similar to: "unclean hangups" can I turn off hook flash?

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

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) ;

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

Hi All, I have just migrated from Asterisk 1.0.0 to Asterisk 1.0.5 and I have an X100P installed. The old asterisk was working, but now the new version isn't picking up any calls! However, I did notice that after installation, I performed modprobe zaptel and modprobe wcfxo and they worked fine, but when I executed ztcfg, I get the following errors: ioctl(ZT_LOADZONE) failed: Invalid

All, Alright, I've looked around the internet, the voip-info.org wiki, and browsed the contents of this mailing list. While I've found a couple of scenarios that are close to this one, I haven't found one that uses my particular card (T100P). Without further delay -- I have successfully configured internal SIP services between a Snom 200 and a Windows X-Lite client and have

I have a bit of a weird problem that I'm having great trouble debugging. I have a TDM400P PCI card with two FXO and two FXS modules. Both FXO modules are connected to BT lines here in the UK. Both BT lines have V23 Caller-ID, which works fine with Asterisk. Both asterisk and zaptel are fresh from CVS. Both FXO modules (channels 3 and 4) are in "group 1" for outgoing calls. My

Good Day All, I am experiencing some weirdness using the E&M channel and hope you can offer a little assistance with the problem I am having. 1) call comes into channel 25 (Second Span first channel of a Digium Quad PRI from SBC-PRI) 2) Call is sent to channel 1 (First Span first channel on the Digium Quad PRI connecting an ADTRAN via E&M Feature Group D) 3) Between rings one and two

A couple of weeks ago I posted a message entitled 'Bridged trunks stuck off hook' about a situation where 2 of my trunks (loopstart pots - but Centrex) are occasionally bridged together. It has occurred to me that what may be happening is that a line hung up by Asterisk might quickly be reused and if the time interval is right, the telco will interpret this as a hook-flash. Is there a way

We are having line noise issues in our Asterisk to legacy PBX integration. All SIP calls originating from IP phones sound crystal clear. All calls that originate from the legacy PBX (Isoetec 228) and route through the Asterisk and out SIP have a lot of line noise. I believe I have it pinned down to these Blue Alarm errors that I can see on the legacy PBX side. zttool shows no alarm but when I

Hello. I'm trying to transfer calls from an analog phone (Zap/1, TDM400P card) to some other terminal connected to my Asterisk PBX. If I make a flash hook pressing the phone hangup button quickly it works as expected, I get a new dialtone and the other side is put on hold. But I would like to use my phone's "R" key instead for some different reasons (it's quite easier to use

In the past I used SetGroup and CheckGroup to figure out if my allowed providers lines are all used or not. Since most of my provider have given me a single line anyway, I wonder if there is a way to check if this (provider) line is taken already. How can I do that? Same is with the phone. How can I see in CLI if a phone is now in use or not? "Sip show peers" shows me just if it is

# 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

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

Help. Why is it that I can call out from my GSBudgetone SIP phone but the audio is "one-way'? Why is it that when I call my asterisk phone number, I get a fast busy?

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

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 everyone, I've been experiencing many problems recently with windows software parsing large text files stored on a samba share (smb version 3.6.3 on ubuntu12.04). The software, which are running on windows XP stations, are crashing quite often. I notice this in the log.smbd at the time of the crash: [2012/08/28 17:25:21.980609, 3] smbd/server.c:629(smbd_parent_housekeeping) Printcap

We're having some difficulty connecting a T1 fax board card to asterisk, with E&M Wink. We've successfully connected it to other E&M Wink T1s, but have been unable to get asterisk to hear it's dialing. Asterisk can dial down to it with no problem, and it hears all the digits, but whenever it sends digits down to asterisk, * always only "gets" the first one. This is

Suppose I have the latest and greatest e2fsprogs. >From reading the docs I understand that 1) boot ext3, clean umount - you can remount ext3 or ext2 2) boot ext3, unclean umount - you can remount ext3 - you can e2fsck and remount ext2 Does this mean that normal linux init script e2fsck will do it, in case fstab says ext2? Or you should make sure to run e2fsck by hand with -f? 3) boot

Hello, If I set my network interface to have "logunclean" along with "dhcp,norfc1918,routefilter,noping,tcpflags", then when I connect to http://welcome.hp.com/country/us/eng/support.htm and choose any of the product I get this. logpkt:LOG:IN=eth0 OUT= MAC=00:a0:cc:5b:09:5f:00:08:e2:32:34:70:08:00 SRC=192.151.11.205 DST=24.24.243.178 LEN=80 TOS=0x00 PREC=0x00 TTL=239 ID=14025