Bill H Posted August 17, 2007 Report Share Posted August 17, 2007 I have been trying for a hour or three to get a particular Dial Plan to work for me. I just cannot seem to find the correct combination. The Object: To have *82 placed ahead of an 11 digit telephone number. The Pattern: Dial 9 and an 11 digit USA telephone number. The Replacement: Dial *82 and then the 11 digit telephone number. ------------------------------------------------------------ I tried 9* as the Pattern and sip:\*\1@\r;user=phone as the Replacement. The \* is supposed to use the * as a literal STAR in the dial plan. It just does not work. Will the PBX accept a literal at this point? I also used sip:77777777\1@\r;user=phone as a test Replacement and it did dial 77777777 and the 11 digit telephone number OK. Then I tried sip:\1\*82@\r;user=phone as a test Replacement and it did the 11 digit telephone number with the *82 appended to it. It looks like the literal can be added after the \1 and not before it. Am I missing something here??? Bill H Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted August 20, 2007 Report Share Posted August 20, 2007 Hmm. I just tried it here and here it was okay. Maybe a problem with space characters? Quote Link to comment Share on other sites More sharing options...
Bill H Posted August 20, 2007 Author Report Share Posted August 20, 2007 Hmm. I just tried it here and here it was okay. Maybe a problem with space characters? I have all the characters packed in tight. I don't see any spaces in the string. Bill H Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted August 20, 2007 Report Share Posted August 20, 2007 When using exactly that dial plan, I get: [5] 2007/08/20 14:01:20: SIP Tr udp:192.168.8.118:5060: INVITE sip:*2123653222@192.168.8.118 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.100:5060;branch=z9hG4bK-80b94d59b29bc3aa6ca9c0e7fb5297f2;rport From: "Susi Sorglos" <sip:41@laptop.pbxnsip.com>;tag=50395 To: <sip:*2123653222@192.168.8.118> Call-ID: 73673963@pbx CSeq: 20439 INVITE Max-Forwards: 70 Contact: <sip:41@192.168.2.100:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.0.2085 Content-Type: application/sdp Content-Length: 274 Quote Link to comment Share on other sites More sharing options...
Bill H Posted August 20, 2007 Author Report Share Posted August 20, 2007 When using exactly that dial plan, I get: [5] 2007/08/20 14:01:20: SIP Tr udp:192.168.8.118:5060: INVITE sip:*2123653222@192.168.8.118 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.100:5060;branch=z9hG4bK-80b94d59b29bc3aa6ca9c0e7fb5297f2;rport From: "Susi Sorglos" <sip:41@laptop.pbxnsip.com>;tag=50395 To: <sip:*2123653222@192.168.8.118> Call-ID: 73673963@pbx CSeq: 20439 INVITE Max-Forwards: 70 Contact: <sip:41@192.168.2.100:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.0.2085 Content-Type: application/sdp Content-Length: 274 Try dialling *82 and then your telephone number. I either get a "Call Failed" or a dial tone when finished dialling. Bill H Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted August 20, 2007 Report Share Posted August 20, 2007 We are not talking about two-stage dialling here? Quote Link to comment Share on other sites More sharing options...
Bill H Posted August 22, 2007 Author Report Share Posted August 22, 2007 We are not talking about two-stage dialling here? I'm not sure if we are or not. I believe that Verizon can accept the whole string (*8212124567890) without a pause after the *82 input. If thats the case then it would look like single stage dialling. But lets say that Verizon needs a pause, after the *82 input, for about 1/2 of a second. Is that possible? Also, I am still not able to get the original Dial Plan to work. Today I will remove all other dial plans to be sure there is no conflicts. Bill H Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted August 22, 2007 Report Share Posted August 22, 2007 But lets say that Verizon needs a pause, after the *82 input, for about 1/2 of a second. Is that possible? Well, that is the question of the PBX gateway. Maybe the PSTN gateway accepts URI like sip:*82,12345678@domain, then it could insert a pause. But there is no pause sending out the INVITE (well it is one packet). But I don't see a reason why the service provider would need a pause. If you can put such a number on a speed dial of a analog phone, it should also work. Quote Link to comment Share on other sites More sharing options...
brandywinetech.com Posted May 7, 2008 Report Share Posted May 7, 2008 I tried and get the same thing as Bill with Callcentric ,. it puts the *82 at the end of the number and not at the beginning .. y Quote Link to comment Share on other sites More sharing options...
brandywinetech.com Posted May 7, 2008 Report Share Posted May 7, 2008 sip:\*82\1@\r;user=phone this actually works with * and no replacement , but now I want to do 9* and get a 9 in the number everytime , Quote Link to comment Share on other sites More sharing options...
brandywinetech.com Posted May 7, 2008 Report Share Posted May 7, 2008 I guess if I post enough i'll answer my own questions eventually , with callcentric .. 9* and sip:\*82\1@\r;user=phone as the Replacement worked fine , y Quote Link to comment Share on other sites More sharing options...
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