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Are there any other wireless IP phones out there other then the Cisco 7920?? -- James Moran <jmoran@potentialtech.com> Potential Technologies

I have a done google seaches on convertion and so far they all failed. Rich adamson and wheely-bin.co.uk Here is what I have Laptop running solarwinds tftp with the following files OS79XX.txt <- POS30201 SIP<phonemac>.cnf.xml RINGLIST.DAT <-Ringer1.pcm Ringer1.pcm The phone (7940) is hardcoded to the TFTP ip of the laptop and has a hardcoded ip is was on firmware version:

...he first place. Cheers, Dean -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Duane Sent: Saturday, 1 May 2004 5:00 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] RE: E164 updater Client John Fraizer wrote: > I would say a more suitable solution in the short term would be for you > to instruct *your* asterisk server to forward your calls to wherever you > will be when you're "mobile". That way, you don't have to worry about > someone not wanting to pay to term...

We just turned up a PRI with NI2 signaling for callerID & Name. We can see the name in the CDR records but, it doesn't show up when a PSTN -> SIP or PSTN -> IAX2 call is received. The phones only get the number. Where do I look to try to fix this problem? It seems to be timing related. John

I have 2 SIP phones (Cisco 7960 & XTen) behind a NAT provided by a Linksys firewall that supports UPnP. The Asterisk server has a public IP. Here are the problems that I am having with this configuration... 1. The 2 SIP phones can call MeetMe and have a conference but cannot call each other. (Yes, they connect but no audio either direction) 2. I have verify=yes in the sip.conf for both

I've tried to setup multiple extension dialing - ie dial 1 number and it rings at a number of sources. For the most part its worked.... Now if someone dials 107 it rings Sip phones at 102 and 107, then goes to voicemail after 40 seconds. exten => 107,1,Dial(SIP/102&SIP/107,40|r) exten => 107,2,Voicemail(u102@pstn) exten => 107,3,Hangup exten => 107,102,Voicemail(b102@pstn)

I am configuring an asterisk server and I want to test the incoming configuration with my FXS handsets. I have the FXS lines able to call eachother and they can connect out the FXO lines. I changed the context for the FXS lines to "incoming" so that they would be able to test the setup for incoming calls. For the incoming context I have: [incoming] exten => s,1,Wait(1) exten

I did some searching to find out what I need to do to get CID&NAME inbound on one of our PRI circuits. Right now, we're only getting number (well, asterisk shows the number in both the name and number portions of CID). Is there anything special I need to do with Asterisk to have it accept the name portion of CID along with the number on a PRI that is provisioned NI2? Thanks, John

I just noticed that incoming calls don't show up in user specific CDR files. For example, if in sip.conf, you have the following entry: [123] callerid="Joe Blow" <123> type=friend username=123 secret=456 mailbox=123@vm-context host=dynamic context=123 canreinvite=no dtmfmode=rfc2833 nat=yes accountcode=customer-name amaflags=billing Calls that the user initiates show up in