search for: codepipe

I just got an email from SIPphone advising that there have been problems with the above firmware and advising to reload from their server. This does in fact reload 1.0.4.35 into the phone. And now voicemail has gone AWOL again. -- Dave Cotton <dcotton@linuxautrement.com>

Hi all, Did I see on the list a while back someone creating a program to reboot the GS phones at a pre set time. If anyone has this program I would love a copy if it is freely distributable. Regards Dave

Guys.. I just saw this for the first time... I did some google and wiki without any luck.. what does a red or yellow alarm mean in zaptel? Feb 23 02:54:16 WARNING[16890]: chan_zap.c:5865 handle_init_event: Detected alarm on channel 2: Red Alarm Feb 23 02:54:24 NOTICE[16890]: chan_zap.c:5860 handle_init_event: Alarm cleared on channel 2 This just happened by itself..

I've already sunk several hours into this without any real progress, so I'd really appreciate any help My task is simple -- establish a connection between a softphone on XP ProSP2 to a Asterisk server on Linux FC4 over a LAN through a Netgear router. The server will then go out to a PSTN termination service. Thus far, the PSTN termination connection works fine -- I've opened up 4569

...e the asterisk on an xbox page!!, but the link seems broken http://nlug.org/mail/nlug<b>2003_12/0094.html I've tried removing the <b> with no luck Anyone know what the link should be ? Thanks Panny ----- Original Message ----- From: "David J Carter" <david.carter@codepipe.com> To: <asterisk-users@lists.digium.com> Sent: Tuesday, February 03, 2004 8:31 PM Subject: RE: [Asterisk-Users] The Smallest Asterisk Server Ever? > Hey I don't know, I paid ?100 ($170) for my XBox, couldn't get a PC for > that. > > The Linux bit is all free, and o...

Hi all Do any of you know i can force asterisk to lookup ip addresses for peers and trunks everytime it tries to make a call. One of the peers has a dynamic ip and is using DynDNS to register host. Now i need to reload asterisk everytime i want to call it thanks liaan __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection

If someone could point me in the right direction I would much appreciate it. Here is my problem: My directions for my sip phone says to dial an ip address 12*34*65*78#. When I dial that into my phone my asterisk server is only picking up some of the numbers in the above example it would pick up 6578. Then of course not find it and ring busy on the phone. The same is true for dialing a regular

...s@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE > > or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > >--__--__-- > >Message: 2 >From: "David J Carter" <david.carter@codepipe.com> >To: "Asterisk User Group" <Asterisk-Users@lists.digium.com> >Date: Tue, 4 May 2004 17:42:39 +0100 >Subject: [Asterisk-Users] Pots Extensions >Reply-To: asterisk-users@lists.digium.com > >Hi all, > >I am either going daft or not reading things right....

Hi, I am trying to forward calls to another * server with IAX Here is What I want to Do 1- Call SERVER1, let say at 51412345678 2- SERVER1 should transfer the call to SERVER2 in a remote location 3- SERVER2 Receive the call and transfer it to the PSTN number. I have one X100P card on each machine. What is happening is that when the remote party picks up the phone, all he can hear is a weird

Hi everyone, I am trying to connect 2 asterisk servers via IAX, but it just fails to do so.. I'm using SIP to connect the IP phones on the LAN at the 2 different physical locations where each server resides and I'm able to communicate on my LAN via SIP without any issues. The problem is that I'm unable to make Asterisk servers talk with each other via IAX.. Here is my issue.

-This is my 'sip.conf' file: ;************************************************************* ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls tos=184 maxexpirey=3600 ; Max length of incoming registration we allow

Hello all, Saturday night, after a couple of shots of bourbon, I realized that I had an old PC sitting in the garage that I could use as an Asterisk gateway if I just blew the dust off it and reloaded it with a modern Linux distribution. In my characteristically impulsive manner, I grabbed it and started cleaning it up so that I could put it in my office without my wife having a fit. The

Hi all, I am trying to get a Menu system to work, and having probs with the internal extensions from the prompts. Below is the extensions.conf section. [mainmenu] ; ;"main menu" context with submenu ; include => default exten => s,1,Answer exten => s,2,Background(hello) exten => s,3,Background(thank_you) exten => s,4,Background(if_you_know_extension) exten =>

Hi all, I have looked through the wiki for any information on how to make an extension autodial another extension when it goes off hook. Anyone done this or know how it's done. regards Dave

Hi, Does anyone know of a provider/terminator of Belfast, Ireland telephone numbers? Thanks in advance Regards Dave

All, I have downloaded and installed openh323 as per the documentation. When the machine now reboots safe_asterisk just keeps restarting. If I start another session and just load asterisk -vvvvvvvgc asterisk loads. If I enter noload chan_h323.so in the modules.conf then safe_asterisk will kick in. Not 100% on Linux yet but I have added the environment variables info into /etc/profile so they

Hi all, I am either going daft or not reading things right. I have just installed, (well 10 hours ago) a TDM40B and 3 X100P cards. I have followed the examples for the conf files to the letter. I can call the pots extensions OK from IAX clients, SIP clients and from the incoming X100P cards. But, if I pick up the handset to make a call all I get is the engaged tone and the following message.

Hi all, I have just been reading an article on the asterisk-doc site about ISDN X-Over cables. The article mentioned the converting of an NT1 to make this possible, has anybody got the information required to modify a BT NT1? Or any information on the BT NT1. Thanks in advance. Regards Dave

Skipped content of type multipart/alternative-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031107/380d67cb/attachment-0001.htm