Call forwarding issue

I’d like to be able to call a DID number ( 8178388105 ) and have it forwarded to an outside number ( 12148844386 )

I use SIP for all freepbx calls… no pots or async lines…

I set up an incoming route ( for 8178388105 ) to go to extension 8105. I have 8105 have a follow me setup to 12148844386#

Now if I just call the 8105 extension from my freepbx connected phone, I get forwarded to the outside number just fine.

If I call from an outside number to the freepbx DID number, the outside
number does not get called. The log shows:
( note the Forbidden part )

[2011-09-08 14:49:06] VERBOSE[4989] pbx.c: – Executing [[email protected]:20] Dial(“Local/[email protected];2”, “SIP/OUT_8100/12148844386,300,”) in $
[2011-09-08 14:49:06] VERBOSE[4989] netsock2.c: == Using SIP RTP TOS bits 184
[2011-09-08 14:49:06] VERBOSE[4989] netsock2.c: == Using SIP RTP CoS mark 5
[2011-09-08 14:49:06] VERBOSE[4989] app_dial.c: – Called SIP/OUT_8100/12148844386
[2011-09-08 14:49:06] WARNING[2638] chan_sip.c: Received response: “Forbidden” from ‘“9726013566” sip:[email protected];tag=as32d0694d’
[2011-09-08 14:49:06] VERBOSE[4989] app_dial.c: – SIP/OUT_8100-0000070a is circuit-busy
[2011-09-08 14:49:06] VERBOSE[4989] app_dial.c: == Everyone is busy/congested at this time (1:0/1/0)
[2011-09-08 14:49:06] VERBOSE[4989] pbx.c: – Executing [[email protected]:21] NoOp(“Local/[email protected];2”, "Dial failed for some reason with DI$
[2011-09-08 14:49:06] VERBOSE[4989] pbx.c: – Executing [[email protected]:22] Goto(“Local/[email protected];2”, “s-CONGESTION,1”) in new stack

Seems like there is some sip configuration thing that I am missing.

jack

Does your trunk provider (trunk OUT_8100 in your logs) require the outgoing Caller ID to be set to your own DID?

If so you must force it on the trunk, because otherwise the forwarded call will take on the Caller ID of the original caller.

I am suspecting this is the case because of the Forbidden response you are receiving from your provider.

  1. Try another VOIP provider.
  2. If you’re using SIP trunks, this won’t work unless you forward UDP ports 10,000 to 20,000 from your router to your PBX.

Not really sure what you are getting at, AdHominem, but if the udp ports are not open/forwarded then no calls would be working… clearly here he is having a specific problem with a scenario not a general RTP audio issue.