Hello there again,
When I setup call forward to external number and I call DID of that extension from external phone call is forwarded but when its answered there is no audio. When I call that extension internally call is forwarded and after answer everything is OK.
There is the same issue with find me/follow me. If I set up to ring my mobile with my extension and I call DID of my extension and answer the call on my mobile there is no audio.
I enabled sip and rtp debug in asterisk and I can see that all SIP packets are as expected but there are no RTP packets at all.
Also I've done tcpdump traces and again that just confirmed no RTP packets at all. For external calls I use sip trunk.
Any idea whats wrong?
PBX manager: 6.0.1.76
Asterisk: 1.6.0.13
Peter
Of course reinvites are off.
Of course reinvites are off. I will try it on asterisk 1.4, maybe it's another bug of asterisk 1.6.0.X... I'll let you know.
Anyway, which asterisk version do you use, which version do you think is most stable with least bugs?
I know that some of the features of asterisk 1.6.0.X and 1.6.1.X are quite nice and usefull but if the whole system doesn't work properly its not worth it...
Peter
im still hanging with 1.4.26
im still hanging with 1.4.26 on my MTE. My home pbx is 1.6.1.x in order to test the parking changes. is it a no-audio or a one-way audio problem? Also try running 1.6.0.latest to see if the problem corrected. Reverting back to 1.4.x will require re-installing the TL module so that the right dialplan gets used.
I just installed thirdlane
I just installed thirdlane MTE ISO with asterisk 1.4.18 on my test mashine. Only registered SIP trunk and 1 extension. Setup call forward always to external number and it's exactly the same... Phone is ringing but after answer there is no-audio (not one-way audio).
no idea what to do now...
Peter
well, I use thirdlane only
well, I use thirdlane only couple of months and I never used call transfer to external number. So maybe it never worked...
Peter
so try this... call into a
so try this...
call into a direct dial # that rings an extension. then do a blind transfer to an outside # like a cell phone.
if that works try this...
set call forwarding on the phone, call the direct dial number again.. see if that works
Blind transfer works fine.
Blind transfer works fine. Rings the phone and I have audio as well. When I set forward on the phone or in PBX manager still no audio...
Peter
are you sure you dont have
are you sure you dont have reinvites enabled on the trunk to your sip provider?
yes, I'm sure, reinvites are
yes, I'm sure, reinvites are disabled on all extensions and sip trunk. I've checked it about million times in PBX manager and in sip.conf as wel...
Peter
OK, I tried - different sip
OK, I tried
- different sip provider - same problem
- 2 providers - one in, second out - same problem
- installed elastix pbx from iso - same problem
- installed switchwox PBX from ISO - same problem
- tried x-lite, mitel, polycom phones, different DIDs - same problem
- PBX installed on server with public IP, then test PC behind NAT - same problem
- forwarded to mobile and landline - same problem
- forward setup in PBX menu and in phone menu - still the same
I just don't know...
Peter
My production server has
My production server has public IP. Most of the tests I've done on this server but I also tested it on another pc which is behing NAT.
Peter
I made small mistake when
I made small mistake when testing it with second sip trunk, firewall wasn't configured correctly. I fixed it and now when I set call forward to external number using second sip trunk (first in, second out) it works fine. But still when I use the same sip trunk to get out there is no audio.
I can't use second sip trunk for this purpose so I need to make it work with just 1 sip trunk...
I test this on server with public IP.
Peter
It seems to be providers
It seems to be providers issue. Just opened ticket with them, hopefully they will come back to me soon. I will let you know.
Peter
It was problem on providers
It was problem on providers side. They had auto-nat enabled on their cisco gateway.
As soon as they turn it off everything works.
Thanks for your help.
Peter
make sure reinvites are turned off