similar to: Configure Asterisk as SIP UA using NAT

Hi, do you have NAT between Asterisk and agent phones? S pozdravem Tomáš Holý Hi Tomas Thanks for replying. Yes, the phones are in one location in a LAN and are then NATed to enable them to contact the Asterisk which is hosted in the cloud. A typical sip.conf phone configuration on the remote server for the site is [general] session-timers=refuse disallow=all allow=g729:20 allow=ulaw

Hello, I have an asterisk box with a peer configured with nat=force_rport,comedia, but asterisk keeps sending the audio to the private IP address and ignoring the client peer nat settings. If I check the "sip show peer extension", I see both symmetric RTP and Force Rport are set to yes, but asterisk seems ignoring them. Force rport : Yes Symmetric RTP: Yes Asterisk is behind a

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 All, I tried to play a little bit with IPv6 to test our VoIP quality software with IPv6 RTP streams. I add "bindaddr=::" to the general section of the sip.conf and netstat shows that Asterisk is listing also on IPv6. My Asterisk server is behind a IPv4 NAT and was working absolutely perfect. But after my bindaddr change I got a problem with external calls. I spend some time to

I have a cloud server... I have a phone in Chicago I have a phone in Indiana. Both are registered to the cloud server - using chan_sip and Asterisk 18.18.0 I can send a pre-recorded message to Chicago it auto answers and hear audio. I can do the same to the phone in indiana. however - when i call from Indiana to Chicago - the phone rings - but I do not get any audio? I have in sip.conf

I'm asking about this scenario: Asterisk(public IP) <--> Internet <--> Router (public IP) <--> SIP client (private IP and NAT) What settings in sip.conf will give this the best fighting chance of working? We already have nat=force_rport,comedia

First I am new to PBX so i might be doing something fundamentally wrong... That being said I got a FreePBX 32bit stable 6.12.65. I am having some issue with the NAT and sound, both phones are ringing but there is sound, I had some talk on IRC: <[TK]D-Fender> Note for elfranne's situation, : nat=force_rport,comedia" should have returned the public IP the call arrived on, but it

What settings have you got for directmedia? Could you try nat=force_rport,comedia directmedia=no -Ashwin -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Luca Bertoncello Sent: 07 June 2015 12:04 To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] Curious problem with NAT

Dear all, I have enabled stun module and configured it in asterisk , but asterisk not using stun returned public ip address for any of the sip requests going out of my network. i have done settings as below res_stun_monitor.conf settings: [general] stunaddr = stun.ideasip.com stunrefresh = 30 stun show status Hostname Port Period Retries Status ExternAddr

Hi, Am making a simple SIP trunk between two Asterisk server, Server 1 sip.conf [usman02] type=peer username=usman02 secret=usman02 host=10.30.2.58 context=man02-trunk port=5060 qualify=yes disallow=all ;allow=g729 allow=g729 ;allow=alaw nat=force_rport,comedia dtmfmode=rfc2833 relaxdtmf=yes insecure=invite,port extensions.conf [man02-trunk] exten => _1X.,1,Dial(SIP/usman02/${EXTEN}) exten

Hi list! Finally I got my wife's phone working in my Asterisk. Unfortunately I have some problems, too... Current situation: - AsteriskNOW with 4 Accounts (00493511111111, 00493512222222, 00493513333333, 5678). This is "for test" and it will be replaced by "the real world", when I got my Asterisk to work... - A second Asterisk (Ubuntu-PBX) on another VM, logging in

I think your phone may be trying to register with the username '1234', while your sip configuration is expecting 'luca'. Can you try changing your phone registration credentials to use 'luca'? Can you give us a sip transcript when you try to place a call from it? On 15-05-28 05:09 PM, Luca Bertoncello wrote: > Darryl Moore <darryl at moores.ca> schrieb: >

On Sun, Jun 7, 2015 at 4:49 AM, Luca Bertoncello <lucabert at lucabert.de> wrote: > 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... :( > >

Hello, I'm trying to get calls working between websocket clients and sip clients. For clients I have sip.js based clients on chrome, Zoipers and a Grandstream phone. Challenge here is I'd like to have Kamailio and rtpengine to handle the bridging between different rtp profiles but Asterisk changes them in the sdp bodies along the way. I'm using Asterisk 11.11.0. Is there a way to

Asterisk Project Security Advisory - AST-2011-013 Product Asterisk Summary Possible remote enumeration of SIP endpoints with differing NAT settings Nature of Advisory Unauthorized data disclosure Susceptibility Remote

Darryl Moore <darryl at moores.ca> schrieb: > Ahh. Seen that before! That suggests to me that you don't have your > sip.conf records setup right. > > What's your sip.conf look like? Well, here what I wrote in my sip.conf: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register =>

Hi All, I have configured WebRTC according to the install document. The clients register correctly. I'm use SIPjs. The clients are able to send messages to the server. The SIP debug shows the messages being received. However I'm stumped for directions on how to route the messages between the clients. Asterisk 11.11.0 Here is my client sip config: [1060] type=friend username=1060 ; The

Ashwin Surendran <Ashwin.Surendran at now-health.com> schrieb: > Have you tried NAT=force_rport ? OK, tried... I can transmit from my phone (aka: I hear my voice on another phone), but I'm not able to receive data (aka: I cannot hear what I say on the other phone). Other suggestion? Thanks Luca Bertoncello (lucabert at lucabert.de)

Hi! I wish you all e Happy New Year first! Allthough, I'm relative new to Asterisk, I got our server up and Running, Softphones, ISDN, and a brand new Snom 821 are working flawlessly. :) Platform is Debian 8/Asterisk Packages (11) from Debian Repo. But I am running into problems setting up 2 older Hardphones, Thomson 2030S. :( with in my sip.conf, I have got for this hardphone: [...]

Hello I have an asterisk 16.2.1 on an ubuntu on AWS, which is experiencing a drop in call. It does not have a certain time, it is random. The audio is flowing normally and the call is dropped. Has anyone ever experienced this? My settings changed below: allowoverlap = no udpbindaddr = 0.0.0.0 tcpenable = no tcpbindaddr = 0.0.0.0 transport = udp, ws, wss srvlookup = yes directmedia = no