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On Wed, Aug 03, 2005 at 11:28:19AM -0500, asterisk-users-request@lists.digium.com wrote: > 10. Inter-Tel AXXESS failure: HDLC Bad FCS (8) on Primary > D-channel of span 1 (Gavin Hamill) > Date: Wed, 3 Aug 2005 15:32:48 +0100 > From: Gavin Hamill <gdh@laterooms.com> > Subject: [Asterisk-Users] Inter-Tel AXXESS failure: HDLC Bad FCS (8) > on Primary D-channel of span 1 > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users@lists.digium.com> > Message-ID: <200508031532.48723.gdh@la...

Hello, I have a little trouble when entering into a domU console (xm console mydomU) : i can''t use the CTRL-C sequence to stop a script/command (like a continuous ping for example) Is there any parameter / tip for that ? Arnaud _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com http://lists.xensource.com/xen-users

Hi, I'm putting together a system to manage agents with Realtime, and without chan_agent. In 1.2.13, there's a handy (although marked as deprecated in apps/Makefile) PGSQL application to let me do this: macro queue-addremove(queuename,penalty) { switch(${MACRO_EXTEN:0:1}) { case I: // Login PGSQL(Connect connid host=XXX user=XXX password=XXX

Nothing much to be said.. I backported ringinuse, autofill and the QueueLog application from 1.4.0 to 1.2.14. Any or all may be applied - order doesn't matter. They have received minimal testing but appear to function correctly. As always with these things, don't blame me if they connect your callers to a phonesex line, etc. http://bum.net/patches/ Cheers, Gavin.

Hi, I'm trying to use ChanIsAvail to build a resilient 'dialout' macro. The logic is simple; try Zap/g1 (a group of two E1s), and if that fails, try locating a channel via DUNDi. Here's a massively cut down version to illustrate the problem I'm having. macro dialout ( dest ) { ChanIsAvail(Zap/g1); noop(Value of AVAILCHAN is ${AVAILCHAN});

Hello. I'm currently evaluating the Sipura SPA-841, and snom 190 phones for use in an Asterisk PBX/call center environment. One feature the SPA-841 has, which I can't figure out how to implement on the snom 190, is the "make/accept calls without registration" feature. Or more specifically, "produce a dial tone even if I'm not registered." I would like to set our

Hullo :) Can someone help me untangle a bit of a mess? I'm trying to set up a demo * server to show off how useful it can be to our business (as an IVR system and VoIP backup if our ISDN30s fail). I've not been able to get NT mode working with our InterTel Axxess PBX, so I've resorted to using normal TE mode and working on the basis the people dial one of the ISDN BRI extension

Hello all, I've seen that the application AgentCallbackLogin()has been set to deprecated in version 1.4. So I've done some tests based on the tutorial "queues-with-callback-members.txt" coming with version 1.4. What's not clear for me is what is happening to agents.conf, it seems that it's no longer needed, and I have to define my agents using variables in

I have been impressed with the quality and meagre bandwidth of the G729 codec from Digium. I am in a testing phase of our roll out, we are using 5 Asterisk PBXs in various countries to provide connectivity for our employees, owners and family. As we are testing, and our setup is somewhat complex due to the peculiarities of our connectivity, there has had to be a lot of changes to servers, cards to

using CVS HEAD :) Some config snippets: extensions.conf: [from-ip] exten => 51,1,Dial(SIP/1301,20,t) exten => 52,1,Queue(ddi831,t) exten => 53,1,Queue(marketing,t) [internal] exten => _13XX,1,Dial(SIP/${EXTEN},20,Tt) queues.conf: [ddi831] strategy=roundrobin timeout=10 announce-frequency=0 announce-holdtime=no member => SIP/1301 [marketing] strategy=roundrobin timeout=10

Hi, Just a quickie - if I want to implement an * solution purely for voice (well, and physical fax machines / dialup modems..) on EuroISDN E1s, is there any benefit to the A104u over the A104c? I'm just trying to decide if the extra ?200 for the A104u is worth it :) Cheers, Gavin.

Yep, another list posting on this topic :) All the messages I've read on this are from people experiencing these errors in quiet times - I get them as soon as I plug a port on our TE410P to an Inter-Tel AXXESS PBX.. and I get them continuously... I'm just sticking an * box in between ISDN30e (we're in the UK so euroisdn) and the PBX.. and whilst the telco ISDN30e side works like

On Wednesday 03 August 2005 17:33, Jens von B?low wrote: > Gavin, > > >> Any ideas/advice would be warmly received right now! > > You are not going to like my response... Erk :) > The only way I could get this to work (luckily I had 2 identical sites and > was busy with the upgrade to the gen2 card) was to downgrade to zaptel > 1.0.7. Alas no - just moved down to

Hi :) I have a little 'slow dialling' problem. When I dial, e.g. 200# for the Asterisk 'echo test' demo application from my PBX extension 1010, I see this in the console the instant I press the # key: -- Starting simple switch on 'Zap/65-1' -- Accepting overlap call from '1010' to '200' on channel 0/3, span 3 then exactly 3 seconds elapses, and

Hi, I've used Agents + Queues before with success, but I can't figure out why this trivial setup is not functioning... stage*CLI> show agents 1306 (gdh) available at '1306@internal' (musiconhold is 'default') 1 agents configured [1 online , 0 offline] and the internal context is simply: [internal] exten => _13XX,1,Dial(SIP/${EXTEN},,h) Now, taking this

Hi, just a note to let people know that I had NetShop make me some E1 crossover cables to replace my own dodgy crimpings, and they work perfectly =) The 3 metre version is ?5.58 and their order code is CS000111/3 (I guess the /3 refers to the length..). They're at www.netshop.co.uk - 01753 691661. Cheers, Gavin.

Hi :) We currently have a single * box with 4-port E1 card terminating 60 channels: [PSTN] | | 2 x E1 [Asterisk] | | 2 x E1 [Legacy PBX] What I'd like to have is this: [PSTN] | \______ | | [*1]- - - -[*2] - DUNDi peering between 2 * boxes | | [Legacy PBX] Whereby a call in either direction would be routed either 'straight through' to/from the PSTN

Hi, I've been trying to implement this [1] on 1.2.13 and whilst my twiddlings seem to work, I just wanted confirmation that I'm not doing something really stupid which could cause a segfault under certain conditions. My chan_queue.c addition is this one line: ast_queue_log(queue, qe->chan->uniqueid, outgoing->chan->name, "RINGNOANSWER", "%d", orig);

Am I doing something really stupid in this AEL macro, or is nesting an 'if' inside a 'switch', inside an 'if' not supported in the 1.2 AEL parser? macro stdexten( ext , dev ) { // First determine if the SIP peer is registered here Set(aretheyhere=${SIPPEER(${ext}:status)}); if(${aretheyhere:0:2}) == "OK") {

Hullo :) I'm using Debian's Asterisk 1.0.7 bristuffed (though I'm only using CAPI for ISDN, and not HFC-S cards) and trying to transfer an incoming SIP call from sipgate.co.uk to any other extension. My phones are AT-320s (PA168S 1.43 firmware) whose documentation says to blind transfer, simply dial the number you want to transfer to, and press 'FWD'... This is what