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Comtec Neil

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About Comtec Neil

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  1. 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
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