[Asterisk-Users] MWI for endpoints not registered at Asterisk
Title: Message 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ouch ... error while writing audio data: : Broken pipe
Title: Message Hi, Can anyone help me with this: I have downloaded latest stable version of Asterisk using the asterisk-update.sh script. Compilation and installation passed well. When I start Asterisk I get the following error: [pbx_realtime.so]Jan 28 09:35:08 WARNING[3253]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/pbx_realtime.so: undefined symbol: ast_load_realtime_multientryJan 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 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Ouch ... error while writing audio data: : Brokenpipe
Title: Message Hi all, Thanks for the information. Yes, I have been downgrading from HEAD to 1.0.5. I have removed the /usr/lib/asterisk/modules and I do not get previous error, but apparently a new one appeared: [cdr_tds.so]Jan 28 15:16:28 WARNING[25289]: loader.c:258 ast_load_resource: libtds.so.3: cannot open shared object file: No such file or directoryJan 28 15:16:28 WARNING[25289]: loader.c:440 load_modules: Loading module cdr_tds.so failed!Ouch ... error while writing audio data: : Broken pipe Do you know what is this related to? Regards,Stojan Sljivic ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail folders
Title: Message Hi, How can I rename existing voicemail folders (INBOX - Inbox; Old - Archive)? Regards, Stojan Sljivic ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail Synchronization
Title: Message Hi, Ihave stress tested the Asterisk Voicemail. We have encountered problem 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 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Stress Test
Title: Message Hi, Is there a free toll for SIP stress testing that supports RTP? Can SIPp be used for such purposes (to send audio)? Regards, Stojan Sljivic ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Stress Test
Hi, Is there any other free tool for SIP testing that has facility to originate audio/media? Regards, Stojan Sljivic -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vamsi Pottangi Sent: Thursday, January 20, 2005 15:43 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Stress Test SIPp has no facility to originate audio/media, it can just send back the media it receives on its RTP port, more like an RTP proxy. ~Vamsi On Thu, 20 Jan 2005 14:55:20 +0100, Stojan Sljivic - Pamet [EMAIL PROTECTED] wrote: Hi, Is there a free toll for SIP stress testing that supports RTP? Can SIPp be used for such purposes (to send audio)? Regards, Stojan Sljivic ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Traditional Telephony Interface Card
Title: Message Hi all, We are located in Europe and we have four analog telephony lines. What hardware is needed to connect Asterisk with these lines? WhatVoIP hard phones operate best with Asterisk? Regards, Stojan Sljivic ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MeetMe Features
Title: Message Hi all, I had a chance to use some call conferencesthat had some very neat functionalities: - When you call you are first 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] modprobe ztdummy - failed
Title: Message Hi all, I have a problem starting the ztdummy. Here is what happens: [EMAIL PROTECTED] /]# modprobe ztdummyNotice: Configuration file is /etc/zaptel.confline 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected FATAL: Error running install command for ztdummy After this, ztdummy is visible with lsmod, but when I try MeetMe, I get following: == Parsing '/etc/asterisk/meetme.conf': FoundDec 7 15:44:01 WARNING[18359]: chan_zap.c:775 zt_open: Unable to open '/dev/zap/pseudo': No such file or directoryDec 7 15:44:01 ERROR[18359]: chan_zap.c:6811 chandup: Unable to dup channel: No such file or directoryDec 7 15:44:01 WARNING[18359]: app_meetme.c:229 build_conf: Unable to open pseudo channel - trying deviceDec 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
