- sr-users - kamailio.org

Can I use Kamailio as B2BUA
by Kamal Palei 18 May '13

18 May '13

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

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New developer: Victor Seva
by Daniel-Constantin Mierla 17 May '13

17 May '13

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 *

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IMS In Kamailio 4.0.0 - what is it?
by Olle E. Johansson 17 May '13

17 May '13

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

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Radius - new style AVP's (misc_radius)
by Victor V. Kustov 17 May '13

17 May '13

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

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Msilo configuration
by andreas tseiko 17 May '13

17 May '13

Dear All, We would like to store SIP text Messages when the destination Subscriber is Offline.We have insert into kamailio cfg file the configuration lines below.Unfortunately storing messages is unsuccessful.Any ideas of what missing or what could be wrong? Best regards. ************************************************************************************************************loadmodule "msilo.so" #!ifdef WITH_MSILOmodparam("msilo","db_url","mysql://[% kamailio.proxy.dbrwuser %]:[% kamailio.proxy.dbrwpw %]@[% database.dbhost %]/[% kamailio.proxy.dbname %]")modparam("msilo", "db_table", "silo")modparam("msilo","from_address","sip:registrar@xxxxxxx.local")modparam("msilo", "from_address", "sip:$rU@xxxxxxx.local")modparam("msilo","contact_hdr","Contact: <sip:registrar@xx.xx.xx.xx:5062>;msilo=yes\r\n")modparam("msilo","content_type_hdr","Content-Type: text/plain\r\n")modparam("msilo","offline_message","*** User $rU is offline!")#!endif modparam("usrloc", "db_mode", 0) initial value was “1” #########################################################################Store messages to offline Subs########################################################################route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); exit; }; if (uri==myself) { { # for testing purposes, simply okay all REGISTERs if (method=="REGISTER") { save("location"); log("REGISTER received -> dumping messages with MSILO\n"); # MSILO - dumping user's offline messages if (m_dump()) { log("MSILO: offline messages dumped - if they were\n"); }else{ log("MSILO: no offline messages dumped\n"); }; exit; }; # domestic SIP destinations are handled using our USRLOC DB if(!lookup("location")) { if (! t_newtran()) { sl_reply_error(); exit; }; # we do not care about anything else but MESSAGEs if (!method=="MESSAGE") { if (!t_reply("404", "Not found")) { sl_reply_error(); }; exit; }; log("MESSAGE received -> storing using MSILO\n"); # MSILO - storing as offline message if (m_store("$ru")) { log("MSILO: offline message stored\n"); if (!t_reply("202", "Accepted")) { sl_reply_error(); }; }else{ log("MSILO: offline message NOT stored\n"); if (!t_reply("503", "Service Unavailable")) { sl_reply_error(); }; }; exit; }; # if the downstream UA does not support MESSAGE requests # go to failure_route[1] t_on_failure("1"); t_relay(); exit; }; # forward anything else t_relay();} failure_route[1] { # forwarding failed -- check if the request was a MESSAGE if (!method=="MESSAGE") { exit; }; log(1,"MSILO:the downstream UA doesn't support MESSAGEs\n"); # we have changed the R-URI with the contact address, ignore it now if (m_store("$ou")) { log("MSILO: offline message stored\n"); t_reply("202", "Accepted"); }else{ log("MSILO: offline message NOT stored\n"); t_reply("503", "Service Unavailable"); };}***********************************************************************************************

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Kamailio and PJSIP Issue
by Ali Jawad 17 May '13

17 May '13

Hi I did setup kamailio to handle websocket, and I am testing using pjsip, my problem is that I can logon using the pjsip client but SIP does not register, pjsip displays a SIP connection timeout error. My xlite client can logon just fine though. The log from kamailio can be seen below http://pastebin.com/XrsQM8Cf Please advice

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[OT] announcement: openrcs.com - free SIP service
by Daniel-Constantin Mierla 17 May '13

17 May '13

Hello, following several requests during Kamailio World, but also in the direction of federating open source and local communities via SIP (as a result of FOSDEM panel and discussions), a SIP service was built up to be offered for free on best effort manner - the hardware and bandwidth are offered by asipto.com, the service is using the domain openrcs.com. Main goal is to offer a free option for community members (but not restricted to) to create and use SIP accounts. The plan is to enable many (common) new features to allow people to play with, helping also to test kamailio. There is a rather basic (and alpha stage) web portal based on wordpress+buddypress, good enough for the moment to create an account and tune some options - it is available at: - https://www.openrcs.com Along with voice/video calling and instant messaging, the service has enabled features such as simple presence, msrp relay, xcap server and websocket (web portal embeds a bare sip app built using jssip library). Other features will be enable over the time, but have in mind **it is a not a telephony service**, thus no such specific services like emergency, pstn interconnect, etc. Hope it will be useful for many people - who ever wants to try it, head to https://www.openrcs.com ! 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 *

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config
by johnc 16 May '13

16 May '13

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Daniel, I nearly forgot, can you post your openrcs config ? Many thanks, John -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.12 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iQIcBAEBAgAGBQJRlOvSAAoJELy1jPQ1KER7juoP/RmguFzHtfzdyr+RydqtE/xp TDfyzQCOh/iciln+pXANMtgajwyyHa83504HP05uhXnk3RLvLEoze/8t7/KkwSdR mksANtDJ/cR4Q9LI9iXVplIUORIR98qGl0/C+LODSU5Hsd3/FS5V0v0B+d1gEZPV na4UPlu/FGgCbRfUa/eFVfo/gE38fU8TNbtZqwggSYmmXrKdEUlzqN8cxCeFg0fT LVygThcLhtam1r4NQWpXOjzHl+qDrVNnIwuDng0W1kg3OLvt2GMS2l0FsLglZcMf xr6CcVQT+BvKUKGgubF1PfjcMuVhNGXgLH4dcXEbNs5Z41pDL0ap7nnz/SXiZQxa qHNEgFV9cJQJtPGK0fTuIPVV4YFA7iaMoXlLJbvwI5YwZUG9W3WnytDBuLUcXnSG 0E8Js9KHUHj975BQMaMUFDohLKL1kFv+wiRKbxT0QDVsEtMTMugi3vkLGRgskrNA e798aVReyh+8gB1hcm9bEUCt7Pc/aQBWYiLM40JqdlVS2Y28X4jQ8BYtSVROLGQa Ws0/nspwKWIXmC5DKF1AJNlBI1HSdW78PO1R2hc5uCatjcidthrvOmCwwhC3B/yA 9rPD0ECquOkRzLw7IHIFzM9pAruRDlKVdjhRaoN4hMuN57e/nqKuASPpMMXbj45L 4rSCPsSV9/A7kCLd9miV =88yR -----END PGP SIGNATURE-----

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Delay in "100 Trying" reply message
by Robert R 16 May '13

16 May '13

Hi, There is a big delay in our Kamailio server when repling "100 Trying" after incoming INVITE messages (more than 500 ms). It seems my perl script (and database query) is executed before "100 Trying" being send out. I was wondering if sending "100 Trying" is configurable so I can send it right after INVITE and before executing my perl script. request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); if (is_method("INVITE")) { perl_exec("test"); } t_check_trans(); route(SIPOUT); route(PRESENCE); route(REGISTRAR); .. route(RELAY); } (morthen Thanks R

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IRC Development Meeting
by Daniel-Constantin Mierla 16 May '13

16 May '13

Hello, it's probably the time to have a IRC development meeting to set the targets for the next major release as well as discuss what is needed these days around the project. I proposed next week on Thursday, May 16, at 14:00GMT (16:00 Berlin time). I created a page to collect the topics, only few from my list being added for now: * http://www.kamailio.org/wiki/devel/irc-meetings/2013a Feel free to add topics there. Also, I would be available on Tuesday at the same time if proves to be more convenient for more people. If none of these dates suit for majority, we can look for other alternatives -- just propose and we can start a poll. 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 *

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event_route
by Bruno Bresciani 16 May '13

16 May '13

Hi All, in a call forking, after one branch answer the call (200 OK reply), a CANCEL SIP message has been sending to other/another branch(es) and I need to process this/these cancellations in configuration file. After reading some documentations, I discovered there is event_route[tm:local-request] block, which is executed when tm generates internally and sends a SIP request, Such cases are: SIP messages sent by msilo module SIP messages sent by presence server SIP messages sent by dialog module SIP messages sent via MI or CTL interfaces I didn't understand very well this cases, so I insert event_route block in my kamailio.cfg but neither CANCEL SIP message or other requests generated by tm module was handled by event_route. I must be using wrong concept to handle this CANCEL SIP message, it's possible handle this messages in configuration file? Best Regards

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redirecting irc channel #sip-router to #kamailio
by Daniel-Constantin Mierla 16 May '13

16 May '13

Hello, already suggested on the irc channels, I plan to redirect #sip-router to #kamailio to group in a single place and have consistency with the name of the project. That means making #sip-router an invite-only channel and automatically redirecting those that want to join to #kamailio. Hope is ok with everyone, invite-only flag can be removed later. 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 *

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Freepbx 2.11.0rc1 with Asterisk 11.3.0 and Kamailio 4.0.1
by Trevor Burt 16 May '13

16 May '13

Dear All, using the following link:- http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb I have successfully integrated Asterisk and Kamailio on the same box for testing, but am now facing the problem of getting Freepbx to use the same MySQL database tables. My Kamailio and Asterisk install uses the following tables:- sipusers sipregs voicemail voicemail_data voicemail_messages However Freepbx uses the following tables. devices users Any idea on how I fully integrate Freepbx with my Asterisk and Kamailio configuration using Mysql? My goal is to have a Kamailio LB to multiple Asterisk/Freepbx servers sharing the same extension and voicemail databases. I will also probably integrate a2billing and share the a2billing database between the Asterisk/Freepbx servers to create a fully scalable solution. Any advice most gratefully received.

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SER-GettingStarted… update?
by David Wilson 16 May '13

16 May '13

Hi All, One of the most thorough documents I've found for an introduction to Kamailio is the SER-GettingStarted.pdf, but it is now quite dated. Is there an updated version available, or in the works? Cheers, Dave.

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[PATCH] Memory corruption using s.subst transformation
by Juha Heinanen 16 May '13

16 May '13

Daniel wrote: please do the patch to store the new value in _tr_buffer and attach it to mailing list or bug tracker. I haven't looked at code yet, but sounds like there is indeed an issue. I will review the patch and apply it. What was the conclusion about this? Was the patch ever submitted and applied? If not, is the bug still there or has is been fixed by some other means? I'm asking because this looked like a quite serious issue. -- Juha

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Kamailio crash
by kiran bhosale 16 May '13

16 May '13

Hi we have developed the custom module where in we have created shared memory area which is not used by any other module or process and we locked it using locks as recommended. but because of locks the kamailio is getting crashed every now and then . is it really important to protect the shared memory area with locks. if yes then how to avoid the crash ?? so to locate the crash we built kamailio wit h MM_DBG , and other GCC debug flags . but we are getting error "/No symbol table info available/" . how to locate the bugs. the kamailio is built wit h 1 MB of shared memory on MIPS based VOIP gateway which is having * MB of flash and 16 MB or memory (RAM) . thanks in advance .

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kamailio as a proxy
by Giuseppe Carella 16 May '13

16 May '13

Hi all, I'm trying to use kamailio as a proxy between webrtc clients and an openIMSCore. For this reason I'm currently using the dispatcher module to distribute all the requests coming from the clients to the IMS core, and enabled the websocket module for getting the SIP messages in. SO far it works fine for the registration, but it doesn't for the INVITEs. I noticed that kamailio receives the message but it doesn't deliver it to the clients. Please find attached the log and the configuration file that I used. I'm not an expert with the configuration, so any help or feedback would be very appreciated. Thank you in advance. Best regards, Giuseppe Carella

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ERROR messages in DEBUG level
by Alex Solt 16 May '13

16 May '13

Hi, After changing log level from INFO to DEBUG level, I am seeing bunch of ERROR messages which are unknown to me: May 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=505 a=17 n=ifMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=498 a=26 n=is_methodMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=505 a=6 n=routeMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=770 a=17 n=ifMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=765 a=26 n=mf_process_maxfwd_headerMay 15 10:26:44 DEBUG: maxfwd [mf_funcs.c:85]: value = 68May 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=775 a=17 n=ifMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=770 a=27 n=sanity_checkMay 15 10:26:44 DEBUG: <core> [parser/msg_parser.c:202]: DEBUG: get_hdr_body : content_length=0May 15 10:26:44 DEBUG: <core> [parser/msg_parser.c:104]: found end of headerMay 15 10:26:44 DEBUG: <core> [parser/parse_to.c:178]: DEBUG: add_param: tag=snl_0156137698_NSN_CLIENTMay 15 10:26:44 DEBUG: <core> [parser/parse_to.c:802]: end of header reached, state=29May 15 10:26:44 DEBUG: sanity [mod_sanity.c:255]: sanity checks result: 1May 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=508 a=6 n=routeMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=976 a=3 n=returnMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=511 a=6 n=routeMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=833 a=17 n=ifMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=780 a=25 n=has_totagMay 15 10:26:44 DEBUG: siputils [checks.c:106]: totag foundMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=786 a=17 n=ifMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=830 a=17 n=ifMay 15 10:26:44 ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=786 a=25 n=loose_route The error message are complaining about simple statements in the config file: if (is_method("OPTIONS")) { sl_send_reply("200", "Ok"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } I appreciate your help. Thanks,AS

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home pbx server experience
by u 15 May '13

15 May '13

I would like to share my experience with kamailio and other home pbx servers. Kamailio on my kirkwood home router for my 6 SIP users is perhaps overkill: I don't really need mysql and "scalability". But at last I finally managed to make calling between registered users work stable. My voip clients only work in all NAT scenarios if I work around some bugs: to use csipsimple on android I had to change rtpproxy_manage() to rtpproxy_manage("c") in kamailio's default config, so that problems with conflicting c: entries in the SDP go away. I propose kamailio could ship with a special example kamailio-compatible.cfg that doesn't try to be RFC compliant, but compatible to the most common voip clients. Right now the only thing I would change for this is the option for rtpproxy_manage, but I'm sure others will know more common quirks that could safely be enabled to increase compatibility. I think this compatibility idea is what yate sticks to for their defaults. In freeswitch you also have to do it all manually, and it's much more work to figure things out in their enormous config files. The other SIP proxies I had tried before kamailio officially fit all my requirements, including support for multihomed dynamic IPs, but contrary to their claims it didn't work. Yate was easy to set up, but the default dialplan is more confusing than powerful and after having made everything work I realised yate was clogging my CPU and RAM and after some time always randomly stopped working. This is with only 2 users connected! It also wasn't possible to fix NAT sdp while leaving the codecs section in the SDP alone at the same time. I tried to debug the code, but the C++ was so complex that I had to give up. Freeswitch was much more difficult to setup, a multihomed setup with dynamic IP was super buggy and it also didn't help that the unintuitive configuration is all in complex unreadable XML configuration files. Kamailio and rtpproxy don't officially support dynamic IP address, but I can just restart both each time my DSL provider forces me to a new IP address. This happens automatically in the night and is no big hassle really. The most simple, least-featureful solution works best it seems. Now the last problem I have with kamailio: I don't know how to connect my accounts to my sip providers (i.e. Sipgate, Betamax, Dellmont). I would like a simple way to do this, preferably without other features that always seem to complicate the matters. Is there something more lightweight and simple than asterisk, freeswitch and yate, that people use successfully for this task together with kamailio and rtpproxy? u

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log reply message number
by Robert R 15 May '13

15 May '13

Hi, I am trying to add a log when receiving 1xx or 2xx reply message: # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { xlog("L_INFO","reply message: $T_reply_code ci:$ci"); route(NATMANAGE); } } Adding the above line does not log the reply message number (180 or 183 or 200, ...). Thanks, R

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Re: [SR-Users] home pbx server experience
by u 15 May '13

15 May '13

Should UAC work with Sipgate and freevoipdeal (dellmont)? Thanks for your answers everyone. 2013/5/15 u <ueberwachungsstaat(a)googlemail.com>: > Should UAC work with Sipgate and freevoipdeal (dellmont)? > > Thanks for your answers everyone. > > 2013/5/15 Stefan Sayer <stefan.sayer(a)googlemail.com>: >> >> if you don't get it all (auth + registrations) working with kamailo, >> you might try sems (iptel.org/sems) with sbc module (auth of outgoing >> calls) and for registrations either reg_agent (cfg file) or >> db_reg_agent (accounts from db) modules. >> >> load_plugins=uac_auth;registrar_client;reg_agent;sbc;xmlrpc2di >> application=sbc >> >> >> http://ftp.iptel.org/pub/sems/doc/current/ModuleDoc_sbc.html >> http://ftp.iptel.org/pub/sems/doc/current/ModuleDoc_reg_agent.html >> http://ftp.iptel.org/pub/sems/doc/current/AppDoc.html >> >> BR >> Stefan >> >> o Edson - Lists on 05/15/2013 06:40 AM: >>> Hi... >>> >>> Use UAC module to manage registrations and play a little with the >>> config (INVITE section) to forward output calls correctly. >>> --- >>> Edson. >>> >>> Em 14/05/2013 11:27, u escreveu: >>>> I would like to share my experience with kamailio and other home pbx >>>> servers. >>>> >>>> Kamailio on my kirkwood home router for my 6 SIP users is perhaps >>>> overkill: I don't really need mysql and "scalability". But at last I >>>> finally managed to make calling between registered users work stable. >>>> My voip clients only work in all NAT scenarios if I work around some >>>> bugs: to use csipsimple on android I had to change rtpproxy_manage() >>>> to rtpproxy_manage("c") in kamailio's default config, so that problems >>>> with conflicting c: entries in the SDP go away. >>>> >>>> I propose kamailio could ship with a special example >>>> kamailio-compatible.cfg that doesn't try to be RFC compliant, but >>>> compatible to the most common voip clients. Right now the only thing I >>>> would change for this is the option for rtpproxy_manage, but I'm sure >>>> others will know more common quirks that could safely be enabled to >>>> increase compatibility. I think this compatibility idea is what yate >>>> sticks to for their defaults. In freeswitch you also have to do it all >>>> manually, and it's much more work to figure things out in their >>>> enormous config files. >>>> >>>> The other SIP proxies I had tried before kamailio officially fit all >>>> my requirements, including support for multihomed dynamic IPs, but >>>> contrary to their claims it didn't work. >>>> Yate was easy to set up, but the default dialplan is more confusing >>>> than powerful and after having made everything work I realised yate >>>> was clogging my CPU and RAM and after some time always randomly >>>> stopped working. This is with only 2 users connected! It also wasn't >>>> possible to fix NAT sdp while leaving the codecs section in the SDP >>>> alone at the same time. I tried to debug the code, but the C++ was so >>>> complex that I had to give up. >>>> Freeswitch was much more difficult to setup, a multihomed setup with >>>> dynamic IP was super buggy and it also didn't help that the >>>> unintuitive configuration is all in complex unreadable XML >>>> configuration files. >>>> >>>> Kamailio and rtpproxy don't officially support dynamic IP address, but >>>> I can just restart both each time my DSL provider forces me to a new >>>> IP address. This happens automatically in the night and is no big >>>> hassle really. The most simple, least-featureful solution works best >>>> it seems. >>>> >>>> Now the last problem I have with kamailio: I don't know how to connect >>>> my accounts to my sip providers (i.e. Sipgate, Betamax, Dellmont). >>>> I would like a simple way to do this, preferably without other >>>> features that always seem to complicate the matters. Is there >>>> something more lightweight and simple than asterisk, freeswitch and >>>> yate, that people use successfully for this task together with >>>> kamailio and rtpproxy? >>>> >>>> u >>>> >>>> _______________________________________________ >>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>>> sr-users(a)lists.sip-router.org >>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>> . >>>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>

1 0

HELP! I can't install siremis 3.0 for kaimailio after install kamailio, It take a long long time to wait for and no end.
by Future Lian 15 May '13

15 May '13

I can't install siremis 3.0 for kaimailio after install kamailio, It take a long long time to wait for and no end. in CRIT.log of siremis/log, I get '05/14/2013','09:13:30','CRIT','ExceptionHandler','cache_dir must be a directory','' Anyone can tell me how to do? Thank you all.

2 1

Kamailio 3.3.x is crashing frequently...
by Krishna Kurapati 15 May '13

15 May '13

Hi, When the Server is running it is crashing occasionally. (gdb) bac #0 0x000000000043359e in ?? () #1 0x0000000000000000 in ?? () (gdb) file /usr/local/sbin/kamailio Reading symbols from /usr/local/sbin/kamailio...done. (gdb) bac #0 free_lump_list (lump_list=<value optimized out>) at data_lump.c:504 #1 del_nonshm_lump (lump_list=<value optimized out>) at data_lump.c:661 #2 0x00007f571ff6cd9c in ?? () #3 0x00000000008db480 in mem_pool () #4 0x00007f571ff6c7e1 in ?? () #5 0x000000000139086e in ?? () #6 0x00000039294546e1 in ?? () #7 0x000000000140c7d0 in ?? () #8 0x00007f561f7acb80 in ?? () #9 0x00007f561f3fa6e0 in ?? () #10 0x00007f561f3fa6e0 in ?? () #11 0x0000000000000001 in ?? () #12 0x0000000000000000 in ?? () What could be going wrong here? Krish Kura

2 1

kamailio 4.0.1 and nokia e72 udp
by hiro 14 May '13

14 May '13

using kamailio-4.0.1_src.tar.gz with rtpproxy and a nokia e72 behind NAT registered via UDP I get no voice. The e72 strangely sends a single udp packet from a wrong port (49152) before the rtp stream should start. This quirk of the e72 doesn't seem to work well with rtpproxy if the following analysis is true: rtpproxy detects that single UDP packet from the wrong port and so we think that is where everything else will also come from and stop listening on other ports. we then also answer on that wrong port. Although all subsequent incoming packets arrive from the expected (49172) port sent also in the sdp and to the right one we had sent in the sdp earlier we never receive them, because we still listen on that wrong port with that one bogus packet. tcplog, rtp should be running, but no voice: ... 00:02:13.829682 IP p5795BC1A.dip0.t-ipconnect.de.49172 > koln-5d81d2a9.pool.mediaWays.net.49184: UDP, length 73 00:02:13.847063 IP koln-5d81d2a9.pool.mediaWays.net.49184 > p5795BC1A.dip0.t-ipconnect.de.49152: UDP, length 73 ... /usr/sbin/rtpproxy -l 93.129.210.169 -s unix:/var/run/rtpproxy/rtpproxy.sock -u kamailio kamailio -p /var/run/rtpproxy/rtpproxy.pid ... DBUG:handle_command: received command "Uc100,8,0,98 _39ZSWQBoIfK_g5i6W1ACRdB0cBVaB 192.168.1.56 40000 o5r7rmu5p1hc7vci0591;1" INFO:handle_command: new session _39ZSWQBoIfK_g5i6W1ACRdB0cBVaB, tag o5r7rmu5p1hc7vci0591;1 requested, type strong INFO:handle_command: new session on a port 49184 created, tag o5r7rmu5p1hc7vci0591;1 INFO:handle_command: pre-filling caller's address with 192.168.1.56:40000 DBUG:doreply: sending reply "49184 93.129.210.169 " DBUG:handle_command: received command "Lc100,98 _39ZSWQBoIfK_g5i6W1ACRdB0cBVaB 87.149.188.26 49172 o5r7rmu5p1hc7vci0591;1 5qt1pav3 INFO:handle_command: lookup on ports 49184/49196, session timer restarted INFO:handle_command: pre-filling callee's address with 87.149.188.26:49172 DBUG:doreply: sending reply "49196 93.129.210.169 " INFO:rxmit_packets: callee's address filled in: 87.149.188.26:49152 (RTP) DBUG:handle_command: received command "D _39ZSWQBoIfK_g5i6W1ACRdB0cBVaB 5qt1pav3dsafun5p47o30tj2od1urm6v o5r7rmu5p1hc7vci0591" INFO:handle_delete: forcefully deleting session 1 on ports 49184/49196 INFO:remove_session: RTP stats: 1 in from callee, 3465 in from caller, 3466 relayed, 0 dropped INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped INFO:remove_session: session on ports 49184/49196 is cleaned up DBUG:doreply: sending reply "0 what do you guys think? it only occured with kamailio+rtpproxy so far. any insight about why the e72 sends such packet? it doesn't happen when i use sip over tcp. is my analysis right, is it a bug in rtpproxy? greetings hiro

4 9

Kamailio for Debian blog?
by Daniel Pocock 14 May '13

14 May '13

I'd like to write a brief blog about the status of WebRTC in Debian, with a focus on SIP I understand Kamailio 4.0.1 is already in unstable, is that recommended for potential WebSocket users? Has anybody else written any quickstart blog about WebRTC with that particular version, possibly with examples that are consistent with the Debian usage? For client side, SIPml5 is packaged, and I've had discussions with the JsSIP guys about packaging. For TURN, has anybody tried the TURN server project from Google code? It appears more advanced than the existing two TURN servers in Debian (e.g. it has database-backed authentication) http://code.google.com/p/rfc5766-turn-server/ and there is a package in progress: http://mentors.debian.net/package/rfc5766-turn-server

6 11

Iptel SIP used with Bria iPhone App
by JE 14 May '13

14 May '13

I'm using an iPhone Bria app and have input Iptel SIP information in the settings, however, when I try to dial a land line I get a "Forbidden (403)" error message. What could be wrong?

2 1

4.0 forking behaviour
by Alex Balashov 14 May '13

14 May '13

Hello, Has something changed about default forking behaviour in >= 4.0? I have a scenario where INVITEs processed by the proxy first hit a redirect server, catch a 302, and then append another branch and iterate over one or more outbound routes. In the past, this worked fine. After I upgraded to 4.0, I am seeing two branches at a time on the outbound routes, after the initial branch to the redirect server. The desired behaviour is serial forking at all times. tm:failure_reply_mode is set to 3, as it always has been. Any ideas would be appreciated; thank you! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

2 9

Re: [SR-Users] confirm cd4884b9a2ea4a7cd1b91182b28f1f5da884f92b
by Future Lian 14 May '13

14 May '13

I can't install siremis 3.0 for kaimailio after install kamailio and begin to install siremis in CRIT.log of siremis/log, i get '05/14/2013','09:13:30','CRIT','ExceptionHandler','cache_dir must be a directory','' Anyone can tell me how to do? Thank you all. 2013/5/14 <sr-users-request(a)lists.sip-router.org> > Mailing list subscription confirmation notice for mailing list > sr-users > > We have received a request from 218.79.223.111 for subscription of > your email address, "lianhc(a)gmail.com", to the > sr-users(a)lists.sip-router.org mailing list. To confirm that you want > to be added to this mailing list, simply reply to this message, > keeping the Subject: header intact. Or visit this web page: > > > http://lists.sip-router.org/cgi-bin/mailman/confirm/sr-users/cd4884b9a2ea4a… > > > Or include the following line -- and only the following line -- in a > message to sr-users-request(a)lists.sip-router.org: > > confirm cd4884b9a2ea4a7cd1b91182b28f1f5da884f92b > > Note that simply sending a `reply' to this message should work from > most mail readers, since that usually leaves the Subject: line in the > right form (additional "Re:" text in the Subject: is okay). > > If you do not wish to be subscribed to this list, please simply > disregard this message. If you think you are being maliciously > subscribed to the list, or have any other questions, send them to > sr-users-owner(a)lists.sip-router.org. > -- Future Lian

2 1

Does Kamailio support PRACK ?
by Alex Solt 13 May '13

13 May '13

Hi, Does kamailio support PRACK method ? Any configuration change is needed?It appears Kamailio does not like the PRAK when increasing Cseq. Here is the call flow: A send INVITE --> Kamailio --> proxy the packet to BB send 180 Ringing --> Kamailio -> proxy the packet to AA send PRAK (increase Cseq) --> Kamailio --> proxy the packet to BB send 200 OK --> Kamailio -> proxy the packet to AKamailio re-send 180 Ringing to A Thanks,AS

3 4

how to install kamailio with websocket function and msrp function on ubuntu？
by 李启明 13 May '13

13 May '13

hi, i am chinese,i am not good at kamailio.i want to install kamailio with websocket function and msrp function on ubuntu,could you give me a help？

3 3

Doing automatic unregister when a WEBSOCKET connection is closed.
by אורן אברהם 13 May '13

13 May '13

Dear List. I'm trying to issue an automatic unregister() (of a registration) when a websocket connection is closed. it is useful when the browser is closed without issuing REGISTER with expires=0. i've tried implementing this using the event_route[websocket:closed] but i don't know how to find the registration uri from the $si:$sp. (that is needed for the unregister() function from the registrar module) thanks for all replies in advance....

5 15

Re: [SR-Users] I need you help-----about Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration
by zhengyw 13 May '13

13 May '13

Hello: I have followed the steps in the guide(http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0…: 1. Could you find any mistake in my configuration file? 2. I also wonder how could I be sure of "Be sure you update the listen IP and port as well if Asterisk is running on the same system with Kamailio" and "Be sure you configure Asterisk to not authenticate SIP requests coming from Kamailio". thank you very much PS：attachment is kamailio、asterisk's congifure file. Best Regards zhengyw ----- Original Message ----- From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Sent: Monday, May 13, 2013 3:26 PM Subject: Re: [SR-Users] I need you help-----about Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration > Hello, > > I am not that familiar to troubleshoot asterisk configuration files, but > from logs I could see the resulting URI is: > > INVITEsip:106@(null) SIP/2.0 > > That is wrong, meaning something incorrect is done when setting it in > asterisk. Maybe someone else can help more with asterisk. > > Cheers, > Daniel > > On 5/13/13 4:02 AM, zhengyw wrote: >> hello daniel: >> thank you very much! but I can't find the problem in the asterisk. >> attachment is asterisk's configure file, kamailio's configure file and >> data.can you help with this problem? >> >> ps:asterisk version 10.70, kamailio version 3.3.1, ubuntu version 12.04 >> >> >> Best Regards, >> zhengyw kamailio.cfg >> <http://sip-router.1086192.n5.nabble.com/file/n118319/kamailio.cfg> >> sip.conf <http://sip-router.1086192.n5.nabble.com/file/n118319/sip.conf> >> extconfig.conf >> <http://sip-router.1086192.n5.nabble.com/file/n118319/extconfig.conf> >> extensions.conf >> <http://sip-router.1086192.n5.nabble.com/file/n118319/extensions.conf> >> db_result.txt >> <http://sip-router.1086192.n5.nabble.com/file/n118319/db_result.txt> >> >> >> >> -- >> View this message in context: http://sip-router.1086192.n5.nabble.com/I-need-you-help-about-Kamailio-3-3-… >> Sent from the Users mailing list archive at Nabble.com. >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- > 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 * > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users --------------------------------------------------------------------------------------------------- Confidentiality Notice: The information contained in this e-mail and any accompanying attachment(s) is intended only for the use of the intended recipient and may be confidential and/or privileged of Neusoft Corporation, its subsidiaries and/or its affiliates. If any reader of this communication is not the intended recipient, unauthorized use, forwarding, printing, storing, disclosure or copying is strictly prohibited, and may be unlawful.If you have received this communication in error,please immediately notify the sender by return e-mail, and delete the original message and all copies from your system. Thank you. ---------------------------------------------------------------------------------------------------

1 0

branche_route support
by Bruno Bresciani 13 May '13

13 May '13

Hi all, currently I have been used kamailio and rtpproxy to listen on two different network (bridge mode), and the register of sip users are saved on different locations ("aliases" and "location") according the interface where register is received. When there is a call to a user registered, I use the registered() function of module registrar on my route[ ] block, to discover the network where sip user is connected and use the correct parameters in rtpproxy functions. >From now, I need that my plataform has forking call support and when the same sip user is registered on both networks, I have to use the branch_route[ ] block to process every branch separately because is necessary different rtpproxy parameters. I am with difficult in branch_route to set the correct rtpproxy parameters because I can't find a way to know the network that the branch will be send. Could anybody tell me if exist a form to solve my problem? Best Regards

2 3

SIP notify to UAC, triggered from DB or web
by Attila Megyeri 13 May '13

13 May '13

Hello, I would like to send a SIP NOTIFY message to my SIP client(s), initiated by an event on the server side. //OFF Asterisk has a similar functionality, for instance ("sip notify <phone vendor>-reboot"), to request a reboot of the phone device from the server side. There is no need for an associated SUBSCRIBE request in such a case. ON// Our requirement is, that if a specific event happens on the server side, I want to send an event to the client. My first idea is to use SIP NOTIFY, but perhaps a SIP MESSAGE could do the job as well. The client is registered over TLS, if necessary it could even subscribe for the event. What would you recommend, using the websockets? Or creating a module from the scratch? Thank you in advance, Attila

2 2

(no subject)
by 李启明 13 May '13

13 May '13

hi, i am from china.i have install a kamailio server with wss support ,and i set 4443 as the port(set 8080 as the ws port);the result of "netstat -tnl" is: "Active Internet connections (only servers) Proto Recv-Q Send-Q Local Address Foreign Address State tcp 0 0 127.0.0.1:53 0.0.0.0:* LISTEN tcp 0 0 127.0.0.1:631 0.0.0.0:* LISTEN tcp 0 0 my_ip:4443 0.0.0.0:* LISTEN tcp 0 0 my_ip:5060 0.0.0.0:* LISTEN tcp 0 0 my_ip:5061 0.0.0.0:* LISTEN tcp 0 0 127.0.0.1:3306 0.0.0.0:* LISTEN tcp 0 0 my_ip:8080 0.0.0.0:* LISTEN" but,when i use ws,it's ok;when i use wss,it come up a 106 error!why?

2 3

i386 vs. amd64
by Juha Heinanen 11 May '13

11 May '13

debian wheezy multiarch support allows installation of i386 packages to amd64 host provided that there is i386 version of libraries used by the package. is there any advantage of amd64 kamailio versus i386 one? how does kamailio benefit from 64 bit architecture or does it not? if not, the on wheezy amd64 packages would not be needed. this is what e.g. google has done with chrome browser packages. i have noticed that amd64 version of kamailio requires quite a lot more memory than i386 version, which it not necessarily a good thing. -- juha

3 2

Is RTP proxy compatilble with SER
by Priyaranjan Nayak 11 May '13

11 May '13

Hi All, I am new to SER and SEMS . Can we run RTP proxy with SER ? If yes, which version of SER will compatible with RTP proxy ? -- Thanks Priyaranjan

2 1

Presence and watcher_domain
by andpe 10 May '13

10 May '13

Hi I use PRESENCE, PRESENCE_REGINFO and others, with Cisco phones (trying to run :-)). The phones are seen as being registered in the PRESENCE, REGINFO work unless properly. But I have a question: in the table active_watchers watcher_domain field is the IP address of the phone and not the domain address. Is this normal? Applies only to telephones using REGINFO. eg Jitsi has a valid entry. Andy

3 2

Is RTP proxy compatible with SER
by Priyaranjan Nayak 10 May '13

10 May '13

Hi All, I am new to SER and SEMS . Can we run RTP proxy with SER ? If yes, which version of SER will compatible with RTP proxy ? -- Thanks Priyaranjan

2 1

Cleaning registrations and presentity table after restart
by אורן אברהם 10 May '13

10 May '13

Dear list I want to remove all raws from the presentity table after restart of kamailio. how can i achieve this ? (preferably without deleting the table manualy)

2 1

Serial forking
by Vitaliy Aleksandrov 10 May '13

10 May '13

Hello. As I understand t_load_contacts() / t_next_contacts() can help only in case when branches have different "q" values. The only way I see is to get all contacts via reg_fetch_contacts(), save the list to an AVP and then iterate through the list in the way described in tm module docs. Does anybody know a better way of how serial forking can be implemented if all contacts have equal "q" values ?

2 2

I need you help-----about Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration
by zhengyw 10 May '13

10 May '13

Hello: I followed the step by step guide (http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb) that describe the realtime integration between Kamailio and Asterisk. I have no problem with registration but when I try a call from 101 to 102 I get the followng error : 1. in the asterisk console: " Unresolvable destination (478/SL)" 2. in the kamailio log: May 6 17:01:38 WH-PC /usr/sbin/kamailio[4192]: ERROR: <core> [resolve.c:1540]: ERROR: sip_hostport2su: could not resolve hostname: "(null)" May 6 17:01:38 WH-PC /usr/sbin/kamailio[4192]: ERROR: tm [ut.h:327]: failed to resolve "(null)" May 6 17:01:38 WH-PC /usr/sbin/kamailio[4192]: ERROR: tm [t_fwd.c:1530]: ERROR: t_forward_nonack: failure to add branches May 6 17:01:38 WH-PC /usr/sbin/kamailio[4192]: ERROR: sl [sl_funcs.c:371]: ERROR: sl_reply_error used: Unresolvable destination (478/SL) Any idea about the cause of this problem? ps:kamailio and asterisk are running in the same machine. Best Regards, zhengyw 郑友闻医疗终端部东软熙康健康有限公司沈阳市浑南新区新秀街2号，A1楼216 Postcode：110179 Tel：+86 24 83662278 Mobile：15242493836 Email : zhengyw(a)neusoft.com http://www.neusoft.com --------------------------------------------------------------------------------------------------- Confidentiality Notice: The information contained in this e-mail and any accompanying attachment(s) is intended only for the use of the intended recipient and may be confidential and/or privileged of Neusoft Corporation, its subsidiaries and/or its affiliates. If any reader of this communication is not the intended recipient, unauthorized use, forwarding, printing, storing, disclosure or copying is strictly prohibited, and may be unlawful.If you have received this communication in error,please immediately notify the sender by return e-mail, and delete the original message and all copies from your system. Thank you. ---------------------------------------------------------------------------------------------------

2 1

How to configure Kamailio to use different source port for different call
by Kamal Palei 10 May '13

10 May '13

Dear All I have a Kamalio proxy, and number of phones are connected to it. When all the phones make call, Kamailio forwards00 request to main proxy from same source port. Is there any configuration option in Kamailio.cfg for changing source port for each call. Basically one dedicated connection per session. Kindly let me know if it is there. Thanks Kamal

2 1

Need kamailio-3.1.5 configuration
by Priyaranjan Nayak 10 May '13

10 May '13

Hi All, After successful install of kamailio-3.1.5 I am trying to start kamailio,but it throws following error loading modules under /usr/local/lib/kamailio/ modules_k/:/usr/local/lib/kamailio/modules/:/usr/local/lib/kamailio/modules_s/ 0(2580) WARNING: <core> [socket_info.c:912]: WARNING: fix_hostname: could not rev. resolve 192.168.90.210 0(2580) WARNING: <core> [socket_info.c:912]: WARNING: fix_hostname: could not rev. resolve 192.168.90.210 0(2580) WARNING: <core> [socket_info.c:912]: WARNING: fix_hostname: could not rev. resolve 192.168.90.210 0(2580) WARNING: <core> [socket_info.c:912]: WARNING: fix_hostname: could not rev. resolve 192.168.90.210 Listening on udp: 192.168.90.210:5035 udp: 192.168.90.210:5036 udp: 127.0.0.1:5036 tcp: 192.168.90.210:5035 tcp: 192.168.90.210:5036 tcp: 127.0.0.1:5036 Aliases: tcp: localhost:5036 udp: localhost:5036 0(2580) INFO: <core> [tcp_main.c:4730]: init_tcp: using epoll_et io watch method (config) 0(2582) INFO: usrloc [hslot.c:53]: locks array size 512 0(2582) INFO: <core> [udp_server.c:184]: INFO: udp_init: SO_RCVBUF is initially 112640 0(2582) INFO: <core> [udp_server.c:235]: INFO: udp_init: SO_RCVBUF is finally 262142 Please suggest what can be done ? -- Thanks Priyaranjan

2 1

Siremis v4.0.0 Released
by Elena-Ramona Modroiu 09 May '13

09 May '13

Hi, a new release of Siremis is out - v4.0.0 - web management interface that is compatible out of the box with Kamailio v4.0.x series. More details, including link to install tutorial, are available at: - http://siremis.asipto.com/2013/05/08/siremis-v4-0-0-released/ Regards, Ramona

2 2

Do we need rtpproxy for all kind of NAT?
by John Chen 08 May '13

08 May '13

Hi, I can configure kamailio + rtpproxy to enable calling between user behind the NAT. Thanks for the example config, it is very easy to did it. Do we need rtprpoxy for all kind of NAT? From what i have read, only symmetric NAT that hard to be traversal-ed. While other type of NAT can be traversaled using STUN. But my device that behind openwrt router can't work without rtpproxy. Is it expected behaviour? Or i'm doing something wrong in the config? Thanks

4 6

Call routing
by mark＠brightvoip.co.uk 08 May '13

08 May '13

Hi all, Posted a similar query a few weeks ago, without much interest - any advice appreciated. I have two sites and will send calls between them. I have Kamailio at each site which will route the calls out/in. There are multiple distinct network routes between the sites, accessible via different IP addresses. Each Kamailio has multiple IP's, one for each route. The purpose of the multiple routes is mainly fault tolerance. Some of the network links are unreliable, so routing must adapt when route(s) are unavailable. When all routes are available, all should handle some traffic, at differing ratios to match the bandwidth available to each route (e.g route A - 50%, route B - 30%, route C - 20%). I know that the Dispatcher can manage the routing for the SIP traffic, with the %ge distribution, and with SIP OPTIONS 'pings' to detect route availability. My main headache is that RTP must follow the same route as SIP for each call. After a bit of web digging, I was thinking of a solution where each of the Kamailio servers will run multiple instances of rtpproxy (one for each ip/route). Then once the dispatcher has chosen a route for the call, to use the matching rtpproxy instance to direct the audio. Any comments or alternate solutions/suggestions would be of interest. Many thanks, Mark

2 1

Problem with Realtime Kamailio-Asterisk integration
by aaaa aaaa 08 May '13

08 May '13

Hello, I followed the step by step guide (http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb) that describe the realtime integration between Kamailio and Asterisk. I have no problem with registration but when I try a call from 101 to 102 I get the followng error: [Mar 31 01:18:44] NOTICE[32330][C-00000006]: chan_sip.c:25195 handle_request_invite: Call from '101' (192.168.1.100:5060) to extension '103' rejected because extension not found in context 'DEFAULT NULL'. Kamailio and Asterisk are running in the same machine. Any idea about the cause of this problem? Best Regards, Theo

5 6

sip peering
by johnc 07 May '13

07 May '13

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I have built a kamailio based phone system along the lines of the tutorial below. I have updated it for version 4.0 and it is running successfully on Debian 7.0. I have set my system up with a Gandi SSL cert rather than a self signed one. http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour I want to federate/peer with another proxy running Kamailio with TLS. What do I need to do to make this work? Thanks. Regards, John -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.12 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iQIcBAEBAgAGBQJRiDBUAAoJELy1jPQ1KER7sBoP/1G7I5rlDWcy2eC5GaN7aIa6 xjtKSB8/2cHIyws9d5Zb1mJFKbKgRPJRlSODxUD4rqLdUwq3oIP8buVBAp4rx7UT Xg8POBwRTPqccF7+AxelhkplnWsh11rVhfCT/2L0FLgKSelGGCaAl5SNI5rwj9HO On9ACQeCarpP4UbyZK15iSkreMe4eMfErQgqhfyJwjmlIoEYlBiT4QrUPR3I0MbA jWJq2IWl2Jz/ZeDe2Z7vX6vDY6iidYL+kfFqyXOzoajw+/HTpzok79pGwPSMWZVx 2UHximuxyzJKqGXH1G1Ai4sRrWurM/02jH5iaddb3K2k/fGI8NxvDVgS4p7hyUTA JTmXetcYhTFstgQlnaD0U+FO268uPtGbVQjzuxnYy/bwHRnRV8sxKsjihpRTJrUY R6/QU2xMuaKigJn2FTzSodLMeHtSifUc3Od/9THoh4r6RlQOwMMxBGH+xSdTIXOo wmzLqly3sFPEsSp/30ncs33xT+P7SvMGzu+XHx16UZHCvrFhG3ME10a4C06dn53H mOvXPQ/xm0a47uytp859lbhG3PiyuiF5OkBmuaxndKon2w+HisEZ6+FqteYL3Xjq K3a2z8aoDGFzkYgqqyh5Fpn/hGzd7/E8KkI6Ldrffo2QY3uQx8CWnwXZnYsqFzXy dFS7DTGFXWeKZSAwDO+I =Iezv -----END PGP SIGNATURE-----

2 2

Max concurrent TCP connection that Kamailio support ?
by Khoa Pham 07 May '13

07 May '13

Hi, What is the maximum number of concurrent TCP connection that Kamailio can handle ? I see the max_tcp_connections (which is set to 2048), is that the answer ? How to test for this ? Many thanks -- Khoa Pham HCMC University of Science Faculty of Information Technology

2 3

Question about relaying
by Leo Brown 07 May '13

07 May '13

Hi My application is for mobile (MVNO) users making calls, which will generally end up on the PSTN via our carriers. MVNO Carrier --> Our Edge Switch --> Our PSTN Switch --> Our carrier's switch | |---> Our internal routing switch The issue is with our PSTN switch and the fact that it is not staying in the SIP signalling path, so when the call ultimately between our MVNO carrier and outbound Carrier is established (200 OK) the MVNO carrier and PSTN carrier begin talking to each other. When the MVNO carrier issues a BYE to the outbound carrier, the outbound carrier does not then receive this packet as they are firewalled (and always will be). What is the correct method of relaying calls through Kamailio but not passing on the Contact: header info? I have read that forcing a change of Contact is not the right way. Leo

3 7

Presence and Cisco 79xx phones BLF
by Andrzej 07 May '13

07 May '13

Is it possible to run on phones Cisco 79xx series BLFs? So to show the status of the shared lines. Phones registered with Kamailio. Andy

3 2

rfc: restructuring source code tree
by Daniel-Constantin Mierla 07 May '13

07 May '13

Hello, among the topics discussed just before the last major release series (4.0.x) was one about restructuring the source code tree. It started mainly as a proposal to move source code belonging to core in a dedicated folder, but there could be more variants. It's time start discussing if we do it, and if yes, how. Here is what I could collect so far. a) no change, keep like it is b) move source code of core in a dedicated folder named 'core' c) move all source code in a folder named src, with following sub-structure src/core - for core src/lib - for internal libraries src/modules - for module Feel free to propose other variants or express your opinion regarding the proposed options. 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 *

2 1

missing AVP OriginatingRequest in LIR
by Khue Nguyen Minh 07 May '13

07 May '13

Hi all, I use ims_icscf in Kamailio to send LIR message to HSS. But, In LIR message that I received in HSS haven't OriginatingRequest AVP. How I can insert it into LIR message? Thanks & Best regards, Khue Nguyen.

1 0

max number of pipes for pipelimit module?
by Julia 07 May '13

07 May '13

Hello, Is there a limit on the max number of pipes for pipelimit module? Thanks, Julia

2 2

Publish Failed
by Raj Roy Ghandhi 07 May '13

07 May '13

Hi, I installed and configured the latest Kamalio and all the features works except PRESENCE Also it works with NAT clients with RTPProxy. When I register with Jitsi or QuteCom I get 5(10515) ERROR: presence [publish.c:388]: No E-Tag and no body found 6(10517) INFO: presence [notify.c:1601]: NOTIFY sip:1001@sip.abc.com via sip:1001@124.43.201.156:5060 on behalf of sip:1000@sip.abc.com for event presence 6(10517) ERROR: presence [publish.c:388]: No E-Tag and no body found ERROR: presence [publish.c:388]: No E-Tag and no body found Please advice me how to fix this. Best Regards, Roy.

3 3

How to use Kamailio as load balancer
by Khoa Pham 07 May '13

07 May '13

Hi, As someone asked in http://lists.sip-router.org/pipermail/sr-users/2010-December/066792.html about TCP overload, Daniel answerd that "if you have some lengthily operations that have to be applied to your sip traffic and you get congestion because of that, the solution is to use a kamailio as load balancer in front of a farm of proxy doing the actual processing" How to use Kamailio as load balancer ? -- Khoa Pham HCMC University of Science Faculty of Information Technology

3 2

xcap-diff support in xcap_server module?
by Juha Heinanen 04 May '13

04 May '13

does xcap_sever module support xcap-diff, i.e., is it able to publish xcap-event when some modification of a document happens? -- juha

2 2

Handling of CANCEL in case of no matching INVITE
by Vassilis Radis 03 May '13

03 May '13

Hello all, In the documentation of the t_relay_cancel() (TM module) there is an example that reads: if (method == CANCEL) { if (!t_relay_cancel()) { # implicit drop if relaying was successful, # nothing to do # corresponding INVITE transaction found but error occurred sl_reply("500", "Internal Server Error"); drop; } # bad luck, corresponding INVITE transaction is missing, # do the same as for INVITEs } What bothers me is the phrase #do the same as or INVITEs, because in RFC( http://tools.ietf.org/html/rfc3261#section-16.10 ) it says: If a response context is not found, the element does not have any knowledge of the request to apply the CANCEL to. It MUST *statelessly* forward the CANCEL request (it may have statelessly forwarded the associated request previously). So aren't we supposed to immediately statelessly forward the CANCEL if t_relay_cancel() did not find the INVITE transaction, instead of doing the same as INVITEs, which could be t_relay() (not stateless)?. Like below: if (method == CANCEL) { if (!t_relay_cancel()) { # implicit drop if relaying was successful, # nothing to do # corresponding INVITE transaction found but error occurred sl_reply("500", "Internal Server Error"); drop; } # bad luck, corresponding INVITE transaction is missing, forward(); } Am I correct or am i missing something? Thank you in advance, Bill

4 7

service uri not found in rls-services document
by Juha Heinanen 03 May '13

03 May '13

my presence server tries to handle subscribe to sip:jh+presence@test.tutpro.com. that service uri is stored in rls-services document of user sip:jh@test.tutpro.com, but for some reason k does not find the service uri. here is some debug: May 1 19:08:43 siika /usr/sbin/pres-serv[27355]: INFO: Handling SUBSCRIBE <sip:jh+presence@test.tutpro.com> May 1 19:08:43 siika /usr/sbin/pres-serv[27355]: INFO: rls [subscribe.c:155]: searching document for user sip:jh@test.tutpro.com May 1 19:08:43 siika /usr/sbin/pres-serv[27355]: INFO: rls [subscribe.c:240]: rls_services document:#012<?xml version='1.0' encoding='UTF-8'?>#012<rls-services xmlns="urn:ietf:params:xml:ns:rls-services"><service uri="sip%3Ajh%2Bpresence%40test.tutpro.com"><resource-list>http%3A//192.98.102.10%3A8080/xcap-root/resource-lists/users/sip%3Ajh%40test.tutpro.com/index/%7E%7E/resource-lists/list%5B%40name%3D%22sipsimple_presence_rls%22%5D</resource-list><packages><package>presence</package></packages></service><service uri="sip%3Ajh%2Bdialog%40test.tutpro.com"><resource-list>http%3A//192.98.102.10%3A8080/xcap-root/resource-lists/users/sip%3Ajh%40test.tutpro.com/index/%7E%7E/resource-lists/list%5B%40name%3D%22sipsimple_dialog_rls%22%5D</resource-list><packages><package>dialog</package></packages></service></rls-services> May 1 19:08:43 siika /usr/sbin/pres-serv[27355]: INFO: rls [subscribe.c:253]: service uri sip:jh+presence@test.tutpro.com not found in rl document for user sip:jh@test.tutpro.com any idea why the service uri is not found even when it appears to be in the document: <service uri="sip%3Ajh%2Bpresence%40test.tutpro.com"><resource-list>http%3A//192.98.102.10%3A8080/xcap-root/resource-lists/users/sip%3Ajh%40test.tutpro.com/index/%7E%7E/resource-lists/list%5B%40name%3D%22sipsimple_presence_rls%22%5D</resource-list><packages><package>presence</package></packages></service> could the escaped chars explain the failure or the fact that resource-list is an uri? -- juha

2 8

extra_id_pv parameter
by Bruno Bresciani 03 May '13

03 May '13

Hi All, I need forking a call to multiple destinations in paralel on different network segments, requiring different rtpproxy parameters. Reading the rtpproxy module documentation I discovered that is possible by setting the "b" parameter on rtpproxy_manage(), rtpproxy_offer(), rtpproxy_answer() and rtpproxy_destroy() functions, but I'm with some doubts... I can't understanding how set the extra_id_pv parameter correctly to use when the "b" parameter is used. The "b" parameter is supported on rtpproxy version 1.2.1? Best Regards

2 3

How to decrypt SIP message
by Khoa Pham 03 May '13

03 May '13

Hi, Currently, I'm using TLS and it works fine. But eventually, TLS is just used to agree upon a secret key. And both client and Kamailio use that secret key to encrypt message. In my solution, I have a fixed secret key. And client encrypt SIP message with that secret key. What code should I insert in Kamalio to decrypt it? -- Khoa Pham HCMC University of Science Faculty of Information Technology

2 4

Re: [SR-Users] service uri not found in rls-services document
by Peter Dunkley 02 May '13

02 May '13

Last I looked (around February) there were some things in Kamailio presence that aren't in OpenSIPS. For example, the separate notifier processes in both presence and rls (I believe one may have been ported across now by Saul - but not both). There are also extensions relating to the built-in XCAP server (particularly in presence_xml and rls) that are in Kamailio and not OpenSIPS. I know that there are some specific RLS fixes and features in Kamailio that only work when you use the built-in XCAP server. I also made some enhancements to the way pidf-manipulation is handled in Kamailio that I do not think have been ported to OpenSIPS. Basically, while OpenSIPS presence may be more advanced in some ways I am quite sure it is less developed in some others. Having spent a lot of time over the course of around 18 months doing work in Kamailio presence I would strongly resist any attempts to junk what is currently there because I know that I need a lot of the stuff that is in the Kamailio implementation. I don't currently have the time to develop it further at the moment either. I think the "junk them and replace with opensips" approach would be a disaster. Regards, Peter > looks like kamailio presence and especially rls is lagging behind in > development as compared to opensips. it properly handles the escapes > that i had trouble with, support external references in presence > rules, xcal-diff, etc. > > what should we do about kamailio presence modules? junk them and > replace with opensips ones or implement the missing stuff ourselves? > i personally don't have resources for the latter. > > -- juha > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > -- Peter Dunkley Technical Director Crocodile RCS Ltd

1 0

Re: [SR-Users] Rtpproxy - IPv4/v6 Translation and earlier removing of session
by Jakub Hrabovský 02 May '13

02 May '13

Thanks for reply, I checked log messages at callee (IPv6 client - Linphone) and result was little surprising for me. This IPv6 client as Callee didnt receive INVITE messages (I thought that yes because of Wireshark at SIP server). So it is my bad somewhere in my local network. But it is little interesting, that in other direction (from IPv6 caller to IPv4 callee) all Reply messages are received at IPv6 client and RTP/UDP packets go in both directions. I checked Iptables and Firewall at all nodes, but nothing (they all are in the same network) and I dont know what filteres messages to IPv6 callee client. btw. the proxy server in the middle is SIP Protocol gateway, which translates messages between IPv4 and IPv6 and packets at this node are OK. thanks for help Jakub Hrabovsky

1 0

Topoh module also masking Via?
by Grant Bagdasarian 01 May '13

01 May '13

Hello, The description of this module says it hides the SIP routing headers, does this also include the Via headers? Grant

3 5

rtpproxy virtual or physical machine
by Grant Bagdasarian 01 May '13

01 May '13

Hello, I understand that machines processing RTP are recommended to be physical machines. Does the same apply for machines just proxy'ing RTP? Also, after how many RTP session would you start noticing issues with RTP when using rtpproxy on a virtual VMware machine? Regard, Grant

2 2

gruu and dead-lock in registrar module
by Andreas Granig 01 May '13

01 May '13

Hi, We're hitting an issue in a deployment where all udp receivers are sitting in FUTEX_WAIT caused by save() -> lock_udomain() and seem to have deadlocked themselves every couple of days. Looking at the code, enable_gruu in registrar is active by default, and in lookup there is a code path /* temp-gruu lookup */ res = ul.get_urecord_by_ruid(_d, ahash, &inst, &r, &ptr); but no lock_udomain is obtained. However, when the execution falls through to the "done:" marker, it does ul.unlock_udomain(_d, &aor); without having called ul.lock_udomain first. 1.) Could someone please review this part? Looks a bit suspicious, although I don't know what implicitly happens in this case. If it were a semaphore and you decrease it to -1 by decrementing it without prior increment, it's essentially causing a dead-lock, but the current locking implementation might work completely different. 2.) Since I have no clue how gruu is supposed to work in detail, and since in our config we don't explicitly handle gruu (no lookup in loose-route, but gruu is enabled by default in registrar and we don't explicitly turned it off), I'm not even sure if we ever hit this code path. I only see that the ruid column in the location table is filled, but in order to get to this part, the ";gr" flag needs to be set in the R-URI for a lookup(), which I don't know whether that happened somehow in some call flows (we only log $ru, which I don't think logs these parameters, right?). Some input is highly appreciated! Andreas

2 4

Re: [SR-Users] Kamailio 3.x and Asterisk Realtime Integration plus dispatcher
by Aldo Antignano 01 May '13

01 May '13

> Sorry, yeah. you should probably ignore that part :-) The same dispatcher >deals with client connections from the "public" network as well as inbound Ok thanks, it's clear now. :) >> In the REGFWD route you send all the registrations to ALL the Asterisks? >> If so, why? > By sending the registration to all the available Asterisk servers then they > all know where each SIP client can be located. If a call comes in from the > PSTN or from a SIP client Kamailio can just be sent it to any of the * > servers and it will find the registered client. This will work even with > multiple load balancers/dispatchers using DNS SRV. In this way should be an option to keep registration in Kamailio only, let Asterisk do its specific job, and configure Asterisk to accept all calls coming from Kamailio as authorized calls by its IP? > In this scenario Kamailio is mainly acting as a load balancer, topology > hider and security/anti-flood front-end to create a scalable and redundant > Asterisk PBX system. Really depends what you are after. These are exactly my goals and was trying to understand which is the best way to achieve this. Any suggestion is welcome, thanks. Anty ------------------------------ Message: 3 Date: Tue, 30 Apr 2013 23:02:20 +0100 From: Barry Flanagan <barry(a)flanagan.ie> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] Kamailio 3.x and Asterisk Realtime Integration plus Message-ID: <CALJb54_1FGmGV4HfmR1zHgcAGqMsSPu6kriTv72W3adyHF3dQw(a)mail.gmail.com> Content-Type: text/plain; charset="utf-8" > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >

2 1

info: monthly mailing list reminder disabled
by Daniel-Constantin Mierla 01 May '13

01 May '13

Hello, I disabled the monthly mailing list membership reminder, being rather annoying when subscribed to lots of mailing lists. The option can be enabled on per subscriber basis, if anyone wants it, then it can be set in personal options at: - http://lists.sip-router.org/cgi-bin/mailman/options/sr-users/ 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 *

1 0

dlg_end_dlg
by Dragos Oancea 01 May '13

01 May '13

Hi everyone, I believe I've run into a problem when using dlg_end_dlg (with mi_datagram ) called from an external perl script with perl_exec() . Kamailio just locks after a while (few seconds) after disconnecting a call. Some parts of my debug log can be found here: http://pastebin.com/ANaTEifC The problem was observed while performing a stress test. Is it a good idea to call dlg_end_dlg() like that or I can kill an established dialog from the routing script in some other way ? Regards, Dragos

2 5

Re: [SR-Users] Kamailio 3.x and Asterisk Realtime Integration plus
by Aldo Antignano 30 Apr '13

30 Apr '13

Thank you very much for your reply Barry. Can you explain us yours ip 10.5.75. and 10.5.76. ? Seems your asterisks have two interfaces, one private and one public? In my infrastructure I have: SIP client -> Kamailio ---> Asterisk 01 ---> Asterisk 02 ... ---> Asterisk N Only Kamailio has a public IP address. In the REGFWD route you send all the registrations to ALL the Asterisks? If so, why? I'm trying to achieve something like: SIP clientA ---> REG ---> Kamailio Is Asterisk01 alive? Yes --> REG ---> Asterisk 01 NO --> Choose next one ---> Asterisk 02 ... ---> Asterisk N ---> Asterisk 01 SIP clientB ---> REG ---> Kamailio Is Asterisl02 alive? Yes --> REG ---> Asterisk 02 NO --> Choose next one ---> Asterisk 03 ... ---> Asterisk N and so on, with a round robin algorithm. When Sip ClientA want to call Sip ClientB, it asks Kamailio, that know that SIP Client B is registered on Asterisk 02 and so send the invite to it. I'm going to work over this in the next few days and let you know the result, if positive I'll write a new complete guide on this topic with your valuable suggestions. Any suggestion is welcome BR, Anty ------------------------------------------------------------------------------------------------------ >> I have read and applied the excellent guide found on: >> http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb >> >> Now I have added to Kamailio the HA/Load Balancer support, with the >> "dispatcher" module. >> This way I have 1 Kamailio and 2 Asterisk machines. >> >> How can I change the routing logic of the sections route[REGFWD] | >> route[FROMASTERISK] route[TOASTERISK] to use the dispatcher module? (in the >> guide above the asterisk binded ip address is cabled in the kamailio config >> code) I have done that. Relevant route entries below. # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; } # for each Asterisk server in setid=2 (External) we send a registration on behalf of the user. sql_xquery("ca", "SELECT SUBSTRING_INDEX(destination,':',-1) AS port, SUBSTRING_INDEX(SUBSTRING(destination,5),':',1) AS address FROM dispatcher WHERE setid = 2", "ra"); $var(i) = 0; while($xavp(ra[$var(i)]) != $null) { $var(rip) = $xavp(ra[$var(i)]=>address); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $xavp(ra[$var(i)]=>port); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); $var(i) = $var(i) + 1; } sql_result_free("ra"); } # Test if coming from Asterisk. We check the dispatcher "ds_is_from_list()" function to see if this is one of our Asterisk IPs route[FROMASTERISK] { if(ds_is_from_list()) { return 1; } else { return -1; } } # Send to Asterisk route[TOASTERISK] { # If call comes in to the .75 iface, we need to send it to the .75 iface of Asterisk as well. # otherwise we send to the .76 iface. We do this by calling different dispatcher sets. This is # because Asterisk needs to use NAT on the .76. (public) interface but not on the .75. if($td=~"10.5.75.") { $var(setid) = 4; xlog("SCRIPT: Call to 10.5.75. ip - using set $var(setid) \n"); } else { $var(setid) = 2; xlog("SCRIPT: Call from $fn to 10.5.76. ip - using set $var(setid) \n"); } # round robin dispatching on set determined above if(!ds_select_dst($var(setid), "4")) { send_reply("404", "No destination"); exit; } t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } Hope this helps. -Barry Flanagan

2 1

Kamailio 3.x and Asterisk Realtime Integration plus Dispatcher module
by Aldo Antignano 30 Apr '13

30 Apr '13

I have read and applied the excellent guide found on: http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb Now I have added to Kamailio the HA/Load Balancer support, with the "dispatcher" module. This way I have 1 Kamailio and 2 Asterisk machines. How can I change the routing logic of the sections route[REGFWD] | route[FROMASTERISK] route[TOASTERISK] to use the dispatcher module? (in the guide above the asterisk binded ip address is cabled in the kamailio config code) Thanks, Anty

2 1

[SR-users] Radius AAA
by Victor V. Kustov 30 Apr '13

30 Apr '13

Hi again! Need tips to some tricks with radius. 1. when receive sip message with REGISTER method - initiate auth with radius. (Authentification) - radius_www_authorize in auth_radius.so 2. when receive INVITE message - ask radius 'bout permissions and parse answer: get permission and next hop IP. (Authorization) ???? misc_radius ???? 3. when call a) before call send Acct-Status-Type = Start with attributes. b) after call send Acct-Status-Type = Stop with attributes. (Accounting) - acc_radius.so 1,3 - clear, I understand how it works. 2 - ? Kamailio must send request to radius server, get answer and route call to server from "next-hop-IP" field. Can I make it? -- С уважением, Виктор В. Кустов ООО "Брянск Связь-ТВ" +7 (952) 966 76 24 JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.8.4-calculate GNU/Linux

2 2

siptrace + flatstore
by Victor V. Kustov 30 Apr '13

30 Apr '13

Hi all loadmodule "siptrace.so" modparam("siptrace", "db_url", "flatstore:/var/log/siptrace") call sip_trace(); nothing in /var/log/siptrace, no errors in logs. whats wrong? -- SY, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.8.4-calculate GNU/Linux

2 2

var vs avp...
by Victor V. Kustov 30 Apr '13

30 Apr '13

Hi! whats difference $var() and $avp()? -- SY, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.8.4-calculate GNU/Linux

7 24

riud access from route block
by Juha Heinanen 30 Apr '13

30 Apr '13

when i call t_relay() on a tcp contact, branch route is executed. in that branch route i call t_on_branch_failure. now if the tcp connection to the contact is broken, for example, due to uas crash, the branch failure route is never executed and thus i cannot get access to $T_reply_ruid. there is $branch(name) pv that gives access to other branched, but not the "main branch". is there any way to get access to ruid of the main branch when t_relay() fails when called from route block? for example, if there is only one contact (the $ru one), which results in t_relay() failure, is there a way to find out the ruid of this contact? -- juha

1 1

kamailio 3.1.4
by Ryan d'Eon 30 Apr '13

30 Apr '13

Hello, I'm trying to support a kamailio install, and they are having a crashing issue. I'm not looking for detailed help on the problem, I just noticed one thing I'd like to check up on here: kamailio 3.1.4 (the version they are running) isn't available for download anymore, though quite a few other old versions are. Are there known bugs? Meaning, should I be worried about this, and go back with the recommendation to upgrade? If so, which is the ideal upgrade path? 3.1.5? 3.1.6? Thanks for any help. Ryan

2 1

!!! about private emails
by Daniel-Constantin Mierla 29 Apr '13

29 Apr '13

Hello, I would like to remind that, at least in my case, writing emails directly without cc-ing the mailing list is not recommended. The rule is also suggested in the mailing lists presentation page: - http://www.kamailio.org/w/mailing-lists/ Unless I asked explicitly for a private email with some specific details, the message will not be noticed and therefore not replied. I use this email address for many public mailing lists and it is intended only for that usage, I am reacting based on the filters I created, unmatched messages getting the lowest priority. There are several common sense as well as technical reasons. Here are some along with other suggestions to improve the likeliness of getting an answer: - we are not answering questions on mailing lists to help only one person, but also other people that may have similar issues in the future -- they can find the answers in the archive with a search engine - the amount of unexpected messages is very high -- it is practically a zero chance to get to the folder with messages that didn't match any filter (thus your message is not going to be replied) - whenever I find time, the first emails I answer are those coming on mailing lists. There are many there as well, people are traveling or having other personal or business projects, so sometime is good to send a reminder if a question does not get an answer - starting with negative approach or no technical content, like you are going to use something else or this application is missing what so ever tutorial giving the solution you need or simply asking for full config of complex requirements, is not helping, making people ignore your messages. Start with what you tried and where you got stuck. Everything you get is for free, nobody here owe you anything. Therefore, if anyone sent me a private message asking about kamailio or continuing a discussion started on mailing lists, then the messages have to be resent on mailing list if one wants an answer. A quick look in my unfiltered folder revealed hundreds of messages during the past year, soon many will be deleted anyhow. Note that you can always mask sensitive data (e.g., replace passwords, IP addresses) before sending logs or config file. 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 24016, 2013 * http://asipto.com/u/katu *

2 1

Rewrite RURI
by Grant Bagdasarian 29 Apr '13

29 Apr '13

Hello, Is it possible to have Kamailio rewrite the Request URI of the INVITE message but sent the INVITE to another address first? For example - Kamailio (10.0.0.1) receives an INVITE - Kamailio rewrites the RURI to 10.0.0.3 - Kamailio sends the INVITE to 10.0.0.2 - The application at 10.0.0.2 sends the INVITE to 10.0.0.3 Regards, Grant

5 5

Re: [SR-Users] Fwd:
by Charles Chance 29 Apr '13

29 Apr '13

Hi, Sincere apologies, not sure how this happened - please ignore, it was not me who sent it :/ Regards, Charles On 2 April 2013 08:21, Charles Chance <charles.chance(a)sipcentric.com> wrote: > http://www.toyosoken.co.jp/ivtduw.php > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > -- *Charles Chance* Managing Director t. 0121 285 4400 m. 07932 063 891 -- www.sipcentric.com Follow us on twitter @sipcentric <http://twitter.com/sipcentric> Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham B7 4EJ.

1 0

Fwd:
by Charles Chance 29 Apr '13

29 Apr '13

http://www.toyosoken.co.jp/ivtduw.php

1 0

siptrace source ip causes martians
by Thilo Bangert 29 Apr '13

29 Apr '13

Hi, i have a kamailio instance, which is connected to two subnets (A and B) and exchanges sip messages between them. I have setup the siptrace module to send hep tracing messages of all sip messages to a host in subnet B. This works really well for sip messages arriving on the IP of subnet B - for those arriving on subnet A however it doesnt work. The hep messages have the interface address of subnet A set as source address. Since the machine receiving the HEP messages also is connected to both subnet A and B, these messages are (correctly) rejected as martians and never reach the sipcapture module on that host. this is on kamailio 3.3.4. Is there a way to force the source IP of outgoing HEP messages in the siptrace module? thanks kind regards Thilo

2 4

kamaillio as SBC
by Vu Ngo thanh 29 Apr '13

29 Apr '13

I am interested in testing Kamaillio acting as an SBC. The SBC as Outbound proxy server, the IPPhone can register to SIP Server via SBC and make a call via SBC. the topology is bellow The IP Phone A, The IP Phone B --------- register ------- FreeSwitch (SBC-Outbound Proxy) -------- SIP Server Could you please help me how to install and configure FreeSwitch as SBC-Outbound Proxy ? Thanks so much

3 2

Montoring/Reporting Solution - Commercial/Open Source
by Ali Jawad 26 Apr '13

26 Apr '13

Hi I am in need of a solution that does provide me with advanced graphing and report for Kamailio, I.e. ASR/ACD over day/month/destination etc. with tresholds and alerts. Does anyone know about a solution that is compatible with Kamailio and does this ? Thanks

4 5

Issue with app_perl - return value, logging and append_branch
by Mark Boyce 26 Apr '13

26 Apr '13

Hi All I may well be me missing something obvious but I have got three odd ones whilst trying to test app_perl module which I'd appreciate some assistance with. Setup -------- Kamailio version : 4.0.0 (x86_64/linux) from Peter Dunkley's YUM Repository Perl version : v5.10.1 OS version : CentOS release 6.4 Kamailio.cfg highlights; loadmodule "app_perl.so" modparam("app_perl", "filename", "/tmp/test.pl") modparam("app_perl", "modpath", "/usr/lib64/kamailio/perl/") # ** would someone like to add the app_ to 5.1 and 5.2 of the manuals please request_route { . . route(PERL); . . } route[PERL] { if ($rU=~"^124" && method=="INVITE") { xlog("L_INFO","perl: match $rm $ru from:$fu ($si:$sp) to:$tu rpid:$re\n"); if (perl_exec("testrouting")) { xlog("L_INFO","perl: true exit\n"); } else { xlog("L_INFO","perl: false exit\n"); } } } /tmp/test.pl has the following in it ... #!/usr/bin/perl use Kamailio qw ( log ); use Kamailio::Constants; use Sys::Syslog qw( :DEFAULT setlogsock); sub testrouting { setlogsock('unix'); openlog($0,'','user'); syslog('info', "PERL Kamailio - Log test"); closelog; return 1; } There is no valid route for 124@ and the above works exactly as I'd expect; -> Phone sends INVITE to 124 <- Proxy responds with AUTH Required -> Phone sends ACK -> Phone sends INVITE to 124 with AUTH -- SysLog gets a "PERL Kamailio - Log test" from perl script -- Kamailio log gets "perl: true exit" <- Proxy responds with "404 Not Found" -> Phone sends ACK ------------------------------------------ First problem is that if I swap the "return 1" in test.pl for "return 0" what I see is; -> Phone sends INVITE to 124 <- Proxy responds with AUTH Required -> Phone sends ACK -> Phone sends INVITE to 124 with AUTH -- SysLog gets a "PERL Kamailio - Log test" from perl script -> Phone resends INVITE to 124 with AUTH -- SysLog gets a "PERL Kamailio - Log test" from perl script -> Phone resends INVITE to 124 with AUTH -- SysLog gets a "PERL Kamailio - Log test" from perl script -- I Hang Up Phone --> Phone send CANCEL --> Phone send CANCEL --> Phone send CANCEL In this second test once perl_exec is called all routing seems to stop. There's nothing from the xlog to say if perl_exec returned true or false. -------------------------------------- The second problem I have is when I add "Kamailio::log(L_INFO," line it never arrives in any of the logs use Kamailio qw ( log ); use Kamailio::Constants; use Sys::Syslog qw( :DEFAULT setlogsock); sub testrouting { setlogsock('unix'); openlog($0,'','user'); syslog('info', "PERL Kamailio - Log test"); closelog; Kamailio::log(L_INFO, "PERL : testrouting\n"); return 1; } -------------------------------------- The third one is that I cannot get append_branch to do anything. If I add to the perl script the lines; my $m = shift; $m->rewrite_ruri("sip:anotheruser\@ourserver"); The call gets diverted as expected. However if I use append_branch; $m->append_branch("sip:anotheruser\@ourserver"); It has no effect at all. -------------------------------------- That's as far as I have experimented with the module so I can't say if everything else is working as expected :-) Anyone have any thoughts? Cheers Mark

2 1

Problem with PUA
by andpe 26 Apr '13

26 Apr '13

Hi I am new to Kamailio. I have a configuration with existing phones. I call between them. However, I have a problem with the configuration of PRESENCE. I guess I'm doing something wrong. Can someone send me a sample setup working. I looked for a long time information, but I could not fix my problem. In the logs I get the information: 6 (19259) ERROR: pua [send_publish.c: 578]: New PUBLISH and no body found-invalid request 6 (19259) ERROR: pua_usrloc [ul_publish.c: 326]: while sending publish [cycle # 12, if constant make sure server lives] Version: Server :: kamailio (4.0.0 (i386/linux)) Build :: mi_core.c compiled on Apr 4 2013 11:20:52 with gcc 4.4.5 Flags :: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES GIT :: 1d009c regards Andy

2 1

RPM packages for v4.0.1
by Daniel-Constantin Mierla 26 Apr '13

26 Apr '13

Hello, rpm packages for v4.0.1 are no available on opensuse build service - more details and download links at: - http://www.kamailio.org/wiki/packages/rpms Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

misc_radius - too many extras
by Victor V. Kustov 26 Apr '13

26 Apr '13

Hi all I've got same error: misc_radius [extra.c:80]: too many extras -> please increase the internal buffer How I get increase buffer in config? -- SY, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.8.4-calculate GNU/Linux

2 1

RLS Module - Getting NOTIFY for offline users
by Attila Megyeri 25 Apr '13

25 Apr '13

Dear Users, We use Kamailio as our SIP proxy and Presence server, leveraging XCAP and RLS functionality. After issuing SUBSCRIBE rls@<domain> I would like to get presence information for the contacts in the RLS list not only when they are online, but also if they are offline. Is this possible? Currently we are getting NOTIFY only for those contacts that are online (presence open) Thank you in advance, Attila

2 2

Kamailio v4.0.1 Released
by Daniel-Constantin Mierla 25 Apr '13

25 Apr '13

Hello, Kamailio SIP Server v4.0.1 stable release is out. This is a maintenance release of the latest stable branch, 4.0, that includes fixes since release of v4.0.0. There is no change to database schema or configuration language structure that you have to do on installations of v4.0.0. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.0.1. For more details about version 4.0.1 (including links and hints to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2013/04/kamailio-v4-0-1-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Re: [SR-Users] Rtpproxy - IPv4/v6 Translation and earlier removing of session
by Jakub Hrabovský 25 Apr '13

25 Apr '13

Hi, 1.) Here INVITE message sent to IPv6 SIP Client, but he doesnt respond: INVITE sip:jano@[2001:4118:300:122:bca7:31ac:ccab:d6f6];line=e7b07b8cb7a33ee SIP/2.0 Record-Route: <sip:[2001:4118:300:122:444A:3FFF:FECC:61A0];lr=on> Record-Route: <sip:[2001:4118:300:122:A00:27FF:FE60:83C];r2=on;lr=on;nat=v46> Record-Route: <sip:158.193.139.207;r2=on;lr=on;nat=v46> Record-Route: <sip:158.193.139.46;lr=on;nat=yes> Via: SIP/2.0/UDP [2001:4118:300:122:444A:3FFF:FECC:61A0];branch=z9hG4bK1c7c.bf9123f4.0 Via: SIP/2.0/UDP [2001:4118:300:122:A00:27FF:FE60:83C];branch=z9hG4bK1c7c.76b1542.0 Via: SIP/2.0/UDP 158.193.139.46;rport=5060;branch=z9hG4bK1c7c.af9123f4.0 Via: SIP/2.0/UDP 10.0.2.15:5060;received=158.193.139.205;rport=55754;branch=z9hG4bK1417756364 From: <sip:jakub@ip02.sip.uniza.sk>;tag=632445346 To: <sip:jano@ip02.sip.uniza.sk> Call-ID: 267806174 CSeq: 21 INVITE Contact: <sip:jakub@158.193.139.205:55754> Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 67 User-Agent: Linphone/3.3.2 (eXosip2/3.3.0) Subject: Phone call Content-Length: 470 P-hint: outbound v=0 o=jakub 123456 654321 IN IP6 2001:4118:300:122:a00:27ff:fe60:83c s=A conversation c=IN IP6 2001:4118:300:122:a00:27ff:fe60:83c t=0 0 m=audio 51652 RTP/AVP 112 111 110 3 0 8 101 a=rtpmap:112 speex/32000/1 a=fmtp:112 vbr=on a=rtpmap:111 speex/16000/1 a=fmtp:111 vbr=on a=rtpmap:110 speex/8000/1 a=fmtp:110 vbr=on a=rtpmap:3 GSM/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-11 a=nortpproxy:yes 2.) I get here cap files of SIPv64 server and Bridge as Enclosure. Here i get my hosts file for wireshark for clear view of SIP entities: 158.193.139.46 SIPserver_IP4 158.193.139.207 Bridge_IP4 2001:4118:300:122:444a:3fff:fecc:61a0 SIPserver_IP6 2001:4118:300:122:a00:27ff:fe60:83c Bridge_IP6 158.193.139.205 WinPC_IP4 2001:4118:300:122:bca7:31ac:ccab:d6f6 IP6_Client 10.0.2.15 IP4_Klient

2 1

Kamailio dialog termination
by Grant Bagdasarian 25 Apr '13

25 Apr '13

Hello, Using the dialog module. According to the documentation a dialog is automatically destroyed when a BYE is received. In case of no BYE, the dialog lifetime is controlled via the default timeout. I'm assuming the dialog module only creates a dialog when a 200 OK is received on the INVITE. So if a CANCEL request is sent for this INVITE or the server answers with anything but a 200 OK, no dialog is created and maintained by the dialog module? Also, when the dialog is terminated, is it guaranteed the dialog entry is removed from memory at all times, but there might be a chance the dialog remains stored in the database table because of some error? I'm using the dialog module to control the number of calls allowed to a certain destination, but it sometimes does occur that an entry in the database is not removed, but I'm not sure if the dialog is also removed from memory when this happens. I hope someone could clear this out for me. Thanks, Grant

3 5

★ SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List, Phillip left a message for you
by Badoo 25 Apr '13

25 Apr '13

You can instantly reply using our message exchange system. Check your message http://eu1.badoo.com/0138644786/in/y9a..eJmaMo/?lang_id=3&g=57&m=61&mid=517… Some other people in the area: If clicking the links in this message does not work, copy and paste them into the address bar of your browser. You have received this email from Badoo Trading Limited (postal address below). http://eu1.badoo.com/impersonation.phtml?lang_id=3&email=sr-users%40lists.s… Badoo Trading Limited is a limited company registered in England and Wales under CRN 7540255 with its registered office at Media Village, 131 - 151 Great Titchfield Street, London, W1W 5BB.

1 0

Planing to release v4.0.1
by Daniel-Constantin Mierla 25 Apr '13

25 Apr '13

Hello, I am considering releasing v4.0.1 next week on Thursday. There are some issues under discussions on mailing list, they will be sorted by that time, but if you are aware of others, please report them on dev mailing list or tracker. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 1

www.kamailio.org rebooted for maintenance
by Henning Westerholt 24 Apr '13

24 Apr '13

Hello, I've needed to reboot www.kamailio.org two times for maintenance (kernel and grub boot loader update) a few minutes ago. The machine which its services is now up again, please let me know if you find some issues because of the change. Thanks and regards, Henning -- Henning Westerholt - Head of IT Operations Internet Access & Communications Tel. +49 721 91374-4483, Fax. +49 721 91374-2734 1&1 Internet AG, Brauerstraße 48, 76135 Karlsruhe, Germany Vorstände: Henning Ahlert, Ralph Dommermuth, Robert Hoffmann, Andreas Hofmann, Markus Huhn, Hans-Henning Kettler, Dr. Oliver Mauss, Jan Oetjen, Martin Witt, Christian Würst. Aufsichtsratsvorsitzender: Michael Scheeren. Amtsgericht Montabaur / HRB 6484

1 0

BYE dialog timeout bad syntax
by Julia 24 Apr '13

24 Apr '13

Hello, The BYE, generated by Kamailio in case of dialog timeout, is malformed: BYE <sip:anonymous@192.168.225.2:50601> sip:anonymous@192.168.225.2:50601 SIP/2.0 Via: SIP/2.0/UDP 172.16.60.113;branch=z9hG4bK77b6.941684b6.0 Via: SIP/2.0/UDP 10.1.1.10;branch=z9hG4bKsr-JfymiMenCtp4urS5CXSnCHJnCXJZiHvRCHYJb6ggNPN9NVDL3 -TWhdW0UvOfN1KPTReRiHy1CHmcs6mZiHS* To: <sip:anonymous;cpc=ordinary@192.168.225.2;tag=01e3a8c0-c1c2-t-1> sip:anonymous;cpc=ordinary@192.168.225.2;tag=01e3a8c0-c1c2-t-1 From: <sip:1255010006@192.168.225.2:5060;user=phone;tag=813142547-1366730270587> sip:1255010006@192.168.225.2:5060;user=phone;tag=813142547-1366730270587 CSeq: 465332801 BYE Call-ID: <mailto:1aebb42c-01e3a8c0-000007c2@192.168.225.2> 1aebb42c-01e3a8c0-000007c2(a)192.168.225.2 Content-Length: 0 User-Agent: kamailio (3.3.2 (x86_64/linux)) Max-Forwards: 69 The bad syntax is located in the "To" header field. According to the RFC 3261 section 25.1, the grammar of the To header field is as follow: To = ( "To" / "t" ) HCOLON ( name-addr / addr-spec ) *( SEMI to-param ) name-addr = [ display-name ] LAQUOT addr-spec RAQUOT addr-spec = SIP-URI / SIPS-URI / absoluteURI to-param = tag-param / generic-param The RFC 3261 section 20.10 explains that: Even if the "display-name" is empty, the "name-addr" form MUST be used if the "addr-spec" contains a comma, semicolon, or question mark. There may or may not be LWS between the display-name and the "<". As the "addr-spec" <sip:anonymous;cpc=ordinary@192.168.225.2> sip:anonymous;cpc=ordinary@192.168.225.2 contains a semicolon, it means that the "name-addr" form must be used. The "To" header field should be encoded as follow: To: < <sip:anonymous;cpc=ordinary@192.168.225.2> sip:anonymous;cpc=ordinary@192.168.225.2>;tag=01e3a8c0-c1c2-t-1 BR, Julia

2 1

Fwd: save accdb records with milisec in timestamp field
by Uri Shacked 24 Apr '13

24 Apr '13

Hi, can anyone tell me if it is possible to save the time and date using accdb (mysql) with milisec as well? the field on the DB is set as TIMESTAMP. BR, Uri

3 5

Re: [SR-Users] Kamailio 4.0 IMS
by Jason Penton 23 Apr '13

23 Apr '13

Thanks Carsten. We have also started some other docs with Richard and Olle. Olle have you managed to publish the doc Rich and I shared with you - at least this is a nice overview of the modules, etc? I know there are still some module docs outstanding from me which I will get to ASAP - apologies for that! Cheers Jason On Tue, Apr 23, 2013 at 11:40 AM, Pavel Segeč <Pavel.Segec(a)fri.uniza.sk>wrote: > Thank you Carsten very much**** > > ** ** > > Palo73**** > > ** ** > > *From:* sr-users-bounces(a)lists.sip-router.org [mailto: > sr-users-bounces(a)lists.sip-router.org] *On Behalf Of *Carsten Bock > *Sent:* Tuesday, April 23, 2013 10:08 AM > *To:* Kamailio (SER) - Users Mailing List > *Cc:* andreea.ancuta.corici(a)fokus.fraunhofer.de; Dragos Vingarzan > *Subject:* Re: [SR-Users] Kamailio 4.0 IMS**** > > ** ** > > Hi,**** > > i was approached by several people after Kamailio World 2013, that our > installation guide is not available anymore.**** > > The "old" version was at some point quite outdated, that is the reason why > i disabled the page. However, due to the great demand i will update the IMS > on Kamailio how-to later this week and publish the updated version (and i > will move it to the Kamailio Wiki so other points can be easily added by > different people).**** > > IMS rocks!**** > > ** ** > > Kind regards, > Carsten**** > > ** ** > > ** ** > > 2013/4/23 Pavel Segeč <pavel.segec(a)fri.uniza.sk>**** > > Sir, > > Thank you for your answer. Aggree all you wrote and I'm aware of that. > Just I'm saying helps us..we use, popularize and love Kamailio project > however we need some help ...that's all... Kamilio 4.0 IMS is a quite new > thing and we are starting to use it during official student classes and > labs, we are gaining experiences and we share them,we reports bugs and so > on, we do not hide our knowledge > > palo73**** > > > > > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users**** > > > > > -- **** > > Carsten Bock > CEO (Geschäftsführer) > > ng-voice GmbH > Schomburgstr. 80 > D-22767 Hamburg / Germany > > http://www.ng-voice.com > mailto:carsten@ng-voice.com > > Office +49 40 34927219 > Fax +49 40 34927220 > > Sitz der Gesellschaft: Hamburg > Registergericht: Amtsgericht Hamburg, HRB 120189 > Geschäftsführer: Carsten Bock > Ust-ID: DE279344284 > > Hier finden Sie unsere handelsrechtlichen Pflichtangaben: > http://www.ng-voice.com/imprint/**** > > ** ** > > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 8255 > (20130422) __________**** > > ** ** > > Tuto spravu preveril ESET NOD32 Antivirus.**** > > ** ** > > http://www.eset.sk**** > > > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 8255 > (20130422) __________ > > Tuto spravu preveril ESET NOD32 Antivirus. > > http://www.eset.sk > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- Jason Penton Senior Manager: Applications and Services Smile Communications Johannesburg, South Africa Phone: +27 83 283 7000 This email is subject to the disclaimer of Smile Communications at http://www.smilecoms.com/disclaimer

1 1

Simulating kamailio config flows
by Andreas Granig 23 Apr '13

23 Apr '13

Hi, I'd like to put a topic up for discussion to test kamailio config logics. The standard way of doing so is to start kamailio as usual and write sipp or sipsak scenarios to perform automated tests. This can get quite complex pretty quickly, as you always have to take care of covering the full transaction on the client side, where in most of the cases you really just want to know which IP/port kamailio would send the message to, and how some headers look like after processing. Also, it doesn't really allow to inspect your variables and avps etc. on the fly other than logging them to syslog and parsing it from there. Let me know if this is stupid and/or a complete overkill, but what about introducing some kind of dummy mode, where you'd pipe a message into kamailio via stdin, and get the resulting message out on stdout (e.g. in ngrep style with ip information as first line, plus the content following), plus a dump of internals (e.g. vars, avps) on stderr as they're being assigned, and once the message is processed, kamailio would just shut down again? There are quite some things to consider, like not doing any timer handling in tm, but would this be something anyone would be interesting to discuss in more detail? If I'm completely on a wrong track, what are suggested alternatives to testing full sip call flows other than running automated UAs against kamailio? Andreas

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Re: [SR-Users] sip_trace unable to capture after next_branches()
by Ramaseshi reddy kolli 23 Apr '13

23 Apr '13

Hi, Once in a while it captures outbound SIP Invite but it captures always 100 trying, 180 and 200 OK and some times it captures inbound SIP INVITE as well. Thanks Ram

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