On Wed, 2012-12-19 at 11:16 -0600, Richard Mudgett wrote:
On Fri, 2012-12-14 at 15:16 +, Ishfaq Malik wrote:
Hi
Can someone else please check the following:
We have installed asterisk 1.8.18.0 onto our development and test
servers. They were previously on 1.8.7.0
When
On Thu, 2012-12-20 at 09:23 +, Ishfaq Malik wrote:
On Wed, 2012-12-19 at 11:16 -0600, Richard Mudgett wrote:
On Fri, 2012-12-14 at 15:16 +, Ishfaq Malik wrote:
Hi
Can someone else please check the following:
We have installed asterisk 1.8.18.0 onto our development and
On 19 December 2012 21:54, Christopher Harrington ch...@acsdi.com wrote:
You probably already know this, but 1.4x is very old (released in 2006)
and is officially end-of-life.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
You might get more help or better behavior by updating
Hii,
we implemented the same senario with modification of meet-me application
and using php and mysql.
When user want to talk, he press some predefined digit on the phone. Admin
can view who raise hands (press digit) in php page and press predeifned
digit and that user get unmuted. As soon as
Hi,
What are the recommended T.38 settings for sending/receiving faxes
from Cisco AS5350XM gateways? The chan_sip.conf file has a remark
about what Cisco is doing wrong and says that the values received from
the gateway should be overridden, but doesn't say what settings to use
for maximum
I have a CentOS 6.3 machine I installed Asterisk 11, worked fine...
I then tried to install on Cents 5.8, seemed to go fine... Then when I
placed a call I got this:
ast_rtp_instance_new: No RTP engine was found. Do you have one loaded?
Did a search and found issues with ARM and this problem
On Thu, Dec 20, 2012 at 2:22 AM, Steve Davies davies...@gmail.com wrote:
On 19 December 2012 21:54, Christopher Harrington ch...@acsdi.com wrote:
You probably already know this, but 1.4x is very old (released in 2006)
and is officially end-of-life.
Jerry Geis wrote:
I have a CentOS 6.3 machine I installed Asterisk 11, worked fine...
I then tried to install on Cents 5.8, seemed to go fine... Then when
I
placed a call I got this:
ast_rtp_instance_new: No RTP engine was found. Do you have one
loaded?
Did a search and found issues
This was a change in v1.8 and is documented in the v1.8 UPGRADE.txt file:
* The PRI channels in chan_dahdi can no longer change the channel name if a
different B channel is selected during call negotiation. To prevent using
the channel name to infer what B channel a call is using and to
On 12/18/2012 10:56 AM, Tim Nelson wrote:
I'm getting this error message on my Asterisk CLI, and in the logs, roughly
every 10-20 seconds:
[2012-12-18 19:19:51] ERROR[24485]: astobj2.c:115 INTERNAL_OBJ: user_data is
NULL
While it doesn't appear to be actually affecting anything, I'm
This was a change in v1.8 and is documented in the v1.8 UPGRADE.txt
file:
* The PRI channels in chan_dahdi can no longer change the channel
name if a
different B channel is selected during call negotiation. To
prevent using
the channel name to infer what B channel a call is using
It is
Action: ExtensionState
Exten: 5551212
Context: fubar
This will return the status of the dialplan exten hint.
/ and
// Action: Command
// Command: ChanIsAvail
// Parameters: DAHDI/1
//
// says Error
// No such command ChanIsAvail
/
ChanIsAvail is a dialplan application not a CLI
The Asterisk 11 part is irrelevant. You need to use an AGI or local
call to use the ChanIsAvail function.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Thursday, December 20, 2012 11:03 AM
To: Asterisk Users
It is
Action: ExtensionState
Exten: 5551212
Context: fubar
This will return the status of the dialplan exten hint.
and Action: Command Command: ChanIsAvail Parameters: DAHDI/1
says Error No such command ChanIsAvail ChanIsAvail is a
dialplan application not a CLI command. It
You should just cache the AMI DAHDIChannel event information in your
program.
If you really must you could use the CLI command pri show channels.
However, it is not intended to be repeatedly run for performance
reasons. It blocks processing of ISDN messages while it is running.
I am not
IMO the local channel call should be the lowest overhead option available.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Thursday, December 20, 2012 11:42 AM
To: Asterisk Users Mailing List - Non-Commercial
You should just cache the AMI DAHDIChannel event information in your
program.
If you really must you could use the CLI command pri show channels.
However, it is not intended to be repeatedly run for performance
reasons. It blocks processing of ISDN messages while it is running.
I am not
IMO the local channel call should be the lowest overhead option available.
What about:
Action: Command
Command: dahdi show channels
I can just look to see if Extension has anything for the Chan I am
interested in?
is
On 12/20/2012 01:00 PM, Jerry Geis wrote:
IMO the local channel call should be the lowest overhead option available.
What about:
Action: Command
Command: dahdi show channels
I can just look to see if Extension has
I have a little dialplan context now...
[check-chanisavail]
exten = s,1,ChanIsAvail(${agi_channel})
exten = s,n,System(/bin/echo ${AVAILCHAN} /tmp/${agi_file})
exten = s,n,Hangup()
and a call file:
Channel: Local/s@check-chanisavail/n
Context: check-chanisavail
Extension: s
Priority: 1
Just for grins, do you have a softphone like xlite that you can try the
outgoing call on? I think it's an outgoing issue, not a polycom one.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent:
Just for grins, do you have a softphone like xlite that you can try the
outgoing call on? I think it's an outgoing issue, not a polycom one.
I do not have a softphone. I have a yealink VP-2009 and same behavior.
Jerry
--
_
Cause 20 means your SIP device is not registered or you do not have an IP
specified for it in your peer.
sip show peers will show that.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Scott Huang
Sent:
The Asterisk Development Team has announced the release of libpri 1.4.14.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/libpri
The release of libpri 1.4.14 resolves several issues reported by the
community and would have not been possible without
Error loading module 'res_rtp_asterisk.so': /usr/lib64/libavformat.so.52:
undefined symbol: av_tree_node_size
This is the error I get when trying to start Asterisk 11 on centos 5.
Asterisk 11 works fine on my centos 6 box - I also verified that on centos 6
I do not have the above mentioend
When doing a 'dahdi show channel X' from the asterisk console, when the line is
not part of a call the echo cancellation line ALWAYS says 'currently OFF'.
Once a call is in progress, it will change to 'ON'. Is this a bug, or is the
behavior by design?
My setup:
Asterisk 10.10.0
Dahdi 2.6.1
On Thu, Dec 20, 2012 at 03:13:01PM -0800, Justin Killen wrote:
When doing a 'dahdi show channel X' from the asterisk console, when the line
is not part of a call the echo cancellation line ALWAYS says 'currently OFF'.
Once a call is in progress, it will change to 'ON'. Is this a bug, or is the
Jerry Geis wrote:
Error loading module 'res_rtp_asterisk.so':
/usr/lib64/libavformat.so.52: undefined symbol:
av_tree_node_size
This is the error I get when trying to start Asterisk 11 on centos 5.
Asterisk 11 works fine on my centos 6 box - I also verified that on
centos 6
I do
This is highly confusing. It would be nice if at least the display gave the
configured value as well.
-Justin Killen
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Russ Meyerriecks
Sent: Thursday, December
I have a box with 12 T1s (4 Te410P cards). The PSTN provider is reporting
slips and ask me to update the clock source. I have my system.conf set as
the following but when I run dahdi_scan only the ports on Card 1 are showing
up with syncsrc=1
system.conf :
span=1,1,0,esf,b8zs
bchan=2-24
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