Not sure if I can give access to the system externally. I know our security policy doesn't allow for stuff like that though.
I'll pop on to the IRC channel - thanks for the help so far, I'm really keen to get this working after tinkering for well over a week with it! Mark. -----Original Message----- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael Collins Sent: 16 March 2009 17:16 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1 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.) > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > Save paper - don't print this email unless you need to. > > -------------------------------------------------------------------------------- > NOTICE from RNID Typetalk > > This communication contains information which is confidential and may also be > privileged. It is for the exclusive use of the addressee. > If you are not the addressee, please note that any distribution, > dissemination, copying or use of this communication or the information in it > is prohibited. 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