<br><font size=2 face="sans-serif">__________________</font>
<br><font size=2><tt>Back in January I started having a problem with my Sipura (and there was<br>
at least one other on the list with the same problem) that if I answer<br>
an incoming call (via X100P) on line 1 of my Sipura, the caller cannot<br>
hear any voice from the internal extension. If the internal user puts<br>
the external user on hold (via flash hook) and returns, both directions<br>
of audio are fine.<br>
</tt></font>
<br><font size=2><tt>Line 2 never has had this problem. For the meantime, I switched the<br>
internal phones so that my wife's favorite phone is line 2 and I told<br>
her to not pick up with line 1. Not a very permanent solution :)<br>
</tt></font>
<br><font size=2><tt>NAT is not an issue as the Sipura and * are on the same network. Is<br>
anyone else having this problem? It looks like other people are using<br>
Sipura (I saw one user with 30 of them ?!) and am surprised that nobody<br>
else is complaining about this problem. I am willing to step through<br>
some sip debug if anyone is interested in the output.<br>
</tt></font>
<br><font size=2><tt>* version: Asterisk CVS-02/08/04-22:22:57<br>
Sipura firmware: 1.0.31 (just upgraded tonight to see if the problem<br>
would go away)<br>
</tt></font>
<br><font size=2><tt>Relevent config sections:<br>
</tt></font>
<br><font size=2><tt>--8<-- sip.conf --8<--<br>
</tt></font>
<br><font size=2><tt>[cordless1]<br>
type=friend<br>
username=cordless1<br>
secret=xxx<br>
host=dynamic<br>
context=cordless1<br>
dtmfmode=info<br>
mailbox=1234<br>
canreinvite=no<br>
disallow=all<br>
allow=alaw<br>
</tt></font>
<br><font size=2><tt>[cordless2]<br>
type=friend<br>
username=cordless2<br>
secret=xxx<br>
host=dynamic<br>
context=cordless2<br>
dtmfmode=info<br>
mailbox=1234<br>
canreinvite=no<br>
disallow=all<br>
allow=alaw<br>
</tt></font>
<br>
<br><font size=2><tt>-- Chris<br>
_______________________________________________</tt></font>
<br>
<br>
<br>
<br>
<br>
<br><font size=2><tt>I had the exact same problem with a Mediatrix 1102....doing a flash hook brought both sides of the conversation together. I found out that my sip.conf file had GSM as the first priority codec and the 1102 doesn't support GSM. I kept that the same but put a "disallow = gsm" statement in my sip entry for the 1102 so g.711ulaw would be the first negotiated codec. That fixed the problem.</tt></font>
<br>
<br><font size=2><tt>VZ</tt></font>
<br>