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

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

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

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

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

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

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]

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

Hi, Starting in Asterisk 1.8.0, Asterisk supports connected line updates. This is fantastic for SIP. How can I prevent them from being sent to a PRI channel? I'm having problems when a call is answered by an internal SIP extension, then transferred (blind or attended) to another internal SIP extension. One of my PRI providers can't handle the ROSE_ETSI_EctInform APDU and drops the

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!

The Asterisk Development Team would like to announce the release of Asterisk 18.18.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/18.18.0 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 18.18.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/18.18.0 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-18.20.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/18.20.0 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-18.20.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/18.20.0 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-20.5.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.5.0 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-20.5.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.5.0 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!