similar to: changethread: can't change device with no technology!

Guys. What does this error mean? -- Playing '/var/spool/asterisk/voicemail/intruder/201/unavail' (language 'sp') -- Playing 'vm-intro' (language 'sp') -- Playing 'beep' (language 'sp') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/intruder/201/INBOX/msg0000 format: wav, 0x812b4f0 -- User ended

I'm running Asterisk HEAD from March 4. I've Googled a bit but I can't figure out what causes this error: app_queue.c:374 in changethread: Can't change device '**Unknown**' with no technology! It doesn't seem to be causing any problems, but I'm curious what causes it. I did a few Google searches and found a lot of people asking about it, but no real answers.

Guys, this error has been driving me nuts and I see no indication anywhere as to what it may mean. Anybody has any clues on this? -- User ended message by pressing # -- Playing 'auth-thankyou' (language 'en') -- Playing 'vm-review' (language 'en') -- Saving message as is -- Playing 'vm-msgsaved' (language 'en') Mar 4 21:02:06

Hi all, I must admit that I am stuck. I have a TDM400P card with two FXS and two FXO modules which I had set up and configured so that it was working beautifully. The only problem was that occasionally it would get itself into a state where outgoing calls would simply be met with a very loud static. A reboot would fix this issue and everything would work fine for a while. Recently however,

Hello, After upgrading to the latest development CVS Head, I am now getting regular errors as follows: Dec 17 17:07:30 WARNING[8092]: app_queue.c:340 changethread: Can't change device with no technology! Also, my ability to answer calls with XTen Pro softphone seems to be a bit flaky now. Any ideas? ===== Jason Goecke www.goecke.net Ph: +31.707.504.634 Mb: +31.707.504.634 Fx:

hi all, i have an asterisk install with a digium 4 port fxo card and cisco 7960 sip phones -- running on a compaq Pentium III (Coppermine) at 800Mhz 256KB cache and 1GB of ram. when a call comes in on zap/1-1 for example, the delay between when zap sees the line going to ring state, and when the desktop telephone rings can be as long as 7000 milliseconds (or about 3 or 4 rings on an ear piece).

I have a nice used zhone channel bank I want to experiment with, but need a T1 interface for my * box to do so. The TE410P looks nice, but more money than I want to spend to experiment, and I don't need 4xT1, only 1xT1. Are there any good 1xT1 PCI cards that are recommended? Regards, -Dorn

Hello , I learn handbook-draft.but I think I don't understand asterisk. where I can find some learning book about asterisk? thank u. B.R. John. -----????----- ???: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]?? asterisk-users-request@lists.digium.com ????: 2004?12?24? 7:51 ???: asterisk-users@lists.digium.com ??: Asterisk-Users Digest, Vol 5,

Hello, I am attempting to write a script to launch a callback based on a dial-in service. I have created this call file: --------------------------- channel: IAX2/user@voipjet/011_valid_number maxretries: 3 retrytime: 5 waittime: 5 context: dialtone extension: 912125551212 priority: 1 --------------------------- Where I first attempt to dial the callback user (channel) and then connect the

Ever since I started using asterisk I see often this error message, can sombody tell me what it means? Feb 26 09:20:40 WARNING[29371]: app_queue.c:374 changethread: Can't change device '**Unknown**' with no technology! __________________________________________________________________ Anton Krall

Would anyone care to offer opinions as to the FXO interface which sucks the least :) I have an application in which it appears I must route certain calls out an analog PSTN line. Presently, I am testing an SPA-3000, but I can't seem to get the echo heard on the IP end of the call down to a non-annoying level. Any suggestions welcomed :) -Dorn

I'd like to get VM_CALLERID to include number in addition to name since often when calls come from cell lines or various other, the name is just a city, state and the number would be more usefull. Is there a way to get the number in the VM_CALLERID string, or is there a second variable I can use in formatting email vmail notifications to get the number? Regards, -Dorn

Hi, I'm wondering - why SIP device state doesn't get updated to anything else, except Not In Use. For queue call (with Local channel) i get: app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use) app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use) app_queue.c: The device state of this queue member, Agent/21168, is still 'Not in

Hello, Could anybody tell me if there is a Linux distribution (or Kernel version) that works better with Asterisk. I am newbie and I don't know if there is a preferred Linux/kernel version for Asterisk. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040904/8798ff25/attachment.htm

hello all, I have asterisk 1.4.26 working on CentOS 5.3 and I have the following problem: - when I restart asterisk all the members of the queue are Invalid. - when I make a call to one of the members, of the queue, and then check the state, it turns to "Not in use" for the called phone, and the queue works fine for that member after. - after doing a module reload of the

Are there any common silent failure modes for email notification from the Voicemail module. I put the email and pager email addresses in my entry in voicemail.conf but no mail gets sent when I leave a voicemail. No obvious error messages either, unless I'm just not looking in the right place. Thanks for any clues :) -Dorn

I just installed 1.4.23.1 with the queue realtime logger backport. Here are my configs: musiconhold.conf [default] mode=files directory=/var/lib/asterisk/moh-native random=yes queues.conf [7703] wrapuptime=0 timeout=15 strategy=rrmemory retry=5 queue-youarenext=queue-youarenext queue-thereare=queue-thereare queue-thankyou=queue-thankyou queue-callswaiting=queue-callswaiting

Or, in full: [Sep 20 17:11:26] WARNING[18373]: app_queue.c:2705 try_calling: The device state of this queue member, SIP/612, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. So, what do I check in UPGRADE.txt? This is with Asterisk 1.4.11

Guys, I saw in the Asterisk documentation (queues.conf) that members can register with technologies such as SIP, Dahdi and Location. But I have a specific need for members to be registered as Khompchannel. Ex: member => Khomp/b0L1/9200 But reloading module app_queue.so when I run the command "queue show", the member registered as Khomp appears as invalid:

When working with call queues, if an agent is logged in via AgentCallbackLogin and the extension they are registered at becomes "unavailable" (from a bad connection, or something of the like), calls routed to that extension seemed to be accepted by it, so if the next action for that extension is to go to voicemail, the caller in the queue is sent to the extensions voicemail. Even worse,