[asterisk-users] Professional Setup..

[Dave Walker]

Greetings,I have a question for those who have done a few professional installs of Asterisk.   Is it taboo to use something like AsteriskNow/FreePBX/Trixbox to get a base installation of Asterisk installed and functional for a small office?  If not then do you always compile from scratch or use CentOS and the yum repositories?   I setup FreePBX for a friend of mine who runs a small call center and I feel guilty profiting from something that so many other people have worked so hard to create.  Ironically www.asterisknow.org has a 'Download - Free" link but I don't see a 'donate' link.    I wouldn't mind demanding my friend contribute some amount to help Asterisk grow but I'm not exactly sure how to do that and what the appropriate amount should be.  I know Digium is a proud sponsor so I buy my phones, cards and refer people to their site if they need to buy equipment or an appliance.  Has this topic come up for conversation in the past and if so then what was the outcome?Thanks!Dave

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Re: [asterisk-users] VoIP over satellite internet

[Don E. Wisdom]

I work on the salmon river in Idaho as a computer/radio tech.
All of the satellite isp's do not have the upstream capability.
Skype barely works. (you have to try upwards of 20 times for it to work)
If I have to make phone calls when I am there I always use the SSB
Radiophone or satellite phone because it is far far far more reliable and
doesn't irritate the living hell out of the person your calling.
I have tried 2-3 different VoIP providers & all have the exact same result.
The other side only hears a few pieces of word or nothing at all and hangs
up.
I have tried this on Starband (360 & 480 modems) & wild blue
Starband also has outages during the day where you cant see their satellite.
Most of the satellite ISP's also have rolling bandwith caps.  (Starbands is
1gig down & 300megs up in a 7 day period for the plans I deal with)
Overall I think its a bad idea.  It most likely will not work.

--Don



On 5/8/09 10:56 PM, "Frank Bulk"  wrote:

> If people don't mind taking turns talking, it will "work".  It's just going
> to be like talking on a CB.  Reminds me of talking to my grandparents in the
> Europe as a child in the early 80's.
> 
> Frank
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Fort
> Sent: Friday, May 08, 2009 10:30 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] VoIP over satellite internet
> 
> Could those on the list who have used or tried to use VoIP over a
> satellite internet connection comment on how well it works or if it
> even works at all in a reliable way.  What is the effect of latency on
> the VoIP path and how much is generally tolerable?  routing via
> satellite adds about a quarter second of latency to the path.  Is that
> too much?
> 
> Eric
> 
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Re: [asterisk-users] Unable to run asterisk CLI commands from php

[Tzafrir Cohen]

On Sat, May 09, 2009 at 10:17:27AM +0530, Sam Hawkin wrote:
> Hi All,
> 
> I am trying to run the asterisk CLI commands from php.
> Some thing like "asterisk -rx reload".
> But it is not working, where as when I try to run
> linux "ls" command it works fine.
> 
> below is sample php code which I am trying to run.
> 
> $command = "sudo asterisk -rx reload";
> 
> $value1 = system($command,$retval1);
> $value2 = exec($command,$retval2);
> 
> echo $value1;
> echo $value2;
> 
> can any one suggest what might be the problem and possible solution to it.

Yes. What error do you actually get? You need to get stderr from hte
command as well for that.

What version of Asterisk do you use, BTW?

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] VoIP over satellite internet

[Frank Bulk]

If people don't mind taking turns talking, it will "work".  It's just going
to be like talking on a CB.  Reminds me of talking to my grandparents in the
Europe as a child in the early 80's.  

Frank

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Fort
Sent: Friday, May 08, 2009 10:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] VoIP over satellite internet

Could those on the list who have used or tried to use VoIP over a
satellite internet connection comment on how well it works or if it
even works at all in a reliable way.  What is the effect of latency on
the VoIP path and how much is generally tolerable?  routing via
satellite adds about a quarter second of latency to the path.  Is that
too much?

Eric

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[asterisk-users] Unable to run asterisk CLI commands from php

[Sam Hawkin]

Hi All,

I am trying to run the asterisk CLI commands from php.
Some thing like "asterisk -rx reload".
But it is not working, where as when I try to run
linux "ls" command it works fine.

below is sample php code which I am trying to run.

$command = "sudo asterisk -rx reload";

$value1 = system($command,$retval1);
$value2 = exec($command,$retval2);

echo $value1;
echo $value2;

can any one suggest what might be the problem and possible solution to it.

Any help is highly appreciated.

Thanks.
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Re: [asterisk-users] VoIP over satellite internet

[Tom Moore]

Voip over satellite can be done if enough bandwidth is reserved properly for
it.
Use the g729 codec and ask for 24 kilobits of upstream cir and you should be
fine.
Also you'll want to mark your packets with the EF tos bit in sip.conf.
If done right the delay isn't too bad. Yes you can tell it is there, but as
long as your latency doesn't fluctuate too much you should be alright.

>From my experience as far as satellite platforms are concerned go with
Idirect when possible. 
I've also used the Viasat systems as well and from all the installs I've
seen the hub operators never really understood how to run the hub to
optimize for voip so I've pretty much given up on that platform. It seems
that Idirect operators are the only ones who know how to make the system
work well.
If you want the best of the best nothing touches C-band though.
Yes its low bandwidth, but C-band you can bet your life on...
Most low end sat services are way over subscribed to start with and when you
add voip in to the mix voip is the first thing to lose.
 
Tom

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Fort
Sent: Friday, May 08, 2009 11:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] VoIP over satellite internet

Could those on the list who have used or tried to use VoIP over a satellite
internet connection comment on how well it works or if it even works at all
in a reliable way.  What is the effect of latency on the VoIP path and how
much is generally tolerable?  routing via satellite adds about a quarter
second of latency to the path.  Is that too much?

Eric

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[asterisk-users] VoIP over satellite internet

[Eric Fort]

Could those on the list who have used or tried to use VoIP over a
satellite internet connection comment on how well it works or if it
even works at all in a reliable way.  What is the effect of latency on
the VoIP path and how much is generally tolerable?  routing via
satellite adds about a quarter second of latency to the path.  Is that
too much?

Eric

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Re: [asterisk-users] IPv6 support?

[Andrew Ruthven]

On Tue, 2009-04-28 at 00:06 +0200, Hans Witvliet wrote:
> Sometime ago i got this status from _the_ guru...
> Russell replied, referencing 1.6.2...
> (but other code might get in-the-way)

Thanks!

> From: Russell Bryant 
> > There has been progress.  It is not yet merged into the main tree, though.  
> > I 
> > would expect it to go in within the first few releases of 1.6 ...

Can anyone point me to where the IPv6 development is currently
happening, I'd be more than happy to start playing with it.

Cheers!

-- 
Andrew Ruthven, Wellington, New Zealand
At work: andrew.ruth...@catalyst.net.nz
At home: and...@etc.gen.nz
GPG fpr: 34CA 12A3 C6F8 B156 72C2  D0D7 D286 CE0C 0C62 B791

Follow the Signs, visit Wellington!  Linux.conf.au 2010
  http://www.lca2010.org.nz


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Re: [asterisk-users] Possible to add Voice delay?

[George Farris]

On Fri, 2009-05-08 at 18:31 -0700, Trevor Peirce wrote:
> George Farris wrote:
> > When I have call forward working or use the voicemail from a SIP phone,
> > the first part of the message is always cut off.  So instead of hearing
> > "call forward cancelled"  I hear "l forward cancelled".
> >
> > Or in voicemail I hear "edian mail" instead of "comedian mail".
> >
> > How can I add a delay so one can hear the entire message or is this a
> > bug in Asterisk?  
> >   
> 
> This is what I've always done:
> 
> 8500,1,Playback(silence/1)
> 8500,n,VoiceMailMain
> 

Yes perfect Dave Platt suggested that as well, works very well.  I have
no idea why I didn't think of it.  Brain fart.

Thanks all



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Re: [asterisk-users] Storage capacity for call recording

[Steve Edwards]

On Fri, 8 May 2009, Michelle Dupuis wrote:

> I want to record calls in wav format.  Can someone tell me how many MB 
> of storage per minute each recording requires (assuming SIP / uLaw codec 
> / full duplex recording)

Your choice of technology and codec are irrelevant to the size of wav file 
produced. Wav files (as produced by Asterisk) are usually "PCM, 16 bit, 
mono 8000 Hz." Asterisk can produce funky wav49 ("GSM 6.10, mono 8000 Hz") 
files but neither monitor() or mixmonitor() support it.

Using the following dialplan snippet:

; test monitor
 exten = *,n,set(TIMEOUT(absolute)=600)
 exten = *,n,monitor(wav:,/tmp/${EPOCH})
;   exten = *,n,mixmonitor(/tmp/${EPOCH}.wav)
 exten = *,n(loop),  playback(demo-congrats)
 exten = *,n,goto(loop)
 exten = *,n,hangup()

I placed 6 simultaneous calls on my over-priced and under-powered Fit-PC 
(500 MHz AMD Geode):

Application Format  CPU File
--- --  --- 
monitor wav 9%  934 kilo-byte per second per leg
monitor gsm 70% 96 kilo-bytes per second per leg
mixmonitor  wav 7%  936 kilo-byte per second
mixmonitor  gsm 36% 96 kilo-bytes per second

Remember that monitor records each leg of the call in a separate file. You 
can use the "m" option or sox to mix them together into a single file at 
the end of the call.

Thanks in advance,

Steve Edwards  sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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Re: [asterisk-users] determination of where a call is placed from (physical location)

[Steve Edwards]

On Fri, 8 May 2009, Eric Fort wrote:

> I am interested in setting up asterisk to record all calls it
> processes.  there are however some legal quirks to doing this that I
> have run across such as one party notification vs 2 party notification
> requirements which depend upon the physical endpoints of a call.  If I
> wish to discretely record all incoming calls except those calling from
> a place requiring 2 party notification (thus avoiding the need for an
> overt announcement) How can I afirmatively determine the physical
> location of the party originating the call and thus the applicable
> legal jurisdiction.

With local number portability, callerid spoofing services, and VOIP 
providers providing DIDs (which are sometimes set as the outgoing 
callerid) in virtually any "geographic" location on earth, I think you're 
out of luck.

Ask your legal beagles if "appearing to originate in a particular physical 
area" is a reasonable effort.

Thanks in advance,

Steve Edwards  sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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Re: [asterisk-users] How to get meetme participants in dialplan?

[Steve Edwards]

On Fri, 8 May 2009, David Backeberg wrote:

> On Fri, May 8, 2009 at 4:54 PM, Steve Edwards  
> wrote:
>> On Fri, 8 May 2009, David Backeberg wrote:
>>> You need a way to keep state. I use a database and AGI for that purpose.
>>
>> I thought about keeping state in the db, but then I'd have to run the AGI
>> when they leave the conference or hangup as well. These callers jump from
>> conference to conference with a single key press and use this feature
>> constantly.
>
> I'm having a better handle on what you're trying to do here. If the
> users are trusted to jump among conferences, can't they also be
> trusted to mute/unmute/hangup themselves? It seems to me that the main
> thing you would want to do for some level of administrative control
> would be to hangup the user's channel.

It's a "free" chat system. Some of the callers are a bunch of bottom 
feeding mouth breathers who have nothing better to do than scream and 
offend the other callers.

> Shouldn't the user channel stay consistent regardless of where they go 
> in the system?

The administrators are "flying blind." All they have is a handset and a 
dialpad. When they go into a conference, if everybody is behaving, they 
move on to the next conference. If somebody is misbehaving, they need a 
way to identify and potentially kick them out of the conference. Muting 
and un-muting the callers 1 by 1 is a bit of a chore, but "ya work with 
what ya got."

Thanks in advance,

Steve Edwards  sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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[asterisk-users] determination of where a call is placed from (physical location)

[Eric Fort]

I am interested in setting up asterisk to record all calls it
processes.  there are however some legal quirks to doing this that I
have run across such as one party notification vs 2 party notification
requirements which depend upon the physical endpoints of a call.  If I
wish to discretely record all incoming calls except those calling from
a place requiring 2 party notification (thus avoiding the need for an
overt announcement) How can I afirmatively determine the physical
location of the party originating the call and thus the applicable
legal jurisdiction.

Eric Fort

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Re: [asterisk-users] How to get meetme participants in dialplan?

[David Backeberg]

On Fri, May 8, 2009 at 4:54 PM, Steve Edwards  wrote:
> On Fri, 8 May 2009, David Backeberg wrote:
>> You need a way to keep state. I use a database and AGI for that purpose.
>
> I thought about keeping state in the db, but then I'd have to run the AGI
> when they leave the conference or hangup as well. These callers jump from
> conference to conference with a single key press and use this feature
> constantly.

I'm having a better handle on what you're trying to do here. If the
users are trusted to jump among conferences, can't they also be
trusted to mute/unmute/hangup themselves? It seems to me that the main
thing you would want to do for some level of administrative control
would be to hangup the user's channel. Shouldn't the user channel stay
consistent regardless of where they go in the system?

And if you know that when they first dial in, shouldn't you be able to
find them later with a:
show channels
?

> Sometimes its a drag being a 1.2 Luddite. Is "meetme concise" a CLI
> command? If so, I'm still stuck parsing the output of an AMI command.

Yes. And true.

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Re: [asterisk-users] Record all calls

[David Backeberg]

On Fri, May 8, 2009 at 5:31 PM, Michelle Dupuis  wrote:
> I'd like to setup a single extension for which all INBOUND and OUTBOUND
> calls are recorded to a wav file.  I took a look at the wiki:
>
> http://www.voip-info.org/wiki/view/Asterisk+record+calls
>
> but it's not too helpful.  Can someone show some sample code in & out
> recording?

You don't need to set up a single extension to do all the work. Any
extension can be set to record, regardless if if that extension is
receiving or dialing. Applications that assist with setting up
recording on an extension include:
Monitor()
MixMonitor()

Take a look at those.

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Re: [asterisk-users] Possible to add Voice delay?

[Trevor Peirce]

George Farris wrote:
> When I have call forward working or use the voicemail from a SIP phone,
> the first part of the message is always cut off.  So instead of hearing
> "call forward cancelled"  I hear "l forward cancelled".
>
> Or in voicemail I hear "edian mail" instead of "comedian mail".
>
> How can I add a delay so one can hear the entire message or is this a
> bug in Asterisk?  
>   

This is what I've always done:

8500,1,Playback(silence/1)
8500,n,VoiceMailMain

Which will answer the line, play back 1 second of silence, then go to 
Comedian Mail without clipping the first little bit.

Regards,

-- 
Trevor Peirce
Digital Conceptions Canada

http://www.digitalcon.ca
1-888-606-3030 / 250 483-0386


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Re: [asterisk-users] Possible to add Voice delay?

[Dan Caescu]

sox original-file.gsm appended-file.gsm pad 1.5

works with wav too.

Dan 
-Original Message-
Sent: Friday, May 08, 2009 6:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Possible to add Voice delay?

Hi all,

This is my first post to the list.

I have searched the net far and wide but can't find an answer to this
problem.

When I have call forward working or use the voicemail from a SIP phone,
the first part of the message is always cut off.  So instead of hearing
"call forward cancelled"  I hear "l forward cancelled".

Or in voicemail I hear "edian mail" instead of "comedian mail".

How can I add a delay so one can hear the entire message or is this a
bug in Asterisk?  

I'm using 1.4.21-2 in Ubuntu 9.04 and Grandstream, Aastra and Polycom
phones all experience this problem.

Thanks
George




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Re: [asterisk-users] chan_mobile and DTMF

[Carlos Ruiz Diaz]

Hello!

I tested the last revision of chan_mobile.c but I am still getting errors
during the call. The last time the error starts when the call was
established but with this new revision it starts sooner during the call
attempting.

The following error is reported:

[May  8 19:40:48] ERROR[6971]: chan_mobile.c:1038 mbl_read: read error 107

Regards.

Carlos.


On Thu, May 7, 2009 at 10:37 AM, Carlos Ruiz Diaz  wrote:

> Thank you.
>
> Please, any patch you think can have possibility to work, just let me know
> and I will do the tests and post the results.
>
> Regards
>
>
> On Thu, May 7, 2009 at 10:33 AM, Matthew Nicholson 
> wrote:
>
>> Sorry.  I am not sure which patch I wanted you to try now.  The issue I
>> posted may be related to your issue.
>>
>> On Wed, 2009-05-06 at 14:53 -0400, Carlos Ruiz Diaz wrote:
>> > I think I misunderstood your mail.
>> > There is no patch available yet, right?
>> >
>> > I went to the page you linked but I did not found a patch file.
>> >
>> > On Wed, May 6, 2009 at 2:45 PM, Carlos Ruiz Diaz
>> >  wrote:
>> > Thank you very much!
>> >
>> > I'll try with the patch and post the results.
>> >
>> >
>> >
>> > On Wed, May 6, 2009 at 12:37 PM, Matthew Nicholson
>> >  wrote:
>> > Try the patch on this bug
>> > http://bugs.digium.com/view.php?id=15042
>> >
>> > I don't get that error with my setup, but others have
>> > seen it.  I am
>> > fairly sure of what is causing it.  Still working on a
>> > fix.
>> >
>> >
>> > On Tue, 2009-05-05 at 22:52 -0400, Carlos Ruiz Diaz
>> > wrote:
>> > > I get a lot errors from chan_mobile when a call is
>> > in progress. More
>> > > than one line inserted every second.
>> > >
>> > > ERROR[6312]: chan_mobile.c:1050 mbl_read: read error
>> > 9
>> > >
>> > > Regards
>> > >
>> > >
>> > >
>> > > On Tue, May 5, 2009 at 9:00 PM, Carlos Ruiz Diaz
>> > >  wrote:
>> > > Thank you for your reply!
>> > >
>> > > I downloaded the latest revision of asterisk
>> > trunk and
>> > > asterisk-addons trunk but it's not working
>> > at all, no key
>> > > pulsation was detected. The last stable
>> > release at least
>> > > detects first key pulsation.
>> > >
>> > > I checked out using:
>> > >
>> > > svn checkout
>> > http://svn.digium.com/svn/asterisk-addons/trunk
>> > > asterisk-addons
>> > >
>> > > and
>> > >
>> > > svn checkout
>> > http://svn.digium.com/svn/asterisk/trunk asterisk
>> > >
>> > > I also tried:
>> > >
>> > > svn co
>> > >
>> >
>> http://svn.digium.com/svn/asterisk-addons/branches/1.6.2
>> > >
>> > > and
>> > >
>> > > svn co
>> > http://svn.digium.com/svn/asterisk/branches/1.6.2
>> > >
>> > > but compiling chan_mobile.c produce errors.
>> > >
>> > > What can I do?
>> > >
>> > > Thanks!
>> > >
>> > >
>> > >
>> > > On Tue, May 5, 2009 at 5:46 PM, Matthew
>> > Nicholson
>> > >  wrote:
>> > > This is a known bug.  It is fixed in
>> > the trunk version
>> > > of chan_mobile.
>> > >
>> > >
>> > > On Tue, 2009-05-05 at 11:38 -0400,
>> > Carlos Ruiz Diaz
>> > > wrote:
>> > > > Hello list,
>> > > >
>> > > > I recently started testing the
>> > chan_mobile addon and
>> > > after a
>> > > > successful installation and
>> > configuration I have a
>> > > couple of problems
>> > > > that I can't fix without your
>> > help.
>> > > >
>> > > > I am using opensuse 11.1, asterisk
>> >  

Re: [asterisk-users] Sangoma a104d and channel banks

[Jim Dickenson]

The problem turned out to be that I had a typo in my include => line so I
was not including the extensions that I intended to in extensions.conf.

Unlike chan_sip, chan_dahdi does not print the nice messages on CLI that
says you have dialed an unknown extension. As I was use to these messages
for SIP I did not associate my problem with unknown extensions.

To find this the console line in logger.conf got debug added to it and I
cranked up the debug level and eventually saw an error about no matching
extension.
-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



> From: Steve Davies 
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Date: Fri, 8 May 2009 10:21:26 +0100
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Subject: Re: [asterisk-users] Sangoma a104d and channel banks
> 
> Hi,
> 
> This may be completely wrong, but I have a feeling it may be related.
> Have you enabled "overlapdialling" in zapata.conf for the channels
> that are on the channelbank? If not, the 1st digit will be sent in,
> not match the dialplan, and be hungup.
> 
> *7xxx is probably working because that matches a built-in feature
> (transfer?), so the system KNOWS already that it needs more digits.
> 
> Cheers,
> Steve
> 
> 2009/5/7 Jim Dickenson :
>> I have more info now.
>> 
>> I have used AMI to watch events, including DTMF events. The phone comes off
>> hook and I see a line on the * CLI saying the line is off hook. I then touch
>> a key and I see the DTMF event in the AMI. At the same time I see a message
>> on CLI that says hangup the line.
> 
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[asterisk-users] Possible to add Voice delay?

[George Farris]

Hi all,

This is my first post to the list.

I have searched the net far and wide but can't find an answer to this
problem.

When I have call forward working or use the voicemail from a SIP phone,
the first part of the message is always cut off.  So instead of hearing
"call forward cancelled"  I hear "l forward cancelled".

Or in voicemail I hear "edian mail" instead of "comedian mail".

How can I add a delay so one can hear the entire message or is this a
bug in Asterisk?  

I'm using 1.4.21-2 in Ubuntu 9.04 and Grandstream, Aastra and Polycom
phones all experience this problem.

Thanks
George




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Re: [asterisk-users] QoS & VPN

[Brent Davidson]

Jeremy Mann wrote:
> Access-list 100 permit ip host  any
>
> Class-map match-any voip
>  Match access-group 100
>
> Policy-map voip
>  Class voip
>   Priority 256
>  Class class-default
>   Fair-queue
>
> Interface fastethernet 0
>  Service-policy output voip
>
>
> Above is what I do to prioritize 256kbit of outbound bandwidth to voip calls, 
> adjust accordingly.  You must also use the qos pre-classify in your ipsec 
> tunnel definitions for this to work, but it does work well.  I know I'm 
> potentially mapping other traffic than voip, but I'm lazy and don't want to 
> classify the rtp and sip and iax ports, rarely does the box do any other 
> traffic than voip as updates occur in off hours.
>
> You'll probably additionally want to match your ipsec keying traffic and give 
> it priority bandwidth, if you're going to push voip through the tunnel you'll 
> find yourself rekeying more often and want to make sure on a saturated link 
> it gets priority so the tunnels don't drop.
>
> If you're on DSL, you probably want to research cascading the Qos, have a 
> root policy that throttles all bandwidth to a certain speed, then a child 
> policy that prioritizes that bandwidth, so you don't saturate your outbound 
> circuit(think in terms of P2P protections).
>
>   
Thank you.  This is EXACTLY what I was looking for.  Do the packet 
counters for show policy-map int fast 0/0 only increment when the 
queuing kicks in or should they be incrementing all the time as packets 
flow?

Thanks again,
Brent



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Re: [asterisk-users] QoS & VPN

[Brent Davidson]

David Backeberg wrote:

On Thu, May 7, 2009 at 3:54 PM, Brent Davidson
 wrote:
  

I've got multiple satellite office all linked back to the main office
via VPN.  Each office has their own asterisk server which registers back
to the main office's Asterisk server.  Each office also has a 1Mb
downstream / 384k - 768k upstream connection.  The branches are using
Speex for their connections back to the main office.  The issue I'm
having is that there are times that I need to VNC in to machines at the
various offices for tech support while the user is also on the phone.
Unfortunately the VNC connection apparently takes priority and makes it
impossible for me to understand anything the person on the phone is
saying, although they can still hear me fine.



VNC is very asymmetric. It doesn't generate much traffic from the
person viewing, and it generates lots of traffic FROM the system being
viewed. This helps explain why the system being viewed side can hear
incoming voice packets, and outbound voice packets that have to
compete with the large amount of outgoing video signal data lose. QoS
may or may not help you here.

  
Well, the fact that our central office has a 10mb downstream / 5mb 
upstream connection (Two 5Mb down 2.5Mb up DSl connections load shared) 
helps with them hearing me clearly too, I'm sure.  I can get the packets 
to them faster than they can get packets to me.

If voice quality is important, you should have a separate connection
dedicated to just voice. The obvious workaround is grab your cell
phone and call them with that. You DO have a way to dial directly to
that office without going over the PIX, right, right? How do you call
the remote office when the PIX goes down?

What will help you is getting a bigger line or separating the voice
traffic from the data traffic completely.

If you are good with ssh, you can also do a compressed ssh tunnel to
encrypt and on-the-fly compress the VNC session. But if this is
Windows good luck with that.
  
Yes, we can dial all satellite office through the PSTN if we really want 
to, but one of the reasons we went to a VOIP system was to cut down on 
the long-distance charges that result from office-to-office calls, and 
to be able to transfer calls from one office to another.  All in all the 
system works as designed, except for the rare occasions that I'm doing 
support with VNC and have a person on the remote extension as well.  But 
just because nobody else has complained yet doesn't mean there aren't 
other conditions that could trigger a poor-quality call.  If I can find 
a solution that works in my worst-case VNC situation then maybe I'll 
prevent a few future issues from ever becoming real problems.


Separating the voice off to it's own connection would defeat the 
cost-cutting reasoning behind the system.



Thanks,
Brent
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Re: [asterisk-users] QoS & VPN

[Frank Bulk - iName.com]

It's been a few years ago, but Network Computing had tests results showing
that VoIP over a VPN was measurably better than outside a VPN.  Why?
Because the latency was low enough that lost UDP packets (within the VPN
tunnel) could be re-transmitted before the jitter buffer had expired.  Since
most jitter buffers are on the order for 10 to 80 msec, if your one-way
latency is any greater than a third of your jitter buffer, it's of no use.
For example, if the one-way latency is 15 msec, the best-case scenario is
that with single-time packet loss, the other packet would arrive at the
destination in ~45 msec.

Frank

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Garth van
Sittert
Sent: Friday, May 08, 2009 10:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] QoS & VPN

I would think that VoIP over VPN is a bad idea as UDP packets need to be 
in realtime not corrected by the TCP of the VPN.

Garth van Sittert
Technical Director
BitCo
08600 24826
www.bitco.co.za



Aurimas Skirgaila wrote:
> Despite the VPN overhead, running VOIP through VPN is good idea 
> because VPN reorders encapsulated UDP packets in correct order. 
> Security matters as well.
>
> I'd suggest to route VNC packets rather over internet than VPN (so do 
> I), as VPN usually has the highest priority.
>
> On Thu, May 7, 2009 at 11:33 PM, Roberto Piola 
> mailto:roberto.pi...@visiant.it>> wrote:
>
> I do not have examples, but if you are using the 1700 series
> router in order to originate the ipsec vpn, you may use command 
> qos pre-classify (please search for it on cco.cisco.com
> )
>
>
> On Thu, May 7, 2009 at 9:54 PM, Brent Davidson
> mailto:br...@texascountrytitle.com>>
> wrote:
>
> I've got multiple satellite office all linked back to the main
> office
> via VPN.  Each office has their own asterisk server which
> registers back
> to the main office's Asterisk server.  Each office also has a 1Mb
> downstream / 384k - 768k upstream connection.  The branches
> are using
> Speex for their connections back to the main office.  The
> issue I'm
> having is that there are times that I need to VNC in to
> machines at the
> various offices for tech support while the user is also on the
> phone.
> Unfortunately the VNC connection apparently takes priority and
> makes it
> impossible for me to understand anything the person on the
> phone is
> saying, although they can still hear me fine.
>
> Our Main office uses a Cisco PIX 506 for the main firewall and VPN
> concentrator.  Each branch office used a Cisco 1700 series
> router with
> IPSec enabled in the IOS.  Is there any sort of QoS I can turn
> on on the
> main router or the branch routers to make sure the voice
> quality takes
> precedence over the VNC?  (Any example configs would be
> greatly appreciated)
>
> Would I be better off routing the voice packets over the
> internet rather
> than the VPN, and could I safely do that without exposing the
> asterisk
> boxes to unnecessary security risks?  (At present all of our
> asterisk
> boxes are behind the firewalls and only talk to each other
> over the
> VPN.  All PSTN connection is done through TDM boards so they
> have no
> direct exposure to the internet.)
>
>
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>
>
>
> -- 
> Mvh,
> Aurimas Skirgaila
> 
>
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Re: [asterisk-users] Storage capacity for call recording

[Steve Totaro]

On Fri, May 8, 2009 at 5:39 PM, Michelle Dupuis  wrote:

>  I want to record calls in wav format.  Can someone tell me how many MB of
> storage per minute each recording requires (assuming SIP / uLaw codec / full
> duplex recording)
>
> Thanks,
> MD
>

1 meg/min is a good rule of thumb.  You will need 2 megs/min until muxed.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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[asterisk-users] Storage capacity for call recording

[Michelle Dupuis]

I want to record calls in wav format.  Can someone tell me how many MB of
storage per minute each recording requires (assuming SIP / uLaw codec / full
duplex recording)
 
Thanks,
MD
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[asterisk-users] Record all calls

[Michelle Dupuis]

I'd like to setup a single extension for which all INBOUND and OUTBOUND
calls are recorded to a wav file.  I took a look at the wiki:
 
http://www.voip-info.org/wiki/view/Asterisk+record+calls
 
but it's not too helpful.  Can someone show some sample code in & out
recording?
 
Thanks,
MD
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Re: [asterisk-users] Configuring SIP Trunk

[Dave Walker]

This looks like a SIP registration problem to me.   You should enable sip debugging.  If the problem/solution does not become obvious then post the results so we can take a look.localhost*CLI> sip show peersName/username  Host    Dyn Nat ACL Port Statusnew/42634          203.196.128.56   5080 OK (112 ms)localhost*CLI> sip set debug ip 203.196.128.56SIP Debugging Enabled for IP: 203.196.128.56localhost*CLI>It may take a several seconds but you should see a bunch of sip traffic come by.   After debugging you can use 'sip set debug off' to turn off sip debugging. I hope this helps.. 


 Original Message 
Subject: [asterisk-users] Configuring SIP Trunk
From: Sathyan M 
Date: Fri, May 08, 2009 6:01 am
To: asterisk-users@lists.digium.com

Hi All,I have searched the various post and not able to find the solution.I am using Asterisk 1.4.21.2 for outgoing calls. Earlier i used ZAP trunk and it works fine. Now i need to move to SIP trunk and configured the same.When i try from softphone i got error as "Call rejected" and in the asterisk i got error as *    -- Executing [90012127773...@from-internal:1] Macro("SIP/2022-083c53f0", "user-callerid|SKIPTTL|") in new stack    -- Executing [...@macro-user-callerid:1] NoOp("SIP/2022-083c53f0", "user-callerid: device 2022") in new stack    -- Executing [...@macro-user-callerid:2] Set("SIP/2022-083c53f0", "AMPUSER=2022") in new stack    -- Executing [...@macro-user-callerid:3] GotoIf("SIP/2022-083c53f0", "0?report") in new stack    -- Executing [...@macro-user-callerid:4] ExecIf("SIP/2022-083c53f0", "1|Set|REALCALLERIDNUM=2022") in new stack    -- Executing [...@macro-user-callerid:5] NoOp("SIP/2022-083c53f0", "REALCALLERIDNUM is 2022") in new stack    -- Executing [...@macro-user-callerid:6] Set("SIP/2022-083c53f0", "AMPUSER=2022") in new stack    -- Executing [...@macro-user-callerid:7] Set("SIP/2022-083c53f0", "AMPUSERCIDNAME=EXTs1") in new stack    -- Executing [...@macro-user-callerid:8] GotoIf("SIP/2022-083c53f0", "0?report") in new stack    -- Executing [...@macro-user-callerid:9] Set("SIP/2022-083c53f0", "AMPUSERCID=2022") in new stack    -- Executing [...@macro-user-callerid:10] Set("SIP/2022-083c53f0", "CALLERID(all)="EXTs1" <2022>") in new stack    -- Executing [...@macro-user-callerid:11] Set("SIP/2022-083c53f0", "REALCALLERIDNUM=2022") in new stack    -- Executing [...@macro-user-callerid:12] ExecIf("SIP/2022-083c53f0", "0|Set|CHANNEL(language)=") in new stack    -- Executing [...@macro-user-callerid:13] NoOp("SIP/2022-083c53f0", "TTL:  ARG1: SKIPTTL") in new stack    -- Executing [...@macro-user-callerid:14] GotoIf("SIP/2022-083c53f0", "1?continue") in new stack    -- Goto (macro-user-callerid,s,23)    -- Executing [...@macro-user-callerid:23] NoOp("SIP/2022-083c53f0", "Using CallerID "EXTs1" <2022>") in new stack    -- Executing [90012127773...@from-internal:2] Set("SIP/2022-083c53f0", "_NODEST=") in new stack    -- Executing [90012127773...@from-internal:3] Macro("SIP/2022-083c53f0", "record-enable|2022|OUT|") in new stack    -- Executing [...@macro-record-enable:1] GotoIf("SIP/2022-083c53f0", "0?2:4") in new stack    -- Goto (macro-record-enable,s,4)    -- Executing [...@macro-record-enable:4] AGI("SIP/2022-083c53f0", "recordingcheck|20090508-171018|1241782818.40") in new stack    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck  recordingcheck|20090508-171018|1241782818.40: Outbound recording not enabled    -- AGI Script recordingcheck completed, returning 0    -- Executing [...@macro-record-enable:5] NoOp("SIP/2022-083c53f0", "No recording needed") in new stack    -- Executing [90012127773...@from-internal:4] Macro("SIP/2022-083c53f0", "dialout-trunk|2|0012127773456||") in new stack    -- Executing [...@macro-dialout-trunk:1] Set("SIP/2022-083c53f0", "DIAL_TRUNK=2") in new stack    -- Executing [...@macro-dialout-trunk:2] ExecIf("SIP/2022-083c53f0", "0|Authenticate|") in new stack    -- Executing [...@macro-dialout-trunk:3] GotoIf("SIP/2022-083c53f0", "0?disabletrunk|1") in new stack    -- Executing [...@macro-dialout-trunk:4] Set("SIP/2022-083c53f0", "DIAL_NUMBER=0012127773456") in new stack    -- Executing [...@macro-dialout-trunk:5] Set("SIP/2022-083c53f0", "DIAL_TRUNK_OPTIONS=tr") in new stack 

Re: [asterisk-users] How to get meetme participants in dialplan?

[Steve Edwards]

On Fri, 8 May 2009, David Backeberg wrote:

> On Thu, May 7, 2009 at 11:44 AM, Steve Edwards 
>  wrote:

>> The meetmeadmin() dialplan function lets you specify a user to mute, 
>> un-mute or kick. But how do you get a list of users in your dialplan?
>
> You need a way to keep state. I use a database and AGI for that purpose.

I thought about keeping state in the db, but then I'd have to run the AGI 
when they leave the conference or hangup as well. These callers jump from 
conference to conference with a single key press and use this feature 
constantly.

>> When a user joins a conference, the user number assigned is "the last 
>> user number +1." If you have a long running conference with callers 
>> joining and leaving all the time, this can grow to be a large number.
>
> Before you add the user to the conference, you can do a: meetme list 
> , parse the first non-space chars of the last line of the 
> output, but as you pointed out, that's messy.

(OT, but FYI,  is not constrained to be a number. This is for a 
"chat house" application. The rooms are named like "en-f1-r9 -- English, 
floor 1, room 9).

This only shows the number of participants, not their user number. The 
user number (ast_conf_user.user_no in app_meetme.c) is the bit displayed 
after "User #:"

Even worse, what if 2 callers join "simultaneously?"

The output of "meetme list" is useless from the standpoint of storing it in a 
DB. Say a 
caller joins and is assigned a user_no of 55. The next caller will get 56. 
If 56 drops off, meetme list shows 55 as the last assigned number but the 
next caller will be assigned 57. And we still have the problem of no easy 
way to get the output of "meetme list" into my dialplan or AGI.

> It's easiest to parse the output of: meetme concise which outputs a 
> nifty csv-style output, but that's not available in 1.2 I don't know 
> which version added meetme concise, but it's definitely in 1.6. You may 
> be able to find the patch and backport it to 1.2?

Sometimes its a drag being a 1.2 Luddite. Is "meetme concise" a CLI 
command? If so, I'm still stuck parsing the output of an AMI command. 
Backporting is even less desirable.

> The closest thing in 1.2 is: meetme and awk for the parties column, 
> after grepping for the conf num. and when you have the user join 
> conference the number of that confererence member will be the 
> incremented current parties number for that room. Then you need to hold 
> on to it in a db.

Yuk. Doing a system() is even uglier :)

And I'd still have the user number problem cited earlier.

>> I want to be able to "mute/unmute/kick caller x" where x ranges from 1 
>> to the number of participants.
>
> This will be a db (maybe through AGI) database lookup, based on what you 
> stored when you joined them to the room. Use that when you want to mute 
> / unmute / kick.

Parsing the AMI output of "meetme list foo" in an AGI works, I was just 
hoping for something more elegant.

Thanks in advance,

Steve Edwards  sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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Re: [asterisk-users] Not receiving voicemail message in mailbox

[Dave Walker]

http://www.voip-info.org/wiki/view/Asterisk+config+voicemail.confmailcmd
Mailcmd allows the administrator to override the default mailer
command with a defined command. Mailcmd takes a string value set to the
desired command line to execute when a user needs to be notified of a
voice mail message. The default command line is: '/usr/sbin/sendmail
-t'. A useful alternative to sendmail is Exim which many people find easier to configure. Examples

 ; - you will need to escape any < or > characters with a \
 mailcmd=/usr/sbin/sendmail -v -t -f asterisk-...@yourdomain.com   ; use -f to prevent r...@localhost.localdomain or similar
 mailcmd=/usr/exim/bin/exim -t
 ; - use the next line for testing
 mailcmd=cat \> /tmp/astvm-mail

 


 Original Message 
Subject: [asterisk-users] Not receiving voicemail message in mailbox
From: jonas kellens 
Date: Fri, May 08, 2009 1:32 am
To: Asterisk Mailing 

   It should be as simple as editing voicemail.conf :  ; Voicemail Configuration ; [general] ; Formats for writing Voicemail.  Note that when using IMAP storage for ; voicemail, only the first format specified will be used. format=wav49|wav|gsm ; Who the e-mail notification should appear to come from serveremail=asterisk-voicemail ; Should the email contain the voicemail as an attachment attach=yes ; Maximum length of a voicemail message in seconds maxmessage=180 ; Minimum length of a voicemail message in seconds for the message to be kept ; The default is no minimum. minmessage=2 ; How many milliseconds to skip forward/back when rew/ff in message playback ; *  during msg playback: Rewind ; #  during msg playback: Fastforward  skipms=3000 ; How many seconds of silence before we end the recording maxsilence=10 ; Silence threshold (what we consider silence: the lower, the more sensitive) silencethreshold=128 ; Max number of failed login attempts maxlogins=3 ; Opmaak e-mail die moet verstuurd worden : emailsubject=Nieuw voicemailbericht in mailbox ${VM_MAILBOX} ! ;emailbody=Beste,\n\n\t Er is een nieuw voicemailbericht nagelaten in mailbox ${VM_MAILBOX} op ${VM_DATE}.\n\n\t Dit bericht is ${VM_DUR} lang en komt van $ {VM_CALLERID} (${VM_CIDNUM}). \n\n\t Je kan het bericht beluisteren door te bellen naar '99' of via het bijgevoegde WAV-bestand (${VM_MESSAGEFILE}).\n\n\t B edankt!\n\n\t\t\t\t--Asterisk--\n ; Set the date format on outgoing mails. ; 24h date format emaildateformat=%A, %d %B %Y at %H:%M:%S ; Attach messages to e-mail notifications? attach = yes ; Context to call back from   callback=from-voicemail  [zonemessages] belgie=Europe/Brussels|'vm-received' Q 'digits/at' R  [voicemail-context]  ; Syntax for new entries looks like this: ; MailboxNumber => password,name,e-mail,pager,options 50 => 4569,Jonas Kellens,jonas.kell...@thecomputerstore.be,,tz=belgie|attach=yes  But I do not receive an e-mail after having left a voicemail message on the voicemailbox 50.  What mail-server does Asterisk uses to send his mail ??? Sendmail is not active on my CentOS-box. I have msmtp + mutt to send me weekly the logfiles. Could it be that ??  Greetingz, Jonas.   ___
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[asterisk-users] Leg-based CDR proposal updated; Major mods

[Steve Murphy]

Hello!

It's me again. I began a fairly large modification to my CDR proposal
some weeks ago, and finally yesterday and this morning got enough
accomplished to allow a commit and some peer review.

Check the docs out via " svn co
http://svn.digium.com/svn/asterisk/team/murf/RFCs "

This is a directory; in it you will find:
CDRfix2.rfc.doc
CDRfix2.rfc.docx
CDRfix2.rfc.pdf

The docx version is the one I actually edit; the doc verison
should be editable in both the windows and linux worlds.
The PDF version is for those who just want to read it.

Basically, I modified the doc to turn Leg-based CDRs into Simple
CDR's with splits (both automatic and dial-plan generated). Instead
of a single time line for complex conversation, broken into consecutive
segments, now you have N time lines (one per participating channel).
A Leg-based CDR system devolves into a Simple CDR system when
all the automatic and dial-plan splitting is turned off. This makes things
easier to implement and understand. At least, for me! (If I don't understand
it, I doubt I can make YOU understand it!)

Also, I gave uniqueID's an overhaul based on the idea of REFERENCING.
I think most users have misunderstood the uniqueID field in the current
CDR system, and don't see the possibilities that solid referencing would
open up. I introduce the idea of uniquely identifying channel 'instances',
which are what Asterisk creates when it creates a channel struct internally.
The
uniqueID field on the CHANNEL struct uniquely identifies that channel
instance
across time for a single server. (Currently).  (barring masquerades). These
are what are
REFERENCED by my destchannel, and channel fields. I provide new fields
called "channel_uniqueid" and "dstchannel_uniqueid" to reference
these fields; not just the channel name (dahdi/1-1 doesn't quite suffice
across time.) (But I left the old fields alone for those who prefer device
info).
CDR unique ID's can/will be generated for external usage,
but are not available to the dialplan or apps within Asterisk. Thus, only
external referencing is possible to individual CDR records. But this should
be good
enough, I think!

This may or may not help with issues brought forward a while back
by greyman and others, -- I'd appreciate hearing about it.

murf

-- 
Steve Murphy, Pres, Consultant
ParseTree Corp
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Re: [asterisk-users] QoS & VPN

[Casey Boone]

Dave Platt wrote:
> OpenVPN doesn't suffer from this problem.  Although it's SSL-based
> (and one might think it does everything through SSL-over-TCP),
> it actually sends the VPN traffic via UDP... it uses TCP only
> for the negotiation and administrative aspects of setting up
> the VPN connection.
> 


UDP is the default, but OpenVPN can be configured for TCP as well


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Re: [asterisk-users] G279 install in 1.6.0.9 ? [SOLVED]

[Olivier]

2009/5/8 Olivier 

> Hello,
>
> Here (http://downloads.digium.com/pub/telephony/codec_g729/README) are
> instructions to install G729 software.
> (I think I followed instructions step by step but g729 license doesn't seem
> to show up).
>
> My question is :
> Is the command bellow still up to date ?
>
> >g729 show


I suddenly remembered that I had once (a long time ago) added a noload =>
codec-g729a.so directive in /etc/asterisk/modules.conf to remove a warning
or error message related to transcoding boards.
Removing helped to have this "g729 show lisense" appear.

Sorry to disturb with this.



>
>
>
> Regards
>
>
>
> PS:
> Here are latest steps:
>
>  # ./benchg729-1.0.6
> Recommended flavor for this system is 'opteron' with an average of 517
> milliseconds.
>
> # wget
> http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.6.0/x86-32/codec_g729a-1.6.0_3.0.3-opteron.tar.gz
> # tar xvf codec_g729a-1.6.0_3.0.3-opteron.tar.gz
> # cp codec_g729a-1.6.0_3.0.3-opteron/codec_g729a.so
> /usr/lib/asterisk/modules/
> # asterisk -rx "restart now"
> # asterisk -rvvv
> Asterisk 1.6.0.9, Copyright (C) 1999 - 2008 Digium, Inc. and others.
> Created by Mark Spencer 
> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for
> details.
> This is free software, with components licensed under the GNU General
> Public
> License version 2 and other licenses; you are welcome to redistribute it
> under
> certain conditions. Type 'core show license' for details.
> =
>   == Parsing '/etc/asterisk/asterisk.conf':   == Found
> Running as group 'asterisk'
>   == Parsing '/etc/asterisk/extconfig.conf': Parsing
> /etc/asterisk/extconfig.conf
>   == Found
> Connected to Asterisk 1.6.0.9 currently running on panoramix (pid = 12075)
> Verbosity is at least 3
> Core debug is at least 3
> -- Remote UNIX connection
> panoramix*CLI> g729 show
> No such command 'g729 show' (type 'help g729 show' for other possible
> commands)
>
>
>
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Re: [asterisk-users] AGI - Ways to create a call

[Steve Edwards]

On Fri, 8 May 2009, David Backeberg wrote:

> On Fri, May 1, 2009 at 11:18 AM, Jimmy Godbout  wrote:
>>> Well, my question is: do you guys have any tip in different ways to 
>>> create a call in Asterisk using AGI + PHP?
>
> You may prefer AMI and the originate command as opposed to making call 
> files. As with many things in Asterisk there's usually multiple ways to 
> do things. Also, if you like PHP you should try out the PHP AMI API. 
> It's very nice, and provides a PHP-ified wrapper for AMI calls. 
> http://www.straw-dogs.co.uk/03/15/asterisk-manager-api-php/
>
> You can then make PHP scripts with AMI and originate, feeding an 
> argument in directly from the web gui. (that's what I do).

You could create the call file and move it to the spool directory all in 
dialplan, skipping AGI altogether.

AMI is a bit more work than a call file -- you need to configure 
manager.conf and then in your AGI you need to log in and read responses.

Thanks in advance,

Steve Edwards  sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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Re: [asterisk-users] QoS & VPN

[Dave Platt]

> I would think that VoIP over VPN is a bad idea as UDP packets need to be 
> in realtime not corrected by the TCP of the VPN.

That depends very much on the VPN in use.

OpenVPN doesn't suffer from this problem.  Although it's SSL-based
(and one might think it does everything through SSL-over-TCP),
it actually sends the VPN traffic via UDP... it uses TCP only
for the negotiation and administrative aspects of setting up
the VPN connection.

As far as I know, OpenVPN makes no attempt at all to re-order
the packets that it encapsulates and transmits.  It simply
accepts the IP packets it is to carry, encrypts them individually,
wraps them in UDP, and retransmits them to its peer.  The peer
receives the UDP, decrypts, and forwards.  No re-ordering.

There may be other VPNs which actually carry all of the
VPN'ed data in a single TCP stream... but I think this is
generally agreed to be a Bad Idea for several reasons.

I run SIP over OpenVPN between my Nokia N810 handheld, and
my Asterisk server at home.  I have not noticed any difference
in call quality between SIP-over-OpenVPN, and non-VPN'ed
SIP, between these two endpoints... except, of course, when
the OpenVPN-encapsulated traffic gets through, and non-VPN'ed
traffic doesn't due to firewall or NATing problems at a
particular wireless network access point.





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[asterisk-users] The efficient way to add MeetMe to pure SIP install ?

[Olivier]

Hello,

Page http://www.voip-info.org/wiki/view/Asterisk+config+meetme.conf seems to
include some old content.

Which is the simplest way to add MeetMe to a pure SIP 1.6.1 install on a
recent kernel ?
Is dahdi_dummy still required ?

Regards
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Re: [asterisk-users] func_odbc.c: Unable to execute query

[arturo arturo]

Thank you very much My problem was the res_odbc.conf file was not
configured... as soon as I enable the DSN it worked


2009/5/8 Tilghman Lesher 

> On Thursday 07 May 2009 18:21:55 arturo arturo wrote:
> > I have an asterisk 1.6.2 installation. I'm trying to configure func_odbc
> to
> > read some mysql tables... but every time I tried I got this message:
> >
> > ERROR[24968] func_odbc.c: Unable to execute query [SELECT bloqueada FROM
> > funciones WHERE extension='750']
>
> What is the output of 'odbc show' from the Asterisk CLI?  What is in your
> odbc.ini, res_odbc.conf, and func_odbc.conf?  Remember that the entry in
> odbc.ini is referenced by res_odbc.conf, and the entry in func_odbc.conf
> references the entry in res_odbc.conf, not odbc.ini.
>
> --
> Tilghman
>
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Re: [asterisk-users] AGI - Ways to create a call

[David Backeberg]

>> -Original Message-
>> From: tiagodura...@gmail.com
>> Sent: Fri, 1 May 2009 11:02:58 -0400
>> To: asterisk-users@lists.digium.com
>> Subject: [asterisk-users] AGI - Ways to create a call
On Fri, May 1, 2009 at 11:18 AM, Jimmy Godbout  wrote:
>> Well, my question is: do you guys have any tip in different ways to
>> create a call in Asterisk using AGI + PHP?

You may prefer AMI and the originate command as opposed to making call
files. As with many things in Asterisk there's usually multiple ways
to do things. Also, if you like PHP you should try out the PHP AMI
API.
It's very nice, and provides a PHP-ified wrapper for AMI calls.
http://www.straw-dogs.co.uk/03/15/asterisk-manager-api-php/

You can then make PHP scripts with AMI and originate, feeding an
argument in directly from the web gui. (that's what I do).

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Re: [asterisk-users] 1.6.1 app_fax: WARNING T.30 ECM carrier not found ??

[David Backeberg]

On Mon, May 4, 2009 at 10:52 PM, sean darcy  wrote:
> Receiving a fax with 1.6.1:
>
>   == Spawn extension (incoming-pstn-line, fax, 1) exited non-zero on
> 'DAHDI/4-1'
>     -- Executing [...@incoming-pstn-line:1] NoOp("DAHDI/4-1", "Fax
> Detected") in new stack
>     -- Executing [...@incoming-pstn-line:2] Goto("DAHDI/4-1",
> "incoming-fax,s,1") in new stack
>     -- Goto (incoming-fax,s,1)
>     -- Executing [...@incoming-fax:1] Set("DAHDI/4-1",
> "FAXFILE=/var/spool/asterisk/fax/20090504_1602-0") in new stack
>     -- Executing [...@incoming-fax:2] ReceiveFAX("DAHDI/4-1",
> "/var/spool/asterisk/fax/20090504_1602-0.tif") in new stack
>     -- Starting simple switch on 'DAHDI/1-1'
>     -- Remote UNIX connection
>     -- Hungup 'DAHDI/1-1'
> [2009-05-04 16:02:39] WARNING[12989]: app_fax.c:128 span_message:
> WARNING T.30 ECM carrier not found
> [2009-05-04 16:02:40] WARNING[12989]: app_fax.c:128 span_message:
> WARNING T.30 ECM carrier not found
> [2009-05-04 16:02:40] WARNING[12989]: app_fax.c:128 span_message:
> WARNING T.30 ECM carrier not found
> [2009-05-04 16:02:46] WARNING[12989]: app_fax.c:128 span_message:
> WARNING T.30 ECM carrier not found
> [2009-05-04 16:02:46] WARNING[12989]: app_fax.c:128 span_message:
> WARNING T.30 ECM carrier not found
>     -- Executing [...@incoming-fax:3] Hangup("DAHDI/4-1", "") in new stack
>
> ECM - error correction mode ( right?) - but the fax is received OK. Any
> reason to worry? Anything to do?

Since you are receiving a fax over an analog line, you can set up
MixMonitor() or Monitor() to record the fax, play it back, listen for
line noise or static or something else that may be happening that's
throwing those warnings from SpanDSP.
Obviously if you're getting a fax you're not as worried as if this
caused the fax to drop. Once you have a recording of the transmission
and the logs of what happens you can submit it to the SpanDSP project
to see what they think.

Incidentally, how verbose do you have to set your CLI to get output
that lists the warnings?

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Re: [asterisk-users] precision of wait dialplan application

[David Backeberg]

On Wed, May 6, 2009 at 8:17 AM, Johann Steinwendtner
 wrote:
> But it seems the Wait(60) lasts longer than 60 seconds:
>
>     -- Executing [...@from_meridian:1] NoOp("DAHDI/29-1", "Test 
> wait") in new stack
>     -- Executing [...@from_meridian:2] Answer("DAHDI/29-1", "") in 
> new stack
>     -- Executing [...@from_meridian:3] NoOp("DAHDI/29-1", "Current 
> timestamp: 20090506135813") in new stack
>     -- Executing [...@from_meridian:4] Wait("DAHDI/29-1", "60") in 
> new stack
>     -- Executing [...@from_meridian:5] NoOp("DAHDI/29-1", "Current 
> timestamp: 20090506135915") in new stack
>     -- Executing [...@from_meridian:6] Hangup("DAHDI/29-1", "") in 
> new stack
>
> What is wrong in this example ?

Could be rounding error. Try it ten times and see if it's always that
wrong. If it's always that wrong, just do a wait(59) :-p

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[asterisk-users] G279 install in 1.6.0.9 ?

[Olivier]

Hello,

Here (http://downloads.digium.com/pub/telephony/codec_g729/README) are
instructions to install G729 software.
(I think I followed instructions step by step but g729 license doesn't seem
to show up).

My question is :
Is the command bellow still up to date ?

>g729 show


Regards



PS:
Here are latest steps:

 # ./benchg729-1.0.6
Recommended flavor for this system is 'opteron' with an average of 517
milliseconds.

# wget
http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.6.0/x86-32/codec_g729a-1.6.0_3.0.3-opteron.tar.gz
# tar xvf codec_g729a-1.6.0_3.0.3-opteron.tar.gz
# cp codec_g729a-1.6.0_3.0.3-opteron/codec_g729a.so
/usr/lib/asterisk/modules/
# asterisk -rx "restart now"
# asterisk -rvvv
Asterisk 1.6.0.9, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer 
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for
details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it
under
certain conditions. Type 'core show license' for details.
=
  == Parsing '/etc/asterisk/asterisk.conf':   == Found
Running as group 'asterisk'
  == Parsing '/etc/asterisk/extconfig.conf': Parsing
/etc/asterisk/extconfig.conf
  == Found
Connected to Asterisk 1.6.0.9 currently running on panoramix (pid = 12075)
Verbosity is at least 3
Core debug is at least 3
-- Remote UNIX connection
panoramix*CLI> g729 show
No such command 'g729 show' (type 'help g729 show' for other possible
commands)
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Re: [asterisk-users] Asterisk 1.6.1.0 can't dial out on Sangoma b600

[Tzafrir Cohen]

On Fri, May 08, 2009 at 12:11:40PM -0400, Mike van der Stoop wrote:
> I have a Sangoma b600de analog card using dahdi 2.1.0.4 and I get the 
> following results (same dialplan, config etc):
> 
> Asterisk 1.6.0.1   => works fine
> Asterisk 1.6.0.9  => can't dial out unless I dial in once or apply patch 
>  >>>==> http://bugs.digium.com/print_bug_page.php?bug_id=14577
> Asterisk 1.6.1.0 => can't dial out, regardless of patch or inbound call 
> first.
> 
> Asterisk incorrectly thinks the line is congested.
> Anyone else experiencing this?

Yes: http://bugs.digium.com/view.php?id=14761

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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[asterisk-users] Asterisk 1.6.1.0 can't dial out on Sangoma b600

[Mike van der Stoop]

I have a Sangoma b600de analog card using dahdi 2.1.0.4 and I get the 
following results (same dialplan, config etc):

Asterisk 1.6.0.1   => works fine
Asterisk 1.6.0.9  => can't dial out unless I dial in once or apply patch 
 >>>==> http://bugs.digium.com/print_bug_page.php?bug_id=14577
Asterisk 1.6.1.0 => can't dial out, regardless of patch or inbound call 
first.

Asterisk incorrectly thinks the line is congested.
Anyone else experiencing this?

Regards and thanks,

Mike v


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Re: [asterisk-users] Override sip.conf settings in extensions.conf? Possible?

[Josh Fuller]

> > Does anyone know if it is possible to override sip.conf settings in
> extensions.conf
> > (for example: session-minse=90) without needing to create an overarching
> peer in sip.conf
> > and selecting it specifically in the dial plan?
> >
> 
> You can do this to some extent starting with Asterisk 1.6.1. With the
> AST_CONFIG
> function, you can change a configuration file from the dialplan. The
> problem is
> that you would also have to reload the configuration file so that the
> change
> would take effect. After the call was completed, you would then have to
> reset
> the value of the option and reload the config file again, since you only
> want
> the option set for one call.
> 
> If this doesn't sound absolutely horrible to you and you want the same
> functionality in Asterisk 1.4, you may be able to get away with simply
> copying
> func_config.c from Asterisk 1.6.1 into Asterisk 1.4's funcs/ directory. I
> haven't tried this myself, so I don't know what tweaks, if any, would be
> required to make the code compile.
> 
> > I'm on the 1.4 stable code base and looking to implement session-timers
> on certain call
> > flows in a modular dial plan.
> >
> 
> (Sorry if I'm not making the correct logical leap here)
> 
> Being able to set the session-timers variables via the dialplan will not
> be
> sufficient in 1.4 in order to enable session timers on certain calls. You
> would
> also have to modify chan_sip.c so that the Asterisk would understand the
> concept
> of session timers and how to properly behave.
> 
> Mark Michelson

Thanks for the fantastic answer, Mark. I'm hesitant to migrate to 1.6 because 
some
of the servers I want to make these changes on are production units.

I think-- based on what you've suggested-- that the best action for me would be
to clone my sip peer definitions in sip.conf and add the specific session-timers
I need for origination in the peer.

Then, I can just call the alternate peer with timers invoked. I've used this 
successfully in the past to create unknown and anonymous calls so I think this 
is 
the easiest course.

I'm not opposed to hacking code but it seems redundant when chan_sip technically
already has the functionality I need.

At any rate, great answer! You did make the correct logical leap, by the way. 
It's
nice to see an answer right from Digium on this question as well. We use your 
cards
in our labs and production and find your hardware very useful. 

Thanks,
Josh Fuller josh.ful...@telus.com 

The views expressed in this e-mail are mine alone and do not necessarily 
reflect the 
views of my employer.


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Re: [asterisk-users] How to get meetme participants in dialplan?

[David Backeberg]

On Thu, May 7, 2009 at 11:44 AM, Steve Edwards
 wrote:
> The meetmeadmin() dialplan function lets you specify a user to mute,
> un-mute or kick. But how do you get a list of users in your dialplan?

You need a way to keep state. I use a database and AGI for that purpose.

> When a user joins a conference, the user number assigned is "the last user
> number +1." If you have a long running conference with callers joining and
> leaving all the time, this can grow to be a large number.

Before you add the user to the conference, you can do a: meetme list ,
parse the first non-space chars of the last line of the output, but as
you pointed out, that's messy.

It's easiest to parse the output of: meetme concise
which outputs a nifty csv-style output, but that's not available in 1.2
I don't know which version added meetme concise, but it's definitely
in 1.6. You may be able to find the patch and backport it to 1.2?

The closest thing in 1.2  is: meetme
and awk for the parties column, after grepping for the conf num.
and when you have the user join conference the number of that
confererence member will be the incremented current parties number for
that room. Then you need to hold on to it in a db.

> I want to be able to "mute/unmute/kick caller x" where x ranges from 1 to
> the number of participants.

This will be a db (maybe through AGI) database lookup, based on what
you stored when you joined them to the room. Use that when you want to
mute / unmute / kick.

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[asterisk-users] AMOOCON debriefing

[randulo]

Anyone who was at AMOOCON and who would deign to join us (ahem, Zoa,
alors?) to hash out what happened and make fun of the presenters,
please join us Friday at 6PM Paris time (5 PM UK) or 12 Noon EDT.

I myself was really pleased to be there and meet so many interesting
and amusing people.

Some recorded discussion is also posted here:
http://sessions.voipUsersConference.org

Join us in a few minutes and every Friday  for more about VoIP
asterisk and the price of fish.

/r

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[asterisk-users] Proxying comparison

[Adrian Marsh]

Hi All,

 

Looking to gauge some opinions on redirect/proxy software.

 

I've two existing A*k servers out on the 'net.  I need to redirect the
traffic going to those two servers, over to a new 3rd one.

 

Unfortunately, when the servers and clients were built, they used
hardcoded IPs, rather than DNS, so a simple DNS update wont work.

 

So I'm looking at IP redirect, or SIP proxying as options.

 

In my lab tests here, using iptables and forwarding the IP packets
seemed to work really well.  However on the hosted servers, it seems I'm
unable to do this with iptables, due to one reason or another on the
hosted platforms (Plesk being the main issue).

 

So - for proxying then, which is the most simplest proxy server to
setup?  I've been playing with siproxd today, and wondering about
OpenSER. But I've not used SIP proxys before.  I just need a very simple
redirect.  Anything inbound SIP redirect over to the new server.

 

Thanks,

 

Adrian

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Re: [asterisk-users] QoS & VPN

[Tilghman Lesher]

On Friday 08 May 2009 10:07:43 Garth van Sittert wrote:
> I would think that VoIP over VPN is a bad idea as UDP packets need to be
> in realtime not corrected by the TCP of the VPN.

Not all VPNs use TCP.  OpenVPN, in particular, uses UDP for the backbone.

-- 
Tilghman

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Re: [asterisk-users] Understanding Codecs

[Adrian Marsh]

Ah... ok thanks for that.  In the end it was an SElinux problem. But I
was curious as to if I was missing some config somewhere. This clears
that up.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: 07 May 2009 15:23
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Understanding Codecs

On Thursday 07 May 2009 03:33:14 Adrian Marsh wrote:
> So where are the codec translations set?

I assume you're talking about the numbers within the table?  They're
calculated at runtime, based upon shortest possible path (in terms of
time)
from one codec to another.  Most codecs translate only to signed linear
audio,
so the translation table tends to be rather simple.  Ulaw to alaw is a
simple
table lookup, which is why it tends to be very fast.

-- 
Tilghman

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Re: [asterisk-users] QoS & VPN

[Garth van Sittert]

I would think that VoIP over VPN is a bad idea as UDP packets need to be 
in realtime not corrected by the TCP of the VPN.

Garth van Sittert
Technical Director
BitCo
08600 24826
www.bitco.co.za



Aurimas Skirgaila wrote:
> Despite the VPN overhead, running VOIP through VPN is good idea 
> because VPN reorders encapsulated UDP packets in correct order. 
> Security matters as well.
>
> I'd suggest to route VNC packets rather over internet than VPN (so do 
> I), as VPN usually has the highest priority.
>
> On Thu, May 7, 2009 at 11:33 PM, Roberto Piola 
> mailto:roberto.pi...@visiant.it>> wrote:
>
> I do not have examples, but if you are using the 1700 series
> router in order to originate the ipsec vpn, you may use command 
> qos pre-classify (please search for it on cco.cisco.com
> )
>
>
> On Thu, May 7, 2009 at 9:54 PM, Brent Davidson
> mailto:br...@texascountrytitle.com>>
> wrote:
>
> I've got multiple satellite office all linked back to the main
> office
> via VPN.  Each office has their own asterisk server which
> registers back
> to the main office's Asterisk server.  Each office also has a 1Mb
> downstream / 384k - 768k upstream connection.  The branches
> are using
> Speex for their connections back to the main office.  The
> issue I'm
> having is that there are times that I need to VNC in to
> machines at the
> various offices for tech support while the user is also on the
> phone.
> Unfortunately the VNC connection apparently takes priority and
> makes it
> impossible for me to understand anything the person on the
> phone is
> saying, although they can still hear me fine.
>
> Our Main office uses a Cisco PIX 506 for the main firewall and VPN
> concentrator.  Each branch office used a Cisco 1700 series
> router with
> IPSec enabled in the IOS.  Is there any sort of QoS I can turn
> on on the
> main router or the branch routers to make sure the voice
> quality takes
> precedence over the VNC?  (Any example configs would be
> greatly appreciated)
>
> Would I be better off routing the voice packets over the
> internet rather
> than the VPN, and could I safely do that without exposing the
> asterisk
> boxes to unnecessary security risks?  (At present all of our
> asterisk
> boxes are behind the firewalls and only talk to each other
> over the
> VPN.  All PSTN connection is done through TDM boards so they
> have no
> direct exposure to the internet.)
>
>
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>
>
>
> -- 
> Mvh,
> Aurimas Skirgaila
> 
>
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Re: [asterisk-users] Override sip.conf settings in extensions.conf? Possible?

[Mark Michelson]

Josh Fuller wrote:
> Hi all...
> 
> Does anyone know if it is possible to override sip.conf settings in 
> extensions.conf 
> (for example: session-minse=90) without needing to create an overarching peer 
> in sip.conf 
> and selecting it specifically in the dial plan?
> 

You can do this to some extent starting with Asterisk 1.6.1. With the 
AST_CONFIG 
function, you can change a configuration file from the dialplan. The problem is 
that you would also have to reload the configuration file so that the change 
would take effect. After the call was completed, you would then have to reset 
the value of the option and reload the config file again, since you only want 
the option set for one call.

If this doesn't sound absolutely horrible to you and you want the same 
functionality in Asterisk 1.4, you may be able to get away with simply copying 
func_config.c from Asterisk 1.6.1 into Asterisk 1.4's funcs/ directory. I 
haven't tried this myself, so I don't know what tweaks, if any, would be 
required to make the code compile.

> I'm on the 1.4 stable code base and looking to implement session-timers on 
> certain call 
> flows in a modular dial plan.
> 

(Sorry if I'm not making the correct logical leap here)

Being able to set the session-timers variables via the dialplan will not be 
sufficient in 1.4 in order to enable session timers on certain calls. You would 
also have to modify chan_sip.c so that the Asterisk would understand the 
concept 
of session timers and how to properly behave.

Mark Michelson

> Thanks,
> Josh Fuller josh.ful...@telus.com
> 

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Re: [asterisk-users] QoS & VPN

[David Backeberg]

On Thu, May 7, 2009 at 3:54 PM, Brent Davidson
 wrote:
> I've got multiple satellite office all linked back to the main office
> via VPN.  Each office has their own asterisk server which registers back
> to the main office's Asterisk server.  Each office also has a 1Mb
> downstream / 384k - 768k upstream connection.  The branches are using
> Speex for their connections back to the main office.  The issue I'm
> having is that there are times that I need to VNC in to machines at the
> various offices for tech support while the user is also on the phone.
> Unfortunately the VNC connection apparently takes priority and makes it
> impossible for me to understand anything the person on the phone is
> saying, although they can still hear me fine.

VNC is very asymmetric. It doesn't generate much traffic from the
person viewing, and it generates lots of traffic FROM the system being
viewed. This helps explain why the system being viewed side can hear
incoming voice packets, and outbound voice packets that have to
compete with the large amount of outgoing video signal data lose. QoS
may or may not help you here.

If voice quality is important, you should have a separate connection
dedicated to just voice. The obvious workaround is grab your cell
phone and call them with that. You DO have a way to dial directly to
that office without going over the PIX, right, right? How do you call
the remote office when the PIX goes down?

What will help you is getting a bigger line or separating the voice
traffic from the data traffic completely.

If you are good with ssh, you can also do a compressed ssh tunnel to
encrypt and on-the-fly compress the VNC session. But if this is
Windows good luck with that.

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[asterisk-users] Override sip.conf settings in extensions.conf? Possible?

[Josh Fuller]

Hi all...

Does anyone know if it is possible to override sip.conf settings in 
extensions.conf 
(for example: session-minse=90) without needing to create an overarching peer 
in sip.conf 
and selecting it specifically in the dial plan?

I'm on the 1.4 stable code base and looking to implement session-timers on 
certain call 
flows in a modular dial plan.

Thanks,
Josh Fuller josh.ful...@telus.com

The views expressed in this e-mail are mine alone and do not necessarily 
reflect the 
views of my employer.


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Re: [asterisk-users] QoS & VPN

[Aurimas Skirgaila]

On Fri, May 8, 2009 at 3:45 PM, Jeff LaCoursiere  wrote:

>
> On Fri, 8 May 2009, Aurimas Skirgaila wrote:
>
> > Despite the VPN overhead, running VOIP through VPN is good idea because
> VPN
> > reorders encapsulated UDP packets in correct order. Security matters as
> > well.
>
> Reorders?  How so?  I think it will maintain the order, only if they have
> arrived in the "correct" order.


UDP doesn't guarantee that over long way packets arrive in correct order,
while TCP based VPN would sort them correctly ;) well, I'm not sure if all
kinds of VPN are SSL/TCP based.
The author mentioned remote offices so this might be useful for him.

>
>
> >
> > I'd suggest to route VNC packets rather over internet than VPN (so do I),
> as
> > VPN usually has the highest priority.
> >
>
> Unless QoS is implemented packets are first come first served.  There is
> no "usually has the highest priority".  Routing one over the Internet
> versus over the VPN won't change that priority.
>

ok.  probably I've misread somewhere about switches which QoS enabled is by
default. By the way we do ask our ISP to prioritize VPN packets and they do.

>
> j
>
> > On Thu, May 7, 2009 at 11:33 PM, Roberto Piola  >wrote:
> >
> >> I do not have examples, but if you are using the 1700 series router in
> >> order to originate the ipsec vpn, you may use command  qos pre-classify
> >> (please search for it on cco.cisco.com)
> >>
> >> On Thu, May 7, 2009 at 9:54 PM, Brent Davidson <
> >> br...@texascountrytitle.com> wrote:
> >>
> >>> I've got multiple satellite office all linked back to the main office
> >>> via VPN.  Each office has their own asterisk server which registers
> back
> >>> to the main office's Asterisk server.  Each office also has a 1Mb
> >>> downstream / 384k - 768k upstream connection.  The branches are using
> >>> Speex for their connections back to the main office.  The issue I'm
> >>> having is that there are times that I need to VNC in to machines at the
> >>> various offices for tech support while the user is also on the phone.
> >>> Unfortunately the VNC connection apparently takes priority and makes it
> >>> impossible for me to understand anything the person on the phone is
> >>> saying, although they can still hear me fine.
> >>>
> >>> Our Main office uses a Cisco PIX 506 for the main firewall and VPN
> >>> concentrator.  Each branch office used a Cisco 1700 series router with
> >>> IPSec enabled in the IOS.  Is there any sort of QoS I can turn on on
> the
> >>> main router or the branch routers to make sure the voice quality takes
> >>> precedence over the VNC?  (Any example configs would be greatly
> >>> appreciated)
> >>>
> >>> Would I be better off routing the voice packets over the internet
> rather
> >>> than the VPN, and could I safely do that without exposing the asterisk
> >>> boxes to unnecessary security risks?  (At present all of our asterisk
> >>> boxes are behind the firewalls and only talk to each other over the
> >>> VPN.  All PSTN connection is done through TDM boards so they have no
> >>> direct exposure to the internet.)
> >>>
> >>>
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> >
> >
> >
> > --
> > Mvh,
> > Aurimas Skirgaila
> >
>
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-- 
Mvh,
Aurimas Skirgaila
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Re: [asterisk-users] Configuring SIP Trunk

[John Novack]

First, move to the last 1.4 version
Several things were broken fixed and broken again and IF I remember 
correctly, some were related to SIP

John Novack

Sathyan M wrote:
> Hi All,
>
> I have searched the various post and not able to find the solution.
>
> I am using Asterisk 1.4.21.2 for outgoing calls. Earlier i used ZAP 
> trunk and it works fine. Now i need to move to SIP trunk and 
> configured the same.
>
> When i try from softphone i got error as "Call rejected" and in the 
> asterisk i got error as
>
> *
>
> -- Executing [90012127773...@from-internal:1] 
> Macro("SIP/2022-083c53f0", "user-callerid|SKIPTTL|") in new stack
> -- Executing [...@macro-user-callerid:1] NoOp("SIP/2022-083c53f0", 
> "user-callerid: device 2022") in new stack
> -- Executing [...@macro-user-callerid:2] Set("SIP/2022-083c53f0", 
> "AMPUSER=2022") in new stack
> -- Executing [...@macro-user-callerid:3] GotoIf("SIP/2022-083c53f0", 
> "0?report") in new stack
> -- Executing [...@macro-user-callerid:4] ExecIf("SIP/2022-083c53f0", 
> "1|Set|REALCALLERIDNUM=2022") in new stack
> -- Executing [...@macro-user-callerid:5] NoOp("SIP/2022-083c53f0", 
> "REALCALLERIDNUM is 2022") in new stack
> -- Executing [...@macro-user-callerid:6] Set("SIP/2022-083c53f0", 
> "AMPUSER=2022") in new stack
> -- Executing [...@macro-user-callerid:7] Set("SIP/2022-083c53f0", 
> "AMPUSERCIDNAME=EXTs1") in new stack
> -- Executing [...@macro-user-callerid:8] GotoIf("SIP/2022-083c53f0", 
> "0?report") in new stack
> -- Executing [...@macro-user-callerid:9] Set("SIP/2022-083c53f0", 
> "AMPUSERCID=2022") in new stack
> -- Executing [...@macro-user-callerid:10] Set("SIP/2022-083c53f0", 
> "CALLERID(all)="EXTs1" <2022>") in new stack
> -- Executing [...@macro-user-callerid:11] Set("SIP/2022-083c53f0", 
> "REALCALLERIDNUM=2022") in new stack
> -- Executing [...@macro-user-callerid:12] 
> ExecIf("SIP/2022-083c53f0", "0|Set|CHANNEL(language)=") in new stack
> -- Executing [...@macro-user-callerid:13] NoOp("SIP/2022-083c53f0", 
> "TTL:  ARG1: SKIPTTL") in new stack
> -- Executing [...@macro-user-callerid:14] 
> GotoIf("SIP/2022-083c53f0", "1?continue") in new stack
> -- Goto (macro-user-callerid,s,23)
> -- Executing [...@macro-user-callerid:23] NoOp("SIP/2022-083c53f0", 
> "Using CallerID "EXTs1" <2022>") in new stack
> -- Executing [90012127773...@from-internal:2] 
> Set("SIP/2022-083c53f0", "_NODEST=") in new stack
> -- Executing [90012127773...@from-internal:3] 
> Macro("SIP/2022-083c53f0", "record-enable|2022|OUT|") in new stack
> -- Executing [...@macro-record-enable:1] GotoIf("SIP/2022-083c53f0", 
> "0?2:4") in new stack
> -- Goto (macro-record-enable,s,4)
> -- Executing [...@macro-record-enable:4] AGI("SIP/2022-083c53f0", 
> "recordingcheck|20090508-171018|1241782818.40") in new stack
> -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
>   recordingcheck|20090508-171018|1241782818.40: Outbound recording not 
> enabled
> -- AGI Script recordingcheck completed, returning 0
> -- Executing [...@macro-record-enable:5] NoOp("SIP/2022-083c53f0", 
> "No recording needed") in new stack
> -- Executing [90012127773...@from-internal:4] 
> Macro("SIP/2022-083c53f0", "dialout-trunk|2|0012127773456||") in new stack
> -- Executing [...@macro-dialout-trunk:1] Set("SIP/2022-083c53f0", 
> "DIAL_TRUNK=2") in new stack
> -- Executing [...@macro-dialout-trunk:2] ExecIf("SIP/2022-083c53f0", 
> "0|Authenticate|") in new stack
> -- Executing [...@macro-dialout-trunk:3] GotoIf("SIP/2022-083c53f0", 
> "0?disabletrunk|1") in new stack
> -- Executing [...@macro-dialout-trunk:4] Set("SIP/2022-083c53f0", 
> "DIAL_NUMBER=0012127773456") in new stack
> -- Executing [...@macro-dialout-trunk:5] Set("SIP/2022-083c53f0", 
> "DIAL_TRUNK_OPTIONS=tr") in new stack
> -- Executing [...@macro-dialout-trunk:6] Set("SIP/2022-083c53f0", 
> "GROUP()=OUT_2") in new stack
> -- Executing [...@macro-dialout-trunk:7] GotoIf("SIP/2022-083c53f0&q

[asterisk-users] DNID Truncated

[Jim Boykin]

We have setup a system with TE110P in E1 mode. Everything works fine
except the DNID (number that was dialed.
) which is truncated to 7 digits. Any idea.

Thanks
Jim

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Re: [asterisk-users] QoS & VPN

[Jeremy Mann]

Access-list 100 permit ip host  any

Class-map match-any voip
 Match access-group 100

Policy-map voip
 Class voip
  Priority 256
 Class class-default
  Fair-queue

Interface fastethernet 0
 Service-policy output voip


Above is what I do to prioritize 256kbit of outbound bandwidth to voip calls, 
adjust accordingly.  You must also use the qos pre-classify in your ipsec 
tunnel definitions for this to work, but it does work well.  I know I'm 
potentially mapping other traffic than voip, but I'm lazy and don't want to 
classify the rtp and sip and iax ports, rarely does the box do any other 
traffic than voip as updates occur in off hours.

You'll probably additionally want to match your ipsec keying traffic and give 
it priority bandwidth, if you're going to push voip through the tunnel you'll 
find yourself rekeying more often and want to make sure on a saturated link it 
gets priority so the tunnels don't drop.

If you're on DSL, you probably want to research cascading the Qos, have a root 
policy that throttles all bandwidth to a certain speed, then a child policy 
that prioritizes that bandwidth, so you don't saturate your outbound 
circuit(think in terms of P2P protections).



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Re: [asterisk-users] "pri show spans" shows nothing

[Jim Boykin]

Thanks. Solved.

On Fri, May 8, 2009 at 5:47 PM, Tzafrir Cohen  wrote:
> On Fri, May 08, 2009 at 05:25:08PM +0530, Jim Boykin wrote:
>> Hi,
>>
>> I am using TE110P card, I configured dahdi (dahdi_scan results below).
>> However, when I issue "pro show spans", it does not give any outputs.
>> Am I missing some configuration?
>
> /etc/asterisk/chan_dahdi.conf
>
> --
>               Tzafrir Cohen
> icq#16849755              jabber:tzafrir.co...@xorcom.com
> +972-50-7952406           mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
>
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[asterisk-users] Numeric Hangup Code

[Venefax]

I am sending SIP or H323 calls to a carrier, and I need to store in the CDR
why the calls are rejected or why they hang up. In SIP, it can be code 503,
500, 488, etc. How do I get the information in my dialplan? I don't mean
$(DIALSTATUS}, but the real numeric code 
F.Alves


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[asterisk-users] Configuring SIP Trunk

[Sathyan M]

Hi All,

I have searched the various post and not able to find the solution.

I am using Asterisk 1.4.21.2 for outgoing calls. Earlier i used ZAP trunk and 
it works fine. Now i need to move to SIP trunk and configured the same.

When i try from softphone i got error as "Call rejected" and in the asterisk i 
got error as 

*

-- Executing [90012127773...@from-internal:1] Macro("SIP/2022-083c53f0", 
"user-callerid|SKIPTTL|") in new stack
-- Executing [...@macro-user-callerid:1] NoOp("SIP/2022-083c53f0", 
"user-callerid: device 2022") in new stack
-- Executing [...@macro-user-callerid:2] Set("SIP/2022-083c53f0", 
"AMPUSER=2022") in new stack
-- Executing [...@macro-user-callerid:3] GotoIf("SIP/2022-083c53f0", 
"0?report") in new stack
-- Executing [...@macro-user-callerid:4] ExecIf("SIP/2022-083c53f0", 
"1|Set|REALCALLERIDNUM=2022") in new stack
-- Executing [...@macro-user-callerid:5] NoOp("SIP/2022-083c53f0", 
"REALCALLERIDNUM is 2022") in new stack
-- Executing [...@macro-user-callerid:6] Set("SIP/2022-083c53f0", 
"AMPUSER=2022") in new stack
-- Executing [...@macro-user-callerid:7] Set("SIP/2022-083c53f0", 
"AMPUSERCIDNAME=EXTs1") in new stack
-- Executing [...@macro-user-callerid:8] GotoIf("SIP/2022-083c53f0", 
"0?report") in new stack
-- Executing [...@macro-user-callerid:9] Set("SIP/2022-083c53f0", 
"AMPUSERCID=2022") in new stack
-- Executing [...@macro-user-callerid:10] Set("SIP/2022-083c53f0", 
"CALLERID(all)="EXTs1" <2022>") in new stack
-- Executing [...@macro-user-callerid:11] Set("SIP/2022-083c53f0", 
"REALCALLERIDNUM=2022") in new stack
-- Executing [...@macro-user-callerid:12] ExecIf("SIP/2022-083c53f0", 
"0|Set|CHANNEL(language)=") in new stack
-- Executing [...@macro-user-callerid:13] NoOp("SIP/2022-083c53f0", "TTL:  
ARG1: SKIPTTL") in new stack
-- Executing [...@macro-user-callerid:14] GotoIf("SIP/2022-083c53f0", 
"1?continue") in new stack
-- Goto (macro-user-callerid,s,23)
-- Executing [...@macro-user-callerid:23] NoOp("SIP/2022-083c53f0", "Using 
CallerID "EXTs1" <2022>") in new stack
-- Executing [90012127773...@from-internal:2] Set("SIP/2022-083c53f0", 
"_NODEST=") in new stack
-- Executing [90012127773...@from-internal:3] Macro("SIP/2022-083c53f0", 
"record-enable|2022|OUT|") in new stack
-- Executing [...@macro-record-enable:1] GotoIf("SIP/2022-083c53f0", 
"0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [...@macro-record-enable:4] AGI("SIP/2022-083c53f0", 
"recordingcheck|20090508-171018|1241782818.40") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
  recordingcheck|20090508-171018|1241782818.40: Outbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing [...@macro-record-enable:5] NoOp("SIP/2022-083c53f0", "No 
recording needed") in new stack
-- Executing [90012127773...@from-internal:4] Macro("SIP/2022-083c53f0", 
"dialout-trunk|2|0012127773456||") in new stack
-- Executing [...@macro-dialout-trunk:1] Set("SIP/2022-083c53f0", 
"DIAL_TRUNK=2") in new stack
-- Executing [...@macro-dialout-trunk:2] ExecIf("SIP/2022-083c53f0", 
"0|Authenticate|") in new stack
-- Executing [...@macro-dialout-trunk:3] GotoIf("SIP/2022-083c53f0", 
"0?disabletrunk|1") in new stack
-- Executing [...@macro-dialout-trunk:4] Set("SIP/2022-083c53f0", 
"DIAL_NUMBER=0012127773456") in new stack
-- Executing [...@macro-dialout-trunk:5] Set("SIP/2022-083c53f0", 
"DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing [...@macro-dialout-trunk:6] Set("SIP/2022-083c53f0", 
"GROUP()=OUT_2") in new stack
-- Executing [...@macro-dialout-trunk:7] GotoIf("SIP/2022-083c53f0", 
"1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing [...@macro-dialout-trunk:9] GotoIf("SIP/2022-083c53f0", 
"0?skipoutcid") in new stack
-- Executing [...@macro-dialout-trunk:10] Set("SIP/2022-083c53f0", 
"DIAL_TRUNK_OPTIONS=") in new stack
-- Executing [...@macro-dialout-trunk:11] Macro("SIP/2022-083c53f0", 
"outbound-callerid|2") in new stack
-- Executing [...@macro-outbound-callerid:1] ExecIf("SIP/2022-083c53f0", 
&qu

Re: [asterisk-users] QoS & VPN

[Jeff LaCoursiere]

On Fri, 8 May 2009, Aurimas Skirgaila wrote:

> Despite the VPN overhead, running VOIP through VPN is good idea because VPN
> reorders encapsulated UDP packets in correct order. Security matters as
> well.

Reorders?  How so?  I think it will maintain the order, only if they have 
arrived in the "correct" order.

>
> I'd suggest to route VNC packets rather over internet than VPN (so do I), as
> VPN usually has the highest priority.
>

Unless QoS is implemented packets are first come first served.  There is 
no "usually has the highest priority".  Routing one over the Internet 
versus over the VPN won't change that priority.

j

> On Thu, May 7, 2009 at 11:33 PM, Roberto Piola 
> wrote:
>
>> I do not have examples, but if you are using the 1700 series router in
>> order to originate the ipsec vpn, you may use command  qos pre-classify
>> (please search for it on cco.cisco.com)
>>
>> On Thu, May 7, 2009 at 9:54 PM, Brent Davidson <
>> br...@texascountrytitle.com> wrote:
>>
>>> I've got multiple satellite office all linked back to the main office
>>> via VPN.  Each office has their own asterisk server which registers back
>>> to the main office's Asterisk server.  Each office also has a 1Mb
>>> downstream / 384k - 768k upstream connection.  The branches are using
>>> Speex for their connections back to the main office.  The issue I'm
>>> having is that there are times that I need to VNC in to machines at the
>>> various offices for tech support while the user is also on the phone.
>>> Unfortunately the VNC connection apparently takes priority and makes it
>>> impossible for me to understand anything the person on the phone is
>>> saying, although they can still hear me fine.
>>>
>>> Our Main office uses a Cisco PIX 506 for the main firewall and VPN
>>> concentrator.  Each branch office used a Cisco 1700 series router with
>>> IPSec enabled in the IOS.  Is there any sort of QoS I can turn on on the
>>> main router or the branch routers to make sure the voice quality takes
>>> precedence over the VNC?  (Any example configs would be greatly
>>> appreciated)
>>>
>>> Would I be better off routing the voice packets over the internet rather
>>> than the VPN, and could I safely do that without exposing the asterisk
>>> boxes to unnecessary security risks?  (At present all of our asterisk
>>> boxes are behind the firewalls and only talk to each other over the
>>> VPN.  All PSTN connection is done through TDM boards so they have no
>>> direct exposure to the internet.)
>>>
>>>
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>
>
>
> -- 
> Mvh,
> Aurimas Skirgaila
>

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Re: [asterisk-users] QoS & VPN

[Aurimas Skirgaila]

Despite the VPN overhead, running VOIP through VPN is good idea because VPN
reorders encapsulated UDP packets in correct order. Security matters as
well.

I'd suggest to route VNC packets rather over internet than VPN (so do I), as
VPN usually has the highest priority.

On Thu, May 7, 2009 at 11:33 PM, Roberto Piola wrote:

> I do not have examples, but if you are using the 1700 series router in
> order to originate the ipsec vpn, you may use command  qos pre-classify
> (please search for it on cco.cisco.com)
>
> On Thu, May 7, 2009 at 9:54 PM, Brent Davidson <
> br...@texascountrytitle.com> wrote:
>
>> I've got multiple satellite office all linked back to the main office
>> via VPN.  Each office has their own asterisk server which registers back
>> to the main office's Asterisk server.  Each office also has a 1Mb
>> downstream / 384k - 768k upstream connection.  The branches are using
>> Speex for their connections back to the main office.  The issue I'm
>> having is that there are times that I need to VNC in to machines at the
>> various offices for tech support while the user is also on the phone.
>> Unfortunately the VNC connection apparently takes priority and makes it
>> impossible for me to understand anything the person on the phone is
>> saying, although they can still hear me fine.
>>
>> Our Main office uses a Cisco PIX 506 for the main firewall and VPN
>> concentrator.  Each branch office used a Cisco 1700 series router with
>> IPSec enabled in the IOS.  Is there any sort of QoS I can turn on on the
>> main router or the branch routers to make sure the voice quality takes
>> precedence over the VNC?  (Any example configs would be greatly
>> appreciated)
>>
>> Would I be better off routing the voice packets over the internet rather
>> than the VPN, and could I safely do that without exposing the asterisk
>> boxes to unnecessary security risks?  (At present all of our asterisk
>> boxes are behind the firewalls and only talk to each other over the
>> VPN.  All PSTN connection is done through TDM boards so they have no
>> direct exposure to the internet.)
>>
>>
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>



-- 
Mvh,
Aurimas Skirgaila
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[asterisk-users] Difference between Transfer and Dial applications

[Aurimas Skirgaila]

Hi,
I wonder what is the difference between Transfer and Dial applications?

Could somebody give me an example of Transfer usage? (documentation and
voip-info looks poor a bit).

I'm using Asterisk 1.2.5 if it matters.

-- 
Mvh,
Aurimas Skirgaila
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[asterisk-users] Can't GOSUB_RESULT with Dial U() option ...

[Olivier]

Hello,

I'm not understanding how to use GOSUB_RESULT in U() option from Dial app
(I'm using 1.6.1)
My extensions.ael is :

context mylocal {
2 => {
Dial(SIP/7530,,U(mynotify));
HangUp();
};
3 => {
Dial(SIP/7531);
HangUp();
};
};

macro mynotify () {
GOSUB_RESULT=ABORT;
};


I thought that depending on value written into GOSUB_RESULT, call would
either abort or continue as usual.
In my experience, call is going on as if I haven't set anything in
mynotify() routine.
(Ultimately, I would also like to use GOTO result code).

Am I missing something ?

Cheers

PS: As a reminder, here is an extract of Dial online doc :

Online doc says :
U(x[^arg]) - Execute via Gosub the routine 'x' for the *called* channel
before connecting
   to the calling channel. Arguments can be specified to the Gosub
   using '^' as a delimiter. The Gosub routine can set the variable
   GOSUB_RESULT to specify the following actions after the Gosub
returns.
   * ABORTHangup both legs of the call.
   * CONGESTION   Behave as if line congestion was encountered.
   * BUSY Behave as if a busy signal was encountered.
   * CONTINUE Hangup the called party and allow the calling
party
  to continue dialplan execution at the next
priority.
   * GOTO:^^ - Transfer the call to the
  specified priority. Optionally, an extension, or
  extension and priority can be specified.
   You cannot use any additional action post answer options in
conjunction
   with this option. Also, pbx services are not run on the peer
(called) channel,
   so you will not be able to set timeouts via the TIMEOUT()
function in this routine.
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Re: [asterisk-users] "pri show spans" shows nothing

[Tzafrir Cohen]

On Fri, May 08, 2009 at 05:25:08PM +0530, Jim Boykin wrote:
> Hi,
> 
> I am using TE110P card, I configured dahdi (dahdi_scan results below).
> However, when I issue "pro show spans", it does not give any outputs.
> Am I missing some configuration?

/etc/asterisk/chan_dahdi.conf 

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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[asterisk-users] "pri show spans" shows nothing

[Jim Boykin]

Hi,

I am using TE110P card, I configured dahdi (dahdi_scan results below).
However, when I issue "pro show spans", it does not give any outputs.
Am I missing some configuration?


active=yes
alarms=OK
description=Digium Wildcard TE110P T1/E1 Card 0
name=WCT1/0
manufacturer=Digium
devicetype=Digium Wildcard TE110P T1/E1
location=PCI Bus 01 Slot 01
basechan=1
totchans=31
irq=209
type=digital-E1
syncsrc=0
lbo=0 db (CSU)/0-133 feet (DSX-1)
coding_opts=HDB3
framing_opts=CCS,CRC4
coding=HDB3
framing=CCS

Thanks
Jim

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Re: [asterisk-users] Sangoma a104d and channel banks

[Jim Dickenson]

I have DTMF detection enabled. I will check if overlap makes a difference.
Right now I have it set to no.
-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



> From: Steve Davies 
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Date: Fri, 8 May 2009 10:22:09 +0100
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Subject: Re: [asterisk-users] Sangoma a104d and channel banks
> 
> Oh, and have you enabled Sangoma's DTMF detection in their config
> file? That is probably also necessary.
> 
> Cheers,
> Steve
> 
> 2009/5/8 Steve Davies :
>> Hi,
>> 
>> This may be completely wrong, but I have a feeling it may be related.
>> Have you enabled "overlapdialling" in zapata.conf for the channels
>> that are on the channelbank? If not, the 1st digit will be sent in,
>> not match the dialplan, and be hungup.
>> 
>> *7xxx is probably working because that matches a built-in feature
>> (transfer?), so the system KNOWS already that it needs more digits.
>> 
>> Cheers,
>> Steve
>> 
>> 2009/5/7 Jim Dickenson :
>>> I have more info now.
>>> 
>>> I have used AMI to watch events, including DTMF events. The phone comes off
>>> hook and I see a line on the * CLI saying the line is off hook. I then touch
>>> a key and I see the DTMF event in the AMI. At the same time I see a message
>>> on CLI that says hangup the line.
>> 
> 
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[asterisk-users] G.722, 1.4 and IAX trunking ...

[Gordon Henderson]

Been playing with G.722 in Asterisk 1.4.24.1 - using the back-ported 
patches from http://carlton.oriley.net/drupal/node/12

Works just fine as far as I can tell - Grandstream phones anyway - playing 
the G722 sound files, and calls between them.

Transcoding seems fine too - calling non G722 devices, it seems to "just 
work"

However Phone A (g722) calls phone B (gsm). Works fine.

Phone A puts B on hold. B hears hold music.

Same call setup, but phone B puts A on hold.

Now, A hears the hold music, but it every 0.5 seconds there is a "blip" 
and it sounds like it's playing at double speed.

I hear the same when calling someone via an IAX trunk which is transcoding 
to ulaw...

My test box is using dahdi-dummy for timing - the remote IAX box has a TDM 
card in it.

Saw this at one point on the console too:

[May  8 10:56:29] WARNING[14662]: format_pcm.c:153 pcm_seek: offset too 
large 1209280, truncating to 969666

So I'm wondering if the patch is not that clever when applied to 1.4, or 
if there really is something amis?

Or should I just jump to 1.6!

Cheers,

Gordon

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Re: [asterisk-users] Sangoma a104d and channel banks

[Steve Davies]

Oh, and have you enabled Sangoma's DTMF detection in their config
file? That is probably also necessary.

Cheers,
Steve

2009/5/8 Steve Davies :
> Hi,
>
> This may be completely wrong, but I have a feeling it may be related.
> Have you enabled "overlapdialling" in zapata.conf for the channels
> that are on the channelbank? If not, the 1st digit will be sent in,
> not match the dialplan, and be hungup.
>
> *7xxx is probably working because that matches a built-in feature
> (transfer?), so the system KNOWS already that it needs more digits.
>
> Cheers,
> Steve
>
> 2009/5/7 Jim Dickenson :
>> I have more info now.
>>
>> I have used AMI to watch events, including DTMF events. The phone comes off
>> hook and I see a line on the * CLI saying the line is off hook. I then touch
>> a key and I see the DTMF event in the AMI. At the same time I see a message
>> on CLI that says hangup the line.
>

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Re: [asterisk-users] Sangoma a104d and channel banks

[Steve Davies]

Hi,

This may be completely wrong, but I have a feeling it may be related.
Have you enabled "overlapdialling" in zapata.conf for the channels
that are on the channelbank? If not, the 1st digit will be sent in,
not match the dialplan, and be hungup.

*7xxx is probably working because that matches a built-in feature
(transfer?), so the system KNOWS already that it needs more digits.

Cheers,
Steve

2009/5/7 Jim Dickenson :
> I have more info now.
>
> I have used AMI to watch events, including DTMF events. The phone comes off
> hook and I see a line on the * CLI saying the line is off hook. I then touch
> a key and I see the DTMF event in the AMI. At the same time I see a message
> on CLI that says hangup the line.

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[asterisk-users] Not receiving voicemail message in mailbox

[jonas kellens]

It should be as simple as editing voicemail.conf :

; Voicemail Configuration
;
[general]
; Formats for writing Voicemail.  Note that when using IMAP storage for
; voicemail, only the first format specified will be used.
format=wav49|wav|gsm
; Who the e-mail notification should appear to come from
serveremail=asterisk-voicemail
; Should the email contain the voicemail as an attachment
attach=yes
; Maximum length of a voicemail message in seconds
maxmessage=180
; Minimum length of a voicemail message in seconds for the message to be
kept
; The default is no minimum.
minmessage=2
; How many milliseconds to skip forward/back when rew/ff in message
playback
; *  during msg playback: Rewind
; #  during msg playback: Fastforward 
skipms=3000
; How many seconds of silence before we end the recording
maxsilence=10
; Silence threshold (what we consider silence: the lower, the more
sensitive)
silencethreshold=128
; Max number of failed login attempts
maxlogins=3
; Opmaak e-mail die moet verstuurd worden :
emailsubject=Nieuw voicemailbericht in mailbox ${VM_MAILBOX} !
;emailbody=Beste,\n\n\t Er is een nieuw voicemailbericht nagelaten in
mailbox ${VM_MAILBOX} op ${VM_DATE}.\n\n\t Dit bericht is ${VM_DUR} lang
en komt van $
{VM_CALLERID} (${VM_CIDNUM}). \n\n\t Je kan het bericht beluisteren door
te bellen naar '99' of via het bijgevoegde WAV-bestand
(${VM_MESSAGEFILE}).\n\n\t B
edankt!\n\n\t\t\t\t--Asterisk--\n
; Set the date format on outgoing mails.
; 24h date format
emaildateformat=%A, %d %B %Y at %H:%M:%S
; Attach messages to e-mail notifications?
attach = yes
; Context to call back from  
callback=from-voicemail

[zonemessages]
belgie=Europe/Brussels|'vm-received' Q 'digits/at' R

[voicemail-context]

; Syntax for new entries looks like this:
; MailboxNumber => password,name,e-mail,pager,options
50 => 4569,Jonas Kellens,jonas.kell...@thecomputerstore.be,,tz=belgie|
attach=yes

But I do not receive an e-mail after having left a voicemail message on
the voicemailbox 50.

What mail-server does Asterisk uses to send his mail ???
Sendmail is not active on my CentOS-box. I have msmtp + mutt to send me
weekly the logfiles. Could it be that ??

Greetingz,
Jonas.
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