I've just bought an OpenVOX B200p ISDN card - and if I remember
correctly from last time I used one of these - it is possible to use
either DAHDI or mISDN with it. Are there any factors to consider when
choosing which software to use? Is one better than the other - or does
one have features
Hello,
Thanks for the help list. I reinstalled the OS and that has fixed the
problem for me. I appreciate your contribution.
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Hello,
I have a dialplan using AGI where a user calls a number and an IVR is
played. When the user presses 1, the system is suppose to call another
number and bridge the call. I am able to do this successfully, but I want
to know the billsecs of the first caller and also the second call from the
On 10/16/2012 08:50 AM, Sebastian Arcus wrote:
I've just bought an OpenVOX B200p ISDN card - and if I remember
correctly from last time I used one of these - it is possible to use
either DAHDI or mISDN with it. Are there any factors to consider when
choosing which software to use? Is one better
I have worked with the B200P before and used the standard mISDN and the
standard DAHDI and both worked fine.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Lists
Sent: 16 October 2012 12:30 PM
To:
On 16/10/12 11:30, Patrick Lists wrote:
On 10/16/2012 08:50 AM, Sebastian Arcus wrote:
I've just bought an OpenVOX B200p ISDN card - and if I remember
correctly from last time I used one of these - it is possible to use
either DAHDI or mISDN with it. Are there any factors to consider when
Thanks Andrew
On 16/10/12 12:08, Andrew Colin wrote:
I have worked with the B200P before and used the standard mISDN and the
standard DAHDI and both worked fine.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
I am having a problem trying to get a particular softphone working on my
setup.
The machine it runs on has more than one interface. When the softphone
registers, it registers fine, and asterisk is given the correct IP for
registration.
Whenever RTP is set-up however, the client gives the
Thomas Kenyon wrote:
I am having a problem trying to get a particular softphone working on my
setup.
The machine it runs on has more than one interface. When the softphone
registers, it registers fine, and asterisk is given the correct IP for
registration.
Whenever RTP is set-up however, the
Joshua Colp wrote:
Thomas Kenyon wrote:
I am having a problem trying to get a particular softphone working on my
setup.
The machine it runs on has more than one interface. When the softphone
registers, it registers fine, and asterisk is given the correct IP for
registration.
Whenever RTP is
I am using asterisk 1.4.43.
When I call over the PRI to a single phone and play my recorded message
its heard just fine.
When I call over the PRI to a single extension (the switch then takes 3
phones offhook in intercom mode)
and play my same recorded message the audio is dropping out and the
Hey,
It's late out here and I'm trying to setup a personal Asterisk server
and have it register to a SIP Provider.
Whilst the account works perfectly find from a softphone I keep
getting a This account is not valid IVR.
The provider appears to be running PortaSIP. Anyone with suggestions?
--
Am 16.10.2012 18:38, schrieb Sahil Gupta:
The provider appears to be running PortaSIP. Anyone with suggestions?
What does your register = line look like in sip.conf? (without the
password)
What does sip show registry show?
What do you see on the Asterisk console (asterisk -vr) when you
At the end of the output for core show channels verbose is a line that
reads 4 active calls. Does anyone know how that number is formatted
if there are more than 999 active calls? Will it have a comma or not?
--
Mitch
--
We recently set up a SIP trunk between an office in NY running Asterisk and
an office in Paris (running Alcatel). All works fine if a SIP phone on the
NY system talks to the Paris PBX. But if something on DAHDI (a PRI or
MeetMe) talks to the Paris PBX, there's a low-volume crackling. This isn't
On Tue, Oct 16, 2012 at 1:31 PM, Richard Kenner ken...@gnat.com wrote:
We recently set up a SIP trunk between an office in NY running Asterisk and
an office in Paris (running Alcatel). All works fine if a SIP phone on the
NY system talks to the Paris PBX. But if something on DAHDI (a PRI or
Mitch Claborn wrote:
At the end of the output for core show channels verbose is a line that
reads 4 active calls. Does anyone know how that number is formatted if
there are more than 999 active calls? Will it have a comma or not?
It will not have a comma.
Cheers,
--
Joshua Colp
Digium, Inc.
cat proc/interrupts?
http://wiki.openvox.cn/index.php/Troubleshooting_of_PRI_cards
I'm sorry that I wasn't clear: the PRI is fine. It's been in use for
years and hasn't caused any problems. What's new is the SIP
connection between the two offices. And another datapoint: the problem
only
I seem to recall seeing somewhere recently where there was a bugfix for
ulaw/alaw conversion which would cause poor audio.
Have you tried updating your Asterisk to the latest of whatever major version
you are running?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
I seem to recall seeing somewhere recently where there was a bugfix
for ulaw/alaw conversion which would cause poor audio.
Hmm. You mean:
https://issues.asterisk.org/jira/browse/ASTERISK-1323
That was quite old, but that is what the noise sounds like.
Have you tried updating your Asterisk
On Tue, Oct 16, 2012 at 4:23 PM, Richard Kenner ken...@gnat.com wrote:
I seem to recall seeing somewhere recently where there was a bugfix
for ulaw/alaw conversion which would cause poor audio.
Hmm. You mean:
https://issues.asterisk.org/jira/browse/ASTERISK-1323
That was quite old, but
How do I use softhangup through the AMI interface?
I am using 1.4.43. Will softhangup hangup a DAHDI channel?
I have found that Action: Hangup does not hangup a DAHDI channel only sip.
Thanks,
jerry
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On Tue, Oct 16, 2012 at 12:40 PM, Ashish Agarwal ashisha...@gmail.comwrote:
Hello,
I have a dialplan using AGI where a user calls a number and an IVR is
played. When the user presses 1, the system is suppose to call
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