On Tue, 2011-02-15 at 05:57 +, A J Stiles wrote:
On Tuesday 15 Feb 2011, Jeff LaCoursiere wrote:
Now this is what I call uptime...
minipbx*CLI show uptime
System uptime: 41 years, 7 weeks, 6 days, 3 hours, 26 minutes, 46 seconds
Last reload: 8 hours, 3 minutes, 51 seconds
On 15 Feb 2011, at 03:39, Jeff LaCoursiere wrote:
minipbx*CLI show uptime
System uptime: 41 years, 7 weeks, 6 days, 3 hours, 26 minutes, 46 seconds
Last reload: 8 hours, 3 minutes, 51 seconds
What's the highest current 'genuine' one on-list?..
klein*CLI core show uptime
System uptime: 2
On Fri, Feb 11, 2011 at 04:37:49PM -0600, Danny Nicholas wrote:
In 500 words or less (if possible), please explain what is a
legal music-on-hold file?
One source of explicitly royalty-free music is the podcasting community:
http://uhort.no/ and http://www.podsafeaudio.com/ both have
On Mon, Feb 14, 2011 at 11:00 PM, Mike Diehl mdi...@diehlnet.com wrote:
Hi all,
I'm trying to get outgoing fax working from an SPA2102 to a PSTN fax machine
via a T.38 enabled trunk. I've got
t38pt_udptl = yes
faxdetect=no
in my sip.conf file. The ATA has all of the T.38 options turned
I am wondering if its possible to have sometime like this:
exten 100 = Dial (g/08039269311)
where g would be a group of SIP extensions and i would be parsing/hard coding
the PSTN numbers into it, so when i dial extension 100, it passes the call to a
group of SIP service provider extensions.
i
Anyway, the answer is: No, it's mathematically impossible to do
that. Even if the passwords were stored encrypted, Asterisk itself
has to be able to get the plaintext passwords to send to the remote
server; so the code to decrypt them must necessarily be located on
the machine. And the
really it's too difficult to understand, please explain more clear
On Tue, Feb 15, 2011 at 5:17 AM, Ricardo Carvalho
rjcarvalho.li...@gmail.com wrote:
Hi,
How can I configure my asterisk server so that I can receive incomming
calls comming from the same IP from where my server also receives
Hi,
I was trying to use SIP session timers with Asterisk 1.8 and two Snom
phones: is it possible to force SIP timers usage only on some specific
SIP channels or is it somehow a global setting?
Thank you very much,
Guido.
--
you can run any function in your hangup extension,
exten = h,1,...
best
On Tue, Feb 15, 2011 at 12:21 PM, Richard Zheng rzh...@gmail.com wrote:
Hi,
In ACD queue, is it possible for the agent to take some actions when the
caller hangs up? For example, to let the agent to enter some
please send your sip.conf, is any NAT procedure implemented in your network?
On Mon, Feb 14, 2011 at 10:16 PM, Ricardo Carvalho
rjcarvalho.li...@gmail.com wrote:
Hi,
I manage an SBC which stands between my company server farm and some SIP
telco trunks. The system works fine, for inbound
On 02/14/2011 10:00 PM, Mike Diehl wrote:
Hi all,
I'm trying to get outgoing fax working from an SPA2102 to a PSTN fax machine
via a T.38 enabled trunk. I've got
t38pt_udptl = yes
faxdetect=no
in my sip.conf file. The ATA has all of the T.38 options turned on, echo
cancellation is off, as well
this command will not work.
what is your main purpose?
do u need to have a conference with a group of sip phones?
best
On Tue, Feb 15, 2011 at 3:13 PM, ayodele abejide ayodeleabej...@hotmail.com
wrote:
I am wondering if its possible to have sometime like this:
exten 100 = Dial
In case of asterisk you simply can't accept registration from an IP which
you have mentioned as static host for IP authentication.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pezhman Lali
Sent: Tuesday, February 15, 2011 5:37 PM
On 02/15/2011 06:18 AM, Richard Kenner wrote:
Anyway, the answer is: No, it's mathematically impossible to do
that. Even if the passwords were stored encrypted, Asterisk itself
has to be able to get the plaintext passwords to send to the remote
server; so the code to decrypt them must
I know there is not a good place for ask this question. but I can not find
in other ways.
Dear,
Do you have any experience for changing the logo of cisco 7905 on sccp
firmware?
best
--
_
-- Bandwidth and Colocation Provided by
You need to use relay request in your SBC instead of forward.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pezhman Lali
Sent: Tuesday, February 15, 2011 5:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
On Tue, Feb 15, 2011 at 07:18:08AM -0500, Richard Kenner wrote:
Anyway, the answer is: No, it's mathematically impossible to do
that. Even if the passwords were stored encrypted, Asterisk itself
has to be able to get the plaintext passwords to send to the remote
server; so the code to
How does that improve things? The reason that works with Cisco routers
is because the code that reads that special key file and uses it to
decrypt the other files is closed-source; nobody can see how it works.
As another poster said, that's not true for Asterisk. If Asterisk had
such a
Right. But it really won't help much (except complicating things) if the
user has decent access to Asterisk.
Yes, but we're talking about cases where the user *doesn't* have access
to Asterisk. At many locations, including mine, Asterisk runs on a
machine dedicated for that purpose and only
Hi,
I want to trunk outbound calls through a sip provider to PSTN, and i want to
write a script to parse the PSTN numbers, so when say extension 100 is dialled
it just starts to dial PSTN numbers through the SIP provider, so it justs an
automated dialer.
Best Regards,
ABEJIDE, Ayodele A.
On Tue, Feb 15, 2011 at 07:54:54AM -0500, Richard Kenner wrote:
Right. But it really won't help much (except complicating things) if the
user has decent access to Asterisk.
Yes, but we're talking about cases where the user *doesn't* have access
to Asterisk. At many locations, including
#include the password (a file the line 'secret=') from a local file on
the file system. The user has no access to it, right?
Right, but we're not talking ONE password, but ANY password. Having
dozens of those files, one for each password, gets to be a real pain
really fast. And you STILL want
On 15 Feb 2011, at 13:17, Richard Kenner wrote:
Of course not! It would be useless if that were the case: the whole
point here would be that you need the master encryption key.
Here's a possible design:
- There's optionally a file in the config
directory called master_key. It contains
On Tue, Feb 15, 2011 at 01:06:16PM +, ayodele abejide wrote:
I want to trunk outbound calls through a sip provider to PSTN, and i
want to write a script to parse the PSTN numbers, so when say
extension 100 is dialled it just starts to dial PSTN numbers through
the SIP provider, so it justs
On Tue, Feb 15, 2011 at 08:17:20AM -0500, Richard Kenner wrote:
#include the password (a file the line 'secret=') from a local file on
the file system. The user has no access to it, right?
Right, but we're not talking ONE password, but ANY password. Having
dozens of those files, one for
Hi all,
my setup is like this:Asterisk 1.8.2.3 + FreePBX 2.8.1 running on a Debian
squeeze box connected with a SIP trunk to Vega 100 VoIP gateway, which, in
turn, is connected by ISDN PRI to a POTS PBX.
When a call is made to a SIP extension from POTS, it proceeds OK, and the
connections
Hello everyone
Is there a possibility to catch an outbound callleg ID for the follovong
scenario: some carrier - --(asterisk1) asterisk2 ?
I can get inbound callid for asterisk1 with a ${SIPCALLID} in
extensions.conf or to look it up in cdrs field (are the same). But how about
I sent a few hours earlier another e-mail to this list detailing a bit more
my problem. Please see it with, it has the following subject: unregistered
trunks and registered phones coming from the same IP
Thanks,
Ricardo.
On Tue, Feb 15, 2011 at 12:36 PM, Pezhman Lali l...@lopl.net wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roger Burton
West
Sent: Tuesday, February 15, 2011 3:09 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] On-Hold Music
On Fri, Feb 11, 2011 at
On Tue, Feb 15, 2011 at 08:39:57AM -0600, Danny Nicholas wrote:
Good suggestion, Roger, but this seems like a slippery slope path.
Today's podcaster could be tomorrows ASCAP/BMI member coming back for you?
Doesn't matter if you use music that has been explicitly released as
royalty-free (usually
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roger Burton
West
Sent: Tuesday, February 15, 2011 8:51 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] On-Hold Music
On Tue, Feb 15, 2011 at
On Tue, Feb 15, 2011 at 09:01:16AM -0600, Danny Nicholas wrote:
Thanks for the tip - got a Norwegian translator for uhort.no?
Anything wrong with
http://translate.google.com/translate?js=nprev=_thl=enie=UTF-8layout=2eotf=1sl=notl=enu=uhort.no
?
R
--
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roger Burton
West
Sent: Tuesday, February 15, 2011 9:06 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] On-Hold Music
On Tue, Feb 15, 2011 at
On Tue, Feb 15, 2011 at 09:09:25AM +, Roger Burton West wrote:
On Fri, Feb 11, 2011 at 04:37:49PM -0600, Danny Nicholas wrote:
In 500 words or less (if possible), please explain what is a
legal music-on-hold file?
One source of explicitly royalty-free music is the podcasting
At the SBC, I delegate the registration to my asterisk server forwarding the
REGISTER requests to it and resetting the contact to itself. This should
allow my asterisk server to forward subsequent messages to the SBC rather
than to the phone client directly. When the SBC receives a 200 OK from the
Hi ALL,
I have install asterisk 1.8.2.3 on my ubuntu 10.x machine with 512 MB memory.
Now i am running sipp tester to check performance but at some point in running
test my asterisk got freez its doing nothing but i can run commands on CLI, But
it doesn't accepting new request this time.
Security through obscurity does not work with open source software.
What a bold statement, are you telling me it works with closed source
software? :P
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F
Sent: Tuesday, February 15, 2011 9:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hide the plain text password
On 02/14/2011 12:04 PM, James Miller wrote:
I did the command listed, and its actually requesting RINGLIST.DAT, so I
changed the filename to match its request but now its showing in the
ring type setting:
Chirp 1
Chirp 2
24 24-ring-tone-1.raw
Att1 ring_att1.pcm
snip
Do you actually have
On 02/15/2011 09:29 AM, C F wrote:
Security through obscurity does not work with open source software.
What a bold statement, are you telling me it works with closed source
software? :P
Depends on your definition of 'works' I guess :-)
With closed source software, it takes rather longer
Hi,
After compiling a installing asterisk 1.8.2.3 I wanted to play with
lua but I noticed that extensions created in extensions.lua was not
being registered with asterisk.
uga1*CLI dialplan show
[ Context 'app_queue_gosub_virtual_context' created by 'app_queue' ]
's' =1. NoOp()
Hello,
could I adjust the Rx and Tx gains for SIP and CAPI? If it is possible, how
should I do it?
Thanks a lot.
best regards,
Felix
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong
Sent: Tuesday, February 15, 2011 11:06 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Adjusting Rx and Tx gains
Hello,
could I adjust the Rx
On Tuesday 15 February 2011 11:06:32 Carlo Pires wrote:
Hi,
After compiling a installing asterisk 1.8.2.3 I wanted to play with
lua but I noticed that extensions created in extensions.lua was not
being registered with asterisk.
uga1*CLI dialplan show
[ Context
Tried that, not possible to play a sound file and prompt users to enter a
number. Is there a way to revive the channel?
On Tue, Feb 15, 2011 at 2:38 AM, Pezhman Lali l...@lopl.net wrote:
you can run any function in your hangup extension,
exten = h,1,...
best
On Tue, Feb 15, 2011 at
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Zheng
Sent: Tuesday, February 15, 2011 11:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] further action after caller in a
On 02/15/2011 11:22 AM, Richard Zheng wrote:
Tried that, not possible to play a sound file and prompt users to enter
a number. Is there a way to revive the channel?
No, there is not. I don't know if there is a simple way to achieve what
you want to achieve with Asterisk at this point,
On Mon, Feb 14, 2011 at 10:31 AM, James Miller paramedi...@gmail.com wrote:
I did that and this is what I got when I tried to play the 24 ringtone:
13:29:16.573318 IP 192.168.1.103.50849 192.168.1.60.69: 39 RRQ Emergency
ring_emergency.pcm octet
That line should read something like:
blah..
On Tue, 15 Feb 2011, A J Stiles wrote:
On Tuesday 15 Feb 2011, Jeff LaCoursiere wrote:
Now this is what I call uptime...
minipbx*CLI show uptime
System uptime: 41 years, 7 weeks, 6 days, 3 hours, 26 minutes, 46 seconds
Last reload: 8 hours, 3 minutes, 51 seconds
Bizarre bug?
I'm
Hi,
I have been having a problem with asterisk crashing when using local channels
and realtime on asterisk 1.8.3-rc2.
The example given here is I think the easiest way to reproduce this problem.
In extensions.conf I have:
[internal]
switch = Realtime/extensions/p
exten = 301,1,Answer()
exten =
I'm scratching my head trying to work out a way of sending a
pre-recorded message as a 'Page' to a list of phones ( Oi! you muppets
you've left the server room door open! or somesuch message :-)
controlled by an external trigger.
I can do a normal page (phones auto-answer on speaker) with
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Russell Brown
Sent: Tuesday, February 15, 2011 12:40 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Paging a message. How?
I'm scratching my
Russell Brown wrote:
I can do a normal page (phones auto-answer on speaker) with SipAddHeader
but that doesn't let me play a pre-recorded message.
Any suggestions?
Dial plan:
[polycom-page]
;*
;* Play the previously recorded page file over
On 11-02-15 01:15 PM, Nic Colledge wrote:
I have been having a problem with asterisk crashing when using local channels
and realtime on asterisk 1.8.3-rc2.
The example given here is I think the easiest way to reproduce this problem.
Generate a backtrace[1] and attached to your message.
[1]
On Tuesday 15 February 2011 12:13:37 Jeff LaCoursiere wrote:
On Tue, 15 Feb 2011, A J Stiles wrote:
On Tuesday 15 Feb 2011, Jeff LaCoursiere wrote:
Now this is what I call uptime...
minipbx*CLI show uptime
System uptime: 41 years, 7 weeks, 6 days, 3 hours, 26 minutes, 46
seconds Last
I found some great pieces of script on the internet that I've combined to allow
Asterisk to send voicemails as an MP3 file, and encode the sender name and
number as well as message number as tags into the MP3 file. I even include a
cover art image which has our company logo and PBX symbol in
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent: Tuesday, February 15, 2011 1:16 PM
To: Asterisk Users List
Subject: [asterisk-users] Voicemail email attachment as MP3, with tags
containing sender name,
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent: Tuesday, February 15, 2011 1:16 PM
To: Asterisk Users List
Subject: [asterisk-users] Voicemail email attachment as
On Tue, Feb 15, 2011 at 10:15 AM, Nic Colledge n...@njcolledge.net wrote:
I have been having a problem with asterisk crashing when using local
channels and realtime on asterisk 1.8.3-rc2.
Nic,
I can reproduce this using the latest SVN for the 1.8 branch. I
don't get the console locking, but
Michelle Dupuis wrote:
If anyone wants to try it out let me know!
I'd also be interested in having a look!
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
--
Hi,
why pstack not working on asterisk ? I believe i compiled asterisk with debug
libraries.
root@ubuntu-test:/usr/local/src/asterisk-1.8.2.3# pstack `pidof asterisk`
624: /usr/sbin/asterisk
'': opening object file: No such file or directory
Could not open object file.
Thanks,
S.
Text below..
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan Thurman
Sent: 15 February 2011 19:39
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Realtime and
I have two Aastra phones, a 6730 and a 6757, both connected to Asterisk
v1.6.2.1. They can call each other's extensions (and make and receive
calls otherwise), but they cannot transfer calls, not even to outside
extensions on the PSTN. The procedure I use is to accept a call on one
phone, press
Text below..
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan Thurman
Sent: 15 February 2011 19:39
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Realtime and
- Original Message -
I found some great pieces of script on the internet that I've
combined to allow Asterisk to send voicemails as an MP3 file, and
encode the sender name and number as well as message number as tags
into the MP3 file. I even include a cover art image which has our
We have deployed about 50 Polycom IP 550's. They have done well, and were easy
to configure.
HTH,
Shane
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Latham
Sent: Saturday, February 12, 2011
Hi,
I've noticed an unusual behavior on the dialplan execution: assume this
DP:
exten = _6XXX,1,NoOp(test1)
exten = _,1,NoOp(test2)
exten = _,2,NoOp(test3)
If I call 6000 then test1 and test3 NoOps get executed, even though the
pattern is different.
I've always thought that if I
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gabriel Ortiz
Lour
Sent: Tuesday, February 15, 2011 2:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Dialplan end of pattern matching question
Gabriel Ortiz Lour wrote:
exten = _6XXX,1,NoOp(test1)
exten = _,1,NoOp(test2)
exten = _,2,NoOp(test3)
Being in the same context, it would look for the next priority that
would match. _,2 would be that priority.
I would suggest:
exten = _6XXX,1,NoOp(test1)
exten =
At 12:12 PM 2/15/2011, you wrote:
I have two Aastra phones, a 6730 and a 6757, both connected to Asterisk
v1.6.2.1. They can call each other's extensions (and make and receive
calls otherwise), but they cannot transfer calls, not even to outside
I'm running 1.6.2.16.1 and have three Aastra
On Tue, Feb 15, 2011 at 2:18 PM, --[ UxBoD ]-- ux...@splatnix.net wrote:
--
I found some great pieces of script on the internet that I've combined to
allow Asterisk to send voicemails as an MP3 file, and encode the sender name
and number as well as message number
On Tue, Feb 15, 2011 at 3:03 PM, Doug Lytle supp...@drdos.info wrote:
Gabriel Ortiz Lour wrote:
exten = _6XXX,1,NoOp(test1)
exten = _,1,NoOp(test2)
exten = _,2,NoOp(test3)
If you're going to keep them in the same context, you can change the second
pattern to exclude 4 digit
At 12:12 PM 2/15/2011, you wrote:
I have two Aastra phones, a 6730 and a 6757, both connected to Asterisk
v1.6.2.1. They can call each other's extensions (and make and receive
calls otherwise), but they cannot transfer calls, not even to outside
I'm running 1.6.2.16.1 and have three Aastra 480i
Ok - I've put the script up on the www.generationd.com web site. Just go to
the Downloads | Asterisk section to pull it down.
I would like to keep control of this script so please send me changes (don't
repost elsewhere) and I'll keep the latest version up for everyone. I'll add a
link to
On Tue, 2011-02-15 at 09:01 +, Steve Howes wrote:
On 15 Feb 2011, at 03:39, Jeff LaCoursiere wrote:
minipbx*CLI show uptime
System uptime: 41 years, 7 weeks, 6 days, 3 hours, 26 minutes, 46 seconds
Last reload: 8 hours, 3 minutes, 51 seconds
What's the highest current 'genuine' one
But when I try to call one extension created with lua I got a message
telling that extension doesnt exist on default context. Am I missing
something?
2011/2/15 Tilghman Lesher tilgh...@meg.abyt.es:
On Tuesday 15 February 2011 11:06:32 Carlo Pires wrote:
Hi,
After compiling a installing
On Tue, 2011-02-15 at 07:18 -0500, Richard Kenner wrote:
Anyway, the answer is: No, it's mathematically impossible to do
that. Even if the passwords were stored encrypted, Asterisk itself
has to be able to get the plaintext passwords to send to the remote
server; so the code to decrypt
kept on reading the thread...
Wouldn't it be better, for asterisk at least, to get rid of all this
identification / authentication stuff?
Keeping config files holding pain passwords or simple md5 isn't the way
to solve this...
Within the unix world those issues have been solved over and over
Yes, nothing changed EXCEPT for the software image the phone pulled down.
All of the files are still in the exact same locations with the exact same
names as they had in 8.9. I'm at a loss as to what's causing this issue and
so apparently is Cisco given they have yet to respond to my follow up
How about encrypt the whole hard drive?
If I built a server and give to other people, there is no easy way to
stop them reset the root password or just mount my drive to read
everything on it. But if build an encrypt OS then it will be secure. My
question here are: 1Is this against Asterisk
At 12:12 PM 2/15/2011, you wrote:
I have two Aastra phones, a 6730 and a 6757, both connected to Asterisk
v1.6.2.1. They can call each other's extensions (and make and receive
calls otherwise), but they cannot transfer calls, not even to outside
I'm running 1.6.2.16.1 and have three Aastra 480i
Problem has been resolved with the assistance of Jonathan. Appears to be an
issue with my text editors not properly tabbing the file correctly.
Regards.
I see blindness, not as a disability, but more of an ability. And Sight
actually, more of a disability because some people with sight tend to
Hi,
I encounter this problem recently after quite some months of my asterisk.
I have a SIP trunk for dial in and out.
When dial-in, it matches the callerid number and decides. If matched, it
will either go into an a very brief IVR. The IVR allows caller to dial
internal extension.
All along it
how did you increase it?
Am 16.02.2011 um 00:11 schrieb Hans Witvliet h...@a-domani.nl:
On Tue, 2011-02-15 at 18:06 +0100, Felix Dong wrote:
Hello,
could I adjust the Rx and Tx gains for SIP and CAPI? If it is
possible, how should I do it?
Thanks a lot.
best regards,
Felix
--
Check if dtmfmode is properly set on SIP trunk ask with your carrier which
dmtfmode they support.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk
asterisk
Sent: Wednesday, February 16, 2011 5:39 AM
To: Asterisk Users Mailing
You may need to share your LUA code and the extension your call is need to
execute.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlo Pires
Sent: Wednesday, February 16, 2011 3:29 AM
To: Asterisk Users
In the past it was set as auto and worked. I change to RFC2833 but did not
work.
How can I check further?
On Wed, Feb 16, 2011 at 10:30 AM, Faisal Hanif fai...@vopium.com wrote:
Check if dtmfmode is properly set on SIP trunk ask with your carrier which
dmtfmode they support.
*From:*
Ask with you SIP carrier which dtmfmode they are using on their end and use
same on asterisk side.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk
asterisk
Sent: Wednesday, February 16, 2011 8:58 AM
To: Asterisk Users Mailing
You can also append add dtmf logging to cosole and see if dtmf is coming
from carrier.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk
asterisk
Sent: Wednesday, February 16, 2011 8:58 AM
To: Asterisk Users Mailing List -
Lis ton message de Mickael avant qu'il ne soit effacé!
Pour lire ton message, suis simplement ce lien:
http://eu1.badoo.com/0199422682/in/pe-wgHsDEkQ/?lang_id=6
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Calu (Maputo, Mozambique)
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Hello,
Are there any gateways which allow me to hook a cellphone to Asterisk and
use that line for routing my calls? Basically, I'm looking to play around a
bit and if I can get to connect a cellphone with Asterisk then that would be
great.
Thanks,
Hitesh
PS: I have tried to search on the web,
How about encrypt the whole hard drive?
If I built a server and give to other people, there is no easy way to
stop them reset the root password or just mount my drive to read
everything on it. But if build an encrypt OS then it will be secure.
It will be more secure. However, you
Hi,
Your question is not clear but below are possible answers to your question,
If you want to attach you cell-phone to asterisk you can simply use
chan_mobile. Using Bluetooth with chan_mobile you can connect your
Cell-Phone as FXO and your handsfree as FXS port to asterisk.
If you
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