search for: diaplay

Displaying 6 results from an estimated 6 matches for "diaplay".

Did you mean: diaplan
2007 Mar 14
4
what happened to asterisk wiki???
Hi im trying access the www.voip-info.org website since yesterday but i cant open it. google search diaplay correct search results but it doesnt open when i click the link. it displays a message about tcp error which says -->"There was a problem communicating with the server". I dont know what the problem is. I just want to ask whether their server is down or not and is everybody having the...
2006 Aug 15
4
escaping html?
Hi I have a wysiwyg html ditor in my app. How do I escape html written to the database and encoding when I display the content> Ty Pieter -------------- next part -------------- An HTML attachment was scrubbed... URL: http://wrath.rubyonrails.org/pipermail/rails/attachments/20060815/d8c50941/attachment.html
2004 Dec 22
2
Matching Caller ID against a database of knowncallers
...004 06:13 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Matching Caller ID against a database of > knowncallers > > > Hi All, > > Is it possible to match caller ID on incoming calls against say text > file of know numbers and diaplay the name rather than the numerical > caller ID? > > I know some handsets such as the SNOM 190 can do this from within the > handset, but I would like it done and updated centrally at the > Asterisk box..... > _______________________________________________ > Asterisk-Users mai...
2003 Aug 22
1
Best SIP phone?
Hello, I would like to request opinions for: What is the best SIP phone? Please give a rating from 1 to 10 (10 being the best) for the following categories: Reliability Ease of deployment with Asterisk Sound quality Please be sure to supply the manufacturer's name and the phone's model number. If you have an opinion on this topic, I would like to hear it. -- Thanks, Tim
2004 Aug 05
0
Strange message, and one-way audio between sip and H.323
...g to use asterisk for converting SIP to H.323 calls. asterisk (0.9.1) runs on the same linux (Redhat 8) box of our gatekeeper (gnugk version 2.0.8). the calls are going out through a cisco gateway. when I make a call from a SIP phone to a PSTN number reachable through the cisco gateway: asterisk diaplays Aug 5 23:24:26 WARNING[1255648560]: chan_oh323.c:2898 alerted_h323_connection: Call ip$localhost/22666 in unexpected state (PLAYONLY). I hear (on the SIP phone) clearly what the other person is saying, but the other person (on the PSTN side) hears nothing from me. gatekeeper and the cisco gat...
2015 Feb 20
1
Excel and Samba Problem
Did you try fort he share: oplocks = no level2 oplocks = no EDV Daniel M?ller Leitung EDV Tropenklinik Paul-Lechler-Krankenhaus Paul-Lechler-Str. 24 72076 T?bingen Tel.: 07071/206-463, Fax: 07071/206-499 eMail: mueller at tropenklinik.de Internet: www.tropenklinik.de -----Urspr?ngliche Nachricht----- Von: Oliver Werner [mailto:oliver.werner at kontrast.de] Gesendet: Freitag, 20. Februar