search for: indignant

...h phone in it's own context, that would be fine. -- Automated Signature: This message is from Flobi of Flobi.com<http://Flobi.com> . Visit my website if you like: http://www.flobi.com/ Please remember to tip your waitress and bartender. They are doing their best to serve you and your indignant, malcontent attitude. -- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050907/8da7a475/attachment.htm

Ok, so I've made progress on 2BCT (2 B-Channel Transfer). I'm assuming that the debug info below shows that XO doesn't have 2BCT enabled on my line, but if anybody can confirm that'll let me be way more indignant. J -- Native bridging DAHDI/1-1 and DAHDI/3-1 > Protocol Discriminator: Q.931 (8) len=28 > Call Ref: len= 2 (reference 801/0x321) (Terminator) > Message type: FACILITY (98) > [1c 15 91 a1 12 02 01 23 06 07 2a 86 48 ce 15 00 08 30 04 02 02 01 93] > Facility (len=23, code...

Address lookup canonical name asterisk.org. aliases addresses 216.27.40.102 Service scan FTP - 21 Error: TimedOut SMTP - 25 Error: ConnectionRefused HTTP - 80 Error: ConnectionRefused POP3 - 110 Error: TimedOut NNTP - 119 Error: TimedOut digium.com is ok though Address lookup canonical name digium.com. aliases addresses 216.207.245.1 Service scan FTP - 21 Error: TimedOut SMTP - 25

Dear all, Has anyone seen this before and can suggest a solution? I've got 3 sip clients behind the router, and they all register with Asterisk using the same IP address. Now, wenn all are registered, all the calls get routed to the client that registered most recently, but not to the correct client. Also, there seems to be some problems registering all 3 clients simultaneously. Could

Hello, I'm still looking for any ideas on this problem: I've got 3 sip clients behind the router, and they all register with Asterisk using the same IP address. Now, wenn all are registered, all the calls get routed to the client that registered most recently, but not to the correct client. Also, there seems to be some problems registering all 3 clients simultaneously. I have

I apologize for the double post. I am curious as to what the usefullness is of the multiple line appearance feature on Polycom phones. I setup our phones to register one line per extension but I hear the IP501's can do three line appearances. Why and how could this feature be applied? Thanks again all. Kenny ______________________________________________________ Click here to

How do I set each extension to play it's own voicemail prompts? I have vm working in that it plays the standard "person at extension 1234 is not available....." and takes the message. I've recorded seperate .gsm files for each user but can not figure out how to use them. - Gary Edison Information Technologies www.EdisonInfo.com P.O. Box 554

Greetings, I am trying to implement distinctive ringing on a Cisco 7960. I have tried setting alert_info to chirp1 or chirp2 before dialing the phone, but it has no affect. If you have successfully implemented distinctive ringing on a 7960, I would really appreciate seeing the snipit of code that works. Thanks in advance Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775)

http://dovecot.org/ It's hot in here. Something like 25-30c for many days now. My brains don't work in that heat. So, this is still not an 1.0-alpha1. I'll put it out when I think the release will actually be fully working :) Changes since test76: - Fixes to FETCHing MIME body parts, especially with message/rfc822 attachments. We hadn't been handling this entirely correctly

I know why SQLite3 is broken. This code: Topic.transaction do |transaction| transaction.rollback! end results in the following database commands: BEGIN; ROLLBACK; COMMIT; In other words, there is a COMMIT when no transaction is in flight. MySQL and Postgres are coll with it, but SQLite3 blows up with indignation. Recorded as a ticket: http://dev.rubyonrails.org/ticket/8030

Hi all! I would like to solve some problems: I have a sip provider that lets me make pstn calls after listening some stuff and entering a pin number: 1) How can I make Asterisk enter the pin number? Then wait 1 second and enter the phone number? I have in extensions.conf: exten => 6*,1,Dial,SIP/2002@myprovider,60,tr I have tried with w (like with ZAP channels) but it does not work, nor

Hi, I'm working with this issue for a while, Now I already solve the dialplan issues, but I still have a question about the Callgroups, I read at www.voip-info.org that , there is a 63 limit of callgroups. And I'm wondering why?? and if the 1.2.0beta version supported more than 63 Groups?? (I did'nt find any Changelog for 1.2) or If not There is any unoficial patch for that ? Thanks

Are there any standards for setting up pbx dialplans for businesses/offices? What I mean is that, which numbers are reserved for a specific use ex. 0 for operator ? Putting Zero for operator in the dialplan seems to be the common practice of businesses. If there is such a standard, * and # are used for what ? -------------- next part -------------- An HTML attachment was scrubbed... URL:

...---------------------------------------- "The simple rights, the civil liberties from generations of struggle must not be just fine words for patriotic holidays, words we subvert on weekdays, but living, honored rules of conduct amongst us...I'm glad the American Civil Liberties Union gets indignant, and I hope this will always be so." -- Senator Adlai E. Stevenson -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: panicoops-09-01-2006.txt URL: <http://lists.centos.org/pipermail/centos/attachments/20060109/a61488c9/attachment-0005.txt...

Hello, CONFIGURATION Syslinux "old": 3.36 Syslinux "new": 5.01 (B)LFS i686-pc-linux-gnu 3.8.0 Machine #1: ASUS P5E-VM HDMI. Latest BIOS. circa 3 years ago. Machine #2: ASUS P8H77-I. Latest BIOS. This year. USB Flash stick: Kingston "Data Traveler" 4GB. Clean (blank) STRANGE (UNEXPLAINED) syslinux BEHAVIOR DIFFERENCE Steps: 1. Set up KNOPPIX 7.05

Hi, i'm testing Dovecot LDA to works with qmail and vpopmail. I have added into a .qmail for a users this line: root at test.com: /home/vpopmail/domains/test.com/0/test# cat .qmail | /var/qmail/bin/preline -f /usr/libexec/dovecot/deliver -d $EXT@$USER This is my dovecont conf: # dovecot -n # 1.1.7: /etc/dovecot.conf # OS: Linux 2.6.24-21-server x86_64 Ubuntu 8.04.1 log_path:

Hi, this is both a statistical and a R question... what would the best way / test to detect an outlier value among a series of 10 to 30 values ? for instance if we have the following dataset: 10,11,12,15,20,22,25,30,500 I d like to have a way to identify the last data as an outlier (only one direction). One way would be to calculate abs(mean - median) and if elevated (to what extent ?) delete the

Following on from my below email, things have taken another bizarre twist.. I have continued getting the error when 2092 tries to listen to messages by dialing 9999. --Executing VoiceMailMain ("SIP/2092-6918", "2092") in new stack --Playing 'vm-password' (language 'en') WARNING: app_voicemail.c:4922 vm_authentication: Unable to read password. Then I

https://bugzilla.mindrot.org/show_bug.cgi?id=1818 Summary: SSH2_MSG_CHANNEL_FAILURE on closed channel Product: Portable OpenSSH Version: 5.1p1 Platform: All OS/Version: All Status: NEW Severity: normal Priority: P2 Component: sshd AssignedTo: unassigned-bugs at mindrot.org ReportedBy:

I noticed that some of my event handlers and scripts evaluated in ajax responses didn''t work in Konqueror (3.5.5). document.getElementsByClassName is not defined in these contexts. I can work around this, by saving document.getElementsByClassName in a (namespaced) global variable once and reassigning it before each use. Still, this misfeature is annoying and not restricted to this