SIP headers

[Comtec Neil]

Hello all, first-time poster so please be gentle!

I've just discovered Vodia and am loving it.  I have an issue with a SIP provider in the UK I'm hoping someone can help me with.  

My SIP provider presents the To: part of a SIP header in a peculiar way.  When a call comes in via SIP it is supposed to present the DDI (Say 01752422954) in the To: part, instead they present the registration username.  It's annoying, but they are a brill provider so I live with it. 

In asterisk I use a little bit of code to process the SIP header:
 

[custom-get-did-from-sip]

exten => _.,1,Noop(Getting DID from SIP header)

exten => _.,n,Set(pseudodid=${SIP_HEADER(To)})

exten => _.,n,Set(pseudodid=${CUT(pseudodid,@,1)})

exten => _.,n,Set(pseudodid=${CUT(pseudodid,:,2)})

exten => _.,n,Set(CALLERID(num)=${CALLERID(num)})

exten => _.,n,Goto(from-trunk,${pseudodid},1)

Is there a similar way of processing the SIP headers in Vodia?  I asked support but I really struggled to articulate my problem.

 

On Vodia, this is the behaviour:

When I set up a DID of 01752422954  I get busy signal when I call 01752422954

When I set up a DID of 0ec7e90dff the call proceed correctly when I call 01752422954
 

Here's my SIP invite:

INVITE sip:0ec7e90dff@178.159.2.99;transport=udp;line=eccbc87e SIP/2.0
Via: SIP/2.0/UDP 193.203.210.39:6060;rport;branch=z9hG4bK-sYn-0-fslogtjv%ho._dyjs!lna!dezu_hggzl
Max-Forwards: 15
Contact: <sip:uaba605f22@193.203.210.39:6060;transport=udp>
From: <sip:07931828049@proxy.voip.co.uk>;tag=55a8a3d28b
To: <sip:01752422954@proxy.voip.co.uk;user=phone>
Call-ID: f8b51a1e-c60e-11e8-9188-1daf01f298ce
CSeq: 113514872 INVITE
Supported: from-change
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL
User-Agent: Synergy/1.9.9.5967
Content-Type: application/sdp
Content-Length: 318
Date: Tue, 02 Oct 2018 06:47:03 GMT

v=0
o=root 1260356252 1260356252 IN IP4 10.200.20.7
s="V4U SBC v2.0"
c=IN IP4 193.203.210.39
t=0 0
m=audio 15620 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

Many thanks,

Neil

 

 

[Vodia PBX]

In the "Routing/Redirection" section of the trunk settings, there are several ways to get this done. There you can set the source to "To-header" (not the request URI). For pulling out your extension in Europe is usually easy because all numbers have a certain prefix that you can put into the "match extension after prefix" mode - for example if 954 would be your extension you could set the prefix to 752422. This way you don't have to set up DID for each and every account and just use a general rule how to find it from the caller-ID.