SUGGESTIONS Strange inbound DID behavior

[atsak]

On an PIAF 2.0.6.5 FPBX 2.11.0.38 Asterisk 11.7.0

Inbound calls with anonymous caller ID are not being set to the correct DID. Call is inbound from a carrier that uses Metaswitch (Integra). Call is returned with no service unless an any any DID catchall is in place. Any ideas why?

    Line 193922: [2014-12-10 19:08:34] VERBOSE[29115][C-0000c783] pbx.c:    -- Executing [anonymous@from-trunk:1] Set("SIP/IntegraIn-000082f5", "__FROM_DID=anonymous") in new stack
    Line 193923: [2014-12-10 19:08:34] VERBOSE[29115][C-0000c783] pbx.c:    -- Executing [anonymous@from-trunk:2] NoOp("SIP/IntegraIn-000082f5", "Received an unknown call with DID set to anonymous") in new stack
    Line 193924: [2014-12-10 19:08:34] VERBOSE[29115][C-0000c783] pbx.c:    -- Executing [anonymous@from-trunk:3] Goto("SIP/IntegraIn-000082f5", "s,a2") in new stack

SIP Debug as so (sanitized but is a normal header packet - for some reason they send two invites but the to field is correct in both cases)

<--- SIP read from UDP:192.168.1.1:5060 --->
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
From: "Anonymous"<sip:[email protected]:5060>;tag=RCRDCAUUCA0.cav.integra.voip+1+287f16+8200d464;isup-oli=00
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 467885723 INVITE
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bKe0kne0206g5g9l46g4b0.1
Allow-Events: message-summary,refer,dialog,line-seize,presence,call-info,as-feature-event
Expires: 180
Organization: MetaSwitch
Supported: resource-priority
Supported: 100rel
Max-Forwards: 69
P-Asserted-Identity: "Anonymous" <sip:[email protected]:5060>
Privacy: id
Remote-Party-ID: <sip:[email protected]:5060>;party=calling;screen=no;privacy=full
Contact: "Anonymous"<sip:[email protected]:5060;transport=udp>;isup-oli=00
Content-Type: application/sdp
Content-Length: 163

 

[atsak]

So no ideas from anyone then or do you need more information ?

 

[wardmundy]

atsak You probably need to handle these in extensions_override_freepbx.conf with something like:

[from-sip-external]
exten => somedid,1,Goto(from-trunk,${DID},1)

Then set the DID up in the regular way in FreePBX with an inbound route to tell it where to go.

 

[atsak]

Thanks wardmundy - but the inbound DID is encoded in the To field in the header. Why would Asterisk ignore it on anonymous CID calls and work just fine on calls with CID? I'm confused why different treatment is needed.

 

[wardmundy]

I don't think it's Asterisk . I think it's the FreePBX design.

 

[atsak]

This was a bug in Asterisk. An upgrade to 11.15 fixed it.