 0 19:22 ? 00:00:00 /usr/sbin/kamailio -P
/var/run/kamailio/kamailio.pid -m 64 -u root -g root
root 20492 20482 0 19:22 ? 00:00:00 /usr/sbin/kamailio -P
/var/run/kamailio/kamailio.pid -m 64 -u root -g root
root 20493 20482 0 19:22 ? 00:00:00 /usr/sbin/kamailio -P
/var/run/kamailio/kamailio.pid -m 64 -u root -g root
root 20495 20429 0 19:23 pts/0 00:00:00 grep kamailio
By the way, in my kamctlr i have the #STORE_PLAINTEXT_PW=1, that means
that it wont store the the passwords in the subscriber table...right¿?
Also, the REGISTER are having the same result:
interface: any
filter: (ip or ip6) and ( port 5060 )
match: 100
#
U 2010/11/19 19:25:14.239925 212.230.253.164:10091 -> 212.230.19.191:5060
REGISTER sip:212.230.19.191 SIP/2.0'
Via: SIP/2.0/UDP 10.134.16.164:10091;branch=z9hG4bK1590232060;rport'
From: ;tag=1251867918'
To: '
Call-ID: 2022066534-1646...@ba.bde.bg.bge'
CSeq: 6878 REGISTER'
Contact:
;reg-id=2;+sip.instance=""'
Authorization: Digest username="101", realm="212.230.19.191",
nonce="4ce5bb380047aa806944ef2f6754dbff9fb1757db9ae",
uri="sip:212.230.19.191", response="1f0f04efa3e0271c6937007dc9fd77e7",
algorithm=MD5'
Max-Forwards: 70'
User-Agent: Grandstream GXV3140 1.0.1.1'
Supported: path'
Expires: 3600'
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO,
REFER, UPDATE, MESSAGE'
Content-Length: 0'
'
#
U 2010/11/19 19:25:18.247925 212.230.253.164:10091 -> 212.230.19.191:5060
REGISTER sip:212.230.19.191 SIP/2.0'
Via: SIP/2.0/UDP 10.134.16.164:10091;branch=z9hG4bK1590232060;rport'
From: ;tag=1251867918'
To: '
Call-ID: 2022066534-1646...@ba.bde.bg.bge'
CSeq: 6878 REGISTER'
Contact:
;reg-id=2;+sip.instance=""'
Authorization: Digest username="101", realm="212.230.19.191",
nonce="4ce5bb380047aa806944ef2f6754dbff9fb1757db9ae",
uri="sip:212.230.19.191", response="1f0f
Re: [SR-Users] Doubts with Registration
You were rigth Iñaki IP tables problem BEst regards and thanks for your help, also thanks to Daniel constantin! 2010/11/19 Iñaki Baz Castillo : > 2010/11/19 Iñaki Baz Castillo : >> 2010/11/19 Danny Dias : >>> And never receiving and answer from Kamailio, attached you will find >>> my kamailio.cfg, if you can check it and tell me what should i check i >>> would appreciate it!!! >> >> Your iptables are blocking the port 5060. Sure. > > Please show the output of: > > iptables -L -n | grep 5060 > > (you can remove/change displayed IP's for privacy). > > > -- > Iñaki Baz Castillo > > -- Ing. Danny Dias www.DannTEL.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] SIP Recorder
Hello my friends, I have a requeriment, which indicates that i have to record every SIP conversation between peers (also for callings to the PSTN); the recording server will be built for our company following this requeriments (also requested for the client): My doubt is: How can i handle sip conversations recording when all the calls are passing through a Proxy Server? I do understand that the media is always peer to peer and the signaling goes through the Proxy, but in this case the media not only has to pass between the peers because it must be recorded. How should i handle this? ___ 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] SIP Recorder
Media NEVER goes through a Proxy core...the question is, how should i record conversations when the calls are all passing through a sip proxy? some lights will be enough for me :) 2011/1/26 Jeremya : > Whoops! some SIP traffic IS peer-to-peer. > > Jeremya wrote: > > Danny Dias wrote: > > > Hello my friends, > > I have a requeriment, which indicates that i have to record every SIP > conversation between peers (also for callings to the PSTN); the > recording server will be built for our company following this > requeriments (also requested for the client): > > My doubt is: How can i handle sip conversations recording when all the > calls are passing through a Proxy Server? I do understand that the > media is always peer to peer and the signaling goes through the Proxy, > but in this case the media not only has to pass between the peers > because it must be recorded. > > How should i handle this? > > ___ > 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 > > > > some media is not peer-to-peer. Especially stuff like BYE and NOTIFY. > Then it is direct to the originator contact address. > > Unless you have both ends set up correctly, or you have 'adjusted' the > SIP traffic, then some stuff may be lost. > > ___ > 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 > > -- Ing. Danny Dias www.DannTEL.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
Re: [SR-Users] SIP Recorder
Thanks Jeremya, but it's a requeriment from the client to record the calls through an external server and not with rtpproxys, my question is how the media should be handled in order to record the conversations if the server is external? Signaling: Phone_A <---> Proxy <---> Phone_B Media: Phone_A <---> SIP RECORDER <---> Phone_B (Changing the SDP headers to send RTP to the IP of the SIP RECORDER). The main problem is that the recording must be made in ACTIVE way, it means, we should record (IN+OUT) in Phone A, and the same in B, 2 recording for each call...the customer says that it's working now in his arquitecture (its analog), and we made the same with the IP technology...resuming: with a sip recorder in the middle of the media should work right? 2011/1/26 Jeremya > Someone correct me if I'm wrong, but I've seen enough examples of > out-of-dialog requests (e.g. BYE) not using the record route to wonder > if this is in fact required for a new dialog. > > I've managed this by setting outbound proxy, but a general rule would help. > > marius zbihlei wrote: > > On 01/26/2011 03:51 PM, Danny Dias wrote: > >> Media NEVER goes through a Proxy core...the question is, how should i > >> record conversations when the calls are all passing through a sip > >> proxy? some lights will be enough for me :) > >> > >> > > > > Hello, > > > > Use Record-Route headers to force in-dialog requests to have the same > > path as the original (also you might want to the a look to Path header > > for REGISTER requests). This will solve the signaling part For Media, > > I think rtpproxy module will achieve what you want (ignore NAT - > > basically all you need is to re-write some media attributes in the > > sdp). The rtpproxy daemon will also be needed. > > > > Cheers, > > > > Marius > >> 2011/1/26 Jeremya: > >> > >>> Whoops! some SIP traffic IS peer-to-peer. > >>> > >>> Jeremya wrote: > >>> > >>> Danny Dias wrote: > >>> > >>> > >>> Hello my friends, > >>> > >>> I have a requeriment, which indicates that i have to record every SIP > >>> conversation between peers (also for callings to the PSTN); the > >>> recording server will be built for our company following this > >>> requeriments (also requested for the client): > >>> > >>> My doubt is: How can i handle sip conversations recording when all the > >>> calls are passing through a Proxy Server? I do understand that the > >>> media is always peer to peer and the signaling goes through the Proxy, > >>> but in this case the media not only has to pass between the peers > >>> because it must be recorded. > >>> > >>> How should i handle this? > >>> > >>> ___ > >>> 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 > >>> > >>> > >>> > >>> some media is not peer-to-peer. Especially stuff like BYE and NOTIFY. > >>> Then it is direct to the originator contact address. > >>> > >>> Unless you have both ends set up correctly, or you have 'adjusted' the > >>> SIP traffic, then some stuff may be lost. > >>> > >>> ___ > >>> 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 > >>> > >>> > >>> > >> > >> > >> > > > > > > ___ > > 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 > -- Ing. Danny Dias www.DannTEL.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
Re: [SR-Users] SIP Recorder
Many thanks Jaremya, The main problem is that both terminals, SHALL (required and must not be changed, because of standards of EUROCAE ED-137 Part3) initiate a session with the recorder server (a commercial one, can't use Asterisk for my disgrace) sending INVITE and receiving the subsequent responses from sip recording server to stablish the session with it...after this, when the media starts to go directly peer to peer (the normal call), the terminals (specials ones) must summarize the IN+OUT audio to the recording server and through rtsp the media should be recorded...it's weird, but thats the requirement :S i mean signaling: A>PROXY>B (the normal procedure) At the same time, this must be done: (I'm not sure how to do this...the proxy could be out of this or not, not sure :() A ---INVITE---> SIP_PROXY ---INVITE---> SIP_RECORDER B ---INVITE---> SIP_RECORDER --INVITE--> SIP_RECORDER Then, The audio will go directly from A to B (because of the normal procedures), and also, A and B, will summarize IN+OUT on each site and send this result through RTSP to the recording server (this is not important to the proxy righ not)...My real doubt is how to stablish the session between the peers A and B to the recording server through the Proxy and also (at the same time) continue with the normal flow of the call (invite from a to b, 200 ok viceversa etc etc...) Should i use some function like t_replicate to send 2 invites like this: A --INVITE--> PROXY --INVITE--> B . . INVITE . RECORDER SERVER But the problem here is that the session between A and PROXY would be OK, but i can't see the way how B should send INVITE to the recorder server.. I hope to be clear on my problem :( and i know it looks very weird, but it's the requirement of the document mentioned above Thanks in advance!!! 2011/1/26 > Danny Dias escribió: > > Thanks Jeremya, but it's a requeriment from the client to record the calls >> through an external server and not with rtpproxys, my question is how the >> media should be handled in order to record the conversations if the server >> is external? >> >> Signaling: Phone_A <---> Proxy <---> Phone_B >> >> Media: Phone_A <---> SIP RECORDER <---> Phone_B (Changing the SDP headers >> to >> send RTP to the IP of the SIP RECORDER). The main problem is that the >> recording must be made in ACTIVE way, it means, we should record (IN+OUT) >> in >> Phone A, and the same in B, 2 recording for each call...the customer says >> that it's working now in his arquitecture (its analog), and we made the >> same >> with the IP technology...resuming: with a sip recorder in the middle of >> the >> media should work right? >> > > > 2 ways of doing that: > > a) > Signaling: A <-> Proxy <-> B2BUA (recorder) <-> B > Media: A <-> B2BUA <-> B > > b) Prefered way > Signaling: A <-> Proxy <-> B > Media: A<-> RTPPROXY <-> B > > At the end, both solutions are THE SAME, what you do is to tell A to send > media to the B2BUA or the RTPPRoxy. > > As a matters of scale ... b) solution is the best one. > > Also, another things to take into account are: > > 1- Transcoding issues (RTPPRoxy does not do transconding, not easly) > 2- Secured RTP (ZRTP, SRTP, etc.) > 3- LAG in audio. > > > > > This message was sent using IMP, the Internet Messaging Program. > > > ___ > 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] SIP Recorder
OOps, made a mistake on tipping.take a look down please... 2011/1/26 Danny Dias > Many thanks Jaremya, > > The main problem is that both terminals, SHALL (required and must not be > changed, because of standards of EUROCAE ED-137 Part3) initiate a session > with the recorder server (a commercial one, can't use Asterisk for my > disgrace) sending INVITE and receiving the subsequent responses from sip > recording server to stablish the session with it...after this, when the > media starts to go directly peer to peer (the normal call), the terminals > (specials ones) must summarize the IN+OUT audio to the recording server and > through rtsp the media should be recorded...it's weird, but thats the > requirement :S > > i mean > > signaling: A>PROXY>B (the normal procedure) > > At the same time, this must be done: (I'm not sure how to do this...the > proxy could be out of this or not, not sure :() > > A ---INVITE---> SIP_PROXY ---INVITE---> SIP_RECORDER > B ---INVITE---> SIP_RECORDER --INVITE--> SIP_RECORDER > B ---INVITE---> SIP_PROXY ---INVITE---> SIP_RECORDER > Then, The audio will go directly from A to B (because of the normal > procedures), and also, A and B, will summarize IN+OUT on each site and send > this result through RTSP to the recording server (this is not important to > the proxy righ not)...My real doubt is how to stablish the session between > the peers A and B to the recording server through the Proxy and also (at the > same time) continue with the normal flow of the call (invite from a to b, > 200 ok viceversa etc etc...) > > Should i use some function like t_replicate to send 2 invites like this: > > A --INVITE--> PROXY --INVITE--> B > . > . INVITE > . > RECORDER SERVER > > > But the problem here is that the session between A and PROXY would be OK, > but i can't see the way how B should send INVITE to the recorder server.. > > I hope to be clear on my problem :( and i know it looks very weird, but > it's the requirement of the document mentioned above > > Thanks in advance!!! > > > > 2011/1/26 > > Danny Dias escribió: >> >> Thanks Jeremya, but it's a requeriment from the client to record the >>> calls >>> through an external server and not with rtpproxys, my question is how the >>> media should be handled in order to record the conversations if the >>> server >>> is external? >>> >>> Signaling: Phone_A <---> Proxy <---> Phone_B >>> >>> Media: Phone_A <---> SIP RECORDER <---> Phone_B (Changing the SDP headers >>> to >>> send RTP to the IP of the SIP RECORDER). The main problem is that the >>> recording must be made in ACTIVE way, it means, we should record (IN+OUT) >>> in >>> Phone A, and the same in B, 2 recording for each call...the customer says >>> that it's working now in his arquitecture (its analog), and we made the >>> same >>> with the IP technology...resuming: with a sip recorder in the middle of >>> the >>> media should work right? >>> >> >> >> 2 ways of doing that: >> >> a) >> Signaling: A <-> Proxy <-> B2BUA (recorder) <-> B >> Media: A <-> B2BUA <-> B >> >> b) Prefered way >> Signaling: A <-> Proxy <-> B >> Media: A<-> RTPPROXY <-> B >> >> At the end, both solutions are THE SAME, what you do is to tell A to send >> media to the B2BUA or the RTPPRoxy. >> >> As a matters of scale ... b) solution is the best one. >> >> Also, another things to take into account are: >> >> 1- Transcoding issues (RTPPRoxy does not do transconding, not easly) >> 2- Secured RTP (ZRTP, SRTP, etc.) >> 3- LAG in audio. >> >> >> >> >> This message was sent using IMP, the Internet Messaging Program. >> >> >> ___ >> 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 >> > > > > -- Ing. Danny Dias www.DannTEL.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] Redundancy between 2 Kamailio servers
Hello my friends, I wonder if someone has done this before? i would like to implement 2 Kamailio servers with redundancy, something like the following: SERVER_A is working as the primary sip proxy (virtual IP as the sip signaling), if it fails, the other server (sleeping) should UP the virtual IP and takes all the traffici guess there is many Linux implementations thah can do this, but i would like to know if there is someone that has done this before and hear some recomendations... I've read some difficulty in the synchronisation of registrations because Kamailio works best when it stores registrations in memory and registrations are constantly changing - they expire and are renewed, as well as new ones joining and old ones leaving. To make the failover solution function seamlessly, it is necessary to synchronise the in-memory registrations between the primary and the backup server . This can be done by forking a copy of the registration request to the backup server, but there are some practical problems in doing this, has anyone do something with this? Thanks in advance! ___ 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] SIP Recorder
Hi Iñaki, 2011/1/27 Iñaki Baz Castillo > 2011/1/26 Danny Dias : > > i mean > > signaling: A>PROXY>B (the normal procedure) > > At the same time, this must be done: (I'm not sure how to do this...the > > proxy could be out of this or not, not sure :() > > A ---INVITE---> SIP_PROXY ---INVITE---> SIP_RECORDER > > B ---INVITE---> SIP_RECORDER --INVITE--> SIP_RECORDER > B ---INVITE---> SIP_PROXY --INVITE--> SIP_RECORDER Each peer must send an INVITE to the sip_recorder server, to stablish a session with it... > Hi, is such SIP_PROXY an instance of Kamailio/SER? > YEs, the proxys are Kamailio. > > -- > Iñaki Baz Castillo > > -- Ing. Danny Dias www.DannTEL.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
Re: [SR-Users] Redundancy between 2 Kamailio servers
Thanks Alex... 2011/1/27 Alex Hermann > On Thursday 27 January 2011, Klaus Darilion wrote: > > Am 27.01.2011 11:21, schrieb Danny Dias: > > > I've read some difficulty in the synchronisation of registrations > because > > > Kamailio works best when it stores registrations in memory and > > > registrations are constantly changing - they expire and are renewed, as > > > well as new ones joining and old ones leaving. To make the failover > > > solution function seamlessly, it is necessary to synchronise the > > > in-memory registrations between the primary and the backup server . > This > > > can be done by forking a copy of the registration request to the backup > > > server, but there are some practical problems in doing this, has anyone > > > do something with this? > > What problems are you referring to? I use this for some years now without > any > problems. > > > I checked for some problems here: http://www.smartvox.co.uk/astfaq_ha_failover_ideas.htm > > 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. > Do you mean that both Kamailio-1 and Kamailio-2 will be as primary server? and the clients will register in the 2 machines? and also they will bind to the ip of the HA? sorry my friend but i do not understand very well, i'm quite new with redundant systems, could you please explain a little? > > > > I think most people either have a database (which is highly-available by > > itself) which is used by both proxies, or every proxy has a local > > database and the synchronization is on DB level (e.g. master-slave > > replication, btw: does somebody know if usrloc DB queries are suitable > > for master-master replication?) > > Last time i tried, they are not, at least not in writeback mode. One proxy > is > expiring records from the DB which the other proxy is trying to update. > Maybe > DB-only mode will work, but that has some practical (performcance) > problems. > -- > Greetings, > > Alex Hermann > > > ___ > 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 > -- Ing. Danny Dias www.DannTEL.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
Re: [SR-Users] Redundancy between 2 Kamailio servers
Do you mean that both Kamailio-1 and Kamailio-2 will be as primary server? and the clients will register in the 2 machines? and also they will bind to the ip of the HA? sorry my friend but i do not understand very well, i'm quite new with redundant systems, could you please explain a little please? 2011/1/27 Alex Hermann > On Thursday 27 January 2011, Klaus Darilion wrote: > > Alex, do you also do NAT keep-alive from the proxies? If yes, are you > > sending them from both servers at the same time? > > No, we require clients to sent nat-keepalives. It is much more efficient. > In > addition, the registrars are not directly accessible by clients, they go > via > load-balancers. Clients keep the NAT binding with the balancer, not the > registrar. > > -- > Greetings, > > Alex Hermann > > > ___ > 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] Redundancy between 2 Kamailio servers
Thanks Iñaki, 2011/1/27 Iñaki Baz Castillo > 2011/1/27 Danny Dias : > > Do you mean that both Kamailio-1 and Kamailio-2 will be as primary > server? > > and the clients will register in the 2 machines? and also they will bind > to > > the ip of the HA? sorry my friend but i do not understand very well, i'm > > quite new with redundant systems, could you please explain a little > please? > > 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? > 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? > Another option without using realtime DB storage is replicating the > REGISTER from one Kamailio to the other (t_replicate method) but it > requires both kamailios being running at the same time (so > net.ipv4.ip_nonlocal_bind must be 1) and kamailios must NOT be managed > by HA. Also it requires some other considerations. > > -- > Iñaki Baz Castillo > > -- Ing. Danny Dias www.DannTEL.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
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
> 2011/1/27 Danny Dias :
> >> 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
>
>
___
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[SR-Users] authentication is not working
Hello my friends,
I'm trying to configure authentication on my Kamailio and is not working at
all :(
I've added the following to the script to make it work: (but it doesn't)
...
loadmodule "auth.so"
loadmodule "auth_db.so"
...
modparam("usrloc", "db_url",
"mysql://kamailio:kamailiorw@localhost/kamailio")
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("auth_db", "db_url",
"mysql://kamailio:kamailiorw@localhost/kamailio")
modparam("auth_db", "load_credentials", "")
...
if (!(method=="REGISTER") && from_uri==myself) /*no multidomain
version*/
{
if (!proxy_authorize("", "subscriber")) {
proxy_challenge("", "0");
exit;
}
if (!db_check_from()) {
sl_send_reply("403","Forbidden auth ID");
exit;
}
consume_credentials();
}
...
if (is_method("REGISTER"))
{
# authenticate the REGISTER requests (uncomment to enable
auth)
if (!www_authorize("", "subscriber"))
{
www_challenge("", "0");
exit;
}
##
if (!db_check_to())
{
sl_send_reply("403","Forbidden auth ID");
exit;
}
if (!save("location"))
sl_reply_error();
exit;
}
But is not working at all...take a look:
#
U 2011/02/03 09:31:04.402891 172.30.140.22:48752 -> 172.30.140.8:5060
REGISTER sip:172.30.140.8 SIP/2.0
Via: SIP/2.0/UDP 172.30.140.22:48752
;branch=z9hG4bK-d87543-9a54af22967ae417-1--d87543-;rport
Max-Forwards: 70
Contact:
To: "1000">
From: "1000">;tag=cd3e2323
Call-ID: MmU0YjM1NThiNTg0ZjhiNGM4ODA4ZmU1YWFiYjBmNTc.
CSeq: 1 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO
User-Agent: X-Lite release 1011s stamp 41150
Content-Length: 0
#
U 2011/02/03 09:31:04.404039 172.30.140.8:5060 -> 172.30.140.22:48752
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.140.22:48752
;branch=z9hG4bK-d87543-9a54af22967ae417-1--d87543-;rport=48752
To: "1000"
>;tag=c97b4d1cb1f3d0da549e06a8d482ef63.34dc
From: "1000">;tag=cd3e2323
Call-ID: MmU0YjM1NThiNTg0ZjhiNGM4ODA4ZmU1YWFiYjBmNTc.
CSeq: 1 REGISTER
Contact: ;expires=3600
Content-Length: 0
Am i missing something in my configuration?
Thanks in advance!!!
___
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Re: [SR-Users] authentication is not working
2011/2/3 Jon Bonilla > El Thu, 3 Feb 2011 11:12:30 +0100 > Danny Dias escribió: > > > Hello my friends, > > > > I'm trying to configure authentication on my Kamailio and is not working > at > > all :( > > > > I've added the following to the script to make it work: (but it doesn't) > > > > . > > Are you asking the same questions in both kamailio and opensips accounts > with > different aliases? > > > different emails...what's the problem? i have a doubt and then i ask...thats the purpose of the forum, if you are not going to help, don't answer... > Why would you do that? Toyima Dias? Really? > > > ___ > 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] authentication is not working
Thanks Daniel...i appreciate your help, i will let you know if i could solve
the problem!
2011/2/3 Daniel-Constantin Mierla
> What version of kamailio are you using? If it is 3.1, then load debugger
> module and enable cfg trace. Then you will see what lines in the
> configuration file are executed.
>
> For older versions (also for 3.1), you can add xlog() lines in your config
> to troubleshoot it.
>
> Cheers,
> Daniel
>
>
> On 2/3/11 3:46 PM, Klaus Darilion wrote:
>
>> Restart Kamailio. Make sure that it is it really restarts:
>>
>> /etc/init.d/kamailio stop
>> ps aux|grep kamailio
>> # if there are some processes left, kill them
>> killall kamailio
>>
>> ps aux|grep kamailio
>> # if there are still some processes left, kill them harder!
>> killall -9 kamailio
>>
>> /etc/init.d/kamailio start
>>
>>
>> make sure Kamailio is really using your configuration file
>>
>> klaus
>>
>>
>> Am 03.02.2011 11:12, schrieb Danny Dias:
>>
>>> Hello my friends,
>>>
>>> I'm trying to configure authentication on my Kamailio and is not working
>>> at
>>> all :(
>>>
>>> I've added the following to the script to make it work: (but it doesn't)
>>>
>>> ...
>>> loadmodule "auth.so"
>>> loadmodule "auth_db.so"
>>> ...
>>> modparam("usrloc", "db_url",
>>> "mysql://kamailio:kamailiorw@localhost
>>> /kamailio")
>>> modparam("auth_db", "calculate_ha1", yes)
>>> modparam("auth_db", "password_column", "password")
>>> modparam("auth_db", "db_url",
>>> "mysql://kamailio:kamailiorw@localhost
>>> /kamailio")
>>> modparam("auth_db", "load_credentials", "")
>>> ...
>>> if (!(method=="REGISTER")&& from_uri==myself) /*no multidomain
>>> version*/
>>> {
>>> if (!proxy_authorize("", "subscriber")) {
>>> proxy_challenge("", "0");
>>> exit;
>>> }
>>> if (!db_check_from()) {
>>> sl_send_reply("403","Forbidden auth ID");
>>> exit;
>>> }
>>> consume_credentials();
>>> }
>>> ...
>>>
>>> if (is_method("REGISTER"))
>>> {
>>> # authenticate the REGISTER requests (uncomment to enable
>>> auth)
>>> if (!www_authorize("", "subscriber"))
>>> {
>>> www_challenge("", "0");
>>> exit;
>>> }
>>> ##
>>> if (!db_check_to())
>>> {
>>> sl_send_reply("403","Forbidden auth ID");
>>> exit;
>>> }
>>> if (!save("location"))
>>> sl_reply_error();
>>> exit;
>>> }
>>>
>>>
>>> But is not working at all...take a look:
>>>
>>> #
>>> U 2011/02/03 09:31:04.402891 172.30.140.22:48752 -> 172.30.140.8:5060
>>> REGISTER sip:172.30.140.8 SIP/2.0
>>> Via: SIP/2.0/UDP 172.30.140.22:48752
>>> ;branch=z9hG4bK-d87543-9a54af22967ae417-1--d87543-;rport
>>> Max-Forwards: 70
>>> Contact:
>>> To: "1000"<
>>> sip%3A1000@172.30.140.8 >>
>>> From: "1000"<
>>> sip%3A1000@172.30.140.8 >>;tag=cd3e2323
>>> Call-ID: MmU0YjM1NThiNTg0ZjhiNGM4ODA4ZmU1YWFiYjBmNTc.
>>> CSeq: 1 REGISTER
>>> Expires: 3600
>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
>>> SUBSCRIBE,
>>> INFO
>>> User-Agent: X-Lite release 1011s stamp 41150
>>> Content-Length: 0
>>>
>>> #
>>> U 2011/02/03 09:31:04.404039 172.30.140.8:5060 -> 172.30.140.22:48752
>>> SIP/2.0 200 OK
>>> Via: SIP/2.0/UDP 172.30.140.22:48752
>>> ;branch=z9hG4bK-d87543-9a54af22967ae417-1--d87543-;rport=48752
>>> To: "1000"<
>>> sip%3A1000@172.30.140.8 >
>>>
>>>> ;tag=c97b4d1cb1f3d0da549e06a8d482ef63.34dc
>>>>
>>> From: "1000"<
>>> sip%3A1000@172.30.140.8 >>;tag=cd3e2323
>>> Call-ID: MmU0YjM1NThiNTg0ZjhiNGM4ODA4ZmU1YWFiYjBmNTc.
>>> CSeq: 1 REGISTER
>>> Contact:>> ;rinstance=fcade2df86ce0ab8>;expires=3600
>>> Content-Length: 0
>>>
>>> Am i missing something in my configuration?
>>>
>>> Thanks in advance!!!
>>>
>>>
>>>
>>>
>>> ___
>>> 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
>>
>
> --
> Daniel-Constantin Mierla
> http://www.asipto.com
>
>
___
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Re: [SR-Users] authentication is not working
Hello, 2011/2/4 Iñaki Baz Castillo > 2011/2/3 Danny Dias : > >> Are you asking the same questions in both kamailio and opensips accounts > >> with different aliases? > > > > different emails...what's the problem? i have a doubt and then i > ask...thats > > the purpose of the forum, if you are not going to help, don't answer... > > You are showing same config file (just replaced "kamailio" with > "opensips") and exactly same SIP traces. So please clarify if you are > using Kamailio or OpenSIPS since they are different. > oooh...different? both are proxys or not? now i'm lost...it's the same trace, it happened with OpenSIPS, i'm using Kamailio and Opensips, but the problem was with Opensips (not problem of the application, the problem was in my hands, bad configuration, etc...), as i said before, i wont ask the same questions to different lists again...but i thought that they would have the exactly behavior in my problem... > If you are using Kamailio and report a problem in OpenSIPS maillist > (or vice versa) then you are providing not valid information (both > projects are not the same, even more in newer versions). > > So what are you using and which exact version? > > For Kamailio 1.5 and for OpenSIPS 1.6.3 (right now Just for testing scenarios...) > Cheers. > > -- > Iñaki Baz Castillo > > ___ 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
