[cisco-voip] sip phone calls between two sites

[venkata sashank]

What if i have call manager express and what can i  do in this scenario. i
understood that its some  codec mismatch. on call manager express i have
specified  g711 codec in sip voip dial peer and also in the ephone( voice
register pool incase of sip) but still no go. The site 2 phone when
initiates the call i get an error message " call failed: not implemented."
does any one have any idea why it happens. thank you in advance.





disconnect at time of answer is a classic example of codec mismatch.

calling site1:xlite to site2:ciscophone and getting g.729 means you are
configured for g.729 between sites.  As g.729 is a licensed codec I do
not believe it is included in xlite natively.  in that case CM will
recognize a code mismatch and allocate a transcoder to compensate.

unfortunately that breaks down when you have 2 xlite phones involved
unless you have transcoders avialable at both sites and transcoders
included in the MRGL of the xlite phone configuration in callmanager.

if xlite supports iLBC you could consider upgrading to a version of CM
that supports iLBC and use that as your low bit rate codec instead of g.729.

to confirm it is a codec issue go into your regions config within CM
and allow g.711 between the 2 sites.  if xlite calls then work then you
are running into a codec mismatch.

/wes

On Thursday, July 30, 2009 9:23:26 AM , venkata sashank
<reachsashank at gmail.com> wrote:
> hi,
> i have 2 sites.Between two sites i am using xlite sip phones . I am
> trying to call the sip phone in site 1 from site2 sip phone. Both
> phones will ring fine, if i pick the call the call will get
> disconnects, calls from sip phone to sccp phone within the same site
> and between different sites will go good .when i check the call
> statistics in the same site when call is established between the sip
> phone and sccp phone they use g711 codec, and call between site 1 sip
> phone to site 2 sccp phone uses g729 codec and the coll goes through(
> i was wondering how it was possible). but between 2 site sip phones
> call cannot be established. In site 2 sip phone when i try to call
> remote sip phone i get an error " call failed: not implemented." any
> suggestions would be welcomed
>
>
>
> Warm Regards,
>  Venkata Sasanka.pathi(-91 950 265 2290),
>  Consultant | unified communications,
>  Locuz Enterprise Solutions Ltd. (A Subsidiary of 3i-Infotech)
>  (office:914066115512),
>  Email address:  sasanka.pathi at locuz.com <mailto:sasanka.pathi at locuz.com>
>  www.locuz.com <http://www.locuz.com>
> ------------------------------------------------------------------------
>
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