sr-users August 2008

sr-users@lists.kamailio.org

144 participants
201 discussions

[Kamailio-Users] multiple extensions/domains

by voipmo

I have been running asterisk servers for a few years now to host multiple organizations. I have been working on setting up OpenSER so that we can have failover and load balancing. My question is how do I keep my different organizations separate? I currently have the following registrations: 1001(a)test.example.com 1002(a)test.example.com 1001(a)another.example.com The problem is that if 1002(a)test.example.com dials 1001 it rings both domains 1001 phones. Am I going about this the right way? Is there a better method? How might I prevent calls between domains? -- View this message in context: http://www.nabble.com/multiple-extensions-domains-tp19173400p19173400.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

15 years, 8 months

[Serusers] Help with Ser and LCR

by Jai Rangi

All, I need little help with LCR module in ser, Say these are the entries on my LCR tables, Prefix, | From_URI | Grp_id | Priority "13612499080" ;"%"; "46"; "1" "1" ;"%"; "1"; "1" "1" ;"%" "2" "2" Now any Say if someone calls 19495556666, Then ser sends the call to Server in Grp_id 1, if the grp_id=1 server does not respond, then SER sends the call to Server with grp_Id 2. Perfect, that exactly what I need. Now if some calls 13612499080", then SER will send the call to server with grp_id 46. If the call failes it should not send the call to server grp_id 1 or two. Right? In my case it is sending the call to server grp_id 1 after the call fails on server grp_id 46. How can I stop that. I want the calls for 13612499080 forwarded to only server grp_id 46 and is that fails then just terminate the call. I will appreciate if someone can provide any help or hint. Thank you, -JP

15 years, 8 months

[Kamailio-Users] AVP to store user contact information

by Asim Riaz

Hi All, I am wondering if there is a way to store the user contact info into avp, relay it to existing Regitrar and store the original contact info into location through a function(which i dont know) once i received successful reply from my sip registrar. Thanks in advance Regards Asim Riaz

15 years, 8 months

[Kamailio-Users] FW: | SMS FALAS | <--- did anyone else on this list get this message?

by Vadim Berezniker

Sorry this is offtopic but.. Did anyone else get this email? I'm just wondering if they scraped emails from this mailing list. -----Original Message----- From: KLIENT I NDERUAR [mailto:info@hi5.com] Sent: Tuesday, August 26, 2008 11:38 AM To: Vadim Berezniker Subject: | SMS FALAS | Klient i/e Nderuar Nese Deshironi te Ndegjoni Muzika ma te rejat ather vizitone Kosova-SMS dhe do te bindeni me muziken qe ofrojm ne. Gjithashtu per Vizitoret e Kosova-SMS ka Startu me ofert te RE. SMS FALAS MUNDSUAR NGA Kosova-SMS 60 SMS NE DIT NGA NR I YT KLIKO PER ME SHUM. --> http://www.kosova-sms.tk

15 years, 8 months

[Serusers] Notice call ending

by inge

Hi all, I look for a workarround to notice the caller that the call will be hang up, by generating a beep. Is someone can help me because I don't find anything about this subject ? Thanks. Regards, Adrien.

15 years, 8 months

[Kamailio-Users] Having Problem routing an ACK

by Stagg Shelton

Hello, I am having an issue routing an ACK back from a new upstream carrier. The basic situation is my openser 1.3.2 receives the INVITE. I do an alias_db_lookup, set some accounting flags, and send the call to t_relay The call makes it to the PBX, as expected, and the PBX sends a 100 trying, 180 ringing. The 180 ringing is sent to the carrier. When the incoming call is answered the PBX sends a 200OK to openser, and that 200OK is sent to the carrier. When the carrier sends the ACK back it makes it to OpenSER, but is not sent to the PBX. The PBX will send several more 200 OK attempts, but each ACK from the carrier stops at openser. From what I can tell from the wireshark packet captures, the Request- Line header in the ACK from the carrier is sip:TELNUMBER@MY_OPENSER_IPADDR I have compared these packets to similar traces from other carriers and the Request-Line header from the other carriers is sip:TELNUMBER@MY_PBX_IPADDR This new carrier is telling me that they are not doing anything wrong with the ACK. RFC3261 seems specific about the construction of the ACK, but at the same time i am getting lost going to all of the other sections trying to identify details. Basically I just need to get the ACK to the PBX. What other information can I provide with this problem. Thanks Stagg Shelton

15 years, 8 months

[Serusers] ser 0.10 doucmentation

by Steven C. Blair

I was wondering if anyone has documentation on the new syntax used in SER 0.10.x. I see the following in the ser-advanced.cfg example file but I do not see these parameters documented anywhere. Thanks,Steve if ($t.fr_inv_timer) { if ($t.fr_timer) { t_set_fr("$t.fr_inv_timer", "$t.fr_timer"); } else { t_set_fr("$t.fr_inv_timer"); } } Senior Network Engineer, Information Systems and Computing Networking and Telecommunications , Suite 221A /6228 University of Pennsylvania Voice:215-573-8396 FAX:215-898-9348

15 years, 8 months

[Kamailio-Users] CDRTool not applying rates

by Brian Del Shasta

I believe my rating tables are configured correctly now thanks to David Villasmil who sent me examples from his working setup. However, I am still unable to see any rates. The rating engine seems to be started and I have no problem identifying and matching calls based on specific criteria. But I still see no rates. I'm wondering if anyone knows the best place to troubleshoot the rating engine would be or if anyone has any other ideas? Thank you in advance, and thanks again David for your help!

15 years, 8 months

Re: [Kamailio-Users] presence fail

by Ricky Gutierrez

in this guide, he doesn't have the module presence_xml, but he has the parameter modparam (" presence ", " force_active ", 1) http://www.kamailio.net/dokuwiki/doku.php/presence:configuration-file it is strange, I believe to be seen in the wiki a same example for openser 1.3.2 greetingss rickygm

15 years, 8 months

Re: [Serusers] call forward no answer (solved)

by caio

append_branch() is missed at ser examples... On Fri, Aug 15, 2008 at 4:22 PM, caio <elcaio(a)gmail.com> wrote: > Hi all, > I can't make call forwarding on no answer works on my ser-0.9.7 installation. > Here you can see a snippet of log, where the call falls on > failure_route and then to route_1 where t_relay is called. > > Log at: http://rafb.net/p/xwTmn195.html > > The call is from sip:579951 to sip:579915(who has enabled the feature > fwdnoanswer to sip:579916. > But I do not know why the INVITE after the append_branch() is still to > R-URI: sip:579915@.... > > After the relay then the call terminates with: > > Aug 15 16:13:25 [/usr/sbin/ser] SIP Reply (status):_ > Aug 15 16:13:25 [/usr/sbin/ser] version: <SIP/2.0>_ > Aug 15 16:13:25 [/usr/sbin/ser] status: <487>_ > Aug 15 16:13:25 [/usr/sbin/ser] reason: <Request Terminated>_ > > I follow the example of ser guide.., but do not work.. > I would appreciate any comment.. > > Regards, > > -- > caiogf > -- caio

15 years, 8 months

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sr-users August 2008