Displaying 20 results from an estimated 8000 matches similar to: "5-7 second delay in connecting outgoing FXO calls"
2011 Jan 14
1
5-7 second delay in connecting outgoing FXO calls
I'm running AsteriskNow 1.7.1 with a OpenVox 2FXO/2FXS card,
SPA942 SIP phone and outgoing SIP and IAX routes.
When I dial local PSTN numbers from the SPA942 using the FXO channels I
observe a 5-7 second delay between when the PSTN number answers the call
and when Asterisk connects the call at my end. There's enough delay
time that I hear an additional ring after the PSTN number has
2010 Sep 07
2
5-7 second connection delay in outgoing FXO calls
I'm running AsteriskNow 1.7.1 with a OpenVox 2/FXO/2FXS card, a Linksys
SPA942 SIP phone and outgoing SIP and IAX routes.
When I dial local PSTN numbers from the SPA942 using the FXO channels I
observe a 5-7 second delay between when the PSTN number answers the call
and when Asterisk connects the call at my end. There's enough delay
time that I hear an additional ring after the PSTN
2009 Oct 28
1
Asterisk 302 Moved Temporarily
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
</head>
<body bgcolor="#ffffff" text="#000000">
Hello,<br>
<br>
I have an * installation that sometimes receives a 302 "Moved
Temporarily" response to an INVITE. * sends the ACK but takes about 30
seconds to start the new INVITE
2006 Aug 01
1
Subscribe
2010 Jun 07
0
No subject
When I load the codec as an Audio Codec the call to AudioCodecInitialize appears to work but if I then get the IsInitialized property that says it is not initialized - so looks like that is my problem, and ties up with the crash in vorbis_synthesis_start(). Looks like vorbis_synthesis_init() has not been called, perhaps.
There's sample code out there that suggests people have got this
2017 Nov 06
3
ORC JIT and multithreading
2013 Aug 30
0
[LLVMdev] Some reflexions about a new HDL language
2015 Nov 22
0
可能您的账户已经被盗用。被不法分子利用发送不良信息！
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content="text/html; charset=utf-8" http-equiv=Content-Type>
<META name=GENERATOR content="MSHTML 8.00.7601.18667"></HEAD>
<BODY>
<TABLE
style="WIDOWS: 1; TEXT-TRANSFORM: none; BACKGROUND-COLOR: rgb(255,255,255); TEXT-INDENT: 0px;
2011 Mar 18
2
Problem routing call to fax machine on DAHDI FXS port
I am running Asterisk 1.6.2.17.2 with a Openvox A400 card with 2FXO/2FXS
modules. I'm trying to set-up things to route analog fax calls from a
FXO port to an analog fax machine on a FXS port on the same card.
Outgoing faxes work just fine. But incoming faces are routed to the
right DAHDI extension, but the call dropped right as the fax machine
rings for the first time. The fax machine
2010 Oct 16
3
Detect incoming fax on PSTN and route to fax machine on DADHI extension?
I'm running an AsteriskNow V1.7.1 with both a PSTN connection and fax
machine. Both are connected to a DAHDI board. I'd like to route
incoming PSTN fax calls to the extension of the fax machine and process
non-fax calls through different dialplan.logic.
What's the best way to go about doing this? I've looked into Fax for
Asterisk, bit I'm not sure that I want it or NVFax
2009 Jun 06
2
Sangoma A200 wt HW echo on 64bit Centos , 4 fxo and crosstalk!
Hi,
Has anyone ever experienced crosstalk with sangoma analogue cards on 64bit?
We have exhausted every test to try and replicate this and find a
solution with Sangoma tech support, but we can not fix it.
We are about to try the card and four *seperate* UK BT lines in a 32bit system.
The current system is a 4gb, dual core cpu with pbx in a flash 1.4,
Zaptel and Asterisk 1.4.21-2
Currently we
2009 May 06
0
astcc - outgoing call does not hangup properly
Hi,
I am using ASTCC and trying to setup a calling card platform.
The problem that I have is that astcc does not hangup calls correctly:
1. If I try to dial a number, call goes through fine. When I hang up
the call from my side I get this:
-- Called 192.168.1.56/1XX6872XXXX (masked a few digits)
-- SIP/192.168.1.56-086c5000 is making progress passing it to
2010 Nov 13
1
CallerID from Samsung PBX line on FXO
Hi,
I've now set up Asterisk to interface with our current Samsung iDCS 100 PBX via an 8SLI analogue extension card in the Samsung and an Openvox A400P04 4-FXO card in the Asterisk box. It all works in that I can place calls in both directions from the office Samsung extensions and Asterisk SIP extensions. The only tricky bit was getting the FXO to detect hang-up from the Samsung correctly -
2007 Jan 11
2
calls to SPA942 disconnect after 15 seconds (chan_sip.c set_destination: can't find address)
Am having a unique problem, calls received on my SPA942 seem to end after 15 seconds, but calls made from this device do not have this problem.
For this device (when receiving calls) I get periodic "chan_sip.c set_destination: can't find address for host"
I have set the "canreinvite=no" in the sip.conf. Does anyone have a sample entry from sip.conf for the Lynksys SPA 942
2007 Feb 14
1
[Fxo] Digium TDM01B vs. OpenVox A400P01?
Hello
If someone had the opportunity of trying those two analog cards, how do
they compare? Digium's sells for $150 while OpenVox's sells for $95.
Thanks.
2010 Nov 24
2
SPA942 on speaker phone does not hang up?
Hello all,
I am using Linksys SPA942 in my current installation activity. I see a
peculiar behavior: A call is made and the SPA942 uses its speaker. When the
far end of a call hangs up , the SPA942 stays off hook, and after a time
plays a fast busy. The user then has to press the line presence button to
hang up the phone.
I think I must be missing some sip.conf parameter. My sip.conf is pretty
2010 Feb 20
0
outgoing callerid problem
Hi,
I have a B410P card with bri_cpe signalling and two Openvox analog card
(A1200p, A800P) with fxo_ks signalling. From the ISDN we have
Point-Point 10 connection with a 10 public phone number range. If I
receive a public call, the asterisk recevies the last two digit from
this range, so it works, I can receive all the 10 numbers. If I'd like
to dial from an exten which I have to
2009 Dec 30
4
Error running Mongrel on Ruby 1.9
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
</head>
<body bgcolor="#ffffff" text="#000000">
<div class="moz-text-html" lang="x-western">
<span class="Apple-style-span"
2008 Mar 05
0
SIP REFER Message, over NAT
Hi people,
I have a few SPA-942 around, all of them work fine except one. The one
behind NAT..
In every phone you can:
* Pickup a Call on one of the line buttons,
* Create a new call on another button
* Press "xferLx" to join those to calls.
This works everywhere except on the one behind NAT. After a lot of
messing around with all the options possible I gave up and subscribed
2011 Mar 21
0
Problem routing call to fax machine on DAHDI FXSport
[18884732963 at from-fax-machine:... - your call is hitting the
from-fax-machine context - yet your 'fax' exten is in the from-pstn-4
context. See the "[2011-03-17 13:40:29.6] NOTICE[8825] chan_dahdi.c:
Fax detected, but no fax extension" line.
When Asterisk detects an incoming fax tone - it tries to automagically
route the call to the 'fax' extension in the SAME