Re: [asterisk-users] Channels freeze on Confbridge

[Andrew Yager]

On Sun, 23 Aug 2020 at 18:23, Antony Stone <
antony.st...@asterisk.open.source.it> wrote:

> On Saturday 22 August 2020 at 22:51:48, Sebastian Nielsen wrote:
>
> > I had a similiar problem, but with calls dropping after 30 sec.
> > It turned out that Android didn't support RP-CID (Reverse Party Caller
> ID)
> > so when I sent the name of the callee to the caller (as some sort of
> > "centralized phonebook function") it caused calls to be dropped as
> android
> > refused to reply on the packets or sent rejections back.
>
> I can see the point you're making here, but what's going to do this after
> 30
> *minutes* of normal call?
>

Have seen plenty of ALGs do weird things like this. 30 minutes is a nice
number, and nice enough that I'd go hunting for ALG issues. It's a good
multiple of 3 minutes, and quite possibly is some big number someone
thought to set in something that "no one would ever hit".

A tcpdump would probably show you what's going on if the logs are otherwise
unclear, and you could also make sure you have sensible RTP timeout rules.

Andrew
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Re: [asterisk-users] Remove ANSI colour trings from log files only

[Andrew Yager]

On Fri, 24 Jul 2020 at 14:23, Tim Požár  wrote:

> You can post process the logs with something like sed.  See:
>
>
> https://superuser.com/questions/380772/removing-ansi-color-codes-from-text-stream



Yeah; we're injesting using filebeat and you can do sed style parsing on
lines into it using one fo the pipeline parsers; it' sjust a bit messy
config wise when using the stnadard docker containers. But if it's not a
"simple" answer it sounds like that's not a bad option.

Andrew
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[asterisk-users] Remove ANSI colour trings from log files only

[Andrew Yager]

Hi,

Is there a way to drop the ANSI colour strings from log files? In
particular, I've got JSON logging throwing logs over to ES, but they have
the ANSI colour escape sequences.

Ideally I don't want to lose coloured logs from the console though, and I
can't "see" a way to do this.

Ast 16 at the moment…

Andrew
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Re: [asterisk-users] Failed to authenticate device message

[Andrew Yager]

Did you check your security log?

There is usually a wealth of info there about who, what, where when and why.

Andrew


On Wed, 22 Jul 2020 at 11:22 pm, Jerry Geis  wrote:

> >exten = i,1,Verbose(Incoming ANONYMOUS SIP call from ${CALLERID(name)}
> >${CALLERID(num)} SRC IP ${CHANNEL(recvip)})
>
>
> Thanks - its not an incoming call - its just a log on the CLI
> There is nothing before it and nothing after - no incoming call.
>
> Jerry
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[asterisk-users] PJSIP AoR vs Endpoint

[Andrew Yager]

Hi,

I realise this is an old question, but I’m struggling to get my head around
it.

The ERD suggests that endpoints can link to multiple AoRs

In what situation would you actually use this? Given that mapping of
inbound calls is primary done to the endpoint, it looks to me like most of
the scenarios where this might be beneficial are actually not possible?

One example I had envisaged was being able to have multiple distinct auth
entries for one endpoint (eg for different devices, such as a WebRTC
device, a TLS device or a UDP device) but the restrictions around device
identification seem to make this not achievable as the auth user or user
needs to be the endpoint name to actually find the correct endpoint.

The answer is somewhat moot; but as I’m thinking about implementation of
some things I’m just trying to make sure I haven’t missed the obvious “yep
- you’d use this here, because it makes X or Y amazing”.

Andrew
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[asterisk-users] 16.11.1 release removed from current

[Andrew Yager]

Hi,

This is not a big issue… I just noticed a build script that was pulling the
16.11.1 release from https://downloads.asterisk.org/pub/telephony/asterisk has
started to fail and went to investigate (I need to test a patch for a
bugifx). It looks like when the 16.12-rc1 RC was released the 16.11.1 was
pulled from the current release directory.

I realise it's probably better practice to pull source tarballs from
old-releases, but is there a reason that the current "released" version was
pulled?

Andrew
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[asterisk-users] Yealink VC200?

[Andrew Ruthven]

Hey,

I'm thinking about buying a couple of Yealink VC200s for some of our
teams, but I haven't been able to find much, if any, information about
the VC200 support for Asterisk and other video conferencing solutions
(Jitsi Meet).

Does anyone know if this kit works okay with Asterisk etc?

Cheers,
Andrew
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Re: [asterisk-users] Dropped calls when all DAHDI lines in use

[Andrew Martin]

- Original Message -
> From: "John Novack SCII_U" 
> To: "Asterisk Users Mailing List, Non-Commercial Discussion" 
> , "Andrew Martin"
> 
> Sent: Monday, October 8, 2018 4:29:41 PM
> Subject: Re: [asterisk-users] Dropped calls when all DAHDI lines in use

> Have you given any thought to moving to at least a current supported version 
> 13?
> Asterisk 11 has been EOL for some time now
> I doubt you will get a resolution to a version no longer supported.
> Moving to the latest version 13 should be relatively quick and painless, and 
> if
> the issue persists you might find more assistance.
> 
> John Novack
> 
> 
> Andrew Martin wrote:
>> Hello,
>>
>> I am running Asterisk 11.17 with DAHDI 2.9.0 and an OpenVox A800P with 8x 
>> analog
>> POTS lines coming into my Asterisk server from the phone company. 
>> Internally, I
>> have about 180 SIP clients defined in sip.conf. What appears to be happening 
>> is
>> that if existing calls are consuming all 8 external lines and a new SIP 
>> client
>> attempts to make a call, an existing call gets dropped. The asterisk log 
>> simply
>> shows this as a normal hangup, so I am not able to easily distinguish 
>> between a
>> normal hangup and this type of dropped call. In testing, I am able to get a 
>> new
>> SIP client to report "service unavailable" when all 8 lines are consumed, yet
>> still drops are reported.
>>
>> I have been unable to find any configuration settings pertaining to 
>> prioritizing
>> existing calls over new calls. What else can I look for to attempt to debug 
>> and
>> fix this so that existing calls are not dropped?
>>
>> Thanks,
>>
>> Andrew
>>
> 
> --
> Dog is my Co-Pilot

John,

Thanks for the reply. Yes, I am planning on moving to version 13 but need to 
find a
solution in the interim. If there are any configuration options that pertain to 
which actions to take with existing calls when new calls come in, I think it is 
likely
that they would be shared between both versions (and I want to make sure I have 
the
correct settings when I switch to version 13 too). Can you advise on any 
tunables
related to handling existing vs new calls?

Thanks,

Andrew

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[asterisk-users] Dropped calls when all DAHDI lines in use

[Andrew Martin]

Hello,

I am running Asterisk 11.17 with DAHDI 2.9.0 and an OpenVox A800P with 8x 
analog 
POTS lines coming into my Asterisk server from the phone company. Internally, I
have about 180 SIP clients defined in sip.conf. What appears to be happening is
that if existing calls are consuming all 8 external lines and a new SIP client
attempts to make a call, an existing call gets dropped. The asterisk log simply
shows this as a normal hangup, so I am not able to easily distinguish between a
normal hangup and this type of dropped call. In testing, I am able to get a new
SIP client to report "service unavailable" when all 8 lines are consumed, yet
still drops are reported.

I have been unable to find any configuration settings pertaining to prioritizing
existing calls over new calls. What else can I look for to attempt to debug and
fix this so that existing calls are not dropped?

Thanks,

Andrew

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

[Andrew Furey]

Ditto; a Gmail issue?

Andrew

On 12 June 2017 at 16:00, Marcelo Terres <mhter...@gmail.com> wrote:

> It is happening the same with me.
>
> Regards,
> Marcelo H. Terres <mhter...@gmail.com>
> IM: mhter...@jabber.mundoopensource.com.br
> https://www.mundoopensource.com.br
> https://twitter.com/mhterres
> https://linkedin.com/in/marceloterres
>
>
> On 12 June 2017 at 08:07, Olivier <oza.4...@gmail.com> wrote:
> > Hello,
> >
> > I'm a faithful reader of this mailing list, for several years now.
> >
> > Lately, I'm receiving emails asking me to re-enable my list subscription
> due
> > to "excessive bouncing".
> >
> > What does this exactly mean and why am I receiving this ?
> > Beside re-enabling my subscription, what can I do to improve things ?
> >
> > Regards
> >
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Re: [asterisk-users] How to detect fake CallerID? (8xx?)

[Andrew Latham]

On Wed, May 10, 2017 at 10:11 AM, Steve Edwards <asterisk@sedwards.com>
wrote:

> I have a 'time and attendance' application. Think janitorial or security
> kind of thing where an employee goes from location to location.
>
> They're supposed to 'clock in' when they get to a site using a phone at
> that site to prove they're there.
>
> Some employees have discovered 'fake caller ID' services can be used to
> say they're on site when they are not.
>
> How can I detect a fake CallerID? The INVITE looks the same to me.
>
> If I have the employees call an 8xx number, can I ask my SIP provider to
> include more headers to show the real ANI? What would that service be
> called?
>
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> https://www.linkedin.com/in/steve-edwards-4244281
>
>
For dangerous material sites a call back was used. They call in and get a
code, the system calls back and asks for the code. Convoluted yes, the call
back was all that was really needed to thwart the fraud. A simple RFID pad
setup could be built to use low usage GSM plan to tag in the RFID on site.
But this is beyond the scope of telephony.

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Re: [asterisk-users] Turn on SIP debugging from DialPlan

[Derek Andrew]

Perfect, exactly what I needed. Thanks.

On Mon, Feb 27, 2017 at 6:32 AM, Igor Zamocky <i...@zamocky.sk> wrote:

> Hi,
>
> If you are ok with starting debug via external system call, why not to use
> something like this (I used to use something similar, it worked):
>
> exten =>* _XXX*,1,System(/usr/sbin/asterisk -rx ‘sip set debug peer *PEER*
> ’)
> same => n,Set(debug_on=1)
> same => n,Dial(SIP/*PEER*/${EXTEN})
>
> exten => *h*,1,GotoIf($[${debug_on} == 1]?undebug)
> same => n,Hangup
> same => n(undebug),System(( sleep 3 \; /usr/sbin/asterisk -rx 'sip set
> debug off' ) &)
> same => n,Set(debug_on=0)
> same => n,Hangup
>
> I don’t know your setup, your dialplan logic, but I’m sure you can adapt
> it to your needs.
>
> I.
>
> On 18 Feb 2017, at 00:26, Rafael dos Santos Saraiva <rafaels...@gmail.com>
> wrote:
>
> Hi
>
> I don't know if works, but you can try this:
>
> System(tcpdump -nq -s 0 -i eth0 -w /tmp/sip.pcap port 5060
> or udp portrange 1-2 &);
> Wait(1);
> Dial(SIP/${EXTEN});
> System(pkill tcpdump);
> Hangup;
>
> Or whitout RTP:
>
> System(tcpdump -nq -s 0 -i eth0 -w /tmp/sip.pcap port 5060
> &);
> Wait(1);
> Dial(SIP/${EXTEN});
>     System(pkill tcpdump);
> Hangup;
>
> Probably the last messages of SIP will be lost, BYE for example.
>
>
>
>
>
> 2017-02-17 20:43 GMT-02:00 Derek Andrew <derek.and...@usask.ca>:
>
>> I have some troublesome numbers that I would like to capture the SIP
>> dialogue when I am calling them. When I am about to dial the number, is
>> there any way to turn on SIP debugging in the dial plan before I make the
>> call? (and turn it off after the call is completed?)
>>
>>
>>
>>
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>>
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>>
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>
>
>
> --
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> Rafael Saraiva
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Re: [asterisk-users] Turn on SIP debugging from DialPlan

[Derek Andrew]

Yes, I agree. Tcpdump is one of my favourite programs. I need to enable it
and disable it from the dialplan though.



On Fri, Feb 17, 2017 at 5:18 PM, Tim Pozar <po...@lns.com> wrote:

> You can tell it to just capture SIP traffic and not the RTP traffic.
> Nice write up of using TCPdump and wireshark can be found here:
>
> https://blog.flowroute.com/2014/04/10/how-to-capture-sip-packets/
>
> BTW, I have found this works really well in trying to debug RTP traffic
> as well.  Wireshark just does the right thing in putting audio back
> together.  Very helpful in tracking down in and out of band DTMF
> problems that we were having with various carriers.
>
> Tim
>
> On 2/17/17 3:07 PM, Derek Andrew wrote:
> > The SIP trace will be adequate but this is on a remote system with
> > limited disk space.
> >
> > I would love to turn on debugging while making the troublesome calls,
> > then turn it off afterward.
> >
> > Tcpdump is great, but starting it and stopping it and keeping all that
> > data would still be an issue.
> >
> > d
> >
> > On Fri, Feb 17, 2017 at 4:56 PM, Tim Pozar <po...@lns.com
> > <mailto:po...@lns.com>> wrote:
> >
> > Why not capture the packets with something like tcpdump and run it
> > through Wireshark?
> >
> > Tim
> >
> > On 2/17/17 2:43 PM, Derek Andrew wrote:
> > > I have some troublesome numbers that I would like to capture the
> SIP
> > > dialogue when I am calling them. When I am about to dial the
> > number, is
> > > there any way to turn on SIP debugging in the dial plan before I
> make
> > > the call? (and turn it off after the call is completed?)
> > >
> > >
> > >
> > >
> > >
> >
> > --
> > 
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> >
> >
> >
> > --
> > Copyright 2017 Derek Andrew (excluding quotations)
> >
> > +1 306 966 4808
> > Communication and Network Services
> > Information and Communications Technology
> > Infrastructure Services
> > *University of Saskatchewan
> > *Peterson 120; 54 Innovation Boulevard
> > Saskatoon,Saskatchewan,Canada. S7N 2V3
> > Timezone GMT-6
> >
> > Typed but not read.
> >
> >
> >
> >
> >
>
> --
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Re: [asterisk-users] Turn on SIP debugging from DialPlan

[Derek Andrew]

But how do you turn on the debugging from the dialplan? What would be cool
is:

same => n,TURN ON DEBUGGING



On Fri, Feb 17, 2017 at 5:09 PM, Victor Villarreal <mefhigos...@gmail.com>
wrote:

> Hi Derek,
>
> SIP debug can be enabled via Asterisk CLI (console) with the command:
>
> asterisk> sip set debug on
>
> If you know via what trunk your call goes, you can use the following
> command instead:
>
> asterisk> sip set debug ip xxx.xxx.xxx.xxx
>
> Where the xxx is the IP of your trunk (voip to pstn provider).
>
> Affter you make all your test, simply issue:
>
> asterisk> sip set debug off
>
> And all the SIP conversation are saved in your full log file.
>
> More info here:
>
> https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
>
> If what you want is test your dialplan, simply use the command:
>
> asterisk> dialplan show xxx@your_context
>
> Where xxx is the number you want to dial, from the context asigned to your
> extension.
>
> Cheers
>
>
> El 17/2/2017 19:44, "Derek Andrew" <derek.and...@usask.ca> escribió:
>
>> I have some troublesome numbers that I would like to capture the SIP
>> dialogue when I am calling them. When I am about to dial the number, is
>> there any way to turn on SIP debugging in the dial plan before I make the
>> call? (and turn it off after the call is completed?)
>>
>>
>>
>>
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>>
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Re: [asterisk-users] Turn on SIP debugging from DialPlan

[Derek Andrew]

The SIP trace will be adequate but this is on a remote system with limited
disk space.

I would love to turn on debugging while making the troublesome calls, then
turn it off afterward.

Tcpdump is great, but starting it and stopping it and keeping all that data
would still be an issue.

d

On Fri, Feb 17, 2017 at 4:56 PM, Tim Pozar <po...@lns.com> wrote:

> Why not capture the packets with something like tcpdump and run it
> through Wireshark?
>
> Tim
>
> On 2/17/17 2:43 PM, Derek Andrew wrote:
> > I have some troublesome numbers that I would like to capture the SIP
> > dialogue when I am calling them. When I am about to dial the number, is
> > there any way to turn on SIP debugging in the dial plan before I make
> > the call? (and turn it off after the call is completed?)
> >
> >
> >
> >
> >
>
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> org/
>
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>
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>http://lists.digium.com/mailman/listinfo/asterisk-users
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[asterisk-users] Turn on SIP debugging from DialPlan

[Derek Andrew]

I have some troublesome numbers that I would like to capture the SIP
dialogue when I am calling them. When I am about to dial the number, is
there any way to turn on SIP debugging in the dial plan before I make the
call? (and turn it off after the call is completed?)
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Re: [asterisk-users] Conference Call like conference done by mobile!

[Andrew Ruthven]

On Wed, 2016-10-05 at 17:34 +0530, Mandar Khire wrote:
> hi,
> I trying to solve one scenario:-
> As I can make call from mobile phone to my friend1. As he accept it,
> I put him on hold, & dial friend2.
> As he also accept it, I put him on hold & follow same procedure till
> friend6.
> The I click on 'Merge call' & I can talk to all 6 friends at a time &
> they can talk each other.
> Can I write This scene by dialplan?How?
> I used Confbridge but its different type of conference.
> Need help.
> Thanks.

Hi Mandar,

Check out the "addcaller" stuff here:

https://wiki.asterisk.org/wiki/display/AST/ConfBridge+Configuration

Essentially you'd have a dialplan where you can call another number
which is then added to the confbridge.

Cheers,
Andrew

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Re: [asterisk-users] Sorcery with templates

[Andrew Ivins]

Thanks for confirming that Joshua. Thought that might be the case. I'll
look at a workaround.

Andrew

On 1 October 2016 at 20:20, Joshua Colp <jc...@digium.com> wrote:

> Andrew Ivins wrote:
>
>> Hi list.
>>
>> I use sorcery to configure an astdb backend to my pjsip endpoints.
>>
>> This works well, but it would be even better if I could set attribute
>> defaults for these endpoints in the config file. The way I do it now
>> forces me to store all endpoint attributes for each endpoint, even when
>> most of them are effectively defaults. If I need to change one, I need
>> to update each endpoint in the astdb.
>>
>>
>> Any suggestions?
>>
>
> There's no way to do this. Templates are purely a config file construct
> (in fact sorcery and other stuff has no idea you've used templates by the
> time it gets the data).
>
> Cheers,
>
> --
> 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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[asterisk-users] Sorcery with templates

[Andrew Ivins]

Hi list.

I use sorcery to configure an astdb backend to my pjsip endpoints.

This works well, but it would be even better if I could set attribute
defaults for these endpoints in the config file. The way I do it now forces
me to store all endpoint attributes for each endpoint, even when most of
them are effectively defaults. If I need to change one, I need to update
each endpoint in the astdb.


Any suggestions?

Andrew
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Re: [asterisk-users] Asterisk Radius CDR

[Andrew Ivins]

You won't see anything in the Asterisk logs because there's nothing to log.
The error happens in the freeradius-client library and returns an integer.
On 29 Sep 2016 17:44, "Willem Offermans" <wil...@offermans.rompen.nl> wrote:

Hello Andrew and asterisk friends,

I suspect that asterisk has problems to deal with the radiusclient in some
way. Therefore it cannot contact the radius server. There should be some
clue in the log files of asterisk, other than ``Unable to create RADIUS
record. CDR not recorded''

As a last resort, extra debug info from the source code can be invoked by
printf commands. But this involves some work.

On Thu, Sep 29, 2016 at 04:38:06PM +0800, Andrew Ivins wrote:
> You don't get anything in the Asterisk logs because freeradius-client
> (formerly radiusclient-ng) returns a single failure code for any failure
> when building a radius request.
>
> Andrew
>
> On 29 September 2016 at 15:55, Willy Offermans <
aster...@offermans.rompen.nl
> > wrote:
>
> > Hi Ahmed and asterisk friends,
> >
> > So asterisk cannot contact the radius server.
> >
> > The radiusclient __can__ contact the radius server.
> >
> > Check in the asterisk log files why asterisk cannot contact the radius
> > server! Be also aware of the user, who is running the daemons. This user
> > might need read access to certain configuration files.
> >
> > On Wed, Sep 28, 2016 at 01:24:58PM -0400, Ahmed Munir wrote:
> > > Hi Andrew and Willy,
> > >
> > > Thanks for sharing the info.
> > >
> > > As for enabling radius server debugging 'radiusd -X', made some test
> > calls
> > > don't see the radiusclient sending data to radius server. However,
using
> > > radtest or radiusclient testing, able to send data to radius server
> > (after
> > > enabling debug).
> > >
> > > For further testing, on my other server  using OpenSIPs, setup the
> > > radiusclient  and data was able to send over to radius server without
any
> > > issue i.e. using same radiusclient config that I'm using for Asterisk
> > > radiusclient.
> > >
> > > Btw, will try to work on Andrew advise and will update you if I make
any
> > > progress.
> > >
> > >
> > >
> > > Date: Wed, 28 Sep 2016 10:09:51 +0200
> > > > From: Willy Offermans <aster...@offermans.rompen.nl>
> > > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > > <asterisk-users@lists.digium.com>
> > > > Subject: Re: [asterisk-users] Asterisk Radius CDR
> > > > Message-ID: <20160928080951.ga4...@vpn.offrom.nl>
> > > > Content-Type: text/plain; charset=us-ascii
> > > >
> > > > Hello Ahmed, Andrew, and asterisk friends,
> > > >
> > > > Some time ago, I ran into similar problems as well :) I can confirm
the
> > > > statement of Andrew: Turn on the logging facilities and you will
find
> > your
> > > > issue most likely.  However, you need also a strategy. ``Radius
client
> > > > testing'' as you mentioned, can mean anything. The point is, can
> > asterisk
> > > > talk to the freeradius server via the client settings? To my
opinion,
> > this
> > > > is easy to test. Maybe the message: ``cdr_radius.c:208 radius_log:
> > Unable
> > > > to create RADIUS record. CDR not recorded'' already implies that
this
> > is
> > > > not possible. I cannot judge it. You can by turning on radiusd -X
and
> > have
> > > > a close look to the output.
> > > >
> > > > On Wed, Sep 28, 2016 at 07:59:13AM +0800, Andrew Ivins wrote:
> > > > > Hi Ahmed,
> > > > >
> > > > > I ran into similar problems. freeradius-client returns the same
error
> > > > code
> > > > > for numerous failure cases, so Asterisk doesn't get an opportunity
> > to log
> > > > > anything useful. If you look here:
> > > > >
> > > > > https://github.com/FreeRADIUS/freeradius-client/blob/master/
> > > > lib/buildreq.c
> > > > >
> > > > > You'll see many instances where it returns ERROR_RC. You are
almost
> > > > > certainly running into one of these. I ended up putting in print
> > debug
> > > > into
> > > > > that file and recompiling. I think in my case it was as simple as
a
> > > > > hostname not resolving. Once you're not working blind, you'll find
> > what
> > > > is
> > > >

Re: [asterisk-users] Asterisk Radius CDR

[Andrew Ivins]

You don't get anything in the Asterisk logs because freeradius-client
(formerly radiusclient-ng) returns a single failure code for any failure
when building a radius request.

Andrew

On 29 September 2016 at 15:55, Willy Offermans <aster...@offermans.rompen.nl
> wrote:

> Hi Ahmed and asterisk friends,
>
> So asterisk cannot contact the radius server.
>
> The radiusclient __can__ contact the radius server.
>
> Check in the asterisk log files why asterisk cannot contact the radius
> server! Be also aware of the user, who is running the daemons. This user
> might need read access to certain configuration files.
>
> On Wed, Sep 28, 2016 at 01:24:58PM -0400, Ahmed Munir wrote:
> > Hi Andrew and Willy,
> >
> > Thanks for sharing the info.
> >
> > As for enabling radius server debugging 'radiusd -X', made some test
> calls
> > don't see the radiusclient sending data to radius server. However, using
> > radtest or radiusclient testing, able to send data to radius server
> (after
> > enabling debug).
> >
> > For further testing, on my other server  using OpenSIPs, setup the
> > radiusclient  and data was able to send over to radius server without any
> > issue i.e. using same radiusclient config that I'm using for Asterisk
> > radiusclient.
> >
> > Btw, will try to work on Andrew advise and will update you if I make any
> > progress.
> >
> >
> >
> > Date: Wed, 28 Sep 2016 10:09:51 +0200
> > > From: Willy Offermans <aster...@offermans.rompen.nl>
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > <asterisk-users@lists.digium.com>
> > > Subject: Re: [asterisk-users] Asterisk Radius CDR
> > > Message-ID: <20160928080951.ga4...@vpn.offrom.nl>
> > > Content-Type: text/plain; charset=us-ascii
> > >
> > > Hello Ahmed, Andrew, and asterisk friends,
> > >
> > > Some time ago, I ran into similar problems as well :) I can confirm the
> > > statement of Andrew: Turn on the logging facilities and you will find
> your
> > > issue most likely.  However, you need also a strategy. ``Radius client
> > > testing'' as you mentioned, can mean anything. The point is, can
> asterisk
> > > talk to the freeradius server via the client settings? To my opinion,
> this
> > > is easy to test. Maybe the message: ``cdr_radius.c:208 radius_log:
> Unable
> > > to create RADIUS record. CDR not recorded'' already implies that this
> is
> > > not possible. I cannot judge it. You can by turning on radiusd -X and
> have
> > > a close look to the output.
> > >
> > > On Wed, Sep 28, 2016 at 07:59:13AM +0800, Andrew Ivins wrote:
> > > > Hi Ahmed,
> > > >
> > > > I ran into similar problems. freeradius-client returns the same error
> > > code
> > > > for numerous failure cases, so Asterisk doesn't get an opportunity
> to log
> > > > anything useful. If you look here:
> > > >
> > > > https://github.com/FreeRADIUS/freeradius-client/blob/master/
> > > lib/buildreq.c
> > > >
> > > > You'll see many instances where it returns ERROR_RC. You are almost
> > > > certainly running into one of these. I ended up putting in print
> debug
> > > into
> > > > that file and recompiling. I think in my case it was as simple as a
> > > > hostname not resolving. Once you're not working blind, you'll find
> what
> > > is
> > > > happening pretty quickly.
> > > >
> > > > Andrew
> > > >
> > > > On 28 September 2016 at 03:32, Ahmed Munir <ahmedmunir...@gmail.com>
> > > wrote:
> > > >
> > > > > I did radius client status testing with radius server, able to
> access
> > > the
> > > > > radius server. However, still getting radius CDR issue after
> setting
> > > debug
> > > > > level 8 even granting 666 access to radiusclient-ng config files.
> > > > >
> > > > > message: cdr_radius.c:208 radius_log: Unable to create RADIUS
> record.
> > > CDR
> > > > > not recorded!
> > > > >
> > > > > Please advise if I missed out anything.
> > > > >
> > > > >
> > > > > Date: Mon, 26 Sep 2016 12:09:34 +0200
> > > > >> From: Willy Offermans <aster...@offermans.rompen.nl>
> > > > >> To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > > >> <asteri

Re: [asterisk-users] Asterisk Radius CDR

[Andrew Ivins]

Hi Ahmed,

I ran into similar problems. freeradius-client returns the same error code
for numerous failure cases, so Asterisk doesn't get an opportunity to log
anything useful. If you look here:

https://github.com/FreeRADIUS/freeradius-client/blob/master/lib/buildreq.c

You'll see many instances where it returns ERROR_RC. You are almost
certainly running into one of these. I ended up putting in print debug into
that file and recompiling. I think in my case it was as simple as a
hostname not resolving. Once you're not working blind, you'll find what is
happening pretty quickly.

Andrew

On 28 September 2016 at 03:32, Ahmed Munir <ahmedmunir...@gmail.com> wrote:

> I did radius client status testing with radius server, able to access the
> radius server. However, still getting radius CDR issue after setting debug
> level 8 even granting 666 access to radiusclient-ng config files.
>
> message: cdr_radius.c:208 radius_log: Unable to create RADIUS record. CDR
> not recorded!
>
> Please advise if I missed out anything.
>
>
> Date: Mon, 26 Sep 2016 12:09:34 +0200
>> From: Willy Offermans <aster...@offermans.rompen.nl>
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> <asterisk-users@lists.digium.com>
>> Subject: Re: [asterisk-users] Asterisk Radius CDR
>> Message-ID: <20160926100934.gb4...@vpn.offrom.nl>
>> Content-Type: text/plain; charset=us-ascii
>>
>>
>> Hello Ahmed,
>>
>> On Fri, Sep 23, 2016 at 04:12:42PM -0400, Ahmed Munir wrote:
>> > Hi,
>> >
>> > I've recently setup Asterisk with Radius CDR by following the document:
>> > https://wiki.asterisk.org/wiki/display/AST/RADIUS+CDR+Backend.
>> >
>> > The issue currently I'm facing is after turning on the debug getting
>> > message: cdr_radius.c:208 radius_log: Unable to create RADIUS record.
>> CDR
>> > not recorded!
>> >
>> > I've checked and grant access 666 to radiusclient config files: servers
>> &
>> > dictionary.digium and 777 to '/var/run/radius.seq'. I've noticed that
>> > /var/run/radius.seq is not getting updated.
>> >
>> >
>> > Further added, in asterisk CLI while running command: cdr show status
>> > getting results below;
>> >
>> > Call Detail Record (CDR) settings
>> > --
>> >   Logging:Enabled
>> >   Mode:   Simple
>> >   Log unanswered calls:   No
>> >   Log congestion: No
>> >
>> > * Registered Backends
>> >   ---
>> > cdr-syslog
>> > Adaptive ODBC
>> > cdr-custom
>> > csv
>> > radius
>> >
>> >
>> > Please advise if I may missed any steps.
>> >
>> > --
>> > Regards,
>> >
>> > Ahmed Munir Chohan
>>
>> I cannot advice you about steps you might have missed, probably none. To
>> my
>> experience, the documentation is not sufficient.
>>
>> I can tell you that freeradius can be run in debug mode: radiusd -X Do
>> this
>> and have a close look to the output.
>>
>> If you cannot find any attempt to connect to the freeradius server you
>> need
>> to have a close look to the asterisk log files as well. Figure out what is
>> going wrong. There should be some clue.
>>
>> I don't understand the grant access settings. Figure out the user which is
>> running asterisk and set the setting appropriately! I remember that I
>> needed the following access setting:
>>
>> -rw-r-  1 root  asterisk  /usr/local/etc/radiusclient-ng/servers
>>
>> So read access for asterisk to the servers file. This was not documented
>> at
>> all, but somehow logical, if you figured it out.
>>
>> --
>> Met vriendelijke groeten,
>> With kind regards,
>> Mit freundlichen Gruessen,
>> De jrus wah,
>>
>> Wiel
>>
>> *
>>  W.K. Offermans
>>
>>Powered by 
>>
>> (__)
>>  \\\'',)
>>\/  \ ^
>>.\._/_)
>>
>>www.FreeBSD.org
>>
>>
>>
> --
> Regards,
>
> Ahmed Munir Chohan
>
>
> --
> _
> -- Bandwidth and Colocation Provi

Re: [asterisk-users] Multiple phones when one is unregistered

[Andrew Ruthven]

On Thu, 2016-09-01 at 10:36 -0700, Dave Platt wrote:
> > 
> Things can become more complicated in a couple of situations:
> 
> (1) If one of the SIP users you specify isn't actually a SIP
> endpoint device, but is a SIP identity on another system (PBX
> or VoIP provider or etc.), then you really don't have any control
> over how that endpoint would handle situations where the called
> user isn't available.  The endpoint might answer with *its*
> voicemail, immediately.
> 
> (2) If you were to dial a Local/ destination rather than a SIP/
> destination, then that dialing operation *is* run back through
> your dialplan, and it might divert the call to voicemail
> instantly.

Another option is what I've had happen recently. I have my main number
dial all the phones in my house, including an old Cisco 7905 that on
busy or no answer would send back a 302 redirect to extension 8000 -
VoiceMail. To make matters worse inbound callers would be dumped into
VoiceMail as though they'd entered it from internally, rather than
external.

While I tried various different ways on the Cisco to stop that
behaviour, I found the only solution was to tell the Dial() command to
ignore the 302 by adding the i flag. Problem solved.

Cheers,
Andrew

-- 
Andrew Ruthven, Wellington, New Zealand
MIITP, ITCP

At work: andrew.ruth...@catalyst.net.nz
At home: and...@etc.gen.nz
Cloud  : NZs only real cloud - https://catalyst.net.nz/cloud
GPG fpr: C603 FC4E 600F 1CEC D1C8  D97C 4B53 D931 E4D3 E863
LCA2017: The Future of Open Source, Hobart, AU - http://linux.conf.au

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Re: [asterisk-users] Calls are dropped after 15 minutes

[Andrew Colin]

 I had a similar issue and i set a timeout which fixed the issue
SIP/trunk/ ${EXTEN},216,t

We only had this on one of our providers the rest we havent had the issue

- Original Message -
From: Steve Edwards 
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Sent: Sat, 30 Jul 2016 20:27:45 +0200 (SAST)
Subject: Re: [asterisk-users] Calls are dropped after 15 minutes

On Sat, 30 Jul 2016, Keith Heppner wrote:

> We have a problem in that calls are dropped after 15 minutes (on both 
> internal and out going calls, incoming calls do not seem to have that 
> limit) How do we fix it?

You may gain some insight from viewing the console output after bumping up 
the debug and verbose levels.

You will probably resolve this by using tcpdump to capture packets and 
wireshark to see what's happening.

I had a problem with a similar description that was resolved by refusing 
SIP session timers.

-- 
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 - Allow * for only some users

[Andrew Ruthven]

Hi John,

Ah ha!  Excellent. That works.

Now for a further tweak, in my stdexten I set voicemail_option with
with b or u, as appropriate and use ${voicemail_option) instead of
option in the call to Voicemail below so the correct prompt is used.

Thank you!

On Thu, 2016-07-21 at 14:53 -0700, John Kiniston wrote:
> I think you almost have it.
> 
> In your vmfwd context have a wildcard match that sends the caller
> back to the originating voicemail and then define specific extensions
> that are allowed to forward.
> 
> 
> [vmfwd]
> exten => _,1,Voicemail(box@context,option)
>  same =>  n,Hangup
> 
> ; Andrew Ruthven
> exten => 7231,1,Set(CALLERID(number)=yyy)
> same => n,Goto(pstn,xxx,1)
> 
> On Thu, Jul 21, 2016 at 2:23 PM, Andrew Ruthven <andrew.ruthven@catal
> yst.net.nz> wrote:
> > Hey,
> > 
> > I have free calling to between DDIs and cellphones on our group
> > plan. I
> > figure it'd be nice to allow staff with those cellphones to be able
> > to
> > forward callers to their VoiceMail to their cellphones using the *
> > feature.
> > 
> > I have a standard extension macro that has VoiceMail support.
> > So far I've done this by duplicating the standard extension macro,
> > and
> > adding this rule (where ARG1 is the extension):
> > 
> >   exten => a,1,Goto(vmfwd,${ARG1},1)
> > 
> > Then in the vmfwd context I have rules like this (I need to set the
> > CALLERID(number) so our SIP provider accepts the call):
> > 
> >   ; Andrew Ruthven
> >   exten => 7231,1,Set(CALLERID(number)=yyy)
> >   exten => 7231,n,Goto(pstn,xxx,1)
> > 
> > Which is working nicely. But, I thought, can I simplify this and
> > just
> > have one macro?
> > 
> > So I've tried doing the following to fold it into my standard
> > extension
> > macro:
> > 
> > 1) Tried using a/_7231 but that didn't match (well, it was worth a
> > try)
> > 2) exten => a,1,Goto(vmfwd,${ARG1},1) works for calls to my
> > extension,
> > but if I disable the 7231 rules in vmfwd, I get:
> > 
> >   [2016-07-22 09:01:07.691] WARNING[11488][C-0420]: pbx.c:6646
> > __ast_pbx_run: Channel 'SIP/192.168.43.254-005a' sent to
> > invalid
> > extension but no invalid handler:
> > context,exten,priority=vmfwd,7231,1
> > 
> >   and the call hangs up, not a very nice user experience.
> > 
> > The second option could work, as long as the user lands back into
> > VoiceMail if there is no valid extension. I thought about using
> > GoSub,
> > but how do I get the caller back into VoiceMail?
> > 
> > I've done a bunch of searching for this, but haven't found any
> > general
> > solutions. Is it possible to do what I'm trying to achieve, or is
> > there
> > a better approach?
> > 
> > This is Asterisk 11.13.
> > 
> > Cheers,
> > Andrew
> > 
> > --
> > 
> > Andrew Ruthven, Wellington, New Zealand
> > MIITP, CITPNZ
> > 
> > At work: andrew.ruth...@catalyst.net.nz
> > At home: and...@etc.gen.nz
> > Card   : http://qr.catalyst.net.nz/907675e1
> > Cloud  : NZs only real cloud - https://catalyst.net.nz/cloud
> > GPG fpr: C603 FC4E 600F 1CEC D1C8  D97C 4B53 D931 E4D3 E863
> > LCA2016: LCA By the Bay, Geelong, AU - lca2016.linux.org
> > 
> > 
> > 
> > 
> > 
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> 
> 
> 
-- 

Andrew Ruthven, Wellington, New Zealand
MIITP, CITPNZ

At work: andrew.ruth...@catalyst.net.nz
At home: and...@etc.gen.nz
Card   : http://qr.catalyst.net.nz/907675e1
Cloud  : NZs only real cloud - https://catalyst.net.nz/cloud
GPG fpr: C603 FC4E 600F 1CEC D1C8  D97C 4B53 D931 E4D3 E863
LCA2016: LCA By the Bay, Geelong, AU - lca2016.linux.org





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[asterisk-users] VoiceMail - Allow * for only some users

[Andrew Ruthven]

Hey,

I have free calling to between DDIs and cellphones on our group plan. I
figure it'd be nice to allow staff with those cellphones to be able to
forward callers to their VoiceMail to their cellphones using the *
feature.

I have a standard extension macro that has VoiceMail support.
So far I've done this by duplicating the standard extension macro, and
adding this rule (where ARG1 is the extension):

  exten => a,1,Goto(vmfwd,${ARG1},1)

Then in the vmfwd context I have rules like this (I need to set the
CALLERID(number) so our SIP provider accepts the call):

  ; Andrew Ruthven
  exten => 7231,1,Set(CALLERID(number)=yyy)
  exten => 7231,n,Goto(pstn,xxx,1)

Which is working nicely. But, I thought, can I simplify this and just
have one macro?

So I've tried doing the following to fold it into my standard extension
macro:

1) Tried using a/_7231 but that didn't match (well, it was worth a try)
2) exten => a,1,Goto(vmfwd,${ARG1},1) works for calls to my extension,
but if I disable the 7231 rules in vmfwd, I get:

  [2016-07-22 09:01:07.691] WARNING[11488][C-0420]: pbx.c:6646
__ast_pbx_run: Channel 'SIP/192.168.43.254-005a' sent to invalid
extension but no invalid handler: context,exten,priority=vmfwd,7231,1

  and the call hangs up, not a very nice user experience.

The second option could work, as long as the user lands back into
VoiceMail if there is no valid extension. I thought about using GoSub,
but how do I get the caller back into VoiceMail?

I've done a bunch of searching for this, but haven't found any general
solutions. Is it possible to do what I'm trying to achieve, or is there
a better approach?

This is Asterisk 11.13.

Cheers,
Andrew

-- 

Andrew Ruthven, Wellington, New Zealand
MIITP, CITPNZ

At work: andrew.ruth...@catalyst.net.nz
At home: and...@etc.gen.nz
Card   : http://qr.catalyst.net.nz/907675e1
Cloud  : NZs only real cloud - https://catalyst.net.nz/cloud
GPG fpr: C603 FC4E 600F 1CEC D1C8  D97C 4B53 D931 E4D3 E863
LCA2016: LCA By the Bay, Geelong, AU - lca2016.linux.org





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Re: [asterisk-users] CALLERID on pjsip doesn't work?

[Andrew Ivins]

Thanks Joshua. That did the trick.

On 4 July 2016 at 19:18, Joshua Colp <jc...@digium.com> wrote:

> Andrew Ivins wrote:
>
>> On 1 July 2016 at 17:41, Joshua Colp <jc...@digium.com
>> <mailto:jc...@digium.com>> wrote:
>>
>>
>> exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>)
>> same => n,Dial(PJSIP/phone123, 30)
>>
>>
>> Your exten line has no priority, is that how it is in your dialplan?
>>
>>
>> Actually no, I stole that line from an earlier email to this list. Mine
>> has a priority.
>>
>> If not you can isolate things a bit further by trying the following:
>>
>> Set(CALLERID(all)=Jon Doe <+123456789>)
>>
>> Or individually:
>>
>> Set(CALLERID(name)=Jon Doe)
>> Set(CALLERID(num)=+123456789)
>>
>>
>> Tried many permutations of this, and the only thing I can get to happen
>> is to make the call present as Anonymous by changing the
>> pres-name/pres-num setting.
>>
>> It's not a production system, dialplan is pretty simple:
>>
>> same => _X.1,Set(CALLERID(name-pres)=allowed)
>> same => n,Set(CALLERID(num-pres)=allowed)
>> same => n,Set(CALLERID(name)=Fred)
>> same => n,Set(CALLERID(num)=6123)
>> same => n,Dial(PJSIP/DEADDEADBEEF, 30)
>> same => n,Hangup()
>>
>> DEADDEADBEEF is the name of the endpoint and the endpoint works. I use
>> MAC addresses and plan to dynamically map extensions to them later on
>> (kind of like user mode in freepbx).
>>
>> In the console, if I log the value of CALLERID, it is what I expect to
>> it to be.
>>
>>
> 
>
> You have from_user set which will override the user in the From header
> which is where callerid would be. You also don't have send_rpid or send_pai
> turned on so there would be no alternate way to send it. Try setting
> send_rpid or send_pai to yes and trying again.
>
>
> --
> 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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Re: [asterisk-users] CALLERID on pjsip doesn't work?

[Andrew Ivins]

On 1 July 2016 at 17:41, Joshua Colp  wrote:
>
>
>> exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>)
>> same => n,Dial(PJSIP/phone123, 30)
>>
>
> Your exten line has no priority, is that how it is in your dialplan?
>

Actually no, I stole that line from an earlier email to this list. Mine has
a priority.


> If not you can isolate things a bit further by trying the following:
>
> Set(CALLERID(all)=Jon Doe <+123456789>)
>
> Or individually:
>
> Set(CALLERID(name)=Jon Doe)
> Set(CALLERID(num)=+123456789)
>

Tried many permutations of this, and the only thing I can get to happen is
to make the call present as Anonymous by changing the pres-name/pres-num
setting.

It's not a production system, dialplan is pretty simple:

same => _X.1,Set(CALLERID(name-pres)=allowed)
same => n,Set(CALLERID(num-pres)=allowed)
same => n,Set(CALLERID(name)=Fred)
same => n,Set(CALLERID(num)=6123)
same => n,Dial(PJSIP/DEADDEADBEEF, 30)
same => n,Hangup()

DEADDEADBEEF is the name of the endpoint and the endpoint works. I use MAC
addresses and plan to dynamically map extensions to them later on (kind of
like user mode in freepbx).

In the console, if I log the value of CALLERID, it is what I expect to it
to be.

In the pjsip debug, the callerid I am trying to set doesn't appear anywhere.

I'm using your Sorcery stuff backing into astb for pjsip, but I've done a
little script to dump it back into text so I can override it in the config
file. Therefore it's a bit verbose. Thanks for looking.

[DEADDEADBEEF]
type=aor
support_path=true
default_expiration=3600
qualify_timeout=3.00
mailboxes=
minimum_expiration=60
outbound_proxy=
voicemail_extension=
maximum_expiration=7200
qualify_frequency=0
authenticate_qualify=false
contact=
max_contacts=1
remove_existing=true

[DEADDEADBEEF]
type=auth
md5_cred=
realm=
auth_type=userpass
password=4D7D9A7F1822
nonce_lifetime=32
username=507B495E565B

[DEADDEADBEEF]
type=endpoint
timers_sess_expires=1800
device_state_busy_at=0
dtls_cipher=
from_domain=
dtls_rekey=0
dtls_fingerprint=SHA-256
direct_media_method=invite
send_rpid=false
pickup_group=
sdp_session=Asterisk
dtls_verify=No
message_context=
mailboxes=
named_pickup_group=
record_on_feature=automixmon
dtls_private_key=
named_call_group=
t38_udptl_maxdatagram=0
media_encryption_optimistic=false
aors=DEADDEADBEEF
rpid_immediate=false
outbound_proxy=
identify_by=username
inband_progress=false
rtp_symmetric=false
transport=transport-udp
rtp_keepalive=0
t38_udptl_ec=none
fax_detect=false
t38_udptl_nat=false
allow_transfer=true
tos_video=0
srtp_tag_32=false
timers_min_se=90
call_group=
sub_min_expiry=0
100rel=yes
direct_media=true
rtp_timeout_hold=0
g726_non_standard=false
dtmf_mode=rfc4733
voicemail_extension=
rtp_timeout=0
dtls_cert_file=
media_encryption=no
media_use_received_transport=false
direct_media_glare_mitigation=none
trust_id_inbound=false
force_avp=false
record_off_feature=automixmon
send_diversion=true
language=
mwi_from_user=
rtp_ipv6=false
ice_support=false
callerid=unknown
aggregate_mwi=true
one_touch_recording=false
cos_video=0
accountcode=
allow=(g722|ulaw|alaw)
rewrite_contact=false
t38_udptl_ipv6=false
tone_zone=
user_eq_phone=false
allow_subscribe=true
rtp_engine=asterisk
auth=DEADDEADBEEF
from_user=DEADDEADBEEF
bind_rtp_to_media_address=false
disable_direct_media_on_nat=false
set_var=
use_ptime=false
outbound_auth=
media_address=
tos_audio=0
dtls_ca_path=
dtls_setup=active
force_rport=false
connected_line_method=invite
callerid_tag=
timers=yes
sdp_owner=-
trust_id_outbound=false
use_avpf=false
context=default
moh_suggest=default
send_pai=false
t38_udptl=false
dtls_ca_file=
callerid_privacy=allowed_not_screened
mwi_subscribe_replaces_unsolicited=false
cos_audio=0
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[asterisk-users] CALLERID on pjsip doesn't work?

[Andrew Ivins]

Asterisk 13.8

Is CALLERID(all) supposed to wok for pjsip? When I do this:

exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>)
same => n,Dial(PJSIP/phone123, 30)

I expect the callerid to be as set, but is always seems to be "phone123",
the name of the endpoint.

Andrew
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[asterisk-users] ARI Push Configuration and duplicate objects

[Andrew Ivins]

Hi,

Asterisk 13.8.0. Can anybody explain why I get two objects whenever I use
ARI Push? (https://wiki.asterisk.org/wiki/display/AST/ARI+Push+Configuration
)

Below is what I see using the auth push example from the wiki, but I get
the same thing for endpoints and aors too.

root@vagrant-ubuntu-wily-64:~# asterisk -r -x "pjsip show auths"
No objects found.

root@vagrant-ubuntu-wily-64:~# curl -X PUT -H "Content-Type:
application/json" -u asterisk:asterisk -d '{"fields": [ { "attribute":
"auth_type", "value": "userpass"}, {"attribute": "username", "value":
"alice"}, {"attribute": "password", "value": "secret" } ] }'
http://localhost:8088/ari/asterisk/config/dynamic/res_pjsip/auth/alice
[
  {
"attribute": "md5_cred",
"value": ""


root@vagrant-ubuntu-wily-64:~#asterisk -r -x "pjsip show auths"
  I/OAuth:
 
 
=

 Auth:  alice/alice
 Auth:  alice/alice
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Re: [asterisk-users] Voicemail using object storage?

[Andrew Ruthven]

I'd say using s3fs (or similar) is an approach, but if VoiceMail had
support baked into it for S3, then the integration would be better.

I'll look into using one the FUSE based approaches as a stop-gap
measure. ;)

On Tue, 2016-02-16 at 13:12 +0100, Olivier wrote:
> Isn't the purpose of s3fs-like addons (see [1]) to let S3 buckets be
> mounted on Linux and thus allow any application like Asterisk make
> use of it ?
> 
> [1] https://github.com/s3fs-fuse/s3fs-fuse
> 
> 2016-02-16 1:05 GMT+01:00 Andrew Ruthven <andrew.ruth...@catalyst.net
> .nz>:
> > Hey,
> > 
> > I've found a bit of chatter about people using hacks to copy
> > voicemail
> > messages into object storage (like S3) after they've been recorded.
> > But
> > I was wondering if any work has been done on the VoiceMail app to
> > actually store and retrieve messages to/from an object store?
> > 
> > Cheers,
> > Andrew
> > --
> > Andrew Ruthven, Wellington, New Zealand
> > MIITP, ITCP
> > 
> > At work: andrew.ruth...@catalyst.net.nz
> > At home: and...@etc.gen.nz
> > Cloud  : NZs only real cloud - https://catalyst.net.nz/cloud
> > GPG fpr: C603 FC4E 600F 1CEC D1C8  D97C 4B53 D931 E4D3 E863
> > LCA2016: LCA By the Bay, Geelong, AU - lca2016.linux.org
> > 
> > 
> > 
> > --
> > ___
> > __
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> > --
> > New to Asterisk? Join us for a live introductory webinar every
> > Thurs:
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> _
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-- 
Andrew Ruthven, Wellington, New Zealand
MIITP, ITCP

At work: andrew.ruth...@catalyst.net.nz
At home: and...@etc.gen.nz
Cloud  : NZs only real cloud - https://catalyst.net.nz/cloud
GPG fpr: C603 FC4E 600F 1CEC D1C8  D97C 4B53 D931 E4D3 E863
LCA2016: LCA By the Bay, Geelong, AU - lca2016.linux.org



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[asterisk-users] Voicemail using object storage?

[Andrew Ruthven]

Hey,

I've found a bit of chatter about people using hacks to copy voicemail
messages into object storage (like S3) after they've been recorded. But
I was wondering if any work has been done on the VoiceMail app to
actually store and retrieve messages to/from an object store?

Cheers,
Andrew
-- 
Andrew Ruthven, Wellington, New Zealand
MIITP, ITCP

At work: andrew.ruth...@catalyst.net.nz
At home: and...@etc.gen.nz
Cloud  : NZs only real cloud - https://catalyst.net.nz/cloud
GPG fpr: C603 FC4E 600F 1CEC D1C8  D97C 4B53 D931 E4D3 E863
LCA2016: LCA By the Bay, Geelong, AU - lca2016.linux.org



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[asterisk-users] Prodding channel Failed

[Andrew Colin]

Hi Guys

 

I am seeing this error a lot in the CLI lately

What does it mean?

Prodding channel SIP/XXX failed

 

 

 

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Re: [asterisk-users] Modify Contact in PJsip

[Andrew Colin]

Hi Joshua
If i put the default_user option per endpoint would it work? 
So what exactly does the contact_user option do?
I know that in freeswitch there is the option extension-in-contact.We  
basically need to achieve the same functionality 
Thanks

 Original message 
From: Joshua Colp <jc...@digium.com> Date: 
2015/10/19  13:03  (GMT+02:00) To: asterisk-users@lists.digium.com 
Subject: Re: [asterisk-users] Modify Contact in PJsip 
On 15-10-19 07:41 AM, Andrew Colin wrote:
> Hi Guys
>
> We are using the wizard to configure our pjsip trunk(see below)
>
> How do we get this setting to work
>
> contact_user=username
>
> We want to change the contact field in the sip invite to display the
> username of the trunk
>

The Contact header can not currently be modified on a per-endpoint basis 
and takes its values from the generated From header. On a global scale 
it could be controlled using the default_user global option. Otherwise 
there's no real way without adding explicit support for it.

-- 
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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Re: [asterisk-users] Modify Contact in PJsip

[Andrew Colin]

Do you know if this can be achieved with the standard sip stack in asterisk?


Kind Regards
Andrew Colin
Converged Telecoms (Pty) Ltd.
Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09)

Switchboard: +27 (0)10 591 4600
Email:  and...@convergedgroup.net
Web:  http://www.convergedgroup.net
75 Witkoppen Road, Northriding, Johannesburg, 2169
P O Box 7246, Weltevredenpark, 1715
This communication is confidential and intended solely for the addressee(s).
Any unauthorized review, use, disclosure or distribution is prohibited. If
you believe this message has been sent to you in error, please notify the
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data corruption, interception, unauthorized amendment, tampering and
viruses, and we only send and receive emails on the basis that we are not
liable for any such corruption, interception, amendment, tampering or
viruses or any consequences thereof.



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp
Sent: Monday, October 19, 2015 2:05 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Modify Contact in PJsip

On 15-10-19 08:17 AM, Andrew Colin wrote:
> Hi Joshua
>
> If i put the default_user option per endpoint would it work?

No, it's a global only option.

>
> So what exactly does the contact_user option do?

It sets the Contact user in an outbound registration so that the URI dialed
by the remote SIP server may contain that user (or may not, depending on
their configuration/deployment).

>
> I know that in freeswitch there is the option extension-in-contact.
> We  basically need to achieve the same functionality

It would require modifying the code and adding support.

-- 
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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Re: [asterisk-users] Modify Contact in PJsip

[Andrew Colin]

Ok thanks Joshua

Do you know what this error means when I dial out in pjsip and the call
fails

Unable to create request with auth.No auth credent als for any realms in
challenge





Kind Regards
Andrew Colin
Converged Telecoms (Pty) Ltd.
Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09)

Switchboard: +27 (0)10 591 4600
Email:  and...@convergedgroup.net
Web:  http://www.convergedgroup.net
75 Witkoppen Road, Northriding, Johannesburg, 2169
P O Box 7246, Weltevredenpark, 1715
This communication is confidential and intended solely for the addressee(s).
Any unauthorized review, use, disclosure or distribution is prohibited. If
you believe this message has been sent to you in error, please notify the
sender by replying to this transmission and delete the message without
disclosing it. Thank you. E-mail including attachments is susceptible to
data corruption, interception, unauthorized amendment, tampering and
viruses, and we only send and receive emails on the basis that we are not
liable for any such corruption, interception, amendment, tampering or
viruses or any consequences thereof.



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp
Sent: Monday, October 19, 2015 2:21 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Modify Contact in PJsip

On 15-10-19 09:12 AM, Andrew Colin wrote:
> Do you know if this can be achieved with the standard sip stack in
asterisk?

If you are referring to chan_sip I don't believe so but it is possible there
is some obscure option or method to do it that I am aware of.

--
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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[asterisk-users] Modify Contact in PJsip

[Andrew Colin]

Hi Guys

 

We are using the wizard to configure our pjsip trunk(see below)

How do we get this setting to work

contact_user=username

 

We want to change the contact field in the sip invite to display the
username of the trunk

 

[trunk_defaults](!)

type = wizard

transport = transport-udp

endpoint/allow_subscribe = no

endpoint/allow = !all,g729

aor/qualify_frequency = 30

registration/expiration = 1800

contact_pattern=xxx

 

[xxx](trunk_defaults)

sends_auth = yes

sends_registrations = yes

endpoint/context = extensions

remote_hosts = xxx.xx.xx.xx

accepts_registrations = no

endpoint/send_rpid = yes

endpoint/send_pai = yes

outbound_auth/username = xxx

outbound_auth/password = xxx

contact_pattern=xxx

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[asterisk-users] PJSIP Dialout error

[Andrew Colin]

Hi Guys

 

I keep getting this "Warning" when I dial out via pjsip and the calls fail

But if I do a pjsip reload it works for 1 minute

 

WARNING[6707]: res_pjsip_outbound_authenticator_digest.c:135
digest_create_request_with_auth_from_old: Unable to create request with
auth.No auth credentials for any realms in challenge.

 

Any ideas?

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Re: [asterisk-users] Storing HANGUPCAUSE in CDR

[Andrew Colin]

You can use this

 

exten => h,1,Set(CDR(userfield)=Hangupcause:${HANGUPCAUSE})

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ross Beer
Sent: Friday, October 9, 2015 1:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: Re: [asterisk-users] Storing HANGUPCAUSE in CDR

 

This was always possible in the past, however does not work in the current
release.
 
I believe this is a bug.
 

  _  

To: asterisk-users@lists.digium.com 

From: cerv...@fpf.slu.cz  
Date: Fri, 9 Oct 2015 10:04:47 +0200
Subject: Re: [asterisk-users] Storing HANGUPCAUSE in CDR

search in archives
save the records to another table like cdr_extended


Dne 7.10.2015 v 15:26 Ross Beer napsal(a):

Hi, 

 

I have the following code that operates when a channel is hung-up:

 

[record-hangupcause]

exten => 1,n,Set(CDR(hangupcause)=${HANGUPCAUSE})

exten => s,n,Return()

 

Before the dial a hangup handler is registered:

 

Set(CHANNEL(hangup_handler_push)=record-hangupcause,s,1)

 

The routine is called and the variables are being set, however not on the
channel's CDR which made the call. I believe this is due to the CDR being
closes as soon as the dial returns. 

 

By changing the cdr option 'endbeforehexten=no' this should keep the CDR
accessible, however all this does is cause another CDR to be created for the
'h' extension.

 

Is there a way to update the CDR so that a result can be stored per dial?

 

Thank you in advance,

 

Ross

 

 

 

 





 

-- 
---
Marek Cervenka
===


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[asterisk-users] Change Contact field in sip invite

[Andrew Colin]

Hi Guys

 

Does anyone know of a way I can change the contact field in the sip invite
to display sip:username:ip instead of sip:did:ip

We need to do this without changing the from field.

I tried using fromuser=username  but that just modifies both the contact and
the from parameter

 

I know in freeswitch they use the parameter extension-In-Contact

 

Has anyone managed to do this before?

 

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[asterisk-users] SIP Phones over VPN Drop Audio One-Way

[Andrew Martin]

Hello,

I am running Asterisk 11 on CentOS 6.x using the DAHDI module with 8x PSTN
analog phone lines for outside connectivity. Internally, I am using several
models of Yealink SIP phones (e.g SIP-T32G) on a dedicated VoIP network,
192.168.0.0/24. I have a few of these Yealink SIP phones configured with an 
OpenVPN certificate so that users working remotely can directly access the phone
system (VPN subnet is 192.168.1.0/24). Note that this is not a NAT; VPN clients 
are able to directly address the Asterisk server and other SIP phones. Last week
the phones connecting over the VPN started dropping audio during the call (e.g 
caller 1 can still hear caller 2, but not vise versa). These calls are between 
two SIP phones (one over the VPN, one internal). The dropouts last for 20 
seconds or more, and sometimes the audio does recover and come back.

I made some changes to the infrastructure last week, but I am not sure that they
are the cause. First, I added echotraining=yes to /etc/asterisk/chan_dahdi.conf
to try and fix echo problem (seems unrelated since the call is all SIP). I also
cleaned up some extraneous firewall rules on the OpenVPN gateway, but I still
allow the VPN phones to connect to the Asterisk server on ports 5000 - 2 for
SIP and RSTP so this also seems unrelated.

I've looked at the syslog on the SIP phones as well as the asterisk output with
sip set debug and rtp set debug on but I don't see anything obviously wrong.
The only sign of a problem I can see is this message when the call is hung up:
pbx.c:   == Spawn extension (dial-extension, 124, 1) exited non-zero on 
'SIP/123-01d9'

Here is an example user in my sip.conf: 
http://pastebin.com/6U2AhyWT

Do you have any ideas about what is causing these dropouts, or what I should
look at next for additional debug information?

Thanks,

Andrew Martin

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Re: [asterisk-users] Queues don't follow dialplan if no members are registered

[Andrew Martin]

- Original Message -
 From: John Kiniston johnkinis...@gmail.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Wednesday, July 29, 2015 11:53:13 AM
 Subject: Re: [asterisk-users] Queues don't follow dialplan if no members are  
 registered
 
 Wow, Looks like they have really increased the options since I last looked.
 
 I just pulled down the Asterisk 13 queues.conf.sample and it's got this in
 it:
 
 ; paused: a member is not considered available if he is paused
 ; penalty: a member is not considered available if his penalty is less than
 QUEUE_MAX_PENALTY
 ; inuse: a member is not considered available if he is currently on a call
 ; ringing: a member is not considered available if his phone is currently
 ringing
 ; unavailable: This applies mainly to Agent channels. If the agent is a
 member of the queue
 ; but has not logged in, then do not consider the member to be available
 ; invalid: Do not consider a member to be available if he has an invalid
 device state.
 ; This generally is caused by an error condition in the member's channel
 driver.
 ; unknown: Do not consider a member to be available if we are unable to
 determine the member's
 ; current device state.
 ; wrapup: A member is not considered available if he is currently in his
 wrapuptime after
 ; taking a call.
 
 An unknown state would be a device that has a valid configuration but isn't
 registered.
 
John,

Thanks for the clarification and your help resolving this issue!

Andrew

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Re: [asterisk-users] Queues don't follow dialplan if no members are registered

[Andrew Martin]

- Original Message -
 From: John Kiniston johnkinis...@gmail.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Tuesday, July 28, 2015 12:12:05 PM
 Subject: Re: [asterisk-users] Queues don't follow dialplan if no members are  
 registered
 
 In your queues.conf do you have a leavewhenempty and joinempty set?
 
 in queues.conf
 [myqueue]
 leavewhenempty = strict
 joinempty = strict
 strategy = ringall
 ringinuse = no
 
 

John,

Thanks for the fast reply! I had joinempty=yes in queues.conf,
which explains why I was seeing this behavior. It looks like the
strict setting is partially-deprecated, so instead I'm using
the following combination:

[myqueue]
musiconhold=default
music=default
strategy=ringall
joinempty=unavailable,invalid,unknown
leavewhenempty=unavailable,invalid,unknown
timeout=18

member = SIP/100
member = SIP/101

Is there any reason that using any of these options would be a
problem, in particular unknown? It is not very well defined
what an unknown state is exactly.

Thanks,

Andrew

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[asterisk-users] Queues don't follow dialplan if no members are registered

[Andrew Martin]

Hello,

I am running Asterisk 11 on CentOS 6.x. I have configured several queues as 
follows in extensions.conf:
exten = s,1,Queue(myqueue,rtnC,18)
same = n,Background(user_unavail)
same = n,WaitExten(10)
exten = 1,1,Voicemail(@my-vm,s)

This rings the phones in the queue for 18 seconds. If no queue members answer,
the caller is then prompted to press 1 and leave a voicemail. This works well
when at least 1 member is registered in the queue, however if no members are
registered in the queue, the Queue() call never seems to return, and thus the
remaining steps in the dialplan never execute. How can I correct this behavior
so that even if the queue has no registered members, the dialplan is still
followed?

Thanks,

Andrew

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Re: [asterisk-users] Call Return

[Andrew Colin]

Hi Aj

Can you perhaps show me an example as to how you would do it as I have tried 
setting it very early but still doesn’t work

Kind Regards

Andrew Colin

Converged Telecoms (Pty) Ltd.

Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09)


Switchboard: +27 (0)10 591 4600
Email: and...@convergedgroup.net

Web: http://www.convergedgroup.net
75 Witkoppen Road, Northriding, Johannesburg, 2169
P O Box 7246, Weltevredenpark, 1715
This communication is confidential and intended solely for the addressee(s). 
Any unauthorized review, use, disclosure or distribution is prohibited. If 
you believe this message has been sent to you in error, please notify the 
sender by replying to this transmission and delete the message without 
disclosing it. Thank you.E-mail including attachments is susceptible to data 
corruption, interception, unauthorized amendment, tampering and viruses, and 
we only send and receive emails on the basis that we are not liable for any 
such corruption, interception, amendment, tampering or viruses or any 
consequences thereof.

-Original Message-
From: A J Stiles [mailto:asterisk_l...@earthshod.co.uk]
Sent: Thursday, July 9, 2015 10:03 AM
To: Andrew Colin; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call Return

On Wednesday 08 Jul 2015, Andrew Colin wrote:
 Hi Guys



 I am trying to write a macro for a call return so for example

 Anyone in the company transfers a call to another extension and it is
 not answered etc it must return to the person who did the transfer

 I have got it working but if the call originates externally for
 example someone calls in to the switchboard and they transfer it then
 it tries to return to the outside caller.

 As doing a return to ${EXTEN}) wont work as that is the external party.

 How do I declare a variable from the extension dialed?
 So for example when 200 dials 201 I can capture the calling party(in
 this case 200) and declare it as a variable?

You need to set a variable quite early in your extension logic, using a Set 
command;

Set(dialled=${EXTEN})

and then later you can retrieve it as ${dialled} .  This variable will 
persist across context jumps, even although ${EXTEN} may have changed.

--
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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[asterisk-users] Call Return

[Andrew Colin]

Hi Guys

 

I am trying to write a macro for a call return so for example

Anyone in the company transfers a call to another extension and it is not
answered etc it must return to the person who did the transfer

I have got it working but if the call originates externally for example
someone calls in to the switchboard and they transfer it then it tries to
return to the outside caller.

 

As doing a return to ${EXTEN}) wont work as that is the external party.

How do I declare a variable from the extension dialed?

So for example when 200 dials 201 I can capture the calling party(in this
case 200) and declare it as a variable?

 

 

 

 

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Re: [asterisk-users] Product CDR/Queue/Meetme

[Andrew Colin]

Hi Helvio
I will be interested to test your product and give you some feedback. .



Sent from my Samsung Galaxy s6 smartphone.

 Original message 
From: Helvio Junior helvio.lis...@gmail.com 
Date: 29/06/2015  20:58  (GMT+02:00) 
To: Abdul Basit basit.e...@gmail.com, Asterisk Users Mailing List - 
Non-Commercial Discussion asterisk-users@lists.digium.com 
Subject: Re: [asterisk-users] Product CDR/Queue/Meetme 

1.8 or higher.

Att,
Hélvio Junior
SafeId - Gestão de identidades e Acessos
+55 41 | 9893-2694, single-sign-on.com.br
helvio.jun...@safetrend.com.br

On 29/06/2015 14:43, Abdul Basit wrote:
 Hi Helviom

 I am interested to evaluate your product.

 What asterisk version you build this product around?

 --
 regards,

 abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445

 On Tue, Jun 23, 2015 at 7:34 PM, Tech Support 
 aster...@voipbusiness.us mailto:aster...@voipbusiness.us wrote:

 Please keep the “me to” emails off the list.

 Regards;

 JV

 *From:*asterisk-users-boun...@lists.digium.com
 mailto:asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com
 mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of
 *Magno Guimarães
 *Sent:* Monday, June 22, 2015 3:55 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] Product CDR/Queue/Meetme

 Hello,





 I am interested, too.





 Att,

 Welinghton




 Citando Mitul Limbani mi...@enterux.in mailto:mi...@enterux.in:

 Hey Helvio,

 Would like to check it out as well.

 Do email me,

 Mitul

 On 22-Jun-2015 9:05 AM, Helvio Junior
 helvio.lis...@gmail.com mailto:helvio.lis...@gmail.com wrote:

 Gentleman,

 Moderators, i don't know if this topic if OFF-Topic, if yes,
 please tell me.

 I had some difficult looking for a Asterisk software that
 provide me some functions (For exemple: CDR, Queue control,
 MeetMe Control) all-in-one. So i decided to develop than.

 In a few weeks i'll deploy a Beta version of this software and
 i'd like to know if is somebody available to try this beta and
 free version?

 If you don't want to try this version but would like to
 see/suggest any feature in this software, let me know.

 Forecast functions to Beta Version:

   * Realtime view for:

   o Queues;
   o Peers (Similar as BLF);
   o Trunk calls/utilization;

   * MeetMe

   o Create, modify, delete and schedule;
   o Real time view of members;
   o Delete members;
   o Mute/Unmute;
   o Send Invite by e-mail (with .VCS file)

   * Dialer

   o Create dialer (by campaign with contacts)
   o Monitoring of campaig, calls, and status;
   o Time control to retry failed call
   o Control of day time to call (commercial time, full
 time, etc...)

   * Charts and reports:

   o Trunk utilization;
   o CDR;
   o Queues (Most common reports and charts, distributions,
 times, etc...)
   o Export to Excel Spreadsheet and PDF File
   o Report Scheduler
   o Much more...

   * REST API for 100% of functionalities;
   * Admin and User Console 100% Web HTML5;
   * Developed in Windows with C#;
   * Integrate with Asterisk using AMI only;
   * Allow manage many Asterisk that you want using same
 instance of this software (One software and one installation);


 Obs.: I'll provide a Full License for everybody that help me
 trying the Beta version.

   

 -- 

   

 Att,

 Hélvio Junior

 SafeId - Gestão de identidades e Acessos

 +55 41 | 9893-2694,single-sign-on.com.br  
 http://single-sign-on.com.br

 helvio.jun...@safetrend.com.br  
 mailto:helvio.jun...@safetrend.com.br

   

 -- 

   

 Att,

 Hélvio Junior

 SafeId - Gestão de identidades e Acessos

 +55 41 | 9893-2694,single-sign-on.com.br  
 http://single-sign-on.com.br

 helvio.jun...@safetrend.com.br  
 mailto:helvio.jun...@safetrend.com.br


 --
 _
 -- Bandwidth and Colocation Provided by
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 Thurs:
 http://www.asterisk.org/hello

 asterisk-users mailing list
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 http://lists.digium.com/mailman/listinfo/asterisk-users


 --
 

Re: [asterisk-users] Retransmission Timeout results in dropped calls after 32 seconds

[Andrew Martin]

- Original Message -
 From: Joshua Colp jc...@digium.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Tuesday, May 12, 2015 5:42:57 PM
 Subject: Re: [asterisk-users] Retransmission Timeout results in dropped 
 calls   after 32 seconds
 
 Andrew Martin wrote:
 
 snip
 
 
  Joshua,
 
  As a mitigation for this problem, could I increase the timerb option in
  sip.conf
  to a large value, say 1 hour (instead of the default 32 seconds)? What
  other
  consequences would there be from this change?
 
 I don't know if chan_sip will allow this, but if it does... it'll keep
 transmitting over and over... it would be better to get to the bottom of
 the problem. Do a packet capture on the machine running Asterisk and see
 where the packet goes. That's the only thing left really. It's also
 possible something got fixed in relation to directmedia between your
 version and latest 11.
 

Joshua,

Looking at the packet capture from the asterisk server during this time,
I see the following sequence of SIP packets:
INVITE (102) - initial call connecting
RINGING (102) - initial call connecting
RINGING (102) - initial call connecting
OK (102) - initial call connecting
ACK (102) - initial call connecting
OK (102) - initial call connecting (seems like a duplicate OK)
INVITE (103) - re-INVITE to go to bypass mode
ACK (102) - initial call connecting (seems like a duplicate ACK)
INVITE (103) - re-INVITE to go to bypass mode (retry #1)
INVITE (103) - re-INVITE to go to bypass mode (retry #2)
INVITE (103) - re-INVITE to go to bypass mode (retry #3)
INVITE (103) - re-INVITE to go to bypass mode (retry #4)
INVITE (103) - re-INVITE to go to bypass mode (retry #5)


Looking at the logs from the yealink phone (http://pastebin.com/aAWs4j6i),
I see a few differences:
INVITE (102) - initial call connecting
TRYING (102) - initial call connecting
RINGING (102) - initial call connecting
INVITE (102) - initial call connecting (seems like a duplicate INVITE)
RINGING (102) - initial call connecting
OK (102) - initial call connecting
ACK (102) - initial call connecting
INVITE (103) - re-INVITE to go to bypass mode
TRYING (103) - re-INVITE to go to bypass mode
OK (103) - re-INVITE to go to bypass mode
ACK (102) - initial call connecting (seems like a duplicate ACK)
ACK (102) -initial call connecting (seems like a duplicate ACK)
INVITE (103) - re-INVITE to go to bypass mode (retry #1)
ACK (102) -initial call connecting (seems like a duplicate ACK)
INVITE (103) - re-INVITE to go to bypass mode (retry #2)
INVITE (103) - re-INVITE to go to bypass mode (retry #3)
INVITE (103) - re-INVITE to go to bypass mode (retry #4)
INVITE (103) - re-INVITE to go to bypass mode (retry #5)
INVITE (103) - re-INVITE to go to bypass mode
INVITE (103) - re-INVITE to go to bypass mode


Most noteworthy is that the phone seems to send the OK for cseq 103, but it
seems that the asterisk server never received this OK, which is why it kept
re-transmitting the INVITE (103). Is this OK supposed to go to the asterisk
server, or to the other phone? If it is supposed to go to the asterisk server,
I suppose the explanation could be network turbulence prevented this OK from 
getting back to the server - does this seem like what happened? If so, what
should be happening differently to ensure that this call doesn't get dropped?

Thanks,

Andrew


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Re: [asterisk-users] Retransmission Timeout results in dropped calls after 32 seconds

[Andrew Martin]

- Original Message -
 From: Joshua Colp jc...@digium.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Wednesday, May 13, 2015 10:50:02 AM
 Subject: Re: [asterisk-users] Retransmission Timeout results in dropped 
 calls   after 32 seconds
 
 Andrew Martin wrote:
  Since some packet loss is a possibility, I assume the protocol has
  mechanisms
  for dealing with it. What should be happening differently in the
  communication
  when packet loss occurs? Should the phone just be re-sending the OK,
  instead of
  printing 0  | ERROR | receive a request with same cseq?? to its log? Or
  should
  Asterisk be starting with a new cseq on each INVITE retry?
 
 The 200 OK should be retransmitted until an ACK is received. It honestly
 looks like the phone can't talk to Asterisk and it's just generally
 screwing up signaling.
 

Thanks for the clarification and help debugging this problem. I will work
with the phone vendor to see if they can resolve this from their end. If you
have any other ideas about how to disable re-INVITEs on the asterisk side,
beyond what I have done already, please let me know.

Thanks,

Andrew

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Re: [asterisk-users] Retransmission Timeout results in dropped calls after 32 seconds

[Andrew Martin]

- Original Message -
 From: Joshua Colp jc...@digium.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Wednesday, May 13, 2015 10:10:25 AM
 Subject: Re: [asterisk-users] Retransmission Timeout results in dropped 
 calls   after 32 seconds
 
 Andrew Martin wrote:
  - Original Message -
 
 snip
 
 
 
  Most noteworthy is that the phone seems to send the OK for cseq 103, but it
  seems that the asterisk server never received this OK, which is why it kept
  re-transmitting the INVITE (103). Is this OK supposed to go to the asterisk
  server, or to the other phone? If it is supposed to go to the asterisk
  server,
  I suppose the explanation could be network turbulence prevented this OK
  from
  getting back to the server - does this seem like what happened? If so, what
  should be happening differently to ensure that this call doesn't get
  dropped?
 
 The traffic is between the phone and Asterisk. As to why, I have no
 idea. The packets aren't getting to Asterisk - that's all I can say. I
 doubt it's network turbulence. Likely getting lost/blocked somewhere.
 
Since some packet loss is a possibility, I assume the protocol has mechanisms
for dealing with it. What should be happening differently in the communication
when packet loss occurs? Should the phone just be re-sending the OK, instead of
printing 0 | ERROR | receive a request with same cseq?? to its log? Or 
should
Asterisk be starting with a new cseq on each INVITE retry?

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Re: [asterisk-users] Retransmission Timeout results in dropped calls after 32 seconds

[Andrew Martin]

- Original Message -
 From: Steve Davies davies...@gmail.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Wednesday, May 13, 2015 11:39:29 AM
 Subject: Re: [asterisk-users] Retransmission Timeout results in dropped 
 calls after 32 seconds
 
 Hi,
 
 In my experience, all Yealink phones work just fine with Asterisk, we have
 hundreds (perhaps even low-thousands) out there with customers on Asterisk
 1.2, 1.6.2, 1.8 and 11.
 
 If you are accurately representing the SIP trace on the phone and the SIP
 trace on Asterisk, then I would strongly suggest a SIP ALG exists in the
 network between the two devices and that SIP ALG does not understand SIP
 properly. The two halves simply do not match, so something must surely be
 interfering.
 
 In my experience it is often an innocent looking Cisco router. Cisco's SIP
 implementation is SIP By Cisco rather than RFC compliant SIP. If that is
 the case Cisco call it a SIP fixup and you just need to disable it.
 
 Hope that helps,
 Steve
 
Steve,

That is an interesting point - the server and the phone are both connected to
Netgear switches where I have enabled their Auto-VoIP feature, which remarks
packets based on protocol (SIP, SCCP, etc) for better QoS:
http://kb.netgear.com/app/answers/detail/a_id/21758

I wonder if this remarking process is modifying another part of the packet too?
Both devices are on the same subnet, so although these switches do route 
traffic as well, that shouldn't be coming into play here.

Andrew

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Re: [asterisk-users] Retransmission Timeout results in dropped calls after 32 seconds

[Andrew Martin]

- Original Message -
 From: Andrew Martin amar...@xes-inc.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Monday, May 11, 2015 4:18:58 PM
 Subject: Re: [asterisk-users] Retransmission Timeout results in dropped 
 calls   after 32 seconds
 
 - Original Message -
  From: Andrew Martin amar...@xes-inc.com
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  asterisk-users@lists.digium.com
  Sent: Monday, May 11, 2015 1:35:07 PM
  Subject: Re: [asterisk-users] Retransmission Timeout results in dropped
  calls   after 32 seconds
 
   That should be all that is required. If that were broken I'd expect
   issue reports to implode - what's the configuration?
   
  
  Here's the sip.conf (only showing a single extension since they're all the
  same):
  [general]
  directmedia=no
  directrtpsetup=no
  dtmfmode=rfc2833
  context=asterisk-internal
  allowsubscribe=no
  qualify=no
  disallow=all
  allow=ulaw
  allow=alaw
  allow=gsm
  localnet=10.10.32.0/255.255.248.0
  localnet=192.168.32.0/255.255.255.0
  
  [146]
  secret=
  host=dynamic
  type=friend
  
  From the aforementioned sip debug capture, 146 is on the 10.10.32.0/21
  network
  and 113 is on the 192.168.32.0/24 network (these are directly route-able so
  no
  NAT is involved). However, I have now been able to reproduce the problem
  between
  two devices directly on the 10.10.32.0/21 network as well.
  
 
 I've gathered the log for this dialog from the SIP phone:
 http://pastebin.com/aAWs4j6i
 
 What I see is that there's an INVITE for CSeq 103, then an OK for CSeq 103,
 then another INVITE is received for CSeq 103, at which point the phone
 reports an error:
 0 | ERROR | receive a request with same cseq??
 
 From the asterisk side, it never seems to receive this OK for CSeq 103, hence
 the reason it sends out the INVITE again.
 
Joshua,

As a mitigation for this problem, could I increase the timerb option in 
sip.conf
to a large value, say 1 hour (instead of the default 32 seconds)? What other
consequences would there be from this change?

Thanks,

Andrew

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Re: [asterisk-users] Retransmission Timeout results in dropped calls after 32 seconds

[Andrew Martin]

- Original Message -
 From: Joshua Colp jc...@digium.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Monday, May 11, 2015 1:24:53 PM
 Subject: Re: [asterisk-users] Retransmission Timeout results in dropped 
 calls   after 32 seconds
 
  Could this perhaps be because the phone doesn't support bypass or
  re-INVITEs?
  Is there a way to disable this functionality and instruct asterisk to just
  stay in the middle of the conversation (bridging or native-bridging) for
  the
  duration of the call? I thought that setting directmedia=no and
  directrtpsetup=no would disable re-INVITEs and force asterisk to use
  bridging
  mode, but perhaps something else is required?
 
 That should be all that is required. If that were broken I'd expect
 issue reports to implode - what's the configuration?
 

Here's the sip.conf (only showing a single extension since they're all the 
same):
[general]
directmedia=no
directrtpsetup=no
dtmfmode=rfc2833
context=asterisk-internal
allowsubscribe=no
qualify=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
localnet=10.10.32.0/255.255.248.0
localnet=192.168.32.0/255.255.255.0

[146]
secret=
host=dynamic
type=friend

From the aforementioned sip debug capture, 146 is on the 10.10.32.0/21 network
and 113 is on the 192.168.32.0/24 network (these are directly route-able so no
NAT is involved). However, I have now been able to reproduce the problem between
two devices directly on the 10.10.32.0/21 network as well.

Thanks,

Andrew

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Re: [asterisk-users] Retransmission Timeout results in dropped calls after 32 seconds

[Andrew Martin]

- Original Message -
 From: Joshua Colp jc...@digium.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Monday, May 11, 2015 12:32:06 PM
 Subject: Re: [asterisk-users] Retransmission Timeout results in dropped 
 calls   after 32 seconds
 
 Andrew Martin wrote:
  - Original Message -
 
 snip
 
 
  By doing a number of test calls today, I have managed to reproduce this
  while
  sip debugging was on, so I have that information available now as well:
  http://pastebin.com/ZJqzdvY3
 
  This was a call from 113 to 146 via a queue. Note that the asterisk server
  is
  at 10.10.32.251. I see the following:
  INVITE sip:146@10.10.32.96:5062 SIP/2.0
  SIP/2.0 180 Ringing
  SIP/2.0 180 Ringing
  SIP/2.0 200 OK
  ACK sip:146@10.10.32.96:5062 SIP/2.0
  INVITE sip:146@10.10.32.96:5062 SIP/2.0
  SIP/2.0 200 OK
  ACK sip:146@10.10.32.96:5062 SIP/2.0
  INVITE sip:146@10.10.32.96:5062 SIP/2.0
  INVITE sip:146@10.10.32.96:5062 SIP/2.0
  INVITE sip:146@10.10.32.96:5062 SIP/2.0
  INVITE sip:146@10.10.32.96:5062 SIP/2.0
  INVITE sip:146@10.10.32.96:5062 SIP/2.0
 
  This appears to start out with a successful SIP conversation (ending with
  the
  first ACK), so it is unclear to me why we have two new sets of INVITEs sent
  afterwards.
 
 Asterisk has sent a re-INVITE to have the media flow directly. The
 device (seems) to respond with the 200 OK (you can tell based on the
 CSeq) for the initial INVITE, and not for the re-INVITE. As Asterisk
 gets no response to its re-INVITE it gives up and terminates the dialog.
 

Could this perhaps be because the phone doesn't support bypass or re-INVITEs?
Is there a way to disable this functionality and instruct asterisk to just 
stay in the middle of the conversation (bridging or native-bridging) for the 
duration of the call? I thought that setting directmedia=no and 
directrtpsetup=no would disable re-INVITEs and force asterisk to use bridging
mode, but perhaps something else is required?

Thanks,

Andrew

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Re: [asterisk-users] Retransmission Timeout results in dropped calls after 32 seconds

[Andrew Martin]

- Original Message -
 From: Andrew Martin amar...@xes-inc.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Monday, May 11, 2015 1:35:07 PM
 Subject: Re: [asterisk-users] Retransmission Timeout results in dropped 
 calls   after 32 seconds

  That should be all that is required. If that were broken I'd expect
  issue reports to implode - what's the configuration?
  
 
 Here's the sip.conf (only showing a single extension since they're all the
 same):
 [general]
 directmedia=no
 directrtpsetup=no
 dtmfmode=rfc2833
 context=asterisk-internal
 allowsubscribe=no
 qualify=no
 disallow=all
 allow=ulaw
 allow=alaw
 allow=gsm
 localnet=10.10.32.0/255.255.248.0
 localnet=192.168.32.0/255.255.255.0
 
 [146]
 secret=
 host=dynamic
 type=friend
 
 From the aforementioned sip debug capture, 146 is on the 10.10.32.0/21
 network
 and 113 is on the 192.168.32.0/24 network (these are directly route-able so
 no
 NAT is involved). However, I have now been able to reproduce the problem
 between
 two devices directly on the 10.10.32.0/21 network as well.
 

I've gathered the log for this dialog from the SIP phone:
http://pastebin.com/aAWs4j6i

What I see is that there's an INVITE for CSeq 103, then an OK for CSeq 103,
then another INVITE is received for CSeq 103, at which point the phone
reports an error:
0 | ERROR | receive a request with same cseq??

From the asterisk side, it never seems to receive this OK for CSeq 103, hence
the reason it sends out the INVITE again.

Thanks,

Andrew

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Re: [asterisk-users] Retransmission Timeout results in dropped calls after 32 seconds

[Andrew Martin]

- Original Message -
 From: Andrew Martin amar...@xes-inc.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Friday, May 8, 2015 5:12:28 PM
 Subject: [asterisk-users] Retransmission Timeout results in dropped calls   
 after 32 seconds
 
 Hello,
 
 I am running Asterisk 11 on CentOS 6.4 with SIP clients (Yealink phones). All
 the SIP clients are on a LAN, so no NAT is involved. I have been experiencing
 an intermittent problem where a call will be successfully answered, but then
 dropped by Asterisk 32 seconds after it is answered (with a Retransmission
 timeout reached on transmission error). Here is an example of this happening
 in the asterisk console:
 http://pastebin.com/7LDwHAJe
 
 This problem only happens a fraction of the time, so I have been unable to
 enable SIP debugging before it happens to get a capture. However, usually the
 caller will just call back immediately and then the call will work without a
 problem. It sounds like SIP Timer B is what causes the call to be dropped if
 an
 ACK to the INVITE is not received within 32 seconds. How can I determine if
 this is the case and how can I resolve this Retransmission timeout problem?
 
 Here is my sip.conf:
 general]
 directmedia=no
 directrtpsetup=no
 dtmfmode=rfc2833
 context=internal
 allowsubscribe=no
 qualify=no
 disallow=all
 allow=ulaw
 allow=alaw
 allow=gsm
 localnet=10.10.32.0/255.255.248.0
 
 
 [123]
 secret=11
 host=dynamic
 type=friend
 

By doing a number of test calls today, I have managed to reproduce this while
sip debugging was on, so I have that information available now as well:
http://pastebin.com/ZJqzdvY3

This was a call from 113 to 146 via a queue. Note that the asterisk server is
at 10.10.32.251. I see the following:
INVITE sip:146@10.10.32.96:5062 SIP/2.0
SIP/2.0 180 Ringing
SIP/2.0 180 Ringing
SIP/2.0 200 OK
ACK sip:146@10.10.32.96:5062 SIP/2.0
INVITE sip:146@10.10.32.96:5062 SIP/2.0
SIP/2.0 200 OK
ACK sip:146@10.10.32.96:5062 SIP/2.0
INVITE sip:146@10.10.32.96:5062 SIP/2.0
INVITE sip:146@10.10.32.96:5062 SIP/2.0
INVITE sip:146@10.10.32.96:5062 SIP/2.0
INVITE sip:146@10.10.32.96:5062 SIP/2.0
INVITE sip:146@10.10.32.96:5062 SIP/2.0

This appears to start out with a successful SIP conversation (ending with the
first ACK), so it is unclear to me why we have two new sets of INVITEs sent
afterwards. 

Also in case it is relevant, the asterisk server has two NICs set up in a bond
with bond-mode 1 (active/backup).

Does this additional debug information provide any clues to why this 
intermittent retransmission timeout error is occurring?

Thanks,

Andrew

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[asterisk-users] Retransmission Timeout results in dropped calls after 32 seconds

[Andrew Martin]

Hello,

I am running Asterisk 11 on CentOS 6.4 with SIP clients (Yealink phones). All
the SIP clients are on a LAN, so no NAT is involved. I have been experiencing
an intermittent problem where a call will be successfully answered, but then
dropped by Asterisk 32 seconds after it is answered (with a Retransmission
timeout reached on transmission error). Here is an example of this happening
in the asterisk console: 
http://pastebin.com/7LDwHAJe

This problem only happens a fraction of the time, so I have been unable to
enable SIP debugging before it happens to get a capture. However, usually the
caller will just call back immediately and then the call will work without a
problem. It sounds like SIP Timer B is what causes the call to be dropped if an
ACK to the INVITE is not received within 32 seconds. How can I determine if
this is the case and how can I resolve this Retransmission timeout problem?

Here is my sip.conf:
general]
directmedia=no
directrtpsetup=no
dtmfmode=rfc2833
context=internal
allowsubscribe=no
qualify=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
localnet=10.10.32.0/255.255.248.0


[123]
secret=11
host=dynamic
type=friend


Thanks!

Andrew Martin

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Re: [asterisk-users] Phones don't stop ringing when queue is answered

[Andrew Martin]

James,

The WaitExten()s just provide a pause between the two Queue() calls to
let the first group of phones finish ringing. In this example I am ringing
the same group (queue_level_1) twice, however in a real-world scenario I 
would ring queue_level_1 and then ring queue_level_2 which each have a 
different list of phones.

Thanks,

Andrew

- Original Message -
 From: James Thomas jthomas...@gmail.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Thursday, May 7, 2015 10:20:10 AM
 Subject: Re: [asterisk-users] Phones don't stop ringing when queue is answered
 
 What purpose do the WaitExten()s serve here? Are you really allowing the
 caller to connect to different extensions in the test-queue context? Have
 you tried without the WaitExten()s?
 
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[asterisk-users] Delayed RTP

[Andrew Colin]

Hi Guys

 

We have a strange issue whereby one phone has delayed rtp

So what happens is when the lady answers the phone for the 1st 1 second
they can not hear her and then everything is fine

 

I am running asterisk 1.8.28.0

Has anyone seen this before?

 

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Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In

[Andrew Martin]

- Original Message -
 From: Guenther Boelter gboel...@gmail.com
 To: asterisk-users@lists.digium.com
 Sent: Tuesday, May 5, 2015 1:05:44 AM
 Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In
 
  Looking into it further, in my case it does not appear to be a
  NATing issue, since running OpenVPN from pfSense means there's no
  NATing occurring between the clients or between the clients and the
  asterisk server.
  
  Although I was unable to reproduce the problems, I did notice some
  packet loss and jitter in sip show channelstats, here is a
  sample: Peer Call ID  Duration Recv: Pack  Lost
  ( %) Jitter Send: Pack  Lost   ( %) Jitter
  192.168.32.26446613544@1  00:03:03 94  004238
  (97.83%) 0. 00  000244 ( 0.00%) 0.
  192.168.32.385b2ebdc92fd  00:03:03 59  01 (
  1.67%) 0. 00  91 ( 0.00%) 0.0028
  
  I was unable to find documentation each of these columns, but the
  high percentage of loss for received packets for 192.168.32.26
  seems suspicious. Do these statistics indicate a problem?
  
  Thanks,
  
  Andrew
 
 Hi Andrew,
 
 is this a linux machine? If so, check your NIC with ifconfig for
 hardware errors.
 
 Guenther
 

Guenther,

Yes, this machine is running CentOS 6.4 (see my original post for more
details). This asterisk server has 2x gigabit NICs set up in a bond with
bond mode 1.

Both ifconfig and ethtool do not report any hardware errors,
although they do show a few checksum errors:
eth0  Link encap:Ethernet  HWaddr 00:11:22:33:44:55
  UP BROADCAST RUNNING SLAVE MULTICAST  MTU:1500  Metric:1
  RX packets:467927100 errors:0 dropped:0 overruns:1 frame:0
  TX packets:304724661 errors:0 dropped:0 overruns:0 carrier:0
  collisions:0 txqueuelen:1000
  RX bytes:131747094082 (122.6 GiB)  TX bytes:93869585242 (87.4 GiB)
  Memory:fb92-fb94

eth1  Link encap:Ethernet  HWaddr AA:BB:CC:DD:EE:FF
  UP BROADCAST RUNNING SLAVE MULTICAST  MTU:1500  Metric:1
  RX packets:41250363 errors:0 dropped:0 overruns:0 frame:0
  TX packets:3467 errors:0 dropped:0 overruns:0 carrier:0
  collisions:0 txqueuelen:1000
  RX bytes:5190889937 (4.8 GiB)  TX bytes:1594075 (1.5 MiB)
  Memory:fb90-fb92

From ethtool -S eth0:
 tx_smbus: 164709
 rx_smbus: 119082408
 dropped_smbus: 104036

 rx_queue_0_packets: 97532982
 rx_queue_0_bytes: 16800645524
 rx_queue_0_drops: 1
 rx_queue_0_csum_err: 0
 rx_queue_0_alloc_failed: 0

 rx_queue_7_packets: 53850556
 rx_queue_7_bytes: 12797600155
 rx_queue_7_drops: 0
 rx_queue_7_csum_err: 41
 rx_queue_7_alloc_failed: 0

Thanks,

Andrew

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Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In

[Andrew Martin]

- Original Message -
 From: Administrator TOOTAI ad...@tootai.net
 To: asterisk-users@lists.digium.com
 Sent: Friday, May 1, 2015 6:42:38 AM
 Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In
 
 Le 01/05/2015 00:05, Andrew Martin a écrit :
  - Original Message -
  From: Administrator TOOTAI ad...@tootai.net
  To: asterisk-users@lists.digium.com
  Sent: Thursday, April 30, 2015 4:43:33 PM
  Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call
  In
 
  I am running Asterisk 11.12.0 on CentOS 6.4. The asterisk server and
  internal phones are located on the 10.10.32.0/21 LAN subnet. I have many
  internal SIP phones, which appear to be working correctly. I have a few
  external phones (Yealink SIP-T32G or other Yealink model) on
  192.168.32.0/24 which have an OpenVPN client configured on them that
  connects back to the LAN network through a pfSense gateway with OpenVPN
  configured on it.
 
  I faced problems with pfsense -no VPN involved- and finally installed
  siproxd on it. Also set the firewall mode to conservative.
 
  Daniel,
 
  Thanks for the information. Do you have an example or documentation on the
  siproxd configuration that you used?
 
 No, just follow the basis of the parameters given by the package. If I
 remember, SIP use the proxy siproxd and RTP is direct.
 

Looking into it further, in my case it does not appear to be a NATing issue,
since running OpenVPN from pfSense means there's no NATing occurring between
the clients or between the clients and the asterisk server.

Although I was unable to reproduce the problems, I did notice some packet loss
and jitter in sip show channelstats, here is a sample:
Peer Call ID  Duration Recv: Pack  Lost   ( %) Jitter 
Send: Pack  Lost   ( %) Jitter
192.168.32.26446613544@1  00:03:03 94  004238 (97.83%) 0. 
00  000244 ( 0.00%) 0.
192.168.32.385b2ebdc92fd  00:03:03 59  01 ( 1.67%) 0. 
00  91 ( 0.00%) 0.0028

I was unable to find documentation each of these columns, but the high 
percentage
of loss for received packets for 192.168.32.26 seems suspicious. Do these 
statistics
indicate a problem?

Thanks,

Andrew




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[asterisk-users] OpenVPN Clients Intermittently Cannot Call In

[Andrew Martin]

Hello,

I am running Asterisk 11.12.0 on CentOS 6.4. The asterisk server and internal 
phones are located on the 10.10.32.0/21 LAN subnet. I have many internal SIP 
phones, which appear to be working correctly. I have a few external phones 
(Yealink SIP-T32G or other Yealink model) on 192.168.32.0/24 which have an 
OpenVPN client configured on them that connects back to the LAN network through 
a pfSense gateway with OpenVPN configured on it. 

Asterisk server LAN IP: 10.10.32.10
My internal test phone: 146 at 10.10.32.96
My external test phone: 265 at 192.168.32.10

My sip.conf for these external users is as follows:
http://pastebin.com/2b9YE7Dz


The dialplan uses this Dial() invocation when dialing either an internal or 
external phone. Note that the max timeout is 12 seconds:
exten = _[12]XX,1,Dial(SIP/${EXTEN},12)


These external phones register correctly, and internal users can call these 
external users, the phones ring immediately, and the call is normal. However, 
if the external users try to dial an internal phone, I've observed some 
different failure modes:
* operating normally: sometimes the call rings immediately, the internal user 
answers, and the audio is present immediately
* ringing delay and no connection even after pickup: sometimes there's a 
significant delay between when the call starts ringing on the external side 
and when it actually starts ringing on the internal user's phone. Consequently, 
the internal user only has 1 or 2 rings to answer. Even if they do answer 
during this time, the line is dead and it goes to voicemail (the next step in 
the dialplan)
* delay before audio is connected after answer: sometimes the internal user 
answers, but there's a delay of 3-10 seconds before either party can hear audio

I've enabled rtp and sip debug for this particular external phone 
(192.168.32.10) and attached console logs from both types of these failures:
* ringing delay and no connection even after pickup: 
http://pastebin.com/fe1khEmF
* delay before audio is connected after answer: http://pastebin.com/uZSMKczk

What else can I try to debug these problems? Since it is intermittent, I am not 
always able to reproduce (sometimes the calls work just fine).

Thanks,

Andrew Martin

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Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In

[Andrew Martin]

- Original Message -
 From: Administrator TOOTAI ad...@tootai.net
 To: asterisk-users@lists.digium.com
 Sent: Thursday, April 30, 2015 4:43:33 PM
 Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In
 
  I am running Asterisk 11.12.0 on CentOS 6.4. The asterisk server and
  internal phones are located on the 10.10.32.0/21 LAN subnet. I have many
  internal SIP phones, which appear to be working correctly. I have a few
  external phones (Yealink SIP-T32G or other Yealink model) on
  192.168.32.0/24 which have an OpenVPN client configured on them that
  connects back to the LAN network through a pfSense gateway with OpenVPN
  configured on it.
 
 I faced problems with pfsense -no VPN involved- and finally installed
 siproxd on it. Also set the firewall mode to conservative.

Daniel,

Thanks for the information. Do you have an example or documentation on the
siproxd configuration that you used?

Thanks,

Andrew

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[asterisk-users] Kamallio registration

[Andrew Colin]

Hi Guys

 

Is it possible to register Kamallio directly to our SIP provider then load
balance the RTP to 2 asterisk servers?

 

We cant do the registration from the asterisk boxes as we want to do it
directly from Kamallio.

Is this possible?

 

 

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Re: [asterisk-users] Script to Program Snom Phones

[Andrew Latham]

On Thu, Apr 9, 2015 at 12:23 PM, Derek Andrew derek.and...@usask.ca wrote:

 SNOM phones can be configured using files on a TFTP server.

 On Thu, Apr 9, 2015 at 11:14 AM, jg webaccounts...@jgoettgens.de wrote:


   Does anyone know how to program Snom phones using a Mac addresses like
 what is done with the Ciscos. I have about 50 extensions to be programmed
 and I am hoping there is a way on Asterisk to program extensions on the
 snom phones. Please assist.


  What do you mean with 50 extensions? Snom phones allow to define a
 directory, where you can export and import a simple text file. There might
 also be a way to automate this using one of the provisioning methods.

 jg




 --
 Copyright 2015 Derek Andrew (excluding quotations)

 +1 306 966 4808
 University of Saskatchewan
 Peterson 120; 54 Innovation Boulevard
 Saskatoon,Saskatchewan,Canada. S7N 2V3
 Timezone GMT-6

 Typed but not read.



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HTTP is the prefered method of provisioning. You can see
http://wiki.snom.com/Settings/setting_server and even the dynamic tools
baked into Asterisk at
https://wiki.asterisk.org/wiki/display/AST/Phone+Provisioning+in+Asterisk


-- 
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Re: [asterisk-users] Script to Program Snom Phones

[Derek Andrew]

SNOM phones can be configured using files on a TFTP server.

On Thu, Apr 9, 2015 at 11:14 AM, jg webaccounts...@jgoettgens.de wrote:


   Does anyone know how to program Snom phones using a Mac addresses like
 what is done with the Ciscos. I have about 50 extensions to be programmed
 and I am hoping there is a way on Asterisk to program extensions on the
 snom phones. Please assist.


  What do you mean with 50 extensions? Snom phones allow to define a
 directory, where you can export and import a simple text file. There might
 also be a way to automate this using one of the provisioning methods.

 jg




-- 
Copyright 2015 Derek Andrew (excluding quotations)

+1 306 966 4808
University of Saskatchewan
Peterson 120; 54 Innovation Boulevard
Saskatoon,Saskatchewan,Canada. S7N 2V3
Timezone GMT-6

Typed but not read.
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Re: [asterisk-users] Asterisk Inbound calls, multiple SIP accounts, calledID

[Andrew Galdes]

Hi Dmitriy and others and thanks for your help so far.

The option match_auth_username=yes seems to have had no effect. From my
reading, this option will try to match the username of the incoming SIP
account to a section heading. If that is how it must work then i can see a
big problem. I'm trying to present the receptionist with a nice display of
which line the call came in on. For example, the receptionist answers calls
for 8 different companies and would like the phone to display the company
name that she should announce to the caller.

Here is a more complete output of an incoming call. I've changed the SIP
numbers to Company1', etc, to hide the numbers.

Connected to Asterisk 10.12.4 currently running on asterisk (pid = 32267)
 Verbosity is at least 12
 asterisk*CLI
 asterisk*CLI
 asterisk*CLI
   == Using SIP RTP CoS mark 5
 -- Executing [s@incoming:1] *Set*(*SIP/Company1-0797*, 
 *thedid=NodePhonesip:compa...@sip.internode.on.net
 sip%3acompa...@sip.internode.on.net*) in new stack
 -- Executing [s@incoming:2] *Set*(*SIP/**Company1**-0797*, 
 *pseudodid=NodePhonesip:** sip:Company2**@sip.internode.on.net
 http://sip.internode.on.net*) in new stack
 -- Executing [s@incoming:3] *Set*(*SIP/**Company1**-0797*, 
 *pseudodid=NodePhonesip:** sip:Company2*) in new stack
 -- Executing [s@incoming:4] *Set*(*SIP/**Company1**-0797*, 
 *pseudodid=** sip:Company2*) in new stack
 -- Executing [s@incoming:5] *GotoIf*(*SIP/**Company1**-0797*, 
 *0?internal,33,1:6*) in new stack
 -- Goto (incoming,s,6)
 -- Executing [s@incoming:6] *GotoIf*(*SIP/**Company1**-0797*, 
 *0?internal,88,1:7*) in new stack
 -- Goto (incoming,s,7)
 -- Executing [s@incoming:7] *GotoIf*(*SIP/**Company1**-0797*, 
 *0?internal,36,1:8*) in new stack
 -- Goto (incoming,s,8)
 -- Executing [s@incoming:8] *GotoIf*(*SIP/**Company1**-0797*, 
 *1?internal,36,1:9*) in new stack
 -- Goto (internal,36,1)
 -- Executing [36@internal:1] *Set*(*SIP/**Company1**-0797*, 
 *CALLERID(name)=SIP/**Company1**-0797*) in new stack
 -- Executing [36@internal:2] *Dial*(*SIP/**Company1**-0797*, 
 *SIP/36,20*) in new stack
   == Using SIP RTP CoS mark 5
 -- Called SIP/36
 -- SIP/36-0798 is ringing
   == Spawn extension (internal, 36, 2) exited non-zero on
 'SIP/Company1-0797'
 asterisk*CLI exit


And here is the sip.conf:

[general]
 match_auth_username=yes
 register=081...:...@sip.internode.on.net/s
 register=082...:...@sip.internode.on.net/s
 register=083...:...@sip.internode.on.net:/s
 register=084...:...@sip.internode.on.net:/s
 register=085...:...@sip.internode.on.net/s
 register=086...:...@sip.internode.on.net/s
 register=087...:...@sip.internode.on.net/s
 register=088...:...@sip.internode.on.net/s

 [Company1]
 username=081...
 fromuser=081...
 secret=...
 canreinvite=no
 qualify=yes
 context=incoming
 type=friend
 insecure=invite,port
 fromdomain=sip.internode.on.net
 host=sip.internode.on.net
 dtmfmode=rfc2833
 disallow=all
 allow=alaw
 allow=ulaw
 allow=g729
 bindport=5060
 bindaddr=0.0.0.0
 nat=yes
 registertimeout=5
 allowoverlap=no
 srvlookup=no
 ubscribecontext=from-sip
 callcounter=yes



[Company2]
 ...
 [Company3]
 ...
 [Company4]
 ...

 And here is some of the extensions.conf file:

[incoming]
 ; Get the DID number from the TO header.
 exten = s,1,Set(thedid=${SIP_HEADER(TO)})
 exten = s,2,Set(pseudodid=${SIP_HEADER(To)})
 exten = s,3,Set(pseudodid=${CUT(pseudodid,@,1)})
 exten = s,4,Set(pseudodid=${CUT(pseudodid,:,2)})


 ; Direct the DID accordingly.
 exten = s,5,GotoIf($[${pseudodid} = 081]?internal,33,1:6)
 exten = s,6,GotoIf($[${pseudodid} = 082]?internal,88,1:7)
 exten = s,7,GotoIf($[${pseudodid} = 083]?internal,36,1:8)
 exten = s,8,GotoIf($[${pseudodid} = 084]?internal,36,1:9)
 exten = s,9,GotoIf($[${pseudodid} = 085]?internal,36,1:10)
 exten = s,10,GotoIf($[${pseudodid} = 086]?internal,89,1:11)
 exten = s,11,GotoIf($[${pseudodid} = 087]?internal,36,1:12)
 exten = s,12,GotoIf($[${pseudodid} = 088]?internal,13,1:13)



-Andrew Galdes


On Thu, Apr 2, 2015 at 3:46 PM, Dmitriy Serov serov@gmail.com wrote:


 This is one of the chronic problems. Try this option in sip.conf:
 match_auth_username=yes

 Carefully read the description, it is better to test in after hours.

 02.04.2015 2:50, Andrew Galdes пишет:

 Hello all,

  I have an Asterisk server (Asterisk 10.12.4) with multiple sip accounts
 with the same service provides. We have 8 phone numbers in total.

  Incoming calls from the public are all correctly directed to appropriate
 office handsets. However, the display on the reception phone (the only one
 i care about) is always showing the same SIP/Account1_0843214321 rather
 than the account representing the number dialed.

  For-instance, if Sam on her mobile calls *08*, Asterisk will
 show a log entry like the following:

  -- Executing [s@incoming:1] Set(SIP/*Account1_08*, 
 thedid=NodePhonesip:*08

Re: [asterisk-users] Asterisk Inbound calls, multiple SIP accounts, calledID

[Andrew Galdes]

Solved it, kinda. It's not cute. I'm sure this is the way NOT to do it but
it does work. For prosperity, the SIP service is through Internode.

Here is my extensions.conf file:

exten = s,1,Set(thedid=${SIP_HEADER(TO)}); ignore this one
exten = s,2,Set(pseudodid=${SIP_HEADER(To)})
exten = s,3,Set(pseudodid=${CUT(pseudodid,@,1)})
exten = s,4,Set(pseudodid=${CUT(pseudodid,:,2)})

exten = s,5,Set(callersname=${IF($[ ${pseudodid} =
081...]?Company1:${callersname})})
exten = s,6,Set(callersname=${IF($[ ${pseudodid}
= 082...]?Company2:${callersname})})
exten = s,7,Set(callersname=${IF($[ ${pseudodid}
= 083...]?Company3:${callersname})})
exten = s,8,Set(callersname=${IF($[ ${pseudodid}
= 084...]?Company4:${callersname})})
exten = s,9,Set(callersname=${IF($[ ${pseudodid}
= 085...]?Company5:${callersname})})
exten = s,10,Set(callersname=${IF($[ ${pseudodid}
= 086...]?Company6:${callersname})})
exten = s,11,Set(callersname=${IF($[ ${pseudodid}
= 087...]?Company7:${callersname})})
exten = s,12,Set(callersname=${IF($[ ${pseudodid}
= 088...]?Company8:${callersname})})

exten = s,13,GotoIf($[${callersname} = Company1]?internal,36,1:14); to
reception
exten = s,14,GotoIf($[${callersname} = Company2]?internal,88,1:15); to
department1
exten = s,15,GotoIf($[${callersname} = Company3]?internal,36,1:16); to
reception
exten = s,16,GotoIf($[${callersname} = Company4]?internal,36,1:17); to
reception
exten = s,17,GotoIf($[${callersname} = Company5]?internal,36,1:18); to
reception
exten = s,18,GotoIf($[${callersname} = Company6]?internal,89,1:19); to
department2
exten = s,19,GotoIf($[${callersname} = Company7]?internal,36,1:20); to
reception
exten = s,20,GotoIf($[${callersname} = Company8]?internal,13,1:21); to
department3

And later in same file:

; Phone 36 reception
 *exten = 36,1,Set(CALLERID(name)=${callersname})*
 exten = 36,n,Dial(SIP/36,20)
 exten = 36,n,VoiceMail(36,u)
 exten = 36,n,Hangup


Ta,


-Andrew Galdes
Managing Director

RHCE, LPI, CCENT

AGIX Linux

Ph: 08 7324 4429
Mb: 0422 927 598

Find us: Website http://www.agix.com.au | LinkedIn
http://au.linkedin.com/in/andrewgaldes | Blog http://agix.com.au/blog |
YouTube http://www.youtube.com/user/andrewgaldes | Google+
http://google.com/+AndrewGaldes

*Platform Architects for High Demand Web Applications.*

On Wed, Apr 8, 2015 at 9:18 AM, Andrew Galdes andrew.gal...@agix.com.au
wrote:

 Hi Dmitriy and others and thanks for your help so far.

 The option match_auth_username=yes seems to have had no effect. From my
 reading, this option will try to match the username of the incoming SIP
 account to a section heading. If that is how it must work then i can see a
 big problem. I'm trying to present the receptionist with a nice display of
 which line the call came in on. For example, the receptionist answers calls
 for 8 different companies and would like the phone to display the company
 name that she should announce to the caller.

 Here is a more complete output of an incoming call. I've changed the SIP
 numbers to Company1', etc, to hide the numbers.

 Connected to Asterisk 10.12.4 currently running on asterisk (pid = 32267)
 Verbosity is at least 12
 asterisk*CLI
 asterisk*CLI
 asterisk*CLI
   == Using SIP RTP CoS mark 5
 -- Executing [s@incoming:1] *Set*(*SIP/Company1-0797*, 
 *thedid=NodePhonesip:compa...@sip.internode.on.net
 sip%3acompa...@sip.internode.on.net*) in new stack
 -- Executing [s@incoming:2] *Set*(*SIP/**Company1**-0797*, 
 *pseudodid=NodePhonesip:** sip:Company2**@sip.internode.on.net
 http://sip.internode.on.net*) in new stack
 -- Executing [s@incoming:3] *Set*(*SIP/**Company1**-0797*, 
 *pseudodid=NodePhonesip:** sip:Company2*) in new stack
 -- Executing [s@incoming:4] *Set*(*SIP/**Company1**-0797*, 
 *pseudodid=** sip:Company2*) in new stack
 -- Executing [s@incoming:5] *GotoIf*(*SIP/**Company1**-0797*, 
 *0?internal,33,1:6*) in new stack
 -- Goto (incoming,s,6)
 -- Executing [s@incoming:6] *GotoIf*(*SIP/**Company1**-0797*, 
 *0?internal,88,1:7*) in new stack
 -- Goto (incoming,s,7)
 -- Executing [s@incoming:7] *GotoIf*(*SIP/**Company1**-0797*, 
 *0?internal,36,1:8*) in new stack
 -- Goto (incoming,s,8)
 -- Executing [s@incoming:8] *GotoIf*(*SIP/**Company1**-0797*, 
 *1?internal,36,1:9*) in new stack
 -- Goto (internal,36,1)
 -- Executing [36@internal:1] *Set*(*SIP/**Company1**-0797*, 
 *CALLERID(name)=SIP/**Company1**-0797*) in new stack
 -- Executing [36@internal:2] *Dial*(*SIP/**Company1**-0797*, 
 *SIP/36,20*) in new stack
   == Using SIP RTP CoS mark 5
 -- Called SIP/36
 -- SIP/36-0798 is ringing
   == Spawn extension (internal, 36, 2) exited non-zero on
 'SIP/Company1-0797'
 asterisk*CLI exit


 And here is the sip.conf:

 [general]
 match_auth_username=yes
 register=081...:...@sip.internode.on.net/s
 register=082...:...@sip.internode.on.net/s
 register=083...:...@sip.internode.on.net:/s
 register=084...:...@sip.internode.on.net

[asterisk-users] Asterisk Inbound calls, multiple SIP accounts, calledID

[Andrew Galdes]

Hello all,

I have an Asterisk server (Asterisk 10.12.4) with multiple sip accounts
with the same service provides. We have 8 phone numbers in total.

Incoming calls from the public are all correctly directed to appropriate
office handsets. However, the display on the reception phone (the only one
i care about) is always showing the same SIP/Account1_0843214321 rather
than the account representing the number dialed.

For-instance, if Sam on her mobile calls *08*, Asterisk will show
a log entry like the following:

-- Executing [s@incoming:1] Set(SIP/*Account1_08*, 
thedid=NodePhonesip:*08*@sip.internode.on.net) in new stack
But Account1_*08* (as the name suggests) has a phone number of 
*08* and not *08*.

So Sam's call will come through and be routed to the correct handset as the
business needs, but it seems that all incoming calls are being labeled as
though coming in on a different account. The effective problem is that the
calledID is now wrong.

I'm after some general advice on how to handle the problem.

Ta,


-Andrew
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[asterisk-users] 4 Port PRI

[Andrew Colin]

Hi Guys

 

I have a 4 port PRI card that I need to setup each port in their own
group.

In chan_dahdi.conf I have the following which works for one port

How do I add the rest of the ports in their own groups so that I can have
different signaling on each?

 

 

[channels]

language=en

switchtype=euroisdn

pridialplan=unknown

resetinterval=600

echocancel=yes

echotraining=yes

;echocancelwhenbridged=no

;rxgain=0

;txgain=0

callerid=asreceived

musiconhold=default

group=1

overlapdial=yes

signalling=pri_cpe

context=extensions

channel = 1-15,17-31

jbenable= yes

jbforce= yes

jbmaxsize= 120

jbimpl= fixed

jbresyncthreshold= 1000

 

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Re: [asterisk-users] 4 Port PRI

[Andrew Colin]

4 Port PRI sangoma a104

From: jg [mailto:webaccounts...@jgoettgens.de]
Sent: Wednesday, March 18, 2015 2:09 PM
To: Andrew Colin; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 4 Port PRI





I have a 4 port PRI card that I need to setup each port in their own group.

In chan_dahdi.conf I have the following which works for one port

How do I add the rest of the ports in their own groups so that I can have 
different signaling on each?





[channels]

language=en

switchtype=euroisdn

pridialplan=unknown

resetinterval=600

echocancel=yes

echotraining=yes

;echocancelwhenbridged=no

;rxgain=0

;txgain=0

callerid=asreceived

musiconhold=default

group=1

overlapdial=yes

signalling=pri_cpe

context=extensions

channel = 1-15,17-31

jbenable= yes

jbforce= yes

jbmaxsize= 120

jbimpl= fixed

jbresyncthreshold= 1000



PRI or BRI? Which card are you using? Typically the installation script or 
procedure lets you configure each span. You seem to have 4 spans for either 
8 or 128 (EuroISDN) channels.

jg

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[asterisk-users] Yealink t26 and T28 Panels

[Andrew Colin]

Hi Guys

 

We have a strange a strange issue at a client they have 3 panels on their
phone and every so often the panels reboot themselves yet the phone stays
on.

We decided to replace the T26 for a T28 to see if it fixes the issue and
still have the exact same issue.

 

Has anyone seen this before?

 

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Re: [asterisk-users] Yealink t26 and T28 Panels

[Andrew Colin]

Originally we used just POE but now each of the 3 panels has its own PSU







From: jg [mailto:webaccounts...@jgoettgens.de]
Sent: Friday, March 13, 2015 11:18 AM
To: Andrew Colin; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Yealink t26 and T28 Panels



Hi!

We have a strange a strange issue at a client they have 3 panels on their 
phone and every so often the panels reboot themselves yet the phone stays 
on.

We decided to replace the T26 for a T28 to see if it fixes the issue and 
still have the exact same issue.



Has anyone seen this before?



I frequently use the newer T48G and T46G phones with the EXP40 expansion 
module. There are issues, if you are logged into the phone via the 
webinterface as an admin. Among other things, the display is not properly 
updated and wrong numbers may get dialed. Some time ago, there was a 
firmware update and I am not aware of any stability issues at the moment.

How do you supply power? 3 expansion modules + the phone and a cheap POE 
switch could be critical. It may not be the power itself, but the correct 
handling of energy saving states.

jg

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[asterisk-users] Strange Polycom Issue

[Andrew Colin]

Hi Guys,

 

We are getting a strange issue on certain polycom phones, sometimes when a
call comes in it just flashes to show there is a call but does not play
any sound.

This problem is very intermittent and happens to maybe 2 out of 10 calls.

 

Has any else experienced this before?

 

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Re: [asterisk-users] BlindXfer Sensitivity

[Andrew Colin]

The strange thing is its only sometimes my dial string is as follows

exten = s,1, Dial (SIP/200,, tT)

For that particular route but obviously s,3 as have Ringing () first etc.
After she pushes ## 6 times it will go thru sometimes.



Sent from Samsung Mobile

div Original message /divdivFrom: Kevin Larsen 
kevin.lar...@pioneerballoon.com /divdivDate:16/02/2015  17:11  
(GMT+02:00) /divdivTo: Andrew Colin and...@convergedgroup.net,Asterisk 
Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com /divdivSubject: Re: [asterisk-users] 
BlindXfer Sensitivity /divdiv
/div Hi Guys 
   
 We have a client running on a polycom vvx400 IP phone on our 
 asterisk 1.8.18 system 
   
 The issue we have is the switchboard lady uses ## to transfer calls 
 but sometimes it just does not work and just plays the DTMF tone to 
 the calling party. 
   
 Is there any way to adjust the sensitivity of the blindxfer feature? 
   
 The polycom Transfer button is useless  as there is a big delay 
 until it apprears 
   
 I would greatly appreciate any advice 

It seems weird that this would be some kind of sensitivity to the DTMF tones. 
The first thing I would look for is on a call that she cannot blind transfer, 
check how the Dial command was used to reach her. Does it have the proper use 
of the tT options (depending on whether she called them or they called her)? I 
would almost bet there is a call path that occurs which doesn't have the proper 
options set to allow the transfer.-- 
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Re: [asterisk-users] BlindXfer Sensitivity

[Andrew Colin]

RFC2833

The strange thing is how asterisk is not registering she has pushed ## on
those Rare occiasions



 On Mon, Feb 16, 2015 at 10:13 AM, Andrew Colin and...@vsave.co.za wrote:

 The strange thing is its only sometimes my dial string is as follows

 exten = s,1, Dial (SIP/200,, tT)

 For that particular route but obviously s,3 as have Ringing () first
 etc.
 After she pushes ## 6 times it will go thru sometimes.


 How is the DTMF being transmitted from the phone to Asterisk? RFC2833,
 in-band, SIP INFO...?

 --
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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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[asterisk-users] Trouble with T38/Dialogic

[Andrew McRory]

Hello, I am working with 1.8.32.2 which I have patched with t38-gateway and
PRACK. t38 is tested and working fine with Zoiper client but I can't get the
t.38 software from Biscom (FAXCOM) to talk. In my first attempts I found
FAXCOM announces that it supports 100rel so I added the PRACK patch hoping
that would do the trick. Now it gets a little further but * complains about
rejecting a non-primary audio stream.

Could this be a problem with 1.8 not liking the second media stream or is
there some more configuration tweaking to be done?

--- CUT --
--- SIP read from UDP:192.168.1.13:5060 ---
INVITE sip:1XX@192.168.1.11 SIP/2.0
From: Biscom
sip:418@192.168.1.13;tag=86c9140-d281eac-13c4-55013-1f571-33180d4a-1f571
To: sip:1XX@192.168.1.11
Call-ID: 73cb2e8-d281eac-13c4-55013-1f571-7ef285e4-1f571
CSeq: 2 INVITE
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-1f571-7a6c245-2c788e59
Supported: 100rel
Max-Forwards: 70
User-Agent: Brktsip/6.6.9B12 (Dialogic)
Contact: sip:192.168.1.13
Authorization: Digest
username=418,realm=asterisk,nonce=37a417d1,uri=sip:1XX@192.168.1.11,response=6cdb70491bdaf9acc75d9b776101d111,algorithm=MD5
Content-Type: application/sdp
Content-Length: 220

v=0
o=418 2209120086 0667748000 IN IP4 192.168.1.13
s=no_session_name
t=0 0
m=audio 56040 RTP/AVP 0
c=IN IP4 192.168.1.13
a=rtpmap:0 pcmu/8000
m=audio 56040 RTP/AVP 8
c=IN IP4 192.168.1.13
a=rtpmap:8 pcma/8000
-
--- (13 headers 10 lines) ---
Sending to 192.168.1.13:5060 (no NAT)
Using INVITE request as basis request -
73cb2e8-d281eac-13c4-55013-1f571-7ef285e4-1f571
Found peer '418' for '418' from 192.168.1.13:5060
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found audio description format pcmu for ID 0
[2015-02-12 23:36:58] WARNING[13727]: chan_sip.c:9305 process_sdp: Rejecting
non-primary audio stream: audio 56040 RTP/AVP 8

--- Reliably Transmitting (no NAT) to 192.168.1.13:5060 ---
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP
192.168.1.13:5060;branch=z9hG4bK-1f571-7a6c245-2c788e59;received=192.168.1.13
From: Biscom
sip:418@192.168.1.13;tag=86c9140-d281eac-13c4-55013-1f571-33180d4a-1f571
To: sip:1XX@192.168.1.11;tag=as7bee4e61
Call-ID: 73cb2e8-d281eac-13c4-55013-1f571-7ef285e4-1f571
CSeq: 2 INVITE
Server: FPBX-12.0.37(1.8.32.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, PRACK, MESSAGE
Supported: replaces, timer, 100rel
Content-Length: 0
--- CUT --

TIA

--
Andrew McRory
Sayso Communications, Inc.
Tallahassee, Florida

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[asterisk-users] BlindXfer Sensitivity

[Andrew Colin]

Hi Guys

 

We have a client running on a polycom vvx400 IP phone on our asterisk
1.8.18 system

 

The issue we have is the switchboard lady uses ## to transfer calls but
sometimes it just does not work and just plays the DTMF tone to the
calling party.

 

Is there any way to adjust the sensitivity of the blindxfer feature?

 

The polycom Transfer button is useless  as there is a big delay until it
apprears

 

I would greatly appreciate any advice

 

 

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Re: [asterisk-users] queue reload command

[Andrew Colin]

Hi



queue reload(queue name) or queue reload all



for example



queue reload reception



From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik
Sent: Thursday, January 8, 2015 2:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] queue reload command



Hi



I'm using asterisk 1.8



Does anyone know how to use the queue reload command. The built in help 
doesn't really help.



queue reload {parameters|membe Reload queues, members, queue rules, or 
parameters



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] One way audio internal

[Andrew Colin]

Hi All

 

We have a strange issue with our hosted asterisk server running on Debian

Internal calls btween extensions using g729 give one way audio

As soon as we change the codec to ALAW the issues goes away.

 

Any ideas how to fix this?

Outbound calls via a trunk work fine with g729

 

Kind Regards

Andrew Colin

Converged Data (Pty) Ltd.

Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09)



 

Direct: +27 (0)10 591 4607

Mobile: +27 (0)82 310 3007
Switchboard: +27 (0)10 591 4600
Email:  mailto:and...@convergedgroup.net and...@convergedgroup.net

Web: http://www.convergedgroup.net
75 Witkoppen Road, Northriding, Johannesburg, 2169
P O Box 7246, Weltevredenpark, 1715
This communication is confidential and intended solely for the
addressee(s). Any unauthorized review, use, disclosure or distribution is
prohibited. If you believe this message has been sent to you in error,
please notify the sender by replying to this transmission and delete the
message without disclosing it. Thank you.E-mail including attachments is
susceptible to data corruption, interception, unauthorized amendment,
tampering and viruses, and we only send and receive emails on the basis
that we are not liable for any such corruption, interception, amendment,
tampering or viruses or any consequences thereof.

 

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Re: [asterisk-users] One way audio internal

[Andrew Colin]

I am using the free g729







Kind Regards

Andrew Colin

Converged Data (Pty) Ltd.

Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09)





Direct: +27 (0)10 591 4607

Mobile: +27 (0)82 310 3007
Switchboard: +27 (0)10 591 4600
Email: and...@convergedgroup.net

Web: http://www.convergedgroup.net
75 Witkoppen Road, Northriding, Johannesburg, 2169
P O Box 7246, Weltevredenpark, 1715
This communication is confidential and intended solely for the addressee(s). 
Any unauthorized review, use, disclosure or distribution is prohibited. If 
you believe this message has been sent to you in error, please notify the 
sender by replying to this transmission and delete the message without 
disclosing it. Thank you.E-mail including attachments is susceptible to data 
corruption, interception, unauthorized amendment, tampering and viruses, and 
we only send and receive emails on the basis that we are not liable for any 
such corruption, interception, amendment, tampering or viruses or any 
consequences thereof.



From: Mitul Limbani [mailto:mi...@enterux.in]
Sent: Friday, November 21, 2014 12:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Andrew Colin
Subject: Re: [asterisk-users] One way audio internal



You probably do not have enough g729 channels license.

On 21-Nov-2014 4:17 PM, A J Stiles asterisk_l...@earthshod.co.uk wrote:

On Friday 21 Nov 2014, Andrew Colin wrote:
 Hi All

 We have a strange issue with our hosted asterisk server running on Debian
 Internal calls btween extensions using g729 give one way audio
 As soon as we change the codec to ALAW the issues goes away.

 Any ideas how to fix this?

 Outbound calls via a trunk work fine with g729

Unless you have serious bandwidth issues, just forget about g.729 and change
to a-law throughout.  A-law is what the PSTN  (in civilised countries)  uses
anyway, so you won't need to transcode  (which chews up processor resources
and risks compromising quality)  for calls to and from the outside world.

If you really need to use g.729 and are outside the USA  (therefore, beyond
the reach of software patents),  there is a free version that you can use --
and this one, better than Digium's offering, comes with the Source Code so 
you
can be sure it isn't doing anything nasty behind the scenes.

But to be honest, you probably are better off just sticking with a-law.

--
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] One way audio internal

[Andrew Colin]

I currently am running on a 

Intel(R) Xeon(R) CPU E5-2670 v2 @ 2.50GHz

 

Codec im using is

 

codec_g729-ast18-icc-glibc-x86_64-core2.so

 

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Re: [asterisk-users] One way audio internal

[Andrew Colin]

I currently am running on a

Intel(R) Xeon(R) CPU E5-2670 v2 @ 2.50GHz



Codec im using is



codec_g729-ast18-icc-glibc-x86_64-core2.so



Kind Regards

Andrew Colin

Converged Data (Pty) Ltd.

Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09)





Direct: +27 (0)10 591 4607

Mobile: +27 (0)82 310 3007
Switchboard: +27 (0)10 591 4600
Email: and...@convergedgroup.net

Web: http://www.convergedgroup.net
75 Witkoppen Road, Northriding, Johannesburg, 2169
P O Box 7246, Weltevredenpark, 1715
This communication is confidential and intended solely for the addressee(s). 
Any unauthorized review, use, disclosure or distribution is prohibited. If 
you believe this message has been sent to you in error, please notify the 
sender by replying to this transmission and delete the message without 
disclosing it. Thank you.E-mail including attachments is susceptible to data 
corruption, interception, unauthorized amendment, tampering and viruses, and 
we only send and receive emails on the basis that we are not liable for any 
such corruption, interception, amendment, tampering or viruses or any 
consequences thereof.



From: Mitul Limbani [mailto:mi...@enterux.in]
Sent: Friday, November 21, 2014 1:04 PM
To: Andrew Colin
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] One way audio internal



Then something to do with your codec selection priority.

On 21-Nov-2014 4:26 PM, Andrew Colin and...@convergedgroup.net wrote:

I am using the free g729







Kind Regards

Andrew Colin

Converged Data (Pty) Ltd.

Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09)





Direct: +27 (0)10 591 4607 tel:%2B27%20%280%2910%20591%204607

Mobile: +27 (0)82 310 3007 tel:%2B27%20%280%2982%20310%203007
Switchboard: +27 (0)10 591 4600 tel:%2B27%20%280%2910%20591%204600
Email: and...@convergedgroup.net

Web: http://www.convergedgroup.net
75 Witkoppen Road, Northriding, Johannesburg, 2169
P O Box 7246, Weltevredenpark, 1715
This communication is confidential and intended solely for the addressee(s). 
Any unauthorized review, use, disclosure or distribution is prohibited. If 
you believe this message has been sent to you in error, please notify the 
sender by replying to this transmission and delete the message without 
disclosing it. Thank you.E-mail including attachments is susceptible to data 
corruption, interception, unauthorized amendment, tampering and viruses, and 
we only send and receive emails on the basis that we are not liable for any 
such corruption, interception, amendment, tampering or viruses or any 
consequences thereof.



From: Mitul Limbani [mailto:mi...@enterux.in]
Sent: Friday, November 21, 2014 12:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Andrew Colin
Subject: Re: [asterisk-users] One way audio internal



You probably do not have enough g729 channels license.

On 21-Nov-2014 4:17 PM, A J Stiles asterisk_l...@earthshod.co.uk wrote:

On Friday 21 Nov 2014, Andrew Colin wrote:
 Hi All

 We have a strange issue with our hosted asterisk server running on Debian
 Internal calls btween extensions using g729 give one way audio
 As soon as we change the codec to ALAW the issues goes away.

 Any ideas how to fix this?

 Outbound calls via a trunk work fine with g729

Unless you have serious bandwidth issues, just forget about g.729 and change
to a-law throughout.  A-law is what the PSTN  (in civilised countries)  uses
anyway, so you won't need to transcode  (which chews up processor resources
and risks compromising quality)  for calls to and from the outside world.

If you really need to use g.729 and are outside the USA  (therefore, beyond
the reach of software patents),  there is a free version that you can use --
and this one, better than Digium's offering, comes with the Source Code so 
you
can be sure it isn't doing anything nasty behind the scenes.

But to be honest, you probably are better off just sticking with a-law.

--
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] Realtime ERROR

[Andrew Colin]

Hi Guys,

 

I have recently moved my database servers to a different database cluster
that runs on haproxy.

Every minute or so I get the following error in asterisk

 

MySQL RealTime: Ping failed (2006).  Trying an explicit reconnect

 

The strange thing is if I do realtime mysql status

It shows as connected just the timer resets.

 

Any idea why this is occurring?

 

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Re: [asterisk-users] Realtime ERROR

[Andrew Colin]

Hi Rainer,

 

I am using roundrobin

 

From: Rainer Piper [mailto:rainer.pi...@soho-piper.de] 
Sent: Thursday, September 25, 2014 6:21 PM
To: Andrew Colin; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Realtime ERROR

 

Am 25.09.2014 um 16:24 schrieb Andrew Colin:

Hi Guys,

 

I have recently moved my database servers to a different database cluster
that runs on haproxy.

Every minute or so I get the following error in asterisk

 

MySQL RealTime: Ping failed (2006).  Trying an explicit reconnect

 

The strange thing is if I do realtime mysql status

It shows as connected just the timer resets.

 

Any idea why this is occurring?

 





Hi Andrew,

what balancing algorithm you use in haproxy.cfg  ?
balance source
balance roundrobin
or
balance leastconn



-- 
Rainer Piper 
Integration engineer 
Koeslinstr. 56 
53123 BONN 
GERMANY 
Phone:  callto:004922897167161 +49 228 97167161 
P2P: sip:rai...@sip.soho-piper.de:5072 (pjsip-test) 
XMPP: rai...@xmpp.soho-piper.de

  _  

No virus found in this message.
Checked by AVG - www.avg.com
Version: 2014.0.4765 / Virus Database: 4025/8267 - Release Date: 09/24/14

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Re: [asterisk-users] PRI answer too fast

[Derek Andrew]

Answer([delay])

If the channel is ringing, answer it, otherwise do nothing. If a delay is
specified, Asterisk will wait this number of milliseconds AFTER answering
the call. If you want to add a delay prior answering, use Wait.

On Mon, Sep 22, 2014 at 9:06 AM, Doug Lytle supp...@drdos.info wrote:

  is there a way to set answer on ring or something so the
  other end at least gets 1 ring.

 First entry in your incoming context should be:

 exten = s,1,Answer(500)

 Doug

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Copyright 2014 Derek Andrew (excluding quotations)

+1 306 966 4808
Information and Communications Technology
University of Saskatchewan
Peterson 120; 54 Innovation Boulevard
Saskatoon,Saskatchewan,Canada. S7N 2V3
Timezone GMT-6

Typed but not read.
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Re: [asterisk-users] is pattern matching inside macro valid?

[Derek Andrew]

There are some issues if you use WaitExten inside a macro.

On Mon, Sep 8, 2014 at 2:48 PM, Anurag Rana anuragrana31...@gmail.com
wrote:

   Can't we use pattern matching inside a macro?
  Because when I am trying to do so call is terminating even for a very
 simple dummy dialplan.

 [demo3]
 exten=98,1,NoOp()
 exten=98,2,Macro(testme)
 exten=h,1,NoOp(terminating call);

 [macro-testme]
 exten=s,1,Playback(Digits/2)
 exten=s,2,WaitExten(15)
 exten=s,3,NoOp()

 exten=_X,1,NoOp(${EXTEN})
 exten=_X,2,Goto(s,3)


  Even for this code when execution reaches the line 2 in macro 'testme'
 it terminates as soon as I input some number.

  Error :

 WARNING[9984][C-000d]: pbx.c:6696 __ast_pbx_run: Invalid extension
 '5', but no rule 'i' or 'e' in context 'demo3'
 -- Executing [h@demo3:1] NoOp(SIP/101-000d, terminating call)
 in new stack
 [Sep  9 02:11:14] NOTICE[9984]: pbx_spool.c:402 attempt_thread: Call
 completed to SIP/101/009871888729

  Anurag Rana
 http://newbie42.blogspot.in/






-- 
Copyright 2014 Derek Andrew (excluding quotations)

+1 306 966 4808
Information and Communications Technology
University of Saskatchewan
Peterson 120; 54 Innovation Boulevard
Saskatoon,Saskatchewan,Canada. S7N 2V3
Timezone GMT-6

Typed but not read.
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Re: [asterisk-users] asterisk multiple ip

[Derek Andrew]

You can achieve your goal with policy based routing (
http://en.wikipedia.org/wiki/Policy-based_routing). You would need to
install the iproute2 package and set up ip rules for routing.

This would allow you to answer endpoints registering on 192.168.10.30 with
the address 192.168.10.30.


On Fri, Aug 29, 2014 at 3:26 AM, Marek Cervenka cerv...@fpf.slu.cz wrote:

 hi,

 i need migrate customers from severeal to one asterisk server with
 multiple ip aliases
 like
 eth0 192.168.10.1
 eth0:1 192.168.10.20
 eth0:2 192.168.10.30

 i must preserve endpoint configuration to these ip adressess

 the problem is if i register to 192.168.10.30, the answer is from
 192.168.10.1

 are there some ways for this scenario?
 1) chan_pjsip?
 2) kamailio in front of asterisk on the same server?
 3) iptables magic?
 4) ...

 thanks

 --
 ---
 Marek Cervenka
 ===


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[asterisk-users] Strange Error

[Andrew Colin]

Hi Guys,

 

Does anyone know what this error means and how to fix it?

 

[Jul  3 11:57:27] WARNING[17040] pbx.c: Don't know what to do with 'SIP/

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Re: [asterisk-users] recording in mp3

[andrew Colin]

Can you explain?


Sent from Samsung Mobile

div Original message /divdivFrom: Tiago Geada 
tiago.ge...@gmail.com /divdivDate:03/07/2014  9:04 PM  (GMT+02:00) 
/divdivTo: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com /divdivSubject: Re: [asterisk-users] 
recording in mp3 /divdiv
/divno need.


mixmonitor has a argument that is a script ran just as the recording is 
finished.

we use this to move the file from ramfs to final destination.

you can use it to use sox and convert it...


On 2 July 2014 18:54, Dave Platt dpl...@radagast.org wrote:

 Problem with this is client needs to listen to the call recordings and my 
 interface will only display .wav or .mp3 so they will moan if they have to 
 wait until the next day for today's recordings

If you're up to writing a bit of shell script, and are running
on Linux, you could automate the conversion process so that it
happens as soon as the recording is completed.

Look at the inotify system service (man section 7) and the
inotifywatch program.  You can tell inotifywatch to monitor
files being written into a specific directory (or set of
directories) and output a series of events when files in this
directory are open or closed.

What you'd probably want to do, is catch the close_write
events (a file has been closed, and it had been opened in
a mode which allows it to be written). When you see a
close_write event for a recording file of the sort that
Asterisk writes, you'd check to see if it's been converted
to your desired format yet.  If not, fire off a separate
task (e.g. via batch) to convert it.

Here's a very simple script I did to do something like this...
run a periodic-processing script a few seconds after files
with a specific name pattern have been touched in any way.
It's not sophisticated enough to look only for close or
close_wait events, but it should give you the idea.

#!/bin/bash

function processevents () {
 action=0
 while true ; do
   if [ $action == 0 ] ; then
   timeout=300
   else
   timeout=5
   fi
   read -t $timeout event
   if [ $? != 0 ] ; then
  action=0
  /data/soundchaser/periodic
   else
  if [[ $event =~ .wav || $event =~ .gotit ]] ; then
  action=1
  fi
   fi
 done
}

cd /data/soundchaser

inotifywait -m /data/soundchaser/public_html/done | processevents


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Re: [asterisk-users] Call rating software

[Andrew Colin]

Can you try maybe assist with this, as I have tried for ages and still cant get 
it right.



-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A J Stiles
Sent: 02 July 2014 11:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call rating software

On Tuesday 01 Jul 2014, andrew Colin wrote:
 Hi Guys
 
 Does anyone know of any good cdr rating software.
 
 I am looking for something that I can pull reports by extension. 
 Not a full billing solution like a2billing.

Have you thought of rolling your own?  It's not hard to write a program in Your 
Favourite Scripting Language™ to pull the records you want from the database 
and create a CSV spreadsheet.  And then have it started by cron, to give you 
regular automatically-generated reports; either e-mailed directly to you, or 
downloadable via a web page.

Alternatively, you could generate your spreadsheet as an .fods  (flat XML) 
file.  This is slightly more effort; but it has the advantage of supporting 
styling of table cells, so your report can be made to look pretty.

--
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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-
No virus found in this message.
Checked by AVG - www.avg.com
Version: 2014.0.4592 / Virus Database: 3986/7781 - Release Date: 07/02/14


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Re: [asterisk-users] Call rating software

[andrew Colin]

Currently I am writing to mysql
With all the default fields in the cdr table in the asterisk databas


Sent from Samsung Mobile

div Original message /divdivFrom: A J Stiles 
asterisk_l...@earthshod.co.uk /divdivDate:02/07/2014  5:50 PM  
(GMT+02:00) /divdivTo: Asterisk Users Mailing List - Non-Commercial 
Discussion asterisk-users@lists.digium.com /divdivSubject: Re: 
[asterisk-users] Call rating software /divdiv
/divOn Wednesday 02 Jul 2014, Andrew Colin wrote:
 Can you try maybe assist with this, as I have tried for ages and still cant
 get it right.

Firstly, have you got CDR working and writing to some sort of database?  We 
use cdr_mysql; although the more modern recommendation is to use cdr_odbc  
(which is more generic, and will work with various database types)  even if 
you are using a MySQL database.


If you haven't got your CDR going into a database, then you need to sort that 
out *first*.

Once you have CDR working, then it's simply a question of determining what SQL 
queries you need to generate to produce your report; then writing a program to 
build up the queries, extract the results and present them in the form you 
want.


-- 
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] recording in mp3

[andrew Colin]

Problem with this is client needs to listen to the call recordings and my 
interface will only display .wav or .mp3 so they will moan if they have to wait 
until the next day for today's recordings


Sent from Samsung Mobile

div Original message /divdivFrom: binary 
dreamer.bin...@gmail.com /divdivDate:01/07/2014  6:09 PM  (GMT+02:00) 
/divdivTo: asterisk-users@lists.digium.com /divdivSubject: Re: 
[asterisk-users] recording in mp3 /divdiv
/divi would go for recording into wav.
then at regular intervals eg every night at 01:00 i would start a script to 
convert the wav to mp3 and then delete the wav files.
it is really easy.



On 30/6/2014 23:30, Scott Griepentrog wrote:
​You will not be able to able to save much space if any by using MP3 instead of 
ulaw or wav -- at least not without expending a lot   of CPU time to 
encode the file at a very low bitrate which sounds pretty bad even with just 
speech.  One of the better space savings options for recordings or voicemail is 
gsm.  Of   course, using an MP3 format just because you ​prefer that is 
understandable.

Additionally, I'm nearly 100% certain that Asterisk does not support encoding 
and directly writing MP3 files.



On Mon, Jun 30, 2014 at 3:11 PM, andrew Colin and...@vsave.co.za   
wrote:
Hey guys

Is it possible to record with mixmonitor straight into mp3.

I am trying to reduce disk space and want my calls to be recorded in mp3 
Instead of wav.




Sent from Samsung Mobile


 Original message 
From: Sameer Rathod
Date:30/06/2014 9:23 PM (GMT+02:00)
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Fwd: Regarding packet2packet bridging


Dear concern,


I want to configure packet2packet bridging in asterisk.
How could I do this any of the tutorial or instructions will help ?

I found the setting the canreinvite=yes  will do the stuff but it is not 
working 

I am using asterisk 12.3 version 

I am very new to asterisk please help me in doing the same.

Thanks in advance.  

-- 
Regards
Sameer Rathod
8109413462 




-- 
Regards
Sameer Rathod
8109413462 


--
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-- 

Scott Griepentrog
Digium, Inc · Software Developer
445 Jan Davis Drive NW · Huntsville, AL 35806 · US
direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029
Check us out at: http://digium.com · http://asterisk.org



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Re: [asterisk-users] recording in mp3

[andrew Colin]

Currently using tikal crystal call recording

Do you guys know of any better ones?



Sent from Samsung Mobile

div Original message /divdivFrom: binary 
dreamer.bin...@gmail.com /divdivDate:01/07/2014  6:33 PM  (GMT+02:00) 
/divdivTo: asterisk-users@lists.digium.com /divdivSubject: Re: 
[asterisk-users] recording in mp3 /divdiv
/divwhat is your interface?



On 1/7/2014 19:13, andrew Colin wrote:
Problem with this is client needs to listen to the call recordings and my 
interface will only display .wav or .mp3 so   they will moan if they 
have to wait until the next day for today's recordings


Sent from Samsung Mobile


 Original message 
From: binary
Date:01/07/2014 6:09 PM (GMT+02:00)
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] recording in mp3

i would go for recording into wav.
then at regular intervals eg every night at 01:00 i would start a script to 
convert the wav to mp3 and then delete the wav files.
it is really easy.



On 30/6/2014 23:30, Scott Griepentrog wrote:
​You will not be able to able to save much space if any by using MP3 instead of 
ulaw or wav -- at least not without expending a lot of CPU time to encode the 
file at a very low bitrate which sounds pretty bad even with just 
speech.  One of the better space savings options for recordings or voicemail is 
gsm.  Of course, using an MP3 format just because you ​prefer that is 
understandable.

Additionally, I'm nearly 100% certain that Asterisk does not support encoding 
and directly writing MP3 files.



On Mon, Jun 30, 2014 at 3:11 PM, andrew Colin and...@vsave.co.za wrote:
Hey guys

Is it possible to record with mixmonitor straight into mp3.

I am trying to reduce disk space and want my calls to be recorded in mp3 
Instead of wav.




Sent from Samsung Mobile


 Original message 
From: Sameer Rathod
Date:30/06/2014 9:23 PM (GMT+02:00)
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Fwd: Regarding packet2packet bridging


Dear concern,


I want to configure packet2packet bridging in asterisk.
How could I do this any of the tutorial or   
instructions will help ?

I found the setting the canreinvite=yes  will do the stuff but it is not 
working 

I am using asterisk 12.3 version 

I am very new to asterisk please help me in doing the same.

Thanks in advance.  

-- 
Regards
Sameer Rathod
8109413462 




-- 
Regards
Sameer Rathod
8109413462 


--
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-- 

Scott Griepentrog
Digium, Inc · Software Developer
445 Jan Davis Drive NW · Huntsville, AL 35806 · US
direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029
Check us out at: http://digium.com · http://asterisk.org






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[asterisk-users] Call rating software

[andrew Colin]

Hi Guys

Does anyone know of any good cdr rating software.

I am looking for something that I can pull reports by extension. 
Not a full billing solution like a2billing.



Sent from Samsung Mobile-- 
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Re: [asterisk-users] recording in mp3

[andrew Colin]

Hey guys

Is it possible to record with mixmonitor straight into mp3.

I am trying to reduce disk space and want my calls to be recorded in mp3 
Instead of wav.




Sent from Samsung Mobile

div Original message /divdivFrom: Sameer Rathod 
sam...@hostnsoft.com /divdivDate:30/06/2014  9:23 PM  (GMT+02:00) 
/divdivTo: asterisk-users@lists.digium.com /divdivSubject: 
[asterisk-users] Fwd: Regarding packet2packet bridging /divdiv
/div
Dear concern,


I want to configure packet2packet bridging in asterisk.
How could I do this any of the tutorial or instructions will help ?

I found the setting the canreinvite=yes  will do the stuff but it is not 
working 

I am using asterisk 12.3 version 

I am very new to asterisk please help me in doing the same.

Thanks in advance.  

-- 
Regards
Sameer Rathod
8109413462 




-- 
Regards
Sameer Rathod
8109413462 

-- 
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Re: [asterisk-users] Attack on Sip server.

[andrew Colin]

Block the ip?

You should only enable sip for your specific clients in iptables.


Sent from Samsung Mobile

div Original message /divdivFrom: arun kumar 
arunvsadni...@gmail.com /divdivDate:27/06/2014  4:42 PM  (GMT+02:00) 
/divdivTo: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com /divdivSubject: Re: [asterisk-users] 
Attack on Sip server. /divdiv
/divHi,

Change the protocol from tcp to udp in iptables.

~Arun

On 27 Jun 2014 20:07, Anurag Rana anuragrana31...@gmail.com wrote:

Hi All.

Someone is attacking on my SIP server.
There are lot of requests coming in and I am not able to stop it because I am 
unable to detect the IP address. 
I used wireshark to capture the packets.

Although I am using very strong password for my SIP users but still is there 
any way to drop these packets and stop this attack.

I tried dropping packet after matching some string (most of the packets from 
attacker contains string 'VaxSIPUserAgent/3.1' ) but it failed. Packets are 
still flowing in. 

iptables -I INPUT 1 -p tcp --dport 5060 -m string --string VaxSIPUserAgent 
--algo bm -j DROP

​Its something like this

Registration from '30 sp:30@my_public_ip:5060 failed for 
'192.168.xxx.xxx:6373' - Wrong Password​

​and there are approx 10 request per minute of this type.

Please suggest some way to stop this.​


-- 
Anurag Rana 
http://newbie42.blogspot.in/
On the trampoline of life's experiences, Striving towards a saintly life in the 
midst of these materialistic turbulences.



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Re: [asterisk-users] Reload problems with 1.8.27 and 11.9.0 - Someone else ?

[Derek Andrew]

Does a reload (not a sip reload) reload everything or does it also require
the sip.conf file to be modified?


On Wed, Apr 30, 2014 at 5:00 AM, Administrator TOOTAI ad...@tootai.netwrote:

 Le 30/04/2014 12:39, Administrator TOOTAI a écrit :

 Le 30/04/2014 12:15, Administrator TOOTAI a écrit :

 Hi,

 after upgrade from 11.8.1 to 11.9.0 on our test server, and from
 1.8.26.1 to 1.8.27 on production one, some CLI commands like sip reload
 or iax2 reload does nothing.

 We opened bug 23683 but it was immediately closed by Matt Jordan,
 telling that he can't reproduce it. But we can.

 Example:

 - switching back to 11.8.1 respectively 1.8.26.1 does the job working
 again (We just run a make install from within this directory)
 - cleaning 11.8.0 source directory -make clean  ./configure  make 
 make install- all is good
 - cleaning 11.9.0 source directory -make clean  ./configure  make 
 make install- problem appears again
 - switching back to 11.8.0 does the job working again (We just run a
 make install from within this directory)

 The first installation of latest version was done by patching the
 previous version, we downloaded the source tar.gz and compile = problem
 stays

 Does anybody else face this problem with latest version? If it was a
 server problem, earlier version should have same behaviour after compiling
 but they don't.

 Server OS is Debian Wheezy 3.2.0-4 amd64 in KVM virtual machine

 Thanks for any hint

 Regards


 We checked on a customer installation made one week ago: they have the
 same problem! It's a Debian Squeeze 2.6.32-5-amd64 on a real server.


 And finally the explanation: if you modify sip.conf file, the reload is
 taken in account, all is good. But if the sip.conf contains includes and
 you modify one of those includes *without modifying* sip.conf, no reload.

 --
 Daniel

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Re: [asterisk-users] stopping unwanted attempts

[Andrew Colin]

Geoip works well to block all countries except your own


Regards
Andrew Colin-mobile
Vsave(PTY)Ltd



 Original message 
From: Eric Wieling ewiel...@nyigc.com 
Date:19/01/2014  8:40 PM  (GMT+02:00) 
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com 
Subject: Re: [asterisk-users] stopping unwanted attempts 


It is far worse when you have multiple phones behind the same public address 
(i.e. NAT).    If any one of the phones has a bad password and the IP gets 
blocked by fail2ban, then all phones at that site would be blocked. 

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Bagnall
Sent: Sunday, January 19, 2014 10:40 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] stopping unwanted attempts

On 19/1/14 2:57 pm, Ron Wheeler wrote:
 fail2ban is so easy to set up, there is no reason not to set it up.

One of the dangers with fail2ban - at least in its default configuration
- is that a legitimate SIP phone with an incorrect password can quite easily 
send dozens of registration attempts in a couple of minutes, thus blocking that 
IP.


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Re: [asterisk-users] stopping unwanted attempts

[Andrew Colin]

Fail2ban works well otherwise you can write your own script im bash or perl to 
block them in iptables


Regards
Andrew Colin-mobile
Vsave(PTY)Ltd



 Original message 
From: Jerry Geis ge...@pagestation.com 
Date:18/01/2014  10:59 PM  (GMT+02:00) 
To: asterisk-users@lists.digium.com 
Subject: [asterisk-users] stopping unwanted attempts 

I see MANY of these in my log files:


[Jan 15 03:06:12] NOTICE[14129] chan_sip.c: Registration from '202 
sip:202@X:5060' failed for '37.8.12.147:26832' - Wrong password
[Jan 15 03:06:19] NOTICE[14129] chan_sip.c: Registration from '5001 
sip:5001@X:5060' failed for '37.8.12.147:21268' - Wrong password
[Jan 15 03:06:23] NOTICE[14129] chan_sip.c: Registration from '30 
sip:30@X:5060' failed for '37.8.12.147:21270' - Wrong password
[Jan 15 03:06:48] NOTICE[14129] chan_sip.c: Registration from '70 
sip:70@X:5060' failed for '37.8.12.147:21328' - Wrong password
[Jan 15 03:06:50] NOTICE[14129][C-0085] chan_sip.c: Call from '' 
(8.33.7.110:5103) to extension '889011972592735467' rejected because extension 
not found in context 'default'.
[Jan 15 03:06:56] NOTICE[14129] chan_sip.c: Registration from '4 
sip:4@X:5060' failed for '37.8.12.147:21272' - Wrong password
[Jan 15 03:07:11] NOTICE[14129] chan_sip.c: Registration from '12001 
sip:12001@X:5060' failed for '37.8.12.147:5060' - Wrong password
[Jan 15 03:34:02] NOTICE[14129][C-0086] chan_sip.c: Call from '' 
(172.246.236.90:5078) to extension '8889011972595301123' rejected because 
extension not found in context 'default'.

What is the correct way to block these idiots so they
don't even get this far.

Thanks,

Jerry

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