Re: [asterisk-users] Asterisk Fax detection *11.7

2014-01-23 Thread Martin 



in the sip.conf i specified

[general]
sendrpid=rpid
trustrpid=yes
language=de
videosupport=yes
callevents=yes
caninvite=yes


There is a typo in the last line above. Should be "canreinvite". AFAIK it's 
obsoleted in favor of directmedia. BTW, try to set it to NO.
BTW, what is the codec order? Fax detection doesn't work reliably over 
compressed codecs (g729 etc...), in my case didn't work at all...

try to add:
directmedia=no
disallow=all
allow=ulaw
allow=alaw

to your peer definition.

Martin 



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Re: [asterisk-users] MeetMe conference splitting

2014-01-23 Thread Rusty Newton 
On Thu, Jan 23, 2014 at 8:09 AM, Igor Dvorzhak  wrote:

> How to move 2 of 3 users in the MeetMe conference to the newly created
> MeetMe conference? Dialplan example is welcome.

Maybe something like an AMI redirect?

https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerAction_Redirect
https://wiki.asterisk.org/wiki/display/AST/AMI+Examples

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Re: [asterisk-users] Mailinglist Digium IP-phones : provisioning Digium D70

2014-01-23 Thread Rusty Newton 
On Wed, Jan 22, 2014 at 9:11 AM, Jonas Kellens  wrote:

> So how do I get the Digium IP-phone to use the md5 digest authentication ??

For the benefit of the archives and those reading the list, but not
the forums - this was answered here
http://forums.digium.com/viewtopic.php?p=195944

"It won't use MD5. It only uses Basic."

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Re: [asterisk-users] type=peer vs type=user (depricated?)

2014-01-23 Thread Rusty Newton 
On Thu, Jan 23, 2014 at 7:01 PM, Rusty Newton  wrote:

> the 1.8,11, or 12 branches. That being said, 12 is rather new and has
> many significant changes that should be considered.[3]

 I meant to reference link [1] of course. :)

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Re: [asterisk-users] type=peer vs type=user (depricated?)

2014-01-23 Thread Rusty Newton 
On Wed, Jan 22, 2014 at 5:56 PM, Michelle Dupuis  wrote:
> I'm looking at setting type=peer vs type=user (in both IAX and SIP conf
> entries), and I found a comment attributed to digium
> (http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer) in 2005 that
> type=user is depricated and that we should only use type=peer
>
> Is that still correct?  Will type=user be phased out, and should even new
> installs of older asterisk versions (eg: 1.6) use type=peer only?
>
> Are people still using type=user for phone sets?  (and type=peer for
> upstream/trunks only)

Howdy!

This is always a confusing part of the chan_sip SIP channel driver.

Rather than try to dig into any history, here is the current
documentation (from sip.conf.sample in the Asterisk source of
1.8,11,12) that you should base your decision to use a particular
"type" on:

"; SIP entities have a 'type' which determines their roles within Asterisk.
; * For entities with 'type=peer':
;   Peers handle both inbound and outbound calls and are matched by
ip/port, so for
;   The case of incoming calls from the peer, the IP address must
match in order for
;   The invitation to work. This means calls made from either
direction won't work if
;   The peer is unregistered while host=dynamic or if the host is
otherise not set to
;   the correct IP of the sender.
; * For entities with 'type=user':
;   Asterisk users handle inbound calls only (meaning they call
Asterisk, Asterisk can't
;   call them) and are matched by their authorization information
(authname and secret).
;   Asterisk doesn't rely on their IP and will accept calls regardless
of the host setting
;   as long as the incoming SIP invite authorizes successfully.
; * For entities with 'type=friend':
;   Asterisk will create the entity as both a friend and a peer.
Asterisk will accept
;   calls from friends like it would for users, requiring only that
the authorization
;   matches rather than the IP address. Since it is also a peer, a
friend entity can
;   be called as long as its IP is known to Asterisk. In the case of
host=dynamic,
;   this means it is necessary for the entity to register before
Asterisk can call it."

Most new work for SIP support in Asterisk is happening around
res_pjsip[1][2]. I don't know that there is any plans to deprecate
type=user going forward in chan_sip.

> Is that still correct?  Will type=user be phased out, and should even new
> installs of older asterisk versions (eg: 1.6) use type=peer only?

New installs of older Asterisk versions? That doesn't sound wise,
seeing as the 1.6 branch doesn't have any support, even for security
issues... A new install of Asterisk should be on a version of Asterisk
supported by the developers.[3] Right now, that would be the latest of
the 1.8,11, or 12 branches. That being said, 12 is rather new and has
many significant changes that should be considered.[3]

[1]: https://wiki.asterisk.org/wiki/display/AST/New+in+12
[2]: https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip
[3]: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

Hope that helps, thanks!

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Re: [asterisk-users] mixmonitor extension

2014-01-23 Thread Patrick Lists 

On 24-01-14 00:37, Marek Cervenka wrote:

can someone confirm that mp3 is unsupported? is patch available?


Iirc mp3 was in asterisk-addons in asterisk 1.4 and iirc in later 
versions of asterisk you can enable format_mp3 in make menuselect.



what about patch for Opus?

uncle google doesnt know


Did you really google?

http://lmgtfy.com/?q=asterisk+opus

Regards,
Patrick

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Re: [asterisk-users] mixmonitor extension

2014-01-23 Thread Marek Cervenka 

can someone confirm that mp3 is unsupported? is patch available?

what about patch for Opus?

uncle google doesnt know

Dne 23.1.2014 16:31, Gareth Blades napsal(a):

On 23/01/14 15:21, Marek Cervenka wrote:

hi,

which file extensios are supported in mixmonitor application?
https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Application_MixMonitor 



can i record to Opus?



"core show file formats" will give you a list of formats your system 
supports together with the filename extension. Not all may be 
supported for writing (mp3 being one example I believe).


> core show file formats
Format Name   Extensions
--    --
slin   mp3mp3
h264   h264   h264
g729   g729   g729
g719   g719   g719
gsmgsmgsm
g726   g726-16g726-16
g726   g726-24g726-24
g726   g726-32g726-32
g726   g726-40g726-40
h263   h263   h263
gsmwav49  WAV|wav49
g722   g722   g722
ulaw   au au
alaw   alaw   alaw|al|alw
ulaw   pcmpcm|ulaw|ul|mu|ulw
siren14siren14siren14
siren7 siren7 siren7
slin192sln192 sln192
slin96 sln96  sln96
slin48 sln48  sln48
slin44 sln44  sln44
slin32 sln32  sln32
slin24 sln24  sln24
slin16 sln16  sln16
slin12 sln12  sln12
slin   slnsln|raw
slin16 wav16  wav16
slin   wavwav
g723   g723sf g723|g723sf
ilbc   iLBC   ilbc
30 file formats registered.





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Re: [asterisk-users] CDR and Transfer, an asterisk scaring bug lasting from 1.4 version...

2014-01-23 Thread Leandro Dardini 
2014/1/23 Matthew Jordan 

> On Thu, Jan 23, 2014 at 12:44 PM, Leandro Dardini 
> wrote:
> > When you use a product which version number is 11 or even 12, you might
> go
> > with the assumption all big bugs are fixed and then you find there is a
> > huge, important, expensive bug still running in the code we are relaying
> > upon...
>
> First, not all versions in 11 are the same. Bugs do get fixed. What
> version of Asterisk 11 are you using?
>

I am using asterisk 11.6 and searching for "CDR transfer" in the issue
tracker return unfixed bugs

https://issues.asterisk.org/jira/browse/ASTERISK-11309
https://issues.asterisk.org/jira/browse/ASTERISK-21822



>
> Second, CDRs are not the same in Asterisk 12. Due to extensive changes
> in the bridging core, CDRs were re-worked heavily. You may want to
> take a look at the notes on the Asterisk wiki [1] for Asterisk 12, as
> well as the CDR specification for Asterisk 12 [2].
>

That seems great! Asterisk 12 really solved the CDR problem when
transferring!


>
> > The problem is simple. If you transfer a call, that dialing will be not
> > reported in the CDR, so no billing will happen. This is a simple example:
>
> And how did you do the transfer? Via DTMF features? Via a particular
> channel driver technology? If so, which channel drivers were involved?
>

Transfer was made using the "transfer" button of the phone and the result
was the same with blind or attended transfer


>
> What kind of transfer was it? Blind? Attended? Failed attended (the
> notorious blonde transfer)?
>
> >
> > Extension 100 calls extension 101
> > After 10 seconds, extension 100 transfer the call to
> > 00VERYEXPENSIVEDESTINATION
> > After 100 seconds, extension 101 hangup the call
> >
> > What do you find in the CDR? Just one record for a call from extension
> 100
> > to extension 101 lasting 10 seconds. What about the 100 seconds call from
> > 100 to 00VERYEXPENSIVEDESTINATION? It will never get billed.
> >
> > How do you manage these cases?
> >
>
> I'm not sure if there is a bug report filed against CDRs for the
> currently maintained branches for lost records during a blind or
> attended transfer that matches your issue. There is ASTERISK-17826,
> which may or may not be your issue: the noted lack of information
> makes it a bit hard to tell. The last issue that I'm aware of that we
> fixed regarding lost CDRs during a transfer was ASTERISK-21394, which
> was fixed in 11.4.0.
>
> So, if you're using a version prior to 11.4.0, you may want to
> consider upgrading. Again, due to the lack of information, it's hard
> to tell whether or not that would help you.
>
> Finally, CDRs in versions of Asterisk prior to 12 are subject to the
> whims of channel masquerades. This has historically made it difficult,
> if not impossible, to guarantee correctness during all transfer
> operations. Additionally, even if we could guarantee a particular set
> of behaviour in all circumstances, the lack of any clear agreement as
> to what a CDR should look like after an attended transfer (or in any
> situation that involved multiple parties) made the problem impossible
> to solve to the satisfaction of everyone. This particular reason is
> why CEL was created. If you continue to have problems with the billing
> records, you may want to consider moving your billing logic to CEL.
>
> Note that since (a) Asterisk 12 re-architected using a consistent
> bridging framework, which killed visible channel masquerades; and (b)
> we decided to not try and please everyone and just defined CDRs for
> how we thought they should work; the behaviour of CDRs in Asterisk 12
> and in future versions should be substantially more predictable.
>
> Matt
>

Thank you a lot! I am going to move ahead with asterisk 12!


>
> [1] https://wiki.asterisk.org/wiki/display/AST/New+in+12
> [2]
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CDR+Specification
>
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> Digium, Inc. | Engineering Manager
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: http://digium.com & http://asterisk.org
>
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Re: [asterisk-users] AMI eventmask question

2014-01-23 Thread Daniel Jenkins 
On Thu, Jan 23, 2014 at 8:46 PM, Matthew Jordan  wrote:

> On Thu, Jan 23, 2014 at 9:31 AM, Michelle Dupuis  wrote:
> > Thanks - I've been through that doc before and couldn't find the info
> > needed, which is why I went to the source code eventually.
> >
> > All events are grouped, and each group is given a name/flag like
> 'system',
> > 'call', etc.  The docs just don't say which events are in which
> group/flag.
> >
> > Perhaps something Digium could add at some point :)
>
> Or someone from the open source community... this is an open source
> project, after all :-)
>
> The managerEventInstance XML elements already have an attribute for
> the manager class, which is populated:
>
> 
> 
> Raised when a parked call times out.
> 
> 
> The parking lot extension.
> 
> 
> 
> The name of the parking lot.
> 
> 
> 
> 
> 
> 
> 
> 
> ParkedCall
> 
> 
> 
>
> You could actually grep the core-en_US.xml file and get all of the
> events that match to a particular class authorization.
>
> It doesn't show up in the CLI due to the xmldoc API not parsing out
> that attribute. The same is true for the wiki documentation; that
> project is up on github [1]. It wouldn't be a large patch to either to
> have that attribute displayed.
>
> Matt
>
> [1] https://github.com/asterisk/publish-docs
>
> --
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> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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>
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>

Thanks Matt, I was going to ask where the tool was that generated from
source, I'll take a look and see if I can contribute that back,

Dan
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Re: [asterisk-users] AMI eventmask question

2014-01-23 Thread Matthew Jordan 
On Thu, Jan 23, 2014 at 9:31 AM, Michelle Dupuis  wrote:
> Thanks - I've been through that doc before and couldn't find the info
> needed, which is why I went to the source code eventually.
>
> All events are grouped, and each group is given a name/flag like 'system',
> 'call', etc.  The docs just don't say which events are in which group/flag.
>
> Perhaps something Digium could add at some point :)

Or someone from the open source community... this is an open source
project, after all :-)

The managerEventInstance XML elements already have an attribute for
the manager class, which is populated:



Raised when a parked call times out.


The parking lot extension.



The name of the parking lot.








ParkedCall




You could actually grep the core-en_US.xml file and get all of the
events that match to a particular class authorization.

It doesn't show up in the CLI due to the xmldoc API not parsing out
that attribute. The same is true for the wiki documentation; that
project is up on github [1]. It wouldn't be a large patch to either to
have that attribute displayed.

Matt

[1] https://github.com/asterisk/publish-docs

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Re: [asterisk-users] CDR and Transfer, an asterisk scaring bug lasting from 1.4 version...

2014-01-23 Thread Matthew Jordan 
On Thu, Jan 23, 2014 at 12:44 PM, Leandro Dardini  wrote:
> When you use a product which version number is 11 or even 12, you might go
> with the assumption all big bugs are fixed and then you find there is a
> huge, important, expensive bug still running in the code we are relaying
> upon...

First, not all versions in 11 are the same. Bugs do get fixed. What
version of Asterisk 11 are you using?

Second, CDRs are not the same in Asterisk 12. Due to extensive changes
in the bridging core, CDRs were re-worked heavily. You may want to
take a look at the notes on the Asterisk wiki [1] for Asterisk 12, as
well as the CDR specification for Asterisk 12 [2].

> The problem is simple. If you transfer a call, that dialing will be not
> reported in the CDR, so no billing will happen. This is a simple example:

And how did you do the transfer? Via DTMF features? Via a particular
channel driver technology? If so, which channel drivers were involved?

What kind of transfer was it? Blind? Attended? Failed attended (the
notorious blonde transfer)?

>
> Extension 100 calls extension 101
> After 10 seconds, extension 100 transfer the call to
> 00VERYEXPENSIVEDESTINATION
> After 100 seconds, extension 101 hangup the call
>
> What do you find in the CDR? Just one record for a call from extension 100
> to extension 101 lasting 10 seconds. What about the 100 seconds call from
> 100 to 00VERYEXPENSIVEDESTINATION? It will never get billed.
>
> How do you manage these cases?
>

I'm not sure if there is a bug report filed against CDRs for the
currently maintained branches for lost records during a blind or
attended transfer that matches your issue. There is ASTERISK-17826,
which may or may not be your issue: the noted lack of information
makes it a bit hard to tell. The last issue that I'm aware of that we
fixed regarding lost CDRs during a transfer was ASTERISK-21394, which
was fixed in 11.4.0.

So, if you're using a version prior to 11.4.0, you may want to
consider upgrading. Again, due to the lack of information, it's hard
to tell whether or not that would help you.

Finally, CDRs in versions of Asterisk prior to 12 are subject to the
whims of channel masquerades. This has historically made it difficult,
if not impossible, to guarantee correctness during all transfer
operations. Additionally, even if we could guarantee a particular set
of behaviour in all circumstances, the lack of any clear agreement as
to what a CDR should look like after an attended transfer (or in any
situation that involved multiple parties) made the problem impossible
to solve to the satisfaction of everyone. This particular reason is
why CEL was created. If you continue to have problems with the billing
records, you may want to consider moving your billing logic to CEL.

Note that since (a) Asterisk 12 re-architected using a consistent
bridging framework, which killed visible channel masquerades; and (b)
we decided to not try and please everyone and just defined CDRs for
how we thought they should work; the behaviour of CDRs in Asterisk 12
and in future versions should be substantially more predictable.

Matt

[1] https://wiki.asterisk.org/wiki/display/AST/New+in+12
[2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CDR+Specification

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[asterisk-users] CDR and Transfer, an asterisk scaring bug lasting from 1.4 version...

2014-01-23 Thread Leandro Dardini 
When you use a product which version number is 11 or even 12, you might go
with the assumption all big bugs are fixed and then you find there is a
huge, important, expensive bug still running in the code we are relaying
upon...

The problem is simple. If you transfer a call, that dialing will be not
reported in the CDR, so no billing will happen. This is a simple example:

Extension 100 calls extension 101
After 10 seconds, extension 100 transfer the call to
00VERYEXPENSIVEDESTINATION
After 100 seconds, extension 101 hangup the call

What do you find in the CDR? Just one record for a call from extension 100
to extension 101 lasting 10 seconds. What about the 100 seconds call from
100 to 00VERYEXPENSIVEDESTINATION? It will never get billed.

How do you manage these cases?

Leandro
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Re: [asterisk-users] AMI eventmask question

2014-01-23 Thread Michelle Dupuis 
Thanks - I've been through that doc before and couldn't find the info needed, 
which is why I went to the source code eventually.

All events are grouped, and each group is given a name/flag like 'system', 
'call', etc.  The docs just don't say which events are in which group/flag.

Perhaps something Digium could add at some point :)

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Jenkins 
[dan.jenkin...@gmail.com]
Sent: Thursday, January 23, 2014 10:25 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] AMI eventmask question




On Thu, Jan 23, 2014 at 3:06 PM, Michelle Dupuis 
mailto:mdup...@ocg.ca>> wrote:
That's an interesting link - I didn't know you could set a per user eventfilter 
in the conf file

However, I'm hoping to do this in the AMI connection for more flexibility.  
Upon login, you can specify the event mask to restrict the type of events sent 
over the AMI connection.  Looking through the Asterisk code I think Newchannel 
falls under the 'call' event flag...but it would be nice to restrict this even 
further (since lots of events fall under 'call').

Ah, I remember what you're talking about now

https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+ManagerAction_Events

That's the link to the Asterisk 12 AMI docs, the specs changed in 12 and some 
events are different but it's the best docs available for the AMI in my opinion 
- the specs didn't change much at all.

So in answer to your question, using EventMask, I don't think this is possible 
as EventMask only works with those blocks of permissions, like call and 
security. If you wanted to only get the one event type then you'd need to do 
that in manager.conf - I may be wrong on that, I haven't looked at the source 
code or anything but from what the docs say, and the docs are generated from 
the source code now, you can only specifiy those larger blocks,

Hope I've helped,

Dan



From: 
asterisk-users-boun...@lists.digium.com
 
[asterisk-users-boun...@lists.digium.com]
 On Behalf Of Daniel Jenkins 
[dan.jenkin...@gmail.com]
Sent: Thursday, January 23, 2014 9:03 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] AMI eventmask question



On Thu, Jan 23, 2014 at 3:25 AM, Michelle Dupuis 
mailto:mdup...@ocg.ca>> wrote:

Hi

I'm creating an AMI client and I only want to get newchannel events (as well as 
responses to any actions I initiate).  What would I set the eventmask to to 
only get the newchannel events?

Are you talking about the event filter in manager.conf?

Using event filters you can say if you want to receive an event or not, 
specific to the event, instead of the classification of system/call etc

Take a look at the event filters in this blog post and you'll see what I mean,

http://hungrygeek.holidayextras.co.uk/2012/05/14/elastix-apply-configuration-changes-problem/



For anyone else looking...is there a table somewhere online that maps events to 
their eventmask categories?  I checked the asterisk wiki and voip-info but 
can't find this...


Dan




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Re: [asterisk-users] ISDN Cause Code 47 Errors

2014-01-23 Thread Dale Noll 
Thanks Richard and Andres.

I had come to the same conclusion, however the provider was fairly snarky
in saying is was my equipment.

We were able to replace the Cisco 2800 with a Cisco 2900 series and the
problem appears to have been resolved.

Thanks again, I always appreciate another set of eyes just in case I missed
something.

Dale



On Wed, Jan 22, 2014 at 11:57 AM, Andres  wrote:

>
> [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI
>> Span: 4 < TEI=0 Call Ref: len= 2 (reference 6918/0x1B06) (Sent from
>> originator)
>> [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI
>> Span: 4 < Message Type: RELEASE COMPLETE (90)
>> [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI
>> Span: 4 < [08 02 80 af]
>> [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI
>> Span: 4 < Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare:
>> 0  Location: User (0)
>> [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI
>> Span: 4 <  Ext: 1  Cause: Resource unavailable, unspecified
>> (47), class = Network Congestion (resource unavailable) (2) ]
>>
>   My guess is your provider did not have a free voice channel to pass
> audio at some leg in the call.  There could be multiple legs in the call
> and one of them had 'Network Congestion'.
>
> [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI
>> Span: 4 Received message for call 0xb750d598 on link 0x9b33f2c TEI/SAPI 0/0
>> [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI
>> Span: 4 -- Processing IE 8 (cs0, Cause)
>> [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI
>> Span: 4 q931.c:8567 post_handle_q931_message: Call 6918 enters state 0
>> (Null).  Hold state: Idle
>> [Jan 14 12:56:04] VERBOSE[13262] sig_pri.c: [Jan 14 12:56:04] Span 4:
>> Processing event PRI_EVENT_HANGUP
>> [Jan 14 12:56:04] VERBOSE[13262] sig_pri.c: [Jan 14 12:56:04] -- Span
>> 4: Channel 0/2 got hangup, cause 47
>>
>
>  Richard
>
>
>
>
>
> --
> Technical Supporthttp://www.cellroute.net
>
>
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> _
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Re: [asterisk-users] mixmonitor extension

2014-01-23 Thread Gareth Blades 

On 23/01/14 15:21, Marek Cervenka wrote:

hi,

which file extensios are supported in mixmonitor application?
https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Application_MixMonitor 



can i record to Opus?



"core show file formats" will give you a list of formats your system 
supports together with the filename extension. Not all may be supported 
for writing (mp3 being one example I believe).


> core show file formats
Format Name   Extensions
--    --
slin   mp3mp3
h264   h264   h264
g729   g729   g729
g719   g719   g719
gsmgsmgsm
g726   g726-16g726-16
g726   g726-24g726-24
g726   g726-32g726-32
g726   g726-40g726-40
h263   h263   h263
gsmwav49  WAV|wav49
g722   g722   g722
ulaw   au au
alaw   alaw   alaw|al|alw
ulaw   pcmpcm|ulaw|ul|mu|ulw
siren14siren14siren14
siren7 siren7 siren7
slin192sln192 sln192
slin96 sln96  sln96
slin48 sln48  sln48
slin44 sln44  sln44
slin32 sln32  sln32
slin24 sln24  sln24
slin16 sln16  sln16
slin12 sln12  sln12
slin   slnsln|raw
slin16 wav16  wav16
slin   wavwav
g723   g723sf g723|g723sf
ilbc   iLBC   ilbc
30 file formats registered.


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[asterisk-users] mixmonitor extension

2014-01-23 Thread Marek Cervenka 

hi,

which file extensios are supported in mixmonitor application?
https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Application_MixMonitor

can i record to Opus?

--
---
Marek Cervenka
===


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Re: [asterisk-users] AMI eventmask question

2014-01-23 Thread Daniel Jenkins 
On Thu, Jan 23, 2014 at 3:06 PM, Michelle Dupuis  wrote:

>  That's an interesting link - I didn't know you could set a per user
> eventfilter in the conf file
>
> However, I'm hoping to do this in the AMI connection for more
> flexibility.  Upon login, you can specify the event mask to restrict the
> type of events sent over the AMI connection.  Looking through the Asterisk
> code I think Newchannel falls under the 'call' event flag...but it would be
> nice to restrict this even further (since lots of events fall under 'call').
>

Ah, I remember what you're talking about now

https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+ManagerAction_Events

That's the link to the Asterisk 12 AMI docs, the specs changed in 12 and
some events are different but it's the best docs available for the AMI in
my opinion - the specs didn't change much at all.

So in answer to your question, using EventMask, I don't think this is
possible as EventMask only works with those blocks of permissions, like
call and security. If you wanted to only get the one event type then you'd
need to do that in manager.conf - I may be wrong on that, I haven't looked
at the source code or anything but from what the docs say, and the docs are
generated from the source code now, you can only specifiy those larger
blocks,

Hope I've helped,

Dan



>  --
> *From:* asterisk-users-boun...@lists.digium.com [
> asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Jenkins [
> dan.jenkin...@gmail.com]
> *Sent:* Thursday, January 23, 2014 9:03 AM
> *To:* Asterisk Users List
> *Subject:* Re: [asterisk-users] AMI eventmask question
>
>
>
> On Thu, Jan 23, 2014 at 3:25 AM, Michelle Dupuis  wrote:
>
>  Hi
>
>
>>  I'm creating an AMI client and I only want to get newchannel events (as
>> well as responses to any actions I initiate).  What would I set the
>> eventmask to to only get the newchannel events?
>>
>
>  Are you talking about the event filter in manager.conf?
>
>  Using event filters you can say if you want to receive an event or not,
> specific to the event, instead of the classification of system/call etc
>
>  Take a look at the event filters in this blog post and you'll see what I
> mean,
>
>
> http://hungrygeek.holidayextras.co.uk/2012/05/14/elastix-apply-configuration-changes-problem/
>
>
>
>>
>> For anyone else looking...is there a table somewhere online that maps
>> events to their eventmask categories?  I checked the asterisk wiki and
>> voip-info but can't find this...
>>
>>
> Dan
>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] AMI eventmask question

2014-01-23 Thread Michelle Dupuis 
That's an interesting link - I didn't know you could set a per user eventfilter 
in the conf file

However, I'm hoping to do this in the AMI connection for more flexibility.  
Upon login, you can specify the event mask to restrict the type of events sent 
over the AMI connection.  Looking through the Asterisk code I think Newchannel 
falls under the 'call' event flag...but it would be nice to restrict this even 
further (since lots of events fall under 'call').

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Jenkins 
[dan.jenkin...@gmail.com]
Sent: Thursday, January 23, 2014 9:03 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] AMI eventmask question



On Thu, Jan 23, 2014 at 3:25 AM, Michelle Dupuis 
mailto:mdup...@ocg.ca>> wrote:

Hi

I'm creating an AMI client and I only want to get newchannel events (as well as 
responses to any actions I initiate).  What would I set the eventmask to to 
only get the newchannel events?

Are you talking about the event filter in manager.conf?

Using event filters you can say if you want to receive an event or not, 
specific to the event, instead of the classification of system/call etc

Take a look at the event filters in this blog post and you'll see what I mean,

http://hungrygeek.holidayextras.co.uk/2012/05/14/elastix-apply-configuration-changes-problem/



For anyone else looking...is there a table somewhere online that maps events to 
their eventmask categories?  I checked the asterisk wiki and voip-info but 
can't find this...


Dan



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Re: [asterisk-users] Register => plain text password

2014-01-23 Thread José Pablo Méndez Soto 
Thanks A. J.




*José Pablo Méndez *


On Wed, Jan 22, 2014 at 3:22 AM, A J Stiles
wrote:

> On Wednesday 22 January 2014, José Pablo Méndez Soto wrote:
> > Hello,
> >
> > Is there anyway to encrypt or scramble a bit the secret used to register
> > with a provider? Im talking about the
> >
> > register => fromuser@fromdomain:secret@host
> >
> > directive in
> > sip.conf
>
> No.
>
> Well.  You *could* scramble it for storage; but that would only lull you
> into
> a false sense of security, because ultimately it would have to be able to
> be
> unscrambled by a program that was already right there on the machine,
> somewhere under /usr/src/ where any competent programmer can look at it.
>
> The client *has* to know the password in plaintext  (or at least, how to
> decrypt the stored, encrypted password),  in order to be able to send it to
> the server.
>
>
> The way things stand, the configuration file with the password in it need
> only
> be readable by the root user.  And you know it has a password in it, so you
> take care with it.
>
>
> Here is an explanation from the developers of the Pidgin IM client, as to
> why
> they store passwords in plaintext in their configuration file:
>
> https://developer.pidgin.im/wiki/PlainTextPasswords
>
> > This clever dude modified the code back in 1.4:
> >
> > http://www.oneharding.com/voip/asterisk_md5_register.html
>
> Unfortunately, that doesn't work.  It just elevates a stolen hash to the
> same
> level of usefulness as a stolen password  (and she even says so much, in
> the
> linked article).
>
> > I imagine that so many years later, and now with the implementation of
> > pjsip this secret could be better protected?
>
> No, because the underlying problem -- that decrypting a stored password
> also
> requires the decryption key; but if the decryption key and encrypted
> password
> are stored on the same machine, then anyone with access to the machine is
> able
> to decrypt the password -- is a limitation of the universe, *not* a
> limitation
> of present-day technology.  There is simply nothing that anybody could
> invent
> that would get around this.
>
> > It is very unsafe to keep the
> > accounts password right out there. Any ideas?
>
> It's hidden behind another password, and that's about as secure as it's
> mathematically possible ever to make it.  And if someone else has root
> access
> to your machine, then I humbly suggest that a SIP password might not be the
> driest lentil you have to soak.
>
>
> --
> AJS
>
> Answers come *after* questions.
>
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[asterisk-users] MeetMe conference splitting

2014-01-23 Thread Igor Dvorzhak 
Hello,

How to move 2 of 3 users in the MeetMe conference to the newly created
MeetMe conference? Dialplan example is welcome.

Best,
Igor
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Re: [asterisk-users] AMI eventmask question

2014-01-23 Thread Daniel Jenkins 
On Thu, Jan 23, 2014 at 3:25 AM, Michelle Dupuis  wrote:

Hi


> I'm creating an AMI client and I only want to get newchannel events (as
> well as responses to any actions I initiate).  What would I set the
> eventmask to to only get the newchannel events?
>

Are you talking about the event filter in manager.conf?

Using event filters you can say if you want to receive an event or not,
specific to the event, instead of the classification of system/call etc

Take a look at the event filters in this blog post and you'll see what I
mean,

http://hungrygeek.holidayextras.co.uk/2012/05/14/elastix-apply-configuration-changes-problem/



>
> For anyone else looking...is there a table somewhere online that maps
> events to their eventmask categories?  I checked the asterisk wiki and
> voip-info but can't find this...
>
>
Dan
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Re: [asterisk-users] Change the preferred audio playback format

2014-01-23 Thread Gopalakrishnan N 
Hope basically depends on the codec Asterisk will playback the file
automatically
On 23 Jan 2014 19:25, "Gareth Blades" 
wrote:

> On 23/01/14 13:38, Ishfaq Malik wrote:
>
>> Hi
>>
>> Is there any way to change the preferred audio playback format in
>> asterisk (I'm using 1.8.25.0)
>> i.e. first check for gsm, if doesn't exits then check for slin?
>>
>
> It should pick whichever source format requires the least cpu to transcode
> into the desired output format.
> So generally that means if there is a source available in the same format
> as the output then it will use it otherwise it will use slin etc...
>
>
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Re: [asterisk-users] Change the preferred audio playback format

2014-01-23 Thread Gareth Blades 

On 23/01/14 13:38, Ishfaq Malik wrote:

Hi

Is there any way to change the preferred audio playback format in 
asterisk (I'm using 1.8.25.0)

i.e. first check for gsm, if doesn't exits then check for slin?


It should pick whichever source format requires the least cpu to 
transcode into the desired output format.
So generally that means if there is a source available in the same 
format as the output then it will use it otherwise it will use slin etc...



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[asterisk-users] Change the preferred audio playback format

2014-01-23 Thread Ishfaq Malik 
Hi

Is there any way to change the preferred audio playback format in asterisk
(I'm using 1.8.25.0)
i.e. first check for gsm, if doesn't exits then check for slin?

Regards

Ish

-- 

Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk

Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
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[asterisk-users] Recording conferences with changing bitrate

2014-01-23 Thread Richard Kenner 
I'm running 10.7.1 (yes, I know it's old, but this may be a problem in
later versions too) and had a conference being recorded via:

Set(CONFBRIDGE(bridge,record_conference)=yes)

The bridge started out at 8KHz despite one HD device.  But when the
second came in (G.722), it switched to 16KHz.  At that point, the recording
file had the bitrate change in the middle.  That seems wrong.  I'd expect
the bitrate of the recording channel to remain unchanged and transcoding
to be used to do the recording.  But it wasn't.

Does this "ring a bell" with anybody?

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