sr-users November 2009

sr-users@lists.kamailio.org

80 participants
108 discussions

by GEMINORIUM crux

hi 1. i want to change contact address .i want to add transport protocol how can i do this ? 2. i am getting Error on $var(query) = "INSERT INTO sip_user(sipuser) VALUES(' "+$fU+"' )"; if(!avp_db_query("$var(query)","$avp(s:test)")){ xlog("ERROR IN Query = $var(query)"); sl_send_reply("400","Bad Request"); exit; } but if i run that query in mysql it give me expected output. why i am getting error ? pls help me ? any suggestion is appreciated. thanks

15 years

[Kamailio-Users] Problem in configuring LCR module

by Masoom Alam

Hi there, I am a first time user of Kamilio 1.5. We are trying to configure the LCR, we get an error in the syslog. The error in the syslog is: Nov 6 15:12:09 kam /sbin/kamailio[5396]: ERROR:lcr:mod_init: AVP fr_inv_timer_avp has not been defined Nov 6 15:12:09 kam /sbin/kamailio[5396]: ERROR:core:init_mod: failed to initialize module lcr Nov 6 15:12:09 kam /sbin/kamailio[5396]: ERROR:core:main: error while initializing modules We have already configured FR_inv_timer_avp parameter like this in the Kamilio.cfg: modparam("tm", "fr_inv_timer_avp", "$avp(i:70") The "tm" module is already loaded in the Kamilio.cfg. Please guide, where we are making problems?

15 years

[Kamailio-Users] Translate "486 Busy Here" to "600 Busy Everwhere"

by geoffreymina＠gmail.com

Hello, I have a need to translate a 486 from my asterisk server to a 600 when sending upstream to my carrier. The problem I am seeing is that when I return 486, the carrier re-presents the call to my next server in the SRV rotation... I really want the 486 to be a final disposition, so I am assuming sending a 600 would solve my issue. I am not sure how to go about forwarding the message, but modifying the SIP code... is that even allowed? This would be something like what I'm trying to do: failure_route[1]{ if(t_check_status("

15 years

Re: [Kamailio-Users] Translate "486 Busy Here" to "600 Busy Everwhere"

by Geoffrey Mina

Apologies... sent directly to Alex and not the list. On Tue, Nov 10, 2009 at 12:02 PM, Geoffrey Mina <geoffreymina(a)gmail.com> wrote: > Alex, > Thanks for the info. We are on the same page about what a proxy > should, and should not do... however, I am strugging with my ITSP > (Level3) a bit on this. Unfortunately, Asterisk is my media end-point > here and there is no mechanism in Asterisk to send back a 600. Level3 > is exhibiting very odd behaivor, and I have a ticket open with them, > but who knows if it will ever go anywhere. > > CallID: 1627795841_10332(a)4.55.4.227 > Level3 --> INVITE --> Kamailio --> Asterisk > Asterisk --> 183 Progress --> Kamailio --> Level3 > Asterisk --> 486 Busy --> Kamailio --> Level3 > Level3 --> ACK --> Kamailio --> Asterisk > > [at this point Level3 IMMEDIATELY sends me another presentation of the > call with a different CallID and the process repeats] > > CallID: 604627041_120490736(a)4.55.20.99 > Level3 --> INVITE --> Kamailio --> Asterisk > Asterisk --> 183 Progress --> Kamailio --> Level3 > Asterisk --> 486 Busy --> Kamailio --> Level3 > Level3 --> ACK --> Kamailio --> Asterisk > > > I am pretty sure I am in the right here, and Level3 shouldn't be > failing over in a 486 scenario... unfortunately my customer doesn't > care why our systems aren't talking properly :) > > Thanks! > > > > On Tue, Nov 10, 2009 at 10:38 AM, Alex Balashov > <abalashov(a)evaristesys.com> wrote: >> geoffreymina(a)gmail.com wrote: >>> >>> Hello, >>> I have a need to translate a 486 from my asterisk server to a 600 when >>> sending upstream to my carrier. The problem I am seeing is that when I >>> return 486, the carrier re-presents the call to my next server in the SRV >>> rotation... I really want the 486 to be a final disposition, so I am >>> assuming sending a 600 would solve my issue. >>> >>> I am not sure how to go about forwarding the message, but modifying the >>> SIP code... is that even allowed? >>> >>> This would be something like what I'm trying to do: >>> >>> failure_route[1]{ >>> if(t_check_status(" >> >> Sure, you can do that. It is not a good idea to do so in principle because >> it violates how proxies should behave; proxies are very thin interoperation >> layers that are designed to statefully pass requests and replies back and >> forth more or less strictly as intended by the endpoints involved, and a >> B2BUA would more ideally fit your goal. Nevertheless, it is possible. >> >> failure_route[1] { >> >> if(t_was_cancelled()) >> exit; >> >> if(t_check_status("486")) { >> t_reply("603", "Declined"); >> exit; >> } >> } >> >> >> -- >> Alex Balashov - Principal >> Evariste Systems >> Web : http://www.evaristesys.com/ >> Tel : (+1) (678) 954-0670 >> Direct : (+1) (678) 954-0671 >> >

15 years

[Kamailio-Users] Double acc records in DB

by Alex Hermann

Hello, I noticed that sometimes I get 2 identical acc records in the database, although I get only 1 in the syslog (I use both db_flag and log_flag). Of course, the second record in the DB shouldn't be there. There are no retransmissions of neither request nor response and only a single branch exists. How is this possible, and more importantly, how can I prevent the double records? This is on Kamailio 1.4+svn. -- Greetings, Alex Hermann

15 years

[Kamailio-Users] Question about translating 486 to 600

by Geoffrey Mina

Hello, I have a need to translate a 486 from my asterisk server into a 600 to my ITSP. I haven't figured out a way to force asterisk to send the 600, so I guess I need to rely on Kamailio to do that for me. The problem is that when the INVITE is sent to me, if I return a 486, my ITSP immediately sends a new INVITE to the next gateway in my SRV records. I want the 486 to be a final disposition in this scenario, so I am guessing I need to send 600 for the carrier to not try another gateway. How would I go about doing this? failure_route[1]{ if(t_check_status("486")){ t_relay("600","Busy Here"); } } I know the above example isn't valid... but that is basically what I need to accomplish. Thanks!

15 years

[Kamailio-Users] avp

by Anders

Hi all, Here I am - back with more "Kamailio 101"-questions! ;-) I need to strip the first character from the uri and write this to the 'acc' table. What I'm doing so far in kamailio.cfg is: modparam("acc", "db_extra", "............., $avp(s:user_ip_tag)"); So I've added the avp-thing in that line. And independently I've created a new column in the 'acc' table called "user_ip_tag". If the above is correct, my next step is to write something to this variable. In the routing, I have this: if (uri=~"^sip:[1-9]00*") { do_something_here_to_take_first_character_of_the_uri_and_put_to_avp...but_how? strip(1); } What I want to do, is to take the 1-9 digit, if it starts with that and has to zeroes after, and put that to the avp - and the strip it off before I move on in the routing. Yes, I have browsed the documentation, but I'm not sharp enough to understand it, it seems. (or alternatively, the documentation is not made for beginners). Thanks!

15 years

[Kamailio-Users] accounting

by Rajesh

Dear all, I would like to know if i can get the duration between two calls in Accounting module..I am in an attempt to do "Billing" module for OpenSER and hence would like to know the same. -- Regards, Rajesh

15 years

SR or Kamailio.

by Vicente

Hi, This is my first message to the list. So, sorry if I say something stupid. I've installed sip-router from GIT repository. It is now compiled and installed. I can see that bins ser, sercmd and ser_mysql.sh has been copied to /usr/local/sbin. ser runs properly but sercmd tries to use /tmp/ser_ctl (ERROR: connect_unix_sock: connect(/tmp/ser_ctl): No such file or directory [2]) while /temp/sip-router_ctl is the socket generated by ser. ser_mysql.sh also fails with error: ./ser_mysql.sh: 326: Bad substitution The INSTALL file on the repository instructs to use serctl utility to watch server health but no binary has been generated with that name. Is it replaced by sercmd? What is the status of sip-router.org? Is it ready for production? Or is it a better option to use Kamailio at the moment? Best regards.

15 years

[Kamailio-Users] How to recognize the caller ID

by Masoom Alam

Hi Every one, Is the following scenario possible and if yes, how? I want to identify the caller ID inside the Kamailio so that a specific caller can be routed to an Asterisk box. Actually, the caller -- a SIP client or an IP phone client is not registered at Kamailio, but Kamailio has to act as a Proxy to a group of Asterisk servers. The sketch in my mind is to use some if-then-else statements within the [route] where user can call a particular number say NXXXXXXX201 and then the Kamailio can strip off the first number and re-route it to an Asterisk server. Regards, MM Alam SERG Pakistan http://serg.imsciences.edu.pk http://www.imsciences.edu.pk/profiles/masoom_alam.html

15 years

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sr-users November 2009