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Hi, Is there a module within the Asterisk standard distribution that provides SNMP features? Is there any third party software for that purpose? Regards, Stojan Sljivic -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050818/918b5ebf/attachment.htm

Good day all How do I get my asterisk and sip to use the password.I'm using x-lite.If I use just the username and no password it still logs on? Here is my sip.conf entry? [101] type=friend callerid="Test User" <101> context = test_1 ; Default context for incoming calls username=101 secret=123456 host=dynamic dtmfmode=inband ; Choices are inband, rfc2833, or info

...G[18359]: app_meetme.c:229 build_conf: Unable to open pseudo channel - trying device Dec 7 15:44:01 WARNING[18359]: app_meetme.c:232 build_conf: Unable to open pseudo device I have used following command to make the ztdummy: make clean make linux26 make install I use Fedora Core 3. Regards, Stojan Sljivic -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041207/b8b2576e/attachment.htm

Hi, I use Asterisk 1.0.5 and TDM04B. When an incoming call over ZAP channel hangs-up, it takes 10 seconds until Asterisk realize that. How can I shorten the time of hang-up detection? Regards, Stojan Sljivic -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050614/de3eec0f/attachment.htm

...WARNING[3253]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/pbx_realtime.so: undefined symbol: ast_load_realtime_multientry Jan 28 09:35:08 WARNING[3253]: loader.c:440 load_modules: Loading module pbx_realtime.so failed! Ouch ... error while writing audio data: : Broken pipe Thanks, Stojan Sljivic -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050128/be65bace/attachment.htm

...roblem with simultaneous calls that are sent to the same mailbox. It occurred that several calls were writing to the same file. It seems that there is a synchronization issue in the Voicemail application. Did someone else find this issue? What would be the solution/workaround for it? Regards, Stojan Sljivic -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050121/1cc6462d/attachment.htm

Hi, We have phones registered at another soft switch, and would like to use Asterisk as a Voicemail system. Is it possible and how to configure Asterisk to send NOTIFY messages (for MWI) to the endpoints that are not registered to the Asterisk? Regards, Stojan Sljivic -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051012/6d183cdb/attachment.htm

...om QAContact: swfdec at lists.freedesktop.org Hello, I'd really like swfdec to support Vimeo videos. It doesn't really have to support HD at first, but just normal would do. Currently, when I try to play a video from Vimeo, the player reports as if there is no video. :/ Thanks, Stojan -- Configure bugmail: http://bugs.freedesktop.org/userprefs.cgi?tab=email ------- You are receiving this mail because: ------- You are the QA contact for the bug. You are the assignee for the bug.

...asked for your name - When someone joins the conference a message "<name> is now joining the conference." is played. - When someone leaves the room a message "<name> has left to conference." is played. How can I set MeetMe/Asterisk to have such features? Regards, Stojan Sljivic -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041209/898a50e0/attachment.htm

...ade these phones to use SIP. How can I get the SIP firmware for my phones. I have tried at Cisco web site but I couldn't find firmware downloads. Can someone help me with that? The phones are currently using following firmware: Application Load ID: P003AM30 Boot Load ID: PC030300 Regards, Stojan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050609/06e9ed93/attachment.htm

Hello, is it possible to change how are caller id on incoming call from isdn, capi lines displayed od sip phones ? ( e.g. SNOM ) standard is 1234567@domain.net. I just want only 1234567 to be displayed. is it possible ? regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/