similar to: Blip every 30 seconds?

Hi, I have an IVR that sounds just fine and dandy over ZAP. However, when I dial in through an 800 number from a provider that I connect to via IAX I get this 'blip' in the sound file. At first I thought it was just packet loss, but it happens at the exact same spot every single time. There are several parts to the IVR menu and several places where this blip happens every single time

Hi, I've noticed on asterisk 1.8.18 I'm hearing the blip of '#' DTMF to end the recording on the recording itself. Is there an easy way to truncate the last 200ms of the recording or so to eliminate this? The DTMF is coming in through rfc2833 and not inband. Thanks. -- James -------------- next part -------------- An HTML attachment was scrubbed... URL:

https://blip.tv/opensuse/episode-5622859 Does anyone have the right tools to extract this video? I refuse to use flash over here, and as it's not on youtube I can't use youtube-dl. Rich. -- Richard Jones, Virtualization Group, Red Hat http://people.redhat.com/~rjones virt-p2v converts physical machines to virtual machines. Boot with a live CD or over the network (PXE) and turn

Hi all, When the Dahdi driver detects DTMF, it seems it's not muting the first 5-15 ms and sometimes the last 2-10 ms of the DTMF tone. This shows up in recorded voicemail greetings -- you hear a very short DTMF '#', or sometimes two blips, at the end of the recording. I have a Mitel SX-200 connected to Asterisk 1.6.0.1 by a couple of Digium cards: a TE420 w/Octasic and pri_net

I have 2 Digium 4 port FXO cards in my system. The system is a P4 2.4Ghz, 512MB RAM, Promise FastTrax 100 TX2 Pro Raid, 80GB RAID1 for storage - whitebox - running RedHat 9. With pretty much any CVS HEAD version we are getting, what I will call, "phantom" calls on some lines. What I mean by a phantom call is that the line will ring, Asterisk will log that the Zap channel has been

Hi, I have a client who has a system with a Sangoma 1 port PRI card with echo canceling in it. For some reason, when the system comes up the PRI will stay up for about 4-5 hours, then drop. "zap show status" shows everything as ok, but we can't make or receive any calls until the system is rebooted. Just restarting asterisk does not fix the problem. I am going to call

http://bugs.freedesktop.org/show_bug.cgi?id=16546 Summary: Blip.tv does not work as expected :) Product: swfdec Version: 0.7.x Platform: x86 (IA32) URL: http://polygamia.pl/prezentacja-life-with-playstation OS/Version: Linux (All) Status: NEW Severity: normal Priority: medium Component:

This may not be the right place to ask, but I have an annoying issue with a Sipura/SPA1000-2.0.10(e) ATA device connected to an Asterisk box. (The system is remote to me, so I've only been able to observe this by dialling into a VoIP phone on-site, then run commands on the box remotely!) First of all it's all working fine connected to an Asterisk box and the user can make/take calls

Hey all, I am trying to figure out a weird problem. I have a two node ctdb samba setup where the first node acts as expected setting the default gateway (startup shutdown takeip recoveryip) if there is a network blip / link down, but the second node does not set the default gateway properly on the initial recovery of the link after the blip has occurred. If I restart the machine or restart the

Nice Ribbit Demo http://blip.tv/file/753401 I think we should get some Asterisk video demo's up on blip.tv as well. Post to this list with the url once you have your demo's up there. Regards, Dean Collins Cognation Pty Ltd dean at cognation.net +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). -------------- next part -------------- An HTML attachment was scrubbed...

On 2/20/2014 6:55 AM, Charles Lepple wrote: > On Feb 19, 2014, at 12:50 PM, Ted Mittelstaedt wrote: > >> Worse, however, is if there's a power failure right near the end of >> the 2-days-off cycle. That happened to me last week - it was a >> short duration 15 second loss - and the upscode2 driver decided it >> needed to issue a forced shutdown. >> >>

Hi, I have a Sangoma 200A (I think that's the model #) analog 4 port card. It works great... however almost everytime after someone hangs up a call they were on.. the system rings the call back in, as though it were a new call coming in. When they pickup no one is there. Can anyone suggest why this is happening, and how I can make it stop?

What codec should be used to connect a * box to a cisco router which has a t1 with 24 trunks coming in? My router voip dial plan looks like this: dial-peer voice 2 voip destination-pattern [1,2,,3,5,8].. session protocol sipv2 session target ipv4:10.x.x.x dtmf-relay cisco-rtp codec g711ulaw no vad ! The problem I have is when more than one call is on it, sometimes the quality gets very

Anyone else around using apcupsd? I seem to be seeing a problem, and I'd like someone to check me on it: I edit /etc/apcupsd/apccontrol to replace the value of SHUTDOWN from /sbin/shutdown to /bin/false (we don't want 3 or 6 servers shutting down over a 2 second or so blip, which it really wants to do). What's happened is that a machine shut down the other day, and looking at it, I

http://bugs.digium.com/bug_view_page.php?bug_id=0001589 Has anyone else heard an audible blip, break or garble between answer and the native bridge attempt using sip? If I change the usleep(500000); to usleep(5000); in rtp.c the proble totally goes away... even the note above it says it needs to be fixed. bkw -------------- next part -------------- An HTML attachment was scrubbed... URL:

All, I've a question on A*k hardware. I'm running 1.2.18 on a Dell DC051 (Intel(R) Celeron(R) CPU 2.80GHz) with 512mb RAM. I'm supporting 60 users (Cisco 7940s each + Xlite PCs). Call loads are low, max of about 10 simultaneous SIP/IAX calls. CPU for A*k rarely goes above 2% as I can tell. Its IP only, no E1/T1 cards present. However, I get complaints of bad voice quality,

Hi folks.. just a few weeks ago I wrote this to someone else: ------------------------ We have several 2900s in production as VoIP servers.. no lockups. On every server I go into the BIOS and: * Disable USB * Disabled uneeded things like Parallel, Serial * Put ETH0 on a seperate IRQ from the Digium card And everything's fine. Dell's do NOT have to share IRQs... go into your BIOS and

Hello list- I'm having an extremely odd issue with an installation of mine. The system is running * 1.2.12.1 and currently handles around 100 handsets. With the exception of a few Grandstream DTA's, all devices are Polycom 320, 430, or 601's. After a recent power outage, I'm having an extremely odd issue with one of the handsets. One of the Polycom 601 units simply reboots every

I am recieving inbound calls to my asterisk box over IAX. And getting "spawn extension....exited non-zero" errors. Im not entirely sure what this means, and would appreciate any help in fixing my problem. FYI: ********** is the inbound phone number x.x.x.x is a remote asterisk box calling my own asterisk box. When I choose it to dial an internal extension using this dialplan: exten

    Some users have complained that their calls drop after about 30 seconds.  Not all, just some.  After looking at the log files the only difference I can find from the dropped calls is the following line: [2020-09-07 11:29:59] VERBOSE[21666][C-00000055] bridge.c: Bridge 14410400-5e04-4358-af0c-45fd71f6f5cd: switching from simple_bridge technology to native_rtp     Most calls just do: