similar to: meetme with ztxen - WARNING[10725]: app_meetme.c:772 build_conf: Unable to open pseudo device

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

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",

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

Hi, I have the following problem that when asterisk receives SIP response 302 it cannot forward the call I get such debug: [Jan 4 10:43:27] WARNING[18671]: channel.c:3281 ast_request: No channel type registered for 'Local' [Jan 4 10:43:27] NOTICE[18671]: app_dial.c:505 wait_for_answer: Unable to create local channel for call forward to 'Local/poczta at routing-sip' (cause = 66)

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, I have the following problem that when someone connects to my conference and is the only member music on hold is played just for one second or less and then stops: [Feb 1 10:38:46] -- Started music on hold, class 'default', on channel 'SIP/sip.touk.pl-0083dad0' [Feb 1 10:38:46] -- Stopped music on hold on SIP/sip.touk.pl-0083dad0 I use the following command to

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

Hi, I'm trying to migrate a Asterisk build-environment from a physical to a paravirtualized machine on Xen 3.0 - on this system Asterisk don't need to run at all, it is only for building RPMs. Host OS and guest OS both are CentOS 5.2. The build of Asterisk, Asterisk-addons and Zaptel works, but MeetMe and some other components are not compiled because Zaptel was not installed on the

Hi, I am trying to utilize MWI with sip channel. when my client sens a SUBSCRIBE to asterisk I get info that user not found: <-------------> [Jan 28 11:49:02] --- (19 headers 0 lines) --- [Jan 28 11:49:02] Creating new subscription [Jan 28 11:49:02] Sending to 192.168.129.38 : 7060 (no NAT) [Jan 28 11:49:02] Found peer 'hellboy' [Jan 28 11:49:02] Looking for hellboy in routing-sip

With 1.6.1, I'm trying to set up a test of meetme for creating dynamic conferences. cat meetme.conf [rooms] conf => 600 extensions.conf: [meetme] exten => 2663,1,MeetMe(,D) exten => 2663,n,Hangup() exten => 2666,1,MeetMe() exten => 2666,n,Hangup() What I'm expecting is to dial 2663, get a conference room number ( 600, I suppose since it's the only room ), and set

Hi, I am using Realtime for sip configuration. When there is an INVITE which arrives at asterisk asterisk makes the following selects: Nov 20 10:37:10] DEBUG[31852]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Everything is fine. [Nov 20 10:37:10] DEBUG[31852]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_conf WHERE name = 'tzl' [Nov

Does Asterisk need a sound card or functional Console/dsp to answer inbound DID number from PRI and playback .gsm files? I can call from any of the SIP extensions on Asterisk and hear audio from Playback(), MeetMe(), or MOH. The problem I am having with calls from my PRI is as follows: I have an Asterisk (CVS-HEAD-08/25/04-20:28:51) currently interfacing a NEAX 2400 IPX with PRI. I have a

Hi, When I have the following extension matching defined: exten => _an_.,1,NoOp(-- Context routing-sip-announcement for ${EXTEN} --) Asterisk doesn't find it when it receives such SIP request: <--- SIP read from 192.168.129.38:7160 ---> INVITE sip:an_hellboy at ms.sip.rd.touk.pl SIP/2.0 Record-Route: <sip:192.168.129.38:7160;lr=on> ... for instance when I use such extension:

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

I was searching for help on how to handle the errors that are returned from the MeetMe application. for instance. 1) if a user tries to join a conference that is locked, allison says that the conference is locked and then comes back to the dialplan, however it does not continue down the dialplan. I have a meetme command on Priority 8, and the CLI says that it returned non zero (as the wiki

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

Hi, I have the following situation I use asterisk as o gateway between networks. What is the reason for such response? What are the criteria for such evaluation? SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.129.74:5160;branch=z9hG4bK17c3.17db29e7.0;received= 192.168.129.74 Via: SIP/2.0/UDP 192.168.129.74;branch=z9hG4bK17c3.23083974.0 Via: SIP/2.0/UDP

Hi all, I have a problem starting the ztdummy. Here is what happens: [root@asterisk /]# modprobe ztdummy Notice: Configuration file is /etc/zaptel.conf line 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