Thanks Sam. Please see below CLI log:
/[root@localhost ~]# asterisk -r
Asterisk 1.6.2.7, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer marks...@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty'
for detail
s.
This is free
Hey,
So far the Dialplan execution is ok, despite the conflicts and some other
mistakes like repeating priorities in it but they're not involved in this
call.
You'r A-leg is H323 endpoint and Destination is on SIP. I'm now thinking
about codec mismatch on first try Tell me this happens every
Hi All,
I am trying to connect my asterisk box with freepbx to PSTN. I
have purchased x100p FXO card and installed in my asterisk server. My
freepbx detected the x100p FXO card and i can see the card specific details
in command line. I have configured the following things.
1. OUTBOUND
Some CLI logs will get you better help on the issue ! also paste the FXO
configurations and how you configured it !
On Wed, Sep 28, 2011 at 2:11 PM, michael k mich...@inapp.com wrote:
Hi All,
I am trying to connect my asterisk box with freepbx to PSTN. I
have purchased x100p FXO
Hello David,
I have this discussion also on the -dev mailing list. and suggested
that we use a database hook to trigger the originate process (pleasee
see Outbound Call Implementation). However, compiling it directly
into asterisk as a realtime moodule insted of using AMI etc...
Cheers,
Nick.
this is related to your carrier's SIP messages as they are sending a sendonly
attribute instead of sendrecv (taking a wild guess here) your asterisk will act
as if the call was placed on hold thus the MOH butts in.
an sip debug log for a similar call will be more helpful?
Tarek Sawah
On Thursday, September 29th, 2011, the Asterisk community services
listed below will be undergoing maintenance (software upgrades and
updates). The services will be shut down at approximately 7:00 PM CDT
(12:00 AM September 30 UTC), and will return no later than 8:00 PM CDT.
We apologize in
I do not know when the recording actually starts but if it start when the agent
answers the call then it might be possible to have the name set in an AGI that
gets run when the agent answers call. If nothing else you can set a variable to
the name you want to have the file have and rename it at
Hello list,
is the field vmexten available when using SIP peers in a realtime
Mysql-DB ?
Thanks.
Kind regards,
Jonas.
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Make sure that you set *async *option to true. If not asterisk will wait
for response on previous calls before making any other calls.
Hope that will help.
On Wed, Sep 28, 2011 at 12:17 AM, Sam Govind govoi...@gmail.com wrote:
If you can post any relevant code sections and CLI output for
this is related to your carrier's SIP messages as they are sending a
sendonly attribute instead of sendrecv (taking a wild guess here) your
asterisk will act as if the call was placed on hold thus the MOH butts in.
an sip debug log for a similar call will be more helpful?
Thanks for the
i have faced this problem with one of the major VoIP whole providers in India
.. they have a new platform with Sonus switches.. which does not support
sendrecv media attribute .. however a work around that may work for you .. is
enabling re-invite on their peer.
let me know if this works out
hello list
i have configured a sip account in order to do an outbound calls and i want
to force a hang up after 1 min for 222 sip
in extensions.conf i have
exten = 222,1,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = 222,n,AbsoluteTimeout(60)
exten =
We have successfully setup and tested integration between Asterisk and MS-SQL.
We are currently running about 70 simultaneous calls throughout the day however
after some time our MS-SQL server (Windows 2008 64bit, SQL Server 2008) starts
to increase it's memory usage exponentially. The MS-SQL
On 11-09-28 01:59 PM, salaheddine elharit wrote:
hello list
i have configured a sip account in order to do an outbound calls and i want
to force a hang up after 1 min for 222 sip
in extensions.conf i have
exten = 222,1,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten =
have a look at the following:
L(x[:y][:z]): Limit the call to 'x'
ms, warning when 'y' ms are left, repeated every 'z' ms) Only 'x' is
required, 'y' and 'z' are optional.
source
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
Tarek Sawah
Information Technology Adviser
Integrated
i have this when
L(x[:y][:z]) - Limit the call to 'x' ms. Play a warning when 'y' ms are
left. Repeat the warning every 'z' ms. The following special
variables can be used with this option:
* LIMIT_PLAYAUDIO_CALLER yes|no (default yes)
but there is no exemple for when i must put X in order to limit the call
can you please give me an exemple
regards
2011/9/28 Tarek Sawah tareksa...@hotmail.com
have a look at the following:
*L(*x[:y][:z]*)*: Limit the call to 'x' ms, warning when 'y' ms are left,
repeated every 'z' ms)
As I read this, the following should be correct:
exten = 222,n,Dial(SIP/${EXTEN},,KkTtL(6))
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine
elharit
Sent: Wednesday, September 28, 2011 1:23 PM
To: Asterisk Users
exten = 222,n,Dial(SIP/${EXTEN},,KkTtLL(6:3:1))
this will call the extension and sets the limit to 6MS which equals 60
seconds.. and will inform the caller of his remaining time when he has only 30
seconds left.. and will repeat the notification every ten seconds (this is an
sorry but the issue still the same there is no hangup after 1Min
regards
2011/9/28 Danny Nicholas da...@debsinc.com
As I read this, the following should be correct:
exten = 222,n,Dial(SIP/${EXTEN},,KkTtL(6))
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
one adjustment i would suggest is using (|) instead of (,)
exten = 222,n,Dial(SIP/${EXTEN}||KkTtL(6))
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
Date: Wed, 28 Sep 2011 18:32:28 +
From:
I am toying with res_ODBC. currently I am using dns=ODBCvalue. Is there a
way to fail this over to another dns value in the event the a primary is
off line.
Thanks
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
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Does anyone have any experience with BinFone for IAX termination?
They good look on the website, but I'm looking for any comments.
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Hi All;
In the zaptel, we were increasing the gain of the voice volume at the hardware
level from the /etc/zaptel and /etc/modprob.conf files, but now we are using
DAHDI, so where to do the same thing?
I am looking actually to increase the volume at hardware level and not software
to avoid
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I use Voip.ms and have a friend who uses BinFone. We both use IAX
He has had some issues lately, but it is unclear if it is binfone or his ISP.
Losing internet connection and BinFone seems to fail to reconnect when his
connection returns.
I have had no complaints with voip.ms
they have an
OK, i am hoping that someone will be able to help me out.
I am using FreePBX 2.8.1.4
I have two asterisk servers connected with a iax trunk.
The trunk is working fine when used via the outbound route setting.
meaning an extension on one server can call a specific extension on
the other server.
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