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 pag
On Fri, Dec 2, 2011 at 12:44 PM, Steve Edwards
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 address.
A very narrow
>
> 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 resour
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
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
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
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 sh
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 clien
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] WARNING[76
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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Ne
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
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 ver
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 abou
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
SI
On 11/26/2011 5:00 PM, C F wrote:
> On Sat, Nov 26, 2011 at 7:50 AM, Gordon Henderson
> 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 you. But you
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. Migr
In article ,
Kamlesh Kumar 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);
> print_r($dialstatus);
>
> AGI Rx << GET VARIABLE DIALSTATUS
> AGI T
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);
AGI Rx << GET VARIABLE DIALSTATUS
AGI Tx >> 200 result=1 (CANCEL)
AGI Rx << Array
AGI Tx >> 5
I believe the syntax is correct because,
If I use
$dd=$dialstatus["code"];
> > $agi->verbose("Status".$dd);
it gives me:
AGI Rx << GET VARIABLE DIALSTATUS
AGI Tx >> 200 result=1 (ANSWER)
AGI Rx << VERBOSE "Status200" 1
If I use
$dd=$dialstatus["result"];
> > $agi->verbose("Status".$dd);
it
In article ,
Kamlesh Kumar 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 DIALSTATUS value, during execution of AGI script, I get empty
Here it is:
AGI Tx >> agi_request: isdcall.php
AGI Tx >> agi_channel: SIP/10036-00a8
AGI Tx >> agi_language: en
AGI Tx >> agi_type: SIP
AGI Tx >> agi_uniqueid: 1322853473.198
AGI Tx >> agi_version: 1.6.2.7
AGI Tx >> agi_callerid: 10036
AGI Tx >> agi_calleridname: 10036
AGI Tx >> agi_callingp
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 wrote:
> Hello,
>
> in /etc/extension.conf
>
> [privoip]
> exten => _00X.,n,AGI(isdcall.php)
>
> R
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
Hi,
How are you calling this AGI in your dialplan !!?
Regards,
Sammy.
On Fri, Dec 2, 2011 at 3:18 PM, Kamlesh Kumar wrote:
> 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
> usin
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:
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