<span class="Apple-style-span" style="border-collapse: collapse; "><div>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.<br>
</div><div><br></div><div><br></div><div><br></div><div><br></div><div><br></div>disconnect at time of answer is a classic example of codec mismatch.<br><br>calling site1:xlite to site2:ciscophone and getting g.729 means you are<br>
configured for g.729 between sites. As g.729 is a licensed codec I do<br>not believe it is included in xlite natively. in that case CM will<br>recognize a code mismatch and allocate a transcoder to compensate.<br><br>unfortunately that breaks down when you have 2 xlite phones involved<br>
unless you have transcoders avialable at both sites and transcoders<br>included in the MRGL of the xlite phone configuration in callmanager.<br><br>if xlite supports iLBC you could consider upgrading to a version of CM<br>
that supports iLBC and use that as your low bit rate codec instead of g.729.<br><br>to confirm it is a codec issue go into your regions config within CM<br>and allow g.711 between the 2 sites. if xlite calls then work then you<br>
are running into a codec mismatch.<br><br>/wes<br><br>On Thursday, July 30, 2009 9:23:26 AM , venkata sashank<br><<a href="mailto:reachsashank@gmail.com" style="color: rgb(42, 93, 176); ">reachsashank@gmail.com</a>> wrote:<br>
> hi,<br>> i have 2 sites.Between two sites i am using xlite sip phones . I am<br>> trying to call the sip phone in site 1 from site2 sip phone. Both<br>> phones will ring fine, if i pick the call the call will get<br>
> disconnects, calls from sip phone to sccp phone within the same site<br>> and between different sites will go good .when i check the call<br>> statistics in the same site when call is established between the sip<br>
> phone and sccp phone they use g711 codec, and call between site 1 sip<br>> phone to site 2 sccp phone uses g729 codec and the coll goes through(<br>> i was wondering how it was possible). but between 2 site sip phones<br>
> call cannot be established. In site 2 sip phone when i try to call<br>> remote sip phone i get an error " call failed: not implemented." any<br>> suggestions would be welcomed<br>><br>><br>><br>
> Warm Regards,<br>> Venkata Sasanka.pathi(-91 950 265 2290),<br>> Consultant | unified communications,<br>> Locuz Enterprise Solutions Ltd. (A Subsidiary of 3i-Infotech)<br>> (office:914066115512),<br>
> Email address: <a href="mailto:sasanka.pathi@locuz.com" style="color: rgb(42, 93, 176); ">sasanka.pathi@locuz.com</a> <mailto:<a href="mailto:sasanka.pathi@locuz.com" style="color: rgb(42, 93, 176); ">sasanka.pathi@locuz.com</a>><br>
> <a href="http://www.locuz.com" target="_blank" style="color: rgb(42, 93, 176); ">www.locuz.com</a> <<a href="http://www.locuz.com" target="_blank" style="color: rgb(42, 93, 176); ">http://www.locuz.com</a>><br>
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