Two very simple solution for your problem:
1-Port redirection in iptables. This I have used for a year or plus and it
worked fine for me. I have redirected 1000 ports to a single port 5060 in
iptables and it worked smooth.
2-There is a script in asterisk source directory to compile portable
Hello All,
Is there any one who can help me to change the From field parameters in
option packets, I have seen that in option packtes asterisk sends its own
information,If you see the below option packet i have highlighted the
asterisk word in from field and in from field tag how can i changed it
On 07/20/2011 05:00 AM, Masood Ahmed wrote:
Hello All, Is there any one who can help me to change the From
field parameters in option packets, I have seen that in option
packtes asterisk sends its own information,If you see the below
option packet i have highlighted the asterisk word in from
user-agent could be set in sip.conf
On Wed, Jul 20, 2011 at 12:43 PM, Alex Balashov
abalas...@evaristesys.comwrote:
On 07/20/2011 05:00 AM, Masood Ahmed wrote:
Hello All, Is there any one who can help me to change the From
field parameters in option packets, I have seen that in option
Good morning,
I am writing a Asterisk dialplan from scratch (for learning and
testing purposes), but i'm having trouble with a algorithm to dial a SIP
group using round-robin. I want that asterisk dial the member of the
group in a circular way, until the call be answered. For example, i have
Try using local channel to accomplish that.
by example :
you have 2 phones in the group and want to dial those phones in
the following fashion. Dial phone 1 first after 15 sec if phone1 does
not pickup dial phone2 :
[group-call]
exten =
On Wednesday 20 Jul 2011, Antonio Modesto wrote:
I am writing a Asterisk dialplan from scratch (for learning and
testing purposes), but i'm having trouble with a algorithm to dial a SIP
group using round-robin. I want that asterisk dial the member of the
group in a circular way, until the
Oh man, so easy, thank you very much!
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
sip.conf
useragent = myasteriskbox
sdpsession = myasteriskbox
On Wed, 2011-07-20 at 12:45 +0300, Israel Gottlieb wrote:
user-agent could be set in sip.conf
On Wed, Jul 20, 2011 at 12:43 PM, Alex Balashov
abalas...@evaristesys.com wrote:
On 07/20/2011 05:00 AM, Masood Ahmed wrote:
Dear Sir,
Can you confirm please if any version of asterisk does support ilbc 20ms
instead of 30 ms sample frequency?
Regards
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us
Hello,
I'm putting Asterisk in to replace an existing IVR and that PBX
system uses * to terminate number input instead of #. I thought it would be
a matter of simply making a new app_read that replaced \# with \* but this
didn't work. Any suggestions (besides bopping the client up
Hello All,
Could some one help me solve this.
I want to configure asterisk as a MRF. The call flow is attached. The
Asterisk server has to receive Invites send 200 OK with updated sdp.
If Asterisk has to register with the proxy for this. I can do the necessary
changes in my APP also.
regards,
Good afternoon,
I am trying to use the System() application but it is always
returning APPERROR in the ${SYSTEMSTATUS} variable, I am trying to run
this command:
System(/bin/sh /var/spool/asterisk/calllog/log.sh ${FromExt}
${exten});
This is the content of the
We have multiple customers running on a single Asterisk 1.4 installation and
therefore require a large number of pickup groups. There seems to be a
limitation of 64 call groups. Can anyone suggest how we work around this?
For example is this limitation removed in a later version, is there a patch,
Are you able to execute: log.sh through the asterisk user?
On Wed, 20 Jul 2011 14:53:53 -0300, Antonio Modesto
mode...@isimples.com.br wrote:
Good afternoon,
I am trying to use the System() application but it is always
returning APPERROR in the ${SYSTEMSTATUS} variable, I am trying to
Are there any plans to include the ISAC codec in Asterisk? Is it possible or
even desirable? Is ISAC open source (nothing indicates it is from the WebRTC
website http://www.webrtc.org)?
--
_
-- Bandwidth and Colocation Provided
The problem was the directory which i was writing the logs, i put the
log file in /var/log/asterisk and it worked.
Thanks.
On Wed, 2011-07-20 at 13:03 -0500, Jorge Gutiérrez wrote:
Are you able to execute: log.sh through the asterisk user?
On Wed, 20 Jul 2011 14:53:53 -0300, Antonio
On Wed, 20 Jul 2011, Antonio Modesto wrote:
System(/bin/sh /var/spool/asterisk/calllog/log.sh ${FromExt} ${exten});
Specifying '/bin/sh' is not necessary.
The system() dialplan application calls the execl() system function with
the command '/bin/sh -c' so specifying '/bin/sh foo' results in
On Wed, 20 Jul 2011, Danny Nicholas wrote:
I’m putting Asterisk in to replace an existing IVR and that PBX system
uses * to terminate number input instead of #.
How about an AGI executing some mix of get data, get option, stream file,
or wait for digit and accumulate the digits yourself.
Sent from my Computer
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Steve Edwards
Sent: Wednesday, July 20, 2011 4:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Wednesday, July 20, 2011 3:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] HELP - Client wants
Hello,
for billing purpose between a multitenant asterisk box and another asterisk,
I am in the need to maintain multiple SIP trunks between them. Usually I use
insecure=invite,port but I had to remove or the trunks will be selected
based on IP address and not with username/password. I had to use
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Wednesday, July 20, 2011 3:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] HELP - Client
On Wed, 20 Jul 2011, Danny Nicholas wrote:
I?m putting Asterisk in to replace an existing IVR and that PBX system
uses * to terminate number input instead of #.
On Wed, 20 Jul 2011, Steve Edwards wrote:
How about an AGI executing some mix of get data, get option, stream
file, or wait for
On Thu, Jul 21, 2011 at 06:29:38AM +1200, CB wrote:
Are there any plans to include the ISAC codec in Asterisk? Is it possible or
even desirable? Is ISAC open source (nothing indicates it is from the WebRTC
website http://www.webrtc.org)?
What do you need it for?
--
Tzafrir
Hi List,
My asterisk box was hacked! Can anyone help on how do I secure my
asterisk box, currently my box is installed with 2 NIC. 1st NIC is for
LAN access and 2nd NIC has a public IP which is registered to our VoIP
Provider.
As I remember I already tried putting our Box on NAT but
26 matches
Mail list logo
