search for: 2003,1

...n 1.7), the code below used to work for converting and extracting based on the Date Time. In version 1.8.1, something changed I know, but I cannot for the life of me figure out what... Data: UserName,RequestDate,PO,OrderDate,ExpDelivDate,Vendor,Total "Woody, Jim",12/19/2002,AP15063,1/7/2003,2/10/2003,Ames ,8570 "Harrold, Paul",12/31/2002,AP15083,1/9/2003,1/10/2003,Ryan ,1039.5 "Vo, Hoang",12/27/2002,AP15055,1/6/2003,1/13/2003,TIDEA,1005.36 "Way, Shawn",1/2/2003,AP15043,1/2/2003,1/9/2003,JS ,1000 "Vo, Hoang",1/7/2003,SO17440,1/8/2003,12/31/2003...

...edHat Enterprise Linux 3AS with kernel 2.4.21-4.EL. The Diva server is running in version 7.5 Can someone help us in reolving the errors with incoming calls? When we try to call to an extension we get these messages in the CLI: -- creating pipe for PLCI=0x401 msn = * == Starting CAPI[contr1/2003]/3 at ,2003,1 failed so falling back to exten 's' == Starting CAPI[contr1/2003]/3 at ,s,1 still failed so falling back to context 'default' Oct 7 16:20:51 WARNING[-1253872720]: pbx.c:1877 ast_pbx_run: Channel 'CAPI[contr1/2003]/3' sent into invalid extension 's' i...

...n not make call outside. heres my dial plan. ? sip.conf ? [general] port = 5060 bindaddr = 0.0.0.0 context = others [2000] type=friend context=outside secret=1234 host=dynamic [2001] type=friend context=outside secret=1234 host=dynamic [2002] type=friend context=outside secret=1234 host=dynamic [2003] type=friend contex=outside secret=1234 host=dynamic ? [2004] type=friend contex=outside secret=1234 host=dynamic [2005] type=friend contex=outside secret=1234 host=dynamic ? [2006] type=friend contex=internal secret=1234 host=dynamic [2007] type=friend contex=internal secret=1234 host=dynamic [2...

OK. What if I have a time series which is collected every Monday, please? What is the proper way to use the start option within the ts command in order to indicate that this is Monday data, please? Thanks again! Sincerely, Erin

...ite=yes allowguest=no allow=ulaw allow=alaw allow=g711 allow=g729 allow=gsm allow=ilbc [2000] type=friend context=my-phones secret= allow=ulaw host=dynamic [2001] type=friend context=my-phones secret= allow=ulaw host=dynamic [2002] type=friend context=my-phones secret= allow=ulaw host=dynamic [2003] type=friend context=my-phones secret= allow=ulaw host=dynamic [xxx.xxx.xxx.yyy] context=pstn-incoming type=friend host=xxx.xxx.xxx.yyy ; IP address of Cisco gateway dtmfmode=rfc2833 disallow=all allow=ulaw insecure=very [1001] context=local-phones type=friend username=1001 secret=secret host=dyn...

Problem I have is this. outside firewall (extension 2003) can call me inside firewall (extension 2000) and all is fine. If I call from inside firewall (extension 2000) to outside firewall (extension 2003) I hear no ringing and person at other end can pick up and I hear for maybe a half second then I go to voicemail. If I add another extension on the ou...

Hello list, I wonder if someone might be able to clear up something for me. I recently set up asterisk and have now managed to get the MeetMe application up and running. When I dial the extension to access the conference/MeetMe application, the only prompt I hear is:"You are currently the only person in this conference." When I use a friend's newly installed asterisk, I hear:

Hi: I am trying to setup a simple auto attendant with Asterisk using SIP extensions. I have an IP trunk to voicepulse and my outgoing calls go over that. I can also receive calls on that voicepulse trunk and want it to an auto attendant. Everything works except on the following: - one of the options is to allow the caller to press the extension that they would like to be connected to. I have

...x included sip [sip] exten => 55,1,VoicemailMain exten => 2001,1,Dial(SIP/2001,20,tr) exten => 2001,2,VoiceMail,u2001 exten => 2001,102,VoiceMail,b2001 exten => 2002,1,Dial(SIP/2002,20,tr) exten => 2002,2,VoiceMail,u2002 exten => 2002,102,VoiceMail,b2002 exten => 2003,1,Dial(SIP/2003,20,tr) exten => 2003,2,VoiceMail,u2003 exten => 2003,102,VoiceMail,b2003 exten => 2004,1,Dial(SIP/2004,20,tr) exten => 2004,2,VoiceMail,u2004 exten => 2004,102,VoiceMail,b2004 exten => 2010,1,Dial(SIP/2010,20,tr) exten => 2010,2,VoiceMail,u2010 exten =...

...2001? exten => 2001,1,Dial(SIP/2001,20) exten => 2001,2,Voicemail(u2001) exten => 2001,102,Voicemail(b2001) exten => 2001,103,Hangup exten => 2002,1,Dial(SIP/2002,20) exten => 2002,2,Voicemail(u2002) exten => 2002,102,Voicemail(b2002) exten => 2002,103,Hangup exten => 2003,1,Dial(SIP/2003,20) exten => 2003,2,Voicemail(u2003) exten => 2003,102,Voicemail(b2003) exten => 2003,103,Hangup exten => 2004,1,Dial(SIP/2004,20) exten => 2004,2,Voicemail(u2004) exten => 2004,102,Voicemail(b2004) exten => 2004,103,Hangup ; ; Define a way so that users can d...

Dear all, I have a question regarding the configuration of *. I have 3 port FXS, and 2 port FXO. I have 4 users that use analog phone connected to FXS (I have 3 phones). I need to limit the user's capability (user A can call International, user B can call long distance, etc). I want to implement the password say to call , he/she needs to puch 9(for outgoing call)2-4 digits password,then

hi do u have the sip phones extensions in the extension.conf and are they in the right context (sip-incoming)??? are the sip phone registering to asterisk?? try stop asterisk and reconnect as asterisk -vvvvvvvc to check see them registering... ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Sistemista WebSolvingJaa Sent: Fri 7/1/2005 6:43 PM

...th data type=friend username=2001 secret=homemonkey host=dynamic ;context=from-sip mailbox=2001 dtmfmode=rfc2833 [2002] ; Duplicate of 2000, except with different auth data type=friend username=2002 secret=homemonkey host=dynamic ;context=from-sip mailbox=2002 dtmfmode=rfc2833 [2003] ; Duplicate of 2000, except with different auth data type=friend username=2003 secret=homemonkey host=dynamic ;context=from-sip mailbox=2003 dtmfmode=rfc2833 [2004] ; Duplicate of 2000, except with different auth data type=friend username=2004 secret=homemonkey ho...

Guys I am having some serious issues with my call queue and Management is breathing down my neck pretty bad, and I am running out of ideas. I have a single queue for my tech support department. I originally was using the AgentCallbackLogin for them and it tested out great on our testing weekends, but it hasn't worked out since. It would only let one of them take calls at a time, no matter

Hi, I've used various patches with asterisk 1.4 to have support for call pickup and notification with good results. Now I'm trying vanilla 1.6.2 with its official support for "dialog-info +xml" notifications with no success. This is what i'm doing: - Phone A has a key configured as type "extension" pointing to Phone B. - In sip.conf I added

Hello All, I have some Polycom IP 500 phones that I would like to have configured for direct dialing to our voice mail system. So far I have been unable to get the hard button labeled Voice Mail to connect to Asterisk without first passing through the message center prompts. I have followed all the Admin Guide instructions regarding the phones .cfg files and using

...=2001 secret=2001 host=dynamic context=from-sip mailbox=101 ;dtmfmod=inband nat=yes reinvite=no canreinvite=no callerid=Mr. Mandar <2001> [2002] type=friend host=dynamic callerid=William Suffill <2002> username=2002 secret=2002 context=from-sip nat=yes mailbox=2002 [2003] ; Duplicate of 2000, except with different auth data type=friend username=2003 secret=2003 host=dynamic context=from-sip mailbox=103 ;dtmfmod=inband reinvite=no canreinvite=no callerid=Mr.Seshu <2003> [2004] ; Duplicate of 2000, except with different auth data type=friend use...

Hi All I am really very interested in starting to use R within our company. I particularly like the open source nature of the product. My company is a medical research company which is part of the University of London. We conduct contract virology research for large pharma companies. My question is how do we validate this software? I wonder if anyone else has had the problem and might be able to