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, how
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
not
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 ne
> 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
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 th
>
>
>
>
>>-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 Asteri
> 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
> Asterisk-
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.
I
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
> extens
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
> extens
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
> extens
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
> extens
nothing in it.
Any steps, any config, any help would be highly appreciated.
Thanks & Regards,
Gurdeep (Guru)
+91-11-35372111
[EMAIL PROTECTED]
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