similar to: directmedia/canreinvite/native bridging question

Hello, I have a question about "directmedia" or "canreinvite", I have experience that whatever I set directmedia=yes or no. After I run sip show settings. all settings looks the same. My question is how I could make sure from "sip show settings" that my "directmedia" configuration is applied. Thanks

On Mon, Dec 15, 2014 at 3:34 AM, Recursive <lists at binarus.de> wrote: > <snip> >> For asterisk 1.6 & 1.8 you would need to set 'canreinvite=no', I don't know what Asterisk 13 will do with this setting. >> > I suspect Asterisk 13 will just ignore it. To make things worse, there seems to be a configuration directive named reinvite (not a typo); I

Hi, I have the following setup: Asterisk <-> Nat <-> Internet <-> Nat <-> 2 x SIP endpoints With directmedia=no I can make a call between the two SIP endpoints; the RTP stream being passed through the Asterisk box. Obviously, this is sub-optimal. I attempted to enable bridging of the call between the 2 endpoints directly, given that they are on the same non-routeable

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

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, when using some non dynamic host eg. host=192.168.111.153 in sip.conf asterisk is not considering specific peer options eg. directmedia=off, transport=tcp if I set host=dynamic and register the sip phone it works as expected. Is this a bug or feature - I wanna disable the usage of directmedia for some peers with fixed ip but wanna allow it in general. Same with transport=tcp. [97]

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

Hi all, i have a little problem to understand this warning message, it's annoying and it cause a lot of spurious in the log files. Im working with this scenario: a sip outgoing trunk (Trunk-out) that support only ulaw and all calls are always routed to this. a list of sip UAs that potentially can use any codec apart g729/g723. I setup the asterisk to do as mediaproxy so directmedia=no and

Hello all, I'm setting up a couple of test boxes and I'm running into a problem. What I need help with is determining whether I'm going something wrong or if I need to post a bug report. I have two asterisk 13.0-beta 3 machines set up with extensions connected to each as such: 3700 ----> AST-A <------> AST-B <---- 3800 & 3801 When I place a call from 3800 to

Hello List, I have some doubt about direct media settings. I have an asterisk 1.8.14 instance running on 172.20.255.50, a soft phone on IP 10.100.210.51 and a gateway at 10.100.210.254 I have set both gateway and peer to "directmedia=yes" but still on gateway I see RTP from asterisk's IP, have tried setting nat=yes/no and also specifying localnet values but not sure where I am

Asterisk 11 CentOS 6.4 Cisco 7971 phones Does chan_skinny support directmedia? Jacob Miles Software Engineer jacob.e.miles at l-3com.com 903.457.4422 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130604/745a7918/attachment-0001.htm> -------------- next part

Background: WAN1 - Fixed IP low latency, low jitter WAN2 - Fixed IP medium latency, higher jitter than I like for good VoIP Firewall/Router not SIP aware NATed LAN Asterisk on server located on LAN. Most, but not all ATA/IP phones on LAN In the past I was running a v1.2 Asterisk which acted as a B2BUA (all RTP streams relayed through Asterisk server) thus presenting only one SIP device to the

Hi All, i am stuck in NAT issue on ec2 cloud computing from last 2-3 days , may be some of you are doing setup and integration on cloud. below is my setup details which may help you to suggest me solution. Asterisk version : 1.6.2.6 1) Kamailio server having public_ip as well local ip .i am using mediaproxy [also tried rtpproxy] . 2) Asterisk server having public_ip as well local ip. setup:

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

Hi, Am Dienstag, den 26.02.2008, 14:01 -0800 schrieb Tanya M. Lattner: > I can move the llvm-gcc4.2 source code up in the list if people think this > is better... but the binaries will still be first and should be. Just make it consistent so people who don't know their way around yet can quickly find what they're looking for. I agree that binaries should be first if they should

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

Hey, we're experiencing a weird problem with Asterisk 1.8.13.1 (1:1.8.13.1~dfsg1-3+deb7). Calls that leave and enter Asterisk via a PBX (sipgate.de) work perfectly fine, almost 100% of the time. However, calls that are routed to sipgate.de, which then routes the call back to our Asterisk instance are "silent" most of the time. What I mean with that is that even though RTP traffic

> I plan to run the test suite, just to establish a known baseline (this > is an amd64 machine, and things tend to be a bit less well-polished than > on stock x86 installations). > Does it make sense to > * first run the test suite with the binaries, > * compile llvm-gcc from sources, > * run the test suite again with the recompiled binaries? What do you plan to use this

Anyone here used these phones? I'm getting more and more frustrated by todays modern crop of routers with their so-called SIP ALGs which are invariably broken, or routers with built-in ATAs which block internal SIP phones from working, so looking to use IAX for some end-users. I already support it for people who want to use (eg) Zoiper and use IAX a lot to plumb boxes together, but never

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