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
P
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 fie
user-agent could be set in sip.conf
On Wed, Jul 20, 2011 at 12:43 PM, Alex Balashov
wrote:
> 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
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
t
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 => group1,Dial(Local/phone1@group-call&Local
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 t
Oh man, so easy, thank you very much!
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as
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
> 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
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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 si
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,
Mo
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 /var/spool/asterisk/cal
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
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 run
> this command:
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)?
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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.
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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 to
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 t
-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 wants
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 dig
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 C
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 unfortun
Hello!
First of all, you should disable unused VoIP protocols. Than remove all
guest accounts from used protocols, disable guest unauth access.
Always use strong passwords for accounts, for users on your system.
Passwords shouldn't be eq username. Move port binds on LAN network for
all active
Thanks. Any link for me to check for the procedure to implement those?
Regards,
Malvin
On 7/21/2011 1:59 PM, Захаров Антон wrote:
Hello!
First of all, you should disable unused VoIP protocols. Than remove
all guest accounts from used protocols, disable guest unauth access.
Always use strong p
On 21.07.2011 09:29, Malvin Rito wrote:
My asterisk box was hacked!
On Thu, 21 Jul 2011, Захаров Антон wrote:
First of all, you should disable unused VoIP protocols.
Once a box has been hacked you cannot trust anything.
Disconnect the box from the network, save whatever DATA ONLY you can
What does "but this didn't work" mean?
Am 20.07.2011 15:25, schrieb Danny Nicholas:
Hello,
I’m putting Asterisk in to
replace an existing IVR and that PBX system uses * to
terminate number input in
Yeap, drop out box is normal idea. But it's strongly wired what type of
hack was. If it was only traffic leak without any footsteps in your
system (shell history, files modification time, logs) I don't think that
box couldn't be used any longer. Try to use port knocking (
http://www.portknockin
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