Thanks
Dial(SIP/101,SIP/102) was a typo on my part, I had & in it's place.
I've read most of the o'really book, it helped me sort what I've got
so far.
The entire internal context I have is as follows:
[globals]
SIPID=XXXXXX
ALL=SIP/UP&SIP/DOWN
[inbound]
exten => ${SIPID},1,Answer()
exten => ${SIPID},n,Dial(${ALL},30,tdr)
exten => ${SIPID},n,VoiceMail([EMAIL PROTECTED])
exten => ${SIPID},n,Hangup()
exten => ${SIPID},103,VoiceMail([EMAIL PROTECTED])
exten => ${SIPID},n,Hangup()
It was almost as though when I answered with the UP phone, the DOWN
phone was going to voicemail and never hanging up, because I was never
able to get a dialtone back, but there was slight background noise
like it was on a call and no-one was talking. if freePBX hadn't killed
all my files, I imagine I would have seen an empty voicemail file. I'm
going to recreate it all and try to check this tonight.
If it makes any difference, I am using a grandstream 386 ATA
Thanks
Chris
On 8/6/06, Hadley Rich <[EMAIL PROTECTED]> wrote:
On Monday 07 August 2006 06:36, Chris Hembrow wrote:
> I am new to asterisk, and learning as I plod along. Currently, I am
> trying to work out how to create a ring group without using AMP.
You should check out the book - 'Asterisk: The Future of Telephony' -
published by O'Reilly it's available to buy or download. It will give you a
good starting point.
> I set my inbound line to ring multiple lines by using
> Dial(SIP/101,SIP/102) but when I answered the call, the lines which
> didn't answer became locked with no dialtone, as though on a call.
That dial line should be Dial(SIP/101&SIP/102) - the comma (or a pipe, |)
separates what to dial from the options to the dial command. typing 'show
application dial' from the Asterisk CLI will get you all the gory details.
> I thought that setting up a ring group might help, but could only find
> references to creating them through AMP.
'Ring Group' is just an AMP term, you are going about it the right way above.
HTH
hads
--
http://nicegear.co.nz
New Zealand's VoIP supplier
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