similar to: Native bridging to SIP endpoints on the same NAT'd network

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

Hi, I have an Asterisk box behind a NAT address and also a Xlite 4 soft phone behind a different NAT network. Asterisk -> Nat -> Internet -> Nat -> Softphone. I can register my softphone to the asterisk box ok via SIP but the RTP stream from the asterisk box is addressed to the private non-routeable address of the softphone when I turn on rtp debuging. How can I configure the rtp

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 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:

We have a system where calls are coming in from telcos via an opensips server and then being redirected out to a regular sip destination. There is no NAT, DTMF features, call recording, or codec translation being performed so I would expect asterisk to issue a reinvite after the call is answered and switch the audio however it is not happening. Here is the sip peer information for the call

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

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, 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

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

how if I use the auto generate once from freepbx ? On Tue, Sep 22, 2015 at 10:12 PM, Ishfaq Malik <ish at pack-net.co.uk> wrote: > > > On 22 September 2015 at 16:04, Thyda ENG <engthyda at gmail.com> wrote: > >> I have many endpoints and each endpoint has some parameter in common so i >> wonder is there any way to config one for all endpoints? Like in my

Hello all: I am trying to bridge two ethernet segments via tinc. Both endpoints are debian stable machines. Unfortunately the package available in stable is 1.0.3, which according to: http://brouwer.uvt.nl/pipermail/tinc/2006-January/001497.html does not work using the switch or hub modes. I looked at installing the unstable package, but it has dependencies that are not available in

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

Here is my setup: * is on a NAT'd subnet, but also has an externally routable IP address. I have a Snom360 that's external to this and behind NAT. The Snom360 can call other phones in * subnet (by their internal extension numbers) and voice is transmitted fine; however, when I attempt to check voicemail (or any * voice recordings for that matter) I can't hear them. The phone just

Hey all, quick question: I have a server running Samba 2.2.3a which has two NAT'd private domains behind it (192.168.1.0/24 and 192.168.88.0/24). The .88.0 subdomain has a wireless AP/router on it (192.168.88.254) which also does NAT (the wireless machines are on 192.168.92.0/24). The router has the IP of the second internal NIC as its gateway; the wireless machines have the

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

I would like to clarify the email I sent yesterday. There are two ethernet segments in two different cities that I would like to operate as one logical network. Both physical lans have a switch/hub, a gateway with one external IP address that NATs traffic and can port forward tinc ports to the internal debian stable machine (where tinc is run), various client computers ('c' in the

Greetings- I recall mention of this bug at some tim elast year, but do not know if anything yet has been done to address it or if it is in any plans... I have multiple servers behind a f/w. I have ports forwarded on the f/w which map to port 22 on the various servers. i.e. ssh -p1001 FIREWALL_IP ---> NON_ROUTEABLE:22 This is fine, but the hostkey negotiation obviously fails... Is

For anyone who is interested, I have a working tftpd (modified wvtftpd) capable of serving configuration, dialplans, and ringtones to Cisco 7960/7940 and ATA-186 devices that are located behind NAT firewalls. As TFTP is not a very firewall/NAT friendly protocol, I had to break some rules to get it to work with these cisco devices. It might cause problems for other TFTP clients, but it works with