if you have enough DSP).
>
> Are you sold on h323 or can you do a full SIP trunk (with MTP) between
> cucm and asterisk?
>
> Thanks,
>
> Ryan
>
>
> ---- Original Message
> From: s m
> Sent: Monday, May 11, 2015 12:25 AM
> To: Ryan Huff
>
>
>
> -------- Original Message
> From: s m
> Sent: Sunday, May 10, 2015 01:19 AM
> To: cisco-voip@puck.nether.net
> Subject: [cisco-voip] how codec transparent works?
>
> hello everybody,
>
> anybody knows how codec transparent works?
>
> i have a
hello everybody,
anybody knows how codec transparent works?
i have a strange problem. i want to set h323 trunk between asterisk and
cisco 2800. it only works when i set codec transparent in dial-peer nodes.
show commands in cisco shows that i have a call with g711alaw but if i set
codec g711alaw
codecs which cisco
and asterisk utilize? dose anyone know anything about it?
On Thu, Apr 30, 2015 at 1:08 PM, wrote:
> how do u know we r all guys? grow up sexist/rasisixt..
>
> - Original Message -
> From: s m
> Date: Thursday, April 30, 2015 3:28 am
> Subjec
you send the full H.245 exchange for a call? That should show us
> where it is failing. We'll want to make sure it gets all the way yo both
> sides sending OpenLogicalChannelAcks.
>
> On Wed, Apr 29, 2015 at 1:14 AM, s m wrote:
>
>> thank you Brian, yes i have set bind a
un "debug h245 asn1" to see if media negotiations as well.
>
> On Tue, Apr 28, 2015 at 3:55 AM, s m wrote:
>
>> hello guys,
>>
>> i want to have h323 trunk between cisco 2800 and asterisk 11.13.1 with
>> ooh323 module. i configured both side and have succe
hello guys,
i want to have h323 trunk between cisco 2800 and asterisk 11.13.1 with
ooh323 module. i configured both side and have successful call from cisco
to asterisk. but when call comes from asterisk to cisco, my phone rings but
no audio is heard and call is disconnected after 5 second. i enab
nscoding you’ll need DSCP resources and some configuration:
>
> http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/transcoding.html
>
>
>
> Cheers,
>
>
>
> *Zoltan Kelemen*
> Emerson
>
>
>
> *From:* cisco-voip [mailto:ci
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-routerFreepbx
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.
