[c-nsp] sip trunk to asterisk

[Jared Mauch]

On Sun, Mar 29, 2015 at 09:05:50AM +0430, s m wrote:
> hello everybody,
> 
> i want to configure a sip trunk between a cisco router and my system which
> has asterisk. this is my scenario:
> 
> Freepbx-----my system-----cisco-router----Freepbx
> 
> my system acts like a router. in cisco, if i set just one codec in
> dial-peers, every thing is ok and i can make a call. but if i set different
> codecs in a voice class codec and assign it to dial-peers, i can make call
> but call is terminated.
> i think there is some difference in sip options (maybe sip headers) between
> cisco and asterisk which causes to codec negotiation fail. as a result of
> it, call terminate.
> 
> any body try it before? any comments or hints are really appreciated.

	What codec are you trying to use?  I've had good success
with using g711ulaw on both sides.

We've had issues with some providers and DTMF working as well
and it seems that Cisco you need to configure the dtmf relay
in about 25 different places to make it all work right, eg:

voice service voip
 dtmf-interworking rtp-nte
 signaling forward unconditional
 h323
  call service stop
 sip
!

and

!
dial-peer voice 1  voip
 preference 1
 destination-pattern my_regex
 session protocol sipv2
 session target ipv4:1.2.3.4
 session transport udp
 dtmf-relay rtp-nte
 codec g711ulaw
 fax-relay ecm disable
 fax rate disable
 fax protocol pass-through g711ulaw
 no vad
!



	- Jared

-- 
Jared Mauch  | pgp key available via finger from jared at puck.nether.net
clue++;      | http://puck.nether.net/~jared/  My statements are only mine.