[asterisk-users] question on SIP and call manager

[Danny Nicholas]

On Asterisk 1.4, Call doesn't line Channel: A&B.  
You could put the second dialplan snippet into a context and do your
callfile like this:
[callccm]
exten => s,1,Dial(SIP/CCMMAIN,10,KkTt)
Exten => s,n,Dial(SIP/CCMSLAVE,10,KkTt)

--
Channel: SIP/104
CallerID: SIP/104
MaxRetries: 1
WaitTime: 60
retryTime: 5
Context: callccm
Extension: s

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jerry Geis
Sent: Thursday, October 15, 2009 7:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] question on SIP and call manager

>
> Here are two ways to address this
>
> 1. Dial(SIP/CCMMAIN&SIP/CCMSLAVE) - this tries both at once
>
> 2. exten => s,1,Dial(SIP/CCMMAIN,10,KkTt)
>    Exten => s,n,Dial(SIP/CCMSLAVE,10,KkTt)
>
> CCMSLAVE only gets called if no one answers CCMMAIN in 10 seconds (2-3
> rings)
>
>   
Danny thats good to know for extensions.conf
but
I am using call files.

echo "Channel: SIP/CCMMAIN/5551212" >  /tmp/call
echo "Context: smvoice-test" >> /tmp/call

Can I do the Channel: SIP/CCMMAIN/5551212&SIP/CCMSLAVE/5551212
in the Channel for the call file?


Jerry


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