similar to: sip.conf host!=dynamic peer specific options (e.g. directmedia=off, transport=tcp) not working!?

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

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

I've got several SIP clients with dynamic IP addresses Asterisk has one public and one private IP address SIP clients might connect to Asterisk from either the internet or the private network (192.168.1.255) - they're portable By default, directmedia/canreinvite is enabled and Asterisk sets up direct media connections between clients. In this case clients on the internet can make calls

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

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

On 11.9, trying to set up a callcentric peer: sip debug: > <--- SIP read from UDP:204.11.192.161:5060 ---> > INVITE sip:1777<myccid>@10.10.11.180:5060 SIP/2.0 > v: SIP/2.0/UDP 204.11.192.161:5060;branch=z9hG4bK-6104e46aaaaef4249814d16a2ffb990d > f: <sip:<calling number>@66.193.176.35>;tag=3606475083-968127 > t:

Ashwin Surendran <Ashwin.Surendran at now-health.com> schrieb: > What settings have you got for directmedia? > > Could you try > > nat=force_rport,comedia > directmedia=no Tried. Peer always unreachable, call not possible... :( Other idea? Thanks Luca Bertoncello (lucabert at lucabert.de)

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

What I have is: * Asterisk 1.8.10.1~dfsg-1ubuntu1, * SPA112 ATA with analog fax in 1-st FXS port connected, * SIP trunk with provider supporting T.38. My network looks like this: * spa112 (192.168.33.200/24) and Asterisk (192.168.5.253/24) in neighbouring LANs, * Asterisk connects to the provider (80.75.130.136) via router (82.200.7.184). Router has full DNAT to Asterisk server. What happens?

I wasn't able to get much out of babytel, beyond the fact that I was, apparently, sending options which is why I'm not getting 200 OK. How can I, generally speaking, ping/telnet or otherwise test the connection to get more data? A connection to this peer directly from a softphone, Jitsi, works fine. linux-k7qk*CLI> linux-k7qk*CLI> sip show peer testcarrier * Name :

On Sat, 15 Aug 2015 16:30:39 +0800 Michael Dupree <michael at easybitllc.com> wrote: > Not 100% ure, but maybe play with the canreinvite or directmedia > settings. Yes! That was it. Just for future searches here is what I did. I added "directmedia = no" in sip.conf. This fixed the issue. I believe that Asterisk was getting confused when one leg was inside NAT and the

Gentlemen, I have 3 faxes attached to an Asterisk. Fax - SPA2102 - Asterisk. 0851711201 and 0851711290 is on our WAN, no NAT. 0197673581 is outside our WAN and needs to be NAT'ed. Sending a fax from 0851711201 to 0851711290, no problem, switches to T38 and fax goes through. Sending a from 0197673581 to 0851711201, no problem as long as i dont enable T38 on 0197673581. But, if i enable T38

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

On 2/23/19 2:39 PM, Social Boh wrote: > *DIrect media with SRTP is not supported. All media when SRTP goes > through Asterisk.* > > So you have to open ports on your firewall and disable directmedia=yes > on your configuration. directmedia is not explicitly enabled; I guess it's the default. Joshua basically says there is no way to control which ports are being used for

I do not want set allowguest=yes. The problem is, there is no official list with ip addresses of Telekom Germany. But I think all ip addresses comes from the ip range 217.0.0.0/13. I have now the following addition to sip.conf. I think it is the only safe option. Or what would you say? [telekom](!) context=from-trunk type=peer defaultuser= authuser= remotesecret= fromdomain=tel.t-online.de

On 8/9/16 12:40 PM, Matt Fredrickson wrote: > On Mon, Aug 8, 2016 at 9:25 AM, Tammy Firefly <tammy-lists at wiztech.biz> wrote: >> Hi All, >> >> We have asterisk 11.23 running sip to vitelity and from there IAX trunks >> split off to where they need to go. We are having a problem getting >> chan_sip to quit ignoring re-invites from Vitelity. Our side ends

Hi, I am getting *Can't provide secure audio requested in SDP offer* with sipml5 client hosted on my local system [1060] ; This will be WebRTC client type=friend username=1060 ; The Auth user for SIP.js host=dynamic ; Allows any host to register secret=sameer ; The SIP Password for SIP.js encryption=yes ; Tell Asterisk to use encryption for this peer avpf=yes ; Tell Asterisk to use AVPF

Hi All, We have asterisk 11.23 running sip to vitelity and from there IAX trunks split off to where they need to go. We are having a problem getting chan_sip to quit ignoring re-invites from Vitelity. Our side ends up sending a reinvite which their side & they do not support us sending a reinvite. Ive tried: canreinvite=no which was supposedly replaced by: directmedia=no Can anyone shed

How to send a test message directly to lmtp, to test proxying?