search for: mutualphone

Original Post ---------------- I have an Asterisk related problem with mutualphone. I can connect to any number with any softphone that I am using (iaxcomm, SJPhone, and a few others). Also I have a Grandstream BT 101. But I cannot call (via Asterisk) to mutualphone destinations. Other destinations go fine. A working phone call (e.g. from iaxcomm) gives the following on the cons...

If this has been covered before - I appologize. We use some Sipura SPA-2000's with the g711 codec and all seems fine (except for the occasional failure to register errors in my asterisk logs - but I will save that for another post). g711 call quality is on par with our Cisco 7960's. However, when using the g729 codec, the call quality on the Sipura device goes downhill on the PSTN side

Hi All, I have been reading through the list quite a bit, and I am going to post this more as a poll than anything else. I am working on setting up a very small business with something that resembles a professional voice system. My idea is to use Asterisk with a SIP provider and SIP clients. I currently have a Vonage account already. So adding the 9.99 a month Soft Phone would be easy.

We are looking for a flatrate long distance provider to integrte with a hosted Asterisk implementation. I would like to have a SIP/IAX handoff to whatever carrier. We want the ability to turn LD to any of our subscribers for a flat rate as well. This request is for USA only. Int'l would be a per minute basis. Any sugggestions are greatly appreciated. Thanks, Ty Carter

Hi, I have an Asterisk Server with a Public IP (No NAT) and I'm trying to register an adapter SIPURA 2000 behind a NAT Linksys Router WRT54G, but It has been impossible. Into the SIPURA Port #1 I have a Termination with MutualPhone and it works perfect. Into the SIPURA Port 2 I have connection to Asterisk Server but I can't reach it. The configuration on both ports is the same. For this reason I'm sure that the problem must be into de SIP.CONF, but I don't know what is the exactly configuration to SIP's beh...

Anyone having problems with LiveVoIP lately? I am seeing failed outgoing calls. Calls that are being routed to wrong numbers. DID's that ring busy. For the pass 2 days I am unable to pass CID. Is anyone else have these problems? Can anyone recommend a Quality VoIP provider?

I think I am missing the whole purposes of *. i see that it can do mainy things, but in laymans temrs I am not sure what it does. I am very proficient in Linux and would like to use * for the following: 1) I would like to get rid of my landline(verizon) and use voip as my main means to communicate on the telephone. I would like to be able to plug in my plain old phone into my linux box and