Syd:

CallerID Name is a funny animal.

I'm not 100% sure that your provider's explanation makes a lot of sense to me because most of the signals sent are digital but allow me to give this little backgrounder on CallerID Name

First of all it is important to note that we are very lucky in Canada. The way the CallerID name system works here is much different than it is in the US. In Canada CallerID Name information is set with every call. In the US it is stored in central databases. Unless you have access to those databases (or your provider will update it for you) you are out of luck when it comes to changing the CallerID Name.

Ok, back to Canada. Chances are that your provider is using ISDN-PRI trunks to their provider. There are 2 ways that CallerID Name can be passed with ISDN-PRI. The first method, commonly used by Nortel DMS switches (think Bell, Allstream), is by passing the name as a parameter in the call-setup message. The other way, commonly used by Lucent 5ESS switches (think Telus), is by passing the name as a Facility IE usually in the progress message.

So the problem is *likely* that your provider's equipment is either configured for the wrong switchtype, doesn't have the CallerID name configured correctly, and/or doesn't support sending CallerID name in the same fashion as their provider.

Asterisk supports sending CallerID Name using both methods. The method can be set via the facilityenable parameter in zapata.conf. But this is only if your connections to your provider are PRI.

So the reality is that there is nothing that you are doing wrong with your CallerID Name via SIP. The problem is just that your provider and their provider are likely either configured wrong or incompatible with each other when it comes to CallerID Name

Regards,
Bill

PS - HAPPY NEW YEAR TO ALL !!!

----- Original Message ----- From: "Syd Carter" <[EMAIL PROTECTED]>
To: "TAUG" <asterisk@uc.org>
Sent: Sunday, December 30, 2007 1:39 PM
Subject: [on-asterisk] Suck it up - No Caller ID - comments?


Hey all.
My caller id _name_ info is not coming across. According to my provider, "we send callerID correctly however the ISP's have old equipment that don't understand the new digital signal therefore they can't understand the call display". I've check my sip dialog and know that I'm sending name and number. I never had this problem with my previous voip service provider. Can anyone provide me with additional insight into "the new digital signal" versus the old one?

Thanks.. Syd


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