[Asterisk-Users] making ASTCC web page secure ???

How do you make the page http://hostname/cgi-bin/astcc-admin/astcc-admin.cgi secure ? , so that only the person administering the calling cards can see the page and make changes to the calling cards, I was thinking of using .htaccess to restrict the access to the page by requiring a password,

[Asterisk-Users] incoming call high failure rate on pickup of call.

Hi, I have a sipura 2000 ATA connected to an asterisk server on the local network, and the POTS line connected to asterisk using a X100P clone, when calling remotely through the X100P (incoming call), the phone attached to the sipura device always rings like it should, however sometimes it does

Re: [Asterisk-Users] Astrerisk + Conversation OneWay

Hi, I have seen this problem before when using sip, I have never seen this probme using IAX2. in the sip soft phone if you are using a proxy you will probaly need to disable the use of stun, but if you are not using a proxy you may need to use stun. anyway if you are behing a firewall you will

Re: [Asterisk-Users] SPA-2000

Hi, I have not implemented any of the spa-2000's yet. Do they work ok with asterisk? Is the 2000 capable of having 2 FXS extensions off each one or is it two fxs ports with the same extension? ___ Asterisk-Users mailing list

Re: [SPAM] RE: [Asterisk-Users] Webmin Module for Asterisk (and thirdlane)

-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users-[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, January 21, 2005 9:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Webmin Module for Asterisk (and

[Asterisk-Users] incoming calls timing out.

I have an stanaphone number for incoming calls, I am using asterisk and it works fine at first, but after a few hours the incoming calls fail they go directly to the stanaphone mailbox. I believe the registration probably times out, or maybe it is the dynamic IP changing. Is there a way to fix

Re: [Asterisk-Users] SPA-2000

Which codecs do you use for the second call? One limitation of the sipura 2000 is that you can not use both ports at the same time with the G729 codec, I belive this may be due to the sipura having an smal CPU that can not handle the load of 2 G729 codecs. Other limitations are the lack

[Asterisk-Users] Stanaphone incoming calls problem.

Hi, I have an stanaphone number, and when I first start asterisk everything works fine, but after a few hours of asterisk running incoming calls fail, and when calling my stanaphone remotely, it takes me directly to the stanaphone voicemail, restarting asterisk fixes the problem for a few hours.

Re: [Asterisk-Users] MeetMe

Hi, You need ztdummy. Follow this link, it worked for me, http://www.voip-info.org/wiki-Asterisk+timer+ztdummy Thanks, Guru. I have just started using * and have been trying to set up MeetMe. So far I have not been able to start a conference. When I dial the conference extension (I am using

Re: [Asterisk-Users] MeetMe

Hi, You need ztdummy. Follow this link, it worked for me, http://www.voip-info.org/wiki-Asterisk+timer+ztdummy Thanks, Guru. I have just started using * and have been trying to set up MeetMe. So far I have not been able to start a conference. When I dial the conference extension (I am using

Re: [Asterisk-Users] MeetMe

Hi, You need ztdummy. Follow this link, it worked for me, http://www.voip-info.org/wiki-Asterisk+timer+ztdummy Thanks, Guru. I have just started using * and have been trying to set up MeetMe. So far I have not been able to start a conference. When I dial the conference extension (I am using

[Asterisk-Users] Asterisk and Cisco Call Manager.

nothing in it. Any steps, any config, any help would be highly appreciated. Thanks Regards, Gurdeep (Guru) +91-11-35372111 [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users