Hello Federico,
Can you please review the Bug Report requirements, and submit a new bug
report for this issue?
https://docs.asterisk.org/Asterisk-Community/Asterisk-Issue-Guidelines/
Also Note:
Before filing a bug report... Your issue may not be a bug or could have
been fixed already. Run
a
commented/disabled example config for the 'mark' user. There is no
default 'open to the world' configuration for mainline asterisk. I
would agree however that the default bindaddr should not be 0.0.0.0 in
manager.conf.sample. I'll put in for a fix for that.
With that being said
Hi Dan,
Your best bet for looking at RTP media specifics is the standards that
define RTP.
Wikipedia has some really good resources on RTP and a list of the
various RFC standards that relate:
https://en.wikipedia.org/wiki/Real-time_Transport_Protocol
On 8/28/23 11:16, Dan Cropp wrote:
/sbin/asterisk -gvc
[Thread debugging using libthread_db enabled]
Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1".
snip
Created by Mark Spencer
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty'
for details.
snip...
etc etc
Hi Federico,
Segfaults are 100% not by design. Typically if something seg faulted,
either there is a logic bug or a component mismatch. The you should
definitely be able to use more than one connection (we use multiple
connections with postgres odbc with no issue).
If Asterisk segfaults
On 8/18/23 12:41, Joshua C. Colp wrote:
On Fri, Aug 18, 2023 at 1:09 PM Mark Murawski
wrote:
I've seen this happen three times in the wild now. I've been
trying to
isolate the source of the issue, but so far it seems like there's not
enough debug output to know why
amount of time
-
INVITE sip:+12011555432@152.X.Y.Z:5060 SIP/2.0^M
Via: SIP/2.0/UDP 192.81.237.20:5060;branch=z9hG4bK489fe.a7c59e79.0^M
From: "MARK MURAWSKI " ;tag=gK0c130ae5^M
To: ^M
Call-ID: 241982955_121107611@4.55.28.225^M
CS
Hi Steve,
You must be using a prebuilt system, maybe a prebuilt Asterisk-based
distribution? Asterisk does not send email by default... Almost
nothing is done by default. Things like sending email have to be
specifically configured to do so in voicemail.conf. If you don't want
to send
Hi Justin,
There's absolutely no detail here regarding the SIP messages going out
and back. You'll need to include the asterisk-side sip debug.
https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
On 8/4/22 20:32, Jerry Geis wrote:
I am running Asterisk 13.30.0
40 core CPU (VM) VMware.
CentOS 7
32 G ram
10G vmx network
Should be plenty of room for anything...
Yes asterisk is running 270% CPU...
Is it not taking advantage of the 40 cores ?
I am bring around 300 SIP endpoints in a muted
On 8/31/22 09:25, Antony Stone wrote:
If I simply do
Tracker="${CDR(uniqueid)}";
it works as required.
It's just not the sort of syntax I've seen in any other language, and it feels
(to me) weird.
^^^ Yup! This is what I was suggesting in my last email. Just add quotes.
Think
On 8/31/22 05:29, Antony Stone wrote:
What I am suggesting is that Tracker=${CDR(uniqueid)} should be converted
by AEL into Set(Tracker=${CDR(uniqueid)}) in order to avoid this sort of
surprise.
On the flip-side... anyone who currently relies on purely
numeric/boolean handling of the current
On 8/30/22 17:51, Mark Murawski wrote:
On 8/30/22 12:34, Antony Stone wrote:
I want.
However writing:
Tracker=${CDR(uniqueid)};
results in:
MSet(Tracker=-1661872057.2349)
systemname is missing.
Hi Antony,
This is not a problem with MSet.
No, it is indeed the documented behaviour
On 8/30/22 12:34, Antony Stone wrote:
I want.
However writing:
Tracker=${CDR(uniqueid)};
results in:
MSet(Tracker=-1661872057.2349)
systemname is missing.
Hi Antony,
This is not a problem with MSet.
No, it is indeed the documented behaviour of MSet "MSet behaves in a
On 8/30/22 11:16, Antony Stone wrote:
If I write in my AEL dialplan:
Set(Tracker=${CDR(uniqueid)});
this results in executing:
Set(Tracker=eagle.domain.com-1661872057.2349)
Just what I want.
However writing:
Tracker=${CDR(uniqueid)};
results in:
On 8/29/22 14:00, aster...@phreaknet.org wrote:
This is a mockup of what the new-style if/else processor would output
26. NoOp(AEL IF("\${DIALSTATUS}" == "BUSY") --
extensions.ael:1405)
27. GotoIf($["${DIALSTATUS}" == "BUSY"]?28:30)
On 8/29/22 09:30, Antony Stone wrote:
It is, although there are ways I think it can be improved - I'm wondering how
best to go about proposing these.
The most obvious for now are:
- please can "a=1;" be converted to use Set() instead of MSet() (especially
since MSet is officially
On 8/29/22 10:15, Antony Stone wrote:
But! What specific reason do you have for wanting Set() instead of
MSet() for all assignments that can't be otherwise just written as an
in-line Set() instead?
I *am* currently writing inline Set() everywhere, but surely the syntax "a=1;"
instead of
On 8/29/22 09:53, Antony Stone wrote:
On Monday 29 August 2022 at 15:35:09, Joshua C. Colp wrote:
MSet is not deprecated.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_MSet
includes the sentence "MSet behaves in a similar fashion to the way Set worked
in 1.2/1.4 and is
On 8/29/22 08:48, Mark Murawski wrote:
On 8/29/22 08:31, Antony Stone wrote:
Hi.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Application_Originate
I need to use Originate() in a dialplan, pointing to another location
in the
same extension of the same context, so for example
On 8/29/22 08:31, Antony Stone wrote:
Hi.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Application_Originate
I need to use Originate() in a dialplan, pointing to another location in the
same extension of the same context, so for example:
On 3/1/22 05:59, Karsten Wemheuer wrote:
Am Dienstag, dem 01.03.2022 um 06:37 -0400 schrieb Joshua C. Colp:
On Tue, Mar 1, 2022 at 6:14 AM Karsten Wemheuer wrote:
Hi *,
i am currently trying to migrate from chan_sip to pjsip. I am using
Asterisk version 18.10.
In chan_sip information about
Hi Antony,
NOW is not a variable...
In the majority of cases (the exceptions are things like CUT)...
variables are utilized by ${}
If NOW was a variable you would see it written as ${NOW}
The word NOW is actually not special. Deep in the Asterisk source (if
you are curious), the flow is
If you're executing /usr/bin/rm directly, shell aliases will have no effect.
On 1/11/22 11:29, Antony Stone wrote:
On Tuesday 11 January 2022 at 17:20:44, Michael Englehorn wrote:
If you're on RHEL or CentOS or one of its descendants,
Oh, now that reminds me that those systems also tend to
Hi Daniel,
This is a production server which is running well over years (asterisk
11-13-16) and this happend with the latest version. Only valid option
you gave is the core show locks. I ask the list before opening a bug
report, as usually.
Please don't let the fact that the system has
Hi,
1) You should change your name on your email client so it doesn't say
"Administrator"
2) Please follow the instructions at
https://wiki.asterisk.org/wiki/display/AST/Installing+Asterisk+From+Source
3) Compile with DEBUG_THREADS and DONT_OPTIMIZE, but note this will
incur a performance
On 1 Apr 2020, at 22:14, Greg Troxel wrote:
>
> I think you need to use tcpdump and turn up firewall debugging.
sngrep is your friend …My bet is UDP vs TCP on firewall rules :-)
Mark
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.
Mark
> On 11 Feb 2020, at 19:22, John T. Bittner wrote:
>
> Guys,
>
> I have a customer that heavily uses modems, the problem they don’t work
> reliably with some of the carriers I have used like Level3.
> This is somewhat expected due to the limits in VoIP so I need
ding to another environment they
should never send us inband but it seems to not be working correctly in the
case.
Regards
Mark.
On Mon, 17 Jun 2019 at 10:11, Floimair Florian
wrote:
> Just a guess, but I suspect that this might be related with strictrtp
> setting in rtp.conf, wh
=yes
internal_timing=yes
progressinband=never
silencesuppression=no
prematuremedia=no
(Per peer)
progressinband=yes
directrtpsetup=no
dtmfmode=rfc2833
directmedia=no
silencesuppression=no
prematuremedia=no
TIA
Mark.
--
Mark Farmer
farm...@gmail.com
On 4/19/2019 1:49 PM, Dovid Bender wrote:
> Mark,
>
> I am using PHP agi and when forking the call does not continue util
> the forked process is done. Am I doing it wrong?
>
>
> On Wed, Apr 10, 2019 at 4:27 PM Mark Wiater <mailto:mark.wia...@greybeam.com>> wrote
On 4/10/2019 3:54 PM, Dovid Bender wrote:
> I have an AGI that can sometimes take time complete. I don't want the
> dialplan to be held up by the agi. Is there any way to call it and have
> Asterisk continue with the dialplan?
>
Is there a reason you can't fork in the AGI and just return to the
Perfect! That was it. ICE support was switched on. It's not needed at all
here so I'll settle for that :)
Thank you Joshua.
On Fri, 5 Apr 2019 at 13:08, Joshua C. Colp wrote:
> On Fri, Apr 5, 2019, at 8:58 AM, Mark Farmer wrote:
> > Thanks for helping with this. Please find reque
Thanks for helping with this. Please find requested logs here:
https://pastebin.com/raw/rY1BSUDb
I have an entry in /etc/hosts that defines both the hostname and the FQDN.
Mark.
On Fri, 5 Apr 2019 at 12:43, Joshua C. Colp wrote:
> On Fri, Apr 5, 2019, at 8:07 AM, Mark Farmer wrote:
>
Hi. Sadly I'm still struggling with this. I've captured the debug output.
I also upgraded to 16.3.0 this morning.
https://pastebin.com/raw/HxbX0uXt
Many thanks
Mark.
On Thu, 4 Apr 2019 at 15:32, Mark Farmer wrote:
> Thanks Joshua.
>
> Hopefully I'll be able to retry tomorrow.
>
&
Thanks Joshua.
Hopefully I'll be able to retry tomorrow.
On Thu, 4 Apr 2019 at 15:30, Joshua C. Colp wrote:
> On Thu, Apr 4, 2019, at 11:27 AM, Mark Farmer wrote:
> > Thanks, I did enable debugging but didn't see any attempts to resolve
> > hostnames. I will give it another
=system
I will retest/debug when ASAP.
Mark.
On Thu, 4 Apr 2019 at 15:20, Joshua C. Colp wrote:
> On Thu, Apr 4, 2019, at 11:18 AM, Mark Farmer wrote:
> > Seems to be res_resolver_unbound.so
> > Reading the documentation now but any hints greatly appreciated!
>
>
Seems to be res_resolver_unbound.so
Reading the documentation now but any hints greatly appreciated!
Mark.
On Thu, 4 Apr 2019 at 15:07, Joshua C. Colp wrote:
> On Thu, Apr 4, 2019, at 11:03 AM, Mark Farmer wrote:
> > Sorry, should have included that.
> >
> > Asteris
Sorry, should have included that.
Asterisk 16.2.1
Mark.
On Thu, 4 Apr 2019 at 14:56, Joshua C. Colp wrote:
> On Thu, Apr 4, 2019, at 10:53 AM, Mark Farmer wrote:
> > As I understand it, delays like this are almost always caused by slow
> > or failing DNS lookups. Running a
dded records to my hosts file and checked using 'genet ahosts hostname'
but still the issue remains.
So how do I figure out what is going wrong please? This is preventing me
from moving from chan_sip to chan_pjsip.
Many thanks
Perfect, thanks.
The dialplan did come from a chan_sip server :) Now updated.
Mark.
On Tue, 2 Apr 2019 at 22:40, Joshua C. Colp wrote:
> On Tue, Apr 2, 2019, at 6:35 PM, Mark Farmer wrote:
> > Hi everyone
> >
> > I’m building an Asterisk 16/PJSIP server and my dialplan
Hi everyone
I’m building an Asterisk 16/PJSIP server and my dialplan uses SIPAddHeader &
SIPRemoveHeader but the apps don’t appear to be installed in v16.
Can anyone tell me where they went and how to get them installed please?
Thanks
Mark.
Mark Farmer
Senior UC Systems Architect
Inter
ing the exact same RTP traffic between provider side
and client side? And was client side captured close to the phone, past the
firewall if there is one?
Mark
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These two phones are not using the same extension, are they?
On 2/6/2019 8:49 AM, basti wrote:
> both phones are registered. and the hardware phone can also make calls.
> but an incoming call is not displayed and also not hearing.
>
> Call Waiting is also disabled.
>
> On 06.02.19 14:07, Cyril
On 8/30/2017 5:03 AM, Steve Davies wrote:
> Mark,
>
> You have cropped the image you inserted above and removed a very
> important part of the line you highlighted. I think is says ",Mark"
> after the time value - You can even see the un-cropped comma in your
> pictur
? Consistently by
480, 3 voice frames?
Will Asterisk just drop the packets that compromise the rewind?
Thanks
Mark
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Check out the new Asterisk community forum
Hi Mike
In this case, if it’s coming from friendly scanner why not drop the packets at
the firewall layer so that Asterisk never sees them?
Mark
> On 15 Aug 2017, at 20:37, mdiehl <mdiehlena...@gmail.com> wrote:
>
> Hi all,
>
> Lately, I've seen an increase in the numb
I've had Lefteris' code running for a few years without a problem.
I don't have a service key but I have entered my API key in the script
in the 'User defined parameters' section. You did that, right? What do
the other user defined parameters in your script look like?
On 7/19/2017 4:37 AM, Rahul
On 7/12/2017 5:30 PM, Holger Freyther wrote:
> I have to copy/mirror/forward the RTP streams for some selected call
> to an external address/port
I'd think that what you want to do might be best done outside of
Asterisk. If you're working with SIP, I'd suggest packet capture tools.
--
On 6/20/2017 8:42 AM, Tech Support wrote:
> I appreciate all the feedback, and replication seems to be a logical
> solution, but I was initially thinking about how to implement a solution
> within Asterisk to write the CDR's to two databases. Is that possible? Now
> I'm just curious.
On 5/31/2017 3:36 PM, Steve Edwards wrote:
I want to capture all SIP messages.
I have about 30 hosts in about 6 colos.
My first thought was dumpcap, but the output file name format bugs me.
What do you use for long term SIP capture?
voipmonitor is what you want.
--
On 5/25/2017 11:11 AM, Tech Support wrote:
I need to be able to tell whether or not the far end extension picked up
might a waitForSilence come in useful here?
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On 4/18/2017 7:40 PM, Ernie Dunbar wrote:
Server network: 192.168.0.0/24
OpenVPN network: 10.8.0.0/24
Asus network: 192.168.1.0/24
The Asterisk SIP registration appears to be responding properly to
this - this is what I see when I do a 'sip show peer' for an Aastra
phone that's connecting
a SIP NOTIFY with one of
cisco-check-cfg, grandstream-check-cfg, polycom-check-cfg,
sipura-check-cfg, snom-check-cfg
and the extension.
I've done it with Yealink phones too, don't have the proper syntax in
front of me though.
Mark
e to open a single
>file containing video and accompanying audio and be able to play those back.
Hi Mark,
Thanks for your reply...
I just tried what you suggested on only got audio. I created a wav
file and put it in the /tmp
directory just like the video.h264 file. So /tmp has video.h264 and
vi
between those two networks. Often what routes, can firewall.
000122.941|sip |4|03|Registration failed User: 165, Error Code:480
Temporarily not available
Mark
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for Asterisk to be able to open a single
file containing video and accompanying audio and be able to play those back.
Mark Michelson
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file contain an address for the
phone to contact?
Mine has voIpProt.server.1.address, but I think you can also use a
reg.x.address in the provisioning files too.
Mark
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I think you had asked what phone works well with VPN's. I've had very
good experiences with Yealink using OpenVPN, never an issue.
I think I've heard that Snom does OpenVPN as well.
Mark
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was
seeing.
I don't think so. At least I don't see that.
Mark
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Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
http
the
original email
exten => 55,1,Verbose(Door buzzer calling)
same => n,Dial(SIP/user1/user2/user3)
should have rung the phones forever as long as one phone was active and
not forwarding or DNDing.
Mark
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-
ethernet
over several pair of copper lines.
The ISDN BRI solution is less than 1/2 the price of the SIP solution.
Any recommendations? Pitfalls?
Mark Engelhardt
- in snowy Vermont!
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,
connecting to both the Broadworks and Plexus platforms in Windstream.
Though all of my asterisk versions have been 1.8.x
Mark
On 2/22/2016 8:20 AM, James Cass wrote:
> Does anyone on this list use Windstream as a SIP trunk provider?
>
> If so, would you mind sharing your peer settings
ICAL URL from gmail and that it's the only one that worked for me. Is
that the URL that you're using?
Did you change your type to ical in calendar.conf?
Mark
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On 07/10/2015 11:53 AM, Rodrigo Pimenta Carvalho wrote:
Hi.
The ASTERISK wiki has a page showing the function PJSIP_HEADER(). However, it
doesn't explain if such function works only over SIP INVITE messages or if it
can be use, for example, to read headers from others types of SIP messages
understand it is because of existing
expire value but would like the previous expire timer to reset and issue a
new registration instead
Regards,
Mark
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New
could help? Set sip debug if it's
a sip trunk?
You'll at least get to see the callerid that Asterisk is putting on the
trunk. That might even help your new VOIP provider do some digging if
could provide the debugging output.
Mark
instead which is working OK but
I’d much rather stick with MixMonitor.
Asterisk version is now 11.17.1
Thanks!
From: Mark Farmer
Reply-To:
asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com
Date: Friday, 10 April 2015 16:34
To: asterisk-users@lists.digium.commailto:asterisk
to handle voicemail? I would
expect anyone who could have taken the call to be able to access the
voicemail, and once one person has dealt with a message it's no
longer a new message to anyone else.
Mark
--
Mark Rogers // More Solutions Ltd (Peterborough Office) // 0844 251 1450
Registered in England
calls at the moment, I
daresay I will want to handle internal calls at some point too,
although nothing stops me having dedicated per-user extensions as well
as per incoming trunk.
Mark
--
Mark Rogers // More Solutions Ltd (Peterborough Office) // 0844 251 1450
Registered in England (0456 0902) 21
Hello,
can someone recommend a good and free Softphone for Windows which does
not display advertisments inside the program?
We have X-Lite but free version display advertisments.
thanks.
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thanks Guys. I like Zoiper. Will try it.
On Sun, Jun 8, 2014 at 5:05 PM, binary dreamer.bin...@gmail.com wrote:
have you tried zoiper or 3cx?
On 9/6/2014 00:01, Mark Robinson wrote:
Hello,
can someone recommend a good and free Softphone for Windows which does
not display
On 10/28/2013 3:59 PM, Ron Wheeler said:
I am reaching the same level of frustration.
I have tried to find the source of the problems.
We have IAX2 to our VoIP provider and SIP phones attached to the Asterisk - No
analogue.
I don't have any problems with IAX, but I hear some do.
We have a
On 10/17/13 23:06, John T. Bittner wrote:
Today I was hacked but caught it very quickly. This is the weird part,
they hacked an IP Auth based account by simply knowing the account name.
How is this possible? I am running Asterisk 11.5.0. Now it’s my fault I
used a dictionary based account name
can
name them whatever you wish. The important piece of information when
determining what type of configuration section it is is the type=
option for the section. With no type= option set, the configuration
section is completely ignored.
Mark Michelson
1. Your softphone is not sending g729
[Jun 3 13:11:27] Capabilities: us - 0x10c (ulaw|alaw|g729), *peer -
audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing)*, combined - 0x4
(ulaw)
I think free version of eyebeam doesn't come with g729, try Microsip or
some other with g729 codec.
If it
Joseph,
I have made a quite a few test calls to 911. They don't charge you and they
don't get upset.
Just let them know right away it is a non-emergency test call, and then let
them know who you are and what you need to verify on their information screen.
Mark Engelhardt
On May 5, 2013
Tried disabling qualify and changing frequency with qualify=yes already, no
luck :(
On Mon, Apr 15, 2013 at 10:11 PM, Mehroz Ashraf
mehroz.ashra...@gmail.comwrote:
I believe qualify parameters does help in doing so. Asterisk forgets about
the peer info when qualify are not acknowledged. You
this is my secondary email
Regards
Zohair
On Mon, Apr 15, 2013 at 10:45 PM, Mark Henry markhenry...@gmail.com wrote:
Tried disabling qualify and changing frequency with qualify=yes already,
no luck :(
On Mon, Apr 15, 2013 at 10:11 PM, Mehroz Ashraf mehroz.ashra...@gmail.com
wrote:
I
Hi,
I have this for UAE,
dateTimeSetting
dateTemplateD/M/YA/dateTemplate
timeZoneArabian Standard Time/timeZone
ntps
ntp
name2.2.2.2/name
ntpModeUnicast/ntpMode
/ntp
Hello List,
I have some doubt about direct media settings.
I have an asterisk 1.8.14 instance running on 172.20.255.50, a soft phone
on IP 10.100.210.51 and a gateway at 10.100.210.254
I have set both gateway and peer to directmedia=yes but still on gateway
I see RTP from asterisk's IP, have
Hello List,
I have some doubt about direct media settings.
I have an asterisk 1.8.14 instance running on 172.20.255.50, a soft phone
on IP 10.100.210.51 and a gateway at 10.100.210.254
I have set both gateway and peer to directmedia=yes but still on gateway
I see RTP from asterisk's IP, have
(DB(Nightop/ext)=107). It’s not as
robust as Mysql or postgres but does seem to do better than the old Berkley
database.
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Mark Robinson
*Sent:* Thursday, October 18, 2012
On 12/27/2012 07:36 PM, Ron Wheeler wrote:
On 27/12/2012 3:14 PM, Carlos Alvarez wrote:
On Thu, Dec 27, 2012 at 12:46 PM, Ernie Dunbar maill...@lightspeed.ca
mailto:maill...@lightspeed.ca wrote:
This past holiday weekend has resulted in some real groaners when
it comes to bugs in our
Carlos,
I think the noise you are hearing might echo cancelation that is broken or set
incorrectly. Maybe the card and asterisk are both trying to echo cancel?
Mark
On Nov 13, 2012, at 1:52 PM, Carlos Chavez wrote:
I have a new install and the customer is complaining that they hear noise
As you may know, asterisk version 10 and high use sqlite. Are any examples
or documentation how to use in dialplan?
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in the dialplan by users are case-insensitive.
Thus if the variable MARK were set, then ${MARK} and ${mark} would
both evaluate to the set value.
2) Variables used internally by Asterisk are case sensitive. So if
some application set a variable called MARK, it would be different
from a variable set by some
Paetec was purchased by Windstream. I was looking one time ago to buy sip
trunk from them that run via T1. If anyone use them with asterisk it would
be nice to hear feedback.
On Sep 26, 2012 9:25 PM, Jared Baxley jared.bax...@gmail.com wrote:
Has anyone had experience using a SIP trunk provided
Hello.
I have asterisk 1.8.18 which connects to ISDN PRI. All phones are sip,
Aastra 6757i. Everything works as expected.
We also have a FAX machine. We need to be able to use that FAX machine to
send or receive faxes. We are planning to have a dedicated did for faxes.
Before, FAX machine was
Thanks Shaun.
On Sep 15, 2012 1:10 AM, Shaun Ruffell sruff...@digium.com wrote:
On Fri, Sep 14, 2012 at 09:42:23PM -0400, Mark Robinson wrote:
I did some research on this subject and still do not understand.
Why we use modules if asterisk can obtain timing directly from
kernel?
Probably
I did some research on this subject and still do not understand. Why we use
modules if asterisk can obtain timing directly from kernel?
On Sep 13, 2012 11:25 PM, Mark Robinson vsysnetw...@gmail.com wrote:
Thanks Shaun. Very usefully head-up
Thanks Shaun. Very usefully head-up.
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http://www.asterisk.org/hello
I know that asterisk on virtual machine require a timing source. What would
you suggest to use for timing? We will plan to use only SIP and IAX2.
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New to
Has anyone attempted to use an Allworx 9212 handset with an Asterisk PBX?
M. Hutter
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On 5/25/2012 3:18 AM, Lee, John (Sydney) said:
-- Executing [*1223*1**1900@incoming:78] Set(SIP/1900-08ee1da8,
DEVSTATE(Custom:cfalw1900)=INUSE) in new stack
I use
'Set(DEVICE_STATE(Custom:var)=BUSY)'
in my 1.4 dialplans to set device state.
mark
there are no ringing channels in context [from-my-sip-provider]
there are no calls to pick up there. However, since [context-100] and
[context-200] both have ringing channels, doing a call pickup in either
of these results in a successful pickup.
Mark Michelson
on?
Thanks,
-Mark
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incorrect. The problem seems due to the
extension not available in your dialplan. Please check carefully in which
context the call is placed and if the extension is defined in that context.
Maybe it can be useful to define a _X. extension to catch all not defined
extensions.
Leandro
[Mark Farmer
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