I have two comments:
a. It maybe doesn't work because of the PCI specifications the box support.
If was manufactured before Jan 2000, it is quite probably that it won't
recognize the Digium cards.
b. From the point of view of load, I see no problems, I think the specs of
the machine are enough for
You don't need the zaptel library if you aren't going to use any digium
cards.
Marco
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
[EMAIL PROTECTED]
Sent: Thursday, February 17, 2005 8:02 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
Go to the ITU website www.itu.org there you can buy all the specifications
you're looking for.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Daniel
Nyström
Sent: Tuesday, February 15, 2005 8:47 AM
To: asterisk-users@lists.digium.com
Subject:
The
complete configuration of such a system requires a lot more of information that
the one you gave.But, at a glance, Asterisk + SER is a good choice for
this kind of venture. Asterisk can serve as the PSTN gateway (ISDN PRI
connections primarily) and Voicemail server. SER can manage the
Have you set your DNS SRV entry for SIP correctly???
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Andrew
White
Sent: Tuesday, February 15, 2005 7:04 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] newbie: help two cisco phones (sip)
Hi,
Dear Xu, my name is Marco Castillo, I'm in Guatemala, Central America, and I
have recently succesfully installed a TE110P here in Guatemala. There are
many implementations of a E1 or T1, but I think that the great majority can
be configured via the zaptel drivers. I will suggest you to buy a card
Remember that SIP uses DNS SRV entries, maybe one of the phones you use
efectively use the DNS SRV entry and the other not. Some VoIP phones have a
flag where you can deactivate this functionality for SIP. If not, make sure
you have in your local DNS a SRV entry for SIP.
Hope this helps.
Marco
.
Marco
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Peter
Svensson
Sent: Thursday, February 10, 2005 6:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] No dialtone in a E1
On Thu, 10 Feb 2005, Marco
Be sure you're using Debian with kernel 2.4.20 or more (if you use the
stable release version of Debian, using dselect will just upgrade your
kernel until 2.4.18, and the zaptel libraries won't be properly compiled).
Using dselect o apt, get the kernel-sources packages and the kernel-heades
Hi, I'm having a little problem when trying to make a call from asterisk. I
connect a SIP phone to asterisk, and in the asterisk box I have a TE110P
card connected to a E1. When a SIP client makes a call through the E1, I
received no dialtone in the SIP client.
In the same manner, when somebody
For
softphones, I used SJPhone, is a very good SIP phone, and I have it working on
asterisk. The setup is kind of tricky, 'cause you must remember to set the sip
register in your local DNS.
For a
good overview and introductory tutorial to asterisk, go to the asterisk home
site
I recently have purchased a new TE110P card, that provides a single T1/E1
port. I have installed it and everything works fine, except for the dial
tones. When I made a call from a SIP phone to a channel in the TE110P, I
receive no dial tone. When I receive a call in a SIP phone from a channel in
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