search for: kamalio

...whatever is needs to. These messages are sent during calls but may also be sent when a call is over. At the moment, I'm using Asterisk to serve these SIP phones so my Asterisk box receives those SIP PUBLISH and discard them with a 489 Bad Event reply. I'm not using or planning to use any Kamalio server. 1. Is there an Asterisk version that would allow me to read (and store) in or out-of-band SIP PUBLISH messages from SIP phones ? 2. Alternatively, is there an Asterisk version that would allow me to relay those messages somewhere ? 3. Would a Kamalio-like box allow me to do this ? 4. Sugge...

...nt when a call is > > over. > > > > At the moment, I'm using Asterisk to serve these SIP phones so my > > Asterisk > > box receives those SIP PUBLISH and discard them with a 489 Bad Event > > reply. > > > > I'm not using or planning to use any Kamalio server. > > > > 1. Is there an Asterisk version that would allow me to read (and store) > > in > > or out-of-band SIP PUBLISH messages from SIP phones ? > > 2. Alternatively, is there an Asterisk version that would allow me to > > relay > > those messages s...

used Kamalio to "supplement" the features that Asterisk either doesn't provide or doesn't provide in as nice a form as the OP desired - can't really speak beyond this as I am not one of them. ------=_NextPart_000_010C_01CB6EAA.3AC2C610 Content-Type: text/html; charset="us-ascii&quo...

...p shows >the invite in the CLI. > > My setup has chan_sip running on 5060 and pjsip (tcp and udp on 5061). > > I'll get PJSIP running on 5060 and see if that makes any difference. UPDATE: I got PJSIP on 5060 and everything is working fine as expected and I can see the calls from Kamalio. Is this a bug with asterisk not recognising the traffic on 5061 even though the SIP messages are being received by the server on that port and I can see it? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachme...

Hi, Does anyone have some good examples of a Kamalio or OpenSips configuration that integrates with Asterisk? Essentially I want to use the SIP router as the UA, but still run all the calls through Asterisk (for dialplan, etc..) I've looked for examples on the project web sites, but I haven't found anything decent yet. Thanks. -- James

Hi All, Do you'll have any recommendations on a Linux based Customer Management and Pre-paid Billing system for Asterisk, Freeswitch or Kamalio? The system should also allow customers to register, login, buy more credit, view call records, etc. Commercial or Open-source are ok as long as they run on Linux. Thanks, A. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/aste...

Hi Guys Is it possible to register Kamallio directly to our SIP provider then load balance the RTP to 2 asterisk servers? We cant do the registration from the asterisk boxes as we want to do it directly from Kamallio. Is this possible? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello! We're looking for a solution to reliably load balance our Asterisk boxes. So far we've been using a hodge-podge of directing different services to different boxes/IPs, but eventually I'd like to consolidate things so we can present a single IP address to the outside world. My question is - how do we go about doing that? I've read a lot of things like load-balancing via

Hello, Our university has to upgrade soon its old Nortel PBX's which holds around 10,000 extensions tied to 5 PBXes. Up to now we thought about commercial solutions but now there is a window openning for open source solution. However, I need examples to convince that this solution is feasible, and preferably at other universities. Are there any pointers for such installations?

...asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > Hello very interesting project you have, however asterisk is not > a registry server, i suggest that you use opensips/opense/kamalio for > your registrar, from where you dispatch to you asterisk servers, inside > a good environment with a controlled network and nice tagged voip flow > you could acheve a good results. > > > _______________________________________________ > -- Bandwidth and Coloc...