sr-users May 2013

sr-users@lists.kamailio.org

95 participants
156 discussions

by Ján Hrnko

Hi, I have a problem when using uri_db module. When I leave everything on default and not specify modparam "uri_db", "db_url", then it will not connect to the database. In the var/log/syslog I have seen that it tries to connect to database okmaialio instead of kamailio. Is it bug, or did I misconfig something? I left the kamctlrc file intact and everything to default... can you confirm this please? Thank you!

10 years, 11 months

remove_body() leaves header Content-Type: application/sdp ?

by Konstantin M.

Hello, I'm using a very ugly code to test remove_body(): request_route { remove_body(); $rd = "192.168.254.85"; t_relay(); } Original INVITE (with SDP): --------------------------------- U 192.168.254.102:5060 -> 192.168.254.104:5060 INVITE sip:123@192.168.254.104 SIP/2.0. Via: SIP/2.0/UDP 192.168.254.102:5060;branch=z9hG4bK50cdeeb8. Max-Forwards: 70. From: "10005" <sip:10005@192.168.254.102>;tag=as78ddcd16. To: <sip:123@192.168.254.104>. Contact: <sip:10005@192.168.254.102:5060>. Call-ID: 79a1c78f01914f1814bfcb7d622cf735@192.168.254.102:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX 1.8.7.1. Date: Sat, 18 May 2013 05:47:32 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 269. . v=0. o=root 1763381182 1763381182 IN IP4 192.168.254.102. s=Asterisk PBX 1.8.7.1. c=IN IP4 192.168.254.102. t=0 0. m=audio 16162 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. --------------------------------- Modified INVITE (after remove_body()): --------------------------------- U 192.168.254.104:5060 -> 192.168.254.85:5060 INVITE sip:123@192.168.254.85 SIP/2.0. Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK6c5c.6160e017.0. Via: SIP/2.0/UDP 192.168.254.102:5060;branch=z9hG4bK50cdeeb8. Max-Forwards: 70. From: "10005" <sip:10005@192.168.254.102>;tag=as78ddcd16. To: <sip:123@192.168.254.104>. Contact: <sip:10005@192.168.254.102:5060>. Call-ID: 79a1c78f01914f1814bfcb7d622cf735@192.168.254.102:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX 1.8.7.1. Date: Sat, 18 May 2013 05:47:32 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. *Content-Type: application/sdp.* Content-Length: 0. . --------------------------------- So, remove_body() is actualy removes SDP from request but leaves Content-Type header. Why ?

10 years, 11 months

SCTP question

by Jignesh Gandhi

Hello, I recently started using SCTP relay feature of Kamailio and am receiving SCTP-SIP and relaying it UDP-SIP and vice versa. What happens , if an SCTP association is broken from the distant end in the middle of the call, is there a way to re transmit the SCTP message via another route ? Thanks, --Jignesh Gandhi

10 years, 12 months

Re: [SR-Users] Msilo configuration‏

by sipatse＠in.gr

10 years, 12 months

Kamailio + Siremis Outbound route

by Tony Turner

Hi Version Kamailio v4.0 + Siremis installed on Debian Wheezy via apt-get install I want to use Kamailio as a proxy edge register to our network. I have installed Kamailio and freeswitch. I can register on Kamailio but I can't route a call from my sip client from Kamailio to freeswitch and out to PSTN Sip client ---- Kamailio-----freeswitch-------SS7 ISDN SIP Gateway --- Carriers If I register direct on Freeswitch I can route out to PSTN but I don't understand Kamailio routing. Can someone let me how I route say from SIP client registered on Kamailio to prefix 01% which goes out to Freeswitch Many Thanks Tony

10 years, 12 months

Kamailio 3.1.x and FreeSWITCH 1.0.6+ for Media Services and SBC Author: Daniel-Constantin Mierla

by Carlton Thompson

http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc#dokuwi… Hello I was trying to use your tutorial on the above page. I am trying to do so using PGQL vs MYSQL. I changed all the mysql parts to pgsql however when i try to start kamailio it fails. I am building the system on alpine linux and attached my kamailio.cfg and kamctlrc. i am able to setup using the default config making necessary changes but when i use yours it fails. Any help would be appreciated. -- Regards, Carlton Thompson, MS Doctoral Student College of Engineering University of Florida Gainesville, FL 32611

10 years, 12 months

Can I use Kamailio as B2BUA

by Kamal Palei

Dear Kamailio experts I have a typical use case where I want Kamailio to behave as a B2BUA. What I mean here is (assume Kamailio is using TCP for SIP call establishment) 1. For each call it should create a separate TCP connection with next proxy in path. 2. When call ends, it should close that connection. If that call is active for 10hrs, then connection should stay alive till 10hrs. Please somebody help. Thanks kamal

10 years, 12 months

New developer: Victor Seva

by Daniel-Constantin Mierla

Hello, I want to announce that a new person got developer GIT write access to repository: Victor Seva. He helped pushing back Kamailio in official Debian packages and lately has submitted many patches to various modules, among upcoming ones is some work on newly added per module debug level (see the logs from yesterday's IRC meeting for more about this feature). His git commit id is: vseva My warm welcome and looking forward to future work within the project! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu *

10 years, 12 months

IMS In Kamailio 4.0.0 - what is it?

by Olle E. Johansson

Jason Penton, Richard Good and Carsten Bock has written an excellent article about the IMS extensions in Kamailio 4.0.0. What it is, what you can do with it and what they're currently working on. Go read it on our web site now to learn more about this work: http://www.kamailio.org/w/2013/05/ims-kamailio/ Have a great weekend! /O

10 years, 12 months

Radius - new style AVP's (misc_radius)

by Victor V. Kustov

Good day! Now I test new style AVP's for radius response storing. Today additional radius responses return in special radpair: SIP-AVP - name, string "email:sr-users@lists.sip-router.org session-timeout#161 nexthopip:h323/0001111@myvoip-gate.kamailio.org" - value. I don't know why developers chose this way. But it's addition kamailio's feature, so let it be. But if radius server responded with standard pairs? Sending Access-Accept of id 60 to 172.16.X.X port 59736 Session-Timeout = 4261674 next-hop-ip = "SIP/00111222333444@cisco-out" SIP-AVP = "email:sr-users@lists.sip-router.org session-timeout#161 next-hop-ip:h323/0001111@myvoip-gate.kamailio.org" session-protocol = "SIP" No chance to get full radius response at this moment. Only parse SIP-AVP pair. This mean that when you wanna integrate K. in existing infrastructure (in my case - billing), u need work with K. and radius. But why, if you already had worked voip devices? I think more flexible solution ADD in misc_radius function to get radpairs as AVP's. See that i got: misc_radius [functions.c:159]: AVP 'Session-Timeout'/0='null'/4261674 has been added misc_radius [functions.c:143]: AVP 'next-hop-ip'/0='SIP/005555777888@cisco-out'/0 has been added misc_radius [functions.c:143]: AVP 'session-protocol'/0='SIP'/0 has been added After radius response i get: $avp(Session-Timeout)=4261674 (integer) $avp(next-hop-ip)="SIP/005555777888@cisco-out" (string) $avp(session-protocol)="SIP" (string) I think its usable and more flexible. I see 3 way to integrate it: 1. additionally with existing radius_load_caller_avps/radius_load_callee_avps (i make same for test) 2. new function in same module, like radius_authorize_request (caller,callee). 3. replace existing style. it's bad for backward compability, of cause. Or it stay my local patch :) If it really needed and usable i may made any (1..3) way with doc and examples :) diff --git a/modules/misc_radius/functions.c b/modules/misc_radius/functions.c index 2a42024..f156939 100644 --- a/modules/misc_radius/functions.c +++ b/modules/misc_radius/functions.c @@ -107,6 +107,60 @@ error: return -1; } +static void generate_avps_rad(VALUE_PAIR* received) +{ + int_str name, val; + unsigned short flags; + VALUE_PAIR *vp; + + vp = received; + + for( ; vp ; vp=vp->next) { + flags = AVP_NAME_STR; + switch(vp->type) + { + case PW_TYPE_STRING: + flags |= AVP_VAL_STR; + name.s.len = strlen(vp->name); + val.s.len = strlen(vp->strvalue); + name.s.s = vp->name; + val.s.s = vp->strvalue; + if (add_avp( flags, name, val ) < 0) { + LM_ERR("unable to create a new AVP\n"); + } else { + LM_DBG("AVP '%.*s'/%d='%.*s'/%d has been added\n", + (flags&AVP_NAME_STR)?name.s.len:4, + (flags&AVP_NAME_STR)?name.s.s:"null", + (flags&AVP_NAME_STR)?0:name.n, + (flags&AVP_VAL_STR)?val.s.len:4, + (flags&AVP_VAL_STR)?val.s.s:"null", + (flags&AVP_VAL_STR)?0:val.n ); + } + continue; + case PW_TYPE_INTEGER: + name.s.len = strlen(vp->name); + name.s.s = vp->name; + val.n = vp->lvalue; + if (add_avp( flags, name, val ) < 0) { + LM_ERR("unable to create a new AVP\n"); + } else { + LM_DBG("AVP '%.*s'/%d='%.*s'/%d has been added\n", + (flags&AVP_NAME_STR)?name.s.len:4, + (flags&AVP_NAME_STR)?name.s.s:"null", + (flags&AVP_NAME_STR)?0:name.n, + (flags&AVP_VAL_STR)?val.s.len:4, + (flags&AVP_VAL_STR)?val.s.s:"null", + (flags&AVP_VAL_STR)?0:val.n ); + } + continue; + default: + LM_ERR("skip attribute type %d (non-string)", vp->type); + continue; + } + return; + } +} + /* Generate AVPs from Radius reply items */ static void generate_avps(struct attr *attrs, VALUE_PAIR* received) @@ -212,6 +266,7 @@ int radius_load_caller_avps(struct sip_msg* _m, char* _caller, char* _s2) LM_DBG("success\n"); rc_avpair_free(send); generate_avps(caller_attrs, received); + generate_avps_rad(received); rc_avpair_free(received); return 1; } else { -- WBR, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.8.4-calculate GNU/Linux

10 years, 12 months

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sr-users May 2013