search for: malcont

Displaying 12 results from an estimated 12 matches for "malcont".

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2005 Sep 07
3
Extensions - Realtime
...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
2004 Aug 07
0
improve energy levels
...h Jacobite conspiracy between the hour for bed and that for breakfast. About the same time, he began to read in his dreams - tales, for the most part, and for the most part after the manner of G. P. R. James, but so incredibly more vivid and moving than any printed book, that he has ever since been malcontent with literature. desnudo 19compaternidad 05 ejemplo ermaduracronome`trica
2005 Sep 07
1
asterisk.org blocked - rejecting connections
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
2005 Sep 07
1
Several SIP clients behind router register with the same IP, messing up call routing, any ideas?
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
2005 Sep 08
1
Additional: Several SIP clients behind router registerwiththe same IP, messing up call routing, any ideas?
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
2005 Sep 08
1
Multiple Line Appearances / Why use this?
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
2005 Sep 08
2
play each person's voicemail
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
2005 Sep 08
2
Distinctive ringing on Cisco 79xx
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)
2005 Sep 06
1
Some problems (SendDTMF, Wait, Parked Calls)
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
2005 Sep 07
3
Hosted PBX (vPBX) and Call/PickUP Groups
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
2005 Sep 08
1
(OT) Dialplan Standards for Business/Offices
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:
2005 Sep 07
1
Eeven Stranger - Asterisk BT100 Password Issue
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