search for: multivoips

I've been working with the old-style (pre-SIP) MultiTech MultiVOIPs, trying to make them work against chan_h323. With the voips in rfc2833 mode, Asterisk can detect DTMF fine from them, but when it sends DTMF to them, they lock up on the second digit, crying about an incoming fax. Has anyone encountered that problem (and found a solution)? Taking the other route...

Anyone has configued Multivoip 3010 with Asterisk ? I am tring to configure this box to work with asterisk . Thanks Imthiyaz -------------------------------------------------------------------- mail2web - Check your email from the web at http://mail2web.com/ .

In the mailing list archives, I found a message that indicates that the IAXy has the ulaw, alaw, and g726 codecs, but I cannot find anything official on Digium's site about it. The Installation Manual has an example iax.conf file that indicates the ulaw codec, so I know that one is good. But we are thinking about using the IAXy over a VPN, to replace our MultiVoip. alaw and ulaw are

We have an application for Asterisk that will require connecting 144 fax ports into the system. Faxes will route externally over a PRI. The 144 ports are for local fax machines within the building. Not all will be faxing simultaneously. We just need to be able to provide ports in the building to plug in lots of fax machines. The plan is to run an Asterisk server for about 100 phones and these

Sorry if I post in this forum, this may be not the right one, but I hope to find in here some experts which could help me out. I have in one location 8 extensions from a Panasonic PBX ?KX-TD1232? connected to FXO Ports on an MultiVoIP Gateway from Multitech. On the other location I have 8 SP5100 IP-Phones from Micronet as remote extensions. Between the locations I have a VPN.

Hi, Can anybody with previous experience with it guide me on how to setup asterisk with analog e&m to connect it to an old style e&m system which uses 4 pair cables on RJ 45 jacks. All the analog cards I know of use RJ 11 jacks. And there is no choice of modernization of the customer equipment. Cable pin out are as follows: 1. M lead 2. E lead 3. Tip1 4. Ring 5. Tip 6. Ring1 7. SG 8.

Last week I was able to do some debugging of the problem I'm having with IAX2/GSM, residential-grade broadband, and VOIP. To summarize, I am having a great learning experience with * and Zap cards, SIP and IAX2. I hit a wall though, when I registered with iaxtel and tried doing VOIP. I spend the better part of a workday with the jitterbuffer and all sorts of settings and finally started to

I am trying to work out IAX <--> IAX communications with my * box. I have a registration with iaxtel and I thought I would start there for my learning. I am able to call the number for Digium's support line (700-428-6000), but the sound is horribly chopping. Some reading revealed the jitterbuffer settings, so I enabled them in iax.conf. I have the following now: ; Inter-Asterisk