-users/2004-September/057110.html
http://www.voip-info.org/wiki-Asterisk+G.729+Licensing
Justin
--
Date: Sat, 2 Sep 2006 16:21:40 +0800
From: Kannaiyan Natesan [EMAIL PROTECTED]
Hi,
I heard of a news, that there is a replacement codec available for
g729
Also any experts confirm that the code does not contain any hacking on
the computer. I will also wait for the confirmation from Digium to use
this code and a clearly defined procedure to pay the license or
royalty fee.
Kannaiyan
On 9/5/06, Raphael Jacquot [EMAIL PROTECTED] wrote:
Kannaiyan
Also any experts confirm that the code does not contain any hacking on
the computer. I will also wait for the confirmation from Digium to use
this code and a clearly defined procedure to pay the license or
royalty fee.
Kannaiyan
On 9/5/06, Raphael Jacquot [EMAIL PROTECTED] wrote:
Kannaiyan
Hey,
Is this code released by Digium?
Looks like directly from digium. Is it GPL with License and Royalty?
Unlimited channels and no restriction !
Author mentioned as Mark Spencer.
If we want to pay the license fees, should we have to Pay to VoiceAge directly?
Hats off.
Hi,
I heard of a news, that there is a replacement codec available for
g729 and accept the g729 codec data for decoding. Anyone familier with
this? Also the good news is that it is noted that it works fine with
asterisk and the g729 encoded data.
Anyone has the link for the free asterisk
support coming shortly.
Kannaiyan Natesan wrote:
Hi,
I heard of a news, that there is a replacement codec available for
g729 and accept the g729 codec data for decoding. Anyone familier with
this? Also the good news is that it is noted that it works fine with
asterisk and the g729 encoded data
support coming shortly.
Kannaiyan Natesan wrote:
Hi,
I heard of a news, that there is a replacement codec available for
g729 and accept the g729 codec data for decoding. Anyone familier with
this? Also the good news is that it is noted that it works fine with
asterisk and the g729 encoded data
PROTECTED] wrote:
PLEASE DON'T CROSS POST!!
Kannaiyan Natesan wrote:
I heard of a news, that there is a replacement codec available for
g729 and accept the g729 codec data for decoding. [...] If there is
any royalty need to pay, is that cheaper than the existing g729
cost?.
G729 is not royalty
/
On 9/2/06, Kannaiyan Natesan [EMAIL PROTECTED] wrote:
STOP !!
I'm least bothered whether g729 works or not or what the developer did
to made it to work.
I'm bothered how it works and what are the details about it. As you
are keen, I'm also keen on the developer who made
/
On 9/2/06, Kannaiyan Natesan [EMAIL PROTECTED] wrote:
STOP !!
I'm least bothered whether g729 works or not or what the developer did
to made it to work.
I'm bothered how it works and what are the details about it. As you
are keen, I'm also keen on the developer who made
It is possible with an ISDN card, (not with analog line)
Zapras application does that.
http://www.voip-info.org/wiki-Asterisk+cmd+ZapRAS
But it is not good to interface with analog modems, since you need to write
a module to handle the soft modem. As of now nothing like that yet to
asterisk.
set the context in zaptel.conf to
context = dontanswer
and in extensions.conf
[dontanswer]
exten = s,1,Hangup
should help.
regards,
kannaiyan
- Original Message -
From: C F [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
1. Have you contacted Broadvoice Technical Support before sending mail in
this list?
2. Broadvoice uses this list for some reason to market their products
creating situations like this. Is there no one to control on this list.
If you don't get proper technical support, you have to decide what
A complete rubbish service in the whole world, which is spoiling the
asterisk mailing list.
I have seen other companies also use asterisk, but they don't do this
gimmicks too for marketing using this mailing list.
Also I had some respect on Olle before, but I lost it now. (When people get
some
I'm not sure about multiple users on the same machine.
It is just to use like a COM Object.
1. Start Skype
2. Intiate the call (with signal conversion and with error codes), once
successful
3. Capture the sound from the audio hardware buffer.
4. Convert it to the any respective codec which I
if you need any let us know regarding the firmwares.
regards,
nkans
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, November 05, 2004 11:21 PM
Subject: [Asterisk-Users] Cisco 7970 Firmware for the 7960G
Hello,
i´m thinking about buying one if the Cisco´s
Check your zaptel.conf and zapata.conf.
It worked for everyone and it should work for you too.
-Kannaiyan
- Original Message -
From: Steve Totaro
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, October 29, 2004 8:20 PM
Subject: Re: [Asterisk-Users] Ambient MD
oo. Is there is anything such so.
When you buy it for you, does that device not belonged to you?
I don't think cisco can catch you if you load linux on that and use that as
a computer.
-Kannaiyan
- Original Message -
From: Danny Zak [EMAIL PROTECTED]
To: Asterisk Users Mailing List
http://www.goods2world.com/downloads/
-Kannaiyan
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You said what is possible and in exists.
Can you try with the settings what you have mentioned.
-Kannaiyan
- Original Message -
From: Andreas Greulich [EMAIL PROTECTED]
To: Asterisk-Users [EMAIL PROTECTED]
Cc: Greulich, Andreas [EMAIL PROTECTED]
Sent: Monday, September 13, 2004 8:35 PM
asterisk is a pbx software. I don't think there is
acompression and uncompression utility except codec conversions. Even in
those cases you can be sure that there will be loss of data as there is no
lossless compression.
If you have any satellite transmission and reception card
which can
-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Kannaiyan Natesan
Sent: Sunday, September 05, 2004 3:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] ChanSpy by anthm and more...
Does it removes the need of external
check rtp.conf
-Kannaiyan
- Original Message -
From:
[EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, September 06, 2004 6:15
PM
Subject: [Asterisk-Users] SIP rtp port
forcing
When a SIP call starts (INVITE
/ 200 OK), asterisk seems to create a
Does it removes the need of external databases (mysql, postgres) or it will
work with existing databases?
-Kannaiyan
- Original Message -
From: Brian West [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
[EMAIL PROTECTED]
Sent: Sunday, September 05, 2004
DigitNetworks (and the like) are selling inferior, driver compatible
hardware, which is ABSOLUTELY NOT the same hardware Digium chose to be
their X100P.
If you can see my previous mails, no user worried about the
hardware whether it is from digium or intel or whether it is having
Does it mean that we cannot talk about Cisco or other FXS products since
IAXy is released??
I hope this list for every member who uses asterisk not Digium's products
users alone.
- Original Message -
From: Jay Milk [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial
of you use every day at home and work?
On Fri, 3 Sep 2004 08:40:59 +0100, Kannaiyan Natesan
[EMAIL PROTECTED] wrote:
Does it mean that we cannot talk about Cisco or other FXS products since
IAXy is released??
I hope this list for every member who uses asterisk not Digium's products
users alone
Can Anybody help how to reject an incoming call using 7960?
-Kannaiyan
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You can use it for interconnection with the local corporate office.
Internet --- Asterisk -- TE405 -- (E1/T1) Channel Banks
FXS Ports to company employees
Which is legally allowed in INDIA.
Not sure of any resellers in INDIA at this moment.
-Kannaiyan
- Original Message -
The hardware is not capable of generating the signals required by the FXS
line.
Actually a X100P card is a modem which interfaces a telephone line and
manipulates the signals on the line with the zaptel driver. It does nothing
with the hardware part on it.
I think some of the hardware geeks
It is certainly possible to play a sound in a channel when the voice stream
is through *.
You can check the source code of app_dial.c which contain such information.
-Kannaiyan
- Original Message -
From: shabanip [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, July 30, 2004 7:48
Eventhough I don't want to misuse the list, I still want to share
information.
My *SIP REGISTRATION* goes successfull with asterisk.
To be sure, I don't want to touch sip.conf until the udp packets receive my
machine.
I don't blame asterisk on my part until then.
I hope ethereal works in
Title: Trouble compiling asterisk-addons MySQL
Did you check whether mysql client libraries and headers were
installed?
-Kannaiyan
http://www.goods2world.com- Your
VoIP Shop
- Original Message -
From:
James
Freire
To: [EMAIL PROTECTED]
Sent: Wednesday, July
Great.
I have got the line a month back. No Incoming calls until now.
It was nice to see, Providing you with world class service and value is
our mission.. Still waiting to get the support.
If you think they are poor and want your money back you can read following,
Try BroadVoice service
I bought broadvoice service a month ago, I'm still yet to receive the
inbound calls.
They were solving it for more than six weeks. I really dunno how difficult
is that.
Outgoing calls get very perfect.
When I ask regarding the problem, they dunno where the call is handled, but
as far as myself
You can also grab a copy from:
http://www.goods2world.com/downloads/
-Kannaiyan
- Original Message -
From: Mark Spencer [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; [EMAIL PROTECTED];
[EMAIL PROTECTED]
Sent: Saturday, July 17, 2004 7:17 AM
Subject: [Asterisk-Users] Asterisk-1.0 RC1
We
As for "doing something better", I would hope to see two
(B things happening ...
(B
(B 1) you begin to use queue_app for your call centre
(B requirements and if you need any assistance, you ask about
(B it here
(B
(B 2) having experienced Asterisk's superior queue management
(B system,
Kannaiyan Natesan wrote:
(B
(B I hope you clearly understand that everyone here
(B **KNOWS**
(B to use alternative software such as SER, what is needed
(B here is
(B the same facility in asterisk.
(B
(B You have not shown us ANY example yet for which this
(B facility is *NEEDED*.
(B
(B
Based upon the analysis I think we need to modify two things,
(B
(B1. chan_sip.c (Registrar)
(B2. app_dial.c (Dial Command for simultaneous dialling, as of now it
(Bsupports simultaneous dialling too)
(B
(BThe registrar of SIP need to collect the array of registrants and the Dial
(Bcommand
ubject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
(B
(B
(B Kannaiyan Natesan wrote:
(B
(B Have you used 5 welcome service in fwd?
(B If not try that. You are invited to join as a volunteer
(B to provide support and talk to new people on fwd.
(B
(B Asterisk can do that m
* No, there's no quick fix for a 100 USD bounty
How much you estimate on quick fix?
-Kannaiyan.
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You need to go through all the messages again.
You feel it as stupid, I sometimes feel * is stupid and could not accept
this simply. To resolve this problem I simply use SER. But it is another
software, that makes me worried.
If you still think this is not at all needed. I better suggest you not
I hope you clearly understand that everyone here **KNOWS** to use
(Balternative software such as SER, what is needed here is the same facility
(Bin asterisk.
(B
(BWhen I see beauty, I want to see all the things in a same figure rather than
(Bsplitted. If you see with splitted face doen't mean
If you talk about alternatives and loosing my hair, the *BEST* solution for
me to go for is SER, which is fast, best and reliable SIP Proxy and we do
that right now for the same purpose. Also the discussion is not about
choosing alternatives but having it on the same.
-Kannaiyan.
-
I accept your views.
(B
(BI have a specific requirements, can you help to attain the same.
(BIn our business we have 25 employees handling customer service.
(B
(BI want to add or remove employees in the customer service so does the
(Bdevices connected to it.
(BI don't want to make any
I explained him a sample need.
(BI don't think asterisk does whatever i want in sip. It is an good PBX.
(B
(BPlease help me to understand. Anywhere am I wrong ? Or as you say is that
(BSIP feature is written?
(B
(B-Kannaiyan.
(B
(B
(B- Original Message -
(BFrom: "usedcanon"
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Kannaiyan Natesan
Sent: Sunday, July 11, 2004 1:15 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] New
Paul,
The question is very simple.
When I call a SIP user, the phone should ring in more than one
extentions. Also more than one phone should be able to register with
asterisk. Right now it is not the case. The last phone which register will
be receiving the calls. This type of
Have you tried 'D' option in dial.
exten = _395X.,1,Dial(SIP/[EMAIL PROTECTED],D(wwPINNOww${EXTEN:3}))
-Kannaiyan
- Original Message -
From: Vassilis Konstantinou [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, July 04, 2004 9:31 AM
Subject: [Asterisk-Users] Using call
have you tried http://www.goods2world.com or http://www.digitnetworks.com
- Original Message -
From: Brian Weaver [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, June 28, 2004 1:16 AM
Subject: [Asterisk-Users] Source for MD3200 modem cards?
It seems that the Apollo MD3200
I have this line in my extensions.conf,
exten = _393.,1,Dial(ZAP/${EXTEN:3},20,tr)
when I make a zap call, it gives me two rings and then makes the zap call.
Is there is a way I can make the call immediate?
Kannaiyan
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Thanks Eric. That works.
Kannaiyan
- Original Message -
From: Eric Wieling [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, June 24, 2004 3:18 PM
Subject: Re: [Asterisk-Users] Delay in Zap Calls?
On Thu, 2004-06-24 at 03:35, Kannaiyan Natesan wrote:
I have this line in my
Any body used patch,
http://bugs.digium.com/bug_view_page.php?bug_id=0001719
to get the callerid for BT Line.
I applied the patch successfully but could not get it to work.
Any help.
Here are the logs:
-- Starting simple switch on 'Zap/1-1'
Jun 17 18:22:31 NOTICE[426000]: chan_zap.c:4811
[snip]
My Zapata.conf:
usecallerid=yes
ukcallerid=yes
Change those two lines to simply usecallerid=uk.
I changed as you said and restarted asterisk. Still doesn't work.
-- Starting simple switch on 'Zap/1-1'
Jun 17 19:24:48 ERROR[262160]: chan_zap.c:4759 ss_thread:
it will default to the US style.
Make sure you have the uk settings in zaptel.conf. Can you see the
callerid
with a std phone on the line?
Chris
- Original Message -
From: Kannaiyan Natesan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, June 17, 2004 7:20 PM
Subject: Re
.
Anybody got it working on the TDM400P yet?
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Kannaiyan
Natesan
Sent: 17 June 2004 19:59
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] BT Caller ID - From Patch ?
I have the following settings
helped me
to solved this problem.
Kannaiyan
- Original Message -
From: Chris Stenton [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, June 17, 2004 9:24 PM
Subject: Re: [Asterisk-Users] BT Caller ID - From Patch ?
On Thu, 2004-06-17 at 21:02, Kannaiyan Natesan wrote:
I get
?
Also I believe the patch is in the driver to detect distinctive ring
tone for home/business calls. Look in the archive to set it up for
a BT line.
Chris
On Thu, 2004-06-17 at 21:02, Kannaiyan Natesan wrote:
I get this error when i receive the call on the line. Also I have Call
Sign
I'm using both for the past one year.
Facing Obsolutely *ZERO PROBLEMS* on both.
As a matter of fact I always like to spend less money.
I don't mind whether you put a DSP chip or no chips.
- Original Message -
From: Matt [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, June 16,
On the hardware page for the X100P card is says it's great for handling
incoming calls. It says nothing about making outgoing calls. Is it at
all possible to use that card to make outgoing calls from Asterisk to
the PSTN lines?
You can use it to make outgoing calls.
Kannaiyan
Title: RE: [Asterisk-Users] Still looking for small fxo sip gateway
The wonder is none of the FXO devices works fine except
asterisk X100P.
I'm not sure what is the stupidity present in that analog
technology.
Kannaiyan
- Original Message -
From:
Kostur,
Andre
To:
As far I have reasearched none of the FXO devices were perfect except Cisco
VIC ones.
If you are looking for reliability I recommend not to use those ones except
cisco.
Kannaiyan
- Original Message -
From: Michael Graves [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, February
You can use,
;sip.conf
register = username:[EMAIL PROTECTED]/extension
to make asterisk as a SIP client.
to forward calls to another client use canreinvite=yes, (if the client
supports reinvite)
and in the extensions.conf
exten = s,1,Dial(SIP/username:[EMAIL PROTECTED])
Kannaiyan
-
act like a normal sip phone?
On Fri, Jan 30, 2004 at 08:20:48AM -, Kannaiyan Natesan wrote:
You can use,
;sip.conf
register = username:[EMAIL PROTECTED]/extension
to make asterisk as a SIP client.
[...]
Can Asterisk act like a normal Sip phone and e.g. connect
I have seen bkw supporting to nufone.net under any circumstances.
Whoever you may be, as a seller you can reject providing services, but I
don't see anything good to fire a customer. Is MBA is one for that?
Is there is any good commision by any chance? from nufone.net. I could not
find any direct
Means RFC3389 support is incomplete. Neither Mark or
developers @ digiumwill not accept it when it gets completed by
anyone.
The best way is to turn off client if possible.
:-)
Please change the settings on your client todisable VAD
settings. That will remove that Notice.
Kannaiyan
You did not just paid for the card alone. You have paid for
support too.
Just request for the support from digium to get it to
work.
Kannaiyan
- Original Message -
From:
Deepakumar JV
To: [EMAIL PROTECTED]
Sent: Thursday, January 29, 2004 6:33
AM
Subject:
the development of asterisk.
Kannaiyan
- Original Message -
From: Doug Meredith [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, January 29, 2004 1:35 PM
Subject: [Asterisk-Users] Re: SIP error
Kannaiyan Natesan [EMAIL PROTECTED] wrote:
Means RFC3389 support is incomplete. Neither
Why don't we make RFC3389 support complete.
Is there is any progress around on that?
Kannaiyan
- Original Message -
From: Brian West [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, January 24, 2004 1:49 AM
Subject: Re: [Asterisk-Users] RFC3389 support issue with DG104S
Its
You are having Cisco 7960G behind NAT.
Try with nat=yes
I'm not sure any other settings will solve that in
asterisk.
I have tried but no luck.
Kannaiyan
- Original Message -
From:
Chris Wilson
To: [EMAIL PROTECTED]
Sent: Saturday, January 24, 2004 8:26
AM
I could not see anything there that is working.
Even the normal SIP connection. (Gives noisy output in the phone)
(Doesn't support stun) It is not NAT friendly.
FXO is utter waste option on this.
I have tried with the filter as what i read in the previous email in the
list, that doesn't works.
Does that connects VoIP to PSTN or only on Fail Over, means it changes from
choosing the PSTN line instead of VoIP line?
Kannaiyan
- Original Message -
From: Senad Jordanovic [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, January 23, 2004 9:44 AM
Subject: RE: [Asterisk-Users]
I have reported to clipcomm, but they were on holidays until end of this
month.
Kannaiyan
- Original Message -
From: Senad Jordanovic [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, January 22, 2004 4:29 PM
Subject: RE: [Asterisk-Users] Standalone FXO device
Kannaiyan Natesan
Can anyone give an idea how much does it cost if we want to buy the Licensed
asterisk source code?
I hope asterisk has two type of licenses,
1. GPL
2. I can buy and develop software on my own.
Am I right ?
Kannaiyan
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Have anyone tried to interface BT's Broadband Voice with asterisk?
Kannaiyan
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BT broadband voice uses ATA-186s configured as MGCP devices.
I think asterisk supports MGCP. I want to configure MGCP with asterisk to
connect to my BT Broadband Voice.
Do you have any idea relating to that.
Kannaiyan
___
Asterisk-Users mailing
-Users] UK BT Interface with asterisk?
Kannaiyan Natesan said:
Have anyone tried to interface BT's Broadband Voice with asterisk?
Kannaiyan
___
No, and not sure of their rates but http://www.telappliant.com/ has good
rates, voice quality
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, January 23, 2004 8:07 PM
Subject: Re: [Asterisk-Users] UK BT Interface with asterisk?
Kannaiyan Natesan said:
Do they offers, free evening and weekend calls? I get from BT.
You can get a free 0870
Is it so hard to put X100P as a ethernet device?
I have been trying FXO devices, but gets me luck.
Kannaiyan
- Original Message -
From: Rich Adamson [EMAIL PROTECTED]
To: Asterisk-a-users-list [EMAIL PROTECTED]
Sent: Friday, January 23, 2004 7:40 PM
Subject: [Asterisk-Users] Mediatrix
If you are not users from mysql database then you can disable in the
makefile.
For this,
USE_MYSQL_FRIENDS=1
change it to
USE_MYSQL_FRIENDS=0
You won't get that error.
Alternatively you can install mysqlclient library to compile it without
errors.
Kannaiyan
- Original Message -
There is no CDR for the call from spool outgoing,
You need to write a patch to solve the same.
Kannaiyan
- Original Message -
From: Iain Stevenson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, January 23, 2004 8:27 PM
Subject: [Asterisk-Users] Back to front logging for calls
Can anyone recommend me a fxo device with SIP or IAX functionality.
I have tried with ,
http://www.clipcomm.co.kr/
They were worster than any device. Device itself costed me $270/- including
shipping but not working.
Kannaiyan
___
Asterisk-Users
in the list.
Kannaiyan
- Original Message -
From: Senad Jordanovic [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, January 22, 2004 12:38 PM
Subject: RE: [Asterisk-Users] Standalone FXO device
Kannaiyan Natesan wrote:
Can anyone recommend me a fxo device with SIP or IAX functionality
Hi,
I think canreinvite=yes won't work in most of the situations.
I have implemented Redirect SIP 300 Message to redirect to the SIP
address you speficy in the sip.conf.
Where you can have ,
register = username:[EMAIL PROTECTED]/extension
[extension]
redirect=yes
I was looking for this info and thought I'd post it here for all.
Codec BR NEB
G.711 64 Kbps 87.2 Kbps
G.729 8 Kbps 31.2 Kbps
G.723.1 6.4 Kbps 21.9 Kbps
G.723.1 5.3 Kbps 20.8 Kbps
G.726 32 Kbps 55.2 Kbps
G.726 24 Kbps 47.2 Kbps
G.728 16 Kbps
Al
--- Kannaiyan Natesan [EMAIL PROTECTED] wrote:
Hi,
I think canreinvite=yes won't work in most of the
situations.
I have implemented Redirect SIP 300 Message to
redirect to the SIP
address you speficy in the sip.conf.
Where you can have ,
register
Is the SIP bin same for IAX as well?
Kannaiyan
- Original Message -
From: Christian Stredicke [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, January 19, 2004 7:08 PM
Subject: RE: [Asterisk-Users] SNOM IAX image
For those who are using snom 200 phones, I think we have a
I have coded chan_sip.c so that you can have
// sip.conf
register = username:[EMAIL PROTECTED]/redirectconfig
[redirectconfig]
redirect=yes
redirecturi=sip:[EMAIL PROTECTED]
redirecturi=sip:[EMAIL PROTECTED]
redirecturi=sip:[EMAIL PROTECTED]
so when you receive a call it will redirect to
/O
Attached is the Debug information with the 300 Redirect implementation
with asterisk,
You can get the source code from
http://www.speak2world.com/asterisk/chan_sip.php
and when you compile and run it, you get the following info in the debug
o/p.
pbx*CLI sip debug
SIP Debugging
I want to share the asterisk capability with the number of calls i pass
through to give a better analysis about asterisk. is there is any existing
module in asterisk or any body is writing the same module which can perform
this.
For Eg.,
P4 Machine
64 MB RAM
zaptel/rip routing/iax routing etc .,
I have seen in the list been receiving the HTML encoded part in the list, is
it possible to strip off the HTML part and to keep the text alone, so that
it will be clean and simple to read, both in the list and in the web.
Is there is any other reason to keep the HTML contents in?
Kannaiyan
Anyone tried Asterisk with Virtual PC ?
I want to have windows and linux on the same machine and to run
simultaneously with asterisk.
Any help.
Kannaiyan
http://www.speak2world.com -- Test your IAX Connection
___
Asterisk-Users mailing list
[EMAIL
Hi,
I don't have a zaptel device for conferencing.
I read from the lists, that
ztdummy and zaprtc need to be installed to get conferencing.
I could able to compile successfully with ztdummy and when i receive the
call it says,
-- Goto (13732,s,1)
-- Executing
Thanks for the reply but it could not solve my problem.
Did you modprobe ztdummy?
modprobe ztdummy
modprobe: Can't open dependencies file /lib/modules/2.4.20-6um/modules.dep
(No such file or directory)
Can you please guide me what should I do for this?
It should return nothing(successfully).
I want to join two calls invoked from asterisk,
Here is my 1.call in /var/spool/asterisk/outgoing,
Channel: IAX2/[EMAIL PROTECTED]/847512,20,tr
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: 13732
Extension: s
Priority: 1
it successfully rings at extension 847512 and I could answer the call.
Hi SW,
The procedure is,
1. Submit the old registration key
2. Get the new License Key. ( Old + New )
3. Then install with a single registration key.
(You can try to ask for the procedure how to destroy your previous
license, that will make asterisk clean)
If you try to
Hi,
Below if the error message which I got from asterisk.
I was trying to fax to asterisk from my fax machine. I really dunno what
is the problem. I use alaw ulaw codec only through my ATA 186. Can anyone
help me what could be the problem.
-- Executing Goto(SIP/-080ef9a0, 13732|s|1) in
I have a problem with IAX call transfer. The call goes successful but
consumes lot of BW in the middle tier.
The actual connection is like this
(NAT) DIAX(IAX2) - *1 -- *2
*1 *2 were public IP with asterisk.
It consumes around 120kbps in total to forward a single GSM call.
I have the
I have my connection as below,
diax(IAX) --- (IAX) * (IAX) -- IAX(*) --- PSTN
In the middle tier of asterisk, it if not completely forwarding the
stream and it consumes the system bandwidth.
I DONT have settings like notransfer=yes
Can you please help me how can I
Title: SquareTrade: My Seal
Anybody guide me, whether I can have instant
hosting and a websetup for asterisk with any providers?
At the basic I want to have sip hosting to my
domain which can handle sip calls and call forwarding etc.,.
Thanks in advance.
Kannaiyan
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