Hello
you mean while placing a video call ? What info am I looking for in the
debug output ?
Kind regards.
J.
On 21-04-17 12:28, Marcelo Terres wrote:
Did you try to activate DEBUG and set the verbosity to a higher level
(100?) to check what Asterisk tells you about?
Regards,
Did you try to activate DEBUG and set the verbosity to a higher level
(100?) to check what Asterisk tells you about?
Regards,
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
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:
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
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
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
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 <
On Wednesday 19 April 2017 at 18:48:29, Atux Atux wrote:
> Hi.
> Here is the output of the command
>
> root@pbx: ~ $ find / -name asterisk -exec ls -ld '{}' \;
>
> drwxr-xr-x 3 root root 4096 Apr 19 17:32 /usr/include/asterisk
>
> drwxr-x--- 3 asterisk asterisk 4096 Apr 19 17:32
Hi.
Here is the output of the command
root@pbx: ~ $ find / -name asterisk -exec ls -ld '{}' \;
drwxr-xr-x 3 root root 4096 Apr 19 17:32 /usr/include/asterisk
drwxr-x--- 3 asterisk asterisk 4096 Apr 19 17:32 /usr/lib/asterisk
-rwxr-xr-x 1 root root 9719880 Apr 19 17:27
On Wed, Apr 19, 2017 at 04:44:39PM +0300, Atux Atux wrote:
> hello there. i am running debian 8 in my swerver and i would like to run
> asterisk as non root.
The Asterisk package included with Debian already does that. Why not
have a look at it?
> i did follow the
>
On Wednesday 19 April 2017 at 15:44:39, Atux Atux wrote:
> hello there. i am running debian 8 in my swerver and i would like to run
> asterisk as non root. i did follow the
> https://www.voip-info.org/wiki-Asterisk+non-root without any success.
Did you do the very first step:
hello there. i am running debian 8 in my swerver and i would like to run
asterisk as non root. i did follow the
https://www.voip-info.org/wiki-Asterisk+non-root without any success. when
i issue
root@PBX: ~ $ asterisk -U asterisk -G asterisk
Privilege escalation protection disabled!
See
Hello
using asterisk 1.8.32.3
I am not able to make a call with video support. I do not know what I am
missing to make this video call.
Codec h264 should be supported.
sip*CLI> core show codecs
Disclaimer: this command is for informational purposes only.
It does not indicate anything
Hi,
We use with success on our production the builtin fax in Asterisk 13, based
on spandsp.
We have several lawyers that use this feature each day with no major issues
to my knowledge.
However, we have enabled T38 on the entire chain and we have a carrier that
handles T38 pretty well.
Before
Dear Saint Michael,
I will be grateful if you could introduce me to the Company that
offers the translation service.
I am really interested in google voice.
Sincerely,
Michael Codjoe
On 29 March 2017 at 17:00, wrote:
> Send asterisk-users mailing
On Sat, Apr 8, 2017 at 7:23 AM, Dan Jenkins wrote:
>
> On Fri, Apr 7, 2017 at 9:44 PM, Teijo wrote:
>
>> Hello,
>>
>> I've been using webrtc (Jscommunicator) with Asterisk occasionally. Only
>> problem until now which remained was that if dtls_rekey
Thank you Dan for this information.
Best regards,
Teijo
8.4.2017, 15:23, Dan Jenkins kirjoitti:
On Fri, Apr 7, 2017 at 9:44 PM, Teijo wrote:
Hello,
I've been using webrtc (Jscommunicator) with Asterisk occasionally. Only
problem until now which remained was that if
On Fri, Apr 7, 2017 at 9:44 PM, Teijo wrote:
> Hello,
>
> I've been using webrtc (Jscommunicator) with Asterisk occasionally. Only
> problem until now which remained was that if dtls_rekey was set to the
> value other than 0, call hanged up when using chrome after the time
Hello,
I've been using webrtc (Jscommunicator) with Asterisk occasionally. Only
problem until now which remained was that if dtls_rekey was set to the
value other than 0, call hanged up when using chrome after the time
where dtls_rekey was set.
I suppose that "bad media description" shown
The Asterisk Development Team would like to announce the release of
Asterisk 13.15.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.15.0 resolves several issues reported by the
community and would have not been
The Asterisk Development Team would like to announce the release of
Asterisk 14.4.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 14.4.0 resolves several issues reported by the
community and would have not been
isk Users Mailing List - Non-Commercial Discussion'
Cc: motty.c...@gmail.com
Subject: [asterisk-users] Asterisk 13.13.1 use_callids = yes Extensions ID
as CallerID
Hello, In Master.csv Asterisk is loggin the Company ID set in
Extensions.conf, but I configured logger.conf to log the EXT ID. For
instan
The Asterisk Development Team has announced security releases for
Certified Asterisk
13.13 and Asterisk 13 and 14. The available security releases are released
as versions 13.13-cert3, 13.14.1, and 14.3.1.
These releases are available for immediate download at
We're investigating
--
George Joseph
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
_
-- Bandwidth and Colocation Provided by
Hello, In Master.csv Asterisk is loggin the Company ID set in
Extensions.conf, but I configured logger.conf to log the EXT ID. For
instance, the SRC in the following line should be my ext. number. Does it
make sense? From my extension 4007 I called 78079745, yet in the log below
the first number
I recently upgraded to Asterisk 14.3.0. When playing a SIP file to a
G722 SIP channel (via chan_sip), I get a crash with the following
traceback. This is reproducable:
#0 0x0036fdc30265 in raise () from /lib64/libc.so.6
#1 0x0036fdc31d10 in abort () from /lib64/libc.so.6
#2
Hello,
I'm working on a (PJ)SIP trunking Asterisk machine with which I'm facing
issues with DTMF.
Installed version is 13.14.0.
1. In outbound calls SDP, I'm seeing these kind of lines:
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
I would expect events to range from 0 to 15, not to 16, as
Hello,
2017-03-17 15:05 GMT+01:00 Olivier :
> Hello,
>
> From a 13.14.0 system:
>
> same = n,Verbose(0,1-CALLERID(num-pres) is ${CALLERID(num-pres)})
> same = n,Set(CALLERID(num-pres)=prohib)
> same = n,Verbose(0,2-CALLERID(num-pres) is now ${CALLERID(num-pres)})
>
Replying
The Asterisk Development Team has announced the release of Asterisk 14.4.0-rc1.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 14.4.0-rc1 resolves several issues reported by the
community and would have not been
The Asterisk Development Team has announced the release of Asterisk 13.15.0-rc1.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.15.0-rc1 resolves several issues reported by the
community and would have not been
Hello,
>From a 13.14.0 system:
same = n,Verbose(0,1-CALLERID(num-pres) is ${CALLERID(num-pres)})
same = n,Set(CALLERID(num-pres)=prohib)
same = n,Verbose(0,2-CALLERID(num-pres) is now ${CALLERID(num-pres)})
I would expect to read "2-CALLERID(num-pres) is now prohib" but I get
Hello list,
We've got an Asterisk crash in one of our servers and the core dump showed
following call tree.
Is this anyhow helpful to someone? Seems like a regular RTP / RTCP handling
that lead to a malloc crash
Grateful for any help!
Cheers,
Patrick
Thread 1 (Thread 0x7f8d6b023700 (LWP
On Sun, Mar 12, 2017, at 02:23 PM, Mike Diehl wrote:
> Hi all,
>
> I'm needing to upgrade Asterisk from 10.x to whatever the recommended
> version
> is that will allow me to continue to use Fax For Asterisk.
>
> I don't have many upgrade windows, I'd like to get the most bang for my
> buck,
>
Hi all,
I'm needing to upgrade Asterisk from 10.x to whatever the recommended version
is that will allow me to continue to use Fax For Asterisk.
I don't have many upgrade windows, I'd like to get the most bang for my buck,
but I can't afford to be a beta tester on this server.
The FFA site
Hi all,
I'm needing to upgrade Asterisk from 10.x to whatever the recommended version
is that will allow me to continue to use Fax For Asterisk.
I don't have many upgrade windows, I'd like to get the most bang for my buck,
but I can't afford to be a beta tester on this server.
The FFA site
I've upgraded to to asterisk 11.25.1 (from 1.8). My local asterisk is
showing it is registered with remote asterisk (same version),
But when I try to make a call I get:
iax2 show registry
Host dnsmgr UsernamePerceived Refresh State
192.168.142.1:4569N
Theory: The carrier is not responding with 100 Trying in the expected time.
Hence, Asterisk is sending the INVITE again.
On Wed, Feb 22, 2017 at 1:00 PM,
wrote:
> Send asterisk-users mailing list submissions to
> asterisk-users@lists.digium.com
>
The Asterisk Development Team has announced the release of Asterisk 14.3.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 14.3.0 resolves several issues reported by the
community and would have not been possible
The Asterisk Development Team has announced the release of Asterisk 13.14.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.14.0 resolves several issues reported by the
community and would have not been possible
Hi, my server is running a fresh install of Asterisk 13.13.1 on CentOS 7. My
extensions.conf file was mostly copied from server running Asterisk 1.8.
That being said! If I dial a number and get a busy signal I get the
following error:
-- SIP/voipeer-084b redirecting info has changed,
1. asterisk callerid issue PJSIP Realtime (Zakir Mahomedy)
> 2. Re: asterisk callerid issue PJSIP Realtime (George Joseph)
>
>
> --
>
> Message: 1
> Date: Wed, 1 Feb 2017 13:50:57 +0000 (UTC)
> From: Zakir
quot;
<asterisk-users@lists.digium.com>
Subject: [asterisk-users] asterisk callerid issue PJSIP Realtime
Message-ID: <1998594554.250932.1485957057...@mail.yahoo.com>
Content-Type: text/plain; charset="utf-8"
I recently rolled out a new server with asterisk 14. ?On the Call
On Wed, Feb 1, 2017 at 6:50 AM, Zakir Mahomedy wrote:
> I recently rolled out a new server with asterisk 14.
> On the Called user phone, the caller ID is the same as the Called User.
>
> eg) ext 406 with callerid 406 calls ext 405 ,
>
> on the caller id on the ext 405
I recently rolled out a new server with asterisk 14. On the Called user phone,
the caller ID is the same as the Called User.
eg) ext 406 with callerid 406 calls ext 405 , on the caller id on the ext
405 phone displaying 405.
We are using realtime PJSIP, I set the callerid field in the
ervers were similar in CPU, Memory
>
>
>
> Any support on this matter is appreciated!
>
>
>
> Thanks,
> Motty
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] *On Behalf Of *kambiz sh
ince the upgrade!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Monday, January 30, 2017 9:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13.
On 01/30/2017 at 05:55 PM Motty Cruz wrote:
> Fresh installed CentOS 7.3 and Asterisk 13.13.1. Download Asterisk from here:
> http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-13-current.tar.gz
>
>
>
>
> I continue to see errors like this:
>
> [2017-01-30 08:37:17]
ytle
Sent: Monday, January 30, 2017 9:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13.13.1
>>> On Jan 30, 2017, at 11:55 AM, Motty Cruz motty.c...@gmail.com wrote:
>>> Fresh installed CentOS 7.3 and Asterisk 13.13.1.
>>> On Jan 30, 2017, at 11:55 AM, Motty Cruz motty.c...@gmail.com wrote:
>>> Fresh installed CentOS 7.3 and Asterisk 13.13.1. Download Asterisk from
>>> here:
>>> http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-13-current.tar.gz
>>>
>>> I continue to see errors like this:
>>>
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13.13.1
On Wed, Jan 25, 2017 at 16:00 Olivier <oza.4...@gmail.com> wrote:
What did you exactly upgade ? Asterisk only ? Asterisk and OS ?
How did you installed Asterisk 1.8 and 13 ? From
On Wed, Jan 25, 2017 at 16:00 Olivier wrote:
> What did you exactly upgade ? Asterisk only ? Asterisk and OS ?
> How did you installed Asterisk 1.8 and 13 ? From source or from package ?
>
> I would be curious to see what would happen after downgrading back to 1.8.
>
>
. Asterisk 13.13.1 (Motty Cruz)
>2. Re: Asterisk 13.13.1 (Olivier)
>
>
> --
>
> Message: 1
> Date: Tue, 24 Jan 2017 12:03:05 -0800
> From: "Motty Cruz" <motty.c...@gmail.com>
> To: "'Asterisk Users Mailing List - Non-Commercial Discussi
What did you exactly upgade ? Asterisk only ? Asterisk and OS ?
How did you installed Asterisk 1.8 and 13 ? From source or from package ?
I would be curious to see what would happen after downgrading back to 1.8.
2017-01-24 21:03 GMT+01:00 Motty Cruz :
> Hello, I recently
Hello, I recently upgraded from Asterisk 1.8 to Asterisk 13. Now users are
starting to complaint about packets loss, conversations are choppy!
I don't even know where to start looking! Choppy conversations happened
within users. I am using sip.conf
[1091]
type=friend
context=sip-phone
On Tue, Jan 24, 2017, at 01:41 PM, Dan Cropp wrote:
> Thank you Joshua.
>
> So there is no way to retrieve header information which may come in on
> subsequent packages?
>
> If not, is there any way to make an Attended Transfer following the
> RFC5589?
> https://tools.ietf.org/html/rfc5589
>
>
-users-boun...@lists.digium.com] On Behalf Of Joshua Colp
Sent: Tuesday, January 24, 2017 11:27 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk 14.2.1 PJSIP - is it possible to
retrieve a PJSIP header To field for the SIP OK response to Trying?
On Tue, Jan 24, 2017, at 01
On Tue, Jan 24, 2017, at 01:25 PM, Dan Cropp wrote:
> I place a call into Asterisk (from SIP phone) and the To header does not
> have a tag. Asterisk then sends it's Trying response, still no tag in
> the To header. The phone then replies with OK, this time the To header
> includes a tag.
>
>
I place a call into Asterisk (from SIP phone) and the To header does not have a
tag. Asterisk then sends it's Trying response, still no tag in the To header.
The phone then replies with OK, this time the To header includes a tag.
Is there any way to retrieve this response To header (including
Hi Alejandro,
The documentation about your question is here:
https://wiki.vtiger.com/vtiger6/index.php/PBX_Manager
After a few seconds of read, I think that VTigerAsteriskConnector can run
on a separate server than Asterisk PBX.
VTigerAsteriskConnector connects to Asterisk via Asterisk Manager
Dear, I have Asterisk 1.8 (installed with Elastix 2.4) and I want to
integrate a Vtiger 6.5 server.
In my PBX I have Asterisk 1.8, Java 1.4 and I have not Java Jetty.
What are the requirements in the Asterisk server in order to install the
VtigerAsteriskConnector package and then integrate the
Hello,
Reading this thread, may I ask if you could get this to work ?
Regards
2016-06-29 6:29 GMT+02:00 Annus Fictus :
> hello,
>
> I'm trying to use Asterisk 13.9.1 with Homer SIP Capture Server.
>
> My hep.conf Asterisk configuration is:
>
> [general]
> enabled = yes
>
On Mon, Dec 19, 2016 at 05:10:42PM +0100, Olivier wrote:
> Thanks for the tip:
> changing to permissive mode made it !
>
> Using methods suggested in [1], do you think its possible and worth the
> effort to configure SELinux to work with Asterisk/Systemd in Enforcing mode
> ?
>
> [1]
On Mon, Dec 19, 2016 at 03:54:47PM +0100, Olivier wrote:
> Hello,
>
> For a new project, I'm adapting existing installation script to CentOS 7.
> I must admit I don't understand how to adapt things to systemd.
>
> Here are my questions:
>
> 1. I don't see any systemd sub-directory in
Le 19/12/2016 à 17:10, Olivier a écrit :
2016-12-19 16:11 GMT+01:00 Jean Aunis >:
Le 19/12/2016 à 15:54, Olivier a écrit :
Running systemctl start asterisk fails with :
Dec 19 15:43:08 foobar systemd: PID file
2016-12-19 16:11 GMT+01:00 Jean Aunis :
> Le 19/12/2016 à 15:54, Olivier a écrit :
>
> Hello,
>
> For a new project, I'm adapting existing installation script to CentOS 7.
> I must admit I don't understand how to adapt things to systemd.
>
> Here are my questions:
>
> 1. I
Le 19/12/2016 à 15:54, Olivier a écrit :
Hello,
For a new project, I'm adapting existing installation script to CentOS 7.
I must admit I don't understand how to adapt things to systemd.
Here are my questions:
1. I don't see any systemd sub-directory in asterisk-13.13.1/contrib.
Do you think
Hello,
For a new project, I'm adapting existing installation script to CentOS 7.
I must admit I don't understand how to adapt things to systemd.
Here are my questions:
1. I don't see any systemd sub-directory in asterisk-13.13.1/contrib.
Do you think such directory and matching Makefile target
ok,
thank you... then I´ll take it as it is
cheers,
yves
Am 18.12.2016 um 13:15 schrieb Larry Moore:
Hi,
I haven't found anything definitive however I expect the TSI that is
sent during initial fax call establishment is stored by the receiving
terminal, see pages 28 & 29 of the English
Hi,
I haven't found anything definitive however I expect the TSI that is
sent during initial fax call establishment is stored by the receiving
terminal, see pages 28 & 29 of the English version of the document at
https://www.itu.int/rec/T-REC-T.30-200509-I/en , I expect the header,
which
Hi,
thanks for your answer. Unfortunately this is, what I already know. I
was wondering, why it is possible to set ID and Header for an outgoing
fax (which will then in turn
be inserted via asterisk on top of the transferred "image") , while it
seems to not be possible to get the Header from
The list of options available are listed here
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_FAXOPT
It doesn't appear that a received header is available unless it is
written into the 'headerinfo' variable after it is received, I haven't
checked for this.
From my days
Hi,
I am using asterisk 11.8 in combination with spandsp to send and receive
T38 Faxes. All works fine, but I do not know
how to get the remoteheader from the fax I receive.
When I send a fax, there are Faxopts to set the localstationid and the
headerinfo, but for receiving, there seems to
Hi all,
Has anyone gotten dovecot 2.2 working with the asterisk 11 imap
voicemail system? I can login via thunderbird or telnet to the imap
server just fine.
The below is logged:
[Dec 16 13:35:47] ERROR[17285]: app_voicemail.c:3176 mm_log: IMAP Error:
Can't open mailbox
The bug tracker includes several issues relating to Path (RFC 3327)
support. It is not clear which version actually included the patch and
which versions are working.
Could anybody update these issues in Jira with a brief comment about the
supported versions?
El 10/12/16 a las 10:15, christopher kamutumwa escribió:
Package kernel-devel-3.10.0-327.36.3.el7.x86_64 already installed and
latest version but i still receive the same error
[root@localhost dahdi-linux-complete-2.11.1+2.11.1]# make
make -C linux all
make[1]: Entering directory
In article ,
christopher kamutumwa wrote:
>
> Package kernel-devel-3.10.0-327.36.3.el7.x86_64 already installed and
> latest version but i still receive the same error
>
> [root@localhost
There are inofficial RPMs for CentOS 7 available if you don't want to
mess with compiling: https://www.tucny.com/telephony/asterisk-rpms
Am 10.12.2016 um 15:47 schrieb christopher kamutumwa:
Hello support
am trying to install dahdi on centos 7 and am doing the make ommand and
below is result
On Sat, Dec 10, 2016 at 8:15 AM, christopher kamutumwa <
chriskamutu...@gmail.com> wrote:
> Package kernel-devel-3.10.0-327.36.3.el7.x86_64 already installed and
> latest version but i still receive the same error
>
How about kernel-headers? That's a separate package on my Fedora system,
not
Package kernel-devel-3.10.0-327.36.3.el7.x86_64 already installed and
latest version but i still receive the same error
[root@localhost dahdi-linux-complete-2.11.1+2.11.1]# make
make -C linux all
make[1]: Entering directory `/usr/src/dahdi-linux-
complete-2.11.1+2.11.1/linux'
make -C
On Sat, Dec 10, 2016 at 7:47 AM, christopher kamutumwa <
chriskamutu...@gmail.com> wrote:
> You do not appear to have the sources for the 3.10.0-327.el7.x86_64 kernel
> installed.
You need to install the kernel-devel package to compile kernel modules.
--Greg
--
Hello support
am trying to install dahdi on centos 7 and am doing the make ommand and
below is result any way out this You do not appear to have the sources for
the 3.10.0-327.el7.x86_64 kernel installed.
uname -a gives me below;
Linux localhost.localdomain 3.10.0-327.el7.x86_64 #1 SMP Thu Nov
The Asterisk Development Team has announced security releases for Asterisk
11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available
security releases are released as versions 11.25.1, 13.13.1, 14.2.1,
11.6-cert16, and 13.8-cert4.
These releases are available for immediate download at:
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
On Sun, Dec 04, 2016 at 08:00:45PM +0100, Mr Dini wrote:
> Hi,
>
> I tried to run the make progdocs, but the first time, it said, I have no
> doxygen installed. So I compiled the latest release and reconfigure the
> asterisk. And after it, ut sucessfully started to build the docs. But it
> took a
On Tue, Dec 06, 2016 at 06:35:07PM +0200, christopher kamutumwa wrote:
> am new to asterisk and am trying to "make all". or dahdi install but in
> only reach this stage below on centos 6.8 . Any idea how to resolve or
> bypass this
>
>
> configure: creating ./config.status
> ./configure: line
am new to asterisk and am trying to "make all". or dahdi install but in
only reach this stage below on centos 6.8 . Any idea how to resolve or
bypass this
configure: creating ./config.status
./configure: line 18858: cannot create temp file for here-document: No such
file or directory
configure:
Okay, in this case will create a ticket tomorrow.
Thanks for Your help!
On Dec 4, 2016 10:25 PM, "Joshua Colp" wrote:
> On Sun, Dec 4, 2016, at 05:13 PM, Mr Dini wrote:
> > No, the disable-xmldoc doesn't disable the whole doc creating procedure.
> >
> > Is there a way to
On Sun, Dec 4, 2016, at 05:13 PM, Mr Dini wrote:
> No, the disable-xmldoc doesn't disable the whole doc creating procedure.
>
> Is there a way to disable it completely?
>
> Regarding the issue... Of course, I Can open a ticket, just I don't know
> about what exactly. I want to compile it without
No, the disable-xmldoc doesn't disable the whole doc creating procedure.
Is there a way to disable it completely?
Regarding the issue... Of course, I Can open a ticket, just I don't know
about what exactly. I want to compile it without doc generate to make the
asterisk module loads up fine.
On
On Sun, Dec 4, 2016, at 03:00 PM, Mr Dini wrote:
> Hi,
>
> I tried to run the make progdocs, but the first time, it said, I have no
> doxygen installed. So I compiled the latest release and reconfigure the
> asterisk. And after it, ut sucessfully started to build the docs. But it
> took a lot of
Hi,
I tried to run the make progdocs, but the first time, it said, I have no
doxygen installed. So I compiled the latest release and reconfigure the
asterisk. And after it, ut sucessfully started to build the docs. But it
took a lot of time, So finally I aborted the process...
Is there a way to
Okay, thanks for the info! Will recompile it with corrected doc support and
see what happens.
On Dec 2, 2016 3:36 PM, "Joshua Colp" wrote:
> On Fri, Dec 2, 2016, at 10:27 AM, Mr Dini wrote:
> > Hi,
> >
> > I compiled the asterisk 14.0.2 to my ARMv5 NAS, however I just have
> >
On Fri, Dec 2, 2016, at 10:27 AM, Mr Dini wrote:
> Hi,
>
> I compiled the asterisk 14.0.2 to my ARMv5 NAS, however I just have
> enough
> time to test it now.
>
> But with the default config (I only edited the http.conf), it won't
> start,
> but gives the following:
>
> Sorcery registered
Hi,
I compiled the asterisk 14.0.2 to my ARMv5 NAS, however I just have enough
time to test it now.
But with the default config (I only edited the http.conf), it won't start,
but gives the following:
Sorcery registered wizard 'bucket'
Sorcery registered wizard 'bucket_file'
Parsing
Hey All,
Slight interlude from your regularly scheduled programming.
For any interested, I will be giving a web broadcast today about
Asterisk 14 and what's new with Asterisk since the 13 release. For
those of you that aren't aware, I'm responsible for day to day
management of the Asterisk
On Wednesday 30 Nov 2016, Emiliano Vazquez wrote:
> i'm using gammu[1] with a 3g dongle and my own chip with my preffer
> provider. It can send over 700 every our and receive to. I don't know if
> you need asterisk and sms in the same way but with this tool you can make
> everything. It has python
i'm using gammu[1] with a 3g dongle and my own chip with my preffer
provider. It can send over 700 every our and receive to. I don't know if
you need asterisk and sms in the same way but with this tool you can make
everything. It has python tools to.
Best regards.
Emiliano.
[1]
Yes, it works !
Thanks :-)
Michele
On 30/11/2016 10:19, Jonathan H wrote:
> I think it might be related to this?
> https://issues.asterisk.org/jira/browse/ASTERISK-26391
>
> I think I remember having to edit logger.conf - this is what mine
> looks like now:
> console => notice,warning,error
>
I think it might be related to this?
https://issues.asterisk.org/jira/browse/ASTERISK-26391
I think I remember having to edit logger.conf - this is what mine
looks like now:
console => notice,warning,error
messages => notice,warning,error
Try that, restart asterisk and see if it works :)
On 30
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