similar to: Latest head giving app_queue.c:340 error

qualify= and mailbox= do not work with the realtime configuration engine. It doesn't matter if you specify them in the database, the thread that handles them will never look at the peers you have defined in the database, only the ones defined in iax.conf. --------------------------- Thank you. Will this be a permanent situation, or be resolved in future releases? ===== Jason Goecke

I have installed the CVS Head as of 12/12/04, as well as the asterisk-addons to ensure that /usr/lib/asterisk/modules/res_config_mysql.so exists. I have configured the following (after building a new DB with the appropriate SQL examples, with mods to drop the invalid keys, on the Wiki): - /etc/asterisk/res_mysql.conf [general] dbhost = 127.0.0.1 dbname = my_db dbuser = my_uname dbpass =

I have installed the CVS Head as of 12/12/04, as well as the asterisk-addons to ensure that /usr/lib/asterisk/modules/res_config_mysql.so exists. I have configured the following (after building a new DB with the appropriate SQL examples, with mods to drop the invalid keys, on the Wiki): - /etc/asterisk/res_mysql.conf [general] dbhost = 127.0.0.1 dbname = my_db dbuser = my_uname dbpass =

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

After I leave a voicemail for an extension and hangup, my asterisk console (with debug turned up quite high) shows two error messages like: WARNING[7664]: app_queue.c:341 changethread: Can't change device with no technology! WARNING[7668]: app_queue.c:341 changethread: Can't change device with no technology! Clues? Thanks! -Dorn

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,

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

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.

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).

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

Hi all, We've been experimenting with the app_queue application, and it works quite well. The only problem we encountered was that outgoing calls (to the operators) aren't logged in CDR. Example, * operators dial a specific number/extension, and AddQueueMember(..) runs (they get added without any problems), and they Hangup. * normal users dial the support/hotline number, get added

Given: Queue(foo|tHnr||bar) where queue foo includes something like IAX2/gw/18005551212 should # transfer work on the remote phone? A read of app_queue.c looks like it ought to work, but all I get is dtmf sent to the caller. (Incidently, I'd really prefer to be able to hit eg * during the announcement to have app_queue continue on as if there were a timeout. Has anyone looked into doing

Before I report this as a bug (and get whacked with more bad karma), I'd like to make sure I'm understanding this "feature." I'm defining a queue with a couple of SIP phones as the memebers -- not agents. queue.conf allows you to set an exit context such that if set (and you use the "T" or "t" option) allows the caller or callee to transfer the call to

Hi, I've just uploaded a patch here: https://issues.asterisk.org/view.php?id=16925 This patch introduces a new parameter; "congestion" to both RINGNOANSWER in queue_log and AgentRingNoAnswer AMI event, which is set to 1 if the call failed to go through because of technical difficulties. And it also is more verbose than app_queue has been earlier, since app_queue usually silently

Anyone seen this? It ocurred on a 'reload app_queue.so' command. Asterisk version is 1.2.9.1. Tried again, but it was not immediately reproducable. Doug. (gdb) bt #0 reload_queues () at app_queue.c:3339 #1 0xb778a7a8 in reload () at app_queue.c:4012 #2 0x0805bb44 in ast_module_reload (name=0x8137cc7 "app_queue.so") at loader.c:257 #3 0x08092b3f in handle_reload (fd=33,

I noticed a few issues with app_queue just wanted to know if its sip related or ata186 related: Ext 111 and Ext 112 are dynamically loged into the queue via AddQueueMember. Call hits queue with fewestcalls routing. Rings ext 111 if 111 doesn't answer. It rings ext 112. If for some reason ext 112 doesn't answer it rings back to 111. Again at this point ext 111 isn't answered it

The new app_groupcount looks great for most applications but it a is a step back for call queueing... since app_queue calls physical interfaces and not extensions, app_groupcont can't be used to limit the calls passed to a dynamically added agent. I presently use the broken sip incominglimit feature (even though it's less than ideal as it also limits outgoing calls preventing

Hi, This is the output from queue show 28: 47 (DAHDI/g0/12345678) (realtime) (Ringing) has taken no calls yet Why is the devicestate "Ringing" when no channels is calling this number, and the queue says "has taken no calls yet"? Is it picking up the general state of a random channel on g0 in dahdi? Or what is happening? It only seems to happen with this particular

--- Jonathan Tew <jonathan@ultracart.com> wrote: We've got the app_queue configured to supposedly allow for a call to be transferred. When the call comes in and an agent answers it (using X-Lite Pro) and then decides to transfer the call (using the SIP phone interface) they get disconnected from their call and after left logged in to the queue system. Obviously we're doing