similar to: detect SIP phone availability before dialing

Displaying 20 results from an estimated 4000 matches similar to: "detect SIP phone availability before dialing"

2005 Oct 17
6
initiate call recording from 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
2005 Oct 05
4
dropped calls when g729 is used on sip leg
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
2006 Mar 13
2
Simple php script to monitor asterisk calls
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
2005 Dec 28
3
voip-info: Asterisk record calls
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
2006 Mar 16
2
Queues Not Reporting Estimated Hold Time
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
2006 Jun 05
4
Local vs. toll Dial Plan
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
2006 Jan 25
4
Setting ringtone on Polycoms
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
2006 Mar 27
1
Master.csv Shell Script
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?
2005 Oct 12
2
Polycom: Button Remapping, HELP!
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
2005 Sep 27
1
blindxfer & atxfer not working?
I'm wondering whether there's a problem with the blindxfer and atxfer commands. I was using Asterisk STABLE and pressing the # key to transfer calls worked fine, except of course when you called up FedEx and they asked "Enter the number of packages, followed by the Pound key". I found on the wiki (http://www.voip-info.org/tiki-index.php?page=Asterisk+config+features.conf) that
2005 Sep 22
1
AgentRecord In and Out streams
How do I combined these in and out wav files on the fly through asterisk to where I hear the whole conversation and only have one wav-file (i.e. : agent-1001-asterisk-478-1127389080-17-in_out.wav) agent-1001-asterisk-478-1127389080-17-in.wav agent-1001-asterisk-478-1127389080-17-out.wav __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best
2005 Oct 04
1
Forcing Codec Usage
Hello, I have VPC (Voice Pulse Connect) and NuFone for providers and I have setup modules.conf with the registered (Digium) G.729 Codec such as: load => codec_g729a.so load => res_crypto.so With both sip/iax2 configuration disallow=all is first and then allow=g729 is next (allow=ulaw,allow=alaw,allow=gsm are next after allow=g729) and it always dials via ulaw. Why is this happening? Josh
2005 Oct 06
2
Mediatrix 1204 and Asterisk
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
2005 Oct 06
2
how do I know what codec is being used
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
2005 Oct 10
1
Outgoing quality
I'm having slight problems with outgoing audio quality on Zap channels. People hear an interrupted voice. Can anyone help..? Regards, Fabrizio Mazzoni Macron SPA
2005 Oct 17
1
How can I get a dialtone calling from outside...
Hi all, How can I configure, in extension.conf, to call and extension and have a dialtone so I can compose a number to dialout? Basically, I want to be able, when I am out of the office, to call in my asterisk box and then dialout from it. Regards, Francois
2006 Jan 05
1
Iaxy Ringtone
Hi all, I have a small query regarding ringing tones on an iaxy2. I have a customer who uses an iaxy to breakout to pstn via our *. However the customer complains that he gets no ringing tone whislt making calls, i just visited the site and can confirm this. I also have another customer who is presently in canada with an iaxy calling thru our * , he doesnt have this issue. I presume that the
2006 Jan 05
3
Remotely reboot SIP Phones ?
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)
2006 Jan 13
2
Asterisk echo & fxotune
Hey all, guess I didn't include enough info in my last query. I'm having massive echo problems on our sip<=>fxo connections & have been reading up on echo cancellation & such. When I try to run fxotune I get the following error: Could not fill input buffer Tuning module 25.Failure! The system has two of the digium 4port fxo cards & one T1 card (which isn't fully
2006 Jan 16
1
I've sent a message to the list 6 hours ago and it's still not showing up
I've sent a message to the list Asterisk-user 6 hours ago and it's still not showing up. I've seen others with questions about the availability of the list. It may be something the moderators want to check out.