[asterisk-users] Issue dialing out
Andre Goree
andre.goree at gmail.com
Sat Jun 15 15:24:21 CDT 2013
On Sat, Jun 15, 2013 at 4:03 PM, Daniel Tryba <daniel at tryba.nl> wrote:
> On Sat, Jun 15, 2013 at 03:02:41PM -0400, Andre Goree wrote:
>> Setting the CID did not work, unfortunately :(
> [...]
>> I'm going to try another number that we have through them in hopes
>> that it'll complete and I'll let you know if that works. Do you have
>> any other suggestions on what you think they might be filtering by?
>>
>> In the trap given to me by the company, they show our system issuing a
>> "disconnect" from our end, rather than their end dropping the call.
>
> Do a "pri set debug" (or whatever it is called in 1.4 (zap?)) Zap/Zap
> bridging should work, it did on my PRIs and still does with DAHDI. Only
> thing I can think of is the TON/NPI might be a problem (but doubt it
> since SIP/Zap works).
>
>
Thanks so much for your suggestions.
I'm running 1.0.x (yes, archaic, and in fact my actual task is
migrating this system to asterisk11+Freepbx -- very fun in and of
itself without regards to this issue...but I digress), and so I needed
to run "pri debug span <span>", which I've now done. I attempted the
call again have pasted the debug output here:
http://pastebin.com/cHHnMfh6
I can't thank you enough for your assistance, and I understand if you
wouldn't want to go through the debug output as it's LONG -- though
I'm thinking most of the pertinent info as towards the end.
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