[Asterisk-Users] IAXy echo?
Mike Hammett
asterisk-users at ics-il.net
Mon Nov 14 21:24:26 MST 2005
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 at lists.digium.com>
To: <asterisk-users at 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 at 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 at alphaque.com>
> Subject: Re: [Asterisk-Users] Can't create iax channel
> To: waynetg at telkomsa.net, Asterisk Users Mailing List - Non-Commercial
> Discussion <asterisk-users at lists.digium.com>
> Message-ID: <437884CC.90702 at 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 at 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 at 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 at alphaque.com>
> Subject: Re: [Asterisk-Users] Can't create iax channel
> To: waynetg at telkomsa.net, Asterisk Users Mailing List - Non-Commercial
> Discussion <asterisk-users at lists.digium.com>
> Message-ID: <437884FF.4050703 at 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 at 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 at routers.com>
> Subject: Re: [Asterisk-Users] How to check how many G729 codec
> licenseinstalled
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <Chameleon.1131974327.adar0 at vegas>
> Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-1
>
> show g729
>
> ------------------------
> From: Mark Quitoriano <markquitoriano at 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 at lists.digium.com>
>
>
>> how can i check how many g729 are being used right now?
>>
>> On 11/13/05, Gentian Bajraktari <g.bajraktari at afb.net.al > wrote:
>>
>> Yes.
>>
>> ----- Original Message -----
>> From: "Angelito Manansala" < lito at voicefidelity.net>
>> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>> <asterisk-users at 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 at 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 at voicevalley.com.au>
>> >> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>> >> <asterisk-users at 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 at 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 at lists.digium.com
>> >> >>> [mailto: asterisk-users-bounces at lists.digium.com] On Behalf Of
>> >> >>> Angelito
>> >> >>> Manansala
>> >> >>> Sent: Sunday, 13 November 2005 10:31 PM
>> >> >>> To: asterisk-users at 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 at 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 at 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 at elitemail.org
>> >> >> skype: bulcrack
>> >> >> _______________________________________________
>> >> >> --Bandwidth and Colocation sponsored by Easynews.com --
>> >> >>
>> >> >> Asterisk-Users mailing list
>> >> >> Asterisk-Users at 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 at 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 at 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 at elitemail.org
>> > skype: bulcrack
>> > _______________________________________________
>> > --Bandwidth and Colocation sponsored by Easynews.com --
>> >
>> > Asterisk-Users mailing list
>> > Asterisk-Users at 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 at 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 at ics-il.net>
> Subject: [Asterisk-Users] IAXy echo?
> To: <asterisk-users at lists.digium.com>
> Message-ID: <015e01c5e91e$6a3d2550$fa01010a at 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 at thevoipconnection.com>
> Subject: RE: [Asterisk-Users] Snom clients deregistering
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> <asterisk-users at lists.digium.com>
> Message-ID: <20051114132640.4C398466F at 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 at openia.com]
>> Sent: Monday, November 14, 2005 5:00 AM
>> To: asterisk-users at 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 at openia.com>
> Subject: Re: [Asterisk-Users] Snom clients deregistering
> To: mike at thevoipconnection.com, Asterisk Users Mailing List -
> Non-Commercial Discussion <asterisk-users at lists.digium.com>
> Message-ID: <43789143.2020804 at 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 at routers.com>
> Subject: RE: [Asterisk-Users] How to check how many G729 codec
> licenseinstalled
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <Chameleon.1131974985.adar0 at 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 at sokhapkin.dyndns.org>
> Subject: Re: [Asterisk-Users] IAXy echo?
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <1131974993.31775.10.camel at 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 at 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 at thevoipconnection.com>
> Subject: RE: [Asterisk-Users] Snom clients deregistering
> To: "'Richard Watson'" <richard at openia.com>, "'Asterisk Users Mailing
> List - Non-Commercial Discussion'" <asterisk-users at lists.digium.com>
> Message-ID: <20051114134012.0DD2C46A8 at 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 at openia.com]
>> Sent: Monday, November 14, 2005 8:30 AM
>> To: mike at 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 at routers.com>
> Subject: Re: [Asterisk-Users] IAXy echo?
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <Chameleon.1131975748.adar0 at 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 at caselaboratories.com" <janvb at caselaboratories.com>
> Subject: Re: [Asterisk-Users] IAXy echo?
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <43789813.7080609 at 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 at 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 at kinex.net>
> Subject: RE: [Asterisk-Users] How to check how many G729 codec
> licenseinstalled
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> <asterisk-users at lists.digium.com>
> Message-ID: <007f01c5e924$21e509c0$bd00000a at 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 at lists.digium.com [mailto:asterisk-users-
>> bounces at 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 at cem-solutions.net>
> Subject: [Asterisk-Users] Configure Asterisk to call from
> softPhone(SIP Channel) to Analog phone(Modem Channel)
> To: <asterisk-users at lists.digium.com>
> Message-ID: <20051114140905.0291246D9 at 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 at 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 at routers.com>
> Subject: RE: [Asterisk-Users] Sipura SPA-2002 Double Ring
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <Chameleon.1131977711.adar0 at 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 at yahoo.com>
> Subject: [Asterisk-Users] OT: Aastra PT 390 Question.
> To: asterisk-users at lists.digium.com
> Message-ID: <20051114142531.49875.qmail at 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 at suburbanbroadband.net>
> Subject: [Asterisk-Users] SIP signaling and canreinvite=yes
> To: <asterisk-users at lists.digium.com>
> Message-ID:
> <07668904BA88BA4E9DA11CDE5B594CB29F066D at 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 at catsanddogs.com>
> Subject: Re: [Asterisk-Users] ISDN card required
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <4378A1BE.7050300 at 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 at pentagon-systems.com>
> Subject: RE: [Asterisk-Users] ISDN card required
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <CE0BC3726DA68C499E60931F0CFA3D500E3B4A at 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 at lists.digium.com
> [mailto:asterisk-users-bounces at 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 at 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 at openia.com>
> Subject: Re: [Asterisk-Users] Snom clients deregistering
> To: asterisk-biz at thevoipconnection.com, Asterisk Users Mailing List -
> Non-Commercial Discussion <asterisk-users at lists.digium.com>
> Message-ID: <4378A3E7.5070506 at 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 at softins.clara.co.uk (Tony Mountifield)
> Subject: [Asterisk-Users] Re: MYSQL issue in UPDATE..
> To: asterisk-users at lists.digium.com
> Message-ID: <dla89d$dah$1 at softins.clara.co.uk>
>
> In article <BAY102-DAV1433B94982FCBD868B1EF2D95A0 at phx.gbl>,
> Mauro Zanin <maurozanin at 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 at softins.co.uk - http://www.softins.co.uk
> Play: tony at mountifield.org - http://tony.mountifield.org
>
>
> ------------------------------
>
> Message: 21
> Date: Mon, 14 Nov 2005 21:06:29 +0800
> From: Stephen Arulraj <stephen.sa.arulraj at solomonstar.com>
> Subject: [Asterisk-Users] Brooktrout MPAC 1200 card with Asterisk
> To: Asterisk-Users at lists.digium.com
> Message-ID: <43788BD5.4060003 at 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 at yahoo.com>
> Subject: [Asterisk-Users] Maximum Number of SIP Phones Supported By
> Asterisk
> To: asterisk-users at lists.digium.com
> Message-ID: <20051114145332.61504.qmail at 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 at 0xdecafbad.com>
> Subject: [Asterisk-Users] asterisk sample size adjustment
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <1131980304.8892.24.camel at 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 at 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 at lists.digium.com>
> Message-ID: <20051114150213.79730.qmail at 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 at 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 at 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 at gmail.com>
> Subject: [Asterisk-Users] connect to gateway h323
> To: Asterisk-Users at lists.digium.com
> Message-ID: <841075b20511140707v1322fc6dy at 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
>
>
> ------------------------------
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> End of Asterisk-Users Digest, Vol 16, Issue 104
> ***********************************************
>
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