similar to: ISDN30 Timing Sources (Jon Morgan)

Hi All, Has anyone hooked up * as an extension/trunk of an Avaya system that has around 2 ISDN30e's. Trying to add 100 extensions to one of our systems, but not sure where to start reading. Thanks. -- Kind Regards, Gavin Henry.

I'm trying to keep the DND status of my Snom phones and the astdb in line but I'm stuck on integrating my gui DND button which talks to * using the manager interface (actually it uses Astmanproxy as the gui host is on a different network to asterisk and can't see the Snom's across the network). All's working fine in my Dialplan; when someone dials the code for DND-on or

Quoth Steve Murphy... >Date: Mon, 12 Jan 2009 08:51:03 -0700 > >QUESTIONS: > >Which of the two would you see being useful to you? Obvious comment really but given LEG based CDR, one can determine the 'Simple' data but you can't work it the other way. I'd therefore find LEG based CDR more useful as the granularity (time on Hold, in Queue, Waiting on pre-xfer ring

Quoth: OCG Technical Support <support at ocg.ca> > >Although we've programmed the softkeys per the manuals, they seem to have no >effect (just dead). For example, our 57i is setup like this: I had similar problems and ended up using the speeddial inband functionality. FWIW, my 57i's setup like so: softkey4 type: speeddial softkey4 label: "*Park" softkey4

I have two Asterisk systems that can route to each other via a VPN with firewalls disabled for testing purposes. Each Server can see (tested via nmap) UDP port 5060 on the other. So... I thought that I could simply use a Dial command in Server A's config to place a SIP call to Server B... but it doesn't seem to work. Server A (192.168.1.33) has: exten => *136,1,Dial(SIP/90 at

Is there a SIP header I can set (for Snom and Yealink phones if that's relevant) or any other mechanism to tell a phone to ignore a particular call from it's missed call list? I have bits of the dialplan that ring groups of phones eg: exten => 200,1,Dial(Sip/112&SIP/113&SIP/114) and I don't want such calls being recorded by the phone as a missed call. Calls to the

Does anyone have any suggestions for a decent receptionists phone? Aastra? Grandstream? Something with (potentially) lots of BLFs, large(ish) screen, headset and most importantly the ability to transfer calls? I've installed five Snom 370s that seemed ideal but my client is very very unhappy as the Snom 370 can't transfer a call correctly if there's another call coming in (details

I'd like to be able to generate a Manager Event from the dialplan but can't seem to find a way to do it. Alternatively, trigger and Event when a record in AstDB gets changed. Can anyone point me in the right direction? Thanks. By way of explanation, I've a app that connects to astmanproxy and I'd like it to know when a call group gets put into Nightservice. Putting the call

I've just started deploying these (well the T28P model) after years of Snom issues and they look pretty good (although the documentation is execrable; if you thought the Snom stuff was obtuse Yealink have got them knocked into a cocked hat!). Anyway, for provisioning I use HTTP with a DHCP entry like:- # # Yealink Phones # group { #

cdr_mysql doesn't set the userfield when it's set inside a macro called from a feature (1.4.25, addons 1.4.8). I have a feature code: autorecord => *1,self,Macro,apprecord The apprecord macro looks like: [macro-apprecord] exten => s,1,Playback(beep) exten =>

Hello Happy New Year to all!! I've just completed porting from Asterisk 1.2 to 1.4. I did this by doing a clean install on a new box, and moving over configuration and scripts where needed. All went surprisingly well! Anyway, one lingering issue is that the function key "lamps" on our Snom phones have all stopped working! We're using a mix of Snom 290/320/360 phones and

Don't you just hate it when something was working and when you come to use it in anger it's broken :-( Something in the, fairly, recent series of Asterisk updates has broken DIGITAL call passthrough. I've an ISDN PBX behind my Asterisk Box (PRI ISDN comes into port 1 of a Digium Wildcard and the PBX it connected to port 2 via an ISDN crossover cable). This PBX used to be able to

Can anyone help with the use of regexten? (* 1.4.3) I've got Asterisk creating extensions for my SIP phones using regexten but I can't seem to figure out how to make use of them once they're registered. Here's my dialplan for from-sip (the SIP's default context): asterisk*CLI> dialplan show from-sip [ Context 'from-sip' created by 'pbx_config' ]

This is concerning an Asterisk 1.4.18 server. We have approximately 70 DID numbers. Incoming calls are placed into the "incoming" context (by zapata.conf) and are routed based on the dialed number. I want to do some manipulation (CallerID name override) to all incoming calls before they are routed. I would prefer to avoid duplicating the necessary code in each DID extension stanza,

Hi, Are there any hardware for ISDN30 ? if yes any problem with this ? is i out-of-box like ISDN2 but with 30 linies ? Do I need more than the cable from my teleprowider and a PCI-card ? /HHA _________________________________________________________________ Find high-speed ‘net deals — comparison-shop your local providers here. https://broadband.msn.com

anybody recommend a supplier in the UK for a pri/isdn30 line (other than BT) thanx very much __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

I'm seeing both inbound and outgoing call failures on our ISDN-30 lines that only seem to go away when I do a "zap restart" or in extremis restart Asterisk (1.4.25 with a Digium TE205P and zaptel 1.4.12.1). If I don't restart zapata or Asterisk the problem rapidly get worse :-( The lines are from BT with LCR from Cable&Wireless (I've tried using the LCR bypass code and

I have a legacy ISDN PBX (Network Alchemy Argent Office) connected to Span 2 of a Digium Wildcard TE205P. Recently Calls from this PBX have been failing with ISDN Cause Code 38 (Network Out of Order!). The problem seems to be getting worse and is now effecting more calls than not (although this could just be because I'm aware of it). Once the ISDN PBX has decided the Network's Out of

>> So it sounds like it should "just work". I'll let you know in a few >> weeks time :) > TDM400P to E1/T1 card faxing fails by design. The lack of > synchronisation between cards means it can *never* work with any > reliability. The hardware will not permit it. >> >> (And I know I've asked this before, but any recommendations for a >>

Hi All, I'd appreciate some help on how to implement "call stealing". That is, where you dial a code to redirect any call on the system to your handset. I'm getting rid of my BRI service and I'm trying to replace the functionality of my existing ISDN2e PBX (Cybergear Gold) with VOIP and Asterisk. On my ISDN PBX, the short-code *46 does this. For example, if I take a call on