similar to: Odd blip when playinv IVR over IAX

Strange issue.... when I record a file from a phone to the asterisk system I get a blip in the recording every 30 seconds. It's a very small blip, but it is there. It seems like it's only if I'm recording, not when I'm playing back that the issue happens. My SATA drives, ETH0, and my Sangoma card are all on seperate IRQs. Any thoughts on what might be causing this and how to

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

Does anyone know of an IAX softphone in Java, whether applet or application? Even the most minimum featureset, just voice and dialing, or even embedded in some other app/let. Preferably GPL. Thanks. -- (C) Matthew Rubenstein

I am playing around with IAX encryption and have had good success. I read somewhere, that trunked packets are not encrypted. Does anybody know if this means the trunk packets themselves are not encrypted but the voice frames in them are encrypted or does this mean that if you are using trunking then encryption of the voice frames will not occur. I have used Wireshark to sniff the packets and it

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:

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

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:

As we have every day, we had a power blip overnight. one, at least, of the servers connected to a SmartUPS via cable, announced that "power exhausted, initiating shutdown" (which I've disabled). The thing is, I know the servers on that UPS draw a ridiculous amount of power, but I don't see that on the others... and this was three seconds, not minutes, after it announced there

Hi. I was wondering if there's a parameter to limit the number of concurrent sessions in IAX, globally or on a per-user basis. That could be needed for security purposes (to prevent dos attacks), to limit bandwidth / cpu usage, or to not allow more than N guest connections, for example. Any other VoIP channel support that? (like SIP, MGCP) Matteo. -- Brancaleoni Matteo

I'm attempting to get two Asterisk 0.4.0 PBXes to communicate with one another using IAX/IAX2 trunks. I've managed to get a semi-functional NAT Firewall working as a PBX (with Asterisk running directly on the firewall itself), but there are issues with bind()ing to various interfaces which is causing outbound SIP issues. To get around these issues, the idea is to do something like

Hello all, The wiki has a fairly detailed description of the the issues involved with encryption of Asterisk calls: http://www.voip-info.org/wiki/view/Asterisk+encryption I'm interested in hearing what is working for people today. I think the ideal solution would be a hard phone that could be plugged in almost anywhere (dsl/cable modem, hotel, etc) and connect securely to a remote

Hi, We are using MozIAX as a softphone with a USB headset and are making outbound calls using IAX with ulaw encoding to our voip provider. We're running asterisk 1.4 Users are complaining that the ring tone generated by asterisk is much louder than the voice call once connected. They are having to turn the volume down to avoid being deafened by the ring tone, but then have an unacceptably

Hi, Just a quick question. Is there a way when making an IAX call to transmit some additional call-data, perhaps in a variable? I could overload callerid-name, but that is nasty and ugly :) Thanks for any suggestions. Regards, Steve

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...

Hi all, All of a sudden all my IAX DIDs have gone down. I couldn't find any reason other than that the ISP is blocking port 4569. DIDs register fine from my home server, but not from office server, which is not behind any NAT. SIP registers fine. I am trying to change IAX port but it apparantly IAX works only on 4569. Changing it in iax.conf doesn't do anything. Changing it is

HEllo list ! When user A calls user B via Asterisk (Users A and B are registered on the same Asterisk server ) and an ISDN PRI, user B phone always shows Asterisk server telephone number. How to hide it and how to forward user A number ? We tried usecallerid, callerid, hidecallerid, restrictcid, usecallingpres in zapata.conf but we always see Asterisk server telephone number ! Thanks

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. >> >>

Hello, Hope everyone is enjoying their holiday! We setup two asterisk servers (From CVS on Wednesday) and set up IAX between the two. Right now they both reside on a switch with a static 192.168.0.x IP address. The first Server is .5 and the second is .30. Our dialplan seems to be working, however on the console we get a flurry of NOTICE and WARNING messages. NOTICE[1116941120]: File

I constantly get the following error messages in /var/log/asterisk/messages: Dec 8 10:52:57 WARNING[1009521664]: File chan_iax.c, Line 3324 (iax_ack_registry): Received unsolicited registry ack from '192.168.0.1' Dec 8 10:52:57 WARNING[1009521664]: File chan_iax.c, Line 4181 (socket_read): Registration failure Where 192.168.0.1 is another asterisk server. Below are the local and

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