similar to: after hours

Ok, the Zhone sucks and the Adtran 750/850 seems to be a little too expensive. Can anyone recommend a decent channelbank that won't break the bank? TIA, -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year

When trying to dial out 982420173 our main number I get the engaged signal before I finish entering the phone number Any ideas ???? Regards Mick

I am not having any luck placing out going calls I dial the number 08 82420173 ( our outside line ) But all I get is engaged signal and log this. Oct 14 08:40:14 DEBUG[16401]: File pbx_wilcalu.c, Line 65 (autodial): Entered Wil-Calu fd=20 Oct 14 08:40:14 DEBUG[8201]: File chan_sip.c, Line 657 (create_addr): Setting NAT on RTP to 0 Oct 14 08:40:14 DEBUG[8201]: File chan_sip.c, Line 548

<FONT face="Default Sans Serif, Verdana, Arial, Helvetica, sans-serif" size=2><div>I am using some of the sample recordings included with asterisk for my conferencing application.&nbsp; They seem to be rather choppy at times, with the overall quality not being quite where I'd like it.&nbsp; I think I saw in a previous mailing list on here that people suggested we

If you ring into * and leave voicemail It does not reset the line Any ideas would be appreciated Regards Mick

When dialling in and dialling my extension, when answered I get " Read_channel ## vpb/1-3: Setting record mode, bridge = 0 WARNING[20499]: File chan_sip.c, Line 1111 (sip_write): Asked to transmit frame type 8, while native formats is 4 (read/write = 4/4) == Spawn extension (default, 1004, 1) exited non-zero on 'vpb/1-3' -- hangup on vpb (vpb/1-3) -- Hungup on vpb/1-3 complete --

Hi all How can I make * ring one phone then if no answer Go to a different extension ?? Any help always appreciated Regards Mick

I would appreciate some help with this. I read the GPL license and basically it says you can do whatever you want with the software (sell, modify) as long as you include the source code, the License and make any changes you make available in the same manner to all others. My questions is this: If I develop an external application (say a Call Center application or a GUI management application)

Is it possible to build a dialplan in which shifting from daytime to nightime is not hour based but phone driven ???

Has anyone got a mobile wireless phone working with * yet ???? Is it possible to use the Cisco 7920 with skinny ???? Regards Mick West

Quick question: My PHP script is now able to connect to the manager port and successfully authenticate using MD5. I would strongly prefer not to do plaintext authentication at all. Would anyone object to plaintext authentication being left out? -- JustThe.net Internet & Multimedia Svcs. [The Fusion of Content & Connectivity] 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve

Hi, I have a Cisco 7920 that I'm trying to get working with my * box. When the phone boots it requests XMLDefault.cnf.xml and SEP<MACADDRESSHERE>.cnf. I assume I set the line number, etc in the latter of the two. However I cannot find any reference to how this file is structured. Anyone know? I assume this is why I'm getting the errors below: Oct 17 19:47:24

Has anyone gotten the voicetronix boards to work with Asterisk, what would it take? Or does anyone know where I can get 4 ports or more fxs PCI cards that do work with asterisk? Brian J. Schrock Network Engineer, RHCE, CCNA Anistone Technologies Phone: 614-798-9106 FAX: 614-573-7165 6926 Avery Rd. Dublin, OH 43017

The demo 1235 extension that Asterisk ships with works fine and the messages are sent to the address I set in voicemail.conf. I guess that means that my configuration is working perfectly so far. But when I set up another extension with a voicemailbox, no mail is sent when a message is left, although I can dial voicemail and listen to the message just fine which I guess rules out voicemailbox

hi, Can anybody pls tell me, how to increase the time gap between 2 digits when you transfer a call. ie, the operator answers the call, and presses hash key to transfer, and then enters the extension number, some times, it timeouts too quickly before the operator enters the whole extension number (may be bcos the operator is slow). I tried the following, but it doesn't seems to be helping

Hi All, This topic has come up before in the Asterisk mailing list many times, so I know that a lot of people have given up in waiting for a FXO card to be approved by the Australian telecommunications authority. My question is: all legalities aside - is anyone using a FXO card in Australia successfully? Thanks in advance. Regards, Gonzalo

I am sure this has been asked before, but why not use Google newsgroup or at least some forum BBS software instead of this cumbersome mailing list process? -- Costas Menico Meezon Software Corp 201-224-8111 costas@meezon.com --

Why does this person have my e-mail address ? ----- Original Message ----- From: <markogift@astriholics.org> To: <hackerwacker@cybermesa.com> Sent: Tuesday, November 23, 2004 1:13 PM Subject: Gift for Mark Spencer > Hello everyone! > > We have been thinking about something that we could do for Mark > Spencer as a holiday gift. We have decided to try to orgranize a

Ok, first off, Asterisk is the coolest piece of software I have EVER had the pleasure of using in my 8 years of running linux !! and I know I haven't even scratched the surface feature wise. Before I get too excited, I wanted to get all you experts to look at the how I implemented my after hours test. The goal is to prevent the phone from ringing afer certain hours, just go to VM.

Hi all ! Have anyone a resource / link for documentation to configure Asterisk to act as a SIP 2 PSTN gateway (ISDN PRI) ? Thx. Regads, Andreas. -- "If you want to pray. Go to the sea." ---------------------------------------------------------------- Andreas Czerniak <cognac@amcs.net> PGPkey http://pgp5.ai.mit.edu:11371/pks/lookup?op=get&search=0xEDB224EC