Re: [SR-Users] Carrierroute, fallback, reinvites

There's probably thousands of setup like that, it works great. On Aug 12, 2017 21:37, wrote: > Hello. Trying to implement fallback route for carrierroute and got sick to > make reinvites working correctly. Until fallback route triggers, everything > is OK. Anyone has working

Re: [SR-Users] RTPEngine recording

Also, the "proc" recording method has an interesting pipeline, involving an ephemeral metadata file and a memory sink exposed through /proc, and a userspace daemon picking up the data and writing it to disk (if this doesn't happen, audio frames going into the sink are discarded). This pipeline

[SR-Users] Carrierroute, fallback, reinvites

Hello. Trying to implement fallback route for carrierroute and got sick to make reinvites working correctly. Until fallback route triggers, everything is OK. Anyone has working config for that purposes (NAT traversal, reinvites, carrierroute ), please? And abstract question - is it generally a

Re: [SR-Users] RTPEngine recording

I have succeeded in prototyping a recording setup using the 'proc' method. However, I've got one issue I can't seem to figure out. On inbound calls only, the inbound (caller) leg on the PSTN side seems to show up interleaved/stuttered in the recording, and also slowed down considerably. This

Re: [SR-Users] Route calls from Kamailio to Asterisk behind NAT

On Fri, Aug 11, 2017 at 09:35:45AM -0700, Benjamin Fitzgerald wrote: > Another amazing tool for this is sipgrep ( > https://github.com/sipcapture/sipgrep). It's like ngrep but for SIP. And here I thought sngrep was the ngrep for SIP :) ___ Kamailio

Re: [SR-Users] Route calls from Kamailio to Asterisk behind NAT

On Sat, Aug 12, 2017 at 08:49:45AM +0300, v...@cell.tel wrote: >Using alias_db - 7XXX calls were succesfully aliased and routed to >Asterisk through user (asterisk server did not accept connections to >sip ports from Internet due to security reasons). >On asterisk side, all