similar to: Contact Directory on Polycom IP-501 phones

Great idea Dean, "I would also suggest that maybe instead of looking to set up within the disaster zones that you consider setting up in the relocation zone (eg where people are being sent to) yes they have payphones there but having a bank of 20 grandstream phones connected a T1 is a far smarter and more effective solution and probably more meaningful." > -----Original Message-----

Hi Jeremy, I think the first place to start on this is some corporate funding by one of the voip providers (you could pretty much guarantee US only calls). Once you have this, the rest (servers, generators, sip phones etc) fairly easy to come by. If you do get this off the ground I would be willing to make a donation to a "asterisk" relief project. I would also suggest that maybe

The national guard and/or army routinely implements VoIP over wireless in situations where comm is lost, I did see an news release that the Guard started this project in the south the day after the disaster hit. The key is not the VoIP infrastructure, that is the easy part (one ss7 Sonus softswitch and a DS3!), the key is distributing IP over a wide area, which is best done on the quick with WiFI

I've got a problem w/ zap calls being dropped right after they are answered. I have a log file: http://pastebin.com/368526 Everything looks OK except for the DEBUG[25563] chan_zap.c: Exception on 9, channel 1 that seems to come up quite often. As soon as the other end of the Zap answers (my cell phone), and I can even hear a half second of noise, the line goes dead and gets hungup. In

I've got an issue w/ 1.2.0beta1, where if I call VoiceMailMain from a sip phone, and then either put in incorrect passwords or just hang up, I never get a Spawn Extension that hangs up the call, and my sip phone is not capable of making any more calls until I restart the daemon. Can anybody help me fix this? -- Jesse Keating GameHouse -- Systems Engineer

When I hang up in voicemail, Asterisk seems to stop responding. (hangup vs pressing # to disconnect). After that, no calls can be made until I restart Asterisk. In IRC, a developer seemed to think it had to do with me using switch => in my dial plan. Basically I never see the calling extension get the -1 signal. Can somebody help me figure out why this is happening and how I can fix it

I am trying to run "scrufs" one of gamehouse games. As all gamehouse games, at start it shows screen where user should register or to choose play free trial. I instaled mozzila plugin or something. This screen shows, but when I try to click on links like "start" or "register" i get an error in terminal Code: >err:mshtml:before_async_open GetWineURL returned NULL

Hello everyone, I have an IAX server (asterisk@home) with a FXO card. I have a trunk connected to a voip provide, asteriskout. When I call my server from a public phone, I want to route this call to the asteriskOUT trunk so that I can make long distance calls. How can I setup a secret password in the extension.conf, so that my asterisk server can allow me to make long distance call ? Thanks

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Adrien Laurent > Sent: 02 August 2005 14:56 > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] How to create a secret code to use > myasterisk@home server's long distance plan from a public phone > >

Hi: Please advice me of a voip provider with reasonable reseller program. I was using voipjet and it has a lot of problems. Did anyone experienced asteriskout.com service? They have good prices. Regards; Chawki Hammoud ____________________________________________________ Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs

Hi All, I''m trying to do something regarding Bandwith Aggregation and I would need to set new options in the IP-header. I had been studying some kernel source that concern to the IP layer, but I don''t really know how to set a new option. Does anyone have any idea to add new custom options into the IP header ? Thank you in advance Felipe Herrera

I tried following the Wiki page regarding the Polycom 501 and having the same extension appear on all 3 line buttons (just like my cisco) but I'm having no luck. Has anyone else had success in doing this? Perhaps someone who has been successful can update the wiki? Thanks, Matthew http://www.voip-info.org/tiki-index.php?page=Polycom+Soundpoint+IP+501

Hi folks, New to the list. Just updated the bootrom and app firmware on a Soundpoint IP 501 as per: http://www.voip-info.org/wiki-Polycom+Phones Updated from: to: APP 1.4.1.0040 1.5.2.0054 BootROM 2.6.1.0003 2.6.2.0032 After I did this, it appears that the Web interface for the phone won't change the settings, nor will it actually reboot the phone now. What do I

Hello, I may have opportunity to obtain a Intel EXPI9404PTLBLK PRO/1000 PT Quad Port ethernet adapter at a significantly reduced price. What I would like to do with it, I want to make sure it is possible and sane before I buy it. -=- Device sits in CentOS box that connects directly to Internet via onboard network adapter. It pretty much only acts as a NAT router + dhcp server + unbound

I saw several threads requesting an Asterisk forum to complement the email list. i.e. http://lists.digium.com/pipermail/asterisk-dev/2004-February/003103.html I recently created an Asterisk forum within TMC's popular VoIP forums for everyone to use. http://voip-forum.tmcnet.com/voip-forum/forum/forum_topics.asp?FID=15

Good morning, my name is Jason. I am using the latest version of Syslinux to boot up Linux from DOS formatted HD. However, I got a problem during booting. Additionally, I am using ramdisk with 'vmluniz' and 'initrd.gz' The screen print out the following error message during booting. ----START---------------------------------------------------------- Syslinux 1.75 2002-06-04

Dear all, Could anybody give me some recommended open source software for wifi management? What I am looking for is a simple and basic tools, for example the software will have these kind of features : - user connect to the wifi network - user got IP from DHCP but he can't browse to anywhere at first. Instead he will redirected to a "login page" - administrator will set the

Hello, Sorry, I know I read this somewhere but now I can't find it when I need it. I'd like to force a call to go out one line if we dial '9' first and then the number. Same for '8' only I will force it out a different line. There is a parameter or a method to allow the dialtone to come back after pressing the first 9... but I can't remember how to do it. Anyone know?

Hello, Can anyone point me in the direction of software to monitor channel usage on voice T1s? Using a TE410. The wiki documentation seems geared to SIP channel usage.... Thanks Charles -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051004/8450d0c2/attachment.htm

I have installed Asterisk and i have configured with two SJPhones; i am able to make calls between these two phones.I am planning to develop a application basically web based application from which the administrator able to trace the call logs or call summary, i mean from which user agent to user agent call is going , and what is the staus, if second user tranfers the call to the third