[Asterisk-Users] Outgoing call spool

Bill Leckey bleckey at tpg.com.au
Sun Sep 28 17:44:43 MST 2003


Andrew Joakimsen wrote:
> No, because asterisk cannot deal with the G723 codec, it can only act as
> a "middle man" of sorts between devices that support it.

Ok, that makes sense.  Could I get the ringing somehow if I changed to 
(say) the G711 codec?

Or, is it possible that this could be done by (say) the SIP RINGING 
message?  I believe that while the remote phone is being rung then the 
originating call is currently in a "call up" state, which means a SIP 
RINGING isn't allowed, but I guess I'm wondering if something like this 
might work?

Thanks,
Bill

> 
> 
> 
> 
>>-----Original Message-----
>>From: asterisk-users-admin at lists.digium.com [mailto:asterisk-users-
>>admin at lists.digium.com] On Behalf Of Bill Leckey
>>Sent: Sunday, September 28, 2003 7:03 PM
>>To: asterisk-users at lists.digium.com
>>Subject: [Asterisk-Users] Outgoing call spool
>>
>>I've been playing with the outgoing call spooling feature a bit lately
>>and it all works as it should with the exception of one irritation.
>>
>>I'm  mostly using SIP to talk to the phones and using G.723.1
>>
>>I copy the call file into the spool/outgoing directory and the
>>originating phone rings.  I pick it up and the remote phone rings.
>>However there is dead silence from the originating earpiece.  Is it
>>possible to somehow generate a ring in the earpiece until the remote
>>phone is picked up?
>>
>>Bill
>>
>>--
>>
>>_______________________________________________
>>Asterisk-Users mailing list
>>Asterisk-Users at lists.digium.com
>>http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> .
> 


-- 




More information about the asterisk-users mailing list