On 7/6/2023 7:08 PM, John Covici wrote:
Hi. I have run into a problem compiling dahdi-linux in kernel
6.1.0-10. Apparently there was a change, so I found a patch to fix
stdbool.h but now I have an implicit declaration of
pci_alloc_consistent in drivers/dahdi/wct4xxp/base.c I don't see any
other
Hi. I have run into a problem compiling dahdi-linux in kernel
6.1.0-10. Apparently there was a change, so I found a patch to fix
stdbool.h but now I have an implicit declaration of
pci_alloc_consistent in drivers/dahdi/wct4xxp/base.c I don't see any
other references to that name anywhere. I am u
On Thu, Jun 8, 2023 at 9:41 AM Yves wrote:
> Hello everyone.
> I allow myself to submit a problem that I can not solve with my VOIP
> provider Orange in France
>
> [2023-06-08 13:19:03] ERROR[185091]:
> res_pjsip/pjsip_configuration.c:1044 from_user_handler: Error
> configuring endpoint 'Biv_Sort
Hello everyone.
I allow myself to submit a problem that I can not solve with my VOIP
provider Orange in France
[2023-06-08 13:19:03] ERROR[185091]:
res_pjsip/pjsip_configuration.c:1044 from_user_handler: Error
configuring endpoint 'Biv_Sortie' - 'from_user' field contains invalid
character '
hi,
i have 2 queues
- queue1
- queue2
1 agent is in both queues
queue strategy is rrmemory
i have 2 calls waiting
call from 12:00 in queue1 from number 777
call from 12:05 in queue2 from number 666
at 12:10 agent is free for next call
i have problem in that newer call (call from 12:05) from q
Any help?
Do I need to post my issue to dev ?
On Wed, Jul 14, 2021 at 10:55 AM SAMPro
wrote:
> Hi
> I need to check the return value of a sub, the sub may return empty so I
> need to check for that. If the return value isn't empty set another
> variable (ARG1) . This is the code I've used in ext
Hi
I need to check the return value of a sub, the sub may return empty so I
need to check for that. If the return value isn't empty set another
variable (ARG1) . This is the code I've used in extension.conf, but didn't
work (the CLI log is after the code).
*Extension.conf:*
[macro-dial]
same => 1
Is `test` your default context (line context= in sip.conf)?
If it is not, then try setting context=test in sip.conf and reload it.
On Fri, Jul 17, 2020 at 8:34 AM John Kiniston
wrote:
> I've got this setup in a test context.
>
> [test]
> exten => s,hint,SIP/7124
> exten => s,1,NoOP(Options to
I've got this setup in a test context.
[test]
exten => s,hint,SIP/7124
exten => s,1,NoOP(Options to $EXTEN)
same => n,Hangup()
exten => _x.,hint,SIP/7124
exten => _X.,1,NoOP(Options to $EXTEN)
same => n,Hangup()
exten => Anonymous,hint,SIP/7124
exten => Anonymous,1,NoOP(Options to $EXTEN)
sa
In article ,
John Kiniston wrote:
>
> I'm implementing a SBC with my Asterisk PBX but the keeps disabling the
> trunk group I've configured and I think it may be because Asterisk is
> returning a 4r04 to the OPTIONS.
>
> I've created a test context and have put in a wildcard pattern match to try
Hey John,
In one installation I have, we use several monitoring tools (nagios based
and custom scripts based) and we have the following:
; Reply OK to SIP:OPTIONS
[public]
exten => s,1,Wait(1)
same => n,Hangup
: For Nagios
exten => nagios,1,Wait(1)
same => n,Hangup
NOTES:
1- We have context=pub
I'm implementing a SBC with my Asterisk PBX but the keeps disabling the
trunk group I've configured and I think it may be because Asterisk is
returning a 4r04 to the OPTIONS.
I've created a test context and have put in a wildcard pattern match to try
and catch those options but it doesn't seem to
On Wed, 3 Jun 2020, Fourhundred Thecat wrote:
On 2020-06-03 17:21, Steve Edwards wrote:
How about:
syslog.local0 = error,verbose,warning
no debugging detail.
syslog.local0 = debug,error,verbose,warning
include debugging detail.
current
> On 2020-06-03 17:21, Steve Edwards wrote:
How about:
syslog.local0 = error,verbose,warning
no debugging detail.
syslog.local0 = debug,error,verbose,warning
include debugging detail.
currently, the above has no effect on logging.
As I
On Wed, 3 Jun 2020, Fourhundred Thecat wrote:
On 2020-06-03 12:18, Tony Mountifield wrote:
In article <88f96e46-e6bb-a7ef-bebb-5588ef6cd...@gmx.ch>,
However, the conversation would then be: should both logging types include
line number and function? should both logging types omit them? should
i
> On 2020-06-03 12:18, Tony Mountifield wrote:
In article <88f96e46-e6bb-a7ef-bebb-5588ef6cd...@gmx.ch>,
However, the conversation would then be: should both logging types include
line number and function? should both logging types omit them? should
it be a configuration option in logger.conf wh
In article <88f96e46-e6bb-a7ef-bebb-5588ef6cd...@gmx.ch>,
Fourhundred Thecat <400the...@gmx.ch> wrote:
> > On 2020-06-02 17:48, Tony Mountifield wrote:
> > In article <94191802-6c9c-bdab-615b-001786a2a...@gmx.ch>,
> > Fourhundred Thecat <400the...@gmx.ch> wrote:
> >> > On 2019-11-16 03:29, Fourh
> On 2020-06-02 17:48, Tony Mountifield wrote:
In article <94191802-6c9c-bdab-615b-001786a2a...@gmx.ch>,
Fourhundred Thecat <400the...@gmx.ch> wrote:
> On 2019-11-16 03:29, Fourhundred Thecat wrote:
case LOGTYPE_SYSLOG:
snprintf(buf, size, "%s[%d]%s: %s:%d in %s: %s",
In article <94191802-6c9c-bdab-615b-001786a2a...@gmx.ch>,
Fourhundred Thecat <400the...@gmx.ch> wrote:
> > On 2019-11-16 03:29, Fourhundred Thecat wrote:
> > Hello,
> >
> > I am logging directly into file and also to syslog.
> > Here is snippet from my /etc/asterisk/logger.conf:
> >
> > mess
> On 2019-11-16 03:29, Fourhundred Thecat wrote:
Hello,
I am logging directly into file and also to syslog.
Here is snippet from my /etc/asterisk/logger.conf:
messages => notice,warning,error,verbose
syslog.local0 => notice,warning,error,verbose
But the logs look different:
VERBOSE[7
Hello,
I am logging directly into file and also to syslog.
Here is snippet from my /etc/asterisk/logger.conf:
messages => notice,warning,error,verbose
syslog.local0 => notice,warning,error,verbose
But the logs look different:
VERBOSE[7609][C-0013] pbx.c:
NOTICE[3042] chan_sip.c: P
On Mon, Oct 7, 2019, at 10:23 AM, John Covici wrote:
> hmmm, is asterisk 16 long term support? I thought only the od
> numbered releases were long term support.
Asterisk 13 and 16 are both LTS releases. The upcoming 17 will be a standard
release. You can always consult the wiki[1] to know what a
hmmm, is asterisk 16 long term support? I thought only the od
numbered releases were long term support.
On Mon, 07 Oct 2019 08:02:51 -0400,
George Joseph wrote:
>
> [1 ]
> [2 ]
> Oh, I forgot to mention that Asterisk 15 went End-Of-Life last Thursday. :)
> You should use Asterisk 16.
>
> O
Oh, I forgot to mention that Asterisk 15 went End-Of-Life last Thursday.
:) You should use Asterisk 16.
On Mon, Oct 7, 2019 at 5:58 AM George Joseph wrote:
>
>
> On Fri, Oct 4, 2019 at 1:19 PM John Covici wrote:
>
>> Hi. I am trying to install asterisk 15.7.4 from git onto a Debian 10
>> sys
Hi. I am trying to install asterisk 15.7.4 from git onto a Debian 10
system and I am running into the following problem. I need to install
meetme (I know its old), and I have dahdi installed and the configure
script answers yes to all the edahdi questions, but the app_meetme
says depends on dahdi
>So the new install is coming along. I hooked up the new box for a couple of
>hours and got a bunch more problems worked out. And yet some still remain. I
>have this subroutine I call occasionally:
>.
>.
>.
>Also, when I installed asterisk it did not set itself up to start when the
>machine boot
Hello Jean-Denis,
Sunday, March 3, 2019, 11:28:02 AM, you wrote:
>> But the third line which should print those 80 characters back to the screen
>> prints an empty string. What might I be doing wrong. It's worked from
>> version 2 or 3 through 13 but it seems to be broken in 16.
> This looks
Le 02/03/2019 à 18:10, Ira a écrit :
> exten => 1,1,set(DB(forwards/calls)=${home_in})
>same => n,set(DB(forwards/number)=1)
>same => n,verbose(${DB(forwards/calls)})
>same => n,return
>
> I can see the code running on the console and it prints out the first line
> with ${home_in} re
Hello Ira,
So the new install is coming along. I hooked up the new box for a couple of
hours and got a bunch more problems worked out. And yet some still remain. I
have this subroutine I call occasionally:
exten => 1,1,set(DB(forwards/calls)=${home_in})
same => n,set(DB(forwards/number)=1)
Hi. I am having a problem when trying to receive calls via en E1
from Telmex using MFC/R2 (MX Variant). Outgoing calls are fine. We
are using a PBXact system with a Digium TE420 (5th Gen) card. Here is a
log from the call:
[10:46:37:707] [Thread: 140631230322432] [Chan 1] - Call start
hi,
quite unlikely (besides of an defect) that the behaviour of your
AudioCodes or Asterisk changed "from alone"... something must have changed.
What does the logs say (from asterisk... do you see register-events? and
from you AudioCodes?)
The AudioCodes Devices can export and restore their con
All;
I have an AudioCodes MP-114 four FXS ATA that recently stopped
registering to my PBX. I'm pulling my hair out here trying to figure out the
root cause without much success. Does anyone have a sample config file that
I could use as a sample? Any insight at all would be greatly appreciated.
Luca Bertoncello schrieb:
> But if I try to call another VoIP-phone it rings but no voice will be
> transferred...
Got it!
A "little" firewall problem... :(
Regards
Luca Bertoncello
(lucab...@lucabert.de)
--
_
-- Bandwidth an
Tzafrir Cohen schrieb:
> This means that you have configured a dahdi channel in
> /etc/asterisk/chan_dahdi.conf . The default configuration does not
> include one. Do you have any DAHDI device on the system?
I think not...
> If /dev/dahdi/channel itself does not exist, it means that the
> kerne
On Thu, Feb 15, 2018 at 06:55:16PM +0100, Luca Bertoncello wrote:
> Hi again!
>
> I tried to attach two VoIP-phones to my new Asterisk 13.14.1 on a Banana PI
> with Armbian/Debian 9.
>
> First test was to call a test service that say the time. Works!
> Second test was to record my voice and play
Hi again!
I tried to attach two VoIP-phones to my new Asterisk 13.14.1 on a Banana PI
with Armbian/Debian 9.
First test was to call a test service that say the time. Works!
Second test was to record my voice and play it again. Works!
Third test was to call the other VoIP-phone. It does NOT work..
Harel Cohen schrieb:
> Is the Sophos a home router or professional one? In many cases what home
Of course the professional firewalls (we have two Sophos in Cluster, to
manage our two SDSLs)
> router does by default needs to be configured manually on professional one.
> E.G. a home router will a
Hi,
Is the Sophos a home router or professional one? In many cases what home
router does by default needs to be configured manually on professional one.
E.G. a home router will allow outgoing sessions and create a return path by
default where professional one won't.
Two things I would look for:
1.
the issue is quiet sure codec based:
[Oct 24 15:32:41] NOTICE[3731]: channel.c:4280 __ast_read: Dropping
incompatible voice frame on SIP/messagenet-028e of format gsm since our
native format has changed to 0x8 (alaw)
shorter:
Dropping incompatible voice frame on SIP/messagenet-028e of fo
Luca Bertoncello schrieb:
Hallo again
> I configured an user for my mobile phone and I can call, but as soon
> as the other party answer, I get this error in Log:
>
> [Oct 24 15:32:41] NOTICE[3731]: channel.c:4280 __ast_read: Dropping
> incompatible voice frame on SIP/messagenet-028e of
You should try another SIP client, just to check it. (Zoiper or
cSipSimple, for example).
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 24 October 2017 at 14:42, Luca
Hi list!
I use Asterisk 1.8.30.0 on an OpenWRT-Router (I know, not the last
version, but I can't upgrade).
It always runned very well, and it runs very well with our home
phones, too, but now I have problems using the native Android
SIP-Client...
I configured an user for my mobile phone a
Hello,
I operate an Asterisk server (v11.13.1) on Debian stable, and it's
rock-solid. The other day, however, I accidentally upgraded the
kernel from the stable 3.16.0 to 4.9.0. Subsequently, audio stopped
working.
Below you can find my analysis while running the 4.9.0 kernel. 888
is a simply Ech
> All;
> I have a problem with regards to “re-parking” calls and I was
> hoping someone could shed some light on the topic. Consider this
scenario:
>
> (1) An inbound call comes in and the attendant answers it
> (2) The attendant places the call on hold and the caller is sent to
> extension
All;
I have a problem with regards to "re-parking" calls and I was hoping
someone could shed some light on the topic. Consider this scenario:
(1) An inbound call comes in and the attendant answers it
(2) The attendant places the call on hold and the caller is sent to
extension 701
(3) Bl
All;
I have a problem with regards to "re-parking" calls and I was hoping
someone could shed some light on the topic. Consider this scenario:
(1) An inbound call comes in and the attendant answers it
(2) The attendant places the call on hold and the caller is sent to
extension 701
(3) Bl
On Fri, Oct 28, 2016 at 02:07:24PM +0200, Jonas Kellens wrote:
> I use PHP 5.6.27.
>
> So I should be looking inside php.ini ?
Web search: php self signed certificate fsockopen
User contributed notes in [1] or the example in [2]
1) http://php.net/manual/en/function.fsockopen.php
2)
http://
On 26-10-16 23:24, Stefan Tichy wrote:
On Wed, Oct 26, 2016 at 04:57:15PM +0200, Jonas Kellens wrote:
if it is indeed manager.conf that I need to edit then the problem is
that I see no param : tlsdontverifyserver=yes
A comment copied from sip.conf.sample:
"If set to yes, don't verify the server
On Wed, Oct 26, 2016 at 04:57:15PM +0200, Jonas Kellens wrote:
> if it is indeed manager.conf that I need to edit then the problem is
> that I see no param : tlsdontverifyserver=yes
A comment copied from sip.conf.sample:
"If set to yes, don't verify the servers certificate when acting as a client.
On 26-10-16 15:03, Dan Jenkins wrote:
On Wed, Oct 26, 2016 at 1:46 PM, Jonas Kellens
mailto:jonas.kell...@telenet.be>> wrote:
Hello
I keep getting the following error when trying to connect to the
Asterisk server using AMI :
$socket = fsockopen("tls://11.22.33.44
On Wed, Oct 26, 2016 at 1:46 PM, Jonas Kellens
wrote:
> Hello
>
>
> I keep getting the following error when trying to connect to the Asterisk
> server using AMI :
>
> $socket = fsockopen("tls://11.22.33.44","5039", $errno, $errstr, 5);
>
> Erorr on CLI :
>
> [Oct 26 14:38:19] ERROR[2992]: tcptls.
Hello
I keep getting the following error when trying to connect to the
Asterisk server using AMI :
$socket = fsockopen("tls://11.22.33.44","5039", $errno, $errstr, 5);
Erorr on CLI :
[Oct 26 14:38:19] ERROR[2992]: tcptls.c:609 handle_tcptls_connection:
Problem setting up ssl connection: er
I saw this on the bug list first and sent a reply, but for the archives
I'll copy it here, too.
REMAINDER() calls libm's remainder(3) or remainderl(3), infix % calls
fmod(3) or fmodl(3).
remainder(3) is defined to round the quotient to the nearest int (always
using round-to-even, notsithstanding
Yes! That's the one. Thank you. That's a good workaround.
The following test dialplan shows the bug (feature?)
exten => 7,1,Verbose(Context: ${CONTEXT} Exten:${EXTEN})
same => n,Set(seconds=57)
same => n,While($[${seconds} <= 400]);
same => n,Set(minutes=$[FLOOR(${seconds} / 60)])
All I can tell you is where -3 comes from.
>From http://www.voip-info.org/wiki/view/Asterisk+Expressions :
REMAINDER(x,y) computes the remainder of dividing x by y. The return value
is x - n*y, where n is the value x/y, rounded to the nearest integer. If
this quotient is 1/2, it is rounded to the n
I'm not mathematically gifted, but shouldn't 957%60 be 15 remainder 57?
Google and my desktop calculator certainly think so.
So where am I going wrong here? The following code
exten => 7,1,Verbose(Context: ${CONTEXT} Exten:${EXTEN})
same => n,Set(myNum=957)
same => n,Set(sec=$[REMAINDER
i'm using Asterisk 1.6.2.9-2+squeeze12
2016-06-30 22:14 GMT+02:00 Richard Mudgett :
>
>
> On Thu, Jun 30, 2016 at 3:00 PM, nik600 wrote:
>
>> Dear all
>>
>> i'm creating an outgoing call to number xxx with this command:
>>
>> http://host:port/mxml?action=Originate&Channel=Local/xxx@to-external
>
On Thu, Jun 30, 2016 at 3:00 PM, nik600 wrote:
> Dear all
>
> i'm creating an outgoing call to number xxx with this command:
>
> http://host:port/mxml?action=Originate&Channel=Local/xxx@to-external
> &Exten=testDTMF&Context=cRETEUNICA&Priority=1
>
> wich points correctly to this portion of dialpl
Dear all
i'm creating an outgoing call to number xxx with this command:
http://host:port/mxml?action=Originate&Channel=Local/xxx@to-external
&Exten=testDTMF&Context=cRETEUNICA&Priority=1
wich points correctly to this portion of dialplan:
[cRETEUNICA]
exten => testDTMF,1,Answer
exten => testDT
Hi list!
I installed Hylafax on a Ubuntu-Server 14.04.
On this server runs Asterisk 11.7.0, too and it was configured like my own
Asterisk server at home, but it does not work... :(
So, I configured Asterisk to connect to Deutsche Telekom and it does!
Then I configured iaxmodem to speak with the
On 2016-02-17 16:28, Richard Mudgett wrote:
On Wed, Feb 17, 2016 at 5:56 PM, Ernie Dunbar
wrote:
On 2016-02-17 15:32, Richard Mudgett wrote:
On Wed, Feb 17, 2016 at 5:15 PM, Ernie Dunbar
wrote:
Hi everyone.
We have an Asterisk server running Debian Squeeze, with Asterisk
v1.8.13.1 (basicall
On Wed, Feb 17, 2016 at 5:56 PM, Ernie Dunbar
wrote:
> On 2016-02-17 15:32, Richard Mudgett wrote:
>
>> On Wed, Feb 17, 2016 at 5:15 PM, Ernie Dunbar
>> wrote:
>>
>> Hi everyone.
>>>
>>> We have an Asterisk server running Debian Squeeze, with Asterisk
>>> v1.8.13.1 (basically, the Debian Stable
On 2016-02-17 15:32, Richard Mudgett wrote:
On Wed, Feb 17, 2016 at 5:15 PM, Ernie Dunbar
wrote:
Hi everyone.
We have an Asterisk server running Debian Squeeze, with Asterisk
v1.8.13.1 (basically, the Debian Stable version for Squeeze, but
with some minor source code changes specific to our s
On Wed, Feb 17, 2016 at 5:15 PM, Ernie Dunbar
wrote:
> Hi everyone.
>
> We have an Asterisk server running Debian Squeeze, with Asterisk v1.8.13.1
> (basically, the Debian Stable version for Squeeze, but with some minor
> source code changes specific to our site). We're trying to upgrade to
> 11.
Hi everyone.
We have an Asterisk server running Debian Squeeze, with Asterisk
v1.8.13.1 (basically, the Debian Stable version for Squeeze, but with
some minor source code changes specific to our site). We're trying to
upgrade to 11.13.1 (The Debian Stable version for Jessie), but I've run
int
On Tue, Sep 1, 2015 at 2:02 AM, Brendan Ord wrote:
> Hello,
>
>
>
> This is a problem with my Cisco CUBE (2811), so apologies for this being
> kind of off-topic. It is acting as a border for my Asterisk 13 server
> though J
>
>
>
> Rather than re-type the details of my problems, I have a post in
Hello,
This is a problem with my Cisco CUBE (2811), so apologies for this being kind
of off-topic. It is acting as a border for my Asterisk 13 server though :)
Rather than re-type the details of my problems, I have a post in the Cisco
community with running-configs and various debugs attached.
On Wednesday 15 Jul 2015, Luca Bertoncello wrote:
> But it seems, that I found the problem, adding:
>
> disallow=all
> allow=g729
>
> to the configuration of the peer for this number...
You need the following;
disallow=all
allow=alaw
in the configuration for *every* device. There is literally
jg schrieb:
> How is the 4th phone configured?
It's not a phone, just a number routed on a phone that receives calls for
other number, too (without any problem).
> You could also enable SIP debugging to get more information about the
> problem.
I already set core set debug 42 and core set verb
I have 4 numbers on my Asterisk 1.8.
3 work perfectly, the 4.th not.
I'm not sure, when it finish to work, since a month ago it runs without any
problem...
Well, if I will be called on this number I can't hear anything and in
Asterisk I see these:
[Jul 15 18:59:55] WARNING[8752]: channel.c:506
Hi list!
I have 4 numbers on my Asterisk 1.8.
3 work perfectly, the 4.th not.
I'm not sure, when it finish to work, since a month ago it runs without any
problem...
Well, if I will be called on this number I can't hear anything and in
Asterisk I see these:
[Jul 15 18:59:55] WARNING[8752]: channel
On Mon, Jun 8, 2015 at 9:56 AM, Igor Potjevlesch wrote:
> Hello!
>
> I've got a little problem with Asterisk (11.14.1), the voicemessages are
> kinda limited to 40 seconds (average) aproximately; because when a message
> reach this long I got a cut in the file (*.wav) after I got this message:
>
Hello!
I've got a little problem with Asterisk (11.14.1), the voicemessages are
kinda limited to 40 seconds (average) aproximately; because when a message
reach this long I got a cut in the file (*.wav) after I got this message:
WARNING[15035][C-21ef]: format_wav_gsm.c:418 wav_read: Short rea
Hi again!
I decided, "just for fun", to install Asterisk on a server of mine (available
in Internet) and to log on my mobile phone on this server.
This Server communicate with my Asterisk at home and if I call a phone at
home from my mobile phone (logged on the Asterisk on the second server), it
ricky gutierrez schrieb:
> compilation problems with the module srtp , check the module
>
> module show like srtp
Now available on OpenWRT... :(
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
--
_
-- Bandwidth and Colocation
2015-06-05 14:29 GMT-06:00 Luca Bertoncello :
> I think it is a problem on Asterisk for OpenWRT... :(
>
> Regards
> Luca Bertoncello
> (lucab...@lucabert.de)
>
compilation problems with the module srtp , check the module
module show like srtp
--
rickygm
http://gnuforever.homelinux.com
--
___
ricky gutierrez schrieb:
> Hi lucas , dou you try this:
>
> https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial
Tested right now.
Same problem...
I think it is a problem on Asterisk for OpenWRT... :(
Regards
Luca Bertoncello
(lucab...@lucabert.de)
--
_
2015-06-05 12:21 GMT-06:00 Luca Bertoncello :
> Hi list!
>
> I'm trying to configure my Asterisk to accept SIP-TLS connections, too.
>
> I followed this HowTo:
>
> http://remiphilippe.fr/sips-on-asterisk-sip-security-with-tls/
>
> But as soon I try to connect to my Asterisk using SIP-TLS I
Hi list!
I'm trying to configure my Asterisk to accept SIP-TLS connections, too.
I followed this HowTo:
http://remiphilippe.fr/sips-on-asterisk-sip-security-with-tls/
But as soon I try to connect to my Asterisk using SIP-TLS I get on
Asterisk-CLI:
== Problem setting up ssl connection
> On 22May, 2015, at 03:51, Jonas Kellens wrote:
>
> Realtime seems to be loaded :
>
> *CLI> realtime mysql status
> general configured for asterisk on socket file /var/lib/mysql/mysql.sock with
> username asterisk.
> MyAsteriskDB connected to MyAsteriskDB@127.0.0.1, port 3306 with username
Hello
I have already several Asterisk servers running with similar
configuration, but now I stumble into a problem.
I have mysql configuration res_config_mysql.conf :
[MyAsteriskDB]
dbhost = 127.0.0.1
dbname = MyAsteriskDB
dbuser = astadmin
dbpass = mysecret
dbport = 3306
dbsock = /var/lib/my
hello every body,
i want to have h323 trunk between cisco 2800 and asterisk 11.13.1 with
ooh323 module. i configured both side and have successful call from cisco
to asterisk. but when call comes from asterisk to cisco, my phone rings but
no audio is heard and call is disconnected after 5 second.
Thanks John,
At first got an error using MySQL-asterisk, but then I removed /etc/ ini
files and used the DSN in /usr/local/etc/odbc.ini, that did the trick for
isql. I must have created the files /etc/ while following a guide online.
Nice!
After some meddling with the Asterisk conf files to have
I notice you have MySQL-asterisk as your definition in your odbc.ini but
you are trying to connect to simply 'MySQL' with your 'isql' command.
Does isql work with 'MySQL-asterisk' as the DSN instead of simply 'MySQL' ?
I have machines that use /etc/odbc.ini and machines that use
/usr/local/etc/od
Hello,
I'm stuck with getting cdr records stored in MySql database. I have a
working realtime environment and have verified that the db connection works
fine when used via res_config_mysql.conf. I'd appreciate Your help on how
to get the odbc connector working as I think there's something wrong wi
If I remember correctly, 9.x firmware dropped UDP support altogether.
On Thu, Jan 22, 2015 at 4:31 AM, Jordan Cook - Gyron Networks <
jordan.c...@gyron.net> wrote:
> > Apparently this is a known problem past v8 firmware:
> >
> http://kaa.kiev.ua/blog/asterisk-and-cisco-7945g-after-sip-firmware-up
> Apparently this is a known problem past v8 firmware:
> http://kaa.kiev.ua/blog/asterisk-and-cisco-7945g-after-sip-firmware-update-
> version-9/
I've done some more playing about and what I've noticed is that even when using
TCP SIP on the 8.x Firmware conferencing doesn’t work - making it use U
Apparently this is a known problem past v8 firmware:
http://kaa.kiev.ua/blog/asterisk-and-cisco-7945g-after-sip-firmware-update-version-9/
On Tue, Jan 20, 2015 at 11:16 AM, Jordan Cook - Gyron Networks <
jordan.c...@gyron.net> wrote:
> > Next step is packet capture to see if there is a clue as t
> Next step is packet capture to see if there is a clue as to the cause of the
> failure in the SIP signalling.
Right, I see the following when running SIP Debug. Looks to me like the phones
are expecting the server to do the conference mixing, which I guess it would do
in CallManager?
<--- SIP
Next step is packet capture to see if there is a clue as to the cause of
the failure in the SIP signalling.
On Tue, Jan 20, 2015 at 10:41 AM, Jordan Cook - Gyron Networks <
jordan.c...@gyron.net> wrote:
> We were using G722 - I thought similarly and tried a call with alaw. Same
> problem occurred
We were using G722 - I thought similarly and tried a call with alaw. Same
problem occurred, any other ideas?
> I'm willing to bet you are forcing the use of G729. 7940 and 7960 phones can
> only do a single G729 channel, and if you require G729 for the second leg of a
> conference, it will fail.
I'm willing to bet you are forcing the use of G729. 7940 and 7960 phones
can only do a single G729 channel, and if you require G729 for the second
leg of a conference, it will fail.
On Tue, Jan 20, 2015 at 10:03 AM, Jordan Cook - Gyron Networks <
jordan.c...@gyron.net> wrote:
> Possibly slight
Possibly slightly off topic, has anyone ever had Cisco 79xx Series phones come
up with "cannot complete conference" errors when trying to conference two calls
together?
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All,
I have a weird situation here and haven't been able to turn up any useful
information in searches, so I thought I'd post to the list.
Essentially, I have a customer who wants us to forward some of their calls
to various cell phones. Normally, I'd use FollowMe() for this (that's how
most of o
Josh, thanks for the feedback. That problem can also occur with dynamic
members, would not be just for those who work with realtime?
tks
2014-06-06 10:14 GMT-03:00 Josh Metzger :
> On Fri, Jun 6, 2014 at 9:03 AM, Eduardo Leones <
> edua...@ypytecnologia.com.br> wrote:
>
>>
>> Guys, I have a p
On Fri, Jun 6, 2014 at 9:03 AM, Eduardo Leones wrote:
>
> Guys, I have a problem. I have a queue on asterisk 1.8 that members are
> added dynamically via the AMI QueueAdd. When you run the CLI a
> "reload app_queue.so" all members who were in the queue disappear. This is
> a bug or some parameter
Guys, I have a problem. I have a queue on asterisk 1.8 that members are
added dynamically via the AMI QueueAdd. When you run the CLI a
"reload app_queue.so" all members who were in the queue disappear. This is
a bug or some parameter that I do not know?
Would have another way to do the reload queu
> e2fsprogs-devel is the package that provides uuid.h on centos 5
I tried that first and it didn't seem to. I'm pretty sure I needed
uuid-dce-devel.
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Richard Kenner skrev 2014-04-27 12:27:
What distro are you building on?
CentOS 5.10.
e2fsprogs-devel is the package that provides uuid.h on centos 5
/niklas
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> What distro are you building on?
CentOS 5.10.
> Both have the libraries listed in install_prereq.
Indeed it has all but 2 or 3 of those libraries (none related to uuid), but
after running that script, it was still missing what it needed for uuid.
Unfortunately, there's no upgrade path from Cen
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