[Asterisk-Users] Cisco 7960 and short delay before voice starts after ring.

[Bisker, Scott (7805)]

I think what James is referring to is the delay once the call already been dialed.  It's not specific to Ciscos, as I'm experiencing the same problem on my polycom phones.  Must be SIP related.

The problem is that once a call is dialed, when the remote party picks up the phone, the first half second is cutoff.  The remote party won't hear the first half second of the call.  I had this happend several times in the last few days.  I've also had a few complaints from users recently.  Here's what it looks like.

SIP phone dials 555-1234 (outside line via PRI)
555-1234 rings
555-1234 answers and says "Hello"
SIP phone hears "o"  or nothing at all.

If 555-1234 is slow to say something, then everything is heard fine.

Caveats.  echotraining and echocancel are enabled on the PRI, however, similiar Zap calls are not affected.





-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Rich Adamson
Sent: Wednesday, March 03, 2004 8:11 AM
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Cisco 7960 and short delay before voice
starts after ring.


> When calling out on a Cisco 7960 there is a short delay before the call 
> gets setup and the other side can hear your voice.
> Anyone know how to compensate for this effect?

Sounds like the 7960 has not been configured with a dialplan that supports
your * dialplan. Look for the dialplan.xml file on your tftp server and
check its contents. Should look something like the following:

<DIALTEMPLATE>
    <TEMPLATE MATCH="0"  Timeout="1" User="Phone"/> <!-- Local operator-->
    <TEMPLATE MATCH="911"  Timeout="0" User="Phone"/> <!-- Local numbers-->
    <TEMPLATE MATCH="3..." Timeout="0" User="Phone"/> <!-- Corporate Dial plan-->
    <TEMPLATE MATCH="4,4......"  Timeout="0" User="Phone"/> <!-- Local numbers-->
    <TEMPLATE MATCH="5,4......"  Timeout="0" User="Phone"/> <!-- Local numbers-->
</DIALTEMPLATE>

The first entry, above, says if the user dialed "0", then wait for one second
to ensure they didn't dial something like "0-555-1212". If no other digits
dialed, the 7960 is supposed to send "0" to asterisk after that 1-second timeout.

The third entry says my local * extensions are four-digit numbers starting with
a "3". If the user dial 3111, the 7960 should immediately send that to * (no
timeout).

Rich


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