similar to: Debug messages.

Hello list, I've got a whole lot of these in my debug log : [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Read factory 0x1cea33a0 and write factory 0x1cea3dd8 both fail to provide 160 samples [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Failed to get 160 samples from read factory 0x1cea33a0 [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Read factory 0x1cea33a0 and write factory 0x1cea3dd8 both

7. Take an Asterisk training course and become a dCAP. As for "we have try to solve it but we're lack of asterisk knowledge" - would you get a plumber to service your car? Why not employ (as in 'pay money') somebody to investigate this further. As Satish pointed out - QoS type issues take a lot of debugging and that usually has to be done on-site. BTW - I doubt any of

One of our client facing this issue, we have try to solve it but we're lack of asterisk knowledge. Anybody can help us? Isn't any problem with asterisk configuration or the problem come from PRI E1 itself? [Apr 11 15:32:48] VERBOSE[9231] chan_dahdi.c: -- Requested transfer capability: 0x00 - SPEECH [Apr 11 15:32:48] DEBUG[6888] channel.c: Avoiding initial deadlock for channel

Hi, I am having a difficulty with getting two realtime user?s to bridge on answer. I have managed successfully to bridge the same two users/channels via the Bridge Manager api command and confirm that the two communicate directly bypassing the asterisk server (I confirmed this with Wireshark). Does anyone have some ideas? I have put some log entries below. I haven?t attached my

Hello list, what does this mean in the debug-log : [Jan 19 15:11:04] DEBUG[1475] audiohook.c: Write factory 0x153cf678 was pretty quick last time, waiting for them. [Jan 19 15:11:04] DEBUG[1701] audiohook.c: Read factory 0x14fe5ef0 was pretty quick last time, waiting for them. [Jan 19 15:11:04] DEBUG[1475] audiohook.c: Read factory 0x153cec40 and write factory 0x153cf678 both fail to provide

I'm having a strange issue, and not really sure where to even begin to troubleshoot it. First let me explain that I have all agents setup locally ( local/100 at agents/n) A call will come in and ring to the agent. When the agent answers the call, they just hear a dial tone. Agent hangs up. Asterisk still shows the agent as 'in use' in queue status. And 'show channels'

Hi every body our Calls are begging dropped for no reason and it starts with the sound quality dropping and then the caller unable to hear our call center agents. Then the call drops or the caller hangs up unable to hear. I could see following lines inside full log ---------------------------------------------------------------------------------- [Jan 20 15:21:35] DEBUG[14982] audiohook.c:

Hello, I see a lot of these messages in the debug log : /[May 3 15:47:09] DEBUG[19081] audiohook.c: Failed to get 160 samples from write factory 0xae17e18 [May 3 15:47:09] DEBUG[19081] audiohook.c: Failed to get 160 samples from write factory 0xae17e18 [May 3 15:47:09] DEBUG[19081] audiohook.c: Read factory 0xae173e0 and write factory 0xae17e18 both fail to provide 160 samples [May 3

I have a very difficult to diagnose problem. We are running Asterisk 1.6.2.11, DAHDI 2.4.0, FreePBX 2.8 on a Centos 5.5 server (Xeon quad core 4gb). Last week we started having a problem where the server will randomly stop sending and receiving calls. Asterisk does not die or crash. You can get the CLI but any command you input will not respond. All phones have "No Service" on their

Dear sir, We have been using asterisk for 4 years. Now we have got problems which occurs during the attended transfer. But we are not always getting this problem. Sometimes it happens. But now we cannot understand why this is happening? problem is:"Failed to play transfer sound! " The log of asterisk is as like as followings: [Mar 25 17:58:40] DEBUG[23168] chan_sip.c: Invalid SIP

> > Sorry, of course cdr.conf not queues.conf. marcus > > Am 01.12.2010 19:16 schrieb "marcus rothe" <synco16 at googlemail.com>: > > > Hi Rodrigo, have you got enabled the appropriate line in queues. Conf? > Regards Marcus > > Thanks very much, I include the line "unansweredy=yes" in the cdr.conf and solve the problem. Thanks again! --

We have a customer where their VM running Asterisk appears to have crashed. Fortunately, we had some debugging enabled. The asterisk messages file has this... (in notepad+ the blank line in the middle is all [NUL][NUL] [NUL][NUL]....) [08/12 15:30:55.880] VERBOSE[6920] app_mixmonitor.c: Begin MixMonitor Recording CBRec/IS__a37ae004-c780-4c7f-88a9-a04402f0ab4e-0000e70f [08/12 15:30:55.881]

Hi to all! I'm trying to create a context for integration with extensions.lua and libsql.mysql, but I'm not getting to run. When I reload the module pbx_lua.so the following error appears: [Feb 24 16:59:29] ERROR[30749]: pbx_lua.c:1249 exec: Error executing lua extension: error loading module 'luasql.mysql' from file '/usr/lib/lua/5.1/luasql/mysql.so':

Hi to all. I'm begin a use the AMI and i have the need to get the uniqueid from the call i have generate using the Action Originate. Anyone can help me? When I generate these commands: action: Originate channel: SIP/101 application: Dial data: SIP/100,120,Ttr The only response I get when the call is answered, is this: Response: Success Message: Originate successfully queued Thanks a

Good morning everyone. I wonder if it is possible, without touching the source code, to Asterisk save the cdr with date in unix time instead of the default date. It's possible? Thanks in advance, -- Rodrigo Lang Opening your mind - Just another Open Source site<http://openingyourmind.wordpress.com/> -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi to all. I make some tests with Asterisk 1.8.2 in Realtime. But i have one problem, the asterisk don't connect in the base and show this message: [Feb 17 11:18:01] WARNING[19061]: res_config_mysql.c:441 realtime_multi_mysql: MySQL RealTime: Invalid database specified: 'asterisk_teste' (check res_mysql.conf) I checked the asterisk config file (res_mysql.conf) and the configuration

Hi to all. I am conducting several tests with the Asterisk manager and I noticed something that I believe to be a problem. When I generate a call with the Action Originate with the Async option true, the event OriginateResponse returns normally. But when I generate a call in the same way, without the Async option, the event OriginateResponse does not come. Is this a bug? It was fixed in some

Good morning everyone. I wonder where I can find a list of the reasons the event OriginateResponse. I found this list [1]. But in my Asterisk has other reasons too. [1] 0 = no such extension or number 1 = no answer 4 = answered 8 = congested or not available Thanks in advanced, -- Rodrigo Lang Opening your mind - Just another Open Source site<http://openingyourmind.wordpress.com/>

Hi to all. I have a problem in the AMI. Sometimes the AMI don't generate the event NewState when the exten of destiny is Ringing and sometimes don't show me the callerid in this events. The event NewState what i refer: Event: Newstate Privilege: call,all Channel: SIP/17-00006fd6 ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 4191920902 CallerIDName: 4191920902 Uniqueid:

Good afternoon list. I am facing a problem with the CDR and Queue_log tables (MySQL). The ABANDON events is being saved correctly in queue_log, but in the table CDR is not saving the registry of such abandoned calls. Apparently the CDR table is functioning normally, I have several records of links in it. From what I noticed, is only the events abandonment that are malfunctioning. With this