similar to: iax encryption

Hi folks, Ok, I've seen this question go unanswered on the mailing list, and I assume it's because no one had the heart to break the bad news to the guy asking, but be honest with me, I can take it. At this time it's flat impossible to have multiple IAX phones behind a NAT without using an * gateway because there's no way to have a client listen on a port besides 4569. Is

Hi All, I was just reading through Info Week while on a flight and happened upon an brief piece about a new VOIP security intiative worked up by a handful of the usual suspects; Alcatel, SMU, NIST, Symantec, etc. All of this begs the question of can't we get just do this as a user community? I understand that the Zultys phone, which I own several, support AES encryption of the RTP stream.

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

We have a customer who is connected to our PSTN gateway using IAX and noticing that even when the traffic from their site is modest their outbound audio has short dropouts. Inbound audio is fine. (They have ADSL so it is expected that outbound audio would be the first to experience problems.) We have several questions to pose to the collective wisdom of this list. Q1: Are there any statistics

Thanks for your quick replies - I had no idea this list was so responsive! One more issue for you - we are trying to build a custom voicemail application and so far everything works great by using AGI to play prompts, get input and record files. However one of our customers has asked for some pretty advanced functionality, specifically the ability to: * stop in the middle of recording (so far so

Hi I'm browsing around the internet looking for signs that the IAX client library and app_meetme support video. I stumbled across this post by SteveK on the 27th of Feb 2006. "My company is looking to hire a full-time developer, who will be working about 25-50% of the time on iaxclient; in particular to finally integrate, build, polish and enhance video in iaxclient, add video

I know this has come up before but... Has anyone done anything to get an IAX client built on Windows? I thought someone had started one, but I haven't heard anything about it since - and that was months ago? Anyone have any idea what the status is? -- Ron Gage - Saginaw, Michigan I am looking for work - resume at http://www.rongage.org/resume.doc Electrical Engineering, Linux Programming,

Asterisk-people, Some of you may have heard that we were working on a simple, cross-platform IAX client library called "iaxclient". We've pretty much been "on vacation" with the project for a while, but recently have made some progress, and now have the library working across platforms, and a simple test client called "testcall", up and running on 3

I'm using Linphone. I tested with Asterisk and Speex only, I created a channel with echo and it worked. It seems to have problem when using app_conference. Jonathan 2006/1/31, Steve Kann <stevek@stevek.com>: > > jonathan blais wrote: > > > Hi, > > > > Does anyone ever used Speex with app_conference in Asterisk ? I'm > > having a hard time to figure

Hi, I'm trying to use iaxcomm. I can place a call from the softphone, but when I place a call to it, when I answer I get ... NOTICE[16401]: File channel.c, Line 1094 (ast_read): Dropping incompatible voice frame on IAX2[paulohm]/3 of format GSM since our native format has changed to ALAW My iax.conf looks like this .. [paulohm] type=friend host=dynamic username=... secret=...

Hi, Didn't get any opinions on the log file I mailed onto the list over the weekend so I am continuing to try and track the cause for the dropped calls.. I have a feeling that its to do with IAX being way too sensitive when it comes to packet loss.. Since it is going across the internet it needs to be more tolerant when it comes to errors and packet loss.. Are there any settings (in the

Just curious, how does Asterisk pack Speex frames in a packet. AFAIK, Linphone just sends raw packets, as specified in the RTP draft. Jean-Marc Le mardi 31 janvier 2006 ? 10:43 -0500, Steve Kann a ?crit : > jonathan blais wrote: > > I'm using Linphone. I tested with Asterisk and Speex only, I created > > a channel with echo and it worked. It seems to have problem when >

We have many IAXy devices in the field now. In all cases, in iax.conf, we have "qualify=yes", so that using "iax2 show peers", we can see whether or not the device is currently online. In some cases, the IAXy device and/or Asterisk are not communicating their qualification, because "iax2 show peers" shows the device as status UNKNOWN. However, when a user picks up

Hi, Does anyone ever used Speex with app_conference in Asterisk ? I'm having a hard time to figure why I always get this error "warning: Invalid mode encountered: corrupted stream?". Jonathan Blais -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20060131/386141a8/attachment.htm

We've changed E1 providers and I'm trying to reconfigure an E400P to make it work with the new lines. They're supposedly "standard" EuroISDN lines (in the UK). I'm initially just trying to get a single line up. I have the following in /etc/zaptel.conf: span=1,0,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 loadzone=uk defaultzone=uk The LED on the back

Jeff, It's in the manual: http://www.speex.org/docs/manual/speex-manual/node10.html (table 3 and 4). However, if you're asking this, you're probably trying to do something wrong, or the hard way. You probably shouldn't be taking speex output, and trying to "count bytes". If you are using the API, then you will just get the bits out, and then you'll know how

Hello, from what I've understood Iax2 should support aes128 encryption. I've found this old info: http://www.voip-info.org/wiki/view/IAX+encryption and this (unanswered?) post http://lists.digium.com/pipermail/asterisk-security/2005-August/000060.h tml Is this the libiax used currently on asterisk http://ftp.digium.com/pub/libiax/ ? I would like to understand if someone is using this in

On 06/22/2011 09:30 AM, Steve Kann wrote: > Speaking of AEC (thought not quite on topic for this thread), > > Has anyone on this list played with the GIPS code that google just > open-sourced? It looks like their AEC also has code to handle > differential sample rates, though I haven't really evaluated it > thoroughly. > > There is really a lot of code in the drop ?

>> Has anyone benchmarked Asterisk on a dedicated single versus dual >> processor machine? > > http://www.astertest.com/ > > Cheers, Philipp The test results that Philipp pointed out show some protocol comparisons that include "iax2 trunking / alaw" and "iax2 / alaw" and concludes that "IAX2 trunking is more than twice as fast as non trunking

On 5/22/23 07:13, Greg Troxel wrote: > gene heskett <gheskett at shentel.net> writes: > >> Which file in the nut I pulled a month ago, should I start with to get >> the new version actually running? I still get - -V=2.7.4 > > It is likely that you have built new nut but old nut is still on your > system. I do a 'sudo make install' to write over the