similar to: Recommendations about infrastructure to use with Asterisk

Hello, all. In our multi-tenant environment, we would like to be able to use the reinvite media redirection within Asterisk for calls within a tenant but not between tenants. We would like inter-tenant calls to be fully proxied by the Asterisk server. I think the answer is, "we can't," but I thought I'd ask anyway. I'd dearly like to remove the substantial traffic

Hello, all. I've stumbled across what seems to be a traffic prioritization issue in a Linux environment and wonder if anyone else has encountered or addressed this issue. We had planned to use expedited forwarding for our RTP and perhaps our SIP packets. Our plan was to set DSCP to 101110 (by the way, I think document http://www.voip-info.org/wiki/view/snom+360 is in error as I'm almost

Hi! REDIRECT lan 3328 tcp www - !192.168.0.0/29,10.0.0.0/16 This rules redirect all traffic web to proxy but howto exclude 1 ip from redirect ?? REDIRECT lan 53 tcp domain - REDIRECT lan 53 udp domain - And in this case howto exclude some ip ? Thx.

Dear all, I have a very strange problem : * external calls work perfectly, * internal calls between some phones too, * but internal call between two similar phones don't work !!! (Snom 710) When we have sound, there are no errors in asterisk. When we do not have sound, there is the following error : * [Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp: No SRTP module

Hi, I'm hoping someone has a suggestion for us. We have an ITSP that sends inbound traffic to us. Unannounced to us last week they started alternately sending traffic from two IP addresses, instead of the one we knew about. Some calls would pass, and others would be dumped as unauthenticated. I added the 2nd IP to the sip.conf file to allow for this, and everything was fine

Hi There, Due to shortage computer, I need to install Apache to my Shorewall box (192.168.1.1) But the real web server is on another box (192.168.1.2) I tried to put rule: DNAT net loc:192.168.168.1 tcp 80 But everytime www connection coming in, it will hit my shorewall Any solution? Cheer Access Yahoo!7 Mail on your mobile. Anytime. Anywhere. Show me how:

Snom default configuration is SRTP enabled. You should disable the SRTP from the phone web GUI configuration Sincerely, Sam Basan From: Mitul Limbani [mailto:mitul at enterux.in] Sent: Thursday, November 12, 2015 5:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] No sound with internal

I'm trying to move the asterisk server to an Amazon Web instance. We have teliax for our sip provider. I'd like for our DID lines to be connected to a users cell phone. Seems simple enough, but I'm getting the dreaded one-way audio, even with nat=yes everyplace I can think of. The dialplan is real easy: [from-teliax-sip] exten => _j.,1,NoOp("From teliax sip with exten

I have an Asterisk 1.6.2 server on a public IP, Cisco 7940 on the localnet, and a trunk to Sipphone/Gizmo/Google Voice. The externhost and localnet parameters are all set correctly in sip.conf. An inbound call from Sipphone works great until the local channel places the call on hold. During hold, the Sipphone user cannot hear music, only silence. The silence continues after the hold, though

I want to use fprobe-ulog (http://fprobe.sourceforge.net/) to generate NetFlow information about traffic going through my router. The question is how to get the logging rules added to the appropriate chains (I''m assuming eth2_in and eth2_out in my case)? I''m using the perl version of shorewall 4.0.6. -- Orion Poplawski Technical Manager 303-415-9701

Hi people! I am not getting smart getting asterisk 1.6 behind a NAT to run. 1. I enabled IP forwarding on debian linux 2. told asterisk in "general" that he is behind NAT and mentioned him his external static IP Adress as well his domain in the outside world. If a client who is connected with a DSL modem calls me, a grandstream module in the LAN behind the router, in the same network

Hello, all. I'm having a deeply frustrating time getting multiple parking lots to work and am wondering what I am doing wrong. I am using Asterisk 1.6.1.1. I defined two separate parking lots in features.conf as follows: [parkinglot_a100] ; SSI context => a100-parking parkpos => 900-920 findslot => next [parkinglot_a10] ; EBC context => a10-parking parkpos => 800-820

Hello, at first, thanks for helping! In the meantime, I have done a lot of research and trial and error, and I could solve that specific problem. Obviously, the dialplan application "Answer" was playing a key role here. My original dialplan snippet (which produced that problem) was: exten => _00., 1, NoOp() same => n, Set(FAXOPT(gateway)=yes) same => n,

Hi, with canreinvite=no and directmedia=no I and getting the message in the logs for all calls "switching from simple_bridge technology to native_rtp" -- Executing [102 at mkg:1] Dial("SIP/101-00000017", "SIP/102") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/102 -- SIP/102-00000018 is ringing -- SIP/102-00000018 answered SIP/101-00000017

Shorewall 4.3.7 is available for testing. ---------------------------------------------------------------------------- P R O B L E M S C O R R E C T E D I N 4 . 3 . 7 ---------------------------------------------------------------------------- 1) Klemens Rutz reported a problem that affects all Shorewall-perl 4.2 and 4.3 versions. The problem: a) Only occurs when

Yiping, Could you describe in a little more detail what your goals are? I agree with Reid and Misha that modifying the instruction definition is usually not advisable but to suggest alternatives, we would need to know more. Also, for some projects it could make sense to change the instruction set. --Vikram http://www.cs.uiuc.edu/~vadve http://llvm.cs.uiuc.edu/ On Oct 20, 2004, at 2:41 PM,

Good day all, I was wondering if I could pick some admin heads here as I have a HUGE project I have been tasked with..... I am asking here since I will be basing everything on Centos, and want it to all play nice together. If anyone feels this is straying off topic, please just reply off list. I do not want to be the cause of one of those threads. I have 3 offices, 1 in Canada, 2 in

On Wed, Oct 20, 2004 at 11:59:45AM -0700, Yiping Fan wrote: > Yeah. We need to have more extra fields in the instruction. Fo > example, during high-level synthesis, we must schedule an instruction > to a certain control step (or cycle), and bind it to be execute on a > certain functional unit, etc. Since we're talking about "execution" and "scheduling", you

Hi John, Thank you for the comments. I am addressing some of them bellow. Regards, Samuel 2014-08-11 9:36 GMT-04:00 John Leidel (jleidel) <jleidel at micron.com>: > Sergey [et.al], thanks for putting this proposal together. Overall, this > looks like a pretty solid approach to providing relatively hardware > agnostic omp target functionality. I had several comments/questions

OK all. I need some help. Looking to deploy asterisk servers and want to get a recommendation on what server to buy. I love Dell's, but from what I see on the list they seem to have some issues. I would like to stay with one brand and need systems for small offices (20 users), medium (50 users) and large (100 users) systems. Thanks for the help. Keith