similar to: Initial review of American Telecom X10001P DECT/SIP phone

I am looking for a way to allow a user to record a call simply by pressing a button or some combination of buttons on their phone. Is this possible? I have read the stuff about the monitor command; however, this doesn't seem to be very interactive for the user. Thanks, Andy -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216

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

http://www.wirelessiq.com/content/newsfeed/7319.html I'm surprised, I thought DECT was already available in the USA from my days selling this at Ericsson Australia back in 1995. Can someone confirm that they aren't already available? Cheers, Dean -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi folks: I know it's come up a few times before, but I need some more detail. I'm looking for a SIP DECT (cordless) phone for North American installations. I've heard only of the Siemens Gigaset S450/C450 phones. Apparently these aren't sold for use in NAm, even though they're supposed to be legal (in the United States, anyway). On top of that, I understand they have some

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

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

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

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

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

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!

Hi List; I read the following sentence: "The CLI prompt is set with the ASTERISK_PROMPT UNIX environment variable" In the following link: http://www.voip-info.org/wiki/index.php page=Asterisk+CLI+prompt The question is: what is the ASTERISK_PROMPT UNIX environment variable and where I can access it to change it? Also where I can find information about it? Regards Bilal Ghayad

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

We are running asterisk with a single POTS line for local calls and a voip line for long distance. Whenever we receive a call on the POTS line it is more than likely, but not always, going to have significant distracting echo. In addition to that there is occasional heavy crackle or static. I have tried to follow the guidelines at :

Dear Group, I have my Asterisk box working with a Mediatrix 1204. I have 2 questions; 1) I do not seem to get a Call ID on the call coming via the Mediatrix 1204. I was wondering if anyone had this configured and if they could share this with me? 2) How do you route a call based on caller ID on Asterisk. At the moment I'm routing calls via DNIS. Thanks and Regards Shad Mortazavi

Hi, This may be a stupid/easy question for many of you. Q. how do I know what codec is being used for a particular call or call leg? Thanks. AK -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051006/5c225e0b/attachment.htm

Hi, Can you give me some councils of remotely rebooting sip phones in asterisk server? How to configure sip_notify.conf and sip.conf? Kind regards, Guan ; Reboot Polycom Phone Event=>check-sync Content-Length=>0 ; Untested (Reboot Sipura Phone) Event=>resync Content-Length=>0 ; Untested (Reboot GrandStream Phone) Event=>sys-control ; Untested (Reboot Cisco Phone)

Where can I investigate the origin of brief silences during calls from/to my SIP phone? Only rare pauses of 0.5-1.0 sec when I'm not able to hear anything. Thanks Mimmus

Can the # be used as a valid key press for a user in a dial plan? if so how can the asterisk recognize it as a valid key press?

Hi, I would like to try the new echo cancelers in zaptel 1.2.3, but don't want to switch to Asterisk 1.2.x just yet. Anyone can tell me if zaptel 1.2.3 will work with Asterisk 1.0.9? Thanks, Andre

Hi all, I installed the Asterisk 1.2.5 and asterisk-addons 1.2.1 of a new Red Hat linux box( Linux version 2.4.20-8smp). I was able to compile both the software but when i start Asterisk, it exits with the following dump. Error Text Start========================= [res_crypto.so] => (Cryptographic Digital Signatures) -- Loaded PUBLIC key 'iaxtel' -- Loaded PUBLIC key