similar to: How to config firewall for RTP/RTCP?

I've been wondering for a long time why my reinvite option is not working with Broadvoice anymore. It happend during the time Broadvoice was having a lot of issues, so I decided to wait. Recently I decided to test the same sip.conf with another VSP (SIPphone) and it worked fine! No issues on the reinvite. Note: clients and server using ULAW (only), no NAT or firewalls, public ip address and

I have been successfully connected (incoming and outgoing) to FWD for a very long time. A few months ago, I changed from SIP-based FWD service to IAX2-based, and that went fine as well, both incoming and outgoing. At the time, I was running Asterisk 1.0.3 Stable. I rarely use the service, so other than noticing that I was always successfully registered to FWD, I didn't make or receive calls

Asterisk Project Security Advisory - AST-2017-008 Product Asterisk Summary RTP/RTCP information leak Nature of Advisory Unauthorized data disclosure Susceptibility Remote Unauthenticated Sessions Severity Critical

Hi: I have a linksys voip gateway connecting to an asterisk server ,when i dial a call from the linksys gateway to asterisk , i see repeated messages of a RTP errors ,and at same time i hear fake ring on the linksys?, This is wht i see on asterisk console?: ? -- Executing [9613070741 at direct:1] Set("SIP/03070741-088bd470", "CALLERID(number)=96170707070") in new stack ??? --

Hello, Has anyone encountered that when Asterisk sends RTCP messages, it stops sending RTP packets until it gets an answer? Can that be fixed? Thanks.

Hello list ! SETUP : Grandstream --sip--> Local Asterisk (NSLU) --iax--> Hosted Asterisk (VirtualDedicatedServer) --sip--> SIPprovider --> my CellPhone PROBLEM : I've noticed that when I put down the horn of my Grandstream it still takes a while for my GSM/CellPhone to stop ringing. INFORMATION : This is the output on the CLI of the local Asterisk-server : [Oct 3 17:40:33]

> What machine are you running (namely what OS)? Debian. > I dont understand your approach. > Why running a 'streamer' behind a nat? > Not enough 'resources' to rent/ rent to buy a ded. Server? > Mean, can't expect to satisfy a lot of listeners this way. :-) I am listening the Muazkhan indications XD: > > Hi Muaz Khan, >> We are adtlantida.tv and

(hello all, this is my first writing. so please bear with me if I'm wrong anywhere.) orry to break too lately, but how is the RTP payload submission is going? could we see the new payload at March IETF? I agree that it would be fairy straightforward to make an RTP payload for ogg vorbis, assuming raw packets, AFAIK. using physical bitstream is, in this case, not adequate by the reasons in

Asterisk is on public IP (as described in the first email) i have 10 years experience in voip, 4 years webrtc in production. i know about ICE/STUN/DTLS-SRTP. yes, not every detail but the basic mechanism but i confess. i dont understand WHY Asterisk SOMETIMES switches destination IP in RTP. this is not only about ICE. its about RTP engine too which is Asterisk specific and Asterisk DEBUG is

hi, i have following topology PSTN - Asterisk ---- internet -----  router - jssip client (wss) Asterisk 13.29.1 on public IP, chan_pjsip for wss, chan_sip/udp for SIP connection to PSTN router - public IP/private IP (NAT) jssip client - private IP - sip over websocket to Asterisk PJSIP ~30% of calls has problem with no audio. reason is that Asterisk is sending RTP to private IP of jssip

with wireshark i need decrypt traffic every call which is time consuming. get debug from pjnat through asterisk is not possible because of technical reasons or nobody did it? in my case its strange that ice candidates are the same good call v=0 o=- 3669976329745317845 2 IN IP4 127.0.0.1 s=- t=0 0 a=msid-semantic: WMS EoNIdKcMZvWBLULGqGPJTDe12ujjFEemeapo m=audio 52421 RTP/SAVPF 8 0 101 c=IN

http://bugzilla.netfilter.org/show_bug.cgi?id=757 Summary: SIP connection helper not setting RTCP conntrack expectation Product: netfilter/iptables Version: linux-2.6.x Platform: i386 OS/Version: Ubuntu Status: NEW Severity: normal Priority: P5 Component: ip_conntrack

And the firewall is in between the phones and both servers or are you registering the phones on a local server and trunking to the other server through the firewall? In terms of nat and Cisco 7960s I've never had a issue registering two of them behind nat to a server on the other side, however, if you called one phone from the other, you'd end up with one way audio. -----Original

Other SIP clients behind the firewall (not using STUN, work). We have a SIP client using STUN and ICE behind a pfSense firewall. The firewall is behaving oddly. REGISTER packets work fine. But when the client tries to make a call, the first INVITE packet from the client pass through the firewall and makes it to the Asterisk server. The Asterisk server sends back a 401 client sends ACK,

hi! I'm working an a asterisk test project at college at the moment. right now we're experiencing two problems. calling our sipphone (optipoint400) from a firefly client leaves us with no audio (no noise...nothing at all...) [the phone is ringing however and the connection seems to be set up] other way round works just fine!! firefly2firefly (stun enabled) also works

This week, I've been really busy with the launch of a new Swedish Voip provider, www.bbtele.se, so I haven't been able to follow the Asterisk community and haven't been very responsive either. My apologies if you've tried to contact me and I did not reply quickly or at all. So to cover up (can't report on what is happening :-) I dedicate this issue of Asterisk Sunday News to

Hi, When we use BudgeTone-100 in our Intranet together with our Asterisk IP PBX everything is working OK. When we try to use the phone behind the Firewall we can't do the connection. When I try to use STUN Server: 128.107.250.38 there is no result. The only way in which I have audio from the one direction (BT-100 to Asterisk) is when I leave blank STUN Server and specify the IP Address in

I've beaten myself bloody dealing with this one... No luck so far. In summary, incoming calls from Gizmo establish, but neither get nor send sound. Outbound calls to Gizmo work fine (well a bit choppy but work) My thought is that the SIP connection is being made fine, but the RTP is getting stopped / blocked / misdone somewhere. Here is the thing: Asterisk 2.5 on Linux (No hardware

I'm trying to understand IAX and whether or not it would solve my difficulties: 'The primary goals for IAX were to minimize bandwidth used in media transmissions, with particular attention drawn to control and individual voice calls, and to provide native support for NAT (Network Address Translation) transparency. Another goal is to be easy to use behind firewalls.'

First, I assume that you will be running NAT at both locations, if that is not the case, then the configuration will change. When you said VPN, are you using PPTP or IPSEC? Microsoft supports PPTP. In order to connect a PC over VPN to the office, which has a PPTP VPN Server, you will need to runs VPN software. After it is run, the PC obtains a new IP address from your office, If you are using