Displaying 5 results from an estimated 5 matches for "port11".
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2004 Apr 12
2
Random disconnect of calls
Hi
I am experiencing some weird behaviour. Calls get disconnected random.
There is no error in the log files.
Sometimes I can talk over 30minutes+ and it is fine. Just earlier I was
only able to talk 2 minutes per session and get disconnected. All I hear
when this happens is a fast busy.
My set up is this: 8 * Grandstream Budge Tone 101. 4 * X100P cards.
Compaq 1Ghz ML Server.
I am running
2008 Jul 07
8
US T1 Hangup Detection
We are in the process of preparing to move our Asterisk server to a
Digital T1 interface card instead of a analog card (via an Adtran
which is now connected to the T1). I did a preliminary test the other
day and hooked the T1 line up to the T1 card, bypassing the Adtran.
This worked rather well I must say. The two issues I ran into are:
1) Caller ID is not working even though I enabled
2004 Apr 08
0
RE: Asterisk-Users digest, Vol 1 #3368 - 12 msgs
...ers@lists.digium.com
Is it true that every time we make a change in the configuration file we =
need to restart the asterisk server. This will not be practical in the =
production environment.=20
Thanks,
--__--__--
Message: 7
Date: Thu, 08 Apr 2004 09:58:51 -0400
From: Thomas Gallaway <rescue@port11.net>
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Restart Asterisk
Reply-To: asterisk-users@lists.digium.com
Jain, Sonal wrote:
>Is it true that every time we make a change in the configuration file we need to restart the asterisk server. This will not be practical in t...
2004 Apr 29
3
Dropped calls -> reproducing scenario
So I think I am able to reproduce the dropped call scenario.
Here is what I do to get a dropped call:
1. Call 1-800-tmobile
2. Go true their IVR and get connected to the customer service IVR
3. Enter my number and SSN
4. press 0
5. Then the audio please hold starts. After about 2-4 seconds the call
gets dropped. (fast busy tone)
The time on my phone will stop running (call time) and I will get
2004 May 19
1
Strange Sip (FWD, SipGate and such) problem
Hi all
I use sipgate and FWD but seem not to get it going. I do not have NAT on
the asterisk box (static ip).
The asterisk box has 2 network interfaces. One internal and one external.
Now when I make an call to a FWD or SipGate number all I get is
-- Executing NoOp("SIP/113-6d2e", "") in new stack
-- Executing Goto("SIP/113-6d2e",