Re: [asterisk-users] buffer-issue when piping live-streams into musiconhold

Michael Kamleitner wrote:
> thx a lot russel...your hack actually works!! :)

Awesome. :)

> Meanwhile I've found something about the musiconhold-conf-option
> "cachertclasses", which might help in starting a separate instance for every
> caller. however, that didn't really work for me... probably this option only
> works for mode=files?!
>
> http://www.asterisk.org/doxygen/trunk/Config_moh.html
> http://lists.digium.com/pipermail/asterisk-commits/2007-November/017911.html

Well, that option only exists in Asterisk trunk, and is only relevant when using
realtime for music on hold. I assume you're probably using one of the released
versions of Asterisk, so this wouldn't be available.

> anyway, thx a lot for your suggestions :)

You're quite welcome. I'm glad I could help out.

--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.

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Re: [asterisk-users] IAX Trunk between two Asterisks

This is my configuration in the extensions.conf,
iax.conf at Site A and Site B, so anyone can help why
the call refused?

Site A:

[IPLink]
type=friend
context=IPLinkIncoming
host=192.168.2.3
usename=IPLink
secret=password
canreinvite=no
nat=no

[SiteBInternal]

exten => _2XX,1,Dial(IAX2/${EXTEN}@IPLink)
exten => _2XX,2,Playback(vm-nobodyavail)
exten => _2XX,3,Hangup()
exten => _2XX,102,Playback(tt-allbusy)
exten => _2XX,103,Hangup()

[IPLinkIncoming]

include => SiteBInternal
include => SiteBExternal

And at Site B:

[IPLink]
type=friend
context=IPLinkIncoming
host=192.168.2.2
usename=IPLink
secret=password
canreinvite=no
nat=no

[SiteAInternal]

exten => _2XX,1,Dial(IAX2/${EXTEN}@IPLink)
exten => _2XX,2,Playback(vm-nobodyavail)
exten => _2XX,3,Hangup()
exten => _2XX,102,Playback(tt-allbusy)
exten => _2XX,103,Hangup()

[IPLinkIncoming]

include => SiteAInternal
include => SiteAExternal

Regards
Bilal

------------------

> Hi All;
>
> I did an IP Trunk using IAX between two Asterisk
> boxes, now Asterisk A can send a call for B but B
> refuse it. The IAX type was configured to be
"friend"
> in the iax.con for Asterisk A and B, is there any
> thing else need to be done to let B accept the call
> from A?
>
> Also, I used an static IP address for the host when
I
> configured the iax client in the iax.conf file.
>
> Any help?
> Regards
> Bilal
>

I used to see this problem when I used to use IAX2.
Sometimes it would
just
go away. I seem to remember using insecure=very to
get it working but
I may
be wrong.

Anyways, post the relevant parts of your IAX2 confs
from both boxes and
someone might be able to spot something right off the
bat.

Thanks,
Steve Totaro

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Re: [asterisk-users] IMAP client in asterisk not trying to contact IMAP server

On Jan 17, 2008 7:55 AM, KodaK <sakodak@gmail.com> wrote:
>
> Thanks, if that was in any of the docs I just completely glossed over
> it. I'll give it
> a shot.

Yes, I skipped over that in the docs. I'm good at that.

Thanks for the help.

I've also written up a quickie how-to on how to enable this on a
trixbox system. Don't know how helpful it is, but it's there.

http://www.trixbox.org/wiki/trixbox-imap

--J(K)

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Re: [asterisk-users] Asterisk desktop tools for OS X

Thanks for your response guys. There are still some issues with the
code (Svn on SourceForge). I am working on getting these fixed up and
will post a message when its ready for download.

I will yell out if I need some Asterisk/Cocoa help. Thanks a lot.

On Jan 18, 2008 7:19 AM, Adrià Vidal <adriavidal@gmail.com> wrote:
> I'm interested too Devraj, please send a copy of if possible to try it.
> Thanks.
>
>
>
> On Jan 17, 2008 12:25 PM, Devraj Mukherjee <devraj@gmail.com> wrote:
> >
> >
> >
> > Hi everyone,
> >
> > I have been long working on a project ( http://asterisktools.org, to be
> > released under GPL) that aims to provide desktop tools for Macs. I am
> > finally getting to the release stages of this application and hope to
> > have an early BETA available next weekend.
> >
> > If there is anybody who is interested in this tool, please send me an
> > email as I am looking for people who can test the application for me
> > before we make a final release.
> >
> > The code is already available via SVN and there are some really cool
> > and thoughtful features.
> >
> > Thanks a lot.
> >
> > --
> > "I never look back darling, it distracts from the now", Edna Mode (The
> > Incredibles)
> >
> >
> > _______________________________________________
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >

http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
>
> --
> --
> Adrià Vidal
> adriavidal@gmail.com
> _______________________________________________
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>

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

--
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Incredibles)

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Re: [asterisk-users] buffer-issue when piping live-streams into musiconhold

thx a lot russel...your hack actually works!! :)

Meanwhile I've found something about the musiconhold-conf-option "cachertclasses", which might help in starting a separate instance for every caller. however, that didn't really work for me... probably this option only works for mode=files?!

http://www.asterisk.org/doxygen/trunk/Config_moh.html
http://lists.digium.com/pipermail/asterisk-commits/2007-November/017911.html

anyway, thx a lot for your suggestions :)

regards,
michael

On Jan 17, 2008 9:52 PM, Russell Bryant < russell@digium.com> wrote:

Michael Kamleitner wrote:
> 10:00 I'm calling the pbx, musiconhold starts correctly to play the
> live-stream (almost live, with very small delay) - that's OK.
> 10:01 I hangup.
>
> -- than I pause for 20 min --
>
> 10:20 I'm calling a second time. However moh now doesn't stream live, but
> starts to continue playing the stream from 10:01. This goes on for about
> 30secs, then the replay stops for a second and continues at the correct
> position (once again, rather "live"). along I get this message at the
> console:

<snip>

> musiconhold.conf:
>
> [default]
> mode=custom
> application=/etc/asterisk/mohstream.sh
>
> mohstream.sh
>
> #!/bin/bash
> /usr/bin/wget -q -O - http://my.stream.com:8000 | /usr/bin/madplay -Q -z -o
> raw:- --mono -R 8000 -a -12 -

Most players don't work quite correctly with Asterisk MOH.  For it to work the
way you expect, the player you are using must throw away the audio when Asterisk
isn't currently reading from the stream.  There was a magic version of mpg123
(0.59r IIRC) that did that, and that is why it was the recommended version.

If you're reading from a raw TCP stream, then you can use the small streamplayer
utility included with Asterisk.  Otherwise, I don't really have a good
suggestion for you right now.  I suppose that you could use some sort of hack to
ensure that music on hold is always playing so that the stream is being serviced.

extensions.conf:

[moh_hack]

exten => hack,1,Answer
exten => hack,n,StartMusicOnHold(default)
exten => hack,n,While(1)
exten => hack,n,Wait(300)
exten => hack,n,EndWhile()

*CLI> originate Local/hack@moh_hack application Echo

--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.

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--
Mag. Michael Kamleitner
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E-Mail: michael.kamleitner@gmail.com
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Phone: +43 699 116 07 923
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Web: http://www.kamleitner.com

Re: [asterisk-users] Asterisk desktop tools for OS X

Hi Tzafrir,

Yes it does use the Manager Interface. It account does require "call"
level access. That may then result in "umlimited access" to Asterisk
(well to originate calls anyway). However I have made real conscious
efforts to filter messages that are being transmitted over the socket
so the application doesn't listen or talk on behalf of a single
extension.

If this is a concern, is every desktop application that integrates
using the Manager Interface a problem for Asterisk administrators?

Also, what is a way around it then? I see desktop tools for Asterisk
being one of the biggest advantages over traditional PBXes.

On Jan 18, 2008 7:19 AM, Adrià Vidal <adriavidal@gmail.com> wrote:
> I'm interested too Devraj, please send a copy of if possible to try it.
> Thanks.
>
>
>
> On Jan 17, 2008 12:25 PM, Devraj Mukherjee <devraj@gmail.com> wrote:
> >
> >
> >
> > Hi everyone,
> >
> > I have been long working on a project ( http://asterisktools.org, to be
> > released under GPL) that aims to provide desktop tools for Macs. I am
> > finally getting to the release stages of this application and hope to
> > have an early BETA available next weekend.
> >
> > If there is anybody who is interested in this tool, please send me an
> > email as I am looking for people who can test the application for me
> > before we make a final release.
> >
> > The code is already available via SVN and there are some really cool
> > and thoughtful features.
> >
> > Thanks a lot.
> >
> > --
> > "I never look back darling, it distracts from the now", Edna Mode (The
> > Incredibles)
> >
> >
> > _______________________________________________
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >

http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
>
> --
> --
> Adrià Vidal
> adriavidal@gmail.com
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>

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

--
"I never look back darling, it distracts from the now", Edna Mode (The
Incredibles)

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Re: [asterisk-users] Polycom Remotely Cancel Call Forward

Kevin Kiely wrote:
>
> I have a remote user on a Polycom IP Phone who has set call forwarding
> by accident and is away from the phone. Does anyone know of a way to
> remotely un-forward the phone? I tried to reboot the phone but that
> didn't work and removing the mac-phone.cfg caused problems
>
Remove the XML element tag from within mac-phone.cfg that it updated with the forwarding information and then reboot it again.

--
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