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Hello all, Has anyone used the SIP version of this phone with Asterisk? I see Polycom has a H.323 and MGCP version also, does anyone know if you flash the phone to swith protocols? Thanks in advance for the info. Ed -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031029/a9377305/attachment.htm

...register my Asterisk server or an individual SIP phone (Cisco 7960 or Polycom IP600) with the Avaya server these 46xx IP telephones use? I don't know what model of the Avaya server the company has purchased, so I have limited info here. Thanks in advance, ================================= Ed Rubright -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031204/f4d5b00c/attachment.htm

I have a x100p card that has call waiting on the line comming into it and then into *..... is there any way i can use call waiting on that line? Michael

Hi, I've seen some reports about ruuning Intel modem with 537 or MD3200 chipset running with Zaptel drivers as a FXO port. Did anybody managed to set up a PCI faxmodem based on Intel536ep chipset to work with * and Zaptel drivers? Modem seemd to work just fine with Linux, but the driver says no;) some more info: Linux 2.4.26 mazuchna:~# cat /proc/pci | grep 536 Communication

...Asterisk server? If so, is there some doc on what needs to be configured in the iptables? 2) If the Asterisk server does indeed have to be outside the firewall with a public IP address, do my SIP phones also have to have public IP address or can they be behind the firewall? Thanks in advance, Ed Rubright -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031114/05127b54/attachment.htm

...ounds terrible!") Any ideas or suggestions? Will the path affect the rxgain setting? Will that need to be re-adjusted? What would cause echo to show up on the TDM400P FXS phones? Is the success people are seeing with the echo patch using the AGRESSIVE suppressor? Thanks in advance, Ed Rubright

...when you are ready to continue. I have compiled zaptel as:: Make clean Make linux26 Make install Does anyone have any thoughts?? Zac Amsler _____ From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Ed Rubright Sent: Monday, March 22, 2004 9:46 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Call waiting You need to send a hook flash to the PSTN line attached to the X100P. This is done by: hook-flash then *0 You'll need to have 3way calling enabled for it to...

I used to get bad echo and also voice breaking up during a conversation, so I upgraded the machine (from a PII 300 to a Athlon 1.8GHz) and have resolved the voice breaking up issues but still seem to have a fairly bad echo on *both* sides of the conversation. Currently I am only using the system for outbound calls, so I am always calling from the digium USB port phone and out to the telco using a

Hi all, We''ve been using Xen 3.0.3-1 on Debian Etch for around 2 years now using standard Xen bridging. Over the past few months, we''ve experienced the periodic disappearance of VIF macs from the FDB of the bridge. It seems to happen at random, primarily with domUs that typically receive less network traffic. When the VIF stops responding, the mac address for it still shows up

I have Asterisk setup in a SOHO environment. I have 2 X100P cards at Zap/1 and Zap/2. I have 1 TDM400P card with Zap/3 - Zap/5. I have subscribed to callwaiting, callerid and calleridcallwaiting from Qwest on the 2 PSTN lines - Zap/1 and Zap/2. My problem is when I'm in an active call to the outside thru Zap/1 or Zap/2, I can't pickup the incoming callwaiting call. I can see the