> On Feb 12, 2015, at 10:42 AM, Matthew Jordan wrote:
>
> On Thu, Feb 12, 2015 at 10:38 AM, David M. Lee <mailto:d...@digium.com>> wrote:
>
> Unfortunately, I doubt the Python test suite would run on non-Linux. I don’t
> even bother trying to run it on Ubuntu; I ha
n’t know
if it’s a problem in the test itself, or in Asterisk.
Unfortunately, I doubt the Python test suite would run on non-Linux. I don’t
even bother trying to run it on Ubuntu; I have a CentOS VM specifically for
running the test suite to avoid platform problems.
--
David M. Lee
Digium, In
Thorsten Göllner writes:
> Am 26.10.2014 00:43, schrieb lee:
>> Hi,
>>
>> how can I make asterisk do something when an outgoing call is picked up?
>>
>>
>> The background is that I would like to record incoming and outgoing
>> phone calls. In ord
Hi,
how can I make asterisk do something when an outgoing call is picked up?
The background is that I would like to record incoming and outgoing
phone calls. In order to do this, I need to play an announcement
telling the person calling or being called that the call will be
recorded.
Here's wh
Meyerriecks
Sent: Saturday, 31 May 2014 3:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Flashing Red Lights in TE420 (5th Gen) Card
On Fri, May 30, 2014 at 4:07 AM, Lee, John (Sydney)
wrote:
> Even without plugging in the ISDN into span 1, al
I have the following software installed in a Centos Box with a TE420 (5th Gen)
card.
. Centos 6.5 64-bit
. Asterisk 1.4.22
. dahdi-linux-complete-2.9.1.1+2.9.1
. libpri-1.4.14.
Even without plugging in the ISDN into span 1, all 4 spans are flashing red.
Plugging an E1 into span 1 makes no differen
Hi,
I have noticed it for a while but I just thought about confirming this with the
Asterisk community.
As the compilation of DAHDI will need to reference Kernel-devel, does it mean
that after DAHDI is installed, we should not yum update kernel because it will
affect the operation of DAHDI?
Than
wanting to upgrade ...
2014-04-15 10:37, Lee, John (Sydney) skrev:
> Hello,
> I have been running Asterisk for the past 5+ years on RedHat and I never
> upgraded it before.
> All my Asterisk software is of the following release:
> 1) Asterisk 1.4.21.2
> 2) Libpri-1.4.4
> 3) Za
about upgrading Asterisk releases.
However, I am bewildered by the myriad of different releases like 1.6, 1.8,
10.x, 11.x, 12.x, 13.x
Can someone please give me some advice as to what release I should upgrade?
Or should I just stick to 1.4.x and just upgrade DAHDI?
Thanks.
Regards,
John Lee
The
//wiki.asterisk.org/wiki/x/1gKfAQ for more details.
>
>
> Looking that up, it says add to asterisk.conf
> [options]
> live_dangerously = yes
>
> After doing this, and stopping and starting I
> still get the message.
>
> Whats up?
You want to avoid danger, so set li
r the request is simply taking that long
to process. Packet loss could cause delays in getting responses, but
usually not for the lengths of times you're talking about.
I know it's not a lot of info, but hopefully you can turn up some
logging or packet captures to narrow down what's
re there are other problems such as line audio quality issues or lots
of non-fax numbers being used accidentally. So your use of SIP (VoIP)
for what should be a lossless data channel is probably a factor there.
Thanks,
Lee.
--
__
o get things the way you want them.
[1]: http://linux.die.net/man/2/open
> Ludovic BOUÉ
--
David M. Lee
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org
--
___
On Nov 29, 2012, at 11:18 AM, Ron Wheeler wrote:
> That is a good answer.
> Thanks.
> Any reason why it is not documented?
It's documented on the Asterisk wiki:
https://wiki.asterisk.org/wiki/display/AST/Contexts,+Extensions,+and+Priorities
> Ron
--
David M. Lee
Digium
: (Set) Options:
> (CDR(userfield)="|usr_r=vieri")
> [Nov 29 10:54:57] WARNING[4838]: pbx.c:1563 pbx_exec: The application
> delimiter is now the comma, not the pipe. Did you forget to convert your
> dialplan? (Set(CDR(userfield)="|usr_r=vieri"
You might want to look at Asterisk-Java, Java library to communicate to
Asterisk via AMI and AGI, https://github.com/srt/asterisk-java
Best Regards,
Rudi Lee
-Original Message-
From: Nweike Onwuyali
Sender: asterisk-users-boun...@lists.digium.com
Date: Thu, 22 Nov 2012 19:45:36
To
ice applications. I believe some exist
for PHP, but I know nothing about them.
> Thanks!
>
> -Ken
Good luck!
--
David M. Lee
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check
Flowroute, which is working awesome, for VOIP calls.
I only have a SIP Phone at home and two Printer/Scanner/Fax Printers.
So on your MFP you'll scan it instead of using the system's "fax"
capability, and then fax it th
ng to take a fax machine out to the parking lot and shoot
it, even talking about this awful waste of time makes my blood boil.
Well, if you were using stand-alone fax machines then that was part of
your problem.
Thanks,
Lee.
--
: https://groups.google.com/forum/?fromgroups=#!forum/adhearsion
--
David M. Lee
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org
--
_
On Sep 10, 2012, at 2:38 AM, Johan Wilfer wrote:
> Thank you David for the feedback.
>
> I reported the following bugs:
>
> https://issues.asterisk.org/jira/browse/ASTERISK-20397 (all bug)
> https://issues.asterisk.org/jira/browse/ASTERISK-20396 (cropped col)
Thanks!
--
D
e?
This is at least looks correct. The 'all' permission is a superset of, well,
all the permissions. The 'write=all' line in manager.conf assigns all of these
permissions to the user.
> Thanks!
>
> --
> Johan Wilfer
--
David M. Lee
Digium, Inc. | Software Developer
you mind being a bit more specific on the Asterisk changes to
which you refer and how they should be implemented in the configuration?
Thanks,
Lee.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Ast
I had done some research but nothing
solid.
Don't try to send faxes over LAN/WAN-strung VoIP channels.
If you want to get rid of your analog lines, that's fine; use an on-line
fax service provider.
Thanks,
Lee.
--
__
info about this error or workaround it.
I' m using asterisk 1.8.11.1, as does the other end.
DCN is disconnect. So DCN in response to DCS means that the receiver
didn't like something about DCS, TCF or possibly TSI so much that it
decided to abort the fax by disconnecting
On 06/07/2012 06:18 AM, Daniel Seagraves wrote:
On Jun 6, 2012, at 10:47 PM, Lee Howard wrote:
> Unless you're going to move to an internet fax service provider you'll
probably not want to attempt to switch your fax line to a VoIP line and still
attempt to fax over it.
ypical. Depending on how much voice traffic you have and how
much of it is local or inbound... switching to a VoIP service may not
actually be a cost-cutting measure.
Thanks,
Lee.
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[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Wiater
Sent: Saturday, 26 May 2012 5:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Function not Registered??
On 5/25/2012 3:18 AM, Lee, John (Sydney) said:
-- Executing [*1223*1**1900
Hi all,
I am running the same Asterisk 1.4.21.2 with the same configuration on all the
servers in the region.
I got this function called func_devstate which I use to control the lights of
the Polycom phones.
This module works well for all the Asterisk servers except this one.
To get it to
On 05/18/2012 04:45 AM, Sebastian Gutierrez wrote:
with FFA I may get 70% of faxes ok.
Nobody that I work with would consider that acceptable.
Lee.
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product you're using.
Thanks,
Lee.
--
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http://www.asterisk.org/h
And what you were doing with the HylaFAX modem config file for the
iaxmodem should have worked to do this. Why it wasn't working can only
be determined by investigating your installation.
Thanks,
Lee.
--
_
-- Bandwidt
do I need to make this possible?
Not interested in HylaFAX with IAXmodems? (I presume that you are using
PSTN circuits and not VoIP.)
Thanks,
Lee.
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N
4579",
"FAXSTATUS: FAILED FAXERROR: Unexpected command after page received
FAXPAGES: 0 @ bitrate 9600") in new stack
FWIW, ping times from the asterisk box and the iax provider are <
15ms, usua
Thanks Sam, John and Justin for your wonderful advice.
Yes, it was the sip.conf parameter "reinvite=" which was causing the
problem.
Setting it to NO will fix it.
Thanks all in asterisk-users mailing list.
The contents of this e-mail are intended for the named addressee only. It
contains inform
I have been deploying Asterisk (open source PABX) in the company which I
work.
So far, all the Asterisk servers do not really talk to each other.
Recently, I am experimenting to dial from one Asterisk server to another
through the WAN and I encountered a no-audio problem although the
callee's phon
> chan_sip does not support specification of the password to be used for
authentication in the dial string itself;
> your ":password" suffix is just being sent to the target system and it
is trying to find a matching extension in the dialplan (and failing).
Thanks Kevin. This is what I reckon fr
I was trying to do a SIP call between two Asterisk servers (1.4.21.2)
1) On the caller server, I coded the following in extensions.conf
Dial(SIP/1166:password@asterisk-callee);
2) On the callee server, I coded the following in sip.conf
[1166]
type=friend; Friends plac
I was trying to do a SIP call between two Asterisk servers (1.4.21.2)
1) On the caller server, I coded the following in extensions.conf
Dial(SIP/1166:password@asterisk-callee);
2) On the callee server, I coded the following in sip.conf
[1166]
type=friend; Friends place calls a
I was trying to do a SIP call between two Asterisk servers (1.4.21.2)
1) On the caller server, I coded the following in extensions.conf
Dial(SIP/1166:password@asterisk-callee);
2) On the callee server, I coded the following in sip.conf
[1166]
type=friend; Friends place calls a
me?
thanks
--
From: "Lee Howard"
Sent: Thursday, September 01, 2011 6:29 PM
To:
Cc: "Asterisk Users Mailing List - Non-Commercial Discussion"
Subject: Re: [asterisk-users] problems with hylafax + iaxmodem +
asterisk1.8.5
Ale
ormally iaxmodem (and probably
therefore HylaFAX) should run on the same system as Asterisk.
Thanks,
Lee.
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Steve Underwood wrote:
On 09/01/2011 11:50 PM, Lee Howard wrote:
kirsten du toit wrote:
You should try disabling ecm..
This seems crazy to me. Why are you recommending it?
Because its the industry standard last resort of anyone who doesn't
understand FAX and is using T.38.
Ev
kirsten du toit wrote:
You should try disabling ecm..
This seems crazy to me. Why are you recommending it?
Lee.
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Steve Totaro wrote:
On Tue, Aug 9, 2011 at 7:22 PM, Lee Howard wrote:
Ryan McGuire wrote:
Unless your network is under load and you are seeing dropped packets
and high jitter, I would absolutely not do T.38. The cheapest and
easiest approach that I have found is to buy yourself an FXS
ay not be any better than that of G.711 fax over
the SIP UDP.
I only recommend faxing over TDM everything else is at your own risk.
Thanks,
Lee.
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New to A
e-force attacker when they have hit on a valid username.
I'm sure there are many other good habits to follow that others here
could share, but those come to mind with respect to the problem you've
experienced.
Thanks,
Lee.
--
Hi Kevin, the ticket below was closed as it doesn't happen with 1.8. It
can't be related to my ODBC connections if others are having it. Should
a new ticket be opened?
Regards
Lee
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-
Seems to be an already reported problem but since no more fixes for 1.6
it's back to 1.6.2.18.2 for me.
https://issues.asterisk.org/jira/browse/ASTERISK-18103
Regards
Lee
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digiu
Hi Eric, are you using ODBC?
Regards
Lee
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric
Wieling
Sent: 18 July 2011 13:54
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re
s a fix to ODBC so I don't
really want to downgrade. I will try and get some traces from one of my test
boxes.
Thanks
Lee
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Davies
Sent: 18 Jul
Hi, is anyone else having problems with the reload command crashing
Asterisk 1.6.2.19? I've run a few tests and 1.6.2.18.2 doesn't have
this problem but 1.6.2.19 after a few reloads is just dumping and
restarting.
T
Hi, can anyone help with this?
Thanks
Lee
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Archer
Sent: 05 July 2011 16:27
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Recording SIP history
Hi all, can
is output to the DEBUG
logging channel
Thanks
Lee
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er)
tied-down to audio channels by putting T.38 into H.323 or UDP/IP SIP
beats me.
Lee.
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o anything with this either.
Can anyone point me in the right direction please?
Thanks
Lee
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ia Telekom???
Hello Lee,
Telekom Malaysia provide PRI lines. We've been actively using their services
for the past few years with success. Let me know if you need contacts.
Regards,
Arstan
On Thu, Jan 20, 2011 at 9:56 AM, Lee, John (Sydney)
wrote:
We are setting up an office in Malaysi
We are setting up an office in Malaysia.
We contacted Telekom Malaysia and are surprised to be told that ISDN-30
is no longer available.
They are yet to give us information of the replacement technology.
Does anyone have any experience and information with this?
Thanks in advance.
--
_
Hi, does QUEUE_PRIO work the Queues and Asterisk 1.6.2? I've found some
documentation on Google but it looks like it's old Asterisk and not
current.
Thanks
Lee
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Hi, try unloading res_timing_dahdi.so then trying again.
Lee
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Davies
Sent: 07 December 2010 12:54
To: Asterisk Users Mailing List - Non-Commercial
ly pay $40 monthly for life than need to
pay $100K to develop this stuff.
> A little
> more searching today turned up this:
>
> http://www.gouloum.fr/code/sm/sm.html
>
> Which is REALLY close to what I need...
And note that it uses sp
r.conf and set
endbeforehexten=no, but this doesn't seem to make any difference.
Does anyone have any ideas or is it a problem with the cdr_adaptive_odbc
module?
Thanks
Lee
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In Asterisk, the funny thing is if a certain component is not installed
properly or the config file has a typo or something, this will be shown
up as a non-existent command in Asterisk command line interface.
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:ast
1:00 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Use modprobe to find E1/T1 jumper
setting
> onPRI card
>
> On 09/29/2010 02:52 AM, Lee, John (Sydney) wrote:
> > Do you mean that if I could define 30 channels in span 1 for
&
:30 +1000, Lee, John (Sydney) wrote
> Does anyone know if I could use modprobe command to find out rather than set
> the jumper on a Digium PRI card?
> I want to find out the jumper settings on the card without opening the box
> which will cause down time.
>
> Thanks.
Does anyone know if I could use modprobe command to find out rather than
set the jumper on a Digium PRI card?
I want to find out the jumper settings on the card without opening the
box which will cause down time.
Thanks.
--
___
Is there a way to specify which IP address to originate calls from in a peer
on sip.conf?
I need to send calls from 10.1.3.10 which is a routed network through
openvpn, but it's using 10.39.0.10 which is a vpn IP address - the asterisk
box is the same box as the vpn bridge for the 10.1.3.0/24 netw
i suppose that depends on the number of eggs and baskets you have... but i'm
guessing not many of either since you're considering using a desktop board
for this...
but, email sangoma support, they will tell you.
On 17 September 2010 12:47, John Novack wrote:
>
>
> Anita Hall wrote:
> > Hi
> >
>
to get accurate cdr's i just use a "border" server to send every call
through that logs cdr... doesn't matter how many times it gets transferred
internally the "border" server still gets accurate records of the whole
call.
On 27 August 2010 21:07, Benny Amorsen
> wrote:
> Carlos Chavez writes:
Thanks for the replies. I am already ignoring the signal but it doesn't
seem to be making much difference on this system with this script. It's
an old legacy script I should hopefully be dropping and writing properly
within the dial plan.
I will keep trying!
Thanks
Lee
-Origin
Hi, I am setting AGISIGHUP=no before running a Perl script via AGI but
it doesn't seem to be doing anything as the script is still exiting on a
hangup and not completing properly. I am using 1.4.35 and have tried
various combinations. Can anyone shed any light on this?
Regards
i do this by having 2 peers setup, one has a call limit of 10 and uses g729,
the rest of the calls get sent to the second peer which uses ulaw.
all calls attempt peer 1 if there's channels available it uses it if not it
just moves through the dialplan to use the second one.
On 19 August 2010 09:1
gust 2010 08:11, Sherwood McGowan wrote:
> On Wed, Aug 18, 2010 at 3:59 PM, Geraint Lee wrote:
> > This is what I ended up doing, working fine now.
> > Cheers
> >
> > On 18 August 2010 08:52, Nasir Iqbal wrote:
> >>
> >> Avoid to use MySQL dialplan applicat
This is what I ended up doing, working fine now.
Cheers
On 18 August 2010 08:52, Nasir Iqbal wrote:
> Avoid to use MySQL dialplan application, instead write an AGI script for
> this purpose
>
> On Tue, Aug 17, 2010 at 4:59 PM, Geraint Lee wrote:
>
>> Right, I'm baffl
Right, I'm baffled.
I have:
exten => s,1,MYSQL(Connect DB1 127.0.0.1 geraint xxx amis2)
exten => s,n,MYSQL(Query NORESULT ${DB1} INSERT\ INTO\ recordings\
(caller_number\,called_number\,date_created\,date_started\,in_use\,server_id)\
VALUES\ (\'${CALLERID(number)}\'\,\'${ARG1}\'\,NOW()\,NOW()\,\'Y
it would be far easier to just use the source...
but...
yum search asterisk
might get you on your way, although i can't see anything that looks like
samples in there.
On 13 August 2010 19:08, Albert Bonomo wrote:
> Hi, I'm trying to install Asterisk with yum.
> I have followed the instruction
I have discussed QoS with our ISP and in order to implement this, I need to
make
sure all VoIP packets are marked in the IP packet header (IPP bits?). Does
Asterisk automatically marks the VoIP packets or do I need to do something in
Asterisk? I need to do this for SIP and H323 protocols.
try looking in extensions.ael
On 25 June 2010 12:25, Eyal Goltzman wrote:
> Hi,
>
>
>
> I have a trivial peace of dialplan for exten 100. I try to change it to
> _1XX and the asterisk act according to a different (Default??) dial plan and
> not the one I want? Is that possible? Where is the oth
Should I log this as a bug since it doesn't work?
Regards
Lee
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Archer
Sent: 20 May 2010 16:28
To: Asterisk Users Mailing List - Non-Commercial Discu
Try a Cisco ASA. It will rewrite the headers if configured properly.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Motiejus Jakštys
Sent: 26 May 2010 14:17
To: Asterisk Users Mailing List - Non-Commercial Di
tension 1234,1,NoOp,hello into test
Failed to add '1234,1,NoOp,hello' extension into 'test' context
Regards
Lee
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Archer
Sent: 19 May 2010
Many thanks.
Regards
Lee
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: 19 May 2010 16:44
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users
Hi, anyone know?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Archer
Sent: 17 May 2010 11:29
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Adding a context from the console
Hi, is it
Hi, is it possible to add a context from the console using the dialplan
command?
Thanks
Lee
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I too, had really bad echoes on a TE121 w/ echo module. When I removed the
module, I haven't had as much echoes as before.
From: Sascha Ferley
To: asterisk-users@lists.digium.com
Sent: Sun, May 16, 2010 12:00:31 PM
Subject: [asterisk-users] Digium TE121P + DA
o AUTH for a successful authentication and a NoOp
shows the correct value again. But when the call ends the variable
going back to the original value.
Thanks
Lee
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I upgraded to 1.6 and tried F and it didn't do the same as the g option. I
will have to use the h extension to finish the logging.
Thanks
Lee
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian
Sent: 10 May 2010
Thanks, I figured it out. I was using 1.4 but have had to move to 1.6.1
Regards
Lee
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: 10 May 2010 17:27
To: Asterisk Users Mailing List
Thanks. Is there no 1.4 equivalent or is this a feature of 1.6 only?
Lee
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian
Sent: 10 May 2010 14:45
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
S
Hi, does anyone know if there is an equivalent dial option for the
source channel to the g option? I've had a good look and can't find
one.
g- Proceed with dialplan execution at the current extension if the
destination channel hangs up.
T
Hi, I have a system using ODBC and connecting to a MS-SQL database.
Does anyone know if it is possible to return a record set consisting of
several rows from SQL back into Asterisk? I have tried using ARRAY but
only the contents of the last row are being stored.
Thanks
Lee
Thanks for replying Noah. I'm using FreePBX web interface and have a "ring
group" that rings 4 phones as the operator. I do know that the context type is
"from-internal" but when it rings as below, the context type becomes
"from-pstn". Can you tell me where exactly to go and change in the Fre
I have a question about the blind transfer
using ##. This works great on our cordless phone, but there have been
occasions that we can't transfer using ##. I was able to reproduce the
issue by doing the following:
1) Call in from the outside line,
2) Ask the operator to transfer me to an exte
gt;>
>
> The patch I'm talking about won't affect t38modem and Hylafax usage at
> all. If the re-INVITE arrives before you have connected the call to
> t38modem, the negotiation process will very likely fail.
Typically HylaFAX users have the c
Can't upgrade the version. how about buying a FXS gateway and be done with
the issue. Go to ebay and search for AudioCodes. You can get 1 FXS port
gateway for around $30 to 2 FXS at $85. Probably the best bet is to convince
the customer to upgrade Asterisk.
__
check the IRQ and make sure the TDM410P has it owns IRQ.
From: Danny Dias
To: asterisk-users@lists.digium.com
Sent: Fri, April 9, 2010 4:52:05 PM
Subject: [asterisk-users] Problems with Fax over TDM410P
Hello my friends...
We are having some problems with the
Figured out my issue. My contacts are in -directory.cfg when it
should be in -directory.xml.
When did Polycom switched from CFG to XML?
From: hin lee
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Fri, March 19, 2010 12:00:14 PM
With the price of FXS gateway, why not just get SIP phones? Polycom 330 is
around $60-$110 a piece.
From: mir shahnawaz
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Tue, March 30, 2010 6:46:20 AM
Subject: Re: [asterisk-users] 24 FXS Por
Mike Diehl wrote:
> I could probably get hylafax configured, but I'm not sure how reliable it is.
>
> If it is considered reliable, would someone let me know?
It's reliable as long as you're not using FoIP (i.e. as long as you're
faxing wi
phone did not pull the new contacts directory. If I
format the phone file system, then it will reflect the new contacts.
From: "Lee, John (Sydney)"
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Wed, March 17, 2010 11:05:09 PM
S
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