it is an old version, but we can't change it now. We are moving to
Asterisk 16 next year, but currently that is our reality.
Any ideas of what could be causing this? Or any ideas of how to debug it?
Thanks.
Regards,
Marcelo H. Terres
https://www.mundoopensource.com.br
https://twitte
ering go to Fosdem next February?
Thanks.
Regards,
Marcelo H. Terres
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres
On Tue, 24 Sep 2019 at 11:10, Jöran Vinzens wrote:
> Good point.
>
> I will try that. We have just started th
Hello Jean-Denis.
I believe the idea is that you answer the survey for each type of scenarios
you are running.
So one for call centre, another one for ivr, etc...
Regards,
Marcelo
On Mon, 11 Mar 2019, 02:10 Jean-Denis Girard, wrote:
> Hi Matt,
>
> I would have loved to participa
Oh, I didn't know that.
Regards,
Marcelo H. Terres
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On Mon, 10 Dec 2018 at 14:50, Floimair Florian wrote:
>
> Alembic currently do
https://wiki.asterisk.org/wiki/display/AST/Managing+Realtime+Databases+with+Alembic
Marcelo H. Terres
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On Fri, 7 Dec 2018 at 13:34, hw wrote:
>
>
The Asterisk source has a tool to create the db
Marcelo
On Thu, 6 Dec 2018, 19:44 Antony Stone On Thursday 06 December 2018 at 17:49:25, hw wrote:
>
> > On 12/05/2018 05:00 PM, Antony Stone wrote:
> > > On Wednesday 05 December 2018 at 15:31:38, hw wrote:
> > >>
You can use the voipmonitor sniffer.
www.voipmonitor.org.
Regards,
Marcelo H. Terres
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On Thu, 6 Dec 2018 at 00:13, Steve Edwards wrote:
>
>
Queue_log
On Sat, 1 Dec 2018, 13:03 hw Hi,
>
> how can I figure out what happens to inbound calls?
>
> The inbound calls I'm particularly interested in make phones that are
> members of a queue ring; when the call isn't picked up, another phone is
> dialed and when the call still isn't picked up,
This simple issues usually are the hardest to find. I know how it is.
Regards,
Marcelo H. Terres
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On Tue, 11 Sep 2018 at 09:47, wrote:
>
>
I have think it should be
context=0705680837
Not
context=[0705680837]
Regards,
On Mon, 10 Sep 2018, 20:43 , wrote:
> On 2018-09-09 10:27, Antony Stone wrote:
>
> > 1. Try removing one of the two commas.
> >
> > 2. Take a copy of your dialplan, and then strip out *everything* except
> > the
Unfortunately, all channels need to be handled by ARI stasis app,
otherwise, you can't use ARI.
Regards,
Marcelo H. Terres
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On Mon, 9 Jul 2018
Hello.
I believe you can do that with ARI, but I am not sure if you can do it
without using ARI to start the call.
Regards,
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
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On Mon
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
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https://twitter.com/mhterres
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On 24 October 2017 at 14:42
(?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?)
Regards,
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
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On 15 September 2017 at 16:18, Bryant Zimmerman wrote:
> Joshua
>
> We are using MariaDB as the databas
Hello Jerry.
Does the Joshua's tips helped you to solve your issues or are you still
facing audios problems?
I am asking you because I need to update some servers but I can't have this
kind of problems.
Thanks.
Regards,
On 5 Sep 2017 2:02 pm, "Joshua Colp" wrote:
> On Tue, Sep 5, 2017, at 09
Probably the best option is to create your own voicemail app using ARI.
Regards,
Marcelo H. Terres
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On 1 September 2017 at 10:50, Tim Turpin wrote
Hello Dovid.
I tried to figure it out, but to be honest I could not find a reason for
the change.
The lines that I sent are the RTP streams detected by Wireshark.
Regards,
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com
98661,2.9025967411766738,0.97877393850963945,""
"1XX.XXX.XXX.XXX",10602,"6X.XXX.XXX.XXX",34170,325951803,"g711A",4949,0,41.
8790815,4.5846492231155924,1.0537488536922062,""
The only thing that I could notice is that the first pack
5.25 is your Asterisk?
Did you try to add a manual Iptables rule?
iptables -I INPUT -j ACCEPT.
This will accept any input packets (just for testing purposes, of course).
Regards,
On 4 Aug 2017 9:27 pm, "Marcelo Terres" wrote:
> Looks like 192.168.5.25 is not responding...
>
Looks like 192.168.5.25 is not responding...
On 4 Aug 2017 8:28 pm, "Jerry Geis" wrote:
> Audio packets are running...
>
> 961 16.150421076 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711
> PCMU, SSRC=0x6A3E0AF1, Seq=28402, Time=73280
> 962 16.170411284 192.168.5.150 -> 192.168.5.25 RTP 214
I don't have much knowledge about freepbx, but if some day I had to use it,
I would prefer to use the Asterisk compiled from source, unless it comes
with an Asterisk package (rpm, supposing it is running CentOS).
On 20 Jul 2017 5:08 pm, "Carlos Chavez" wrote:
> On 7/20/1
I tried and face the same problem. I also installed the libcpg-dev but the
problem persists.
I will test it later if I had time.
Regards,
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in
Which version of Asterisk are you using? Are you compiling it with the
bundle pjproject ?
--with-pjproject-bundled
Regards,
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
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On 19
Did you installed the dev package? corosync-dev
Marcelo H. Terres
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On 19 July 2017 at 14:46, Ryan, Travis wrote:
> I want to use corosync
loud.google.com/speech> can be downloaded here
<https://github.com/zaf/asterisk-speech-recog/archive/cloud_api.zip>.
You can also clone the project with Git <https://git-scm.com/> by running:
$ git clone git://github.com/zaf/asterisk-speech-recog
On 19 Jul 2017 11:36 am, "
If you clone zaf repository you will find a branch called cloud_api that
works with the new version.
In the link that you sent you will find information about it.
Regards,
On 19 Jul 2017 11:36 am, "Rahul MathuR" wrote:
> Hi Marcelo,
>
> Thanks for replying, I do not know
Did you already tried the cloud_api branch?
Regards,
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
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https://twitter.com/mhterres
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On 19 July 2017 at 10:17, Rahul MathuR wrote:
> Hi Jonathan
>
&g
This is the pjsip library.
Is it possible to you to update pjsip for the latest version to test if it
solves the problem?
On 18 Jul 2017 3:52 pm, "Carlos Chavez" wrote:
> I am getting frequent segfaults on a new Asterisk installation. So far
> the only message I see is:
>
> Jul 18 09:02:42 pbx
BLF with pjsip is a little bit different.
Did you read the https://wiki.asterisk.org/wiki/display/AST/Configuring+
res_pjsip+for+Presence+Subscriptions?
On 16 Jul 2017 3:38 am, "Ryan, Travis" wrote:
> I have servers setup in versions 11 and 13. Between two 11 servers, I had
> no issues sharing
Please open a Ticket (https://issues.asterisk.org), to let them know that
they need to update the documentation in Wiki and also handle this
situation when using Alembic in Debian 9 (could happens in other Distros
too).
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
https
8 Jul 2017 9:17 am, "Antony Stone"
wrote:
On Saturday 08 July 2017 at 07:15:08, Marcelo Terres wrote:
> There are no sip show channels on AMI. Also, the output that you sent is
> not a AMI output. Are u using AMI ou running commands on console?
I'm using AMI.
I have a
There are no sip show channels on AMI. Also, the output that you sent is
not a AMI output. Are u using AMI ou running commands on console?
Running commands on console and parsing the output is the worst way to
obtain data, first because it is not easily parseable.
Second, it doesn't show you all
Take a look on that:
https://issues.asterisk.org/jira/browse/ASTERISK-20532
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 30 June 2017 at 22:23, Jonathan H wrote
You should try to limit it in your sip trunks (is you are using SIP
trunks, of course)
Marcelo H. Terres
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On 30 June 2017 at 15:41, Marcelo Terres
This limit is only valid for inbound calls:
Sets a maximum number of simultaneous inbound channels. No limit is
set by default.
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
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On
Well, you could create and AGI and run it after the normal CDR was inserted.
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
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On 20 June 2017 at 13:42, Tech Support wrote:
>
Yes, let's stop to use our gmail accounts because JUST THE DIGIUM
MAILING LIST is bouncing.
All other mailman servers must be wrongly configured, and the Digium
server is the only one that is correct. Perfect!
:-D
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
It is happening the same with me.
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 12 June 2017 at 08:07, Olivier wrote:
> Hello,
>
> I'm a faithful
And it is worst (and that could be the reason of the error).
127.0.0.1 is configured in 2 interfaces (lo and venet0), just with
different network masks.
Marcelo H. Terres
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https
Well, based on the config that you sent, your server just have the
localhost IP (127.0.0.1)
Marcelo H. Terres
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On 6 June 2017 at 18:54, andre castro
I think you need to configure the IPs in your server. You just have localhost...
Marcelo H. Terres
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On 6 June 2017 at 16:27, andre castro wrote
Looks like it comes com pbx_dundi.c.
Why are you using dundi?
Regards,
Marcelo H. Terres
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On 6 June 2017 at 18:43, Marcelo Terres wrote:
> Wh
Which Asterisk version are you using?
Marcelo H. Terres
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On 6 June 2017 at 18:32, andre castro wrote:
> Any ideas.
> After configuring
Try to use the echo app. If you can listen your echo, so it is
something in the network.
Regards,
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
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On 6 June 2017 at 14:18, andre
, length 65
14:20:45.376559 IP 192.168.141.1.4569 > 192.168.141.7.4569: UDP, length 53
14:20:45.376630 IP 192.168.141.7.4569 > 192.168.141.1.4569: UDP, length 12
^C
12 packets captured
12 packets received by filter
0 packets dropped by kernel
Thelma
On 06/05/2017 02:17 PM, Marcelo Terres wrote:
>
Try tcpdump
On 5 Jun 2017 9:41 pm, wrote:
Doesn't matter how much I increase the verbose output
asterisk -vvr
asterisk will not even print a single line.
How to find out if my firewall has this port open?
https://www.grc.com
is reporting that my port is 4569 is in Stealth mode (so it is
You can use tcpdump in your server to verify if it is receiving the
packets.
tcpdump -ni any port 4569
So you have more than one ip in the server?
On 5 Jun 2017 9:13 pm, wrote:
> No, I don't think it is IP table issue, I've not upgraded dd-wrt for a
> while and it was zoiper was working OK wit
Is it enabled in the iax.conf file?
Regards,
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
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On 5 June 2017 at 13:48, wrote:
> Does asterisk listen on port 4569 by defa
You can save individual calls with voipmonitor too, and it save the
info in a mysql db, allowing you to search the pcap files easily.
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
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+Variables
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 29 May 2017 at 10:06, Jonas Kellens wrote:
> Hello
>
> thank you for your answer.
>
> However thi
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerAction_DialplanExtensionRemove
Marcelo H. Terres
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On 8 May 2017 at 16:13, Antony Stone
https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+ManagerAction_DialplanExtensionAdd
Is it enough?
Regards,
Marcelo H. Terres
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On 8 May 2017 at
You can create your own dynamic features.
https://wiki.asterisk.org/wiki/display/AST/Custom+Dynamic+Features
If it supports AGI (I'm not sure of that), it would be a good method
do to that, probably.
Regards,
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
Ah, ok.
Everytime you install a package you need to run configure again to
allow detection of new lib.
Regards,
Marcelo H. Terres
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On 20 April 2017
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
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On 20
ii libltdl7:amd64 2.4.6-0.1
amd64System independent dlopen wrapper for GNU
libtool
Also, I really don't remember of having any kind of problems with odbc support.
Did you have all this packages (or equivalents) installed too?
Regards,
Marc
I suppose that you enable the video support on sip.conf, right?
Regards,
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
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On 19 April 2017 at 13:18, Jonas Kellens wrote:
> He
You just need to read the email :-)
It is a common procedure to most mailing lists.
Regards,
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
https
What version of Asterisk are you using?
When I go to cdr_adaptative_odbc in Asterisk 14 it depends of res_odbc
and res_odbc depends on: generic_odbc(E), res_odbc_transaction(M),
ltdl(E)
Regards,
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
You need unixodbc and odbcinst packages too, to configure the odbc.
[]s
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On 13 April 2017 at 19:41, Pierre Couderc wrote:
> I
You can configure the features in the features.conf file, but some
features like DND and call forward are not available, so, or you use
the SIP client own functionalities for that (if available), or you
will have to develop your own features.
Regards,
Marcelo H. Terres
IM: mhter
Zoiper?
On 15 Feb 2017 6:46 p.m., "Motty Cruz" wrote:
> Hello, I have a user that prefers Soft SIP phone install on his laptop,
> for security reasons I have enable TLS on our Asterisk server to support
> TLS authentication, It works well with hard phones. Has anybody in this
> forum use SIP Sof
Thanks Joshua.
Marcelo H. Terres
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On 14 February 2017 at 14:01, Joshua Colp wrote:
> On Tue, Feb 14, 2017, at 09:57 AM, Marcelo Terres wrote:
>
Same problem with me.
I downloaded the file in 2 different places and had the same error...
Marcelo H. Terres
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On 14 February 2017 at 08:42
Hello Valter.
Probably you will get more informations about that in the asterisk-dev
mailing list.
Regards,
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
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On 18 January 2017 at
I think that you need the dev files too. In Debian 8, the package is
libmysqlclient-dev.
But Debian 8 uses libmysqlclient-18. Where did you get the 20 ?
Regards,
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
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https
Hello.
As I promised during the talk, this is the post with diaplans and tools
that I used.
https://www.mundoopensource.com.br/astricon-2016-asterisk-xmpp-talk/
Regards.
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
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Confirmed, I open an issue about it:
[JIRA] (ASTERISK-26431) Queues doesn't appear when using realtime configuration
Marcelo H. Terres
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On Sat
ly this is the problem, but I don't know why
it is happening.
My extconfig.conf:
queues => odbc,asterisk
queue_members => odbc,asterisk
And my tables:
asterisk=# SELECT COUNT(name) from queues;
count
---
3
(1 row)
asterisk=# SELECT COUNT(queue_name) from qu
[]s
Marcelo H. Terres
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--
_
-- Bandwidth and Colocation Provided by http://ww
Hello Jonathan,
https://issues.asterisk.org/jira/browse/ASTERISK-26391
Regards,
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
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On Tue, Sep 20, 2016 at 4:19 AM, Jonathan H wrote
That's the same behavior that I noticed.
I'll open an issue about it soon.
Marcelo H. Terres
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On Tue, Sep 20, 2016 at 4:19 AM, Jonatha
Thanks Joshua.
Marcelo H. Terres
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On Mon, Sep 19, 2016 at 7:53 PM, Joshua Colp wrote:
> Marcelo Terres wrote:
>>
>> Hey dev team, kud
nups
But after upgrade to Asterisk 14 rc1, even if I use the sample
configuration files, I didn't get any information when call rasterisk.
root@rtc:/usr/local/src/asterisk-14.0.0-rc1# rasterisk
rtc*CLI>
Is that expected?
Regards,
Marcelo H. Terres
IM: mhter...@jabber.mundoopensou
One more thing about my last email: I think that you forgot to update
the configs/samples/res_odbc.conf.sample file, because it still
contains idlecheck and limit parameters.
Regards,
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https
7;idlecheck' options are
deprecated. Please see UPGRADE.txt for information
I know that odbc configuration changes, but there is no UPGRADE file
containing this information. In fact, there is no UPGRADE-14.txt file.
Regards,
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
1.4.22-1ubuntu4.14.04.2
amd64static library, header files, and docs for
libunbound
ii libunbound2:amd641.4.22-1ubuntu4.14.04.2
amd64library implementing DNS resolution and
validation
Any ideas?
Regards,
Marcelo H. Terres
IM: mhter...@jabber.mu
Thanks Joshua.
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres
On Sat, Aug 13, 2016 at 11:12 AM, Joshua Colp wrote:
> Marcelo Terres wrote:
>>
>> I'm tryin
1.4.22-1ubuntu4.14.04.2
amd64static library, header files, and docs for
libunbound
ii libunbound2:amd641.4.22-1ubuntu4.14.04.2
amd64library implementing DNS resolution and
validation
Any ideas?
Marcelo H. Terres
IM: mhter...@jabber.mundoope
Why don't you use the bundle option in Asterisk compilation ?
./configure --with-pjproject-bundled
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 Thu, Aug 11,
Going to AstriCon 2016 ?
Don't miss my talk about how to use XMPP and Asterisk to improve the
user experience.
https://astricon2016.sched.org/event/7Zje/using-asterisk-and-xmpp-to-provide-greater-tools-to-your-customers-and-your-users
Regards,
Marcelo H. Terres
IM:
Whatapp is developed in Erlang and uses a modified XMPP protocol, FunXMPP.
What do you want to do, exactly?
[]s
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres
On Fri, Jul 29
https://www.mundoopensource.com.br/yealink-t21p_e2-com-bugs-no-firmware-bugs-in-the-yealink-t21p_e2-firmware/
[]s
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres
_. ?
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 Tue, Jul 26, 2016 at 11:39 AM, Jerry Geis wrote:
> It seems I am not getting any digits coming over a SIP tr
No problems with authentication during invite after reboot?
I'm using insecure=no in SIP configuration.
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 Thu, J
Hello.
Anybody in the list is using this IP phone?
Regards,
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres
the
video later.
The goal is about 2500 USD (more informations in
https://www.catarse.me/aporte_financeiro_para_palestrar_na_astricon_2016_b06f).
If the project not reach the necessary amount, all values will be
returned for the donators.
Thanks all.
Marcelo H. Terres
IM: mhter...@jabbe
https://www.mundoopensource.com.br/yealink-t21p_e2-com-bugs-no-firmware-bugs-in-the-yealink-t21p_e2-firmware/
Regards,
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres
ts or suggestions are welcomed.
Regards,
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres
--
_
-- Bandwidth and Coloc
Hello.
I developed a little project (a PoC) to "integrate" Asterisk IVRs with
"other softwares", allowing that data already entered in IVR can be used in
other stages of a customer service, for example.
The main goal is to provide more efficiency and interoperability between
different solutions i
Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Marcelo H Terres
--
Hi.
I can't find X-RTP-Stat SIP header in my packets. I'm using Asterisk 13.6
and PJSIP.
Is there something that I should configure to Asterisk add this header?
Thanks.
Marcelo Hartmann Terres
Fones: +55 51 3024-3568 | +55 11 4063-8864 | +55 92 3090-0115
Propus - TI alinhada a negóci
your informations I'll be able to make it
better and develop new features.
http://www.mundoopensource.com.br/xmpp-asterisk-integration-practical-example-part-2/
http://www.mundoopensource.com.br/astdemo-en/
Thanks a lot.
Regards,
Marcelo H. Terres
mhter...@gmail.co
://www.mundoopensource.com.br/plugin-b9-openfire/
I hope you enjoy this version and I hope listen your feedbacks.
Regards,
Marcelo H. Terres
mhter...@gmail.com
Openfire-BR owner mailing list
http://www.mundoopensource.com.br
--
_
-- Bandwidth and
http://www.mundoopensource.com.br/versao-0-4-plugin-serverinfo-lancada/
(portuguese)
http://www.mundoopensource.com.br/serverinfo-plugin-openfire/
Regards,
Marcelo H. Terres
mhter...@gmail.com
IM: marc...@jabber.mundoopensource.com.br
http://www.mundoopensource.com.br
http
You always need to use your jabber domain in jabberid.
Regards,
Marcelo H. Terres
mhter...@gmail.com
IM: marc...@jabber.mundoopensource.com.br
http://www.mundoopensource.com.br
http://offtopicsandfun.blogspot.com
http://biertasters.blogspot.com
http://twitter.com/mhterres
On Mon, Oct 13, 2014
Retrieves the numeric status associated with the buddy identified by
jid. If the buddy does not exist in the buddylist, returns 7.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_JABBER_STATUS_res_xmpp
Regards,
Marcelo H. Terres
mhter...@gmail.com
IM: marc
Hey everybody.
Another XMPP+Asterisk example:
http://www.mundoopensource.com.br/en_page_send-xmpp-message-extensions-logged-asterisk-queue/
[]s
Marcelo H. Terres
mhter...@gmail.com
Openfire-BR mailing list's owner
IM: marc...@jabber.mundoopensource.com.br
http://www.mundoopensource.com.br
People ask me about process_xmpp_msg.agi script, so you can find it in my blog:
http://www.mundoopensource.com.br/en_page_xmpp_asterisk_pratical_example/
Regards,
Marcelo H. Terres
mhter...@gmail.com
IM: marc...@jabber.mundoopensource.com.br
http://www.mundoopensource.com.br
http
http://www.mundoopensource.com.br/en_page_xmpp_asterisk_pratical_example/
Regards,
Marcelo H. Terres
mhter...@gmail.com
Openfire-BR owner list
IM: marc...@jabber.mundoopensource.com.br
http://www.mundoopensource.com.br
http://offtopicsandfun.blogspot.com
http://biertasters.blogspot.com
http
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