similar to: kill sip user

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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091019/f6aa2510/attachment.htm
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. -------------- next
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. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090910/af1f9656/attachment.htm
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 ? -------------- next part -------------- An HTML attachment was scrubbed... URL:
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