Re: [asterisk-users] IAX2 getting stuck

[Kseniya Blashchuk]

Hmmm.. So if you are sure that the poke packets leave the network interface
(I would still check with tcpdump as well, maybe a firewall issue?) then it
makes sense to check the other side to make sure the poke packets reach
other servers.
I mean with tcpdump you may see if there are incoming packets from your
peers on the interface. If there are, then they are dropped or ignored by
your servers. If no, then it's better to check the other side.
you may try smth like 'tcpdump -npi  host  and port 4569'
Do you have a firewall configured on this server?

On Fri, Apr 21, 2017, 12:36 AM Carlos Chavez  wrote:

> On 4/20/17 2:37 PM, Kseniya Blashchuk wrote:
>
> If SIP goes to the same provider then yes. Still I would check a packet
> capture for better understanding. BTW, did you try iax debug?
>
> чт, 20 апр. 2017 г. в 19:46, Carlos Chavez :
>
>> On 4/20/17 12:45 AM, Kseniya Blashchuk wrote:
>>
>> Can it happen that the routes lead the traffic through another interface?
>> Did you try a packet capture with tcpdump? Do the packets really leave the
>> usb adapter? Can asymmetric routing be in effect?
>> Maybe there were some static routes that disappeared when the adapter was
>> unplugged...
>>
>> On Thu, Apr 20, 2017, 12:41 AM Antony Stone <
>> antony.st...@asterisk.open.source.it> wrote:
>>
>>> On Wednesday 19 April 2017 at 23:35:24, Carlos Chavez wrote:
>>>
>>> > On 4/19/17 4:23 PM, Antony Stone wrote:
>>> > >
>>> > > You say the USB ethernet adapter got unplugged and then
>>> reconnected...
>>> > >
>>> > > 1. What's the name of the network device for this adapter?  Is it the
>>> > > same name as it previously had?
>>> > >
>>> > > 2. What does 'ifconfig' say the IP address is for this adapter?
>>> > >
>>> > > 3. What do you have in /etc/asterisk/iax.conf for 'bindaddr' and
>>> > > 'bindport'?
>>> > >
>>> > > 4. Do you have SIP connections on the same network interface, and are
>>> > > those working as normal?
>>> > >
>>> > >
>>> > > Antony.
>>> >
>>> > 1- No changes to device names.  eth0 is the main link to the network,
>>> > eth1 (also internal) goes to a SIP provider and eth2 (the USB adapter)
>>> > goes to another SIP provider.  All IAX trunks use eth0
>>> >
>>> > 2- ifconfig gives the proper IP and netmask for all interfaces
>>> >
>>> > 3- We do not specify bindaddr or bindport in the config file as the
>>> > default is to bind to 0.0.0.0
>>> >
>>> > 4- We had to make new SIP trunks to replace the IAX2 trunks to all
>>> > servers.  The SIP trunk is working with no problems.  Except for two
>>> SIP
>>> > links to PSTN all internal extensions use the same network interface.
>>>
>>> Ugh :(
>>>
>>> Sorry, I have no more ideas, then.
>>>
>>> I hope someone else comes into this thread with a helpful suggestion.
>>>
>>>
>>> If routing was the problem then the SIP trunk would not work.
>> Usually IAX2 is a little more forgiving about routing than SIP.
>>
>> The new SIP trunks are replacing the IAX2 trunks to our other
> Asterisk servers and use exactly the same network paths, that is why I know
> it is not a network infrastructure issue.  We did turn on IAX debug and we
> only se the server trying to poke the other servers but there is not
> response or any incoming traffic.
>
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Re: [asterisk-users] log incoming calls without answering

[Brandon B.]

On 2017-04-20 02:33 PM, Fabio Moretti wrote:

Yes, I'll definitely do the test before set up the whole proyect, but
the point basically is: it is possibile for asterisk to log a call
without answering it? How to do it in the dialplan? Or I'm wasting time
because an analog line who enter asterisk is always answered?


Have you considered the legal implications of what you are trying to 
accomplish? You are describing a pen register 
. Asterisk is not suitable 
for this purpose, even if it technically can be used in a fashion. You 
might want look into high impedance telephone line equipment which solve 
the technical issues.


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Re: [asterisk-users] log incoming calls without answering

[Pete Mundy]

> On 21/04/2017, at 9:33 am, Fabio Moretti  wrote:
> 
> the point basically is: it is possibile for asterisk to log a call
> without answering it? How to do it in the dialplan? Or I'm wasting time
> because an analog line who enter asterisk is always answered?

Yes.

Something similar to this will work (untested, possibly with syntax errors, but 
gives the idea):

exten => s,1,NoOp(Inbound call from callerID $CALLERID(num))
exten => s,n,Hangup

The 'Hangup' will simply tell the FXS box (which you've used to terminate the 
line) that Asterisk isn't accepting the call.

One problem could be the FXS box re-inviting Asterisk over and over (causing 
duplicate NoOp logs), but you could add a Wait in there for 2-3 minutes before 
'Hangup'. Not many calls would ring for more than 2-3 minutes. In fact the PSTN 
may limit the number of rings (I have a feeling here in NZ it's about 250 rings 
or so).

exten => s,1,NoOp(Inbound call from callerID $CALLERID(num))
exten => s,n,Wait(180)
exten => s,n,Hangup

Pete



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Re: [asterisk-users] log incoming calls without answering

[Pete Mundy]

Fabio, this doesn't answer your question directly and it's not Asterisk related 
in any way, but it's another way to engineer a solution to the problem and I've 
seen it done before.

Many analog modems will decode the caller ID on the analog line and provide it 
as part of the 'RING' notification on their serial port. This can be used as 
input on a script on a computer to log the caller ID and datestamp.

For the Asterisk solution you propose to be able to work, the analog line has 
to support caller ID. If it does that then it will work with any of these 
modems too.

It's a purely passive connection to the line (just bridge it over the pair) and 
it may be cheaper/simpler/easier for you to build.

Food for thought?

Pete



> On 21/04/2017, at 8:26 am, Fabio Moretti  wrote:
> 
> Hi,
> 
> I've some analogic lines and I'm asked if it's possible to program an 
> asterisk for "checking" the inbound calls without answering them, doing 
> something like this:
> 
> analog line 1 -+-- asterisk
>   |
>   \__ analog phone
> 
> when a call enter, asterisk sense it and store its values (callerid, date and 
> time, etc) somewhere, but nothing more, people will answer using the old 
> analog phone.
> The goal is to have a log of the inbound calls without touching the old 
> analog system (it's shared between different subjects).
> 
> I'm pretty sure it's something possible, but how to tell asterisk: "ok, call 
> this AGI, and then don't answer and do nothing more".
> 
> Any idea?
> 
> Thanks


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Re: [asterisk-users] log incoming calls without answering

[James Cloos]

> "FM" == Fabio Moretti  writes:

FM> when a call enter, asterisk sense it and store its values (callerid,
FM> date and time, etc) somewhere, but nothing more, people will answer
FM> using the old analog phone.  The goal is to have a log of the inbound
FM> calls without touching the old analog system (it's shared between
FM> different subjects).

IIUC, the pots line has both some number of analog phones a/o fax
machines on it, plus a fxo->sip gateway, yes?

You can route the sip portion to asterisk and have the dialplan log
everything but never answer.

You may want to call the Ringing dialplan application, but even that
may not be required.  OTOH, calling Ringing should prevent the gateway
from assuming that the asterisk machine never saw the INVITE.

Eventually, when the other extension answers, the fxo->sip gateway will
cancel the sip call just like it would if the caller hung up.

(There is a possibility that any given gateway may not cancel the sip
call until the analog call is completed; you need to test.)

-JimC
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Re: [asterisk-users] log incoming calls without answering

[John Novack_PIV2]

Fabio Moretti wrote:

Il 20/04/2017 18:09, kevin.lar...@pioneerballoon.com ha scritto:

I honestly don't know if you can do what you want without some piece
of equipment picking up the line. What I would do is get an analog
line, an analog phone, an analog to sip device (there are many to
choose from) and a basic asterisk instance. I would then make a small
test setup where the analog line goes to a splitter. One side of the
splitter goes to your analog phone. One side goes to your analog to
SIP converter and then into your asterisk instance via your ethernet
network. Use your cell phone to call the number of your analog line
and see if it works. You would have to code a basic dialplan on the
asterisk side and set up the trunk to your converter, which I am
assuming you know how to do.

Yes, I'll definitely do the test before set up the whole proyect, but
the point basically is: it is possibile for asterisk to log a call
without answering it? How to do it in the dialplan? Or I'm wasting time
because an analog line who enter asterisk is always answered?


I used to do something similar, though not for 60 lines. I had ports off a 
channel bank bridged to a propitiatory ( think Vonage ) device, or even a 
copper PSTN line.
Not sure where you are in the world, but different analog lines deliver caller 
id differently, so timing can become an issue, and you will have no clue when 
the call is either answered or abandoned.

Here in the US it is between the first and second ring
You would need to work through the timing, as if the line rings several times 
you need to detect the ring, capture the CLID, then ignore any additional rings 
for x period of time as there will only be one CLID string sent per call.


Once I converted all lines to VOIP it all became much easier.
If your analog lines are NOT in the US or Canada, you may run into different 
issues, or it might even become easier.


John Novack

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Re: [asterisk-users] IAX2 getting stuck

[Carlos Chavez]

On 4/20/17 2:37 PM, Kseniya Blashchuk wrote:

If SIP goes to the same provider then yes. Still I would check a 
packet capture for better understanding. BTW, did you try iax debug?


чт, 20 апр. 2017 г. в 19:46, Carlos Chavez >:


On 4/20/17 12:45 AM, Kseniya Blashchuk wrote:


Can it happen that the routes lead the traffic through another
interface? Did you try a packet capture with tcpdump? Do the
packets really leave the usb adapter? Can asymmetric routing be
in effect?
Maybe there were some static routes that disappeared when the
adapter was unplugged...


On Thu, Apr 20, 2017, 12:41 AM Antony Stone
> wrote:

On Wednesday 19 April 2017 at 23:35:24, Carlos Chavez wrote:

> On 4/19/17 4:23 PM, Antony Stone wrote:
> >
> > You say the USB ethernet adapter got unplugged and then
reconnected...
> >
> > 1. What's the name of the network device for this
adapter?  Is it the
> > same name as it previously had?
> >
> > 2. What does 'ifconfig' say the IP address is for this
adapter?
> >
> > 3. What do you have in /etc/asterisk/iax.conf for
'bindaddr' and
> > 'bindport'?
> >
> > 4. Do you have SIP connections on the same network
interface, and are
> > those working as normal?
> >
> >
> > Antony.
>
> 1- No changes to device names.  eth0 is the main link to
the network,
> eth1 (also internal) goes to a SIP provider and eth2 (the
USB adapter)
> goes to another SIP provider.  All IAX trunks use eth0
>
> 2- ifconfig gives the proper IP and netmask for all interfaces
>
> 3- We do not specify bindaddr or bindport in the config
file as the
> default is to bind to 0.0.0.0
>
> 4- We had to make new SIP trunks to replace the IAX2 trunks
to all
> servers.  The SIP trunk is working with no problems. 
Except for two SIP

> links to PSTN all internal extensions use the same network
interface.

Ugh :(

Sorry, I have no more ideas, then.

I hope someone else comes into this thread with a helpful
suggestion.


If routing was the problem then the SIP trunk would not work. 
Usually IAX2 is a little more forgiving about routing than SIP.


The new SIP trunks are replacing the IAX2 trunks to our other 
Asterisk servers and use exactly the same network paths, that is why I 
know it is not a network infrastructure issue.  We did turn on IAX debug 
and we only se the server trying to poke the other servers but there is 
not response or any incoming traffic.


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Re: [asterisk-users] log incoming calls without answering

[Fabio Moretti]

Il 20/04/2017 18:09, kevin.lar...@pioneerballoon.com ha scritto:
>
> I honestly don't know if you can do what you want without some piece
> of equipment picking up the line. What I would do is get an analog
> line, an analog phone, an analog to sip device (there are many to
> choose from) and a basic asterisk instance. I would then make a small
> test setup where the analog line goes to a splitter. One side of the
> splitter goes to your analog phone. One side goes to your analog to
> SIP converter and then into your asterisk instance via your ethernet
> network. Use your cell phone to call the number of your analog line
> and see if it works. You would have to code a basic dialplan on the
> asterisk side and set up the trunk to your converter, which I am
> assuming you know how to do.

Yes, I'll definitely do the test before set up the whole proyect, but
the point basically is: it is possibile for asterisk to log a call
without answering it? How to do it in the dialplan? Or I'm wasting time
because an analog line who enter asterisk is always answered?







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Re: [asterisk-users] log incoming calls without answering

[kevin . larsen]

> I've already proposed your solution (is the most reasonable) but they
> have more than 60 analogs lines (no faxes) and some of them terminate in
> appliances like alarms, etc, so the solution must not touch in any way
> the connection between the line and his termination: doing a analog to
> digital conversion, passing it to asterisk and the convert it back to
> analog is prone to problems (what if asterisk crashes? or if a gateway
> fail?).
> I can split the existing lines (there are no complex things like adsl or
> digital signaling), convert the branches to digital and terminate then
> into an asterisk machine, so any failure will not affect the old
> circuit, but of course I've to configure asterisk to ONLY LOG calls and
> nothing more.
> 
> This is what they want:
> - line 1 ring
> - line 1 is splitted in two, the first branch (let's say the "analog"
> branch) go to an analog phone, that rings
> - the second branch go through a gateway and then to asterisk
> - asterisk log (with an AGI for example) "line 1 rings at  from 
"
> no more is required from asterisk, if someone answer the analog phone or
> not is not my business.
> 
Ok, so I would agree with them that a conversion to digital and back again 
would tend to break things like fax lines and alarm lines. My analog lines 
in my facilities are there because a lot of alarm systems just don't work 
with SIP at all. It's something the alarm companies are going to have to 
figure out in the next decade or so as the Telcos are moving away from 
copper and switched networks and towards fiber and packet based networks.

I honestly don't know if you can do what you want without some piece of 
equipment picking up the line. What I would do is get an analog line, an 
analog phone, an analog to sip device (there are many to choose from) and 
a basic asterisk instance. I would then make a small test setup where the 
analog line goes to a splitter. One side of the splitter goes to your 
analog phone. One side goes to your analog to SIP converter and then into 
your asterisk instance via your ethernet network. Use your cell phone to 
call the number of your analog line and see if it works. You would have to 
code a basic dialplan on the asterisk side and set up the trunk to your 
converter, which I am assuming you know how to do.

This would at least give you a fairly low cost way to test to see if you 
can trigger what you want on the Asterisk side without also triggering the 
line itself to be answered. I would also note that you would only be able 
to log incoming calls this way. I can't see a way you would be able to 
detect an outgoing call from the analog extension.

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Re: [asterisk-users] log incoming calls without answering

[Fabio Moretti]

Il 20/04/2017 17:32, kevin.lar...@pioneerballoon.com ha scritto:
>
> This gets kinda Rube Golberg-ish, but convert the incoming analog line
> to sip, route it through asterisk and have asterisk do its thing
> before converting it back to analog to send to the phone. Only problem
> is you get a lot of extra hardware involved in the mix to make it
> work. It will be a lot of expense and trouble, so you need to make
> sure that whatever part you want asterisk to play is worth that
> effort. Also, I wouldn't touch a fax line in this manner.
>
> If you could give a bit more info on what you want asterisk to do, we
> could maybe give better advice on how to solve your problem.

Hi Kevin,

I've already proposed your solution (is the most reasonable) but they
have more than 60 analogs lines (no faxes) and some of them terminate in
appliances like alarms, etc, so the solution must not touch in any way
the connection between the line and his termination: doing a analog to
digital conversion, passing it to asterisk and the convert it back to
analog is prone to problems (what if asterisk crashes? or if a gateway
fail?).
I can split the existing lines (there are no complex things like adsl or
digital signaling), convert the branches to digital and terminate then
into an asterisk machine, so any failure will not affect the old
circuit, but of course I've to configure asterisk to ONLY LOG calls and
nothing more.

This is what they want:
- line 1 ring
- line 1 is splitted in two, the first branch (let's say the "analog"
branch) go to an analog phone, that rings
- the second branch go through a gateway and then to asterisk
- asterisk log (with an AGI for example) "line 1 rings at  from "
no more is required from asterisk, if someone answer the analog phone or
not is not my business.







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Re: [asterisk-users] Voicemail asking for login

[James Cloos]

> "DC" == D'Arcy Cain  writes:

DC> I did debug 10 and saved the console output into files which I
DC> compared side by side.  No material difference.

In that case I'd add more debug statements to apps/app_voicemail.c (in
vm_exec()), including a log at the start of what is in *data and args.

Looking at it, it only plays vm-whichbox when ast_strlen_zero(data),
which implies that the args to Voicemail are not making it through.

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Re: [asterisk-users] log incoming calls without answering

[kevin . larsen]

> From: Fabio Moretti 
> To: Asterisk Users Mailing List - Non-Commercial Discussion 
> 
> Date: 04/20/2017 03:26 PM
> Subject: [asterisk-users] log incoming calls without answering
> Sent by: asterisk-users-boun...@lists.digium.com
> 
> Hi,
> 
> I've some analogic lines and I'm asked if it's possible to program 
> an asterisk for "checking" the inbound calls without answering them,
> doing something like this:
> 
> analog line 1 -+-- asterisk
>|
>\__ analog phone
> 
> when a call enter, asterisk sense it and store its values (callerid,
> date and time, etc) somewhere, but nothing more, people will answer 
> using the old analog phone.
> The goal is to have a log of the inbound calls without touching the 
> old analog system (it's shared between different subjects).
> 
> I'm pretty sure it's something possible, but how to tell asterisk: 
> "ok, call this AGI, and then don't answer and do nothing more".
> 
> Any idea?
> 
> Thanks

This gets kinda Rube Golberg-ish, but convert the incoming analog line to 
sip, route it through asterisk and have asterisk do its thing before 
converting it back to analog to send to the phone. Only problem is you get 
a lot of extra hardware involved in the mix to make it work. It will be a 
lot of expense and trouble, so you need to make sure that whatever part 
you want asterisk to play is worth that effort. Also, I wouldn't touch a 
fax line in this manner.

If you could give a bit more info on what you want asterisk to do, we 
could maybe give better advice on how to solve your problem.

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[asterisk-users] log incoming calls without answering

[Fabio Moretti]

Hi,

I've some analogic lines and I'm asked if it's possible to program an asterisk 
for "checking" the inbound calls without answering them, doing something like 
this:

analog line 1 -+-- asterisk
   |
   \__ analog phone

when a call enter, asterisk sense it and store its values (callerid, date and 
time, etc) somewhere, but nothing more, people will answer using the old analog 
phone.
The goal is to have a log of the inbound calls without touching the old analog 
system (it's shared between different subjects).

I'm pretty sure it's something possible, but how to tell asterisk: "ok, call 
this AGI, and then don't answer and do nothing more".

Any idea?

Thanks






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Re: [asterisk-users] Voicemail asking for login

[D'Arcy Cain]

On 2017-04-20 04:07 PM, James Cloos wrote:

I enable full log and run 'core set debug 9' before doing a pair of
tests.

(The full log is easier to grep than the console output.)

Then compare a working vs stocktrans2 side by side.


I did debug 10 and saved the console output into files which I compared 
side by side.  No material difference.


--
D'Arcy J.M. Cain
Vybe Networks Inc.
http://www.VybeNetworks.com/
IM:da...@vex.net VoIP: sip:da...@vybenetworks.com

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Re: [asterisk-users] Voicemail asking for login

[James Cloos]

I enable full log and run 'core set debug 9' before doing a pair of
tests.

(The full log is easier to grep than the console output.)

Then compare a working vs stocktrans2 side by side.

-JimC
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Re: [asterisk-users] "Your call is not allowed. P U A M I"

[BJ Weschke]

Google search reveals a fairly dated reference to the same carrier switch
tag message being delivered in a Skype for Business forum thread.

https://social.microsoft.com/Forums/en-US/e25b3198-b5a0-4a43-9328-4a1aff5f6ed0/1800-number-dialing-issue?forum=communicationsservertelephony

On Thu, Apr 20, 2017 at 3:43 PM, Antony Stone <
antony.st...@asterisk.open.source.it> wrote:

> On Thursday 20 April 2017 at 21:29:58, Steve Edwards wrote:
>
> > Not an Asterisk question, but...
> >
> > A bunch of our 8xx numbers started playing this recording when dialed.
> Our
> > provider (Inteliquent) says it's not them.
>
> Where are Inteliquent feeding the calls (assuming they connect instead of
> playing that message) to?
>
> Are they a SIP trunk provider, supposedly passing calls to your PBX (in
> which
> case it's either them or your PBX, so there shouldn't be a lot of
> discussion)?
>
> Does Inteliquent have any record of the calls being placed IN to the 8xx
> numbers (if they do, this eliminates the possibility of message being
> played
> by the callER's service provider)?
>
> Does it make any difference which carrier you use to make the call?
>
> > Does anybody know who is playing it and what it means?
>
> I've certainly never heard (of) it.
>
>
> Antony.
>
> --
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>
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> list;
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> me.
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Re: [asterisk-users] "Your call is not allowed. P U A M I"

[Antony Stone]

On Thursday 20 April 2017 at 21:29:58, Steve Edwards wrote:

> Not an Asterisk question, but...
> 
> A bunch of our 8xx numbers started playing this recording when dialed. Our
> provider (Inteliquent) says it's not them.

Where are Inteliquent feeding the calls (assuming they connect instead of 
playing that message) to?

Are they a SIP trunk provider, supposedly passing calls to your PBX (in which 
case it's either them or your PBX, so there shouldn't be a lot of discussion)?

Does Inteliquent have any record of the calls being placed IN to the 8xx 
numbers (if they do, this eliminates the possibility of message being played 
by the callER's service provider)?

Does it make any difference which carrier you use to make the call?

> Does anybody know who is playing it and what it means?

I've certainly never heard (of) it.


Antony.

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Re: [asterisk-users] IAX2 getting stuck

[Kseniya Blashchuk]

If SIP goes to the same provider then yes. Still I would check a packet
capture for better understanding. BTW, did you try iax debug?

чт, 20 апр. 2017 г. в 19:46, Carlos Chavez :

> On 4/20/17 12:45 AM, Kseniya Blashchuk wrote:
>
> Can it happen that the routes lead the traffic through another interface?
> Did you try a packet capture with tcpdump? Do the packets really leave the
> usb adapter? Can asymmetric routing be in effect?
> Maybe there were some static routes that disappeared when the adapter was
> unplugged...
>
> On Thu, Apr 20, 2017, 12:41 AM Antony Stone <
> antony.st...@asterisk.open.source.it> wrote:
>
>> On Wednesday 19 April 2017 at 23:35:24, Carlos Chavez wrote:
>>
>> > On 4/19/17 4:23 PM, Antony Stone wrote:
>> > >
>> > > You say the USB ethernet adapter got unplugged and then reconnected...
>> > >
>> > > 1. What's the name of the network device for this adapter?  Is it the
>> > > same name as it previously had?
>> > >
>> > > 2. What does 'ifconfig' say the IP address is for this adapter?
>> > >
>> > > 3. What do you have in /etc/asterisk/iax.conf for 'bindaddr' and
>> > > 'bindport'?
>> > >
>> > > 4. Do you have SIP connections on the same network interface, and are
>> > > those working as normal?
>> > >
>> > >
>> > > Antony.
>> >
>> > 1- No changes to device names.  eth0 is the main link to the network,
>> > eth1 (also internal) goes to a SIP provider and eth2 (the USB adapter)
>> > goes to another SIP provider.  All IAX trunks use eth0
>> >
>> > 2- ifconfig gives the proper IP and netmask for all interfaces
>> >
>> > 3- We do not specify bindaddr or bindport in the config file as the
>> > default is to bind to 0.0.0.0
>> >
>> > 4- We had to make new SIP trunks to replace the IAX2 trunks to all
>> > servers.  The SIP trunk is working with no problems.  Except for two SIP
>> > links to PSTN all internal extensions use the same network interface.
>>
>> Ugh :(
>>
>> Sorry, I have no more ideas, then.
>>
>> I hope someone else comes into this thread with a helpful suggestion.
>>
>>
>> If routing was the problem then the SIP trunk would not work.
> Usually IAX2 is a little more forgiving about routing than SIP.
>
> --
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> Carlos Chávez+52 (55)8116-9161 <+52%2055%208116%209161>
>
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[asterisk-users] "Your call is not allowed. P U A M I"

[Steve Edwards]

Not an Asterisk question, but...

A bunch of our 8xx numbers started playing this recording when dialed. Our 
provider (Inteliquent) says it's not them.


Does anybody know who is playing it and what it means?

--
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] Voicemail asking for login

[D'Arcy Cain]

On 2017-04-20 12:52 PM, J Montoya wrote:

On Thursday 20 Apr 2017, D'Arcy Cain wrote:

On 2017-04-20 12:23 PM, D'Arcy Cain wrote:

Here is the full dialplan for stocktrans2.


I reduced this to the following and I still have the error.

exten => stocktrans2,1,Verbose(0,Entering extension stocktrans2)
 same => n(VoiceMail),Set(CDR(userfield)=VoiceMail)
 same => n,Verbose(0,${CALLERID(all)} going into voice mail for
stocktrans2)
 same => n,VoiceMail(stocktrans2@VoiceMail,u)
 same => n,Hangup()


O.K., so -- assuming that extension "darcy" behaves properly if you simplify
it similarly -- nothing before there can be causing the problem.


Actually, it also failed too when I reduced it.  Turns out that that's 
why I needed to set "_ACCOUNT".  Here is the actual reduced version for 
stocktrans2.  It still fails but substituting "darcy" for "stocktrans2 
works OK.


exten => stocktrans2,1,Verbose(0,Entering extension stocktrans2)
same => n(VoiceMail),Set(CDR(userfield)=VoiceMail)
same => n,Verbose(0,${CALLERID(all)} going into voice mail for 
stocktrans2)

same => n,Set(_ACCOUNT=stocktrans2)
same => n,VoiceMail(stocktrans2@VoiceMail,u)
same => n,Hangup()


What is in your [VoiceMail] context?  Are "stocktrans2" and "darcy" separate
extensions, or is there a catch-all?  What is in the "a" extension  (which
gets called when the * key is pressed) ?


Nothing in [VoiceMail] context except the mailboxes but this precedes 
the context:


[general]
attach=yes
maxsilence=10
maxlogins=3
serveremail=n...@vex.net
format=wav49
fromstring=Vybe Networks Voice Mail
nextaftercmd=yes
forcename=yes
pollmailboxes=yes
pollfreq=5

emailbody=Dear ${VM_NAME}:\n\n\tjust wanted to let you know you were 
just left a ${VM_DUR} long message (number ${VM_MSGNUM})\nin mailbox 
${VM_MAILBOX} from ${VM_CALLERID}, on ${VM_DATE}, so you might\nwant to 
check it when you get a chance.  Thanks!\n\n\t\t\t\t-- Vybe Networks\n


They are separate extensions.  I do not do catch-alls if I can help it. 
Since I generate the configs it is no big deal to expand everything in 
each extension.


; voice mail
  exten => a,1,Verbose(${ACCOUNT} entering mailbox)
same => n,Set(CDR(userfield)=${ACCOUNT})
same => n,VoicemailMain(${ACCOUNT}@VoiceMail)
same => n,Hangup

The VoiceMail extension, as are all the extensions, is in the 
[LocalSets] context.


--
D'Arcy J.M. Cain
Vybe Networks Inc.
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IM:da...@vex.net VoIP: sip:da...@vybenetworks.com

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Re: [asterisk-users] Voicemail asking for login

[J Montoya]

On Thursday 20 Apr 2017, D'Arcy Cain wrote:
> On 2017-04-20 12:23 PM, D'Arcy Cain wrote:
> > Here is the full dialplan for stocktrans2.
> 
> I reduced this to the following and I still have the error.
> 
> exten => stocktrans2,1,Verbose(0,Entering extension stocktrans2)
>  same => n(VoiceMail),Set(CDR(userfield)=VoiceMail)
>  same => n,Verbose(0,${CALLERID(all)} going into voice mail for
> stocktrans2)
>  same => n,VoiceMail(stocktrans2@VoiceMail,u)
>  same => n,Hangup()

O.K., so -- assuming that extension "darcy" behaves properly if you simplify 
it similarly -- nothing before there can be causing the problem.

What is in your [VoiceMail] context?  Are "stocktrans2" and "darcy" separate 
extensions, or is there a catch-all?  What is in the "a" extension  (which 
gets called when the * key is pressed) ?

-- 
JM

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] asterisk as non root

[Antony Stone]

On Thursday 20 April 2017 at 18:31:03, Atux Atux wrote:

> root@PBX: /var/www/html $ /etc/init.d/asterisk start
> [ ok ] Starting asterisk (via systemctl): asterisk.service.

I'm somewhat puzzled that your root-user prompt is "$"
instead of the more normal "#", but never mind...

> root@PBX: /var/www/html $ ps aux | grep asterisk
> asterisk  1007  0.7  2.3  67128 23748 ?Ssl  Apr19   8:49 
> /usr/sbin/asterisk -U asterisk -G asterisk

So, the first column of that output shows you that asterisk is
running as the user "asterisk".

On my Debian system I only have "-U asterisk" without the "-G asterisk".

> root  4186  0.0  0.1   4192  1992 pts/0S+   17:30   0:00 grep asterisk

...and the grep command was run by "root"

> root@PBX: /var/www/html $ /usr/sbin/asterisk –rx "sip show peers"
> Privilege escalation protection disabled!
> See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details.
> Asterisk already running on /var/run/asterisk/asterisk.ctl.  Use 'asterisk
> -r' to connect.

Who does "ls -l" show you that file /var/run/asterisk/asterisk.ctl
is owned by?

On my machine it's:

srwxrwx--- 1 asterisk asterisk 0 Apr 11 10:32 /var/run/asterisk/asterisk.ctl


Antony.

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Re: [asterisk-users] IAX2 getting stuck

[Carlos Chavez]

On 4/20/17 12:45 AM, Kseniya Blashchuk wrote:

Can it happen that the routes lead the traffic through another 
interface? Did you try a packet capture with tcpdump? Do the packets 
really leave the usb adapter? Can asymmetric routing be in effect?
Maybe there were some static routes that disappeared when the adapter 
was unplugged...



On Thu, Apr 20, 2017, 12:41 AM Antony Stone 
> wrote:


On Wednesday 19 April 2017 at 23:35:24, Carlos Chavez wrote:

> On 4/19/17 4:23 PM, Antony Stone wrote:
> >
> > You say the USB ethernet adapter got unplugged and then
reconnected...
> >
> > 1. What's the name of the network device for this adapter?  Is
it the
> > same name as it previously had?
> >
> > 2. What does 'ifconfig' say the IP address is for this adapter?
> >
> > 3. What do you have in /etc/asterisk/iax.conf for 'bindaddr' and
> > 'bindport'?
> >
> > 4. Do you have SIP connections on the same network interface,
and are
> > those working as normal?
> >
> >
> > Antony.
>
> 1- No changes to device names.  eth0 is the main link to the
network,
> eth1 (also internal) goes to a SIP provider and eth2 (the USB
adapter)
> goes to another SIP provider.  All IAX trunks use eth0
>
> 2- ifconfig gives the proper IP and netmask for all interfaces
>
> 3- We do not specify bindaddr or bindport in the config file as the
> default is to bind to 0.0.0.0
>
> 4- We had to make new SIP trunks to replace the IAX2 trunks to all
> servers.  The SIP trunk is working with no problems. Except for
two SIP
> links to PSTN all internal extensions use the same network
interface.

Ugh :(

Sorry, I have no more ideas, then.

I hope someone else comes into this thread with a helpful suggestion.


If routing was the problem then the SIP trunk would not work. 
Usually IAX2 is a little more forgiving about routing than SIP.


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Carlos Chávez
+52 (55)8116-9161

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Re: [asterisk-users] asterisk as non root

[Atux Atux]

root@PBX: /var/www/html $ /etc/init.d/asterisk start
[ ok ] Starting asterisk (via systemctl): asterisk.service.
root@PBX: /var/www/html $ ps aux | grep asterisk
asterisk  1007  0.7  2.3  67128 23748 ?Ssl  Apr19   8:49
/usr/sbin/asterisk -U asterisk -G asterisk
root  4186  0.0  0.1   4192  1992 pts/0S+   17:30   0:00 grep
asterisk
root@PBX: /var/www/html $ /usr/sbin/asterisk –rx "sip show peers"
Privilege escalation protection disabled!
See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details.
Asterisk already running on /var/run/asterisk/asterisk.ctl.  Use 'asterisk
-r' to connect.
root@PBX: /var/www/html $



On Thu, Apr 20, 2017 at 1:36 PM, Antony Stone <
antony.st...@asterisk.open.source.it> wrote:

> On Thursday 20 April 2017 at 12:31:14, Atux Atux wrote:
>
> > Hi. thanks a lot for your replies. I did stop the services and i did
> issued
> > the  the "chown" and "chmod" commands listed in the guide.
> > It is necessary to compile it, instead if using the apt-get version
> > What am i missing?
>
> Let's go back to basics for a moment - you say this is a Debian system -
> in my
> experience Debian already runs Asterisk as the "asterisk" user and not as
> root, so let's see what you have.
>
> 1. Start Asterisk (probably using "/etc/init.d/asterisk start", or maybe
> "service asterisk start")
>
> 2. Check who it's running as: "ps aux | grep asterisk"
>
>
> Antony.
>
>
> --
> What makes you think I know what I'm talking about?
> I just have more O'Reilly books than most people.
>
>Please reply to the
> list;
>  please *don't* CC
> me.
>
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Re: [asterisk-users] Voicemail asking for login

[D'Arcy Cain]

On 2017-04-20 12:23 PM, D'Arcy Cain wrote:


Here is the full dialplan for stocktrans2.


I reduced this to the following and I still have the error.

exten => stocktrans2,1,Verbose(0,Entering extension stocktrans2)
same => n(VoiceMail),Set(CDR(userfield)=VoiceMail)
same => n,Verbose(0,${CALLERID(all)} going into voice mail for 
stocktrans2)

same => n,VoiceMail(stocktrans2@VoiceMail,u)
same => n,Hangup()

--
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Vybe Networks Inc.
http://www.VybeNetworks.com/
IM:da...@vex.net VoIP: sip:da...@vybenetworks.com

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Re: [asterisk-users] Voicemail asking for login

[D'Arcy Cain]

On 2017-04-20 05:14 AM, J Montoya or A J Stiles wrote:

This is just screaming "configuration mismatch" -- or, possibly, "latent bug
whereby things parsed in separate places should be treated the same, but are
actually getting treated differently".


I really don't want to be the "my system isn't working so there must be 
a bug in Asterisk" guy but I am certainly starting to suspect it.



I think we are going to need to see your dialplan logic, and maybe your
voicemail.conf, in order to work out what is different between this one user
and all the others.  You might even need to use `hd` to examine the files, just
in case there is a stray non-printing character spoiling things.


Non-printing characters show up in vi so I would have seen that by now.

Every extension is build by a script that takes information from the 
database and does substitutions which is what makes this so baffling. 
Every extension is built exactly the same way.  Here are two voicemail 
entries, the failing one and mine that works.  The only sanitation I did 
was for the password.


stocktrans2 => ,Angelica Douglas,stocktra...@vex.net
darcy => ,Vybe Networks - D'Arcy,da...@vex.net

Here is the full dialplan for stocktrans2.

exten => stocktrans2,1,Verbose(0,Entering extension stocktrans2)
same => n,Goto(DialCell)
same => n,GotoIf($["x" = "x"]?DialAlt)

same => n(DialAll),Verbose(0,${CALLERID(all)} Calling ${EXTEN} and ALL)
same => n,Dial(SIP/stocktrans2/thinktel//907084,30)
same => n,Goto(VoiceMail)

same => n(DialAlt),Verbose(0, ${CALLERID(all)} Calling ${EXTEN} and 
SoftPhone)

same => n,Dial(SIP/stocktrans2/907084,30)
same => n,Goto(VoiceMail)

same => n(DialCell),GotoIf($["x" = "x"]?DialDesk)
same => n,GotoIf($["${CALLERID(ani)}" = ""]?DialDesk)
same => n,Verbose(0,${CALLERID(all)} Calling "${EXTEN}" and cell "")
same => n,Dial(SIP/stocktrans2/thinktel/,30)
same => n,Goto(VoiceMail)

same => n(DialDesk),Verbose(0,${CALLERID(all)} Calling ${EXTEN})
same => n,Dial(SIP/stocktrans2,30)

same => n(VoiceMail),Set(CDR(userfield)=VoiceMail)
same => n,Verbose(0,${CALLERID(all)} going into voice mail for 
stocktrans2)

same => n,Set(_ACCOUNT=stocktrans2)
same => n,VoiceMail(stocktrans2@VoiceMail,u)
same => n,Hangup()

Here is mine.

exten => darcy,1,Verbose(0,Entering extension darcy)
same => n,GotoIf($["${DEVICE_STATE(SIP/901001)}" = 
"UNAVAILABLE"]?DialCell)

same => n,GotoIf($["x4168035991" = "x"]?DialAlt)

same => n(DialAll),Verbose(0,${CALLERID(all)} Calling ${EXTEN} and ALL)
same => n,Dial(SIP/darcy/thinktel/4168035991/901001,30)
same => n,Goto(VoiceMail)

same => n(DialAlt),Verbose(0, ${CALLERID(all)} Calling ${EXTEN} and 
SoftPhone)

same => n,Dial(SIP/darcy/901001,30)
same => n,Goto(VoiceMail)

same => n(DialCell),GotoIf($["x4168035991" = "x"]?DialDesk)
same => n,GotoIf($["${CALLERID(ani)}" = "4168035991"]?DialDesk)
same => n,Verbose(0,${CALLERID(all)} Calling "${EXTEN}" and cell 
"4168035991")

same => n,Dial(SIP/darcy/thinktel/4168035991,30)
same => n,Goto(VoiceMail)

same => n(DialDesk),Verbose(0,${CALLERID(all)} Calling ${EXTEN})
same => n,Dial(SIP/darcy,30)

same => n(VoiceMail),Set(CDR(userfield)=VoiceMail)
same => n,Verbose(0,${CALLERID(all)} going into voice mail for darcy)
same => n,Set(_ACCOUNT=darcy)
same => n,VoiceMail(darcy@VoiceMail,u)
same => n,Hangup()

There are some minor differences based on whether they have set up an 
alternate phone but either way it gets to the line that set the CDR 
userfield.  They both ultimately hit one or the other of these lines.


same => n,VoiceMail(stocktrans2@VoiceMail,u)
same => n,VoiceMail(darcy@VoiceMail,u)

The only "Set" command that might change the environment is the setting 
of "_ACCOUNT" which they both do.  Not sure why I even do that.  Perhaps 
I was planning some other feature that I never finished.  Or does 
VoiceMail() use it?


Someone (in private email for some reason so I won't give his name) 
suggested DumpChan() but I can't seem to make that work, even if I load 
app_dumpchan.


--
D'Arcy J.M. Cain
Vybe Networks Inc.
http://www.VybeNetworks.com/
IM:da...@vex.net VoIP: sip:da...@vybenetworks.com

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

[Pierre Couderc]

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 
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  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 
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  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 
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

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Re: [asterisk-users] Backport of Stretch's asterisk.service file into Jessie: successful start not detected by systemd [SOLVED]

[Olivier]

Basically, adding libsystemd-dev on Jessie before recompiling (./configure,
make, ...) allowed Asterisk to notify systemd it has successufully started.

For reference, please note this feature requires Asterisk 13.12.0 and above.

Thank you very much, Tzafrir, for your help !
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Re: [asterisk-users] Asterisk 1.8.32.3 : no video (h.264)

[Jonas Kellens]

Hello

in sip.conf I have ;

videosupport=yes




Kind regards.

J.


On 20-04-17 13:09, Marcelo Terres wrote:

I suppose that you enable the video support on sip.conf, right?

Regards,
Marcelo H. Terres 
IM: mhter...@jabber.mundoopensource.com.br
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 << 47) (0x8000)  audiotestlaw
(G.711 

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

[Pierre Couderc]

Thank you for this list, which helps me to be sure that the "good" 
packages are installed !


I have checked that all these packages are installed now.

Now,  I shall restart the install from the beginning and check again.

Thank you.

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 
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  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 
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  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 
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

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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 
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  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 
>> 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  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 
 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
>>>
>>>
>>>
>>> --
>>> _
>>> -- 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

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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] Backport of Stretch's asterisk.service file into Jessie: successful start not detected by systemd

[Tzafrir Cohen]

On Thu, Apr 20, 2017 at 12:02:36PM +0200, Olivier wrote:
> Hello,
> 
> I've been tasked to enable automatic Asterisk restart on failure on a
> Jessie platform (running latest Asterisk 13.15.0).
> 
> I build a dedicated Jessie VM on which I installed Asterisk from source.
> I configured a couple of files in /etc/asterisk directory.
> I positively checedk that with simple config, Asterisk could sucessively
> start using an /etc/init.d/asterisk file and sysv-systemd compatibility
> tools.
> 
> Then I copied a /etc/systemd/system/asterisk.service file with the
> following content:

[snip]

> [Service]
> Type=notify

[snip]

> After creating above asterisk.service file, running "systemctl
> daemon-reload" and "systemctl start asterisk", I could observe:
> - asterisk is starting OK,
> - asterisk prints to its log file a line such as:
> [Apr 20 11:23:22] VERBOSE[770] asterisk.c: Asterisk Ready.
> 
> but, at the same time, I got:
> april 20 11:23:22 jessievm asterisk[770]: [Apr 20 11:23:22] NOTICE[770]:
> app_queue.c:9095 reload_queues: No call queueing config fil
> april 20 11:24:48 jessievm systemd[1]: asterisk.service start operation
> timed out. Terminating.
> april 20 11:24:48 jessievm asterisk[770]:
> april 20 11:24:48 jessievm systemd[1]: Failed to start Asterisk PBX.

'Type=notify' means that the service is only considered ready once is
notifies systemd so through a special socket, using sd_notify(3). You
told systemd that Asterisk will notify it when ready, but Asterisk never
did. Thus systemd decided that service has failed.

Make sure you have libsystemd-dev installed. Alternatively, remove that
line (to keep the default Type=simple).

When built with systemd support, asterisk updates systemd about its
status using sd_notify(). This means that it is only considered as
started when it has loaded all the modules. This can make startup
ordering simpler.

BTW: if anybody wants, you can use sd_notify to set the status (as shown
in 'systemctl status asterisk.service') to an arbitrary string.
Basically just use ast_sd_notify() anywhere in the code. It becomes a
no-op if there's no systemd support.

-- 
   Tzafrir Cohen
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com

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Re: [asterisk-users] asterisk as non root

[Antony Stone]

On Thursday 20 April 2017 at 12:31:14, Atux Atux wrote:

> Hi. thanks a lot for your replies. I did stop the services and i did issued
> the  the "chown" and "chmod" commands listed in the guide.
> It is necessary to compile it, instead if using the apt-get version
> What am i missing?

Let's go back to basics for a moment - you say this is a Debian system - in my 
experience Debian already runs Asterisk as the "asterisk" user and not as 
root, so let's see what you have.

1. Start Asterisk (probably using "/etc/init.d/asterisk start", or maybe 
"service asterisk start")

2. Check who it's running as: "ps aux | grep asterisk"


Antony.


-- 
What makes you think I know what I'm talking about?
I just have more O'Reilly books than most people.

   Please reply to the list;
 please *don't* CC me.

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Re: [asterisk-users] asterisk as non root

[Atux Atux]

Hi. thanks a lot for your replies. I did stop the services and i did issued
the  the "chown" and "chmod" commands listed in the guide.
It is necessary to compile it, instead if using the apt-get version
What am i missing?



On Wed, Apr 19, 2017 at 10:47 PM, Antony Stone <
antony.st...@asterisk.open.source.it> wrote:

> On Wednesday 19 April 2017 at 18:48:29, Atux Atux wrote:
>
> > Hi.
> > Here is the output of the command
> >
> > root@pbx: ~ $  find / -name asterisk -exec ls -ld '{}' \;
> >
> > drwxr-xr-x 3 root root 4096 Apr 19 17:32 /usr/include/asterisk
> >
> > drwxr-x--- 3 asterisk asterisk 4096 Apr 19 17:32 /usr/lib/asterisk
> >
> > -rwxr-xr-x 1 root root 9719880 Apr 19 17:27 /usr/src/asterisk-11.25.1/
> main/asterisk
> >
> > drwxrwxr-x 3 1013 users 4096 Apr 19 16:56 /usr/src/asterisk-11.25.1/
> include/asterisk
> >
> > -rwxr-xr-x 1 root root 9719880 Apr 19 17:32 /usr/sbin/asterisk
>
> Okay, those look reasonable to me - however I'm surprised at some which
> are missing:
>
> /var/log/asterisk
> /var/spool/asterisk
> /var/run/asterisk
>
> Did you *stop* Asterisk before trying to change it to run as non-root?
>
> I think Tzafrir Cohen's comments are very well worth following.
>
>
> Antony.
>
> --
> "There has always been an underlying argument that we should open up our
> source code more broadly. The fact is that we are learning from open source
> and we are opening our code more broadly through Shared Source.
>
> Is there value to providing source code? The answer is unequivocally yes."
>
>  - Jason Matusow, head of Microsoft's Shared Source Program, in response
> to leaks of Windows source code on the Internet.
>
>Please reply to the
> list;
>  please *don't* CC
> me.
>
> --
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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:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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[asterisk-users] Backport of Stretch's asterisk.service file into Jessie: successful start not detected by systemd

[Olivier]

Hello,

I've been tasked to enable automatic Asterisk restart on failure on a
Jessie platform (running latest Asterisk 13.15.0).

I build a dedicated Jessie VM on which I installed Asterisk from source.
I configured a couple of files in /etc/asterisk directory.
I positively checedk that with simple config, Asterisk could sucessively
start using an /etc/init.d/asterisk file and sysv-systemd compatibility
tools.

Then I copied a /etc/systemd/system/asterisk.service file with the
following content:

[Unit]
Description=Asterisk PBX
Documentation=man:asterisk(8)
Wants=network-online.target
After=network-online.target

[Service]
Type=notify
ExecStart=/usr/sbin/asterisk -g -f -U asterisk
ExecReload=/usr/sbin/asterisk -rx 'core reload'
Restart=no
RestartSec=1
WorkingDirectory=/var/lib/asterisk

[Install]
WantedBy=multi-user.target

Please, note that this file has the same content as file
/lib/systemd/system/asterisk.service in Debian Stretch expect for line
Restart=on-failure I changed to Restart=no, for the moment.


After creating above asterisk.service file, running "systemctl
daemon-reload" and "systemctl start asterisk", I could observe:
- asterisk is starting OK,
- asterisk prints to its log file a line such as:
[Apr 20 11:23:22] VERBOSE[770] asterisk.c: Asterisk Ready.

but, at the same time, I got:
april 20 11:23:22 jessievm asterisk[770]: [Apr 20 11:23:22] NOTICE[770]:
app_queue.c:9095 reload_queues: No call queueing config fil
april 20 11:24:48 jessievm systemd[1]: asterisk.service start operation
timed out. Terminating.
april 20 11:24:48 jessievm asterisk[770]:
april 20 11:24:48 jessievm systemd[1]: Failed to start Asterisk PBX.


If I'm not mistaken:
- lastest "Asterisk Ready" line is visible in /var/log/asterisk/full but
invisible in Systemd journal,
- things work as if Systemd was not notified of Asterisk successful start
and than decided to stop Asterisk,
- man page [1] mentions support of type=notify in Debian Jessie
- on Stretch, I used asterisk 13.14.0 while I used asterisk 13.15.0, so I
supposed this issue doesn't come from Asterisk itself but the way I
configured my system.


Systemd versions are:
Jessie:
systemd 215
+PAM +AUDIT +SELINUX +IMA +SYSVINIT +LIBCRYPTSETUP +GCRYPT +ACL +XZ
-SECCOMP -APPARMOR

Stretch:
systemd 232
+PAM +AUDIT +SELINUX +IMA +APPARMOR +SMACK +SYSVINIT +UTMP +LIBCRYPTSETUP
+GCRYPT +GNUTLS +ACL +XZ +LZ4 +SECCOMP +BLKID +ELFUTILS +KMOD +IDN


My questions are:
1. Is it possible to use systemd's Restart=on-failure option to
automaticaly restart Asterisk, given Jessie systemd version ?
2. If positive, how ?
3. If negative, which option would you recommend (Monit ? ) ?

Best regards


[1] https://manpages.debian.org/jessie/systemd/systemd.service.5.en.html
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Re: [asterisk-users] Voicemail asking for login

[J Montoya or A J Stiles]

On Wednesday 19 Apr 2017, D'Arcy Cain wrote:
> Yes and  [using something like "1571"]  works just fine for us.  The problem
> is that we are trying
> to deal with the situation where someone calls themselves from another
> phone (internal or external) to pick up their messages.  In every other
> case it asks for their password (which is always numeric) and goes into
> the VM.  This one extension asks for a mailbox.

This is just screaming "configuration mismatch" -- or, possibly, "latent bug 
whereby things parsed in separate places should be treated the same, but are 
actually getting treated differently".

I think we are going to need to see your dialplan logic, and maybe your 
voicemail.conf, in order to work out what is different between this one user 
and all the others.  You might even need to use `hd` to examine the files, just 
in case there is a stray non-printing character spoiling things.

-- 
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] tcpbind and source IP address

[Kseniya Blashchuk]

Hi!
The issue did not reproduce with pjsip. As for ppa - somebody recommended
me ppa:sapian/asterisk. Does anybody use it maybe?


вт, 18 апр. 2017 г. в 2:24, Ludovic Gasc :

> Hi,
>
> I recommend you to install from sources, especially because the latest
> Asterisk 13 has several bugfixes for pjsip.
> To my knowledge, nobody proposes ppa or Debian backports for Asterisk.
>
> Wazo has Debian packages, but it's only for Debian Jessie and with extra
> patches for Wazo.
>
> --
> Ludovic Gasc (GMLudo)
> Lead Developer Architect at ALLOcloud
> https://be.linkedin.com/in/ludovicgasc
>
> 2017-04-16 21:36 GMT+02:00 Kseniya Blashchuk :
>
>> Hi!
>>
>> Unfortunately pjsip is broken in Ubuntu Asterisk installed from repo. Yes
>> I also thought to try with pjsip, just to know if it's also affected. I'll
>> try to make a test next days.
>>
>> On Sun, Apr 16, 2017, 8:18 PM Ludovic Gasc  wrote:
>>
>>> Hi Kseniya,
>>>
>>> You might test with chan_pjsip: We have less production experience with
>>> chan_pjsip than chan_sip, however, for now, we are more and more confident
>>> in this new stack while we're digging in documentation and we're testing on
>>> production.
>>>
>>> However, I've no idea if you'll have the same issue with pjsip, but more
>>> chances of support on the issues tracker of Asterisk to have help.
>>>
>>> Regards.
>>>
>>>
>>> --
>>> Ludovic Gasc (GMLudo)
>>> Lead Developer Architect at ALLOcloud
>>> https://be.linkedin.com/in/ludovicgasc
>>>
>>> 2017-03-13 14:41 GMT+01:00 Kseniya Blashchuk :
>>>
 Ok, thank you for the assistance!

 пн, 13 мар. 2017 г. в 16:38, Joshua Colp :

> On Mon, Mar 13, 2017, at 10:32 AM, Kseniya Blashchuk wrote:
> > Tested with latest Asterisk 14.3.0 on Ubuntu 16 kernel
> 4.4.0-66-generic
> > and
> > Centos 7 kernel 3.10.0-514.10.2.el7.x86_64. Absolutely the same
> behavior.
> > Joshua, maybe you can advice what can be done further?
>
> You can file an issue but chan_sip is a community supported module, so
> there is no guarantee of when it would be looked at and resolved.
> Ultimately though someone has to spend the time to replicate what is
> going on, look into the code, and understand what is going on.
>
> --
> 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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>
> 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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>>>
>>> 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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>>> 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
>>
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>> 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:
>