Dial Plan Problem

[Bill H]

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

[Vodia PBX]

Hmm. I just tried it here and here it was okay. Maybe a problem with space characters?

[Bill H]

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

[Vodia PBX]

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

[Bill H]

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

[Vodia PBX]

We are not talking about two-stage dialling here?

[Bill H]

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

[Vodia PBX]

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.

[brandywinetech.com]

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

[brandywinetech.com]

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 ,

[brandywinetech.com]

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