Hi,
We are facing an issue with asterisk in the case of call-Transfer scenarios.
Our requirement is to identify whether an incoming call is a fresh incoming
call or a Transferred call from some other clients.
We have a setup, where in the asterisk1.6 (as SIP server) is running in Linux
Marco
Did you get to the bottom of this.
I've just come across the same problem today also with a B410P after
upgrading from debian lenny( 2.6.26-2-686 ) to squeeze ( 2.6.32-5-686 ).
On reboot I often get BUG: soft lockup - CPU#0 stuck for 61s[swapper:0]
The only fix is to power off
but
Hi,
how many UDP ports is required for 1 call. and why .
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
--
_
-- Bandwidth and Colocation Provided by
On 02/21/2012 07:30 AM, virendra bhati wrote:
Hi,
how many UDP ports is required for 1 call. and why .
A 'call' is too ambiguous to answer your question. Is this a voice call,
a video/voice call, a FAx call, a T.140 text call, or something else?
--
Kevin P. Fleming
Digium, Inc. | Director
right now it's only voice call.
But thanks for segregate the call.
Now i want to know about all calls used port too.
On Tue, Feb 21, 2012 at 7:06 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 02/21/2012 07:30 AM, virendra bhati wrote:
Hi,
how many UDP ports is required for 1 call.
As many ports as required by the nature of the call, i.e. the protocol(s) used
for the bearer.
--
This message was painstakingly thumbed out on my mobile, so apologies for
brevity and errors.
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
On 02/21/2012 07:51 AM, Alex Balashov wrote:
As many ports as required by the nature of the call, i.e. the
protocol(s) used for the bearer.
For an IAX2 call, the answer is 'zero' for all of those call types (at
least the ones that are supported in IAX2, not all of them are).
For protocols
asterisk-users@lists.digium.com wrote:
Please click the link the NACHA site and update your user
account:ID0664474
Interesting. Came from tipas...@gmail.com
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither
What release are you trying this with?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant
Zimmerman
Sent: Monday, February 20, 2012 5:34 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Park and PARKINGDYNAMIC
Danny
I am on 1.8.x
I also have 1.10 boxes up but have not tried it there yet. According to the
change logs it should work from 1.8 and up but it does not appear to do so.
I have been going through the source code trying to figure it out as there
are no real doc's on it as of yet. If I can
Is it possible to define a customize the which sound file is played when I
send a caller to VoiceMailMain()?
By default the sound file is vm-login.codec.
Is there a way to specify which sound file is played per context or some
other way to play a different sound file in place of vm-login?
I
I believe this is what you want. Instead of this
Exten = _X.,123,Voicemail(100)
Do
Exten = _X.,123,playback(your-message)
Exten = _X.,123,voicemail(100,s)
Per the instructions, (100) plays the standard message, (100,b) plays busy
(100,u) plays unavailable and (100,s) plays nothing
Danny,
This seems to be a solution for sending people to leave a voicemail, I
need a solution for VoiceMailMain() when people call in to get their
messages, change greeting etc.
If I use the s option with VoiceMailMain it just skips checking the
passcode according to the docs.
Thanks for your
There was a kludgy solution posted a while back that might work for you.
Since Asterisk is multi-lingual you could do this
Exten = _X.,123,Set(CHANNEL(language)=fr)
Exten = _X.,124,Voicemailmain()
This assumes you aren't using fr(French). Just copy
/var/lib/asterisk/sounds/en to
From: Todd Routhier fonema...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, February 21, 2012 11:30:34 AM
Subject: Re: [asterisk-users] Define custom vm-login sound file per
VM context?
Danny,
This seems to be a
Wow, that makes me wonder if I could do something like:
Set(CHANNEL(language)=Cust327)
Then create a Language folder named Cust327 and have it just work.
Weee... :-)
Of course that leads me to think that I could have whole sets of custom
sounds for all of Asterisk based on setting this Language
If I recall correctly, it does have to be a real country and a two-letter
code, but that still gives you hundreds of variants for this kludge.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Todd Routhier
Sent: Tuesday, February 21,
OK, this will work and is probably a better solution than the language
idea. Although, the language idea just sounds easier and a little more fun
:-)
Hmm, I think I will try the language solution and see if it works with a
fake country/language code like Cust327 or whatever.
Just wonder if that
2012-02-21 19:20, Todd Routhier skrev:
OK, this will work and is probably a better solution than the language
idea. Although, the language idea just sounds easier and a little more
fun :-)
Hmm, I think I will try the language solution and see if it works with
a fake country/language code
Wow, that looks like good stuff.
On Tue, Feb 21, 2012 at 12:24 PM, Johan Wilfer li...@jttech.se wrote:
2012-02-21 19:20, Todd Routhier skrev:
OK, this will work and is probably a better solution than the language
idea. Although, the language idea just sounds easier and a little more
fun
Hello,
I'm trying to send a fax with sendafax aplication and receive the fax with
the receiveFax aplication on the same Asterisk Server (1.8..8.2).
All work fine but the PBX always use T30 protocol.
Is thes a variable or setting to configure Asterisk to send and receive this
fax with T38
At my wits end with this, and can't proceed any further so I'm hoping
someone has seen this and can assist. I can not get streaming
musiconhold to work with Asterisk.
My Asterisk version is 1.8.8.0 and the mpg123 version is 1.9.1, OS is
CentOS 5.7. When I call the musiconhold class (default
There is a bug in up to version 1.8.9 with external moh sources and dahdi timers
-Original Message-
From: Stephen Brown stephen.brow...@gmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Tue, 21 Feb 2012 15:34:19
To: Asterisk Users Mailing List - Non-Commercial
On Tue, Feb 21, 2012 at 2:34 PM, Stephen Brown stephen.brow...@gmail.comwrote:
At my wits end with this, and can't proceed any further so I'm hoping
someone has seen this and can assist. I can not get streaming musiconhold
to work with Asterisk.
My Asterisk version is 1.8.8.0 and the mpg123
Ok I now have the basics for dynamic parking working but for some reason
when a caller calls in and is parked with a transfer the return call dials
the sip peer of the caller and not hte peer of the last party that parked
the call. Anyone have any ideas on this? Also when a call is transfered
On 02/21/2012 02:55 PM, Bryant Zimmerman wrote:
Ok I now have the basics for dynamic parking working but for some reason when
a
caller calls in and is parked with a transfer the return call dials the sip
peer
of the caller and not hte peer of the last party that parked the call. Anyone
Is it just me, or is doing a blind transfer to a parking lot not such a
great idea? If I'm a receptionist, I'm going to want to know the lot number
to tell somebody to pick up the call?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Jason
Thank you for the response. It looks as if I am running up against this bug
as well. I was also using park more like park and announce and that was
giving me issues. I will watch this bug report and make some modifications
to my park scenarios.
Thanks
Bryant
On Feb 14, 2012, at 17:13 , isr...@gmail.com wrote:
On the snom too
Create a conferance and then press the transfer button. That will join the
parties and release the receptionist
Hmm...You can do that with just hitting the transfer button, or is there more?
I'm using a Snom 870 with
Danny
I see our point, but we are trying to transfer to a know spot using (BLF)
My issue is that it does not appear to work as it keeps looping the call
back to the callers extension. I think I may have figured out a way arround
that but it will take some more testing. To be sure.
Thanks
How much time are you giving to pick up the lot? I think the default is
like 30 seconds.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant
Zimmerman
Sent: Tuesday, February 21, 2012 4:29 PM
To: Asterisk Users Mailing List -
Danny
We are setting our default to 3 min, but we will allow our users to adjust
their setting to what they want for their lot. From our perspective the
best time really depends on use case. Say you needed to park a callers for
a shop floor. That park time would need to be greater than a
On 2/21/2012 3:38 PM, isr...@gmail.com wrote:
There is a bug in up to version 1.8.9 with external moh sources and dahdi
timers
Do you have a link to the bug report? I was unable to find anything but
it's possible I'm not looking hard enough ;)
Share with us your musiconhold.conf
that bug is running since the start of 1.8 and has been fixed in 1.8.9
https://issues.asterisk.org/jira/browse/ASTERISK-17474
i know it says that after the first time asterisks starts it works but
thats true only if the moh was loaded before the timing
its a long story but the fix is finally in
DAHDI it is are there any known workarounds? I use the FreePBX
distro and they are a bit behind, so no telling when they will update.
On 2/21/2012 6:45 PM, Israel Gottlieb wrote:
that bug is running since the start of 1.8 and has been fixed in 1.8.9
You could preload the res_moh (don't remember the full name) but that will only
help until the next reload which is the next time you'll click the orange bar
Or use a different timer which could get you into other problems
Maybe some else has a other idea
-Original Message-
From:
On 02/17/2012 03:28 AM, Frank Church wrote:
Should a Linksys Sipura 2102 be configured with nat=yes even if it is on
the local network?
I have been having some troubles with a Linksys Sipura 2100 series,
which suffers from NO AUDIO after a few calls.. Because it is on the
same subnet as
On 02/21/2012 05:34 PM, Stephen Brown wrote:
application=/usr/bin/mpg123 -q -s -f 8192 --mono -r 8000
/var/lib/asterisk/sounds/music/Rolling In The Deep.mp3
Probably unrelated to your issue, but you're going to want to quote that
filename.
--
You do need to wait until FreePBX updates the Asterisk.
Use yum to install the modules you need.
-Vladimir
On 2/21/2012 7:04 PM, Stephen Brown wrote:
DAHDI it is are there any known workarounds? I use the FreePBX
distro and they are a bit behind, so no telling when they will update.
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