[asterisk-users] Congestion in Outgoing call through PRI

[Richard Lyman]

Octavio Ruiz wrote:
> On Wed, Sep 3, 2008 at 10:33 AM, Richard Lyman <pchammer at dynx.net> wrote:
>   
>> Octavio Ruiz wrote:
>>     
>
>   
>>> On Sat, Aug 30, 2008 at 12:17 PM, Shariq Khan <shariqrazakhan at gmail.com> wrote:
>>> The output of a
>>> CLI>      pri intese debug
>>> at Asterisk CLI before make a test call would be very useful, libPRI
>>> 1.4.7 is just fine.
>>>       
>
>   
>> I am amazed no one else have suggested trying a different phone type
>> like an IAX2 softphone. (if i am right, this will work)
>>     
>
> For me is complete clear that
>
>     -- Zap/1-1 is circuit-busy
>     -- Hungup 'Zap/1-1'
>   == Everyone is busy/congested at this time (1:0/1/0)
>
> the Zap channel is the one which returns the congestion status, not
> the other leg (whatever the technology is).
> Anyway, if he try both options nobody is going to be hurt.
>
> I forgot completely mention (and carefully read their  zaptel.conf
> configuration and see dchan=16 declared rather than hardhdlc=16 )
> that  probably their issue is already solved and documented just right
> here: http://wiki.sangoma.com/Asterisk-FAQ#hardhdlc  Shariq, can you
> tell us your wanrouter + zaptel version?
>
>   
If i remember correctly you also had a Zap/1 PROGRESS and PROCEEDING 
message just above this where it said 'passing to SIP/xxx'.

So, that means it wasn't the Zap side that caused the drop.

Please, just do the test with an IAX2 softphone.  It is *only* a test!