search for: audiohook

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 15:...

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 fail t...

...] DEBUG[27462] app_queue.c: Queue 'assist' Leave, Channel 'IAX2/wally-10884' [Nov 25 17:47:38] VERBOSE[27464] logger.c: == Begin MixMonitor Recording IAX2/wally-10884 [Nov 25 17:47:38] DEBUG[27462] features.c: bridge answer set, chan answer set [Nov 25 17:47:38] DEBUG[27464] audiohook.c: Failed to get 160 samples from write factory 0x7f5aa4095fa0 [Nov 25 17:47:38] DEBUG[27464] audiohook.c: Failed to get 160 samples from write factory 0x7f5aa4095fa0 [Nov 25 17:47:39] DEBUG[27464] audiohook.c: Failed to get 160 samples from write factory 0x7f5aa4095fa0 [Nov 25 17:47:39] DEBUG...

...DEBUG[6914] chan_dahdi.c: Queuing frame from PRI_EVENT_PROCEEDING on channel 0/2 span 1 [Apr 11 15:32:48] VERBOSE[9231] app_dial.c: -- DAHDI/2-1 is proceeding passing it to SIP/2130-000006fb [Apr 11 15:32:48] DEBUG[9231] rtp.c: Ooh, format changed from unknown to ulaw [Apr 11 15:32:48] DEBUG[9232] audiohook.c: Failed to get 160 samples from read factory 0xb7817dd0 [Apr 11 15:32:48] DEBUG[9231] rtp.c: Created smoother: format: 4 ms: 20 len: 160 [Apr 11 15:32:48] DEBUG[6915] chan_dahdi.c: Queuing frame from PRI_EVENT_PROGRESS on channel 0/4 span 2 [Apr 11 15:32:48] VERBOSE[9226] app_dial.c: -- DAHDI/35-...

...DEBUG[6914] chan_dahdi.c: Queuing frame from PRI_EVENT_PROCEEDING on channel 0/2 span 1 [Apr 11 15:32:48] VERBOSE[9231] app_dial.c: -- DAHDI/2-1 is proceeding passing it to SIP/2130-000006fb [Apr 11 15:32:48] DEBUG[9231] rtp.c: Ooh, format changed from unknown to ulaw [Apr 11 15:32:48] DEBUG[9232] audiohook.c: Failed to get 160 samples from read factory 0xb7817dd0 [Apr 11 15:32:48] DEBUG[9231] rtp.c: Created smoother: format: 4 ms: 20 len: 160 [Apr 11 15:32:48] DEBUG[6915] chan_dahdi.c: Queuing frame from PRI_EVENT_PROGRESS on channel 0/4 span 2 [Apr 11 15:32:48] VERBOSE[9226] app_dial.c: -- DAHDI/35-...

...ound 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: Write factory 0x7f32f4005940 was pretty quick last time, waiting for them. [Jan 20 15:21:35] DEBUG[14988] audiohook.c: Write factory 0x7f32dc00a0d0 was pretty quick last time, waiting for them. [Jan 20 15:21:35] DEBUG[15013] audiohook.c: Write factory 0x7f32f0016660 was pretty quick last time, w...

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...

...t 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' shows that the channel is still active. I have enabled full logging, and noticed this when it happens: [Mar 31 12:12:27] DEBUG[25854] audiohook.c: Failed to get 160 samples from write factory 0xacfbce0 [Mar 31 12:12:27] DEBUG[25854] audiohook.c: Failed to get 160 samples from write factory 0xacfbce0 [Mar 31 12:12:27] DEBUG[25854] audiohook.c: Failed to get 160 samples from write factory 0xacfbce0 [Mar 31 12:12:27] DEBUG[25854] audiohook...

...times 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 message - rejected , no callid, len 366 [Mar 25 17:58:40] DEBUG[13617] audiohook.c: Write factory 0x1bfa000c was pretty quick last time, waiting for them. [Mar 25 17:58:40] DEBUG[13579] audiohook.c: Read factory 0x1bd61fe0 was pretty quick last time, waiting for them. [Mar 25 17:58:40] DEBUG[23168] chan_sip.c: Setting SIP_ALREADYGONE on dialog 5bd1acee539e699b4f5e79c94a348361 a...

...n 2019-04-04 13:41:15 UTC We also had debugging enabled and things were output to our debug file for 17 more seconds.... The blank line in my e-mail is once again a line with all [NUL}... (in notepad+ the blank line in the middle is all [NUL][NUL] [NUL][NUL]....) [08/12 15:31:12.776] DEBUG[6781] audiohook.c: Read factory 0x7f079389bff8 and write factory 0x7f079389ca38 both fail to provide 160 samples [08/12 15:31:12.777] DEBUG[6709] audiohook.c: Failed to get 160 samples from read factory 0x7f07937066d8 [08/12 15:31:12.777] DEBUG[6709] audiohook.c: Read factory 0x7f07937066d8 and write factory 0x7f0...

On 21-11-16 15:17, Matthew Jordan wrote: > > On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > Hello > > when using Asterisk version 13.12.2 I notice that it takes up to > 30 seconds (sometimes even longer) for a call queue to call its > members. > >

...well. One of the others is running in the same hardware and environment but does not have this problem. In the log files the only weird thing I see is: [Sep 6 11:09:48] DEBUG[24238] chan_dahdi.c: Write returned -1 (Resource temporarily unavailable) on channel 49 [Sep 6 11:09:48] DEBUG[24239] audiohook.c: Read factory 0x2aaaac5d3c60 was pretty quick last time, waiting for them. [Sep 6 11:09:48] DEBUG[24288] chan_dahdi.c: Write returned -1 (Resource temporarily unavailable) on channel 54 [Sep 6 11:09:48] DEBUG[24452] audiohook.c: Write factory 0x2aaacc67ad08 was pretty quick last time, waiting f...

Hello, My configuration is : Card 0 - kernel dummy sound card Card 1 - my soundcard I have a jackd running in background. My jackd launch command is : jackd --port-max 16 --realtime --no-mlock -d alsa --playback hw:1,0 --capture hw:1,0 --rate 8000 --period 1024 --shorts --inchannels 2 --outchannels 2 --dither triangular & 1 ) I open asterisk with chan_alsa.so connected (with asoundrc) to

...he MixMonitorID can be stored as a channel variable using the 'i' MixMonitor option and is returned upon creation if this option is used. As part of this change, if no MixMonitorID is specified in the manager action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor audiohooks on the channel. Previous behavior would set the flag on the first MixMonitor audiohook found. - ### pbx_dundi: Add PJSIP support. DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the pjsip_outgoing_endpoint option to be set in dundi.conf. - ### test.c: Fix counting of...

...he MixMonitorID can be stored as a channel variable using the 'i' MixMonitor option and is returned upon creation if this option is used. As part of this change, if no MixMonitorID is specified in the manager action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor audiohooks on the channel. Previous behavior would set the flag on the first MixMonitor audiohook found. - ### pbx_dundi: Add PJSIP support. DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the pjsip_outgoing_endpoint option to be set in dundi.conf. - ### test.c: Fix counting of...

Hello, ChanSpy can not be used on a Channel that is being recorded with MixMonitor. How can I verify if a channel which I want to spy on, is currently not being recorded ?! Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120202/7954fe9e/attachment.htm>

Hi all, I read on serveral pages that chan_spy is not part of asterisk anymore as on http://www.voip-info.org/wiki-Asterisk+cmd+ChanSpy on the bottom of this page. I have a testing server with debian-testing and debian packages for asterisk installed. In the modules directory /usr/lib/asterisk/modules is a app_chanspy.so already there. The currently installed version is 1.4.13. So, it is

...n in asterisk to manipulate the audio in a call? I would like to, for example change the voice of one caller but without manipulating the audio that comes from another caller. I have read about something called JACK but i don't know if i can use it for this (or how to use it). I am playing with audiohooks now, but I don't think I can change only my voice - whole audio in a call gets manipulated. Thanks, Slawek -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100420/d0b876b5/attachment.htm

Good morning to all. In my Asterisk console i have a lot of this messages: [Dec 14 10:50:52] DEBUG[12790]: audiohook.c:215 audiohook_read_frame_both: Read factory 0x8afae68 and write factory 0x8afb884 both fail to provide 160 samples [Dec 14 10:50:52] DEBUG[12790]: audiohook.c:221 audiohook_read_frame_both: Write factory 0x8afb884 was pretty quick last time, waiting for them. Someone can tell me what this messa...

...he MixMonitorID can be stored as a channel variable using the 'i' MixMonitor option and is returned upon creation if this option is used. As part of this change, if no MixMonitorID is specified in the manager action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor audiohooks on the channel. Previous behavior would set the flag on the first MixMonitor audiohook found. - ### pbx_dundi: Add PJSIP support. DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the pjsip_outgoing_endpoint option to be set in dundi.conf. - ### test.c: Fix counting of...