Re: [asterisk-users] outbound SIP trunk hunting (or any fxo for that matter)

[Motiejus Jakštys]

On Mon, Aug 23, 2010 at 5:06 PM, Infra  wrote:
>
> On Aug 7, 2007 'Mojo' wrote:
>
> Nicholas Blasgen wrote:
>> I've got 4 SIP phone lines with a call-limit of 2 for each.  I've
>> written a handy macro to allow my users to dial a phone number and the
>> macro will figure out the next available line to use by first checking
>> if the GROUP() is over 2 and then checking to see if ChanIsAvail() as a
>> backup, and if it can't use the line for either reason it goes to the
>> next line.  The problem is that there are enough situations that the
>> Macro gets called twice without much time seperation.  Both macros check
>> the group() number, it comes back as free, they check the line
>> availability and it's open, and they try dialing.  But because they both
>> started at more or less the same instant, they've both at the same stage
>> in the macro and sometimes (maybe 10% of the time) a macro will try
>> dialing on a line that's already in use.
>>
>> My question is this.  Is it possible to tell Asterisk to execute part of
>> a macro as a block without allowing any other commands to be processed
>> during that time?  Some way to LOCK the dialplan (as you'd do in SQL).
>> I want my macro to be able to execute the part of the code that checks
>> line status and then sets the GROUP() without allowing any other
>> dialplans from running during that time.  Anyone know if this is a
>> current feature?
>>
>> --
>> /Nick
>
> What would be a correct way to do this in 1.4.x?

I suppose AGI is synchronous, you can implement the dialplan locking in AGI.
AGI(lock.pl)
do your GROUP() things
AGI(unlock.pl)
To make the locking atomic you may use posix semaphores. Look for APIs
in Perl, C, Python...

Motiejus

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Re: [asterisk-users] IAX2 - Separate Signaling and Media?

[Tim Nelson]

- "Tim Nelson"  wrote:
> Greetings all-
> 
> Here's an odd question. Supposedly, IAX2 now has the ability to
> operate with signaling and media in separate streams, very much like
> SIP. I've read about this feature here[1] and there[2], but I have yet
> to see how to actually implement or test it. There are no options in
> the iax.conf sample configs with Asterisk.
> 
> All suggestions welcome, except those telling me to jump off a bridge
> because separated signaling and media makes IAX pointless when
> compared to SIP. :-)
> 

Ugh, and let me specify references as originally intended:

[1] http://tools.ietf.org/search/rfc5456
[2] http://www.voip-info.org/wiki/view/IAX+versus+SIP

--Tim

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[asterisk-users] IAX2 - Separate Signaling and Media?

[Tim Nelson]

Greetings all-

Here's an odd question. Supposedly, IAX2 now has the ability to operate with 
signaling and media in separate streams, very much like SIP. I've read about 
this feature here[1] and there[2], but I have yet to see how to actually 
implement or test it. There are no options in the iax.conf sample configs with 
Asterisk.

All suggestions welcome, except those telling me to jump off a bridge because 
separated signaling and media makes IAX pointless when compared to SIP. :-)

--Tim

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Re: [asterisk-users] sip probe syntax

[Lyle Giese]

Matt Kershnar wrote:
> If anyone has any info on this it'd be much appreciated - haven't
> found much about this topic anywhere. We are setting up sip probe
> monitor to make sure that our Asterisk boxes are up and functional (or
> at least responding to the sip protocol) and we need to determine the
> appropriate probe syntax for the probe requests to the Asterisk boxes.
> These boxes are running on various platforms and asterisk versions so
> we'd like to keep this as universal as possible. If we need to be
> platform/version specific any advice would be helpful. Thanks!
>
http://exchange.nagios.org/directory/Plugins/Network-Protocols/*-VoIP/SIP/check_sip-sipsak/details

Lyle Giese
LCR Computer Services, Inc.


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[asterisk-users] sip probe syntax

[Matt Kershnar]

If anyone has any info on this it'd be much appreciated - haven't found much
about this topic anywhere. We are setting up sip probe monitor to make sure
that our Asterisk boxes are up and functional (or at least responding to the
sip protocol) and we need to determine the appropriate probe syntax for the
probe requests to the Asterisk boxes. These boxes are running on various
platforms and asterisk versions so we'd like to keep this as universal as
possible. If we need to be platform/version specific any advice would be
helpful. Thanks!
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Re: [asterisk-users] Asterisk, HylaFax and Cardiff

[Doug Lytle]

Don Kelly wrote:
> Thanks, Doug,
>
> The Cardiff Teleforms application is unattended. The Winprint link that I
> looked at shows that it uses a print dialog. Does it have an automagic mode,
> too?
>

Not that I'm aware of, but it's been a while since I've looked to see if 
there has been updates.

Doug

-- 
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."


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Re: [asterisk-users] Asterisk, HylaFax and Cardiff

[Don Kelly]

Thanks, Doug,

The Cardiff Teleforms application is unattended. The Winprint link that I
looked at shows that it uses a print dialog. Does it have an automagic mode,
too?

--Don



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Monday, August 23, 2010 3:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk, HylaFax and Cardiff

Don Kelly wrote:
> I looked at http://www.hylafax.org/content/Desktop_Client_Software and
> visited several websites before posting this.
>
> Nothing I saw said they would emulate WinFax print-to-fax.
>

Not being familiar with WinFax, we currently use Winprint HylaFAX:  It 
creates a fax printer under Windows that allows printing directly to 
HylaFAX+

http://winprinthylafax.sourceforge.net/

Doug

-- 

Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety."


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Re: [asterisk-users] How to debug this specific issue?

[Steve Davies]

On 23 August 2010 18:32, Stefan Schmidt  wrote:
> Steve Davies schrieb:
>> I need suggestions please on how to determine where it is locking, and why.
>>
>> Many thanks,
>> Steve
>>
>>
> hello,
>
> have you allready tried strace ?
> you could just easily start asterisk with this command:
>
> strace asterisk -

Yes, I tried this. Output just stops along with everything else and
there are no clues.

> or whatever options you want.
> maybe you could see some more information with this.
> you could also try
>
> time asterisk -v

Interesting. I'll try that.

> to see if its user or system time you loose.
> and finaly using a second terminal and try rasterisk -x"core show locks"
> for example will also give you some information.

The whole system is locked. I cannot run anything :(

Cheers,
Steve

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Re: [asterisk-users] Asterisk, HylaFax and Cardiff

[Doug Lytle]

Don Kelly wrote:
> I looked at http://www.hylafax.org/content/Desktop_Client_Software and
> visited several websites before posting this.
>
> Nothing I saw said they would emulate WinFax print-to-fax.
>

Not being familiar with WinFax, we currently use Winprint HylaFAX:  It 
creates a fax printer under Windows that allows printing directly to 
HylaFAX+

http://winprinthylafax.sourceforge.net/

Doug

-- 

Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."


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Re: [asterisk-users] can't build resODBC on SUSE 11.3

[Tilghman Lesher]

On Monday 23 August 2010 14:58:31 Tim Panton wrote:
> On 23 Aug 2010, at 18:07, Warren Selby wrote:
> > On Mon, Aug 23, 2010 at 11:34 AM, Tim Panton  wrote:
> > What is  menuselect actually looking for when it blocks me from selecting
> > res_odbc ?
> >
> > I've got unixOdbc installed and working. I also have
> > /usr/lib64/libltdl.so.3 - so I'm confused as it is claiming these are the
> > pre-requisites ?
> >
> > How can I best track down what it _thinks_ is missing ?
> >
> > (This is on asterisk 1.8 svn trunk - but I don't think that is important,
> > I think it is a package number issue)
> >
> > Thanks in advance,
> >
> > Tim.
> >
> >
> > Tim Panton - Web/VoIP consultant and implementor
> > www.westhawk.co.uk
> >
> >
> > You need to install the -devel packages of libtool-ltdl and unixODBC.
>
> Ah, libtool was what I was missing - thanks!

Actually, the only reason why libtool was set to be a separate dependency was
due to a bug in Fedora Core 7, where UnixODBC required ltdl to link, but did
not require it as an RPM dependency.  Now that that bug has long since been
fixed, that Asterisk dependency could probably be removed.

-- 
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Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Asterisk, HylaFax and Cardiff

[Don Kelly]

I looked at http://www.hylafax.org/content/Desktop_Client_Software and
visited several websites before posting this.

Nothing I saw said they would emulate WinFax print-to-fax.

I'd really appreciate hearing about any direct experiences.



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Monday, August 23, 2010 3:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk, HylaFax and Cardiff


On Mon, 23 Aug 2010, Don Kelly wrote:

> 
> I?m looking for a way to use our implementation of HylaFax on Asterisk
with Cardiff (an
> old installation of Cardiff document stuff).
> 
> Is someone doing that?
> 
> If no one has direct experience, is there a HylaFax client that emulates
WinFax
> print-to-fax?
>

Lots!

http://www.hylafax.org/content/Desktop_Client_Software

j


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Re: [asterisk-users] Asterisk voicemail server - gsm notifications

[Paul Belanger]

On Mon, Aug 23, 2010 at 11:57 AM, Matt  wrote:
> Has anyone successfully implemented Asterisk as a voicemail server for a
> GSM/cellular system and worked out a way to send notifications of new
> messages to the phones?
>
Yes, look at externnotify within voicemail.conf

-- 
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Re: [asterisk-users] DAHDI not detecting caller hangup

[Paul Belanger]

On Mon, Aug 23, 2010 at 11:56 AM, --[ UxBoD ]--  wrote:
> Odd problem have just noticed in that when I call into the PBX DAHDI detects 
> the call and hands it off to the extension, if I then hang up it still 
> continues to process through the dialplan.
>
It is common for telco not to provide a disconnect tone for Analog.
You'll need to confirm one is there, either ask your telco or have
Asterisk record the line.  Then update your indications.conf with your
disconnect tone.

Also be sure you set a TIMEOUT in your dialplans.

-- 
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Re: [asterisk-users] DAHDI not detecting caller hangup

[Paul Belanger]

On Mon, Aug 23, 2010 at 11:56 AM, --[ UxBoD ]--  wrote:
> Odd problem have just noticed in that when I call into the PBX DAHDI detects 
> the call and hands it off to the extension, if I then hang up it still 
> continues to process through the dialplan.
>
It is common for telco not to provide a disconnect tone for Analog.
You'll need to confirm one is there, either ask your telco or have
Asterisk record the line.  Then update your indications.conf with your
disconnect tone.

Also be sure you set a TIMEOUT in your dialplans.

-- 
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Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com

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Re: [asterisk-users] All phones ringing when temporary loss of Internet

[Paul Belanger]

On Mon, Aug 23, 2010 at 1:03 PM, --[ UxBoD ]--  wrote:
> This is a real strange one and trying to phantom it out.  One of our clients 
> is connected to our Asterisk installation, from two sites, via VPN which 
> works great. Every so often one of the sites VPN tunnel goes does for say a 
> couple of seconds. When that happens all the extensions, including both 
> sites, ring which is bizarre. Has anybody seen this before ? I only see two 
> places in the dial plan where all phones are called; and neither should be 
> tripped :(
>
Enable some SIP debugs and reproduce the problem.  Should be simple
enough to resolve.

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Re: [asterisk-users] Asterisk, HylaFax and Cardiff

[Jeff LaCoursiere]

On Mon, 23 Aug 2010, Don Kelly wrote:



I’m looking for a way to use our implementation of HylaFax on Asterisk with 
Cardiff (an
old installation of Cardiff document stuff).

Is someone doing that?

If no one has direct experience, is there a HylaFax client that emulates WinFax
print-to-fax?



Lots!

http://www.hylafax.org/content/Desktop_Client_Software

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[asterisk-users] Asterisk, HylaFax and Cardiff

[Don Kelly]

I'm looking for a way to use our implementation of HylaFax on Asterisk with
Cardiff (an old installation of Cardiff document stuff).

 

Is someone doing that?

 

If no one has direct experience, is there a HylaFax client that emulates
WinFax print-to-fax?

--Don

Don Kelly

PCF Corp
People Come First
651 842-1000
888 Don Kell(y)
651 842-1001 fax

 

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Re: [asterisk-users] can't build resODBC on SUSE 11.3

[Tim Panton]

On 23 Aug 2010, at 18:07, Warren Selby wrote:

> On Mon, Aug 23, 2010 at 11:34 AM, Tim Panton  wrote:
> What is  menuselect actually looking for when it blocks me from selecting 
> res_odbc ?
> 
> I've got unixOdbc installed and working. I also have /usr/lib64/libltdl.so.3 
> - so I'm confused
> as it is claiming these are the pre-requisites ?
> 
> How can I best track down what it _thinks_ is missing ?
> 
> (This is on asterisk 1.8 svn trunk - but I don't think that is important,
> I think it is a package number issue)
> 
> Thanks in advance,
> 
> Tim.
> 
> 
> Tim Panton - Web/VoIP consultant and implementor
> www.westhawk.co.uk
> 
> 
> You need to install the -devel packages of libtool-ltdl and unixODBC.

Ah, libtool was what I was missing - thanks!

Tim.

Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk



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Re: [asterisk-users] IAX2 and other modules load error on 1.8.0-beta3

[Tilghman Lesher]

On Monday 23 August 2010 13:56:10 Ira wrote:
> At 11:28 AM 8/23/2010, you wrote:
> >In the future (and this goes for everybody, not just you), if you do not
> >understand a request made by a developer, PLEASE ask that question,
> > instead of giving us no feedback whatsoever.
> >
> >I'll defer to Paul's excellent set of instructions as to how to test a
> >proposed patch.
>
> But the automatically generated messages I got didn't say anything
> indicating that a response is needed, nor did they give a hint what
> to do with the info.

Sorry that that wasn't clear.  I need you to patch your installation,
recompile, and reinstall Asterisk, to verify that the patches fix the problem
reported.

> And I've no idea what "Paul's excellent set of 
> instructions" is or where I might find them?

Paul also reported to your previous message with a set of instructions.  If
you've already deleted his message, it is available in the archives:
http://lists.digium.com/pipermail/asterisk-users/2010-August/252735.html

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Re: [asterisk-users] IAX2 and other modules load error on 1.8.0-beta3

[Ira]

At 11:28 AM 8/23/2010, you wrote:
>I'll defer to Paul's excellent set of instructions as to how to test a
>proposed patch.

I found them. I don't use IAX2 and so it ended up in the recycle bin.

Ira 


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Re: [asterisk-users] IAX2 and other modules load error on 1.8.0-beta3

[Ira]

At 11:28 AM 8/23/2010, you wrote:
>In the future (and this goes for everybody, not just you), if you do not
>understand a request made by a developer, PLEASE ask that question, instead
>of giving us no feedback whatsoever.

>I'll defer to Paul's excellent set of instructions as to how to test a
>proposed patch.


But the automatically generated messages I got didn't say anything 
indicating that a response is needed, nor did they give a hint what 
to do with the info. And I've no idea what "Paul's excellent set of 
instructions" is or where I might find them?

Ira 


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Re: [asterisk-users] IAX2 and other modules load error on 1.8.0-beta3

[Tilghman Lesher]

On Monday 23 August 2010 12:19:38 Ira wrote:
> At 09:26 AM 8/23/2010, you wrote:
> >There's already an issue open for this, AND there is a patch posted, BUT
> > the reporter needs to verify that the patch(es) fix the issue for him:
> > https://issues.asterisk.org/view.php?id=17707
>
> And how was I supposed to now that?  I being the reporter.
>
> I hate to seem stupid, but when I got the email I looked there but
> have no idea what I'm supposed to do or how to do it. What is a patch
> and what do I do with it?

In the future (and this goes for everybody, not just you), if you do not
understand a request made by a developer, PLEASE ask that question, instead
of giving us no feedback whatsoever.  We're not trying to stump you or make
you feel stupid, honest; we just need to verify that a proposed solution fixes
the problem reported and not some other, unrelated problem.

I'll defer to Paul's excellent set of instructions as to how to test a
proposed patch.

-- 
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Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org

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[asterisk-users] channel stay up when extension unreachable

[Anton Raharja]

Hi,

We are using asterisk 1.4.34, ubuntu 10.4, below is suspicious activity
recorded in our full log. Could you help us to explain what had
happened. Thanks.

=== my friend, 801, from his room did a test by dialing echo test in
freepbx, *43:

[Aug 20 14:42:46] VERBOSE[14427] logger.c: -- Executing
[...@from-internal:1] Answer("SIP/801-03f5", "") in new stack
[Aug 20 14:42:46] VERBOSE[14427] logger.c: -- Executing
[...@from-internal:2] Wait("SIP/801-03f5", "1") in new stack
[Aug 20 14:42:47] VERBOSE[14427] logger.c: -- Executing
[...@from-internal:3] Playback("SIP/801-03f5", "demo-echotest") in
new stack
[Aug 20 14:42:47] VERBOSE[14427] logger.c: -- 
Playing 'demo-echotest' (language 'en')
[Aug 20 14:43:07] VERBOSE[14427] logger.c: -- Executing
[...@from-internal:4] Echo("SIP/801-03f5", "") in new stack

=== electricity down in 801's room and 801 became unreachable:

[Aug 20 14:46:45] NOTICE[8052] chan_sip.c: Peer '801' is now
UNREACHABLE!  Last qualify: 7

=== after 25 minutes power restored and 801 re-registered. 801 continue
testing, dialed several other destinations, also dialed *43 several
times. He didn't noticed any suspicious log and didn't bother to check
it coz 801 worked, calls were made and seems to be completed normally.

[Aug 20 15:13:04] VERBOSE[8052] logger.c: -- Registered SIP '801' at
xxx.xxx.xxx.xxx port 1806
[Aug 20 15:13:04] VERBOSE[8052] logger.c: -- Saved useragent "X-Lite
release 1104o stamp 56125" for peer 801
[Aug 20 15:13:04] NOTICE[8052] chan_sip.c: Peer '801' is now Reachable.
(17ms / 2000ms)

=== tonight, in our server, I noticed that I have a channel associated
with 801 elapsed for at least 81 hours after a "core show channels",
while theres no way 801 still available or making that call

=== later on after soft hangup:

[Aug 23 23:52:01] VERBOSE[14427] logger.c:   == Spawn extension
(from-internal, *43, 4) exited non-zero on 'SIP/801-03f5'
[Aug 23 23:52:01] VERBOSE[14427] logger.c: -- Executing
[...@from-internal:1] Macro("SIP/801-03f5", "hangupcall") in new stack
[Aug 23 23:52:01] VERBOSE[14427] logger.c: -- Executing
[...@macro-hangupcall:1] ResetCDR("SIP/801-03f5", "w") in new stack
[Aug 23 23:52:01] DEBUG[14427] app_macro.c: Executed application: ResetCDR
[Aug 23 23:52:01] VERBOSE[14427] logger.c: -- Executing
[...@macro-hangupcall:2] NoCDR("SIP/801-03f5", "") in new stack
[Aug 23 23:52:01] DEBUG[14427] app_macro.c: Executed application: NoCDR
[Aug 23 23:52:01] VERBOSE[14427] logger.c: -- Executing
[...@macro-hangupcall:3] GotoIf("SIP/801-03f5", "1?skiprg") in new stack
[Aug 23 23:52:01] VERBOSE[14427] logger.c: -- Goto
(macro-hangupcall,s,6)
[Aug 23 23:52:01] DEBUG[14427] app_macro.c: Executed application: GotoIf
[Aug 23 23:52:01] VERBOSE[14427] logger.c: -- Executing
[...@macro-hangupcall:6] GotoIf("SIP/801-03f5", "1?skipblkvm") in new
stack
[Aug 23 23:52:01] VERBOSE[14427] logger.c: -- Goto
(macro-hangupcall,s,9)
[Aug 23 23:52:01] DEBUG[14427] app_macro.c: Executed application: GotoIf
[Aug 23 23:52:01] VERBOSE[14427] logger.c: -- Executing
[...@macro-hangupcall:9] GotoIf("SIP/801-03f5", "1?theend") in new stack
[Aug 23 23:52:01] VERBOSE[14427] logger.c: -- Goto
(macro-hangupcall,s,11)
[Aug 23 23:52:01] DEBUG[14427] app_macro.c: Executed application: GotoIf
[Aug 23 23:52:01] VERBOSE[14427] logger.c: -- Executing
[...@macro-hangupcall:11] Hangup("SIP/801-03f5", "") in new stack
[Aug 23 23:52:01] VERBOSE[14427] logger.c:   == Spawn extension
(macro-hangupcall, s, 11) exited non-zero on 'SIP/801-03f5' in macro
'hangupcall'
[Aug 23 23:52:01] VERBOSE[14427] logger.c:   == Spawn extension
(from-internal, h, 1) exited non-zero on 'SIP/801-03f5'

anton


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Re: [asterisk-users] IAX2 and other modules load error on 1.8.0-beta3

[Paul Belanger]

On Mon, Aug 23, 2010 at 1:19 PM, Ira  wrote:
> I hate to seem stupid, but when I got the email I looked there but
> have no idea what I'm supposed to do or how to do it. What is a patch
> and what do I do with it?
>
Not stupid, in fact I would like to write documentation about testing
patches on the tracker.

Basically you need to do the following:

$ svn co http://svn.digium.com/svn/asterisk/branches/1.8 asterisk-18
$ cd asterisk-18
$ wget 'https://issues.asterisk.org/file_download.php?file_id=27079&type=bug'
-O - | patch -p0
$ wget 'https://issues.asterisk.org/file_download.php?file_id=27080&type=bug'
-O - | patch -p0
$ ./contrib/script/live_ast configure
$ make
$ ./contrib/script/live_ast install
$ ./contrib/script/live_ast samples
$ ./live/asterisk -vc

This will allow you to test the patch, besure to copy any existing
.config files into:

$ asterisk-18/live/etc/asterisk/

-- 
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com

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Re: [asterisk-users] How to debug this specific issue?

[Stefan Schmidt]

Steve Davies schrieb:
> I need suggestions please on how to determine where it is locking, and why.
>
> Many thanks,
> Steve
>
>   
hello,

have you allready tried strace ?
you could just easily start asterisk with this command:

strace asterisk -

or whatever options you want.
maybe you could see some more information with this.
you could also try

time asterisk -v

to see if its user or system time you loose.
and finaly using a second terminal and try rasterisk -x"core show locks" 
for example will also give you some information.

best regards

steve


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[asterisk-users] Transfer to non registered extension creates call hangup

[Rushikesh]

Hi list,


I have a small problem happening due to call transfer.

Whenever the call gets transfered to a remote extension ( which is not 
registered to asterisk ) it results in hangup().

When asterisk tries to dial the other extension it results in failure 
making the call cut down :(

Is there any way by which I can send the call back to the person who 
transfered it?


Regards,
Rushikesh

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Re: [asterisk-users] IAX2 and other modules load error on 1.8.0-beta3

[Ira]

At 09:26 AM 8/23/2010, you wrote:
>There's already an issue open for this, AND there is a patch posted, BUT the
>reporter needs to verify that the patch(es) fix the issue for him:
>https://issues.asterisk.org/view.php?id=17707

And how was I supposed to now that?  I being the reporter.

I hate to seem stupid, but when I got the email I looked there but 
have no idea what I'm supposed to do or how to do it. What is a patch 
and what do I do with it?

Ira 


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Re: [asterisk-users] asterisk + openBTS

[Tim Panton]

Last time I looked, no OpenBTS does not (yet) support handoff between base 
stations during a call. 

Handoff between calls can be done using SIP registrations to a central 
asterisk. 

Tim. 

Sent from my iPhone

On 23 Aug 2010, at 13:42, equis software  wrote:

> Do you know if OpenBTS support handoff?
> 
> Thanks
> 
> 
> On Fri, Aug 20, 2010 at 12:32 PM, Steve Totaro 
>  wrote:
> On Fri, Aug 20, 2010 at 10:41 AM, Tim Panton  wrote:
> >
> >
> >
> > On 19 Aug 2010, at 20:59, Randy R wrote:
> >
> >> On Thu, Aug 19, 2010 at 12:37 PM, Alan Lord (News)  
> >> wrote:
> >>> On 19/08/10 18:20, equis software wrote:
>  I want to know about asterisk and openBTS
> >>> This island runs it's GSM network on OpenBTS: http://www.niueisland.com/
> >>>
> >>> This was the place he presented about.
> >>>
> >>> Read the blog here: http://openbts.sourceforge.net/NiuePilot/
> >>
> >> and more about the installation here:
> >>
> >> http://vuc.me/2010/island-telephony-adventure/
> >>
> >
> >
> > I was part of the team that went to Niue to install OpenBTS,
> > I'm happy to answer questions if you have them,
> > although I'm not the radio guy - asterisk is more my thing :-)
> >
> > Tim.
> >
> > Tim Panton - Web/VoIP consultant and implementor
> > www.westhawk.co.uk
> 
> In all reality, Asterisk could be substituted with any other platform.
> 
> All the magic happens in the USRP, OpenBTS, and the cellular phones.
> Asterisk is merely handling the routing and voice, same as it ever
> was.  It is just the top of the stack.
> 
> I have two USRPs and a handful of daughter boards, and yes I have two
> flex 800s that have been physically altered so they can also be flex
> 1800s with a simple command line.  These are the boards you want for
> GSM (Cellular).
> 
> There is also a project to be able to listen into phone calls (thanks
> to the French making encryption so weak) besides a ton of other
> applications that can be dreamed up.
> 
> You can do passive radar, track people that have cell phones powered
> on,  RFID (Free tolls anyone?), WiFi, heck, you can even kill people
> with certain types of pacemakers.
> 
> While OpenBTS is cool and is on topic with Asterisk, read up on
> GNURadio and all the projects and applications you can come up with.
> It is really cool technology.
> 
> Start here http://gnuradio.org/redmine/projects/show/gnuradio but you
> can easily find things like this
> http://tech.mit.edu/V128/N30/subway/Defcon_Presentation.pdf or come up
> with your own with a bit of imagination and skillz.
> 
> Thanks,
> Steve Totaro
> 
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Re: [asterisk-users] can't build resODBC on SUSE 11.3

[Warren Selby]

On Mon, Aug 23, 2010 at 11:34 AM, Tim Panton  wrote:

> What is  menuselect actually looking for when it blocks me from selecting
> res_odbc ?
>
> I've got unixOdbc installed and working. I also have
> /usr/lib64/libltdl.so.3 - so I'm confused
> as it is claiming these are the pre-requisites ?
>
> How can I best track down what it _thinks_ is missing ?
>
> (This is on asterisk 1.8 svn trunk - but I don't think that is important,
> I think it is a package number issue)
>
> Thanks in advance,
>
> Tim.
>
>
> Tim Panton - Web/VoIP consultant and implementor
> www.westhawk.co.uk
>


You need to install the -devel packages of libtool-ltdl and unixODBC.

-- 
Thanks,
--Warren Selby
http://www.selbytech.com
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[asterisk-users] All phones ringing when temporary loss of Internet

[--[ UxBoD ]--]

Hi,

This is a real strange one and trying to phantom it out.  One of our clients is 
connected to our Asterisk installation, from two sites, via VPN which works 
great. Every so often one of the sites VPN tunnel goes does for say a couple of 
seconds. When that happens all the extensions, including both sites, ring which 
is bizarre. Has anybody seen this before ? I only see two places in the dial 
plan where all phones are called; and neither should be tripped :(
-- 
Thanks, Phil

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[asterisk-users] How to debug this specific issue?

[Steve Davies]

Hi,

I am happy with the usual GDB backtrace methods and so forth, but have
an issue that I cannot work out how to trace on 1.6.2.10.

If I use either the Bridge() app, or the manager Action: Bridge() in a
certain scenario (Basically to bridge 2 SIP channels, like an attended
transfer, resulting in 2 other SIP channels being discarded) then the
whole server locks solid. The console stops, the network stops,
something is hammering the box and nothing (including debug tools)
seem to be able to do anything about it.

If I 'nice' asterisk to lowest priority, and 'nice' a copy of 'top' to
highest priority, everything still locks. After a short period, the
box recovers, seemingly due to the 60 second RTP timer. Anything that
was being logged is lost.

My theory is the I am somehow causing a frame loop internal to
Asterisk by setting up some type of illegal bridge, but I used the
same code on 1.2 (I backported Bridge()) and it works just fine.

I need suggestions please on how to determine where it is locking, and why.

Many thanks,
Steve

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Re: [asterisk-users] Dahdi install gone wrong

[Todd Reese]

  I've made the system work by overlaying the old trixbox config in 
/etc/asterisk.  BUT this is a disaster waiting to happen with this client.

I'm having a hard time deciphering the trixbox extensions*.conf files in 
order to make a simple system where the client won't muck it up.

On 8/23/2010 11:37 AM, Cassius Smith wrote:
>* -Original Message-
>* From: Todd Reese
>* Reply-to: Asterisk Users Mailing List - Non-Commercial
>  Discussion
>* To: asterisk-users@lists.digium.com
>* Subject: [asterisk-users] Dahdi install gone wrong
>* Date: Mon, 23 Aug 2010 10:26:58 -0400
>*
>* Hi All,
>*
>*
>* I've got a project installing a Digium TDM800P card with 8 FXO's
>  in an
>* Asterisk box.
>*
>*
>* The computer is running Slackware 13.1 and I've installed the
>  current
>* Dahdi and Asterisk 1.6.2.11.
>*
>*
>* I've installed several boxes that are pure VOIP but, I haven't
>  installed
>* a Dahdi interface and I'm stumped.  I've got it to the point of
>  Dahdi
>* seeing the card and Asterisk recognizing dahdi but, I can't see
>  any
>* channels for the calls to come in on.
>*
>* I've had to borrow files from an old config of Trixbox (the
>  machine was
>* underpowered) to get to the point where I am in my setup.
>*
>* I would like to inquire some help from the group to get me up
>  and
>* receiving calls on the card.
>*
>*
>* Regards,
>*
>* Todd Reese
>*
>* Include:
>*
>*
>* chan_dahdi.conf==
>*
>*
>* ; Configuration file
>*
>* [trunkgroups]
>*
>* [channels]
>*
>* language=en
>* context=from-zaptel
>* signalling=fxs_ks
>* rxwink=300  ; Atlas seems to use long (250ms) winks
>* ;
>* ; Whether or not to do distinctive ring detection on FXO lines
>* ;
>* ;usedistinctiveringdetection=yes
>*
>* usecallerid=yes
>* hidecallerid=no
>* callwaiting=yes
>* usecallingpres=yes
>* callwaitingcallerid=yes
>* threewaycalling=yes
>* transfer=yes
>* cancallforward=yes
>* callreturn=yes
>* echocancel=yes
>* echocancelwhenbridged=no
>* ;echotraining=800
>* rxgain=0.0
>* txgain=0.0
>* group=0
>* callgroup=1
>* pickupgroup=1
>* immediate=no
>*
>* ;faxdetect=both
>* faxdetect=incoming
>* ;faxdetect=outgoing
>* ;faxdetect=no
>*
>* ;Include setup-pstn configs
>* #include dahdi-channels.conf
>*
>* group=1
>*
>* ;Include PBXconfig configs
>* #include chan_dahdi_additional.conf
>*
>*
>*
>* dahdi-channels.conf=
>*
>* ; Autogenerated by /usr/sbin/dahdi_genconf on Wed Aug 18
>  20:25:02 2010
>* ; If you edit this file and execute /usr/sbin/dahdi_genconf
>  again,
>* ; your manual changes will be LOST.
>* ; Dahdi Channels Configurations (chan_dahdi.conf)
>* ;
>* ; This is not intended to be a complete chan_dahdi.conf. Rather,
>  it is
>* intended
>* ; to be #include-d by /etc/chan_dahdi.conf that will include the
>  global
>* settings
>* ;
>*
>* ; Span 1: WCTDM/0 "Wildcard TDM800P Board 1" (MASTER)
>* ;;; line="1 WCTDM/0/0 FXSKS  (SWEC: MG2)"
>* signalling=fxs_ks
>* callerid=asreceived
>* group=0
>* context=from-pstn
>* channel =>  1
>* callerid=
>* group=
>* context=default
>*
>* ;;; line="2 WCTDM/0/1 FXSKS  (SWEC: MG2)"
>* signalling=fxs_ks
>* callerid=asreceived
>* group=0
>* context=from-pstn
>* channel =>  2
>* callerid=
>* group=
>* context=default
>*
>* ;;; line="3 WCTDM/0/2 FXSKS  (SWEC: MG2)"
>* signalling=fxs_ks
>* callerid=asreceived
>* group=0
>* context=from-pstn
>* channel =>  3
>* callerid=
>* group=
>* context=default
>*
>* ;;; line="4 WCTDM/0/3 FXSKS  (SWEC: MG2)"
>* signalling=fxs_ks
>* callerid=asreceived
>* group=0
>* context=from-pstn
>* channel =>  4
>* callerid=
>* group=
>* context=default
>*
>* ;;; line="5 WCTDM/0/4 FXSKS  (SWEC: MG2)"
>* signalling=fxs_ks
>* callerid=asreceived
>* group=0
>* context=from

[asterisk-users] can't build resODBC on SUSE 11.3

[Tim Panton]

What is  menuselect actually looking for when it blocks me from selecting 
res_odbc ?

I've got unixOdbc installed and working. I also have /usr/lib64/libltdl.so.3 - 
so I'm confused
as it is claiming these are the pre-requisites ?

How can I best track down what it _thinks_ is missing ?

(This is on asterisk 1.8 svn trunk - but I don't think that is important,
I think it is a package number issue)

Thanks in advance, 

Tim. 


Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk




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Re: [asterisk-users] Make a transfer for external line.

[Gustavo Duarte]

  Here are the output requested.


asteriscoII*CLI>  dialplan show s...@from-pstn
[ Context 'from-pstn' created by 'pbx_config' ]
   's' => 1. Answer()
[pbx_config]
 2. Background(main-menu)
[pbx_config]
 3. WaitExten()
[pbx_config]

-= 1 extension (3 priorities) in 1 context. =-

Thanks in advance.

Gustavo Duarte.



On 23/08/2010 11:08, Paul Belanger wrote:
> On Mon, Aug 23, 2010 at 9:28 AM, Gustavo Duarte  wrote:
>> [b] -- Unable to find extension '' in context 'from-pstn' [/b]
>>
>> Please let me to know if you need configuration files.
>>
> This is a configuration problem:
>
> *CLI>  dialplan show s...@from-pstn
>

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Re: [asterisk-users] Dahdi install gone wrong

[Todd Reese]

 They are FXO modules and yes, the lines are coming in from the telco.

On 8/23/2010 12:05 PM, Doug Dawson wrote:


The card you installed has FXO or FXS modules in it ? are you 
getting your lines directly from the telco co???



Doug D


On Mon 23/08/10 8:37 AM , Cassius Smith cass...@cassius.org sent:

* -Original Message-
* From: Todd Reese mailto:trees...@gmail.com>>
* Reply-to: Asterisk Users Mailing List - Non-Commercial
Discussion mailto:asterisk-users@lists.digium.com>>
* To: asterisk-users@lists.digium.com

* Subject: [asterisk-users] Dahdi install gone wrong
* Date: Mon, 23 Aug 2010 10:26:58 -0400
*
* Hi All,
*
*
* I've got a project installing a Digium TDM800P card with 8 FXO's
in an
* Asterisk box.
*
*
* The computer is running Slackware 13.1 and I've installed the
current
* Dahdi and Asterisk 1.6.2.11.
*
*
* I've installed several boxes that are pure VOIP but, I haven't
installed
* a Dahdi interface and I'm stumped. I've got it to the point of
Dahdi
* seeing the card and Asterisk recognizing dahdi but, I can't see
any
* channels for the calls to come in on.
*
* I've had to borrow files from an old config of Trixbox (the
machine was
* underpowered) to get to the point where I am in my setup.
*
* I would like to inquire some help from the group to get me up
and
* receiving calls on the card.
*
*
* Regards,
*
* Todd Reese
*
* Include:
*
*
* chan_dahdi.conf==
*
*
* ; Configuration file
*
* [trunkgroups]
*
* [channels]
*
* language=en
* context=from-zaptel
* signalling=fxs_ks
* rxwink=300 ; Atlas seems to use long (250ms) winks
* ;
* ; Whether or not to do distinctive ring detection on FXO lines
* ;
* ;usedistinctiveringdetection=yes
*
* usecallerid=yes
* hidecallerid=no
* callwaiting=yes
* usecallingpres=yes
* callwaitingcallerid=yes
* threewaycalling=yes
* transfer=yes
* cancallforward=yes
* callreturn=yes
* echocancel=yes
* echocancelwhenbridged=no
* ;echotraining=800
* rxgain=0.0
* txgain=0.0
* group=0
* callgroup=1
* pickupgroup=1
* immediate=no
*
* ;faxdetect=both
* faxdetect=incoming
* ;faxdetect=outgoing
* ;faxdetect=no
*
* ;Include setup-pstn configs
* #include dahdi-channels.conf
*
* group=1
*
* ;Include PBXconfig configs
* #include chan_dahdi_additional.conf
*
*
*
* dahdi-channels.conf=
*
* ; Autogenerated by /usr/sbin/dahdi_genconf on Wed Aug 18
20:25:02 2010
* ; If you edit this file and execute /usr/sbin/dahdi_genconf
again,
* ; your manual changes will be LOST.
* ; Dahdi Channels Configurations (chan_dahdi.conf)
* ;
* ; This is not intended to be a complete chan_dahdi.conf. Rather,
it is
* intended
* ; to be #include-d by /etc/chan_dahdi.conf that will include the
global
* settings
* ;
*
* ; Span 1: WCTDM/0 "Wildcard TDM800P Board 1" (MASTER)
* ;;; line="1 WCTDM/0/0 FXSKS (SWEC: MG2)"
* signalling=fxs_ks
* callerid=asreceived
* group=0
* context=from-pstn
* channel => 1
* callerid=
* group=
* context=default
*
* ;;; line="2 WCTDM/0/1 FXSKS (SWEC: MG2)"
* signalling=fxs_ks
* callerid=asreceived
* group=0
* context=from-pstn
* channel => 2
* callerid=
* group=
* context=default
*
* ;;; line="3 WCTDM/0/2 FXSKS (SWEC: MG2)"
* signalling=fxs_ks
* callerid=asreceived
* group=0
* context=from-pstn
* channel => 3
* callerid=
* group=
* context=default
*
* ;;; line="4 WCTDM/0/3 FXSKS (SWEC: MG2)"
* signalling=fxs_ks
* callerid=asreceived
* group=0
* context=from-pstn
* channel => 4
* callerid=
* group=
* context=default
*
* ;;; line="5 WCTDM/0/4 FXSKS (SWEC: MG2)"
* signalling=fxs_ks
* callerid=asreceived
* group=0
* context=from-pstn
* channel => 5
* callerid=
* group=
* context=default
*
* ;;; line="6 WCTDM/0/5 FXSKS (SWEC: MG2)"
* signalling=fxs_ks
* callerid=asreceived
* group=0
* context=from-pstn
* channel => 6
* callerid=
* group=
* context=default
*
* ;;; line="7 WCTDM/0/6 FXSKS (SWEC: MG2)"
* signalling=fxs_ks
* callerid=asreceived
* group=0
* context=from-pstn
* channel => 7
* callerid=
* group=
* context=default
*
* ;;; line="8 WCTDM/0/7 FXSKS (SWEC: MG2)"
* signalling=fxs_ks
* callerid=asreceived
* group=0
* context=from-pstn
* channel => 8
* callerid=
* group=
* context=default
*
*
* =system.conf===

Re: [asterisk-users] IAX2 and other modules load error on 1.8.0-beta3

[Tilghman Lesher]

On Monday 23 August 2010 10:35:26 Miguel Molina wrote:
> I've installed asterisk 1.8.0-beta3, and found this errors related to
> several modules:
>
> [Aug 23 08:31:54] WARNING[3883]: loader.c:429 load_dynamic_module: Error
> loading module 'chan_iax2.so': /usr/lib/asterisk
> /modules/chan_iax2.so: undefined symbol: ast_aes_set_decrypt_key
> [Aug 23 08:31:54] WARNING[3883]: loader.c:819 load_resource: Module
> 'chan_iax2.so' could not be loaded.
> [Aug 23 08:31:54] WARNING[3883]: loader.c:429 load_dynamic_module: Error
> loading module 'pbx_dundi.so': /usr/lib/asterisk
> /modules/pbx_dundi.so: undefined symbol: ast_check_signature_bin
> [Aug 23 08:31:54] WARNING[3883]: loader.c:819 load_resource: Module
> 'pbx_dundi.so' could not be loaded.
> [Aug 23 08:31:54] WARNING[3883]: loader.c:429 load_dynamic_module: Error
> loading module 'func_aes.so': /usr/lib/asterisk/
> modules/func_aes.so: undefined symbol: ast_aes_set_decrypt_key
> [Aug 23 08:31:54] WARNING[3883]: loader.c:819 load_resource: Module
> 'func_aes.so' could not be loaded.
>
> What are the requirements for these modules? Or is this an issue that
> needs to be reported on the bugtracker?

There's already an issue open for this, AND there is a patch posted, BUT the
reporter needs to verify that the patch(es) fix the issue for him:
https://issues.asterisk.org/view.php?id=17707

Failing a reporter, someone else who has the problem needs to test the patch
and verify the fix.

-- 
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Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Dahdi install gone wrong

[Doug Dawson]

 
 The card you installed has FXO or FXS modules in it ? are you getting
your lines directly from the telco co???

 Doug D

 On Mon 23/08/10 8:37 AM , Cassius Smith cass...@cassius.org sent:
  * -Original Message-
 * From: Todd Reese 
 * Reply-to: Asterisk Users Mailing List - Non-Commercial
 Discussion 
 * To: asterisk-users@lists.digium.com [3]
 * Subject: [asterisk-users] Dahdi install gone wrong
 * Date: Mon, 23 Aug 2010 10:26:58 -0400
 * 
 * Hi All,
 * 
 * 
 * I've got a project installing a Digium TDM800P card with 8 FXO's
 in an 
 * Asterisk box.
 * 
 * 
 * The computer is running Slackware 13.1 and I've installed the
 current 
 * Dahdi and Asterisk 1.6.2.11.
 * 
 * 
 * I've installed several boxes that are pure VOIP but, I haven't
 installed 
 * a Dahdi interface and I'm stumped. I've got it to the point of
 Dahdi 
 * seeing the card and Asterisk recognizing dahdi but, I can't see
 any 
 * channels for the calls to come in on.
 * 
 * I've had to borrow files from an old config of Trixbox (the
 machine was 
 * underpowered) to get to the point where I am in my setup.
 * 
 * I would like to inquire some help from the group to get me up
 and 
 * receiving calls on the card.
 * 
 * 
 * Regards,
 * 
 * Todd Reese
 * 
 * Include:
 * 
 * 
 * chan_dahdi.conf==
 * 
 * 
 * ; Configuration file
 * 
 * [trunkgroups]
 * 
 * [channels]
 * 
 * language=en
 * context=from-zaptel
 * signalling=fxs_ks
 * rxwink=300 ; Atlas seems to use long (250ms) winks
 * ;
 * ; Whether or not to do distinctive ring detection on FXO lines
 * ;
 * ;usedistinctiveringdetection=yes
 * 
 * usecallerid=yes
 * hidecallerid=no
 * callwaiting=yes
 * usecallingpres=yes
 * callwaitingcallerid=yes
 * threewaycalling=yes
 * transfer=yes
 * cancallforward=yes
 * callreturn=yes
 * echocancel=yes
 * echocancelwhenbridged=no
 * ;echotraining=800
 * rxgain=0.0
 * txgain=0.0
 * group=0
 * callgroup=1
 * pickupgroup=1
 * immediate=no
 * 
 * ;faxdetect=both
 * faxdetect=incoming
 * ;faxdetect=outgoing
 * ;faxdetect=no
 * 
 * ;Include setup-pstn configs
 * #include dahdi-channels.conf
 * 
 * group=1
 * 
 * ;Include PBXconfig configs
 * #include chan_dahdi_additional.conf
 * 
 * 
 * 
 * dahdi-channels.conf=
 * 
 * ; Autogenerated by /usr/sbin/dahdi_genconf on Wed Aug 18
 20:25:02 2010
 * ; If you edit this file and execute /usr/sbin/dahdi_genconf
 again,
 * ; your manual changes will be LOST.
 * ; Dahdi Channels Configurations (chan_dahdi.conf)
 * ;
 * ; This is not intended to be a complete chan_dahdi.conf. Rather,
 it is 
 * intended
 * ; to be #include-d by /etc/chan_dahdi.conf that will include the
 global 
 * settings
 * ;
 * 
 * ; Span 1: WCTDM/0 "Wildcard TDM800P Board 1" (MASTER)
 * ;;; line="1 WCTDM/0/0 FXSKS (SWEC: MG2)"
 * signalling=fxs_ks
 * callerid=asreceived
 * group=0
 * context=from-pstn
 * channel => 1
 * callerid=
 * group=
 * context=default
 * 
 * ;;; line="2 WCTDM/0/1 FXSKS (SWEC: MG2)"
 * signalling=fxs_ks
 * callerid=asreceived
 * group=0
 * context=from-pstn
 * channel => 2
 * callerid=
 * group=
 * context=default
 * 
 * ;;; line="3 WCTDM/0/2 FXSKS (SWEC: MG2)"
 * signalling=fxs_ks
 * callerid=asreceived
 * group=0
 * context=from-pstn
 * channel => 3
 * callerid=
 * group=
 * context=default
 * 
 * ;;; line="4 WCTDM/0/3 FXSKS (SWEC: MG2)"
 * signalling=fxs_ks
 * callerid=asreceived
 * group=0
 * context=from-pstn
 * channel => 4
 * callerid=
 * group=
 * context=default
 * 
 * ;;; line="5 WCTDM/0/4 FXSKS (SWEC: MG2)"
 * signalling=fxs_ks
 * callerid=asreceived
 * group=0
 * context=from-pstn
 * channel => 5
 * callerid=
 * group=
 * context=default
 * 
 * ;;; line="6 WCTDM/0/5 FXSKS (SWEC: MG2)"
 * signalling=fxs_ks
 * callerid=asreceived
 * group=0
 * context=from-pstn
 * channel => 6
 * callerid=
 * group=
 * context=default
 * 
 * ;;; line="7 WCTDM/0/6 FXSKS (SWEC: MG2)"
 * signalling=fxs_ks
 * callerid=asreceived
 * group=0
 * context=from-pstn
 * channel => 7
 * callerid=
 * group=
 * context=default
 * 
 * ;;; line="8 WCTDM/0/7 FXSKS (SWEC: MG2)"
 * signalling=fxs_ks
 * callerid=asreceived
 * group=0
 * context=from-pstn
 * channel => 8
 * callerid=
 * group=
 * context=default
 * 
 * 
 * =system.conf=
 * 
 * 
 * # Autogenerated by /usr/sbin/dahdi_genconf on Sun Aug 22
 19:34:02 2010
 * # If you edit this file and execute /usr/sbin/dahdi_genconf
 again,
 * # your manual changes will be LOST.
 * # Dahdi Configuration File
 * #
 * # This file is parsed by the Dahdi Configurator, dahdi_cfg
 * #
 * # Global data
 * 
 * loadzone = us
 * defaultzone = us
 * 
 * # Span 1: WCTDM/0 "Wildcard TDM800P Board 1" (MASTER)
 * fxsks=1
 * #echocanceller=mg2,1
 * fxsks=2
 * #echocanceller=mg2,2
 * fxsks=3
 * #echocanceller=mg2,3
 * fxsks=4
 * #echocanceller=mg2,4
 * fxsks=5
 * #echocanceller=mg2,5
 * fxsks=6
 * #echocanceller=mg2,6
 * fxsks=7
 * #echocanceller=mg2,7
 * fxsks=8
 * #echocanceller=mg2,8

 Can you include the germane part of the dialplan also?

 -- 
 __

[asterisk-users] Asterisk voicemail server - gsm notifications

[Matt]

Has anyone successfully implemented Asterisk as a voicemail server for a
GSM/cellular system and worked out a way to send notifications of new
messages to the phones?
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[asterisk-users] DAHDI not detecting caller hangup

[--[ UxBoD ]--]

Hi,

Odd problem have just noticed in that when I call into the PBX DAHDI detects 
the call and hands it off to the extension, if I then hang up it still 
continues to process through the dialplan.

This is what I have in chan_dahdi.conf:

[channels]
language=en
echocancel=yes
usecallerid=yes
cidsignalling=v23
sendcalleridafter = 2
hanguponpolarityswitch=yes
rxgain=2.0
txgain=3.0
progzone=uk
signalling=fxs_ks
callerid=asreceived
group=0
context=inbound-dahdi
channel => 1
callerid=
group=
context=inbound-dahdi

and using Asterisk 1.6.2.11 and DAHDI 2.3.0.1.  Dahdi_scan looks okay as-well:

[1]
active=yes
alarms=OK
description=Wildcard TDM400P REV E/F Board 5
name=WCTDM/4
manufacturer=Digium
devicetype=Wildcard TDM400P REV E/F
location=PCI Bus 02 Slot 02
basechan=1
totchans=4
irq=169
type=analog
port=1,FXO
port=2,none
port=3,none
port=4,none
-- 
Thanks, Phil

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Re: [asterisk-users] Dahdi install gone wrong

[Cassius Smith]

  * -Original Message-
  * From: Todd Reese 
  * Reply-to: Asterisk Users Mailing List - Non-Commercial
Discussion 
  * To: asterisk-users@lists.digium.com
  * Subject: [asterisk-users] Dahdi install gone wrong
  * Date: Mon, 23 Aug 2010 10:26:58 -0400
  * 
  * Hi All,
  * 
  * 
  * I've got a project installing a Digium TDM800P card with 8 FXO's
in an 
  * Asterisk box.
  * 
  * 
  * The computer is running Slackware 13.1 and I've installed the
current 
  * Dahdi and Asterisk 1.6.2.11.
  * 
  * 
  * I've installed several boxes that are pure VOIP but, I haven't
installed 
  * a Dahdi interface and I'm stumped.  I've got it to the point of
Dahdi 
  * seeing the card and Asterisk recognizing dahdi but, I can't see
any 
  * channels for the calls to come in on.
  * 
  * I've had to borrow files from an old config of Trixbox (the
machine was 
  * underpowered) to get to the point where I am in my setup.
  * 
  * I would like to inquire some help from the group to get me up
and 
  * receiving calls on the card.
  * 
  * 
  * Regards,
  * 
  * Todd Reese
  * 
  * Include:
  * 
  * 
  * chan_dahdi.conf==
  * 
  * 
  * ; Configuration file
  * 
  * [trunkgroups]
  * 
  * [channels]
  * 
  * language=en
  * context=from-zaptel
  * signalling=fxs_ks
  * rxwink=300  ; Atlas seems to use long (250ms) winks
  * ;
  * ; Whether or not to do distinctive ring detection on FXO lines
  * ;
  * ;usedistinctiveringdetection=yes
  * 
  * usecallerid=yes
  * hidecallerid=no
  * callwaiting=yes
  * usecallingpres=yes
  * callwaitingcallerid=yes
  * threewaycalling=yes
  * transfer=yes
  * cancallforward=yes
  * callreturn=yes
  * echocancel=yes
  * echocancelwhenbridged=no
  * ;echotraining=800
  * rxgain=0.0
  * txgain=0.0
  * group=0
  * callgroup=1
  * pickupgroup=1
  * immediate=no
  * 
  * ;faxdetect=both
  * faxdetect=incoming
  * ;faxdetect=outgoing
  * ;faxdetect=no
  * 
  * ;Include setup-pstn configs
  * #include dahdi-channels.conf
  * 
  * group=1
  * 
  * ;Include PBXconfig configs
  * #include chan_dahdi_additional.conf
  * 
  * 
  * 
  * dahdi-channels.conf=
  * 
  * ; Autogenerated by /usr/sbin/dahdi_genconf on Wed Aug 18
20:25:02 2010
  * ; If you edit this file and execute /usr/sbin/dahdi_genconf
again,
  * ; your manual changes will be LOST.
  * ; Dahdi Channels Configurations (chan_dahdi.conf)
  * ;
  * ; This is not intended to be a complete chan_dahdi.conf. Rather,
it is 
  * intended
  * ; to be #include-d by /etc/chan_dahdi.conf that will include the
global 
  * settings
  * ;
  * 
  * ; Span 1: WCTDM/0 "Wildcard TDM800P Board 1" (MASTER)
  * ;;; line="1 WCTDM/0/0 FXSKS  (SWEC: MG2)"
  * signalling=fxs_ks
  * callerid=asreceived
  * group=0
  * context=from-pstn
  * channel => 1
  * callerid=
  * group=
  * context=default
  * 
  * ;;; line="2 WCTDM/0/1 FXSKS  (SWEC: MG2)"
  * signalling=fxs_ks
  * callerid=asreceived
  * group=0
  * context=from-pstn
  * channel => 2
  * callerid=
  * group=
  * context=default
  * 
  * ;;; line="3 WCTDM/0/2 FXSKS  (SWEC: MG2)"
  * signalling=fxs_ks
  * callerid=asreceived
  * group=0
  * context=from-pstn
  * channel => 3
  * callerid=
  * group=
  * context=default
  * 
  * ;;; line="4 WCTDM/0/3 FXSKS  (SWEC: MG2)"
  * signalling=fxs_ks
  * callerid=asreceived
  * group=0
  * context=from-pstn
  * channel => 4
  * callerid=
  * group=
  * context=default
  * 
  * ;;; line="5 WCTDM/0/4 FXSKS  (SWEC: MG2)"
  * signalling=fxs_ks
  * callerid=asreceived
  * group=0
  * context=from-pstn
  * channel => 5
  * callerid=
  * group=
  * context=default
  * 
  * ;;; line="6 WCTDM/0/5 FXSKS  (SWEC: MG2)"
  * signalling=fxs_ks
  * callerid=asreceived
  * group=0
  * context=from-pstn
  * channel => 6
  * callerid=
  * group=
  * context=default
  * 
  * ;;; line="7 WCTDM/0/6 FXSKS  (SWEC: MG2)"
  * signalling=fxs_ks
  * callerid=asreceived
  * group=0
  * context=from-pstn
  * channel => 7
  * callerid=
  * group=
  * context=default
  * 
  * ;;; line="8 WCTDM/0/7 FXSKS  (SWEC: MG2)"
  * signalling=fxs_ks
  * callerid=asreceived
  * group=0
  * context=from-pstn
  * channel => 8
  * callerid=
  * group=
  * c

[asterisk-users] IAX2 and other modules load error on 1.8.0-beta3

[Miguel Molina]

Hi everyone,

I've installed asterisk 1.8.0-beta3, and found this errors related to 
several modules:

[Aug 23 08:31:54] WARNING[3883]: loader.c:429 load_dynamic_module: Error 
loading module 'chan_iax2.so': /usr/lib/asterisk
/modules/chan_iax2.so: undefined symbol: ast_aes_set_decrypt_key
[Aug 23 08:31:54] WARNING[3883]: loader.c:819 load_resource: Module 
'chan_iax2.so' could not be loaded.
[Aug 23 08:31:54] WARNING[3883]: loader.c:429 load_dynamic_module: Error 
loading module 'pbx_dundi.so': /usr/lib/asterisk
/modules/pbx_dundi.so: undefined symbol: ast_check_signature_bin
[Aug 23 08:31:54] WARNING[3883]: loader.c:819 load_resource: Module 
'pbx_dundi.so' could not be loaded.
[Aug 23 08:31:54] WARNING[3883]: loader.c:429 load_dynamic_module: Error 
loading module 'func_aes.so': /usr/lib/asterisk/
modules/func_aes.so: undefined symbol: ast_aes_set_decrypt_key
[Aug 23 08:31:54] WARNING[3883]: loader.c:819 load_resource: Module 
'func_aes.so' could not be loaded.

What are the requirements for these modules? Or is this an issue that 
needs to be reported on the bugtracker?

Have a nice day.

Regards,

-- 
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center


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Re: [asterisk-users] problem with mssql and Asterisk 1.8.0 beta3

[Tilghman Lesher]

On Monday 23 August 2010 09:34:23 Paul Belanger wrote:
> On Mon, Aug 23, 2010 at 10:03 AM, Tilghman Lesher  
wrote:
> > For this one, you need to ensure that res_odbc.so loads before
> > res_config_odbc.so.  Actually, the load order in 1.8 is such that, unless
> > you're using static realtime, you should not be using the 'preload'
> > directive at all, and everything will just naturally load in the right
> > order.
>
> However, using 'preload' _should_ not be an issue, should it?

Preload isn't an issue, as long as you have all prerequisites.  If you're
missing one, you'll get an error emitted as to those symbols which are
lacking.  However, unless you're using static realtime for configuration files
which load prior to any modules (i.e. parts of the core), then using the
preload directive is unnecessary.

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] problem with mssql and Asterisk 1.8.0 beta3

[unserossi]

> For this one, you need to ensure that res_odbc.so loads before



> res_config_odbc.so.  Actually, the load order in 1.8 is such that, unless

> you're using static realtime, you should not be using the 'preload' directive

> at all, and everything will just naturally load in the right order.

>

However, using 'preload' _should_ not be an issue, should it?



@Oliver, do you mind posting your modules.conf file? I'd like to try a test.



-- 

No Problem Paul.
Here it is.

 


modules.conf
Description: Binary data
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Re: [asterisk-users] mapping of disconnect reasons

[Harel Cohen]

Sorry for the late response.
Philipp,
I've checked the file below and also the suggested voip-info link. None of 
those describe how or why Asterisk assumed that 402 should be mapped to "NORMAL 
TERMINATION" status. Both places refer to how Asterisk status should be mapped 
to SIP cause and not vice-versa. Could you (or someone) please take another 
look to locate the correct file?
Thanks
Harel

--

Message: 4
Date: Wed, 04 Aug 2010 15:20:05 +0200
From: Philipp von Klitzing 
Subject: Re: [asterisk-users] mapping of disconnect reasons
To: Asterisk Users Mailing List - Non-Commercial Discussion

Message-ID:
<4c598525.14932.1d6...@klitzing.pool.informatik.rwth-aachen.de>
Content-Type: text/plain; charset=US-ASCII

> The mapping in Asterisk 1.4.24 is the problem: 402 "Payment Required"
> is mapped to 16 "Normal termination" instead of 21 "Call Rejected".
> Could you direct me to the relevant file of code where these mappings 
> are done? Before reporting a bug I would like to confirm whether this 
> issue has been addressed on newer releases.

Look in channels/chan_sip.c and search for "3398"

See also:
http://www.voip-
info.org/wiki/index.php?page=Asterisk+variable+hangupcause

Philipp




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Re: [asterisk-users] problem with mssql and Asterisk 1.8.0 beta3

[unserossi]

> [Aug 23 15:06:12] WARNING[7180]: loader.c:387 load_dynamic_module: Error

> loading module 'res_config_odbc.so':

> /usr/lib/asterisk/modules/res_config_odbc.so: undefined symbol:

> ast_odbclear_cache



For this one, you need to ensure that res_odbc.so loads before 

res_config_odbc.so.  Actually, the load order in 1.8 is such that, unless

you're using static realtime, you should not be using the 'preload' directive

at all, and everything will just naturally load in the right order.



--

I uncommented the two lines in modules.conf (this is what you meant?)

preload => res_odbc.so
preload => res_config_odbc.so

but asterisk still does not want to start as long as the dsn is set to enabled 
in res_odbc.conf.

I have my sip users/peers configured in realtime, don't know if this is called 
'static realtime'?
With 1.6 it works without preloading the modules, with 1.8 I still get

[Aug 23 16:48:06] NOTICE[3130]: loader.c:1098 load_modules: 2 modules will be 
loaded.
  == Parsing '/etc/asterisk/res_odbc.conf':   == Found
[Aug 23 16:48:06] NOTICE[3130]: res_odbc.c:1471 odbc_obj_connect: Connecting 
mssql
asterisk: net.c:348: tds_select: Assertion `timeout_seconds >= 0' failed.




 
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Re: [asterisk-users] problem with mssql and Asterisk 1.8.0 beta3

[Paul Belanger]

On Mon, Aug 23, 2010 at 10:03 AM, Tilghman Lesher  wrote:
> For this one, you need to ensure that res_odbc.so loads before
> res_config_odbc.so.  Actually, the load order in 1.8 is such that, unless
> you're using static realtime, you should not be using the 'preload' directive
> at all, and everything will just naturally load in the right order.
>
However, using 'preload' _should_ not be an issue, should it?

@Oliver, do you mind posting your modules.conf file? I'd like to try a test.

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[asterisk-users] Dahdi install gone wrong

[Todd Reese]

  Hi All,


I've got a project installing a Digium TDM800P card with 8 FXO's in an 
Asterisk box.


The computer is running Slackware 13.1 and I've installed the current 
Dahdi and Asterisk 1.6.2.11.


I've installed several boxes that are pure VOIP but, I haven't installed 
a Dahdi interface and I'm stumped.  I've got it to the point of Dahdi 
seeing the card and Asterisk recognizing dahdi but, I can't see any 
channels for the calls to come in on.

I've had to borrow files from an old config of Trixbox (the machine was 
underpowered) to get to the point where I am in my setup.

I would like to inquire some help from the group to get me up and 
receiving calls on the card.


Regards,

Todd Reese

Include:


chan_dahdi.conf==


; Configuration file

[trunkgroups]

[channels]

language=en
context=from-zaptel
signalling=fxs_ks
rxwink=300  ; Atlas seems to use long (250ms) winks
;
; Whether or not to do distinctive ring detection on FXO lines
;
;usedistinctiveringdetection=yes

usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
;echotraining=800
rxgain=0.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no

;faxdetect=both
faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no

;Include setup-pstn configs
#include dahdi-channels.conf

group=1

;Include PBXconfig configs
#include chan_dahdi_additional.conf



dahdi-channels.conf=

; Autogenerated by /usr/sbin/dahdi_genconf on Wed Aug 18 20:25:02 2010
; If you edit this file and execute /usr/sbin/dahdi_genconf again,
; your manual changes will be LOST.
; Dahdi Channels Configurations (chan_dahdi.conf)
;
; This is not intended to be a complete chan_dahdi.conf. Rather, it is 
intended
; to be #include-d by /etc/chan_dahdi.conf that will include the global 
settings
;

; Span 1: WCTDM/0 "Wildcard TDM800P Board 1" (MASTER)
;;; line="1 WCTDM/0/0 FXSKS  (SWEC: MG2)"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 1
callerid=
group=
context=default

;;; line="2 WCTDM/0/1 FXSKS  (SWEC: MG2)"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 2
callerid=
group=
context=default

;;; line="3 WCTDM/0/2 FXSKS  (SWEC: MG2)"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 3
callerid=
group=
context=default

;;; line="4 WCTDM/0/3 FXSKS  (SWEC: MG2)"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 4
callerid=
group=
context=default

;;; line="5 WCTDM/0/4 FXSKS  (SWEC: MG2)"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 5
callerid=
group=
context=default

;;; line="6 WCTDM/0/5 FXSKS  (SWEC: MG2)"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 6
callerid=
group=
context=default

;;; line="7 WCTDM/0/6 FXSKS  (SWEC: MG2)"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 7
callerid=
group=
context=default

;;; line="8 WCTDM/0/7 FXSKS  (SWEC: MG2)"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 8
callerid=
group=
context=default


=system.conf=


# Autogenerated by /usr/sbin/dahdi_genconf on Sun Aug 22 19:34:02 2010
# If you edit this file and execute /usr/sbin/dahdi_genconf again,
# your manual changes will be LOST.
# Dahdi Configuration File
#
# This file is parsed by the Dahdi Configurator, dahdi_cfg
#
# Global data

loadzone= us
defaultzone = us

# Span 1: WCTDM/0 "Wildcard TDM800P Board 1" (MASTER)
fxsks=1
#echocanceller=mg2,1
fxsks=2
#echocanceller=mg2,2
fxsks=3
#echocanceller=mg2,3
fxsks=4
#echocanceller=mg2,4
fxsks=5
#echocanceller=mg2,5
fxsks=6
#echocanceller=mg2,6
fxsks=7
#echocanceller=mg2,7
fxsks=8
#echocanceller=mg2,8


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Re: [asterisk-users] Opensource Speech recognition for Asterisk

[Paul Belanger]

On Sat, Aug 21, 2010 at 7:09 PM, Zeeshan Zakaria  wrote:
> Your experience could be different and better then most, and you have
> certainly complete right of your own opinion.
>
Speech recognition is only as effective as your grammars and they are
never 100%. The require lots tuning and analyzing to be effective.  My
clients I works with simply believe once the IVR is developed and
installed, the work end there; This is never the case.

We actively consult with a Speech Linguist when are in the tuning
phase of the grammars.  Countless hours listening to recorded calls,
rebuilding grammars, in an attempt to increase accuracy.  Lots of hard
work, but when you actively improve grammars, recognition rates
increase.

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Re: [asterisk-users] How to prevent soft hangup from being necessary ?

[Paul Belanger]

On Mon, Aug 23, 2010 at 7:14 AM, Olivier  wrote:
> So, which tools are available to automatically detect that SIP channels are
> up without but no RTP media is flowing in or from them ?
>
Make sure you explicitly call Hangup() and implement max timeouts for
your channels.

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Re: [asterisk-users] Make a transfer for external line.

[Paul Belanger]

On Mon, Aug 23, 2010 at 9:28 AM, Gustavo Duarte  wrote:
>    [b] -- Unable to find extension '' in context 'from-pstn' [/b]
>
> Please let me to know if you need configuration files.
>
This is a configuration problem:

*CLI> dialplan show s...@from-pstn

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[asterisk-users] outbound SIP trunk hunting (or any fxo for that matter)

[Infra]

On Aug 7, 2007 'Mojo' wrote:

Nicholas Blasgen wrote:
> I've got 4 SIP phone lines with a call-limit of 2 for each.  I've
> written a handy macro to allow my users to dial a phone number and the
> macro will figure out the next available line to use by first checking
> if the GROUP() is over 2 and then checking to see if ChanIsAvail() as a
> backup, and if it can't use the line for either reason it goes to the
> next line.  The problem is that there are enough situations that the
> Macro gets called twice without much time seperation.  Both macros check
> the group() number, it comes back as free, they check the line
> availability and it's open, and they try dialing.  But because they both
> started at more or less the same instant, they've both at the same stage
> in the macro and sometimes (maybe 10% of the time) a macro will try
> dialing on a line that's already in use.
>
> My question is this.  Is it possible to tell Asterisk to execute part of
> a macro as a block without allowing any other commands to be processed
> during that time?  Some way to LOCK the dialplan (as you'd do in SQL).
> I want my macro to be able to execute the part of the code that checks
> line status and then sets the GROUP() without allowing any other
> dialplans from running during that time.  Anyone know if this is a
> current feature?
>
> --
> /Nick

What would be a correct way to do this in 1.4.x?

Thanks,

Michael

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Re: [asterisk-users] How to do barging using asterisk server.

[Paul Belanger]

On Mon, Aug 23, 2010 at 2:58 AM, Janu Mukherjee  wrote:

> system. Press 1 to trade 2 to selland so on". I want to stop this and
> press 1 or talk even before the prompt finishes. How to achieve this. I was
> told that this is similar to barging in asterisk but i could not get info on

*CLI> core show application Background

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Re: [asterisk-users] WaitExten() always times out

[Miguel Molina]

El 20/08/10 16:14, Kathryn Jones escribió:
> Thanks for all the help, but I still can't find what's wrong.
>
> I enabled console => notice,warning,error,debug,dtmf like Miguel 
> suggested. The output is attached.
>
> I noticed that the rtp.c session never starts, which as I understand 
> is what catches the dtmf tone, but I could not find how to start it :s.
>
> The Answer() and waitExten(5,m) didn't fix my problem. I hope someone 
> can help me see the problem after looking at the attached console output.
The following line brought my attention:

[Aug 20 16:50:04] DEBUG[5319]: channel.c:1882 __ast_answer: Didn't receive a 
media frame from SIP/xx.xx.xxx.xx-0026 within 500 ms of answering. 
Continuing anyway



Are your sure that RTP audio (media) is correctly received in asterisk? 
I suspect network or firewall problems. Also, you said that you were 
going to receive calls from the PSTN, but are you testing from a SIP 
endpoint?

Regards,

-- 
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center


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Re: [asterisk-users] problem with mssql and Asterisk 1.8.0 beta3

[Tilghman Lesher]

On Monday 23 August 2010 08:14:43 unsero...@aol.com wrote:
> Hi all,
>
> I am testing with Asterisk 1.8.0 beta3 using realtime with a mssql server
> using freetds and unixodbc, which works with 1.6.1.20.
>
> With the same config in 1.8 I get an error when trying to start asterisk
> which says:
>
>
> [Aug 23 15:06:12] WARNING[7180]: loader.c:387 load_dynamic_module: Error
> loading module 'res_config_odbc.so':
> /usr/lib/asterisk/modules/res_config_odbc.so: undefined symbol:
> ast_odbc_clear_cache

For this one, you need to ensure that res_odbc.so loads before 
res_config_odbc.so.  Actually, the load order in 1.8 is such that, unless
you're using static realtime, you should not be using the 'preload' directive
at all, and everything will just naturally load in the right order.

> [Aug 23 15:06:12] NOTICE[7180]: res_odbc.c:1471 
> odbc_obj_connect: Connecting mssql asterisk: net.c:348: tds_select:
> Assertion `timeout_seconds >= 0' failed.

This is internal to the freetds driver and is therefore not a bug in Asterisk.

-- 
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Check us out at: www.digium.com & www.asterisk.org

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[asterisk-users] Make a transfer for external line.

[Gustavo Duarte]

   Hi all,

We have an asterisk version v1.6.1.20 with a TDM400 board (2 FXS and 2 
FXO).
We want to do a transfer "blind" and "attended" from a line external 
connected to one FXO.

We have made configuration, and transfers from internal lines (FXS) work 
fine but from (FXO) not.

We have made 2 test, one work fine from FXS and the other form FXO no.

Test 1, work fine:

1) A (FXS-1) --- call > B(FXS-2)
2) B (FXS-2) press #1 (blind transfer) after that press dtmf  9 (to 
dial  FXO-1).
3) B hungup.
4) A  connected to  linea externa FXO-1.


Test 2, did't work.

1) A (FXS-1) --- call > B(FX0-1)
2) B (FX0-1) digita #1 (blind transfer), listen tone and when press any 
dtm the call hungup.

Bellow you can see, log when we try a blind transfer from FXO.

[Aug 18 09:16:58] DEBUG[4756]: features.c:2053 ast_feature_interpret: 
Feature interpret: chan=SIP/gustavo-0002, peer=DAHDI/65-1, code=#1, 
sense=2, features=2, dynamic=#
[Aug 18 09:16:58] DEBUG[4756]: features.c:1948 feature_interpret_helper: 
Feature detected: fname=Blind Transfer sname=blindxfer exten=#1
[Aug 18 09:16:58] DEBUG[4756]: rtp.c:2648 ast_rtp_new_source: Setting 
the marker bit due to a source update
 -- Started music on hold, class 'default', on channel 
'SIP/gustavo-0002'
[Aug 18 09:16:58] DEBUG[4756]: channel.c:3710 set_format: Set channel 
DAHDI/65-1 to write format ulaw
 --  Playing 'pbx-transfer.ulaw' (language 'es')
[Aug 18 09:16:58] DEBUG[4756]: channel.c:3710 set_format: Set channel 
DAHDI/65-1 to write format slin
[Aug 18 09:16:59] DEBUG[3245]: rtp.c:1246 ast_rtcp_read: Got RTCP report 
of 108 bytes
[Aug 18 09:17:03] DEBUG[3245]: rtp.c:1246 ast_rtcp_read: Got RTCP report 
of 108 bytes
[b] -- Unable to find extension '' in context 'from-pstn' [/b]

Please let me to know if you need configuration files.

Thanks in advance.

Gustavo Duarte.

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[asterisk-users] problem with mssql and Asterisk 1.8.0 beta3

[unserossi]

Hi all,

I am testing with Asterisk 1.8.0 beta3 using realtime with a mssql server using 
freetds and unixodbc, which works with 1.6.1.20.

With the same config in 1.8 I get an error when trying to start asterisk which 
says:


[Aug 23 15:06:12] WARNING[7180]: loader.c:387 load_dynamic_module: Error 
loading module 'res_config_odbc.so': 
/usr/lib/asterisk/modules/res_config_odbc.so: undefined symbol: 
ast_odbc_clear_cache
[Aug 23 15:06:12] NOTICE[7180]: res_odbc.c:1471 odbc_obj_connect: Connecting 
mssql
asterisk: net.c:348: tds_select: Assertion `timeout_seconds >= 0' failed.

When I set 


[mssql]
enabled => no

in res_odbc.conf asterisk starts.

Have there been changes in res_odbc or is this a bug?

Thanks for advise.

Oliver
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Re: [asterisk-users] asterisk + openBTS

[equis software]

Do you know if OpenBTS support handoff?

Thanks


On Fri, Aug 20, 2010 at 12:32 PM, Steve Totaro <
stot...@totarotechnologies.com> wrote:

> On Fri, Aug 20, 2010 at 10:41 AM, Tim Panton  wrote:
> >
> >
> >
> > On 19 Aug 2010, at 20:59, Randy R wrote:
> >
> >> On Thu, Aug 19, 2010 at 12:37 PM, Alan Lord (News) <
> alansli...@gmail.com> wrote:
> >>> On 19/08/10 18:20, equis software wrote:
>  I want to know about asterisk and openBTS
> >>> This island runs it's GSM network on OpenBTS:
> http://www.niueisland.com/
> >>>
> >>> This was the place he presented about.
> >>>
> >>> Read the blog here: http://openbts.sourceforge.net/NiuePilot/
> >>
> >> and more about the installation here:
> >>
> >> http://vuc.me/2010/island-telephony-adventure/
> >>
> >
> >
> > I was part of the team that went to Niue to install OpenBTS,
> > I'm happy to answer questions if you have them,
> > although I'm not the radio guy - asterisk is more my thing :-)
> >
> > Tim.
> >
> > Tim Panton - Web/VoIP consultant and implementor
> > www.westhawk.co.uk
>
> In all reality, Asterisk could be substituted with any other platform.
>
> All the magic happens in the USRP, OpenBTS, and the cellular phones.
> Asterisk is merely handling the routing and voice, same as it ever
> was.  It is just the top of the stack.
>
> I have two USRPs and a handful of daughter boards, and yes I have two
> flex 800s that have been physically altered so they can also be flex
> 1800s with a simple command line.  These are the boards you want for
> GSM (Cellular).
>
> There is also a project to be able to listen into phone calls (thanks
> to the French making encryption so weak) besides a ton of other
> applications that can be dreamed up.
>
> You can do passive radar, track people that have cell phones powered
> on,  RFID (Free tolls anyone?), WiFi, heck, you can even kill people
> with certain types of pacemakers.
>
> While OpenBTS is cool and is on topic with Asterisk, read up on
> GNURadio and all the projects and applications you can come up with.
> It is really cool technology.
>
> Start here http://gnuradio.org/redmine/projects/show/gnuradio but you
> can easily find things like this
> http://tech.mit.edu/V128/N30/subway/Defcon_Presentation.pdf or come up
> with your own with a bit of imagination and skillz.
>
> Thanks,
> Steve Totaro
>
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Re: [asterisk-users] EMail on Missed Call

[Sherwood McGowan]

On Mon, Aug 23, 2010 at 5:35 AM, --[ UxBoD ]--  wrote:
> Hi,
>
> Running Asterisk 1.6.2.11 and wondering what would be the best way to send an 
> email when a missed call has occurred ? I believe you can modify [stdexten] 
> is this still the case in V1.6 ?
> --
> Thanks, Phil
>
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>

You'll just want to put a test for the value of ${DIALSTATUS} after
the Dial command (since it'll go there if the call times out). Check
for NOANSWER, and then act accordingly

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Re: [asterisk-users] How to prevent soft hangup from being necessary ?

[Sherwood McGowan]

On Mon, Aug 23, 2010 at 6:14 AM, Olivier  wrote:
> Hi,
>
> From time to time, I have to manually kill some "frozen" calls with soft
> hangup commands.
>
> (As far as I can tell, those freezes occurred after network breakdown (VPN
> or ethernet link between 2 LAN switches).
> So at this point, I would say I can't do much to keep those network
> breakdown to happen).
>
> So, which tools are available to automatically detect that SIP channels are
> up without but no RTP media is flowing in or from them ?
>
> Regards
>
>
>
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Odd, the RTPTimeout and RTPHoldTimeout settings in sip.conf should be
handling that for you

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[asterisk-users] How to prevent soft hangup from being necessary ?

[Olivier]

Hi,

>From time to time, I have to manually kill some "frozen" calls with soft
hangup commands.

(As far as I can tell, those freezes occurred after network breakdown (VPN
or ethernet link between 2 LAN switches).
So at this point, I would say I can't do much to keep those network
breakdown to happen).

So, which tools are available to automatically detect that SIP channels are
up without but no RTP media is flowing in or from them ?

Regards
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[asterisk-users] EMail on Missed Call

[--[ UxBoD ]--]

Hi,

Running Asterisk 1.6.2.11 and wondering what would be the best way to send an 
email when a missed call has occurred ? I believe you can modify [stdexten] is 
this still the case in V1.6 ?
-- 
Thanks, Phil

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[asterisk-users] How to do barging using asterisk server.

[Janu Mukherjee]

Hi All,

I have this requirement. I have an xlite client registered with asterisk
server. And from this when i dial an extension say xxx it invokes an AGI
script which gives me a series of instructions like "Welcome to this IVR
system. Press 1 to trade 2 to selland so on". I want to stop this and
press 1 or talk even before the prompt finishes. How to achieve this. I was
told that this is similar to barging in asterisk but i could not get info on
this and i see different barging examples like spying on to an extension.I
am not getting a clear picture on how to make the above thing work. Please
help me in this regard.


Thanks in Advance,
Jahnavi.
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