sr-users May 2013

sr-users@lists.kamailio.org

95 participants
156 discussions

by Raj Roy Ghandhi

Hi Friends, I am trying to use the Kamalio 4 and SIPML5 and so for it is working fine. But when I subscribe for presentity of another then I receive all the entries of that user in the presentity table in mysql db. It seem that it I receive the entries that are not expired. Is there a way to get the latest status of the user or just avoid duplicate entries in the presentity table. Please advice. Best Regards, Roy.

11 years, 2 months

kamailio 4.0.1 websocket support

by Rupayan Dutta

Dear All, I have build and installed kamailio 4.0.1 from source(tarball release) in Cent Os 5.8 in i386 machine.I can start kamailio but when i want to connect kamailio using SipMl5 client which uses sip over websocket the client can't connect with the server.I am attaching my generated kamailio.cfg file with this mail.Is there any additional configuration needed for websocket support in kamailio 4.0.1?I have located the example websocket.cfg file in kamailio source but totally confused what modification to be done in the end of the file(e.g Main SIP request routing logic blocks). Note: I moified *modules.lst and include db_mysql auth auth_db sl nathelper tm xhttp tls msrp websocket modules *

11 years, 2 months

Re: [SR-Users] Kamailio + Siremis Outbound route ---- Re Sent Due to bad English

by Tony Turner

(re-sent due to bad English) Hi, I have 3 registered test users, how can I configure Siremis to do the trunk to freeswitch using LCR or Carrierroute rather than using the code below. I am keen to be able to setup Inbound + Outbond trunks via Siremis. Do you know if there is a manual for Siremis or a how to / step by step? if($rU =~"^01") { $ru = "sip:" + $rU + "@__FREESWITCHIP__"; route(RELAY); exit; } Currently with the above code if a user phones one of the other extensions it tries to route out to the PSTN network rather than the extension, is that because I have put the above code in the wrong place in the config so it never gets to the code to route to the extension? (routing to PSTN is fine) Or do I need an if else statement wrap checking if local user, please can you give me some idea of the code ... Thanks Tony From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] Sent: 20 May 2013 16:19 To: tony.turner(a)nodemax.com; Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio + Siremis Outbound route Hello, if you want to send all calls that arrive to kamailio having the prefix 01 to freeswitch: if($rU =~"^01") { $ru = "sip:" + $rU + "@__FREESWITCHIP__"; route(RELAY); exit; } Be sure calls are authenticated at that point and, if needed, the call is not actually coming from freeswitch. Cheers, Daniel On 5/20/13 11:33 AM, Tony Turner wrote: Hi Version Kamailio v4.0 + Siremis installed on Debian Wheezy via apt-get install I want to use Kamailio as a proxy edge register to our network. I have installed Kamailio and freeswitch. I can register on Kamailio but I can't route a call from my sip client from Kamailio to freeswitch and out to PSTN Sip client ---- Kamailio-----freeswitch-------SS7 ISDN SIP Gateway --- Carriers If I register direct on Freeswitch I can route out to PSTN but I don't understand Kamailio routing. Can someone let me how I route say from SIP client registered on Kamailio to prefix 01% which goes out to Freeswitch Many Thanks Tony _______________________________________________ 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 *

11 years, 2 months

Re: [SR-Users] Kamailio + Siremis Outbound route

by Daniel-Constantin Mierla

Hello, if you want to send all calls that arrive to kamailio having the prefix 01 to freeswitch: if($rU =~"^01") { $ru = "sip:" + $rU + "@__FREESWITCHIP__"; route(RELAY); exit; } Be sure calls are authenticated at that point and, if needed, the call is not actually coming from freeswitch. Cheers, Daniel On 5/20/13 11:33 AM, Tony Turner wrote: > > Hi > > Version Kamailio v4.0 + Siremis installed on Debian Wheezy via apt-get > install > > I want to use Kamailio as a proxy edge register to our network. > > I have installed Kamailio and freeswitch. > > I can register on Kamailio but I can't route a call from my sip client > from Kamailio to freeswitch and out to PSTN > > Sip client ---- Kamailio-----freeswitch-------SS7 ISDN SIP Gateway --- > Carriers > > If I register direct on Freeswitch I can route out to PSTN but I don't > understand Kamailio routing. > > Can someone let me how I route say from SIP client registered on > Kamailio to prefix 01% which goes out to Freeswitch > > Many Thanks > > Tony > > > > _______________________________________________ > 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 *

11 years, 2 months

LCR Module - Rule_id

by Eloy Coto Pereiro

Hi, It Is possible get the rule_id after execute next_gw() function? Cheers

11 years, 2 months

Calls remain stuck in state 1

by Giany

Hello, We are using dipatcher to limit the concurrent number of calls, problem is that from time to time calls remain stuck in state 1 and it breaks our concurrent limits..I was not able to make a kamailio log with high debug as it happens randomly. Attached is a tcpdump flow: Conv.| Time | serverA | Provider | | | | RemoteEnd | 112 |938.355 | INVITE SDP (g729 g711U GSM X-NSERTPType-100 te...hone-eventRT) | |(5060) ------------------> (5050) | | 112 |938.357 | 100 Trying| | | | |(5050) ------------------> (5060) | | 113 |938.420 | INVITE SDP (g711U g729 telephone-eventRTPType-...)| | |(5050) ------------------> (5060) | | 113 |938.422 | 100 trying -- your call is important to us | | |(5060) ------------------> (5050) | | 113 |938.422 | INVITE SDP (g711U g729 telephone-eventRTPType-...)| | |(5060) ------------------> (1416) | | 113 |938.908 | INVITE SDP (g711U g729 telephone-eventRTPType-...)| | |(5060) ------------------> (1416) | | 113 |939.908 | INVITE SDP (g711U g729 telephone-eventRTPType-...)| | |(5060) ------------------> (1416) | | 113 |941.906 | INVITE SDP (g711U g729 telephone-eventRTPType-...)| | |(5060) ------------------> (1416) | | 113 |945.907 | INVITE SDP (g711U g729 telephone-eventRTPType-...)| | |(5060) ------------------> (1416) | | 112 |948.358 | CANCEL | | |SIP Request | |(5060) <-------------------------------------- (61016) | 112 |948.359 | CANCEL | | |SIP Request | |(5060) ------------------> (5050) | | 112 |948.359 | 200 canceling | |SIP Status | |(5060) --------------------------------------> (61016) | 112 |948.359 | 487 Request Terminated | |SIP Status | |(5050) ------------------> (5060) | | 112 |948.359 | 200 OK | | |SIP Status | |(5050) ------------------> (5060) | | 112 |948.359 | ACK | | |SIP Request | |(5060) ------------------> (5050) | | 112 |948.360 | 487 Request Terminated | |SIP Status | |(5060) --------------------------------------> (61016) | 113 |948.360 | CANCEL | | |SIP Request | |(5050) ------------------> (5060) | | 113 |948.360 | 200 canceling | |SIP Status | |(5060) ------------------> (5050) | | 112 |948.365 | ACK | | |SIP Request | |(5060) <-------------------------------------- (61016) | dialog:: hash=136:689416016 state:: 1 ref_count:: 3 timestart:: 0 timeout:: 0 callid:: 745eed805cad6b9a3e1727d169cf3461@serverA:5050 from_uri:: sip:fromnumber@serverA:5050 from_tag:: as53f6bee4 caller_contact:: sip:fromnumber@serverA:5050 caller_cseq:: 102 caller_route_set:: caller_bind_addr:: udp:serverA:5060 callee_bind_addr:: to_uri:: sip:internalnr@serverA:5060 to_tag:: callee_contact:: callee_cseq:: callee_route_set:: As you see the remoteEnd does not answer at all to this request (due to network issue most likely) and the provider sends a CANCEL after approx 3 seconds. >From what I see the INVITE that is sent from asterisk towards kamailio remains stuck (938.420). We are using Kamailio 3.1.6. Any idea what could be the reason for this? Thank you.

11 years, 2 months

"SER Getting Started" - what's the copyright and license?

by Olle E. Johansson

http://kamailio.org/docs/ser-getting-started/SER-GettingStarted.pdf I'm looking for the authors of this document to find out what license it is published under. It would be nice to be able to use it and update it instead of starting a new "getting started" from scratch. Please contact me if you get this message! Sorry for cross-posting. /O

11 years, 2 months

Re: [SR-Users] Simulating kamailio config flows

by Daniel-Constantin Mierla

On 4/22/13 5:30 PM, Juha Heinanen wrote: > Andreas Granig writes: > >> 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? > in addition to the message, you would need to be able to tell which > ip addr it is coming from and which ip/port in kamailio it is going to. > also, on the output side, kamailio would need to tell which > proto/ip/port it would use to send the message out. Details about going out can be printed on onsend_route. In this route block, one can execute drop and nothing is sent to the wire. It can be a config started with a special define specified with -A parameter. Enabling debugger module with cfg trace should give valuable information about what has been executed from config. From what Andreas suggesting, printing the value of variables as they are assigned is missing, probably can be added by hooking in the interpreter when doing the assignment operation. Grouping all above under some global/command line parameter can be useful to make it easier to do a dry run. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

11 years, 2 months

add_path() and save()

by אורן אברהם

Dear list. when using registrar module with path support does the following code work: add_path(); save(); or maybe the path is added only for outgouing messages and the save() won't be affected ?

11 years, 2 months

nat_keepalive() and fix_contact() are incompatible

by David

Hello, So I want to setup a Kamailio SIP Proxy( version 4 ) that will do NAT signalling handling. With the help of nat_traversal and path modules. Attached is my kamailio.cfg file. My problem is I want to do keep alive on users who are behind NAT, so I called nat_keepalive();. The script will work fine, but as soon as I enable either add_contact_alias() or fix_contact(), the keep alive doesn't activate. I modified nat_traversal.c to identify the issue and found that it is comparing the contact header from the request and the reply to find the expiration. Trouble is, the request contact header was taken from before I called fix_contact() and the reply header has the contact from after the fix_contact() call. So my traces show that on the return trip the callback is called, I bet the stack trace looks like this : 1. Some method in tm 2. __tm_reply_in() defined at nat_traversal.c:1358 3. get_register_expire called from nat_traversal.c:1377 and defined at nat_traversal.c:878 4. STR_MATCH_STR called from nat_traversal.c:937 and defined at nat_traversal.c:77 STR_MATCH_STR returns false because the headers are different. I added else { LM_ERR("failed to match because request %s is not reply %s", contact->uri.s, r_contact->uri.s ) ; } to the code and see : May 21 12:18:52 kamailio-dev-nat /usr/sbin/kamailio[30163]: ERROR: nat_traversal [nat_traversal.c:946]: failed to match because request sip:username@mytld#015#012#015#012 is not reply sip:username@myip:59080>;expires=60;received="sip:64.18.189.196:5060"#015#012Server: CenseredUA 5.2-rc1#015#012Content-Length: 0#015#012#015 So as you can see the original Contact header has the server's domain name and the response has the IP address that set by fix_contact(). So I think that nat_keepalive() should not be comparing the Contact from the original request but using the contact header from the modified request. Any ideas what to do next ? Thanks, David

11 years, 2 months

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sr-users May 2013