So I have a SIP phone number e.g 1111 that will reach the ‘from-internal’ context if dial. Inside the context I have some extensions like PJSIP/1000, PJSIP/1001, PJSIP/1002. Is it possible that i could dial 1111*1000 and it will bring me directly to PJSIP/1000?
I’m having difficulty understanding this, although part of the problem may be confusing SP user names with Asterisk (dialplan) extensions.
I suppose a literal answer would be:
[from-internal]
exten => _1111*XXXX,1,Dial(PJSIP/${EXTEN:5})
but I don’t see what that adds over the, common:
[from-internal]
exten => _XXXX,1,Dial(PJSIP/${EXTEN})
which makes me think I have misunderstood.
Both are subject to the security issues of making SIP user names the same as the primary extension for the device.
Most phones will also allow you to dial the full SIP URI,and completely bypass the PBX.
Hi,
sorry, I didn’t make myself clear.
The number is a normal telephone number, which is registered with the SIP trunking provider so people could reach us without having to have sip phones.
If they dial 1111, they would reach the whole context… Could they just dial something like 1111*XXXX and reach the XXXX extension?
If they were going to provide a direct in dialling service, they would sell it as such. In any case, I think Germany is the only country that adds direct in dialling digits after the standard number length; most companies include it within the standard number length.
Also, in practice, service providers use the SIP registration contact address in the request URI, which is what determines the initial extension. Were the do provide direct in dialling, the number typically has to be parsed out of the To header.
You either need to buy a service with direct in dialling, or use an automated attendant, i.e present secondary dial tone, from your dialplan, after the call has been answered, and read in additional digits. Background() and WaitExten, are typically used for this. The caller will need to wait for the secondary proceed signal, before keying extra digits. (People using analogue DTMF phones may be able to dead reckon its time by including pauses in the dialling.
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I’ll put that slightly differently.
The answer is yes, but it is an additional service you will have to buy. Generally one would expect it to be offered by ITSPs that provide services for PABXes, but probably not for those that provide services for individuals.
There will be no delimiter. The digits will be part of the PSTN number, and, in most countries, the overall number will not be distinguishable from a normal PSTN number. (In fact the ITSP may be using a similar service, but over ISDN, to get the numbers that it sub-allocates to customers.)
It is unlikely that it will be presented in the request URI, so you will probably have to parse it out of the To header (normally the request URI user will be what you provided in the register request).
I assume your service provider connection is SIP. If it is ISDN, you can expect the service to be available, and the digits will be in the incoming number, and therefore directly usable as an Asterisk extension.
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