search for: pcphonelin

> Hi! > >> We are in the process of an Asterisk call center deployment using IAX2 >> G711 ulaw softphones. Outbound sound quality is terrible. RESPONSE: This is Bill McCready from PCPhoneline.com . To address your sound quality issues, you may want to check out our VTA1000 Skype+SIP+IAX2 Tri-Mode Gateway which plugs into the USB port of a Windows computer. You can use it with any good cordless or regular phone. This is our very latest version of the VTA1000 and it supports Skype +...

http://www.pcphoneline.com/skype "The VPT1000 is NOT a simple last generation USB phone audio device but is rather a next generation integrated gateway and USB phoneset with simultaneous dual mode Skype and SIP calling support. Skype is not forecast to have "SkypeIn" available until June 2005 but you c...

...continue to use analog phone devices via an FXO/FXS box. However, I am not clear on whether I can expect these devices to provide "call waiting" features and caller ID features to the connected analog phone. Ideally I would use one of the cheaper devices such as that available from www.pcphoneline.com Any clues? A.

Hi all, I have a question that I was hoping someone could answer for me. I would like to find a USB phone that works with Asterisk... preferably one that does not have any issues with NAT. Can anyone point me to something suitable ? We are essentially planning to do a beta for a few hundred customers on our network and would like them to be able to call one another @ no charge. Because

I have quite successfully set up the Services button to work on the 7940 running SIP. I have a metric-imperial converter, a foreign exchange rate calculator, a calendar etc available to users. The XML is really fussy though. Simon -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Rich Adamson Sent: Thursday, 1

Hi People, Ok i've tried everything I can think of but cant get this to work. Can someone please give me an example of their sip.conf settings and also the details of the settings in their grandstream phone to allow: 1. Grandstream phone to register with asterisk when on same lan. 2. Grandstream phone to register with asterisk when phone is behind a nat. Regards, Aaron.

Not to sound like a broken record but...... Is there any word on the availability of the FXO cards for the TDM400P? I have an application that would benefit. If it has been dropped please let me know. Thanks Gene Kochanowsky

I have a PCPHONELINE SIS phone set it up to asterisk Registered SIP '205' at 24.172.221.22 port 2770 expires 120 (Port changes every time) Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 24.172.221.22(24.172.221.22 is my phones IP) Anyone have an idea what the problem is? Jeff

And if you want to use it with windows telephony software, such as answering machine or modem communications software, you can probably take the drivers for the Intel MD3200 based modem, modify the .inf for the Digium vendor and device ID. I have not tried this, but since the MD3200 modem works that way in Linux, the X100P may work that way in Windows. Then you would have a $100 winmodem! Let

Had a good question for the list, it seems whenever I work in an Asterisk console or on the machine normally I get jitters on any audio going through it. Especially if you did file copies or a 'ps ax' for example. I was wondering if there was a proper way to 'nice' the asterisk proc's? Cisco does this for example to it's "EXEC" and icmp processes, I tried

Hello All I am looking for a PC based switchboard application. Cisco CallManager has a web attendant console that allows you to use the PC to transfer calls and the like and I was wondering if there was a similar program compatible with *. Thank you in advance Keith D'Atrio -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello all asterisk users! Question: Does anybody know about any good USB handset that would understand SIP and Asterisk and will run with Linux? I have found tons of them, but they are mainly only supported in Windows environment. I would like to set up new phone system in our company that would be based on asterisk acting as PBX and SIP. If you have any suggestions, please let me know. Any

...Users] sip change? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <412F760F.60800@accessgate.net> > Content-Type: text/plain; charset="us-ascii" > > has anyone tested the vst1000 SIP phone from pcphoneline ? > http://www.pcphoneline.com/ > > Doug > > Jerry Roy wrote: > >> Hi All, >> >> Looking for a recommendation. I was hoping to purchase a * "KIT" for a >> small office. I have 4 lines and 4 extensions need phones so I need 4 >> phones....