Displaying 20 results from an estimated 500 matches similar to: "T38 Passthrough testing -- unknown media type error"
2006 Mar 15
0
T.38 Passthrough testing -- IAX problem
Trying out SVN-oej-t38passthrough-r12677 on a server that also needs to
pass some calls to another using IAX and attempts to use the Dial
command results in multiple messages "Out of idle IAX2 threads for I/O,
pausing!".
Since this server needs to support IAX I'll have to back out this
version and find another idle server to use to play with the T.38 code.
g.
--
George
2006 Mar 17
0
One-Way SIP Audio with SVN Codebase
Please tell me the obvious mistake I'm making here. (And yes, I well
know about NAT and one-way audio problems in general.)
I want to try the new T.38 passthrough stuff, downloaded it, built it,
tested it with an SPA-2100 and can hear announcements fine but echo test
shows no audio outbound (i.e. SPA to Asterisk).
Registered the second SPA-2100 channel with an Asterisk 1.0.10 server in
2009 Apr 08
1
Call Pickup Works w/Linksys ATA, not with Cisco 7940G
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type">
</head>
<body bgcolor="#ffffff" text="#000000">
I have an Asterisk 1.4.18 with a mix of cordless phones connected using
Linksys SPA2102 ATAs and Cisco 7940G
2006 Apr 23
0
New backport of T.38 fax passthrough functionality to asterisk-1.2.7.1
(This is a shameless copy-paste from the note I posted on
http://bugs.digium.com/view.php?id=5090)
I have again backported the whole T.38 shebang to the stable branch. The port was
based on two versions of the t38passthrough branch: r19125, the latest
unconflicted automerge, and r13623, the latest version without the new chan_sip
flag structure. Basically, the port contains everything that
2007 Jun 08
0
Replacing SX-2000 Centigram Voicemail with Asterisk?
We have a customer with an obsolete Centigram voicemail system who would
like to replace it with Asterisk.
Any one with experience doing this or information on the signalling and
trunking used to connect the Mitel SX-2000 to the Centigram server?
--
George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102
www.netvoice.ca www.ip-centrex.ca
www.digium.ca
2004 Dec 06
0
CVS HEAD h323 no longer builds?
Attempts to perform a "make all" in /usr/src/asterisk/channels/h323
fails with countless errors of the form:
/usr/src/pwlib/include/ptlib/ptime.h:152: macro or `#include' recursion
too deep
In file included
A "make all" using the stable branch builds with the same pwlib code but
of course the h323 code in the stable branch doesn't work.
So it seems those of us who
2005 Aug 24
0
Distorted Sound from E1
We're having a problem with an E1 trunk in Mexico into an IVR server and
would appreciate any suggestions.
Hardware: Digium TE110P jumpered for E1
zaptel.conf:
span=1,1,0,ccs,hdb3
# clear=1-30
bchan=1-15
bchan=17-31
dchan=16
loadzone = us
defaultzone=us
Circuit status is fine: Status: Provisioned, Up, Active
Calls are accepted by Asterisk without any
2006 Jan 19
0
AudioCodes Unreliable DTMF Detection
We're trying to use some AudioCodes MP104 FXO units as gateways to
Asterisk but cannot get them to reliably detect DTMF. Some landline
calls get most digits but some are duplicated. Some cell phone calls get
0% DTMF recognition.
Anyone with experience with these units have any suggestions? ABP
Technical Support has been unable to diagnose the problem and is now
sending random guesses and
2006 Mar 17
1
One-Way SIP Audio with SVN Codebase (CANCEL)
I wrote earlier:
> Please tell me the obvious mistake I'm making here....
The problem was a lack of sleep. Sorry to have troubled the list.
--
George Pajari, netVOICE communications 604 484 VOIP (484 8647 x102)
Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102)
www.netvoice.ca www.ip-centrex.ca
www.digium.ca www.grandstream.ca
2007 Jul 12
0
No subject
"Annoying that people aren't following the directions and only entering 3
digits, but we've had some high level meetings here with a string of clients
coming through in an unusually compressed frequency. And I've had 5
complaints over 2 days that callers couldn't find Jane Smith."
-
George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102
2008 Feb 18
0
Vancouver - Asterisk Event Feb 18 (Monday)
The Vancouver Linux User Group is holding a "Virtualization Round Table"
Monday (Feb 18) evening at the BC Institute of Technology discussing
some of the different approaches to server virtualization. I'll be
speaking about using OpenVZ to provide virtual servers used to host
multiple instances of Asterisk (the technology behind our Virtual
Private Asterisk Server or VPAS
2005 May 09
2
AGI - How to Make Calls and Bridge to Original Incoming
I need to accept an incoming call, make a series of outgoing calls, and
once I find someone willing to accept the call, bridge the original
incoming call to the outgoing call.
Using Dial from an AGI script isn't enough because once the Dial'ed
number connects, the call is immediately bridged and I need to ask the
called party if they will accept the call.
I can see a couple of
2009 Jan 15
1
Patton SmartNode 4638 and ISDN2e
Hello
Does anyone have any experience with configuring BT (British Telecom)
ISDN2e lines to work with Patton SmartNodes?
I have a Patton SmartNode 4638, which is now connected to 3 x ISDN2e
lines - and in turn connected to our internal LAN. I'm having huge
issues configuring the SmartNode to successfully "see" the ISDN channels
- and to be honest, I'm lost as to how to then
2007 Dec 31
1
PRI Crapping Out Regularly
We have a server with a TE120 on a partial PRI trunk that several times
a day declares the PRI trunk down and stops handling calls until the
asterisk is stopped, the zaptel/te120 modules reloaded, and asterisk
started.
Just before things go down, the log shows the following error:
ERROR[9424] chan_zap.c: Write to 28 failed: Unknown error 500
at which point a "show pri spans"
2007 May 16
5
Microsoft's Move Into IP PBX Market
From c|net News:
"On Monday,Microsoft and nine leading phone manufacturers--Asustek
Computer, GN, LG-Nortel, NEC, Plantronics, Plycom, Samsung, Tatung, and
Vitelix--announced the public beta program for Microsoft Office
Communications Server 2007 and Microsoft Office Communicator 2007."
http://news.com.com/8301-10784_3-9719931-7.html?part=rss&subj=news&tag=2547-1_3-0-20
--
2010 Apr 30
0
Problems with t38modem and bitrate sent to t38-termination service
Hi all the people in the list!
I'm new on this list, this is my first post.
I configured asterisk 1.6 with freepbx 2.7 and dahdi to send faxes with
t38modem conected to hylafax as a sip extension of asterisk.
Everything is supposed to be configured fine, the faxes start sending, but
at the middle of the transaction, it fails. The T.38 termination provider
told me that they were receiving
2008 Jan 29
1
PRI Alarms, Comes Back, But Asterisk Won't Touch It!
Here is the scenario: Asterisk 1.14.13; zaptel 1.4.6; Digium TE120P
(same problem with various previous versions; same problem with
different TE120P cards).
The customer has a partial (10 B-Channel) PRI that when it is busy
(eight or more B channels in use), tends to fail as shown below...
[Jan 26 23:00:31] ERROR[31893] chan_zap.c: Write to 28 failed: Unknown
error 500
[Jan 26 23:00:31]
2010 May 09
1
B410P and Patton smartnode : any success ?
Hi,
1. Has someone met any success at all, connecting a Digium B410P to a Patton
Smartnode 4638 (with latest 5.3 firmware) ?
2. If positive, then, which signalling was used on both sides ?
My project's goal is use a Patton Smartnode 4638 to act as telco BRI lines,
from a B410P-enabled asterisk box.
In my testings, I can see channel is respectively up (with patton web server
status page)
2009 Feb 02
2
Configuring Patton SmartNode with ISDN2e and Asterisk
Hello
Does anyone have any experience with configuring BT (British Telecom)
ISDN2e lines to work with Patton SmartNodes - and then Asterisk?
I have a Patton SmartNode 4638, which is now connected to 3 x ISDN2e
lines - and in turn connected to our internal LAN. I'm having huge
issues configuring the SmartNode to successfully "see" the ISDN channels
- and to be honest, I'm
2005 Oct 12
2
Patton SmartNode
Does anybody have any experience using a Patton SmartNode as a SIP/Telco
gateway with Asterisk? They seem really inexpensive and appear to
support all of the necessary features, but I don't have any experience
with their products, so I don't know if they are any good. We are
currently using a Cisco 2600 w/ PRI card and it works fine, but I was
looking for someone else as a possible