Re: [OpenSIPS-Users] timeout between the invite and the sip message after the "180 trying"?

ok On Wed, Apr 25, 2018 at 9:43 AM, Abdoul Osséni <abdoul.oss...@gmail.com> wrote: > Yes > > between 100 trying or 180 ringing? > or > between 100 trying or 183 sdp progress? > > > Abdoul > > 2018-04-25 15:10 GMT+02:00 Russell Treleaven <rtrelea...@bun

Re: [OpenSIPS-Users] timeout between the invite and the sip message after the "180 trying"?

Do you mean 100 trying or 180 ringing? On Wed, Apr 25, 2018, 9:03 AM Abdoul Osséni wrote: > Hello list, > > Is it possible to set the timeout between the invite and the sip message > after the "180 trying"? > > Best regards > Abdoul. >

Re: [OpenSIPS-Users] Remote ip change in-dialog

What mobile sip client supports this? On Mon, Oct 10, 2016 at 6:50 AM, Saioa Perurena wrote: > Hi, > > I've the following problem, any advice will be welcome. > > A calls B, A changes his ip because of network change (3G to wifi for > example) sends again an

Re: [OpenSIPS-Users] How to overcome SIP ALG on Wi-Fi routers

Look for the fragmentation flag. On May 17, 2016 1:08 PM, "Nabeel" wrote: > In that case, the answer to your question seems to be that the UDP packets > did not reach the OpenSIPS server, because nothing was added to the > OpenSIPS logs using debug level 4. All of this

Re: [OpenSIPS-Users] How to overcome SIP ALG on Wi-Fi routers

TCP works for me. On May 2, 2016 8:43 PM, "Nabeel" wrote: > Thanks for the suggestions of using TLS or changing the port. I changed > the port, but some routers are still able to mess with the SIP headers. I > would have used TLS, if not for two reasons: > > 1. ICE

Re: [OpenSIPS-Users] [BOOK] Building Telephony Systems with OpenSIPS - 2.1 version

Never mind I see that it is now available from Oreilly. On Tue, Feb 9, 2016 at 10:33 AM, Russell Treleaven <rtrelea...@bunnykick.ca> wrote: > Hi, > > Will this book become available through O’Reilly Media? > I try to buy from them if possible. > > Sincerely, > > Rus

Re: [OpenSIPS-Users] Secretary/Boss

I will let others fill in the details but I think this is what you want to achieve on a high level. A user calls the Boss's number which is actually routed to the assistant's phone. Assistant answers call and transfers the call to the Boss's "real" extension

Re: [OpenSIPS-Users] SIP and RTP Proxy without local user base

As an alternative you could try bypass media. https://freeswitch.org/confluence/display/FREESWITCH/Bypass+Media+Overview On Fri, Apr 24, 2015 at 11:17 AM, Roman Dissauer ro...@dissauer.net wrote: Dear all, I’m running a centralized Freeswitch based PBX for use on several sites. All phones