sr-users September 2008

sr-users@lists.kamailio.org

95 participants
156 discussions

[Kamailio-Users] CANCEL can't be forward!
by BERGANZ François 10 Sep '08

10 Sep '08

hello a problem with my CANCEL frames! user-------openser------asterisk invite>>>>>>>|>>>>>>invite ringing<<<<<<|<<<<<<ringing cancel>>>>>>>| in fact, I don't know where forward the cancel in stateless! I have no information in the frame to forward it to the right asterisk (one for g711, one for g729...) have you an idea? thank you all Cordialement BERGANZ François <http://www.acropolistelecom.net> http://www.acropolistelecom.net TEL/FAX : 33 (0) 1 70 72 50 15

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[Kamailio-Users] failure route problem - duplicate invites
by Matthew McGuire 10 Sep '08

10 Sep '08

I'm having a problem with the route sections below. When my proxy receives an Invite to proxy to a PSTN gateway, the proxy is sending 2 invites very fast to the gateway in the first route block which ends up creating a mess with the gateway. Either I have something wrong in the routing code or I have an SER/OS/hardware timing problem. Any suggestions ? route[1] { xlog("L_INFO", "Sending to route 1\n%mb\n"); rewritehostport ("1.1.1.1:5060"); append_branch(); t_on_failure("1"); t_relay(); } failure_route[1] { if(t_check_status("487")) { break; } xlog( "L_INFO", "failure on route 1\n%mb\n"); append_branch(); route(2); break; } route[2] { xlog("L_INFO", "Sending to route 2\n%mb\n"); rewritehostport ("2.2.2.2:5060"); append_branch(); t_on_failure("2"); t_relay(); } failure_route[2] { if(t_check_status("487")) { break; } xlog( "L_INFO", "failure on route 2\n%mb\n"); append_branch(); route(3); break; } route[3] { xlog("L_INFO", "Sending to rout 3\n%mb\n"); rewritehostport ("3.3.3.3:5060"); t_relay(); } Thanks, Matt

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[Serusers] help required
by Engr. Muhammad Noman Saeed 10 Sep '08

10 Sep '08

Dear All, I need your help for establishing SER as a SIP server for my organization. I have downloaded it at my fedora 8.0 machine, but as i am new with linux and SER it is hard to find me how to install/configure. kindly help me as a new SER user. regards -- ========================= Noman =========================

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[Serusers] how can i config ser.cfg
by 邱磊 10 Sep '08

10 Sep '08

hi friends: i need some help! frist,my asterisk receive a sip message from sip proxy via SER.when my asterisk send 200 to SER and then receive ACK from proxy directly! when asterisk send BYE to the proxy and can hangup the call. but now i redirect the BYE to SER, i found the SER can not send the message to proxy! what can i do in the ser.cfg? thanks in advance 2008-09-10 邱磊

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[Kamailio-Users] add number to an avp ?
by BERGANZ François 09 Sep '08

09 Sep '08

I need to add 1 to an avp like : $avp(i:681)=$avp(i:681)+1; but don't work! how to? thank you Cordialement BERGANZ François http://www.acropolistelecom.net TEL/FAX : 33 (0) 1 70 72 50 15

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[Serusers] Dubt on RFC 4484 and SIP Proxy Server behavior
by Francesco la Torre 09 Sep '08

09 Sep '08

Hi all, I've a doubt regarding a statement at the end of page 5 "... Some important design decisions are associated with carrying assertions in a SIP request. If an assertion is carried by value, or uses a MIME-based content indirection system, then proxy servers will be unable to inspect the assertion themselves. If the assertion were referenced in a header, however, it might be possible for the proxy to acquire and inspect the assertion itself. There ..." Why a proxy server will be unable to inspect the assertion themself ? SER has functions by which it could be read message body, does this behavior not respect any rfc ? I'm missing something :-S Thanks in advance, Francesco la Torre

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[Kamailio-Users] [Fwd: Re: Unconditional call forward]
by Stefano Palleschi 09 Sep '08

09 Sep '08

Hi Gustavo, thanks for your reply. I've changed the *avp type* from "2" to "0" but I get always the same error: ERROR:avpops:load_avps: incomplet uri <sip:81.29.176.57> Now I'm going to reboot the server... If I'll have news I'll let you know. Regards, Stefano -------- Messaggio Originale -------- Oggetto: Re: [Kamailio-Users] Unconditional call forward Data: Wed, 3 Sep 2008 10:38:42 -0300 Da: Gustavo Mistrinelli <gmistrinelli(a)gmail.com> A: Stefano Palleschi <stefano.palleschi(a)okcom.it> CC: users(a)lists.kamailio.org Referenze: <48BE6299.5000805(a)okcom.it> Hi Stefano, Please check *avp type* that you are using, use type 0 instead of 2 See below the different types for avps: 0 - *AVP* with string name and string value 1 - *AVP* with string name and integer value 2 - *AVP* with integer name and string value 3 - *AVP* with integer name and integer value Regards, Gustavo On Wed, Sep 3, 2008 at 7:10 AM, Stefano Palleschi <stefano.palleschi(a)okcom.it <mailto:stefano.palleschi@okcom.it>> wrote: Hello all, i'm trying to do an unconditional call forward with openser. I've added the instructions below in the script: if (avp_db_load("$ru/username", "$avp(s:callfwd)")) { avp_pushto("$ru", "$avp(s:callfwd)"); xlog("$avp(s:callfwd)"); route(1); exit; } and i've created a record on the usr_preferences table of openser database: mysql> select * from usr_preferences; +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ | uuid | username | domain | attribute | value | type | modified | +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ | 1 | 0666620996 | 81.29.176.57 <http://81.29.176.57> | callfwd | sip:0666653218@81.29.176.57 <mailto:sip%3A0666653218@81.29.176.57> | 2 | 2008-09-03 09:51:33 | +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ 1 row in set (0.00 sec) like showed, i would like forward the call that arrive at the sip client 0666620996 to the pstn number 0666653218. At openser restarting I get always this error: ERROR:avpops:load_avps: incomplet uri <sip:81.29.176.57 <http://81.29.176.57>> without make any call. Why in the uri there isn't the first part (username@)? Thanks in advance. Stefano Palleschi _______________________________________________ Users mailing list Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users -- Gustavo Mistrinelli

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[Kamailio-Users] 302 redirect inside redirect
by Kent Pirlo 09 Sep '08

09 Sep '08

Ok, I am trying to get a redirect server working correctly, only cache is I need to hit another redirect server first, and grab info from the contact in the that 302 reply before processing and sending back a 300 to the original switch. call flow would be something like this: call comes into kamailio, 1) kamailio sends the invite to another proxy(1), 2) that proxy returns a 302,(if it times out it moves on to step 5) 3) kamailio sends an ACK back to that proxy(1), 4) kamailio looks at the contact header of the 302 reply, 5) depending on that info, I decide where to tell the originating switch where to go, 5) kamailio sends originating switch a 300 redirect with new contact header. 6) originating switch sends back ACK. I have tried doing this with a first step going to route(1) , which does a "t_on_reply" and "t_relay" to the second proxy. The onreply route, graps the contact saves it into variable and then sends to route(2), Route(2) looks at the variable, does some other lookups and sends t_reply("300","redirect"). The switch then sends the ACK, but kamailio says its not a local ACK and ignores it.. Then it keeps sending 300's back to the switch... Please help.. Thanks in Advance. kent

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[Kamailio-Users] Documentation Site Down
by Jon Farmer 09 Sep '08

09 Sep '08

Hi When I go to http://kamailio.org/docs/ i get a 404 error. Any ideas anyone? Regards Jon

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[Kamailio-Users] Forward call
by Thiago Rondon 09 Sep '08

09 Sep '08

Folks, My problem for forward call, that I need to change username/password and realm to do my forward. So, I try this: modparam("uac","from_restore_mode", "auto") modparam("uac","auth_realm_avp","$avp(s:uac_realm)") modparam("uac","auth_username_avp","$avp(s:uac_username)") modparam("uac","auth_password_avp","$avp(s:uac_password)" route { ... if(uri=~"^sip:0[0-9]{10}@") { xlog("L_INFO","uac_atual: $avp(s:uac_username) !!"); $avp(s:uac_username)="username"; $avp(s:uac_realm)="1.1.1.1"; $avp(s:uac_password)="password"; xlog("L_INFO","uac_trocado $avp(s:uac_username) !!"); route(3); exit; } .. } ... route[3] { t_on_failure("3"); # reset flag to mark no authentication yet performed resetflag(7); # forward to PSTN uac_replace_from("username","sip:username@1.1.1.1"); rewritehostport("1.1.1.1:5060"); xlog("L_INFO","rewritehost !!!!!!!!"); t_relay(); } .. So, when I do xlog, in the first entry, uac_username don't display, why ? And its strange, because in one supplier its ok, and in another its not run. But I don't undestand why uac_username is empty in the first xlog! Thanks, -Thiago Rondon

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sr-users September 2008