Displaying 20 results from an estimated 900 matches similar to: "kill sip user"
2009 Aug 14
1
i have a error in ivr
i call to my tollfree number buy my CLI send the next error:
Aug 14 08:15:22 WARNING[25931]: format_wav.c:169 check_header: Unexpected
freqency 22050
Aug 14 08:15:22 WARNING[25931]: file.c:441 ast_filehelper: Unable to open
file on /var/lib/asterisk/sounds/procall3.wav
Aug 14 08:15:22 WARNING[25931]: file.c:828 ast_streamfile: Unable to open
procall3 (format ulaw): No such file or directory
Aug
2010 Nov 24
2
Avoided deadlock Error
My Asterisk is Asterisk 1.2.30.2 currently running on viciserver the problem
is :
Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided
deadlock for '0x861f6d8', 9 retries!
Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided
deadlock for '0x85a6420', 9 retries!
Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided
2009 Mar 16
3
Help Inbound number
i create inbound number but i calling and send this error:
[Mar 16 11:41:12] NOTICE[30847]: chan_sip.c:14383 handle_request_invite:
Call from '101396_procall' to extension '8888246463' rejected because
extension not found.
but the extensin existed
--
Bayardo S?nchez Garc?a
Web Developer - Internet Portals - Asterisk Support - Windows Server Support
- Proxi Support
E-mail:
2009 Jan 21
1
recording failed
I have a problem when I call a good record but I make a call to return to
the same number I erased the previous record, and I replaced with the last
call
--
Bayardo S?nchez Garc?a
Web Developer - Internet Portals
Linux User: #418392
Ubuntu User #14171
America Central - Managua, NI (505) 249-2853 - 4886876
IM msn messenger: bjsanchezg at hotmail.com
Skype: bayardo.sanchez
This email is intended
2009 Apr 15
2
inbound filed
i create inbound confi my confi is:
[incoming]
exten=> 18888246463,,1,Dial(SIP/8003,60,rT)
exten=> 6463,1,Dial(SIP/8003,60,rT)
exten=> 18888246463,,n,Wait(5)
exten=> 18888246463,,n,Hangup
but y calling and send this error in my CLI:
[Apr 15 09:58:48] NOTICE[26985]: chan_sip.c:14383 handle_request_invite:
Call from '101396_procall' to extension '8888246463' rejected
2009 Jan 18
2
Recordin call in asterisk
I need help need recording all call for my pbx but i am a novato in asterisk
my confi for record is:
exten=>_NXXXXXXXXX,n,Set(CALLFILENAME=CLIENTE-${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${UNIQUEID})
exten => _NXXXXXXXXX,n,MixMonitor(${CALLFILENAME}.gsm,m)
exten => _NXXXXXXXXX,n,Dial(${TRUNK_CLIENTE}/${EXTEN})
--
Bayardo S?nchez Garc?a
Web Developer - Internet
2009 Jan 26
3
I need help
i have a problem need help
== Spawn extension (DLPN_everything, 2095773777, 2) exited non-zero on
'SIP/8022-b7225740'
-- Got SIP response 503 "Service Unavailable" back from 74.63.41.218
-- SIP/voipms4-09ab0c38 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'SIP/8011-b724f888' status is 'CONGESTION'
2009 Jan 31
3
Is http://downloads.digium.com/pub/ down???
Anyone else having problems connecting to
http://downloads.digium.com/pub/ ??
Jonn
2009 Oct 18
4
Customising Firmware
Hi,
Does anyone have any advice on customising firmware of an SPA921 so that
it can be locked to a sip provider and display logos on the config
pages.
Many thanks
Dan Journo
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2009 Oct 05
5
Networking Concept
Hello,
I would like to know how Asterisk deal in this case:
Assume I have a Main Asterisk Server located in UK, and another box that
have PSTN interfaces located in China, now the purpose is to FW calls
through PSTN.
Assuming I have a client who is calling from Japan to my main switch in UK
and he is calling China, (japan have latency around 500ms to UK and 100ms to
China), how asterisk
2009 Dec 09
1
Problem with Asterisk and SPA-3000
Hello everybody,
I have a very strange issue with a Linksys SPA-3000 (1 fxo + 1 fxs) used
as PSTN gateway to asterisk in a small office. Everything works just
fine, except that sometimes, and it seems that only for long incoming
calls, the IVR menu appears on the middle of the call(like a three way
call, call goes on with prompts playing over the parties). Dialing an
extension at the prompt at
2009 Jan 25
5
soft phone
hi
wich soft phone do you recomend but i need this feature it must ask for user
name and password when it start.
i know xline and zoipper but they dont have that i can acomplish this whit
twinkle but i need it for Windows :-(
any ideas?
thanks
--
(\__/)
(='.'=)This is Bunny. Copy and paste bunny into your
(")_(")signature to help him gain world domination.
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2009 Oct 08
4
No sound on voicemail from analog line
Hello.
I have a server installed with asterisk 1.6. I have a PSTN line that comes in to one of those clone cards. Everything seem to be working fine. The only problem I have is that I can't get voicemails coming from the PSTN line. All other: SIP, IAX work fine. I can hear those ok but, when it comes to a call that comes in from PSTN I get no sound.
What can cause that problem?
Thanks in
2009 Sep 10
2
ASR & ACD
Is there any program Asterisk users use to calculate ASR and ACD ??
Thanks for any comments.
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2009 Oct 11
5
Call Recording and Posting
Hello,
I'm working on a call recording solution. I would like recordings to
either be automatically uploaded via FTP, or posted to a URL for
processing by our main server.
Is Asterisk capable of doing this or will I have to create a separate
application that monitors a temp directory for new recordings?
I ask because I don't have any experience in Linux programming, so I
2011 Mar 12
2
Restrict file types to be saved in a samba server
Hi,
I have a Samba server, it's main goal is to store documents of all users
of the network. Certain users abuses and save mp3, mov, jpg, gif and
other files that must be saved in other file server, so I need to
restrict the those type files and allow my users save only office files
like .doc, .docx, .xls, .ppt, .pdf
thanks for your help.
Bayardo.
2009 Sep 09
2
Call getting stucked !!
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 ?
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2009 Sep 15
3
Which is best provider for G.729
hello
I dont want to disgrace any company but i want to know from
your(user)experience which one is good in case of g.729 (performace etc)
is it Howler(http://www.howlertech.com/products/howlets)
OR its Digium (
http://store.digium.com/productview.php?category_id=5&product_code=8G729CODEC&main_category_id=5
)
plz note i dont want to degrade any company... But to know what experience
you
2009 Sep 30
1
How to finish a Meetme
Hi people, I want to make a meetme between 2 numbers.
First I enter the number1 into the meetme. It is waiting for the other number, but the other number never entered, so, how can I finish the meetme from the dialplan?. Is it posible by using MeetmeAdmin and kick all the users?
Thanks,
Anahi Ludue?a
_________________________________________________________________
Descubre
2009 Oct 10
1
Asterisk to Asterisk access voicemail - not working
Asterisk to Asterisk voicemail not working (accessing voicemail from another asterisk).
PSTN to Asterisk is working, but not between two asterisk :-(
I've tried setting my asterisk dtmf to rfc2833, inband it is not working.
The other Asterisk Linksys is set dtmf = auto
--
Joseph