[Asterisk-Users] Sip phones transfer not working.

[mattf]

Hello,

You can try the doublehash.patch linked on this bug:

http://bugs.digium.com/bug_view_page.php?bug_id=0000885

it makes it so you have to dial two hashes ## in quick succession to trigger
a transfer. It works very well, but the problem is that it won't work with
asterisk 0.7.1 and no one wants to fix it so it will. 

If anyone out there can fix this patch or make it a config option I would be
very greatful.

Thanks,

MATT---


-----Original Message-----
From: Ariel Batista [mailto:abatista at avionica.com]
Sent: Wednesday, January 21, 2004 12:07 PM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Sip phones transfer not working.


I have a Cisco 7960 & IpDialogs that I am not able to use the transfer
button on it.  What happens is that it puts the call on hold and then it
gives you a dial tone.  You can dial but it will not transfer the call.
What we are trying to do is transfer to extension 700 for parking so another
person can pick up the line.  We can not use the # key to do this due to we
have several IVR's that use the # key as part of it's menu.  

Does anyone have a example settings that we can use.  Every thing else works
on the SIP phones.  

I did some search and found a patch.  But I have never added any patches to
my Asterisk server and since I am new to linux and Asterisk I really do not
want to work with patches?  If this patch works has it been put into a CVS?



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