The call volume is going to be very low. If I understand this correctly, I
would create a trunk under Gateways in sipX for my Asterisk system and create
the other end in Asterisk accordingly, rather than calling it an Unmanaged
Gateway.
And to answer another question, yes the sipX and Asterisk
, November 06, 2012 2:40 PM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Cannot Xfer Calls Received from Unmanaged Trunks
The call volume is going to be very low. If I understand this correctly, I
would create a trunk under Gateways in sipX for my Asterisk system and
create the other
Some development on this topic...
Watching the SIP debug in Asterisk, I see that when I try to transfer the call
it actually asks Asterisk to dial the target extension and Asterisk has no clue
how to deal with that since it owns the 1xxx group while sipX owns 2xxx. If I
put in a line that says
When using a SIP trunk you will need to have Asterisk point to port 5080.
On Tue, Nov 6, 2012 at 10:25 PM, Chris Parker cparke...@me.com wrote:
Some development on this topic...
Watching the SIP debug in Asterisk, I see that when I try to transfer the
call it actually asks Asterisk to dial
That part seemed to work, but I kept getting sipx not being found as a peer,
even though my context in sip.conf was [sipx]
On Nov 7, 2012, at 0:10, Josh Patten jpat...@ezuce.com wrote:
When using a SIP trunk you will need to have Asterisk point to port 5080.
On Tue, Nov 6, 2012 at 10:25
Kind of stumped with this one...
I have Vitelity as my SIP trunk, which is configured in Asterisk to answer with
an IVR and perform some other functions.
If a call is passed from my AA in Asterisk or the DID is configured to call an
extension that belongs to my sipX box (Polycom phones), I
An unmanaged gateway is just that. Can I assume that the address for both
systems are on the same subnet? Unmanaged gateways would assume that the
other and knows how to handle the SIP REFER method.
Asterisk as a sip trunking system is not exactly compliant.
If REFER is not supported, then the
The biggest red flags are Asterisk and CUCM
The SIP stacks on these platforms aren't complete and REFER support is
usually lacking. Trunking with these platforms requires a session border
controller to interface with these platforms. How many active calls do you
expect between these systems? If
