similar to: i have a error in ivr

Displaying 20 results from an estimated 400 matches similar to: "i have a error in ivr"

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 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:
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 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 Sep 29
2
kill sip user
I have a user but I need to give that user only kill and disable all connection cut calls what is the command in the CLIC -- Bayardo S?nchez Garc?a Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxy Support - Linux Server E-mail: bayardo.sanchez at gmail.com Linux User: #418392 America Central - Managua, NI (505) 2249-2853 - 84886876 IM msn messenger:
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 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
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.
2011 Mar 17
1
Upgrading system on file server
Hi, I have an old version of Suse runing a Samba. I will upgrade this box from Suse 9.3 -> Suse 11.3. I know that there are a lot of risk but my top fear is about Samba. This is a production server and network users authenticate with this server. We do not have roaming profile, but I know that if I install from scratch I lose my domain, SID number changes and I have to reconfigure all the
2005 Oct 11
1
migrated to new ver on voip connection vs1 server voicemail problems
I migrated to a new version of the voip connection vs1 server software and I am now getting these errors when I try to call my voicemail. Any thoughts? The files are there, so I don't get it. Oct 11 19:57:26 WARNING[6587]: format_wav.c:140 check_header: Not a wav file 49 Oct 11 19:57:26 WARNING[6587]: file.c:418 ast_filehelper: Unable to open fd on
2006 May 16
0
Re: [Astlinux-users] British English Female files ready for download
Mark, While these samples are pretty good they do not work "out of the box" - there are a couple of issues: 1. the samples are 44100 samples/second and Asterisk needs them to be at 8000 samples/second. This is what happens if you prune out all of the Amercian voicemail prompts and substitute yours: Asterisk 1.2.7, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark
2005 Oct 11
5
help with broken voicemail
I can not figure out what the heck is going on. I went back to my old version and I still get errors when the voicemail system tries to load any of the greetings, unavail messages, etc. the normal voicemail prompts work, but any user recording don't work. Leaving a new message appears to work, but the system wont replay them, it throws errors. Here is an example of the errors: Oct 11
2006 Oct 16
0
Weird problem with beep.wav!
This is really doing my head in! For some reason, my asterisk box can't playback beep.wav. I have this extension defined in my internal context: '10001' => 1. Answer() [pbx_config] 2. Wait(2) [pbx_config] 3. Record(/tmp/asterisk/10001:gsm) [pbx_config]
2009 Jan 31
3
Is http://downloads.digium.com/pub/ down???
Anyone else having problems connecting to http://downloads.digium.com/pub/ ?? Jonn
2011 Mar 14
1
Setting up samba server
hi! I'm currently working with my new samba server [samba2.domain.com] which I will use for file sharing and I have an existing samba/ldap [smb/ldap.domain.com] as domain controller which I will use for authentication. below is my smb2.domain.com smb.conf [global] workgroup = domain.com server string = Linux File Server security = SERVER password
2008 Nov 11
7
music on hold
hii guys: i get the message from the asterisk: Started music on hold, class 'default', on Local/s at skype-web-callback-dial-263to263-1775,1 [2008-11-11 14:32:41] WARNING[1781]: format_wav.c:156 check_header: Unexpected freqency 11025 [2008-11-11 14:32:41] WARNING[1781]: file.c:322 fn_wrapper: Unable to open format wav [2008-11-11 14:32:41] WARNING[1781]:
2010 Oct 12
1
sound file debug
Hi gang, I have a "fun" one for you. I'm not getting the quality of sound I want out of GSM, so I'm trying to make my files into .WAV (.wav) format. Here is the "file" output for 5 files: file *.WAV cents.WAV: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz dollars.WAV: RIFF (little-endian) data, WAVE audio, Microsoft
2004 Sep 03
7
Dropping incompatible voice frame
Hi: i have a problem. Mi extensions.conf: exten => _N.,1,Setvar(VOICEMAILREQ=${EXTEN}) exten => _N.,2,SetAccount(${customer}) exten => _N.,3,SetCDRUserField(${VOICEMAILREQ:1}) exten => _N.,4,ResponseTimeout(5) exten => _N.,5,Background(ifyou) exten => _N.,6,Background(silence/1) exten => _N.,7,Background(ifyou) exten => _N.,8,Background(silence/5) exten
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
2005 May 19
2
Voicemail wav49 format problem
I have the voicemail format set to wav49 in my voicemail.conf file. When retrieving voicemails, the first message plays back ok - but then Asterisk hangs up and the log shows the following error. Any idea what's up? May 19 12:57:24 VERBOSE[7860]: Asterisk Ready. May 19 13:48:51 WARNING[7860]: Not a wav file 49 May 19 13:48:51 WARNING[7860]: Unable to open fd on