-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Wednesday, August 20, 2014 9:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dispatching calls
I doubt PBX settings (other than CallerID) would break calling to only one
specific carrier.
Have you tried pri show span X?
From one of our boxes:
pbx*CLI pri show span 1
Primary D-channel: 24
Status: Up, Active
Switchtype: National ISDN
Type: CPE
Remote type: Unknown node type
Overlap Dial:
with the end user in this early media situation?
Thanks in advance.
rv
2014-08-07 20:02 GMT-04:00 Eric Wieling
ewiel...@nyigc.commailto:ewiel...@nyigc.com:
Generally the only thing you are allowed to do before answer is send audio.
You can’t receive audio and can’t receive DTMF. I assume
Generally the only thing you are allowed to do before answer is send audio.
You can’t receive audio and can’t receive DTMF. I assume it is to prevent
people from doing exactly what you are trying to do --- trying to have two way
communications without paying for the call.
From:
From sip.conf.sample in 11.10.0
;use_q850_reason = no ; Default no
; Set to yes add Reason header and use Reason header if
it is available.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Lam
Sent:
Making LinkedID available in the dialplan would also be useful.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rafael dos Santos
Saraiva
Sent: Tuesday, July 22, 2014 1:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
1:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call Identifier Logging
On Tue, Jul 22, 2014 at 12:45 PM, Eric Wieling
ewiel...@nyigc.commailto:ewiel...@nyigc.com wrote:
Making LinkedID available in the dialplan would also be useful.
LinkedID
Where is this documented?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steven Wheeler
Sent: Tuesday, July 22, 2014 2:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call Identifier
Depends on the carrier. Verizon Wireless appears to activly block SIP.
G729 codec is needed on 3G and is a good idea on 4G. I use TLS and SRTP to
work around carrier stupidity. I also use a non-standard port for TLS. It
mostly works much of the time. Don’t get BRIA, every time your
Does your Supermicro system have the Intel Card of Sorrow, aka Intel 82574L?
If so, see:
http://www.zdnet.com/intel-ethernet-controller-vulnerable-to-packet-of-death-710984/
http://blog.krisk.org/2013/02/packets-of-death-update.html
-Original Message-
From:
setting cause 17?
-Justin
From:
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Wednesday, July 09, 2014 4:38 PM
To: Asterisk Users Mailing List
If you use Playtones you should put an Answer and a Wait(1) before the Playtones
I recommend using the Hangup app instead. Busy would be Hangup(17).
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen
Sent: Wednesday, July
Generally if you want to send a cause 17 to the caller you would use Hangup(17)
and let the caller's switch generate the busy tone.
If the dialplan has already answered the call, then you might want to use Busy
or Playtones.
From: asterisk-users-boun...@lists.digium.com
If you are executing database put Agora modele/IVR/AstreinteNagios/1
${ASTR_State} while in the Asterisk CLI, that won't work. You cannot access
DIALPLAN variables from the CLI.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
I think you will find that direct audio between two endpoints does not work
when NAT is involved.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sameer Rathod
Sent: Tuesday, July 08, 2014 11:18 AM
To: Asterisk Users Mailing List -
Hi Eric,
I am behind nat
Is there any solution for the same.
My goal is to deduct the balance
for the call but free my asterisk server from audio packet load.
On Tue, Jul 8, 2014 at 7:51 PM, Eric Wieling
ewiel...@nyigc.commailto:ewiel...@nyigc.com wrote:
I think you will find that direct audio
Once set, settings apply to all following channel = lines until the setting is
changed.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ethy H. Brito
Sent: Tuesday, July 08, 2014 1:30 PM
To:
This is a common issue and is covered in the mailing list archives multiple
times.
Do a Google search for something like:
site:lists.digium.com fail2ban
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steven Howes
Sent: Friday,
It depends on your carrier.With some carriers, such as Verizon SIP, you do
this using P-Asserted-Identity. With Verizon SIP, if they can’t figure out how
to bill the call, it will be rejected.
With Level 3 SIP, you can use From: or PAID but if the number you present to
them is not on your
You need to talk to your carrier.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Positively
Optimistic
Sent: Thursday, June 26, 2014 10:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
Something like memcachedb is also an option.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gregory Malsack
Sent: Wednesday, June 25, 2014 5:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
There are two common types of echo.
Accoustic Echo: This is caused by microphone picking up audio from the
speaker. This echo cannot generally be removed by echo cancelers. The
solution to accoustic echo is to prevent the microphone from picking up audio
from the speaker (or handset or
A is a valid DTMF digit, chances are your PBX is detecting the digit wrong.
If you have relaxdtmf enabled, disable it. If that doesn't help, play with
the audio gains. Too loud or too soft can cause DTMF issues.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Generally in the UPGRADE.txt file which came in the tarball. A pretty version
is here https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+12
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Mudgett
Sent:
Using Set(MASTER_CHANNEL(CDR(remoteUid))=foo); might do what you want
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mikael Fredin
Sent: Tuesday, June 10, 2014 11:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
line so it is more specific
than Re: Contents of asterisk-users digest...
Today's Topics:
1. Shorten time between DTMF (CDR)
2. Re: Shorten time between DTMF (Eric Wieling)
--
Message: 1
Date: Fri, 6 Jun 2014 13:04
Which EXACT parameter did you change in asterisk.conf?
Changing DTMF duration for DAHDI is done in chan_dahdi.conf.
SIP DTMF duration and inter-digit duration is generally set on the phone.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
How many g729 Licenses do you have?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matteo Campana
Sent: Wednesday, June 04, 2014 10:48 AM
To: asterisk-users
Subject: [asterisk-users] Renegotiate SIP audio codec after call is up
Hi
Have you tried RetryDial()?
---
Documentation for Polycom phones can be found at http://help.nyigc.net/
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of jg
[webaccounts...@jgoettgens.de]
Sent: Tuesday,
The reason I suggested that option is because in Q.931-land Display Name can be
sent in the call setup or as a facility message immediately after the call
setup. I don't know about Q.SIG-land.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Have you tried playing with the facilityenable setting in chan_dahdi.conf?
chan_dahdi.conf.sample should have some info on that option.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Armen K
Sent: Friday,
Inband audio, such as ringing, busy, intercept, silence, etc require
transcoding. We solved the issue on our Asterisk installs by purchasing a
hardware transcoding card (for G729, but the card supports several codecs in
hardware. In my experience transcoding happens, accept it and move
Try the card in another machine with a different brand of motherboard. If it
works you know it is a hardware issue.
Do you have an actual T-1 plugged into your card? If not, try that and see if
there is any difference.
-Original Message-
From:
If the attacks are direct (rather than through Asterisk) and you have a Polycom
phone, check around page 522 of the firmware 4.0 admin guide.
If the attacks are directed at your Asterisk then you should use fail2ban to
dynamically block attackers.
If the attacks are coming to your phone via
Not that I'm aware of.
SIPAddHeader won't help you. Asterisk only sends the extra headers when you
use the Dial app.
You'll need to install a SIP Proxy in front of Asterisk if you want to
manipulate the SIP headers.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
I believe Polycom phones support Multicast for paging and intercom without any
Asterisk involvement. Check the Admin Guide.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Josh Metzger
Sent: Thursday, May
Most FXS ATAs do not support supervision so they don't work well when plugged
into a PBX's analog FXO (aka CO) ports.
If the Mitel can provide supervision on analog phone ports (i.e. FSX) then you
could use an ATA with FXO ports. If the Mitel does not support supervision on
analog phone
In my experience DNS issues will cause Asterisk to take a long time to reload
and could stop Asterisk for working at all.
List all the IPs of the box in /etc/hosts and make sure /etc/resolv.conf points
to a working nameserver. See if that helps at all.
-Original Message-
From:
Does the script generate an error when run outside of Asterisk? An AGI should
simply wait for input when run outside of Asterisk.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A
Sent:
/US_email_disclosure. Edward D. Jones Co., L.P. d/b/a
Edward Jones, 12555 Manchester Road, St. Louis, MO 63131 © Edward Jones. All
rights reserved.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Doesn't MixMonitor use sox to combine the incoming and outgoing recordings?
If so, I'd expect MixMonitor to add MORE delay, not less.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Josh Metzger
Sent:
I would be very surprised is anyone uses WatchGuard SIP ALG. For the past 12
years the advice has always been Disable SIP ALG and let Asterisk do the NAT
fixup itself on any firewall, regardless of brand.I wish you the best of
luck.
-Original Message-
From:
I had little problem converting my AEL scripts from 1.4 to 11
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
wanting to upgrade ...
On 17/4/14 4:53 pm, Eric Wieling wrote:
I had little problem converting my AEL scripts from 1.4 to 11
Did they have lots of macros in them?
If so, then you, sir, are a better man than I, and I take my hat off to you :-)
(and any hints you might want to share
All significant changes should be listed in the UPGRADE*.txt included in the
Asterisk source code.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce Ferrell
Sent: Thursday, April 17, 2014 4:15 PM
To:
So few people use Asteisk on OSX that I doubt anyone will answer.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Manu
Sent: Monday, April 14, 2014 4:13 AM
To: Asterisk Users Mailing List - Non-Commercial
This doesn't fix the issue, but a work around might be to try using file names
without the any : in them
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan White
Sent: Thursday, April 10, 2014 2:56
Here is an example from one of my production dialplans
same =
n,ExecIf(${REGEX(^1205|^1256|^1850|^1718|^1212|^1917|^1347|^1646|^1929
${CALLERID(num)})}]?Hangup)
Assuming you meant 0-9 and not the literal X (which means nothing special in
regular expressions):
same =
1) put a maximum number of loops in your IVR to terminate calls which are dead
or gone.
2) put maximum message length in voicemail.conf (ever tried to delete a 4 day
long voicemail?)
3) Call sometimes get stuck. This is life.
-Original Message-
From:
I have an iptables file which blocks all traffic except traffic from networks
allocated by ARIN or are Legacy networks. I pulled the information from
http://www.iana.org/assignments/ipv4-address-space/ipv4-address-space.xhtml
My iptables script can be found at the link below.
972 is Israel
See: http://en.wikipedia.org/wiki/List_of_country_calling_codes#Ordered_by_code
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis
Sent: Wednesday, March 26, 2014 11:05 AM
To:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chad Wallace
Sent: Wednesday, March 26, 2014 6:31 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Numbers hackers call
If this is to 972
I would not say happy, since there is no happiness in a world with T.38, but
Level 3 supports T.38.Level 3 is wholesale only as far as I know.
Vitelity has some fax service stuff too.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
H.323 is a communications protocol like SIP. H261 is a codec like ulaw or
gsm. You do not need H323 unless you are using the H323 protocol INSTEAD
of SIP.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf
In no specific order:
Download the Asterisk tarball you want to use and study all the
UPGRADE*.txt files included in it.
Buy or download the latest ATFOT book, study it.
Install Asterisk into a test box, even a VM is OK for testing, study the
output of core show applications and
This is an excerpt from a script I use for post processing received faxes.
You need the PHP process extension, on CentOS that is the php-process package.
end of code which interacts with asterisk
declare(ticks=1);
// become a daemon so we don't tie up asterisk resources while we process
Often it is P-Asserted-ID, but depends on the carrier. You should be asking
your carrier how to do this. Be careful, if the carrier doesn't like your CID
spoofing they might bill the call to a default number on the account.
I suspect it is the destination which is rejecting the call because
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Thursday, March 13, 2014 1:39 PM
To: rwhee...@artifact-software.com; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Brummell
Sent: Thursday, March 13, 2014 2:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
rwhee...@artifact-software.com
Subject: Re:
Try setting the sip.conf entry to friend, not peer and not user.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Boykin
Sent: Tuesday, March 11, 2014 10:34 AM
To: Asterisk Users Mailing List -
can help me. Attach updated logs: http://pastebin.com/HmKEDAUq
Thanks,
On Thu, Dec 19, 2013 at 8:48 AM, Eric Wieling ewiel...@nyigc.com wrote:
See sip.conf.sample in the Asterisk tarball for documentation of valid
settings.
-Original Message-
From
Because sometimes marketing overcomes technical correctness.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Rechberger
Sent: Monday, March 10, 2014 7:39 PM
To: asterisk-users@lists.digium.com
Asterisk transcodes at many other points. Inband ringing, audio mixing for
conferences, beep tones. It is naive to think you can passthrough g729 and
never transcode without spending significant amounts of time tracking down each
instance Asterisk would have to transcode.
Over the
Why would you use anything other than Digium's fully licensed and fully
compatable with Asterisk modules?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jayson Devor
Sent: Friday, February 28, 2014 4:04 PM
For 23 channels I recommend a hardware transcoding card.
We use http://www.sangoma.com/products/d100-30-400-sessions/ I think Digium
also has a transcoding card also.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
In my experience when you run out of g729 licenses additional calls will fail.
Simple as that. Make sure you run out of licenses.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tony Mountifield
Sent:
To be fair NAT is rewriting your SIP packet source port. This happens all day,
on almost every NAT device out there.Stop thinking it is purely a port
rewriting issue, something else is going on.
Have you set localnet and externip in sip.conf. Maybe the NAT device has a
short UDP
A reboot of the system after hours appears to have solved the issue.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jg
Sent: Wednesday, February 19, 2014 11:15 AM
To: Asterisk Users Mailing List -
I can't imagine it working any other way.
Either your phones are on static IP addresses or they must register to inform
Asterisk the IP associated with the peer entry in sip.conf.Unless you have
chan_psychic.so Asterisk won't know the IP of the phones unless you tell it.
-Original
Attach the packet capture to your Jira bug report or post it online somewhere.
Hopefully someone will look at it.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Tuesday, February 18, 2014
No. Asterisk will accept calls from unregistered devices, but you have to
enable guests I sip.conf and hope your dialplan is secure. No sane person does
this.
Asterisk cannot send calls to a device unless it knows the address from a
register or from a host= entry for the peer.
You may
PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Host = Dynamic in a Register Free Setup
On 2/18/2014 2:09 PM, Eric Wieling wrote:
No. Asterisk will accept calls from unregistered devices, but you have
to enable guests I sip.conf and hope your
We are seeing the message PRI Error on span 1: Received MDL/TEI management
message, but configured for mode other than PTMP on one of our Asterisk boxes
on a PRI. A Google search turns up a number of hits for this error, but they
are all for BRI not PRI.
I'm reasonably sure there are no
I was not aware Wireshark worked on PRI spans. What interface should I tell it
to watch? 8-|
Card is: wanpipe: AFT-A101-SH PCI T1/E1 card found (HDLC (DS) rev.37), cpu(s) 1
chan_dahdi.conf has a timestamp of Jun 25 of last year
# cat /etc/asterisk/chan_dahdi.conf
;autogenerated by
!
[2014-02-13 14:40:06] WARNING[2932] chan_dahdi.c: PRI Error on span 1: Received
MDL/TEI managemement message, but configured for mode other than PTMP!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric
Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PRI Error on span 1: Received MDL/TEI management
message, but configured for mode other than PTMP
Hi,
http://wiki.sangoma.com/wanpipe-wireshark-pcap-pri-bri-wan-t1-e1-tracing
On 18 February 2014 21:34, Eric Wieling ewiel
My recent experiences with Sangoma tech support have been less than good. I
admit the issue was rather wieid. We had an issue about a year ago where
Sangoma PCI cards would not work in the servers we were purchasing unless you
used a PCI/PCI-X converter/riser card. Everything would load
Asterisk is a B2BUA -- think of it as two calls, one inbound call from your
switch to Asterisk and one for outbound call from Asterisk to the destination.
Using SIPAddHeader or similar is the proper way to copy headers from the
inbound call to the outbound call in Asterisk.
-Original
How about: SipAddHeader(${SIP_HEADER(P-Asserted-Identity)})
Might have some issues with the ; character being see as start of comment.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Cameo
Sent: Monday,
-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Crash in asterisk 10.0.0.
Hi Eric Wieling,
Thanks for your reply. what is the reason for that crash?? . when i read the
core dump i found something like signal 11.
what it means because of signal 11 asterisk crashed . Before upgrading i
Upgrade to 11.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gopalakrishnan N
Sent: Friday, February 14, 2014 12:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
We use extensions.conf, AEL, and AGI scripts.Debugging AEL scripts can
beinteresting, but worth it. I also like being able to program in a real
language
Our extensions.conf handles the incoming call initially, an AGI is then run
which talks to the database and does the heavy lifting.
Are you absolutely sure you need to use the outboundproxy setting rather than
using a peer/friend?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Lemberger
Sent: Tuesday, January 21, 2014 7:53 PM
To:
Make sure you do NOT have any *bindaddr options set in your sip.conf. If you
do, you are telling Asterisk to not allow the OS to pick the source IP and
hence the routing.
The *bindaddr options are seldom useful.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
It is far worse when you have multiple phones behind the same public address
(i.e. NAT).If any one of the phones has a bad password and the IP gets
blocked by fail2ban, then all phones at that site would be blocked.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Original message
From: Eric Wieling
Date:19/01/2014 8:40 PM (GMT+02:00)
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] stopping unwanted attempts
It is far worse when you have multiple phones behind the same public address
(i.e. NAT).If any
-Original Message-
progressinband=never
Setting progressinband=never when you wantprogress (ringing) to be sent as
audio (inband) has always confused me.
Can anyone shed some light on that?
--
_
-- Bandwidth and
Generally exten h is not run when the callee hangs up. See also the g option
to Dial.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
richard.seg...@marisec.ca
Sent: Tuesday, January 14, 2014 10:27 AM
To:
Ask your carrier to test the circuit. Often HDLC errors, especially with
modern cards, are caused by a dirty T-1 not a PBX or card issue.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Serafini
Sent:
See https://wiki.asterisk.org/wiki/display/AST/Asterisk+Security+Framework
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Al lists
Sent: Wednesday, January 08, 2014 10:11 PM
To: Asterisk Users Mailing List
Sometimes you need to disable B-channel restarts, see chan_dahdi.conf.sample
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mitesh Thakkar
Sent: Tuesday, January 07, 2014 2:35 PM
To:
This is a classic symptom of having reinvites and/or direct media enabled on
Asterisk or SIP ALG enabled on the router.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Cameo
Sent: Monday, January 06,
Which firmware version? 4.1.x is only for use with MS Link server. A symptom
of running 4.1.x firmware with a non-MS server is the phone will not show
buddies.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf
signalling=pri_cpe
callgroup=1
;pickupgroup=1
immediate=no
channel = 1-15,17-31
group=2
callgroup=2
switchtype=qsig
signalling=pri_net
callerid=5
immediate=no
channel = 32-46,48-52
2013/12/19 Eric Wieling ewiel...@nyigc.com
The basic idea is dial using your main outbound dahdi group
in attached file my dialplan
thanks and regards
2013/12/20 Eric Wieling ewiel...@nyigc.com
You must write dialplan code to do what you want. Assuming you
are not using a GUI with Asterisk, post your dialplan used for outgoing calls
Inbound call hunting is handled by your carrier, not Asterisk.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Salaheddine
Elharit
Sent: Thursday, December 19, 2013 12:52 PM
To: Asterisk Users Mailing List -
the calls from to servers
i ask about outbound calls not inbound round-robin
best regards
2013/12/19 Eric Wieling ewiel...@nyigc.com
Inbound call hunting is handled by your carrier, not Asterisk.
-Original Message-
From: asterisk-users-boun
Calls dropping after 20 seconds is often directmedia enabled when it should not
be enabled or RTP keepalives enabled when they should not be enabled. Dropping
around 20 mins is often Session Timers being enabled when they don't work for
the specific environment.
-Original Message-
ports.
On Wednesday, December 18, 2013, Eric Wieling wrote:
Calls dropping after 20 seconds is often directmedia
enabled when it should not be enabled or RTP keepalives enabled when they
should not be enabled
If the device is not registering then you have to specify the port as well as
the ip in the database entry for the peer.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik
Sent: Thursday, December
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