[SR-Users] Help with Asterisk RT integration
Hi I'm trying to integrate a (K) front end cluster with an Asterisk back end cluster and Asterisk RT (legacy system) I've followed the recipe at http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb (with one minor exception …) if (from_uri!=myself && uri!=myself) became if (uri!=myself) as with the original line in place we were able to spoof traffic from 3rd party sites and route out onto the PSTN (which I thought was bad) … anyway … The problem I'm seeing currently is that when a call is passed down a SIP trunk to an end user on the (K) platform we're losing the DNID Asterisk delivers the call to SIP/account/DNID (K) however just tries to deliver the call to DNID@domain which comes back as not found. I've had limited success hand coding aliases into the alias_db however we're still missing the DNID info and having to scrape it from the SIP To field (with a) limited success and b) fears of a support nightmare if we try and move existing customers onto the new platform). If we were to do this for real I'd either have to modify usrloc or try some perl_exec magic I'm guessing … (although it might be possible with a view on the existing sql database … but I'd rather not have to do that if there is a simpler way) I confess I'm struggling to get my head around the config and docs and hoped someone could point me in the right direction? For legacy reasons Asterisk needs to be in the critical path on this particular build … what I'm looking for is a simple recipe and some helpful pointers on how to implement it that will allow enable me to swing (K) into the path between our end user SIP devices and the existing asterisk back ends without losing the ability to deliver hundreds of numbers down a single SIP trunk to a subscriber, and that doesn't require them to make any changes on their end as they will still see the equivalent of SIP:${DNID}@example.com arriving on their PBX This should be simple, but I'm obviously missing something :) Help and pointers gratefully received -- Jon Morby FidoNet - the internet made simple! 10 - 16 Tiller Road, London, E14 8PX tel: 0845 004 3050 / fax: 0845 004 3051 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] NAPTR, SRV and sips vs. transport=tls
Iñaki Baz Castillo writes: > sips is a bad hack and a useless pseudo-security mechanism. An IETF > fail. Don't you think that? yes, in my opinion too sips should never been included in rfc3261. -- juha ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] NAPTR, SRV and sips vs. transport=tls
2012/12/11 Olle E. Johansson : > In addition there is a lot of missing pieces to get SIPS: to work. LIke how a > proxy > can signal back to the originating UA that it could not set up a TLS > connection because > the certificate of the next hop was bad/expired/not signed by approved CA or > something else. And there are more issues (I hate to remember them) that make SIPS unfeasible. > After ten years, I think SIPS as a uri scheme is a lost cause. This does NOT > mean that > TLS is a lost cause, but I think we can't leave the decision about security > to the end point > user - and they can't decide whether or not they want to place a request for > "secure signalling" in their > call setup. The WebRTC way is better, just make every call more secure. Well, WebRTC just defines the media plane (which MUST be SRTP-DTLS) but the signaling plane is up to the application/web provider, which can be as secure (or insecure) as any SIP or HTTP deployment. Cheers. -- Iñaki Baz Castillo ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] NAPTR, SRV and sips vs. transport=tls
2012/12/11 Juha Heinanen : > Iñaki Baz Castillo writes: > >> transport=tls has NEVER been real, no one RFC mentions it. > > transport=tls is very real. many sip UAs and proxies support it. I know that and I never said the opposite. I said that no one RFC standarizes transport=tls. > funny that you now care about RFCs, when in the same message you don't > care about them regarding sips. sips is a bad hack and a useless pseudo-security mechanism. An IETF fail. Don't you think that? -- Iñaki Baz Castillo ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] MySql error
Hello, On 12/11/12 3:20 PM, Hermes F. wrote: I'm trying to create MySql database using the following |/usr/local/sbin/kamdbctl create I followed the instructions on http://kb.asipto.com/kamailio:install:3.3.x-from-git-centos5x but I receive the following error message: "ERROR: database engine not specified, please setup one in the config script" Can someone help me please ? | |you have to update the kamctlrc and set DBENGINE: http://kb.asipto.com/kamailio:install:3.3.x-from-git-centos5x#mysql_database Cheers, Daniel | -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Regarding Kamailio Registrar
You are right after reboot registrar module takes registrations from a db. If someone send UnRegister during outage, kamailio will keep active registration until occurence of expiration time that was calculated during registration. To speed up user location cleaning you can decrease the "expire" value in registrations. Dear all, We are using Kamailio SIP Server in our application. We want to know 1 thing: We are using Kamailio SIP server as registrar and proxy server. For registration database, we are using dbtext module. We maintain record of SIP Extensions who have registered in Our proprietary file too. Now when we boot the Kamailio SIP server, At that time our file should be empty i.e. previously registered Extension entries should be deleted or it must be kept as it is. But if during power off, any extension had sent REGISTER request to Unregister then it was not executed as kamailio SIP server was not ON. But at boot time, as its entry was present, Kamailio SIP server will Consider it a registered extension. Please guide us for this situation. Thank you, Amit Shah. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] MySql error
I'm trying to create MySql database using the following /usr/local/sbin/kamdbctl create I followed the instructions on http://kb.asipto.com/kamailio:install:3.3.x-from-git-centos5x but I receive the following error message: "ERROR: database engine not specified, please setup one in the config script" Can someone help me please ? ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Unable to set custom RPID
Hello, have you set the value of RPID in the appropriate avp? http://kamailio.org/docs/modules/stable/modules_k/siputils.html#id2537282 Cheers, Daniel On 12/11/12 3:00 PM, Mino Haluz wrote: Hi, what things should I set in order to set my custom RPID. What I am doing: remove_hf("Remote-Party-ID"); append_rpid_hf("", ";party=calling;id-type=subscriber;screen=no"); but RPID is only removed, second command does not append any rpid. Thanks Mino ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@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 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Unable to set custom RPID
Hi, what things should I set in order to set my custom RPID. What I am doing: remove_hf("Remote-Party-ID"); append_rpid_hf("", ";party=calling;id-type=subscriber;screen=no"); but RPID is only removed, second command does not append any rpid. Thanks Mino ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Planning next minor release - v3.3.3
Hello, I am considering releasing next minor version from branch 3.3 at the beginning of next week, Monday or Tuesday (Dec 17 or 18). For outstanding issues, please use the tracker or send to sr-dev mailing list. Developers that pushed fixed in the master branch should backport them in 3.3 before the next week starts. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users