similar to: zapata.conf for a BT phone line with a TDM422P

Hi, Is there any documentation on the fields prewink, preflash, wink, flash, rxwink, rxflash, start and debounce in zapata.conf? The "Recall" button on my phone doesn't seem to trigger a transfer via my shiny new TDM40B. However, tapping the hook does, but only if I tap it for long enough. Presumably the "Recall" button's timing is too short? Further, most users who

Dear All, Thank you for taking the time to read this post - I am *confused!* as to why my asterisk setup does not work as it should. I have an ISDN 30 connection for telephony, a Sangoma card, and asterisk installed. Incoming calls, and outgoing calls work 100%. Making an international call, results in silence, or the error message all circuits are busy Numbers being passed to the trunk for

Hi I am in the process of porting the Ogg Vorbis libraries and player(s) to Symbian OS (EPOC32) which runs on Psion machines and Nokia 9210 to name some machines. They are all using an ARM core. I managed to successfully recompile libogg, libvorbis and libvorbisfile with only minor modifications. They include: * removal of static data, which is not supported by EPOC DLLs. All lookup tables are

If anyone can help me with this I'll be soooo grateful :) We have a isdn30 line, with a DDI range. We also have 2 business units that have separate 0870 numbers that are mapped onto 2 DDI numbers. I would like to be able to present these 0870 numbers from the business units so that the correct number is displayed on a callerid, or when 1471 is dialled. BT claim that I can only have a

# export USB_DEBUG=5 # /usr/local/ups/bin/usbhid-ups -a CP550SLG -DDDDD Network UPS Tools - Generic HID driver 0.34 (2.4.3) USB communication driver 0.31 0.000000 debug level is '5' 0.000426 upsdrv_initups... usb_set_debug: Setting debugging level to 5 (on) usb_os_init: Found USB VFS at /dev/bus/usb usb_os_find_busses: Found 001 usb_os_find_busses: Found 002

My daughter has a BT account, with a BT supplied single-port router. I'd like to replace it with a standard router, but the settings appear to be totally hidden. If anyone reading uses BT, could you please tell me where to find the info? Maybe off-list, to save bandwidth for others/ Thanks Anne

Hello, I have a Dell PowerEdge 2850 that periodically (6 times in the last 24 hours) freezes, requiring a power cycle in order to come back. The machine is running a Xen 2.0 stable source install. Here is the appropriate Grub entry: title Xen 2.0 / XenLinux 2.6 kernel /boot/xen-2.0.gz dom0_mem=131072 module /boot/vmlinuz-2.6-xen0 root=/dev/sda1 ro console=tty0 Dom0 boots happily, and new

I'm having problems with unclean hangups (being read as a flash instead of a hangup?). Can I turn off hook flash recognition in asterisk, but still have the flash button on the analog phone operational? Could I use these settings in zapata.conf to fix my problem? *prewink*: Sets the pre-wink timing. *preflash*: Sets the pre-flash timing. *wink*: Sets the wink timing. *rxwink*: Sets the

I get the following message on a Centos 5 system (really a Trixbox 2.4 build on Centos 5): Jan 14 00:12:28 sip2 kernel: hub 1-0:1.0: connect-debounce failed, port 1 disabled What does this mean? This message occurs about 30 times/sec for about 45 sec. Then my Bluetooth token starts up. Jan 14 00:12:28 sip2 kernel: hub 1-0:1.0: connect-debounce failed, port 1 disabled Jan 14 00:13:00 sip2

Everyone - thought I would pass on a useful piece of information. Finally got a solution to my phantom ringing problem. Problem - the zhone is triggered into detecting ringing by the Automatic Line Insulation Tests (ALIT or LIT) run nightly automatically by the telco. Here it is twice between 8pm and 9pm on my particular lines. My first approach before I knew specifically the buzzword for

Ah, but the interesting thing is that they are coming out with a 'clam' version like the 5MX. Details are limited at the moment, but that could mean the combination of 5MX usability with a supported linux distro. I am drooling in anticipation. Sounds like a "I've finally finished my PhD and deserve a treat" situation to me :) Dave On Thu, Mar 06, 2003 at 09:17:20AM

Does anyone know the default E&M Wink timings for Nortel DID ports? The default settings on Asterisk are: ; prewink: Pre-wink time (default 50ms) ; preflash: Pre-flash time (default 50ms) ; wink: Wink time (default 150ms) ; flash: Flash time (default 750ms) ; start: Start time (default 1500ms) ; rxwink: Receiver wink time (default 300ms) ;

I have a situation where I am receiving DID calls using Immediate Start Pulse signalling on a Loop Start trunk. The line terminates on a Newbridge Mainstreet 3624 channel bank, which provides battery etc. The channel is converted and routed to Asterisk. The lines are configured as follows: /etc/asterisk/zapata.conf ; Channels 1-24 service MainStreet 3624 channel bank context=infrom-did group=1

Hello This is a fun one for the list... Twice now, the Police have contacted us to say they have had a silent call then hangup from our landline number to the 999 service. As a matter of course, they follow up these calls in case someone is in distress. Nobody here was in distress - well, no more than normal! The Police aren't hugely happy when we tell them it must be a mistake. Thing

From a 'sip debug': Sip read: SIP/2.0 100 Trying From: "Evert"<sip:[username]@[my ext. IP]>;tag=as6e18534e To: <sip:[dialled number]@[SIP server of VoIP provider]> Call-ID: 6cbf41c25281f08b2e7bbc5043061975@[my ext. IP] CSeq: 102 INVITE Via:SIP/2.0/UDP [my ext. IP]:5060;branch=z9hG4bK4fd1045b Content-Length:0 7 headers, 0 lines Sip read: SIP/2.0 403 Forbidden

Hey all, I have an Asterisk-box with an E100P and a PRI (Euro-ISDN) coming into it from a Meridian-switch. The incoming numbers on this PRI all start with the same digit and the last part of the dialled number is signalled to Asterisk digit by digit, until Asterisk signals that the number is complete and the call rings. All works well, unless I have 2 or more numbers which start with the same

I have recently built my first asterisk system and am very impressed with its capabilities. However, I have run into one problem that hopefully someone can help me with. I am trying to use the DIALSTATUS function to route incoming calls to the appropriate Voice Mail (busy or unavailable) or to an Unavailable Number recording if the number is not assigned. However, I find that DIALSTATUS

I have two Adtran 750's connecting our analog phones to asterisk. On occasion, I get a channel that gets "stuck" off hook. 'show channels' shows: Zap/27-1 (longdistance s 1 ) Rsrvd (None) (None) And will just stay like that until the phone is manually picked up and hung up again (or asterisk is stopped/started). I guess this is a function of an unclean hangup (being

Hi Is it possible with Asterisk to tell the called party which number was dialled by the caller? Or in place of the number dialled have a description such as 'Sales' or 'Accounts'? Ideally, I would like to show a description corresponding to the number dialled followed by CIDName. How might this be set up? Currently my extensions.conf is: exten => xx,1,LookupCIDName

Hi all! I'm trying to have asterisk route all outgoing calls out via my VOIP provider. exten => _NXXXXXXX,1,Dial,SIP/BYEXTENSION@VOIP seems to have them to in the direct direction. However, debug shows that my asterisk doesn't identify itself correctly: Sip read: SIP/2.0 100 Trying From: "Evert"<sip:asterisk@[my IP]>;tag=as0aca53fa To: <sip:[dialled