similar to: bridging - Didn't get a frame from channel

I upgraded from 1.0.9 to 1.2.1. In 1.0.9 everything worked perfect. Now, I call in my IVR, and after navigating in menus when I get dialtone for dialing extension, Sound is choppy and I get bunch of messagess: -- Silence suppression is disabled (option_silence_suppression=0 chan->timingfd=30) -- Silence suppression is disabled (option_silence_suppression=0 chan->timingfd=30) -- Silence

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

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 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

Hi all, i got a lot of this messages if only one caller is in a meetme conference and it playing a MusicOnHold Sound. If a second Caller entry the Conference the messages ended. DEBUG[11794] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=61 What does this message mean? Thanx for answers Daniel

Hello List - Here's the setup: Mediatrix 1102 ATA (t38enabled) <--> Asterisk 1.4.1 <--> IP <--> SIP GW <--> TDM The T38 call comes up perfect - I see the initial invite, followed by G711, Re-Invite, T38 establishes, Fax Completes, T38 Stops, Call Down. here's the problem - I see the following in my console: [Mar 19 05:09:38] WARNING[4745] chan_sip.c: Can't

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

Hi, I've installed * 1.2.1 with latest bristuff patches (0.3.0-PRE-1f). When I activate music-on-hold on a SIP-to-SIP connection, the music sounds like in a fast-forward play mode. On the *-console I can see much lines like this: -- Silence suppression is disabled (option_silence_suppression=0 chan->timingfd=18) What's going on? With bristuff 0.3.0-PRE-1d everything works fine (but

Hi Folks, I'm trying to use my mobile as a trunk via bluetooth - calls done in a softphone go thru GSM network and calls destinated to my mobile are answered at the softphone. I have asterisk configured to do so but I'm facing an issue - Audio is audible but it?s not intelligible. I feel like the audio is breaking. Below is the asterisk log. I also get lots of ?hci_scodata_packet: hci0

I'm tying to use park and announce for call park on Asterisk 1.4.2 but I'm having trouble getting it to work properly. This use to work with an older version of Asterisk. A telephone on the PSTN calls an IP phone. The IP phone is assigned extension 3-8396. 3-8396 answers the call and attempts to perform a blind transfer to x700, the parking lot number. The transfer gets to Asterisk,

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 all on a debian amd64 i've installed (from source) asterisk 1.4.30 On the system we have in average 50 concurrent calls in queue and 40 sip members. I'm experiencing an apparently random problem: sometimes some users receive 2 calls from asterisk, apparently ignoring the ringinuse=no settings. It appears on users that are members of many queues As you can see from the log, the

Hello, I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended Transfer

Hello, I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended Transfer

Hello, how are you? I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended

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, I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended Transfer

I am doing a little application to originate a call through Asterisk via AMI (Perl Asterisk::Manager). It logs in successfully, does an originate command with Exten: 0020 (which is set up to answer and wait for 60 then hang up) Channel: SIP/5101234567 at test-host (which comes to my desktop machine also running Asterisk). At the target machine I see only a CANCEL to which it immediately responds

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'