similar to: Forwarding a call to another FXO port

What kind of channelbank/FXS port are you connecting to? I've seen problems connecting to some of the older versions of the Adtran Total Access 750's. I wouldn't doubt there would be problems on other channelbanks with older firmwares. Of course, no firmware on CAC AB1's I have the AAstra 480, Adtran 750 Channelbank (updated firmware), T100P card, and it worked fine on the

Tim Thompson Commercial Sales Engineer http://www.amatechtel.com (806) 722-2227 Would anyone have an idea on how I would be able to take the mic in on the computer and put it as the "talking party" for a conference room. I would then be able to set up a "listen only" profile for others to get in on. Reason for doing this is for 'shut-in's' for my Church.

I was playing with a Sprint Vision phone recently and noticed when viewing the low level ReadyLink configuration screens that there are references to SIP registrars and the like. Does anyone happen to know if Sprint's implementation of ReadyLink truly is SIP based, and if so, managed to get it to interoperate with Asterisk. If so, it would prove to be an interesting paging mechanism and

I've searched through the lists, but can't seem to find a reference to someone that said they had a time and temperature application for *. Voip-info wiki would be a good place for it as a Festival example!!! Can someone remind me where it's at? TIA. Tim Thompson

I have noticed that when I try to connect multiple G729 VoIP devices into a MeetMe conference that I can only add up to the number of G729 licenses I have. Now I would think that because all the devices are G729, this wouldn't be the case and the only license that would ever be used would be if a non G729 device or Zap channel was a part of the Meetme conference. This is apparently note the

Greetings, I've been having a large number of deadlock issues lately on chan_sip occurring only on VIA EPIA ML6000 boards. I'm curious if anyone else is having similar issues. My Config (have multiple systems all running the same hardware with the same problem) VIA EPIA ML6000 1GB RAM 80GB HDD Various Digium Cards (T1 and TDM cards) Trixbox 1.2.2 (though running stock asterisk code)

I haven't seen anybody so far post a complex fix for the udev problems on FC3 with the latest kernel. On that note, I have a temporary fix to allow zaptel to load somewhat normally. I found that modifying the zaptel script to 1) load, unload, then load the driver modules and 2) insert a pause between modules seems to allow things to work. This assumes you have followed the instructions

We currently have a 15-phone system using Asterisk, a combination of analog phones/Grandstream HandyTone-286 and Grandstream BT101s, and 4 X100Ps connected to analog lines. The system works well except for the occasional echo problem. I have all the echo parameters configured, removed all the extra incoming analog lines except to the PBX, etc. following all the advice on the wiki and on the

Are their any issues with Asterisk and Redhat Enterprise? I have see one or two posts with issues concerning compiling zaptel drivers but that is about it. Just looking for some consensus to if any problems exist with it. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040422/56d03b72/attachment.htm

does anonybody know what is the status of gsm codec in snom phones ? they were some issuses in archives, some problems so i would like to know what is the actual status. best regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ majo at sunteq dot sk

Hi folks! Based upon your experience on the field what wifi sip phone would you reccomend ? A customer asked for a wireless * install and I'm looking for advice, tnx Alessio Focardi [[*] - Interconnessioni Italy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060628/05b2fb30/attachment.htm

Hi, How difficult is to setup and maintain an Asterisk PBX with phones from multiple vendors? Is it even worth considering or is it safer to pick one vendor for phones and stick with them? I am more concerned about proprietary DHCP extensions, firmware upgrades etc..If anyone has any thoughts or experiences they would like to share I would be more than happy to hear from them. Thanks -Ravi

i am just curious how many * systems are in the real world with more than one user. do you run a certain version? you dont update CVS do you? any admins running a system of over twenty? over fifty? over one-hundred? i deal with 3com and nec systems all day (i am cerified in the 3com nbx advanced network telephony, elite voice mail, CCNA, A+, nec ipk, and soon to be "asterisk school of

Greetings, I have seen a few postings in the past regarding the interop of Asterisk and the Cisco 7920 WiFi phone. To date, I have not seen a definitive method to getting the phone working. Assuming someone has this actually working, can that person step up and answer these questions. 1) What Channel is it working with (chan_skinny or chan_sccp)? 2) If code was used that is not a part of a

Hi. There does not seem to be a php53-eaccelerator in standard Centos yum channels, from what I can see. That is a mainstay for us. Has anyone found that any particular php53-eacclerator from other locations play well with it? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Ok, Here is what I got working: A call comes in from a Zap line. 5 SIP extension ring if nobody picks up, the call is transfered to a cell phone number. That works. I not want to add a playback of a file ("Please waite while you are being transfered") before transfering the call to the cell phone. How can I do this? Andre

I tried it w/ mine as well and it hung up on me because I just have Voicemail running not Voicemail2. It seems as though you have Voicemail2 because it's trying to play the Unavialable message. Just a thought though. Does it do the samething w/ [qout-phillyq] exten => 0,1,Voicemail(u1) exten => 0,2,Goto(default,s,1) Tim Thompson http://www.amatechtel.com (806) 722-2227

Just cross pairs 1-2 and 4-5 1 <--> 4 2 <--> 5 Never done an E1, but I think it's the same for a T1 4 wire cable. Tim Thompson Commercial Sales Engineer http://www.amatechtel.com (806) 722-2227 -----Original Message----- From: bam [mailto:bam@cqm.co.uk] Sent: Monday, December 29, 2003 9:12 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] E100P pinouts

> On May 20, 2015, at 7:13 AM, Neil Henning <llvm at duskborn.com> wrote: > > On 20/05/15 08:37, Owen Anderson wrote: >> >>> On May 19, 2015, at 7:32 PM, Sean Silva <chisophugis at gmail.com <mailto:chisophugis at gmail.com>> wrote: >>> >>> >>> >>> On Tue, May 19, 2015 at 4:05 PM, Owen Anderson <resistor at

I used Apache Directory Studio to remove the "RID Set" node and after that a simple "samba-tool computer delete DC2" worked just fine. Perhaps the version of Samba in TurnKey Linux V16.1 has a bug (4.9.5-Debian)? Likely demotion should remove leaf nodes from domain controllers as part of the process. I have now recreated DC2 using TurnKey Linux V17.1 (4.17.6-Debian) and so