fc30.i686/drivers/dahdi/xpp/xpp_usb.ko] undefined!
BUILDSTDERR: ERROR: "__udivdi3" [/builddir/build/BUILD/dahdi-linux-kmod-3.0.0/
_kmod_build_5.2.8-200.fc30.i686/drivers/dahdi/xpp/xpp_usb.ko] undefined!
BUILDSTDERR: ERROR: "__moddi3" [/builddir/build/BUILD/dahdi-linux-kmod-3.0.0
00.fc29.x86_64
>
> The same kernel packages as the 5.1 kernels.
>
> sean
Hi Sean. Unfortunately I can only add a +1 for the DAHDI kernel modules, but
can confirm that the SipWise rtpengine kernel module also fails to build. I'm
waiting to try on 5.2.8 to see if anything is different
s://messinet.com/post/voip/2013/09/10/leaving-the-dundi-e.164-network/
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i-linux-kmod/dahdi-linux-kmod.spec
It looks like you're using F24, so you might be able to rebuild using the
SRPMs https://messinet.com/pub/fedora/linux/updates/26/SRPMS/
--
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t; I'm trying to replace it with
>
> PJSIP_CONTACT(${CHANNEL(contact)},user-agent}) but I'm not getting any data
> returned when I query ${CHANNEL(contact)}
>
> Is there a different function I should use to get my needed user agent of
> the active call?
How about ${PJSIP_HEADER(read,User-A
nk.com/forum/showthread.php?tid=8330=39161#pid39161
> It's a classic NAT situation: the phone system is in a droplet at digital
> ocean, but my phones are here at home behind a NAT. I see only 3 NAT
> related options:
>
> force_rport
> rtp_symm
George, thanks so much for the help on this. The wizards did the trick!
Sincerely,
Anthony Critelli
B.S. Applied Networking and Systems Administration, 2014
www.acritelli.com
(845) 283-4117
On Mon, Feb 8, 2016 at 10:08 PM, George Joseph <george.jos...@fairview5.com>
wrote:
>
>
>
question
is: what would the trunk configuration look like on the other Asterisk
server? Would it be the same, minus unique things like IP addresses?
https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Configuration+Examples
Thanks so much for the help.
Sincerely,
Anthony Critelli
www.acritelli.com
}) monitor OK
exten = 200,1,Tranfer(SIP/
mailto:SIP/65...@toto.home.local65...@toto.home.local
mailto:SIP/65...@toto.home.local) monitor not ok
do you have an idea ??
Thank you
ANTHONY--
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callerid to following:
exten = 6003,n,Originate(SIP/6003@asterisk,app,meetme,6003,x)
I use this patch
https://messinet.com/rpms/browser/asterisk/asterisk-12-app_originate_callerid.patch
because of https://issues.asterisk.org/jira/browse/ASTERISK-23016
-A
--
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On Thursday, December 25, 2014 03:53:44 PM Dmitry Melekhov wrote:
25.12.2014 15:46, Anthony Messina пишет:
On Thursday, December 25, 2014 11:48:12 AM Dmitry Melekhov wrote:
I want to change call files, which has caller id in them, to call
originate from dial plan.
But I don't see
:
https://issues.asterisk.org/jira/browse/ASTERISK-24441
-A
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Hello asterisk users,
Compiled asterisk-11.13.0 on openSUSE 13.1, however Channel driver chan_sip
is XXX in menuselect --- it depends on: chan_local(M), res_crypto(M),
res_http_websocket(M)
chan_local is [*] chan_local in menuselect,
res_crypto is in Resource Modules, Depends on:
List - Non-Commercial Discussion
Subject: Re: [asterisk-users] No chan_sip in compiled asterisk-11.13.0
Anthony Azzopardi wrote:
So this means that openssl(E) is holding everything?
Can someone give me some help on this?
On my Debian install, openssl shows as:
openssl
for the report.
Thanks for the quick turnaround. It is much appreciated. -A
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resolves this issue as well.
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solved, permissions problem. Asterisks run with user asterisk at default, I
changed to asteriskpbx as the book says ;)
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anthony
Azzopardi
Sent: 03 September 2014 20:57
Hello asterisk-users,
Just compiled and installed 11.12.0 however when I try to connect with
rasterisk I get:
Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl
exist?)
It seems that asterisk.ctl is not created.
--
=2ce57c334633881bb4d1baaeb6ae1e63c032abdc
It's what Fedora uses as well. This should work properly in EL7. Hopefully
in not too long, I'll have Asterisk 13 builds for EL7, though I need to figure
out a few dependency issues: https://messinet.com/rpms/
-A
--
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On Thursday, August 14, 2014 03:15:16 AM Paul Greenberg wrote:
Hi Anthony,
That script does not work. My guess is that it is related to the way
asterisk interacts with CentOS environment.
Best Regards,
Paul Greenberg, Esq.
On Wednesday, August 13, 2014 12:11:42 PM Carlos Chavez wrote
,
Anthony.
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On Sunday, March 30, 2014 02:24:35 PM Anthony Messina wrote:
On Sunday, March 30, 2014 07:07:47 PM Tzafrir Cohen wrote:
On Fri, Mar 28, 2014 at 07:57:54PM -0500, Anthony Messina wrote:
On Friday, March 28, 2014 07:43:48 PM Anthony Messina wrote:
Unfortunately, after
http
On Sunday, March 30, 2014 07:07:47 PM Tzafrir Cohen wrote:
On Fri, Mar 28, 2014 at 07:57:54PM -0500, Anthony Messina wrote:
On Friday, March 28, 2014 07:43:48 PM Anthony Messina wrote:
Unfortunately, after
http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commitdiff;h=6cebc1c
'
make: *** [install-firmware] Error 2
-A
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On Friday, March 28, 2014 07:43:48 PM Anthony Messina wrote:
Unfortunately, after
http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commitdiff;h=6cebc1c1fb1
2cc0661f3810ef47ad33206b2e398
I am unable to build DAHDI-Linux in a mock chroot for packaging
purposes. I believe
of the devices.
http://messinet.com/trac/wiki/Asterisk/Message
-A
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On Monday, January 28, 2013 08:06:38 AM Anthony Messina wrote:
On Monday, January 28, 2013 01:55:09 PM Steven Howes wrote:
Who do I need to poke to get the yum repository / RPM files updated? The
dahdi RPMs are not up to date with the CentOS kernel versions any more,
it's making doing
.fc18
As a side note, I've been working out how to move forward with kernel module
signing in Koji, as I've upgraded to Fedora 18. So far, the prospects for
signed kernel modules are looking good. Though I wish Digium would just get
DAHDI into the upstream kernel already :/
-A
--
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or --strip option?
You'll need to use.the -p or --strip option^^
But in your case, both you and DAHTOOL-60-f17.diff will need to be in the
2.6.1+2.6.1/tools/ directory before you issue:
patch -p1 DAHTOOL-60-f17.diff
--
Anthony - http://messinet.com - http://messinet.com/~amessina
the channel.
This did not happen to me in Asterisk 10. After removing the traditional
Hangup() at the end, and restarting Asterisk, the messages route properly for
me. -A
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On 12/02/2011 11:37 AM, Anthony Messina wrote:
I've just connected my new Android (Motorola RAZR) phone to Asterisk
using CSipSimple and have discovered that on any call between CSipSimple
and an Asterisk DAHDI or IAX2 channel, the 'other' end of the call will
hear a rhythmic tapping as if my
Hi Albert,
we currently use QueueMetrics to monitor and report on call center
statistics...
regards
Anthony
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Generic PLC: Enabled
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Hi Jean,
I suggest Queuemetrics. There are many out there but this one is good for
monitoring and reporting.
I know there's a free version you can try.
All the best
Anthony
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On 08/20/2011 07:00 AM, Tim King wrote:
exten = h,n,System(/usr/bin/php /var/lib/asterisk/bin/faxnotify.php
do you need the -f option to php?
exten = h,n,System(/usr/bin/php -f /var/lib/asterisk/bin/faxnotify.php
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8F89
On 08/06/2011 09:49 PM, Bruce Ferrell wrote:
Errors follow:
http://lists.digium.com/pipermail/asterisk-users/2011-July/264993.html
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/AsteriskFAXGateway
I have some time next week if it needs some tweaks to work with Asterisk
1.4. -A
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/AsteriskFAXGateway
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.
I use http://www.gafachi.com/ for outbound T.38.
I have had excellent service from both.
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kernel module. Or, if you are
using a RedHat/Fedora based distro, you're welcome to use the
dahdi-linux and dahdi-linux-kmod RPMS I build here. I include OSLEC
with the dahdi-linux-kmod build.
http://messinet.com/rpms/
--
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jump to the failed
extension?
You need to define the 'failed' extension in your context to have the
${REASON} variable set (I've found).
exten = failed,1,NoOp(Failure reason is: ${REASON})
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am using Asterisk
1.6 with this AGI.
-A
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, the Bridge command is used to bridge the
original (with the matching DB entry) call-- the call that is coming in from
GV through ipKall.
I suppose you don't need that AGI and could probably do this using Curl in the
dialplan.
-A
--
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[gv-out]
exten = _X.,1,AGI(gv/gv.agi,call)
same = n,While($[${DB_EXISTS(gv/channel)} = 1])
same = n,Wait(0.3)
same = n,EndWhile()
same = n,Hangup()
And the AGI (written in Bash) is here:
http://messinet.com/trac/wiki/AsteriskGVGateway
http://messinet.com/trac/browser/gv/gv.agi
--
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in there but my eyes are glazing ove
Believe me, I've glazed over the Bash man page for quite some time to get that
interface going ;)
If you're interested in mail to fax (and back), give it a shot. I could use
some testers.
Have a good night. -A
--
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the document open? the solution might be different for
every reader of that document.
the previously proposed web link-based solution would provide you with the
greatest reach.
perhaps we aren't exactly sure what you are trying to accomplish. what is
your end goal?
--
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the uniqueid was)
Then I could go on to say
T=$agi_uniqueid
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works well also.
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phones. -A
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exten = 1234,1,SIPAddHeader(Alert-Info: info=Bellcore-dr2) ; Double Ring
same = n,Dial(SIP/...
[internal-context]
; Calls routed from within the system
exten = 1234,1,Dial(SIP/... ; No special ring
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in the current context.
you could use DUNDi for this and avoid external DB and/or AGI. -a
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Hello,
I was planning on using a call file to test my IVR on a regular basis to
ensure it is operational
Channel: local/1...@from-internal
Application: SendDTMF
Data: ww12345678#1w1234#w1ww
But what ever I try so far the IVR does not seem to
You could read the source code, but based on it's name I would say it is a
library responsible for zone specific tone generation. Many parts of the world
have different tone patterns than the U.S. and Asterisk is used worldwide. A
better question is, why are you concerned by it?
e159 is the vendorid for the TDM400P. You'll see all the drivers that
use e159. Then lsmod | grep those drivers other than wctdm. If you see
one loaded, blacklist it.
sean
thanks, sean! that worked for me:
http://messinet.com/trac/rpms/changeset/141
--
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On Sunday 07 March 2010 05:10:02 pm sean darcy wrote:
Good. Glad it we figured it out. BTW, is your src.rpm for dahdi-linux
available?
sean
Here you go. -A
http://messinet.com/pub/fedora/linux/updates/12/SRPMS/dahdi-
linux-2.2.1-2.fc12.src.rpm
--
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just finished the same thing with, unfortunately, the same result
as you on both i686 and x86_64.
I'll keep googling :)
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the adsl came up and dns could be done, everything
worked fine again
I can confirm that exact same behavior: 1.6.1.12
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This is often caused by the dahdi module not loading, check
/var/log/asterisk/messages for the reason, or better yet, from the cli load the
module manually and see the error in real time. If I had to guess I would say
it is a configuration error.
Thank you and have a nice day,
Anthony Francis
Hello List
I'm trying to figure out what is wrong between my asterisk and my NEC 2400
pbx
We have been trying to link them with a spare PA-24DTG
from the NEC, I'm able to call an extension on the Asterisk, however the
extension rings, and then immediatly hangs up
I traced it back to the debug of
them for the kernels I use (Fedora 12 x86_64 and i686.PAE) but you can
check http://messinet.com/trac/rpms/ and checkout the svn-build-rpm tool to
build from an svn checkout if you already have a build setup configured.
--
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were using CentOS. You'll need to change some of the
definitions at the top of the file to match whatever version of dahdi-tools
you have installed (if CentOS has them). If not, the Fedora specs and patches
are here: http://cvs.fedoraproject.org/viewvc/rpms/dahdi-tools/
--
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also provides packages for RHEL5, if those would work.
http://atrpms.net/dist/el5/
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you'll need to patch and compile.
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You need to wait at least 1 second on an incoming POTS line for CID info, add a
wait(1) as the first step on incoming connections.
Thank you and have a nice day,
Anthony Francis
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Arun
If asterisk enters the answered state at any point in the call, then the call
disposition becomes answered.
Thank you and have a nice day,
Anthony Francis
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Szasz Szabolcs
Sent: Tuesday
Where do you get FFA? I have not seen this, what is the minimum version of
Asterisk that you need? Sorry about the questions.
Thank you and have a nice day,
Anthony Francis
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com
Hi,
I have just installed asterisk, I want to send registration request to
192.168.4.3:6090 and the domain should be test1.net
I have added the following line to sip.conf
register = 897...@test1.net:pazzwrd:897...@192.168.4.3:6090
now the problem is that the SIP Request is appearing as
.
Does anyone know which version of libpri and zaptel I should be using? I cannot
find a good reference to this.
Thank you and have a nice day,
Anthony Francis
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asterisk-users
at this
OID
you may need to do export MIBS=+ASTERISK-MIB snmpwalk ...
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to the something unique outside of Asterisk itself, such as as the
external bash process id $$ or the process id combined with the date in
nanoseconds.
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Description
a call from a call file or the AMI using
a local channel.
Channel: Local/s...@sendfax
Exten: number to be dialed
Context: outbound
Priority: 1
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Description
Andy Rosen wrote:
... figure out a good way to log which conference ID that is being used.
The only way I have found to do this is in the events, the conference
enter event has the unique id of the call, which will tie it to the cdr,
and the conference number.
Hope this helps!
Anthony
with the firmware of those phones as newer versions of Aastra phones
(5Xi) work without the modification.
I have several Aastra 480i CT phones on three separate Asterisk 1.6.1.6 on
Fedora 11 (asterisk-1.6.1.6-1.fc11.x86_64) and do not see this problem.
--
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/recording is in the spot for
[,MOH_Class]
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On Wednesday 12 August 2009 08:30:33 am harry R wrote:
Or maybe can suggest another CDR GUI ?
i began work on this a while ago...
http://messinet.com/trac/webcdr+/
it's what i use now, though i'd like to add more features, etc.
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Klaus Darilion wrote:
FYI: I checked the sources and Asterisk does write CDRs only if the call
in answered locally or forwarded to an outgoing channel.
Thus, as workaround I wrapped the extensions behind Dial(Local/...)
regards
klaus
Klaus Darilion schrieb:
Hi!
I just found out
allowing a codec called xxx.xxx.xxx.0/255.255.255.0, which probably
isn't what you want.
Your looking for host=dynamic.
Anthony
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AstriCon 2009 - October 13 - 15 Phoenix
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I agree, I personally do this using the queue events from the AMI. Make
sure you turn on queue events in queues.conf!
Anthony
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http://www.digium.com/
Isn't that requirement a little hypocritical since Asterisk is heavily
corporate sponsored?
Just asking,
Anthony
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:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 4
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
-- Called 180018002662...@tf.voipmich.com
-- SIP/tf.voipmich.com-140f2228 is circuit-busy
--
Anthony - http
???
Thanks and regards
srinivas antarvedi
try module reload res_crypto.so or restart your asterisk servers.
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Tilghman Lesher wrote:
On Thursday 11 June 2009 03:59:01 Steve Howes wrote:
On 11 Jun 2009, at 08:59, BERGANZ François wrote:
In my dialplan, I do s,n,DIAL(…)
If my called phone response and after hangup, asterisk execute the
h,1,…
But, if I the caller hangup at ringing (cancel),
# right now, but there
was a manager bug that was fixed in following versions of asterisk.
the patch that does the fix is simple:
http://cvs.fedoraproject.org:80/viewvc/rpms/asterisk/F-10/0016-Fix-a-reversed-
logic-ast_strlen_zero.patch?revision=1.1view=markup
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the local calls
(incurring no local-toll or long distance charges) which areband a and
band b.
https://messinet.com/trac/telephony-tools/wiki/LocalCallingAreaGrabber
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the
agentcallbacklogin command did.|
I totally agree, I have never seen any example that makes it work. If
somebody shows me how to do it without using Voicemail I will let you know.
Thanks
David
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Mr. David Anthony O'Reilly, B.Sc Comp (Hons)
M.Sc MOB
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Mr. David Anthony O'Reilly, B.Sc Comp (Hons)
M.Sc MOB Postgraduate @ University College Cork, Ireland - M.Sc (Mob) - 2009
Computer Science Graduate of The University of Dublin, Trinity College -
B.Sc (Comp) 2008
Email: oreil...@tcd.ie/d...@student.cs.ucc.ie
Tel
you wanted.
you need to have your script exit with something other than 0 if you'd like to
have AGISTATUS not be SUCCESS.
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note-I use extensions.ael but I am sure any code that is for extensions.conf
will be easily convertable as I love AEL
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Mr. David Anthony O'Reilly, M.Sc (Mob), B.Sc Comp (Hons)
M.Sc MOB Postgraduate @ University College Cork, Ireland - M.Sc (Mob
Anyone knows what should be the configuration of the new solution of
Digium for fax in order to send and receive faxes from PSTN to a fax
machine through an ATA implementing T38 protocol?
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with their DTMF detection on their PRI's.
Anthony
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Anthony Francis wrote:
Jeff LaCoursiere wrote:
On Fri, 17 Apr 2009, Jeff LaCoursiere wrote:
On Fri, 17 Apr 2009, Jeff LaCoursiere wrote:
I went ahead and switched to SIP just for grins, and made sure
dtmfmode=rfc2833 is in the peer config on both sides
into my work
voicemail out of my asterisk box at home.
try setting callprogress=no
by the way, for anyone else, might there be a way to enable/disable
callprogress from the dialplan?
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Bayardo Sanchez wrote:
tollfree calls was working fine but stopped working without any reason
Oh, there's a reason.
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bindaddr = 0.0.0.0
I would set this to the ethernet interface IP address, I believe this may be
your issue.
Registration is only for receiving calls, if you are not seeing information on
the dial, then the phone is not talking to the server. I would make sure of
the settings in the
SIP wrote:
IPKall still exists.
http://www.ipkall.com
No customer service, and the number has to be used every month or you
lose it. But it's there. And free. And good.
N.
Dean Collins wrote:
Does IPKALL still exist?
I am after a free SIP trunk – who is still giving these away
Or you could use the domain feature, where you set a default context
per domain, that overrides the one in the general section.
/Olle
Olle,
That's the point. The SIP context precedence right now is default, peer,
domain. That precedence doesn't make sense.
The context precedence should
It took me a while to understand what you were saying ... more clarity
to your emails!
I was trying to be clear and complete. So many times if you forget to mention
1 thing or another, or are too long, you get non-helpful comments back. But I
will try harder. Right now Asterisk is as
Ok, this is where it gets interesting. Consider the case of a PBX
which has its own MOH source and is talking via Asterisk to another
PBX.
If that PBX wants to put the call on hold while sending its own MOH,
you would probably argue that it should not send a re-INIVTE at all,
but should
Regarding compression with g.729/gsm/etc. and Asterisk
If we convert all the voice files to the corresponding format g.729/gsm/etc.
and we send digits using RFC 3261 and we do not need silence detection, is
there still a need to decompress the media stream ?
If doable how to make
Okay, I am not understanding if I have this correct or not.
I have a requirement to allow guests into a PBX from different domains.
However, I can not allow the guests into the default context because each
domain has its own IVR. So I end up setting the domain context. I also need
to
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