Any chance you can give one of us access to this system? Best thing to
do would be to join #openzap on irc.freenode.net.
-MC (IRC: mercutioviz)

On Mon, Mar 16, 2009 at 9:53 AM, Mark Tabron
<mark.tab...@rnid-typetalk.org.uk> wrote:
> Quick update on this. We've had the Euro ISDN line checked by BT and it all 
> checks out ok - engineers were able to originate and make calls into the 
> equipment on the end of the line our comms room.
>
> So, it looks like either Wanpipe / FS can't use the circuit but do report it 
> as being up. Changed all the usual stuff like patch cables so I'm really at a 
> dead end as to what this could be.
>
> Any ideas? Pastebin debug output is in my reply below.
>
> -----Original Message-----
> From: freeswitch-users-boun...@lists.freeswitch.org 
> [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Mark 
> Tabron
> Sent: 13 March 2009 14:16
> To: freeswitch-users@lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Problem dialing out via E1
>
> I've not used Asterisk or Yate before. I've picked this project up from 
> another colleague who is on long term leave, but I know he did look at 
> Asterisk before deciding FS was more suited to our requirements (replacement 
> PBX for an ageing Meridian).
>
> Thanks for the reply and pointers towards debugging. I've uploaded our output 
> as directed from Openzap dumps plus the complete FS debug that appears when 
> placing an outside call. Hopefully it can help to provide a possible answer!
>
> http://pastebin.freeswitch.org/7751
>
> Will setup an IRC client and see if I can log onto the channel.
>
> Thanks again!
>
> -----Original Message-----
> From: freeswitch-users-boun...@lists.freeswitch.org 
> [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael 
> Collins
> Sent: 12 March 2009 16:50
> To: freeswitch-users@lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Problem dialing out via E1
>
>> My first post to the list. I'm a bit of a newb to FreeSwitch (and linux) so
>> apologies if some of my terminology isn't quite correct.
>
> Welcome to FS! Just out of curiosity, have you ever used Asterisk or YATE?
>
>>
>>
>>
>> Recently had a 9 channel ISDN30 (euro - q931) installed by BT (UK). We've
>> hooked it up to our FreeSwitch setup with a Sangoma A101 card. Light on the
>> card is green and wanrouter is installed and up in TDM_API mode, with the
>> connection status showing as connected.  Configured Openzap for 9 b and 1 d
>> channel as described in Freeswitch Wiki. Then created a diaplan to fire off
>> any calls preceded by 9 to the next available openzap channel.
>
> Looks good so far...
>
>> The problem I have is when I initiate an external call (using 9xxxxxxx) from
>> an extension I can see Freeswitch allocating the call to the next available
>> channel but then the just sits there and times out after 1 minute. With the
>> cause stated as ORIGINATOR_CANCEL (guessing this is the time out)
>
> okay, some debugging info will be useful. Please read this wiki page first:
> http://wiki.freeswitch.org/wiki/Reporting_Bugs
>
> It has lots of useful information for how to gather log information,
> how to use the pastebin, etc.
>
> Specifically for this issue you'll need to use the pastebin because
> there will be so much information. Here are some pointers:
>
> To see what's happening with openzap you'll need to use the "oz list"
> and "oz dump 1" at the command line (CLI). You'll also need to turn on
> debugging so that PRI messages show up. You'll need to capture the
> output on the CLI and put it into the pastebin.
> (http://pastebin.freeswitch.org).
>
> Welcome to the wonderful world of telephony debugging!
> -MC
>
> P.S. - We have a few IRC channels where you can join to get more
> real-time support:
> #freeswitch and #openzap on irc.freenode.net. (More details are in the
> wiki page I mentioned above.)
>
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