[Asterisk-Users] IAXy echo?

Mike Hammett
Mon, 14 Nov 2005 21:24:26 -0000

Will ask them to check the speaker volumes.

Not sure if you meant outside of my case, but in my case it's less than 15
ms.


----
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com


----- Original Message -----
From: <asterisk-users-request@lists.digium.com>
To: <asterisk-users@lists.digium.com>
Sent: Monday, November 14, 2005 9:07 AM
Subject: Asterisk-Users Digest, Vol 16, Issue 104


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> Today's Topics:
>
>   1. Re: Can't create iax channel (Dinesh Nair)
>   2. Re: Can't create iax channel (Dinesh Nair)
>   3. Re: How to check how many G729 codec licenseinstalled
>      (Rich Adamson)
>   4. IAXy echo? (Mike Hammett)
>   5. RE: Snom clients deregistering (Michael Crown)
>   6. Re: Snom clients deregistering (Richard Watson)
>   7. RE: How to check how many G729 codec licenseinstalled
>      (Rich Adamson)
>   8. Re: IAXy echo? (Sergey Okhapkin)
>   9. RE: Snom clients deregistering (The VoIP Connection)
>  10. Re: IAXy echo? (Rich Adamson)
>  11. Re: IAXy echo? (janvb@caselaboratories.com)
>  12. RE: How to check how many G729 codec licenseinstalled (Sean Cook)
>  13. Configure Asterisk to call from softPhone(SIP Channel) to
>      Analog phone(Modem Channel) (ashok)
>  14. RE: Sipura SPA-2002 Double Ring (Rich Adamson)
>  15. OT: Aastra PT 390 Question. (Richard Reina)
>  16. SIP signaling and canreinvite=yes (Damon Estep)
>  17. Re: ISDN card required (Kristof Hardy)
>  18. RE: ISDN card required (Lee Archer)
>  19. Re: Snom clients deregistering (Richard Watson)
>  20. Re: MYSQL issue in UPDATE.. (Tony Mountifield)
>  21. Brooktrout MPAC 1200 card with Asterisk (Stephen Arulraj)
>  22. Maximum Number of SIP Phones Supported By Asterisk (nr k)
>  23. asterisk sample size adjustment (trixter aka Bret McDanel)
>  24. Re: Can't make calls from Asterisk IAX to other IAX
>      (chawki hammoud)
>  25. connect to gateway h323 (Reli Loin)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Mon, 14 Nov 2005 20:36:28 +0800
> From: Dinesh Nair <dinesh@alphaque.com>
> Subject: Re: [Asterisk-Users] Can't create iax channel
> To: waynetg@telkomsa.net, Asterisk Users Mailing List - Non-Commercial
> Discussion <asterisk-users@lists.digium.com>
> Message-ID: <437884CC.90702@alphaque.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
>
>
> On 11/10/05 15:02 Wayne Gemmell said the following:
>> When trying to call from this side to that side I get the following
>>
>>     -- Executing Dial("SIP/301-2d50",
>> "IAX2/wayne:password@homebase.hidden.com/204") in new stack
>> Nov 10 08:37:21 WARNING[30785]: channel.c:455 ast_best_codec: Don't
know
>> any
>> of 0xf800 formats
>> Nov 10 08:37:21 WARNING[30785]: channel.c:455 ast_best_codec: Don't
know
>> any
>> of 0xf800 formats
>> Nov 10 08:37:21 WARNING[30785]: chan_iax2.c:7745 iax2_request: Unable
to
>> create translator path for unknown to ulaw on IAX2/wayne-5
>
> there's your problem right there. what codecs are the SIP peer set to
use
> ?
> apparently, asterisk cant translate between ulaw and the unknown codec.
>
> --
> Regards,                           /\_/\   "All dogs go to
heaven."
> dinesh@alphaque.com                (0 0)    http://www.alphaque.com/
> +==========================----oOO--(_)--OOo----==========================+
> | for a in past present future; do
> |
> |   for b in clients employers associates relatives neighbours pets; do
> |
> |   echo "The opinions here in no way reflect the opinions of my $a
$b."
> |
> | done; done
> |
> +=========================================================================+
>
>
> ------------------------------
>
> Message: 2
> Date: Mon, 14 Nov 2005 20:37:19 +0800
> From: Dinesh Nair <dinesh@alphaque.com>
> Subject: Re: [Asterisk-Users] Can't create iax channel
> To: waynetg@telkomsa.net, Asterisk Users Mailing List - Non-Commercial
> Discussion <asterisk-users@lists.digium.com>
> Message-ID: <437884FF.4050703@alphaque.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
>
>
> On 11/10/05 17:36 Wayne Gemmell said the following:
>> On Thursday 10 November 2005 10:55, Jason Walker wrote:
>>
>>>The statement of zaptel being required is strange...I use IX
trunking
>>>exclusively for my servers. Two of them have no zaptel/Digium
hardware
>>>and
>>>the trunk calls are fine.
>>
>> I don't know where I read it, apparently it is needed for timing or
>> something,
>> could be in the old handbook or hitchikers guide to asterisk as I
havn't
>> got
>> far enough into the new handbook to comment.
>
> IAX trunking works even without digium cards as long as the ztdummy pseudo
> timer module is loaded.
>
> --
> Regards,                           /\_/\   "All dogs go to
heaven."
> dinesh@alphaque.com                (0 0)    http://www.alphaque.com/
> +==========================----oOO--(_)--OOo----==========================+
> | for a in past present future; do
> |
> |   for b in clients employers associates relatives neighbours pets; do
> |
> |   echo "The opinions here in no way reflect the opinions of my $a
$b."
> |
> | done; done
> |
> +=========================================================================+
>
>
> ------------------------------
>
> Message: 3
> Date: Mon, 14 Nov 2005 07:18:39 -0600
> From: Rich Adamson  <radamson@routers.com>
> Subject: Re: [Asterisk-Users] How to check how many G729 codec
> licenseinstalled
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users@lists.digium.com>
> Message-ID: <Chameleon.1131974327.adar0@vegas>
> Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-1
>
> show g729
>
> ------------------------
>  From: Mark Quitoriano <markquitoriano@gmail.com>
>  Subject: Re: [Asterisk-Users] How to check how many G729 codec
> licenseinstalled
>  Date: Mon, 14 Nov 2005 19:13:29 +0800
>  To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users@lists.digium.com>
>
>
>> how can i check how many g729 are being used right now?
>>
>> On 11/13/05, Gentian Bajraktari <g.bajraktari@afb.net.al > wrote:
>>
>>     Yes.
>>
>>     ----- Original Message -----
>>     From: "Angelito Manansala" <
lito@voicefidelity.net>
>>     To: "Asterisk Users Mailing List - Non-Commercial
Discussion"
>>     <asterisk-users@lists.digium.com >
>>     Sent: Sunday, November 13, 2005 1:23 PM
>>     Subject: Re: [Asterisk-Users] How to check how many G729 codec
>>     licenseinstalled
>>
>>     >         g723   gsm  ulaw  alaw  g726 adpcm  slin lpc10  g729
speex
>> ilbc
>>     >
>> 23     -     -     -     -     -     -     -     -     -     -     -
>>     >    gsm     -     -     3     3     4     3     2     9     -
-
>> 131
>>     >   ulaw     -     5     -     1     3     2     1     8     -
-
>> 130
>>     >   alaw     -     5     1     -     3     2     1     8     -
-
>> 130
>>     >   g726     -     6     3     3     -     3     2     9     -
-
>> 131
>>     >  adpcm     -     5     2     2     3     -     1     8     -
-
>> 130
>>     >   slin     -     4     1     1     2     1     -     7     -
-
>> 129
>>     >  lpc10     -     8     5     5     6     5     4     -     -
-
>> 133
>>     >
>> 29     -     -     -     -     -     -     -     -     -     -     -
>>     >
>> ex     -     -     -     -     -     -     -     -     -     -     -
>>     >   ilbc     -     9     6     6     7     6     5
>> 2     -     -     -
>>     >
>>     >
>>     > this means i have no g729 codec installed..
>>     >
>>     > thanks guys!
>>     >
>>     > :p
>>     >
>>     >
>>     > On 11/13/05, Gentian Bajraktari <
g.bajraktari@afb.net.al> wrote:
>>     >> Do:
>>     >> *CLI> show translations
>>     >>
>>     >> If you see - (lines) on the G729 row/columns than you do
not have
>> any
>>     >> G729
>>     >> support.
>>     >>
>>     >>
>>     >> RG.
>>     >>
>>     >> ----- Original Message -----
>>     >> From: "Sahil Gupta"
<sgupta@voicevalley.com.au>
>>     >> To: "Asterisk Users Mailing List - Non-Commercial
Discussion"
>>     >> <asterisk-users@lists.digium.com>
>>     >> Sent: Sunday, November 13, 2005 1:03 PM
>>     >> Subject: Re: [Asterisk-Users] How to check how many G729
codec
>>     >> licenseinstalled
>>     >>
>>     >>
>>     >> > Right :)
>>     >> >
>>     >> > Regards,
>>     >> >
>>     >> >
>>     >> > Sahil Gupta
>>     >> > VoiceValley
>>     >> >
>>     >> > On Sun, 13 Nov 2005, Angelito Manansala wrote:
>>     >> >
>>     >> >> *CLI> show g729
>>     >> >> No such command 'show g729' (type
'help' for help)
>>     >> >>
>>     >> >> this means i have no g729 codec installed, right?
>>     >> >>
>>     >> >> On 11/13/05, Zafer Khodr
<khodrz@optusnet.com.au> wrote:
>>     >> >>> That's easy...
>>     >> >>> Just go into asterisk cli and type  "
show g729  "
>>     >> >>> It will tell you how many are active and how
many you have in
>> total
>>     >> >>>
>>     >> >>>
>>     >> >>> Regards
>>     >> >>> Zafer
>>     >> >>>
>>     >> >>> -----Original Message-----
>>     >> >>> From: asterisk-users-bounces@lists.digium.com
>>     >> >>> [mailto:
asterisk-users-bounces@lists.digium.com] On Behalf Of
>>     >> >>> Angelito
>>     >> >>> Manansala
>>     >> >>> Sent: Sunday, 13 November 2005 10:31 PM
>>     >> >>> To: asterisk-users@lists.digium.com
>>     >> >>> Subject: [Asterisk-Users] How to check how
many G729 codec
>> license
>>     >> >>> installed
>>     >> >>>
>>     >> >>> Guys, is the any CLI commands or info files
where you can
>> check how
>>     >> >>> many g729 codec
>>     >> >>> license installed.
>>     >> >>>
>>     >> >>>
>>     >> >>> Regards,
>>     >> >>> Lito
>>     >> >>>
_______________________________________________
>>     >> >>> --Bandwidth and Colocation sponsored by
Easynews.com --
>>     >> >>>
>>     >> >>> Asterisk-Users mailing list
>>     >> >>> Asterisk-Users@lists.digium.com
>>     >> >>>
http://lists.digium.com/mailman/listinfo/asterisk-users
>>     >> >>> To UNSUBSCRIBE or update options visit:
>>     >> >>>
http://lists.digium.com/mailman/listinfo/asterisk-users
>>     >> >>>
>>     >> >>>
>>     >> >>>
>>     >> >>>
_______________________________________________
>>     >> >>> --Bandwidth and Colocation sponsored by
Easynews.com --
>>     >> >>>
>>     >> >>> Asterisk-Users mailing list
>>     >> >>> Asterisk-Users@lists.digium.com
>>     >> >>>
http://lists.digium.com/mailman/listinfo/asterisk-users
>>     >> >>> To UNSUBSCRIBE or update options visit:
>>     >> >>>
http://lists.digium.com/mailman/listinfo/asterisk-users
>>     >> >>>
>>     >> >>
>>     >> >>
>>     >> >> --
>>     >> >> Best Regards,
>>     >> >> Angelito Manansala
>>     >> >> www.voicefidelity.net
>>     >> >> Mobile: +639175425807
>>     >> >> DID: (+63) 44 7906770
>>     >> >> msn: bulcrack@elitemail.org
>>     >> >> skype: bulcrack
>>     >> >> _______________________________________________
>>     >> >> --Bandwidth and Colocation sponsored by
Easynews.com --
>>     >> >>
>>     >> >> Asterisk-Users mailing list
>>     >> >> Asterisk-Users@lists.digium.com
>>     >> >>
http://lists.digium.com/mailman/listinfo/asterisk-users
>>     >> >> To UNSUBSCRIBE or update options visit:
>>     >> >>
http://lists.digium.com/mailman/listinfo/asterisk-users
>>     >> >>
>>     >> > _______________________________________________
>>     >> > --Bandwidth and Colocation sponsored by Easynews.com
--
>>     >> >
>>     >> > Asterisk-Users mailing list
>>     >> > Asterisk-Users@lists.digium.com
>>     >> >
http://lists.digium.com/mailman/listinfo/asterisk-users
>>     >> > To UNSUBSCRIBE or update options visit:
>>     >> >
http://lists.digium.com/mailman/listinfo/asterisk-users
>>     >> >
>>     >> >
>>     >>
>>     >>
>>     >> _______________________________________________
>>     >> --Bandwidth and Colocation sponsored by Easynews.com --
>>     >>
>>     >> Asterisk-Users mailing list
>>     >> Asterisk-Users@lists.digium.com
>>     >> http://lists.digium.com/mailman/listinfo/asterisk-users
>>     >> To UNSUBSCRIBE or update options visit:
>>     >>
http://lists.digium.com/mailman/listinfo/asterisk-users
>>     >>
>>     >
>>     >
>>     > --
>>     > Best Regards,
>>     > Angelito Manansala
>>     > www.voicefidelity.net
>>     > Mobile: +639175425807
>>     > DID: (+63) 44 7906770
>>     > msn: bulcrack@elitemail.org
>>     > skype: bulcrack
>>     > _______________________________________________
>>     > --Bandwidth and Colocation sponsored by Easynews.com --
>>     >
>>     > Asterisk-Users mailing list
>>     > Asterisk-Users@lists.digium.com
>>     > http://lists.digium.com/mailman/listinfo/asterisk-users
>>     > To UNSUBSCRIBE or update options visit:
>>     >   http://lists.digium.com/mailman/listinfo/asterisk-users
>>     >
>>     >
>>
>>     _______________________________________________
>>     --Bandwidth and Colocation sponsored by Easynews.com --
>>
>>     Asterisk-Users mailing list
>>     Asterisk-Users@lists.digium.com
>>     http://lists.digium.com/mailman/listinfo/asterisk-users
>>     To UNSUBSCRIBE or update options visit:
>>        http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>> --
>> Regards,
>> Mark Quitoriano, CCNA
>> http://www.atamanetworks.com
>>
>> Fan the flame...
>> http://www.spreadfirefox.com/?q=user/register&r=19441
> ---------------End of Original Message-----------------
>
>
>
>
> ------------------------------
>
> Message: 4
> Date: Mon, 14 Nov 2005 07:21:43 -0600
> From: "Mike Hammett" <asterisk-users@ics-il.net>
> Subject: [Asterisk-Users] IAXy echo?
> To: <asterisk-users@lists.digium.com>
> Message-ID: <015e01c5e91e$6a3d2550$fa01010a@ICS.local>
> Content-Type: text/plain; charset="iso-8859-1"
>
> I've got two customers on the same broadband provider.  Same Asterisk
box
> on my end.  Same CLEC.
>
> One has an IAXy and the other has an Asterisk box with an array of devices
> (Grandstream, Cisco, ATCOM, xten, etc.).
>
> The people behind the Asterisk box have had no audio quality issues.  The
> person with the IAXy often encounters an echo.  The echo is only heard on
> the remote side and it only contains the remote caller's voice.  This
echo
> has been heard with the remote side being varying LECs.  The echo is not
> always there.  I'd almost say that the echo is not there more than it
is.
>
> Troubleshooting next step?
>
> I haven't changed out the IAXy because I don't have any other ATAs
to put
> in place.
>
>
> ----
> Mike Hammett
> Intelligent Computing Solutions
> http://www.ics-il.com
>
> -------------- next part --------------
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>
> ------------------------------
>
> Message: 5
> Date: Mon, 14 Nov 2005 08:26:42 -0500
> From: "Michael Crown" <mike@thevoipconnection.com>
> Subject: RE: [Asterisk-Users] Snom clients deregistering
> To: "'Asterisk Users Mailing List - Non-Commercial
Discussion'"
> <asterisk-users@lists.digium.com>
> Message-ID: <20051114132640.4C398466F@lists.digium.com>
> Content-Type: text/plain; charset="us-ascii"
>
> Does the phone ocasionally prompt the user for a password? -Mike
>
>> -----Original Message-----
>> From: Richard Watson [mailto:richard@openia.com]
>> Sent: Monday, November 14, 2005 5:00 AM
>> To: asterisk-users@lists.digium.com
>> Subject: [Asterisk-Users] Snom clients deregistering
>>
>> -----BEGIN PGP SIGNED MESSAGE-----
>> Hash: SHA1
>>
>> Hi all,
>>
>> I have a server currently running Asterisk 1.0.7 placed out
>> in the wild (i.e. not behind NAT).
>>
>> I have groups of sip clients all behind various NAT firewalls
>> (mainly adsl routers).
>>
>> Up to now I've mainly used Sipuras and not had any serious
problems.
>> Recently I've been experimenting with Snom phones and I have
>> encountered  problems where the Snoms register fine initially
>> but after a while (which could be anything from 2minutes to
>> 45 minutes) they lose their registration. Sample snom
>> configuration in sip.conf follows:
>>
>> [888120]
>> type=friend
>> username=888120
>> mailbox=888120
>> canreinvite=no
>> nat=yes
>> secret=secret
>> host=dynamic
>> qualify=yes
>> context=sipdemo
>> subscribecontext=sipdemo
>>
>> I've experimented with several different adsl routers and was
>> surprised at the difference this can make, however the
>> problem is still there to a greater or lesser extent.
>>
>> I've also tried using a Stun server following recommendation here:
>>
>> http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_aud
>> io_asterisk.html
>>
>> Again this makes a difference, but doesn't entirely solve the
>> problem - there are still occasions where the Snom is
>> unreachable or unknown.
>>
>> The implication seems to be that if asterisk does not send
>> keepalives often enough then the way through the nat is lost.
>>
>> I've also tried lowering the expiry time of the asterisk
>> sessions (in increments down to 30 seconds) in the hope that
>> it would result in more activity and keep the firewall open,
>> but it didn't help.
>>
>> Another strange factor is using the BLF on snoms - the
>> situation seems to be worse with those enabled, but that
>> might not be relevant.
>>
>> So I guess I have a few questions:
>>
>> 1) Has anyone had this happen before and what, if any, was
>> the solution?
>>
>> 2) How do I increase the frequency with which asterisk sends
>> keepalives?
>>
>> 3) Does SER handle this better - would placing this outside
>> the NAT help handle connections from inside?
>>
>> 4) Do newer versions of asterisk handle this better?
>>
>> 5) Any other suggestions?
>>
>> TIA.
>>
>> - --
>> Richard Watson
>> -----BEGIN PGP SIGNATURE-----
>> Version: GnuPG v1.4.1 (GNU/Linux)
>> Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
>>
>> iD8DBQFDeGAzP05lUVhVYk0RAkM1AKCepBdfTkLoqwNlnbMpH3CWGTWCcwCeOFlE
>> jbKdXnKHNqG7951KlctSfek>> =ttdo
>> -----END PGP SIGNATURE-----
>>
>>
>
>
>
> ------------------------------
>
> Message: 6
> Date: Mon, 14 Nov 2005 13:29:39 +0000
> From: Richard Watson <richard@openia.com>
> Subject: Re: [Asterisk-Users] Snom clients deregistering
> To: mike@thevoipconnection.com, Asterisk Users Mailing List -
> Non-Commercial Discussion <asterisk-users@lists.digium.com>
> Message-ID: <43789143.2020804@openia.com>
> Content-Type: text/plain; charset=UTF-8
>
> Michael Crown wrote:
>> Does the phone ocasionally prompt the user for a password? -Mike
>
> Yes it does!!!!
>
> How did you know?
>
>
>
> ------------------------------
>
> Message: 7
> Date: Mon, 14 Nov 2005 07:26:15 -0600
> From: Rich Adamson  <radamson@routers.com>
> Subject: RE: [Asterisk-Users] How to check how many G729 codec
> licenseinstalled
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users@lists.digium.com>
> Message-ID: <Chameleon.1131974985.adar0@vegas>
> Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-1
>
>> Easy:
>> > show g729
>>
>> This will show total in use and total available channels for g729
>> doesnt work for me, maybe its a version difference.
>> I do have g729 loaded, and that was verified.
>
> Using cvs-head...
>
> If you have the digium licensed g729, the 'show g729' looks like:
> show g729
> 0/0 encoders/decoders of 6 licensed channels are currently in use
>
> If you loaded a different g729 codec (unlicensed, but available on the
> internet), the response will be "No such command..."
>
>
>
>
> ------------------------------
>
> Message: 8
> Date: Mon, 14 Nov 2005 08:29:53 -0500
> From: Sergey Okhapkin <sos@sokhapkin.dyndns.org>
> Subject: Re: [Asterisk-Users] IAXy echo?
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users@lists.digium.com>
> Message-ID: <1131974993.31775.10.camel@sergei.leapstone.com>
> Content-Type: text/plain
>
> Lower speaker volume on the phone connected to IAXy.
>
> On Mon, 2005-11-14 at 07:21 -0600, Mike Hammett wrote:
>> I've got two customers on the same broadband provider.  Same
Asterisk
>> box on my end.  Same CLEC.
>>
>> One has an IAXy and the other has an Asterisk box with an array of
>> devices (Grandstream, Cisco, ATCOM, xten, etc.).
>>
>> The people behind the Asterisk box have had no audio quality issues.
>> The person with the IAXy often encounters an echo.  The echo is only
>> heard on the remote side and it only contains the remote caller's
>> voice.  This echo has been heard with the remote side being varying
>> LECs.  The echo is not always there.  I'd almost say that the echo
is
>> not there more than it is.
>>
>> Troubleshooting next step?
>>
>> I haven't changed out the IAXy because I don't have any other
ATAs to
>> put in place.
>>
>>
>> ----
>> Mike Hammett
>> Intelligent Computing Solutions
>> http://www.ics-il.com
>>
>>
>> _______________________________________________
>> --Bandwidth and Colocation sponsored by Easynews.com --
>>
>> Asterisk-Users mailing list
>> Asterisk-Users@lists.digium.com
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> ------------------------------
>
> Message: 9
> Date: Mon, 14 Nov 2005 08:40:10 -0500
> From: "The VoIP Connection"
<asterisk-biz@thevoipconnection.com>
> Subject: RE: [Asterisk-Users] Snom clients deregistering
> To: "'Richard Watson'" <richard@openia.com>,
"'Asterisk Users Mailing
> List - Non-Commercial Discussion'"
<asterisk-users@lists.digium.com>
> Message-ID: <20051114134012.0DD2C46A8@lists.digium.com>
> Content-Type: text/plain; charset="us-ascii"
>
> There is a setting on the "Advanced" page called "Challenge
Response on
> Phone". Turn this setting to "Off" and your problem will be
solved. Also,
> we
> usually set the "Proposed Expiry" to 1 minute On the
"SIP" page when
> phones
> are behind a NAT.
>
> -Mike
>
>> -----Original Message-----
>> From: Richard Watson [mailto:richard@openia.com]
>> Sent: Monday, November 14, 2005 8:30 AM
>> To: mike@thevoipconnection.com; Asterisk Users Mailing List -
>> Non-Commercial Discussion
>> Subject: Re: [Asterisk-Users] Snom clients deregistering
>>
>> Michael Crown wrote:
>> > Does the phone ocasionally prompt the user for a password? -Mike
>>
>> Yes it does!!!!
>>
>> How did you know?
>>
>
>
>
> ------------------------------
>
> Message: 10
> Date: Mon, 14 Nov 2005 07:41:11 -0600
> From: Rich Adamson  <radamson@routers.com>
> Subject: Re: [Asterisk-Users] IAXy echo?
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users@lists.digium.com>
> Message-ID: <Chameleon.1131975748.adar0@vegas>
> Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-1
>
>
>> I've got two customers on the same broadband provider.  Same
Asterisk box
>> on my end.
> Same CLEC.
>>
>> One has an IAXy and the other has an Asterisk box with an array of
>> devices
> (Grandstream, Cisco, ATCOM, xten, etc.).
>>
>> The people behind the Asterisk box have had no audio quality issues.
The
>> person with
> the IAXy often encounters an echo.
>> The echo is only heard on the remote side and it only contains the
remote
>> caller's
> voice.  This echo has been heard with the
>> remote side being varying LECs.  The echo is not always there.  I'd
>> almost say that
> the echo is not there more than it is.
>>
>> Troubleshooting next step?
>>
>> I haven't changed out the IAXy because I don't have any other
ATAs to put
>> in place.
>
> Best guess... the iaxy doesn't have an echo can in it, and probably
relies
> on asterisk to do the cancellation.
>
>
>
>
> ------------------------------
>
> Message: 11
> Date: Mon, 14 Nov 2005 14:58:43 +0100
> From: "janvb@caselaboratories.com"
<janvb@caselaboratories.com>
> Subject: Re: [Asterisk-Users] IAXy echo?
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users@lists.digium.com>
> Message-ID: <43789813.7080609@caselaboratories.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> You will also experience this if the latency between the Asterix PABX
> and IAXy is so high that echo cancel don't work.
>
> Jan
> Rich Adamson wrote:
>
>>>I've got two customers on the same broadband provider.  Same
Asterisk box
>>>on my end.
>>>
>>>
>>Same CLEC.
>>
>>
>>>
>>>One has an IAXy and the other has an Asterisk box with an array of
>>>devices
>>>
>>>
>>(Grandstream, Cisco, ATCOM, xten, etc.).
>>
>>
>>>
>>>The people behind the Asterisk box have had no audio quality issues.
The
>>>person with
>>>
>>>
>>the IAXy often encounters an echo.
>>
>>
>>>The echo is only heard on the remote side and it only contains the
remote
>>>caller's
>>>
>>>
>>voice.  This echo has been heard with the
>>
>>
>>>remote side being varying LECs.  The echo is not always there.
I'd
>>>almost say that
>>>
>>>
>>the echo is not there more than it is.
>>
>>
>>>
>>>Troubleshooting next step?
>>>
>>>I haven't changed out the IAXy because I don't have any
other ATAs to put
>>>in place.
>>>
>>>
>>
>>Best guess... the iaxy doesn't have an echo can in it, and probably
relies
>>on asterisk to do the cancellation.
>>
>>
>>_______________________________________________
>>--Bandwidth and Colocation sponsored by Easynews.com --
>>
>>Asterisk-Users mailing list
>>Asterisk-Users@lists.digium.com
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>>
>>
>>
>
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> ------------------------------
>
> Message: 12
> Date: Mon, 14 Nov 2005 09:03:04 -0500
> From: "Sean Cook" <scook@kinex.net>
> Subject: RE: [Asterisk-Users] How to check how many G729 codec
> licenseinstalled
> To: "'Asterisk Users Mailing List - Non-Commercial
Discussion'"
> <asterisk-users@lists.digium.com>
> Message-ID: <007f01c5e924$21e509c0$bd00000a@benchmarksystems.com>
> Content-Type: text/plain; charset="US-ASCII"
>
> Are you running the g729 module from digium?  Registered?
>
> Sean
>
>> -----Original Message-----
>> From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-
>> bounces@lists.digium.com] On Behalf Of trixter aka Bret McDanel
>> Sent: Monday, November 14, 2005 7:50 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: RE: [Asterisk-Users] How to check how many G729 codec
>> licenseinstalled
>>
>> On Mon, 2005-11-14 at 07:46 -0500, Sean Cook wrote:
>> > Easy:
>> > > show g729
>> >
>> > This will show total in use and total available channels for g729
>>
>> doesnt work for me, maybe its a version difference.
>>
>> I do have g729 loaded, and that was verified.
>>
>> --
>> Trixter http://www.0xdecafbad.com     Bret McDanel
>> UK +44 870 340 4605   Germany +49 801 777 555 3402
>> US +1 360 207 0479 or +1 516 687 5200
>> FreeWorldDialup: 635378
>
>
>
> ------------------------------
>
> Message: 13
> Date: Mon, 14 Nov 2005 19:37:51 +0530
> From: "ashok" <ashok.reddy@cem-solutions.net>
> Subject: [Asterisk-Users] Configure Asterisk to call from
> softPhone(SIP Channel) to Analog phone(Modem Channel)
> To: <asterisk-users@lists.digium.com>
> Message-ID: <20051114140905.0291246D9@lists.digium.com>
> Content-Type: text/plain; charset="us-ascii"
>
>
> Hi *users,,
>
> I'm researching on Asterisk PBX phone system initially I was
successfull
> in
> configuring 2 SIP users with DIAL rules in extension.conf and
> configured 2X-Lite softphones to use my proxy
> Registered successfully also able to dial and communicate.
>
> Now i am trying to dial from softphone to analog phone
> connected to Internal Modem of my proxy but ended up
> with errors while loading asterisk -vvvvgc
>
> Asterisk Dynamic Loader Starting:
>  == Parsing '/etc/asterisk/modules.conf': Found
> [chan_modem.so] => (Generic Voice Modem Driver)
>  == Parsing '/etc/asterisk/modem.conf': Found
>  == Loading modem driver chan_modem_slamr.soNov 14
> 15:02:15 WARNING[8042]: loader.c:258
> ast_load_resource:
> /usr/lib/asterisk/modules/chan_modem_slamr.so: cannot
> open shared object file: No such file or directory
> Nov 14 15:02:15 ERROR[8042]: chan_modem.c:968
> load_module: Failed to load driver chan_modem_slamr.so
>  == Unregistered channel type 'Modem'
> Nov 14 15:02:15 WARNING[8042]: loader.c:345
> ast_load_resource: chan_modem.so: load_module failed,
> returning -1
>  == Unregistered channel type 'Modem'
> Nov 14 15:02:15 WARNING[8042]: loader.c:391
> load_modules: Loading module chan_modem.so failed!
>
> Any idea how to generate chan_modem_slamr.so file???
>
> [root@localhost slmodem-2.9.10]# more
> /etc/modules.conf
> alias eth0 8139too
> alias eth1 via-rhine
> alias usb-controller ehci-hcd
> alias usb-controller1 usb-uhci
> alias sound-slot-0 via82cxxx_audio
> post-install sound-slot-0 /bin/aumix-minimal -f
> /etc/.aumixrc -L >/dev/null 2>&1 || :
> pre-remove sound-slot-0 /bin/aumix-minimal -f
> /etc/.aumixrc -S >/dev/null 2>&1 || :
> alias char-major-212 slamr
> alias char-major-213 slusb
>
> Internal modem :- Smartlink chipset v.92 internal pci
> Modem
>
> Pls suggest me how do I write DIAL rule so that user 2000 registerd to
> proxy
> via Softphone can dial 2001 to analog phone.
>
> Thanks in advance.
>
> Warm Regards
> ashok
>
>
>
> ------------------------------
>
> Message: 14
> Date: Mon, 14 Nov 2005 07:48:17 -0600
> From: Rich Adamson  <radamson@routers.com>
> Subject: RE: [Asterisk-Users] Sipura SPA-2002 Double Ring
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users@lists.digium.com>
> Message-ID: <Chameleon.1131977711.adar0@vegas>
> Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-1
>
>> >> I recently implemented a Sipura SPA-2002 with one of my
Asterisk
>> >> installations.  On internal calls, the SPA generates ringtone
as
>> >> expected. However, when I dial out via my IAX-based service
>> >> provider, I hear
>> >> both the telco-generated ringtone as well as the SPA-generated
>> >> ringtone.  Sometimes, the SPA continues to generate the
ringtone even
>> >> after the call has been answered.
>> >
>> > I don't have a spa-2002, but do use a spa3k. I doubt very much
the
>> > sipura device is actually providing ringback tone, and I don't
recall
>> > any parameters that would enable/disable such an item. (The Admin
>> > manual does not mention it either.)
>> >
>> > You might check your extensions.conf entry for dialing your
provider
>> > to see if you have an "r" in that line. If so, remove
it.
>>
>> The SPA-2002 is definitely generating the additional ringback.  I
>> verified
>> this by temporarily changing the frequency of the ringback in the
SPA's
>> "Regional" settings.
>>
>> I also verified that I am not using the "r" option in the
Dial command.
>> If
>> I were, however, only the Asterisk-generated ringback would be heard,
and
>> then only until the call supervised (i.e. I would not be hearing two
>> distinct ring signals, and the ringback would not occasionally persist
>> for
>> the duration of a call while still hearing the called party).
>>
>> This problem is present only with the SPA-2002, and none of the other
SIP
>> devices connected to this Asterisk server.  I have also tried making
>> outbound calls via different service providers, all with the same
>> results.
>
> If I had this problem, I'd use ethereal to observe the sip traffic to
> the box and look for a control packet containing "RING". If that
is
> coming from your asterisk box "after" a call in progress, then
asterisk
> isn't functioning properly.
>
> If you don't see that packet, then I'd be on the horn to sipura
support.
> (Make sure you're running the latest firmware for the box as that will
> always be their first suggestion.)
>
>
>
>
>
> ------------------------------
>
> Message: 15
> Date: Mon, 14 Nov 2005 06:25:31 -0800 (PST)
> From: Richard Reina <rf_reina@yahoo.com>
> Subject: [Asterisk-Users] OT: Aastra PT 390 Question.
> To: asterisk-users@lists.digium.com
> Message-ID: <20051114142531.49875.qmail@web53809.mail.yahoo.com>
> Content-Type: text/plain; charset=iso-8859-1
>
> Does anyone know how to put an Aastra PT 390 in
> headset mode, so it will only give a dial tone when
> you are ready ?  Right now I can't figure how to keep
> it hung up?  If I hit googbye it merely flashes (give
> me a dial tone again).
>
> Any help would be greatly appreciated?
>
>
>
>
> __________________________________
> Yahoo! FareChase: Search multiple travel sites in one click.
> http://farechase.yahoo.com
>
>
> ------------------------------
>
> Message: 16
> Date: Mon, 14 Nov 2005 07:28:48 -0700
> From: "Damon Estep" <damon@suburbanbroadband.net>
> Subject: [Asterisk-Users] SIP signaling and canreinvite=yes
> To: <asterisk-users@lists.digium.com>
> Message-ID:
> <07668904BA88BA4E9DA11CDE5B594CB29F066D@ns1.soho.soho-systems.com>
> Content-Type: text/plain; charset="us-ascii"
>
> After reviewing many other posts as well as wiki information on
> canreinvite and asterisk media path I am not clear on whether asterisk
> still manages sip signaling after a reinvite has been issued between a
> peer and a UA.
>
>
>
> Here are the details;
>
>
>
> UA <g.711u> Asterisk <g.711u> SIP long distance provider.
>
> The SIP LD provider uses a session border controller to ensure that all
> sip traffic originates from my asterisk IP address.
>
> The SIP LD provider will accept RTP streams from any source.
>
>
>
> Due to an issue when sending faxes with * in the media stream, I want to
> remove asterisk from the media stream for specific UAs (faxes complete
> successfully without asterisk in the stream, tested by setting the UA to
> the asterisk IP address).
>
>
>
> In theory, if canreinvite=yes, codecs match (g.711u) and there are no
> dial options that require asterisk to remain in the stream, the
> re-invite should be issued and the UA and the peer should be the
> endpoints of the RTP streams.
>
>
>
> Questions;
>
>
>
> Does it work? I am having trouble getting it to work that way.
>
> Is the sip signaling all handled by asterisk in this case? - required by
> my providers session border controller.
>
>
>
> I guess what I am asking is can asterisk function as a SIP PROXY when
> configured correctly?
>
>
>
> Any examples or limitations I might have missed?
>
>
>
> Thank you!
>
>
>
> Damon
>
>
>
>
>
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> ------------------------------
>
> Message: 17
> Date: Mon, 14 Nov 2005 15:39:58 +0100
> From: Kristof Hardy <kristof.hardy@catsanddogs.com>
> Subject: Re: [Asterisk-Users] ISDN card required
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users@lists.digium.com>
> Message-ID: <4378A1BE.7050300@catsanddogs.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Lee Archer wrote:
>> Can anyone point me in the direction of a quality, works with Asterisk,
>> BRI card.  I need minimum 2 port/4 channel.
>
> Ack. Like Mark pointed out, I also used Junghanns.net cards, works fine.
>
> Cheers.
>
>
>
> ------------------------------
>
> Message: 18
> Date: Mon, 14 Nov 2005 14:46:33 -0000
> From: "Lee Archer" <lee.archer@pentagon-systems.com>
> Subject: RE: [Asterisk-Users] ISDN card required
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users@lists.digium.com>
> Message-ID:
> <CE0BC3726DA68C499E60931F0CFA3D500E3B4A@exsystems.Systems.local>
> Content-Type: text/plain; charset="us-ascii"
>
> Thanks to all.  I'll probably go with the quadBri card they do.
>
> Regards
>
> Lee
>
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com
> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kristof
> Hardy
> Sent: 14 November 2005 14:40
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] ISDN card required
>
> Lee Archer wrote:
>> Can anyone point me in the direction of a quality, works with
>> Asterisk, BRI card.  I need minimum 2 port/4 channel.
>
> Ack. Like Mark pointed out, I also used Junghanns.net cards, works fine.
>
> Cheers.
>
> _______________________________________________
> --Bandwidth and Colocation sponsored by Easynews.com --
>
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> ------------------------------
>
> Message: 19
> Date: Mon, 14 Nov 2005 14:49:11 +0000
> From: Richard Watson <richard@openia.com>
> Subject: Re: [Asterisk-Users] Snom clients deregistering
> To: asterisk-biz@thevoipconnection.com, Asterisk Users Mailing List -
> Non-Commercial Discussion <asterisk-users@lists.digium.com>
> Message-ID: <4378A3E7.5070506@openia.com>
> Content-Type: text/plain; charset=UTF-8
>
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> The VoIP Connection wrote:
>> There is a setting on the "Advanced" page called
"Challenge Response on
>> Phone". Turn this setting to "Off" and your problem will
be solved. Also,
>> we
>> usually set the "Proposed Expiry" to 1 minute On the
"SIP" page when
>> phones
>> are behind a NAT.
>
> That doesn't seem to have helped entirely.
>
> The "Password" prompt no longer appears but the phone still
becomes
> UNREACHABLE then UNKNOWN after a few minutes.
>
> In the system information on the phone it reports "Registration
Failed".
> However a few minutes later it logs itself back in.
>
> I have two identical snoms on the bench here and they both do the same
> thing, logging in and operating fine, before eventually (but not
> necessarily at the same time) losing registration and stopping for a few
> minutes.
>
>
>
>
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.1 (GNU/Linux)
> Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
>
> iD4DBQFDeKPnP05lUVhVYk0RAqTfAJYtZqmp1dCRLDhu3C1jHRCeUk5LAJ42z2rV
> 5Jr8qm+Ruyvv3h2L3jOjUA=> =PlHs
> -----END PGP SIGNATURE-----
>
>
> ------------------------------
>
> Message: 20
> Date: Mon, 14 Nov 2005 14:51:25 +0000 (UTC)
> From: tony@softins.clara.co.uk (Tony Mountifield)
> Subject: [Asterisk-Users] Re: MYSQL issue in UPDATE..
> To: asterisk-users@lists.digium.com
> Message-ID: <dla89d$dah$1@softins.clara.co.uk>
>
> In article <BAY102-DAV1433B94982FCBD868B1EF2D95A0@phx.gbl>,
> Mauro Zanin <maurozanin@hotmail.com> wrote:
>> Hi Everybody,
>> I'm trying to execute a MYSQL(UPDATE..............................)
sql
>> command over a table I have previously red. I get a timeout and no
update
>> happens.
>> I use  * 1.0.9.
>> I wonder if MYSQL set of commands allows Update...
>
> Yes, I use UPDATE within MYSQL() successfully.
>
> If you post the complete extract from your dialplan, starting with the
> "MYSQL(Connect..." up to the "MYSQL(Disconnect...",
then we might be able
> to suggest where the problem lies.
>
> Cheers
> Tony
> --
> Tony Mountifield
> Work: tony@softins.co.uk - http://www.softins.co.uk
> Play: tony@mountifield.org - http://tony.mountifield.org
>
>
> ------------------------------
>
> Message: 21
> Date: Mon, 14 Nov 2005 21:06:29 +0800
> From: Stephen Arulraj <stephen.sa.arulraj@solomonstar.com>
> Subject: [Asterisk-Users] Brooktrout MPAC 1200 card with Asterisk
> To: Asterisk-Users@lists.digium.com
> Message-ID: <43788BD5.4060003@solomonstar.com>
> Content-Type: text/plain; charset="us-ascii"
>
>
> I have a 4 port brooktrout PCI E1/T1 blade card (MPAC 1200) that was
> used for some carrier server. Will Asterisk support this? Has anyone
> used this successfully before? Thanks! Stephen
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> ------------------------------
>
> Message: 22
> Date: Mon, 14 Nov 2005 06:53:32 -0800 (PST)
> From: nr k <nrkrishnan2005@yahoo.com>
> Subject: [Asterisk-Users] Maximum Number of SIP Phones Supported By
> Asterisk
> To: asterisk-users@lists.digium.com
> Message-ID: <20051114145332.61504.qmail@web33814.mail.mud.yahoo.com>
> Content-Type: text/plain; charset=iso-8859-1
>
> Hi All
>
> Can anybody tell me the maximum number of SIP Phones
> supported by Asterisk.
>
> regards
> ramakrishnan.n
>
>
>
>
> __________________________________
> Start your day with Yahoo! - Make it your home page!
> http://www.yahoo.com/r/hs
>
>
> ------------------------------
>
> Message: 23
> Date: Mon, 14 Nov 2005 06:58:24 -0800
> From: trixter aka Bret McDanel <trixter@0xdecafbad.com>
> Subject: [Asterisk-Users] asterisk sample size adjustment
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users@lists.digium.com>
> Message-ID: <1131980304.8892.24.camel@rufus.home.tld>
> Content-Type: text/plain; charset="us-ascii"
>
> Is there any way to adjust the sample size asterisk uses for VoIP
> codecs?  From what I have gathered it uses a fixed 20ms sample size for
> all codecs.  While some require at least this, some can be configured
> for less.  This results in more overhead, but can be tweaked to provide
> more efficient transfer on the backbone links due to ATM framing
> properties.
>
> If anyone has any information on how to change the sample size I would
> appreciate hearing about it, because I cant find anything with google.
> Asterisk is a particularly bad google term since it is used as a
> footnote market, wildcard, etc :P
>
>
> --
> Trixter http://www.0xdecafbad.com     Bret McDanel
> UK +44 870 340 4605   Germany +49 801 777 555 3402
> US +1 360 207 0479 or +1 516 687 5200
> FreeWorldDialup: 635378
> http://www.sacaug.org/ Sacramento Asterisk Users Group
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> ------------------------------
>
> Message: 24
> Date: Mon, 14 Nov 2005 07:02:12 -0800 (PST)
> From: chawki hammoud <cyhammoud@yahoo.com>
> Subject: Re: [Asterisk-Users] Can't make calls from Asterisk IAX to
> other IAX
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users@lists.digium.com>
> Message-ID: <20051114150213.79730.qmail@web90104.mail.scd.yahoo.com>
> Content-Type: text/plain; charset=iso-8859-1
>
> Sorry, I just saw the post.
>
> Yes, it's the same format
>
> Regards;
> Chawki
>
> --- Matt Riddell <matt.riddell@sineapps.com> wrote:
>
>> chawki hammoud wrote:
>> > Hi:
>> >
>> > I have been having this problem for sometime that
>> I am
>> > not able to solve and I hope someone can help.
>> >
>> > I can make VOIP calls between my Asterisk box and
>> my
>> > VOIP provider using sip channel without a problem.
>> But
>> > when I attempt to make a call using IAX, the call
>> get
>> > accepted and then get a hangup message:
>>
>> is this the same number format you send when using
>> sip: 0017046872001
>>
>> --
>> Cheers,
>>
>> Matt Riddell
>> _______________________________________________
>>
>> http://www.sineapps.com/news.php (Daily Asterisk
>> News - html)
>> http://freevoip.gedameurope.com (Free Asterisk Voip
>> Community)
>> http://www.sineapps.com/rssfeed.php (Daily Asterisk
>> News - rss)
>>
>> _______________________________________________
>> --Bandwidth and Colocation sponsored by Easynews.com
>> --
>>
>> Asterisk-Users mailing list
>> Asterisk-Users@lists.digium.com
>>
> http://lists.digium.com/mailman/listinfo/asterisk-users
>> To UNSUBSCRIBE or update options visit:
>>
>>
> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
>
> __________________________________
> Yahoo! FareChase: Search multiple travel sites in one click.
> http://farechase.yahoo.com
>
>
> ------------------------------
>
> Message: 25
> Date: Mon, 14 Nov 2005 16:07:46 +0100
> From: Reli Loin <kerpox@gmail.com>
> Subject: [Asterisk-Users] connect to gateway h323
> To: Asterisk-Users@lists.digium.com
> Message-ID: <841075b20511140707v1322fc6dy@mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> Hello,
>
> Hello, I do not arrive has to connect me has a gateway h323, in
> termination of call.
>
> i have one ip for a termination call xxx.xx.xx.xx,
>
> I do not know if the problem comes from my parameters oh323.conf or the
> gateway
>
>
> i using a latest version asterisk (asterisk 1.2rc1),openh323 latest
> version mimas patch,
> and pwlib latest version and asterisk-0h323-0.7.3
>
> my config files.
>
> -----oh323.conf-------------------
>
> h245Tunnelling=yes
> ;
> ; Enable early H.245 messages in call SETUP message.
> ;
> h245inSetup=yes
> ;
> ; Set jitter buffer (in milliseconds, 20...10000).
> ;
> jitterMin=20
> jitterMax=100
> ;
> ; Set IP Type-of-Service byte for RTP channels.
> ; Valid values for this option are:
> ;   lowdelay, throughput, reliability, mincost, none
> ; Moreover, an integer (in decimal or hex format) may be entered.
> ;
> ipTos=none
> ;
> ; Set the maximum number of inbound/outbound/simultaneous
> ; H.323 connections.
> ;
> outboundMax=100
> inboundMax=100
> simultaneousMax=100
> ;
> ; Call Rate Limiter params (ingress direction). When the total number
> ; of active calls is above 'crlThreshold' then the rate of the
incoming
> ; H.323 calls is restricted in a way where no more than
'crlCallNumber'
> ; calls are allowed in 'crlCallTime' milliseconds, thus limiting
the rate
> ; of incoming calls to:
> ;     'crlCallNumber' / ('crlCallTime' / 1000)
Calls-per-Sec.
> ;
> ;crlCallNumber=20
> ;crlCallTime=20000
> ;crlThreshold=30
> ;
> ; Set the bandwidth limit for H.323 connections.
> ; The value is in Kbps.
> ;
> ;bandwidthLimit=1024
> ;
> ; Set tracing options for the wrapper library and for the
> ; OpenH323 library.
> ; libTraceFile can be 'stdout' or a full path name to the
tracefile.
>
> ; Only the trace info for OpenH323 is logged in libTraceFile.
> ;
> wrapLibTraceLevel=0
> libTraceLevel=0
> libTraceFile=stdout
> ;
> ; Disable gatekeeper or specify a gatekeeper. The gatekeeper's ID is
> the zone name.
> ; Valid values for this option are:
> ;   DISABLE,
> ;   DISCOVER,
> ;   <gatekeeper's DNS name>,
> ;   <gatekeeper's ip>,
> ;   GKID:<gatekeeper's id>
> ;   <gatekeeper's id>@<gatekeeper's name or address>
> ;
> ;gatekeeper=192.168.1.6
> gatekeeper=xxx.xxx.xxx.xx
> ;
> ; Set the gatekeeper password. If used, it enables H.235 access to
> gatekeeper.
> ;
> ;gatekeeperPassword=secret
> ;
> ; Set the gatekeeper registration timeout. Before the expiration of
> ; the timeout, a re-registration is attempted.
> ;
> gatekeeperTTL=60
> ;
> ; Set the mode for sending user-input (DTMF)
> ; Valid values for this option are:
> ;   Q931        -   Q.931 Keypad Information Element
> ;   STRING      -   H.245 string
> ;   TONE        -   H.245 tone
> ;   RFC2833     -   RFC2833
> ;   INBAND      -
> ;
> userInputMode=TONE
> ;
> ; AMA flags (default, omit, billing, documentation)
> ;
> amaFlags=default
> ;
> ; Account code
> ;
> ;
> accountCode=H323
> ;
> ; Default language
> ;
> language=en
> ;
> ; Default Music-On-Hold class
> ;
> musiconhold=default
> ;
> ; Set the default context of H.323 calls.
> ;
> context=h323
>
> ;-----------------------------------------
> ; Configure H.323 aliases, prefixes and
> ; related ASTERISK's contexts
> ;-----------------------------------------
> [register]
> ;
> ; Aliases/prefixes associated with the default context
> ; defined in section [general].
> ;
> alias=asterisk
> alias=123
> ;
> ; Aliases/prefixes routed in "all-aliases" context.
> ;
> context=all-aliases
> alias=ASTERISK
> alias=666
> ;
> ; Aliases/prefixes routed in "more-aliases" context.
> ;
> context=more-aliases
> alias=665
> ;
> ; Aliases/prefixes routed in "all-prefixes" context.
> ;
> context=all-prefixes
> gwprefix=00
> gwprefix=01
> ;
>
>
> ; Aliases/prefixes routed in "more-stuff" context.
> ;
> context=more-stuff
> alias=664
> gwprefix=02
>
> ;-----------------------------------------
> ; Specify and configure CODEC related
> ; options
> ;-----------------------------------------
> [codecs]
> ;
> ; Define the codec list of the channel driver.
> ; Every "codec" option may have a "frames" option
> ; associated with it.
> ; Valid values for the "codec" option are:
> ;   G711U       -   G.711 u-Law
> ;   G711A       -   G.711 A-Law
> ;   G7231       -   G.723.1(6.3k)
> ;   G72316K3    -   G.723.1(6.3k)
> ;   G72315K3    -   G.723.1(5.3k)
> ;   G7231A6K3   -   G.723.1A(6.3k)
> ;   G7231A6K3   -   G.723.1A(6.3k)
> ;   G726        -   G.726(32k)
> ;   G72616K     -   G.726(16k)
> ;   G72624K     -   G.726(24k)
> ;   G72632K     -   G.726(32k)
> ;   G72640K     -   G.726(40k)
> ;   G728        -   G.728
> ;   G729        -   G.729
> ;   G729A       -   G.729A
> ;   G729B       -   G.729B
> ;   G729AB      -   G.729AB
> ;   GSM0610     -   GSM 0610
> ;   MSGSM       -   Microsoft GSM Audio Capability
> ;   LPC10       -   LPC-10
> ; Number of frames in RTP packet (if not specified) is 1.
> ;
> codec=G711A
> frames=20
> codec=G711U
> frames=20
> ;codec=GSM0610
>
> ;frames=4
> ;codec=G7231
> ;frames=2
> codec=G729
> frames=2
>
> [h323terminate]
> type=peer
> host=xx.xxx.xxx.xxx
> dtmfcodec=99
>
> ----------------------------------------------
>
>
> oh323 show conf in asterisk cli
>
> Configuration of OpenH323 channel driver
> ------------------------------------------
> Version: 0.7.3
> Listening on address: 0.0.0.0:1720
> Gatekeeper used:  No gatekeeper
> FastStart/H245Tunnelling/H245inSetup: ON/ON/ON
> Supported formats in pref. order: alaw<0> ulaw<1> g729<2>
> Jitter buffer limits (min/max): 20-100 ms
> TCP port range: 10000 - 20000
> UDP (RAS) port range: 10000 - 20000
> UDP (RTP) port range: 10000 - 20000
> IP Type-of-Service value: 0
> User input mode: tone
> Max number of inbound H.323 calls: 100
> Max number of outbound H.323 calls: 100
> Max number of simultaneous H.323 calls: 100
> Max call rate (ingress direction): 1.00/30
> Default language: en
> Default music class: default
> Default context: h323
>
>
> ------------------------------
>
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>
>
> End of Asterisk-Users Digest, Vol 16, Issue 104
> ***********************************************
>