[Asterisk-Users] No "native bridge" on outbound SIP channels
Eric Bishop
asterisk.eric at gmail.com
Sat Jan 14 23:23:08 MST 2006
Yes the 7960 is also set only to use alaw. I was under the impression though
that nat=yes did not effect this. And if it does why does it native bridge
ok on inbound calls with the same nat=yes
On 1/15/06, Jonathan Feally <vulture at netvulture.com> wrote:
>
> I'm guessing that you have a similar entry in your sip.conf for the 7960??
> The 7960 has a setting for preferred codec. It defaults to g711 U-Law. You
> might try changing this setting also as the 7960 doesn't know that you only
> want to speak A-Law. You will also want to make sure that the nat settings
> are disabled on both devices as they are on the same network. nat=never is a
> better choice than nat=no. You might also check your extensions.conf to
> verify that the calling from 1760 to 7960 is the same as from 7960 to 1760.
> You could also try moving both devices to using U-Law instead.
>
> -Jon
>
> Eric Bishop wrote:
>
> Hi all,
>
> I have a Cisco 1760 gateway and and Cisco 7960 VoIP phone running via
> Asterisk. Both are running g711A codecs and SIP. On inbound calls I get a
> native bridge, however on outbound calls I never get a native bridge. With
> other SIP gateways I do get a native bridge on the outbound call. My
> sip.conf is as follows:
>
> [cisco1760]
> type=friend
> context=incoming
> host=192.168.0.55
> insecure=yes
> nat=no
> canreinvite=no
> dtmfmode=rfc2833
> disallow=all
> allow=alaw
>
> I have also confirmed while on an outbound calls that both are using the
> exact same codecs. sip show channels shows
>
> pbx*CLI> sip show channels
> Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last
> Message
> 192.168.0.55 123456789 4ea2e1314cd 00102/00000 alaw No
> Tx: ACK
> 192.168.0.58 200 0013c427-f4 00101/00102 alaw No Rx:
> ACK
> 2 active SIP channels
>
>
> Anyone have an idea what's going on?
>
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