sr-users February 2009

sr-users@lists.kamailio.org

76 participants
98 discussions

[Kamailio-Users] lookup("location") problem
by BERGANZ François 19 Feb '09

19 Feb '09

Hello, I do as the example in the module registrar doc: lookup("location"); switch ($?) { case -1: case -3: sl_send_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Not Found"); exit; }; And SER report me ERROR:core:pv_parse_spec: wrong char [$/36] in [$?] at [0 (0)] Know you why? Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

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[Kamailio-Users] call keepalived using rtimer?
by Noel R. Morais 19 Feb '09

19 Feb '09

Hi Guys, I would like to know if is possible to make a keepalive for active calls (dialog module)? Well, I think it is but is there a better way than making a selec into the dialog table and building the INVITEs? Thanks, Noel

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[Kamailio-Users] SIP URI Extracting
by Ramu 19 Feb '09

19 Feb '09

Hi, I would like to extract uri part from SIP Request uri. Say I received a INVITE sip:1234567890@ip Now I would like to take 4567890 the uri and send it as a new INVITE. Ramu

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[Kamailio-Users] LCR in Kamailio 1.4
by Asim Riaz 19 Feb '09

19 Feb '09

Hello, I am using LCR, for some reason load_gws() function is not finding the gateway in DB, it was working in the start then added some more routes and after that everything stopped working, I am not sure where to start troubleshooting, any idea ? kamctl lcr show lcr routes +----+--------+----------+--------+----------+ | id | prefix | from_uri | grp_id | priority | +----+--------+----------+--------+----------+ | 9 | 0 | | 1 | 1 | +----+--------+----------+--------+----------+ lcr gateways +------------+---------------+------+------------+-----------+--------+-------+------+-------+ | gw_name | ip_addr | port | uri_scheme | transport | grp_id | strip | tag | flags | +------------+---------------+------+------------+-----------+--------+-------+------+-------+ | XXXX | XXXX | 5060 | 1 | 1 | 1 | 0 | | 0 | +------------+---------------+------+------------+-----------+--------+-------+------+-------+ my routing script if(!load_gws()) { xlog("L_INFO", "[ROUTE-X] Unable To Find Gateway "); sl_send_reply("503", "unable to find gateway"); exit; } Thanks in advance. Asim

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[Kamailio-Users] related free voip providers
by Daniel-Constantin Mierla 19 Feb '09

19 Feb '09

Hello, I posted a summary of three free voip providers running sip-router.org related sip servers. It is based on my experience as user of all these (an other) services. The providers listed on the page are not ranked, just presented in alphabetic order, pretty summary about each one. I plan a longer version separately, one by one. It is a good starting point for those willing to see what can be done with Kamailio (OpenSER) and/or SIP Express Router (SER). Another benefit is for users looking to get an independent SIP address, not related to current job or location. I created the post after some private inquires, but I think it worth for public audience: http://miconda.wordpress.com/2009/02/17/top-three-free-voip-services/ In the same spirit, you can check the second edition of (my personal) Kamailio (OpenSER) awards: http://miconda.wordpress.com/2009/01/19/kamailio-openser-2008-awards/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] SIREMIS 0.9.2 released - Support for Carrierroute
by Elena-Ramona Modroiu 19 Feb '09

19 Feb '09

Hi all, a new version of SIREMIS, web management interface for Kamailio (OpenSER) and SIP-Router.org, is out as 0.9.2. Main improvements refer to addition of support to manage the carrierroute module: * management of carrierroute table * management of carrierfailureroute table * management of carrier_name table * management of carrier_domain table View screenshots: http://www.asipto.com/gallery/v/siremis/ Demo site (it works on a database with random data, username: admin, password: admin): http://siremis.asipto.com/demo/ Download: http://siremis.asipto.com/pub/downloads/siremis/ Install/Upgrade documentation: http://siremis.asipto.com/install/ Regards, Ramona

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[Kamailio-Users] Config
by NYam 19 Feb '09

19 Feb '09

Dear Sir, I have problem . wenn i root:~ # kamctl start INFO: Starting Kamailio : ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed root:~ # kamailio -c Feb 17 11:01:49 [10767] ERROR:core:main: loading config file(/usr//etc/kamailio/kamailio.cfg): No such file or directory Feb 17 11:01:49 [10767] WARNING:core:fm_free: free(0) called Feb 17 11:01:49 [10767] WARNING:core:fm_free: free(0) called root:~ # kamailio -f /etc/kamailio/kamailio.cfg then i see udp: 127.0.0.1 [127.0.0.1]:5060 ********************* ********************** tcp: 127.0.0.1 [127.0.0.1]:5060 ok Where is Problem ? can you help me? with kind regards Nyam

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[Kamailio-Users] Kamailio 1.4.3 Carrierroute BYE does not
by unknown＠1.org 19 Feb '09

19 Feb '09

Hello: I have this scenario: I already rad about the bye2bye, the dialog module, i was not able to see any light there. Here is the scenario A--GW--------Kamailio(carrierroute)---------GW1---B |--------------GW2 |--------------GW3 (etc) the capture |<-------------BYE---- -<----NOTHING HERE<<<-| the By does go back if the B user hangs it is not even attempted in the kamailio, per the traces. the BY generates records in the CRD from B side if i hang manually the bye generates CDR records from the A side. Again the Bye is understood by the kamailio and generates the CDR records, that is Not the problem, the Problem is i have the user A in silence, i am talking about the Signaling Only, not Media, no NAT. I even tried the mediaproxy (just trying) the config is like the: http://voipembedded.com/resources/openser_cr.cfg i added if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails exit; } route(10); } else { /* uncomment the following lines if you want to enable presence */ ##if (is_method("SUBSCRIBE") && $rd == "your.server.ip.address") { ## # in-dialog subscribe requests ## route(200); ## exit; ##} if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); xlog("L_WARN", "[$mi] discarding ACK\n"); exit; } this is the main difference for now, i changed to use as a basis example the Example 1.45. Configuration example - Routing to user tree. same results. i tried adding the record_route in the invite also, same result. At this point i have no clue what i am missing. Please let me know if this should work or not, and if yes, based in that basic example, what i should add.

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[Kamailio-Users] A2billing Project <-> Openser
by BERGANZ François 18 Feb '09

18 Feb '09

Hello, I use Asterisk2billing to do prepaidit works fine! But, now, I need to auth my users in the openser and forward the call by loadbal to asterisk(s) which use Asterisk2billing. Anybody know how to do? Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

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[Kamailio-Users] SIP-T
by Kaeman Chris 18 Feb '09

18 Feb '09

Hello, Does opensips support SIP-T? Thank you! KC Add more friends to your messenger and enjoy! Go to http://messenger.yahoo.com/invite/

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sr-users February 2009