similar to: Cell phones and DTMF

The owner of my company just asked me for an Asterisk brochure. Has anyone seen such a creature? I know of some really informative websites, but I think a pdf would be priceless at this point. Thanks, Bob McDowell EMAIL PRIVELEGED & CONFIDENTIAL CLIENT COMMUNICATION    *** PRIVILEGED AND CONFIDENTIAL CLIENT COMMUNICATION *** This e-mail message and all attachments, if any, may

I ran into a weird one last night. If I use 'service asterisk start' I have problems (see below). If I exclusively use 'asterisk -vvvvvvvvvvvvc' everything works normally. It happens like this: 1) 'service asterisk start' 2) Use asterisk normally, etc, etc - eventually change something that requires a restart 3) Issue either a CLI 'stop now' or a 'service

My Asterisk server believes that a Digium TE405P and sound are incompatible. Basically, no matter what else I do to the machine it terms of hardware, if the TE405P is installed, none of the playback/background/etc commands work. MOH works fine. So far, I have tried: 1) Seven different PCI soundcards with different chipsets. (Go go Computer Junk Store!) 2) Disabling each and every device

Forget the sound card. It isn't related. The subject above should have read 'TE405P No Voice Problem' or something similar. It appears to be a zaptel timing issue, but I have found a workaround. For those of you just tuning in, here is the story: I have a CentOS/Intel 865 box currently running Asterisk 1.2.7.1 and zap 1.2.5, both compiled from the source available off the main

I think I'm very close to getting IAXModem and Hylafax going, but my current inbound hylafax logs show this: Feb 23 10:09:37.98: [ 3638]: MODEM <Empty line> Feb 23 10:09:37.98: [ 3638]: MODEM TIMEOUT: waiting for v.21 carrier Two questions - 1) Does anyone know what step I missed here? (I.e. please help!) 2) Is there a document I should be working off of? Google doesn't seem to

I have gone through some archive about Nortel MICS (Meridian ?)+ Asterisk Integration but I'm not sure whether same as my case . 70 telephone sets | | Nortel MICS 4.1 --------- Asterisk | PSTN I have read the David Gomillion's Guide and got the idea . However, my plan is slightly different from what he did , I need to use Nortel MICS to connect to PSTN (I

Looks like we will be forced to make a move on our voicemail system as Nortel has declared Meridian Mail an end of life product. Frustrating thing is that it would seem their only reason for it is so they can force our hand to move to Call Pilot. Is there any documentation and feedback out there of how people have used Asterisk as a voicemail system for a legacy PBX? We've got about

I'd like to use native moh instead of with mpg123... for some reason the processes never bloody die. For native moh to not spawn an external player, I'd need to convert the default supplied moh sound files in /var/lib/asterisk/mohmp3 to ulaw and g729 format. Anyone know of a free, easy way to convert them? Thanks, Doug.

List, how can I put fax server functionality on Asterisk? * as a reliable fax server for 500-1000 fax/day (mostly incoming)? Fax server should be like HylaFax, i.e. stable, low maintenance and functionality like receiving fax as email with PDF attachment, sending faxes per WHFC. Faxing with spandsp using bri_stuff (BeroNet/Junghanns quadBRI ISDN cards) shortens some faxes, or faxes loose lines,

This isn't an asterisk question, more of a telco question. Hopefully someone on this list involved with telco can answer it. I have a client who is using asterisk with analog phone lines. There is a bad clicking noise on the line. "Click.... Click.... Click". We have narrowed it down to being the farmer's fencer next door. (And by next door I mean 2 acres away).

Hi everyone, I am fairly new to the idea of VoIP, although I've been reading about it off and on for the last few years. Now it is starting to look mature enough to consider implementing it, but there is one thing that I haven't been able to get a clear answer on... With Vonage, you are using the Vonage network - it is their responsibility to route your call to the endpoint, which is

Hi I was given the task to try and build a VOIP solution to an office building with multiple tenants in all sizes and shapes. Some of them will require auto attendants and some will simply want direct lines to their phones. The question I have is: Can asterisk be configured to handle multiple auto attendants? TIA Paolo

I was just getting to work on fax for my * system, so I thought I would bring everything up to date since there would be some new compilations involved. yum update gave me kernel-2.6.16-1.2069_FC4 but after recompiling zaptel, I kept getting "FATAL module zaptel not found" Chased this for an hour with multiple recompiles and reboots. Finally dropped back to 2.6.15-1.1833_FC4,

I have a very strange problem I'm hoping someone has encountered already. I've scoured the internet for an answer to this one. My phone company provides no disconnect supervision. Hence I'm forced to use the busydetect feature. I have a TDM400 with two FXO ports. If I call from an internal extension to a landline and then hangup the landline Asterisk detects the busy signal

I realize that this isn't the biz list, so I'll make this short. If you have a good vendor of IP hardware that you'd recommend someone check out, please send me their info off-list. My e-mail is 'bmcdowell@federalprotection.com'. Any tips would be much appreciated. Thanks, Bob McDowell

Editing subject line to reflect current status. On 1/26/06, Nilesh Londhe <lvnilesh@gmail.com> wrote: > Since T616 is not answering (and incoming calls are going to Cingular > voicemail after 30 sec,) I suspect the problem focus area is... > > > -- Executing Answer("BLT/T616", "") in new stack > > Is

Is there any known reason that the DISA() routine should behave differently than WaitExten() as far as recognizing DTMF tones? If not, I suspect there's a bug here. Try it yourself--two DID's on our PRI, numbers below let you test each routine: It is my observation that some setups/phones DO and some DO NOT express this variance. --I could not show any variance on a sprint mobile phone

I am trying to integrate a Nortel Norstar system with an Asterisk service using a TDM04B card (4 fxo). So far everyting works from Asterisk to Nortel. The problem is when someone dials from the Nortel PBX to the Asterisk server. Asterisk answers the call and provides a dialtone (with DISA) but appartently the DTMF tones are not passed to asterisk and the call cannot proceed. This only

Hello, I am using Asterisk SVN, a cellular phone, and chan_mobile to run a small home PBX with two analog telephones connected to a Linksys ATA using SIP. It works great (except for some Bluetooth adapter bugs that I am still trying to beat...seems the misaligned audio detection still needs work), but I have encountered an interesting issue. If I am using an automated system that accepts input

Thank you! I run --times when I use rsync (I actually use the -a flag) but the times do not transfer over and if I run rsync dryrun with -i I can see that it wants to transfer the files because of times. When I run rsync a second time with your suggestion the times do transfer over. I don't know why... B ________________________________________ From: rsync [rsync-bounces at lists.samba.org]