asterisk users - Mar 2017 - Optimizing forwarded and redirected calls by avoiding signaling and media data redirection

Hi!

I'm having a setup where my asterisk PBX connects to PSTN via a single
SIP trunk.  Now, when I transfer or redirect incoming calls from the SIP
trunk to another number which is routed through the SIP trunk, my
asterisk stays on the way; it just dials out the new destination number
the call is transferred/redirected to and connects the newly dialed
channel to the existing incoming channel.

Since these two channels are in the same SIP trunk, would it be possible
to tell the trunk SIP server to not involve my asterisk anymore, both
for signaling and media data?  Or is this inherently not possible via SIP?

Regards,
Sree

Hi!

Apparently this is possible; my asterisk server is doing this when my
SIP phone redirects the call with a SIP REFER message.  The phone is
excluded from the call after it transfers the call.

I'll contact my ITSP if their trunk can also do this.

Regards,
Sree
On 03/09/2017 11:03 PM, Sree Harsha Totakura wrote:
> Hi!
> 
> I'm having a setup where my asterisk PBX connects to PSTN via a single
> SIP trunk.  Now, when I transfer or redirect incoming calls from the SIP
> trunk to another number which is routed through the SIP trunk, my
> asterisk stays on the way; it just dials out the new destination number
> the call is transferred/redirected to and connects the newly dialed
> channel to the existing incoming channel.
> 
> Since these two channels are in the same SIP trunk, would it be possible
> to tell the trunk SIP server to not involve my asterisk anymore, both
> for signaling and media data?  Or is this inherently not possible via SIP?
> 
> Regards,
> Sree
>