Re: [asterisk-users] Outbound T.38 via RTP with pjsip does not work as expected

2017-04-06 Thread Michael Maier 
On 04/06/2017 at 08:33 PM, Joshua Colp wrote:
> On Thu, Apr 6, 2017, at 03:15 PM, Michael Maier wrote:
>> Hello!
>>
>> I'm trying to send a fax via T.38 to a destination, which should be T.38
>> capable. My provider supports T.38, too. Unfortunately, it doesn't work.
>> This means:
>>
>> Call is started and SDP is negotiated w/ alaw. Callee sends reinvite -
>> for alaw again (and not for T.38)!! After about 30s, callee hangs up
>> because of missing data (this is true, because I don't send alaw coded
>> fax data.
>>
>> Tracing the signaling shows, that the callee doesn't have any
>> possibility to recognize, if I'm supporting T.38, because it is never
>> sent during Invite process.
>>
>> I'm missing the media feature tag sip.fax in the contact header. Did I
>> miss some configuration?
> 
> This is not currently supported in either chan_sip or chan_pjsip.
> There's no configuration which will enable it. It would need to be
> written. Have you confirmed this is what is needed by them?

No - I have to confirm it. But this may take some time :-).


Thanks,
Michaal

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Re: [asterisk-users] restart system from extension

2017-04-06 Thread Steve Edwards 

Please don't top post.


On Thu, Apr 06, 2017 at 08:16:34PM +0300, Atux Atux wrote:


hi. i would like to be able to reboot the system from my extension. is 
that possible? if yes, how?


Στις 6 Απρ 2017 8:25 μ.μ., ο χρήστης "Tzafrir Cohen" 
 έγραψε:



System('sudo /sbin/reboot')


On Thu, 6 Apr 2017, Atux Atux wrote:


Could you give some more details please?


While there may be many ways, sudo is the most obvious. I'd suggest 
sticking with it and learning it before researching other methods if you 
find a use case that sudo can't handle.


Sudo allows a user to run specific (or any) commands as another user. For 
example, allow the asterisk user to run a command as apache or root. This 
ability can be configured to run with or without a password.


If you're running Asterisk as root, which you shouldn't do, you don't need 
sudo since root can do almost everything.


If you're running Asterisk as asterisk, you can create a file named 
asterisk in /etc/sudoers.d/ that tells sudo what commands you want sudo to 
allow the user named asterisk, what user that command will be executed as, 
and if a password is needed. And more -- wade through 'man sudoers' for 
all the gory details.


In it's simplest form, which is not the most secure, this file would look 
like:


asterisk ALL=(ALL) NOPASSWD: ALL

This means the user asterisk can run all commands as any user on any host 
without needing a password. Kind of scary, right?


The best way to edit this file is with the 'visudo' command because it 
will help prevent you from creating an invalid file. This can be very 
important when you are editing your file (for your login) remotely.


You should take the time to create a 'sudoers' file for the asterisk user 
that allows the single '/sbin/reboot' command. Do as I say, not as I've 
done :)


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Re: [asterisk-users] Outbound T.38 via RTP with pjsip does not work as expected

2017-04-06 Thread Joshua Colp 
On Thu, Apr 6, 2017, at 03:15 PM, Michael Maier wrote:
> Hello!
> 
> I'm trying to send a fax via T.38 to a destination, which should be T.38
> capable. My provider supports T.38, too. Unfortunately, it doesn't work.
> This means:
> 
> Call is started and SDP is negotiated w/ alaw. Callee sends reinvite -
> for alaw again (and not for T.38)!! After about 30s, callee hangs up
> because of missing data (this is true, because I don't send alaw coded
> fax data.
> 
> Tracing the signaling shows, that the callee doesn't have any
> possibility to recognize, if I'm supporting T.38, because it is never
> sent during Invite process.
> 
> I'm missing the media feature tag sip.fax in the contact header. Did I
> miss some configuration?

This is not currently supported in either chan_sip or chan_pjsip.
There's no configuration which will enable it. It would need to be
written. Have you confirmed this is what is needed by them?

-- 
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Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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[asterisk-users] Outbound T.38 via RTP with pjsip does not work as expected

2017-04-06 Thread Michael Maier 
Hello!

I'm trying to send a fax via T.38 to a destination, which should be T.38
capable. My provider supports T.38, too. Unfortunately, it doesn't work.
This means:

Call is started and SDP is negotiated w/ alaw. Callee sends reinvite -
for alaw again (and not for T.38)!! After about 30s, callee hangs up
because of missing data (this is true, because I don't send alaw coded
fax data.

Tracing the signaling shows, that the callee doesn't have any
possibility to recognize, if I'm supporting T.38, because it is never
sent during Invite process.

I'm missing the media feature tag sip.fax in the contact header. Did I
miss some configuration?

That's my setup:
Hylafax sends fax to t38modem and t38modem is connected via SIP to
asterisk as extension. The extension is bound to an outbound route,
which uses the t.38 capable ISP.

pjsip.endpoint.conf:

[ISP]
t38_udptl=yes
t38_udptl_nat=no # there is no nat necessary
t38_udptl_ec=fec

[t38endpoint]
t38_udptl=yes
t38_udptl_ec=fec


Any idea?


Thanks,
Michael

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Re: [asterisk-users] restart system from extension

2017-04-06 Thread Atux Atux 
Could you give some more details please?

Στις 6 Απρ 2017 8:25 μ.μ., ο χρήστης "Tzafrir Cohen" <
tzafrir.co...@xorcom.com> έγραψε:

> On Thu, Apr 06, 2017 at 08:16:34PM +0300, Atux Atux wrote:
> > hi. i would like to be able to reboot the system from my extension. is
> that
> > possible? if yes, how?
>
> System('sudo /sbin/reboot')
>
> You need to allow that in a sudoers file, of course. This may or may not
> be a good idea.
>
> There are a host of other methods to permit unplivilidged users /
> processes to run do specific priviliged actions.
>
> --
>Tzafrir Cohen
> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com
>
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> org/
>
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Re: [asterisk-users] Define SIP fromuser field in Dial()-command

2017-04-06 Thread Jonas Kellens 

Hello


in what way does this set the 'fromuser' field in the SIP INVITE ?



Kind regards.


J.

On 05-04-17 22:05, Pete Mundy wrote:

Hi Jonas

Does the information at this link help?

http://the-asterisk-book.com/1.6/funktionen-callerid.html

Pete


On 5/04/2017, at 8:11 pm, Jonas Kellens > wrote:


Hello

anyone have some useful input on this ?



Thanks.


On 03-04-17 10:25, Jonas Kellens wrote:

Hello

how can I set the fromuser field of the SIP INVITE when using the 
Dial()-command in the dialplan ?





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Re: [asterisk-users] Commit dialplan & other config. in memory to disk?

2017-04-06 Thread Steve Edwards 

On Thu, 6 Apr 2017, Nathan Anderson wrote:

In the future, if I were to ever run into a similar situation, is there 
any way to request or instruct Asterisk to write the current dialplan 
that is in memory and other important config files (e.g., users.conf) to 
disk in a *different* location than where it originally read them from 
when it started up?  I could have saved myself a crap-ton of work if 
this were possible...


On Thu, 6 Apr 2017, John Novack wrote:

I run a cron job every early morning to do just that as well as the 
MySql data files so if need be I can recreate a machine and have call 
records and whatever not more than a day stale.


Ditto.

Each host emails all of it's configuration files and database dumps in a 
tarball every day. This has saved mine and my client's bacon on more than 
1 occasion.


It also comes in handy when a client says 'I liked it better the way it 
was last Monday' or when a client says 'why am I being charged for x hours 
work last Tuesday.' I can restore the tarballs and a quick diff shows 
everything I did -- down to the time-stamped commands in my .bash_history.


Back to your specific problem...

No. There is no easy way to recover configurations from a running 
Asterisk. The misleadingly named 'dialplan save' will not save you. It 
will trash any file in it's way with a useless rendition of your dialplan. 
It will show your globals and your application calls, but without any of 
the parameters.


Seriously, was this some disgruntled developer's middle finger as they 
stormed out of the office?


I've found myself in your position, not by hardware failure, but by a 
fat-fingered 'rm' and had the better part of a days work at stake.


I used the 'script' shell command and the Asterisk 'dialplan show' command 
and then ran that through a PHP script that recovered enough.


You're welcome to the script at:

http://www.sedwards.com/recover-show-dialplan.php

--
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-
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https://www.linkedin.com/in/steve-edwards-4244281

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Re: [asterisk-users] restart system from extension

2017-04-06 Thread Tzafrir Cohen 
On Thu, Apr 06, 2017 at 08:16:34PM +0300, Atux Atux wrote:
> hi. i would like to be able to reboot the system from my extension. is that
> possible? if yes, how?

System('sudo /sbin/reboot')

You need to allow that in a sudoers file, of course. This may or may not
be a good idea.

There are a host of other methods to permit unplivilidged users /
processes to run do specific priviliged actions.

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

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[asterisk-users] restart system from extension

2017-04-06 Thread Atux Atux 
hi. i would like to be able to reboot the system from my extension. is that
possible? if yes, how?
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Re: [asterisk-users] feature codes

2017-04-06 Thread Atux Atux 
hi. thanks for the reply. when you say configure them in features.conf?

On Thu, Apr 6, 2017 at 11:57 AM, Marcelo Terres  wrote:

> 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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Re: [asterisk-users] Commit dialplan & other config. in memory to disk?

2017-04-06 Thread Tzafrir Cohen 
On Thu, Apr 06, 2017 at 09:54:25AM +, Nathan Anderson wrote:
> 'lo,
> 
> So yesterday, one of our clients had the misfortune of having the disk that 
> their Asterisk config (*.conf) was stored on take a dirt nap.  Of course, 
> Asterisk was still running at the time, and everything continued to work 
> (except for voicemail, which was stored on the same disk) right up until I 
> shut down Asterisk to investigate what was going on.  Because the disk was 
> dead, though, I couldn't start Asterisk back up after that, and OF COURSE the 
> backups were not firing off correctly so now we are faced with regenerating 
> the config again (including dialplan) from scratch.
> 
> In the future, if I were to ever run into a similar situation, is there any 
> way to request or instruct Asterisk to write the current dialplan that is in 
> memory and other important config files (e.g., users.conf) to disk in a 
> *different* location than where it originally read them from when it started 
> up?  I could have saved myself a crap-ton of work if this were possible...

I'm not sure this works. But it's worth a shot: (bind-?)mount a writable
file system at /etc/asterisk .

Be sure to umount it quickly enough after the write.

-- 
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Re: [asterisk-users] Issues with Siren14 codec in Asterisk 14.3.0

2017-04-06 Thread Joshua Colp 
On Thu, Apr 6, 2017, at 12:17 PM, Richard Kenner wrote:
> > I would say this is a bug in func_speex and not in codec_siren14. This
> > is because the datalen is zero. 
> 
> Ah!  So, like?
> 
> *** func_speex.c.orig   2017-02-13 15:00:19.0 -0500
> --- func_speex.c2017-04-06 11:16:03.0 -0400
> ***
> *** 185,189 
> }
>   
> !   speex_preprocess(sdi->state, frame->data.ptr, NULL);
> snprintf(source, sizeof(source), "%s/speex", frame->src);
> if (frame->mallocd & AST_MALLOCD_SRC) {
> --- 185,190 
> }
>   
> !   if (frame->data.ptr && frame->datalen)
> ! speex_preprocess(sdi->state, frame->data.ptr, NULL);
> snprintf(source, sizeof(source), "%s/speex", frame->src);
> if (frame->mallocd & AST_MALLOCD_SRC) {

Yes, although really you only need to check datalen. If that is set to
non-zero then there is data.

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Re: [asterisk-users] Issues with Siren14 codec in Asterisk 14.3.0

2017-04-06 Thread Richard Kenner 
> I would say this is a bug in func_speex and not in codec_siren14. This
> is because the datalen is zero. 

Ah!  So, like?

*** func_speex.c.orig   2017-02-13 15:00:19.0 -0500
--- func_speex.c2017-04-06 11:16:03.0 -0400
***
*** 185,189 
}
  
!   speex_preprocess(sdi->state, frame->data.ptr, NULL);
snprintf(source, sizeof(source), "%s/speex", frame->src);
if (frame->mallocd & AST_MALLOCD_SRC) {
--- 185,190 
}
  
!   if (frame->data.ptr && frame->datalen)
! speex_preprocess(sdi->state, frame->data.ptr, NULL);
snprintf(source, sizeof(source), "%s/speex", frame->src);
if (frame->mallocd & AST_MALLOCD_SRC) {


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Re: [asterisk-users] Issues with Siren14 codec in Asterisk 14.3.0

2017-04-06 Thread Joshua Colp 
On Thu, Apr 6, 2017, at 10:57 AM, Richard Kenner wrote:
> I'm seeing Asterisk crashes with the following frame at func_speex.c:188:
> 
> (gdb) p *frame
> $6 = {frametype = AST_FRAME_VOICE, subclass = {integer = 0, 
> format = 0xe2f9e20, frame_ending = 0}, datalen = 0, samples = 640, 
>   mallocd = 1, mallocd_hdr_len = 232, offset = 64, 
>   src = 0x2ac07413e7f8 "siren14tolin32", data = {ptr = 0x3cab9378, 
> uint32 = 1017877368, pad = "x\223\253<\000\000\000"}, delivery = {
> tv_sec = 1491485582, tv_usec = 407272}, frame_list = {next = 0x0}, 
>   flags = 0, ts = 0, len = 0, seqno = 0}
> 
> frame->data.ptr is an out-of-range address.
> 
> Does this ring a bell to anybody?  Without sources of the Siren14 codec,
> how would you recommend we debug this?

I would say this is a bug in func_speex and not in codec_siren14. This
is because the datalen is zero. There is no data on the frame (which is
why frame->data.ptr is garbage, although it could be improved to be
NULL) but there are samples, so it's an interpolated frame. The
func_speex module should be changed to handle this.

I'd suggest filing an issue on the issue tracker for it[1].

[1] https://issues.asterisk.org/jira

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[asterisk-users] Issues with Siren14 codec in Asterisk 14.3.0

2017-04-06 Thread Richard Kenner 
I'm seeing Asterisk crashes with the following frame at func_speex.c:188:

(gdb) p *frame
$6 = {frametype = AST_FRAME_VOICE, subclass = {integer = 0, 
format = 0xe2f9e20, frame_ending = 0}, datalen = 0, samples = 640, 
  mallocd = 1, mallocd_hdr_len = 232, offset = 64, 
  src = 0x2ac07413e7f8 "siren14tolin32", data = {ptr = 0x3cab9378, 
uint32 = 1017877368, pad = "x\223\253<\000\000\000"}, delivery = {
tv_sec = 1491485582, tv_usec = 407272}, frame_list = {next = 0x0}, 
  flags = 0, ts = 0, len = 0, seqno = 0}

frame->data.ptr is an out-of-range address.

Does this ring a bell to anybody?  Without sources of the Siren14 codec,
how would you recommend we debug this?

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Re: [asterisk-users] Commit dialplan & other config. in memory to disk?

2017-04-06 Thread John Novack 



Nathan Anderson wrote:

'lo,

So yesterday, one of our clients had the misfortune of having the disk that 
their Asterisk config (*.conf) was stored on take a dirt nap.  Of course, 
Asterisk was still running at the time, and everything continued to work 
(except for voicemail, which was stored on the same disk) right up until I shut 
down Asterisk to investigate what was going on.  Because the disk was dead, 
though, I couldn't start Asterisk back up after that, and OF COURSE the backups 
were not firing off correctly so now we are faced with regenerating the config 
again (including dialplan) from scratch.

In the future, if I were to ever run into a similar situation, is there any way 
to request or instruct Asterisk to write the current dialplan that is in memory 
and other important config files (e.g., users.conf) to disk in a *different* 
location than where it originally read them from when it started up?  I could 
have saved myself a crap-ton of work if this were possible...

Thanks,

-- Nathan


Isn't this a task for Linux and a cron job rather than asterisk?
Simplest thing would be to copy to another machine even off site, all the confs 
and whatever else would help you resurrect a machine more quickly.
Backups not "firing off correctly"  means what?

When changes are made to the dialplan a copy off site or at least off machine 
is in order

I run a cron job every early morning to do just that as well as the MySql data 
files so if need be I can recreate a machine and have call records and whatever 
not more than a day stale.


John Novack

--

Dog is my Co-pilot


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Re: [asterisk-users] Commit dialplan & other config. in memory to disk?

2017-04-06 Thread Jean Aunis 
You can execute something like asterisk -rx "dialplan show" > 
some_file.conf, but unfortunately the result cannot be directly parsed 
by Asterisk. Still it will give you a readable snapshot of your current 
dialplan.



Le 06/04/2017 à 11:54, Nathan Anderson a écrit :

'lo,

So yesterday, one of our clients had the misfortune of having the disk that 
their Asterisk config (*.conf) was stored on take a dirt nap.  Of course, 
Asterisk was still running at the time, and everything continued to work 
(except for voicemail, which was stored on the same disk) right up until I shut 
down Asterisk to investigate what was going on.  Because the disk was dead, 
though, I couldn't start Asterisk back up after that, and OF COURSE the backups 
were not firing off correctly so now we are faced with regenerating the config 
again (including dialplan) from scratch.

In the future, if I were to ever run into a similar situation, is there any way 
to request or instruct Asterisk to write the current dialplan that is in memory 
and other important config files (e.g., users.conf) to disk in a *different* 
location than where it originally read them from when it started up?  I could 
have saved myself a crap-ton of work if this were possible...

Thanks,

-- Nathan




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[asterisk-users] Commit dialplan & other config. in memory to disk?

2017-04-06 Thread Nathan Anderson 
'lo,

So yesterday, one of our clients had the misfortune of having the disk that 
their Asterisk config (*.conf) was stored on take a dirt nap.  Of course, 
Asterisk was still running at the time, and everything continued to work 
(except for voicemail, which was stored on the same disk) right up until I shut 
down Asterisk to investigate what was going on.  Because the disk was dead, 
though, I couldn't start Asterisk back up after that, and OF COURSE the backups 
were not firing off correctly so now we are faced with regenerating the config 
again (including dialplan) from scratch.

In the future, if I were to ever run into a similar situation, is there any way 
to request or instruct Asterisk to write the current dialplan that is in memory 
and other important config files (e.g., users.conf) to disk in a *different* 
location than where it originally read them from when it started up?  I could 
have saved myself a crap-ton of work if this were possible...

Thanks,

-- Nathan

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Re: [asterisk-users] feature codes

2017-04-06 Thread 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.
>
>
> --
> _
> -- 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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[asterisk-users] feature codes

2017-04-06 Thread Atux Atux 
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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