similar to: modprobe ztdummy - failed

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:

Hi all, I had a chance to use some call conferences that 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

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_multientry Jan 28

Hi, I have 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

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

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

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

Upon setting up and configuring the my extension.conf, meetme.conf and following the instruction outlined at this web page: http://www.voip-info.org/wiki-Asterisk+timer+ztdummy I get the following errors when calling the meetme number. Executing Wait("SIP/216.53.118.2-f41196e0", "1") in new stack -- Executing MeetMe("SIP/216.53.118.2-f41196e0",

I would us Meetme for conferance SIP-->SIP fist. my Meetme.conf: [rooms] conf => 9999 my extensions.conf: exten => 9999,1,MeetMe(9999) But : == Parsing '/etc/asterisk/meetme.conf': Found Jun 16 10:33:22 WARNING[12100]: chan_zap.c:916 zt_open: Unable to open '/dev/zap/pseudo': No such file or directory Jun 16 10:33:22 ERROR[12100]: chan_zap.c:6969 chandup: Unable

I am trying to set up a simple conference call capability with asterisk My meetme.conf [general] [rooms] conf => 61 conf => 62 conf => 63 conf => 63 My extensions.conf exten => 60,1,Answer() exten => 60,2,MeetMe(,EMxp) When I enter extension 60 I enter a conference - I get repeated "you are entering conference 6 1 that is not a valid conference number you are

Hello Users, Is Without Zaptel interface Installed, conference Bridge is worked or not. Why it need, For SIP conferences through OpenSER.... Please Help me For me its Giving Some Errors and warnings. *== Parsing '/etc/asterisk/meetme.conf': Found Oct 25 18:16:13 WARNING[12281]: chan_zap.c:913 zt_open: Unable to open '/dev/zap/pseudo': No such file or directory Oct 25

Hello, I'm trying to set up a conference room. When I dial it's extension, I get an audible error saying "Not a valid conference room, please try again" followed by a disconnect. I've got debug sip peer 1001 (my X-Lite client) and I see this in the logs: (I'm pretty sure it has something to do with ztdummy, but I dunno... I have the port installed, but I

Hello Users, Good Morning, I'm doing on Conference Bridge with Asterisk + OpenSER with CBMySql modules. And I'm not Using the Zapptel Cards. 9001 ----------> dial 19001(conference Users)-------openSER ---------> Asterisk ------------------------------------------------------------------------ *In Extension.conf * [from-sip] exten =>

Hi list, we have a dual Xeon server with 2GB RAM running Debian Etch 2.6.18.4-686 kernel. The server has 2 B410P cards plugged in. No other card. We installed Asterisk 1.4 trunk with zaptel trunk, ran make b410p, the install mISDN1.1.0 (for bug 9064) configure and compile Asterisk with chan_misdn, all is fine. In misdn-init.conf we have added option=1,master_clock. Asterisk is up and

http://bugs.freedesktop.org/show_bug.cgi?id=15623 Summary: Support for Vimeo.com Product: swfdec Version: unspecified Platform: x86 (IA32) OS/Version: Linux (All) Status: NEW Severity: enhancement Priority: high Component: plugin AssignedTo: swfdec at lists.freedesktop.org ReportedBy:

Hi, I have asterisk installed in the xen virtual server. I installed zaptel 1.4.2.1 and patched it to have ztxen module. I loaded ztxen module but when I try to invoke or call to my meetme application I get the following warning and negative result of connecting to conference: [Feb 5 17:46:13] WARNING[10725]: app_meetme.c:772 build_conf: Unable to open pseudo device [Feb 5 17:46:13] --

Hi! i have an error using the meetme aplication, and just dont work.. my meetme.conf is: [rooms] conf = 700 i calling from a sip phone, the extension number is 600. there is the error: Executing [700@numberplan-custom-1:1] MeetMe("SIP/600-09111e58", "700|MI") in new stack WARNING[20055]: channel.c:3024 ast_request: No channel type registered for 'zap' WARNING[20055]:

Hello- I've been trying to scour both * and linux user group archives for a solution to this particular problem, but I am just plan stuck. I got the latest zaptel sources from cvs, uncommented ztdummy.o in Makefile, ran make; make install then, did depmod -a. All is well up until now. When i do modprobe ztdummy, i get this error: dv@sidetrack:/usr/local/src/zaptel# depmod -a

Hi, I've install zaptel and I don't have a Digium card installed in the machine. So I want to install ztdummy to have Music On Hold working. I've follow these instruction http://www.voip-info.org/wiki-Asterisk+timer+ztdummy and at the point "modprobe ztdummy" it's failing. I'm getting these messages: root@dubic:~# modprobe ztdummy FATAL: Error inserting ztdummy

Hello All, I upgraded Asterisk 1.8.10 to Asterisk 1.8.22 since upgrading I can't get meetme feature to work when dial meetme extension, can you please help? It always worked before, also I do not have dahdi installed on this machine, never did. -- Executing [104 at sipphones:1] MeetMe("SIP/101-00000813", "104") in new stack == Parsing