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07-12-2007, 11:12 PM | #1 |
Junior Member
Join Date: Jul 2007
Posts: 1
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How do I force A PSTN call for mobiles
Hi,
I have finally managed to configure my Speedtouch 780 to use voxalot and am making UK landline calls through VoipStunt (cos they give you 300 mins a week FREE). However i have NO IDEA what to do when it comes to dialling plans... and I was wondering how I set up my system so that whenever I dial a number beginning with 07 (for mobiles) it uses the normal BT line. Also how would I do the above if I wanted to use a provider that provides cheaper calls than BT (but not a VOIP service). someone like where you have to dial 1282 or something before the number you require. If anyone could give me a step by step that would be great because as i said im really thick when it comes to dialling plans and dont understand ANYTHING that looks like <:0><--007> etc etc (I dont even know where to put it). So a real basic approach and answer would be great. Thanks guys gals.... Carl. |
07-13-2007, 09:08 AM | #2 |
Junior Member
Join Date: Apr 2007
Posts: 4
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I have a Speedtouch 780, and I have set it so I can dial a prefix (9) to send a call via PSTN. Instructions in this thread: The Unofficial Be Users Forum
However, I don't use BT for calls to UK mobiles, as there are much cheaper alternatives. I have just started using vopicheap for all UK landline and mobile calls. The landline calls are free (limits apply) and the mobile calls are 6.4p/min at all times. The relevant 2 dial plans: Pattern (for UK mobiles): _07. Pattern (for landlines): _0[12]. Replacement: 0044${EXTEN:1} Provider: Voipcheap There, that wasn't too difficult, was it? :-) Cheers |
07-14-2007, 12:59 AM | #3 | |
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Join Date: Mar 2006
Posts: 96
Thanks: 8 Thanked 26 Times in 19 Posts |
380 Alternative Service
Quote:
I don't have any experience with doing it, but I understand it would involve creating or editing the dial plan on your ATA. The way sending calls to the PSTN is done with the bundled SIP VoIP services in Japan (I don't know how widespread this is as a standard) is by having the SIP server send a "380 Alternative Service" reply back to the ATA. For example, the bundled SIP VoIP services in Japan don't handle calls to Japanese freecall numbers, of the form 0120xxxxxx (such calls are automatically routed to the PSTN). If I dial a 0120xxxxxx number, the ATA sends this SIP INVITE to the server (as normal): Code:
INVITE sip:0120xxxxxx@ipphone.plala.or.jp;user=phone SIP/2.0 Via:SIP/2.0/UDP 60.36.xxx.yyy:5060;branch=z9hG424F7C3 From:<sip:050xxxxxxxx@ipphone.plala.or.jp>;tag=830B2A235781 To:<sip:0120xxxxxx@ipphone.plala.or.jp;user=phone> Contact:<sip:60.36.xxx.yyy> Session-Expires:180;refresher=uac Content-Type:application/sdp Call-ID:100000002@60.36.xxx.yyy CSeq:1 INVITE Min-SE:180 User-Agent: Fletsphone/2.3 (SV3 03.04; NTTEAST/NTTWEST) Date:Thu, 23 Jun 2007 14:12:27 GMT Max-Forwards:70 Supported:timer Content-Length:147 v=0 o=- 1142531234 1142531234 IN IP4 60.36.xxx.yyy s=- c=IN IP4 60.36.xxx.yyy t=0 0 m=audio 5004 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 Code:
SIP/2.0 380 Alternative Service Via:SIP/2.0/UDP 60.36.xxx.yyy:5060;branch=z9hG424F7C3 From:<sip:050xxxxxxxx@ipphone.plala.or.jp>;tag=830B2A235781 To:<sip:0120xxxxxx@ipphone.plala.or.jp;user=phone> Contact:<sip:60.36.xxx.yyy> Call-ID:100000002@60.36.xxx.yyy CSeq:1 INVITE Content-Type: text/plain Content-Length: 26 Date: Sun, 26 Mar 2006 00:13:15 GMT Alternative Service = PSTN For people who have such ATAs, it would perhaps be useful if a VoXaLot dial plan option could send a "380 Alternative Service" back to the ATA, for seemless integration with the PSTN. |
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