On 10/23/2014 05:00 PM, Matthew Jordan wrote:
On Thu, Oct 23, 2014 at 3:32 PM, Dave Fullerton
dfullertaster...@shorelinecontainer.com
mailto:dfullertaster...@shorelinecontainer.com wrote:
Hello all,
I'm setting up a couple of test boxes and I'm running into a
problem. What I
On 10/22/2014 03:55 PM, Tim Nelson wrote:
- Original Message -
Greetings-
Working with the T.38 gateway functionality that is sparsely
documented [1], I'm attempting to get the following functional:
Asterisk calling system - Asterisk system in T.38 Gateway Mode (box
in question)
Hello all,
I'm setting up a couple of test boxes and I'm running into a problem.
What I need help with is determining whether I'm going something wrong
or if I need to post a bug report. I have two asterisk 13.0-beta 3
machines set up with extensions connected to each as such:
3700
On 07/17/2014 09:46 AM, Dave Fullerton wrote:
Hello all,
I'm running into an issue with Asterisk 12.4 and IMAP voicemail. I
have asterisk set up to connect to my Dovecot IMAP server and I can
leave and retrieve messages from my inbox and old messages. However, I
am unable to move messages
Hello all,
I'm running into an issue with Asterisk 12.4 and IMAP voicemail. I
have asterisk set up to connect to my Dovecot IMAP server and I can
leave and retrieve messages from my inbox and old messages. However, I
am unable to move messages between folders. I get a message from
asterisk
On 06/27/2013 10:37 AM, Andrew Latham wrote:
On Thu, Jun 27, 2013 at 10:34 AM, A J Stiles
asterisk_l...@earthshod.co.uk wrote:
On Thursday 27 June 2013, Eric Cooper wrote:
I'd like to protect my expensive Digium FXO cards from spikes on my
three incoming PSTN lines. Does anyone have any
Not sure how I should officially report this, but I'm getting a compile
error with DAHDI-linux 2.7 when I define CONFIG_DAHDI_NET in
include/dahdi/dahdi_config.h. I am able to compile successfully when I
leave it undefined, but I need to be able to use the network support.
snipped
On 06/10/2013 11:53 AM, Shaun Ruffell wrote:
On Mon, Jun 10, 2013 at 11:33:16AM -0400, Dave Fullerton wrote:
Not sure how I should officially report this...
You should feel free to open issues at http://issues.asterisk.org.
but I'm getting a compile error with DAHDI-linux 2.7 when I define
On 05/04/2013 08:43 PM, Mike Diehl wrote:
Hi all.
I just installed bunch of IP450's and everything went well and my
customer is happy except that they are unable to transfer calls to
other extenstions.
They can dial them directly just fine.
However, when the user is in a call and presses
Daniel,
The bootom is not part of the SIP application that you downloaded.
You need to download the appropriate bootrom from the link Kevin
supplied. Before you do any more though, you really need to download the
SoundpointIP Admin guide here:
On 12/06/2012 04:09 PM, Tim Nelson wrote:
I have a site with Polycom handsets on all the desks, mostly IP650s, some
IP550s, and some IP450s as well.
I need to update the firmware on the IP450s. However, the firmware simply won't
load.
The latest firmware (4.0.3 Rev F) supports all phones at
On 07/26/2012 04:28 PM, Tim Nelson wrote:
Greetings-
I've got a handful of Polycom IP 550 handsets connected to an Asterisk 1.8.12.0
system. Everything is running smoothly with few problems. However, I have an
issue that maybe someone could shed light on...
Many of the phones have 'buddy
efforts aren`t wasted.
Mike
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Dave Fullerton
Sent: Monday, February 13, 2012 9:39 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Polycom firmware
On 02/10/2012 05:30 PM, Mike wrote:
Hi,
I just moved many Polycom phones from firmware v3 to 4.0.1b. Anto-Answer
simply stopped functioning. I can downgrade and make it work, upgrading
kills it again. There obviously is a difference in how the newer
firmware is treating this auto answer sip
-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dave Fullerton
Sent: Thursday, November 17, 2011 11:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] polycom soundpint ip650 question
No, this is all done
On 07/29/2011 06:56 AM, --[ UxBoD ]-- wrote:
Hi,
compiling up a new installation of Asterisk 1.8.5 on CentOS 6 X86_64 and
am seeing the following when running the make:
/usr/bin/ld: skipping incompatible /usr/lib/libpam.so when searching for
-lpam
/usr/bin/ld: skipping incompatible
Actually, I don't think that has been the case for quite a while. Anyone
can get the latest firmware directly from polycom. Including, 3.3.1F
http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html
On 02/24/2011 03:32 PM, Mike wrote:
Sorry, I realize my tone might not go down
On 08/03/2010 10:48 AM, Joel Maslak wrote:
I've been replacing an old Toshiba DK switch with an Asterisk solution. I'm
needing a solution for fax machines that works as well as a POTS line from
my carrier. If the POTS line is the solution, I'll keep it, but I'd rather
move away from that.
On 07/08/2010 10:19 AM, Zeeshan Zakaria wrote:
That's why I specifically mentioned Cat5 networks, because giga bit networks
which use four pairs are called Cat6 networks.
This is true that Cat5 networks are also used with gigabit hardware, but
technically it is wrong. Cat6 hardware uses
On 06/15/2010 12:48 PM, Samantha wrote:
Hey Guys
I have hylafax working about 95%
The problem is I have a DID for fax 0742244224
When I receive a fax I see in the log file
n 16 02:44:19] VERBOSE[3679] logger.c: -- Executing
[0742244...@from-sip-external:1] NoOp(SIP/5060-0a2f7308,
On 06/07/2010 01:27 PM, Michelle Dupuis wrote:
I'm succesfully using IAXmodem for faxing (using hylafax) with Asterisk. I
would like a little more control for outbound calls using IAXmodem, but I'm
not sure how to do it. It looks like dialing out over IAXmodem bypasses the
dialplan
On 03/24/2010 03:56 PM, Miguel Molina wrote:
Gergo Csibra escribió:
Hello Asterisk,
This is only a test, because I can't start new thread in this list...
If you can send an email, you can start a new thread on this list.
What's the point of all this?
He was probably having the same
Jeff LaCoursiere wrote:
I had a customer ask me about time/date information being sent to his
analog (attached to a Linksys SPA2102) answering machine. I didn't know
that POTS could carry this information. Is this something Asterisk could
send over SIP?
Cheers,
j
Time and date
Jeff LaCoursiere wrote:
On Thu, 4 Mar 2010, Dave Fullerton wrote:
Jeff LaCoursiere wrote:
I had a customer ask me about time/date information being sent to his
analog (attached to a Linksys SPA2102) answering machine. I didn't know
that POTS could carry this information. Is this something
Sascha Ferley wrote:
Hi,
I am in need of ordering a new server here for our asterisk solution. Since
the corporate standard is Dell we need to stick to a dell server. We used to
deploy 2900III without any issues, however now they are almost not available
any more and are looking at a new
That's a bit misleading. Yes calls that travel over a PRI will be using
ulaw, but only over the PRI leg of the call. The SIP leg can still be
using G.729 with asterisk transcoding between the two legs.
Ben, You haven't shown us the contents of your sip.conf file for the
peers you are working
__
Roland Schorr Tower
www.rolandschorr.com
b...@rolandschorr.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Dave Fullerton
Sent: Tuesday, December 15, 2009 11:05 AM
To: Asterisk Users Mailing
Jon Moore wrote:
On Tue, Nov 10, 2009 at 11:43 AM, Doug Lytle supp...@drdos.info wrote:
Jon Moore wrote:
I changed the line in extensions.conf to be SET(CALLERID(ALL)=Corner
Homecare2703653903)
To match what it appears I'm getting from ATT, only the 10 digit number.
We've got ATT out of the
To my knowledge DAHDI does not use libpri, only asterisk.
In my experience you can upgrade libpri and restart asterisk, just like
you did, to make the upgrade take effect. As to what the proper thing
to do is, it's probably better to recompile asterisk after upgrading libpri.
-Dave
Karl Fife
Jason Parker wrote:
Doug Lytle wrote:
Dave Fullerton wrote:
Note num and not number I don't know if that was a change from 1.4
to 1.6 or if Doug mistyped it.
Not a mistype. I've been using number all along, but looking at the
docs shows that I've been incorrect. It must concatenate
I didn't read every post in that thread, but I don't think that's what
he's asking about.
I think what he wants (and I would like too) is something like iaxmodem
that instead of connecting to hylafax you connect to pppd or minicom or
the like. I'd love to be able to provide one or two channels
Olivier wrote:
Hi,
Most (if not all) IP phones support provisioning through DHCP/TFTP.
The trouble is some phones seem to require to store their config files in
TFTP root directory.
This makes this TFTP root directory a bit messy.
What are the best practices or tricks to manage this TFTP
Joseph wrote:
Does anybody know here I can find old Sipura firmware 2.0.13 for SPA-3000
I have Cisco 3.1.20 but it is not working as it suppose to.
http://www.totek.ca/index.php?option=com_contenttask=viewid=151Itemid=39
___
-- Bandwidth and
Michelle Dupuis wrote:
I actually see the TOS setting in iax.conf, but the default (commented out)
is EF - which doesn't even match a valid bit combination according to
voip-info wiki
If this is the right place, what TOS value are people using succesfully over
an ADSL connection?
be DSCP
compatible...or I need to do more reading :)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dave Fullerton
Sent: Thursday, October 01, 2009 3:01 PM
To: Asterisk Users List
Subject: Re: [asterisk-users
Giedrius Augys wrote:
Hello,
I have one problem and I need to disable dtmf (disable rfc2833, info and
inband) on one (other peers must support dtmf) SIP peer . Is it possible?
Workaround would be use g729 codec with dtmfmode=inband.
Maybe there is better solution?
Thanks for help.
Mike wrote:
Hi,
I have been trying a (really simple) push application for the Polycom
microbrowser, using a Polycom 650 with 3.2 firmware.
I can't do anything, I always get Push message cannot be displayed back
from the Polycom phone, and all I am sending is the Polycom example
communicating with the phone via telnet to debug the problem?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dave Fullerton
Sent: Thursday, September 24, 2009 1:15 PM
To: Asterisk Users Mailing List - Non
Jason Baker wrote:
Echo Cancellation: 128 taps unless TDM bridged, currently ON
The currently ON is telling you that the echo canceller is active.
You could try changing echotraining to no in chan_dahdi.conf as well.
What were you running before you upgraded?
So, Asterisk doesn't
Jason Baker wrote:
I recently upgraded my Asterisk system to Dahdi and now I have an echo
problem.
I am running Asterisk 1.4.25 with Dahdi Complete 2.2.0, on a Digium
TE121B PCI express card with a HARDWARE echo cancellation module. All
this is housed on a CentOS 5.5 box, 2.6.18 Kernel. My
C F wrote:
We have setup asterisk to handle our calls before between telco and an
Avaya definity. The PRI keeps locking up every so often.
In addition I keep getting this error when trying to call the avaya:
-- Channel 0/2, span 1 got hangup request, cause 102
-- Hungup 'Zap/2-1'
C F wrote:
I need to add a second T1 to an asterisk system. However the first
card is in a PCI-e slot, and the only available slot is a PCI card.
Could that work?
TIA
Technically, you should be able to run two cards in two different type
slots with no problem. You will double the number of
Tzafrir Cohen wrote:
On Tue, Aug 18, 2009 at 10:00:55AM -0400, Dave Fullerton wrote:
Here's my $0.02. If you don't want an echo canceller, specify
echocanceller=none,x-y and have dahdi_cfg print a warning (at any
verbosity level) when an echo canceller is not specified for a channel
Jeff LaCoursiere wrote:
On Wed, 19 Aug 2009, Dave Fullerton wrote:
Tzafrir Cohen wrote:
On Tue, Aug 18, 2009 at 10:00:55AM -0400, Dave Fullerton wrote:
Here's my $0.02. If you don't want an echo canceller, specify
echocanceller=none,x-y and have dahdi_cfg print a warning (at any
verbosity
Kevin P. Fleming wrote:
Dave Fullerton wrote:
It is true that this method would require more configuration work and
that it would probably throw people off who were used to the old method.
However, I don't agree that it leaves more room for error. The current
system, IMHO, has a certain
Kevin P. Fleming wrote:
Dave Fullerton wrote:
The dahdi_scan tool will tell you whether hardware echocans are present
or not, among other methods.
I tried that, but I didn't see anything that specified whether the echo
canceller was present. Here's the output, can you tell me what I
Kevin P. Fleming wrote:
Jeff LaCoursiere wrote:
On Tue, 18 Aug 2009, Kevin P. Fleming wrote:
[snip]
Note: It is *mandatory* to configure an echo canceler for the
system's channels using dahdi_cfg unless the interface cards in use
have echo canceler modules available and enabled.
Shashi Dookhee wrote:
Hi all,
I'd like to setup a really lean Asterisk installation that essentially has a
full ISDN PRI (ATT, T1, 23 B-chans, 1 D-chan, BZ8S, 5ESS, National dialplan)
on a Digium TE207P adapter that all it does is convert the ISDN channels to
SIP/IAX channels. Then I
Tzafrir Cohen wrote:
On Thu, Jul 09, 2009 at 05:31:18PM -0400, Steve Totaro wrote:
You have a small typo:
exten = _.,1,Dial(Zap,g1,${EXTEN})
exten = _.,2,Dial(SIP,Provider,${EXTEN})
exten = _.,1,Dial(Zap/g1/${EXTEN})
exten = _.,2,Dial(SIP/Provider/${EXTEN})
('/' instead of ',')
Steve Totaro wrote:
On Fri, Jul 10, 2009 at 10:45 AM, Dave Fullerton
dfullertaster...@shorelinecontainer.com wrote:
Tzafrir Cohen wrote:
On Thu, Jul 09, 2009 at 05:31:18PM -0400, Steve Totaro wrote:
You have a small typo:
exten = _.,1,Dial(Zap,g1,${EXTEN})
exten = _.,2,Dial(SIP
Steve Totaro wrote:
On Fri, Jul 10, 2009 at 11:40 AM, Dave Fullerton
dfullertaster...@shorelinecontainer.com wrote:
Steve Totaro wrote:
On Fri, Jul 10, 2009 at 10:45 AM, Dave Fullerton
dfullertaster...@shorelinecontainer.com wrote:
Tzafrir Cohen wrote:
On Thu, Jul 09, 2009 at 05:31
Kevin P. Fleming wrote:
Hose wrote:
I have a feeling that the issue is between transcoding of ulaw to g.722
and it's too loud during the transcoding - anyway to adjust the levels?
There was a flaw in Asterisk's G.722 transcoder module that was fixed
recently (on May 15, 2009), so any
Olivier wrote:
Hi,
I'm wondering how I could notify to a dumb analog phone that a voicemail
message is waiting.
My goal would be to change the tone that is heard just before user starts to
dial.
Any idea on that ?
Yea, it's called stutter dial tone. For DAHDI channels just specify the
jonas kellens wrote:
On Fri, 2009-07-03 at 11:58 +0100, Mike wrote:
tempest:~# lspci
00:00.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN
interface
I don't think this is you TDM-card...
This is mine :
04:05.0 Ethernet controller: Digium, Inc. TDM400P (rev 11)
Mike wrote:
Folks,
I have a Xen Asterisk VM with a TDM400 card. When I try to run
dahdi_cfg, I get:
tempest:~# dahdi_cfg -vvv
DAHDI Tools Version - 2.2.0
DAHDI Version: 2.2.0
Echo Canceller(s):
Configuration
==
Channel map:
Channel 01: FXO Kewlstart
- Original Message -
From: Dave Fullerton dfullertaster...@shorelinecontainer.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, June 23, 2009 8:39 AM
Subject: Re: [asterisk-users] Minimizing downtime during updates
Karl Fife wrote:
I was about to ask this question when I figured out the answer by combing
through the makefile.
I am posting this anyway because I think it's good to know, and I didn't
find any threads that speak to it when I searched the list history.
My Question was:
When updating
John A. Sullivan III wrote:
Hello, all. I've stumbled across what seems to be a traffic
prioritization issue in a Linux environment and wonder if anyone else
has encountered or addressed this issue.
We had planned to use expedited forwarding for our RTP and perhaps our
SIP packets. Our
James A. Shigley wrote:
snip
The odd thing is that I can send the call down one of my other PRI ports
to our Amtelco Infinity system. (via exten=
9819213,1,Dial(${inf}/409${EXTEN}). I've tried everything I can think of
and googled for a good while trying to find an explanation for got
Jim Gottlieb wrote:
I'm evaluating using Polycom phones for our call center and I've set
up my first phone (a SoundPoint 560) to give it a try.
The phone is working and can successfully place and receive calls.
But every minute, there's an error in the log file:
chan_sip.c:
Asterisk Development Team wrote:
The Asterisk Development Team is pleased to announce the first release
candidate of Asterisk 1.4.26. Asterisk 1.4.26-rc1 is available for
immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
This release is primarily a fix for an issue
Danny Nicholas wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dave Fullerton
Sent: Wednesday, May 20, 2009 4:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk
Danny Nicholas wrote:
Hi Listers,
I'm running 1.4.25-rc1 on opensuse 11.0 with
dahdi-linux-2.1.0.3, dahdi-tools-2.1.0.2, libpri-1.4.7 and snapdsp.0.0.2.
Incoming calls work fine. Outgoing calls made directly (exten =
s,1,Dial(DAHDI/G1) then number work fine. The problem I
Daniel Bareiro wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Tzafrir.
El miércoles 20 de mayo del 2009 a las 10:00:46 -0300,
Tzafrir Cohen escribió:
On Wed, May 20, 2009 at 07:03:15AM -0300, Daniel Bareiro wrote:
Hint: you don't need to set 'signalling' for analog
Cary Fitch wrote:
Can any one suggest a little code to either ring a cell phone when a new VM
message is recorded, or send a text message?
Basically outside sales people want to know they have a new message, but
don't want to be interrupted to take a forwarded call.
While a message by
Hello list,
I wondered if those of you using Polycom phones could recommend a decent
cord detangler. I've had quite a few handsets get the tabs broken off in
the jack from cord detanglers due to the recessed nature of the jack.
This seems like it would work but I wanted some opinions before I
Jeff LaCoursiere wrote:
Hmm, let me rephrase that (now that I have googled a bit). I am having
trouble with DTMF tones over two IAX trunks:
Polycom501---ast---[IAX]---ast2---[IAX]---provider
Both IAX trunks were ulaw, and that worked fine. I recently changed the
first leg to be g729
Marshall Henderson wrote:
Hi fellow Asterisk users!
I've got a PRI being used with a bunch of iaxmodems/Hylafax. I
currently have each individual channel of the PRI in its own context
that rings a specific iaxmodem. However, when a fax is complete on
that modem and another call comes into
Marco Sambo wrote:
Mhmm. Thaht's strange!
modinfo oslec
--
modinfo: could not find module oslec
and
modinfo dahdi_echocan_oslec
--
filename: /lib/modules/2.6.26-1-486/dahdi/dahdi_echocan_oslec.ko
license:GPL
author: Tzafrir Cohen
to and
you'll get the echo module with DADHI. It requires you download 2.6.28
but not that you are running 2.6.28.
2009/4/1 Marco Sambo derwid...@gmail.com
But I don't have also echo
modinfo echo
modinfo: could not find module echo
2009/4/1 Dave Fullerton dfullertaster
Hey All,
ATT is installing a PRI in a couple weeks and while I've been doing
homework on PRI's for the last few weeks there's something I'm still
confused about. After being asked how many digits I wanted them to send
us (10) was how many digits will you outpulse to us, 4, 7 or 10? I asked
Gordon Henderson wrote:
Is there a way to set/clear a BLF LED on a phone from the dialplan?
I want to use one as an indicator of some state in the PBX - in this case
it's night mode but I can think of other applications.
I have BLFs working just fine for normal stuff, just wonderin if I
Hello all.
I have Asterisk connected to an Adit 600 channel bank with a TE110P and
the channel bank is connected to a PBX providing dialtone to the PBX
with fxo_ks signalling. When a call between the PBX and Asterisk
completes there is a momentary battery drop/reversal or something that
Rosa De Santis wrote:
Hello all.
Please, I'd like to know if somebody can help me with this problem.
I have successfully configured a PBX with Asterisk 1.4 and a Digium analog
card with 4 ports.
This PBX has a lot of incoming and outgoing calls, and works perfect in
general, but there
Aqua Man wrote:
after installing asterisk 1.4.23.1 and dahdi-linux-2.1.0.4 and at CLI
module load chan_dahdi.so receive the following:
signalling must be specified before any channels are.
CLI Warning [4663]: chan_dahdi.c:11627 process_dahdi: Ignoring signalling
Hello All,
I'm trying to connect Asterisk to an Executone phone system with an
analog DID card and I'm hoping someone can help me figure out what I'm
doing wrong. The Executone DID card provides battery to the telco, when
the telco wishes to dial a DID it goes off-hook, waits for a wink from
Dave Fullerton wrote:
Hello All,
I'm trying to connect Asterisk to an Executone phone system with an
analog DID card and I'm hoping someone can help me figure out what I'm
doing wrong. The Executone DID card provides battery to the telco, when
the telco wishes to dial a DID it goes off
Jeff LaCoursiere wrote:
On Fri, 20 Feb 2009, Danny Nicholas wrote:
This is just a hack, but why don't you schedule a sip notify
polycom-restart during lunch hour? You could run it from a cron job
using this line for each phone:
Asterisk -rx sip notify polycom-check-cfg 100 replacing 100
Hello all,
I have two machines I'm connecting with TDMOE (dahdi dynamic spans) and
I have a question about timing parameters. By my understanding one
machine should be the source of the timing and the other a slave of that
timing.
So on machine A I have the following in system.conf:
Lee Wilson wrote:
Hey Everyone,
I would like to start testing/playing with PRI channels but I don't have
access to a PRI line. Is it possible to do the equivilent of a crossover
between two PRI Cards (say Digium's TE120P)?
What I was thinking is that I could set one asterisk box up
Alan Lord (News) wrote:
Hi all,
I built my first asterisk using the traditional (?) .conf files and
constructs.
I recall reading books at the time about AEL but it seemed new and
untested so I left it alone. Now, I'm interested to poll the audience
here to see if I should look into
Tzafrir Cohen wrote:
On Wed, Jan 21, 2009 at 06:35:58PM -0600, troxlinux wrote:
Hi list, I install dahdi-linux successfully with the module of oslec
for the echo, but when I specify it in the system.conf the echo
canceller oslec it shows me errors:
DAHDI_ATTACH_ECHOCAN failed on channel 4:
Vincent Li wrote:
On Thu, 22 Jan 2009, troxlinux wrote:
I have dahdi-linux-2.1.0.3 in centos 5.2 and the last version oslec svn
I have installed oslec and loaded, but it doesn't work me with dahdi
modinfo oslec
filename: /lib/modules/2.6.18-92.1.22.el5/kernel/net/ipv4/oslec.ko
Jeremy Mann wrote:
Looking for two things:
1. Anyone that has dialplan logic for an executive assistant. My owners
want their extensions to ring on her phone, and be very obvious to her which
extension is ringing. They also want her to have presense. She's got
Polycom IP 650
Daniel Varella wrote:
Hi everybody,
Happy New Year !
I'm trying to detect if a call was answered by a machine (linke
voicemail systems) or a human.
I would like to use AMD (Answering Machine Detect) command, but
with my configuration it was not possible get there.
Follow
Vincent wrote:
Hello
I'm contemplating building an Asterisk voice server out of the compact
Asus EeeBox:
http://www.asus.com/products.aspx?l1=24l2=165
But they're so compact, they don't have a PCI slot to handle an analog
phone line. I'd like to minimize footpring and cables: Besides
Lee, John (Sydney) wrote:
Calling all Polycom gurus:
I am using Polycom IP601 phones with Asterisk 1.4.21.2
In all Polycom phones, I set the following in sip.cfg.
dialplan dialplan.impossibleMatchHandling=2
/dialplan
(I leave the digitmap unchanged because I thought setting
Philipp Kempgen wrote:
Philipp Kempgen schrieb:
Klaus Darilion schrieb:
I use an if condition in extensions.ael to check if a channel variable
is defined and if defined I add a certain header:
context toNormaleRufe {
_X. = {
if (${NUMBER}) {
Brent Davidson wrote:
Philipp Kempgen wrote:
Brent Davidson schrieb:
macro outside-dial ( num ) {
if (${DB_EXISTS(Office/${CALLERID(num)})}) {
TRUNK=Zap/r2;
} else {
TRUNK=Zap/r1;
}
Dial(${TRUNK}/${num},,Ttok);
}
[Dec 23 12:16:22] WARNING[2994]:
Mr. James W. Laferriere wrote:
Hello Tzafir ,
On Thu, 18 Dec 2008, Tzafrir Cohen wrote:
On Thu, Dec 18, 2008 at 12:48:59PM -0500, Matt Watson wrote:
after you have configured zaptel manually the first time, copy the
menuselect.makeopts file that is generated in the root directory of
Christian wrote:
Hi all,
I am trying to isntall the v1.6 version of Asterisk on my Intrepid
system, but I get an error after I have typed make:
[CC] manager.c - manager.o
manager.c: In function ‘action_getvar’:
manager.c:1732: error: ‘SENTINEL’ undeclared (first use in this function)
Brent Davidson wrote:
I have several branch offices all running Asterisk PBX's that register
to each other via SIP so that calls can be transferred from office to
office. Everything is working great on the office to office transfers,
but I'd like to somehow make the CallerID more useful.
Brent Davidson wrote:
Dave Fullerton wrote:
Check the entries for office1 and office2 servers in sip.conf. If they
have a callerid= entry comment it out and do a SIP reload. When it is
set asterisk overrides the caller ID sent to it.
-Dave
There aren't any callerid= entries in any of my
Brent Davidson wrote:
Dave Fullerton wrote:
Brent Davidson wrote:
Dave Fullerton wrote:
Check the entries for office1 and office2 servers in sip.conf. If
they have a callerid= entry comment it out and do a SIP reload. When
it is set asterisk overrides the caller ID sent
Mike wrote:
Hi,
Say I wanted to know what context a SIP registration is using to dial out in
my dialplan, what would I do?
For example, I have phones on a local-calls-only context (as defined in
sip.conf), others in unrestricted-calls. In my dialplan, I`d like to act
on that knowledge.
Mike wrote:
Hi,
Is there a way to page a Polycom phone that is already in use (if, of
course, the call isn't on speakerphone already)?
I've never been able to find a way. Any attempt I made either put the
existing call on hold to auto-answer the page or the page just rang at
the
Is anyone else having difficulty compiling 1.6.0.2?
It bombs out when compiling manager.c
manager.c: In function 'action_getvar':
manager.c:1732: error: 'SENTINEL' undeclared (first use in this function)
manager.c:1732: error: (Each undeclared identifier is reported only once
manager.c:1732:
Tzafrir Cohen wrote:
On Tue, Dec 02, 2008 at 01:22:16PM -0500, Dave Fullerton wrote:
Is anyone else having difficulty compiling 1.6.0.2?
It bombs out when compiling manager.c
manager.c: In function 'action_getvar':
manager.c:1732: error: 'SENTINEL' undeclared (first use in this function
Olivier wrote:
Hello,
Groups in asterisk are summarized here (
http://www.voip-info.org/wiki/view/Channels+and+Groups).
Is there any difference between DAHDI/G1/0123456789 and DAHDI/g1/0123456789
(as I've been advised in another thread, to switch from one notation to the
other and I can't
There are also settings which will turn on local echo cancellation for
the handset, headset and/or speaker phone. I don't recall their names at
the moment. They are off by default on the handset and headset unless
you're using a very recent (3.0+) SIP app.
Tim Nelson wrote:
I'm not sure about
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