sr-users January 2016

sr-users@lists.kamailio.org

86 participants
112 discussions

kamailio_call_disconnecting issue
by yasir ay 25 Jan '16

25 Jan '16

Hello, Hi team I have a kamailio SIP server for call Routing,my call flow is like VoipDialer--->Kamailio-Routing Server-->Softswitch-->Telco/PSTN Mobile caller----------------------------------------------------------------------->calle Now Am facing an issue for a call from voip to telco after answering the call i tried to disconnect.but call is not disconneting..what will i do in kamailio.cfg pls help me to re-write the script am using tls also Regards AmanIdo

2 1

using myself in equation with from_uri
by Dirk Teurlings - SIGNET B.V. 25 Jan '16

25 Jan '16

Hi, The documentation states the following. https://www.kamailio.org/wiki/cookbooks/4.2.x/core#myself if(uri==myself) { log("the request is for local processing\n"); }; Now this could also be used to do something like this: (also used in some examples) if(from_uri==myself) { log("the request is for local processing\n"); }; Using this the from_uri could be: sip:sip.voipexample.net:5061 myself in this case only has sip.voipexample.net:5060 as an Aias Now the equation wouldn't match, because the port 5061 is non-default. Would it be possible to match the from_uri to myself ignoring the port in from_uri? The only thing I can think of is using a regular expression. somthing like this: from_uri=~"sip:.+@sip\.voipexample\.net(:([0-9]{1,4}|[1-5][0-9]{4}|6[0-4][0-9]{3}|65[0-4][0-9]{2}|655[0-2][0-9]|6553[0-5])|)" But this skips the use of myself entirely. Any insight on this would be greatly appreciated! Cheers, Dirk

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poros@@lists.sip-router.org
by Poros Mitra 25 Jan '16

25 Jan '16

1 0

Fosdem 2016
by Daniel-Constantin Mierla 22 Jan '16

22 Jan '16

Hello, anyone going to Fosdem? If no unexpected weather conditions messing the transport system, I am going to be there. It would be good to know who else we can meet there. If there are enough people interested, we can try organizing a social event (dinner/drinks) on Saturday evening. Last year we did it together with Jitsi team, but they are not coming this year. Perhaps there are other folks around the RTC dev room we can ask. On the other hand, it is harder to find a place if it is a large group, even harder to put all together if there are different groups/coordinators. Also, it is not easy to discuss withing a large group, as typically for a dinner we were seated to tables. Therefore organizing for a smaller group just for Kamailio friends could be a good alternative. Let's see first the number of participant and the level of interest for various options. This being an announcement, was crossposted to few mailing lists, but as most of the people are on sr-users, I think it would be good to reply and follow up on this discussion only there. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

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Kamailio and openLDAP integration.
by Safdar Khan 22 Jan '16

22 Jan '16

Hi guys, I have tried to integrate kamailio and ldap.It seems like both are taking to each other. Here the purpose of ldap is for back-end support for sip user authentication rather than kamailio database.However i am able to log in but could not make call between sip users.I could not figure out what i am missing. Here is the routing block for ldap in my kamailio.cfg > request_route { > route(LDAPAUTH); > > } --- route[LDAPAUTH] > { > if(is_method("REGISTER")) > { > if(is_present_hf("Authorization")) > { > if > (!ldap_search("ldap://sipaccounts/ou=MyUsers,dc=example,dc=com?cn,userPassword?one?(cn=$fU)")) > { > switch ($retcode) > { > case -1: > sl_send_reply("404", "User Not Found"); > exit; > case -2: > sl_send_reply("500", "Internal server error"); > exit; > default: > exit; > } > } > ldap_result("cn/$avp(username)"); > ldap_result("userPassword/$avp(password)"); > route(REGISTRAR); > if (!pv_www_authenticate("$td", "$avp(password)", "4")) > { > www_challenge("$td", "1"); > exit; > } > sl_send_reply("200", "ok"); > exit; > } > else > { > www_challenge("$td", "1"); > exit; > } > } > }

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SIP Trunking using Kamailio IMS
by indranil chowdhury 21 Jan '16

21 Jan '16

Hello, Is SIP trunking possible with the free version of Kamailio or we need to upgrade to business license? Kindly get back to us with the details of using SIP trunking using Kamailio. We have a requirement in our organisation regarding the same which uses IMS and SIP trunking. In this setup we would initially like to test with 2 LTE EPCs connected to two Kamailio IMS with a SIP trunk between them. Thanks and Regards, Indranil

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topoh quirk with 200 OK for CANCEL
by Alex Balashov 21 Jan '16

21 Jan '16

Hi, I'm using topoh and am running into lots of these errors: Jan 19 20:34:52 csrpus1 /usr/local/sbin/kamailio[3641]: ERROR: [R-DEFAULT-STATELESS-REPLY:nmhfrms2ib9Cnm9kS00Qtbtot5PMiQ0Ni1QKS5iKibuotel14Ygc] !> Received stateless reply 200 (CANCEL) from xxx.xxx.xxx.xxx:5060 Jan 19 20:34:53 csrpus1 /usr/local/sbin/kamailio[3639]: ERROR: tm [t_lookup.c:932]: t_reply_matching(): matching transaction found but callids don't match (received: nmhfrms2ib9Cnm9kS00Qtbtot5PMiQ0Ni1QKS5iKibuotel14Ygc stored: 51899BC6-569EE4330005CE1B-7D47C700_bleg) The call topology is: UAC --> Kamailio (+ topoh) --> PSTN GW Topoh is behaving precisely as expected, but it looks like we might have hit a TM case that topoh does not accommodate somehow: 1. A CANCEL comes in from the UAC and Kamailio sends a CANCEL for its branch to the PSTN GW. 2. Kamailio sends a '200 cancelling' for the UAC's branch backward. 3. The PSTN GW replies with a 200 OK for the CANCEL (PSTN GW branch) that contains the topoh-spoofed Call-ID, but for some reason Kamailio does not match this reply, as indicated by the above error. 4. Because the 200 OK reply was stateless and not matched to a CANCEL transaction, this causes Kamailio to retransmit the CANCEL to the PSTN GW, at which point it receives: 481 Call Leg/Transaction Does Not Exist 5. #4 repeats once more. I don't see any real harm from this situation, but it seems like something that needs fixing. The 487 Request Terminated final reply is processed and relayed correctly to the UAC, and the dialog ends appropriately everywhere. FWIW, I thought this might have to do with async processing of the initial INVITE (through mqueue + rtimer), so I turned it off, but it didn't make any difference. My CANCEL handling in the main request route looks like this: if(is_method("CANCEL")) { set_rtpengine_set("1"); rtpengine_delete(); if(!t_relay_cancel()) { sl_send_reply("500", "Internal Server Error"); exit; } xlog("L_ERR", "[R-MAIN:$ci] !> " "Corresponding INVITE transaction for CANCEL " "does not (or, no longer) exists\n"); exit; } So, it just looks like TM is not matching, and therefore not absorbing the 200 OK reply to its CANCEL to the upstream gateway -- when topoh is enabled. Any insights would be appreciated! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 11

RFC: removing (disabling) fork=no
by Daniel-Constantin Mierla 21 Jan '16

21 Jan '16

Hello, wondering if anyone is using fork=no -- some old docs suggest it is suitable for debugging, but actually kamailio doesn't work properly in this mode, leading to more troubles than benefits (e.g., having reports of invalid issues, like tcp not working in this mode). In first phase I would disable setting this value, with a warning if set to no, because most of the configs out there have fork=yes. Removing it could be considered in the future. Note that this fork=no is different than don't daemonize controlled with -D, which will stay being useful for some init.d systems. Comments or other suggestions? Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

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Re: [SR-Users] Fwd: Kamailio and NAT
by Nelson Migliaro 20 Jan '16

20 Jan '16

I am thinking that I cat get the port during the first call and then use that port for the rest of calls. Maybe that first call will fail but after that, all calls will go fine. Example of a Trying showing received port: 52548 ¿Do you know what pseudovariable represent that value? I tested with all variables I could find but I could not find anyone. Thank you --------------------------------------------------------------------------------------------------------------------------------------------------------- 2016/01/19 17:38:38.981987 VENDOR-IP:5060 -> KAMAILIO-IP:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP PUBLIC-IP:52548;branch=z9hG4bK8b5c.b122371ac0ac2f3ef3204b0f192cb16c.1;rport=52548 Via: SIP/2.0/UDP ASTERISK-IP:5060;received=ASTERISK-IP;branch=z9hG4bK4082d124;rport=5060 From: 888888888 <sip:888888888@ASTERISK-IP>;tag=as34f971fe To: <sip:999999999@VENDOR-DOMAIN> Call-ID: 0b5307fa290674a97b47970643fce42a@ASTERISK-IP:5060 CSeq: 102 INVITE Server: kamailio Content-Length: 0 2016-01-15 9:50 GMT+01:00 Nelson Migliaro <eng.migliaro(a)gmail.com>: > Hello Daniel, > > Yes, I am registered to the vendor. > > Regards, > > Nelson.- > > 2016-01-15 7:58 GMT+01:00 Daniel-Constantin Mierla <miconda(a)gmail.com>: > >> Ahh, I thought Asterisk is in the public internet, but actually you >> connect to a provider (vendor), which seems to run Kamailio as well. >> >> Using information from 100 trying is too late, as the INVITE was already >> sent... so one more question before trying to propose a solution. Do you >> have to register to the provider? >> >> Cheers, >> Daniel >> >> >> On 14/01/16 18:51, Nelson Migliaro wrote: >> >> Yes it is possible, but is there an easy way to workaround the issue >> using Kamailio. >> >> Because I have the port because vendor is sending that info in Trying: >> >> 2016/01/13 20:10:15.842055 VENDOR-IP:5060 -> PRIVATE-IP-KAMAILIO:5060 >> SIP/2.0 100 trying -- your call is important to us >> Via: SIP/2.0/UDP PUBLIC-IP:52548;branch= >> z9hG4bKdd74.992e238037882e809653f713a5a580a9.1;rport=*52548* >> >> I need to find the way to discover the port used by firewall (maybe >> getting that info from Trying) and then advertise that port. >> >> >> >> 2016-01-14 18:32 GMT+01:00 Daniel-Constantin Mierla < <miconda(a)gmail.com> >> miconda(a)gmail.com>: >> >>> Not really up to date with all Asterisk features -- do you know if you >>> can append a custom header to a SIP response that is going to be generated >>> by Asterisk? Eventually the reply for an OPTIONS request. >>> >>> Cheers, >>> Daniel >>> >>> >>> On 14/01/16 17:19, Nelson Migliaro wrote: >>> >>> Yes, I manage all devices, even the internet router but it does not >>> allow static pat. >>> >>> 2016-01-14 16:07 GMT+01:00 Daniel-Constantin Mierla <miconda(a)gmail.com>: >>> >>>> Do you control the Asterisk? If yes, depending on Asterisk capabilities >>>> of building replies, you may be able to do some automation to detect the >>>> external port. >>>> >>>> Cheers, >>>> Daniel >>>> >>>> On Thu, Jan 14, 2016 at 3:47 PM, Nelson Migliaro < >>>> <eng.migliaro(a)gmail.com>eng.migliaro(a)gmail.com> wrote: >>>> >>>>> There is not a public Kamailio, only one Kamailio behind NAT, >>>>> >>>>> Right now the configuration is: >>>>> >>>>> Asterisk <-> Kamailio (Private IP + advertise public IP + RTP Proxy ) >>>>> <-> Internet router (public IP + symmetric na) <-> Internet >>>>> >>>>> Regards, >>>>> >>>>> 2016-01-14 15:43 GMT+01:00 Daniel-Constantin Mierla < >>>>> <miconda(a)gmail.com>miconda(a)gmail.com>: >>>>> >>>>>> Is the kamailio behind nat communicating with another kamailio on a >>>>>> public IP? >>>>>> >>>>>> Cheers, >>>>>> DAniel >>>>>> >>>>>> On Thu, Jan 14, 2016 at 1:33 PM, Nelson Migliaro < >>>>>> <eng.migliaro(a)gmail.com>eng.migliaro(a)gmail.com> wrote: >>>>>> >>>>>>> Thank you Daniel for your answer, >>>>>>> >>>>>>> As you mention, there is a symmetric nat and router does not allow a >>>>>>> static NAT. >>>>>>> >>>>>>> By sniffing traffic I can see the port is using new but in case it >>>>>>> change, how can automate the process of advertising the correct port? >>>>>>> >>>>>>> Cheers! >>>>>>> >>>>>>> >>>>>>> ---------- Forwarded message ---------- >>>>>>> From: Daniel-Constantin Mierla < <miconda(a)gmail.com> >>>>>>> miconda(a)gmail.com> >>>>>>> Date: 2016-01-13 23:28 GMT+01:00 >>>>>>> Subject: Re: [SR-Users] Kamailio and NAT >>>>>>> To: "Kamailio (SER) - Users Mailing List" < >>>>>>> <sr-users(a)lists.sip-router.org>sr-users(a)lists.sip-router.org> >>>>>>> >>>>>>> >>>>>>> Hello, >>>>>>> >>>>>>> it looks like you have a symmetric nat router, so the allocated port >>>>>>> is randomly selected. >>>>>>> >>>>>>> If you don't control the nat router to set a static forwarding rule >>>>>>> or it doesn't provide the option to set static forwarding, then you are >>>>>>> pretty much left with sniffing the traffic to discover the external port >>>>>>> and advertise it. >>>>>>> >>>>>>> Cheers, >>>>>>> Daniel >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> On 13/01/16 20:31, Nelson Migliaro wrote: >>>>>>> >>>>>>> Hello, >>>>>>> >>>>>>> I finally were able to run my Kamailio behind NAT but in order to >>>>>>> accomplish that I included: >>>>>>> >>>>>>> listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548 >>>>>>> >>>>>>> 52548 is the port my internet router change when doing NAT >>>>>>> (5060->52548). I found this port sniffing traffic >>>>>>> >>>>>>> Conclusions at this point are: >>>>>>> >>>>>>> >>>>>>> ---------------------------------------------1-------------------------------------------------------------------------------------------------- >>>>>>> If I use this line: >>>>>>> >>>>>>> listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:5060 it does not work >>>>>>> :( >>>>>>> >>>>>>> When I dial a call, INVITE / ACK / Trying / OK goes fine because >>>>>>> they are part of the same transaction >>>>>>> When remote party disconnects the call, BYE goes to PUBLIC-IP port >>>>>>> 5060 and router blocks de request. I assume vendor sends BYE to 5060 >>>>>>> because it is a new transaction >>>>>>> >>>>>>> -----------------------------------------------2-------------------------------------------------------------------------------------------------- >>>>>>> >>>>>>> If I use this line: >>>>>>> >>>>>>> listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548 it work !!!!!! >>>>>>> >>>>>>> When I dial a call, INVITE / ACK / Trying / OK goes fine because >>>>>>> they are part of the same transaction >>>>>>> When remote party disconnects the call, BYE goes to PUBLIC-IP port >>>>>>> 52548 and router forward the request to Kamailio. Since there is an open >>>>>>> connection. >>>>>>> >>>>>>> I need to find the way to find the way to advertise the public port >>>>>>> internet router is doing NAT (PAT). >>>>>>> >>>>>>> >>>>>>> --------------------------------------------------------------------------------------------------------------------------------------------------- >>>>>>> This trace is a call that worked fine because I included line: >>>>>>> >>>>>>> listen=udp:SOURCE-IP:5060 advertise PUBLIC-IP:52548 >>>>>>> >>>>>>> >>>>>>> This trace is an INVITE with this line: listen=udp:SOURCE-IP:5060 >>>>>>> advertise PUBLIC-IP:52548 >>>>>>> 2016/01/13 20:10:15.793568 PRIVATE-IP-KAMAILIO:5060 -> VENDOR-IP:5060 >>>>>>> INVITE sip:NUM-DESTINATION@VENDOR-IP SIP/2.0 >>>>>>> Record-Route: < >>>>>>> sip:PUBLIC-IP:52548;lr=on;ftag=as3b72a453;vsf=AAAAAAEECQkCAgsNAXBeL0NPXVQfU0suMTY5LjIzMQ--;vst=AAAAAAAAAAAAAAAAAABCUEIAX1lKWF5MF0tB >>>>>>> A-;nat=yes> >>>>>>> Via: SIP/2.0/UDP >>>>>>> PUBLIC-IP:52548;branch=z9hG4bKdd74.992e238037882e809653f713a5a580a9.0 >>>>>>> Via: SIP/2.0/UDP >>>>>>> PRIVATE-IP-SOFTPHONE:5060;received=PRIVATE-IP-SOFTPHONE;branch=z9hG4bK2f4e76ba;rport=5060 >>>>>>> Max-Forwards: 69 >>>>>>> From: NUM-SOURCE <sip:NUM-SOURCE@PRIVATE-IP-KAMAILIO>;tag=as3b72a453 >>>>>>> To: <sip:NUM-DESTINATION@sip.VENDOR-IP> >>>>>>> Contact: >>>>>>> <sip:NUM-SOURCE@PRIVATE-IP-SOFTPHONE:5060;alias=PUBLIC-IP~5060~1> >>>>>>> Call-ID: 329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060 >>>>>>> CSeq: 102 INVITE >>>>>>> User-Agent: Kamailio >>>>>>> Date: Wed, 13 Jan 2016 19:10:15 GMT >>>>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, >>>>>>> INFO, PUBLISH, MESSAGE >>>>>>> Supported: replaces, timer >>>>>>> Content-Type: application/sdp >>>>>>> Content-Length: 255 >>>>>>> >>>>>>> >>>>>>> Trying..... >>>>>>> >>>>>>> 2016/01/13 20:10:15.842055 VENDOR-IP:5060 -> PRIVATE-IP-KAMAILIO:5060 >>>>>>> SIP/2.0 100 trying -- your call is important to us >>>>>>> Via: SIP/2.0/UDP >>>>>>> PUBLIC-IP:52548;branch=z9hG4bKdd74.992e238037882e809653f713a5a580a9.1;rport=52548 >>>>>>> Via: SIP/2.0/UDP >>>>>>> PRIVATE-IP-SOFTPHONE:5060;received=PRIVATE-IP-SOFTPHONE;branch=z9hG4bK2f4e76ba;rport=5060 >>>>>>> From: NUM-SOURCE <sip:NUM-SOURCE@PRIVATE-IP-KAMAILIO>;tag=as3b72a453 >>>>>>> To: <sip:NUM-DESTINATION@VENDOR-IP> >>>>>>> Call-ID: 329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060 >>>>>>> CSeq: 102 INVITE >>>>>>> Server: kamailio >>>>>>> Content-Length: 0 >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> And finally a BYE >>>>>>> >>>>>>> 2016/01/13 20:10:28.545526 VENDOR-IP:5060 -> PRIVATE-IP-KAMAILIO:5060 >>>>>>> BYE sip:34982298000@PRIVATE-IP-SOFTPHONE:5060;alias=PUBLIC-IP~5060~1 >>>>>>> SIP/2.0 >>>>>>> Via: SIP/2.0/UDP >>>>>>> VENDOR-IP;branch=z9hG4bK26d8.847e6e14eef37e2cfc8b5e81d33de73d.0 >>>>>>> From: <sip:675896262@PRIVATE-IP-KAMAILIO>;tag=gK0293ed93 >>>>>>> To: "NUM-SOURCE" <sip:NUM-SOURCE@VENDOR-IP>;tag=as3b72a453 >>>>>>> Call-ID: 329950447629810f7bdeaeed0cc034e1@PRIVATE-IP-SOFTPHONE:5060 >>>>>>> CSeq: 28731 BYE >>>>>>> Max-Forwards: 69 >>>>>>> Route: < >>>>>>> sip:PUBLIC-IP:52548;lr=on;ftag=as3b72a453;vsf=AAAAAAEECQkCAgsNAXBeL0NPXVQfU0suMTY5LjIzMQ--;vst=AAAAAAAAAAAAAAAAAABCUEIAX1lKWF5MF0tBMzA-;na >>>>>>> yes> >>>>>>> Reason: Q.850;cause=16 >>>>>>> Content-Length: 0 >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> ----------------------------------------------------------------------------------------------------------------------------------------------------------------------- >>>>>>> >>>>>>> ----------------------------------------------------------------------------------------------------------------------------------------------------------------------- >>>>>>> >>>>>>> Finally, It is finally working because I hardcoded NAT´d port. >>>>>>> I would like to find a way to avoid setting the port in "hard". >>>>>>> >>>>>>> Thank you >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>>>>>> Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >>>>>>> list >>>>>>> <sr-users(a)lists.sip-router.org>sr-users(a)lists.sip-router.org >>>>>>> <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> >>>>>>> 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>sr-users(a)lists.sip-router.org >>>>>>> <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> >>>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Daniel-Constantin Mierla - <http://www.asipto.com> >>>>>> http://www.asipto.com >>>>>> <http://twitter.com/#%21/miconda>http://twitter.com/#!/miconda - >>>>>> <http://www.linkedin.com/in/miconda>http://www.linkedin.com/in/micond >>>>>> >>>>>> _______________________________________________ >>>>>> 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 >>>>> >>>>> >>>> >>>> >>>> -- >>>> Daniel-Constantin Mierla - <http://www.asipto.com>http://www.asipto.com >>>> http://twitter.com/#!/miconda - http://www.linkedin.com/in/micond >>>> <http://www.linkedin.com/in/miconda> >>>> >>>> _______________________________________________ >>>> 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 listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >>> -- >>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>> Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu >>> >>> >>> _______________________________________________ >>> 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 Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >> Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu >> >> >

2 2

Registration to sip provider
by Evren Akbulut 19 Jan '16

19 Jan '16

Hi, I am a newbie. As I read I understand I need to use UAC module for registration but cannot figure how to trigger the registration request. I enable the UAC module and it seem to be working. How can I trigger the REGISTER to a SIP provider account? And is it possible to use multiple accounts for different provider registrations? Can someone help about UAC usage please. Regards Evren Akbulut

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sr-users January 2016