asterisk users - Aug 2004 - DID Questions

I'll try to keep this short and sweet, I don't want to confuse you, or
myself.. So here goes

I had a DID, it went down, I got a new DID (800#) and I changed the
exten=>###-###-#### to the new 800 number in extensions.conf

I'd done lots of changes to the config files and nothing seems to be
doing anything, When I dail the 800 number all I get is a fast busy
signal.  When I log into the CLI, it shows me nothing when I dail the
number. It used to show me pages and pages of info when someone dialed
in. I'm new to asterisk/voip and I'm lost.

Seems like your provider hasn't configured the DID properly and its not even
hitting your system. Check with them.

Mohammed Salim
EZZI Telecom

 

 

-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Mike Roberts
Sent: Monday, August 16, 2004 4:25 PM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] DID Questions

I'll try to keep this short and sweet, I don't want to confuse you, or
myself.. So here goes

I had a DID, it went down, I got a new DID (800#) and I changed the
exten=>###-###-#### to the new 800 number in extensions.conf

I'd done lots of changes to the config files and nothing seems to be
doing anything, When I dail the 800 number all I get is a fast busy
signal.  When I log into the CLI, it shows me nothing when I dail the
number. It used to show me pages and pages of info when someone dialed
in. I'm new to asterisk/voip and I'm lost.
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Use sip debug, that should print all the trace info for the call, id you
know the ip or the originating gateway/proxy, you cal also use sip debug
<IP>



-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Mike Roberts
Sent: Monday, August 16, 2004 6:38 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] DID Questions

Is there anyway to test if this call is touching my servers? Everyone is
telling me the DID is fine. But I can't confirm that for sure. And I
don't want to go ahead and start making changes to my config when the
DID isn't even working in the first place.
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I agree with Mohammed. 

There are two ways to handle a 1877 call by the PSTN provider, when some one
dials the number.

1) Teminate the call to another PSTN Number like 1-732-452-9222
2) By redirecting the call to a destination IP Address(i.e the IP address of a
gateway) running similar protocol ex:SIP or H323
   In order to do this the PSTN provider has to convert the TDM call to a VOIP
call with the required translation in the T1 PRIs of their VOIP Access Gateways,
and then forward it to the destination IP.
   Check whether your provider can send it to your gateway as SIP call and not
as H323 call, as all providers will not be able to terminate the calls as SIP.
Most of the providers I know of are having Old CISCO AS5300/5400 equipment that
can only translate abd send calls as H323.

Seshu Kanuri


-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Mohammed
Salim
Sent: Monday, August 16, 2004 11:59 PM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] DID Questions


Which device did you assign the 877 DID to? Because if you call that device
using another device registered to your asterisk box (i.e. your xlite
softphone) then its not hitting your provider at all.  The asterisk sees the
call come from a registered sip device and finds that there is a registered
device that has the assigned DID and routes the call to it, completing it.
However, when you call using PSTN, that's when the provider has to route the
call to your asterisk pbx. And it seems to me, they aren't doing that
properly.

Unless I'm totally not understanding what your trying to say but I've
been
in these situations many times and its usually the provider's fault.


Mohammed Salim
EZZI Telecom

 

 


-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Mike Roberts
Sent: Monday, August 16, 2004 6:56 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] DID Questions

I'm using SIP and I'm not getting anything in the CLI when I call over
the PSTN. But over my Sip Softphone I can call it,  no problem.
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