similar to: VoicePluse DID problem

>From samles config. When user change uavarible message from firefly login by IAX got 3 file .WAV .gsm . wav But no sound record. When login by antek 804 GW (SIP mode) record success. but message can't play when someone need to leave voice mail to this box. But when delete .WAV . gsm and leave only .wav its' work. Is problem in asterisk Or client Dome C.

I just upgrade quintum A800 with new SIP firmware ---------- Product Name: Tenor Analog A800 Multipath Switch - 8 ports (Rev. B) Gatekeeper Status: Mini GK Calls Allowed: 8 Feature Bit Status: -PS/+RB/-ER Languages allowed: 1 Serial Number: A002-00308F Ethernet Address: 00-30-E1-00-30-8F IP Address: 10.101.0.10 Subnet Mask: 255.255.255.0 Default Gateway: 10.101.0.1 System Software Version:

Anyone using asterisk with VoicePlus ? Looking for a new provider and wanted to get a little feedback

Hello: I am new to the list. I am trying to set up asterisk with voicepulse. I have a voicepulse username + password, and SIP DID. When I login to voicepulse, I have this under my devices tab: Devices *Login:* Sysxxxxxxx *Password:* xxxxxxxxxx *Context:* VPWS *Connects to:* gw5.voicepulse.com My question is: Do I need a 2.4.x kernel? Currently I am running Debian/stable stock 2.2.x ? Has

I have a linux computer with a sound card, I want to place a call to a cell phone using asterisk and my voiceplus account. I can use kphone (linux app) to call voiceplus, but I am hoping to learn to use some of the cooler features of asterisk. I am lost. On my debian installation there is no man page... Powerful package, I'm excited by the possibilities, but I don't know where to

Greetings, I have been experimenting with Asterisk for a few weeks and finally decided to take the plunge and purchase a few DIDs for inbound calling. Our attempts at IAX/IAX2 connectivity with VoicePulse have been less than successful. We get "Registration Refused" errors from Asterisk whenever we launch the server. The front-line support folks at VoicePulse suggested that we are

Dear Sirs, Anybody ever tried running SIP up against Voicepulse? On their http://connect.voicepulse.com they claim they support both SIP and IAX, but I can't seem to get SIP running. I have as mentioned before on this list - huge problems getting any timing devices running on some of my machines, so IAX is not really an option right now. If I try I get a "Service Unavailable"

We are looking for another solution besides Samba, which will allow our application developers using Windows 95/98/NT to access share on our Sun Box running Solaris 2.6 operating systems. Our supervisor feels the Samba installation process is more complex than what we really need. Any suggestions? Steven Hawks NCR Corporation IT Services - GAD Sales and Business Infrastructure * 1529 Brown St.,

Ladies and Gentleman, I have setup Asterisk and AMP. They are working with out error. But now I need to get everything going. I have Voicepluse and they give sample iax.conf and extensions.conf files but that does me a little good as I am using AMP. Is there some docs somewhere on how to confgure once I have AMP up and running? I am not a telephony guy and alot of this looks like greek to

Dear sir, after 3 month with asterisk i start SIP A-Z provider for test my solution. Please take a look and test Or send live traffic he he http://msarn.com Dome C. ----------------------------------------------------------------

I'm trying to use Jeremy's suggestion of dialling using just the peer name instead of user:pass@peer but I'm running into some really funky issues. It does the same thing with both VoicePulse and another * server I have. [voicepulse] type=peer host=gw5.voicepulse.com trunk=yes user=USERNAME pass=PASSWORD and in my dialplan: Dial(IAX2/voicepulse/${EXTEN:2}@VPWS,90,r) The log shows

I am wondering if there is a way to create a SIP/IAX group of outgoing lines like Zap groups. I am currently using the following method, but would like to use features such as ?g2? that would list all the accounts for a SIP or IAX connection. exten => _1NXXNXXXXXX,1,Dial(SIP/account_name:Password@gw5.voicepulse.com/${EXTEN }) exten =>

hello, after playing with an asterisk configuration for voip for a few weeks i'm trying to get outbound dialing with voicepulse going - i've cut down the asterisk to a very minimal install (1 SIP client) to try to localize the problem. The SIP client works fine (SIP and * on the same NAT) and could access the demo from samples before i removed it, and can call itself - so i am

I can ping it just fine. I am on gw5.voicepulse.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040615/054c83f4/attachment.htm

Hello metafor users, I'm using metafor to perform a single-effect summary estimate of the raw proportion of patients experiencing a post-operative complication, and I'm interested in seeing if this proportion differs between the three most commonly used surgical techniques. The software is working as expected, but I would like to double check on the interpretation of my mixed-effect model

I came across a problem when one of our clients was not able to access any of the servers on our network. This person has never connected to us before and now for this first time was trying to do it from his home is Houston, TX using earthlink cable service provided by Time Warner. All this information, I think, is important because when I started examining my shorewall logs I found out

Has anyone been able to make an html template for the voicemail emails. We would love to customize them beyond just plain text. We have dome some Google searches and have not been able to come up with much. I believe that Switchvox has customized the voicemail email into html. Has anyone ever tried this? Thanks, /Josh -------------- next part -------------- An HTML attachment

All, I've almost got my Asterisk PBX setup, but I've having some problems with the VoicePulse IAX trunk. On outbound calls, when dialing a PSTN number through the IAX2 trunk, music on hold (moh, using the m option in the dial command) does not work. The console states that "stop sound" on IAX2 channel. Ring works, but only without the r option. MOH works when trying to dial a

I am looking for a SIP Soft Video phone, which I can use with Asterisk. If you have one installed (regardless if free or purchased) please tell me which one, the settings in Asterisk and your experience with it. bye Ronald

The Register is carrying a article written by Kevin Poulsen of Securtiy Focus, calling asterisk "..the most powerful tool for manipulating and accessing CPN data.." > http://www.theregister.co.uk/2004/07/07/hackers_gut_voip/ I hope NuFone doesn't drop asterisk-set-able callerid's after this article; i've been wanting that feature from voicepluse for a long time.