Hello,
I tried to search the answer of my problem but unable to get resolution so
sending to you guys. I'm using asterisk 1.6.2.7 and writing the AGI scripts
using PHP. I'm unable to retrieve the DIALSTATUS value, during execution of AGI
script, I get empty value.
Extracts from AGI Script:
Hi,
How are you calling this AGI in your dialplan !!?
Regards,
Sammy.
On Fri, Dec 2, 2011 at 3:18 PM, Kamlesh Kumar kamlesh_...@hotmail.comwrote:
Hello,
I tried to search the answer of my problem but unable to get resolution so
sending to you guys. I'm using asterisk 1.6.2.7 and writing
Hello,
in /etc/extension.conf
[privoip]
exten = _00X.,n,AGI(isdcall.php)
Regards,
Kamlesh
Date: Fri, 2 Dec 2011 16:16:27 +0500
From: govoi...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] DIALSTATUS Values
Hi,
How are you calling this AGI in your dialplan
Can you also paste the Asterisk Console logs around the part where AGI is
dialing and after the dialing part ! make sure AGi debug is enabled as well.
On Fri, Dec 2, 2011 at 4:24 PM, Kamlesh Kumar kamlesh_...@hotmail.comwrote:
Hello,
in /etc/extension.conf
[privoip]
exten =
Here it is:
SIP/10036-00a8AGI Tx agi_request: isdcall.php
SIP/10036-00a8AGI Tx agi_channel: SIP/10036-00a8
SIP/10036-00a8AGI Tx agi_language: en
SIP/10036-00a8AGI Tx agi_type: SIP
SIP/10036-00a8AGI Tx agi_uniqueid: 1322853473.198
SIP/10036-00a8AGI Tx
In article snt142-w45a64e4743de653da591...@phx.gbl,
Kamlesh Kumar kamlesh_...@hotmail.com wrote:
I tried to search the answer of my problem but unable to get resolution so
sending to you
guys. I'm using asterisk 1.6.2.7 and writing the AGI scripts using PHP. I'm
unable to
retrieve the
I believe the syntax is correct because,
If I use
$dd=$dialstatus[code];
$agi-verbose(Status.$dd);
it gives me:
SIP/10036-00b2AGI Rx GET VARIABLE DIALSTATUS
SIP/10036-00b2AGI Tx 200 result=1 (ANSWER)
SIP/10036-00b2AGI Rx VERBOSE Status200 1
If I use
In addition to my reply:
I used to fetch the value using print_r function but that also tells that there
is no value in data section.
$dialstatus=$agi-get_variable(DIALSTATUS);
print_r($dialstatus);
SIP/10036-00b8AGI Rx GET VARIABLE DIALSTATUS
SIP/10036-00b8AGI Tx 200 result=1
In article snt142-w54267269808afd17bccd5891...@phx.gbl,
Kamlesh Kumar kamlesh_...@hotmail.com wrote:
In addition to my reply:
I used to fetch the value using print_r function but that also tells that
there is no value
in data section.
$dialstatus=$agi-get_variable(DIALSTATUS);
On Thursday 01 December 2011, Hans Witvliet wrote:
On Thu, 2011-12-01 at 14:02 +, A J Stiles wrote:
On Thursday 01 December 2011, gincantalupo wrote:
Hi all,
any idea about how to replace Skype For Asterisk?
Thank You.
Giorgio
1. Migrate your Skype users over
On 11/26/2011 5:00 PM, C F wrote:
On Sat, Nov 26, 2011 at 7:50 AM, Gordon Henderson
gordon+aster...@drogon.net wrote:
On Sat, 26 Nov 2011, Terry Brummell wrote:
Install Configure Fail2Ban then the host will be blocked from
connecting. And no, it's not new.
I don't need Fail2Ban, thank
Fail2ban assumes that #1 your environment is (wide) open and #2 you will
need to update iptables on an instant response to attack basis. If you
are open enough, even fail2ban isn't going to really help. If you have a
sufficiently written set of iptables rules (or you aren't allowing external
I've just connected my new Android (Motorola RAZR) phone to Asterisk
using CSipSimple and have discovered that on any call between CSipSimple
and an Asterisk DAHDI or IAX2 channel, the 'other' end of the call will
hear a rhythmic tapping as if my voice stream is being chopped up in
equal parts
On Fri, 2 Dec 2011, Jim Lucas wrote:
How is using Fail2Ban less resource intensive then me writing (by hand)
iptable rules?
It depends on how you define resources and how much of those resources you
have.
Gordon (based on my understanding of his posts) does a lot of Asterisk
systems on
On 12/2/2011 12:44 PM, Steve Edwards wrote:
On Fri, 2 Dec 2011, Jim Lucas wrote:
How is using Fail2Ban less resource intensive then me writing (by
hand) iptable rules?
It depends on how you define resources and how much of those resources
you have.
Gordon (based on my understanding of his
Hello all,
I recently found this when looking an IAX trunk:
context=*
Does it have a special meaning or is it the same like 'default'?
Thanks,
Elder
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Hello,
I have been trying to playback a video file via Playback() in Asterisk
1.8.7.1 but the file format seems to fail.
[2011-12-02 18:46:24] WARNING[7665]: file.c:653 ast_openstream_full: File
/etc/asterisk/cp-10fps-QCIF-20Kbps.h263 does not exist in any format
[2011-12-02 18:46:24]
I am receiving requests to register to my Asterisk extensions. I have the
full SIP packets. I also do see what extension is being tried to be
registered. Is there ANY WAY to know what password is being attempted?
I think the appropriate term would be decode the base64 response I get from
the
Hi,
Has anyone succeded using DHCP Option 43 and Aastra phones to set the
configuration server from a pfSense router or any other router?
Sorry, if not directly related to Asterisk but I am sure the collective
knowledge will pay off.
I am specifically wondering what the Number, Type and Value
Hello,
Is there a php or any other program to analyse Asterisk CDR which is stored
in asteriskcdrdb. I want to know outbound and inbound channels and not
the internal calls channels as well which is what CDR Stats does currently.
It doesn't differentiate between those.
Someone might have done a
On 12/02/2011 05:24 PM, asterisk jobs wrote:
I am receiving requests to register to my Asterisk extensions. I have
the full SIP packets. I also do see what extension is being tried to be
registered. Is there ANY WAY to know what password is being attempted?
I think the appropriate term would be
hi folks.
when i use regular PSTN(sip phone - asterisk - PRI) to call
certain numbers and when that number is unavailable. i usually
hear an early media message saying blahblah is unavailable,
please try again. but when i use skype connect(sip phone - asterisk
- skype connect). i just hear ring
As the Authorization header clearly states, this value is created using an
MD5 Digest (hash). Since it is a digest function, it is not reversible. It
is impossible to recover the password that was used during the calculation
of the response value (although given enough time and CPU resources,
On Fri, Dec 2, 2011 at 12:44 PM, Steve Edwards
asterisk@sedwards.com wrote:
Gordon (based on my understanding of his posts) does a lot of Asterisk
systems on very limited hardware hosts. His approach uses iptables features
to limit the number of SIP INVITES and REGISTERS per second per IP
When a caller calls my google voice phone number, I must answer, wait and
press one to accept. Sometimes even that does not work.
I have tried a few different things to get asterisk to place the call in an
answered state and send the DTMF 1 with the Dial macro.
I found Malcom Davenports wiki
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