similar to: initiate call recording from phone.

I need to find a way to have the Polycom phones automatically park calls. Right now my users hit #70# (I know the last # is optional but it speeds it up.) to park a call. Personally I think this is easy, but my users would like one button to do this for them. The Polycom has buttons we don't need (Transfer & Services), it would be nice if I could remap one of those buttons to dial

I can not figure out what the heck is going on. I went back to my old version and I still get errors when the voicemail system tries to load any of the greetings, unavail messages, etc. the normal voicemail prompts work, but any user recording don't work. Leaving a new message appears to work, but the system wont replay them, it throws errors. Here is an example of the errors: Oct 11

I migrated to a new version of the voip connection vs1 server software and I am now getting these errors when I try to call my voicemail. Any thoughts? The files are there, so I don't get it. Oct 11 19:57:26 WARNING[6587]: format_wav.c:140 check_header: Not a wav file 49 Oct 11 19:57:26 WARNING[6587]: file.c:418 ast_filehelper: Unable to open fd on

I am running CVS-HEAD-04/12/05-21:44:31 and I am curious if it is possible to customize the email message sent to the pager email address. Thanks, Andy -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216 agoss@ntad.com

Whenever we call IBM, the call counter on the phone never starts and in the CLI the zap channel never gets the answered signal from the PRI. See below. -- Executing Dial("SIP/5933-645d", "Zap/g1/18004267378") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/18004267378 At this point, I am in IBM's menu system. However the call never

Thanks for the reply. Forgive me for being na?ve, however have jumped in to this asterisk project at work due to some circumstances beyond my control and I don't know a lot about carriers and how this all works. I am figuring it out, but it's a lot of trial by fire. As far as I know, we only use 1 carrier for our system. We have a PRI from NuVox and we use 7 channels for our asterisk

Hiya, hope I don't bore anybody with this. There are certainly a lot of monitor-y things out there and they just didn't fit my need, so maybe this will fit someone's besides mine. http://horanappraisals.com/asterisk/pbxmonitor/ contains two files. one is a php script called pbxmonitor, and one is a flat file of extensions to extension name mappings of internal users. It

Hi, I have lookedaround and don't see this anywhere. Is there a way to tell the ip500 to not make the aural MWI blips?

Hello - I have 8 polycom 501s all setup great using ulaw. We have put them through a pretty rigorous torture over the last 4 months, and they've performed famously. No dropped calls ever. We invested in some g729 licenses. changed my ipmid.cfg so that g729 is priority 1 and ulaw is priority 2. I added allow=g729 to my extension's sip.conf entry, where existed before disallow=all

> Watch the output of 'pri debug span 1' on the Asterisk server while > placing the call - bug #4468 (http://bugs.digium.com/view.php?id=4468) > might be relevant. Yes, this is exactly what is happening. Thanks a lot. I am thinking about adding a special case for the IBM 800 number since it is the only one my company is complaining about. Currently I have this in my dialplan:

On this page http://www.voip-info.org/wiki-Asterisk+record+calls there is "Example by Mojo". I have done everything he said and I have sox package installed. [root@pbx recordings]# sox -help sox: Version 12.17.7 ... When I open this web page http://10.0.0.26/recordings/index.php I get this: No Recordings Found And there are recordings in /var/spool/asterisk/monitor Do I have to do

I am running 1.2.5 with a simple queue and have announce-holdtime = yes in queues.conf for that queue. The person is being told their posistion in the queue and the CLI says the estimated hold time, but it never plays it for the caller. It worked previously, i am not sure when it stopped, i think after 1.2.1. Is this a known bug? I dont want to report it to the bug tracker if its already been

Hello all, I need some help with my Cisco 7960 enabling TFTP. Does anyone know what numbers to press in the menu? Or can I enable this through telnet? Many thanks, Christian

Ok asked this earlier with no response so I will phrase it a different way. I am sure someone had to deal with this and there is a "best way." I want to let Asterisk make the decision on best path based on local exchange - xxx-yyy - where xxx is one of my local area codes and xxx is exchange designator. The problem is that the list is rather large. Maybe 50-100. The idea is that I can

i have ordered 500s from tritechcoa.com several times over the past 4 months. great service and delivery, and the prices are lowish, only problem is, they add a $20 handling fee per phone, on top of phone price, and shipping, making the lower price not as good -----Original Message----- From: Mojo with Horan & Company, LLC [mailto:mojo@horanappraisals.com] Sent: Friday, July 15, 2005 12:01

Hi, I'm having trouble setting the ringtone on my Polycom 501. The relevant entry in extensions.conf is: exten => 801,hint,SIP/creative1 exten => 801,1,SetVar(ALERT_INFO="Test") exten => 801,2,Dial(SIP/creative1,20,Ttr) In the sip.cfg: <alertInfo voIpProt.SIP.alertInfo.1.value="Test" voIpProt.SIP.alertInfo.1.class="13"/> and <TEST

If anyone could tell me what this error is all about, I would be very grateful. Oct 12 01:42:53 WARNING[2724]: app.c:1109 ast_lock_path: Failed to lock path '/var/spool/asterisk/voicemail/default/5933/INBOX': Operation not permitted Oct 12 01:42:53 WARNING[2724]: app.c:1109 ast_lock_path: Failed to lock path '/var/spool/asterisk/voicemail/default/5933/Old': Operation not

Hi all, Configuration: Analog phone connected to TDM400p. I'd like the phone to give a half-ring (chirp) periodically when there is a message waiting. Can this be done? How is it configured? The visible "Message waiting" indicator and the stutter dial tone are working fine, but are not sufficient for me. Thanks!

Im not looking for anything super detailed, just something to run through the master.csv file and give total time per account code. . . .does anyone out there have a script like this I could work from?

Originally posted by: mailto: Hi all Now I'm making IVR sequance that is customised [mainmanu]. I wish to notify invaid command like a following exten => i,1,playback('your command is ...') exten => i,2,playback(${EXTEN}) ; <---- Say 'i' oops! ;-( exten => i,3,playback(' is incorrect! please again ') # This exten lines are figure for instruction. # I