similar to: Poor call quality – line drop, chopping sound, like robotic voice, Both party could not hear caller voice

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

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

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

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

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]

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

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

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'

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

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. > >

Hi, I'm trying to use reshape but I cannot quite understand how it works. Could somebody help me on this? Example, my data is something like: mydat <- data.frame(tree= 1:10, serra=rep(1:2, c(5,5)), bt01= 101:110, bt02= 201:210, bt03= 301:310, mm01= 9101:9110, mm02= 9201:9210, mm03= 9301:9310) > mydat tree serra bt01 bt02 bt03 mm01 mm02 mm03 1 1 1 101 201 301 9101 9201

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

The Asterisk Development Team would like to announce security release Asterisk 20.3.1. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.3.1 and https://downloads.asterisk.org/pub/telephony/asterisk The following security advisories were resolved in this release:

The Asterisk Development Team would like to announce security release Asterisk 20.3.1. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.3.1 and https://downloads.asterisk.org/pub/telephony/asterisk The following security advisories were resolved in this release:

Hi, While compiling 1.8.8.1, I met the following error: [AR] hash/hash.o hash/hash_bigkey.o hash/hash_buf.o hash/hash_func.o hash/hash_log2.o hash/hash_page.o hash/ndbm.o btree/bt_close.o btree/bt_conv.o btree/bt_debug.o btree/bt_delete.o btree/bt_get.o btree/bt_open.o btree/bt_overflow.o btree/bt_page.o btree/bt_put.o btree/bt_search.o btree/bt_seq.o btree/bt_split.o btree/bt_utils.o db/db.o

I have written a few different summary functions. I want to calculate the statistics by groups and I am having trouble getting the output as a dataframe. I have attached one example with a small dataset that calculates summary stats and percentiles, I have others that calculate upper confidence limits etc. I would like the output to be converted to a dataframe with one of the columns as the

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

The Asterisk Development Team would like to announce the release of Asterisk 20.3.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.3.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of Asterisk 20.3.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.3.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!