[Asterisk-Users] MWI for endpoints not registered at Asterisk

2005-10-12 Thread Stojan Sljivic - Pamet
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

[Asterisk-Users] Ouch ... error while writing audio data: : Broken pipe

2005-01-28 Thread Stojan Sljivic - Pamet
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]:

RE: [Asterisk-Users] Ouch ... error while writing audio data: : Brokenpipe

2005-01-28 Thread Stojan Sljivic - Pamet
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

[Asterisk-Users] Voicemail folders

2005-01-24 Thread Stojan Sljivic - Pamet
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

[Asterisk-Users] Voicemail Synchronization

2005-01-21 Thread Stojan Sljivic - Pamet
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.

[Asterisk-Users] SIP Stress Test

2005-01-20 Thread Stojan Sljivic - Pamet
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

RE: [Asterisk-Users] SIP Stress Test

2005-01-20 Thread Stojan Sljivic - Pamet
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

[Asterisk-Users] Traditional Telephony Interface Card

2004-12-13 Thread Stojan Sljivic - Pamet
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

[Asterisk-Users] MeetMe Features

2004-12-09 Thread Stojan Sljivic - Pamet
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

[Asterisk-Users] modprobe ztdummy - failed

2004-12-07 Thread Stojan Sljivic - Pamet
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