June 2012 - sr-users - kamailio.org

AVPs - how many values on stack
by Olle E. Johansson 13 Jun '12

13 Jun '12

I got this question today that I can't find the answer on: How many values are there in an avp? If I have pushed X values into the AVP - is there a function that counts them for me? /O

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release of v3.3.0 scheduled for Monday, June 18, 2012
by Daniel-Constantin Mierla 13 Jun '12

13 Jun '12

Hello, as branch 3.3 was created and several web pages updated/created already for v3.3.0, the plan is to make the full release next Monday, on June 18, 2012. Packaging specs look very good thanks to Jon and Peter. The few issues left open should be worked meanwhile. For other opinions, reply with your comments here. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

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About sdpops
by Grégoire Vandendeurpel 13 Jun '12

13 Jun '12

Hello, I'm using Kamailio 3.2.3 and SDPOPS module. I just have a question about the configuration: What do you mean by "ANY_ROUTE" in configuration ?? Cause I don't know where in the configuration , I have to use the differents functions of SDPOPS. If I do something like this: request_route { if ($fU == "103") { sdp_remove_codecs_by_id("0,3,8"); #sdp_remove_media("audio"); } It does not work . So please help me about where I have to put this in?? Thanks in advance -- *Grégoire Vandendeurpel, * * * *IT Sector*

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sipcapture logging
by Konstantin M. 13 Jun '12

13 Jun '12

Hello, Is it possible to log SIP sessions using sipcapture module to log files by log format instead of database ? Thanks!

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Re: [SR-Users] Kamailio Unexpectedly Terminating
by Nathaniel L. Keeling III 12 Jun '12

12 Jun '12

Yes, This is a 64bit system T-Mobile. America’s First Nationwide 4G Network Jason Penton <jason.penton(a)smilecoms.com> wrote: >from the logs he has sent the mem addresses all look 64 bit. Sparc AFAIK is >only 64 bit anyway > >Cheers >Jason > >On Tue, Jun 12, 2012 at 8:50 AM, Daniel-Constantin Mierla <miconda(a)gmail.com >> wrote: > >> Hello, >> >> is it 64bit architecture? >> >> Cheers, >> Daniel >> >> >> On 6/11/12 9:12 PM, Akan wrote: >> >>> Here is the information requested: >>> >>> System = SunOS >>> Node = -f >>> Release = 5.10 >>> KernelID = Generic_141444-09 >>> Machine = sun4u >>> BusType = <unknown> >>> Serial = <unknown> >>> Users = <unknown> >>> OEM# = 0 >>> Origin# = 1 >>> NumCPU = 1 >>> >>> One machine is a Sun Fire V120, 2g memory, UltraSPARC-IIe 650MHz, >>> UltraAX-i2 >>> The other machine is a Sun Netra T1 200, 1g memory, UltraSparc-IIe >>> 500MHz, UltraAX-i2 >>> >>> Thanks >>> >>> Nathaniel L Keeling >>> >>> >>> >>> On 6/11/2012 8:33 AM, Daniel-Constantin Mierla wrote: >>> >>>> Hello, >>>> >>>> I committed a patch that should make it work when realm_prefix is not >>>> set for registrar module: >>>> >>>> http://git.sip-router.org/cgi-**bin/gitweb.cgi/sip-router/?a=**commit;h= >>>> **d42379da90f2ec87cb5dbb00ebb563**c7528ec910<http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=d42379d…> >>>> >>>> For the moment I have no idea why SIGBUS is triggered, i tried to access >>>> the structure directly, not via defines, and there is still a problem when >>>> realm_prefix is set. It's ok accessing the .len field but not .s of the >>>> same structure. The field .s is the first, making me think it's not >>>> alignment issue, or it is something very specific to this system. >>>> >>>> Overall, I am not familiar with sparc/solaris systems, maybe if you send >>>> here full details about OS version, architecture and CPU type, others using >>>> similar systems can give hints. >>>> >>>> Jason reported that on solaris with x86, this problem was not >>>> encountered. >>>> >>>> Cheers, >>>> Daniel >>>> >>>> On 6/4/12 8:57 AM, Daniel-Constantin Mierla wrote: >>>> >>>>> Hello, >>>>> >>>>> still strange with the crash on that if condition. One of your previous >>>>> reports showed a different line, when assigning a pointer. Also, I kind of >>>>> understood that save() was ok, but lookup() not. Is not the case anymore as >>>>> the backtrace shows. >>>>> >>>>> SIGBUS can occur because of alignment in memory which is thrown in >>>>> solaris, does not happen in linux flavours. I am thinking of putting the >>>>> prefix field first so the access to it is at beginning of the straucture, >>>>> just to see if that is the problem, although is should not be. >>>>> >>>>> I will add a patch soon, otherwise I am out of ideas for the moment >>>>> with the SIGBUS, I would need access to the machine to do some testing by >>>>> myself. >>>>> >>>>> Cheers, >>>>> Daniel >>>>> >>>>> On 6/2/12 10:36 PM, Akan wrote: >>>>> >>>>>> Ok, I reloaded the servers with v3.2 from git without performing the >>>>>> checkout and with just the master branch. Reran my tests on 2 servers and >>>>>> Kamailio terminated with a core dump. I have included the full trace of one >>>>>> of the servers. The other trace has the same results. >>>>>> >>>>>> Thanks >>>>>> >>>>>> Nathaniel >>>>>> >>>>>> On 5/30/2012 2:25 PM, Daniel-Constantin Mierla wrote: >>>>>> >>>>>>> Hello, >>>>>>> >>>>>>> On 5/30/12 9:10 PM, Akan wrote: >>>>>>> >>>>>>>> This was a re-pull from the git master. I had deleted all of the old >>>>>>>> files and issued these commands: >>>>>>>> >>>>>>>> git clone --depth 1 git://git.sip-router.org/sip-**router<http://git.sip-router.org/sip-router>kamailio >>>>>>>> cd kamailio >>>>>>>> git checkout -b 3.2 origin/3.2 >>>>>>>> >>>>>>>> This is the source that I got. Could this be a problem within git? >>>>>>>> >>>>>>> do not checkout branch 3.2, just try with the master branch, as said, >>>>>>> not all patches are backported to stable 3.2. >>>>>>> >>>>>>> For sake of completeness, here are some guidelines: >>>>>>> http://www.kamailio.org/wiki/**install/devel/git<http://www.kamailio.org/wiki/install/devel/git> >>>>>>> >>>>>>> Cheers, >>>>>>> Daniel >>>>>>> >>>>>>> >>>>>> >>>>>> ______________________________**_________________ >>>>>> 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<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<http://www.linkedin.com/in/miconda> >>>>> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - >>>>> http://asipto.com/u/katu >>>>> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - >>>>> http://asipto.com/u/kpw >>>>> >>>>> >>>>> >>>> -- >>>> Daniel-Constantin Mierla -http://www.asipto.com >>>> http://twitter.com/#!/miconda -http://www.linkedin.com/in/**miconda<http://www.linkedin.com/in/miconda> >>>> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - >>>> http://asipto.com/u/katu >>>> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - >>>> http://asipto.com/u/kpw >>>> >>>> >>>> No virus found in this message. >>>> Checked by AVG - www.avg.com <http://www.avg.com> >>>> Version: 2012.0.2178 / Virus Database: 2433/5061 - Release Date: 06/10/12 >>>> >>>> >>> >>> >>> ______________________________**_________________ >>> 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<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<http://www.linkedin.com/in/miconda> >> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - >> http://asipto.com/u/katu >> >> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - >> http://asipto.com/u/kpw >> >> >> >> >> ______________________________**_________________ >> 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<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> >> > >This email is subject to the disclaimer of Smile Communications (PTY) Ltd. at http://www.smilecoms.com/disclaimer > >_______________________________________________ >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

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High availability design with Kamailio
by Reinhard, Carel (GD) 12 Jun '12

12 Jun '12

Dear all, hope to get some information from you guys! For a redundant service we have to build up SIP proxy/registrar server in a high available scenario. Therefore, we will have 2 SIP proxy/registrar on two completely independent servers. They should be SIP dialog stateful and replicate all the session/call states between each other. If one server fails, the other should have all the SIP dialog information for call handling and accounting. How can this done in a stable and reliable way? Is this feature mature enough to support enterprise requirements? The only information I have is to build it up with the DISPATCHER module, but it seems more to be a stateless load-balancer instead of a HA module. Who has experience with HA failover designs with Kamailio and can give me some hints? Thank you very much in advance Kind regards Carel Carel Reinhard Security and VoIP Engineer SRG SSR Switzerland Telefon direkt +41 31 350 94 07 Carel.Reinhard(a)srgssr.ch<mailto:Carel.Reinhard@srgssr.ch>

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Shared memory fragmentation
by Morten Isaksen 12 Jun '12

12 Jun '12

Hi, We have 2 Kamailio 3.0.3 servers that has been running with carrierroute for about 2 years without any problems. They have 128 MB shared memory and modparam("carrierroute", "config_source", "db") modparam("carrierroute", "db_url", "<DBURL>") modparam("carrierroute", "fetch_rows", 500) The carrierroute table is about 91K lines and have been growing slowly. Suddenly we get this ERROR: carrierroute [cr_data.c:585]: could not allocate shared memory from available pool after a few "kamctl cr reload". I increased the shared memory to 256 MB but with the same result. I have now increased it to 512 MB and it seems to work better now. I have noticed this. After a restart the shmem counters is like this: shmem:total_size = 536870912 shmem:used_size = 28486752 shmem:real_used_size = 40147128 shmem:max_used_size = 41135424 shmem:free_size = 496723784 shmem:fragments = 555 And after the first "kamctl cr reload" it is like this: shmem:total_size = 536870912 shmem:used_size = 28619016 shmem:real_used_size = 51842768 shmem:max_used_size = 76993616 shmem:free_size = 485028144 shmem:fragments = 722063 Notice the increase in fragments. Sequentials "kamctl cr reload" does not change the fragments allot. Any ideas? -- Morten Isaksen

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maddr parameter in contact header and fix_nated_contact
by Morten Isaksen 12 Jun '12

12 Jun '12

Hi, I have this setup: Mediation server with internal ip -- NAT firewall -- Kamailio proxy -- PSTN gateway The mediation server sends a contact header with maddr=<internal IP> fix_nated_contact only changes the host part and leaves the maddr intact. When the PSTN gateway sends a PRACK or a BYE it puts the <internal IP> in the RURI, and kamailio then try to forward the message directly to the <internal IP> and not the NAT ip. >From RFC 3261 19.1.1: The maddr parameter indicates the server address to be contacted for this user, overriding any address derived from the host field. When an maddr parameter is present, the port and transport components of the URI apply to the address indicated in the maddr parameter value. [4] describes the proper interpretation of the transport, maddr, and hostport in order to obtain the destination address, port, and transport for sending a request. I can see that OpenSIPS has fixed this. http://sourceforge.net/tracker/?func=detail&atid=1086410&aid=3312423&group_… Would it be possible to get this patch into Kamailio also? -- Morten Isaksen

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Install Kamailio v3.3.x from GIT
by Daniel-Constantin Mierla 12 Jun '12

12 Jun '12

Hello, now that the branch 3.3 is available, I created the tutorial about how to install Kamailio v3.3.x from GIT: * http://www.kamailio.org/wiki/install/3.3.x/git Feedback and improvements are welcome! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

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git branch 3.3 created
by Daniel-Constantin Mierla 11 Jun '12

11 Jun '12

Hello, GIT branch 3.3 was created to hold the release series 3.3.x -- version 3.3.0 will be out in max. 1 week, a decision after collecting eventual issues still open to be reported today. Here are few details about how to use it. Download and use branch 3.3 from git: git clone --depth 1 git://git.sip-router.org/sip-router kamailio cd kamailio git checkout -b 3.3 origin/3.3 Pushing patches: - only bug fixes or documentation improvements have to be committed to branch 3.3 - first commit to master branch (if the bug applies there as well), cherry-pick the commit to branch 3.3 and push to remote repository Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

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development for future version 3.4.0 is open
by Daniel-Constantin Mierla 11 Jun '12

11 Jun '12

Hello, short note to let everyone know that new features can be committed again to the master branch -- they will be part of future version 3.4.0 (if no other numbering release will be decided). Testing phase of version 3.3.0 within master is ended, a dedicated branch being created for it (details in a separate message). Happy coding, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

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Capturing a users call (SIP+RTP+T.38)
by Klaus Darilion 11 Jun '12

11 Jun '12

Hi all! Do you know any comfortable tools to filter out a certain users call? e.g. searching in SIP packets for the user pattern, get the media port out of SDP and capture also RTP+T.38 I found pcapsipdump but have not tried it yet. Any suggestions or do I have to write my own tool? thanks Klaus

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branching 3.3 - to be used for 3.3.x releases
by Daniel-Constantin Mierla 11 Jun '12

11 Jun '12

Hello, I am going to create the GIT branch 3.3 in short time -- this branch will be used for releasing 3.3.x versions. After that master can take again commits of new features, to be part of next major release 3.4.x. Branch 3.3 will take only fixes. From my point of view, I have two issues that I am trying to solve and the 3.3.0 is ready to go out, namely: - a solaris sigbus - pua_dialoginfo issue reported on mailing list Otherwise, nothing else critcal is out there, to my knowledge. Report them if you have other ones. Packaging specs were updated, so we are pretty much ready to release when no relevant issues is open. The wiki page with what is new in this release was updated a bit as well, add if you find something missing at: - http://www.kamailio.org/wiki/features/new-in-devel Also, contribute to the upgrade guidelines: - http://www.kamailio.org/wiki/install/upgrade/3.2.x-to-3.3.0 Cheer, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

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About kamailio 3.3 msrp module and sylkserver
by Min Wang 11 Jun '12

11 Jun '12

Hi I am wondering what are the major differences between kamailio 3.3 msrp module and sylkserver (http://sylkserver.com/features.phtml ) ? thanks. min

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Dispatcher PSTN
by Kr0m 11 Jun '12

11 Jun '12

Hello I am not able to dial to pstn with kamailio, the call is routed to my pstn-gw(asterisk), but the final phone rings 4 or 5 seconds and then it is hanged up. My outbound route is: route[PSTN] { if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } route(TOASTERISK); exit; return; } route[TOASTERISK] { sl_send_reply("100","Trying"); uac_replace_from("$fn","sip:$fn@$fd"); route(NATMANAGE); ds_select_dst("1","4"); t_on_failure("1"); t_relay(); } failure_route[1] { ds_mark_dst("i"); if (!ds_next_dst()) { t_reply("503", "Service unavailable: no more dst"); exit; } route(TOASTERISK); } With a traffic capture i can see the traffic returning to my kamailio server. Any suggestion will be appreciated.

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Kamailio and Nonce Count
by patrice.bodeven＠orange.com 11 Jun '12

11 Jun '12

Hello, I try to understand the nonce count handling. By parameter "modparam("auth", "nonce_expire", 21600)" I can define the validity of the Nonce in the time. But I want also to define the validity of the Nonce in the number of use. For exemple, I want to limit the reuse of the nonce at 64 times (including register, Invite) But I can see the definition of the nonce_count for all the SIP users on Kamailio and not user per user. If I understand when I use modparam("auth", "nonce_count", 1) # enable nonce_count support modparam("auth", "nc_array_order", 20) # 1M in-flight nonces, using 1Mb memory modparam("auth", "nid_pool_no", 4) I define a memory to store the nonce values for all the SIP users. So when User A use nonce x for REGISTER and nonce y for INVITE, nonce x is still valid ! My understanding is it correct ? Is anyway to define a nonce validity in number of times per user ? Patrice B _________________________________________________________________________________________________________________________ Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration, France Telecom - Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci. This message and its attachments may contain confidential or privileged information that may be protected by law; they should not be distributed, used or copied without authorisation. If you have received this email in error, please notify the sender and delete this message and its attachments. As emails may be altered, France Telecom - Orange is not liable for messages that have been modified, changed or falsified. Thank you.

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Kamailio 3.2 + RTP proxy - no errors - no media
by Lukas Lani 10 Jun '12

10 Jun '12

Hi, I am trying to use RTPproxy 1.2.1 in bridge mode between IPv4 and IPv6 together with Kamailio 3.2. I configured and ran it on server together with Kamailio. I start it with parameters: -l 192.168.6.2 -6 /2001:6666:6666:6666::2 -s udp:localhost:7722 SIP messages are correctly exchanged, SDP looks to be translated, but when RTP session starts, I see RTP flow only from UC1 to RTPproxy and nothing more. On RTPproxy server I see 4 open ports for IPv4 address but no one for IPv6 address. When I tried to change start parameters to -6 2001:6666:6666:6666::2 -l /192.168.6.2 -s udp:localhost:7722 I could see 4 open ports for IPv6 address but no one for IPv4 address. Can anybody help? do you know about any known error? can it be related to Kamailio or RTPproxy? thank you, Lukas Lani

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Looking for RTP Proxy in TCP
by Austin Einter 08 Jun '12

08 Jun '12

Hi All Now I am using Kamailio 3.1.5 and RTP proxy 1.1. Looks both are compatible and working fine. The RTP Proxy basically sends/receives RTP packets over UDP. Is there any RTP Proxy available that does send/receive of RTP packets over TCP and also should be compatible with Kamailio 3.1.5. If you have any information in this regard, kindly share. Thanks a lot. Austin

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lcr from_uri not matching
by Ben WIlliams 08 Jun '12

08 Jun '12

Has anyone experienced this problem where lcr only works when from_uri is null. When I change it to .* the match fails. Thanks Ben

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Re: [SR-Users] Issue using Kamailio as a proxy in front of Asterisk
by Daniel-Constantin Mierla 07 Jun '12

07 Jun '12

Do not write private emails, keep the mailing list cc-ed for the topics started there. Cheers, Daniel On 6/7/12 12:22 PM, Dominik Mauritz wrote: > Daniel, Stoyna, > > the solutions you described look very interesting to me. I will have a > look at both and pick the one that fits best. > > Thanks guys. > > Rgs, > Dominik > > > > Am 07.06.12 10:57, schrieb Daniel-Constantin Mierla: >> Hello, >> >> On 6/6/12 9:57 PM, Stoyan Mihaylov wrote: >>> We use Jitsi as SIP client, and openxcap along with camailio to >>> handle presence. Then jitsi know if account is online or offline. >>> Our Asterisk dont know nothing about accounts (it accepts all calls >>> from kamailio). There I run AGI scripts, which can check kamailio >>> tables - and I can know if account is online or offline. Of course >>> this do not work if account is "forced" to offline. >>> But may be there is better solution. >>> >>> On Wed, Jun 6, 2012 at 10:04 PM, Dominik Mauritz >>> <dominik.mauritz(a)web.de <mailto:dominik.mauritz@web.de>> wrote: >>> >>> I have already tried that. I defined SIP-Accounts in Asterisk >>> with host=<Kamailio-IP> (instead of host=dynamic). This solves the >>> described problem but it also has side effects: >>> >>> - You don't have the correct presence status on your phone (e. >>> g. xlite) indicating wether the account is online or offline >>> - Asterisk always sends invites to Kamailio on incoming calls >>> even if there is no phone registered to the account >>> >>> These are not the worst things in the world, but it is maybe not >>> the best solution possible. >>> >>> >>> >>> Am 06.06.12 20:41, schrieb Stoyan Mihaylov: >>> >>> We use also Kamailio in front of Asterisk - but I forward >>> only calls to Asterisk - register/unregister stay in Kamailio. >>> Asterisk dont know which device is registered, and which is not. >>> >>> >>> On Wed, Jun 6, 2012 at 8:20 PM, Dominik Mauritz >>> <dominik.mauritz(a)web.de <mailto:dominik.mauritz@web.de> >>> <mailto:dominik.mauritz@web.de <mailto:dominik.mauritz@web.de>>> wrote: >>> >>> Hi All, >>> >>> some days ago I installed Kamailio as a front end for >>> Asterisk following this tutorial: >>> >>> http://kb.asipto.com/asterisk:__realtime:kamailio-3.1.x-__asterisk-1.6.2-as… >>> <http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb> >>> >>> >>> >>> I added Multi Domain support and adjusted some other things >>> to fit my environment. Almost everything is working perfectly now. >>> One small issue is left: >>> >>> With Kamailio in front of Asterisk I have one feature (next >>> to other cool things) I was missing for years. I am now able to >>> register more than one device on the same SIP account. This is >>> nothing new for Kamailio users but Asterisk doesn’t support this. >>> >>> If I register two phones on one SIP account with Kamailio >>> everything is fine. They are able to make outgoing calls and both >>> ring on incoming calls. But if one phone unregisters Kamailio >>> forwards the unregister request to Asterisk and Asterisk sets this >>> account to offline. Now a call comes in but Asterisk is not sending >>> out an invite because for Asterisk the phone is offline. >>> >>> I wonder if it is possible to forward an unregister request >>> from Kamailio to Asterisk only if the last endpoint registered with >>> one SIP account unregisters. If there is more than one endpoint >>> using the same Account Kamailio should not forward the unregister >>> request to Asterisk. >>> >>> Any idea? >>> >> you can use reg_fetch_contacts(...) in your config to find out how >> many contacts are for an user and based on that do not send the >> un-register to asterisk, see: >> >> http://kamailio.org/docs/modules/stable/modules_k/registrar.html#id2498205 >> >> http://kamailio.org/docs/modules/stable/modules_k/registrar.html#id2498441 >> >> >> Cheers, >> Daniel >> >> -- >> Daniel-Constantin Mierla -http://www.asipto.com >> http://twitter.com/#!/miconda -http://www.linkedin.com/in/miconda >> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 >> -http://asipto.com/u/katu >> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 >> -http://asipto.com/u/kpw >> >> >> > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

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record calls audio and video
by Saul Waizer 07 Jun '12

07 Jun '12

Hello List, I am looking for a way to record the audio and video of a call. I am using Kamailio (latest) on Ubuntu. Does anyone know if there's any existing modules that allow for recording or any other solution that can provide recording of calls on Kamailio? Thank you!

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Siremis v3.2.1 Released
by Elena-Ramona Modroiu 07 Jun '12

07 Jun '12

Hi, Siremis v3.2.1 is out, the web management interface for Kamailio SIP Server. It is the last release with new features planned to be compatible with Kamailio v3.2.x. Next major releases for Siremis will target upcoming Kamailio v3.3.x. This time new features relate to web pages that allow management of own profile by SIP users, such as changing SIP password, management of speed dials or views for accounting, cdrs and location records. There was also added a form for doing user registrations (e.g., can be used for allowing public registrations). You can read more details about the release as well as details about installation at: * http://siremis.asipto.com/ Regards, Ramona

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Limiting calls from users in prepaid scenerio
by Aft nix 07 Jun '12

07 Jun '12

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all, We are thinking about launching a different "usage policy". The usage policy will be prepaid in some sense. but instead of limiting "number of minutes" we are thinking about limiting "number of calls". That means if you pay this X amount of doller, you can make Y amount of calls. calls can last whatever the user wants given that its under 1 hr. I've seen how to limit concurrent calls using SER, but the usage i have described, is not seen by me. So please help me out with this thing. My initial understanding about how it can be done : 1) when a dialog is confirmed with ACK(we have both callid, from tag, to tag), we can make an entry to the database by increasing "number of calls" field by one. 2) A field in database will be used as flag for usage limit. when "number of calls" exceeds the usage limit, it will flag it as "finished usage limit". 3) when a INVITE arrives, a check will be made with $fU against DB. With this framework in mind, I have read "db_mysql" module. And to my surprise, no direct the database functions are not exported to the cfg. I don't know if any module in kamailio exports mysql_queury() to cfg, which i need for my purpose. And may be there is some built in module which provides a higher level layer which will enable myself achieving my goal. If in the end i need "exported" mysql functions, then i can develop it myself. So which way i should go? Thanks in advance - -- - -aft -----BEGIN PGP SIGNATURE----- Version: OpenPGP.js v0.1 Comment: http://openpgpjs.org wsBcBAEBAgAQBQJPzc6dCRCJVJ6A/SK8awAARAkH/R5rqfiTPqBP+zZr8rgM 0JE42l9jxYq5OteD13U4CvcrUklXZlgDf2jwr1F7ndd1rlV3yKXAT2u61Bz5 LwNneRZydl3xnPmt2+TV6Qkcgy8RXgyRMntP9iLeQOXWJNW7D637h7tLL11d O50s86ofgJT+dfTICbomNZjLZRr1DvYOdcByG3LHcYx033G6bIFlMqJ0ZO+G cBdgtJcLrwo6iPn8Gv+5aJDy2kRUphrF7cVCzFzLPnNVZpvXNNcv7LB9bkUH cClcLDVAbsdw0TPd9Z6DNJSxFlHy2uzwNWB9mhpVJD1toT8Geb4iCOi+ZG8P WT5fgNwSoZlEm0BBBHwMUdk= =fcdD -----END PGP SIGNATURE-----

2 2

Issue using Kamailio as a proxy in front of Asterisk
by Dominik Mauritz 07 Jun '12

07 Jun '12

Hi All, some days ago I installed Kamailio as a front end for Asterisk following this tutorial: http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb I added Multi Domain support and adjusted some other things to fit my environment. Almost everything is working perfectly now. One small issue is left: With Kamailio in front of Asterisk I have one feature (next to other cool things) I was missing for years. I am now able to register more than one device on the same SIP account. This is nothing new for Kamailio users but Asterisk doesn’t support this. If I register two phones on one SIP account with Kamailio everything is fine. They are able to make outgoing calls and both ring on incoming calls. But if one phone unregisters Kamailio forwards the unregister request to Asterisk and Asterisk sets this account to offline. Now a call comes in but Asterisk is not sending out an invite because for Asterisk the phone is offline. I wonder if it is possible to forward an unregister request from Kamailio to Asterisk only if the last endpoint registered with one SIP account unregisters. If there is more than one endpoint using the same Account Kamailio should not forward the unregister request to Asterisk. Any idea? Thanks, Dominik

3 4

kamailio as SBC
by Mino Haluz 06 Jun '12

06 Jun '12

Hi, I know that kamailio is SIP proxy, but is there any way how to implement kamailio as SBC like OpenSIPS with B2BUA module ? I tried OpenSIPS with this module, but it does not work with mediaproxy module. Mino

5 4

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sr-users June 2012