Hi All,
I need a help on g729 codec.Is there any tool which can convert g711 codec into
g729 codec and supports batch processing ?
Thanks in advance
vivek
--- On Fri, 11/7/08, Edgar Guadamuz [EMAIL PROTECTED] wrote:
From: Edgar Guadamuz [EMAIL PROTECTED]
Subject: Re: [asterisk-users
Hi,
I've just followed
http://www.voip-info.org/wiki/view/Asterisk+Speaks+with+Google+Talkinstructions
from wiki,
And i always get my jabber (GoogleTalk account for asterisk server) not
registred:
Hi,
I would like to seek an opinion or list of providers in USA or particularly
in California. We would need someone who can offer maximum ports and lowest
rates.
Thanks very much,
Vivek
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Hi,
try adding this in your stdtime/localtime.c
#define _POSIX_PTHREAD_SEMANTICS
#undef TM_ZONE
#undef TM_GMTOFF
if this does not work just google it, there are workaround for this problem
Thanks,
Vivek
On 12/2/07, Mike Clark [EMAIL PROTECTED] wrote:
I submiited to the list
I am not sure if this fits in your requirement but try dial command.
--Vivek
On 11/29/07, Olivier [EMAIL PROTECTED] wrote:
Hi,
I would like to originate my first call from CLI.
As I'm new to this, I'm wondering if it's possible.
When I type originate from CLI, I've got
Hi,
x-lite has extensive debug facility you can turn that on in the advanced
options, that probably will give better understanding as what is going on
from x-lite side. i also have experienced the same but that involved
firewall and NAT issues.
Thanks,
Vivek
On 11/30/07, Newbie [EMAIL
well, then i would recommend to see full log in debug mode that might give
some clue. if you have not done this before you can uncomment line starting
with full= in the logger.conf... the log will be the usual
/var/log/asterisk/ directory.
Thanks,
Vivek
On 11/30/07, Newbie [EMAIL PROTECTED
- Original Message -
*From:* Vivek Shrivastava [EMAIL PROTECTED]
*To:* Newbie [EMAIL PROTECTED]
*Cc:* Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
*Sent:* Saturday, December 01, 2007 11:50 AM
*Subject:* Re: [asterisk-users] Registration state
you can also look at this...
http://www.asteriskguru.com/tutorials/idefisk_20_free.html
I has this error initially with Asterisk server when I try to register.
Device does not match ACL
got it resolved by setting Caller ID Name : users exten
On 11/30/07, Vivek Shrivastava [EMAIL
yup with chan_oss
On 11/30/07, Olivier [EMAIL PROTECTED] wrote:
2007/11/30, Vivek Shrivastava [EMAIL PROTECTED]:
I am not sure if this fits in your requirement but try dial command.
Do you mean, dialing both extensions one after the other and then, bridge
them ?
Or do you mean using
looks like something wrong with the dial plan in the extensions.conf.. i
would recommend start debug on and see the content of full log may be
that give some clue.
Thanks,
Vivek
On 11/30/07, Russell Brown [EMAIL PROTECTED] wrote:
I have two Asterisk systems that can route to each other via
you can try Cain Abel ( to route calls) and Wireshark to record all the
calls.
On 11/29/07, Adam Moffett [EMAIL PROTECTED] wrote:
I'm pretty sure asterisk won't do that without modification. You'll
need to do packet sniffing and decode the datathere may be products
that do this, but
We are using only voip chanels with 400-500 channels. Although we are still
in begining phase but i have not seen any problem as such.
Thanks,
Vivek
On 11/20/07, Mark Adams [EMAIL PROTECTED] wrote:
I wanted to see if anyone has set up a large amount of out bound only
voip channels?
We
Hi Ryan,
Are the SIP and RTP ports are randomly selected or there are specific ports
for these? Unchecking
random port selection option on the device/softphone may help.
--Vivek
On 11/10/07, Ryan Newington [EMAIL PROTECTED] wrote:
Hi Luki,
Thanks for your advice. I've checked the firewall
Hi Ryan,
I was just wondering if they need to be according rtp.conf. ( or you may
need to modify rtp.conf)
Regards,
Vivek
On 11/11/07, Ryan Newington [EMAIL PROTECTED] wrote:
Hi Vivek,
The SIP port is set to the standard port 5060. The RTP ports as far as I
know are random ephemeral
I would recommed to convert that to gsm format
http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk
On 11/11/07, Michael Schwartz [EMAIL PROTECTED] wrote:
I'm using Asterisk 1.4.13, the latest released version. The linux platform
is FC7.
I setup my
well i think rtp port range is defined in rtp.conf and correct me if i am
wrong, these ports must be opened/forwarded to communicate.
http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html
Let me know if you need more information.
Thanks,
Vivek
On 11/11/07, Ryan
if your router and UA have syslog facility you can use RouterSyslog also.
You can use Cain and Able with wireshark for switched network.
Thanks,
Vivek
On 11/9/07, Alan Lord [EMAIL PROTECTED] wrote:
Steve Edwards wrote:
snip /
Examples of what I'd like to see:
1) A SIP telephone
Well, unfortunately i did not dig much into why/how it worked with
openvpn, but it did work for me with default setup.I think you may need to
set constant ports instead of random ports.
Thanks,
Vivek
On 11/9/07, bilal ghayyad [EMAIL PROTECTED] wrote:
Hi Friends;
Actually I would appreciate
I think you can save/get the number in variable and then assign it to
callerid. I am doing similar and working for me.
Thanks,
Viv
On 11/8/07, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote:
Is there any way to see the called number when a call gets redirected to
the 'a' extension from
Hi,
i am facing some problem configuring 2 Grandstrem phones 101 behind NAT. I
have put server ip as 192.x.x.x and OutGoing proxy as 72.x.x.x, i have
forwarded ports on both Grandstream and Asterisk sides, and using those
ports on Grandstream for SIP and RTP with random ports =no. This setup is
yeah i found openvpn helpful in NAT cases.
-Vivek
On 11/6/07, Baji Panchumarti [EMAIL PROTECTED] wrote:
after a copious loss of follicles :-), I finally got outbound working.
Basically the channel statement in the call file needs to have the
number to be called. For eg., in test.call
Hi,
Yes, i have used it for T.38 faxing.
Thanks,
Vivek
On 10/26/07, Nasir Iqbal [EMAIL PROTECTED] wrote:
Hi,
Have you tried Callweaver http://www.callweaver.org
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Hello friends,
I am trying to install asterisk 1.4.0 . I am configuring it as follows:-
./configure --prefix=/home/vivek/downloads/install/asterisk/
But still while running 'make install', it tries to install it in
/var/lib/asterisk/ and stops because of failing permissions.
I have provided
Hello friends,
I am trying to install asterisk 1.4. I am configuring it as follows:-
./configure --prefix=/home/vivek/downloads/install/asterisk/
But still while running 'make install', it tries to install it in
/var/lib/asterisk/ and stops because of failing permissions.
I have provided
I am getting this warning:-
Oct 23 15:47:22 WARNING[2124]: db.c:171 ast_db_put: Unable to put value '
192.168.1.12:5060:300:15553695861:sip:[EMAIL PROTECTED]:5060' for key '23'
in family 'SIP/Registry
I checked the file permissions. They are proper. There doesnot seem to be a
visible error. No
.
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
All science is either physics or stamp collecting.
-- Ernest Rutherford
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asterisk-users mailing
I checked the file permissions. They are proper. There doesnot seem to be a
visible error. No change has been done in any conf files for the past 4 months.
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
All science is either physics or stamp collecting
,Set(sipcid = ${CUT(SIPCALLID,-,1)}) --- evaluates to E305CEC5
I want this hex value in int. But i cant think of a clean solution.
Please help.
Thanks in advance.
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
All science is either physics or stamp
Hi Michael,
Thanks a lot. I am working on an agi script and it does it. Thanks a lot again.
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
All science is either physics or stamp collecting.
-- Ernest Rutherford
Michiel van Baak wrote
:10:34 WARNING[30029]: channel.c:787 channel_find_locked: Avoided
initial deadlock for '0x81bbd78', 10 retries!
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
All science is either physics or stamp collecting.
-- Ernest Rutherford
touch the file ilbc.o and
ilbc.so. But it wouldnot help. Please suggest how do I go further with this?
Thankyou all in advance.
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
All science is either physics or stamp collecting.
-- Ernest
Hello friends, does anyone know if there is a gui for asterisk provided with
the asterisk source or has to downloaded from somewhere else.
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
All science is either physics or stamp collecting
Hi all,
How do I download the development branch of asterisk 1.4. I am eagerly
waiting for it.
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
All science is either physics or stamp collecting.
-- Ernest Rutherford
Hello friends,
I want to use the background(playfile) application without the channel
being answered. I dont want playback because I would like the callee to dial
the number while the file is being played. but I dont know how do i do that.
With warm regards.
Vivek J. Joshi.
[EMAIL
dont know what to do. Has anyone
got an audiocodes with asterisk working. Please help me with some
configurations in audiocodes
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
All science is either physics or stamp collecting.
-- Ernest
.
Does anyone know to do this or has done this before?
Please share your experiences please.
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
--New opinions often appear first as jokes and fancies, then as blasphemies and
treason, then as questions open
which I want to
implement.
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
--New opinions often appear first as jokes and fancies, then as blasphemies and
treason, then as questions open to discussion, and finally as established
truths.
Joseph Tanner
regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
--Sweat saves blood, blood saves lives, and brains saves both.
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Asterisk-Users mailing list
To UNSUBSCRIBE or update options
whereby I can detect the dropped
packets or enable their queueing or buffering?
Please help, I am running out of ideas.
Thanking you all.
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
--Sweat saves blood, blood saves lives, and brains saves both
help me how do I track this.
Thanks all for reading this mail.
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
--Sweat saves blood, blood saves lives, and brains saves both.
___
--Bandwidth and Colocation provided
regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
--Sweat saves blood, blood saves lives, and brains saves both.
___
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Asterisk-Users mailing list
To UNSUBSCRIBE or update options
if you could help me.
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
--Sweat saves blood, blood saves lives, and brains saves both.
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
--Sweat saves blood, blood saves lives
could help me.
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
--Sweat saves blood, blood saves lives, and brains saves both.
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Dear Paul H.,
Thanks my dear friend, that worked.
Thanks a lot for the help.
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
--Sweat saves blood, blood saves lives, and brains saves both.
___
--Bandwidth
connect.
Thanks for reading this.
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
--Sweat saves blood, blood saves lives, and brains saves both.
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Asterisk
, rename the conflicting
verisons of chan_h323.so, or chan_oh323.so, or chan_ooh323.so from asterisk
modules to something else.
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
--Sweat saves blood, blood saves lives, and brains saves both
Hello Friends,
I was trying to dial agents from a normal extension. My extensions.conf is
configured as
exten = 11,1,AgentCallbackLogin
exten = 12,1,Dial(Agent/12) ;; configured in agents.conf as agent =
12,12, vivek
exten = 13,1,Dial(SIP/13) ,, is configured in sip.conf
to do it but was
unsuccessful.
Please tell me if there is a tweak or a workaround for this.
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
--Optimism is a mania for saying things are well when one is in hell
Thanks a lot Mr. Alexander Lopez for your prompt attension.
I tried the same thing but it wouldnot happen. I use it as:-
exten = 12,1,Dial(Agent/12)
exten = 12,2,Hangup
where agent 12 is configured as :-
agent = 12,12, vivek
After the agent is logged in on extension no12 as follows
Callback
= 12,12,vivek
I am not able to figure out why would not it dial agent 12.
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
--Optimism is a mania for saying things are well when one is in hell.
Alexander Lopez wrote:
Can you tell me how agent 12 is logging
Thanks Mr.Miano
Thanks a lot. Now I think I wont have to bother about balming all my problems
to zapata. I have also succeeded quite a bit and installed a basic PBX system
without it.
Thanks a lot again.
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd
asterisk.org ?
I think I dont because I dont use a digium card but do I have to still
confugure for FXO and FXS ports?
Kindly help me solving my doubt.
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
--Truth springs from argument amongst friends
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
--Truth springs from argument amongst friends.
[EMAIL PROTECTED] wrote:
Hello friends,
I have a strange problem. I am using asterisk 1.2 and asterisk addons 1.2. I
have three SIP phones and one H323
are in caller groups 1
pickupgroup=1 ; We can do call pick-p for call group 1
;; rest of the sip users are configured in the same way.
Help will be very much appreciated. Kindly help. I am totally confused as to
where the fault is.
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED
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