[asterisk-users] Delays on conferences

[Marcelo Terres]

Hello.

We are having a weird issue with conferences.

Let me explain:

A enters conference room.
B enters conference room.

When B talks, A can listen it immediately. When A talks, took almost a
second to B receives the audio.

Scenario:
Asterisk 11 with meetme.
CentOS 6/7, Dahdi 2.9/2.11

I know 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://twitter.com/mhterres
https://linkedin.com/in/marceloterres
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Re: [asterisk-users] Load issues using AGI

[Marcelo Terres]

Hello Jöran, how are you?

your issues got me very curious and concerned, because I will be working on
something similar of what you are doing soon.

If you don't mind, please keep us updated with our discoveries, especially
that one related to the last suggestion of Tony.

PS: are you considering 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 the work to port our perl AGI to
> Java fastAGI. We will eliminate some of the AGI and see how performance
> improves.
>
> In terms of the perl speed, i will try your suggestion.
>
> Thanks
> Jöran
>
> Tony Mountifield  schrieb am Di., 24. Sep. 2019,
> 11:23:
>
>> In article > vlrrepr10gmkqqs5dptob9x66lx5ooct6uh4oyrst...@mail.gmail.com>,
>> Jöran Vinzens  wrote:
>> >
>> > @john, we using Perl. To see if it is a problem with the perl i had put
>> an
>> > "exit 0" just at the first lines  so there is no logic done at the AGI.
>> > It's only the start up and return from AGI what produces the most of the
>> > load. Nevertheless, we will try what you just posted.
>>
>> Even if you put "exit 0" at the top of the script, the perl interpreter
>> will
>> still need to compile the whole script (and any modules it uses) before it
>> executes the "exit 0".
>>
>> Try commenting out or removing the rest of the script.
>>
>> Cheers
>> Tony
>> --
>> Tony Mountifield
>> Work: t...@softins.co.uk - http://www.softins.co.uk
>> Play: t...@mountifield.org - http://tony.mountifield.org
>>
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Re: [asterisk-users] Asterisk Usage Survey

[Marcelo Terres]

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 survey, but I feel it does
> apply to my situation: as an integrator, I'm installing Asterisk for
> call centers, PBX, IVR... so I can not answer the first question of the
> survey ;) I also have dfferent versions installed.
>
> This is not a negative comment, I just want to express that the survey
> does not seem to apply to me; and many people on the Asterisk lists may
> be in a situation similar as mine.
>
>
> Thanks,
> --
> Jean-Denis Girard
>
> SysNux   Systèmes   Linux   en   Polynésie  française
> https://www.sysnux.pf/   Tél: +689 40.50.10.40 / GSM: +689 87.797.527
>
> Le 08/03/2019 à 05:35, Matthew Fredrickson a écrit :
> > Hey All,
> >
> > For those of you that do not know me, my name is Matthew Fredrickson
> > and I’m the project lead for the Asterisk project. First off, I wanted
> > to thank all of you that contribute in various ways to the project –
> > whether it be at a developmental level, answering questions on forums
> > and mailing lists, contributing documentation, or just generally
> > advocating for it within your sphere of influence. It takes so many
> > people’s efforts to make the project what it is and to sustain such a
> > large and vibrant user and developer community.
> >
> > We created a general survey inquiring how people utilize Asterisk. It
> > should only take about 10-15 minutes, but would help us understand
> > better how our users are utilizing Asterisk and help us to understand
> > if there are important areas of Asterisk that we underemphasize from a
> > development perspective. If you don’t mind filling it out, it would be
> > greatly appreciated.
> >
> > Thanks *so* much again for your time, and best wishes to each of you
> > in your efforts.
> >
> > https://goo.gl/forms/xL1VUHRsf95saly13
> >
> > Matthew Fredrickson
> >
>
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Re: [asterisk-users] how to use a database

[Marcelo Terres]

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 queue_logs.
> As of now it only covers configuration, voicemail and cdr.
>
>
>
>
> With best regards
>
> Florian Floimair
> Innovation - Software-Development
>
> COMMEND INTERNATIONAL GMBH
> A-5020 Salzburg, Saalachstraße 51
> http://www.commend.com 
>
> Security and Communication by Commend
>
> FN 178618z | LG Salzburg
>
> Am 07.12.18, 15:56 schrieb "asterisk-users im Auftrag von hw" 
> :
>
> On 12/07/2018 03:36 PM, Administrator TOOTAI wrote:
> > Le 07/12/2018 à 14:32, hw a écrit :
> >
> > [...]
> >>
> >> Queues seem to be the only way to have several phones ring at once, or
> >> are there other ways?
> >
> > Dial(SIP/Phone1/Phone2&.../Phonex,,)
> >
>
> Good to know, thanks!
>
>
> What are the entries needed in the queue_members table when using odbc?
> Alembic made the primary key so that each queue can only have one entry
> (What is an interface here?), and there's probably a reason for that.
> How do you enter several members for a queue?  Asterisk seems to either
> rather crash than to create a queue, or to do nothing.
>
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Re: [asterisk-users] how to use a database

[Marcelo Terres]

https://wiki.asterisk.org/wiki/display/AST/Managing+Realtime+Databases+with+Alembic

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, 7 Dec 2018 at 13:34, hw  wrote:
>
> On 12/06/2018 10:26 PM, Marcelo Terres wrote:
> > The Asterisk source has a tool to create the db
>
> Which one is that?
>
>
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Re: [asterisk-users] how to use a database

[Marcelo Terres]

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.
> > >
> > > You need to create that for yourself.
> > >
> > > Asterisk can write to database tables, but doesn't help you set them
> up,
> > > for reasons I can't comment on.
> >
> > How do I know what the schema needs to be?  Does anybody have a scheme
> > for the queue_log table (and maybe others)?
>
> A Google search for "asterisk queue_log schema" gives results such as:
>
> http://work.mikeboylan.com/posts/2012/03/asterisk-queuelog-to-mysql.html
>
> https://gist.github.com/melvinlee/f57da3080dff40f71631
>
> https://stackoverflow.com/questions/30161384/asterisk-11-queue-log-to-mysql
>
> https://www.voip-info.org/asterisk-queuelog-on-mysql/
>
> > Do I get to see the queries that are being used to write this data,
>
> No.
>
> > or do I need to form them myself and enter them into some configuration
> file?
>
> No.
>
> > >> How dynamic are changes made in the database?
> > >
> > > If by "dynamic" you mean "quickly used" then the answer is
> "immediately".
> >
> > There's a note in some configuration file saying that dynamic extensions
> > are deprecated and suggesting to use func_odbc instead.  This func_odbc
> > seems to be the most awkward way anyone could think of for this, though.
>
> I use func_odbc in plenty of situations, but I'm not familiar with it
> being
> recommended for managing queues.
>
> Without seeing the "note in some configuration file" that you refer to,
> though,
> I don't know what to say about this.
>
> > >> For example, if I want to have an extension 'foobar' and want to ring
> > >> different devices depending on some factors (like time of day, for
> > >> example), can I modify the entry in the database for the device to
> ring
> > >> from 'bar' to 'baz', and baz will ring instead of bar from thereon?
> > >
> > > Yes.
> >
> > And IIUC the extension would use something like
> > 'Dial(SIP/ODBC_PICK_USER(...))' after defining a query for my
> > ..._PICK_USER function in func_odbc.conf to return what to dial
> > depending on the argument(s) supplied?
>
> No comment; I don't use this feature myself.
>
> > How do I make asterisk reload func_odbc.conf?  Or is that not needed?
>
> Not needed.  The whole point of configs in database tables is that they
> take
> effect immediately without having to tell Asterisk to reload anything.
>
> We did start off just talking about getting queue_log into a database
> table,
> though.
>
> > >> Is it possible to use configuration from both the database and the
> files
> > >> at the same time?
> > >
> > > Yes.
> >
> > So far, that seems to work fine :)
>
> Good.
>
>
> Antony.
>
> --
> Software development can be quick, high quality, or low cost.
>
> The customer gets to pick any two out of three.
>
>Please reply to the
> list;
>  please *don't* CC
> me.
>
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Re: [asterisk-users] Capture SIP all the time

[Marcelo Terres]

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, 5 Dec 2018, Saint Michael wrote:
>
> > Is there a way to configure the old SIP channel to stay in sip set debug
> > all the time, without human intervention and also at boot time, by
> > default?
>
> If your goal is capture all SIP traffic, there may be other tools better
> suited.
>
> For example, tcpdump, dumpcap, or pcapsipdump can capture SIP packets.
> pcapsipdump can even capture the RTP along with the SIP so you can listen
> to the call if that doesn't make your bosses and coworkers freak out.
>
> I like to capture all of the SIP traffic in a pool of files that
> expire after 30 days. Then, when somebody says 'hey, my call didn't
> connect yesterday' I have something to work with.
>
> sngrep is a great tool for searching for calls and displaying decoded
> dialogs.
>
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
>  https://www.linkedin.com/in/steve-edwards-4244281
>
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Re: [asterisk-users] figuring out what happened to a call

[Marcelo Terres]

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, asterisk hangs up.
>
> I want to know the following:
>
>
> + Who's calling?
>
> + What did the caller dial?
>
> + Is an inbound call being picked up or not?
>
> + Which phone picks it up?
>
> + Which of the phones that could be rung for the call are busy so that
>they can not be used to pick up the call?
>
> + How long has a call been going on for (for both the ones that were
>picked up and the ones that weren't)?
>
>
> I could only figure out who is calling and might be able to figure out
> what the caller dialed.  There seems to be no way to tell how a call is
> being dealt with, though.
>
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Re: [asterisk-users] failed to find existing extension

[Marcelo Terres]

This simple issues usually are the hardest to find. I know how it is.

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, 11 Sep 2018 at 09:47,  wrote:
>
> Indeed, just corrected it.
>
> It had to be something I did myself (as it was working before tidying
> up) but never thought such a mistake was hard to pinpoint.
> Thanks very much, now I can continue exploring * again.
>
> Hans
>
>
>
> On 2018-09-10 22:15, Antony Stone wrote:
> > 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 simple :)
> >
>
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>
> Check out the new Asterisk community forum at: https://community.asterisk.org/
>
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Re: [asterisk-users] failed to find existing extension

[Marcelo Terres]

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
> > one context and the one number you want to match:
> >
> > [0705680837]
> > exten => 31705680837,1,NooP( Incoming 31705680837 on CC)
> >   same => n,Answer();
> >   same => n,Background(dit_is_het_nummer_van_de_familie_witvliet)
> >   same => n,Wait(1)
> >   same => n,Hangup()
> >
> > 3. Of course, make sure your sip.conf points this provider at the
> > context
> > (although that does seem to be working from the console output).
> >
> > 4. Turn on debug logging, not just verbose.
> >
> >
> > Regards,
> >
> >
> > Antony.
> >
>
>
> Tried, but looks identical:
> (the comma's were infliced by my move from 'extension' towards 'same')
>
> Ok, removing most of my lines:
> pbx*CLI> dialplan show 0705680837
> [ Context '0705680837' created by 'pbx_config' ]
>'31705680837' =>  1. NooP( Incoming 31705680837 on CC)
> [extensions.conf:610]
>  2. Answer()
> [extensions.conf:611]
>  3.
> Background(dit_is_het_nummer_van_de_familie_witvliet)
> [extensions.conf:612]
>  4. Wait(1)
> [extensions.conf:613]
>  5. Hangup()
> [extensions.conf:614]
>
> -= 1 extension (5 priorities) in 1 context. =-
> pbx*CLI>
>
> Debug & verbose at 7, but still:pbx*CLI>
>
>== Using SIP RTP CoS mark 5
> > 0x7f49ac23c470 -- Strict RTP learning after remote address set
> to: 185.29.203.27:43036
> [Sep 10 13:16:17] NOTICE[835][C-002d]: chan_sip.c:26513
> handle_request_invite: Call from '77707057984' (185.29.203.27:5060) to
> extension '31705680837' rejected because extension not found in context
> '[0705680837]'.
> pbx*CLI>
>
> Part of my sip.conf
>
> [0705680837]
> type=peer
> context=[0705680837]
> nat=force_rport,comedia
> username=77707057984
> fromuser=0705680837
> host=185.29.203.27
> secret=*
> canreinvite=no
> dtmfmode=inband
> insecure=invite
> disallow=all
> allow=alaw,gsm,ulaw
> qualify=yes
> relaxdtmf=yes
> directmedia=no
>
>
>
> /var/log/asterisk/messages:
> [Sep  9 11:50:18] NOTICE[835][C-002a] chan_sip.c: Call from
> '77707057984' (185.29.203.27:5060) to extension '31705680837' rejected
> because extension not found in context '[0705680837]'.
> [Sep  9 11:50:49] NOTICE[835][C-002b] chan_sip.c: Call from
> '77707057984' (185.29.203.27:5060) to extension '31705680837' rejected
> because extension not found in context '[0705680837]'.
> [Sep 10 10:39:03] NOTICE[835] chan_sip.c: Peer '0705680837' is now
> Lagged. (2022ms / 2000ms)
> [Sep 10 10:39:13] NOTICE[835] chan_sip.c: Peer '0705680837' is now
> Reachable. (12ms / 2000ms)
> [Sep 10 13:09:54] NOTICE[835][C-002c] chan_sip.c: Call from
> '77707057984' (185.29.203.27:5060) to extension '31705680837' rejected
> because extension not found in context '[0705680837]'.
> [Sep 10 13:16:17] NOTICE[835][C-002d] chan_sip.c: Call from
> '77707057984' (185.29.203.27:5060) to extension '31705680837' rejected
> because extension not found in context '[0705680837]'.
>
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Re: [asterisk-users] How to steal an answered call?

[Marcelo Terres]

Unfortunately, all channels need to be handled by ARI stasis app,
otherwise, you can't use 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 Mon, 9 Jul 2018 at 11:22, 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.br
> https://twitter.com/mhterres
> https://linkedin.com/in/marceloterres
>
> On Mon, 9 Jul 2018 at 04:17, David Cunningham  
> wrote:
> >
> > Hello,
> >
> > I'm familiar with Pickup/PickupChan for taking a ringing call, but does 
> > anyone know how a phone can "steal" an already answered call from another 
> > phone? Our users have decided that call parking is too long-winded and 
> > don't want to use that.
> >
> > For example: phone A calls phone B, phone B answers the call, phone C dials 
> > something to "steal" the call from B, and finally A and C are talking.
> >
> > Searching on voip-info.org shows a "BristuffSteal" command but it's very 
> > out of date (Asterisk 1.2).
> >
> > Thanks in advance for any suggestions.
> >
> > Kind regards,
> >
> > --
> > David Cunningham, Voisonics Limited
> > http://voisonics.com/
> > USA: +1 213 221 1092
> > New Zealand: +64 (0)28 2558 3782
> > --
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> >
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> >
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Re: [asterisk-users] How to steal an answered call?

[Marcelo Terres]

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 Mon, 9 Jul 2018 at 04:17, David Cunningham  wrote:
>
> Hello,
>
> I'm familiar with Pickup/PickupChan for taking a ringing call, but does 
> anyone know how a phone can "steal" an already answered call from another 
> phone? Our users have decided that call parking is too long-winded and don't 
> want to use that.
>
> For example: phone A calls phone B, phone B answers the call, phone C dials 
> something to "steal" the call from B, and finally A and C are talking.
>
> Searching on voip-info.org shows a "BristuffSteal" command but it's very out 
> of date (Asterisk 1.2).
>
> Thanks in advance for any suggestions.
>
> Kind regards,
>
> --
> David Cunningham, Voisonics Limited
> http://voisonics.com/
> USA: +1 213 221 1092
> New Zealand: +64 (0)28 2558 3782
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>
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>
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Re: [asterisk-users] Problem using Android SIP-Client

[Marcelo Terres]

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 Bertoncello  wrote:
> 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 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 format gsm since our
> native format has changed to 0x8 (alaw)
>
> and I can't hear anything...
>
> This is the configuration of the user:
>
> [00491771234567]
> fullname = 00491771234567
> secret = MYVERYSECRET
> dahdichan = 1
> hassip = yes
> hasiax = no
> hash323 = no
> hasmanager = no
> callwaiting = no
> context = default
> host = dynamic
> dtmfmode=rfc2833
> canreinvite=no
> sendrpid=pai
> type=friend
> ;nat=force_rport,comedia
> nat=yes
> qualify=yes
> qualifyfreq=60
> ;transport=Auto
> avpf=no
> force_avp=no
> icesupport=no
> encryption=no
> callgroup=1
> pickupgroup=1
> dial=SIP/00491771234567
> allow = all
>
> Any idea?
> The user worked very well with my old mobile phone (Android 4), I __THINK__
> the problem happens since I use my new phone with Android 7...
>
> Thanks
> Luca Bertoncello
> (lucab...@lucabert.de)
>
>
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Re: [asterisk-users] Realtime pjsip issues

[Marcelo Terres]

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
(?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?)

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 15 September 2017 at 16:18, Bryant Zimmerman  wrote:
> Joshua
>
> We are using MariaDB as the database storage.
> We have recreated the database tables with alembic.
>
> Test 1:
>
> We enable tables for aors, auths and endpoints only.
> With cache turned off the end point registers successfully
> We have no way to get any feed back as pjsip show/list returns no objects
> found.
> pjsip send notify cmd endpoint -- does not work as it says there is no
> endpoint.
> endpoint can send a call as it appears to be registered, we have no way to
> confirm this form the console but calls come in.
>
>
> Test 2:
>
> We enable cache on the endpoints, auth and aors in the sorcery.conf
>
> endpoint/cache =
> memory_cache,object_lifetime_stale=600,object_lifetime_maximum=1800,expire_on_reload=yes,full_backend_cache=yes
> auth/cache=memory_cache,expire_on_reload=yes
> aor/cache =
> memory_cache,object_lifetime_stale=1500,object_lifetime_maximum=1800,expire_on_reload=yes,full_backend_cache=yes
>
> We now get an error:
>
> [2017-09-15 11:02:04] WARNING[3375]: res_pjsip_registrar.c:744
> registrar_on_rx_request: AOR '6162480909-300' has no configured
> max_contacts. Endpoint '6162480909-300' unable to register
> The aors entry has the max_contacts set to 1 but the error still occurs.
>
> pjsip show/list shows the endpoint shows endpoints, aors, auths  but
> registration fails
>
>
> Test 3:
>
> We enable cache on the endpoints, auth and aors in the sorcery.conf
>
> endpoint/cache =
> memory_cache,object_lifetime_stale=600,object_lifetime_maximum=1800,expire_on_reload=yes
> auth/cache=memory_cache,expire_on_reload=yes
> aor/cache =
> memory_cache,object_lifetime_stale=1500,object_lifetime_maximum=1800,expire_on_reload=yes
>
> Endpoint registers
> pjsip show/list endpoints works the first time and fails there after.
>
> UBNTU-ROSSI-GUEST*CLI> pjsip show endpoints
>  Endpoint:  
>   
> I/OAuth:
> 
> Aor:  
> 
>   Contact:   
>  
>   Transport:
> 
>Identify:
> 
> Match:  
> Channel:  
>   
> Exten:   CLCID: 
> ==
>  Endpoint:  6162480909-300   Not in use
> 0 of inf
>  InAuth:  6162480909-300/6162480909-300
> Aor:  6162480909-300 1
>   Contact:  6162480909-300/sip:6162480909-300@192.168. 0475d46ff2
> Unknown nan
>   Transport:  udp-nat   udp  0  0  0.0.0.0:5060
>
> Objects found: 1
> UBNTU-ROSSI-GUEST*CLI> pjsip show endpoints
> No objects found.
>
> pjsip show/list shows the endpoint fails ever time after the first.
>
> Test 4:
>
> Test 1: with the addition of the contacts entry as realtime in sorcery.conf
> We get error on registration attempt:
>
> [2017-09-15 11:16:07] WARNING[3591]: res_config_odbc.c:120 custom_prepare:
> SQL Prepare failed! [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 (?,
> ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?)]
> [2017-09-15 11:16:07] ERROR[3591]: res_pjsip_registrar.c:432
> register_aor_core: Unable to bind contact
> 'sip:6162480909-300@192.168.201.105:59758' to AOR '6162480909-300'
>
> Registration has failed at this point.
>
>
> I can offer the following:
> A dump of the database schema that alembic is creating.
> extconfig.config
> sorcery.conf
>
> Thanks
> Bryant
>
> 
> From: "Joshua Colp" 
> Sent: Friday, September 15, 2017 9:56 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Realtime pjsip issues
>
> On Fri, Sep 15, 2017, at 10:37 AM, Bryant Zimmerman wrote:
>> Joshua
>>
>> That is the interesting part of it. We took our configs and database
>> tables from our working 13.12.2 deployments and tried to use them with
>> our
>> new 13.17.1 deployments and we are having issues where the tables are not
>> working. On the new server asterisk keeps saying it can't find the AORS
>> entries when we purge the sorcery memory cache it starts finding the aors
>> but then it says it cant find the auths.
>>
>> The wired thing is when it says it can't find the aors and auths entries
>> it does not show it is looking for the values in the aors and auth fields
>> from the endpoints tables. 

Re: [asterisk-users] Asterisk 11.25.2

[Marcelo Terres]

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 09:56 AM, Jerry Geis wrote:
> > My setup using 11.25.1 was working. When I installed 11.25.2 I now get
> > "sort of" working.
> >
> > I am using NAT in the setup. When I have an internal phone and call out I
> > get audio both ways.
> > But when I call IN my phone rings but I have no audio.
> >
> > Is there a new setting I need to tweek ?
>
> You can try setting "strictrtp" to "no" in rtp.conf and seeing if that
> resolves the issue. If it does then getting a packet capture of the
> traffic could confirm why we are dropping the media. It may be that the
> source is changing without telling us, which the security fix protects
> against.
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
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>
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Re: [asterisk-users] Asterisk Voicemail changes

[Marcelo Terres]

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 10:50, Tim Turpin  wrote:
> Thank you.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of J Montoya or A 
> J Stiles
> Sent: Friday, September 01, 2017 3:47 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Asterisk Voicemail changes
>
> On Friday 01 Sep 2017, Tim Turpin wrote:
>> Is there a way that I can modify the source code for the voicemail
>> application?  I need to change some of the options in the user’s
>> interface to make it work like an existing system that I’m replacing.
>
> $ vi /usr/src/asterisk-*/apps/app_voicemail.c
>
> --
> JM or AJS
>
> Note:  Originating address only accepts e-mail from list!  If replying off- 
> list, change address to asterisk1list at earthshod dot co dot uk .
>
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Re: [asterisk-users] VoIP monitor and multiple RTP streams

[Marcelo Terres]

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 <mhter...@gmail.com>
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres

On 14 August 2017 at 17:52, Dovid Bender <do...@telecurve.com> wrote:

> Marcelo,
>
> You need to look at the box changing the SSRC and figure out why it's
> changing it. Where are you seeing the multiple rows 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
>> 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 address and port).
>>
>> Example:
>>
>> "Source Address","Source Port","Destination Address","Destination
>> Port","SSRC","Payload","Packets","Lost","Max Delta (ms)","Max
>> Jitter","Mean Jitter","Status"
>> "6X.XXX.XXX.XXX",34170,"1XX.XXX.XXX.XXX",10602,277011456,"g7
>> 11A",7289,0,21.3036449,21.265543809819981,0.073286945955809715,""
>> "1XX.XXX.XXX.XXX",10602,"6X.XXX.XXX.XXX",34170,2020146713,"g
>> 711A",2099,0,36.2968661,2.9025967411766738,0.97877393850963945,""
>> "1XX.XXX.XXX.XXX",10602,"6X.XXX.XXX.XXX",34170,325951803,"g7
>> 11A",4949,0,41.8790815,4.5846492231155924,1.0537488536922062,""
>>
>> The only thing that I could notice is that the first packet that had the
>> new SSRC (325951803) has the marker bit on, but I could not find a reason
>> for the SSRC change.
>>
>> Any ideas of what could be causing that?
>>
>> Thanks.
>>
>> 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
>>
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>>
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>>
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>>
>
>
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[asterisk-users] VoIP monitor and multiple RTP streams

[Marcelo Terres]

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 address and port).

Example:

"Source Address","Source Port","Destination Address","Destination
Port","SSRC","Payload","Packets","Lost","Max Delta (ms)","Max Jitter","Mean
Jitter","Status"
"6X.XXX.XXX.XXX",34170,"1XX.XXX.XXX.XXX",10602,277011456,"g711A",7289,0,21.
3036449,21.265543809819981,0.073286945955809715,""
"1XX.XXX.XXX.XXX",10602,"6X.XXX.XXX.XXX",34170,2020146713,"g711A",2099,0,36.
2968661,2.9025967411766738,0.97877393850963945,""
"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 packet that had the
new SSRC (325951803) has the marker bit on, but I could not find a reason
for the SSRC change.

Any ideas of what could be causing that?

Thanks.

Regards,

Marcelo H. Terres 
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres
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Re: [asterisk-users] Change OS from CentOS 6 to 7

[Marcelo Terres]

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 192.168.5.25 is not responding...
>
> On 4 Aug 2017 8:28 pm, "Jerry Geis" <jerry.g...@gmail.com> 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 PT=ITU-T G.711
>> PCMU, SSRC=0x6A3E0AF1, Seq=28403, Time=73440
>> 963 16.190381989 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711
>> PCMU, SSRC=0x6A3E0AF1, Seq=28404, Time=73600
>> 964 16.210387990 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711
>> PCMU, SSRC=0x6A3E0AF1, Seq=28405, Time=73760
>> 965 16.230353530 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711
>> PCMU, SSRC=0x6A3E0AF1, Seq=28406, Time=73920
>> 966 16.250362957 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711
>> PCMU, SSRC=0x6A3E0AF1, Seq=28407, Time=74080
>> 967 16.270375476 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711
>> PCMU, SSRC=0x6A3E0AF1, Seq=28408, Time=74240
>> 968 16.290361413 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711
>> PCMU, SSRC=0x6A3E0AF1, Seq=28409, Time=74400
>> 969 16.310380701 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711
>> PCMU, SSRC=0x6A3E0AF1, Seq=28410, Time=74560
>> 970 16.330372239 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711
>> PCMU, SSRC=0x6A3E0AF1, Seq=28411, Time=74720
>> 971 16.350381239 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711
>> PCMU, SSRC=0x6A3E0AF1, Seq=28412, Time=74880
>> 972 16.370378599 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711
>> PCMU, SSRC=0x6A3E0AF1, Seq=28413, Time=75040
>> 973 16.390376810 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711
>> PCMU, SSRC=0x6A3E0AF1, Seq=28414, Time=75200
>> 974 16.410491478 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711
>> PCMU, SSRC=0x6A3E0AF1, Seq=28415, Time=75360
>> 975 16.430377388 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711
>> PCMU, SSRC=0x6A3E0AF1, Seq=28416, Time=75520
>> 976 16.450314213 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711
>> PCMU, SSRC=0x6A3E0AF1, Seq=28417, Time=75680
>>
>> Jerry
>>
>>
>> On Fri, Aug 4, 2017 at 3:04 PM, Jerry Geis <jerry.g...@gmail.com> wrote:
>>
>>> Hi all,
>>>
>>> I had a box with CentOS 6... I backed up, installed C7. restored my
>>> backups,
>>> put back on asterisk 11.25.1 put back my configs and ran a test... All
>>> seems good, my device activates like audio is ready to come out - but no
>>> audio. CLI looks like everything is running - just no audio...
>>>
>>> The device is registered.
>>> no errors on startup.
>>> using the same hardware as before all on local network no NAT issues
>>> I tried turning off the firewall - not help.
>>>
>>>
>>> Thoughts on why no audio, or things to look at. ?
>>>
>>> Thanks
>>>
>>> Jerry
>>>
>>
>>
>> --
>> _
>> -- 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:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
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Re: [asterisk-users] Change OS from CentOS 6 to 7

[Marcelo Terres]

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 PT=ITU-T G.711
> PCMU, SSRC=0x6A3E0AF1, Seq=28403, Time=73440
> 963 16.190381989 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711
> PCMU, SSRC=0x6A3E0AF1, Seq=28404, Time=73600
> 964 16.210387990 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711
> PCMU, SSRC=0x6A3E0AF1, Seq=28405, Time=73760
> 965 16.230353530 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711
> PCMU, SSRC=0x6A3E0AF1, Seq=28406, Time=73920
> 966 16.250362957 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711
> PCMU, SSRC=0x6A3E0AF1, Seq=28407, Time=74080
> 967 16.270375476 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711
> PCMU, SSRC=0x6A3E0AF1, Seq=28408, Time=74240
> 968 16.290361413 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711
> PCMU, SSRC=0x6A3E0AF1, Seq=28409, Time=74400
> 969 16.310380701 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711
> PCMU, SSRC=0x6A3E0AF1, Seq=28410, Time=74560
> 970 16.330372239 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711
> PCMU, SSRC=0x6A3E0AF1, Seq=28411, Time=74720
> 971 16.350381239 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711
> PCMU, SSRC=0x6A3E0AF1, Seq=28412, Time=74880
> 972 16.370378599 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711
> PCMU, SSRC=0x6A3E0AF1, Seq=28413, Time=75040
> 973 16.390376810 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711
> PCMU, SSRC=0x6A3E0AF1, Seq=28414, Time=75200
> 974 16.410491478 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711
> PCMU, SSRC=0x6A3E0AF1, Seq=28415, Time=75360
> 975 16.430377388 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711
> PCMU, SSRC=0x6A3E0AF1, Seq=28416, Time=75520
> 976 16.450314213 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711
> PCMU, SSRC=0x6A3E0AF1, Seq=28417, Time=75680
>
> Jerry
>
>
> On Fri, Aug 4, 2017 at 3:04 PM, Jerry Geis  wrote:
>
>> Hi all,
>>
>> I had a box with CentOS 6... I backed up, installed C7. restored my
>> backups,
>> put back on asterisk 11.25.1 put back my configs and ran a test... All
>> seems good, my device activates like audio is ready to come out - but no
>> audio. CLI looks like everything is running - just no audio...
>>
>> The device is registered.
>> no errors on startup.
>> using the same hardware as before all on local network no NAT issues
>> I tried turning off the firewall - not help.
>>
>>
>> Thoughts on why no audio, or things to look at. ?
>>
>> Thanks
>>
>> Jerry
>>
>
>
> --
> _
> -- 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:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
-- 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:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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Re: [asterisk-users] Asterisk 13.16.0 segfault

[Marcelo Terres]

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" <cur...@telecomab.mx> wrote:

> 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.mundoopensource.com.br
> https://www.mundoopensource.com.br
> https://twitter.com/mhterres
> https://linkedin.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?
>>
>> On 18 Jul 2017 3:52 pm, "Carlos Chavez" <cur...@telecomab.mx> wrote:
>>
>>> I am getting frequent segfaults on a new Asterisk installation.  So far
>>> the only message I see is:
>>>
>>> Jul 18 09:02:42 pbxbogota kernel: asterisk[26799]: segfault at 188 ip
>>> 7fb2d535723f sp 7fb25a11b5c0 error 4 in
>>> libasteriskpj.so.2[7fb2d52e5000+18]
>>> Jul 18 09:17:00 pbxbogota kernel: asterisk[27453]: segfault at 188 ip
>>> 7f4afea0c23f sp 7f4a7f7e35c0 error 4 in
>>> libasteriskpj.so.2[7f4afe99a000+18]
>>> Jul 18 09:22:57 pbxbogota kernel: asterisk[28471]: segfault at 188 ip
>>> 7f2eb611923f sp 7f2e3aec25c0 error 4 in
>>> libasteriskpj.so.2[7f2eb60a7000+18]
>>> Jul 18 09:25:49 pbxbogota kernel: asterisk[28949]: segfault at 188 ip
>>> 7fc5758dd23f sp 7fc4fa6245c0 error 4 in
>>> libasteriskpj.so.2[7fc57586b000+18]
>>> Jul 18 09:31:17 pbxbogota kernel: asterisk[29203]: segfault at 188 ip
>>> 7f5f29abb23f sp 7f5eae8285c0 error 4 in
>>> libasteriskpj.so.2[7f5f29a49000+18]
>>>
>>> Since this is a Freepbx distro does could the problem be related to
>>> their flavor of Asterisk?  I have several other plain Asterisk servers
>>> running on this version without any problems.  Any recommendations on how
>>> to debug this?
>>>
>>> My solution to this is going to be compiling Asterisk manually
>> instead of using their pre packaged version as debugging will take a lot
>> more time.
>>
>
> The Freepbx distro still uses a separate pjproject as far as I know.
> When I compile my own I always use the bundled version now.
>
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez+52 (55)8116-9161 <+52%2055%208116%209161>
>
>
> --
> _
> -- 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:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
-- 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:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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Re: [asterisk-users] corosync and Asterisk 13

[Marcelo Terres]

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 <mhter...@gmail.com>
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres

On 20 July 2017 at 02:29, Ryan, Travis <ry...@oscarwinski.com> wrote:

> I’m pretty sure it’s an issue with Ubuntu 16.04 and the version (2.3.5)
> that it installs and Asterisk’s ./configure doesn’t recognize it. Any ideas
> how to figure this out?
>
>
>
> I have proved it on two separate 16.04 servers and neither shows up. But
> 14.04 and Asterisk 13 does show.
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] *On Behalf Of *Ryan, Travis
> *Sent:* Wednesday, July 19, 2017 3:26 PM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
> *Subject:* Re: [asterisk-users] corosync and Asterisk 13
>
>
>
> I have an 14.04 server with Asterisk 13.5 that will recognize it. It has
> corosync and corosync-dev version 2.3.3 on it.
>
>
>
> The one I can’t get to recognize it is 16.04 ubuntu, with corosync and
> -dev version 2.3.5. Won’t be recognized by ./configure for Asterisk 13.17,
> 13.16, or Asterisk Certified 13.13.
>
>
>
> I REALLY need some help figuring this out. 
>
>
>
> Thanks!
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com <asterisk-users-boun...@lists.digium.com>] *On
> Behalf Of *Ryan, Travis
> *Sent:* Wednesday, July 19, 2017 1:13 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
> *Subject:* Re: [asterisk-users] corosync and Asterisk 13
>
>
>
> Anyone else using corosync with Asterisk 13 and Ubuntu 16.04 or higher?
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com <asterisk-users-boun...@lists.digium.com>] *On
> Behalf Of *Ryan, Travis
> *Sent:* Wednesday, July 19, 2017 10:13 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion <
> 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
> Behalf Of *Marcelo Terres
> *Sent:* Wednesday, July 19, 2017 10:05 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
> *Subject:* Re: [asterisk-users] corosync and Asterisk 13
>
>
>
> Did you installed the dev package?
> corosync-dev
>
>
>
>
>
>
> 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 19 July 2017 at 14:46, Ryan, Travis <ry...@oscarwinski.com> wrote:
>
> I want to use corosync and installed it via ubuntu repository. I guess
> there is a version 1 and 2 of corosync. For some reason ./configure for
> Asterisk (13) isn’t recognizing I have corosync installed. I can’t enable
> the res_corosync module in menuselect.
>
>
>
> Any ideas?
>
>
>
> Thanks!
>
>
>
> Travis
>
>
> --
> _
> -- 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:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --
> _
> -- 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:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
-- 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:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk 13.16.0 segfault

[Marcelo Terres]

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.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.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?
>
> On 18 Jul 2017 3:52 pm, "Carlos Chavez" <cur...@telecomab.mx> wrote:
>
>> I am getting frequent segfaults on a new Asterisk installation.  So far
>> the only message I see is:
>>
>> Jul 18 09:02:42 pbxbogota kernel: asterisk[26799]: segfault at 188 ip
>> 7fb2d535723f sp 7fb25a11b5c0 error 4 in
>> libasteriskpj.so.2[7fb2d52e5000+18]
>> Jul 18 09:17:00 pbxbogota kernel: asterisk[27453]: segfault at 188 ip
>> 7f4afea0c23f sp 7f4a7f7e35c0 error 4 in
>> libasteriskpj.so.2[7f4afe99a000+18]
>> Jul 18 09:22:57 pbxbogota kernel: asterisk[28471]: segfault at 188 ip
>> 7f2eb611923f sp 7f2e3aec25c0 error 4 in
>> libasteriskpj.so.2[7f2eb60a7000+18]
>> Jul 18 09:25:49 pbxbogota kernel: asterisk[28949]: segfault at 188 ip
>> 7fc5758dd23f sp 7fc4fa6245c0 error 4 in
>> libasteriskpj.so.2[7fc57586b000+18]
>> Jul 18 09:31:17 pbxbogota kernel: asterisk[29203]: segfault at 188 ip
>> 7f5f29abb23f sp 7f5eae8285c0 error 4 in
>> libasteriskpj.so.2[7f5f29a49000+18]
>>
>> Since this is a Freepbx distro does could the problem be related to their
>> flavor of Asterisk?  I have several other plain Asterisk servers running on
>> this version without any problems.  Any recommendations on how to debug
>> this?
>>
>> My solution to this is going to be compiling Asterisk manually
> instead of using their pre packaged version as debugging will take a lot
> more time.
>
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez+52 (55)8116-9161 <+52%2055%208116%209161>
>
>
> --
> _
> -- 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:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
-- 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:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] corosync and Asterisk 13

[Marcelo Terres]

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  wrote:

> I want to use corosync and installed it via ubuntu repository. I guess
> there is a version 1 and 2 of corosync. For some reason ./configure for
> Asterisk (13) isn’t recognizing I have corosync installed. I can’t enable
> the res_corosync module in menuselect.
>
>
>
> Any ideas?
>
>
>
> Thanks!
>
>
>
> Travis
>
> --
> _
> -- 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:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Integration of Google Speech API V2

[Marcelo Terres]

Development snapshots that support Speech API v2 are available in either zip
<https://github.com/zaf/asterisk-speech-recog/zipball/master> or tar
<https://github.com/zaf/asterisk-speech-recog/tarball/master> formats.

A version that supports the new Cloud Speech API
<https://cloud.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, "Rahul 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?
>
>
>
> On Wed, Jul 19, 2017 at 3:41 PM, Marcelo Terres <mhter...@gmail.com>
> wrote:
>
>> Did you already tried the cloud_api branch?
>>
>> 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 19 July 2017 at 10:17, Rahul MathuR <rahul.ultim...@gmail.com> wrote:
>>
>>> Hi Jonathan
>>>
>>> Thanks !
>>> That would indeed be wonderful, at this point I really do not care
>>> whether I need to use Python or Lua or JS.
>>>
>>> I was following http://zaf.github.io/asterisk-speech-recog/
>>> but hit a road end with (for the lack of sane word ) copulating Google's
>>> Key
>>>
>>>
>>>
>>> On Wed, Jul 19, 2017 at 2:28 PM, Jonathan H <lardconce...@gmail.com>
>>> wrote:
>>>
>>>> Yes! But I can only tell you if you can use Python, as I used Google's
>>>> own demo code.
>>>>
>>>> If you can hold on for half an hour, I'll remove personal info and put
>>>> a version up on Github if you're interested?
>>>>
>>>>
>>>> On 19 July 2017 at 09:37, Rahul MathuR <rahul.ultim...@gmail.com>
>>>> wrote:
>>>>
>>>>> Hi,
>>>>>
>>>>> I'm trying to integrate Google cloud speech recognition v2 in it. I
>>>>> can get the audio recorded, have created Service key and API key but
>>>>> whenever I try to access it, I just get 403 access denied. I am at my wits
>>>>> end here.
>>>>>
>>>>> Has anybody tried it ? were you successful ? Could you please guide me
>>>>> how to do it ?
>>>>> I'll be grateful to you if this works !
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Warm Regds.
>>>>> MathuRahul
>>>>>
>>>>> --
>>>>> _
>>>>> -- 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:
>>>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>>>
>>>>> asterisk-users mailing list
>>>>> To UNSUBSCRIBE or update options visit:
>>>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>
>>>>
>>>>
>>>> --
>>>> _
>>>> -- 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:
>>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>>
>>>> asterisk-users mailing list
>>>> To UNSUBSCRIBE or update options visit:
>>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>
>>>
>>>
>>>
>>> --
>>> Warm Regds.
>>> MathuRahul
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> Check out the new Asterisk community forum at:
>>> https:/

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

[Marcelo Terres]

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" <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?
>
>
>
> On Wed, Jul 19, 2017 at 3:41 PM, Marcelo Terres <mhter...@gmail.com>
> wrote:
>
>> Did you already tried the cloud_api branch?
>>
>> 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 19 July 2017 at 10:17, Rahul MathuR <rahul.ultim...@gmail.com> wrote:
>>
>>> Hi Jonathan
>>>
>>> Thanks !
>>> That would indeed be wonderful, at this point I really do not care
>>> whether I need to use Python or Lua or JS.
>>>
>>> I was following http://zaf.github.io/asterisk-speech-recog/
>>> but hit a road end with (for the lack of sane word ) copulating Google's
>>> Key
>>>
>>>
>>>
>>> On Wed, Jul 19, 2017 at 2:28 PM, Jonathan H <lardconce...@gmail.com>
>>> wrote:
>>>
>>>> Yes! But I can only tell you if you can use Python, as I used Google's
>>>> own demo code.
>>>>
>>>> If you can hold on for half an hour, I'll remove personal info and put
>>>> a version up on Github if you're interested?
>>>>
>>>>
>>>> On 19 July 2017 at 09:37, Rahul MathuR <rahul.ultim...@gmail.com>
>>>> wrote:
>>>>
>>>>> Hi,
>>>>>
>>>>> I'm trying to integrate Google cloud speech recognition v2 in it. I
>>>>> can get the audio recorded, have created Service key and API key but
>>>>> whenever I try to access it, I just get 403 access denied. I am at my wits
>>>>> end here.
>>>>>
>>>>> Has anybody tried it ? were you successful ? Could you please guide me
>>>>> how to do it ?
>>>>> I'll be grateful to you if this works !
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Warm Regds.
>>>>> MathuRahul
>>>>>
>>>>> --
>>>>> _
>>>>> -- 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:
>>>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>>>
>>>>> asterisk-users mailing list
>>>>> To UNSUBSCRIBE or update options visit:
>>>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>
>>>>
>>>>
>>>> --
>>>> _
>>>> -- 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:
>>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>>
>>>> asterisk-users mailing list
>>>> To UNSUBSCRIBE or update options visit:
>>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>
>>>
>>>
>>>
>>> --
>>> Warm Regds.
>>> MathuRahul
>>>
>>> --
>>> _
>>> -- 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:
>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>> --
>&

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

[Marcelo Terres]

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  wrote:

> Hi Jonathan
>
> Thanks !
> That would indeed be wonderful, at this point I really do not care whether
> I need to use Python or Lua or JS.
>
> I was following http://zaf.github.io/asterisk-speech-recog/
> but hit a road end with (for the lack of sane word ) copulating Google's
> Key
>
>
>
> On Wed, Jul 19, 2017 at 2:28 PM, Jonathan H 
> wrote:
>
>> Yes! But I can only tell you if you can use Python, as I used Google's
>> own demo code.
>>
>> If you can hold on for half an hour, I'll remove personal info and put a
>> version up on Github if you're interested?
>>
>>
>> On 19 July 2017 at 09:37, Rahul MathuR  wrote:
>>
>>> Hi,
>>>
>>> I'm trying to integrate Google cloud speech recognition v2 in it. I can
>>> get the audio recorded, have created Service key and API key but whenever I
>>> try to access it, I just get 403 access denied. I am at my wits end here.
>>>
>>> Has anybody tried it ? were you successful ? Could you please guide me
>>> how to do it ?
>>> I'll be grateful to you if this works !
>>>
>>>
>>>
>>> --
>>> Warm Regds.
>>> MathuRahul
>>>
>>> --
>>> _
>>> -- 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:
>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>> --
>> _
>> -- 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:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> Warm Regds.
> MathuRahul
>
> --
> _
> -- 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:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Asterisk 13.16.0 segfault

[Marcelo Terres]

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 pbxbogota kernel: asterisk[26799]: segfault at 188 ip
> 7fb2d535723f sp 7fb25a11b5c0 error 4 in
> libasteriskpj.so.2[7fb2d52e5000+18]
> Jul 18 09:17:00 pbxbogota kernel: asterisk[27453]: segfault at 188 ip
> 7f4afea0c23f sp 7f4a7f7e35c0 error 4 in
> libasteriskpj.so.2[7f4afe99a000+18]
> Jul 18 09:22:57 pbxbogota kernel: asterisk[28471]: segfault at 188 ip
> 7f2eb611923f sp 7f2e3aec25c0 error 4 in
> libasteriskpj.so.2[7f2eb60a7000+18]
> Jul 18 09:25:49 pbxbogota kernel: asterisk[28949]: segfault at 188 ip
> 7fc5758dd23f sp 7fc4fa6245c0 error 4 in
> libasteriskpj.so.2[7fc57586b000+18]
> Jul 18 09:31:17 pbxbogota kernel: asterisk[29203]: segfault at 188 ip
> 7f5f29abb23f sp 7f5eae8285c0 error 4 in
> libasteriskpj.so.2[7f5f29a49000+18]
>
> Since this is a Freepbx distro does could the problem be related to their
> flavor of Asterisk?  I have several other plain Asterisk servers running on
> this version without any problems.  Any recommendations on how to debug
> this?
>
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez
> dCAP #1349
> +52 (55)8116-9161
>
> --
> _
> -- 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:
>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] BLF sharing between Asterisk 11 and 13

[Marcelo Terres]

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 BLF, and assigning the hints on my Cisco 525G2 phones.
>
>
>
> I’ve upgraded to 13 on one of these servers, and now can’t share BLF. I
> get something like…
>
>
>
> [2017-07-15 22:35:49] NOTICE[3483]: res_pjsip/pjsip_distributor.c:347
> log_unidentified_request: Request from '"Travis Ryan" '
> failed for '10.1.2.XXX:5060' (callid: 8c79c540-c0710...@10.1.2.xxx) - No
> matching endpoint found
>
>
>
> How do I make a server allow another extension on another server see it’s
> BLF/Hints?
>
> --
> _
> -- 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:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Asterisk realtime - Error with index length in alembic script

[Marcelo Terres]

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

On 12 July 2017 at 13:11, Floimair Florian  wrote:

> Nevermind guys!
>
> I just found out the solution myself:
>
> MariaDB in Debian uses utf8mb4 as default character set (see here:
> https://mariadb.com/kb/en/mariadb/differences-in-
> mariadb-in-debian-and-ubuntu/).
>
> I needed to uncomment the lines with utf8mb4 in /etc/mysql/maria.db.conf
> in the following files:
>
> 50-client.cnf (1 line)
> 50-mysql-clients.cnf (1 line)
> 50-server.cnf (2 lines)
>
>
>
> With best regards
>
> Florian Floimair
>
>
> -Ursprüngliche Nachricht-
> Von: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] Im Auftrag von Floimair Florian
> Gesendet: Mittwoch, 12. Juli 2017 13:50
> An: Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
> Betreff: [asterisk-users] Asterisk realtime - Error with index length in
> alembic script
>
> Hi!
>
> I just tried setting up Asterisk realtime database following the wiki
> article https://wiki.asterisk.org/wiki/display/AST/Setting+up+
> PJSIP+Realtime on a Debian 9 machine (which switched from MyQSL to
> MariaDB).
>
> One has to install mariadb-plugin-connect, python-mysqldb and alembic
> packages (alembic does not work when installed via pip).
> Additionally - since MariaDB by default does not have a root user password
> set and running mysql -u root requires sudo as well - you need to execute
> the following:
> sudo mysql_secure_installation
> sudo mysql_upgrade -p --force
>
> So far so good. I run into problems when running alembic when I get to the
> following change:
> https://linkprotect.cudasvc.com/url?a=https://e96a0b8071c_
> increase_pjsip_column_size.py=E,1,dGJHzJtuX7eYDELI39tEC4ecYafZjs
> CUjWDL5p09DOWe28cNAbd_GFmJLD2jBZfffS-vYPvUH1CUUjR7gX1rtdvm5NFCTV_
> tDVCtQerGg6RZ=1
> mariadb fails this operation with error "Specified key was too long; max
> key length is 767 bytes" when it tries to increase some fields to
> varchar(255).
>
> Any idea how to solve this? Do I maybe have to switch to a different
> encoding for this to work?
>
> Thanks in advance
>
>
>
> With best regards
>
> Florian Floimair
>
> COMMEND INTERNATIONAL GMBH
> A-5020 Salzburg, Saalachstraße 51
> https://linkprotect.cudasvc.com/url?a=http://www.commend.com=E,1,
> AMmXtJHsI4hmbBwyuM3M1xbRoGx-jiTaCiP7NxokUuMnadAIjb8xkI9P1ki_
> t2xzN78KaSN07DIpAamHT5VUTD5fbWuRqBVMk8ZAvCXz4t9wk89RgGU28EY,=1
>
> Security and Communication by Commend
>
> FN 178618z | LG Salzburg
>
> -Ursprüngliche Nachricht-
> Von: https://linkprotect.cudasvc.com/url?a=https://asterisk-
> users-boun...@lists.digium.com=E,1,7s7_D_Myc9BrXsqexg-
> b_jeGW99IlnqrhZCMhGKzBBE0m7-4lzl4Pqf0FBhPDU7YvysBh3XyuK7jq
> olYZryc5Pv214OOwiAf7rFVSlR6XZKzTS_0oyqQLA,,=1
> https://linkprotect.cudasvc.com/url?a=https://[mailto:
> asterisk-users-boun...@lists.digium.com=E,1,
> b02t9WMuMstwiWAHz0XrrZjHTVSQwnEy5yxXJi5pqNE6eqJ_ZzijQ4_
> PsoLa3tnaco3BYXQ5Ck2OHfmk_Dm4EHbE77z220o2c-VzuvBbEcq7PCY,=1] Im
> Auftrag von Thomas
> Gesendet: Montag, 10. Juli 2017 14:07
> An: https://linkprotect.cudasvc.com/url?a=https://asterisk-
> us...@lists.digium.com=E,1,LICqKTGOt1JJCqd7cLtDeAYTRlaeW-
> 0IaAjeofhcEGlqHiUa9FX1v_0Z61fjn6Cglc1LwJESdZ5CsnB1ZeUM
> Pn7gV2z5agkz3kh8onHV0Oxnmn9c4DjH6CBU=1
> Betreff: [asterisk-users] ConfBridge increase talking volume as standard
>
> Hello,
>
> is it possible to increase talking volume for caller in ConfBridge as
> standard without need to press buttons after joining an conference room.
>
> best regards
> Thomas
>
> --
> _
> -- Bandwidth and Colocation Provided by https://linkprotect.cudasvc.
> com/url?a=http://www.api-digital.com=E,1,3cxgVpYz6rj8HJh87TiGg9vmNONVb8
> R9gj8CUtsKQo4J7XZd3A8P5Q2lgkuqRb7I2h0ILWV9fb2VVtM_
> fLD5Wkjc1g637rszrIIFlYV5gEq-t1OY5td0MjI,=1 --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>https://linkprotect.cudasvc.com/url?a=http://lists.digium.
> com/mailman/listinfo/asterisk-users=E,1,8wdFEBPawJhe3Q7z9CDJspktcWZmak
> 6_F7Qwy-KlgT8Y3RKfi8rz6GZboEEYt3CZnt4-JjH7gKYVY79x72M6dUv0yXmSnCduYV
> FcBMK4FJNKC_QOHqI9aSjt=1
>
> --
> _
> -- Bandwidth and Colocation Provided by https://linkprotect.cudasvc.
> com/url?a=http://www.api-digital.com=E,1,VVFR1vt-VTtY8TkmHZUWfvj_
> 

Re: [asterisk-users] AMI column widths

[Marcelo Terres]

You are using AMI to run CLI commands and that's the problem.

Try to use the equivalent AMI actions to get the information that you want.

My suggestion : get all channels in use (CoreShowChannels) and then filter
just the SIP, since there is not an action to do exactly what you need.


On 8 Jul 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 console?

I'm using AMI.

I have a connection to the Asterisk server on port 5038, initated with:

Action: Login
Username: x
Secret: y
Events: off

I receive back:

Response: Success
Message: Authentication accepted

I then issue:

Action: Command
Command: SIP show channels

and I get back:

Response: Follows
Privilege: Command
Peer User/ANR Call ID  Format
 Hold
Last MessageExpiry Peer

plus the data I quoted previously.

> Running commands on console and parsing the output is the worst way to
> obtain data, first because it is not easily parseable.

And also because it is very inefficient with connection setups, I believe.

> Second, it doesn't show you all data.
>
> Third, you can have these truncate problems, because that's not intention
> of CLI.
>
> Using proper AMI Actions you will probably achieve your goals
>
> https://wiki.asterisk.org/wiki/display/AST/AMI+Actions

Hm, I don't see anything there which will give me a list of the SIP channels
currently in use - what command should I be using for that?


Thanks,


Antony.

> On 7 Jul 2017 10:32 pm, Antony Stone wrote:
>
> Hi.
>
> I'm trying to get a list of the channels currently in use on an Asterisk
> server (1.8.32.1 if it matters) over AMI.
>
> I send the command "sip show channels", and I get back a response along
the
> lines of (* used to protect the innocent...):
>
> Peer User/ANR Call ID  Format   Hold
>  Last MessageExpiry Peer
> *8.22.*0.340203564  0221e874158bb62  0x4 (ulaw)   No
>  Tx: ACKSIPtrunkNu
> *.1*.19.70 (None)   2021549013484-1  0x0 (nothing)No
>  Rx: OPTIONS
> *.34.*.208 200101   712173267@192.1  0x4 (ulaw)   No
>  Rx: ACK200101
> *.1*.19.70 (None)   149831567021051  0x0 (nothing)No
>  Rx: REGISTER   
>
> So, firstly, the "Call ID" column is clearly truncated, because it should
> show more than is indicated above,
> but more importantly for me, the "Peer" column is truncated, and what
> should show as "SIPtrunkNumber8"
> is only shown as "SIPtrunkNu".
>
> How can I get the full column widths of these items shown in the output?
>
> Note that it is not a solution just to say "don't call it
> 'SIPtrunkNumber8'; call it 'SIPtrunk8' instead", because
> this name has also been modified slightly to conceal the real name of the
> trunk, which is actually longer
> than "SIPtrunkNo8", but still with the most important information at the
> end.
>
> What I'm looking for is how to get the *full* details of all the channels
> shown.
>
> I have checked, and there is no "verbose" option to the "sip show
channels"
> command.
>
>
> Thanks,
>
>
> Antony.

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Re: [asterisk-users] AMI column widths

[Marcelo Terres]

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 data.

Third, you can have these truncate problems, because that's not intention
of CLI.

Using proper AMI Actions you will probably achieve your goals

https://wiki.asterisk.org/wiki/display/AST/AMI+Actions

Regards,


On 7 Jul 2017 10:32 pm, "Antony Stone" 
wrote:

Hi.

I'm trying to get a list of the channels currently in use on an Asterisk
server (1.8.32.1 if it matters) over AMI.

I send the command "sip show channels", and I get back a response along the
lines of (* used to protect the innocent...):

Peer User/ANR Call ID  Format   Hold
 Last MessageExpiry Peer
*8.22.*0.340203564  0221e874158bb62  0x4 (ulaw)   No
 Tx: ACKSIPtrunkNu
*.1*.19.70 (None)   2021549013484-1  0x0 (nothing)No
 Rx: OPTIONS
*.34.*.208 200101   712173267@192.1  0x4 (ulaw)   No
 Rx: ACK200101
*.1*.19.70 (None)   149831567021051  0x0 (nothing)No
 Rx: REGISTER   

So, firstly, the "Call ID" column is clearly truncated, because it should
show more than is indicated above,
but more importantly for me, the "Peer" column is truncated, and what
should show as "SIPtrunkNumber8"
is only shown as "SIPtrunkNu".

How can I get the full column widths of these items shown in the output?

Note that it is not a solution just to say "don't call it
'SIPtrunkNumber8'; call it 'SIPtrunk8' instead", because
this name has also been modified slightly to conceal the real name of the
trunk, which is actually longer
than "SIPtrunkNo8", but still with the most important information at the
end.

What I'm looking for is how to get the *full* details of all the channels
shown.

I have checked, and there is no "verbose" option to the "sip show channels"
command.


Thanks,


Antony.

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

[Marcelo Terres]

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:
> OK, I give up and come grovelling, "Fork" was suggested at 18:23, it's
> now 22:20 and I have been through 4 different methods, all block with
> a 2 second delay before returning to dialplan.
>
> Here are just some of the examples I have tried, as as per the
> suggestions, I am closing all possible outputs in the forked process.
>
> https://docs.python.org/3.5/library/multiprocessing.html
> https://docs.python.org/3.5/library/multiprocessing.html?highlight=multiprocessing#multiprocessing.Process.join
> https://stackoverflow.com/questions/19747371/python-exit-commands-why-so-many-and-when-should-each-be-used
> https://stackoverflow.com/questions/27624850/launch-a-completely-independent-process
> https://stackoverflow.com/questions/13612434/why-are-the-methods-sys-exit-exit-raise-systemexit-not-working
> https://stackoverflow.com/questions/43280947/os-fork-share-local-variable-with-parent
> https://stackoverflow.com/questions/24052217/may-someone-explain-the-following-os-fork-example-to-me
> http://www.python-course.eu/forking.php
> https://pymotw.com/3/subprocess/
> http://code.activestate.com/recipes/186101-really-closing-stdin-stdout-stderr/
>
> This is the most likely looking code based on the examples. I would
> really, really appreciate a couple of pointers as to where I might be
> going wrong:
>
> #! /usr/bin/env python3
> # -*- coding: utf-8 -*-
>
> import multiprocessing as mp
> import time
> import sys
> import os
>
> #from asterisk.agi import AGI
> #agi = AGI()
>
> def f(name):
> sys.stdin.close()
> sys.stdout.close()
> sys.stderr.close()
> os.close(0)   # close C's stdin stream
> os.close(1)   # close C's stdout stream
> os.close(2)   # close C's stderr stream
> time.sleep(2)
> f = open('/var/lib/asterisk/agi-bin/tns/testing/testout.txt', 'w')
> f.write(name)
> f.close()
>
>
> if __name__ == '__main__':
> print('before process')
> mp.set_start_method('fork')
> q = mp.Queue()
> p = mp.Process(target=f, args=('asterisk',))
> p.start()
> sys.exit()
>
> On 30 June 2017 at 19:59, J Montoya or A J Stiles
>  wrote:
>> On Friday 30 Jun 2017, Jonathan H wrote:
>>> What's the simplest, easiest quickest least-code way of firing off an AGI
>>> with some variable, and then returning to the dialplan?
>>
>> You have to use the "fork" command.  This starts a copy of the process with
>> all the same internal state including variables and filehandles.  The command
>> returns a non-zero value  (which is the PID of the child process; you may 
>> need
>> this, if you plan to outlive your children and have to clear their entries
>> from the process table)  to the parent process, and zero to the child 
>> process.
>> So in the parent, you exit and return to the dialplan; and in the child, you
>> close STDIN, STDOUT and STDERR  (so no process is waiting for you to produce
>> output),  then just take your time doing what you have to.  The parent is
>> already long dead by this time, so exiting goes nowhere.
>>
>>> I've seen people talking about fastAGI, stasis, python ASYNC... all seems
>>> rather complicated. I feel I must be missing something embarrassingly
>>> obvious - isn't there just the equivalent of the unix shell's "&"?!
>>
>> Yes, fork!  That is what the "&" operator is using "under the bonnet".
>>
>> --
>> JKLM
>>
>> Note:  Originating address only accepts e-mail from list!  If replying off-
>> list, change address to asterisk1list at earthshod dot co dot uk .
>>
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>>
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Re: [asterisk-users] asterisk.conf ignored?

[Marcelo Terres]

You should try to limit it in your sip trunks (is you are using SIP
trunks, of course)
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 30 June 2017 at 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...@jabber.mundoopensource.com.br
> https://www.mundoopensource.com.br
> https://twitter.com/mhterres
> https://linkedin.com/in/marceloterres
>
>
> On 30 June 2017 at 14:15, Stefan Viljoen <viljo...@verishare.co.za> wrote:
>> Hi all
>>
>>
>>
>> I’m trying to limit the maximum concurrent calls on my Asterisk to try and
>> mitigate another problem I posted about earlier.
>>
>>
>>
>> I’ve edited
>>
>>
>>
>> /etc/asterisk/asterisk.conf
>>
>>
>>
>> And uncommented this line, and put a value of 60 in there:
>>
>>
>>
>> maxcalls = 60
>>
>>
>>
>> in an effort to limit my Asterisk to 60 simultaneous calls.
>>
>>
>>
>> I did a
>>
>>
>>
>> core reload
>>
>>
>>
>> in the CLI after doing that.
>>
>>
>>
>> Any idea why my running instance totally ignores this setting? I still goes
>> right ahead and services unlimited numbers of simultaneous calls - we have
>> 90 extensions or so and it will happily service 90 simultaneous calls in
>> spite of asterisk.conf clearly stating
>>
>>
>>
>> maxcalls = 60
>>
>>
>>
>> The “maxload” specification is also ignored, load can go anywhere the
>> Asterisk instance keeps taking more calls despite load exceeding, for
>> example
>>
>>
>>
>> maxload = 10
>>
>>
>>
>> in /etc/asterisk/asterisk.conf
>>
>>
>>
>> What am I doing wrong that the asterisk binary is apparently ignoring
>> settings in /etc/asterisk/asterisk.conf?
>>
>>
>>
>> Thanks
>>
>>
>>
>> Stefan
>>
>>
>>
>>
>>
>>
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>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
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Re: [asterisk-users] asterisk.conf ignored?

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


On 30 June 2017 at 14:15, Stefan Viljoen  wrote:
> Hi all
>
>
>
> I’m trying to limit the maximum concurrent calls on my Asterisk to try and
> mitigate another problem I posted about earlier.
>
>
>
> I’ve edited
>
>
>
> /etc/asterisk/asterisk.conf
>
>
>
> And uncommented this line, and put a value of 60 in there:
>
>
>
> maxcalls = 60
>
>
>
> in an effort to limit my Asterisk to 60 simultaneous calls.
>
>
>
> I did a
>
>
>
> core reload
>
>
>
> in the CLI after doing that.
>
>
>
> Any idea why my running instance totally ignores this setting? I still goes
> right ahead and services unlimited numbers of simultaneous calls - we have
> 90 extensions or so and it will happily service 90 simultaneous calls in
> spite of asterisk.conf clearly stating
>
>
>
> maxcalls = 60
>
>
>
> The “maxload” specification is also ignored, load can go anywhere the
> Asterisk instance keeps taking more calls despite load exceeding, for
> example
>
>
>
> maxload = 10
>
>
>
> in /etc/asterisk/asterisk.conf
>
>
>
> What am I doing wrong that the asterisk binary is apparently ignoring
> settings in /etc/asterisk/asterisk.conf?
>
>
>
> Thanks
>
>
>
> Stefan
>
>
>
>
>
>
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> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
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Re: [asterisk-users] Writing CDR's to two database servers

[Marcelo Terres]

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 Support  wrote:
> I appreciate all the feedback, and replication seems to be a logical 
> solution, but I was initially thinking about how to implement a solution 
> within Asterisk to write the CDR's to two databases. Is that possible? Now 
> I'm just curious.
> Thanks Much;
> John V.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Antony Stone
> Sent: Monday, June 19, 2017 01:08 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Writing CDR's to two database servers
>
> On Monday 19 June 2017 at 18:12:35, Sebastian Gutierrez wrote:
>
>> use replication
>
> 1. Agreed - use replication.
>
> 2. If you want an HA (High Availability, not dependent on a single Master DB 
> server replicating to a slave) solution, consider setting up Master-Master 
> replication, with an LVS (Linux Virtual Server) HA machine in front of the 
> two, so that writes can go to either server using only a single IP address 
> configured in Asterisk.
>
> Then, if one fails, you can still write to (and read from) the other, repair 
> the failed one, and restore replication.
>
>
> Antony
>
>> > On Jun 19, 2017, at 17:47, Tech Support  wrote:
>> >
>> > All;
>> >
>> > I know that there are probably several solutions to this problem, but
>> > what I am trying to do is provide some redundancy for my customers
>> > CDR data. I know that doing simple backups of MySQL is probably the
>> > easiest way to go, but I’m thinking that there may be some benefit
>> > to simultaneously writing the CDR data to multiple servers at once.
>> > However, I’m drawing a blank on this one. Has anyone else done this
>> > before? Any insight at all would be greatly appreciated.
>> >
>> > Thanks Much;
>> > John V.
>
> --
> Atheism is a non-prophet-making organisation.
>
>Please reply to the list;
>  please *don't* CC me.
>
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Re: [asterisk-users] OT: Explain where mailing list bouncing comes from ?

[Marcelo Terres]

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


On 16 June 2017 at 22:37, James B. Byrne  wrote:
>
> On Fri, June 16, 2017 12:28, Tim S wrote:
>
> Whether it is intentional or not these messages railing against the
> list operators has a decided tone of condescension which is not
> warranted.  The fact of the matter is that DMARC is broken by design
> and the unpleasant effects that adoption of it has on mailing-list
> traffic were well hashed out on the ITEF mailing lists before it was
> adopted anyway.  What was predicted there has come to pass.
>
> DMARC conflicts with the existing SMTP RFCs in several ways, none of
> which I will elaborate here but all of which may be discovered by
> perusing the relevant threads on the ITEF mailing lists.  Some mailing
> list management software, notably Mailman, since has been modified to
> 'work around' the problems with DMARC if so configured by the list
> owners.  But only at the cost of violating the SMTP RFCs themselves.
> Do not take my word for it.  Raise these issues on the Postfix mailing
> list and discover what response you get from Viktor and Wietse.
>
> The driving force behind DMARC was YAHOO's shoddy security of their
> own users' accounts.  With Hotmail and similar ilk close behind. It is
> a completely inappropriate, and in my opinion ill-thought-out,
> technical solution to what is essentially an internal security problem
> at some email providers, albeit very large ones.  In general it is an
> example of what is called 'externalising your costs'.
>
> The appropriate answer has been provided: lose the
> gmail/hotmail/yahoo/freemail account and administer your own domain
> for personal email. Configure the spf and dkim settings on your own
> domain as required to suit your needs and not those of someone else.
>
> --
> ***  e-Mail is NOT a SECURE channel  ***
> Do NOT transmit sensitive data via e-Mail
>  Do NOT open attachments nor follow links sent by e-Mail
>
> James B. Byrnemailto:byrn...@harte-lyne.ca
> Harte & Lyne Limited  http://www.harte-lyne.ca
> 9 Brockley Drive  vox: +1 905 561 1241
> Hamilton, Ontario fax: +1 905 561 0757
> Canada  L8E 3C3
>
>
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Re: [asterisk-users] OT: Explain where mailing list bouncing comes from ?

[Marcelo Terres]

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 reader of this mailing list, for several years now.
>
> Lately, I'm receiving emails asking me to re-enable my list subscription due
> to "excessive bouncing".
>
> What does this exactly mean and why am I receiving this ?
> Beside re-enabling my subscription, what can I do to improve things ?
>
> Regards
>
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>
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Re: [asterisk-users] asterisk server - no sound

[Marcelo Terres]

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 <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 18:54, 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 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.mundoopensource.com.br
>> https://twitter.com/mhterres
>> https://linkedin.com/in/marceloterres
>>
>>
>> On 6 June 2017 at 16:27, andre castro <an...@andrecastro.info> wrote:
>>> Thanks Anthony.
>>>
>>> I did it on the server, according to
>>> https://www.voip-info.org/wiki/view/port+forwarding
>>>
>>> However after doing it, when running Asterisk I get the following message
>>> sudo asterisk -vvr
>>> No ethernet interface found for seeding global EID. You will have to set
>>> it manually.
>>> Unable to access the running directory (No such file or directory).
>>> Changing to '/' for compatibility.
>>>
>>> How and where can it be set?
>>>
>>> My server ifconfig:
>>>
>>> loLink encap:Local Loopback
>>>   inet addr:127.0.0.1  Mask:255.0.0.0
>>>   inet6 addr: ::1/128 Scope:Host
>>>   UP LOOPBACK RUNNING  MTU:65536  Metric:1
>>>   RX packets:113895058 errors:0 dropped:0 overruns:0 frame:0
>>>   TX packets:113895058 errors:0 dropped:0 overruns:0 carrier:0
>>>   collisions:0 txqueuelen:0
>>>   RX bytes:36041459269 (33.5 GiB)  TX bytes:36041459269 (33.5 GiB)
>>>
>>> venet0Link encap:UNSPEC  HWaddr
>>> 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00
>>>   inet addr:127.0.0.1  P-t-P:127.0.0.1  Bcast:0.0.0.0
>>> Mask:255.255.255.255
>>>   inet6 addr: ::2/128 Scope:Compat
>>>   inet6 addr: 2a01:488:66:1000:5c33:846e:0:1/128 Scope:Global
>>>   UP BROADCAST POINTOPOINT RUNNING NOARP  MTU:1500  Metric:1
>>>   RX packets:158483849 errors:0 dropped:0 overruns:0 frame:0
>>>   TX packets:272193853 errors:0 dropped:230 overruns:0 carrier:0
>>>   collisions:0 txqueuelen:0
>>>   RX bytes:61233254724 (57.0 GiB)  TX bytes:106403959440 (99.0 GiB)
>>>
>>> venet0:0  Link encap:UNSPEC  HWaddr
>>> 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00
>>>   inet addr:server.ip.add.r  P-t-P:server.ip.add.r
>>> Bcast:server.ip.add.r  Mask:255.255.255.255
>>>   UP BROADCAST POINTOPOINT RUNNING NOARP  MTU:1500  Metric:1
>>>
>>>
>>>
>>> On 06/06/2017 05:09 PM, Antony Stone wrote:
>>>> On Tuesday 06 June 2017 16:57:07 andre castro wrote:
>>>>
>>>>> On 06/06/2017 04:36 PM, Antony Stone wrote:
>>>>>>
>>>>>> Tell us about your networking arrangement - are both phones and the
>>>>>> Asterisk machine on the same network?
>>>>>
>>>>> Nop. They are in 2 different networks. The phones in one and the
>>>>> Asterisk machine in another.
>>>>
>>>> Okay, that is why you have audio between the two phones, then - they can 
>>>> see
>>>> each other directly, on the same network, and nothing is interfering with 
>>>> the
>>>> traffic between them.
>>>>
>>>>>> Is there a router in between any of them?
>>>>>
>>>>> Yes. In the phones network.
>>>>>
>>>>>> Is there any NAT involved?
>>>>>
>>>>> Yes in the phones' network. They both have different private IP address
>>>>> and one public IP.
>>>>
>>>> Okay, I suspect that this NATting router is not passing the UDP packets 
>>>> from
>>>> the server back to the phones correctly, based on the SIP connection (when 
>>>> the
>>>> phone makes the call).
>>&

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

[Marcelo Terres]

Well, based on the config that you sent, your server just have the
localhost IP (127.0.0.1)
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 18:54, 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 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.mundoopensource.com.br
>> https://twitter.com/mhterres
>> https://linkedin.com/in/marceloterres
>>
>>
>> On 6 June 2017 at 16:27, andre castro <an...@andrecastro.info> wrote:
>>> Thanks Anthony.
>>>
>>> I did it on the server, according to
>>> https://www.voip-info.org/wiki/view/port+forwarding
>>>
>>> However after doing it, when running Asterisk I get the following message
>>> sudo asterisk -vvr
>>> No ethernet interface found for seeding global EID. You will have to set
>>> it manually.
>>> Unable to access the running directory (No such file or directory).
>>> Changing to '/' for compatibility.
>>>
>>> How and where can it be set?
>>>
>>> My server ifconfig:
>>>
>>> loLink encap:Local Loopback
>>>   inet addr:127.0.0.1  Mask:255.0.0.0
>>>   inet6 addr: ::1/128 Scope:Host
>>>   UP LOOPBACK RUNNING  MTU:65536  Metric:1
>>>   RX packets:113895058 errors:0 dropped:0 overruns:0 frame:0
>>>   TX packets:113895058 errors:0 dropped:0 overruns:0 carrier:0
>>>   collisions:0 txqueuelen:0
>>>   RX bytes:36041459269 (33.5 GiB)  TX bytes:36041459269 (33.5 GiB)
>>>
>>> venet0Link encap:UNSPEC  HWaddr
>>> 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00
>>>   inet addr:127.0.0.1  P-t-P:127.0.0.1  Bcast:0.0.0.0
>>> Mask:255.255.255.255
>>>   inet6 addr: ::2/128 Scope:Compat
>>>   inet6 addr: 2a01:488:66:1000:5c33:846e:0:1/128 Scope:Global
>>>   UP BROADCAST POINTOPOINT RUNNING NOARP  MTU:1500  Metric:1
>>>   RX packets:158483849 errors:0 dropped:0 overruns:0 frame:0
>>>   TX packets:272193853 errors:0 dropped:230 overruns:0 carrier:0
>>>   collisions:0 txqueuelen:0
>>>   RX bytes:61233254724 (57.0 GiB)  TX bytes:106403959440 (99.0 GiB)
>>>
>>> venet0:0  Link encap:UNSPEC  HWaddr
>>> 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00
>>>   inet addr:server.ip.add.r  P-t-P:server.ip.add.r
>>> Bcast:server.ip.add.r  Mask:255.255.255.255
>>>   UP BROADCAST POINTOPOINT RUNNING NOARP  MTU:1500  Metric:1
>>>
>>>
>>>
>>> On 06/06/2017 05:09 PM, Antony Stone wrote:
>>>> On Tuesday 06 June 2017 16:57:07 andre castro wrote:
>>>>
>>>>> On 06/06/2017 04:36 PM, Antony Stone wrote:
>>>>>>
>>>>>> Tell us about your networking arrangement - are both phones and the
>>>>>> Asterisk machine on the same network?
>>>>>
>>>>> Nop. They are in 2 different networks. The phones in one and the
>>>>> Asterisk machine in another.
>>>>
>>>> Okay, that is why you have audio between the two phones, then - they can 
>>>> see
>>>> each other directly, on the same network, and nothing is interfering with 
>>>> the
>>>> traffic between them.
>>>>
>>>>>> Is there a router in between any of them?
>>>>>
>>>>> Yes. In the phones network.
>>>>>
>>>>>> Is there any NAT involved?
>>>>>
>>>>> Yes in the phones' network. They both have different private IP address
>>>>> and one public IP.
>>>>
>>>> Okay, I suspect that this NATting router is not passing the UDP packets 
>>>> from
>>>> the server back to the phones correctly, based on the SIP connection (when 
>>>> the
>>>> phone makes the call).
>>>>
>>>> SIP is on UDP 5060; audio is on UDP 10,000 - 20,000.
>>>>
>>>> If it's a Linux router, you need to 

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

[Marcelo Terres]

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 castro  wrote:
> Thanks Anthony.
>
> I did it on the server, according to
> https://www.voip-info.org/wiki/view/port+forwarding
>
> However after doing it, when running Asterisk I get the following message
> sudo asterisk -vvr
> No ethernet interface found for seeding global EID. You will have to set
> it manually.
> Unable to access the running directory (No such file or directory).
> Changing to '/' for compatibility.
>
> How and where can it be set?
>
> My server ifconfig:
>
> loLink encap:Local Loopback
>   inet addr:127.0.0.1  Mask:255.0.0.0
>   inet6 addr: ::1/128 Scope:Host
>   UP LOOPBACK RUNNING  MTU:65536  Metric:1
>   RX packets:113895058 errors:0 dropped:0 overruns:0 frame:0
>   TX packets:113895058 errors:0 dropped:0 overruns:0 carrier:0
>   collisions:0 txqueuelen:0
>   RX bytes:36041459269 (33.5 GiB)  TX bytes:36041459269 (33.5 GiB)
>
> venet0Link encap:UNSPEC  HWaddr
> 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00
>   inet addr:127.0.0.1  P-t-P:127.0.0.1  Bcast:0.0.0.0
> Mask:255.255.255.255
>   inet6 addr: ::2/128 Scope:Compat
>   inet6 addr: 2a01:488:66:1000:5c33:846e:0:1/128 Scope:Global
>   UP BROADCAST POINTOPOINT RUNNING NOARP  MTU:1500  Metric:1
>   RX packets:158483849 errors:0 dropped:0 overruns:0 frame:0
>   TX packets:272193853 errors:0 dropped:230 overruns:0 carrier:0
>   collisions:0 txqueuelen:0
>   RX bytes:61233254724 (57.0 GiB)  TX bytes:106403959440 (99.0 GiB)
>
> venet0:0  Link encap:UNSPEC  HWaddr
> 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00
>   inet addr:server.ip.add.r  P-t-P:server.ip.add.r
> Bcast:server.ip.add.r  Mask:255.255.255.255
>   UP BROADCAST POINTOPOINT RUNNING NOARP  MTU:1500  Metric:1
>
>
>
> On 06/06/2017 05:09 PM, Antony Stone wrote:
>> On Tuesday 06 June 2017 16:57:07 andre castro wrote:
>>
>>> On 06/06/2017 04:36 PM, Antony Stone wrote:

 Tell us about your networking arrangement - are both phones and the
 Asterisk machine on the same network?
>>>
>>> Nop. They are in 2 different networks. The phones in one and the
>>> Asterisk machine in another.
>>
>> Okay, that is why you have audio between the two phones, then - they can see
>> each other directly, on the same network, and nothing is interfering with the
>> traffic between them.
>>
 Is there a router in between any of them?
>>>
>>> Yes. In the phones network.
>>>
 Is there any NAT involved?
>>>
>>> Yes in the phones' network. They both have different private IP address
>>> and one public IP.
>>
>> Okay, I suspect that this NATting router is not passing the UDP packets from
>> the server back to the phones correctly, based on the SIP connection (when 
>> the
>> phone makes the call).
>>
>> SIP is on UDP 5060; audio is on UDP 10,000 - 20,000.
>>
>> If it's a Linux router, you need to make sure you are allowing FORWARDed 
>> traffic
>> which matches ESTABLISHED, RELATED.
>>
>> If it's not a Linux router, you need to find out how to get it to support SIP
>> and RTSP.
>>
>>
>> Good luck,
>>
>>
>> Antony.
>>
>
> --
> oo.io
> bibliotecha.info
>
> --
> _
> -- 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:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
_
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Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] asterisk server - no sound

[Marcelo Terres]

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 Terres <mhter...@gmail.com> wrote:
> Which Asterisk version are you using?
>
> 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 18:32, andre castro <an...@andrecastro.info> wrote:
>> Any ideas.
>> After configuring  port forwarding on the server (machine making nat) to
>> forward connections originated from external clients to the machine
>> running asterisk, as explained in
>> https://www.voip-info.org/wiki/view/port+forwarding
>> My peers were unable to register.
>>
>>
>> And When running Asterisk I am getting:
>> No ethernet interface found for seeding global EID. You will have to set
>> it manually.
>> Unable to access the running directory (No such file or directory).
>> Changing to '/' for compatibility.
>>
>> Any advice what to do next?
>>
>> thanks
>> a
>>
>> On 06/06/2017 05:27 PM, andre castro wrote:
>>> Thanks Anthony.
>>>
>>> I did it on the server, according to
>>> https://www.voip-info.org/wiki/view/port+forwarding
>>>
>>> However after doing it, when running Asterisk I get the following message
>>> sudo asterisk -vvr
>>> No ethernet interface found for seeding global EID. You will have to set
>>> it manually.
>>> Unable to access the running directory (No such file or directory).
>>> Changing to '/' for compatibility.
>>>
>>> How and where can it be set?
>>>
>>> My server ifconfig:
>>>
>>> loLink encap:Local Loopback
>>>   inet addr:127.0.0.1  Mask:255.0.0.0
>>>   inet6 addr: ::1/128 Scope:Host
>>>   UP LOOPBACK RUNNING  MTU:65536  Metric:1
>>>   RX packets:113895058 errors:0 dropped:0 overruns:0 frame:0
>>>   TX packets:113895058 errors:0 dropped:0 overruns:0 carrier:0
>>>   collisions:0 txqueuelen:0
>>>   RX bytes:36041459269 (33.5 GiB)  TX bytes:36041459269 (33.5 GiB)
>>>
>>> venet0Link encap:UNSPEC  HWaddr
>>> 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00
>>>   inet addr:127.0.0.1  P-t-P:127.0.0.1  Bcast:0.0.0.0
>>> Mask:255.255.255.255
>>>   inet6 addr: ::2/128 Scope:Compat
>>>   inet6 addr: 2a01:488:66:1000:5c33:846e:0:1/128 Scope:Global
>>>   UP BROADCAST POINTOPOINT RUNNING NOARP  MTU:1500  Metric:1
>>>   RX packets:158483849 errors:0 dropped:0 overruns:0 frame:0
>>>   TX packets:272193853 errors:0 dropped:230 overruns:0 carrier:0
>>>   collisions:0 txqueuelen:0
>>>   RX bytes:61233254724 (57.0 GiB)  TX bytes:106403959440 (99.0 GiB)
>>>
>>> venet0:0  Link encap:UNSPEC  HWaddr
>>> 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00
>>>   inet addr:server.ip.add.r  P-t-P:server.ip.add.r
>>> Bcast:server.ip.add.r  Mask:255.255.255.255
>>>   UP BROADCAST POINTOPOINT RUNNING NOARP  MTU:1500  Metric:1
>>>
>>>
>>>
>>> On 06/06/2017 05:09 PM, Antony Stone wrote:
>>>> On Tuesday 06 June 2017 16:57:07 andre castro wrote:
>>>>
>>>>> On 06/06/2017 04:36 PM, Antony Stone wrote:
>>>>>>
>>>>>> Tell us about your networking arrangement - are both phones and the
>>>>>> Asterisk machine on the same network?
>>>>>
>>>>> Nop. They are in 2 different networks. The phones in one and the
>>>>> Asterisk machine in another.
>>>>
>>>> Okay, that is why you have audio between the two phones, then - they can 
>>>> see
>>>> each other directly, on the same network, and nothing is interfering with 
>>>> the
>>>> traffic between them.
>>>>
>>>>>> Is there a router in between any of them?
>>>>>
>>>>> Yes. In the phones network.
>>>>>
>>>>>> Is there any NAT involved?
>>>>>
>>>>> Yes in the phones' network. They both have different private IP address
>>>>> and one public IP.
>>>>
&g

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

[Marcelo Terres]

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:
> Any ideas.
> After configuring  port forwarding on the server (machine making nat) to
> forward connections originated from external clients to the machine
> running asterisk, as explained in
> https://www.voip-info.org/wiki/view/port+forwarding
> My peers were unable to register.
>
>
> And When running Asterisk I am getting:
> No ethernet interface found for seeding global EID. You will have to set
> it manually.
> Unable to access the running directory (No such file or directory).
> Changing to '/' for compatibility.
>
> Any advice what to do next?
>
> thanks
> a
>
> On 06/06/2017 05:27 PM, andre castro wrote:
>> Thanks Anthony.
>>
>> I did it on the server, according to
>> https://www.voip-info.org/wiki/view/port+forwarding
>>
>> However after doing it, when running Asterisk I get the following message
>> sudo asterisk -vvr
>> No ethernet interface found for seeding global EID. You will have to set
>> it manually.
>> Unable to access the running directory (No such file or directory).
>> Changing to '/' for compatibility.
>>
>> How and where can it be set?
>>
>> My server ifconfig:
>>
>> loLink encap:Local Loopback
>>   inet addr:127.0.0.1  Mask:255.0.0.0
>>   inet6 addr: ::1/128 Scope:Host
>>   UP LOOPBACK RUNNING  MTU:65536  Metric:1
>>   RX packets:113895058 errors:0 dropped:0 overruns:0 frame:0
>>   TX packets:113895058 errors:0 dropped:0 overruns:0 carrier:0
>>   collisions:0 txqueuelen:0
>>   RX bytes:36041459269 (33.5 GiB)  TX bytes:36041459269 (33.5 GiB)
>>
>> venet0Link encap:UNSPEC  HWaddr
>> 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00
>>   inet addr:127.0.0.1  P-t-P:127.0.0.1  Bcast:0.0.0.0
>> Mask:255.255.255.255
>>   inet6 addr: ::2/128 Scope:Compat
>>   inet6 addr: 2a01:488:66:1000:5c33:846e:0:1/128 Scope:Global
>>   UP BROADCAST POINTOPOINT RUNNING NOARP  MTU:1500  Metric:1
>>   RX packets:158483849 errors:0 dropped:0 overruns:0 frame:0
>>   TX packets:272193853 errors:0 dropped:230 overruns:0 carrier:0
>>   collisions:0 txqueuelen:0
>>   RX bytes:61233254724 (57.0 GiB)  TX bytes:106403959440 (99.0 GiB)
>>
>> venet0:0  Link encap:UNSPEC  HWaddr
>> 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00
>>   inet addr:server.ip.add.r  P-t-P:server.ip.add.r
>> Bcast:server.ip.add.r  Mask:255.255.255.255
>>   UP BROADCAST POINTOPOINT RUNNING NOARP  MTU:1500  Metric:1
>>
>>
>>
>> On 06/06/2017 05:09 PM, Antony Stone wrote:
>>> On Tuesday 06 June 2017 16:57:07 andre castro wrote:
>>>
 On 06/06/2017 04:36 PM, Antony Stone wrote:
>
> Tell us about your networking arrangement - are both phones and the
> Asterisk machine on the same network?

 Nop. They are in 2 different networks. The phones in one and the
 Asterisk machine in another.
>>>
>>> Okay, that is why you have audio between the two phones, then - they can see
>>> each other directly, on the same network, and nothing is interfering with 
>>> the
>>> traffic between them.
>>>
> Is there a router in between any of them?

 Yes. In the phones network.

> Is there any NAT involved?

 Yes in the phones' network. They both have different private IP address
 and one public IP.
>>>
>>> Okay, I suspect that this NATting router is not passing the UDP packets from
>>> the server back to the phones correctly, based on the SIP connection (when 
>>> the
>>> phone makes the call).
>>>
>>> SIP is on UDP 5060; audio is on UDP 10,000 - 20,000.
>>>
>>> If it's a Linux router, you need to make sure you are allowing FORWARDed 
>>> traffic
>>> which matches ESTABLISHED, RELATED.
>>>
>>> If it's not a Linux router, you need to find out how to get it to support 
>>> SIP
>>> and RTSP.
>>>
>>>
>>> Good luck,
>>>
>>>
>>> Antony.
>>>
>>
>
> --
> oo.io
> bibliotecha.info
>
> --
> _
> -- 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:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
_
-- 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:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users 

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

[Marcelo Terres]

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 2017 at 14:18, andre castro  wrote:
> hello folks,
> this might be a simple question...
>
> I just installed asterisk in a debian server.
> All seems to be running fine, but the audio sent by the server.
> If I have one of my registered peers call and extension (102) that plays
> back audio (extension.conf and sip.conf coffee-pasted below), Asterisk
> answers and prints no errors.
> Its `sip show channels` prints:
>
> PeerUser/ANRCall IDFormatHoldLast MessageExpiry
>Peer
> peer.ip1001 1...-5060   (ulaw)  No Rx: ACK
>1001
>
> But I hear nothing at the peer's end.
>
> When one peer calls another, sound comes through just fine.
> So my hunch is that is something to do with the audio supplied by the
> server.
> Do I need to have alsa installed??
> Any hint?
>
> sip.conf:
>
> [general]
> context = unauthenticated
> bindport = 5060
> bindaddr = 0.0.0.0
> tcpbindaddr = 0.0.0.0
> tcpenable = yes
> videosupport = no
> textsupport=yes
> alwaysauthreject=yes
> allowguest=no
>
> [1001] ; grandstream 1
> context = home
> type = friend
> callerid = One <1001>
> secret = XYZ
> host = dynamic
> mailbox = 1001
> disallow = all
> allow = ulaw
> transport = udp
> dtmfmode=auto   ; accept touch-tones from the devices, negotiated
> automatically
> nat=force_rport
>
> [1005] ; mobile
> context = home
> type = friend
> callerid = Five <1005>
> secret = XYZ
> host = dynamic
> mailbox = 1005
> disallow = all
> allow = ulaw
> transport = udp
> dtmfmode=auto   ; accept touch-tones from the devices, negotiated
> automatically
> nat=force_rport
>
>
> extensions.conf:
> [home]
> exten = 102,1,Answer()
> same =  n,Wait(1)
> same =  n,Playback(beep)
> same =  n,Wait(1)
> same =  n,Playback(im-sorry)
> same =  n,Wait(1)
> same =  n,Playback(number-not-answering)
> same =  n,Wait(1)
> same =  n,Hangup()
>
> exten => 1001,1,Dial(SIP/1001) ; grandstream phone
> exten => 1005,1,Dial(SIP/1005) ; mobile
>
>
>
>
> --
> oo.io
> bibliotecha.info
>
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Re: [asterisk-users] IAX port 4569

[Marcelo Terres]

There are two different networks connecting in the tcpdump, 10 and 192.

In this example that you send, are you trying to connect from different ips
in different server ips?

Regards,




On 5 Jun 2017 9:23 pm, <the...@sys-concept.com> wrote:

Yes, it is working!

tcpdump -ni any port 4569
dropped privs to tcpdump
tcpdump: verbose output suppressed, use -v or -vv for full protocol decode
listening on any, link-type LINUX_SLL (Linux cooked), capture size 262144
bytes
14:20:42.184521 IP 10.0.0.108.4569 > 10.0.0.100.4569: UDP, length 53
14:20:42.184921 IP 10.0.0.100.4569 > 10.0.0.108.4569: UDP, length 37
14:20:42.190529 IP 10.0.0.108.4569 > 10.0.0.100.4569: UDP, length 83
14:20:42.190639 IP 10.0.0.100.4569 > 10.0.0.108.4569: UDP, length 71
14:20:42.191378 IP 10.0.0.108.4569 > 10.0.0.100.4569: UDP, length 12
14:20:45.320191 IP 192.168.141.7.4569 > 192.168.141.1.4569: UDP, length 31
14:20:45.338718 IP 192.168.141.1.4569 > 192.168.141.7.4569: UDP, length 65
14:20:45.338875 IP 192.168.141.7.4569 > 192.168.141.1.4569: UDP, length 82
14:20:45.357173 IP 192.168.141.1.4569 > 192.168.141.7.4569: UDP, length 40
14:20:45.357331 IP 192.168.141.7.4569 > 192.168.141.1.4569: 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, Marcelo Terres wrote:
> 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, <the...@sys-concept.com> 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 with my previous version of
>> asterisk.
>>
>> After upgrade to 11.25.1 it stop working.
>> I'm sure port forwarding on dd-wrt is working OK as I have port 80 and
>> 443 open.
>>
>>
>> Thelma
>> On 06/05/2017 07:12 AM, Christopher van de Sande wrote:
>>> Another might be to make sure iptables isn't blocking the connection.
>>>
>>> You can run
>>> iptables -L -n -v
>>> To see if its set to block any ports.
>>>
>>>
>>> On June 5, 2017 9:06:55 AM EDT, the...@sys-concept.com wrote:
>>>> I'm getting:
>>>> netstat -a |grep 4569
>>>> udp0  0 0.0.0.0:45690.0.0.0:*
>>>>
>>>> Should I be getting localhost IP?
>>>>
>>>> Thelma
>>>>
>>>> On 06/05/2017 06:48 AM, the...@sys-concept.com wrote:
>>>>> Does asterisk listen on port 4569 by default?
>>>>>
>>>>> I'm running version Asterisk 11.25.1 and have a problem registering
>>>>> Zoiper (IAX) to Asterisk.
>>>>> I'm getting an error:
>>>>> Registration refused
>>>>>
>>>>
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>>>>
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>>>
>>>
>>
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>
>
>

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

[Marcelo Terres]

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 closed) :-/


Thelma
On 06/05/2017 02:19 PM, Victor Villarreal wrote:
> I think you need to increase verbose output and search in
> /var/log/asterisk/full for any error message related to IAX2 registration
> or simil.
>
> 2017-06-05 17:12 GMT-03:00 :
>
>> 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 with my previous version of
>> asterisk.
>>
>> After upgrade to 11.25.1 it stop working.
>> I'm sure port forwarding on dd-wrt is working OK as I have port 80 and
>> 443 open.
>>
>>
>> Thelma
>> On 06/05/2017 07:12 AM, Christopher van de Sande wrote:
>>> Another might be to make sure iptables isn't blocking the connection.
>>>
>>> You can run
>>> iptables -L -n -v
>>> To see if its set to block any ports.
>>>
>>>
>>> On June 5, 2017 9:06:55 AM EDT, the...@sys-concept.com wrote:
 I'm getting:
 netstat -a |grep 4569
 udp0  0 0.0.0.0:45690.0.0.0:*

 Should I be getting localhost IP?

 Thelma

 On 06/05/2017 06:48 AM, the...@sys-concept.com wrote:
> Does asterisk listen on port 4569 by default?
>
> I'm running version Asterisk 11.25.1 and have a problem registering
> Zoiper (IAX) to Asterisk.
> I'm getting an error:
> Registration refused
>

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>>>
>>>
>>>
>>
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>> org/
>>
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>
>
>
>
>

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Re: [asterisk-users] IAX port 4569

[Marcelo Terres]

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 with my previous version of
> asterisk.
>
> After upgrade to 11.25.1 it stop working.
> I'm sure port forwarding on dd-wrt is working OK as I have port 80 and
> 443 open.
>
>
> Thelma
> On 06/05/2017 07:12 AM, Christopher van de Sande wrote:
> > Another might be to make sure iptables isn't blocking the connection.
> >
> > You can run
> > iptables -L -n -v
> > To see if its set to block any ports.
> >
> >
> > On June 5, 2017 9:06:55 AM EDT, the...@sys-concept.com wrote:
> >> I'm getting:
> >> netstat -a |grep 4569
> >> udp0  0 0.0.0.0:45690.0.0.0:*
> >>
> >> Should I be getting localhost IP?
> >>
> >> Thelma
> >>
> >> On 06/05/2017 06:48 AM, the...@sys-concept.com wrote:
> >>> Does asterisk listen on port 4569 by default?
> >>>
> >>> I'm running version Asterisk 11.25.1 and have a problem registering
> >>> Zoiper (IAX) to Asterisk.
> >>> I'm getting an error:
> >>> Registration refused
> >>>
> >>
> >> --
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> >> https://community.asterisk.org/
> >>
> >> New to Asterisk? Start here:
> >>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> >>
> >> asterisk-users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >>   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
>
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> org/
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Re: [asterisk-users] IAX port 4569

[Marcelo Terres]

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 asterisk listen on port 4569 by default?
>
> I'm running version Asterisk 11.25.1 and have a problem registering
> Zoiper (IAX) to Asterisk.
> I'm getting an error:
> Registration refused
>
> --
> Thelma
>
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>
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>
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Re: [asterisk-users] OT: Want to capture all SIP messages

[Marcelo Terres]

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
https://linkedin.com/in/marceloterres


On 2 June 2017 at 17:00, ewieling  wrote:
>
> I use pcapsipdump.  It has the added advantage of splitting the captures
> into individual calls
>
>
>
> On 06/01/2017 06:09 AM, Tony Mountifield wrote:
>>
>> In article ,
>> Steve Edwards  wrote:
>>>
>>> On Wed, 31 May 2017, Steve Edwards wrote:
>>>
 I want to capture all SIP messages.

 I have about 30 hosts in about 6 colos.

 My first thought was dumpcap, but the output file name format bugs me.

 What do you use for long term SIP capture?
>>>
>>> A little more specificity...
>>>
>>> I'd like the capture to be in a series of files that can be 'rotated' or
>>> 'aged out' so that I can always have x days of traffic on hand but not
>>> have to prune the files to keep the storage requirements reasonable.
>>
>> On most of my systems I have a script sip-capture:
>>
>> ---
>> #!/bin/sh
>>
>> DATE=`date '+%Y%m%d-%H%M%S'`
>> FILE=sip-`hostname -s`-$DATE.pkt
>>
>> cd /var/tmp
>>
>> tcpdump -C 8 -i any -n -p -s 0 -w $FILE udp port 5060 > >/dev/null 2>&1 &
>> ---
>>
>> I start it in /etc/rc.d/rc.local for want of anywhere better.
>>
>> Being in /var/tmp, cron.daily/tmpwatch deletes files older than 30 days.
>> I could just have easily put them somewhere else and used the -W option
>> to tcpdump to remove old files on a rolling basis.
>>
>> Cheers
>> Tony
>
>
>
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Re: [asterisk-users] Best way to know a call is being transfered

[Marcelo Terres]

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.

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Variables

Regards,
Marcelo H. Terres 
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https://www.mundoopensource.com.br
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On 29 May 2017 at 10:06, Jonas Kellens  wrote:
> Hello
>
> thank you for your answer.
>
> However this does not help me to know when a call is being transfered.
>
> My question is simple : if A calls B, and then B tranfers (unattened or
> attended) the call to C, how can I know this happens ?? I see it happening
> on the CLI, but how can I "catch" this, for example in the dialplan logic ?
> Or through AMI perhaps ?
>
>
>
> Kind regards.
>
> J.
>
>
>
> Op 29-05-17 om 10:16 schreef Jonathan H:
>
>> Well, once you've upgraded to a version of Asterisk which didn't
>> become "EOL - DO NOT USE - NO FIXES" (!) almost 2 years ago, then you
>> might be able use logging which was introduced 5 years ago in Asterisk
>> 11. Although the "transfers" section in the info below says it "can be
>> a little tricky...". Read on!
>>
>> https://wiki.asterisk.org/wiki/display/AST/Call+Identifier+Logging
>>
>> 
>>
>> Call ID Logging (which has nothing to do with caller ID) is a new
>> feature of Asterisk 11 intended to help administrators and support
>> givers to more quickly understand problems that occur during the
>> course of calls. Channels are now bound to call identifiers which can
>> be shared among a number of channels, threads, and other consumers.
>>
>> Transfers
>>
>> Transfers can be a little tricky to follow with the call ID logging
>> feature. As a general rule, an attended transfer will always result in
>> a new call ID being made because a separate call must occur between
>> the party that initiates the transfer and whatever extension is going
>> to receive it. Once the attended transfer is completed, the channel
>> that was transferred will use the Call ID created when the transferrer
>> called the recipient.
>>
>> Blind transfers are slightly more variable. If a SIP peer 'peer1'
>> calls another SIP peer 'peer2' via the dial application and peer2
>> blind transfers peer1 elsewhere, the call ID will persist. If on the
>> other hand, peer1 blind transfers peer2 at this point a new call ID
>> will be created. When peer1 transfers peer2, peer2 has a new channel
>> created which enters the PBX for the first time, so it creates a new
>> call ID. When peer1 is transferred, it simply resumes running PBX, so
>> the call is still considered the same call. By setting the debug level
>> to 3 for the channel internal API (channel_internal_api.c), all call
>> ID settings for every channel will be logged and this may be able to
>> help when trying to keep track of calls through multiple transfers.
>>
>>
>> On 29 May 2017 at 08:17, Jonas Kellens  wrote:
>>>
>>> Hello
>>>
>>> using Asterisk 1.8.32.3.
>>>
>>> What is the best way of knowing a call is being transfered (attended and
>>> unattended) ? And also knowing whereto (sip user) the call is being
>>> transfered and who is the transferer ?
>>>
>>> So I can log this information.
>>>
>>>
>>>
>>> Kind regards.
>>>
>>> J.
>>>
>>>
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>>>
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>>>
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>
>
>
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Re: [asterisk-users] Dial an extension to modify dialplan

[Marcelo Terres]

https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerAction_DialplanExtensionRemove
Marcelo H. Terres <mhter...@gmail.com>
IM: mhter...@jabber.mundoopensource.com.br
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https://linkedin.com/in/marceloterres


On 8 May 2017 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?
>
> Is there a similar call to delete an extension, or to modify an existing one?
>
> On the basis that the OP already has extension 2000 defined, he would need to
> delete this and replace it with a new definition, or alter the current
> definition, to get the required results.
>
> Simply being able to add a new extension to an existing dialplan isn't quite
> enough.
>
>
> Antony.
>
>> On 8 May 2017 at 15:35, Frank Vanoni <mailingl...@linuxista.com> wrote:
>> > Hello
>> >
>> > I have the following scenario:
>> >
>> > [mynicecontext]
>> > exten => 2000,1,Dial(SIP/deviceA/deviceB/deviceC)
>> >
>> > As expected, by dialing 2000, all three devices will ring. And that's
>> > fine.
>> > However, there are situations where I only want "deviceA" and "deviceB"
>> > to ring. I would like to have an extension to dial in order to modify
>> > the dialplan.
>> >
>> > Here is what I did...
>> >
>> > In extensions.conf:
>> >
>> > -- snip -
>> > [mynicecontext]
>> > #include "ringdevice.conf
>> >
>> > exten => 2000,1,GoTo(ringdevice,ring,1)
>> >
>> > exten => 4000,1,System(/bin/cat /etc/asterisk/twodevices.txt
>> >
>> >> /etc/asterisk/ringdevice.conf)
>> >
>> > exten => 4000,2,Wait(3)
>> > exten => 4000,3,System(/usr/sbin/asterisk -rx "dialplan reload")
>> > exten => 4000,4,Playback(service)
>> >
>> > exten => 4001,1,System(/bin/cat /etc/asterisk/alldevices.txt
>> >
>> >> /etc/asterisk/ringdevice.conf)
>> >
>> > exten => 4001,2,Wait(3)
>> > exten => 4001,3,System(/usr/sbin/asterisk -rx "dialplan reload")
>> > exten => 4001,4,Playback(service)
>> > -- end snip -
>> >
>> > twodevices.txt contains
>> > exten => ring,1,Dial(SIP/deviceA)
>> >
>> > alldevices.txt contains
>> > exten => ring,1,Dial(SIP/deviceA/deviceC)
>> >
>> > By dialing 4000 or 4001, the dialplan is modified and reloaded
>> > accordingly.
>> >
>> > Is there a better solution?
>> >
>> > Frank
>
> --
> 3 logicians walk into a bar. The bartender asks "Do you all want a drink?"
> The first logician says "I don't know."
> The second logician says "I don't know."
> The third logician says "Yes!"
>
>Please reply to the list;
>  please *don't* CC me.
>
> --
> _
> -- 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:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

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

[Marcelo Terres]

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


On 8 May 2017 at 15:35, Frank Vanoni  wrote:
> Hello
>
> I have the following scenario:
>
> [mynicecontext]
> exten => 2000,1,Dial(SIP/deviceA/deviceB/deviceC)
>
> As expected, by dialing 2000, all three devices will ring. And that's
> fine.
> However, there are situations where I only want "deviceA" and "deviceB"
> to ring. I would like to have an extension to dial in order to modify
> the dialplan.
>
> Here is what I did...
>
> In extensions.conf:
>
> -- snip -
> [mynicecontext]
> #include "ringdevice.conf
>
> exten => 2000,1,GoTo(ringdevice,ring,1)
>
> exten => 4000,1,System(/bin/cat /etc/asterisk/twodevices.txt
>> /etc/asterisk/ringdevice.conf)
> exten => 4000,2,Wait(3)
> exten => 4000,3,System(/usr/sbin/asterisk -rx "dialplan reload")
> exten => 4000,4,Playback(service)
>
> exten => 4001,1,System(/bin/cat /etc/asterisk/alldevices.txt
>> /etc/asterisk/ringdevice.conf)
> exten => 4001,2,Wait(3)
> exten => 4001,3,System(/usr/sbin/asterisk -rx "dialplan reload")
> exten => 4001,4,Playback(service)
> -- end snip -
>
> twodevices.txt contains
> exten => ring,1,Dial(SIP/deviceA)
>
> alldevices.txt contains
> exten => ring,1,Dial(SIP/deviceA/deviceC)
>
> By dialing 4000 or 4001, the dialplan is modified and reloaded
> accordingly.
>
> Is there a better solution?
>
> Frank
>
>
> --
> _
> -- 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:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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>http://lists.digium.com/mailman/listinfo/asterisk-users

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

[Marcelo Terres]

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


On 4 May 2017 at 16:27, Daniel Journo  wrote:
> Hi,
>
>
>
> Is it possible to set up a feature code to move both a caller and callee to
> a meeting room?
>
> If yes, what should I be looking at?
>
>
>
> Bonus question, is it possible to then automatically dial a 3rd person and
> invite them to the meeting room?
>
>
>
> The client wants to do this with the push of a couple of buttons only.
>
>
>
> Many thanks
>
> Dan
>
>
>
>
> --
> _
> -- 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:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

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_
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  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[Marcelo Terres]

Ah, ok.

Everytime you install a package you need to run configure again to
allow detection of new lib.

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 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:
>>
>> Good question.
>>
>> I am running Asterisk 14 on Ubuntu 16.04 and I had this packages
>> installed:
>>
>> ii  libodbc1:amd64   2.3.1-4.1
>>   amd64ODBC library for Unix
>> ii  odbc-postgresql:amd641:09.06.0200-1.pgdg14.04+1
>>   amd64ODBC driver for PostgreSQL
>> ii  odbcinst 2.3.1-4.1
>>   amd64Helper program for accessing odbc ini files
>> ii  odbcinst1debian2:amd64   2.3.1-4.1
>>   amd64Support library for accessing odbc ini files
>> ii  unixodbc 2.3.1-4.1
>>   amd64Basic ODBC tools
>> ii  unixodbc-dev 2.3.1-4.1
>>   amd64ODBC libraries for UNIX (development files)
>>
>> ii  libltdl-dev:amd642.4.6-0.1
>>   amd64System independent dlopen wrapper for GNU
>> libtool
>> 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,
>> 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 19 April 2017 at 17:50, Pierre Couderc <pie...@couderc.eu> wrote:
>>>
>>> Than you very much.
>>> I use asterisk 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 Terres wrote:
>>>>
>>>> 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 <mhter...@gmail.com>
>>>> IM: mhter...@jabber.mundoopensource.com.br
>>>> https://www.mundoopensource.com.br
>>>> https://twitter.com/mhterres
>>>> https://linkedin.com/in/marceloterres
>>>>
>>>>
>>>> On 17 April 2017 at 23:36, nous <pie...@couderc.eu> wrote:
>>>>>
>>>>> Thank you, but unixodbc and odbcinst are installed... end even
>>>>> unixodbc-dev
>>>>>
>>>>> But I get the same need for "generic odbc(E)".
>>>>>
>>>>>
>>>>>
>>>>> 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
>>>>>> https://www.mundoopensource.com.br
>>>>>> https://twitter.com/mhterres
>>>>>> https://linkedin.com/in/marceloterres
>>>>>>
>>>>>>
>>>>>> On 13 April 2017 at 19:41, Pierre Couderc <pie...@couderc.eu> wrote:
>>>>>>>
>>>>>>> I use debian stretch and I have installed unixodbc-dev
>>>>>>>
>>>>>>> but I have a dependency on genreric_odbc in make menuselect
>>>>>>>
>>>>>>> What am I missing ? Is there an howto ?
>>>>>>>
&g

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

[Marcelo Terres]

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 <mhter...@gmail.com>
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.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:
>>
>> I suppose that you enable the video support on sip.conf, right?
>>
>> 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 19 April 2017 at 13:18, Jonas Kellens <jonas.kell...@telenet.be> wrote:
>>>
>>> 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 about your configuration.
>>>  INTBINARY  HEX   TYPE   NAME
>>> DESCRIPTION
>>>
>>> ---
>>>1 (1 <<  0)(0x1)  audio   g723
>>> (G.723.1)
>>>2 (1 <<  1)(0x2)  audiogsm
>>> (GSM)
>>>4 (1 <<  2)(0x4)  audio   ulaw
>>> (G.711 u-law)
>>>8 (1 <<  3)(0x8)  audio   alaw
>>> (G.711 A-law)
>>>   16 (1 <<  4)   (0x10)  audio   g726aal2
>>> (G.726 AAL2)
>>>   32 (1 <<  5)   (0x20)  audio  adpcm
>>> (ADPCM)
>>>   64 (1 <<  6)   (0x40)  audio   slin
>>> (16
>>> bit Signed Linear PCM)
>>>  128 (1 <<  7)   (0x80)  audio  lpc10
>>> (LPC10)
>>>  256 (1 <<  8)  (0x100)  audio   g729
>>> (G.729A)
>>>  512 (1 <<  9)  (0x200)  audio  speex
>>> (SpeeX)
>>> 1024 (1 << 10)  (0x400)  audio   ilbc
>>> (iLBC)
>>> 2048 (1 << 11)  (0x800)  audio   g726
>>> (G.726 RFC3551)
>>> 4096 (1 << 12) (0x1000)  audio   g722
>>> (G722)
>>> 8192 (1 << 13) (0x2000)  audio siren7
>>> (ITU
>>> G.722.1 (Siren7, licensed from Polycom))
>>>16384 (1 << 14) (0x4000)  audiosiren14
>>> (ITU
>>> G.722.1 Annex C, (Siren14, licensed from Polycom))
>>>32768 (1 << 15) (0x8000)  audio slin16
>>> (16
>>> bit Signed Linear PCM (16kHz))
>>>65536 (1 << 16)(0x1)  image   jpeg
>>> (JPEG
>>> image)
>>>   131072 (1 << 17)(0x2)  imagepng
>>> (PNG
>>> image)
>>>   262144 (1 << 18)(0x4)  video   h261
>>> (H.261 Video)
>>>   524288 (1 << 19)(0x8)  video   h263
>>> (H.263 Video)
>>>  1048576 (1 << 20)   (0x10)  video  h263p
>>> (H.263+ Video)
>>>  2097152 (1 << 21)   (0x20)  video   h264
>>> (H.264 Video)
>>>  4194304 (1 << 22)   (0x40)  video  mpeg4
>>> (MPEG4 Video)
>>>  8388608 (1 << 23)   (0x80)  videounknown
>>> (unknown)
>>> 16777216 (1 << 24)  (0x100)  videounknown
>>> (unknown)
>>> 33554432 (1 << 25)  (0x200)   textunknown
>>> (unknown)
>>> 67108864 (1 << 26)  (0x400)   textred
>>> (T.

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

[Marcelo Terres]

Good question.

I am running Asterisk 14 on Ubuntu 16.04 and I had this packages installed:

ii  libodbc1:amd64   2.3.1-4.1
 amd64ODBC library for Unix
ii  odbc-postgresql:amd641:09.06.0200-1.pgdg14.04+1
 amd64ODBC driver for PostgreSQL
ii  odbcinst 2.3.1-4.1
 amd64Helper program for accessing odbc ini files
ii  odbcinst1debian2:amd64   2.3.1-4.1
 amd64Support library for accessing odbc ini files
ii  unixodbc 2.3.1-4.1
 amd64Basic ODBC tools
ii  unixodbc-dev 2.3.1-4.1
 amd64ODBC libraries for UNIX (development files)

ii  libltdl-dev:amd642.4.6-0.1
 amd64System independent dlopen wrapper for GNU
libtool
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,
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 19 April 2017 at 17:50, Pierre Couderc <pie...@couderc.eu> wrote:
> Than you very much.
> I use asterisk 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 Terres wrote:
>>
>> 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 <mhter...@gmail.com>
>> IM: mhter...@jabber.mundoopensource.com.br
>> https://www.mundoopensource.com.br
>> https://twitter.com/mhterres
>> https://linkedin.com/in/marceloterres
>>
>>
>> On 17 April 2017 at 23:36, nous <pie...@couderc.eu> wrote:
>>>
>>> Thank you, but unixodbc and odbcinst are installed... end even
>>> unixodbc-dev
>>>
>>> But I get the same need for "generic odbc(E)".
>>>
>>>
>>>
>>> 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
>>>> https://www.mundoopensource.com.br
>>>> https://twitter.com/mhterres
>>>> https://linkedin.com/in/marceloterres
>>>>
>>>>
>>>> On 13 April 2017 at 19:41, Pierre Couderc <pie...@couderc.eu> wrote:
>>>>>
>>>>> I use debian stretch and I have installed unixodbc-dev
>>>>>
>>>>> but I have a dependency on genreric_odbc in make menuselect
>>>>>
>>>>> What am I missing ? Is there an howto ?
>>>>>
>>>>> Thanks
>>>>> PX
>>>>>
>>>>> --
>>>>> _
>>>>> -- 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:
>>>>>https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>>>
>>>>> asterisk-users mailing list
>>>>> To UNSUBSCRIBE or update options visit:
>>>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>>
>>>
>>> --
>>> _
>>> -- 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:
>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
&

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

[Marcelo Terres]

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 Kellens  wrote:
> 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 about your configuration.
> INTBINARY  HEX   TYPE   NAME
> DESCRIPTION
> ---
>   1 (1 <<  0)(0x1)  audio   g723
> (G.723.1)
>   2 (1 <<  1)(0x2)  audiogsm   (GSM)
>   4 (1 <<  2)(0x4)  audio   ulaw
> (G.711 u-law)
>   8 (1 <<  3)(0x8)  audio   alaw
> (G.711 A-law)
>  16 (1 <<  4)   (0x10)  audio   g726aal2
> (G.726 AAL2)
>  32 (1 <<  5)   (0x20)  audio  adpcm
> (ADPCM)
>  64 (1 <<  6)   (0x40)  audio   slin   (16
> bit Signed Linear PCM)
> 128 (1 <<  7)   (0x80)  audio  lpc10
> (LPC10)
> 256 (1 <<  8)  (0x100)  audio   g729
> (G.729A)
> 512 (1 <<  9)  (0x200)  audio  speex
> (SpeeX)
>1024 (1 << 10)  (0x400)  audio   ilbc
> (iLBC)
>2048 (1 << 11)  (0x800)  audio   g726
> (G.726 RFC3551)
>4096 (1 << 12) (0x1000)  audio   g722
> (G722)
>8192 (1 << 13) (0x2000)  audio siren7   (ITU
> G.722.1 (Siren7, licensed from Polycom))
>   16384 (1 << 14) (0x4000)  audiosiren14   (ITU
> G.722.1 Annex C, (Siren14, licensed from Polycom))
>   32768 (1 << 15) (0x8000)  audio slin16   (16
> bit Signed Linear PCM (16kHz))
>   65536 (1 << 16)(0x1)  image   jpeg   (JPEG
> image)
>  131072 (1 << 17)(0x2)  imagepng   (PNG
> image)
>  262144 (1 << 18)(0x4)  video   h261
> (H.261 Video)
>  524288 (1 << 19)(0x8)  video   h263
> (H.263 Video)
> 1048576 (1 << 20)   (0x10)  video  h263p
> (H.263+ Video)
> 2097152 (1 << 21)   (0x20)  video   h264
> (H.264 Video)
> 4194304 (1 << 22)   (0x40)  video  mpeg4
> (MPEG4 Video)
> 8388608 (1 << 23)   (0x80)  videounknown
> (unknown)
>16777216 (1 << 24)  (0x100)  videounknown
> (unknown)
>33554432 (1 << 25)  (0x200)   textunknown
> (unknown)
>67108864 (1 << 26)  (0x400)   textred
> (T.140 Realtime Text with redundancy)
>   134217728 (1 << 27)  (0x800)   text   t140
> (Passthrough T.140 Realtime Text)
>   268435456 (1 << 28) (0x1000)   textunknown
> (unknown)
>   536870912 (1 << 29) (0x2000)   textunknown
> (unknown)
>  1073741824 (1 << 30) (0x4000)  (unk)unknown
> (unknown)
>  2147483648 (1 << 31) (0x8000)  (unk)unknown
> (unknown)
>  4294967296 (1 << 32)(0x1)  audio   g719   (ITU
> G.719)
>  8589934592 (1 << 33)(0x2)  audiospeex16
> (SpeeX 16khz)
> 17179869184 (1 << 34)(0x4)  audiounknown
> (unknown)
> 34359738368 (1 << 35)(0x8)  audiounknown
> (unknown)
> 68719476736 (1 << 36)   (0x10)  audiounknown
> (unknown)
>137438953472 (1 << 37)   (0x20)  audiounknown
> (unknown)
>274877906944 (1 << 38)   (0x40)  audiounknown
> (unknown)
>549755813888 (1 << 39)   (0x80)  audiounknown
> (unknown)
>   1099511627776 (1 << 40)  (0x100)  audiounknown
> (unknown)
>   219902322 (1 << 41)  (0x200)  audiounknown
> (unknown)
>   4398046511104 (1 << 42)  (0x400)  audiounknown
> (unknown)
>   8796093022208 (1 << 43)  (0x800)  audiounknown
> (unknown)
>  17592186044416 (1 << 44) (0x1000)  audiounknown
> (unknown)
>  35184372088832 (1 << 45) (0x2000)  audiounknown
> (unknown)
>  70368744177664 (1 << 46) (0x4000)  audiounknown
> (unknown)
> 140737488355328 (1 << 

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

[Marcelo Terres]

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 <mhter...@gmail.com>
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 09:32, Irfan PHEERUNGGEE
<i.pheerung...@thymbusiness.fr> wrote:
> Can someone tell me how do I stop receiving mails from asterisk-users please?
> How 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 List - Non-Commercial Discussion 
> <asterisk-users@lists.digium.com>
> Objet : Re: [asterisk-users] How to build with cdr_adaptive_odbc ?
>
> 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 <mhter...@gmail.com>
> IM: mhter...@jabber.mundoopensource.com.br
> https://www.mundoopensource.com.br
> https://twitter.com/mhterres
> https://linkedin.com/in/marceloterres
>
>
> On 17 April 2017 at 23:36, nous <pie...@couderc.eu> wrote:
>> Thank you, but unixodbc and odbcinst are installed... end even
>> unixodbc-dev
>>
>> But I get the same need for "generic odbc(E)".
>>
>>
>>
>> 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
>>> https://www.mundoopensource.com.br
>>> https://twitter.com/mhterres
>>> https://linkedin.com/in/marceloterres
>>>
>>>
>>> On 13 April 2017 at 19:41, Pierre Couderc <pie...@couderc.eu> wrote:
>>>>
>>>> I use debian stretch and I have installed unixodbc-dev
>>>>
>>>> but I have a dependency on genreric_odbc in make menuselect
>>>>
>>>> What am I missing ? Is there an howto ?
>>>>
>>>> Thanks
>>>> PX
>>>>
>>>> --
>>>> 
>>>> _
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>>>> https://community.asterisk.org/
>>>>
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>>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>>
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>>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
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Re: [asterisk-users] How to build with cdr_adaptive_odbc ?

[Marcelo Terres]

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 <mhter...@gmail.com>
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres


On 17 April 2017 at 23:36, nous <pie...@couderc.eu> wrote:
> Thank you, but unixodbc and odbcinst are installed... end even unixodbc-dev
>
> But I get the same need for "generic odbc(E)".
>
>
>
> 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
>> https://www.mundoopensource.com.br
>> https://twitter.com/mhterres
>> https://linkedin.com/in/marceloterres
>>
>>
>> On 13 April 2017 at 19:41, Pierre Couderc <pie...@couderc.eu> wrote:
>>>
>>> I use debian stretch and I have installed unixodbc-dev
>>>
>>> but I have a dependency on genreric_odbc in make menuselect
>>>
>>> What am I missing ? Is there an howto ?
>>>
>>> Thanks
>>> PX
>>>
>>> --
>>> _
>>> -- 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:
>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
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> Check out the new Asterisk community forum at:
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>
> New to Asterisk? Start here:
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>
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Re: [asterisk-users] How to build with cdr_adaptive_odbc ?

[Marcelo Terres]

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 Couderc  wrote:
> I use debian stretch and I have installed unixodbc-dev
>
> but I have a dependency on genreric_odbc in make menuselect
>
> What am I missing ? Is there an howto ?
>
> Thanks
> PX
>
> --
> _
> -- 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:
>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users

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Re: [asterisk-users] feature codes

[Marcelo Terres]

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


On 6 April 2017 at 08:46, Atux Atux  wrote:
> hi.
>
> i am running asterisk 11 and i am stuck with the feature codes. how do i
> setup them.
> Now the system has.
>
> PBX*CLI> features show
> Builtin Feature Default Current
> --- --- ---
> Pickup *8 *8
> Blind Transfer # #
> Attended Transfer
> One Touch Monitor
> Disconnect Call * *
> Park Call
> One Touch MixMonitor
>
> Dynamic Feature Default Current
> --- --- ---
> (none)
>
> Feature Groups:
> ---
> (none)
>
> Call parking (Parking lot: default)
> 
> Parking extension : 700
> Parking context : parkedcalls
> Parked call extensions: 701-750
> Parkingtime : 45000 ms
> Comeback to origin : yes
> Comeback context : parkedcallstimeout (comebacktoorigin=yes, not used)
> Comeback dial time : 30
> MusicOnHold class : default
> Enabled : Yes
> PBX*CLI>
>
> My extensions.conf is:
> exten => _2X,1,Dial(SIP/CYTA/${EXTEN})
> exten => _2X,1,Busy()
> exten => _69,1,Dial(SIP/voda/${EXTEN})
> exten => _69,1,Busy()
> [code]
>
> I would like to be able to transfer calls, blind/attended transfer, call
> forward, DND. I would appreciate any help available please.
>
>
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>
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Re: [asterisk-users] Soft SIP phones that support TLS - Asterisk version 13.13.1

[Marcelo Terres]

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 Soft phones with TLS authentication enabled? Any suggestions?
>
>
>
> Thanks,
> Motty
>
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Re: [asterisk-users] 14.3.0 download archive corrupt - cannot extract

[Marcelo Terres]

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 14 February 2017 at 14:01, Joshua Colp <jc...@digium.com> 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...
>
> An issue was filed for tracking this[1] and it will be resolved later
> today.
>
> [1] https://issues.asterisk.org/jira/browse/ASTERISK-26791
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
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>
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Re: [asterisk-users] 14.3.0 download archive corrupt - cannot extract

[Marcelo Terres]

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 2017 at 08:42, Jonathan H  wrote:
> Hi there;
>
> 2 linux boxes and Windows all report an error and the archive is not
> extractable.
>
> Wget reports the size as follows:
>
> 2017-02-14 08:36:21 (7.29 MB/s) - ‘asterisk-14-current.tar.gz’ saved
> [40653605/40653605]
>
> It starts un-tarring but then
>
> asterisk-14.3.0/bridges/bridge_native_rtp.c
> asterisk-14.3.0/sounds/
> asterisk-14.3.0/sounds/asterisk-core-sounds-en-gsm-1.5.tar.gz
>
> gzip: stdin: invalid compressed data--format violated
> tar: Unexpected EOF in archive
> tar: Unexpected EOF in archive
> tar: Error is not recoverable: exiting now
>
>
>
>
>
>
>
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Re: [asterisk-users] Developing Asterisk Modules

[Marcelo Terres]

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 18 January 2017 at 19:51, Valter Nogueira  wrote:
> I want to develop an Asterisk module, but I have several doubts when
> building Asterisk plus my module - which is inside src/apps directory.
>
> -How to compile and link additional .c programs, like chan_sip does?
>
> -How to compile and link libpq inside an Asterisk module?
>
> -How do I debug Asterisk modules?
>
>
> Thanks
>
> Valter
>
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Re: [asterisk-users] Asterisk 11.23 with libmysqlclient20 on Debian 8

[Marcelo Terres]

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
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres


On Mon, Oct 3, 2016 at 7:54 PM, Victor Villarreal  wrote:
> Hi List!
>
> I'm facing a problem while compiling Asterisk-11 on a Debian 8 server.
>
> The mysql-server version installed is 5.7 and come from the official mySQL
> community repo for Debian.
>
> After compile, install and execute Asterisk, the comman "lsof -p `pidof
> asterisk` | grep mysql" don't produce any output. Like if confgure script
> don't found the mysql lib.
>
> With libmysqlclient18 every is Ok. How can I use libmysqlclient20 with
> Asterisk ?
>
> Thanks in advance, and best regards.
>
> root@nodo1:/usr/src/asterisk-11.23.0# ls -lh /usr/lib/x86_64-linux-gnu/ |
> grep mysql
> -rw-r--r-- 1 root root 5,7M ago 25 09:37 libmysqlclient.a
> lrwxrwxrwx 1 root root   20 ago 25 09:37 libmysqlclient.so ->
> libmysqlclient.so.20
> lrwxrwxrwx 1 root root   24 ago 25 09:37 libmysqlclient.so.20 ->
> libmysqlclient.so.20.3.2
> -rw-r--r-- 1 root root 4,2M ago 25 09:37 libmysqlclient.so.20.3.2
> -rw-r--r-- 1 root root  18K ago 25 09:37 libmysqlservices.a
>
> --
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> https://www.github.com/mefhigoseth
> ...:::[ God Rulz ! ]:::...
>
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[asterisk-users] Using Asterisk and XMPP - AstriCon 2016

[Marcelo Terres]

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
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres
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Re: [asterisk-users] Problems with queues realtime configuration

[Marcelo Terres]

Confirmed, I open an issue about it:

[JIRA] (ASTERISK-26431) Queues doesn't appear when using realtime configuration


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, 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 realtime configuration but the system is not
> loading the queues.
>
> Let me show you what happens when I'm trying to load the module
> app_queue.so (core set debug 100):
>
> [Oct  1 18:33:15] DEBUG[18028] loader.c: Registering module app_queue
> [Oct  1 18:33:15] DEBUG[18028] config.c: Parsing /etc/asterisk/queuerules.conf
> [Oct  1 18:33:15] VERBOSE[18028] config.c: Parsing
> '/etc/asterisk/queuerules.conf': Found
> [Oct  1 18:33:15] DEBUG[18028] config.c: Parsing /etc/asterisk/queues.conf
> [Oct  1 18:33:15] VERBOSE[18028] config.c: Parsing
> '/etc/asterisk/queues.conf': Found
> [Oct  1 18:33:15] DEBUG[18028] res_odbc.c: Reusing ODBC handle
> 0x1f1a3d0 from class 'asterisk'
> [Oct  1 18:33:15] DEBUG[18028] res_odbc.c: Found queue_name column
> with type 12 with len 80, octetlen 255, and numlen (0,0)
> [Oct  1 18:33:15] DEBUG[18028] res_odbc.c: Found interface column with
> type 12 with len 80, octetlen 255, and numlen (0,0)
> [Oct  1 18:33:15] DEBUG[18028] res_odbc.c: Found membername column
> with type 12 with len 80, octetlen 255, and numlen (0,0)
> [Oct  1 18:33:15] DEBUG[18028] res_odbc.c: Found state_interface
> column with type 12 with len 80, octetlen 255, and numlen (0,0)
> [Oct  1 18:33:15] DEBUG[18028] res_odbc.c: Found penalty column with
> type 4 with len 10, octetlen -1, and numlen (0,10)
> [Oct  1 18:33:15] DEBUG[18028] res_odbc.c: Found paused column with
> type 4 with len 10, octetlen -1, and numlen (0,10)
> [Oct  1 18:33:15] DEBUG[18028] res_odbc.c: Found uniqueid column with
> type 4 with len 10, octetlen -1, and numlen (0,10)
> [Oct  1 18:33:15] DEBUG[18028] res_odbc.c: Releasing ODBC handle
> 0x1f1a3d0 into pool
> [Oct  1 18:33:15] DEBUG[18028] res_odbc.c: Reusing ODBC handle
> 0x1f1a3d0 from class 'asterisk'
> [Oct  1 18:33:15] DEBUG[18028] res_config_odbc.c: Skip: 0; SQL: SELECT
> * FROM queue_members WHERE interface LIKE ? AND queue_name LIKE ?
> ORDER
> BY interface
> [Oct  1 18:33:15] DEBUG[18028] res_config_odbc.c: Parameter 1
> ('interface LIKE') = '%'
> [Oct  1 18:33:15] DEBUG[18028] res_config_odbc.c: Parameter 2
> ('queue_name LIKE') = '%'
> [Oct  1 18:33:15] DEBUG[18028] res_odbc.c: Releasing ODBC handle
> 0x1f1a3d0 into pool
> [Oct  1 18:33:15] NOTICE[18028] app_queue.c: No entries were found for
> ringinuse/ignorebusy in queue_members table. Using 'ringinuse'
> [Oct  1 18:33:15] VERBOSE[18028] pbx_app.c: Registered application 'Queue'
> [Oct  1 18:33:15] VERBOSE[18028] pbx_app.c: Registered application
> 'AddQueueMember'
> [Oct  1 18:33:15] VERBOSE[18028] pbx_app.c: Registered application
> 'RemoveQueueMember'
> [Oct  1 18:33:15] VERBOSE[18028] pbx_app.c: Registered application
> 'PauseQueueMember'
> [Oct  1 18:33:15] VERBOSE[18028] pbx_app.c: Registered application
> 'UnpauseQueueMember'
> [Oct  1 18:33:15] VERBOSE[18028] pbx_app.c: Registered application 'QueueLog'
> [Oct  1 18:33:15] VERBOSE[18028] manager.c: Manager registered action Queues
> [Oct  1 18:33:15] VERBOSE[18028] manager.c: Manager registered action
> QueueStatus
> [Oct  1 18:33:15] VERBOSE[18028] manager.c: Manager registered action
> QueueSummary
> [Oct  1 18:33:15] VERBOSE[18028] manager.c: Manager registered action QueueAdd
> [Oct  1 18:33:15] VERBOSE[18028] manager.c: Manager registered action
> QueueRemove
> [Oct  1 18:33:15] VERBOSE[18028] manager.c: Manager registered action 
> QueuePause
> [Oct  1 18:33:15] VERBOSE[18028] manager.c: Manager registered action QueueLog
> [Oct  1 18:33:15] VERBOSE[18028] manager.c: Manager registered action
> QueuePenalty
> [Oct  1 18:33:15] VERBOSE[18028] manager.c: Manager registered action
> QueueMemberRingInUse
> [Oct  1 18:33:15] VERBOSE[18028] manager.c: Manager registered action 
> QueueRule
> [Oct  1 18:33:15] VERBOSE[18028] manager.c: Manager registered action
> QueueReload
> [Oct  1 18:33:15] VERBOSE[18028] manager.c: Manager registered action 
> QueueReset
> [Oct  1 18:33:15] VERBOSE[18028] pbx_functions.c: Registered custom
> function 'QUEUE_VARIABLES'
> [Oct  1 18:33:15] VERBOSE[18028] pbx_functions.c: Registered custom
> function 'QUEUE_EXISTS'
> [Oct  1 18:33:15] VERBOSE[18028] pbx_functions.c: Registered custom
> function 'QUEUE_MEMBER'

[asterisk-users] Problems with queues realtime configuration

[Marcelo Terres]

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
app_queue.so (core set debug 100):

[Oct  1 18:33:15] DEBUG[18028] loader.c: Registering module app_queue
[Oct  1 18:33:15] DEBUG[18028] config.c: Parsing /etc/asterisk/queuerules.conf
[Oct  1 18:33:15] VERBOSE[18028] config.c: Parsing
'/etc/asterisk/queuerules.conf': Found
[Oct  1 18:33:15] DEBUG[18028] config.c: Parsing /etc/asterisk/queues.conf
[Oct  1 18:33:15] VERBOSE[18028] config.c: Parsing
'/etc/asterisk/queues.conf': Found
[Oct  1 18:33:15] DEBUG[18028] res_odbc.c: Reusing ODBC handle
0x1f1a3d0 from class 'asterisk'
[Oct  1 18:33:15] DEBUG[18028] res_odbc.c: Found queue_name column
with type 12 with len 80, octetlen 255, and numlen (0,0)
[Oct  1 18:33:15] DEBUG[18028] res_odbc.c: Found interface column with
type 12 with len 80, octetlen 255, and numlen (0,0)
[Oct  1 18:33:15] DEBUG[18028] res_odbc.c: Found membername column
with type 12 with len 80, octetlen 255, and numlen (0,0)
[Oct  1 18:33:15] DEBUG[18028] res_odbc.c: Found state_interface
column with type 12 with len 80, octetlen 255, and numlen (0,0)
[Oct  1 18:33:15] DEBUG[18028] res_odbc.c: Found penalty column with
type 4 with len 10, octetlen -1, and numlen (0,10)
[Oct  1 18:33:15] DEBUG[18028] res_odbc.c: Found paused column with
type 4 with len 10, octetlen -1, and numlen (0,10)
[Oct  1 18:33:15] DEBUG[18028] res_odbc.c: Found uniqueid column with
type 4 with len 10, octetlen -1, and numlen (0,10)
[Oct  1 18:33:15] DEBUG[18028] res_odbc.c: Releasing ODBC handle
0x1f1a3d0 into pool
[Oct  1 18:33:15] DEBUG[18028] res_odbc.c: Reusing ODBC handle
0x1f1a3d0 from class 'asterisk'
[Oct  1 18:33:15] DEBUG[18028] res_config_odbc.c: Skip: 0; SQL: SELECT
* FROM queue_members WHERE interface LIKE ? AND queue_name LIKE ?
ORDER
BY interface
[Oct  1 18:33:15] DEBUG[18028] res_config_odbc.c: Parameter 1
('interface LIKE') = '%'
[Oct  1 18:33:15] DEBUG[18028] res_config_odbc.c: Parameter 2
('queue_name LIKE') = '%'
[Oct  1 18:33:15] DEBUG[18028] res_odbc.c: Releasing ODBC handle
0x1f1a3d0 into pool
[Oct  1 18:33:15] NOTICE[18028] app_queue.c: No entries were found for
ringinuse/ignorebusy in queue_members table. Using 'ringinuse'
[Oct  1 18:33:15] VERBOSE[18028] pbx_app.c: Registered application 'Queue'
[Oct  1 18:33:15] VERBOSE[18028] pbx_app.c: Registered application
'AddQueueMember'
[Oct  1 18:33:15] VERBOSE[18028] pbx_app.c: Registered application
'RemoveQueueMember'
[Oct  1 18:33:15] VERBOSE[18028] pbx_app.c: Registered application
'PauseQueueMember'
[Oct  1 18:33:15] VERBOSE[18028] pbx_app.c: Registered application
'UnpauseQueueMember'
[Oct  1 18:33:15] VERBOSE[18028] pbx_app.c: Registered application 'QueueLog'
[Oct  1 18:33:15] VERBOSE[18028] manager.c: Manager registered action Queues
[Oct  1 18:33:15] VERBOSE[18028] manager.c: Manager registered action
QueueStatus
[Oct  1 18:33:15] VERBOSE[18028] manager.c: Manager registered action
QueueSummary
[Oct  1 18:33:15] VERBOSE[18028] manager.c: Manager registered action QueueAdd
[Oct  1 18:33:15] VERBOSE[18028] manager.c: Manager registered action
QueueRemove
[Oct  1 18:33:15] VERBOSE[18028] manager.c: Manager registered action QueuePause
[Oct  1 18:33:15] VERBOSE[18028] manager.c: Manager registered action QueueLog
[Oct  1 18:33:15] VERBOSE[18028] manager.c: Manager registered action
QueuePenalty
[Oct  1 18:33:15] VERBOSE[18028] manager.c: Manager registered action
QueueMemberRingInUse
[Oct  1 18:33:15] VERBOSE[18028] manager.c: Manager registered action QueueRule
[Oct  1 18:33:15] VERBOSE[18028] manager.c: Manager registered action
QueueReload
[Oct  1 18:33:15] VERBOSE[18028] manager.c: Manager registered action QueueReset
[Oct  1 18:33:15] VERBOSE[18028] pbx_functions.c: Registered custom
function 'QUEUE_VARIABLES'
[Oct  1 18:33:15] VERBOSE[18028] pbx_functions.c: Registered custom
function 'QUEUE_EXISTS'
[Oct  1 18:33:15] VERBOSE[18028] pbx_functions.c: Registered custom
function 'QUEUE_MEMBER'
[Oct  1 18:33:15] VERBOSE[18028] pbx_functions.c: Registered custom
function 'QUEUE_MEMBER_COUNT'
[Oct  1 18:33:15] VERBOSE[18028] pbx_functions.c: Registered custom
function 'QUEUE_MEMBER_LIST'
[Oct  1 18:33:15] VERBOSE[18028] pbx_functions.c: Registered custom
function 'QUEUE_GET_CHANNEL'
[Oct  1 18:33:15] VERBOSE[18028] pbx_functions.c: Registered custom
function 'QUEUE_WAITING_COUNT'
[Oct  1 18:33:15] VERBOSE[18028] pbx_functions.c: Registered custom
function 'QUEUE_MEMBER_PENALTY'
[Oct  1 18:33:15] VERBOSE[18028] loader.c: Loaded app_queue.so =>
(True Call Queueing)

As you can see, there is no mention of table queues, just of table
queue_members and probably 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-users] Asterisk and XMPP - AstriCon 2016

[Marcelo Terres]

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

Marcelo H. Terres 
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres

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

[Marcelo Terres]

Hello Jonathan,

https://issues.asterisk.org/jira/browse/ASTERISK-26391

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 Tue, Sep 20, 2016 at 4:19 AM, Jonathan H <lardconce...@gmail.com> wrote:
> Great! Thanks, team, but just before I file a bug..
>
> No matter how many v and d I put, when I now do "dialplan reload" in
> v14, it just says "Dialplan reloaded".
>
> Previously, it used 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> wrote:
>> Marcelo Terres wrote:
>>>
>>> I noticed another different behaviour.
>>>
>>> In older versions, when I call rasterisk, I receive some informations
>>> about it. Fox example:
>>>
>>> [root@pbx2 ~]# rasterisk
>>> Asterisk 11.22.0, Copyright (C) 1999 - 2013 Digium, Inc. and others.
>>> Created by Mark Spencer<marks...@digium.com>
>>> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty'
>>> for details.
>>> This is free software, with components licensed under the GNU General
>>> Public
>>> License version 2 and other licenses; you are welcome to redistribute it
>>> under
>>> certain conditions. Type 'core show license' for details.
>>> =
>>> Running as user 'asterisk'
>>> Running under group 'asterisk'
>>> Connected to Asterisk 11.22.0 currently running on pbx2 (pid = 1461)
>>> pbx2*CLI>
>>> Disconnected from Asterisk server
>>> Asterisk cleanly ending (0).
>>> Executing last minute cleanups
>>>
>>>
>>> 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?
>>
>>
>> I don't know of anything which explicitly disabled this. Please file an
>> issue[1].
>>
>> [1] https://issues.asterisk.org/jira
>>
>> --
>> Joshua Colp
>> Digium, Inc. | Senior 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 http://www.api-digital.com --
>>
>> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>>  http://www.asterisk.org/community/astricon-user-conference
>>
>> New to Asterisk? Start here:
>>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>   http://www.asterisk.org/community/astricon-user-conference
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

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

[Marcelo Terres]

That's the same behavior that I noticed.

I'll open an issue about it soon.
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 Tue, Sep 20, 2016 at 4:19 AM, Jonathan H <lardconce...@gmail.com> wrote:
> Great! Thanks, team, but just before I file a bug..
>
> No matter how many v and d I put, when I now do "dialplan reload" in
> v14, it just says "Dialplan reloaded".
>
> Previously, it used 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> wrote:
>> Marcelo Terres wrote:
>>>
>>> I noticed another different behaviour.
>>>
>>> In older versions, when I call rasterisk, I receive some informations
>>> about it. Fox example:
>>>
>>> [root@pbx2 ~]# rasterisk
>>> Asterisk 11.22.0, Copyright (C) 1999 - 2013 Digium, Inc. and others.
>>> Created by Mark Spencer<marks...@digium.com>
>>> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty'
>>> for details.
>>> This is free software, with components licensed under the GNU General
>>> Public
>>> License version 2 and other licenses; you are welcome to redistribute it
>>> under
>>> certain conditions. Type 'core show license' for details.
>>> =
>>> Running as user 'asterisk'
>>> Running under group 'asterisk'
>>> Connected to Asterisk 11.22.0 currently running on pbx2 (pid = 1461)
>>> pbx2*CLI>
>>> Disconnected from Asterisk server
>>> Asterisk cleanly ending (0).
>>> Executing last minute cleanups
>>>
>>>
>>> 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?
>>
>>
>> I don't know of anything which explicitly disabled this. Please file an
>> issue[1].
>>
>> [1] https://issues.asterisk.org/jira
>>
>> --
>> Joshua Colp
>> Digium, Inc. | Senior 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 http://www.api-digital.com --
>>
>> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>>  http://www.asterisk.org/community/astricon-user-conference
>>
>> New to Asterisk? Start here:
>>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>   http://www.asterisk.org/community/astricon-user-conference
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

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

[Marcelo Terres]

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:
> Marcelo Terres wrote:
>>
>> 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. 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.
>
>
> This actually happened between Asterisk 13.7.0 and 13.8.0, with a subsequent
> change reverting it somewhat and moved back into Asterisk. I've created an
> issue[1] to clean this up, both the upgrade and the sample config.
>
> [1] https://issues.asterisk.org/jira/browse/ASTERISK-26389
>
> -- Joshua Colp
> Digium, Inc. | Senior 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 http://www.api-digital.com --
>
> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>  http://www.asterisk.org/community/astricon-user-conference
>
> New to Asterisk? Start here:
>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users

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  http://www.asterisk.org/community/astricon-user-conference

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  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[Marcelo Terres]

I noticed another different behaviour.

In older versions, when I call rasterisk, I receive some informations
about it. Fox example:

[root@pbx2 ~]# rasterisk
Asterisk 11.22.0, Copyright (C) 1999 - 2013 Digium, Inc. and others.
Created by Mark Spencer <marks...@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty'
for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=
Running as user 'asterisk'
Running under group 'asterisk'
Connected to Asterisk 11.22.0 currently running on pbx2 (pid = 1461)
pbx2*CLI>
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups


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 <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 7:49 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_odbc.conf.sample file, because it still
> contains idlecheck and limit parameters.
>
> 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 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 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. 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 <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 12:49 PM, Asterisk Development Team
>> <asteriskt...@digium.com> wrote:
>>> The Asterisk Development Team has announced the first release candidate of
>>> Asterisk 14.0.0. This release candidate is available for immediate
>>> download at http://downloads.asterisk.org/pub/telephony/asterisk
>>>
>>> The release of Asterisk 14.0.0-rc1 resolves several issues reported by the
>>> community and would have not been possible without your participation.
>>> Thank you!
>>>
>>> The following are the issues resolved in this release candidate:
>>>
>>> Bugs fixed in this release:
>>> ---
>>>  * ASTERISK-26375 - res_pjsip_transport_management: Log message
>>>   states seconds, but time value is milliseconds (Reported by
>>>   Joshua Colp)
>>>  * ASTERISK-26364 - res_pjsip: Don't assume a request will have
>>>   target addresses (Reported by Joshua Colp)
>>>  * ASTERISK-26349 -  13.11.1 res_pjsip/pjsip_distributor.c: Request
>>>   'REGISTER' failed (Reported by Dmitry Melekhov)
>>>  * ASTERISK-26264 - res_pjsip: Crash when applying ACL from
>>>   non-existent endpoint (Reported by nappsoft)
>>>  * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets)
>>>   (Reported by Etienne Lessard)
>>>  * ASTERISK-26341 - ARI: Stopping a media playlist only stops the
>>>   current media URI being played back, and not the whole list
>>>   (Reported by Matt Jordan)
>>>
>>> For a full list of changes in this release candidate, please see the
>>> ChangeLog:
>>>
>>> http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.0.0-rc1
>>>
>>> Thank you for your continued support of Asterisk!
>>>
>>> --
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>>

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

[Marcelo Terres]

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 <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 7:44 PM, Marcelo Terres <mhter...@gmail.com> wrote:
> 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. 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 <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 12:49 PM, Asterisk Development Team
> <asteriskt...@digium.com> wrote:
>> The Asterisk Development Team has announced the first release candidate of
>> Asterisk 14.0.0. This release candidate is available for immediate
>> download at http://downloads.asterisk.org/pub/telephony/asterisk
>>
>> The release of Asterisk 14.0.0-rc1 resolves several issues reported by the
>> community and would have not been possible without your participation.
>> Thank you!
>>
>> The following are the issues resolved in this release candidate:
>>
>> Bugs fixed in this release:
>> ---
>>  * ASTERISK-26375 - res_pjsip_transport_management: Log message
>>   states seconds, but time value is milliseconds (Reported by
>>   Joshua Colp)
>>  * ASTERISK-26364 - res_pjsip: Don't assume a request will have
>>   target addresses (Reported by Joshua Colp)
>>  * ASTERISK-26349 -  13.11.1 res_pjsip/pjsip_distributor.c: Request
>>   'REGISTER' failed (Reported by Dmitry Melekhov)
>>  * ASTERISK-26264 - res_pjsip: Crash when applying ACL from
>>   non-existent endpoint (Reported by nappsoft)
>>  * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets)
>>   (Reported by Etienne Lessard)
>>  * ASTERISK-26341 - ARI: Stopping a media playlist only stops the
>>   current media URI being played back, and not the whole list
>>   (Reported by Matt Jordan)
>>
>> For a full list of changes in this release candidate, please see the
>> ChangeLog:
>>
>> http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.0.0-rc1
>>
>> Thank you for your continued support of Asterisk!
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>>   http://www.asterisk.org/community/astricon-user-conference
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users

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

[Marcelo Terres]

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


On Mon, Sep 19, 2016 at 12:49 PM, Asterisk Development Team
 wrote:
> The Asterisk Development Team has announced the first release candidate of
> Asterisk 14.0.0. This release candidate is available for immediate
> download at http://downloads.asterisk.org/pub/telephony/asterisk
>
> The release of Asterisk 14.0.0-rc1 resolves several issues reported by the
> community and would have not been possible without your participation.
> Thank you!
>
> The following are the issues resolved in this release candidate:
>
> Bugs fixed in this release:
> ---
>  * ASTERISK-26375 - res_pjsip_transport_management: Log message
>   states seconds, but time value is milliseconds (Reported by
>   Joshua Colp)
>  * ASTERISK-26364 - res_pjsip: Don't assume a request will have
>   target addresses (Reported by Joshua Colp)
>  * ASTERISK-26349 -  13.11.1 res_pjsip/pjsip_distributor.c: Request
>   'REGISTER' failed (Reported by Dmitry Melekhov)
>  * ASTERISK-26264 - res_pjsip: Crash when applying ACL from
>   non-existent endpoint (Reported by nappsoft)
>  * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets)
>   (Reported by Etienne Lessard)
>  * ASTERISK-26341 - ARI: Stopping a media playlist only stops the
>   current media URI being played back, and not the whole list
>   (Reported by Matt Jordan)
>
> For a full list of changes in this release candidate, please see the
> ChangeLog:
>
> http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.0.0-rc1
>
> Thank you for your continued support of Asterisk!
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>   http://www.asterisk.org/community/astricon-user-conference
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

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

[Marcelo Terres]

.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)
>  * ASTERISK-25725 - core: Incorrect XML documentation may result in
>   weird behavior (Reported by Joshua Colp)
>  * ASTERISK-25722 - ASAN & testsute: stack-buffer-overflow in
>   sip_sipredirect (Reported by Badalian Vyacheslav)
>  * ASTERISK-25709 - ARI: Crash can occur due to race condition when
>   attempting to operate on a hung up channel (Reported by Mark
>   Michelson)
>  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
>   by Badalian Vyacheslav)
>  * ASTERISK-25685 - infrastructure: Run alembic in Jenkins build
>   script (Reported by Joshua Colp)
>  * ASTERISK-25712 - Second call to already-on-call phone and
>   Asterisk sends "Ready" (Reported by Richard Mudgett)
>  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
>   (Reported by Badalian Vyacheslav)
>  * ASTERISK-25179 - CDR(billsec,f) and CDR(duration,f) report
>   incorrect values (Reported by Gianluca Merlo)
>  * ASTERISK-25611 - core: threadpool thread_timeout_thrash unit
>   test sporadically failing (Reported by Joshua Colp)
>  * ASTERISK-25686 - PJSIP: qualify_timeout is a double, database
>   schema is an integer (Reported by Marcelo Terres)
>  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
>   (Reported by Corey Farrell)
>  * ASTERISK-25696 - bridge_basic: don't cache xferfailsound during
>   a transfer (Reported by Kevin Harwell)
>  * ASTERISK-25697 - bridge_basic: don't play an attended transfer
>   fail sound after target hangs up (Reported by Kevin Harwell)
>  * ASTERISK-25683 - res_ari: Asterisk fails to start if compiled
>   with MALLOC_DEBUG  (Reported by yaron nahum)
>  * ASTERISK-24097 - Documentation - CHANNEL function help text
>   missing 'linkedid' argument (Reported by Steven T. Wheeler)
>  * ASTERISK-25690 - Hanging up when executing connected line sub
>   does not cause hangup (Reported by Joshua Colp)
>  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
>   reload' cause a crash (Reported by Sean Bright)
>  * ASTERISK-25632 - res_pjsip_sdp_rtp: RTP is sent from wrong IP
>   address when multihomed (Reported by Olivier Krief)
>  * ASTERISK-25637 - Multi homed server using wrong IP (Reported by
>   Daniel Journo)
>  * ASTERISK-25394 - pbx: Incorrect device and presence state when
>   changing hint details (Reported by Joshua Colp)
>  * ASTERISK-25640 - pbx: Deadlock on features reload and state
>   change hint. (Reported by Krzysztof Trempala)
>  * ASTERISK-25681 - devicestate: Engine thread is not shut down
>   (Reported by Corey Farrell)
>  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
>   shutdown (Reported by Corey Farrell)
>  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
>   Corey Farrell)
>  * ASTERISK-25675 - Endpoint not listed as Unreachable (Reported by
>   Daniel Journo)
>  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
>   by Corey Farrell)
>  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
>   Farrell)
>  * ASTERISK-25668 - res_pjsip: Deadlock in distributor (Reported by
>   Mark Michelson)
>  * ASTERISK-25664 - ast_format_cap_append_by_type leaks a reference
>   (Reported by Corey Farrell)
>  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
>   ADD_VENDOR_CODE (Reported by Aaron An)
>  * ASTERISK-25137 - endpoint stasis messages are delivered twice
>   (Reported by Vitezslav Novy)
>  * ASTERISK-25116 - res_pjsip:  Two PeerStatus AMI messages are
>   sent for every status change (Reported by George Joseph)
>  * ASTERISK-25641 - bridge: GOTO_ON_BLINDXFR doesn't work on
>   transfer initiated channel (Reported by Dmitry Melekhov)
>  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
>   Brandon)
>  * ASTERISK-25625 - res_sorcery_memory_cache: Add full backend
>   caching (Reported by Joshua Colp)
>  * ASTERISK-25601 - json: Audit reference usage and thread safety
>   (Reported by Joshua Colp)
>  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
>   sungtae kim)
>  * ASTERISK-25615 - res_pjsip: Setting transport async_operations >
>   1 causes segfault on tls transports (Reported by George Joseph)
>  * ASTERISK-25442 - using realtime (mysql) queue members are never
>   updated in wait_our_turn function (app_queue.c)  (Reported by
>   Carlos Oliva)
>  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
>   thread of aster

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

[Marcelo Terres]

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:
> Marcelo Terres wrote:
>>
>> 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 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?
>
>
> The version and capability check for unbound was too strict and has been
> tweaked since the initial beta1 release. The next beta (or rc) will have the
> fix, and it's confirmed to work against Ubuntu 14.04.
>
> --
> Joshua Colp
> Digium, Inc. | Senior 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 http://www.api-digital.com --
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>   http://www.asterisk.org/hello
>
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Re: [asterisk-users] Asterisk 14.0.0-beta1 Now Available

[Marcelo Terres]

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


On Wed, Jul 27, 2016 at 6:02 PM, Asterisk Development Team
 wrote:
> The Asterisk Development Team has announced the first beta of
> Asterisk 14.0.0. This beta is available for immediate
> download at http://downloads.asterisk.org/pub/telephony/asterisk
>
> The release of Asterisk 14.0.0-beta1 resolves several issues reported by the
> community and would have not been possible without your participation.
> Thank you!
>

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Re: [asterisk-users] PJSIP not detected

[Marcelo Terres]

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, 2016 at 4:08 PM, Saint Michael  wrote:
> I installed PJSIP from the project
> git clone https://github.com/asterisk/pjproject pjproject
> cd pjproject
> make uninstall & make distclean
> ./configure --libdir=/usr/lib64 --prefix=/ --enable-shared --disable-sound
> --disable-resample --disable-video --disable-opencore-amr
> --with-external-srtp
> make dep && make && make install && ldconfig && ldconfig -p | grep pj
>
> and it is there, but the configure for Asterisk 13.11.0-rc1 does not detect
> it and it cannot compile it.
> What am I doing wrong? The box is Ubuntu 14.04 LTS
>
>
>
> --
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[asterisk-users] AstriCon 2016 - XMPP and Asterisk

[Marcelo Terres]

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: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres

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Re: [asterisk-users] WhatsApp feature on Asterisk

[Marcelo Terres]

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, 2016 at 6:15 AM, Yves biganiro  wrote:
> Can anyone put light on whatsapp features   and how it can be operated .
> What are the technology that need to be installed ,
>
> Regards
>
> Yves
>
>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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[asterisk-users] Yealink T21P E2 bug solved

[Marcelo Terres]

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

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Re: [asterisk-users] SIP trunk

[Marcelo Terres]

_.  ?

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 trunk.
>
> How can I match "anything" or "nothing" and start my extension.
>
> Usually I have something like:
> exten => 55,1,Goto(,yyy,1)
>
> but if 55 does not come across and it appears to be no digits
> coming across how do I match that that and just start.
>
> I thought about _X but that says digits. I dont think I am getting any
> digits
> I just want *anything* coming across to start the call.
>
> Basically ANY call coming across the trunk just do the same as 55 above.
>
> Thanks,
>
> Jerry
>
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Re: [asterisk-users] Asterisk and Yealink T21P E2

[Marcelo Terres]

No problems with authentication during invite after reboot?

I'm using insecure=no in SIP configuration.

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 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 <mhter...@gmail.com>
>> IM: mhter...@jabber.mundoopensource.com.br
>> https://www.mundoopensource.com.br
>> https://twitter.com/mhterres
>> https://linkedin.com/in/marceloterres
>>
>
> Sure.  Tons of them.
>
> --
> Jeff LaCoursiere
> 312 962 5250 desk
> 815 546 6599 cell
>
>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users

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[asterisk-users] Asterisk and Yealink T21P E2

[Marcelo Terres]

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

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[asterisk-users] AstriCon 2016 - crowdfunding for participation

[Marcelo Terres]

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 about it in
https://astricon2016.sched.org/event/7Zje/using-asterisk-and-xmpp-to-provide-greater-tools-to-your-customers-and-your-users.

If you want to help, don't be shy, I'm certain you won't regret. :-)
And even if you won't attend the conference, you can always see 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...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres

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[asterisk-users] Yealink T21P_E2 - bugs in firmware

[Marcelo Terres]

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

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

[Marcelo Terres]

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 structure of plugins and
monitoring agents that allows system administrators to develop their
own features.

You can find more informations (including source code for download) in
my blog at 
https://www.mundoopensource.com.br/xybot-xmpp-bot-that-allows-users-to-interact-with-asterisk/.

Any doubts 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

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

[Marcelo Terres]

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 in a heterogeneous enterprise scenario.

Despite the fact that I started this project to integrate Asterisk IVRs
with customer service softwares, this is a multipurpose project that can be
used with any kind of software that you want.

The project uses the Asterisk's ARI API and XMPP (PubSub) to deliver the
information.

You can find more informations (including source code for download) in my
blog at
https://www.mundoopensource.com.br/delivering-asterisk-ivr-data-to-softwares-using-xmpp/
.

Any doubts or suggestions are welcomed.

Regards,
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Re: [asterisk-users] PJProject Bundled Update

[Marcelo Terres]

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
> > introduced with Asterisk 13.8.0.
> >
> > More info on the Asterisk Wiki
> > <
> https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject#BuildingandInstallingpjproject-bundled>
> and
> > in this email thread
> >  >.
> >
> > Since then I've fixed a few issues related to older versions of Debian
> > and CentOS which you can in these 2 patches.
> > https://gerrit.asterisk.org//2516
> > https://gerrit.asterisk.org/2449
> >
> > Any other feedback?  I'd like to get an idea of how many folks have
> > tried it.
> >
> > Thanks
> > george
> >
>
> Built on fedora 23.  speexdsp-devel is required. It provides speex_echo.h .
>
> Haven't actually run it, but it built.
>
> Thanks for all the work. This much easier. Maybe i'll switch to 13.
>
> sean
>
>
>
>
>
>
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[asterisk-users] X-RTP-Stat SIP header

[Marcelo 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 negócios
Service | Telecom | Tech | Data Science
www.propus.com.br
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[asterisk-users] A little OT: AstDemo - Openfire and Asterisk integration

[Marcelo Terres]

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 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.com
IM: marc...@jabber.mundoopensource.com.br
http://www.mundoopensource.com.br
http://offtopicsandfun.blogspot.com
http://biertasters.blogspot.com
http://twitter.com/mhterres
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[asterisk-users] OT: Openfire - new version of B9 plugin released

[Marcelo Terres]

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 define a password to protect socket connections.

You can read the release notes here:
http://www.mundoopensource.com.br/versao-0-3-plugin-b9-lancada/
(portuguese only).
Changelog is here: http://files.mundoopensource.com.br/plugins/b9_changelog.html
More informations here: http://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

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

[Marcelo Terres]

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://offtopicsandfun.blogspot.com
http://biertasters.blogspot.com
http://twitter.com/mhterres

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Re: [asterisk-users] JABBER_STATUS CODE 7

[Marcelo Terres]

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 at 6:06 PM, ricky gutierrez xserverli...@gmail.com wrote:
 2014-10-13 14:44 GMT-06:00 Matthew Jordan mjor...@digium.com:

 The error message is pretty explicit about what you asked it to look for:

 {quote}
 acf_jabberstatus_read: Resource alcides of buddy operadora@172.16.8.59
 was not found.


 strange, I put the fqdn to ejabberd, and now , not shows the code 7

 [Oct 13 14:53:08] WARNING[4609][C-000f]: res_xmpp.c:1617
 acf_jabberstatus_read: Could not find buddy in list:
 'operad...@xmpp.domain.com'
 -- Executing [0@locales:1] Set(SIP/5002-0010, STATUS=) in new 
 stack
 -- Executing [0@locales:2] GotoIf(SIP/5002-0010,
 0?disponible:nodisponible) in new stack
 -- Goto (locales,0,6)
 -- Executing [0@locales:6] JabberSend(SIP/5002-0010,
 ejabberd,operad...@xmpp.domain.com,Llamada perdida de5002) in new
 stack

 --- XMPP sent to 'ejabberd' ---
 message type='chat' to='operad...@xmpp.domain.com'
 from='aster...@xmpp.domain.com/asterisk-xmpp'bodyquot;Llamada
 perdida de5002quot;/body/message
 -
 -- Executing [0@locales:7] Hangup(SIP/5002-0010, ) in new stack
   == Spawn extension (locales, 0, 7) exited non-zero on 'SIP/5002-0010'

 --- XMPP received from 'operadora' ---
 message from='aster...@xmpp.domain.com/asterisk-xmpp'
 to='operad...@xmpp.domain.com' type='chat'bodyquot;Llamada perdida
 de5002quot;/body/message
 -
 -- Executing [s@messages1:1] NoOp(Message/ast_msg_queue,
 Mensaje hacia usuarios XMPP) in new stack
 -- Executing [s@messages1:2] JabberSend(Message/ast_msg_queue,
 ejabberd,allan@172.16.8.59,Llamada perdida de5002) in new stack

 --- XMPP sent to 'ejabberd' ---
 message type='chat' to='allan@172.16.8.59'
 from='aster...@xmpp.domain.com/asterisk-xmpp'bodyquot;Llamada
 perdida de5002quot;/body/message
 -
 -- Executing [s@messages1:3] NoOp(Message/ast_msg_queue, Estado
 del mensaje ) in new stack
 -- Executing [s@messages1:4] Hangup(Message/ast_msg_queue, )
 in new stack
   == Spawn extension (messages1, s, 4) exited non-zero on
 'Message/ast_msg_queue'


 Do you have a buddy operadora@172.16.8.59 with a resource of alcides?
 Based on the provided output, it does not appear as if you have that
 buddy/resource combination, in which case the result of 7 is what I
 would expect.

 I have put it in both

 Client: alcides
 Buddy:  ce...@xmpp.domain.com
 Resource: asterisk-xmpp
 node: http://www.asterisk.org/xmpp/client/caps
 version: asterisk-xmpp
 Google Talk capable: no
 Jingle capable: yes
 Buddy:  aster...@xmpp.domain.com
 Resource: asterisk-xmpp
 node: http://www.asterisk.org/xmpp/client/caps
 version: asterisk-xmpp
 Google Talk capable: no
 Jingle capable: yes
 Buddy:  operad...@xmpp.domain.com
 Resource: 36500272461413222444766262
 node: http://pidgin.im/
 version: I22W7CegORwdbnu0ZiQwGpxr0Go=
 Google Talk capable: no
 Jingle capable: yes
 Resource: asterisk-xmpp
 node: http://www.asterisk.org/xmpp/client/caps
 version: asterisk-xmpp
 Google Talk capable: no
 Jingle capable: yes

 Client: operadora
 Buddy:  ce...@xmpp.domain.com
 Resource: asterisk-xmpp
 node: http://www.asterisk.org/xmpp/client/caps
 version: asterisk-xmpp
 Google Talk capable: no
 Jingle capable: yes
 Buddy:  operad...@xmpp.domain.com
 Resource: asterisk-xmpp
 node: http://www.asterisk.org/xmpp/client/caps
 version: asterisk-xmpp
 Google Talk capable: no
 Jingle capable: yes
 Resource: 36500272461413222444766262
 node: http://pidgin.im/
 version: I22W7CegORwdbnu0ZiQwGpxr0Go=
 Google Talk capable: no
 Jingle capable: yes
 Buddy:  ejabb...@xmpp.domain.com
 Buddy:  alci...@xmpp.domain.com
 Resource: asterisk-xmpp
 node: http://www.asterisk.org/xmpp/client/caps
 version: asterisk-xmpp
 Google Talk capable: no

Re: [asterisk-users] JABBER_STATUS CODE 7

[Marcelo Terres]

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...@jabber.mundoopensource.com.br
http://www.mundoopensource.com.br
http://offtopicsandfun.blogspot.com
http://biertasters.blogspot.com
http://twitter.com/mhterres


On Thu, Oct 9, 2014 at 7:28 PM, ricky gutierrez xserverli...@gmail.com wrote:
 anyone here?

 2014-10-01 8:09 GMT-06:00 ricky gutierrez xserverli...@gmail.com:
 Hi all,I hope to find a solution with the help of the list, I'm trying
 to get the status of my extensions with ejabberd , the idea is to
 visualize my users ejabberd incoming calls or missed.

 I'm testing with my operator extension with this code but only get the
 missed call notification does not show me where the call is coming.

 my piece of code

 [operadora]
 exten = 
 0,1,Set(STATUS=${JABBER_STATUS(ejabberd,operadora@172.16.8.59/alcides)})
 same= n, GotoIf($[0${STATUS} = 1]?disponible:nodisponible)
 same= n(disponible),
 JabberSend(ejabberd,operadora@172.16.8.59,Llamada Entrante
 ${CALLERID(num)})
 same= n,Dial(SIP/5001)
 same= n,Hangup()
 same= n(nodisponible),
 JabberSend(ejabberd,operadora@172.16.8.59,Llamada perdida de
 ${CALLERID(num)}
 )
 same= n,Hangup()



 look the log

 Oct  1 08:04:10] NOTICE[4789][C-0028]: res_xmpp.c:1631
 acf_jabberstatus_read: Resource alcides of buddy operadora@172.16.8.59
 was not found.
 -- Executing [0@locales:1] Set(SIP/5002-0029, STATUS=7) in new 
 stack
 -- Executing [0@locales:2] GotoIf(SIP/5002-0029,
 0?disponible:nodisponible) in new stack
 -- Goto (locales,0,6)
 -- Executing [0@locales:6] JabberSend(SIP/5002-0029,
 ejabberd,operadora@172.16.8.59,Llamada perdida de 5002) in new
 stack

 [Oct  1 08:04:34] WARNING[13482][C-0005]: pbx.c:6646
 __ast_pbx_run: Channel 'Message/ast_msg_queue' sent to invalid
 extension but no invalid handler: context,exten,priority=default,s,1

 not work for me, and I think this should work asterisk receiving presence 
 status

 --- XMPP received from 'operadora' ---
 presence from='operadora@172.16.8.59/12233853371412171752845116'
 to='operadora@172.16.8.59/asterisk-xmpp'showchat/showpriority1/priorityc
 xmlns='http://jabber.org/protocol/caps' node='http://pidgin.im/'
 hash='sha-1' ver='I22W7CegORwdbnu0ZiQwGpxr0Go='/x
 xmlns='vcard-temp:x:update'photo//x/presence
 -

 any idea?

 regardss


 --
 rickygm

 http://gnuforever.homelinux.com



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 http://gnuforever.homelinux.com

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

[Marcelo Terres]

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
http://offtopicsandfun.blogspot.com
http://biertasters.blogspot.com
http://twitter.com/mhterres

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

[Marcelo Terres]

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://twitter.com/mhterres

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

[Marcelo Terres]

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://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 list
 IM: marc...@jabber.mundoopensource.com.br
 http://www.mundoopensource.com.br
 http://offtopicsandfun.blogspot.com
 http://biertasters.blogspot.com
 http://twitter.com/mhterres

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