similar to: Ring tone changes when asterisk answers the call

Hello, Over the past several weeks, we have been having an intermittant problem with our deployment of a TDM400P card (4 fxo). ?We have tried many things, and the problem still re-occurs. The Problem: Occasionally (every 48 hours), the TDM400p card will stop answering incoming calls on ALL fxo ports. ?Attempts to send outbound calls on any Zap channel will result in hearing a loud

ztdummy successfully compiles under kernel 2.6, but when I load it I get ztdummy: Unknown symbol fill_td ztdummy: Unknown symbol insert_td_horizontal ztdummy: Unknown symbol uhci_devices ztdummy: Unknown symbol uhci_interrupt ztdummy: Unknown symbol alloc_td ztdummy: Unknown symbol unlink_td ztdummy: Unknown symbol delete_desc I had a quick look at the source, and it looks like these function

I followed an asterisk howto at http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html and everything was setup perfectly. I logged into the voicemail, changed my greeting, but it still played the old one. I checked the proper folder in /var/spool/asterisk/voicemail/ and I found 3 gsm files that had the default messages, and the 3 wav's that I recorded. I have format=wav set in the

Hi all, I curious to know if other UIP200 users have this same issue: You flash (XFER button) to consult-transfer a caller to another extension. If the transfer target party is unavailable (ie: voicemail), there appears to be no way to get the original caller back. If it's a known limitation, has anyone come up with a functional work around? Thank -- ..................................

I couldn't be happier with the simplicity of this - but it's a hack! Hi all, I'm currently using a SIP client (BT101) to connect via DSL to a remote instance of Asterisk. - Asterisk has a private IP behind my OFFICE router. - The SIP client has a private IP behind my HOME router. I'm doing this _without_ the use of STUN or proxy servers. Here's how it works: -

Hello all, The "Secret Agent" final release of the Asterisk Management Portal is now available for download: http://amp.coalescentsystems.ca/ This exciting new release adds a great deal of functionality and flexibility. Thank you for all the contributions and feedback! 1.10.007 - Added AMP Users (multi-department, basic multi-tenant) - Added incremental upgrade script

Hi, We've recently deployed 6 Uniden UIP200 phones (running firmware 4.54). We've been having some serious problems: 1) All the phones randomly reboot themselves. Typically when trying to answer or initiate a call. 2) All the phones will disconnect from a calls with the PSTN after 2-3 minutes. 3) The phones are unable to interact with a remote IVR (digit presses are not received at

Hello all, Asterisk Management Portal 1.10.009 has now been released. This exciting new version has several notable additions (listed below). The AMP homepage is http://amp.coalescentsystems.ca. Here you'll find links to the download, install guide, and documentation wiki. As usual, please use amportal-users mailing list for discussions about AMP:

I compiled libtiff version 3.6.1 and spandsp and spandsp version k. When trying to load asterisk I get the folloein error: Jun 7 10:15:03 WARNING[16384]: loader.c:408 load_modules: Loading module app_dtmftotext.so failed! Ouch ... error while writing audio data: : Broken pipe [root@zapata root]# Warning, flexible rate not heavily tested! Please help! -- Manuel Marin Garcia TRANSTELCO S.A.

I have Asterisk CVS-HEAD-05/25/04-17:13:22, Copyright (C) 1999-2004 Digium. Usign exclusively digium hardware. 3 TDM400P cards. 1 4xFXO 1 4xFXS 1 1xFX0 & 3xFXS When * is attending FXO calls, bridged to FXS calls, natively ofcourse, at a random time, the call hangus up. Also, for example, if a call is done, and an other extension hangup, there are some probability that the other extension

Hello, Since outgoinglimit is EOL'd, I've implemented SetGroup/GetGroupCount to ensure that SIP clients will only have a single call at any time. Works perfectly for simple calls using Dial(). I'm now struggling to find a way to similarily limit 2nd calls to SIP clients that are Agents, who receive their calls from a Queue(). Is there any way to accomplish this (without writing

Does anyone know how (if possible) to do three way calling on the UIP-200? There doesn't seem to be much info about this phone, but all the feature lists I've read says it can do conference calls. I can't seem to do it, though. Any help would be appreciated. -- Seth "et lux in tenebris lucet" Mattinen sethm@rollernet.us

I'm trying to set up basic monitoring for a specific extension (5004) to record all outgoing and incoming calls and save them as WAV files. I've set this in the extensions.conf file: exten => 5004,1,Answer exten => 5004,2,Wait,1 exten => 5004,3,SetVar(CALLFILENAME=/var/spool/asterisk/MONITOR-${TIMESTAMP}-${CA LLERIDNUM}) exten => 5004,4,Monitor,wav|${CALLFILENAME} But it

I got a Uniden UIP200 and started to configure it and I am lost.... I have a tftp server setup on my * server and have the files unidencom.txt and uniden<mac>.txt there. But it doesn't quite work yet. It registers as a sip phone (sip show peers), but I cann't dial it and the display shows #1 disconnected all the time. It has firmware version BS4.59a in it. I have no idea if I

hello, yesterday the uniden uip200 phone was recommended to someone. i am looking for an alternative to grandstream bt-100 because i can not do a supervised tranfer with it. here my questions: 1) does the uip200 support supervised transfers? 2) can i buy the phones in europe, especially in germany? thanks in advance, jan goericke

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031223/62cf404e/attachment.htm -------------- next part -------------- Hello When I tried loading TDM400P module using insmod command, I get following error messages. Is there some problem with my asterisk installation. Please advise. Thanks Tony $insmod wcfxs Using

I am unable to compile the zaptel drivers on the latest kernel for fc 3, I get the following errors which are listed below if anyone has any suggestions on how I can solve this issue aside from trying a different distro, please don't hesitate to offer. Thanks in advance. [root@asterisk-test2 zaptel]# make linux26 make -C /lib/modules/`uname -r`/build SUBDIRS=/usr/src/zaptel modules make[1]:

Hello. I've seen several posts talking about line quality using Digium cards that are sharing IRQs or on machines where X is running but after trying all of those fixes I am still having a problem with line static on outoing calls. BTW, calls that are from one extension to another extension have no static, however, they have occasional clicks and pops. At any rate, I was wondering if

OK, last showstopper that I just can't puzzle my way through - parking calls with the snom phones. I get the two phones connected, I hit transfer on one, the other phone goes to MOH and the first phone gives me DT, so I dial 700 and hit the OK button. Call transferred, the SNOM hangs up before I have a chance to hear which extension it parked to. Is there a way to make the SNOM phones

Hi! I have just bought the X100P and TDM400P cards to install on my computer to implement the PBX. I also downloaded the newest softwares asterisk_ver0.5.0, libpri_ver0.4.0, and zaptel_ver0.7.0) to install on my computer (Red Hat Linux 8.0). All packages are compiled well. When I use "modprobe" to load drivers (modprobe zaptel, modprobe wcfxo, modprobe wcfxs), the first two (zaptel,