Send Asterisk-Users mailing list submissions to
asterisk-users@lists.digium.com
To subscribe or unsubscribe via the World Wide Web, visit
http://lists.digium.com/mailman/listinfo/asterisk-users
or, via email, send a message with subject or body 'help' to
[EMAIL PROTECTED]
You can reach the person managing the list at
[EMAIL PROTECTED]
When replying, please edit your Subject line so it is more specific
than "Re: Contents of Asterisk-Users digest..."
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? ([EMAIL PROTECTED])
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 <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Can't create iax channel
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
Discussion <asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
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:[EMAIL PROTECTED]/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."
[EMAIL PROTECTED] (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 <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Can't create iax channel
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
Discussion <asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
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."
[EMAIL PROTECTED] (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 <[EMAIL PROTECTED]>
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: <[EMAIL PROTECTED]>
Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-1
show g729
------------------------
From: Mark Quitoriano <[EMAIL PROTECTED]>
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 <[EMAIL PROTECTED] > wrote:
Yes.
----- Original Message -----
From: "Angelito Manansala" < [EMAIL PROTECTED]>
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 < [EMAIL PROTECTED]> 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" <[EMAIL PROTECTED]>
>> 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 <[EMAIL PROTECTED]> 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: [EMAIL PROTECTED]
>> >>> [mailto: [EMAIL PROTECTED] 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: [EMAIL PROTECTED]
>> >> 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: [EMAIL PROTECTED]
> 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" <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] IAXy echo?
To: <asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
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 --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20051114/1c3520b3/attachment-0001.htm
------------------------------
Message: 5
Date: Mon, 14 Nov 2005 08:26:42 -0500
From: "Michael Crown" <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] Snom clients deregistering
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
<asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="us-ascii"
Does the phone ocasionally prompt the user for a password? -Mike
-----Original Message-----
From: Richard Watson [mailto:[EMAIL PROTECTED]
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 <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Snom clients deregistering
To: [EMAIL PROTECTED], Asterisk Users Mailing List -
Non-Commercial Discussion <asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
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 <[EMAIL PROTECTED]>
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: <[EMAIL PROTECTED]>
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 <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] IAXy echo?
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
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" <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] Snom clients deregistering
To: "'Richard Watson'" <[EMAIL PROTECTED]>, "'Asterisk Users Mailing
List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
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:[EMAIL PROTECTED]
Sent: Monday, November 14, 2005 8:30 AM
To: [EMAIL PROTECTED]; 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 <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] IAXy echo?
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
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: "[EMAIL PROTECTED]" <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] IAXy echo?
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
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
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20051114/d9b92041/attachment-0001.htm
------------------------------
Message: 12
Date: Mon, 14 Nov 2005 09:03:04 -0500
From: "Sean Cook" <[EMAIL PROTECTED]>
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: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="US-ASCII"
Are you running the g729 module from digium? Registered?
Sean
-----Original Message-----
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] 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" <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] Configure Asterisk to call from
softPhone(SIP Channel) to Analog phone(Modem Channel)
To: <asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
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???
[EMAIL PROTECTED] 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 <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] Sipura SPA-2002 Double Ring
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
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 <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] OT: Aastra PT 390 Question.
To: asterisk-users@lists.digium.com
Message-ID: <[EMAIL PROTECTED]>
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" <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] SIP signaling and canreinvite=yes
To: <asterisk-users@lists.digium.com>
Message-ID:
<[EMAIL PROTECTED]>
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
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20051114/91dbf128/attachment-0001.htm
------------------------------
Message: 17
Date: Mon, 14 Nov 2005 15:39:58 +0100
From: Kristof Hardy <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] ISDN card required
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
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" <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] ISDN card required
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Message-ID:
<[EMAIL PROTECTED]>
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: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] 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 <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Snom clients deregistering
To: [EMAIL PROTECTED], Asterisk Users Mailing List -
Non-Commercial Discussion <asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
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: [EMAIL PROTECTED] (Tony Mountifield)
Subject: [Asterisk-Users] Re: MYSQL issue in UPDATE..
To: asterisk-users@lists.digium.com
Message-ID: <[EMAIL PROTECTED]>
In article <[EMAIL PROTECTED]>,
Mauro Zanin <[EMAIL PROTECTED]> 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: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
------------------------------
Message: 21
Date: Mon, 14 Nov 2005 21:06:29 +0800
From: Stephen Arulraj <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] Brooktrout MPAC 1200 card with Asterisk
To: Asterisk-Users@lists.digium.com
Message-ID: <[EMAIL PROTECTED]>
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
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20051114/3c777596/attachment-0001.htm
------------------------------
Message: 22
Date: Mon, 14 Nov 2005 06:53:32 -0800 (PST)
From: nr k <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] Maximum Number of SIP Phones Supported By
Asterisk
To: asterisk-users@lists.digium.com
Message-ID: <[EMAIL PROTECTED]>
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 <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] asterisk sample size adjustment
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
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
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: application/pgp-signature
Size: 189 bytes
Desc: This is a digitally signed message part
Url :
http://lists.digium.com/pipermail/asterisk-users/attachments/20051114/5dba5267/attachment-0001.pgp
------------------------------
Message: 24
Date: Mon, 14 Nov 2005 07:02:12 -0800 (PST)
From: chawki hammoud <[EMAIL PROTECTED]>
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: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=iso-8859-1
Sorry, I just saw the post.
Yes, it's the same format
Regards;
Chawki
--- Matt Riddell <[EMAIL PROTECTED]> 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 <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] connect to gateway h323
To: Asterisk-Users@lists.digium.com
Message-ID: <[EMAIL PROTECTED]>
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@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
End of Asterisk-Users Digest, Vol 16, Issue 104
***********************************************
