similar to: using broadvoice and vonage hardware withAsterisk

Re-sent because had wrong subject line on first post, sorry. I have just installed * for a small office (P4 3Ghz, 1MB RAM, RH9). It's replacing an analog PBX, and for now, all incoming calls arrive on 10 FXO's. Outgoing calls are via Voicepulse. Phones are SIP, Cisco 7940G's. My problem is random disconnects on both incoming and outgoing calls. The phones are behind a firewall;

Hello Marc, Could you post the install details for the Diva PCI? I have a Diva Pro PCI 3.0 and I am looking to get it working... I am running Debian Testing. So far it appears there are a few things floating around but I am not certain which direction to go. I know I need to load the firmware first. I did manage to load the modules ok. Thanks, Greg -----Original Message----- From:

-----BEGIN PGP SIGNED MESSAGE----- Greetings, I've just about got Asterisk up and running and am wondering the following. Currently, I subscribe to both Vonage and Broadvoice and as such, I've got a Sipura and Cisco ATA186. Although I'm sure this is expressly prohibited somewhere in my service agreements, can I reprogram these devices to access my own asterisk server rather than

Hi all. I've been lurking here for a couple weeks just trying to get an idea of how to install asterisk. I'm running Debian with a custom kernel version 2.4.18. I think I've got all the dependencies installed - debian packages readline4 and openssl are installed. It's a very barebones install with only the packages relevant to building a kernel and asterisk installed. The

I'm getting these "calls" at 16 and 46 minutes after every hour. The SIP phone rings, and if we pick up, we get a dial tone. If we don't pick up, we get the dial tone in a voicemail message. An analog phone connected to the incoming POTS line doesn't ring (whether or not * remains connected to the line). It's like the horror movie where the babysitter is getting

My compile of asterisk is bombing out in /usr/src/asterisk/res . I'm running debian with kernel 2.4.21 (custom). Zaptel and libpri appear to make okay. Any suggestions? When I reboot I don't see any indication that the zaptel modules are loading. I don't know if the final bits of a successful asterisk make do anything with that so I'm not sure if I have deeper problems than

Hi, I am sorry if this message has been reposted, but for some reason I am having problems with posting it. I configured asterisk and zaptel modules with fedora2. I want to be able to load the zaptel wcfxo and wcfxs modules. For now I will use only the Wildcard TDM400P card. I am able to load the modules but I cant configure them using ztcfg or zttool because the tools are compiled to use the

My users are reporting that some voicemail messages are being cut off in the middle of being played back. The recordings are OK (they play fine when forwarded to e-mail, and they can often be accessed OK during a later call to voicemail). I found nothing in the archives on this -- ideas anyone? RH9, P4, CVS-HEAD-09/02/04-08:44:34, aggressive echo suppression turned on. Jim Shilliday IT

Does anyone currently use Vonage with Asterisk? I've tried to set it up but it looks like Asterisk (at least the version that I have) does not handle well the SIP call dialog, sending a BYE with the wrong tag. As a result, when I hang up, Vonage sends back a 400 Bad Request and the call on the PSTN side does not hang up. I know that Vonage does a lot of nasty stuff which impacts UA's

Hello everyone. I've seen postings for connecting asterisk to vonage but I'm still having trouble achieving that. I have a vonage softphone and I'm trying to register to vonage using asterisk. I have not had any luck. I am behind a firewall. I've successfully gotten xlite to connect and work from the same network. When I change the port setting in [general] to 5061, I am able to

hello I am going to be setting up a stun server on windows how do I enable it to work withasterisk? thanks hank -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050628/6995e8d5/attachment.htm

Hi all, I'm just wondering about these VoIP services -- do you have to sign up one account -per- client that will be using the service? I've got multiple extensions behind my Asterisk box, and I want to be able to allow all my staff to place calls via the provider. So if I sign up for one account, will multiple users behind my Asterisk box be able to make calls, using that same

Hi, I'm trying to get Asterisk to connect to Vonage (softphone acct) to allow me to place and receive calls. I have successfully configured Asterisk to route inbound calls and send them to the correct extension, but I can't get outbound calls to work. I have Asterisk successfully registering with Vonage, but when an INVITE is sent out, I get a "404 Not Found" back from Vonage

i have tried to connect my asterisk server to vonage like this: Sip.conf: register => 1<yournumber>:<secret>@atlas-east.vonage.net:5060 [vonage] type=friend username=1<yournumber> secret=<secret> host=atlas-east.vonage.net port=5060 allow=all maxexpirey=15 dtmfmode=inband fromuser=1<yournumber> fromdomain=atlas-east.vonage.net canreinvite=no nat=yes

This for the archives in case it may help someone: I was able to upgrade two Cisco 7940's from firmware P0030301MFG2 to SIP 7.1 as follows: 1. Installed 7.1 images from the Cisco zip file to the TFTP server. 2. Specified "image_version: P0S3-07-1-00" in SIP<MAC>.cnf and SIPDefault.cnf 3. For the older of the two phones, renamed P003-07-1-00.bin to P0S3-07-.bin, making it

I stand corrected! However you do get my point ... The bigger the company the worse it is. Having to deal with these guys is a nightmare. The company that brings me out in spots is Rogers Cable (24/7). They have this electronic air-head called "Gertrude" or something, (an android) who can't understand the word "NO" and has trouble with "YES" (actually like my

Vonage doesn't sell just a softphone account -- or at least they didn't about six months ago when I was a Vonage customer. But they do allow a softphone as an add-on to an ATA-based account. Because the softphone account works with openly available soft clients, it also works with asterisk. The big "secret" is that they use port 5061, rather than port 5060. > > I

My previous post hasn't even made it to the list yet (am I being moderated?), but I got Vonage's Softphone service working for inbound and outbound calls. Keep in mind that there's currently no perceived limit on simultaneous inbound calls, which makes this a wonderful solution for Asterisk (at least for my use). Below is a sanitized snippet from my working sip.conf; your mileage may

I just got setup with vonage.ca with the motorola ata unit.. I fired up ethreal and checked out what's flying over the network... The sniff below would lead me to believe that it might be possible to have asterisk spoof the User-Agent field and register itself? Any thoughts/feedback? Thanks. > > No. Time Source Destination Protocol Info >

I setup a vonage account last year, and cancelled it last night when I put my asterisk box together and signed up for a Broadvoice account to use with it. Now I would like to use my Linksys router as an MTA, but realize it is still programmed with all of vonage's proprietary information and I do not know how to clear it. I understand that just pushing the reset button will not do it.