[c-nsp] sip trunk to asterisk

[s m]

thank you guys for your answers.
 i have no problem if i use just one codec as you mentioned Jared. my call
hangs up when i have voice class codec with more than one codec. i trace
all debug message and think that my cisco router can not transcode codecs
to each other. so when the codec of receiving call is different with the
first priority in cisco router, cisco can not transcode it, therefore call
hangs up. do you think i am true or not??? if yes would you please tell me
how configure my router to transcode between different codecs (for example
between alaw and ulaw)? my cisco router is 2800.

thank you
SAM

On Mon, Mar 30, 2015 at 11:51 PM, Jared Mauch <jared at puck.nether.net> wrote:

> 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.
>