...@lists.digium.com] On Behalf Of michael k
Sent: Thursday, October 06, 2011 11:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PSTN connectivity
Hi All,
I got a busy message like all lines are currently busy and please
try again later in call to ZAP trunk
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PSTN connectivity
Hi All,
I got a busy message like all lines are currently busy and please
try again later in call to ZAP trunk. Please help me to resolve this issue
== Using SIP RTP TOS bits
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michael k
Sent: Thursday, October 06, 2011 8:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PSTN connectivity
Hi All,
I got a busy message like all lines are currently busy and please
try
-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *michael k
*Sent:* Thursday, October 06, 2011 8:46 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] PSTN connectivity
** **
Hi All
Hi,
Please see the sample.
A ) Analog HardwareType Ports Action FXO Ports 1
Edithttp://192.168.1.134/admin/config.php?type=setupdisplay=dahdidahdi_form=analog_signallingports=fxo
FXS
Ports --
B) Analog FXO PortsPort 1: *Loop Start* Group: *0* Context: * from-analog*
*
C ) ZAP Trunk (DAHDI
The Call at this point is not even looking for FXO/Dahdi/Zap.. See the CLI.
there is some misconfiguration in FreePBX and your dialled number is not
hitting any dial-able rule. See your FreePBX guide.
On Thu, Sep 29, 2011 at 11:01 AM, michael k mich...@inapp.com wrote:
Hi,
Please see the
Can you please figure out the configuration issue in my freepbx ?
On Thu, Sep 29, 2011 at 11:35 AM, Sam Govind govoi...@gmail.com wrote:
The Call at this point is not even looking for FXO/Dahdi/Zap.. See the CLI.
there is some misconfiguration in FreePBX and your dialled number is not
Actually its easier. I haven't worked on FreePBX lately so what I remember
is here: You've created a Zap trunk. Focus on the Dial Rule - You can keep
it empty as well. Then you've created an outbound route its dial-rule is
important.
But the funny thing which I didn't mention before is that
Thanks for the update. but how do i resolve this issue ? can you help me
please ?
On Thu, Sep 29, 2011 at 1:00 PM, Sam Govind govoi...@gmail.com wrote:
Actually its easier. I haven't worked on FreePBX lately so what I remember
is here: You've created a Zap trunk. Focus on the Dial Rule - You
michael k wrote:
Thanks for the update. but how do i resolve this issue ? can you help me please
?
Can you PLEASE take this to the FreePBX support group?
It seems obvious to most that therein lies the problem
You are thinking you wish to dial out through the X100, but Asterisk is
On Thu, Sep 29, 2011 at 7:51 AM, michael k mich...@inapp.com wrote:
Thanks for the update. but how do i resolve this issue ? can you help me
please ?
You didn't provide a full CLI trace of the outgoing call, you only supplied
the hangup portion of the call. Please try again.
Also, what are
Hey Warren I thought that these are the complete CLI logs for one call. It
started like == Using SIP RTP CoS mark 5 and from-internal priority-1
..So that seemed legit to me. Yeah I too suspect that dialing rules are not
being matched and thats why Gotoif's are failing.
On Thu, Sep 29, 2011 at
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
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