similar to: Difference between Asterisk and Asterisk@home

Hello, I am just building my first Asterisk server. Looking for a couple of good quality ip phones. I like the Cisco 7960, are they easy to configure to work with Asterisk? What are the alternatives, with a good speaker phone, and simple clean and stylish look like the Cisco? I appreciate your advice. Kind regards Michael Felder IT Medic Australia Pty. Ltd. P: 03 9557 2213 F: 03 9557 2214

Hello, My asterisk currently will dial an outside number after I dial the number and press send on the phone. How can I get it setup so I have to press '0' for an outside line. Kind regards Michael Felder IT Medic Australia Pty. Ltd. P: 03 9557 2213 F: 03 9557 2214 M: 0419 568 217 E: Michael@ITMedic.com.au http://www.ITMedic.com.au Keeping your computer systems healthy.

Hello, Is it possible to connect the Panasonic KX-T7665 IP phone to Asterisk? Kind regards Michael Felder IT Medic Australia Pty. Ltd. P: 03 9557 2213 F: 03 9557 2214 M: 0419 568 217 E: Michael@ITMedic.com.au http://www.ITMedic.com.au Keeping your computer systems healthy.

Hello, Does anybody have the latest Boot ROMs for the IP500 and IP 600 Polycom phones. I have one of each and can't find the Boot ROM v 3 anywhere to download. I would also love a good sample phone.cfg and sip.cfg files from an Aussie asterisk user to look at. Also the ip500 is having problems trying to load the bootrom 2.6.2 ? Any ideas? Kind regards Michael Felder IT Medic

Hello, I am looking at the Polycom phones. The ip600 has a very nice screen, is that the only real advantage over the ip500 and ip300.. Is it worth the extra dollars? Kind regards Michael Felder IT Medic Australia Pty. Ltd. P: 03 9557 2213 F: 03 9557 2214 M: 0419 568 217 E: Michael@ITMedic.com.au <blocked::mailto:Michael@ITMedic.com.au> http://www.ITMedic.com.au

hello is there a gui that would allow me to configure everything from phones, to extentions, to voice mail to basicly everything that asterisk can do? I did go to www.voip-info.org and none of the guis I saw there do the trick and the ones that come close aren't downloadable just wanted to see status on this thanks hank ---------------------------------------- My Inbox is protected by

Good Afternoon, I have an Asterisk box that is acting like it is passing "core show" before every command I type. For example, if I type sip, I will get "No such command 'sip' (type 'core show help sip' for other possible commands). Any ideas? -- -jayson

Haven't seen email since the 29th.. just testing. -------------- next part -------------- A non-text attachment was scrubbed... Name: mhess.vcf Type: text/x-vcard Size: 288 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050801/96555713/mhess.vcf

Hi all, I am handed a project to setup *. The requirement is that it can handle 8 T1s. Half of the calls coming into the system will be routed to SIP extensions (with transcoding). The machine we have in our disposal is a new dual Xeon 3.2gHz server with 2g of ram and an dual 1000mb nic. Voice will be coming in from the PSTN (through 2 quad digium cards) in g711ulaw, and most of the time will

Hi gang, We are moving our 1.4 asterisk with DAHDI over to 10.0 with SIP. Everything is going nicely except that I can't get NV_FAXDETECT to compile properly into 10.0. Because of this, I will have to have my receptionist manually transfer incoming faxes. Any suggestions? Thanks in Advance Danny Nicholas -------------- next part -------------- An HTML attachment

Hello, I have Asterisk 1.6.2.9-2, the directive "sendrpid" does not work! I placed this in my peer: (sip.conf) sendrpid=yes trustrpid=yes or sendrpid=yes trustrpid=no (and restarted Asterisk) and the line "Remote-Party-ID" does not appear in my sip debug! Please help me, Mickael. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello - I am looking at connecting 7 - 10 locations together using Asterisk and possibly some VoIP gateway appliances. I need to insure best voice quality as these trunks will be used primarily for customer calls. I am considering implementing a full T1 frame relay circuit to each location which can be done for a reasonable cost. DSL and Cable are currently at each location and setup for

Sending works but on receive it keeps failing - reporting back 'training' failure. I am using Asterisk 1.6 with T38. What should I post to the list to assist diagnoses? Michael

Anybody here having trouble receiving email from the list on Gmail? I havn't received anything since Friday July 29.

Hi, Sorry for maybe not a very list related topic, but I have always been curious if there is information on how many Asterisk based PBXs are operating Worldwide? Thanks and hope the community will not reject my curiosity! :) Best Regards, Vallu Sevana Oy -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have compiled zaptel on Mandrake following everything I have always done on Fedora. It is 2.6 udev so... I had to modify the 01-devfs.rules Make linux26 Make Make install... Everything appears to compile correctly but it says module not found when doing "modprobe zaptel" Is this a rights issue? Jordan Novak -------------- next part -------------- An HTML attachment was

I'm trying to setup asterisk to accept call from Teliax, request the 10 digit number from user, then dial it thru the VoIPJet. If I'm not wrong I will be charged by both providers because both connection is active during conversation. So my question is can I set the things so that I pay only to VoIPJet? Specific configuration snippets will be greatly appeciated. Thank you.

Hi, Has anyone met any success, installing localized (ie non-english) menus within SIP firmware enabled Cisco 7941 ? Those phones seem to be trying to download localized menus from Cisco Call Manager but as they are managed by an Asterisk server, I'm looking for a workaround. Any advice ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

We have written some monitoring and stat collection scripts that use asterisk -rx "command" The script runs once a min and logs data and posts any critical notifications. Everything is working well with this method but we get the -- Remote UNIX connection / disconnect message once a min and we would like to suppress it. Is it possible without reducing the verbose logging level.

Please help me test my setup by dialing 800.564.0215 and listen to the queue for a bit. I have a quad port T1 with NFAS setup. I can dial-out but I cannot dial any 800 numbers (Global Crossing says I need LDS service and that will be a couple weeks) so I cant test it myself. I need at least 24 callers to feel comfortable enough that it is working properly. Thanks, Steve Totaro