similar to: Dead Call But Still Active

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

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

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, 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

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:

So I'm using the READ() application within an IVR, and having a strange issue, and wondering if anyone else has had this problem. When calling from an outside line, and entering the digits during the read() part of my dialplan, it's accepting some of the digits twice, though it's only keyed in once. When testing the dialplan internally, it accepts only the digits that I key in.

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 saw some of the heat about the $20 bounty earlier. So I don't want to put a low bounty out. Quote me a bounty, and I'll see if I can get it approved by management. :-) I'm in need of getting this bug fixed. Bug has all of the details, but basically 1.4.22 broke it all. I've waited as long as I can - hoping the bug would 'resolve itself' - but now I'm putting a

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

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

I'm wondering if there's a way to set which periodic-announce file is played from my dialplan, much like setting the monitor-filename. Something like this: exten => s,n, Set(PERIODIC_ANNOUNCE=foo) This would be a great feature if it doesn't already exist. Or perhaps there's a better way to do this. Thanks for your time. -- Regards, Robert Broyles

Hi, I just got and setup a new ip500 yesterday and it worked for about 15 minutes. Then it froze during a reboot.Now, when power cycled, the logo comes on for 3 seconds and then the screen is blank and nothing further happens. 468* factory reset doesn't work. I am about to send the phone back, but wonderd if anyone had a suggestion first.

Is there a way to set "na.rm=TRUE" as a default, so that this does not have to be re-specified for all of the functions requiring this option? Thanks in advance, Stephanie Broyles sbroyl@lsuhsc.edu [[alternative HTML version deleted]]

Hi. We install ruby, rail, rubygems, rails, mod_fastcgi and fastCGI nad we get from this URL http://bobcares.com/article38.html ans everything install with no errors, but when we try to run "test" we get this: Error from apache error_log: Premature end of script headers: /home/user/public_html/rails/dispatch.cgi /usr/local/lib/ruby/site_ruby/1.8/rubygems.rb:46:in

Is there a way to override the queue wrapup time on the fly? I would like to allow a longer wrapup time for my agents, but if they are already done with closing up the call ticket, I would like them to be able to dial an extension or something to override the wrapup. Is there a way to do that? -- Regards, Robert Broyles

Hi everyone, This is off topic and is for GS technical support really but it seems that there are a lot of Budge Tone 100/101/102 users out there. I've got a Budge Tone-100 (101 - without the extra 10base ethernet connetion?) here. I changed the configuration through its web based interface and I clicked the reboot link. But then something went wrong and ever since then it doesn't

>> Oh I don't doubt the basics, red book is red book and bits are >> identically replicable and re rippable bits. > > I don't see any problem with taking innovation as far as is practical > and saying "it's finished, no more updates". Sure, basics :) Again, I'm meaning in regard to about bugs, docs, porting and nits. > If I want to do freedb