similar to: IPv6 and IPv4 NAT not working

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

Dear list, I'm trying to get Asterisk to work dual-stack on Linux and I'm left with a question. Imagine that a user (on the road) connects to Asterisk from various places. Many of them probably don't have IPv6 support yet. However, his house and office do have IPv6 connectivity. I would like to make sure that whenever IPv6 is available, the connection will be made over IPv6, but

Hi I have installed Asterisk 11.13.1 on Fedora running in VirtualBox. The VB network interface is configured to use NAT. The host machine is Windows 7 and is connected to a SIP server using a VPN connection. I have configured ?externaddr?, ?localnet? and ?nat=force_rport,comedia?. Asterisk registration is successful, I see in Wireshark the packets send between Asterisk and SIP server. However,

Is there anything in the Asterisk logs? Which side sends the BYE? Were you able to capture the traffic with sngrep/wireshark to see if any side stopped sending/getting RTP? What did the other side see? On Mon, Sep 21, 2020 at 3:22 PM Roberto < roberto.medola at gasparimsantos.com.br> wrote: > Hello > I have an asterisk 16.2.1 on an ubuntu on AWS, which is experiencing a > drop in

Roberto Check your router if ALG or similar feature is enabled. Disable and test. Also, on SNGREP check if both parties are getting ACK correctly after RTP starts. *--* *Atenciosamente,* *Luciano Moreira**(85)99974-2750* *__Logic Telecom* *0800-085-7799 | (85)4042-7799 | **(11)4210-7799* Em ter., 22 de set. de 2020 às 13:35, Roberto < roberto.medola at gasparimsantos.com.br>

Hello. Thanks for the reply. Yes. In the traffic analyzed, the BYE is sent by the originator of the call, but there is no "human" hangup, but the asterisk one. BYE is sent, received and confirmed. I don't know how I could investigate the reason for this BYE. Em 21/09/2020 17:12, Dovid Bender escreveu: > Is there anything in the Asterisk logs? Which side sends the BYE? Were

I found this configuration file on Wiki for FWD behind firewall ; SIP Configuration for Asterisk ; [general] disallow=all allow=ulaw port=5060 ; Port to bind to bindaddr=0.0.0.0 ; Address to bind SIP channel to externip=xxx.xxx.xxx.xxx localnet=172.16.1.0 localmask=255.255.255.0 context=inbound-sip ; Default context for incoming calls maxexpirey=180 defaultexpirey=160 tos=reliability

Hello, I am doing some tests with asterisk on a dual-stack environment. I have some doubts regarding asterisk binding addresses on a server with 2 network cards. According to asterisk documentation: /; With the current situation, you can do one of four things:/ /; a) Listen on a specific IPv4 address. Example: bindaddr=192.0.2.1/ /; b) Listen on a specific IPv6 address.

i've got * and phones in small home network all behind NAT. Outbound to iconnect proxy works great. Now to get in/out working with another carrier. Carrier2, Commpartners, i have working with one of the phones and a soft phone without * just fine. Next I register the phone with * fine. Create a trunk, but it the trunk fails to register... help I'm getting the following msg during

I'm running asterisk on a Freebsd with 2 Nic's. Inside NIC is 192.168.5.x where the phones are. Outside NIC used to be a public IP with the ISP's device set to bridging, but the new WiMAX router only offers me the public ip 94.18.x.x on the outside, and forwarding everything to 192.168.1.50 on the "Outside NIC" Some of the phones are being disconnected with Asterisk

Hi people! I am not getting smart getting asterisk 1.6 behind a NAT to run. 1. I enabled IP forwarding on debian linux 2. told asterisk in "general" that he is behind NAT and mentioned him his external static IP Adress as well his domain in the outside world. If a client who is connected with a DSL modem calls me, a grandstream module in the LAN behind the router, in the same network

Hi. I've just bought SIP telephony service from a Swedish telco. I've managed to make and receive calls with kphone. Now I want to set up asterisk to be able to add fancy features like voice mail and recording conversations. But first I have to get the basic setup right. I'm running asterisk and kphone on the same machine, behind at NAT-router. When I make a call (from my regular

Hello. I have an Asterisk server (ViciDialNow) set up behind NAT. I can manage to make outbound calls, but the communication drops off after 30 seconds or so. I'd really appreciate having some assistance from the mailing list on this issue. So, I'm having an Asterisk server behind a firewall and Zoiper softphones on SIP connecting to Asterisk on the same local area network. The

Issue is what subject says. Here is the background. Version: 11.11.0 Topology: Asterisk Box at our Data Center behind Cisco Firewall. Everything works fine from remote offices over a VPN. Issue is sales team would like to connect up to our Asterisk box remotely (offnet). Common enough solution, I'm guessing. So, I've opened all the correct holes on the firewall and hammered out

This has probably been answered somewhere, but I'm stumped. I have two Zap channels (FXS and FXO), both working fine. I can call from Zap/1 to Zap/2 and reverse. I've also configured SIP channels, both inside and outside of my firewall. Inside can call outside, and outside can call inside. Also, both inside and outside can make and receive calls to/from Zap/1 & Zap/2. What

We have just updated our office server to Asterisk 11.4.0 from 1.8.15 and internally everything is working fine. The problem we are having is that we cannot use any external phone connected through the Internet. This used to work fine with 1.8 but since the upgrade whenever you register any phone from an outside network the phone tries to register using the servers internal IP. I endo up

I have a multi-homed machine (quite a few IP addresses on one of the interfaces) For SIP I found that using externaddr in sip.conf would make it much more reliable with ICE and RTP using the correct IP Is there an equivalent setting for XMPP / motif.conf? I saw bindaddr in gtalk.conf but it doesn't appear to be mentioned in the source code for chan_motif

Hi all, I am running an Asterisk server behind a NAT. I want to forward the calls from PSTN to a SIP phone (no nat and also an asterisk). I set the externip and localnet in sip.conf already. I opened the ports in my firewall. (I changed SIP port from 5060 to 5065 and limited the rtp port to 12000-13000) However, I just can't call out. I've always received SIP/2.0 404 Not Found. My

I had been running an older patched CVS to get VOIP working with NAT and everything had been running fine. I just built * on a new box with CVS-01/18/04-12:19:25. And now I can get remote SIP users to register. Has anything major changed... [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to externip = 69.132.68.17 ; Address

I've been googling and talking with Libretel about my setup and the fact that incoming calls to my asterisk box through the Libretel number reach my box (I hear the greeting being played) but then don't accept DTMF. Here is a rough diagram of my setup: Asterisk | server | NAT <------------ Libretel | router | Note that there are NO SIP