similar to: AGI -> GET DATA not working on current stable cvs (anyone else?)

Hello all, I feel dumb asking this, but does anyone have a link to the SIP firmware for the 7912 on Cisco's site? I have a SmartNet contract, but I just can't find the link (you can search for "7960 sip firmware" and find that fast). Thanks for the help, Jeb Campbell jebc@c4solutions.net

I'm replacing a Merlin for a client and they have a PagePal Intercom that I would like to reuse. Here is what I know about it: It has a screw-down wires that goto rj-11 (This was told to me over the phone) that went into one of the Merlin ports. I tried bring it up with fxo_ks and fxo_ls (assuming it was analog and autoanswered) but no luck. I would be happy to replace if anyone knows of

I understand that all work is going into xen3, but I had wanted to note that aoe (drivers/block/aoe) is giving me trouble on xen 2.0.6 (so we can keep and eye on xen3). Specifically I can''t see nor export AOE devices. As a quick background on AOE, it is not IP (not routable, etc), but works with broadcasts and packets to MAC addresses (see http://www.coraid.com). (for anyone who

Hi All, I'm using an X100P to connect to PSTN ( context=from-pstn ). I'm trying to get fax detection to work. Using the simplest dialplan, I cannot get * to detect fax tones: [from-pstn] exten => s,1,Answer exten => fax,1,Goto(ext-fax,999,1) The fax is never detected (ie: Goto never executed) All that I see on the * -vvvvc console is: -- Executing Answer("Zap/1-1",

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

Hi, I'm looking at setting up a small production system - predominantly for voice mail and IVR (with a few extensions and hold music MP3's). I've found a couple of IBM X330 servers, with dual 1.13Ghz P3 processors. My question is; is a dual 1.13Ghz P3 server sufficient to run for real-life demands? I come from a Unix/Mac background, so I'm not swayed by the '3Ghz'

I'm thinking about using asterisk as an IVR system with an existing avaya index system. I've got 2x PRI 30 lines coming in to the Index, and I have 4 spare PRI cards in the Index. I was thinking about using a QUAD PRI card from Digium and having the calls come into the Index then transfer to Asterisk for IVR then back to the Index. That way if we get 60 inbound calls we'd in

I have a very annoying problem, when doing rake db:migrate, I get the error message: Missing these required gems: will_paginate ~> 2.3.11 But the gem is not missing, It runs very well in my app. To make the rake pass I have to comment out the gem in my environment.rb file. environment.rb: config.gem "will_paginate", :version => "~> 2.3.11", :source =>

I have as large customer that would like to repalce all their Avaya PBXs with a OpenSer/Asterisk solution. Now the plan is to replace their 12,000 Avaya desk sets with Polycoms. My question is has anyone successfully used with OpenSer and or Asterisk, if so I would like to talk with you about hiring you to build this in our lab envirnment. Bob G. bobg at techie.com- --

Hi I''m a new xen user. I gather from the mailing lists that this problem occurs with excessive console output. However, I believe dom1 was sitting idle at the login prompt when I noticed top on dom0 was showing python at 99% cpu. xend would not respond to restarts. Reboot fixed it. What debugging output can i give you, if any, if this happens again? sam elstob

Hi all: I've been developing and testing on a tdm400p card and it's been going well. As you probably know, the tdm400p needs an ide power supply, but the dell poweredge 2600 that this card is destined for eventually has all the power supplied on the backplane with no ide cables. The thing is, on the motherboard in the server, there is an ide ribbon connector, and beside that, something

Hello Everyone, I have released AstLinux 0.2.2. There are way too many improvements to list here, but here is a short summary: Linux 2.4.27, iptables, mini_httpd (with PHP & SSL), phpconfig, AstShape traffic shaping, tftp server, OpenSSH, proftpd, Soekris Net4801 and Pentium-MMX and higher x86 support. There is actually WAY more software, but I couldn't possibly list it all. It

Hello all, I hope someone can help me with this old Asterisk version. I have to run this version because of a custom IVR written on it. Porting it would take much too long and we'd have to hire a consultant because of all the hooks it has into Oracle databases and real-time information. We have a brand-new Avaya phone system in place and we will be cutting over to it in late March 2019.

Thanks for the reply John. About 85-90% of what this box has to do is just handle calls, but it also has options to transfer calls to the main phone system, which up to now has been another asterisk box. For example, you can hit 6 to be transferred to the Lost & Found Department. I do have allowguest set to “yes” already, but of course I also have type=peer and the other stuff for a sip

Does anyone Know if tha avaya 4621 IP phone work wiht asteisk? if it work it has featuras working Thanks Ignacio -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051014/d3a65784/attachment.htm

I am not interested in open source solutions. I want to know how much the propriety systems cost in terms of licensing. Specially Avaya now a days per extension. Exclusive or Inclusive of the hardware for 10 agents as noted. Thanks On Thu, Sep 2, 2010 at 8:18 AM, Muhammad Shomail Haider <msh0786 at gmail.com>wrote: > Hi Bruce, > > It all depends what exactly you are in need of. A

Hi ! I am trying to connect Asterisk with Avaya Definity. I use this tutorial to do this http://cyril-constantin.blogspot.com/2008/04/howto-connect-avaya-to-asterisk.html The comunication between avaya and asterisk is fine but without sound. I can call from Asterisk to Avaya and extension ring or Avaya to Asterisk and extension ring too but I cant hear anything Example Asterisk ---> Avaya --

Samba has a shadow_copy module to serve read-only snapshots to windows clients. I tried to use snapshotting in freebsd 5 (haven''t tried 6), but couldn''t get it stable. I''m really looking forward to getting it going on Solaris. Anyway you can read about the format of the link here: http://us5.samba.org/samba/docs/man/Samba-HOWTO-Collection/VFS.html#id2618182 When it

Hi, I've been attempting to write a gui wrapper to launch ssh -X user at machine application I'm trying to launch ssh and connect to it with pipes so that my front end can enter the password if required (either from a cache or by popping up a dialogue box). I've been having problems with pipes though reading from ssh's stdout, for when it asks for the password. Before I go

Hello! We have asterisk connected over PRI no our phone network, so I'm avaya PBX user. Asterisk connects to another avaya system over h323. Connection can be shown as avaya--PRI-asterisk--h323-avaya When I do call as avaya user I see name of remote end avay user, i.e. connected line info. As I see in debug remote side send is as 14:07:29:758 Received H.2250 Message = { 14:07:29:758