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21 participants
37572 discussions

Pick Domain name from Location Table
by aawaise 21 Apr '14

21 Apr '14

Hi All, When my kamailio server receives an "invite" packet to make a call. I want the kamailio server to make a decision about call routing. Whether to route call to other kamailio server or route it to local user. I want to base this decision on "domain" of called user. This "domain" is to be picked from "Location List" in database. Both servers are using the same database. I have two tasks here. One is to pick the domain of user from "Location Table" of Siremis. And secondly, route this call to other server on basis of the string I get from "Domain" of "Location Table". Need help regarding changes in kamailio.cfg for this. Any help will be highly appreciated. Thanks, Regards, Aawaise -- View this message in context: http://sip-router.1086192.n5.nabble.com/Pick-Domain-name-from-Location-Tabl… Sent from the Users mailing list archive at Nabble.com.

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PCSCF configuration towards pcrf
by Dees 19 Apr '14

19 Apr '14

Hi, Very new to kamailio, following instruction have copied the examples file for pcscf and ran kamaillo. I have installed ims modules, but I don't see to see any connection initiation towards to PCRF? Any clues on what I am doing wrong. Thanks. Dees root@kamalio:/etc/bind# more /etc/hosts 127.0.0.1 localhost 10.209.96.85 pcscf.kamailio-ims.org kamalio #10.209.96.92 pcrf.kamailio-ims.org root@kamalio:/etc/bind# ping pcrf.kamailio-ims.org PING pcrf.kamailio-ims.org (10.209.96.92) 56(84) bytes of data. 64 bytes from 10.209.96.92: icmp_req=1 ttl=64 time=0.392 ms

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event_route[tm:branch-failure] question
by Juha Heinanen 18 Apr '14

18 Apr '14

in branch route i execute: t_on_branch_failure("contact"); xlog("L_INFO", "Routing $rm <$ru> to contact\n"); and in branch_failure route i have: event_route [tm:branch-failure:contact] { if (t_check_status("486")) { xlog("L_INFO", "Got 486 response to <$ru>\n"); }; and i get to syslog: Apr 13 21:44:07 siika /usr/sbin/sip-proxy[17475]: INFO: Routing INVITE <sip:test-0x11107b0@192.98.102.30:5064> to contact Apr 13 21:44:11 siika /usr/sbin/sip-proxy[17437]: INFO: Got 486 response to <sip:test@test.tutpro.com> why has $ru in branch_failure route changed from $ru of the branch to original request uri (aor)? i would expect to see in $ru of branch_failure route the $ru of the branch. is this a bug? if not, is there any means in branch_failure route to get access to the branch that failed. -- juha

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kamailio db
by Slava Bendersky 18 Apr '14

18 Apr '14

Hello Everyone, Is possible run kamailio without database ? I got ldap working and extension is registered, but can't place calls between two extensions. Getting 404 all the time. Slava.

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invite always contains "s"
by Vasco Mendes 18 Apr '14

18 Apr '14

Hi, I created an incoming trunk between iptel and my asterisk. Whatever the account is on iptel I always have in the from of received INVITE < sip :s@mypublicip :5060 >. < s > is not a recognized DID on my asterisk. Why do I receive this < s > in the from ?? Many thanks for your help, Vasco --- Ce courrier électronique ne contient aucun virus ou logiciel malveillant parce que la protection avast! Antivirus est active. http://www.avast.com

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Editings in kamailio.cfg for routing decision
by aawaise 18 Apr '14

18 Apr '14

Hi All, When a user sends an "invite" packet to make a call. I want to make a decision about call routing. Whether to route call to other server or route it to local user. I want to base this decision on "domain" of called user. This "domain" is to be picked from "Location List" in database. Both servers are using the same database. What changes are to be made in kamailio.cfg ?? Actually in my environment my user can connect to any of the servers. I want to know to which server it is connected and then make the call routing decision. Any help will be highly appreciated. Thanks, Regards, Aawaise -- View this message in context: http://sip-router.1086192.n5.nabble.com/Editings-in-kamailio-cfg-for-routin… Sent from the Users mailing list archive at Nabble.com.

1 0

Custom fr timer for dialog's OPTIONS keep alives
by Vassilis Radis 17 Apr '14

17 Apr '14

Hello, I would like to try customizing the fr_timer for the OPTIONS keep-alive msgs transactions that the dialog module sends. I tried to call t_set_fr() function from inside the dialog modules's code (in dlg_req_within.c , in dlg_send_ka() function after the call to d_tmb. t_request_within(&uac_r); But it seems that this tm function is not available from inside other module's code. Is the above is the correct place to try and do it, and how can I programmatically ,inside a module's code, adjust fr_timer? On a side note, i followed the call flow from dlg_send_ka() down to the t_uac_prepare() function of tm module, where the fr_timer is set. In there, there is this comment+code: /* init timers hack, new_cell->fr_timer and new_cell->fr_inv_timer * must be set, or else the fr will happen immediately * we can't call init_new_t() because we don't have a sip msg * => we'll ignore t_set_fr() or avp timer value and will use directly the * module params fr_inv_timer and fr_timer -- andrei */ new_cell->fr_timeout=cfg_get(tm, tm_cfg, fr_timeout); new_cell->fr_inv_timeout=cfg_get(tm, tm_cfg, fr_inv_timeout); new_cell->end_of_life=get_ticks_raw()+lifetime; Does this mean that it cant be done in a straighforward way? Any advice on how I can achieve the customized fr_timers for those messages? Thanks.

2 2

Basic question regarding Routing
by aawaise 17 Apr '14

17 Apr '14

Hello, I want to ask some basic questions. Anyone please guide me to links with answers. I am sure such basics must be explained online. 1. What is exactly achieved by "route(RELAY)" ?? Routing calls from local users to other domains ?? 2. If user 1 is caller and user 2 is being called. Which pseudo variable will be used to catch value of called user i.e user 2 and his domain ?? I want to pick domain name from location table in database and not from the authorization module. Thanks, Cheers, Aawaise -- View this message in context: http://sip-router.1086192.n5.nabble.com/Basic-question-regarding-Routing-tp… Sent from the Users mailing list archive at Nabble.com.

2 1

Routing decision based on domain entry in Location Table
by aawaise 17 Apr '14

17 Apr '14

Hello, I have deployed two kamailio servers, both sharing the same database. I want my kamailio servers to make a decision on the routing of incoming call by seeing the domain of "called user" from location table of online users. If the "called user" is connected to same server, the call should be routed to user directly. In other case if the user is connected to other server, the call should be forwarded to other user for further processing. Will someone please guide me how can this be achieved ?? P.S : I am using kamailio-3.1.1. Thanks, Regards, Aawaise. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Routing-decision-based-on-domain-en… Sent from the Users mailing list archive at Nabble.com.

2 4

rls multipart body does not contain watcherinfo+xml
by Mihai Marin 16 Apr '14

16 Apr '14

Hello Sirs, I'm trying to accomplish a rls subscribe but I have some miss-understandings regarding the multipart publish response that should come. I'm using the integrated xcap from the kamailio with all the xml files added. My problem is that event if I request a multipart containing application/pidf+xml, application/rlmi+xml, application/watcherinfo+xml I only get the first two. The request contains the following headers: Event: presence Supported: eventlist Accept: multipart/related, application/pidf+xml, application/rlmi+xml, application/watcherinfo+xml The response only sends the first two: --4kSPpRGyJQct4DIhyGnyobqO Content-Transfer-Encoding: binary Content-ID: <1397508890.sip:bob@93.187.138.214.681064155> Content-Type: application/rlmi+xml;charset="UTF-8" <?xml version="1.0"?> <list uri="sip:bob@93.187.138.214" xmlns="urn:ietf:params:xml:ns:rlmi" version="4" fullState="false"><resource uri="sip:alice@93.187.138.214"><instance id="Scf8UhwQ" state="active" cid="1397508890.sip:alice@93.187.138.214.1552580371"/></resource></list> --4kSPpRGyJQct4DIhyGnyobqO Content-Transfer-Encoding: binary Content-ID: <1397508890.sip:alice@93.187.138.214.1552580371> Content-Type: application/pidf+xml <?xml version="1.0" encoding="UTF-8" standalone="no"?> <presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" entity="sip:alice@93.187.138.214"> <dm:person id="p6850"> <rpid:activities/> <rpid:status-icon>http://93.187.138.214:5060/xcap-root/oma_status-icon/users/sip:alice@93.187…</rpid:status-icon> </dm:person> <tuple id="t7673"> <status> <basic>open</basic> </status> <contact>sip:alice@93.187.138.214</contact> <note>Online</note> </tuple> </presence> --4kSPpRGyJQct4DIhyGnyobqO-- What do I missing? I need the body part with the watchers info in oder to see the state of the watcher (active, pending) - to accomplish the authorization for users. Thank you in advance for you help. Best regards, Mihai M

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