similar to: HELP NEEDED! - Asterisk GUI

Hi, I have been doing various testing with asterisk and its been going great. However I am a bit feedup of using vi for editing configs, and would rather do it from any machine on my LAN. I am running debian and * via xorcom rapid on a test PC at the minute. Hence phpconfig would be great, however I am having difficulty getting it to work. I have searched the message boards and the wiki, and

No, I have apache 1.3.33 and mod_ssl 2.8.22 installed. Do I need to have apache2-mod_php installed? Rgds, Julius. > DO you have apache2-mod_php installed ? > > Which distro are you using ? > >> -----Original Message----- >> From: asterisk-users-bounces@lists.digium.com >> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of >> Julius Kidubuka >>

> When I do click on the phpconfig.php link from > http://ip-of-machine/phpconfig/, it returns a page with the actual > contents of that file (phpconfig.php) and doesn't load the page. See some > of the output below; Try a simple php-script in this directory. Something like this, name it test.php <?PHP print (test) ?> If it does not print "test", and you see the

> -----Urspr?ngliche Nachricht----- > Von: Mojo with Horan & Company, LLC [mailto:mojo at horanappraisals.com] > Gesendet: Dienstag, 25. M?rz 2008 23:23 > An: Asterisk Users Mailing List - Non-Commercial Discussion > Betreff: Re: [asterisk-users] Asterisk parking hold and > transferdigittimeout > > It seems that the dialplan comes into play. If your parking >

Hi, anyone out there with the same problems and a possible solution to the following? The functions callparking and hold use the same transferdigittimeout in features.conf. While I think 3 to 5 seconds are enough to let the user "find" their keys on the phone, the double ammount of time ( 2 x 5 secs) you have to wait before a call is parked and the parkposition is announced, is

> We have our Asterisk server running smoothly with a SIP BRI gateway > for inbound calls. However if the Internet connection goes down and a > DNS server becomes unreachable Asterisk basically does not function. > By this I mean it does not answer call coming in from the gateway > (which is on the local LAN) and you can't even reload it - just hangs > there. If I change the

Hello, I have SER and Asterisk running on the same box. I want SER to listen on port 5060 (it is) and Asterisk to listen on port 5062. I have configured my phones to register with x.x.x.x:5060 (SER) and Asterisk will purely act as a voicemail server at the moment. However I cannot get Asterisk to listen on a different port. It is my understanding that I just need to set the port in sip.conf

I'm having a problem getting my queue to function as it should. After 20 seconds or so, it should prompt the user with a message "thanks for holding..... press # to leave a message or stay on the line to continue holding". I set up the "context" in the queues.conf file, so if a user presses a digit, they should be able to leave. But I get a SIP BUSY message. Here are my

Hi, I have installed two X-Lite phones and they're able to login successfully. The two phones plus the Asterisk system are all on the same LAN with private addresses assigned to each of them. When a call is initiated and is picked up on the other end, there is completely no sound at all (as in the line goes dead). The codecs set in the softphones are g711u, g711a, GSM, iLBC and SPX.

Hi, When I do click on the phpconfig.php link from http://ip-of-machine/phpconfig/, it returns a page with the actual contents of that file (phpconfig.php) and doesn't load the page. See some of the output below; <?PHP /** * * Asterisk configuration file interface script * * * * * * phpconfig:,v 1.0 2003/07/03 17:19:37 * Authors: Dave Packham <dave.packham@utah.edu> *

Here is a part of a working sip.conf for Asterisk with SNOM190 Phones. Try using host=dynamic and have a closer look at the configuration in the snom 190. Also, try using dtmfmode=rfc2833 . [general] realm = hallinux2.gwsnettech.local port = 5060 bindaddr = 0.0.0.0 context = default disallow=all allow=alaw allow=ulaw allow=gsm register => 081503:xxxxxx@sipgate.de/081503 language=de tos=0x04

hello i need to know how to enable the feature in the agents.conf to make the users got to press # to answer the call when is in the queue and the agent is logged in. at this time the call enters the queue and the agents who is logged in only beeps once and then the call enters automatically. can anybody help me?? TIA Edgar

I need to be able to send an sms alert to one's mobile/cell phone. For instance, when I receive a voicemail message in my inbox, I also want to be able to get a message on my cell phone alerting me of this e-mail. How possible is this? And if it is, what do I need to do to get the service up and running? Ideas are most welcome. Thanks, Julius.

Hello I have a Junghanns ISDN BRI card for incoming calls and use SIP Polycom IP300 phones. My config files look like this: features.conf pickupextn = *8 zapata.conf context=frompstnisdn group=1 callgroup=1 pickupgroup=1 I also edited sip.conf like this: group=1 callgroup=1 pickupgroup=1 But on internal and incoming calls if I dial *8 from any phone I cannot pickup. Do I need to add

Hi all, I am able to receive voicemail in my mail box but when I try to play the audio file attachment, I hear nothing at all (yet the caller on the other end does leave a voicemail message)! Anyone had a similar problem before? Ideas are welcome! Note: I am using Asterisk@Home 0.6 Thanks in advance, -- Rgds, Julius Kidubuka. "My advice to you is get married: if you find a good wife

Does anybody have any experience using a Patton SmartNode as a SIP/Telco gateway with Asterisk? They seem really inexpensive and appear to support all of the necessary features, but I don't have any experience with their products, so I don't know if they are any good. We are currently using a Cisco 2600 w/ PRI card and it works fine, but I was looking for someone else as a possible

Hello all. I'm having an odd problem getting * and sipgate to work together. From Sipgate support I have gotten this repsonse to my query: ===== Your Asterisk is registering incorrectly with our servers. It registers like this: sip:s@217.XXX.XXX.XXX:5076 The "s" should be your SIP ID. Anything else is rejected. I don't know where you can find this setting, but from our

Hello everyone, I am trying to install windows xp as a guest OS. It runs through the installer just fine, but every time I reboot, it keeps booting back to the installer. Here is my config: ------------<code>--------------------------------------- kernel="/usr/lib/xen/boot/hvmloader" builder="hvm" memory=512 device_model=''/usr/lib/xen/bin/qemu-dm''

After getting my feet wet with asterisk@home, I want to set up a second asterisk box to add a call shop billing and other add-ons such as LCR. My question is as follows. Is there a backup program that will save to a tape drive or a USB CD Writer so if I mess up an install I don't have to go through a complete reinstall? I saw a few programs out there but they required X windows and from

Hi, I am looking for a simple solution to do this. I wish to have the user to enter their preferred method of connection i.e. for the cheapest solution to their desktop phone or mobile phone, then plan callfile based on the number that user provided and dial to the user. Any suggestions? CK -------------- next part -------------- An HTML attachment was scrubbed... URL: