Assuming this is outbound, no problem. For inbound, I don't think so
either. Can you be a little more specific?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Muhammad Ali
Sent: Friday, March 23, 2012 4:30 AM
To:
suggestions?
Thanks in advance
Danny Nicholas
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http
-Commercial Discussion
Subject: Re: [asterisk-users] Problem installing asterisk 10.1.3 on SUSE
Danny Nicholas wrote:
libstdc++.so: file not recognized: File format not recognized
Google shows:
Very likely you try to link a 64-bit executable with a 32-bit library or
vice versa.
http
Of Patrick Lists
Sent: Thursday, March 22, 2012 11:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Problem installing asterisk 10.1.3 on SUSE
On 22-03-12 16:47, Danny Nicholas wrote:
So is Asterisk 10 supposed to be 32 or 64 bit? P.S. a pox on sqlite3
-boun...@lists.digium.com] On Behalf Of A J Stiles
Sent: Thursday, March 22, 2012 12:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Problem installing asterisk 10.1.3 on SUSE
On Thursday 22 March 2012, Danny Nicholas wrote:
Confusion? I'm looking
Dumb question - you did module parkandannounce reload?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant
Zimmerman
Sent: Monday, March 19, 2012 8:25 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Remove
Which Technology are you using for the call (DAHDI/SIP/other)?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty.cruz
Sent: Monday, March 19, 2012 10:42 AM
To: asterisk-users@lists.digium.com
Subject:
is Polycom soundpoint ip 450.
Rining continues in the background after successfully answered, is a ramdom
not constantly.
Thanks,
-Motty
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Monday
AFAIK the Asterisk GUI still does this. If not, it is a simple thing to do
in PERL, C, etc.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad
Sent: Monday, March 19, 2012 4:35 PM
To:
I've had pretty good experience with VoicePulse.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jake Wicke
Sent: Thursday, March 15, 2012 10:46 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Reliable SIP Trunk
According to the specifications, it should connect with little difficulty.
http://www.voipsupply.com/polycom-cx3000
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant
Zimmerman
Sent: Wednesday, March 14, 2012 12:32 PM
To:
Depending on your Asterisk version, add hideconnect = yes to asterisk.conf
and restart.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Munir
Sent: Wednesday, March 14, 2012 2:13 PM
To: asterisk-users@lists.digium.com
Subject:
#1 you might need a progress() statement after answer
#2 what does sip show peer xxx look like on this peer?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
Sent: Tuesday, March 13, 2012 4:18 PM
To:
Ping the phones, then run arp.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
resea...@businesstz.com
Sent: Tuesday, March 13, 2012 4:52 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Getting
To the best of my knowledge, your best options, not necessarily in order
are:
1. Vestec ASR
2. Lumenvox ASR
3. google ASR (there was a good post in February about how to use this)
4. Sphynx ASR
Options 1 and 2 are/were recommended by Digium.
-Original Message-
From:
AFAIK, this is a shell count (The count is kept in shell memory for the
running asterisk process). You handicap potential answer by not stating
your Asterisk version or your technology (SIP/DAHDI/T1/etc). If you are
just using SIP trunks, SIP RELOAD might do it.
From:
AFAIK, it works in the 1.8 and 10.X branches (I have used it in 10.0.2)
There was a known issue with some certificates that used multiple levels
IIRC.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gavin Henry
Nothing against fail2ban but in this case I think the route drop solution
is more appropriate.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of equis software
Sent: Wednesday, March 07, 2012 9:52 AM
To: Asterisk Users Mailing List -
I think (since I opened this particular can-o-worms) that it depends on your
bootrom/sip level.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P.
Fleming
Sent: Monday, March 05, 2012 10:26 AM
To:
I would suggest VoicePulse. They seem to have a wide presense.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Royce Souther
Sent: Monday, March 05, 2012 10:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
That depends on too many things to answer in a short reply, but if you do it
the right way, yes.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Varella
Sent: Monday, March 05, 2012 10:59 AM
To: Asterisk
Since you are using AMI, I would assume you are using one of the AMI
interfaces from CPAN or somewhere. If this is the case you could do
something like this:
my $astman = new Asterisk::Manager;
$astman-user('mickey');
$astman-secret('mouse');
my
This one is simple. Open /usr/sbin/safe_asterisk and put # in first
character of line 86 and 102. Or modfy /etc/sudoers to allow your sudo to
execute ulimit.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Munir
Sent:
Asterisk Now should serve your needs nicely.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Moffett
Sent: Wednesday, February 29, 2012 11:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Danny,
I would like to know do I need to worry about this message? And why I'm
getting this ulimit message? Please provide reason briefly
From: Danny Nicholas da...@debsinc.com
Subject: Re: [asterisk-users] Getting Ulimit Message after restart
asteriskservice
To: 'Asterisk Users
I would say that this is correct
http://support.freepbx.org/forum/freepbx/general-help/freepbx-vs-asterisknow
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Moffett
Sent: Wednesday, February 29, 2012
My first two guesses are that encryption is hosing you or that the
single-channel nature of IAX2 may have something to do with it. IAX2
talks on 1 channel, SIP uses twisted pair connotation on two channels
(as I understand it).
-Original Message-
From:
Maybe his google-fu is phisher-fu!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez
Sent: Monday, February 27, 2012 12:53 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] CDR
That would be AMI
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alper Tekinalp
Sent: Friday, February 24, 2012 6:32 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] View Uniqueid of Active Calls
Hi.
As title says
Why not use AMD (Answering Machine Detect)?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
Sent: Friday, February 24, 2012 3:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
+1 Dale - p.s. the grep -P '\d\d\d\d' killed the output on my 1.4 box.
P.P.S if you change grep -P (Useragent|Contact) to grep -P
(Username|Contact|Username) it produces a nice 4 line report like this:
Def. Username: Danny Nicholas
Useragent: PolycomSoundPointIP-SPIP_501-UA/3.1.2.0392
My short google-fu session says no - It's not universal for Polycom either
(the phone has to support microbrowser).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik
Sent: Thursday, February 23, 2012
You don't state the Asterisk version you are running, but personal
experience tells me you'd better invest in some Rogaine if you're depending
on the built-in stuff from AGI for DTMF input. I have personally wasted
weeks trying it.
From: asterisk-users-boun...@lists.digium.com
22, 2012 9:14 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] AGI: blocking script until playback complete
On 22/2/12 2:55 pm, Danny Nicholas wrote:
You don't state the Asterisk version you are running, but personal
experience tells me you'd better invest in some Rogaine
I think it's a warning as opposed to a bug. If the call were happening
all in Tecnology (SIP/DAHDI/etc), the warning would be because your
channel didn't support the codec (I can't do alaw so I'm gonna talk in
slin). The LOCAL channel by definition (AFAIK) doesn't specifically
support/deny any
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Wednesday, February 22, 2012 2:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] codec mismatch
While it is never actually safe to assume anything about Asterisk, the
general setting for seconds is milliseconds.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Wednesday, February 22,
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
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
am playing the sound file I need before sending them to VoiceMailMain but
then Comedian Mail! plays right after of course.
--Todd
On Tue, Feb 21, 2012 at 10:59 AM, Danny Nicholas da...@debsinc.com wrote:
I believe this is what you want. Instead of this
Exten = _X.,123,Voicemail(100
be a real country/language code and
not something made up.
--Todd
On Tue, Feb 21, 2012 at 11:37 AM, Danny Nicholas da...@debsinc.com wrote:
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
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
arround that
but it will take some more testing. To be sure.
Thanks
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
_
From: Danny Nicholas da...@debsinc.com
Sent: Tuesday, February 21, 2012 4:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
Did you set CHANNEL(reversecharge) somewhere?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rafael dos Santos
Saraiva
Sent: Friday, February 17, 2012 10:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Santos
Saraiva
Sent: Friday, February 17, 2012 11:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Block Collect Calls on ISDN trunk
This is a variable received from the isdn channel.
Att,
Rafael Saraiva
2012/2/17 Danny Nicholas da
/5132083300-4'
rssr305*CLI -- Hungup 'DAHDI/i1/5132083300-4'
-- Hungup 'DAHDI/i1/5132083300-4'
rssr305*CLI
Att,
Rafael Saraiva
2012/2/17 Danny Nicholas da...@debsinc.com
I would put a Verbose statement after Proceeding to verify the value returned
from ISDN channel, like
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Block Collect Calls on ISDN trunk
Which version do you recommend? Mine is 1.4.12.
Att,
Rafael Saraiva
2012/2/17 Danny Nicholas da...@debsinc.com
From what I read, your libpri may be out of date.
From
4520 is for DUNDI. Obviously your install uses H323 in some flavor.
Mgcp-callagent is for jitterbuffering? And sieve and complex-main I have no
clue (perhaps H323 tag-alongs)
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas
You could register the agent to a SIP extension with followme. When the
queue went to ring the SIP extension, followme would send the call on to the
mobile/land line.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
As I read this, this is a regular attended transfer.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Phil Frost
Sent: Tuesday, February 14, 2012 2:33 PM
To: Asterisk Users Mailing List - Non-Commercial
I think you can do the same thing with most Polycom phones.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andres
Sent: Tuesday, February 14, 2012 4:11 PM
To: Asterisk Users Mailing List - Non-Commercial
Did the 4.0.1b update overwrite sip.ld on these phones? If I recall
correctly you have to tweak that file to make auto-answer work correctly.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brian ipt
Sent: Friday, February 10, 2012
If the MOH thing is really true, a more realistic test would be to run
playback(demo-instruct). Since I know that I will eventually cross this
bridge in real life/real time, I devised this test on my Asterisk 10.0 box
Dialplan (in default context)
exten = 3366,1,answer()
exten =
Not a complaint, per se, just a question. Why are the LTS versions odd
(11, 13, 15, etc) and the non-LTS (10, 12, etc) even? As I read the chart,
Digium/Asterisk is committing to a new LTS version every 2 years?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Only trust the wiki if it explicitly refers to your current version (and
then you should still test it).
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez
Sent: Tuesday, February 07, 2012 10:46 AM
To: Asterisk Users
Just my .02 - I don't think DUNDI will be deprecated any time soon (they
tend to warn you about that at least 6 months out). I think it is in
Extended because it isn't used in a majority of installs.
From: asterisk-users-boun...@lists.digium.com
Perhaps someone with too much time on their hands could parse the CHANGES
files and make a nice spreadsheet/PDF that puts all of the changes into a
chart form?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
You are mis-understanding the concept - the noanswer option is playing the
file as you requested, but since you aren't answering the call, no channel
is established to actually present the sound to you.
From: asterisk-users-boun...@lists.digium.com
Local/queue/?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro
Cabrera Obed
Sent: Monday, February 06, 2012 1:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
-users] Custom extension: dial a queue
No, Local/queue/ don't work at all :(
2012/2/6, Danny Nicholas da...@debsinc.com:
Local/queue/?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Alejandro Cabrera
Install binutils-devel - this includes libbfd.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant
Zimmerman
Sent: Monday, February 06, 2012 11:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Assuming your longtime task is in an AGI, you could do this:
AGI
Print STDOUT SET MUSIC ON HOLD DEFAULT\n
.. agi logic ..
.. end of agi logic ..
Print STDOUT SET MUSIC ON HOLD OFF\n
-Original Message-
From: asterisk-users-boun...@lists.digium.com
I don’t think that’s possible in the Asterisk framework. As I understand
it, the follow-me logic would transfer the call outside of Asterisk control.
If this is not a correct assumption, you could try MixMonitor on these
calls.
From: asterisk-users-boun...@lists.digium.com
Use the Group() function to track queue usage and availability.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nathan Pryor
Sent: Friday, February 03, 2012 11:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
What happens if you do a SIP RELOAD instead of restarting Asterisk?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Sent: Thursday, February 02, 2012 1:14 PM
To: asterisk-users@lists.digium.com
Maybe you should change your values in udptl.conf? By default the range is
4000 to 4099, but is effectively 4001 to 4099 because the protocol doesn't
use even numbers by default, so it runs out of entries in 500 tries.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Agreed - I think the solution is a patch to udptl.c to reset the counter
instead of dying with this message (just my opinion).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Sent: Thursday, February
Alternately, you could use a SIP channel with followme or the newer releases
have some Bluetooth capabilities.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Tuesday, January 31, 2012 8:46 AM
Why the short life on Asterisk 10?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P.
Fleming
Sent: Tuesday, January 31, 2012 11:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P.
Fleming
Sent: Tuesday, January 31, 2012 5:13 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Proposed changes to Asterisk release and
support cycles
On 01/31/2012 04:07 PM, Danny Nicholas wrote:
Why the short
Subject: Re: [asterisk-users] Proposed changes to Asterisk release and
support cycles
On 01/31/2012 05:14 PM, Danny Nicholas wrote:
1.8 and 11 forward all seem to have a timeline of around 5 years. 10
only runs for two. Since the code is available that isn't a biggie to
me, but the appearance
Try changing qualify=yes to qualify=90 on your TCP peers.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant
Zimmerman
Sent: Friday, January 27, 2012 9:45 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] sip
Did you install sqlite3-devel?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
Sent: Friday, January 27, 2012 3:26 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] upgraded
Just a WAG, but I'm guessing they may have a limited number of lines and
don't want one phone hogging 2-3 at a time.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Lists
Sent: Thursday, January 26,
AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ChanSpy : how to know channel name ?
This could work, yes.
But the context is not always the same.
Also ${CHANNELS(miq8) will return nothing...
Jonas.
On 01/24/2012 08:47 PM, Danny Nicholas wrote
You don’t state which version this is for, but it seems like a simple patch
would be for voicemail to play sorry-mailbox-full.wav (standard sound). In
lieu of all that, you could do a quick-and-dirty AGI to read /v/l/a/m and play
the message back since voicemail is one of the larger modules
You don’t state which version this is for, but it seems like a simple patch
would be for voicemail to play sorry-mailbox-full.wav (standard sound). In
lieu of all that, you could do a quick-and-dirty AGI to read /v/l/a/m and play
the message back since voicemail is one of the larger modules
Strip off the -x. Just listen to SIP/miq8 and SIP/375382280 in your
example.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, January 24, 2012 7:47 AM
To: Asterisk Users Mailing List - Non-Commercial
AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ChanSpy : how to know channel name ?
Hello,
OK thanks. But, I want to listen to the conversation (not just 1 channel out
of 2 channels). How then do I use ChanSpy ?
On 01/24/2012 03:41 PM, Danny
I would personally rather use a stand-alone daemon to query the mailboxes and
send an email to the box owner when he or she reaches a tolerance level rather
than depend on an overloaded application that is running God-only-knows what
modifications to the original intent (IMAP, Real-Time, Active
the channel name ?
On 01/24/2012 03:53 PM, Danny Nicholas wrote:
I would try chanspy(sip/miq8,b) - the b flag denotes to only listen to a
bridged call which (it seems to me) should pick up both sides.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun
the channel name so I can ChanSpy
the correct channel ?
On 01/24/2012 04:13 PM, Danny Nicholas wrote:
It's not random. The Channel Name is Tech/peer-sequence (sequence is in
hex). You can control (to a degree) the peer portion in
sip.conf/users.conf.
From: asterisk-users-boun...@lists.digium.com
SIP/375382280-2 default
SIP/miq8-2419 sub-uitGSM
SIP/3749378004- default
SIP/instlpr0-2 sub-uitinternation
Can you tell me what is the extension ? How will I know the context ? The
context is not always the same...
On 01/24/2012 04:32 PM, Danny Nicholas wrote:
You
Question 1 - no
The format is this
#define FORMAT2 %-25.25s %-39.39s %-3.3s %-10.10s %-3.3s %-8s %-11s
%-32.32s %s\n
Question 2
debsphone2*CLI core show channels concise
SIP/1104-051b!default!99!2!Up!Playback!tt-monkeys!1104!!!3!3!(None)!1327
35.1307
debsphone2*CLI core show
For the most part, if it worked in 1.8.5 it should work in 1.8.8.1 unless
specifically noted in changes.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Monday, January 23, 2012 2:03 PM
To:
You want your program to live in /usr/local/bin. /etc/init.d is where the
bash scripts that run programs that live elsewhere are housed. It is not a
good practice to put executeables there. For example, /etc/init.d/asterisk
runs /usr/sbin/safe-asterisk. The scenario I typically use is that
What version of Asterisk are you trying to implement this in?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mahesh katta
Sent: Tuesday, January 17, 2012 1:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
You aren't opening the line in the 123 call. In the 200 call, the
Answer() opens the output audio channel. In the 123 call you are plunging
into the SayDigits() function without opening the channel. Some functions
will generate their own Answer() if not present, others will not.
From:
the same problem, but how do I
change this?
On Mon, Jan 16, 2012 at 4:26 PM, Danny Nicholas da...@debsinc.com wrote:
You aren't opening the line in the 123 call. In the 200 call, the
Answer() opens the output audio channel. In the 123 call you are plunging
into the SayDigits() function without opening
A should transfer C to a local channel that plays the IVR then returns the
call to A.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mahesh katta
Sent: Monday, January 16, 2012 9:56 AM
To: Asterisk Users Mailing List - Non-Commercial
configure the between call oneside
IVR
I was tried it but its not going.. with same
Best Regards,
Mahesh Katta
On Mon, Jan 16, 2012 at 9:32 PM, Danny Nicholas da...@debsinc.com wrote:
A should transfer C to a local channel that plays the IVR then returns the
call to A.
From: asterisk-users
Two more offerings - #1 - add DTMF parameter so function can be stopped by
pressing a digit or digits other than * or # - #2 - add an option to
silence the beep. If you were using this in an IVR and wanted to say
press 1 or say help for help, silencing the beep before recording would
(IMO) make
What about this
http://support.avaya.com/css/P8/documents/100102120
or this?
http://www.ingate.com/files/Solving_Firewall-NAT_Traversal.pdf
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthias Apitz
Sent:
You could use a parking lot instead of attended transfer?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez
Sent: Tuesday, January 10, 2012 11:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Why don't you just use vovida-linux from sourceforge?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant
Zimmerman
Sent: Tuesday, January 10, 2012 3:12 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Linux Stun
O.P. doesn't state his Asterisk version, but in 10.0(beta) I had a similar
problem where sqlite3 couldn't create the new Asterisk DB. From what I read
in the archives, we really could use a guru to thoroughly pound these DB
statements to make them a bit more bullet-proof.
-Original
raw -r 44100 jan01.sln44
Not a biggie if no, since this is a decent work-around.
Thanks
Danny Nicholas
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
-users] 44Khz files in Asterisk 10
On Mon, 9 Jan 2012 13:59:07 -0600
Danny Nicholas da...@debsinc.com wrote:
Hi gang,
I'm thrilled to be able to use a better quality sound
in Asterisk 10, but have to change my wav files to sln44 to get the
benefit. Is there some conf
: Danny Nicholas da...@debsinc.com
Subject: Re: [asterisk-users] asterisk - AGI (perl) - sqlplus
(oracle)
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Message-ID: 00ca01cccb06$911e8300$b35b8900$@debsinc.com
Content-Type: text/plain
I found this on another post and cleaned it up - might help
#!/usr/local/bin/perl
use strict;
use IO::Socket;
my $target = shift; #192.168.0.255;
my $target_port = 4569;
socket(PING, PF_INET, SOCK_DGRAM, getprotobyname(udp));
# Build Packet ...
# Names from ethereal filter of registration
301 - 400 of 2418 matches
Mail list logo
