hI
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To
/lib/asterisk/sounds/30-minutes-of-silence.gsm ;*
*
*
*
*
*
*
On Fri, Mar 5, 2010 at 4:36 AM, David @ULC ucoms2...@gmail.com wrote:
I believe we GSM of 8 bit for Asterisk ?
On Fri, Mar 5, 2010 at 4:35 AM, David @ULC ucoms2...@gmail.com wrote:
Record a muted channel for 30 minutes like
I need to create 30 mins of GSM file for Asterisk .
Silent / Blank file.
Whats the best way to create it ?
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, 2010 at 4:21 AM, David @ULC ucoms2...@gmail.com wrote:
I need to create 30 mins of GSM file for Asterisk .
Silent / Blank file.
Whats the best way to create it ?
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I believe we GSM of 8 bit for Asterisk ?
On Fri, Mar 5, 2010 at 4:35 AM, David @ULC ucoms2...@gmail.com wrote:
Record a muted channel for 30 minutes like this:
exten = s,1,Answer(1)
exten = s,n,Progress()
exten = s,n,record(silence_long.gsm|1800|s)
exten = s,n,hangup
*Code:*
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing Playback(Local/91441425477...@default-b9f2,1,
sip-silence) in new stack
-- Playing 'sip-silence' (language 'en')
-- Executing AGI(Local/91441425477...@default-b9f2,1, agi://
127.0.0.1:4577/call_log) in new stack
I changed my VOIP, and now things are ok.
But didnt understand, how can VOIP can affect it ?
On Wed, Feb 24, 2010 at 11:53 PM, David @ULC ucoms2...@gmail.com wrote:
*Code:*
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing Playback(Local/91441425477...@default-b9f2,1
Its crazy
I made it working .
Today I had to reinstall all due to soem reason.
Now, when I am trying, its NOT coming.
Same CPU, Same Lan, Same Windows which acts as Internet Gateway.
CALL Doesnt hit my Asterisk.
http://i50.tinypic.com/1z3axrc.jpg
http://i45.tinypic.com/23mr5uq.jpg
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How can I ask IPKall to send call to my Asterisk ?
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???
On Thu, Feb 18, 2010 at 2:45 AM, David @ULC ucoms2...@gmail.com wrote:
I dont have a Static IP.
How can I ask IPKall to send call to my Asterisk ?
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Looks like IdeaSip need STATIC ip else it doesnt work.
.
On Thu, Feb 18, 2010 at 3:02 AM, David @ULC ucoms2...@gmail.com wrote:
Ok
I can use
Dyndns.org
I registered myself.
easy.selfip.com
https://www.dyndns.com/account/services/hosts/easydialnow.selfip.com
successfully activated
for '11012012...@66.54.140.46' timed out, trying again (Attempt
#18)
When I try to Ping from my CentOS , I can ping 66.54.140.46.
On Thu, Feb 18, 2010 at 3:11 AM, David @ULC ucoms2...@gmail.com wrote:
Looks like IdeaSip need STATIC ip else it doesnt work.
.
On Thu, Feb 18, 2010 at 3:02 AM, David
Feb 17 19:19:04 NOTICE[2554]: chan_sip.c:10077 handle_response_peerpoke:
Peer '11012012600' is now TOO LAGGED! (2567ms / 2000ms)
On Thu, Feb 18, 2010 at 5:34 AM, David @ULC ucoms2...@gmail.com wrote:
hmmm Ok..
Is this a Asterisk Question ?
I have a setting as :
Global Settings
proxy.ideasip.com, port 5060
-- Got SIP response 479 Please don't use private IP addresses back
from 208.97.25.11
On Thu, Feb 18, 2010 at 5:34 AM, David @ULC ucoms2...@gmail.com wrote:
hmmm Ok..
Is this a Asterisk Question ?
I have a setting as :
Global Settings
http://i50.tinypic.com/120rwya.jpg
On Thu, Feb 18, 2010 at 7:12 AM, David @ULC ucoms2...@gmail.com wrote:
So, this will change :
register = 11012012600:passw...@proxy.ideasip.com/11012012600
[ideasip]
type=friend
secret=password
username=11012012600
host=proxy.ideasip.com
insecure
208.97.25.11
I cant use Ideasip ???
On Thu, Feb 18, 2010 at 7:12 AM, David @ULC ucoms2...@gmail.com wrote:
So, this will change :
register = 11012012600:passw...@proxy.ideasip.com/11012012600
[ideasip]
type=friend
secret=password
username=11012012600
host=proxy.ideasip.com
insecure
I use IdeaSip with IPKall.
How may channels are open when we use IdeaSip ?
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No way ??
On Thu, Nov 19, 2009 at 8:59 AM, David @ULC ucoms2...@gmail.com wrote:
Anyway to Increase Volume gain in Asterisk ?
USING g729 codec.
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Anyway to Increase Volume gain in Asterisk ?
USING g729 codec.
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Looking for Genuine VPS Server for 250 ports on Rent.
Anyone can help ?
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asterisk-users
Ideasip is down today ?
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I want to use IPKall with Asterisk.
Now, I want my calls to land at 2 different locations , not connected with
each other.
If I want to configure IPKall DID number in Asterisk , I need to specify IP
on IPKall.
How can I make it enable so that calls can land up at both locations ?
I am GETTING tired of CALLS getting stucked !!!
Any help ?
On Thu, Sep 10, 2009 at 2:52 AM, David @ULC ucoms2...@gmail.com wrote:
8186223080 : No entry in the log file !!!
On Thu, Sep 10, 2009 at 2:06 AM, David @ULC ucoms2...@gmail.com wrote:
Local/718186223...@d 718186223...@default
I am using asterisk.
I also have an access to VOIPSwitch ver 2 where I can see live calls.
Many times I have seen that my calls are getting strucked and then it gets
disconneected after 59 mins ( as settings are done accordingly in
VOIPSwitch)
What could be the reason ?
had ?
On Wed, Sep 9, 2009 at 11:41 PM, David @ULC ucoms2...@gmail.com wrote:
I am using asterisk.
I also have an access to VOIPSwitch ver 2 where I can see live calls.
Many times I have seen that my calls are getting strucked and then it gets
disconneected after 59 mins ( as settings
Local/718186223...@d 718186223...@default Up
Dial(SIP/18186223...@sip209||t
I see this in my Asterisk when I do
show channels
On Thu, Sep 10, 2009 at 1:49 AM, David @ULC ucoms2...@gmail.com wrote:
I don't know where is the problem. May be with VOIPSwitch OR may be with
Asterisk..
Call
:(none)
On Thu, Sep 10, 2009 at 2:06 AM, David @ULC ucoms2...@gmail.com wrote:
Local/718186223...@d 718186223...@default Up
Dial(SIP/18186223...@sip209||t
I see this in my Asterisk when I do
show channels
On Thu, Sep 10, 2009 at 1:49 AM, David @ULC ucoms2...@gmail.com wrote
I see this : /etc/asterisk/logger.conf
[logfiles]
console = notice,warning,error
messages = notice,warning,error,debug,verbose
On Thu, Sep 10, 2009 at 2:11 AM, David @ULC ucoms2...@gmail.com wrote:
*Details :*
* SIP Call
Direction: Outgoing
Call-ID
8186223080 : No entry in the log file !!!
On Thu, Sep 10, 2009 at 2:06 AM, David @ULC ucoms2...@gmail.com wrote:
Local/718186223...@d 718186223...@default Up
Dial(SIP/18186223...@sip209||t
I see this in my Asterisk when I do
show channels
On Thu, Sep 10, 2009 at 1:49 AM, David @ULC
Below link show the download link for Call Aanalyzer and install procedure :
http://www.757.org/~joat/wiki/index.php/Viewing_CDR_Data_with_Asterisk_CDR_Analyzer
But how to create DB in mysql and what wld be he structure ?
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DB created.
and Restarted asterisk.
But No data getting stored in cdr table.
Anything step by step to install that application ?
On Thu, Sep 10, 2009 at 3:29 AM, David @ULC ucoms2...@gmail.com wrote:
Below link show the download link for Call Aanalyzer and install procedure
:
http
Asterisk-Addons : No
On Thu, Sep 10, 2009 at 3:47 AM, David @ULC ucoms2...@gmail.com wrote:
DB created.
and Restarted asterisk.
But No data getting stored in cdr table.
Anything step by step to install that application ?
On Thu, Sep 10, 2009 at 3:29 AM, David @ULC ucoms2
*I am getting below CLI in my asterisk :*
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing AGI(SIP/cc101-b7910cc0, agi://127.0.0.1:4577/call_log)
in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial(SIP/cc101-b7910cc0,
I have 2 sips configured :
1) register =sama:xx...@209.51.191.xxx:5060
2) register =sama:xx...@209.51.192.xxx:5060
Both are active.
5060 port will be same or different ?
On Wed, Sep 9, 2009 at 12:29 AM, David @ULC ucoms2...@gmail.com wrote:
*I am getting below CLI in my asterisk
How to extract that CDR from asterisk ?
On Sat, Aug 29, 2009 at 2:11 AM, David @ULC ucoms2...@gmail.com wrote:
From Asterisk, I need a List of Numbers , asterisk dialed out.
I am looking for status of each number dialed out.
Whether its failed or successful .
Any way
Ok Got it.
Any 3rd party Interface which can get me all these result in a front end ?
On Mon, Aug 31, 2009 at 8:19 PM, David @ULC ucoms2...@gmail.com wrote:
How to extract that CDR from asterisk ?
On Sat, Aug 29, 2009 at 2:11 AM, David @ULC ucoms2...@gmail.com wrote:
From Asterisk, I need
To view the post and reply , I always to use below link..
http://lists.digium.com/pipermail/asterisk-users/2009-August/thread.htmlhttp://lists.digium.com/pipermail/asterisk-users/2009-February/thread.html
Any better way to access the forum ?
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On Mon, Aug 24, 2009 at 11:39 PM, David @ULC ucoms2...@gmail.com wrote:
Oh my god..
Today its saying there is NOONE to take your call.I am using IdeaSIP
What could be the reasons ?
It was working perfectly till saturday .
On Thu
From Asterisk, I need a List of Numbers , asterisk dialed out.
I am looking for status of each number dialed out.
Whether its failed or successful .
Any way ?
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Oh my god..
Today its saying there is NOONE to take your call.I am using IdeaSIP
What could be the reasons ?
It was working perfectly till saturday .
On Thu, Aug 20, 2009 at 11:54 PM, David @ULC ucoms2...@gmail.com wrote:
IdeaSIP worked perfect for me.
On Thu, Aug 20, 2009 at 11:27 PM
you're not logged in means ?
On Mon, Aug 24, 2009 at 11:39 PM, David @ULC ucoms2...@gmail.com wrote:
Oh my god..
Today its saying there is NOONE to take your call.I am using IdeaSIP
What could be the reasons ?
It was working perfectly till saturday .
On Thu, Aug 20, 2009 at 11
We all know the FWD is NO more available.
How to set up IPKALL so that my Inbound number rings on my eyebeam or xlite
?
Any alternative for FWD ?
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On Thu, Aug 20, 2009 at 11:54 PM, David @ULC ucoms2...@gmail.com wrote:
IdeaSIP worked perfect for me.
On Thu, Aug 20, 2009 at 11:27 PM, David @ULC ucoms2...@gmail.com wrote:
We all know the FWD is NO more available.
How
IdeaSIP worked perfect for me.
On Thu, Aug 20, 2009 at 11:27 PM, David @ULC ucoms2...@gmail.com wrote:
We all know the FWD is NO more available.
How to set up IPKALL so that my Inbound number rings on my eyebeam or xlite
?
Any alternative for FWD
exten = _X48600XXX,1,MeetMeAdmin(${EXTEN:2},T,${EXTEN:0:1})
exten = _X48600XXX,2,Hangup
exten = _X38600XXX,1,MeetMeAdmin(${EXTEN:2},t,${EXTEN:0:1})
exten = _X38600XXX,2,Hangup
exten = _X28600XXX,1,MeetMeAdmin(${EXTEN:2},m,${EXTEN:0:1})
exten = _X28600XXX,2,Hangup
exten =
Any more suggestions ?
On Wed, Jul 1, 2009 at 8:30 AM, David @ULC ucoms2...@gmail.com wrote:
Thanks for the Reply,
I was waiting online for someone to reply : -)
Here is my Extension file : [ Where should I enter those line ? ]
exten = 8600099,1,Meetme(8600099)
exten = 8600100,1,Meetme
*
*
*/var/lib/asterisk/sounds/silence/1*
*
*
*1 is the folder or the filename ?*
On Wed, Jul 1, 2009 at 8:30 AM, David @ULC ucoms2...@gmail.com wrote:
Thanks for the Reply,
I was waiting online for someone to reply : -)
Here is my Extension file : [ Where should I enter those line
sound file is intact
Yes. I checked it with my other server.
On Wed, Jul 1, 2009 at 9:14 PM, David @ULC ucoms2...@gmail.com wrote:
*
*
*/var/lib/asterisk/sounds/silence/1*
*
*
*1 is the folder or the filename ?*
On Wed, Jul 1, 2009 at 8:30 AM, David @ULC ucoms2...@gmail.com wrote
, Jul 1, 2009 at 9:14 PM, David @ULC ucoms2...@gmail.com wrote:
*
*
*/var/lib/asterisk/sounds/silence/1*
*
*
*1 is the folder or the filename ?*
On Wed, Jul 1, 2009 at 8:30 AM, David @ULC ucoms2...@gmail.com wrote:
Thanks for the Reply,
I was waiting online for someone to reply
When I login to the asterisk, I just hear the HALF of the welcome message :
You are currently the instead of You are currently the only person in
the conference
Thats also, I hear it after 60 secs or so..
Asterisk 1.2.27
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, Jul 1, 2009 at 6:19 AM, David @ULC ucoms2...@gmail.com wrote:
When I login to the asterisk, I just hear the HALF of the welcome message :
You are currently the instead of You are currently the only person in
the conference
Thats also, I hear it after 60 secs or so..
Asterisk 1.2.27
What the best website and book to start learning asterisk ?
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I am from the eastern part of India and there is No institute to have a
formal training for Asterisk.
I would like to open One Institute dedicated for Training. ( Though I am
also learning )
Any advice is highly appreciated.
On Mon, Jun 22, 2009 at 10:50 PM, David @ULC ucoms2...@gmail.com wrote
Which one to download for CentOS ?
http://www.wireshark.org/download.html#thirdparty
On Sat, May 23, 2009 at 1:01 AM, David @ULC ucoms2...@gmail.com wrote:
We are facing alot of problem in the DTMF. At times we are unable to do the
verification because whenever we press the numbers
Thanks a lot but which one to download before installing ?
On Sat, May 23, 2009 at 1:01 AM, David @ULC ucoms2...@gmail.com wrote:
We are facing alot of problem in the DTMF. At times we are unable to do the
verification because whenever we press the numbers for verification it does
We are facing alot of problem in the DTMF. At times we are unable to do the
verification because whenever we press the numbers for verification it does
not detects and at times it detects the wrong number for instance if the
customer is having the phone no. as 1234567890 it will detect 123467890
Can this be due to G729 codec ?
If yes, how to Uninstall g729 ?
Asterisk 1.2.27 is the version.
On Sat, May 23, 2009 at 1:01 AM, David @ULC ucoms2...@gmail.com wrote:
We are facing alot of problem in the DTMF. At times we are unable to do the
verification because whenever we press
1)
disallow=all
allow=g729
allow=ulaw
context=default
dtmfmode=rfc2833
2) We use SIP.
3) IVR 3rd party verification.
4) VOIP
On Sat, May 23, 2009 at 1:01 AM, David @ULC ucoms2...@gmail.com wrote:
We are facing alot of problem in the DTMF. At times we are unable to do the
verification
No
On Sat, May 23, 2009 at 1:01 AM, David @ULC ucoms2...@gmail.com wrote:
We are facing alot of problem in the DTMF. At times we are unable to do the
verification because whenever we press the numbers for verification it does
not detects and at times it detects the wrong number for instance
We are using asterisk and sometime when our guys are on call , they hear
some voice of person and amazingly that person is NOT from our center.
Any one faced this kind of thing ?
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No such pattern.
It happens ad hoc.
On Tue, May 19, 2009 at 11:46 PM, David @ULC ucoms2...@gmail.com wrote:
We are using asterisk and sometime when our guys are on call , they hear
some voice of person and amazingly that person is NOT from our center.
Any one faced this kind of thing
I am usIng VOIP.
On Tue, May 19, 2009 at 11:46 PM, David @ULC ucoms2...@gmail.com wrote:
We are using asterisk and sometime when our guys are on call , they hear
some voice of person and amazingly that person is NOT from our center.
Any one faced this kind of thing
Some at 5:34 pm EST DAILY, all my call get disconnect.
I tried RE-INSTALLATION, I tried Reinstallation on a virgin HDD, but its
same.
I tried changing VOIP provider I tried changing Internet Provider..But no
help..
What could be the reason ?
Here are my enties of crontab :
### recording
Current time is
Wed May 20 02:05:41 EDT 2009 on the server
On Wed, May 20, 2009 at 3:56 AM, David @ULC ucoms2...@gmail.com wrote:
Some at 5:34 pm EST DAILY, all my call get disconnect.
I tried RE-INSTALLATION, I tried Reinstallation on a virgin HDD, but its
same.
I tried changing VOIP
Is it that server disconnect all calls at 1am EDT as things are configured
in that way ?
On Wed, May 20, 2009 at 4:08 AM, David @ULC ucoms2...@gmail.com wrote:
Current time is
Wed May 20 02:05:41 EDT 2009 on the server
On Wed, May 20, 2009 at 3:56 AM, David @ULC ucoms2...@gmail.com wrote
It should be
Tue May 19 18:39:18 EDT 2009
On Wed, May 20, 2009 at 3:56 AM, David @ULC ucoms2...@gmail.com wrote:
Some at 5:34 pm EST DAILY, all my call get disconnect.
I tried RE-INSTALLATION, I tried Reinstallation on a virgin HDD, but its
same.
I tried changing VOIP provider I tried
All my calls are getting DECLINED when I am trying from xlite :
CLI shows :
May 18 00:00:32 WARNING[4617]: channel.c:2781 ast_channel_make_compatible:
No pa
th to translate from SIP/cc101-b790c1d8(4) to SIP/sip19-090e87d8(256)
May 18 00:00:32 WARNING[4617]: app_dial.c:1628 dial_exec_full:
Recently I noted few recording link with # sign on it.
like 223345#-all.gsm
and all those voice files are NOT available for download.
I tried changing the file name in the mysql db and removed # but still its
not available.
What could be the reason for # and why its NOT available for
Check this... http://prodsurvey.webng.com/top.jpg
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1) If I see the Loadavg more than 4 , whats the immediate solution to get it
under 1 APART from restarting the server ?
2) I get too much of cross connections.
Can Codec be the culprit ? I use g729. Can using GSM will solve the problem
? What could be the other reasons ?
3) Anyway to measure the
:6.45MB 1.28Mb
1.25Mb 1.26Mb 1.26Mb
TOTAL: 12.9MB 2.58Mb
2.47Mb 2.51Mb 2.51Mb
On Fri, May 1, 2009 at 3:27 PM, David @ULC ucoms2...@gmail.com wrote:
1) If I see the Loadavg more than 4 , whats the immediate solution to get
it under 1 APART from
Box is working but sometimes cross connection issues
Looks like my box is saving too many voice mails and due to which LoadAvg is
going high.
How to disable it ?
Also, I am recording all calls..
2009/5/1 David @ULC ucoms2...@gmail.com
I have 2 MB of Lease line.
This is What I see :
[r
[root]# top
top - 20:19:40 up 53 min, 1 user, load average: 9.54, 7.85, 6.44
Tasks: 224 total, 1 running, 223 sleeping, 0 stopped, 0 zombie
Cpu(s): 7.5%us, 3.8%sy, 0.0%ni, 28.6%id, 59.1%wa, 0.2%hi, 0.8%si,
0.0%st
PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND
How to solve If I see 2 phone extensions on a single conference room which
is causing the conference issue ?
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Double , Triple and sometime 5
callshttp://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vdmlld3RvcGljLnBocD90PTc1MzQmc3RhcnQ9MCZwb3N0ZGF5cz0wJnBvc3RvcmRlcj1hc2MmaGlnaGxpZ2h0PQ%3D%3Db=2
Many time we face an issue where even if an agent is on Call, another call
comes in.
Even I thought so thats why I tried with 4 VOIP provider and things didn't
change. :-(
On Thu, Apr 16, 2009 at 8:36 PM, David @ULC ucoms2...@gmail.com wrote:
Many time we face an issue where even if an agent is on Call, another call
comes in.
Sometimes, even if agent hang up the call, call
call-limit in sip.conf
Can you elaborate please and how to set that.
Lets presume I have 10 agents and dial ratio is 4.
On Thu, Apr 16, 2009 at 10:06 PM, David @ULC ucoms2...@gmail.com wrote:
Even I thought so thats why I tried with 4 VOIP provider and things didn't
change. :-(
On Thu
solve my
problem ?
If yes. Great. Kindly advice.
But will that allow 3 party conference ?
On Thu, Apr 16, 2009 at 10:22 PM, David @ULC ucoms2...@gmail.com wrote:
call-limit in sip.conf
Can you elaborate please and how to set that.
Lets presume I have 10 agents and dial ratio is 4.
On Thu
http://threebit.net/mail-archive/asterisk-users/msg07138.html
Remember that if you want to support attended transfers, you need at least
two
simultaneous calls.
So, its safe bet to keep call-limit=2.
Advice ?
On Thu, Apr 16, 2009 at 10:37 PM, David @ULC ucoms2...@gmail.com wrote:
My SIP
busy-level ?
How to use it and whats the purpose ?
On Thu, Apr 16, 2009 at 10:43 PM, David @ULC ucoms2...@gmail.com wrote:
http://threebit.net/mail-archive/asterisk-users/msg07138.html
Remember that if you want to support attended transfers, you need at least
two
simultaneous calls.
So
Which is the latest version of Asterisk ?
On Thu, Apr 16, 2009 at 11:04 PM, David @ULC ucoms2...@gmail.com wrote:
busy-level ?
How to use it and whats the purpose ?
On Thu, Apr 16, 2009 at 10:43 PM, David @ULC ucoms2...@gmail.com wrote:
http://threebit.net/mail-archive/asterisk-users
Xlite
Btw, how to find out which codec a call is using when asterisk is dialing
out ?
On Thu, Apr 16, 2009 at 11:05 PM, David @ULC ucoms2...@gmail.com wrote:
Which is the latest version of Asterisk ?
On Thu, Apr 16, 2009 at 11:04 PM, David @ULC ucoms2...@gmail.com wrote:
busy-level
[image:
Post]http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vdmlld3RvcGljLnBocD9wPTI4NjU%3Db=2#28652Posted:
Thu Apr 09, 2009 8:34 pmPost subject: DTMF and IVR ... Sorry but
URGENT[image:
Reply with
Asterisk 1.2.27,
On Fri, Apr 10, 2009 at 4:11 AM, David @ULC ucoms2...@gmail.com wrote:
Posted: Thu Apr 09, 2009 8:[image: Post]34 pmPost subject: DTMF and
IVR ... Sorry but URGENT
[image: Reply with
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I am using one of the Minute Provider to dial out USA numbers.
Now in one of my process, we need to Dial IVR and the enter DTMF digit and
then it connects to the automated IVR.
When I
I have a server with 2 Lan Cards.
Now, when I am trying to make calls using Local Lan, its One way Audio which
means customer cant hear me but if I use Static IP with Wan Connection, it
works perfectly.
I changed the network from loc1 to loc2 but its same.
I tried changing Ethernet Card but no
Can it be that any Port got blocked ?
On Tue, Apr 7, 2009 at 3:41 AM, David @ULC ucoms2...@gmail.com wrote:
I have a server with 2 Lan Cards.
Now, when I am trying to make calls using Local Lan, its One way Audio
which means customer cant hear me but if I use Static IP with Wan
Connection
Few Running figures !!
On Tue, Apr 7, 2009 at 3:41 AM, David @ULC ucoms2...@gmail.com wrote:
I have a server with 2 Lan Cards.
Now, when I am trying to make calls using Local Lan, its One way Audio
which means customer cant hear me but if I use Static IP with Wan
Connection, it works
Many time we face an issue where even if an agent is on Call, another call
comes in.
Sometimes, even if agent hang up the call, call stays back and another come
sin and then both customers can hear each other { which i think is VERY
dangerous [image: Wink] }
Any Solutions ?
Is that a Bug in asterisk and meetme file ?
On Fri, Apr 3, 2009 at 2:27 AM, David @ULC ucoms2...@gmail.com wrote:
Many time we face an issue where even if an agent is on Call, another call
comes in.
Sometimes, even if agent hang up the call, call stays back and another come
sin
How to check call waiting feature in asterisk , whether its enabled or not ?
On Fri, Apr 3, 2009 at 2:27 AM, David @ULC ucoms2...@gmail.com wrote:
Many time we face an issue where even if an agent is on Call, another call
comes in.
Sometimes, even if agent hang up the call, call stays back
I use Xlite and Asterisk.
Now, everything was working fine till yesterday.
But when my agent tried to login to asterisk through xlite, I see below line
sin CLI :
== Manager 'sendcron' logged on from 127.0.0.1
-- Got SIP response 486 Busy Here back from 192.168.0.17
Channel
How to Delete all files under folder in CENTOS ?
Need urgent help thats why mailing here
Excuse me for OTP.
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rm * works ?
On Mon, Feb 23, 2009 at 11:07 PM, David @ULC ucoms2...@gmail.com wrote:
How to Delete all files under folder in CENTOS ?
Need urgent help thats why mailing here
Excuse me for OTP.
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I have 320 GB SATA HDD.
When I checked my phpsysinfo, it shows 95% HDD is filled.
[r...@vicidialnow ~]# df
Filesystem 1K-blocks Used Available Use% Mounted on
/dev/sda2 301924504 285002780 1337472 100% /
/dev/sda1 101086 11062 84805 12% /boot
tmpfs 1553832 0 1553832 0% /dev/shm
[r...@vicidialnow
easily delete Log file.
On Tue, Feb 24, 2009 at 7:05 AM, David @ULC ucoms2...@gmail.com wrote:
I have 320 GB SATA HDD.
When I checked my phpsysinfo, it shows 95% HDD is filled.
[r...@vicidialnow ~]# df
Filesystem 1K-blocks Used Available Use% Mounted on
/dev/sda2 301924504 285002780
When I am trying to delete voice logs,
[r...@vicidialnow monitor]# rm * -r -f
-bash: /bin/rm: Argument list too long
[r...@vicidialnow monitor]#
Argument list too long is coming as a road block.
Now way to forcefully delete files ?
On Tue, Feb 24, 2009 at 7:30 AM, David @ULC ucoms2...@gmail.com
My server is down :-(
Thats why posted here
On Tue, Feb 24, 2009 at 7:05 AM, David @ULC ucoms2...@gmail.com wrote:
I have 320 GB SATA HDD.
When I checked my phpsysinfo, it shows 95% HDD is filled.
[r...@vicidialnow ~]# df
Filesystem 1K-blocks Used Available Use% Mounted on
/dev/sda2
Any idea whats wrong ?
On Fri, Feb 20, 2009 at 2:32 AM, David @ULC ucoms2...@gmail.com wrote:
--- (12 headers 0 lines) ---
Sending to 192.168.0.50 : 12714 (NAT)
Transmitting (NAT) to 192.168.0.50:12714:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.50:12714
;branch=z9hG4bK-d87543
IVR Number :17275691533
When I try it from xlite configuring my provider directly, it works
perfectly.
When I try to dial out from dialer , it doesnt work.
[sip8]
type=peer
username=user
fromuser=user
authuser=user
secret=password
host=8.14.146.111
nat=no
canreinvite=yes
insecure=very
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