If SIP goes to the same provider then yes. Still I would check a packet
capture for better understanding. BTW, did you try iax debug?
чт, 20 апр. 2017 г. в 19:46, Carlos Chavez :
> On 4/20/17 12:45 AM, Kseniya Blashchuk wrote:
>
> Can it happen that the routes lead the
Google search reveals a fairly dated reference to the same carrier switch
tag message being delivered in a Skype for Business forum thread.
https://social.microsoft.com/Forums/en-US/e25b3198-b5a0-4a43-9328-4a1aff5f6ed0/1800-number-dialing-issue?forum=communicationsservertelephony
On Thu, Apr 20,
I enable full log and run 'core set debug 9' before doing a pair of
tests.
(The full log is easier to grep than the console output.)
Then compare a working vs stocktrans2 side by side.
-JimC
--
James Cloos OpenPGP: 0x997A9F17ED7DAEA6
--
> "DC" == D'Arcy Cain writes:
DC> I did debug 10 and saved the console output into files which I
DC> compared side by side. No material difference.
In that case I'd add more debug statements to apps/app_voicemail.c (in
vm_exec()), including a log at the start of
Il 20/04/2017 18:09, kevin.lar...@pioneerballoon.com ha scritto:
>
> I honestly don't know if you can do what you want without some piece
> of equipment picking up the line. What I would do is get an analog
> line, an analog phone, an analog to sip device (there are many to
> choose from) and a
Il 20/04/2017 17:32, kevin.lar...@pioneerballoon.com ha scritto:
>
> This gets kinda Rube Golberg-ish, but convert the incoming analog line
> to sip, route it through asterisk and have asterisk do its thing
> before converting it back to analog to send to the phone. Only problem
> is you get a lot
On Thursday 20 April 2017 at 21:29:58, Steve Edwards wrote:
> Not an Asterisk question, but...
>
> A bunch of our 8xx numbers started playing this recording when dialed. Our
> provider (Inteliquent) says it's not them.
Where are Inteliquent feeding the calls (assuming they connect instead of
On 2017-04-20 04:07 PM, James Cloos wrote:
I enable full log and run 'core set debug 9' before doing a pair of
tests.
(The full log is easier to grep than the console output.)
Then compare a working vs stocktrans2 side by side.
I did debug 10 and saved the console output into files which I
> From: Fabio Moretti
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Date: 04/20/2017 03:26 PM
> Subject: [asterisk-users] log incoming calls without answering
> Sent by: asterisk-users-boun...@lists.digium.com
>
> Hi,
>
On 4/20/17 2:37 PM, Kseniya Blashchuk wrote:
If SIP goes to the same provider then yes. Still I would check a
packet capture for better understanding. BTW, did you try iax debug?
чт, 20 апр. 2017 г. в 19:46, Carlos Chavez >:
On 4/20/17
> I've already proposed your solution (is the most reasonable) but they
> have more than 60 analogs lines (no faxes) and some of them terminate in
> appliances like alarms, etc, so the solution must not touch in any way
> the connection between the line and his termination: doing a analog to
>
Not an Asterisk question, but...
A bunch of our 8xx numbers started playing this recording when dialed. Our
provider (Inteliquent) says it's not them.
Does anybody know who is playing it and what it means?
--
Thanks in advance,
Fabio Moretti wrote:
Il 20/04/2017 18:09, kevin.lar...@pioneerballoon.com ha scritto:
I honestly don't know if you can do what you want without some piece
of equipment picking up the line. What I would do is get an analog
line, an analog phone, an analog to sip device (there are many to
Hi,
I've some analogic lines and I'm asked if it's possible to program an asterisk
for "checking" the inbound calls without answering them, doing something like
this:
analog line 1 -+-- asterisk
|
\__ analog phone
when a call enter,
Fabio, this doesn't answer your question directly and it's not Asterisk related
in any way, but it's another way to engineer a solution to the problem and I've
seen it done before.
Many analog modems will decode the caller ID on the analog line and provide it
as part of the 'RING' notification
Hmmm.. So if you are sure that the poke packets leave the network interface
(I would still check with tcpdump as well, maybe a firewall issue?) then it
makes sense to check the other side to make sure the poke packets reach
other servers.
I mean with tcpdump you may see if there are incoming
On 2017-04-20 02:33 PM, Fabio Moretti wrote:
Yes, I'll definitely do the test before set up the whole proyect, but
the point basically is: it is possibile for asterisk to log a call
without answering it? How to do it in the dialplan? Or I'm wasting time
because an analog line who enter asterisk
> "FM" == Fabio Moretti writes:
FM> when a call enter, asterisk sense it and store its values (callerid,
FM> date and time, etc) somewhere, but nothing more, people will answer
FM> using the old analog phone. The goal is to have a log of the inbound
FM> calls without
> On 21/04/2017, at 9:33 am, Fabio Moretti wrote:
>
> the point basically is: it is possibile for asterisk to log a call
> without answering it? How to do it in the dialplan? Or I'm wasting time
> because an analog line who enter asterisk is always answered?
Yes.
Hello
in sip.conf I have ;
videosupport=yes
Kind regards.
J.
On 20-04-17 13:09, Marcelo Terres wrote:
I suppose that you enable the video support on sip.conf, right?
Regards,
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
Thank you very much, Marcello. You got it. The point is to restart
.configure AFTER installing these pakcages.
PC
On 04/20/2017 01:13 PM, Marcelo Terres wrote:
Good question.
I am running Asterisk 14 on Ubuntu 16.04 and I had this packages installed:
ii libodbc1:amd64
Good question.
I am running Asterisk 14 on Ubuntu 16.04 and I had this packages installed:
ii libodbc1:amd64 2.3.1-4.1
amd64ODBC library for Unix
ii odbc-postgresql:amd641:09.06.0200-1.pgdg14.04+1
amd64ODBC driver
Thank you for this list, which helps me to be sure that the "good"
packages are installed !
I have checked that all these packages are installed now.
Now, I shall restart the install from the beginning and check again.
Thank you.
PC
On 04/20/2017 01:13 PM, Marcelo Terres wrote:
Good
On Thu, Apr 20, 2017 at 12:02:36PM +0200, Olivier wrote:
> Hello,
>
> I've been tasked to enable automatic Asterisk restart on failure on a
> Jessie platform (running latest Asterisk 13.15.0).
>
> I build a dedicated Jessie VM on which I installed Asterisk from source.
> I configured a couple of
I suppose that you enable the video support on sip.conf, right?
Regards,
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres
On 19 April 2017 at 13:18, Jonas
Basically, adding libsystemd-dev on Jessie before recompiling (./configure,
make, ...) allowed Asterisk to notify systemd it has successufully started.
For reference, please note this feature requires Asterisk 13.12.0 and above.
Thank you very much, Tzafrir, for your help !
--
On Wednesday 19 Apr 2017, D'Arcy Cain wrote:
> Yes and [using something like "1571"] works just fine for us. The problem
> is that we are trying
> to deal with the situation where someone calls themselves from another
> phone (internal or external) to pick up their messages. In every other
>
On Thursday 20 April 2017 at 12:31:14, Atux Atux wrote:
> Hi. thanks a lot for your replies. I did stop the services and i did issued
> the the "chown" and "chmod" commands listed in the guide.
> It is necessary to compile it, instead if using the apt-get version
> What am i missing?
Let's go
Hi. thanks a lot for your replies. I did stop the services and i did issued
the the "chown" and "chmod" commands listed in the guide.
It is necessary to compile it, instead if using the apt-get version
What am i missing?
On Wed, Apr 19, 2017 at 10:47 PM, Antony Stone <
Hi!
The issue did not reproduce with pjsip. As for ppa - somebody recommended
me ppa:sapian/asterisk. Does anybody use it maybe?
вт, 18 апр. 2017 г. в 2:24, Ludovic Gasc :
> Hi,
>
> I recommend you to install from sources, especially because the latest
> Asterisk 13 has
Hello,
I've been tasked to enable automatic Asterisk restart on failure on a
Jessie platform (running latest Asterisk 13.15.0).
I build a dedicated Jessie VM on which I installed Asterisk from source.
I configured a couple of files in /etc/asterisk directory.
I positively checedk that with
root@PBX: /var/www/html $ /etc/init.d/asterisk start
[ ok ] Starting asterisk (via systemctl): asterisk.service.
root@PBX: /var/www/html $ ps aux | grep asterisk
asterisk 1007 0.7 2.3 67128 23748 ?Ssl Apr19 8:49
/usr/sbin/asterisk -U asterisk -G asterisk
root 4186 0.0 0.1
On 2017-04-20 12:23 PM, D'Arcy Cain wrote:
Here is the full dialplan for stocktrans2.
I reduced this to the following and I still have the error.
exten => stocktrans2,1,Verbose(0,Entering extension stocktrans2)
same => n(VoiceMail),Set(CDR(userfield)=VoiceMail)
same =>
On 4/20/17 12:45 AM, Kseniya Blashchuk wrote:
Can it happen that the routes lead the traffic through another
interface? Did you try a packet capture with tcpdump? Do the packets
really leave the usb adapter? Can asymmetric routing be in effect?
Maybe there were some static routes that
On Thursday 20 April 2017 at 18:31:03, Atux Atux wrote:
> root@PBX: /var/www/html $ /etc/init.d/asterisk start
> [ ok ] Starting asterisk (via systemctl): asterisk.service.
I'm somewhat puzzled that your root-user prompt is "$"
instead of the more normal "#", but never mind...
> root@PBX:
On 2017-04-20 05:14 AM, J Montoya or A J Stiles wrote:
This is just screaming "configuration mismatch" -- or, possibly, "latent bug
whereby things parsed in separate places should be treated the same, but are
actually getting treated differently".
I really don't want to be the "my system isn't
On 2017-04-20 12:52 PM, J Montoya wrote:
On Thursday 20 Apr 2017, D'Arcy Cain wrote:
On 2017-04-20 12:23 PM, D'Arcy Cain wrote:
Here is the full dialplan for stocktrans2.
I reduced this to the following and I still have the error.
exten => stocktrans2,1,Verbose(0,Entering extension
On Thursday 20 Apr 2017, D'Arcy Cain wrote:
> On 2017-04-20 12:23 PM, D'Arcy Cain wrote:
> > Here is the full dialplan for stocktrans2.
>
> I reduced this to the following and I still have the error.
>
> exten => stocktrans2,1,Verbose(0,Entering extension stocktrans2)
> same =>
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