similar to: asterisk 1.8.22 - : app_meetme.c:1248 build_conf: Unable to open DAHDI pseudo devic

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 was switching from zaptel to dahdi and got latest SVN from everything. Compiling works fine. kernel module dahdi_dummy is loaded. /dev/dahdi/pseudo exists Trying to go into a meetme does not work: [Aug 11 14:04:45] -- Executing [8001 at client_int_sgmobile:1] MeetMe("SCCP/6000-00000001", "444|dcIM") in new stack [Aug 11 14:04:45] WARNING[4184]: app_meetme.c:775

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]:

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

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

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",

Hello ! In file app_meetme.c we can read A ZAPTEL INTERFACE MUST BE\n" "INSTALLED FOR CONFERENCING FUNCTIONALITY.\n" I receive message, when I try conference WARNING[28686]: File app_meetme.c, Line 151 (build_conf): Unable to open pseudo channel -- Playing 'conf-invalid' Does it means that I cannot establish conference without any hardware zaptel interface ??? What

Ok.. I got * and SIP working internally now .. still wrestling with connecting two remote * pbx's together.. I'll save that for another day though :) I setup Meetme on this new * PBX, but when I try to dial to join the conference, I hear a recording saying the conference is invalid or something to that effect. Here's a copy of my log files: == Parsing

Hi All, I've had a great time exploring * on & off over the past month and have had a deal of success, very impressed with this product... I have an issue on a debian unstable system with the zap interface. I've compiled and loaded the ztdummy successfully, but when attempting to connect to a conference room I get failure. I get the error chan_zap.c:757 zt_open: Unable to open

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

I'm running 2.4.5 and app_meetme never plays conf-hasleft or conf-hasjoined with user names. I looked at app_meetme.c, but couldn't determine the cause. Any suggestions are greatly appreciated. exten => 600,1,MeetMe(600|i) I get the following: -- Executing MeetMe("SIP/jon-21f8", "600|aciMps") in new stack == Parsing '/etc/asterisk/meetme.conf': Found

Hi, I don't have a zaptel device for conferencing. I read from the lists, that ztdummy and zaprtc need to be installed to get conferencing. I could able to compile successfully with ztdummy and when i receive the call it says, -- Goto (13732,s,1) -- Executing MeetMe("SIP/-08118800", "1234") in new stack == Parsing

Is this me or what? -- Playing 'demo-congrats' -- Executing MeetMe("H323:996", "") in new stack -- Playing 'conf-getconfno' == Parsing '/etc/asterisk/meetme.conf': Found WARNING[17425]: File app_meetme.c, Line 151 (build_conf): Unable to open pseudo channel -- Playing 'conf-invalid' -- Playing 'conf-getconfno'

I have ztdummy installed: Module Size Used by ztdummy 3464 0 zaptel 218756 1 ztdummy crc_ccitt 2176 1 zaptel ohci_hcd 16388 0 floppy 49028 0 pcspkr 2180 0 piix 8580 0 [permanent] ehci_hcd 24456 0 uhci_hcd 26256 0 rtc

I do not have any zaptel hardware on the Asterisk box, I could not have meetme functioning. I did modify the Makefile in zaptel directory on line 168 by including ztdummy as one of the modules to compile in. The error message from the concole: -- Executing MeetMe("SIP/1002-e9ca", "4700") in new stack == Parsing '/etc/asterisk/meetme.conf': Found

When I call meetme: exten => 1000,1,Answer exten => 1000,n,Meetme(|||d) Asterisk is complaing with: -- Executing Answer("IAX2/xxx.yyy.142.204:4569-2", "") in new stack -- Executing MeetMe("IAX2/xxx.yyy.142.204:4569-2", "|||d") in new stack -- Playing 'conf-getconfno' (language 'en') Warning, flexible rate not

I try to use meetme app after reading manual i compile and install zaptel with ztdummy when i make lsmod i have ztdummy 2532 0 (unused) wcusb 20064 0 (unused) zaptel 179168 4 [ztdummy wcusb] usb-uhci 26348 0 [ztdummy] usbcore 51616 0 [wcusb usb-uhci] after it i recompile asterisk and after it i have

Hello, I want to share mailbox between two extensions Ext. 101 Ext. 102 I want the messages to go to mailbox 101, when when checked mailbox from extension 102 to be able to clear the bliking red light. here is extensions.conf exten => 102,hint,SIP/${EXTEN} exten => 102,1,Dial(SIP/101&SIP/102,20,t) exten => 102,2,Voicemail(101,u) exten => 102,102,Voicemail(101,b) exten =>

> > Hi, You can achieve this by integrate CCM and asterisk using SIP trunk. In CCM you can create SIP trunk, After creating SIP trunk in between CCM and asterisk, you have to configure dialplan on CCM to pass the calls to asterisk. One the caller id comes to Asterisk you have to use extension.conf to route the calls. You can also try with freepbx GUI to configure inbound route, it makes

Hi, if I dial meetme from extension 200 directly it works ok - I get moh as only user (first trace). If I dial to other local extension and trasfer from there I get second trace... Apparent difference between those two is warning : Jan 27 11:06:33 WARNING[6133]: res_musiconhold.c:466 moh_alloc: No class: random What this could mean ? Direct Call log-----------------------------------------: