Re: [asterisk-users] remove

[Matt Fredrickson]

Check the footer at the bottom of this message for instructions on how
to unsubscribe :-)

Matthew Fredrickson

On Fri, Jun 1, 2018 at 12:11 PM, David Mutterer  wrote:
>
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[asterisk-users] Testing...

[Matt Fredrickson]

Test from non-digium email.

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[asterisk-users] Test from Digium address

[Matt Fredrickson]

Testing again.

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[asterisk-users] One more test

[Matt Fredrickson]

I need to send one more test.  Here it is!

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[asterisk-users] More testing

[Matt Fredrickson]

More testing.  Test test test. :-)

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[asterisk-users] Test

[Matt Fredrickson]

Testing again :-)

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[asterisk-users] AstriCon Approaching, Super Earlybird Pricing Expires In 3 Days

[Matt Fredrickson]

Hey All,

So one of the jobs that I get to do as head of the Asterisk project is
to help inform people about the yearly conference we have about
Asterisk named Astricon.

For those who are not familiar with it, AstriCon is a fantastic event
for anyone that is serious about Asterisk. This year, it's back in
Orlando, Florida, on October 9-11. Just as a heads up, the Super
Earlybird discount on a full AstriCon pass finishes on April 30, so
it's a good time to register to get the best deal:
https://www.asterisk.org/community/astricon-user-conference/register

AstriCon is a great chance to mix with all your favorite Asterisk
Community members and key members of the Asterisk development team
while you learn the latest developments, watch some crazy Dangerous
Demos and just have a whole bunch of fun! The Expo floor is always
worth a visit too, with many in the Asterisk Ecosystem present to show
your their offerings.

Also, by attending you help to financially support the Asterisk
project, as revenue from attendance is directly attributable to the
project and it makes it easier for me to justify Asterisk related
expenses such as hiring more Asterisk developers.

Hope to see many of you there!

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Re: [asterisk-users] Wanted: WebRTC tutorial

[Matt Fredrickson]

On Tue, Apr 24, 2018 at 10:54 AM, Bruce Ferrell  wrote:
> A while back (last year maybe?), there was a Digium blog post on setting up
> WebRTC.
>
> I was never able to get that working.
>
> I was working with Asterisk 15 on a RHEL derived distro and had no idea of
> where to go to shoot the failure.
>
> Has anyone got a tutorial with trouble shooting?

Great question!  I'm assuming you're talking about the SFU blog post -
http://blogs.asterisk.org/2017/09/20/asterisk-15-multi-stream-media-sfu/
?

I'd be curious as to what difficulties you ran into.  We actually need
to try to consolidate the information in that post with the webrtc
setup page on the wiki -
https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support

You might try those two pages if you haven't yet.  If you have
already, perhaps posting your specific challenges that you encountered
here might be helpful.

Thanks!

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[asterisk-users] Asterisk Community Services Outages

[Matt Fredrickson]

Dear Asterisk Community,

For the past 24 hours or so, Digium’s upstream provider has had a few
outages that have affected Asterisk community services, including
Asterisk.org, the mailing lists, and potentially other services.  We
apologize for any inconvenience that it has caused.  Hopefully things
are back up and running, but please let me know if you see anything
that’s still down.

Thanks so much for your patience.

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Re: [asterisk-users] Strange problem with PRI on 64-bit?

[Matt Fredrickson]

On Tue, Apr 3, 2018 at 4:38 PM, Tony Mountifield  wrote:
> In article 
> 

Re: [asterisk-users] Strange problem with PRI on 64-bit?

[Matt Fredrickson]

On Tue, Apr 3, 2018 at 5:44 AM, Tony Mountifield  wrote:
> I have some more investigation to do on this, but I wanted to see if anyone
> here had any insight into the issue I've run into.
>
> The hardware is a HP DL360 G6 with a TE420 gen 5 4-port T1 PRI card. It is one
> of several systems that have been running without issue since 2010/2011. They
> have all been running CentOS 4 32-bit with Zaptel 1.4.12.1 (with patch for gen
> 5 card), libpri 1.2.8 and asterisk 1.2.32.
>
> Having taken this particular system out of production, I updated it to CentOS
> 6.9 32-bit, with DAHDI 2.11.1, LibPRI 1.6.0 and Asterisk 11.25.3 (this version
> of Asterisk is required at the moment due to custom modifications).
> This appears to work fine.
>
> In order to reduce the number of different versions we support, I reinstalled
> the OS using the 64-bit version of CentOS 6.9 instead, and rebuilt, using
> the same versions as above.
>
> However, for reasons I don't understand, the 64-bit version was logging
> frequent PRI errors every few minutes:
>
> [Apr  1 03:40:52] VERBOSE[8989] chan_dahdi.c: PRI Span: 2 TEI=0 MDL-ERROR 
> (A): Got supervisory frame with F=1 in state 7(Multi-frame established)
> [Apr  1 03:40:58] VERBOSE[8988] chan_dahdi.c: PRI Span: 1 TEI=0 MDL-ERROR 
> (A): Got supervisory frame with F=1 in state 7(Multi-frame established)
> [Apr  1 03:44:06] VERBOSE[8990] chan_dahdi.c: PRI Span: 3 TEI=0 MDL-ERROR 
> (A): Got supervisory frame with F=1 in state 7(Multi-frame established)
> [Apr  1 03:46:38] VERBOSE[8990] chan_dahdi.c: PRI Span: 3 TEI=0 MDL-ERROR 
> (A): Got supervisory frame with F=1 in state 7(Multi-frame established)
> [Apr  1 03:47:20] VERBOSE[8988] chan_dahdi.c: PRI Span: 1 TEI=0 MDL-ERROR 
> (A): Got supervisory frame with F=1 in state 7(Multi-frame established)
> [Apr  1 03:47:24] VERBOSE[8989] chan_dahdi.c: PRI Span: 2 TEI=0 MDL-ERROR 
> (A): Got supervisory frame with F=1 in state 7(Multi-frame established)
>
> This left the PRIs in strange states - trying to make a call failed with 
> cause 101.
>
> So I re-installed the 32-bit OS again, and rebuilt, and the above MDL-ERRORs
> were no longer present, and the system operated normally again.
>
> So my question is: does anyone have any clues why there would be a difference
> in PRI behaviour between 32-bit and 64-bit builds? Has anyone else run into
> anything similar?


That does seem quite odd.  If I remember right, those messages would
come up if it looked like the other end hadn't received a message when
it thought it should have.  I can't think of anything that would
particularly impact 64 bit systems versus 32 bit systems in that
domain (ISDN real time message timing, etc).  Are you sure there's
nothing else different (kernel version or something else like that)?
Maybe also run a patlooptest on the spans in question to make sure
that they're running cleanly.

Matthew Fredrickson

>
> Cheers
> Tony
> --
> Tony Mountifield
> Work: t...@softins.co.uk - http://www.softins.co.uk
> Play: t...@mountifield.org - http://tony.mountifield.org
>
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>
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>
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Re: [asterisk-users] More testing - sorry guys

[Matt Fredrickson]

Thanks :-)

On Wed, Mar 28, 2018 at 3:52 PM, Markus Weiler
<markus_wei...@mailworks.org> wrote:
> I received it :-)
>
>
> Am 28.03.2018 um 22:44 schrieb Matt Fredrickson:
>>
>> Just a test.
>>
>
>
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> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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[asterisk-users] Sorry for interruption of service

[Matt Fredrickson]

Hey All,

Just as a public service announcement, we had a 12-16 hour window with
mailing list service interruption.

Last night we scheduled a time to update the mailing list server but
today found some problems impacting mailing service after the updates.
Due to this discovery, we quickly reverted the updates to restore
mailing list service.  Things should be back up and running now.

I apologize again for the inconvenience.  You might need to resend any
messages that were sent in the last half day or so.

-- 
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Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

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[asterisk-users] More testing - sorry guys

[Matt Fredrickson]

Just a test.

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Re: [asterisk-users] AMI potential memory leak

[Matt Fredrickson]

On Wed, Mar 21, 2018 at 4:03 PM, Dan Cropp  wrote:
> We are communicating with Asterisk via AMI.  Running Asterisk version
> 13.18.5 on an Ubuntu box.
>
>
>
> If you look at the event response, the Result field is filled with random
> characters.  I’m not sure what to do because that is a completely random
> result.  It makes no sense.
>
>
>
>
>
> We send the following command to asterisk via AMI
>
> Action: AGI
>
> ActionID: C44415
>
> Channel: SIP/192.168.40.105-1338
>
> CommandID: C44415
>
> Command: asyncagi break
>
>
>
> Asterisk processes it and the Asterisk log shows it sends the following for
> the AsyncAGIEXEC event…
>
> This is coming directly from the Asterisk log so it’s clearly Asterisk
> filling the Result with what appears to be random characters.
>
>
>
> [03/21 15:44:27.793] DEBUG[1213] manager.c: Examining AMI event:
>
> Event: AsyncAGIExec
>
> Privilege: agi,all
>
> Channel: SIP/192.168.40.105-1338
>
> ChannelState: 6
>
> ChannelStateDesc: Up
>
> CallerIDNum: 1234
>
> CallerIDName: 
>
> ConnectedLineNum: 
>
> ConnectedLineName: 
>
> Language: en
>
> AccountCode: 11
>
> Context: ABC
>
> Exten: 3002
>
> Priority: 8
>
> Uniqueid: 1521665055.4920
>
> Linkedid: 1521665055.4920
>
> Result:
> %F7%EF%F0%F4z%7C%EE%EF%F6%FCkWDDLO%5Cm%EF%7B__%60h%FE%E0%D4%D5%D9%DA%DD%E2%E8%E1%DC%DE%E0%F3%EC%EF%7C%ED%ED%EC%FAx~oov%F9%F2%F6%EB%ED%FA%FD%F9%FEvjXGFMKWg%FF%7Cdidew%E8%D9%D9%D8%D6%DA%DF%E3%E3%E2%E5%E9%EA%E1%E6%EF%E9%F8%F8%F6%FD%F8%7C~%7D%FC%F9%F7%ED%F4%F6%F9oh%60JFMKP%5Do%FFgtjch%7B%E3%DF%DB%D6%D8%DB%E6%EC%E7%E6%E5%DF%DD%E0%EA%F6%ED%EC%F7%E8%E4%EB%F2%F8%F9%7Cu%FC%F8%FAwhXIIIHQ%5Dksw%7Fgfl%7F%EA%E2%DA%D8%DC%DE%DE%E2%E4%DF%E0%E3%E9%E4%E8%EC%E6%E3%E1%E4%E0%E3%EE%F4%FAxy%FCj%5BLIJFLU%5Bdm%FFljws%FD%ED%E2%DC%DD%DA%D9%DC%DE%DD%E1%E7%E9%E8%E6%EC%E5%E1%E5%E0%DF%E2%E9%ED%F1xyo%5EOKMHJSU%5Bfvml%7F%7B%7B%F0%EB%EA%E2%DD%DD%DB%DB%DC%DD%E1%E0%E1%E3%E3%DF%DF%DF%DE%E0%E5%E9%F6%7CmcTKMJIOTYboon%7Ctt%FA%F7%F3%E9%DE%E1%DC%D9%DB%DE%E1%DE%E6%E7%DF%E1%DF%DD%DC%DE%E1%E1%EE%FAx%5DRMLHJOPWblo%7C%FA%7B~%F9%FC%F7%E9%E1%E1%DC%D9%DC%DC%DB%DF%E3%E0%DF%E0%DE%DC%DF%E1%E1%EE%F9r%5DQMKHKOQXdmt%F8%F6%FC%FB%FFy%F5%EA%E5%DF%DA%D9%DA%D8%DB%DE%DD%DE%DF%DF%DF%E3%EA%EB%ECrkbSPOLLOST_jk%F9%F3%FB%F9%FF%FE%F9%EF%E9%E2%DF%DA%DA%DA%D9%D9%DB%DC%DC%DF%E3%E4%EB%F4%FFmdYRPLKNOTZdl%7D%EE%F6%EE%EB%EF%EE%EA%E3%E7%DE%DB%DD%D8%D9%D9%D9%D9%DC%DF%E1%EE%FAxf_XROMMNQUYbgx%FE%FB%EC%EE%ED%E4%E4%E0%DD%DD%DB%DA%D9%DA%D9%D9%DD%DD%E2%EC%F9tg%5EZSPONORUZ%5Efmw%FC%F3%ED%E7%E2%E1%DC%DB%DB%D8%D9%D9%D8%DA%DB%DD%E2%E9%F6uh%60%5BVSQOOQSX%5B_jq%F9%ED%E8%E1%DF%DC%DA%DA%DA%DA%D9%DB%DC%DC%E3%E6%EE%FEtha%5EYWVTTUWY%5Eckx%FA%EC%E7%E2%DF%DD%DC%DD%DC%DC%DF%DE%E2%E6%E7%F1%F8%7Boid_%5C%5BZY%5BZ%5B%5E%60flu%FF%F4%EC%E8%E3%E1%E1%DF%DF%E1%E0%E4%E7%E9%EE%F5%FByqmhfca%60%5E__%60bfhlqz%FE%F7%EF%EE%EB%E8%EA%E9%E9%EB%EA%ED%EF%F3%F7%FC%7Cysookkkiijiklmqry%7B%7F%FD%F9%F7%F5%F4%F5%F5%F6%F7%F8%F8%FA%FD%FF%7Dxxtrsooonoposttw%7By%7C%7D~%7F%FE%FF%FE%FD%7F%FE%7F%FE%7F%FF%7F%FE%7F%7D~%7B%7Bzvwwvvwvuwvxwyxxyy%7B%7C%7C%7D%7D%7F%7F%7F%FD%7F%FF~%7D%7D%7B%7Bzywyvwvvvwxxy%7By~%7C%7D%7F%7D~~%7C~%FF%7D%7C%FF~%7F%7F~~%7Dz%7C%7Czxywwxxwxxwy%7Bx%7Bz%7C%7D%7C%FF~%7F%7F~%FF~%7F%7D~%7C%7D%7D~%7D%7D%7C%7B%7B%7B%7Bxzyzyzzz%7Cz%7D~%7C%7C~%7C~%7F~~%FF%7D%7F%7F%7D%7F~%7C%7C%7B%7By%7Byy%7B%7B%7B%7B~%7D%7D~~%7F%7F%FE%7F%FF
>
> CommandId: C44415

That seems kind of yucky.  How reproducible is it?


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Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

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[asterisk-users] Test

[Matt Fredrickson]

Testing, 1, 2, 3.

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[asterisk-users] Test

[Matt Fredrickson]

This is a test.

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[asterisk-users] FOSDEM

[Matt Fredrickson]

Hey All,

For any interested in potentially meeting up to talk about Asterisk
and other fun things, Ben Ford from Digium's Asterisk development team
and myself will be in Brussels for FOSDEM Feb 3-4.

I hope to see many of you there!

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

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[asterisk-users] Recent Video Interview and Upcoming Webinar about Asterisk 15

[Matt Fredrickson]

Hey Everybody,

As a project, we would like to do a better job of getting additional
information about new developments in Asterisk to the community.  I
think this is something I have struggled with in the past (to some
extent) and would like to improve upon in the future.

For anybody interested, there's a fun 15 minute video interview
featuring Matt Jordan and myself discussing Asterisk 15 and what's new
with it.  It can be found at:

https://www.youtube.com/watch?v=0XSDOPftNpM=youtu.be

For those who enjoyed the video or would like to get a bit of a deeper
dive into Asterisk 15, the annual Asterisk 15: Under the Hood
presentation will be this Tuesday, December 5th.  It covers some of
the newer features in Asterisk 15 at a lower level and is intended for
a fairly technical audience.  It's usually very well attended.
Details and registration information can be found at:

http://bit.ly/2BnrpmF

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Digium, Inc. | Asterisk Project Lead
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

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[asterisk-users] Asterisk EOL Announcement

[Matt Fredrickson]

Dearly Beloved,

We have gathered here today to mourn the passing of a deeply regarded
branch of Asterisk - Asterisk 11.  As of today, it has officially
reached its end of life.  It was a good branch, having served 5 years
faithfully in the service of its users.  As far as history goes,
11.0.0 was born on November 28th 2012.  It had 1458 commits in its 5
year life, and some will try to use it even after its useful end of
life.  Now, mostly due to the fact that it is no longer with us,
Asterisk 11 will become one of the “great” releases of Asterisk,
joining the ranks of all the other “great” branches such as 1.0, 1.2,
1.4, 1.6, 1.8, and 10.  Please join with me now for a moment of
silence for it, as it passes to the great beyond.

In all seriousness, if you didn’t get the humor in the above message,
today is the day that Asterisk 11 officially goes end of life.  For
the last year, Asterisk 11 has been in security fix only mode, meaning
it stopped receive regular bug fixes a year ago, and has only received
security related fixes.  Today that all ends and not even security
fixes will going into that branch.  If you haven’t gotten off of it
yet, there is no better time than the present.

If you’re curious about the dates and times associated with life
cycles transitions on Asterisk branches, you can read more at
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

Last of all, thanks to all of you who contribute to the Asterisk
project, whether it be bug reports, bug fixes, new feature
development, or helping other users by answering questions on the
mailing lists, forums, and other venues.  At the end of the day, it’s
the quality of the user and development community that make Asterisk
such a great project.

Best wishes.

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Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

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[asterisk-users] Asterisk 14 Security Fix Only Mode

[Matt Fredrickson]

Hey all,

For those who may not be aware Asterisk 14 transitioned from bug fix mode
to security-fix-only mode a few weeks ago (Sept 26th). For those of you
that are still on this release, it's a good time to consider building an
upgrade plan for moving to 15.x.x.  I sincerely apologize for the late
notification.  Somehow, this was overlooked in the push to get 15.0.0
released.

You can read more about this process and the relevant dates on the wiki at
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

Best wishes, and sorry again about the confusion on this.

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[asterisk-users] Asterisk 15 Beta Released

[Matt Fredrickson]

It is with great pleasure I wish to inform you of the first beta
release of the new Asterisk 15 branch. It's a very exciting time to be
a user of Asterisk! Asterisk 15 is arguably the biggest release of
Asterisk that has happened in the last 10 or so years. There has been
a lot of work done in the Asterisk core to better support newer
multi-stream video and WebRTC related technologies.  For those who are
interested, much of this will be covered in blog posts at
http://blogs.asterisk.org/ over the next month or two.

Typically, when a new major branch of Asterisk is created (13, 14,
15...), there are a few months of testing on the new branch that
occurs prior to release in order to find regressions and other issues
that may cause a first official release from the branch to be dead on
arrival for a significant number of users. With today's release of
15.0.0-beta1, this process has begun. Please feel free to start
testing this version of Asterisk in as many adverse environments as
possible. Any bugs should be reported on the Asterisk issue tracker at
https://issues.asterisk.org/

As a side note, due to many of the core changes in the 15 branch that
have been made since Asterisk 14 was released, it has been decided
that Asterisk 15 will not be an LTS release. For those of you who are
not familiar with the differences between LTS versus standard
releases, you can find more information here [1].

Thanks to all the many Asterisk community members for providing so
much help and support to make Asterisk the great open source project
that it is.

P.S. Binary codecs and other modules distributed by Digium are not
immediately available for 15.0.0-beta1, but should be shortly.

Best wishes to all, and happy testing!

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

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Re: [asterisk-users] Moving call DAHDI from channel X to Y.

[Matt Fredrickson]

On Sun, Jul 30, 2017 at 8:34 PM, Daniel Harper  wrote:
> I am seeing the in the asterisk logs that channels (PRI ISDN)  are
> being moved ..
>
> [Jul 29 16:31:48] VERBOSE[16125] logger.c: -- Moving call
> (DAHDI/57-1) from channel 57 to 58.
>
> I then see the moved channels with a "0:" in front of it.
>
> [Jul 29 16:31:48] VERBOSE[26691] logger.c: -- Hungup 'DAHDI/0:58-1'
>
> Any ideas why this could be happening?
>
> I believe these messages are coming from chan_dahdi.c and the
> "pri_fixup_principle" function.
>

This is a normal thing that can happen when doing B-channel selection
as a part of the call setup process.  If one side disagrees with the
initial choice for B-channel, the other side can request that it be
moved in the reply.  It's more likely to happen on busy PRIs with
bidirectional (ingress and egress) traffic.

Hope that helps!

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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

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[asterisk-users] Asterisk 11 EOL 6 Month Notice

[Matt Fredrickson]

Greetings,

This is your friendly 6 month warning that Asterisk 11 will be
reaching an official end of life state on October 25, 2017.  As many
of you know, for the past 6 months Asterisk 11 has been in security
fix only mode.  This means it currently does not receive bug fixes,
but it does receive applicable security fixes and will continue to do
so for the next 6 months.

If you are still running Asterisk 11, this is a great time to start
preparing for a move to the latest LTS branch, 13, as a forward
migration path.  You can read more about Asterisk's branch and
versioning policy as well as relevant version sunset information on
the Asterisk wiki [1].

Best wishes to you all, and happy upgrading. :-)

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

-- 
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Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

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Re: [asterisk-users] Asterisk 13.15.0, webrtc, Google chrome 58 beta and "bad media description"

[Matt Fredrickson]

On Sat, Apr 8, 2017 at 7:23 AM, Dan Jenkins  wrote:

>
> On Fri, Apr 7, 2017 at 9:44 PM, Teijo  wrote:
>
>> Hello,
>>
>> I've been using webrtc (Jscommunicator) with Asterisk occasionally. Only
>> problem until now which remained was that if dtls_rekey was set to the
>> value other than 0, call hanged up when using chrome after the time where
>> dtls_rekey was set.
>>
>> I suppose that "bad media description" shown in Chrome's window which
>> causes call to fail, has appeared with Chromes newer versions (currently 58
>> beta installed) or with Asterisk 13.15.0. Audio codec I'm using is Opus.
>>
>> Has somebody else encountered this problem, or more better resolved it?
>>
>> Best regards,
>>
>> Teijo
>>
>> --
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>>
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>> https://community.asterisk.org/
>>
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>>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
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>>
>
>
> Hi Teijo
>
> Take a read of https://nimblea.pe/monkey-business/2017/01/19/webrtc-
> asterisk-and-chrome-57/ :)
>

13.15.0 should address rtcp-mux issues.

If there are still issues outstanding, it might be worth reporting a bug on
issues.asterisk.org.

Best wishes :-)

-- 
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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Re: [asterisk-users] 100% CPU after upgrade.

[Matt Fredrickson]

On Mon, Apr 3, 2017 at 4:45 PM, Mike Diehl  wrote:
> Those are all rational questions, so here we go:
>
> We upgraded from 11.x, though the system was a backup server, so it was never
> actually used.
>
> The system is a 2.4Gh quad-core Xenon with 4G of RAM, so it should have plenty
> of power for what I'm asking it to do.  The system is configured via RT using
> a local Mysql database.
>
> We only use the native SIP channel driver at this time.
>
> I honestly don't see any reason for this server to eat 100% of it's cpu, and
> am hesitant to roll it out to production until I understand why it is.

I don't either.  Is there any Asterisk logging that indicates
something that might be going on?  If you can't see anything, try
increasing the core debug level and core verbose level (core set
verbose 10, core set debug 10) at the Asterisk CLI and see if you get
anything more out of logging to see what's going on.

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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

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Re: [asterisk-users] 100% CPU after upgrade.

[Matt Fredrickson]

One thing you didn't mention was what version you previously upgraded
from...  Also, more information about the system in general would
help.  (Endpoints, is it realtime or flat file configured, if
realtime, what type of database, what channel drivers (SIP or PJSIP,
and others).

Matthew Fredrickson

On Fri, Mar 31, 2017 at 12:08 PM, Mike Diehl  wrote:
> Hi all,
>
> I've upgraded to Asterisk 13.14.0 and now I'm seeing that Asterisk is using 
> 100% CPU.
>
> I have one AMI agent connected that is acting rationally.  I've got a hand 
> full of SIP (RT) registrations.  There is no other call activity.
>
> I've tried to unload various modules; nothing resolved the issue.
>
> Any suggestions?
>
> --
> Mike Diehl
>
>
>
>
> --
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> Check out the new Asterisk community forum at: https://community.asterisk.org/
>
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Re: [asterisk-users] Bounty on Google Voice

[Matt Fredrickson]

On Wed, Mar 29, 2017 at 11:45 AM, Saint Michael  wrote:
> The channel motif and res_xmpp do not work. But there is one company that
> does make it work and charges $US 6 for a lifetime connection to your own
> free Google Voice number, from SIP. I wonder if anybody would be able to fix
> Asterisks libraries so people of low income would not have to pay a third
> party for this basic translation service.

Hey,

Sorry to hear about your difficulty with this code.  At this time, I'm
not aware of an active maintainer/owner of those modules.  As I'm sure
you're already aware, Asterisk is an open source project worked on by
people with different interests and motivations.  Some work on it
purely for the fun, others work on it because they have a business
interest of some sort.  Bug bounties are a great way to get the
attention of people in both categories, and are usually posted to the
asterisk-dev mailing list.  You can learn more about the guidelines
prior to posting at this wiki page:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Bug+Bounties

Hope that helps a bit, and best wishes in getting your bug fixed.

-- 
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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

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Re: [asterisk-users] UniMRCP and Asterisk 14

[Matt Fredrickson]

On Thu, Mar 23, 2017 at 8:27 PM, Richard Kenner  wrote:
> When I look at the lastest UniMRCP manual, they only mention as high as
> Asterisk 13.  Does anybody know if I need to do anything to allow it
> to work on Asterisk 14 and, if so, what that is?

I can't speak for the MRCP guys, but from a difference perspective,
swapping MRCP from Asterisk 13 to Asterisk 14 shouldn't be too
difficult.  Most of the changes between the two shouldn't affect most
people's use cases, including projects such as MRCP.  I'd definitely
check with their discussion forums though, since it seems that they
don't monitor the asterisk-users mailing lists.

-- 
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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

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Re: [asterisk-users] Manager events showing in CLI

[Matt Fredrickson]

Try doing a `core set debug 0` at the Asterisk CLI.  That should
disable it.  Or remove debug from your console output in logger.conf.

Best wishes,
Matthew Fredrickson

On Sun, Mar 26, 2017 at 5:35 PM, Telium Technical Support
 wrote:
> I did that too – no debug related settings in there!  That’s why I’m
> stumped.
>
>
>
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Victor
> Villarreal
> Sent: Sunday, March 26, 2017 2:06 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Subject: Re: [asterisk-users] Manager events showing in CLI
>
>
>
> Ok,
>
>
>
> Please, check your manager.conf and logger.conf for any clue about debugging
> options, into the Asterisk configuration directory.
>
>
>
> El 26 mar. 2017 14:52, "Telium Technical Support" 
> escribió:
>
> I tried that but it had no effect.  Still see things like:
>
>
>
> [2017-03-26 13:49:39] DEBUG[2088]: manager.c:5693 match_filter: Examining
> AMI event:
>
> Event: SuccessfulAuth
>
> Privilege: security,all
>
> EventTV: 2017-03-26T13:49:39.407-0400
>
> Severity: Informational
>
> Service: SIP
>
> EventVersion: 1
>
> AccountID: 221essionID: 0x7fa0cc005cc8
>
> LocalAddress: IPV4/UDP/192.168.67.4/5060
>
> RemoteAddress: IPV4/UDP/192.168.67.26/5060
>
> UsingPassword: 1
>
>
>
>
>
> [2017-03-26 13:49:39] DEBUG[1882]: chan_sip.c:9196 __find_call: = Looking
> for  Call ID: 280f68000ff289291b366a1242530ce8@192.168.67.4:5060 (Checking
> To) --From tag as494dfc4b --To-tag 4155795028
>
> [2017-03-26 13:49:39] DEBUG[1882]: chan_sip.c:4419 __sip_ack: Stopping
> retransmission on '280f68000ff289291b366a1242530ce8@192.168.67.4:5060' of
> Request 102: Match Found
>
> [2017-03-26 13:49:39] DEBUG[1882]: chan_sip.c:6725 sip_destroy: Destroying
> SIP dialog 280f68000ff289291b366a1242530ce8@192.168.67.4:5060
>
> [2017-03-26 13:49:39] DEBUG[1882]: chan_sip.c:4275 __sip_autodestruct: Auto
> destroying SIP dialog 'cbf5d92f6844702b'
>
> [2017-03-26 13:49:39] DEBUG[1882]: chan_sip.c:6725 sip_destroy: Destroying
> SIP dialog cbf5d92f6844702b
>
> [2017-03-26 13:49:39] DEBUG[2088]: manager.c:6138 process_message: Running
> action 'Command'
>
> [2017-03-26 13:49:39] DEBUG[1951]: manager.c:6138 process_message: Running
> action 'Command'
>
>
>
> cli> manager set debug off
>
>
>
>
> --
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> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
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>
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> Check out the new Asterisk community forum at:
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>
> New to Asterisk? Start here:
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Re: [asterisk-users] Issue with handling of 480 DND

[Matt Fredrickson]

Response inline.

On Fri, Jan 6, 2017 at 12:47 PM, Markus Weiler 
wrote:

> Nobody any idea?
>
> It would be really helpful,
>
> Markus
>
>
>
>
> Am 06.01.2017 um 12:07 schrieb Markus Weiler:
>
>> Hi List,
>>
>> we're calling a sip phone from our Asterisk Server, and try to add logic
>> depending on the dialstatus
>>
>> Stripped down example;
>>
>> exten = 494X,n,Dial(SIP/4120089,15,w)
>> exten = 494X,n,Goto(98-${DIALSTATUS},1)
>> exten = 494X,n,Hangup()
>>
>>
>> .
>> exten = 98-BUSY,1,NoOp(Busy)
>> exten = 98-BUSY,n,ExecIf($["${Voicemail}" =
>> "1"]?Playback(/home/4120/mitarbeiter/ab))
>> 
>> exten = 98-NOANSWER,1,NoOp(noanswer)
>> exten = 98-NOANSWER,n,ExecIf($["${Voicemail}" =
>> "1"]?Playback(/home/4120/mitarbeiter/ab))
>> 
>>
>> Íf the phone call times out, the call is sent to 98-NOANSWER and then
>> answered as expected.
>> If the User presses DND on his phone the call is sent to 98-BUSY which
>> is identical but then the call is hung up. This behaviour is
>> unexpected/unwanted.
>>
>> We tried to figure out what the difference is and think it's how
>> Asterisk handles the "480 Do Not Disturb" from the phone
>> (xxx.xxx.xxx.xxx).
>> It is passed to our main incoming server (zzz.zzz.zzz.zzz) as "181 call
>> is being forwarded".
>>
>> Is this a bug or a feature? :-) How could we handle this correctly?
>>
>
I believe that this is a consequence of the fact that when chan_sip
receives a "480 Do Not Disturb" it also queues a redirecting_update frame
on the channel. My guess is that the redirecting update is probably
triggering the "181 Call is being forward as well".  If you add the 'I'
flag to your dial, I believe it should suppress bridging of the redirection
information and I would think that would also cause the 181 not to be
sent.  If it's not that flag, I'd check the documentation for app_dial, as
I'm pretty sure there's a flag that should suppress that redirection from
from being bridged to the calling channel.

Hope that helps.

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[asterisk-users] Asterisk 14 web broadcast

[Matt Fredrickson]

Hey All,

Slight interlude from your regularly scheduled programming.

For any interested, I will be giving a web broadcast today about
Asterisk 14 and what's new with Asterisk since the 13 release.  For
those of you that aren't aware, I'm responsible for day to day
management of the Asterisk project now that Matt Jordan has been moved
into the CTO role at Digium.

You can get info about it at:

http://bit.ly/2gDkFrh

It will be live today at 8AM, 2PM, and 9PM CDT.

Hope to see many of you there!

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Re: [asterisk-users] Force hangup not working on stuck channel

[Matt Fredrickson]

Also, it looks like in
https://issues.asterisk.org/jira/browse/ASTERISK-21762 there might be
a workaround (see the last comment at the bottom).

Matthew Fredrickson

On Fri, Nov 4, 2016 at 2:01 PM, Matt Fredrickson <cres...@digium.com> wrote:
> On Thu, Nov 3, 2016 at 11:16 AM, Carlos Chavez <cur...@telecomabmex.com> 
> wrote:
>> I am unable to force a hangup on a channel that has been stuck for over two
>> days:
>>
>> IAX2/from-CD-11006   oficina  27701 Up  Dial
>> IAX2/to-CD/2883   3467130007  46:24:59 Sotelo  Sotelo
>> IAX2/to-CD-20713
>>
>> I have tried "hangup request IAX2/from-CD-11006" several times but no joy.
>> I also see the following in the CLI:
>>
>> [Nov  3 10:05:54] WARNING[2879]: chan_iax2.c:4936 handle_call_token: Too
>> much delay in IAX2 calltoken timestamp from address X.X.X.X
>>
>> This is an IAX2 trunk between two Asterisk 1.8 servers (I know it is old but
>> new client so haven't had time yet to upgrade to 13).  Because this channels
>> is stuck
>>  all other calls between servers are not working.  The only way I have found
>> to resolve the problem is to stop and restart Asterisk.  This is obviously a
>> great inconvinience so is there a way for force iax to unload even if there
>> are channels in use?  Or any other way to kill these stubborn channels?
>
> If doing a soft hangup on them doesn't work, the only other way I know
> to do it is to restart Asterisk.  Sorry about the bad news :-(
>
> There's a part of me that's curious as to why the channel is stuck,
> but there's another part of me that says "1.8... run away quickly".
>
> You could try to replicate it with a modern branch (i.e. 13/14) and
> see if it still exists.  At the very least that'd leave you the option
> of posting a bug on the issue tracker about it.  Also, a lot of bugs
> have been fixed since 1.8, so it's quite possible that this issue is
> resolved as well.
>
> --
> Matthew Fredrickson
> Digium, Inc. | Engineering Manager
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA



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Re: [asterisk-users] Force hangup not working on stuck channel

[Matt Fredrickson]

On Thu, Nov 3, 2016 at 11:16 AM, Carlos Chavez  wrote:
> I am unable to force a hangup on a channel that has been stuck for over two
> days:
>
> IAX2/from-CD-11006   oficina  27701 Up  Dial
> IAX2/to-CD/2883   3467130007  46:24:59 Sotelo  Sotelo
> IAX2/to-CD-20713
>
> I have tried "hangup request IAX2/from-CD-11006" several times but no joy.
> I also see the following in the CLI:
>
> [Nov  3 10:05:54] WARNING[2879]: chan_iax2.c:4936 handle_call_token: Too
> much delay in IAX2 calltoken timestamp from address X.X.X.X
>
> This is an IAX2 trunk between two Asterisk 1.8 servers (I know it is old but
> new client so haven't had time yet to upgrade to 13).  Because this channels
> is stuck
>  all other calls between servers are not working.  The only way I have found
> to resolve the problem is to stop and restart Asterisk.  This is obviously a
> great inconvinience so is there a way for force iax to unload even if there
> are channels in use?  Or any other way to kill these stubborn channels?

If doing a soft hangup on them doesn't work, the only other way I know
to do it is to restart Asterisk.  Sorry about the bad news :-(

There's a part of me that's curious as to why the channel is stuck,
but there's another part of me that says "1.8... run away quickly".

You could try to replicate it with a modern branch (i.e. 13/14) and
see if it still exists.  At the very least that'd leave you the option
of posting a bug on the issue tracker about it.  Also, a lot of bugs
have been fixed since 1.8, so it's quite possible that this issue is
resolved as well.

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Re: [asterisk-users] What's the smallest, lightest Asterisk you can build? Does size even matter?

[Matt Fredrickson]

On Tue, Nov 1, 2016 at 6:00 PM, Jonathan H  wrote:
> All I need is PJSIP, ulaw, alaw, wav, astdb and all the dialplan functions.
>
> I don't need any other DB layer, I have no hardware, and I was wondering
> what the smallest build possible was.
>
> I experimented, but everything relied on other things. And then I
> wondered... is there actually any point? Is there anything to be gained?
>
> Will it matter more when there are lots of concurrent calls, or should I
> just not worry, leave all the options in makemenu, make it easy on myself
> and build the full thing each time?

For most modules there isn't a big point to disabling them as there
isn't an lot of ongoing CPU activity for a module not being used.
That being said, there are some things you can disable that can
improve your performance, but they're going to be application
dependent.  I believe that disabling CDRs, for example, can make a big
difference on heavily loaded systems and some people don't use them.
CDRs take some amount of work at the end to stitch together the notion
of a call from the various call related events that occur.  HEP also
is a big offender.  CELs and AMI can also make some difference when
disabled, but nothing on the order of CDRs and HEP.

Hope that helps!

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Re: [asterisk-users] RS485 Audio device

[Matt Fredrickson]

On Fri, Oct 28, 2016 at 7:09 PM, Jerry Geis  wrote:
> Hi All,
>
> Is there any devices or pair of devices that do audio over RS485
> and then convert to SIP for us in asterisk?
> Of course a speaker and push button at the other end.
>
> Is there anything like that out there?

Ok, I'll bite.  How does one do audio over RS485?

I've never worked with RS485, but from some brief googling it looks
like it's a fancy version of RS232.  I'm not sure where you'd get
(analog) audio from on RS232.

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[asterisk-users] Asterisk 11 - Security Fix Only Notice

[Matt Fredrickson]

Hey All,

This is a friendly notice that as of today Asterisk 11 has entered
security fix only mode.  From this point onward additional releases of
Asterisk 11 will no longer be made unless there is a security fix
being applied to the branch.  Users of Asterisk 11 are encouraged to
move to one of the newer major versions, Asterisk 13 (LTS) or Asterisk
14, as soon as possible.

For more information on Asterisk versions and their supported lifetimes,
please see the following wiki page:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

Thank you for your continued support of Asterisk!

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Re: [asterisk-users] Got bitten by the 255 char variable limit - how best to work around it?

[Matt Fredrickson]

On Sat, Oct 22, 2016 at 8:05 PM, Jonathan H  wrote:
> I loop through a list in Asterisk which is generated by a Python AGI
> and I've just been bitten by a variable limit I didn't realise existed
> before.
>
> The only way I can think of working around this is to get Python to
> write the list to file, then do a FILE_COUNT_LINE to get the number of
> items (needed first) and then iterate through them by doing FILE to
> read one line at a time.
>
> Would rather stay away from polluting ASTDB, and don't want to install
> MySQL as I don't need it for anything else.
>
> Ideally, though, is there a workaround for the variable limit? Thanks!

I was curious about this and started looking through the code to
remember how this all worked.  I don't see anything specifically that
would truncate setting a channel variable internally, but perhaps it's
a limitation due to how AGI is processed or buffers used for
parsing/interpreting there.  There are a number of fixed sized buffers
used in the AGI parsing code.  I'm curious, what version of Asterisk
are you using?

I'm not aware of any workarounds other than those you mentioned.  If
you wanted to get really into it, you could poke around in
res/res_agi.c and see if you can figure out which buffer is limiting.
YMMV there though.

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Re: [asterisk-users] Audio when card is in condition yellow

[Matt Fredrickson]

Usually a card is supposed to send yellow alarm (so it's transmitted)
when it detects LOS (loss of signal) on the T1, or essentially a red
alarm condition is detected.  So if yellow is being sent, it means
that at least one end is not able to sync up the line, which means
you'll have junk/garbled audio potentially.

Hope that helps,
Matthew Fredrickson

On Fri, Oct 21, 2016 at 1:51 PM, Jerry Geis  wrote:
> With asterisk 11.23.0 I have about 121 SIP devices connected.
> normally things sound fine when speaking a message on these devices (using
> conference bridge).
>
> Currently the TE122 card is in condition yellow normally it is not.
> I sent a message to all devices and it was garbled on many of them.
> Is this a result of card in COND YELLOW?
>
> If so what can I do about it while the card is in cond yellow.
> The other end is currently down, no ETA on coming back.
>
> Thanks,
>
> Jerry
>
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Re: [asterisk-users] queue_log/cel sqlite

[Matt Fredrickson]

On Thu, Oct 20, 2016 at 9:45 AM, marek cervenka <cerva...@gmail.com> wrote:
>
> Dne 20/10/2016 v 16:32 Matt Fredrickson napsal(a):
>>
>> On Thu, Oct 20, 2016 at 4:50 AM, marek cervenka <cerva...@gmail.com>
>> wrote:
>>>
>>> i tested this
>>>
>>> # cat /etc/asterisk/extconfig.conf
>>> [settings]
>>> queue_log => sqlite3,cdrDb
>>>
>>> # cat /etc/asterisk/res_config_sqlite3.conf
>>> [cdrDb]
>>> dbfile = /var/lib/asterisk/realtime.sqlite3
>>>
>>> sqlite3 /var/lib/asterisk/realtime.sqlite3
>>>
>>> CREATE TABLE "queue_log" ("time" TEXT, "data1" TEXT, "data2" TEXT,
>>> "data3"
>>> TEXT, "data4" TEXT, "data5" TEXT, "event" TEXT, "agent" TEXT, "queuename"
>>> TEXT, "callid" TEXT);
>>>
>>> and it works
>>>
>>> sqlite> select * from queue_log;
>>> 2016-10-20 11:40:36.628804||QUEUESTART|NONE|NONE|NONE
>>> 2016-10-20 11:40:36.690313||CONFIGRELOAD|NONE|NONE|NONE
>>>
>>> column types needs modification to something more appropriate
>>>
>>> can someone with confluence access ad info to
>>>
>>>
>>> https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration ?
>>
>> Which info are you referring to?  The table schema?
>>
>
> ideally add "correct" sql schema for sqlite to asterisk repo and link it to
>
> https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
>
>
> it was hard for me to find if queue_log can be logged with sqlite. imho it
> will be usefull document the example configuration for others
> but i'm not sure where is the best place
> maybe https://wiki.asterisk.org/wiki/display/AST/Queue+Logs ?

My suggestion would be to add a comment to the page with your proposed
changes.  That would be the best place to start, for the next time
someone works on that page.

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Re: [asterisk-users] queue_log/cel sqlite

[Matt Fredrickson]

On Thu, Oct 20, 2016 at 4:50 AM, marek cervenka  wrote:
> i tested this
>
> # cat /etc/asterisk/extconfig.conf
> [settings]
> queue_log => sqlite3,cdrDb
>
> # cat /etc/asterisk/res_config_sqlite3.conf
> [cdrDb]
> dbfile = /var/lib/asterisk/realtime.sqlite3
>
> sqlite3 /var/lib/asterisk/realtime.sqlite3
>
> CREATE TABLE "queue_log" ("time" TEXT, "data1" TEXT, "data2" TEXT, "data3"
> TEXT, "data4" TEXT, "data5" TEXT, "event" TEXT, "agent" TEXT, "queuename"
> TEXT, "callid" TEXT);
>
> and it works
>
> sqlite> select * from queue_log;
> 2016-10-20 11:40:36.628804||QUEUESTART|NONE|NONE|NONE
> 2016-10-20 11:40:36.690313||CONFIGRELOAD|NONE|NONE|NONE
>
> column types needs modification to something more appropriate
>
> can someone with confluence access ad info to
>
> https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration ?

Which info are you referring to?  The table schema?

Matthew Fredrickson


>
>
> is there somebody using it in production?
> thanks
>
> Dne 20/10/2016 v 10:16 marek cervenka napsal(a):
>
>> hi,
>>
>> is it possible log cel/queue_log to sqlite?
>>
>> via odbc?
>>
>> any experience?
>>
>> marek
>>
>>
>>
>
>
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[asterisk-users] Asterisk 11 - Security Fix Mode

[Matt Fredrickson]

Hello Everyone,

As many of you are already aware, we are rapidly approaching the time
when the Asterisk 11 branch will go into what is known as security fix
only mode.  Up to this point, bug fixes have been included and merged
into the 11 branch.  For Asterisk 11, this new phase of life shall
begin October 25th of this year.

This means that from a development perspective, the Asterisk
development team will not be putting effort into bug fixes for the 11
branch after the 25th of October.  Security related patches will be
merged for another year before completely putting the branch to rest.
During the course of that year, releases will be made as needed and as
security related patches are merged.

For any questions, you can either reply to this message or look at the
Asterisk Versioning Policy wiki page [1] as it explains most of this
process in greater detail.

Thanks so much again for all of your support and effort.  Asterisk
would not be what it is if it were not for all the great contributors
that are and have been involved.

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

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Re: [asterisk-users] Farewell

[Matt Fredrickson]

Best of wishes to you in your retirement!  It's been a great 10 years, and
I'm personally looking forward to the great things coming in the next 10.

Matthew Fredrickson

On Wed, Aug 17, 2016 at 8:37 AM, Vincent Medina  wrote:

> I just wanted to wish all of you good luck I'm officially retired and will
> be removing my name from the list. I can attest that this list has been a
> great help throughout my career. I have deployed probably over 100
> installations over a 10-year period.
>
> Any of you newcomers this list the most valuable tool you can have.
>
>
>
> Sincerely,
>
> Vincent Medina
> Information Systems Director
> APCN, Inc.
>
> (305)785-3355
>
> Sent using www.apcn.net Internet Services.
>
>
>  Original message 
> From: Dario Estupinan 
> Date: 08/17/2016 8:53 AM (GMT-05:00)
> To: Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
> Subject: Re: [asterisk-users] Realtime SIP peers do not register any more
> after upgrade to Asterisk 13
>
> REMOVE ME please.
>
> 2016-08-15 15:16 GMT-05:00 Jonas Kellens :
>
>> Hello
>>
>> after I have upgraded from Asterisk 12 to asterisk-certified-13.8-cert1
>> none of my realtime SIP peers (saved in MySQL DB) register anymore.
>>
>>
>> [Aug 15 22:03:43] NOTICE[30098]: chan_sip.c:28451
>> handle_request_register: Registration from ''
>> failed for '78.119.140.190:5076' - Wrong password
>> [Aug 15 22:04:13] NOTICE[30098]: chan_sip.c:28451
>> handle_request_register: Registration from ''
>> failed for '78.119.140.190:5072' - Wrong password
>> [Aug 15 22:04:43] NOTICE[30098]: chan_sip.c:28451
>> handle_request_register: Registration from ''
>> failed for '78.119.140.190:5062' - Wrong password
>> [Aug 15 22:04:46] NOTICE[30098]: chan_sip.c:28451
>> handle_request_register: Registration from ''
>> failed for '78.119.140.190:5060' - Wrong password
>> [Aug 15 22:04:53] NOTICE[30098]: chan_sip.c:28451
>> handle_request_register: Registration from ''
>> failed for '78.119.140.190:5060' - Wrong password
>>
>>
>> Is this a known problem ??
>>
>>
>> Second question I have : can I get the complete list of columns that can
>> be used in realtime database for sip peers somewhere (update for Ast 13) ?
>> Are columns like dtlsenable, dtlsverify, dtlscertfile, dtlscafile,
>> dtlssetup possible ??
>>
>>
>>
>>
>> Thanks for the help.
>>
>>
>> Kind regards.
>>
>> Jonas.
>>
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>
>
>
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>
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> *Líder de NOC+*
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>
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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

[Matt Fredrickson]

On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens <jonas.kell...@telenet.be> wrote:
> My main reason not to upgrade to Ast 13 is because I'm afraid of losing
> functionality as there are certain functions deprecated/replaced. This can
> also cause headache :-)
>
> I will do so if there is no other option.
>
> But still, I don't see why Ast 13 would differ so much in this case ? If ICE
> and NAT is working (not causing problems) why should Ast 13 bring me audio
> and Ast 12 don't ??

If you want to minimize grief, start with 13 - WebRTC has been a
moving target for the last 5 years, it is not an old, mature standard
like ISDN or SIP.  If you find interop problems in an older version of
Asterisk with WebRTC, it's likely that it has been fixed in 13, and if
it hasn't the most likely place to obtain the fix will be in 13.

After you get the WebRTC part working, then you can move back the
versions of Asterisk you're using to see if it still works.

As far as ICE not working goes, if the browser you're talking to is
not on the same network as the Asterisk server, it's *possible* you
might need a true TURN server as well, instead of just an ICE server.

Matthew Fredrickson

>
>
>
> On 11-08-16 16:25, Jonathan H wrote:
>
> I'm genuinely fascinated why you are insisting on using a version of
> Asterisk almost 3 years old, for which EOL support ended last year.
>
> Is there any particular reason you cannot or will not use the current
> version as others have suggested?
>
> Also, I see you are using Doubango and WebRTC, but in the logs, I see WS and
> WSS.
>
> You NEED to be using 100% WSS otherwise you've not got a hope in hell of
> anything working with WEBRTC.
> Check the console of the web browser you are trying to make the call from
> (CTRL-SHIFT-I in Chrome on Windows, for example).
>
> Also, you'll need to be using valid certificates - self-signed certificates
> won't work for any current implementation of WebRTC that I know of,
> certainly not if anything involves current versions of Chrome or Firefox.
> That said, LetsEncrypt certs work fine for this, so no need to spend out on
> one.
>
> Switch to Asterisk 13.10 and save yourself a whole lotta headache.
>
> On 11 August 2016 at 15:09, Jonas Kellens <jonas.kell...@telenet.be> wrote:
>>
>> Hello
>>
>> Using Asterisk 12.8.2.
>>
>
>
>>
>> On 10-08-16 22:03, Matt Fredrickson wrote:
>>>
>>> My suggestion is to verify and debug against Asterisk 13 first, and
>>> then you can try backing down versions, rather than reverse.  WebRTC
>>> is a rapidly moving target, and has required ongoing changes that may
>>> not have made it into older and feature frozen versions of Asterisk.
>
>
>
>
>
>
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Re: [asterisk-users] Asterisk & Vitelity Invite issues

[Matt Fredrickson]

Wait a second, I thought in your original email that you said that
Asterisk was generating reinvites.  It sounds now like you're saying
that the remote side is initiating reinvites instead.

My understanding is that the canreinvite/directmedia option only
influences Asterisk's behavior with regards to generating reinivites.
If it receives a reinvite, I don't think these options will do
anything about that.  In fact, I'd guess that not properly responding
to a received reinvite is going to potentially break things from the
SIP perspective.

Matthew Fredrickson


On Wed, Aug 10, 2016 at 4:53 PM, Tammy Firefly <tammy-li...@wiztech.biz> wrote:
>
>
> On 8/9/16 12:40 PM, Matt Fredrickson wrote:
>> On Mon, Aug 8, 2016 at 9:25 AM, Tammy Firefly <tammy-li...@wiztech.biz> 
>> wrote:
>>> Hi All,
>>>
>>> We have asterisk 11.23 running sip to vitelity and from there IAX trunks
>>> split off to where they need to go.  We are having a problem getting
>>> chan_sip to quit ignoring re-invites from Vitelity.  Our side ends up
>>> sending a reinvite which their side & they do not support us sending a
>>> reinvite.  Ive tried:
>>>
>>> canreinvite=no which was supposedly replaced by:
>>>
>>> directmedia=no
>>>
>>> Can anyone shed any light on this matter?  I'd love to get this fixed.
>>>
>>
>> Those options *should* influence chan_sip's reinvite behavior - at
>> least they have from my experiences working with chan_sip.  Do you
>> know what is triggering the reinvite in the first place, or does it
>> look like a normal media reinvite?
>>
>
>
> every 15 minutes vitelity sends a re-invite to keep the call going.  I
> have a packet capture from it if you'd like it feel free to email me off
> list @ tamara.wis...@wiztech.biz
>
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Re: [asterisk-users] Original Callerid on transfer in asterisk 13

[Matt Fredrickson]

How are you attempting to view the original CallerId?

Matthew Fredrickson

On Wed, Aug 10, 2016 at 2:59 PM, Israel Gottlieb  wrote:
> Hi
> Is there any configuration change in asterisk 13.9.1 to show original
> callerid on a transfer
> In asterisk 11.21 it works as expected
>
> Thanks
>
>
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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

[Matt Fredrickson]

My suggestion is to verify and debug against Asterisk 13 first, and
then you can try backing down versions, rather than reverse.  WebRTC
is a rapidly moving target, and has required ongoing changes that may
not have made it into older and feature frozen versions of Asterisk.

Matthew Fredrickson

On Wed, Aug 10, 2016 at 3:01 PM, Jonas Kellens <jonas.kell...@telenet.be> wrote:
> Hello
>
> thank you for your answer.
>
> I don't understand how there are many tutorials and examples on the web
> where every time the outcome is a working setup. Very strange I feel now
> after my personal experience with Asterisk 11 and webRTC.
>
> You also say Asterisk 13. How about Asterisk 12 then ??
>
>
>
> Kind regards.
>
>
>
> On 10-08-16 21:53, Matt Fredrickson wrote:
>
> I don't see an ice-ufrag or ice-pwd line in the response from
> Asterisk, correlating with your suspicion that there is no ICE.  Are
> you sure that the stun server you're using (the google one) still
> works?  I haven't tried that server in a while, but I distantly seem
> to recall that maybe they shut it down.
>
> Asterisk 13 is a better place to be as well.  Asterisk 11 hasn't been
> feature updated in a while, and it could be that it could be a number
> of patches/fixes behind with regards to webrtc support, particularly
> with regards to interoperating with a modern browser version.
>
> Hope that helps,
> Matthew Fredrickson
>
> On Wed, Aug 10, 2016 at 5:02 AM, Jonas Kellens <jonas.kell...@telenet.be>
> wrote:
>
> On 10-08-16 08:52, Ludovic Gasc wrote:
>
> For WebRTC, I recommend you to use Asterisk 13+.
>
> Have a nice day.
>
> Ludovic Gasc (GMLudo)
> http://www.gmludo.eu/
>
>
>
>
> Hello
>
> then why is there an option in sip.conf and rtp.conf " icesupport=yes" ??
>
> This is no answer to my question.
>
> So again : what am I missing to get ICE support on my Asterisk 11.23.0 ??
>
>
>
> Kind regards.
>
>
>
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>
>
>
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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

[Matt Fredrickson]

I don't see an ice-ufrag or ice-pwd line in the response from
Asterisk, correlating with your suspicion that there is no ICE.  Are
you sure that the stun server you're using (the google one) still
works?  I haven't tried that server in a while, but I distantly seem
to recall that maybe they shut it down.

Asterisk 13 is a better place to be as well.  Asterisk 11 hasn't been
feature updated in a while, and it could be that it could be a number
of patches/fixes behind with regards to webrtc support, particularly
with regards to interoperating with a modern browser version.

Hope that helps,
Matthew Fredrickson

On Wed, Aug 10, 2016 at 5:02 AM, Jonas Kellens  wrote:
>
> On 10-08-16 08:52, Ludovic Gasc wrote:
>
> For WebRTC, I recommend you to use Asterisk 13+.
>
> Have a nice day.
>
> Ludovic Gasc (GMLudo)
> http://www.gmludo.eu/
>
>
>
>
> Hello
>
> then why is there an option in sip.conf and rtp.conf " icesupport=yes" ??
>
> This is no answer to my question.
>
> So again : what am I missing to get ICE support on my Asterisk 11.23.0 ??
>
>
>
> Kind regards.
>
>
>
> --
> _
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Re: [asterisk-users] EAGI script with missing audio on /dev/fd/3

[Matt Fredrickson]

On Tue, Aug 2, 2016 at 11:42 AM, nik600  wrote:
> Dear all
>
> i'm trying to access to the input audio raw stream with a very basic EAGI
> script:
>
>
> #!/bin/sh
> echo "EXEC Queue 2001"
> cat  /dev/fd/3 > /tmp/pippo
>
> This is my dialplan:
>
> exten => 001,NoOp(test)
> exten => 001,n,Answer
> exten => 001,n,EAGI(/tmp/my-eagi.agi)
>
>
> When i call, the script is executed and the call goes in queue, i can hear
> the MOH, the file /tmp/pippo is created but it is empty.
>
> Any idea or suggestion?

If you take out the "echo "EXEC Queue 2001" part of it, do you get
audio in the file?

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Re: [asterisk-users] Asterisk & Vitelity Invite issues

[Matt Fredrickson]

On Mon, Aug 8, 2016 at 9:25 AM, Tammy Firefly  wrote:
> Hi All,
>
> We have asterisk 11.23 running sip to vitelity and from there IAX trunks
> split off to where they need to go.  We are having a problem getting
> chan_sip to quit ignoring re-invites from Vitelity.  Our side ends up
> sending a reinvite which their side & they do not support us sending a
> reinvite.  Ive tried:
>
> canreinvite=no which was supposedly replaced by:
>
> directmedia=no
>
> Can anyone shed any light on this matter?  I'd love to get this fixed.
>

Those options *should* influence chan_sip's reinvite behavior - at
least they have from my experiences working with chan_sip.  Do you
know what is triggering the reinvite in the first place, or does it
look like a normal media reinvite?

-- 
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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

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Re: [asterisk-users] AstriCon 2016 - XMPP and Asterisk

[Matt Fredrickson]

Looking forward to seeing you there, and hopefully to seeing your talk!

Matthew Fredrickson

On Tue, Aug 2, 2016 at 11:02 AM, Marcelo Terres  wrote:
> Going to AstriCon 2016 ?
>
> Don't miss my talk about how to use XMPP and Asterisk to improve the
> user experience.
>
> https://astricon2016.sched.org/event/7Zje/using-asterisk-and-xmpp-to-provide-greater-tools-to-your-customers-and-your-users
>
> Regards,
>
> Marcelo H. Terres 
> IM: mhter...@jabber.mundoopensource.com.br
> https://www.mundoopensource.com.br
> https://twitter.com/mhterres
> https://linkedin.com/in/marceloterres
>
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Re: [asterisk-users] Force out-bond call to specific CIC

[Matt Fredrickson]

Yes, as far as I remember, in your dial string, simply use a
Dial(DAHDI/X/1234567) where X is the dahdi device channel number.

Hope that helps.
Matthew Fredrickson

On Wed, Jul 13, 2016 at 5:22 AM, Mehdi Shirazi  wrote:
> Hi
>
> How is it possible to use Dial application to force out-bond call use
> "specified channel" number in one E1 or specified CIC (SS7) ?
>
> Regards
> M.Shirazi
>
>
>
>
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Re: [asterisk-users] Unable to create channel DAHDI

[Matt Fredrickson]

Looks like the hookstate is listed as offhook.  I don't think
chan_dahdi will attempt to make a call out a device that is offhook.

Hope that helps,
Matthew Fredrickson

On Tue, Jun 7, 2016 at 1:36 PM, Brent Davidson
 wrote:
> In trying to troubleshoot the Delay after Answer problem I had before (which
> seems to be fixed), I have somehow created a new problem:
>
> Outgoing calls are now failing with the following message:
>
> [Jun  7 13:28:09] WARNING[9247][C-]: app_dial.c:2429 dial_exec_full:
> Unable to create channel of type 'DAHDI' (cause 0 - Unknown)
>
> But I DO have working dahdi as incoming calls are working correctly.
>
> CLI> dahdi show channels
>Chan Extension   Context Language   MOH Interpret
> BlockedIn Service Description
>  pseudo defaultdefault
> Yes
>   3 mainmenu   default
> Yes
>   4 mainmenu   default
> Yes
> CLI> dahdi show status
> Description  Alarms  IRQbpviol CRCFra
> Codi Options  LBO
> Wildcard AEX410  OK  0  0  0  CAS
> Unk   0 db (CSU)/0-133 feet (DSX-1)
> CLI> dahdi show channel 3
> Channel: 3
> Description:
> File Descriptor: 14
> Span: 1
> Extension:
> Dialing: no
> Context: mainmenu
> Caller ID:
> Calling TON: 0
> Caller ID name:
> Mailbox: none
> Destroy: 0
> InAlarm: 0
> Signalling Type: FXS Kewlstart
> Radio: 0
> Owner: 
> Real: 
> Callwait: 
> Threeway: 
> Confno: -1
> Propagated Conference: -1
> Real in conference: 0
> DSP: no
> Busy Detection: yes
> Busy Count: 8
> Busy Pattern: 0,0,0,0
> TDD: no
> Relax DTMF: yes
> Dialing/CallwaitCAS: 0/0
> Default law: ulaw
> Fax Handled: no
> Pulse phone: no
> HW Gains (RX/TX): Disabled/Disabled
> SW Gains (RX/TX): 0.00/0.00
> Dynamic Range Compression (RX/TX): 0.00/0.00
> DND: no
> Echo Cancellation:
> 128 taps
> (unless TDM bridged) currently OFF
> Wait for dialtone: 0ms
> Actual Confinfo: Num/0, Mode/0x
> Actual Confmute: No
> Hookstate (FXS only): Offhook
> CLI>
>
> dahdi show channel 4
> Channel: 4
> Description:
> File Descriptor: 15
> Span: 1
> Extension:
> Dialing: no
> Context: mainmenu
> Caller ID:
> Calling TON: 0
> Caller ID name:
> Mailbox: none
> Destroy: 0
> InAlarm: 0
> Signalling Type: FXS Kewlstart
> Radio: 0
> Owner: 
> Real: 
> Callwait: 
> Threeway: 
> Confno: -1
> Propagated Conference: -1
> Real in conference: 0
> DSP: no
> Busy Detection: yes
> Busy Count: 8
> Busy Pattern: 0,0,0,0
> TDD: no
> Relax DTMF: yes
> Dialing/CallwaitCAS: 0/0
> Default law: ulaw
> Fax Handled: no
> Pulse phone: no
> HW Gains (RX/TX): Disabled/Disabled
> SW Gains (RX/TX): 0.00/0.00
> Dynamic Range Compression (RX/TX): 0.00/0.00
> DND: no
> Echo Cancellation:
> 128 taps
> (unless TDM bridged) currently OFF
> Wait for dialtone: 0ms
> Actual Confinfo: Num/0, Mode/0x
> Actual Confmute: No
> Hookstate (FXS only): Offhook
>
> The Hookstates always say offhook for some reason, though I'm not sure why.
>
> My setup:
>
> Server running Asterisk 13.9.1, Dahdi 2.11.1 w/ OSLEC
> Server is CentOS 7
> Quad core CPU with 16GB Ram
> 2 Snom 300 phones.
> NO NAT.  Server and phone are on the same subnet with only a gigabit switch
> between them.
> Digium AEX410P analog card with 2 incoming analog PSTN lines
>
> Any ideas?
>
>
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Re: [asterisk-users] Advices on how to evaluate voice quality in a mixed Dahdi/SIP environment ?

[Matt Fredrickson]

On Wed, May 18, 2016 at 9:44 AM, Olivier  wrote:
> I've got the following setup:
>
> PSTN  ITSP  SDSL Modem-Router  Gateway -
> Asterisk with B410P --- SIP Phones

Wow.

> Both SDSL Modem-Router and Gateway are managed by my ITSP.
>
> Some calls coming from PSTN and forwarded to an other PSTN number have a
> poor voice quality.

How are you forwarding them?  Is it in such a way that you remain in
the audio path, or do you get out of the audio path in the forward?

> How can I best illustrate this ?

It depends on what let has the bad audio.  If it's on the SIP side
(RTP to RTP) a pcap file will show you your perspective of audio
losses.  Received RTCP reports should show you the other side's
perspective of audio losses as well.

> A friend advised me to simply record incoming DAHDI channel, for instance.
> How can I then translate record WAV file into meaningful figures ?

If DAHDI is still in the picture in the forward scenario, that would
be another place to monitor the audio.

> More generaly, what would you suggest ?

Try to capture each leg (IP side, using tcpdump/wireshark) and on
DAHDI using dahdi_monitor or something equivalent.  Figure out if any
of your legs of audio quality issues.  If you don't see anything, it's
something at their end.

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Re: [asterisk-users] Avaya Phones and Asterisk

[Matt Fredrickson]

On Fri, May 20, 2016 at 8:54 AM, Diogo Cosito  wrote:
> Dear gentlemen, how are you?
> I wonder if anyone has experience with Avaya devices, 9608G and 9641GS
> models, running on SIP and using TCP transport.
> The calls work well, but the callerid only "pass" number of the extension or
> external number, without the name (configured correctly in sip.conf and
> testing with other devices UDP works fine, like Yealink, Eyebeam, etc),
> device contacts list does not work and also the hint does not work 
> can anybody help me?

So if you use UDP transport everythng works fine?

Can you post a packet capture of this happening?  Also, what version
of Asterisk and your sip.conf?

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Re: [asterisk-users] pjsip segfault problem

[Matt Fredrickson]

Have you tried updating to pjproject version 2.5.x?  It should have
the patch that you listed in your other email, which I believe should
be included in that branch.

Hope that helps, and best of luck.

Matthew Fredrickson

On Thu, May 26, 2016 at 4:11 AM, Marek Červenka  wrote:
> hi,
>
> after switch from 13.7 + pjproject 2.4.5 to 13.9.1 pjproject bundled i have
> problem with segfault  (centos 6)
>
> Program terminated with signal 11, Segmentation fault.
> #0  0xb7665695 in check_cached_response (sess=0xafbd688c, packet=0xb07676d8,
> pkt_size=132, options=1, token=0xafecc2bc, parsed_len=0x0,
> src_addr=0xb0e47a20, src_addr_len=16)
> at ../src/pjnath/stun_session.c:1287
> 1287if (t->msg_magic == msg->hdr.magic &&
>
>
> it was only once after 2 days.  i dont know how to repeat it now :(
>
> any similiar experience?
>
>
> --
> ---
> Marek Cervenka
> ===
>
>
> --
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Re: [asterisk-users] T.38 with Audiocodes gateway

[Matt Fredrickson]

On Fri, Apr 29, 2016 at 1:34 AM, Olivier  wrote:
> Hello,
>
> I'm helping a colleague (*) which has the following setup:
>
> ITSP ---  --- Asterisk 13 ---  --
> Audiocodes MP-112 ---   --- Fax machine
>
> My issue is the following :
> Audiocodes gateway reject INVITEs with 488 Not Acceptable Here
>
> It seems this gateway requires t38 settings to be present in SDP body in the
> very first INVITE.
>
> My questions are the following:
>
> 1. I expected T.38 to exclusively work with reINVITE where calls are
> established as normal voice calls (PCMA/PCMU in SDP, for instance) and then
> upgraded to T.38 (when CNG is detected, for instance).
> Have you ever heard of T.38 sessions being established right from the start
> (ie with T.38 settings in the first INVITE) ?

No.  It would seem to be extremely broken if it denies a call based on
a lack of T.38 sdp parameters on the initial INVITE.

> 2. Is it possible to configure Asterisk to pass T.38 settings in SDP in the
> first INVITE it sends ?
>
> 3. Any suggestion with Audiocodes gateway ?

Look for T.38 settings maybe?  See if there is something keeping you
from sending an initial invite with non-T.38 SDP?

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Re: [asterisk-users] PRI error "ROSE REJECT"

[Matt Fredrickson]

Perhaps it's taking a bit longer in the network for the media path to
open after the CONNECT, which would explain why the first digit is not
being detected.

Also, what changed last week?

I don't see the ROSE REJECT message anywhere in the pri debug -
perhaps you didn't catch it.

Matthew Fredrickson

On Fri, Mar 25, 2016 at 9:15 PM, Carlos Chavez <cur...@telecomabmex.com> wrote:
> On 2016-03-25 16:02, Matt Fredrickson wrote:
>>
>> PRI debug of the entire call would be great, also, switchtype would be
>> awesome as well.
>>
>> Thanks!
>>
>> Matthew Fredrickson
>>
>> On Thu, Mar 24, 2016 at 4:07 PM, Carlos Rojas <crt.ro...@gmail.com> wrote:
>>>
>>> Hi
>>>
>>> Did you activate the pri debug on the cli asterisk?
>>>
>>> On Thu, Mar 24, 2016 at 12:59 PM, Carlos Chavez <cur...@telecomabmex.com>
>>> wrote:
>>>>
>>>>
>>>> We've been having some problems with an E1 PRI line for a few days.  We
>>>> get the following errors:
>>>>
>>>> [Mar 24 10:13:39] ERROR[22009] chan_dahdi.c: PRI Span: 2 ROSE REJECT:
>>>> [Mar 24 10:13:39] ERROR[22009] chan_dahdi.c: PRI Span: 2INVOKE
>>>> ID:
>>>> 316
>>>> [Mar 24 10:13:39] ERROR[22009] chan_dahdi.c: PRI Span: 2PROBLEM:
>>>> Invoke: Unrecognized Operation
>>>>
>>>> The telephone company says that everything is fine on their side,
>>>> obviously.  The problems started a few days ago when a user reported
>>>> that
>>>> incoming calls get dropped when you try to dial a particular extension
>>>> from
>>>> the main IVR.  We are using Asterisk 1.8.15-cert2 on a CentOS 6.7
>>>> server,
>>>> DAHDI 2.6.1 and libpri 1.4.  Any recommendations?
>>>>
>>>> --
>
>
> system.conf:
> # Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER)
> span=1,2,0,cas,hdb3
> cas=1-15:1101
> cas=17-31:1101
>
> # Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2"
> span=2,1,0,ccs,hdb3
> # termtype: te
> bchan=32-46
> dchan=47
> bchan=48-62
>
> loadzone= mx
> defaultzone = mx
>
> chan_dahdi.conf:
> language=es
> context=e1-incoming
> usecallerid=yes
> hidecallerid=no
> callwaiting=no
> canpark=no
> usecallingpres=no
> callwaitingcallerid=no
> threewaycalling=no
> transfer=yes
> cancallforward=no
> callreturn=no
> echocancel=yes
> echocancelwhenbridged=no
> echotraining=yes
> rxgain=0.0
> txgain=0.0
> accountcode=E1
> amaflags=default
> signalling=pri_cpe
> pridialplan=unknown
> prilocaldialplan=unknown
> switchtype=euroisdn
> overlapdial=no
> immediate=no
> group=2
> faxdetect=no
> callerid=asreceived
> mohinterpret=default
> mohsuggest=default
> dahdichan=32-46,48-62
>
> Here is the pri debug:
> -- Accepting call from '55' to '5732' on channel 0/1, span 2
> -- Executing [5732@e1-incoming:1] Goto("DAHDI/i2/55-3c",
> "menu-gci,s,1") in new stack
> -- Goto (menu-gci,s,1)
> -- Executing [s@menu-gci:1] Wait("DAHDI/i2/55-3c", "2") in new
> stack
> -- Executing [s@menu-gci:2] Answer("DAHDI/i2/55-3c", "") in new
> stack
> PRI Span: 2 q931.c:4683 q931_connect: Call 48 enters state 8 (Connect
> Request).  Hold state: Idle
> PRI Span: 2
> PRI Span: 2 > DL-DATA request
> PRI Span: 2 > Protocol Discriminator: Q.931 (8)  len=14
> PRI Span: 2 > TEI=0 Call Ref: len= 2 (reference 48/0x30) (Sent to
> originator)
> PRI Span: 2 > Message Type: CONNECT (7)
> PRI Span: 2 TEI=0 Transmitting N(S)=98, window is open V(A)=98 K=7
> PRI Span: 2
> PRI Span: 2 > Protocol Discriminator: Q.931 (8)  len=14
> PRI Span: 2 > TEI=0 Call Ref: len= 2 (reference 48/0x30) (Sent to
> originator)
> PRI Span: 2 > Message Type: CONNECT (7)
> PRI Span: 2 > [18 03 a9 83 81]
> PRI Span: 2 > Channel ID (len= 5) [ Ext: 1  IntID: Implicit  Other(PRI)
> Spare: 0  Exclusive  Dchan: 0
> PRI Span: 2 >   ChanSel: As indicated in following
> octets
> PRI Span: 2 >   Ext: 1  Coding: 0  Number Specified
> Channel Type: 3
> PRI Span: 2 >   Ext: 1  Channel: 1 Type: CPE]
> PRI Span: 2 > [1e 02 81 82]
> PRI Span: 2 > Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU)
> standard (0)  0: 0  Location: Private network serving the local user (1)
> PRI Span: 2 >   Ext: 1  Progress Description:
> Called equipment is non-ISDN. (2) ]
> 

Re: [asterisk-users] PRI error "ROSE REJECT"

[Matt Fredrickson]

PRI debug of the entire call would be great, also, switchtype would be
awesome as well.

Thanks!

Matthew Fredrickson

On Thu, Mar 24, 2016 at 4:07 PM, Carlos Rojas  wrote:
> Hi
>
> Did you activate the pri debug on the cli asterisk?
>
> On Thu, Mar 24, 2016 at 12:59 PM, Carlos Chavez 
> wrote:
>>
>> We've been having some problems with an E1 PRI line for a few days.  We
>> get the following errors:
>>
>> [Mar 24 10:13:39] ERROR[22009] chan_dahdi.c: PRI Span: 2 ROSE REJECT:
>> [Mar 24 10:13:39] ERROR[22009] chan_dahdi.c: PRI Span: 2INVOKE ID:
>> 316
>> [Mar 24 10:13:39] ERROR[22009] chan_dahdi.c: PRI Span: 2PROBLEM:
>> Invoke: Unrecognized Operation
>>
>> The telephone company says that everything is fine on their side,
>> obviously.  The problems started a few days ago when a user reported that
>> incoming calls get dropped when you try to dial a particular extension from
>> the main IVR.  We are using Asterisk 1.8.15-cert2 on a CentOS 6.7 server,
>> DAHDI 2.6.1 and libpri 1.4.  Any recommendations?
>>
>> --
>> Telecomunicaciones Abiertas de México S.A. de C.V.
>> Carlos Chávez
>> dCAP #1349
>> +52 (55)9116-91161
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>   http://www.asterisk.org/hello
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
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Re: [asterisk-users] OPUS support in Asterisk 13

[Matt Fredrickson]

On Thu, Mar 24, 2016 at 9:09 AM, Chirag Desai  wrote:
> Hi all,
>
> Sorry if this has been asked before. I searched a lot, but found conflicting
> answers, so hoping for some clarification.
>
> My question is does Asterisk 13 support OPUS? If so which version exactly?

Sort of - it supports OPUS pass through officially, but does not
support OPUS transcoding.  I'm not certain when pass through support
went in though, but I believe it was prior to cutting of the 13
branch.

There are some unofficial patches that add OPUS support to Asterisk
but I cannot point you to which ones are best to use, unfortunately.

Hope that helps!

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Re: [asterisk-users] accept DMTF tone during ringing

[Matt Fredrickson]

For what channel driver, and what use case?

It's my understanding that in the traditional telephone network
(ISDN/SS7/analog), prior to a call being answered, you were not necessarily
guaranteed a two way media path.  Sometimes it was available (there are few
stories of large companies who somehow talked their telco provider into
allowing it so users could traverse their IVR prior to a billable answer)
but I don't *think* it's guaranteed.

>From a SIP standpoint, there are many ways it could not work as well,
depending on the type of DTMF being sent.  If you're using inband DTMF of
any sort (audio inband or RFC2833-ish) if you don't have a bidirectional
media path established through NATs and such, you could miss digits as well.

Matthew Fredrickson


On Sun, Nov 8, 2015 at 3:50 PM, hadi  wrote:

>
> Hi,
>
>
> How to accept DMTF tone during ringing mode? Its possible.
>
> Regards
>
> -Hadi.Salem
>
>
>
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Re: [asterisk-users] Fax and Asterisk

[Matt Fredrickson]

- Lee Howard [EMAIL PROTECTED] wrote:
 Andrew Nowrot wrote:
 
  I am trying to build reliable fax solution with asterisk, iaxmodem
 and 
  hylafax. I am attempting to do this on Compaq DL-360 with 2 pentium
 3 
  1.2 GHz (512 cache) and 2GB of RAM. I am using a Sangoma A101. After
 
  installing the newest zaptel and wanpipe-3.1.0 beta I did zttest and
 
  it didn't give me good results:
 
  99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 
  99.975586%
  99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 
  99.975586%
 
 
 Are you having trouble with fax?  Rumor is it that the Sangoma
 hardware 
 isn't as needy this way as is the Diguim.  I'm not sure about that,
 though.

That should not be relevant anymore.  You should see approximately the same 
reliability with both cards, as long as the card and timing is configured 
correctly.

Matthew Fredrickson


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Re: [Asterisk-Users] AstriCon 2006 Location

[Matt Fredrickson]

On Sun, Sep 18, 2005 at 11:32:00AM -0500, Brian Capouch wrote:
 Senad J wrote:
 If you are looking for the maximum number of cheap flights from around
 the world, and plenty of convention and room space, the answer is Las
 Vegas :-)
 
 
 I would definitively agree!
 
 
 Yes, but what would one do there?
 
 One who doesn't gamble, drink, or carouse, that is.
 
 I am making my first trip to LV later this Fall, and I dread it.  I 
 can't imagine what I'll be able to find to do when I'm not at the 
 conference.

It's ok, I don't either  :-)  I was actually kind of wondering the same
thing.  I'm sure there's something to do that doesn't involve all of that.

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Re: [Asterisk-Users] Seperate Incoming calls on TDM02?

[Matt Fredrickson]

On Thu, Sep 15, 2005 at 12:04:41PM -0400, C. Hatton Humphrey wrote:
 I have a TDM02B to bring in two POTS lines for my incoming calls; I
 need to point each line to a different IVR... is there somewhere that
 can I can look to get this setup working?
  
 Basically, each line is for a different business. I know that for a
 DID the routing is simple but I'm not seeing where I can match up a
 DID with a Zap channel.
 
 I'm currently looking into the zapata.conf file to do this as it is my
 understanding that the control can be taken care of there.  My system
 is running [EMAIL PROTECTED] 1.5.

Yeah, in your zapata.conf just give each channel a different context setting.

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Re: [Asterisk-Users] PRI echo

[Matt Fredrickson]

On Mon, Sep 12, 2005 at 08:48:31PM -0600, Gabriel Gunderson wrote:
  Upgrade! Upgrade! Upgrade!  At the very least, upgrade to
  the last 1.0.x.  Even better would be to use the CVS-HEAD
  or the latest 1.2beta release of libpri, zaptel, and asterisk.
  
  There are new features that help TREMENDOUSLY with echo.  Try
  the new echo canceller out (KB1, recently made default).  Your
  bugs will probably go away, and your echo problems will likely
  (at the very least) be improved.
 
 Is there any reason that you shouldn't run 1.2 beta of libpri and
 1.0.9 of asterisk.  I'm at a place where we can't risk upgrading
 asterisk right now but would love better echo cancellation on our PRI.
  Would it even make a difference or are the bulk of improvements in
 asterisk itself?

As a general principle, you should not mix and match versions like this.
For this specific circumstance, you MIGHT be able to use zaptel from
head, IIRC., but I personally don't recommend it.  libpri from head will
DEFINITELY not work with 1.0.x asterisk.  Many things have changed since
then.  Just try upgrading.  I think a lot of people are set in the mold
that the 1.0.x releases are more stable than the 1.2.x coming release
branch.  Feature wise, it probably is, but I feel (at least about libpri
and zaptel related issues) that 1.2.x/HEAD is by far more stable and reliable
than 1.0.x.

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Re: [Asterisk-Users] PRI echo

[Matt Fredrickson]

On Sat, Sep 10, 2005 at 12:51:21AM -0700, Jason Kim wrote:
 My configuration is pri*(te405p)---iaxclient.
 My * version is 1.0.7 running on tyan dual opteron
 board.
 I have several problems.

Upgrade! Upgrade! Upgrade!  At the very least, upgrade to
the last 1.0.x.  Even better would be to use the CVS-HEAD
or the latest 1.2beta release of libpri, zaptel, and asterisk.

There are new features that help TREMENDOUSLY with echo.  Try
the new echo canceller out (KB1, recently made default).  Your
bugs will probably go away, and your echo problems will likely
(at the very least) be improved.

 
 1) inbound echo
 For outbound call(iaxclient--pri), there is almost no
 echo. But for inbound(pri--iaxclient), I can hear
 distinct echo. Can Sangoma a104 or digium te406p help
 this problem? 
 
 2)Today i received te406p. I know T1/E1 jumper. But
 how can i change the configuration of te406p for echo
 cancel mode selection?
 
 3) asterisk crash
 (gdb) bt
 #0  0x002a973f15dd in q921_transmit_iframe
 (pri=0x2a9cd0ecf0, buf=0x40ffeac0, len=9, 
 cr=1) at q921.c:384
 #1  0x002a973f701c in q931_xmit (pri=0x2a9cd0ecf0,
 h=0x40ffeac0, len=9, cr=1)
 at q931.c:1848
 #2  0x002a973f720f in send_message
 (pri=0x2a9cd0ecf0, c=0x2a9cd12810, msgtype=77, 
 ies=0x2a97500570) at q931.c:1888
 #3  0x002a973f7b31 in q931_release
 (pri=0x2a9cd0ecf0, c=0x2a9cd12810, cause=16)
 at q931.c:2141
 #4  0x002a973f78eb in pri_disconnect_timeout
 (data=0x2a9cd12810) at q931.c:2092
 #5  0x002a973f309b in __pri_schedule_run
 (pri=0x2a95aa1da0, tv=0x40ffefa0)
 at prisched.c:97
 #6  0x002a973f30f8 in pri_schedule_run
 (pri=0x2a95aa1da0) at prisched.c:109
 #7  0x002a9729e77a in pri_dchannel
 (vpri=0x2a9cd0ecf0) at chan_zap.c:7415
 #8  0x00307f305f81 in start_thread () from
 /lib64/tls/libpthread.so.0
 #9  0x00307e6c3af3 in thread_start () from
 /lib64/tls/libc.so.6
 #10 0x in ?? ()
 
 4)chan_iax2
 Some times asterisk log file is filled with strange
 message as follow.
 Sep  8 13:36:41 NOTICE[4339]: I should never be
 called!
 Sep  8 13:36:41 NOTICE[4339]: I should never be
 called!
 And some times it's overloading my sata HDD and 
 disable ssh connection.
 
 iax.conf
 ---
 [general]
 port=5036
 disallow=all
 tos=0x04
 qualify=no
 
 [agent]
 type=friend
 username=agent
 secret=agent
 context=agent
 host=dynamic
 notransfer=yes
 callerid=20005000
 
 zapata.conf
 ---
 [channels]
 context=default
 switchtype=euroisdn
 pridialplan=national
 prilocaldialplan=national
 nationalprefix=
 
 signalling=pri_cpe
 usecallerid=yes
 hidecallerid=no
 callwaiting=yes
 callwaitingcallerid=yes
 callprogressdetect=yes
 threewaycalling=yes
 transfer=yes
 cancallforward=yes
 echocancel=yes
 echocancelwhenbridged=yes
 echotraining=800
 rxgain=2.0
 txgain=-4.0
 
 group=1
 channel = 1-15
 channel = 17-31
 channel = 32-46
 channel = 48-62
 --
 
 Thanks, have a great holiday!
 
 Regards,
 Jason
 
 
 
   
   
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Re: [Asterisk-Users] -- PROGRESS with cause code 34 received?

[Matt Fredrickson]

On Wed, Sep 07, 2005 at 01:47:49PM +0200, Roy Sigurd Karlsbakk wrote:
 hi
 
 i get these messages every now and then
 
 -- PROGRESS with cause code 34 received
 
 wtf is this?

Nothing to see here, move along :-)

Seriously though, it's basically just and interesting message to see.  The cause
code IE withing the progress message was set to 34 (You can look up what that 
means
in the Q.931 spec).

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Re: [Asterisk-Users] One way echo canceling?

[Matt Fredrickson]

On Thu, Sep 01, 2005 at 12:40:16PM -0400, Doug Lytle wrote:
 When there is a call on zap 1, from a sip phone on the remote office 

 
 I have not seen it myself, but I have heard that some people have ahd 
 trouble with
 overlapdial and echo cancellation.  I have not been able to confirm 
 whether or not
 this is actually a bug.  One possible fix is to disable overlapdial and 
 see if echo
 cancellation is enabled after this.  If it is, this might be a bug in 
 chan_zap.c
 
 Turning off overlapdial did indeed fix it.  It now shows as being enabled.

Try updating to latest HEAD.  I just fixed it this morning.  It should work
with overlapdial enabled now.

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Re: [Asterisk-Users] One way echo canceling?

[Matt Fredrickson]

On Wed, Aug 31, 2005 at 09:46:37PM -0400, Doug Lytle wrote:
 When there is a call on zap 1, from a sip phone on the remote office 
 side and typing 'zap show channel 1'  shows echo cancel is on, doing the 
 same thing from the Definity to a SIP phone shows echo cancel off.  
 Shouldn't it be on during a call on both the incoming and outgoing legs 
 as long as it comes accross the PRI?  Some (Myself included) have noted 
 a slight echo on the Definity to SIP leg of the connection.
 
 My zapata.conf is below:
 
 switchtype = national
 context = incoming
 signalling = pri_cpe
 echocancel=yes
 echotraining = yes
 echocancelwhenbridged=yes
 overlapdial = yes
 group = 1
 channel = 1-23

I have not seen it myself, but I have heard that some people have ahd trouble 
with
overlapdial and echo cancellation.  I have not been able to confirm whether or 
not
this is actually a bug.  One possible fix is to disable overlapdial and see if 
echo
cancellation is enabled after this.  If it is, this might be a bug in chan_zap.c

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Re: [Asterisk-Users] TDM04b and echo

[Matt Fredrickson]

On Wed, Aug 31, 2005 at 08:45:34PM -0500, chris gamble wrote:
 the echo isnt horrible most of the time, and seems extremely random in
 that i can call a number once without echo, then dial the same number a
 second time and get echo.
 
 things i am currently considering (and would like to know if these might
 be useful)
 1 upgrade from 1.09 ( asterisk at home ) to 1.2 cvs code base

That is worth a shot.  There are a few new echo-related features that have
been added:

1.) fxotune - try this first.  There is a file called README.fxotune that
explains how to use it.  It is primarily for doing echo related line tuning
(which in your case possibly won't help).

2.) Also, there is a new echo canceller in CVS-HEAD zaptel that has received a
lot of positive feedback.  Look in zconfig.h for ECHO_CAN_KB1 for further
information.

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Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared

[Matt Fredrickson]

On Sun, Aug 28, 2005 at 02:52:20PM -0400, Andrew Kohlsmith wrote:
 On Sunday 28 August 2005 11:59, Steve Underwood wrote:
  I don't follow why knowing that impedance mismatch is the problem has
  stopped you making fxotune fix it. :-\ Where you the one who asked me
  how to make fxotune work well on IRC? Someone asked a while ago, and
  said they were working on a faster tuning algorithm for fxotune. I've
  forgotten who.
 
 I thought fxotune set up the built-in FIR filter in the DAA and nothing more. 
  
 I'm really uncertain how a little filter is going to help with impedance 
 matching, as it's still the same frequency ranges that need to get through to 
 be digitized.
 
 I have, however, been known to be mistaken on more than one occassion.  :-)

fxotune currently only does tuning with the AC impedance functions on the 
Si3050.

If there is continued line-related echo problems, there is always the option of
adding the onboard digital hybrid tuning to the mix, but it is unable to fix the
problem as much as doing the AC impedance tuning.  The onboard hybrid is more
for that last step of tweaking, only to be used after doing the line adjustment
for the AC impedance.

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Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared

[Matt Fredrickson]

On Mon, Aug 29, 2005 at 03:24:01PM -0500, Ric Moseley wrote:
 Are changes to the zapata.conf  file read on the fly or do you have to
 restart the asterisk process?

It doesn't make any changes to the zapata.conf file.  It has it's own config
file that you have to set it up to load from before you start asterisk.  See
the README.fxotune file in zaptel CVS HEAD.

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Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared

[Matt Fredrickson]

On Fri, Aug 26, 2005 at 02:00:54PM -0600, Rich Adamson wrote:
 Relative to the fxotune app, it would appear the app is specific
 to the v2.4 kernels (/dev/zap*), which the v2.6 kernels don't use

It should with 2.4 and 2.6.  2.6 kernels with properly configured udev
rules should create the /dev/zap/* entries dynamically.

 (but rather the udev equivalent). (When I had * running on a v2.4
 kernel, the output from fxotune never deviated from all zero's. So
 I'm assuming the default chipset values were already tweaked by the
 chipset manufacturer to US telco lines. If that is true, then 
 running fxotune in the US has very little value.)

Sometimes in the US you still deal with line impedance issues.  In fact,
I was told by an engineer that worked for the company that designs the
line interface part that the bulk of echo problems (with line interface 
parts such as this) are related to AC impedance mismatches (which is one
reason why I haven't done the digital hybrid tuning portion of fxotune
still).  It should work the same regardless of which kernel (2.4 or 2.6)
you are using.

If it doesn't, and you have udev setup correctly, something is fundamentally
wrong in the setup.

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Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared

[Matt Fredrickson]

Ok, fxotune is a work in progress so to speak.  I fixed something in it about
a week ago that may help it adjust to the line better (whereas before I'm not
sure that it was at all).  Try the latest CVS-HEAD version of fxotune as your
first step.  (oh, after you use fxotune you should turn off your gain settings
in zapata.conf).

Second step is to try the new echo canceller that was added to CVS-HEAD.  Look
in zconfig.h and try the KB1 echo canceller.  I have received many good reports
that it has cured practically all echo on all of the systems that I have heard
feedback from.

If all of this doesn't work, you probably have a serious hardware line issue
that you should resolve with your telco.

---
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Re: [Asterisk-Users] HDLC/Zaptel/Kernel 2.6.11(.9)

[Matt Fredrickson]

On Tue, Aug 23, 2005 at 11:47:19AM -0500, Matt Schulte wrote:
   I'm having a heck of a time getting hdlc to work on kernel
 2.6.11.9 .. I compiled hdlc, hdlc_gen, hdlc_cisco, hdlc_raw, into the
 kernel (note into, and not 'modules').

Kevin can correct me if I got this wrong, but IIRC, he noticed this problem
and traced something similar to this back to the cisco hdlc encapsulation
layer in the kernel.  For some reason, his version of the kernel (I think it
was 2.6.12) had problems with the cisco encapsulation.  Try a different version
of the kernel and see if that works.

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Re: [Asterisk-Users] Asterisk 1.0.9: TE411P replacement for TE410P 1stgen causes crashes

[Matt Fredrickson]

On Tue, Aug 23, 2005 at 08:03:42PM +0200, [EMAIL PROTECTED] wrote:
 I replaced a TE410P Rev C 1st Generation Firmware with a TE411P
 without any cfg changes (zaptel/zapata).
 
 As a result Asterisk crashes on outbound from PRI4 going to PRI1 calls:
 
 Aug 23 18:22:00 WARNING[4693]: chan_zap.c:7545 zt_pri_error: PRI: !! Got a 
 UA, but i'm in state 1
 Aug 23 18:22:00 WARNING[4693]: chan_zap.c:7545 zt_pri_error: PRI: !! Got a 
 UA, but i'm in state 1
 
 Urgent handler
 Urgent handler
 -- Accepting overlap voice call from '4123655105' to '555115' on channel 
 0/26, span 4
 -- Starting simple switch on 'Zap/119-1'
 Urgent handler
 Urgent handler
 Ouch ... error while writing audio data: : Broken pipe
 Junk at the beginning 49443303
 Warning, flexibel rate not heavily tested!
 Segmentation fault
 
 Are there any changes to the config neccessary?

There shouldn't be.

 Are there known issues with this Asterisk version?

Yes.  You should update your libpri/zaptel/asterisk to the latest CVS-STABLE
release.  This last week we commited quite a few random fixes for the TE411P
that could have affected this particular issue.  If not, it might be quicker
handled if you contact digium support.

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Re: [Asterisk-Users] any ISDN/PRI signaling experts out there?

[Matt Fredrickson]

On Fri, Aug 19, 2005 at 07:01:00AM -0600, Damon Estep wrote:
 On the same setup, if I connect another PRI device to it that emulates
 switch side signaling and includes the CNAM as a Display IE in the
 setup, the SIP invite is properly formatted and * receives the calling
 party name.
 
 Does anyone here have enough experience with ISDN PRI signaling to
 comment with some level of authority on this?

Asteris/libpri can process and handle either style of caller name delivery
(GR-1367 and Display IE).  If they do now send the name information in the
SETUP message you may have to do a delay in your dialplan before you access
that information.  You'd have to hook your Asterisk machine directly up
to the 5E for that to work though.

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Re: [Asterisk-Users] TE411P problem

[Matt Fredrickson]

On Fri, Aug 12, 2005 at 09:03:42PM -0500, Tim Connolly wrote:
 You might start by running /usr/src/zaptel/zttest. See if you stay at 100%.
 That's going to be the first thing digium checks. You might also run the
 autosupport script and take a look at it for anything obvious.
 
 I'm having lots of stability problems with my 411's. I'm not blaming the 411
 yet, just seems odd that I ran for months on a TE110P with a peak of 10-15
 calls, and now my box kernel panics each time it hits the same load.
 Granted, its got 4 PRI's now, but still only 10-15 calls will kill it. Does
 seem to kill the echo as long as the zttest comes back clean.

FWIW, we just fixed a somewhat major bug in the driver for the quad span cards
that caused kernel panics on the card when you would receive a digit on an
unconfigured channel.  You might try updating against origcvs.digium.com
(one of the cvs mirrors doesn't seem to be mirroring correctly) and see if
that fixes your panics.

-- 
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Re: [Asterisk-Users] TE411P problem

[Matt Fredrickson]

On Fri, Aug 12, 2005 at 09:03:42PM -0500, Tim Connolly wrote:
 I'm having lots of stability problems with my 411's. I'm not blaming the 411
 yet, just seems odd that I ran for months on a TE110P with a peak of 10-15
 calls, and now my box kernel panics each time it hits the same load.
 Granted, its got 4 PRI's now, but still only 10-15 calls will kill it. Does
 seem to kill the echo as long as the zttest comes back clean.

Right now, we're trying to work out some issues that we have seen
in customers machines similar to this.  If the kernel panics are caused by the
TE411P driver (wct4xxp) then you might want to try calling Digium tech support
about this so that we can help you get it fixed.

-- 
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Re: [Asterisk-Users] PRI dropped calls w/ asterisk dropped between pstn norstar

[Matt Fredrickson]

On Wed, Aug 10, 2005 at 08:50:50PM -0400, Gary Reuter wrote:
 I dropped an asterisk server with a TE405P between a Norstar Meridian
 PBX and it's PRI PSTN connection.  Everything seemed to work fine
 using a pass-thru-type dialplan configuration... except now we've
 realised that outbound calls to celphones get dropped upon connect,
 but not on every call (almost always the first try, but not the
 second).  All other calls to non-celphones had no problems at all.
 I've had to reconnect the legacy PBX directly to the PSTN because it
 was too much of a problem.
 Asterisk was simply passing calls on one side to the other.  The 'pri
 debug span' and regular logging doesn't appear have anything unusual
 or obvious.  Limited testing could not reproduce the problem when
 calling from a SIP-phone out through the same PRI.
 The only things I can think of are not having 'transfer=no' anywhere
 in my zapata.conf (although none of the docs indicate this applies to
 a PRI), or having set the wrong switch type (set to dms100 when I
 maybe should have set national).
 
 I'm hoping someone else has encountered this and has a 'quick' fix or
 explanation, or can at least suggest what I should be looking for
 specifically in the pri debug output.

If you can post your `pri debug span X` (where X is the span in question)
we might be able to help you a little better.

-- 
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Re: [Asterisk-Users] ISDN DID

[Matt Fredrickson]

On Tue, Aug 09, 2005 at 05:20:15PM -0500, Panitaxx wrote:
 thanks for your response. here is the log of one call:
 
 Enabled debugging on span 1
 
 Asterisk*CLI 
 
  Protocol Discriminator: Q.931 (8)  len=33
  Call Ref: len= 2 (reference 72/0x48) (Originator)
  Message type: SETUP (5)
  [a1]
  Sending Complete (len= 1)
  [04 03 90 90 a3]
  Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer
 capability: 3.1kHz audio (16)
   Ext: 1  Trans mode/rate: 64kbps,
 circuit-mode (16)
   Ext: 1  User information layer 1: A-Law (35)
  [18 03 a9 83 8d]
  Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0,
 Exclusive Dchan: 0
 ChanSel: Reserved
Ext: 1  Coding: 0   Number Specified   Channel Type: 3
Ext: 1  Channel: 13 ]
  [1e 02 84 83]
  Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard
 (0) 0: 0   Location: Public network serving the remote user (4)
Ext: 1  Progress Description: Calling
 equipment is non-ISDN. (3) ]
  [6c 0b 00 83 39 31 35 34 35 31 39 30 30]
  Calling Number (len=13) [ Ext: 0  TON: Unknown Number Type (0)  NPI:
 Unknown Number Plan (0)
Presentation: Presentation allowed of
 network provided number (3) '915451900' ]
 -- Making new call for cr 72
 -- Processing Q.931 Call Setup
 -- Processing IE 161 (cs0, Sending Complete)
 -- Processing IE 4 (cs0, Bearer Capability)
 -- Processing IE 24 (cs0, Channel Identification)
 -- Processing IE 30 (cs0, Progress Indicator)
 -- Processing IE 108 (cs0, C
 alling Party Number)
 -- Going to extension s|1 because of Complete received

I don't see a number specified here.  Do you have overlapdial=yes enabled?

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Re: [Asterisk-Users] PRI problem

[Matt Fredrickson]

On Tue, Jul 12, 2005 at 10:59:42PM +0800, matt001 wrote:
 currently we are able to use our USA sip phone to conenct into the E1 box, 
 but still unable to dial out to chinese phone numbers. They said from their 
 ISDN switch console, it shows D channel not connected to the voip server yet.
 
 here si the sip debug msg, we got a Message type: DISCONNECT (69) and unable 
 to dial any numbers.
 
 Jul 12 12:56:26 WARNING[1523]: chan_zap.c:1931 pri_find_dchan: No D-channels 
 available!  Using Primary on channel anyway 16!
 -- Making new call for cr 32771
  Protocol Discriminator: Q.931 (8)  len=47
  Call Ref: len= 2 (reference 3/0x3) (Originator)
  Message type: SETUP (5)
  [04 03 80 90 a3]
  Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer 
  capability: Speech (0)
   Ext: 1  Trans mode/rate: 64kbps, circuit-mode 
  (16)
   Ext: 1  User information layer 1: A-Law (35)
  [18 04 e9 81 83 81]
  Channel ID (len= 6) [ Ext: 1  IntID: Explicit, PRI Spare: 0, Exclusive 
  Dchan: 0
 ChanSel: Reserved
Ext: 1  DS1 Identifier: 1
Ext: 1  Coding: 0   Number Specified   Channel Type: 3
Ext: 1  Channel: 1 ]
  [28 08 4a 69 61 6e 20 4c 69 75]
  Display (len= 8) [ Jian Liu ]
  [6c 04 21 81 31 30]
  Calling Number (len= 6) [ Ext: 0  TON: National Number (2)  NPI: 
  ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation permitted, user number 
  passed network screening (1) '10' ]
  [70 0d a1 30 31 33 39 30 31 30 33 35 34 33 36]
  Called Number (len=15) [ Ext: 1  TON: National Number (2)  NPI: 
  ISDN/Telephony Numbering Plan (E.164/E.163) (1) '013901035436' ]
 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Initiated, peerstate 
 Overlap sending
  Protocol Discriminator: Q.931 (8)  len=9
  Call Ref: len= 2 (reference 3/0x3) (Originator)
  Message type: DISCONNECT (69)
  [08 02 81 90]
  Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   Location: 
  Private network serving the local user (1)
   Ext: 1  Cause: Normal Clearing (16), class = Normal Event 
  (1) ]
 NEW_HANGUP DEBUG: Destroying the call, ourstate Disconnect Request, peerstate 
 Disconnect Indication

Have you played with the overlapdial settings any to see if that helps?

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Re: [Asterisk-Users] ISDN PRI No Audio

[Matt Fredrickson]

On Wed, Jul 06, 2005 at 05:24:06PM -0500, Andy Brezinsky wrote:
 [Span 3 D-Channel 0] Channel ID (len= 6) [ Ext: 1  IntID: Explicit, PRI 
 Spare: 0, Exclusive Dchan: 0
 [Span 3 D-Channel 0]ChanSel: Reserved
 [Span 3 D-Channel 0]   Ext: 1  DS1 Identifier: 2
 [Span 3 D-Channel 0]   Ext: 1  Coding: 0   Number 
 Specified   Channel Type: 3
 [Span 3 D-Channel 0]   Ext: 1  Channel: 24 ]
  [1e 02 81 83]

Make sure that your span map is correctly done.  It looks like the destination
b channel is channel 24 on span 2.  Make sure that you have your DS1s plugged in
in the correct order and it's using the right DS1 for this.  The channel that 
chan_zap
picked for that was 48, so make sure also that they are not numbering the DS1 
identifier
beginning with 0.  You might want to see if you need to adjust your spanmap and 
related
config in zapata.conf for all of this.

-- 
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Re: [Asterisk-Users] ISDN PRI No Audio

[Matt Fredrickson]

On Wed, Jul 06, 2005 at 12:42:52PM -0500, Andy Brezinsky wrote:
 Console Output:
-- Accepting call from '414944' to '80094042XX' on channel 2/24, 
 span 4
-- Executing Wait(Zap/48-1, 3) in new stack
-- Executing Answer(Zap/48-1, ) in new stack
-- Executing Playback(Zap/48-1, tt-monkeys) in new stack
-- Playing 'tt-monkeys' (language 'en')
-- Executing Read(Zap/48-1, TEST||2) in new stack
-- Accepting a maximum of 2 digits.
-- User entered nothing.
-- Executing SayDigits(Zap/48-1, ) in new stack
-- Playing 'digits/1' (language 'en')
-- Playing 'digits/1' (language 'en')
-- Playing 'digits/1' (language 'en')
-- Playing 'digits/1' (language 'en')
-- Executing Hangup(Zap/48-1, ) in new stack
  == Spawn extension (default, 80094042XX, 6) exited non-zero on 'Zap/48-1'
-- Hungup 'Zap/48-1'
 
 The Problem:
 No audio and no DTMF tones are passed.  We cannot hear the test audio, 
 we cannot send digits back and we cannot hear the digits being said.  
 GBLX setup a tap and said we were not sending them any audio at all so 
 we're fairly certain this problem is on our end and not theirs. 
 
 We're at a loss here and can't really figure out what's wrong.  Can 
 someone provide some insight into this problem?

Hmm... I wonder if this has anything to do with some of those NFAS changes that
I made a month or two ago.  Make sure that you're running latest CVS-HEAD on 
asterisk
and libpri and see if it still does it.  If it does, can you provide the output 
that 
you gave above along with the `pri debug span X` that goes along with it?

-- 
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Re: [Asterisk-Users] Gizmo: Skype done right?

[Matt Fredrickson]

On Thu, Jun 30, 2005 at 01:31:55PM -0700, Jerry Glomph Black wrote:
 I've just submitted this as a Slashdot story, too.
 I have absolutely no connection with any of the principals, I just think they
 are doing the right thing.   This could have a major impact on the Asterisk
 community, and VoIP usage in general.
 
 Michael Robertson, of mp3.com fame, has been battling for open standards in 
 the
 IP telephony world, in addition to his better-known Lindows (now Linspire, at
 http://www.linspire.com) venture to promote Linux on the desktop.  His
 sipphone.com VoIP operation works great for me, but Michael has been long
 concerned about the totally closed and proprietary nature of Skype (as well 
 as a
 lot of the misleading hype surrounding it).
 
 Today his crew released Gizmo (at http://www.gizmoproject.com) (a tentative
 name until a better one is found) which has the main benefits of Skype, PLUS 
 it
 is layered upon SIP, DUNDI, and the existing sipphone.com infrastructure,
  ^^^

Looks like they already messed up... If they're going to redo all of this 
anyway,
they might as well use a protocol like IAX where you don't have NAT problems.

Matthew Fredrickson
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Re: [Asterisk-Users] Setting Caller ID after Dial

[Matt Fredrickson]

On Wed, Jun 29, 2005 at 12:37:34PM -0500, Eric Wieling aka ManxPower wrote:
 Bryce Chidester wrote:
 The CallerID that is seen by others on calls originating from your  PRI 
 is set by your PRI provider; you have no control from Asterisk  about 
 this as it gets overridden by the provider. You must contact  your 
 carrier and ask them to set the CallerID for all PRI lines to  the 
 desired name/number.
 
 Wrong!
 
 Your carrier can allow you to set your own Caller*ID NUMBER.  Some 
 carriers do this by default, some on request, some refuse to do so.
 
 Your carrier lets us set our Caller*ID NUMBER to anything we want.
 
 The Caller*ID NAME you have no control over.  It will show up as 
 whatever name is associated with the number.

Some carriers will also let your set your name too ;-)

Matthew Fredrickson
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Re: [Asterisk-Users] INBAND DTMF G729 ASTERISK

[Matt Fredrickson]

On Fri, Jun 24, 2005 at 11:59:51AM -0400, Julio Arruda wrote:
 Andrew Kohlsmith wrote:
 On Friday 24 June 2005 10:58, [EMAIL PROTECTED] wrote:
 
 But there are some products that supports DTMF inband on G729. Ok, it will
 not work in most cases(like everyone told) but why Asterisk dont support
 it? Is this hardcoded, or is possible to try it out?
 
 
 Asterisk can do it too, it's just not reliable on any platform.  Set 
 dtmfmode=inband and use the g729 codec; that's all there is to it.  You 
 will be disappointed though.
 
 I think that IAX, as one example, won't allow this ? Have a faint memory 
 of some error message when trying it (maybe was ILBC+iax ?)

IAX always sends DTMF out of band.  That's why SIP sucks.  Too many options,
too many ways to mess something up.

Matthew Fredrickson
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Re: [Asterisk-Users] OT: MAX TNT and PRI calling name (CNAM) facility message

[Matt Fredrickson]

On Thu, Jun 23, 2005 at 12:20:34AM -0600, Kevin Blackham wrote:
 Does anyone have a MAX/APX with working ingress PRI calling name?
 
 I recently acquired a MAX TNT on the cheap and it's integrating fine
 except for one thing.  In the 11.0.0 release notes, it is stated that
 ISDN calling name will, if present and permitted by presentation
 flags, be added to the From: and Remote-Party-ID: headers of the
 INVITE.  I'm not able to make this happen.  Pcap captures show it is
 indeed in neither header, and I suspect the MAX is sending the INVITE
 before it receives this data.  Debug traces show it does receive the
 message, but due to limitations of the CLI, I cannot correlate whether
 it's received before or after the INVITE is dispatched.  It works
 great direct to Asterisk (of course) via TE410P on the same NI-2
 spans.
 
 My FACILITY message that contains the CNAM wanders in from 100 to
 400ms after the initial SETUP.  I can't seem to find any way to get
 the MAX to stall for a half-second before invoking the INVITE (if
 that's even the issue).  Is my provider too slow?  Is there another
 valid way for CNAM to be provided during the SETUP message, assuming
 my provider can stall the call setup until the SS7 query is returned?
 (google for Q.931 docs not helping me much there either)

That's one of the (many) ways that caller name is provided.  In fact, it's
pretty much the most common way that I've seen for ISDN PRI.  I don't
know if you're provider supports it, but sometimes you can get it in the
SETUP message.  I'm not sure what level of control they have though.

Matthew Fredrickson
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Re: [Asterisk-Users] INBAND DTMF G729 ASTERISK

[Matt Fredrickson]

On Thu, Jun 23, 2005 at 03:19:06PM -0400, [EMAIL PROTECTED] wrote:
 Why don't Asterisk support inband DTMF with G729? Is  there a way to do
 that!?

I think the answer to that is quite obvious;  G.729 is a lossy voice oriented
compression technique.  Any inband DTMF data will be essentially useless by
the time it gets encoded.

 Are you using RFC2833? Doesn't it a security hole?

Sorry, don't know if I quite get your point here.  Care to expound on what
you're talking about?

Matthew Fredrickson
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Re: [Asterisk-Users] New Asterisk Implementation

[Matt Fredrickson]

On Wed, Jun 22, 2005 at 05:32:22PM +0200, harry gaillac wrote:
 look at ser projects:
 asterisk is limited to 250 channels 

What kind of crack are you smoking?  There are people that have set
up more than 250 channel systems.

Matthew Fredrickson
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Re: [Asterisk-Users] New Asterisk Implementation

[Matt Fredrickson]

On Wed, Jun 22, 2005 at 03:56:55PM -0500, Andrew Latham wrote:
 He is talking about ZAP channels. This is correct. No one should want
 to use that many ZAP channels.

You can use more than 250 zap channels.  That limitation has been removed
a long time ago.  chan_zap.c doesn't have to use the device files directly
for channels above the limit of /dev/zap/ device entries.

 
 On 6/22/05, Matt Fredrickson [EMAIL PROTECTED] wrote:
  On Wed, Jun 22, 2005 at 05:32:22PM +0200, harry gaillac wrote:
   look at ser projects:
   asterisk is limited to 250 channels
  
  What kind of crack are you smoking?  There are people that have set
  up more than 250 channel systems.
  
  Matthew Fredrickson
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 -- 
 sig
 Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)
 WWW: http://lathama.com
 Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED]
 If any of the above are down we have bigger problems than my email!
 /sig
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Re: [Asterisk-Users] Clicks in audio with TE100P PRI

[Matt Fredrickson]

On Thu, Jun 09, 2005 at 12:51:30AM -0300, Alejandro G wrote:
 I should tell you that the TE100P is connected to another E1 board (not a
 live E1) from Natural Microsystems which acts as a gateway to PSTN. This
 board works as a PRI master but I don't think that this could be the problem
 as long as using other phones or in LAN it works perfectly and the voice is
 clear with no clicks o sound looses.

Do you find that these clicks occur at the same time concurrently with
increased hard drive activity?

If so, and if you have an IDE hardrive, try doing a `hdparm -u1 
/dev/yourhardrivedevice`

Matthew Fredrickson
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Re: [Asterisk-Users] SS7

[Matt Fredrickson]

On Tue, Jun 07, 2005 at 11:30:27AM -0400, Matt wrote:
 Hi,
 Has anyone used the SS7 link from Digium?  If so, how did it work for
 you?   Any issues?  Anything to be aware of?  Do I just need a T1 card
 like the PRI card I have now from Digium?

FYI, I don't think that Digium has an SS7 link :-)  You might want to check
with Markku.  His email address is at the bottom of the page on the wiki.

Matthew Fredrickson
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Re: [Asterisk-Users] TE410P

[Matt Fredrickson]

On Tue, Jun 07, 2005 at 07:18:32PM +0100, Tony Hoyle wrote:
 Juan Pablo Abuyeres wrote:
 lspci -v says:
 02:08.0 Communication controller: Unknown device d161:0410 (rev 02)
 Flags: bus master, medium devsel, latency 32, IRQ 52
 Memory at dd20 (32-bit, non-prefetchable) [size=128]
 
 There's nothing in the PCI device database (www.pcidatabase.com) for 
 manufacturer 0xd161... possibly you have a broken card?
 
 A TE410P would be 10EE:0314

That's the new digium vendor id (digi ~= d161)

Matthew Fredrickson
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Re: [Asterisk-Users] Philips - [QSIG] - Alcatel - [H323] - Asterisk -[SIP] - Users.

[Matt Fredrickson]

On Wed, May 04, 2005 at 04:26:01PM +0200, Andreas Sikkema wrote:
 DT wrote:
 
  Firstly we have to connect our Asterisk system to a Philips PBX
  throught QSIG protocol (interfaces S0), but we doesn't find any
  documentation about the support of QSIG and S0 interfaces by
  Asterisk.   
  
  [PSTN/ISDN] --- Philips -[QSIG over S0]- Asterisk -[SIP]-
  Final users. 
  
  Is it possible?
  does Asterisk support QSIG and S0 interfaces?

Asterisk has rudimentary support for Q.SIG.  Call setup and teardown
works, along with a few of the the more basic features.  Check the
sample zapata.conf from CVS-HEAD.

Matthew Fredrickson
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Re: [Asterisk-Users] Zap/PRI: received AOC-E charging

[Matt Fredrickson]

On Tue, Apr 26, 2005 at 09:04:48AM -0500, Matthew Boehm wrote:
 Trying to make a call via our PRI: (CVS everything,
 CVS-NHEAD-04/23/05-16:08:12)
 
 -- Executing Dial(IAX2/[EMAIL PROTECTED],
 Zap/R2/2815699900|30) in new stack
 -- Requested transfer capability: 0x00 - SPEECH
 -- Called R2/2815699900
 -- Channel 0/19, span 2 got hangup
 -- Channel 0/19, span 2 received AOC-E charging 0 units
 Apr 26 09:06:49 WARNING[10040]: chan_zap.c:7457 zt_pri_error: PRI: Call
 Reference Length not supported: 0
 -- Zap/43-1 is circuit-busy
 -- Hungup 'Zap/43-1'
 
 Any idea on what AOC-E means? Here is a full pri debug:

It looks like something is triggering the new AOC code in libpri.  Are you
sure that this is the complete debug?  I only saw a facility IE for calling 
name,
but I didn't any Advice of Charge info there.

Matthew Fredrickson
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Re: [Asterisk-Users] Local Echo

[Matt Fredrickson]

On Tue, Apr 12, 2005 at 05:43:18PM -0700, Noah Silverman wrote:
 Great suggestion.  I'll try it ASAP.
 
 Where do I get fxotune?

It's in CVS-HEAD zaptel.   You'lll need to use the CVS-HEAD zaptel drivers
as well, since there is a new IOCTL for doing echo tuning.

Matthew Fredrickson
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Re: [Asterisk-Users] Local Echo

[Matt Fredrickson]

On Tue, Apr 12, 2005 at 07:03:26PM -0700, Bashir Ullah - www.Lamsre.Com wrote:
 hi
 
 i did not find fxotune under zapte-1.0.6 , please let me know is it
 different module , need to install seperate, please show me the way , i am
 having same echo problem and finding its solution for mt tdm fxo.

It's in CVS-HEAD zaptel.  You'll need to use the CVS-HEAD drivers as well for
it to work.

Matthew Fredrickson
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Re: [Asterisk-Users] TE110P - NT-Mode ?

[Matt Fredrickson]

On Tue, Apr 12, 2005 at 12:15:02PM +0200, Henry Jensen wrote:
 Hello,
 
 I still try to connect a TE110P card to a TMS2 card in a Siemens HiPath
 3750.
 
 The TMS2 card can be used to connect to an NT (Amtsanschluss)
 or to connect to another S2M-Line (PRI). When connecting to another PRI,
 I can select between CorNet (proprietary), ECMA-QSIG and ISO-QSIG.
 It seems that Asterisk supports none of these protocols.

You should be able to generate/receive calls on Q.SIG (ECMA).  A handful
of the more advanced supplementary services are somewhat supported as well
(mostly data gather type stuff, receive of callername, receive of call diversion
information, etc).  MWI is also supported in libpri, but I haven't written
the Asterisk portion of it yet.

Matthew Fredrickson
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Re: [Asterisk-Users] Local Echo

[Matt Fredrickson]

On Tue, Apr 12, 2005 at 10:16:16AM -0700, Noah Silverman wrote:
 I have a strange echo problem.
 
 When speaking on the phone with someone, I hear MY OWN voice with a
 sever echo.  The other party sounds perfect, and they can hear me
 perfectly.  It is as if only the sidetone has an echo.
 
 I'm running * on a dedicated box, small LAN, and am using 4 FXO cards to
  connect the box to PTSN lines.  My phones are Polycom IP500 SIP phones.
 
 The only echo cancellation stuff that I can find relates to cancelling
 echo between my system and the PTSN lines.  Since the call is perfect,
 I don't see how this would apply.
 
 Any suggestions??

If you're using a TDM card, you might see if the fxotune program will help.

It does impedance tuning of the card and finds the line impedance that has
the lowest mean power (i.e. least echo).  I've been working on it for a while
and some people have had some success with it.

Matthew Fredrickson
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