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...----------------------------------------------------------------------- -------------------------------------------------- /etc/asterisk/ ------------------------------------------------------------------------ -------------------------------------------------- ; December 30, 2003 08:06 EST mwless@mindworks.net ; ; Asterisk can be found on http://www.asterisk.org ; ; Brent D. Franks - mwless@mindworks.net ; ; This file configures: ; 4 Wildcard X100P PCI FXO Cards ; [channels] language=en ; Put all 4 Lines into one group, so we can make and place ; calls on the first available line on the box...

On Sun, 2003-03-16 at 19:24, Ted Faber wrote: > On Fri, Mar 14, 2003 at 02:48:26AM -0500, Andrew J Caines wrote: > > Starting the server as normal with xinit gives normal startup messages, then fails > > with a signal 11: > > I have this problem, too. Same nv server. I'm happy to provide > details. One that might help is that this is an Athelon XP 1800. > >

Hello, Has anyone on the list been able to successfully setup a Polycom Soundpoint 500 IP phone? I am getting failed registrations, and the Polycom documentation is not very precise. Their web interface isn't helping much either. Thanks in advance, Brent -------------- next part -------------- An HTML attachment was scrubbed... URL:

...to the list to solicit feedback (and if my assumptions are correct, hopefully help others out)... If anyone out there knows anything about wiring up an Echo Can, and my outline below is incorrect, please let the me/list know.. I have posted the Documents I received from Tellabs at: http://regent.mindworks.net/echo/ If you look at Page 6 of the Shelf PDF, you can see there are four 50 pin connectors. These connectors are as follows: Receive In (RX-in?) Send Out (TX-out?) Send In (TX-in?) Receive Out (RX-out?) Am I off base yet? Now, in order to hook this up between a Cha...

Hello, I have searched the lists, and have seen this message posted numerous times, however there never seem to be any replies to it. I was curious if anyone had figured out the WARNING: File chan_iax2.c, Line 5466 (set_config): Ignoring port for now Everything seems to work just fine, however when I try to change iax.conf port to 4569 it doesn't seem to like that either. Thanks, Brent

Hello, We have 15 Soundpoint IP 500 phones and recently call quality has deteriorated. On some calls there is a static-buzzing of sorts that occurs when users talk. It can be picked up on SIP-SIP calls and SIP-ZAPTEL (Channel BANK<<-T100P). It basically sounds really weird whenever someone talks, it sounds like a bee buzzing or something. Very hard to explain. Also, there will be echo

...bchannels and 1 dchannel connected to a Class 3 switch and its been up for about 2 weeks solid. I do believe that the zaptel software restarts the channels every so often but I've never gotten a red alarm before. -Matthew ----- Original Message ----- From: "Brent Franks" <mwless@mindworks.net> To: <asterisk-users@lists.digium.com> Sent: Friday, December 17, 2004 11:28 AM Subject: [Asterisk-Users] Red Alarm / Alarm Cleared Zaptel Issue (bug?) > Hello, > > About every 2 or 3 days I notice in the messages log file: > > Dec 17 08:39:27 WARNING[1220]: Detected a...

Hello, About every 2 or 3 days I notice in the messages log file: Dec 17 08:39:27 WARNING[1220]: Detected alarm on channel 2: Red Alarm Dec 17 08:39:27 WARNING[1220]: Detected alarm on channel 3: Red Alarm Dec 17 08:39:27 WARNING[1220]: Detected alarm on channel 4: Red Alarm Dec 17 08:39:27 WARNING[1220]: Detected alarm on channel 5: Red Alarm Dec 17 08:39:27 WARNING[1220]: Detected alarm on

Hello, I'm looking for a similar feature... Dial a number via ZAP/g1 after the line gets answered wait 10 seconds send DTMF Regards, Marc -- Network Manager Marc Storck LuxAdmin.Org mstorck@luxadmin.org Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352

Hello, After reading the lists and taking reccomendations from TC, I have finally given up on the echo can built into asterisk. I am sick of hearing complaints from users, so the money spent on a hardware echo can will be worth its weight in gold. I am curious however, about some setup and component requirements. It seems as if every telecom place I call, either never calls back or doesn't

...m at my house before (without an * system) where I pick up too quickly for it to ring. Are there any methods to mitigate this problem? I can see that Asterisk is recognizing the call, but it still opens a zap channel for the outgoing call. Any help would be appreciated. Thanks, Brent D. Franks Mindworks Internet Services

I am getting ready to install Asterisk and I was looking into the Polycom IP600 phones. I spoke with Polycom sales to verify the multiple line appearance and they said it would work. More specifically, if lines 1-3 all contain the same SIP registration info, the Polycom will only send out 1 SIP registration to the server and then handle the calls ringing on multiple lines. I was wondering if

I've just installed an X100P, built the kernel module, and tried to use it to make an outgoing call (via a phone connected to an ATA-186). However, I just get a reorder tone and see this on the console: -- Executing Dial("SIP/ata1-4409", "Zap/1/5551212") in new stack NOTICE[1200825920]: File app_dial.c, Line 481 (dial_exec): Unable to create channel of type

...ate that. That is why I >> do my research until I hit a wall, then I will ask here. I appreciate >> whatever you can spare time for. >> >> Thanks! >> Wiley >> >> =20 >> >> -----Original Message----- >> From: Brent Franks [mailto:mwless@mindworks.net]=20 >> Sent: Monday, July 19, 2004 10:26 AM >> To: asterisk-users@lists.digium.com >> Subject: RE: [Asterisk-Users] Polycom IP 500 Voicemail >> >> =20 >> >>> Thank you! >>> >>> Can you tell me more about the dial plan feature?...

Hello All, I have some Polycom IP 500 phones that I would like to have configured for direct dialing to our voice mail system. So far I have been unable to get the hard button labeled Voice Mail to connect to Asterisk without first passing through the message center prompts. I have followed all the Admin Guide instructions regarding the phones .cfg files and using

Hello, On the Polycom IP 500 Phones, when I press the mic mute button, the mic on the speaker or headset goes muted. However when I press the mic mute button again, the call is terminated by asterisk. Asterisk shows a: WARNING[1236268096]: File channel.c, Line 1296 (do_senddigit): Unable to handle DTMF tone 'f' for 'SIP/####-####' I am using reinvite=no on the phones. After

Has anyone had any success using more than one or two X100P cards? I have 4 in a system, and channels 2 3 and 4 all seem to work just fine. Channel 1 however is acting up. I get random red alarms, disconnects, etc. I have checked the /proc/interrupts and everything is sitting on it's own IRQ. Also checked memory addresses and everything looks good there. Not looking for anything specific,

Does * support Called Party Identification? Say for example, I dial extension 2000, SIP sends back John Doe from the sip.conf file where extension 2000 is defined? Would this violate the SIP RFC? - Brent

Hello, I am seeing this error: Jan 24 20:02:53 WARNING[1175660480]: Ring/Off-hook in strange state 6 on channel 3 on one of my phone lines that comes into my channel bank. It only happens on one line, and it still seems to work. Also, I have had two different channel banks installed (ADIT 600 & Adtran TA 750) and see the same error message on incoming phone calls on that line. I don't

Hi everyone, This one has me baffled.... We installed 16 Polycom IP 500 Phones in January on an Asterisk Server: 3.0Ghz 768MB of Ram 30GB Harddrives 10/100/1000 NIC Our Lan setup is like this: 3 Netgear FS526T Switches 2 10/100/1000 Mbps copper gigabit ports 24 10/100 Mbps ports 100Mbps out to phones the 1Gig ports connect the switches together, and one gig port goes to the