[asterisk-users] Optimizing forwarded and redirected calls by avoiding signaling and media data redirection

Sree Harsha Totakura totakura at net.in.tum.de
Sat Mar 11 08:41:47 CST 2017


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.

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

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