search for: port11

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",