similar to: Error in MeetMe modules ?

I've got a question about codec_dahdi witrh a system running Asterisk 1.6.0.6 and DAHDI 2.1.0.4 with a TE410P card. The system is used primary to route calls between different PRI connections, so no transcoding between codecs is happening as far as I am aware. 1) How can I use codec_dahdi? Would it be useful when passing a call from one dahdi channel to another dahdi channel? 2) I'm

I have a CentOS release 4.7 box running asterisk-1.4.26.1 with dahdi-linux-2.2.0.2 and dahdi-tools-2.2.0 I regularly get these messages, is this something i should be worried about? [Aug 21 01:05:07] VERBOSE[4343] logger.c: codec_g726.so => (ITU G.726-32kbps G726 Transcoder) [Aug 21 01:05:07] ERROR[4343] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory [Aug

I have asterisk 1.8.2.3 + A102D Sangoma card 2 port T1. when i am starting asterisk i am getting this error on console. func_callerid.so => (Party ID related dialplan functions (Caller-ID, Connected-line, Redirecting)) == Registered application 'PrivacyManager' app_privacy.so => (Require phone number to be entered, if no CallerID sent) == Registered custom function

Hi actually, i test a new Asterisk Server and i want add Mysql Realtime SIP. I read on the wiki: =================================================== Database Config put the following in res_mysql.conf [general] dbhost = 127.0.0.1 dbname = asterisk dbuser = myuser dbpass = mypass dbport = 3306 Values in sip.conf or iax.conf like in older versions of * are no longer used. Database Table Lets

After updating to the latest CVS Feb 17 15:20:03 WARNING[15317]: config.c:819 read_config_maps: The 'sipfriends' table is obsolete, update your config to use sipusers and sippeers, though they can point to the same table. == Binding sipusers to mysql/asterisk/sip == Binding sippeers to mysql/asterisk/sip Feb 17 15:20:03 WARNING[15317]: config.c:823 read_config_maps: The

I have a Linux-Vserver guest running CentOS 5.5 with Asterisk 1.6 installed from the asterisk.org and digium.com repositories. I have Asterisk starting (service asterisk start) but see errors about dahdi in /var/log/asterisk/messages. ... ERROR[25658] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory Linux-Vservers don't allow, under normal circumstances,

Hello list! I?m having an issue when reloading Asterisk, I?ve had this problem in Asterisk 1.6.1.6 so I upgrade to 1.6.2.1 version, but I still have the same error. For example, I send a "reload" in Asterisk CLI and this is the output: isb152*CLI> reload == Parsing '/etc/asterisk/extconfig.conf': == Found == Parsing '/etc/asterisk/manager.conf': == Found

Hi all, Trying to do what should be a basic info retrieval from my asterisk network -- get the list of IAXPeers and break down by IP, name, and Status etc. Asterisk 1.0 Manager unfortunately. Using PHP. Easy to get the response -- but parsing it is impossible! That table format throws me for a loop ... Any idea of an easy way to seperate out the columns of that horrid 'table' reponse

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-----------------------------------------:

I am getting the following warnings on the CLI when loading Asterisk 1.6.1-rc1: [Feb 12 12:32:34] NOTICE[22261]: timing.c:59 ast_install_timing_functions: Multiple timing modules are loaded. You should only load one. [Feb 12 12:32:34] ERROR[22261]: codec_dahdi.c:398 find_transcoders: Failed to open /dev/dahdi/transcode: No such file or directory [Feb 12 12:32:33] WARNING[22261]:

Hi, I've just done up a quick proggy to show me the status of my IAX peers from my windows box. It plugs into the simple manager proxy. You can see more information (including a screenshot) at: http://www.sineapps.com/news.php?rssid=384 You can download it directly from: http://www.sineapps.com/down/IAXPeers.zip Could you please have a look and let me know your thoughts. -- Cheers,

Is there any reason why IAXPeers output is different from SIPPeers output? The response has no Eventlist: start Ej. Response: Success Eventlist: start Message: Peer status list will follow Event: PeerEntry Channeltype: SIP ObjectName: 1001 ChanObjectType: peer IPaddress: 192.168.175.1 IPport: 63772 Dynamic: yes Natsupport: no VideoSupport: no TextSupport: no ACL: no Status:

Hi! I am proving Asterisk 1.2.24 in realtime with MySQL 5.0.27 using Idefisk softphones. I followed the steps of "how to" of voip-org but always have this error: Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime:

I just upgraded to 1.6.2.rc-1 after running betas 2 and 3 with no problems and while everything seems fine i get these message at startup and than all is well. Should I be worried or do i need to let the team know about this? Also, is not finding "/dev/dahdi/transcode" a problem I should be worried about? And lastly conf2ael always segfaults when I try to run it. it did run once

Hi, I am setting up a small call center using *. I have ZAP setup for incoming calls and IAX setup for agents. Agents login using AgentCallbackLogin. When customers call, it's getting picked up and when queue is trying to call back the agents, I am getting error. I am using CVS HEAD, and updated just now. The error is: -- Executing Answer("Zap/1-1", "") in new

I have an Ubuntu system and went into Synaptic and checked asterisk for installation. Once installed, I started it with /usr/sbin/asterisk -vvvgc and got the following output with several errors and notices. Do I need to do more or are these ok? I expected to have some conf files in /etc/asterisk but there is nothing there. Thanks! Created by Mark Spencer <markster@digium.com>

Hello, I have transcoding card TC400P installed in server running Debian with Asterisk 1.4.23. Everything seams to be fine and after I boot up server I see in dmesg: 7.590966] Zapata Telephony Interface Registered on major 196 [ 7.590966] Zaptel Version: 1.4.12.1 [ 7.590966] Zaptel Echo Canceller: MG2 [ 7.610963] zttranscode: Loaded. [ 7.618969] wctc4xxp: tc400b0: Attached to

When starting *, I get the following when the chan_iax2.so loads: [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2)) == Manager registered action IAXpeers == Parsing '/etc/asterisk/iax.conf': Found WARNING[16384]: File chan_iax2.c, Line 4980 (set_config): Ignoring port for now == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) == Using TOS

Hey, whenever I try to load, I get these errors Jan 15 16:37:24 ERROR[7573]: chan_iax2.c:7486 load_module: Unable to bind to 0.0.0.0 port 4569: Address already in use Jan 15 16:37:24 WARNING[7573]: loader.c:345 ast_load_resource: chan_iax2.so: load_module failed, returning -1 == Manager unregistered action IAXpeers == Unregistered channel type 'IAX2' Jan 15 16:37:24 WARNING[7573]:

I signed up for an IAXTEL account and have been trying, unsuccessfully, to get it working. In IAX.CONF I have: [iaxtel_out] type=peer host=iaxtel.com username=USERNAME secret=SECRET auth=rsa inkeys=iaxtel [iaxtel] type=friend context=incoming host=iaxtel.com auth=rsa inkeys=iaxtel However, when I start Asterisk, I get the following warning: [chan_iax2.so] => (Inter Asterisk eXchange