[Freeswitch-users] Problem dialing out via E1

[Michael Collins]

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.Tabron at 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-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mark Tabron
> Sent: 13 March 2009 14:16
> To: freeswitch-users at 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-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
> Sent: 12 March 2009 16:50
> To: freeswitch-users at 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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