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You will probably want canreinvite=yes on your sip entries unless you
are going to be using monitoring or some other feature in which
asterisk needs to hear the conversation. Also, Is asterisk answering
the call from the 7960 or is the 1760 doing it through the dial cmd? If
asterisk answers the call, then this could be part of the problem.<br>
<br>
Can you send an output of the console for a call from 1760 -> 7960
with a <br>
show channel for each SIP device, and then the same thing for 7960-1760.<br>
<br>
-Jon<br>
<br>
Eric Bishop wrote:<br>
<blockquote
cite="mid4acda1b40601142223x2056199es850f5d18bc0e8bce@mail.gmail.com"
type="cite">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<br>
<br>
<br>
<br>
<br>
<div><span class="gmail_quote">On 1/15/06, <b
class="gmail_sendername">Jonathan Feally</b> <<a
href="mailto:vulture@netvulture.com">vulture@netvulture.com</a>>
wrote:</span>
<blockquote class="gmail_quote"
style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
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.<br>
<br>
-Jon<br>
<br>
Eric Bishop wrote:
<blockquote
cite="http://mid4acda1b40601142200k5a87951do5c67f8ee9a90bb95@mail.gmail.com"
type="cite">
<div><span class="e" id="q_108ccb8feba85fcc_1">Hi all,<br>
<br>
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:<br>
<br>
[cisco1760]<br>
type=friend<br>
context=incoming<br>
host=<a href="http://192.168.0.55" target="_blank"
onclick="return top.js.OpenExtLink(window,event,this)">192.168.0.55</a><br>
insecure=yes<br>
nat=no<br>
canreinvite=no<br>
dtmfmode=rfc2833<br>
disallow=all <br>
allow=alaw<br>
<br>
I have also confirmed while on an outbound calls that both are using
the exact same codecs. sip show channels shows<br>
<br>
pbx*CLI> sip show channels<br>
Peer
User/ANR Call ID Seq
(Tx/Rx) Form Hold Last
Message <br>
<a href="http://192.168.0.55" target="_blank"
onclick="return top.js.OpenExtLink(window,event,this)">192.168.0.55</a>
123456789
4ea2e1314cd 00102/00000 alaw
No Tx:
ACK <br>
<a href="http://192.168.0.58" target="_blank"
onclick="return top.js.OpenExtLink(window,event,this)">192.168.0.58</a>
200
0013c427-f4
00101/00102 alaw No Rx:
ACK <br>
2 active SIP channels<br>
<br>
<br>
Anyone have an idea what's going on?<br>
</span></div>
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