Hi John,
there is no firewall:
snom -- pbx -- patton -- pstn
It happens ONLY with IVRs. Normal calls are fine. How can it be?
I call my pbx from the customer pbx: when I directly call my phone
it works, when I call a test ivr it does not
Run tcpdump with portrange 1-2 (or what range of ports you use for
rtp) and dial test ivr to see what happening.
2011/10/17 gincantalupo gincantal...@fgasoftware.com
**
Hi John,
there is no firewall:
snom -- pbx -- patton -- pstn
It happens ONLY with IVRs. Normal calls are fine.
Is the ivr using early media?
-Original Message-
From: Anton Kvashenkin anton.juga...@gmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Mon, 17 Oct 2011 12:08:51
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Reply-To: Asterisk
2011/10/11 Kevin P. Fleming kpflem...@digium.com
On 10/11/2011 09:57 AM, Olivier wrote:
It would be great if list-based subscriptions could be added to Asterisk
features.
They can be, if someone wrote the code to handle them.
I'm reading this list daily and strangely, I didn't see
On Sat, Oct 01, 2011 at 07:02:48PM -0300, Gustavo Santos wrote:
Hello,
I need to convert the dahdi_monitor output to dBm0, so I can measure Echo
Return Loss in dB.
dahdi_monitor defaults to recording a signed linear PCM file. But latest
versions (as of 2.4.0) will output a wav file if the
Hi,
found where the problem is.I tried with a Grandstream phone and it
works!!!
The problem is my (crappy) Snom phone.
I'm investigating the probhope to find the cause asap.
Sorry for wasting your time, guys. :)
Giorgio
On 10/14/2011 12:21 PM, gincantalupo wrote:
Hi all,
I'm
On 17 October 2011 11:01, gincantalupo gincantal...@fgasoftware.com wrote:
Hi,
found where the problem is.I tried with a Grandstream phone and it
works!!!
The problem is my (crappy) Snom phone.
I'm investigating the probhope to find the cause asap.
FYI: snom firmware 7.3.30 is
Hi List,
I am using an application chanspy() in asterisk which Is used for spy the
channels which is mention on chanspy(). It's working fine.
But I want to make this spy as per my need. Like make a group of some
channels then spy on these specify channels. I read that thee is an option
in
Hi list.
I need to do an integration with my asterisk server with my PABX (Siemens
HiPath 3550 ver. 4.0). I have the asterisk server, 1 trunk with 30 channel
with ISDN signaling. The scenario is:
Call - modem - PABX IP - Analog PABX
The PABX IP is not communicating with the analog
show channels is giving me a line like:
Local/call_out@playfile-981a,2!local-playfile!call_out!4!Up!MeetMe!PA0008|1qt!3175661010!!3!2!(None)
I extract from that the following ID:
Local/call_out@smvoice-local-public-address-playfile-981a
I am trying to do the AMI Action: Hangup command (which I
In article 4e9c3cbf.1070...@pagestation.com,
Jerry Geis ge...@pagestation.com wrote:
show channels is giving me a line like:
Local/call_out@playfile-981a,2!local-playfile!call_out!4!Up!MeetMe!PA0008|1qt!3175661010!!3!2!(None)
I extract from that the following ID:
On 10/17/2011 10:33 AM, Jerry Geis wrote:
show channels is giving me a line like:
Local/call_out@playfile-981a,2!local-playfile!call_out!4!Up!MeetMe!PA0008|1qt!3175661010!!3!2!(None)
I extract from that the following ID:
Local/call_out@smvoice-local-public-address-playfile-981a
I am
Hi folks,
I'm having an issue with an asterisk 1.4.36 with an E1 card that is not
forwarfing the early media a remote SIP end-point is creating.
--incoming E1 call--asterisk 1.4.36SIP endpoint (which happens to be
an asterisk 1.6.20).
I've checked signalling and the remote end-point returns
Hi List,
What is the diffidence between A Generic SIP Device and
Generic ZAP Device while we create an extension in FreePBX ?
Mic
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New to
Asterisk Project Security Advisory - AST-2011-012
Product Asterisk
Summary Remote crash vulnerability in SIP channel driver
Nature of Advisory Remote crash
The Asterisk Development Team has announced a security release for Asterisk 1.8.
The available security release is released as version 1.8.7.1.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases
The release of Asterisk 1.8.7.1
A Generic SIP device would be a SIP trunk, hardphone or softphone. A
Generic ZAP device would be a Zaptel/DAHDI device like a TDM400P or OBI110.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michael k
Sent: Monday, October 17, 2011
Hello,
Trying to upgrade (from asterisk18-1.8.6.0-1) to the latest RPM
version from Asterisk repo I found that asterisknow-version is needed
by package asterisk18-core-1.8.7.0-2
How could this be explained?
Best regards,
Ioan
#
[root@localhost ~]# yum update asterisk18* -x
On 10/17/2011 02:22 PM, Ioan Indreias wrote:
Hello,
Trying to upgrade (from asterisk18-1.8.6.0-1) to the latest RPM
version from Asterisk repo I found that asterisknow-version is needed
by package asterisk18-core-1.8.7.0-2
How could this be explained?
Best regards,
Ioan
The
On Mon, Oct 17, 2011 at 10:37 PM, Jason Parker jpar...@digium.com wrote:
On 10/17/2011 02:22 PM, Ioan Indreias wrote:
The asterisknow-version package contains the repository files (see
/etc/yum.repos.d/) for the repositories on packages.asterisk.org and
packages.digium.com. Installing this
You could possibly get the IP from sip show peers then curl back to that
address.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Robins
Sent: Friday, October 14, 2011 7:31 PM
To: Asterisk Users Mailing
Hello list,
A client is asking to setup an asterisk based conferencing solution
which could handle 10,000 participants (in one single conference or
combined in multiple conferences), and later could be scaled to handle
up to 50,000 participants. All callers will be over SIP, using g711.
I
On Mon, 17 Oct 2011, VisionVoIP wrote:
A client is asking to setup an asterisk based conferencing solution
which could handle 10,000 participants (in one single conference or
combined in multiple conferences), and later could be scaled to handle
up to 50,000 participants. All callers will be
Hi List,
Is there any way by whcih I can make group of user as per my requiremt and
start spy on these channels whic Chanspy ?
-
Thanks and regards
Virendra Bhati
+91-9172341457
Software Engineer
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