You can create trunk/route specific dial command parameters.
Regards,
Faisal Hanif
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Davies
Sent: Friday, August 24, 2012 8:40 PM
To: Asterisk Users Mailing
On Sat, Aug 25, 2012 at 12:04:48AM +, ebusic wrote:
> I have the same problem with OpenVOX D210E card
>
> These one is not working:
>
> dahdi: Telephony Interface Registered on major 196
> dahdi: Version: 2.6.1
[snip]
> and these one working fine:
>
> dahdi: Telephony Interface Registered
I have the same problem with OpenVOX D210E card
These one is not working:
dahdi: Telephony Interface Registered on major 196
dahdi: Version: 2.6.1
ACPI: PCI Interrupt :03:00.0[A] -> GSI 16 (level, low) -> IRQ 169
wct4xxp :03:00.0: Firmware Version: c01a
wct4xxp :03:00.0: FALC Fram
> Using
>
>
> exten => h,n,set(CDR(llp)=${CHANNEL(rtpqos,audio,local_lostpackets)})
>
> gives me
>
>
> [Aug 24 12:08:10] WARNING[12087]: sip/dialplan_functions.c:221
> sip_acf_channel_read: Unrecognized argument
> 'rtpqos,audio,local_lostpackets' to CHANNEL
>
> [Aug 24 12:08:10] WARNING[12087
On 24 August 2012 15:34, Faisal Hanif wrote:
> Steve Davies wrote:
>>Hi SIP Gurus,
>>
>>I've tried to find the relevant RFCs, but am struggling. I can find
>>the odd opinion online, but was wondering if anyone could give a
>>definitive answer.
>>
>>If a SIP call is initiated (INVITE) and receives
Thank you Eric, but this is still the old documentation:
>
> rtpqos - R/O Get QOS information about the RTP stream
>
> This option takes two additional arguments:
>
> Argument 1:
>
> 'audio' Get data about the audio stream
>
> 'video' Get d
pbx*CLI> core show function CHANNEL
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stefan at WPF
Sent: Friday, August 24, 2012 6:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [aste
- Original Message -
> A simply PHP based thing would be OK. Maybe I should look more
> specifically for that or can anyone here recommend a PHP based CDR
> viewer?
> Meanwhile I ended up building a mysql view, for private purposes it
> does the job. A real solution would still be nice, t
hi,
you can simply avoid this by using local ring r option in dial command.
azterisk pass local ring voice to caller and will not bridge b leg audio until
b leg is answered.iin
Regards,
Faisal Hanif
(sent from phone)
Steve Davies wrote:
>Hi SIP Gurus,
>
>I've tried to find the relevant RFCs
Hi SIP Gurus,
I've tried to find the relevant RFCs, but am struggling. I can find
the odd opinion online, but was wondering if anyone could give a
definitive answer.
If a SIP call is initiated (INVITE) and receives either a "180 with
SDP", or a "183 with SDP", then the remote party will start to
One trick you can do is to accept all calls into the dial plan and then do IP
lookups and call pattern checks to determine if the call is good to go past
your sidewalk code. You need to make sure this code is very efficient so that
you can lock out bogus callers and attackers. If you use this
I ended up writing a basic parsing script that lets me search the full log,
based on some unique identifier (eg, my own extension "vlog 2027"). It then
digs out the associated A*k log number for each line that's it, and lists them
out. Then I choose the 'call' and it re-filters by that call only
Actually, you could look for WARNING or ERROR and probably find what you
needed.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stefan at WPF
Sent: Friday, August 24, 2012 8:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discu
Thank you Danny, but the problem is that I don't know what exactly I shall
look for. I think there's no specific word in the log that clearly
identifies this kind of problem? ):
2012/8/24 Danny Nicholas
> Not the best solution, but you could do a “quick and dirty” crawler to
> query /var/log/ast
Not the best solution, but you could do a "quick and dirty" crawler to query
/var/log/asterisk/full in PHP or PERL or your language of choice. Even in a
4K-5K calls per day environment this process usually takes less than 1
minute to run.
From: asterisk-users-boun...@lists.digium.com
[mailto:a
If somebody is calling me using a wrong configured SIP phone, he gets back
an error message from my Asterisk server. That's ok, however I'd also like
to know that I missed a call. However there's no CDR entry created in that
case and checking the asterisk logs manually is not that great... Any way
I will test this allso. Thanks.
- Original Message -
From: "Eric Wieling"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, 23 August, 2012 9:32:26 PM
Subject: Re: [asterisk-users] sip trunk failing to register causes sip
phones to become unreachabl
Hi,
Thanks. I will try this.
Regards,
John
- Original Message -
From: "Warren Selby"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, 23 August, 2012 9:24:48 PM
Subject: Re: [asterisk-users] sip trunk failing to register causes sip phones
to become unrea
A simply PHP based thing would be OK. Maybe I should look more specifically
for that or can anyone here recommend a PHP based CDR viewer?
Meanwhile I ended up building a mysql view, for private purposes it does
the job. A real solution would still be nice, though.
2012/8/23 Tim Nelson
> - Or
Hi Hans,
On 24-08-12 10:13, Hans Witvliet wrote:
Hi all,
After making a nice demo-setup for one of our innivationmanagers, he
came up with a completely different stratagy ;-(
Well if you could create it then obviously it's no longer innovative so
they had to come up with something else :-)
Ok,
This is something to do with folder layouts.
I have:
/var/lib/asterisk/sounds - uk files
/var/lib/asterisk/sounds/digits -uk/us digits
/var/lib/asterisk/sounds/jp - Japanese files
/var/lib/asterisk/sounds/jp/digits - Japanese digits
I read the 1.4 notes on :
http://www.voip-
Ok... I'm baffled..
I took a copy of my machine and put it in a virtual machine, then upgraded the
VM to 1.4.44 to experiment, and unknowingly let it install the default US GSM
sounds again.
My code runs, but, it still plays the US digits when the debug says the below.
You can see its set
Using
exten => h,n,set(CDR(llp)=${CHANNEL(rtpqos,audio,local_lostpackets)})
gives me
> [Aug 24 12:08:10] WARNING[12087]: sip/dialplan_functions.c:221
>> sip_acf_channel_read: Unrecognized argument
>> 'rtpqos,audio,local_lostpackets' to CHANNEL
>
> [Aug 24 12:08:10] WARNING[12087]: func_channel.
Hi Chris, Thanks for replying,
I've got it set in the context in extensions.conf:
[TokyoReception]
exten => s,1(TOKYORECEPTION),Answer
exten => s,n,Set(CHANNEL(language)=jp) ; set japanese by default
exten => s,n,SET(LOOP=0)
exten => s,n,SET(LANG=JP)
It could be something fixed between 1
Hi all,
After making a nice demo-setup for one of our innivationmanagers, he
came up with a completely different stratagy ;-(
They want to have an Ejabberd server, with xmpp-clients.
When you see a contact coming online, just point and click for making a
phone call.
Sounds/looks nice and do-able
The built in "file convert" function has saved my bacon when I need to convert
quickly.
Apart from that, there are a ton of examples with external programs on
voip-info.org:
http://www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk
http://www.voip-info.org/wiki/view/Asteris
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