I'm using Asterisk 13.4.0 and DAHDI 2.10.2. I've got a FXO line that I
use for in and outgoing PSTN calls. Unfortunately I'm getting a lot of
spam calls on the number.
I had the extension configured to forward incoming calls to 2 SIP
extensions or go to voicemail. But now I'm getting loads
I'm thinking of condensing some of my boxes down to KVM virtual machines
running under SmartOS. My Asterisk box is running Centos 6.4 and I'd like
to include it.
Is anyone running Asterisk on a virtual machine under SmartOS? Does DAHDI
work?
Thanks in advance.
Frank
--
My AsteriskNow box was updated to Centos 5.6 (2.6.18-238.5.1.el5) and
DAHDI doesn't want to load. I've tried building it from the sources, but
get this error message:
CC [M]
/root/Desktop/dahdi-linux-complete-2.4.1.1+2.4.1/linux/drivers/dahdi/xpp/card_bri.o
In file included from
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
My outgoing FXO calls are answered but have no audio in either direction
if I have callprogress=no in chan_dahdi.conf. If I change to
callprogress=yes then the audio returns. My chan_dahdi.conf file is
listed below. Can anyone point-out why callprogress=no isn't working?
#cat /tmp/a
My outgoing FXO calls are answered but have no audio in either direction
if I have callprogress=no in chan_dahdi.conf. If I change to
callprogress=yes then the audio returns. My chan_dahdi.conf file is
listed below. Can anyone point-out why callprogress=no isn't working?
#cat /tmp/a
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
I updated an AsteriskNow system to 2.6.18-194.26.1.el5 with yum update.
Upon reboot dahdi modules cannot be found. In yum.log I see that
kmod-mISDN, kmod-dahdi-linux and kmod-dahdi-linux-fwload-vpmadt032 were
all deleted during the update.
I reinstalled the deleted packages but the dahdi modules
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
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
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
On 9/7/2010 9:05 PM, asterisk-users-requ...@lists.digium.com wrote:
Subject: [asterisk-users] 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
I have 2 FXO channels from which I want to route incoming calls to
different contexts in extensions.conf. I edited the context entries in
dahdi-channels.conf and created matching entries in extensions.conf.
One channel is routed to the new context as I want, but the other
channel is stuck going
I've moved from trixbox to AsteriskNow. Does anyone know if there's
something like the PBX Status screen for AsteriskNow?
A module the shows the status of SIP and IAX2 registry and peers, etc
for all individual entries?
The FreePBX System Status screen shows when something fails to register
I'm trying to get a MP-114 FXS/FXO gateway working with Asterisk. It
registers fine and I can call between the MP-114 and other extensions,
but I'm not having much luck with the FXO ports. syslog shows the
problem to be in the MP-114 configuration.
Can anyone help?
I'm trying to get Asterisk working with Zaptel support. The Zaptel
driver packages that I can find are too old to work with current
versions of Asterisk. Has anyone ported anything recent?
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I'm running an Asterisk box that's connected to the world via 5MB
down/384kB up cable internet service. I've noticed that the sound
quality for both IAX and SIP calls sometimes starts to suffer. IVR
prompts and MOH frequently have slight pauses from the outside, but
sound fine from inside
My wife really likes the fit and feel of my SPA-942. Anyone know of a
POTS telephone with similar rugged construction?
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I'm looking for pointers towards building and running the zaptel drivers
under Solaris 10.
Can anyone help?
Frank
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I've been trying to get Asterisk 1.4.X running under Solaris 10 x86 with
limited success.
I can build Asterisk and get it started but have run in to a problem
with a segmentation fault with the help command in the CLI.
When I start Asterisk:
# ./asterisk -vvvgc
Asterisk 1.4.9, Copyright (C)
Message: 1
Date: Tue, 15 May 2007 23:01:24 -0400
From: Frank Tarczynski [EMAIL PROTECTED]
Subject: [asterisk-users] Asterisk 1.4.4 reproducibly dumps core on
Solaris 10
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1
I'm having a problem with asterisk-1.4.4 dumping core under Solaris 10
with a SIGSEGV error.
gdb gives this stack trace:
#0 0xfebd4d0c in strlen () from /usr/lib/libc.so.1
#1 0xfec2a386 in _ndoprnt () from /usr/lib/libc.so.1
#2 0xfec2d4bb in vsnprintf () from /usr/lib/libc.so.1
#3
I have built Asterisk 1.4.4 on my Solaris 10 x86 box:
LDFLAGS='-R/usr/sfw/lib -R/opt/csw/lib -L/opt/csw/lib -L/usr/sfw/lib'
CPPFLAGS=-I/opt/csw/include ./configure -with-curl=/opt/csw
--without-oss --without-vpb --prefix=/opt/asterisk-1.4
The build and install go fine but the asterisk
I'm looking to build the zaptel drivers on a Solaris 10 X86 box. I've
found the driver source code on
https://svn.sunlabs.com/svn/solaris-asterisk but this source is posted
along with Asterisk 1.2.7.1 Does anyone know of a fresher version? Is
this code considered somewhat ready for prime
I'm having a problem with my IAXy not always connecting to my Asterisk box.
When I pick-up the phone plugged in to the IAXy I get a busy signal. I
have to hang-up the phone and wait a few seconds after the orange LED goes
out and then try again.
When this happens I don't see any connection
I'm running Asterisk 1.2.13 on a Solaris 10 X86 box behind an IPCop
firewall on a 5Mbps down/512 up cable connection.
I'm having sound quality problems when users call in for voicemail and
with music on hold. The sound is choppy and muffled while souding pretty
good for calls inside the network.
Dovid B wrote:
Do you have the issues locally ? Are you using Ztdummy ?
- Original Message - From: Frank Tarczynski
[EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, October 23, 2006 10:48 PM
Subject: [asterisk-users] Where to best start looking for voicemail/moh
I'm running Asterisk 1.2.12.1 on a Solaris 10 box. I've built mpg123
but it doesn't want to play well under Solaris so I want to replace it
madplay.
I've edited app/app_mp3.c and res/res_musiconhold.c to change the calls
for mpg123 to madplay with the appropriate options.
The madplay
Message: 9
Date: Fri, 29 Sep 2006 08:23:29 -0700
From: Ken Godee [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Replacing mpg123 with madplay under
Solaris?
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL
I'm looking for a VOIP provider in Panama that will support outging DIDs
and SIP or preferably IAX.
Can anyone help?
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I'm having a problem where the very first words of the Asterisk voicemail
system prompt are distorted into a loud ear-splitting beep. When I dial my
VoiceMailMain extension I get this loud beep followed by the rest of the
initial voicemail system prompt. After that everything works fine. I've
I'm having a problem where the very first words of the Asterisk voicemail
system prompt are distorted into a loud ear-splitting beep. When I dial my
VoiceMailMain extension I get this loud beep followed by the rest of the
initial voicemail system prompt. After that everything works fine. I've
I'm looking for a recent asterisk package for the Linksys WRT54G.
Has anyone know of a 1.2.X build for this box?
Thanks,
Frank
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,Congestion
Message: 21 Date: Mon, 7 Nov 2005 14:25:50 -0500 (EST) From: Frank
Tarczynski [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Help
with dialplan to allow breakout to DISA To:
asterisk-users@lists.digium.com Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain;charset=iso-8859-1 Yes
I'm trying to set-up a dialplan for incoming calls that allows a breakout
by pressing something like *. Users would then be able to get an inside
dial tone for voicemail, outgoing calls, etc.
I've been struggling with Waitexten(), Disa() in the dialplan but not
having much luck.
Are there any
:
exten = *, 1, Authenticate(PASSWORD)
exten = *, 2, DISA(no-password|DESTINATION_CONTEXT)
exten = *, 3, Hangup
It seems to work fine...
-Rusty
On 11/7/05, Frank Tarczynski [EMAIL PROTECTED] wrote:
I'm trying to set-up a dialplan for incoming calls that allows a
breakout
by pressing
I'm using an IAXy witha current CVS-head build of Asterisk.
The IAXy has an extensions.conf entry somethng like this:
exten = 1,1,Ringing
exten = 1,2,Answer
exten = 1,3,Voicemail(u1)
exten = 1,4 Hangup
This works fine for calls routed to extension 1. But if a second call is
routed to the IAXy
I am trying to use a SIP provider for outgoing and incoming calls under
Asterisk. I am running a recent CVS-head 1.09 build and the SIP
provider is using a SPA-3000. I can register with the SIP provider's
server and outgoing calls seem to work just fine.
But I cannot get incoming calls to
I am trying to use a SIP provider for outgoing and incoming calls under
Asterisk. I am running a recent CVS-head 1.09 build and the SIP
provider is using a SPA-3000. I can register with the SIP provider's
server and outgoing calls seem to work just fine.
But I cannot get incoming calls to work
Does anyone know of a IAX termination/DID provider in Panama? (507
country code).
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I need some help generating configuration files for Asterisk. Since I'm
running under Solaris I'm having trouble with some of the utilities that
are more linux-centric.
Can anyone recommend a free/low-cost package to generate conf files that
is not linux-dependent and will handle a IAX2 and
I'm new to asterisk and need some help with getting a SIP connection
working.
I am trying to establish a termination point/DID number in another
country. I am currently running Asterisk CVS-HEAD. My foreign provider
uses SIP and authenticates via IP address. I am not required to
register my
I'm new to asterisk and need some help with getting a SIP connection
working.
I am trying to establish a termination point/DID number in another
country. I am currently running Asterisk CVS-HEAD. My foreign provider
uses SIP and authenticates via IP address. I am not required to
register my
I'm new to asterisk and need some help with ideas to handle this
configuration question.
I am trying to establish a termination point/DID number in another
country. I am currently running Asterisk CVS-HEAD. My foreign provider
uses SIP and authenticates via IP address. I am not required to
I'm new to asterisk and need some help with ideas to handle this
configuration question.
I am trying to establish a termination point/DID number in another
country. I am currently running Asterisk CVS-HEAD. My foreign provider
uses SIP and authenticates via IP address. I am not required to
Message: 26
Date: Mon, 29 Aug 2005 15:26:31 +0800
From: chris [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] error compiling on solaris 10
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain;
Message: 11
Date: Sun, 28 Aug 2005 11:46:29 +0800
From: chris [EMAIL PROTECTED]
Subject: [Asterisk-Users] error compiling on solaris 10
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain;
I'm trying to set-up H.323 support under Asterisk. I built a recent CVS
release and the ooh323c code from the asterisk-addons. Everything built
and installed and the H.323 stuff loads OK when asterisk starts.
What is the easiest way to check if the H.323 code is working? I've
edited the
I'm running a recent CVS build under Solaris 10.
In the shell than I'm running the Asterisk console I have TZ=US/Eastern
and in my voicemail.conf I have tz=eastern and
eastern=America/New_York|'vm-received' Q 'digits/at' IMp.
The voicemail envelope information seems to be exactly 4 hours
I've built Asterisk from recent CVS sources on a Solaris 10 X86 box. I
tweaked the makefile to get the build to run using gcc. And most
recently ran into va_args problems with new code in asterisk/utils.c.
It seems to run OK and register with my VoIP provider, but I'm still
having trouble
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