search for: copilotconsulting

...y to transfer the call to 112, and then it hangs up. I can try calling in from outside and parking the call and it hangs up too. It is SO weird that I can transfer to voicemail ok but nothing else! Here is a sip debug of me calling in, talking to someone, and having them transfer me: http://www.copilotconsulting.com/typescript I was running a recent CVS version of * but have gone back to CVS-v1-0-12/21/04-16:31:58 just to make sure it isn't some sort of problem with latest cvs. Any clues are greatly appreciated! -- Tracy Reed http://copilotcom.com This message is cryptographically signed for yo...

...verything back in and use up to the maximum RAM configured for that domain if they get busy and need it but let it swap out the rest of the time so other busier domains can use the physical RAM. This is feature #1 on my Xen wishlist. Is there any work going into this area? -- Tracy R Reed http://copilotconsulting.com 1-877-MY-COPILOT _______________________________________________ Xen-devel mailing list Xen-devel@lists.xensource.com http://lists.xensource.com/xen-devel

Good day all I'm still looking for a SIP client that will work on fedora core 1? Thanks

...g on here? I have googled and checked the wiki many times and cannot find anyone with this problem but it has happened to me twice now. I have two unusable phones until I get this fixed. The other 6 flashed just fine. -- Tracy Reed The attachment is a digital signature. http://copilotconsulting.com More info: http://copilotconsulting.com/sig -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040827/fca498d...

Hello, As I understand * don't supports T.38 in Zap channels (please correct me if I'm wrong, BTW is there plans for such support?) I believe it's should support T.38 in "pass-thru" mode. I mean setup like this: Hardware gate with T.38 ------ Asterisk ------ Hardware gate with T.38 But I had troubles with this setup (no faxing) while two gates conneted directly with same

...page: http://www.voip-info.org/tiki-index.php?page=Asterisk+Request+to+schedule+in+the+past so that is where I have added my findings on this problem so that others don't have to go through what I went through. -- Tracy Reed The attachment is a digital signature. http://copilotconsulting.com More info: http://copilotconsulting.com/sig -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040808/d8c17fe...

Is there any way to play background music on a sip phone while the phone is not in use like many legacy pbx's offer? Could you take 7960 and use the 6th line in a similar fashion to the all setup maybe? Thoughts ideas? -- respectfully, Joseph - (606) 477-2355 x140 ------=============

I have a client who is looking at replacing their PBX, and I'm interested in putting together an Asterisk solution for them. One feature that would really, really get their attention is if I could do the "Vonage" thing, where if a PSTN caller dials a direct extension (coming in over PRI) both the user's deskset _and_ an external number (their cell phone) would ring, with

...vig.com on a x86_64 running Linux From openh323 version.h: #define MAJOR_VERSION 1 #define MINOR_VERSION 15 From pwlib version.h: #define MAJOR_VERSION 1 #define MINOR_VERSION 8 -- Tracy Reed http://copilotcom.com This message is cryptographically signed for your protection. Info: http://copilotconsulting.com/sig -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20041122/90c85495/attachment.pgp

...h nat=yes in sip.conf) which is bizarre and contrary to what I have read where IAX should be NAT-safe and SIP not. I have dreams of a world fully converted to IPv6 where NAT no longer exists. Alas, it is but a dream. -- Tracy Reed The attachment is a digital signature. http://copilotconsulting.com More info: http://copilotconsulting.com/sig

When we receive or make a call to the outside - they can hear us, but we cant hear them. It may work 1 of 20 times. I have set canreinvite=no and looked at many sites but cannot track down this problem. Current setup: Isdn Eicon Diva card / Capi -> Asterisk --> network. I have tried adjusting the RTP port in rtp.conf with the Cisco default ports, no luck. Anyone had this

...lling to pay $$$ for an extra set of eyes to get this resolved fast. It's probably something quick and easy and we are just missing it. Email me ASAP if you are willing to help. -- Tracy Reed http://copilotcom.com This message is cryptographically signed for your protection. Info: http://copilotconsulting.com/sig -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20041124/d8f13d0b/attachment.pgp

Can some people post some configurations they've implemented when deploying an * system for let's say 25-50 stations and maybe a larger 200 station system? I would assume some kind of chassis with some DSP boards and some kind of system board with a hard drive for running the system and storing the voice mails - obviously I'm interested in specific chassises and boards used and

...e on only if the phone is busy. If the phone is offline for some reason the light is dark. But the status of the light never changes even when I am making calls on one of the phones. -- Tracy Reed http://copilotcom.com This message is cryptographically signed for your protection. Info: http://copilotconsulting.com/sig -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20041204/24046f67/attachment.pgp

I cannot get the transfer button to work on a Snom 190, I cannot get the # to work either. Any ideas? Regards Thorben -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041207/5ac06747/attachment.htm

I'm considering that board as a mail and voip gateway for home use. In view of all those statements about how little resources asterisk needs, did anybody already try running asterisk on it? Thanks, Bruno.

Hi All, Can someone explain to me the difference between g711's ulaw and alaw codecs? Is it just different header info or is the actual payload in each encoded differently? I have thus far noe been able to find any difinative information onthe matter. All I've managed to find out that they are "similar", they sound the same and that it doesn't matter which you use. Could

...I don't know if it matters but there is no zaptel hardware at all in this box, pure voip. Anyone have any idea where the bottleneck could be or any tuning tweaks we could make? -- Tracy Reed http://copilotcom.com This message is cryptographically signed for your protection. Info: http://copilotconsulting.com/sig -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20041130/126134f3/attachment.pgp

I have several 7960 phones with SIP image (7.3) and Asterisk 1.0.1 on FreeBSD. When I have 2 active SIP calls on the 7960 phone there are no available lines for additional calls. I tried to configure 2 lines to the same SIP server but it's still limited to 2 calls. How to utilize all lines? -- Called user -- SIP/user-acc6 is ringing -- SIP/user-acc6 answered SIP/x.x.x.x-09a9a000 --

Hello, We are looking for commercial solution SS7 with Asterisk. It does not need to be "build-in" with Asterisk. Could anybody suggest something? Thank you in advance. Bart