[asterisk-users] SIP Qualify is not working with Postgres

Hello. I'm using Asterisk 1.4.22 with Postgres 8.3 in a Ubuntu 8.04 Server. I configured Asterisk to get sip from Postgres, and set qualify for all sips as yes, but the sip show peers command show the status of the peers as UNKNOWN srvcentral*CLI sip show peers Name/username Host

[asterisk-users] DTMF over IAX trunk ignoring last digit

Hello. Scenario: 9 servers connectec to each other over IAX trunks. Users used to call to remote extensions and remote conferences (meetme) via IAX. Problem: all extensions from one server (just one) when try to attend remote conferences had problems with PIN validation. If they use their local

[asterisk-users] XMPP + Asterisk integration - a practical and simple example

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

Re: [asterisk-users] XMPP + Asterisk integration - a practical and simple example

://offtopicsandfun.blogspot.com http://biertasters.blogspot.com http://twitter.com/mhterres On Fri, Aug 29, 2014 at 11:51 AM, Marcelo Terres mhter...@gmail.com wrote: http://www.mundoopensource.com.br/en_page_xmpp_asterisk_pratical_example/ Regards, Marcelo H. Terres mhter...@gmail.com Openfire-BR owner

[asterisk-users] More XMPP + Asterisk integration: Send a XMPP message to all extensions logged in an Asterisk queue

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

Re: [asterisk-users] JABBER_STATUS CODE 7

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:

Re: [asterisk-users] JABBER_STATUS CODE 7

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

[asterisk-users] OT: Openfire: ServerInfo Plugin - version 0.4 released

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

[asterisk-users] OT: Openfire - new version of B9 plugin released

Hey people. I just released B9 version 0.3. This version contains new commands (create conference, invite conference), but the major feature is socket connection that can be configured in a console admin page. In the page you can enable socket connections, change ip and port for binding and

[asterisk-users] A little OT: AstDemo - Openfire and Asterisk integration

Hi, I continued the developing of Openfire and Asterisk integration projects, and now I'm here to invite you to test AstDemo, that allows VoIP operations directly in XMPP clients. So if use Openfire and wanna test AstDemo, please send me some feedback, suggestions and bug reports. With your

[asterisk-users] X-RTP-Stat SIP header

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 negócios Service

Re: [asterisk-users] PJProject Bundled Update

Build with success in Ubuntu 14.04 LTS. I just need to install some packages (libspeex and libgsm. Em sex, 1 de abr de 2016 20:47, sean darcy escreveu: > On 03/31/2016 11:57 AM, George Joseph wrote: > > As you know, the ability to use a bundled version of pjproject was > >

[asterisk-users] Delivering Asterisk IVR data to softwares using XMPP

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

[asterisk-users] XyBot - a XMPP bot that allows users to interact with Asterisk

Hi. I'm here to invite you all to test another PoC that I developed and that uses Asterisk and XMPP, called XyBot. XyBot is a XMPP bot written in python and its main goal is to enable users to interact with asterisk directly from their XMPP client. Xybot was built to provide a expandable

[asterisk-users] Yealink T21P E2 bug solved

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

[asterisk-users] Asterisk and Yealink T21P E2

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 --

Re: [asterisk-users] Asterisk and Yealink T21P E2

terres On Thu, Jul 14, 2016 at 3:42 PM, Jeff LaCoursiere <j...@jeff.net> wrote: > > On 07/14/2016 02:14 PM, Marcelo Terres wrote: >> >> Hello. >> >> Anybody in the list is using this IP phone? >> >> Regards, >> >> Marcelo H. Terres

Re: [asterisk-users] WhatsApp feature on Asterisk

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

Re: [asterisk-users] SIP trunk

_. ? 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

[asterisk-users] AstriCon 2016 - XMPP and Asterisk

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:

Re: [asterisk-users] PJSIP not detected

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

Re: [asterisk-users] Asterisk 14.0.0-beta1 Now Available

I'm trying to compile it with unbound but I'm getting the following error: "The UNBOUND installation appears to be missing or broken." Ubuntu 14.04.5 LTS \n \l root@rtc:/usr/local/src/asterisk-14.0.0-beta1# dpkg -l | grep -i unboun ii libunbound-dev:amd64

Re: [asterisk-users] Asterisk 14.0.0-beta1 Now Available

Thanks Joshua. Marcelo H. Terres <mhter...@gmail.com> 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 <jc...@digium.com> wrote: > Marce

[asterisk-users] AstriCon 2016 - crowdfunding for participation

Hello everybody. Well, I know that this is not the purpose of the list, but I started a crowdfunding project to allow me attend AstriCon 2016, as a speaker. My talk "Using Asterisk and XMPP to provide greater tools to your customers and your users" was approved and you can get more information

[asterisk-users] Yealink T21P_E2 - bugs in firmware

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

Re: [asterisk-users] Asterisk 14.0.0-beta1 Now Available

* ASTERISK-25702 - PjSip realtime DB and Cache Errors since > upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2 (Reported by > Nic Colledge) > * ASTERISK-25730 - build: make uninstall after make distclean > tries to remove root (Reported by George Joseph) > * AST

Re: [asterisk-users] 14.3.0 download archive corrupt - cannot extract

Same problem with me. I downloaded the file in 2 different places and had the same error... Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On 14 February

Re: [asterisk-users] 14.3.0 download archive corrupt - cannot extract

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... > > An issue was filed for tracking this[1] and it will be resolved later > today. > > [1] https://issues.asterisk.org

Re: [asterisk-users] Soft SIP phones that support TLS - Asterisk version 13.13.1

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

Re: [asterisk-users] Developing Asterisk Modules

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 https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On

Re: [asterisk-users] Asterisk 14.0.0-rc1 Now Available

o, and I could scroll back and see > if there were any obvious errors in the dialplan. > > Is this and intended change, something I've done wrong, or a bug that > needs filing? > > Thanks! > > On 20 September 2016 at 00:37, Joshua Colp <jc...@digium.com> wrote: >>

Re: [asterisk-users] Asterisk 14.0.0-rc1 Now Available

to give some info, and I could scroll back and see > if there were any obvious errors in the dialplan. > > Is this and intended change, something I've done wrong, or a bug that > needs filing? > > Thanks! > > On 20 September 2016 at 00:37, Joshua Colp <jc...@digium.com&g

Re: [asterisk-users] Asterisk 14.0.0-rc1 Now Available

Hey dev team, kudos for the good job. Just one information. When I started Asterisk after upgrade to version 14, I received this information: [Sep 19 19:40:57] WARNING[22694]: res_odbc.c:525 load_odbc_config: The 'pooling', 'shared_connections', 'limit', and 'idlecheck' options are deprecated.

Re: [asterisk-users] Asterisk 14.0.0-rc1 Now Available

https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On Mon, Sep 19, 2016 at 7:44 PM, Marcelo Terres <mhter...@gmail.com> wrote: > Hey dev team, kudos for the good job. > > Just one information. > > When I started Asterisk after upgrad

Re: [asterisk-users] Asterisk 14.0.0-rc1 Now Available

Thanks Joshua. Marcelo H. Terres <mhter...@gmail.com> IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On Mon, Sep 19, 2016 at 7:53 PM, Joshua Colp <jc...@digium.com> wrote: > Marce

Re: [asterisk-users] Asterisk 14.0.0-rc1 Now Available

ource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On Mon, Sep 19, 2016 at 7:49 PM, Marcelo Terres <mhter...@gmail.com> wrote: > One more thing about my last email: I think that you forgot to update > the configs/samples/res_

Re: [asterisk-users] Asterisk 11.23 with libmysqlclient20 on Debian 8

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

[asterisk-users] Asterisk and XMPP - AstriCon 2016

Hello. This is the link of the slides of my talk presented yesterday in AstriCon, about Asterisk and XMPP. As soon as the video is available, I'll share it too. http://pt.slideshare.net/mhterres/astricon-2016-using-asterisk-and-xmpp-to-provide-greater-tools-to-your-customers-and-your-users []s

Re: [asterisk-users] Problems with queues realtime configuration

in.com/in/marceloterres On Sat, Oct 1, 2016 at 2:42 PM, Marcelo Terres <mhter...@gmail.com> wrote: > Hello. > > I'm using Asterisk 14.0.2 and I'm not sure exactly when this problem > starts to happen, but maybe somebody here can help me with it. > > I'm using queues with realtim

[asterisk-users] Using Asterisk and XMPP - AstriCon 2016

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 https://www.mundoopensource.com.br

[asterisk-users] Problems with queues realtime configuration

Hello. I'm using Asterisk 14.0.2 and I'm not sure exactly when this problem starts to happen, but maybe somebody here can help me with it. I'm using queues with realtime configuration but the system is not loading the queues. Let me show you what happens when I'm trying to load the module

Re: [asterisk-users] How to build with cdr_adaptive_odbc ?

You need unixodbc and odbcinst packages too, to configure the odbc. []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 13 April 2017 at 19:41, Pierre

Re: [asterisk-users] How to build with cdr_adaptive_odbc ?

terres On 20 April 2017 at 13:16, Pierre Couderc <pie...@couderc.eu> wrote: > 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: >> >>

Re: [asterisk-users] Asterisk 1.8.32.3 : no video (h.264)

in.com/in/marceloterres On 20 April 2017 at 12:42, Jonas Kellens <jonas.kell...@telenet.be> wrote: > Hello > > in sip.conf I have ; > > videosupport=yes > > > > > Kind regards. > > J. > > > > On 20-04-17 13:09, Marcelo Terres wrote: >>

Re: [asterisk-users] How to build with cdr_adaptive_odbc ?

terisk 14, and yes, menuselect shows me the need for > generic_odbc(E), res_odbc_transaction(M) and ltdl(E) > > but what does this imply under debian ? > > I have unixodbc installed an tested and too libltdl-dev ! > But what am I missing ? > > > On 04/19/2017 10:10 AM, Marcelo

Re: [asterisk-users] Asterisk 1.8.32.3 : no video (h.264)

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

Re: [asterisk-users] feature codes

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:

Re: [asterisk-users] How to build with cdr_adaptive_odbc ?

t;. > > > > On 17/04/2017 10:48, Marcelo Terres wrote: >> >> You need unixodbc and odbcinst packages too, to configure the odbc. >> >> []s >> Marcelo H. Terres <mhter...@gmail.com> >> IM: mhter...@jabber.mundoopensource.com.br >>

Re: [asterisk-users] How to build with cdr_adaptive_odbc ?

to get myself out of the list? > > > Irfan > -Message d'origine- > De : asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] De la part de Marcelo Terres > Envoyé : mercredi 19 avril 2017 10:11 > À : Asterisk Users Mailing Lis

Re: [asterisk-users] VoIP monitor and multiple RTP streams

ws in MySQL or wireshark? > > > On Mon, Aug 14, 2017 at 11:24 AM, Marcelo Terres <mhter...@gmail.com> > wrote: > >> Hello. >> >> Is someone here using VoIPmonitor? >> >> I am using just the sniffer and I found some pcap files that contain some >>

[asterisk-users] VoIP monitor and multiple RTP streams

Hello. Is someone here using VoIPmonitor? I am using just the sniffer and I found some pcap files that contain some odd streams. For example, I have a file with 3 streams, but the weird stuff is that 2 streams are the same (e.g., have the same source address and port and same destination

Re: [asterisk-users] AMI column widths

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

Re: [asterisk-users] AMI column widths

2017 9:17 am, "Antony Stone" <antony.st...@asterisk.open.source.it> 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 conso

Re: [asterisk-users] Asterisk 13.16.0 segfault

in.com/in/marceloterres On 19 July 2017 at 17:03, Carlos Chavez <cur...@telecomab.mx> wrote: > On 7/19/17 2:37 AM, Marcelo Terres wrote: > > This is the pjsip library. > > Is it possible to you to update pjsip for the latest version to test if it > solves the problem? >

Re: [asterisk-users] corosync and Asterisk 13

> asterisk-users@lists.digium.com> > *Subject:* Re: [asterisk-users] corosync and Asterisk 13 > > > > yessir > > > > *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com <asterisk-users-boun...@lists.digium.com>] *On

Re: [asterisk-users] Asterisk 13.16.0 segfault

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: >

Re: [asterisk-users] BLF sharing between Asterisk 11 and 13

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

Re: [asterisk-users] Integration of Google Speech API V2

l MathuR" <rahul.ultim...@gmail.com> wrote: > Hi Marcelo, > > Thanks for replying, I do not know what this branch is. > Could you please let me know. > > Also, I enabled google cloud speech API only from the console. Do I need > more API enabled? > > > &g

Re: [asterisk-users] Integration of Google Speech API V2

Thanks for replying, I do not know what this branch is. > Could you please let me know. > > Also, I enabled google cloud speech API only from the console. Do I need > more API enabled? > > > > On Wed, Jul 19, 2017 at 3:41 PM, Marcelo Terres <mhter...@gmail.com> &

Re: [asterisk-users] Integration of Google Speech API V2

Did you already tried the cloud_api branch? 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 July 2017 at 10:17, Rahul MathuR

Re: [asterisk-users] corosync and Asterisk 13

Did you installed the dev package? corosync-dev 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 July 2017 at 14:46, Ryan, Travis

Re: [asterisk-users] Asterisk 13.16.0 segfault

e: > On 7/20/17 8:47 AM, Marcelo Terres wrote: > > Which version of Asterisk are you using? Are you compiling it with the > bundle pjproject ? > > --with-pjproject-bundled > > Regards, > > Marcelo H. Terres <mhter...@gmail.com> > IM: mhter...@jabber.mund

Re: [asterisk-users] Asterisk realtime - Error with index length in alembic script

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:

Re: [asterisk-users] asterisk.conf ignored?

15:41, Marcelo Terres <mhter...@gmail.com> wrote: > 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 <mhter...@gmail.com> > IM: mhter...@j

Re: [asterisk-users] Simplest way of executing a non-blocking (async) python AGI script?

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,

Re: [asterisk-users] asterisk.conf ignored?

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 https://www.mundoopensource.com.br https://twitter.com/mhterres

Re: [asterisk-users] Change OS from CentOS 6 to 7

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" <mhter...@gmail.com> wrote: > Looks like 192.168.5.2

Re: [asterisk-users] Change OS from CentOS 6 to 7

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

Re: [asterisk-users] Writing CDR's to two database servers

Well, you could create and AGI and run it after the normal CDR was inserted. Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On 20 June 2017 at 13:42, Tech

Re: [asterisk-users] Best way to know a call is being transfered

Unfortunately, the transfer AMI events were introduced just in Asterisk13. But, you can set the __TRANSFER_CONTEXT variable and probably the __GOTO_ON_BLINDXFR (this one I never used) to control the transfer in your own way.

Re: [asterisk-users] OT: Want to capture all SIP messages

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 https://www.mundoopensource.com.br https://twitter.com/mhterres

Re: [asterisk-users] IAX port 4569

Is it enabled in the iax.conf file? 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 5 June 2017 at 13:48, wrote: > Does

Re: [asterisk-users] OT: Explain where mailing list bouncing comes from ?

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

Re: [asterisk-users] OT: Explain where mailing list bouncing comes from ?

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: >

Re: [asterisk-users] Dial an extension to modify dialplan

https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+ManagerAction_DialplanExtensionAdd Is it enough? Regards, Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres

Re: [asterisk-users] Dial an extension to modify dialplan

at 16:13, Antony Stone <antony.st...@asterisk.open.source.it> wrote: > On Monday 08 May 2017 at 15:44:47, Marcelo Terres wrote: > >> https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+ManagerAction_Dialpl >> anExtensionAdd >> >> Is it enough? > >

Re: [asterisk-users] Feature Code to Meeting Room

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

Re: [asterisk-users] IAX port 4569

168.141.1.4569: UDP, 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, M

Re: [asterisk-users] IAX port 4569

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

Re: [asterisk-users] *****SPAM***** Re: IAX port 4569

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

Re: [asterisk-users] asterisk server - no sound

Which Asterisk version are you using? Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On 6 June 2017 at 18:32, andre castro wrote: >

Re: [asterisk-users] asterisk server - no sound

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 https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On 6 June

Re: [asterisk-users] asterisk server - no sound

Looks like it comes com pbx_dundi.c. Why are you using dundi? Regards, Marcelo H. Terres <mhter...@gmail.com> IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On 6 June 2017 at 18:43, Marcelo

Re: [asterisk-users] asterisk server - no sound

I think you need to configure the IPs in your server. You just have localhost... Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On 6 June 2017 at 16:27, andre

Re: [asterisk-users] asterisk server - no sound

at 18:54, andre castro <an...@andrecastro.info> wrote: > I am using version: 14.5.0 > No, Im not using Dundi. > Can you a bit more informative when you say I "need to configure the IPs > in your server"? > thanks! > a > On 06/06/2017 07:47 PM, Marcelo Ter

Re: [asterisk-users] asterisk server - no sound

thanks! > a > On 06/06/2017 07:47 PM, Marcelo Terres wrote: >> I think you need to configure the IPs in your server. You just have >> localhost... >> Marcelo H. Terres <mhter...@gmail.com> >> IM: mhter...@jabber.mundoopensource.com.br >> https://www

Re: [asterisk-users] Realtime pjsip issues

Hello. Did you ps_contacts table has all columns listed here? INSERT INTO ps_contacts (id, via_addr, qualify_timeout, call_id, reg_server, path, endpoint, via_port, authenticate_qualify, uri, qualify_frequency, user_agent, expiration_time, outbound_proxy) VALUES (?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?,

Re: [asterisk-users] Asterisk 11.25.2

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,

Re: [asterisk-users] Asterisk Voicemail changes

Probably the best option is to create your own voicemail app using ARI. 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 1 September 2017 at

Re: [asterisk-users] Problem using Android SIP-Client

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

Re: [asterisk-users] failed to find existing extension

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

Re: [asterisk-users] failed to find existing extension

ote: > > On Monday 10 September 2018 at 21:54:33, Marcelo Terres wrote: > > > >> I have think it should be > >> > >> context=0705680837 > >> > >> Not > >> > >> context=[0705680837] > > > > Ha! You're right... so

Re: [asterisk-users] How to steal an answered call?

:22, Marcelo Terres wrote: > > 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 > https://www.mundoopensource.com.

Re: [asterisk-users] How to steal an answered call?

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 https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On

Re: [asterisk-users] how to use a database

On 12/06/2018 10:26 PM, Marcelo Terres wrote: > > The Asterisk source has a tool to create the db > > Which one is that? > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] Capture SIP all the time

You can use the voipmonitor sniffer. www.voipmonitor.org. 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, 6 Dec 2018 at 00:13, Steve Edwards wrote: > > On Wed,

Re: [asterisk-users] how to use a database

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: > > >> I don't see a table for that. >

Re: [asterisk-users] how to use a database

Oh, I didn't know that. 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 Mon, 10 Dec 2018 at 14:50, Floimair Florian wrote: > > Alembic currently doesn't cover

Re: [asterisk-users] figuring out what happened to a call

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

Re: [asterisk-users] Asterisk Usage Survey

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 participate to the

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