[asterisk-users] ekiga register problems

returning newbie. Trying to register ekiga for the first time to my asterisk server only. [204] user=204 context=internal type=friend secret=xxx insecure=very canreinvite=no host=dynamic disallow=all allow=ulaw allow=alaw nat=no Can anyone tell me what I am missing? I am not

RE: [asterisk-users] ekiga register problems

:17 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] ekiga register problems On Mon, May 28, 2007 at 10:45:25PM -0400, Brad Sumrall wrote: returning newbie. Trying to register ekiga for the first time to my asterisk server only. [204] user=204 context=internal type

[asterisk-users] auto/forced call

Can anyone guide me to a how to on automating a call? I know a little piece of code (normally python) has to be place some where and then a file has to be mv into the spooler. Where do I get the run down? I have a button on another application that sends an email and I want it to also send a

[asterisk-users] Asterisk GUI issue, minor

I installed the asterisk GUI, Asterisk web manager, it loads fine, but if I go to the AGI section, I get a permission denied Obviously apache cannot access the /etc/asterisk directory. I added apache as group, but still the same problem. Suggestion any one?

RE: [asterisk-users] Polycom Provisioning Problems

Access the phones through the web interface, Compare version numbers with the phones that work Compare only with other 501 phones Make sure all settings are identical, most polycom web interfaces will loose there setting adjustments if you click on another tab, so do one page at a time, click

RE: [asterisk-users] Asterisk Pix firewalls

Pix usually uses NAT, A quick fix is to simply forward the ports in your NAT statements. If the pix is new, call Cisco and cheat like I do so often! Brad _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Nuñez Sent: Wednesday, April 25, 2007 9:31 AM To:

RE: [asterisk-users] SLES?

This question should really be asked at Linux. Basically FC, Red hat, Centos and SUSE are all the same. Some minor security defaults and a few directory changes. Last time I check (it has been some time now); All of the above on their enterprise level basically only supported the install, updates

RE: [asterisk-users] call dispatching - legacy application

Then you take the number you get from your database and put it into the asterisk spooler. Remember, the temp file you create has to be moved to the spooler using the mv command. Nothing else works. There might be one other step, I am not sure with Asterisk 1.4. I had a friend help me do it

RE: [asterisk-users] No Audio with SIP to only one provider whenswitching servers

I would not rule your firewall out as the problem! Port 5060 is only the authentication port, the rtp stream is normally 10,000 thru 20,000. Some of your phone may have STUN modules on them. Open 10,000 thru 20,000 and 5060 on the firewall. Stick some holes in it for testing purposes. Verify

RE: [asterisk-users] Marketing 101

Personally, I look for specialty applications. Work smart not hard! I myself am looking for outstanding marketers for a fire hot industry / telecom application. I have all of the correct duckies in a row, just need to send it to the market the correct way. [EMAIL PROTECTED] _

RE: [asterisk-users] Polycom SP 601 Reboot Issue- Help!

Hard reset the phone first! Provision and see if it is fixed. No? Upgrade software (watch out for provisioning changes). Still rebooting? Downgrade software. Still rebooting? You now have a new door stop! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of

RE: [asterisk-users] Random Asterisk deaths

test -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wayne Jensen Sent: Wednesday, April 25, 2007 7:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Random Asterisk deaths Asterisk 1.2.13 (newest available

RE: [asterisk-users] SLA Appearance between 2 Cisco 7960's (SIP)

I am very confident the 7960G has a sip load. I know for sure the regular 7960 does and the G just means gigabit interface. The 7970 was the only one that didn't because of all the color interface/touch screen, and Cisco was still pushing call manager big time, so skinny was the only load

[asterisk-users] Need help making a voice record server $$$

Hey there folks, Looking to my favorite mailing list for assistance and have a few bucks to pay you for your time. Me: Played with asterisk for a while in the early days and getting stuck on silly stuff on a time sensitive project for a friend. Project: PSTN incoming call to asterisk and then

RE: [asterisk-users] Re: Question about DSP in Digium card

Whether it is IAX, SIP, H323 or ? These are authentication handshakes to establish an rtp stream. SIP = user name and password in a standardized IP packet IAX = same H.323 = same Is also has to do with what codec are supported as well. As far as NAT is concerned! Yep,

[asterisk-users] Refresher course needed!

Hello everyone My name is Brad, I am an old Asterisk Vet of the very early days just coming back to join the group. Ok, for starters, I feel like the monkey with the light bulb looking at extensions.conf and sip.conf. It has been some time. A friend ask me to set up a asterisk server that

[asterisk-users] Zaptel silly issue

I am geet this error, I assume because I have zero digium hardware installed. This is to be an entirely web based PBX. Can anyone point me to an easy 123 for installing zaptel in dummy form? I need music on hold for a VPS server. Brad -Original Message- From: [EMAIL PROTECTED]