Hi List,
Yes you are right but I want to cross check to outside world to. How they
will support me in such case...
:)
On Thu, Jun 16, 2011 at 11:23 AM, Alex Balashov
abalas...@evaristesys.comwrote:
I thought the idea was that Asterisk Engineers already know the answers
to such questions?
Hi,
I am using asterisk1.2
In this, my dialplan is going large , so i need to configure this small
pieces for this, i did in my extensions.conf
when I dial the 123 its not going , means that file is not reading. is there
any parameters to add any where ? please tell me
this #include is not working
Hi all,
I have 100 DID's which is 4578900 to 4578999 , and i have 5001 to 5099
extensions. when incomming call come to this DID no. (4578901) that time
5001 extestinsion should ring.
below my dial plan is not getting any result , inthat has any mistakes.
please help.
exten =
Fail2ban
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Thursday, June 16, 2011 11:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to secure our Asterisk server
On Thu, Jun 16, 2011 at 12:24:15PM +0530, mahesh katta wrote:
Hi,
I am using asterisk1.2
In this, my dialplan is going large , so i need to configure this small
pieces for this, i did in my extensions.conf
when I dial the 123 its not going , means that file is not reading. is there
any
On Thu, Jun 16, 2011 at 12:48:39PM +0530, mahesh katta wrote:
Hi all,
I have 100 DID's which is 4578900 to 4578999 , and i have 5001 to 5099
extensions. when incomming call come to this DID no. (4578901) that time
5001 extestinsion should ring.
below my dial plan is not getting any result ,
Hi,
I hope this is not rude of my part. I normally avoid answering mails
that relate in such way commercially to hardware.
This list is non-commercial If you want to ask questions of commercial
nature, please use Asterisk-biz:
http://lists.digium.com/mailman/listinfo/asterisk-biz
Please
On Thu, Jun 16, 2011 at 12:26:13PM +0500, Faisal Hanif wrote:
Fail2ban
Fail2Ban protects you from one (or two) specific types of attacks:
someone trying to establish many connections to your PBX (e.g.: SIP
register, or SIP invite) in order to try to guess a username/password
combination the hard
On Thu, Jun 16, 2011 at 11:26:21AM +0800, asterisk asterisk wrote:
Hi,
I am looking for a simple solution to do this.
I wish to have the user to enter their preferred method of connection i.e.
for the cheapest solution to their desktop phone or mobile phone, then plan
callfile based on
On Thursday 16 Jun 2011, mahesh katta wrote:
Hi all,
I have 100 DID's which is 4578900 to 4578999 , and i have 5001 to 5099
extensions. when incomming call come to this DID no. (4578901) that time
5001 extestinsion should ring.
below my dial plan is not getting any result , inthat has any
Hi all.
I've got a Grandstream GXP2xxx that is working fine in gerneral. However, when
I try to transfer a call, my dialplan doesn't have access to any of the
channel's variables. It seems that this is a known issue in 1.6.x.
Has it been fixed in 1.8.x? Is 1.8 stable enough for
Hi,
This Friday Chris Matthieu of SMSified.com will explain how to send
SMS from your apps. As usual, there will be talk about Asterisk,
questions, answers, and comments about telephony, networks, VoIP and
even some OT. All are welcome to join the weekly average of 35-60
callers live. If you
-- Executing MixMonitor(Zap/3-1,
/var/spool/asterisk/astrec/20110616-122003-0559566768-4578924-1308212403.2225.gsm|av(0)V(0))
in new stack
-- Executing Dial(Zap/3-1, SIP/50{EXTEN:-2}||tTo) in new
stack
== Begin MixMonitor Recording
Zap/3-1
== Parsing '/etc/asterisk/manager.conf':
Found
On Thursday 16 Jun 2011, mahesh katta wrote:
-- Executing Set(Zap/3-1, Dest=50{EXTEN:-2}) in new
stack
-- Executing MixMonitor(Zap/3-1,
/var/spool/asterisk/astrec/20110616-122003-0559566768-4578924-1308212403.2
225.gsm|av(0)V(0)) in new stack
-- Executing Dial(Zap/3-1, SIP/50
hi, everyone
i want to use sipp to auto answer the ivr， to simulate the keypad send
digital sequence， so i try to send DTMF by application/dtmf-relay, but i
have got this error message in the asterisk CLI, Could you help me? Thanks!
[Jun 16 17:11:34] WARNING[26321]: rtp.c:3207
Hi,
If I am right then will you discuss about the sending sms with asterisk into
that conference ?
On Thu, Jun 16, 2011 at 1:41 PM, randulo rand...@randulo.com wrote:
Hi,
This Friday Chris Matthieu of SMSified.com will explain how to send
SMS from your apps. As usual, there will be talk
Dears;
I am sure that you have experience with Cisco IP Phones. I need to be sure if
someone used Cisco 7942G in skinny firmware with Asterisk 1.8 and how it was
(if fine or it has a problem).
Are the below the only 3 needed files to be placed in the tftpboot directory:
I've no experience with that phone model or protocol. But if you run a tftp
trace you'll see what files the phone is looking for.
Check my old thread on pbxinaflash forums for details.
i
-- Original Message --
Received: 04:59 AM COT, 06/16/2011
From: bilal ghayyad bilmar...@yahoo.com
On Thu, Jun 16, 2011 at 11:52 AM, virendra bhati virbh...@gmail.com wrote:
If I am right then will you discuss about the sending sms with asterisk into
that conference ?
We can if someone wants to, that's how the VUC works.
:r
--
Hi all,
I will really appreciate if you can spend some time to share your experience
or point me in right direction.
I have been told to prepare a single box Asterisk system (No Distributed
architecture) for following features.
-Asterisk 1.8
-300 SIP extensions (sip.conf)
-8 port PRI card (E1)
hello list,
i have asterisk 1.4 with IAX and sip i have configured the MixMonitor in
order to record the conversation
i can record all the calls inbound and outbound without problem.
but when i receive an inbound call from customer in IAX(1000) and i want to
transfer the call to other phone
Check the option of 'd' in Dial().
d: Allow the calling user to dial a 1 digit extension while waiting for a
call to be answered. Exit to that extension if it exists in the current
context, or the context defined in the ${EXITCONTEXT} variable,if it exists.
[SATISH]
On Wed, Jun 15, 2011 at
Hey Elliot;
Would you mind posting your dialplan for your Google Voice config? I am
having a hell of a time getting it to do *anything*.
Perhaps I am just fat-fingering.
Would you mind? Thanks in advance.
Glen
On 6/13/2011 19:02, Elliot Murdock wrote:
Hello,
I am using 1.8.4.2 and while
On 16/06/11 07:36 AM, salaheddine elharit wrote:
hello list,
i have asterisk 1.4 with IAX and sip i have configured the MixMonitor in
order to record the conversation
but when i receive an inbound call from customer in IAX(1000) and i want
to transfer the call to other phone SIP(223)
the
thanks for your response
the call is going to IAX(1000), i have i DID Number when the customer call
this number 0520XX the call is goint to agent
IAX. in my dialplan i have
exten = 223,1,MixMonitor(blah.wav)
exten = 223,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten = 223,n,Dial(SIP/223)
and
Could someone point me in the right direction of how to create a Fast AGI
script without memory leaks? I was told i need to clear the result set for
mysql queries, Im not sure how to do that. My script is a simple perl script
of 70 lines doing database lookups and executing dial and voicemail
I'd
Tzafrir, Whats up with this 1.2 vs 1.8 signature?
On Thu, Jun 16, 2011 at 3:38 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
Hi,
I hope this is not rude of my part. I normally avoid answering mails
that relate in such way commercially to hardware.
This list is non-commercial If you
On 16/06/11 09:20 AM, salaheddine elharit wrote:
thanks for your response
the call is going to IAX(1000), i have i DID Number when the customer
call this number 0520XX the call is goint to agent
IAX. in my dialplan i have
exten = 223,1,MixMonitor(blah.wav)
exten =
On 06/16/2011 07:58 AM, Mike Diehl wrote:
Well, I ran a simple test by trying to configure the second port to use the DNS
SRV record, as described below.
Here is what I have: (sanitized)
==
Proxy_2_ diehlnet.com/Proxy_2_
i have asterisk 1.4 and also i have aheeva applicaton also installed in my
server
in the consol this call may be monitored or recorded
best regrads
2011/6/16 Leif Madsen leif.mad...@asteriskdocs.org
On 16/06/11 09:20 AM, salaheddine elharit wrote:
thanks for your response
the call is
Try switching the Set and MixMonitor commands so the AUDIOHOOK_INHERIT will
be in effect when Mixmonitor starts
exten = 223,1,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten = 223,n,MixMonitor(blah.wav)
exten = 223,n,Dial(SIP/223)
From: asterisk-users-boun...@lists.digium.com
On 15/06/2011 8:15 PM, Matteo Campana wrote:
HI list,
no idea?? :)
There not much substance in the information provided for an assessment
to be made.
I would suggest you capture the network traffic between UAC (g711)
Asterisk UAS ensuring the snap length is large enough to capture the
We experience the same thing. The solution we use is to not change codecs in
the middle of a call. I assumed it was an issue with our upstream.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Larry
On 6/14/2011 5:08 AM, Paul Hayes wrote:
On 13/06/11 19:44, Mike Diehl wrote:
Hi all,
I'm trying to provision my PAP2T's to use a SVR lookup to find the
Asterisk
server. I'm using a provisioning file that contains an element like:
Proxy_1_ _sip._udp.example.com/Proxy_1_
However, the PAP
On Thursday 16 June 2011 8:10:38 am Marius Pedersen wrote:
On 06/16/2011 07:58 AM, Mike Diehl wrote:
Well, I ran a simple test by trying to configure the second port to use
the DNS SRV record, as described below.
Here is what I have: (sanitized)
On 06/16/2011 04:49 PM, Mike Diehl wrote:
On Thursday 16 June 2011 8:10:38 am Marius Pedersen wrote:
On 06/16/2011 07:58 AM, Mike Diehl wrote:
Well, I ran a simple test by trying to configure the second port to use
the DNS SRV record, as described below.
Here is what I have: (sanitized)
hi Danny
thank you for your response i switched the MixMonitor and i still have the
same result
any help please
2011/6/16 Danny Nicholas da...@debsinc.com
Try switching the Set and MixMonitor commands so the AUDIOHOOK_INHERIT will
be in effect when Mixmonitor starts
exten =
On Thu, 16 Jun 2011, vip killa wrote:
Could someone point me in the right direction of how to create a Fast
AGI script without memory leaks? I was told i need to clear the result
set for mysql queries, Im not sure how to do that. My script is a simple
perl script of 70 lines doing database
Hello,
I do not use the skinny firmware. By the way, questions like this are best
shared with the asterisk-users group mailing list, so that a large segment
of the Asterisk community can benefit from the questions and answers.
Cassius Smith
--
On 6/16/11 4:59 AM, bilal ghayyad
Since AUDIOHOOK_INHERIT is a backport from 1.8, something may be amiss in
the 1.4 IAX rendition. I assume your install would not be friendly for a
1.8 upgrade?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine
elharit
Sent:
The Asterisk version is 1.8.3.2
The Cisco IP Phone is 7942G and it is running now skinny.
The used TFTP is tftp-server which is installed in fedora.
I placed the following two files (but look like it was not taken from the
TFTP, as
nothing appeared in the messages), but I am able to to
I have two calls (626 and 542) coming into the same phone (524).
SIP/524-05b5!smvoice-sip!!1!Up!AppDial!(Outgoing
Line)!_2XX!!3!9!SIP/542-05b4
SIP/542-05b4!smvoice-sip!_2XX!8!Up!Dial!SIP/524|30|!542!!3!9!SIP/524-05b5
SIP/524-05b3!smvoice-sip!!1!Up!AppDial!(Outgoing
On 16/06/11 19:12, Cassius Smith wrote:
Hello,
I do not use the skinny firmware. By the way, questions like this are best
shared with the asterisk-users group mailing list, so that a large segment
of the Asterisk community can benefit from the questions and answers.
Cassius Smith
Agreed
--
Hi All
Just upgraded from 1.6? to 1.8.4.1
I ised to be able to get a digital call working across a bridged isdn
channel in 1.6 and 1.4 using the following;-
exten = _X.,1,gotoif($[${TRANSFERCAPABILITY}=DIGITAL]?5:)
exten = _X.,2,dial(DAHDI/g1/${EXTEN})
exten = _X.,3,Noop(${CHANNEL})
exten =
I'm using ISDN30 for a bridged application
in all the old versions of asterisk the time slot number is shown in the
channels and dstchannel fields of the cdr
I understand this has chaned in 1.8,is there a way of getting the time slot
information stored somewhere at the end of the call so this
After a Good Call from a PSTN phone if I do a sip prune realtime peer
9013XX9XX8 (9013XX9XX8 being the phone number of the Agent/Member) then I
can call the number again and not get the issue. So this has something to
do with the stuff that is put in my peer table after a call.
Any ideas?
On
It is possible to log queue in mysql without turning on realtime asterisk?
Thanks!
[]'sf.rique
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar
I'm using ISDN30 for a bridged application
in all the old versions of asterisk the time slot number is shown in
the channels and dstchannel fields of the cdr
I understand this has chaned in 1.8,is there a way of getting the time
slot information stored somewhere at the end of the call so
Hi All
Just upgraded from 1.6? to 1.8.4.1
I ised to be able to get a digital call working across a bridged isdn
channel in 1.6 and 1.4 using the following;-
exten = _X.,1,gotoif($[${TRANSFERCAPABILITY}=DIGITAL]?5:)
exten = _X.,2,dial(DAHDI/g1/${EXTEN})
exten =
Do anyone know how to receiving incoming call from GV number associated with
an non gmail.com account? I have custom domains under google and would like
to receiving calls via asterisk.
The google chat function is missing in these GV accounts.
On Thu, Jun 16, 2011 at 11:30 AM, asterisk asterisk
Only GV numbers that can terminate to a Google Chat Account can be connected
directly to asterisk.
Otherwise you will need to get a free SIP Account, and route calls to it.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Can this non gmail.com GV number be terminated at some sip accounts so that
I can bridge to it via asterisk as client?
On Fri, Jun 17, 2011 at 11:48 AM, William Stillwell
will...@stillwellsoft.com wrote:
Only GV numbers that can terminate to a Google Chat Account can be
connected directly to
Thanks. Will need some time to look into.
On Thu, Jun 16, 2011 at 3:56 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Thu, Jun 16, 2011 at 11:26:21AM +0800, asterisk asterisk wrote:
Hi,
I am looking for a simple solution to do this.
I wish to have the user to enter their
I believe you can send a GV # to any US Phone number.
That is beyond the scope of this list.
But in order to directly terminate the GV # into Asterisk (without using
SIP) you must be able to terminate the GV # into a Google Chat Account.
as what is being done is theoretically making
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