similar to: Asterisk@home - problem getting console output ...

Hello, I'm having problem with a wildcard TE110P. As soon as I load the module (wcte11xp for kernel 2.6.10), it spawns a yellow error with or without an E1 plugged-in. Any one managed to set it up in France? Here are my files: zaptel.conf: span=1,0,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 zapata.conf: [channels] language=fr context=default switchtype=euroisdn pridialplan=unknown

We had added a lot to this release to our one button install of Asterisk. Now you can have even more features automatically installed and configured. Asterisk 1.0.7 AMP 1-10-007 Flash Operator Panel 0.20 Redesigned WebMeetme weather agi scripts Midnight Commander We have added some of our most requested features. - Web Meetme is now installed by default and the meetme2

People on the list tend to think you can't make many cards work on a regular desktop. If you're willing to wait a couple of week I might have an answer for you. _____ From: Robert Augustyn [mailto:augustynr@yahoo.com] Sent: Saturday, December 11, 2004 7:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Will Adtran TSU 600 work with *?

Hi, I am considering A101/102/104 cards for my asterisk installations. Has any of you used these or any Sangoma cards in such environment? Any thoughts? How do they stack up against Digium cards? Any input would be greatly appreciated. robert

Hello, Has anyone Knowledge about softphone IAX for pocket PC totally free? Tkanks for all. -- Sandra Salmer?n Ntutumu <makevuy@ehas.org> Tlf. Analog: +34 914888405 / M?vil: 653574298 Tlf. IP desde FWD: 656212. Ext: 10 / Tel. IP desde EHAS: 010010 Fundaci?n EHAS: Enlace Hispanoamericano de Salud - www.ehas.org Telemedicina rural para zonas aisladas de pa?ses en desarrollo

I want to buy a new server to run Asterisk and after looking at prices for the Athlon XP 3000+ it costs the same as an Athlon 64 at the same speed rating. I was wondering if Zaptel/Asterisk will compile/work on an Athlon 64? -- Carlos Chavez Director de Tecnolog?a Telecomunicaciones Abiertas de M?xico S.A. de C.V.

Hello, I've signed up for a NuFone account, and added the following instructions to my config files per NufFones directinos: iax.conf [NuFone] type=peer host=switch-1.nufone.net secret=password extensions.conf (under the [default] context) exten => _1NXXNXXXXXX,1,Dial,IAX2/f00b3r@NuFone/${EXTEN} I then get this message in the CLI: -- Executing Dial("SIP/jake-fe5d",

Hi, I am working on developing a professional Marketing Materials for my systems. I plan on using a very good(expensive) company to do that so splitting the costs with several people would be nice. Let me know if you are interested on taking part in it. robert -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, Occasionally, DIDs from different providers stop working for some reason. I would like to be able to monitor situations like that and react before any of my clients start going ballistic on me. Any ideas? Scripts you know of or wrote and willing to share? Any info?would be greatly appreciated. ? Robert ? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I am planning couple small business installations and wader what should I use for 2 to 6 lines a gateway or pci card? Any comments greatly appreciated on pros and cons and brands. Thanks, robert

Hi, I have a strange problem, when I try to connect to les.net from our local asterisk server through Procurve router I seems to be connecting on any port above 1024 and when I reload sip the port is changing too ... So I never get 5060? Any ideas on what is going on and how to resolve it? ? Sincerely, Robert Augustyn 519-997-3106 ext:802 www.linqone.com ? ? -------------- next part

Hi all, Is using a Citel gateway with Asterisk a good solution for reusing of the old Nortel digital phones? Would love to get some input from actual users. Any/all opinions welcome. robert -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070510/bc5fc18f/attachment.htm

Hi, We have zabbix running and would love to be able to monitor our asterisk box with it. I believe that some sort of SNMP is build in 1.4+ correct? Where do I find more info or a how to on what is supported and how to use it? Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Just installed V 0.4 of asterisk@home Programmed up 3 sip budgetone extensions, they call call each other fine. Tried to dial '9' for an outside line through an X100P to a packet8 ATA but got 'all circuits are busy now'. Here is the console output. == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/30-8d25' -- Executing

Hi, I need to be able to unable and disable iax2 trunks from the cli? Is there a way to do it if so how? Sincerely, Robert Augustyn -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081128/7ee24fa1/attachment.htm

Can anybody provide DIDs for Chatham, ON? Usage based preferred, but flat-rate is not an issue. ? ? Contact off list. ? Thanks for your time, ? ? Sincerely, Robert Augustyn ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100529/e8ac734e/attachment.htm

Hi, Do you need a special cable and power for that unit or you can run it of PoE Switch? Thanks. robert

Hi all, We have put together a firmware for a range of inexpensive routers. It has been configured to provide optimum VoIP performance. We have internally tested it for number of months and it looks very good. You should be able to run it easily with 20+ phones on local network ( we still did not hit the upper limit ) assuming that you have bandwidth. Your VoIP will get prioritized over other

Hi, I have two servers ( A and B) in different towns. Both servers have pstn attached to them. Now I need to have calls coming to both servers to be answered on server A and then distributed between two sites. What is the best way of doing ? Having all calls to B forwarded to A on telco?s end? Having calls from B forwarded to A through internet? ? I tried to call forward the line on

Hi,   A question on something which normally should be easy !   I perform a linear regression using lm function:   > reg1 <- lm (a b+c+d, data = database1)   Then I try to perform the Chow (1960) test (structural change test) on my regression. I know the breakpoint date. I try the following code like it is described in the “Examples” section of the “strucchange” package :   > sctest(reg1,