similar to: Getting No Antenna bar when behind a NAT

how do i troubleshoot no ring tone. It was working and all i added was the lines below now it doesn't ring. Edit sip_nat.conf for proper NAT: localnet=192.168.1.0/255.255.255.0 externhost=pbx.DOMAIN.com (Set your external hostname name here) externrefresh=10 fromdomain=DOMAIN.com (Set your external domain name here) nat=yes qualify=yes canreinvite=no Add extra codecs to

It's my first post here, so I'll cut to the chase I have 2 Asterisk servers and want to connect them using sip on one and pjsip on the other one. One is running at home and another at a VPS. The first one will be the client (with dynamic ip) and the 2nd the server. The client uses sip and the server pjsip. This is the client's sip.conf [general] context = default allowguest = no

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

I'm trying to install a program from a CD-ROM that is included with a publication called the ARRL Antenna Book, 19th Edition which claims compatibility with Windows up to 2000. Running the setup program I get the following (I am executing it from the CD-ROM): $ wine ABSETUP.EXE Disabling HW TCL support Disabling HW TCL support Disabling HW TCL support Disabling HW TCL support wine: Unhandled

Hello; Is there any modules or channels or integration between asterisk and any of the following: whatsapp, facebook, viber, yahoo and hotmail messanger? Regards Bilal

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

Hi, I am try to configure Asterisk as PBX system with two interfaces as shown below. One interface pointing to the local subnet with a SIP phone and another interface pointing to the external ISP SIP Sever. SJPhone(X.X.141.32)<--------->(Y.Y.47.149)local-intf-|Asterisk|external- intf(Z.Z.247.106)<-------->(w.w.158.26)ISP-SIP-Server----OutsideWorld I am able to setup a call from the

Is there anything special that anyone here has had to do to get an Aastra phone (on the Internet) to talk to Asterisk behind a PIX firewall? Ports 10000-20000 UDP are open on the PIX and forwarding to the Asterisk server. The Asterisk server's RTP.CONF is set to use 10000-20000. The phone registers, and will place AND receive calls, however, no audio is passed. The phone is an Aastra

Dear all, On a new server (postfix dovecot postfixadmin Centos) I did define quota=0 in postfixadmin However suddenly a user with more than 9Gb of mail got his mailbox new/cur empty and maillog shows: Sep 3 15:43:56 mail16 dovecot: lda(brouwerb at scholarium.nl): Error: sieve: msgid=<alpine.LRH.2.20.1509031543050.16381 at streaming2.antenna.nl>: failed to store into mailbox

Hi list! Since the internal calls work as expected and I can register my Asterisk on an external provider, I'd like to add a new feature and allow my mobile phone to connect to my Asterisk and manage calls. Well, first of all, my Asterisk is NOT direct on Internet available, but behind a NAT. So I configured my sip.conf: localnet=192.168.200.0/24 externhost=myhost.noip.com externrefresh=180

Hi all, I'm getting one way audio when calling over the SIP trunk i.e. end device B (remote end of SIP trunk) can hear device A (softphone registered with Asterisk) but device A can't hear device B. Even though I configured same NAT configurations on other servers and they are working good. The NAT configuration is listed below; localnet=130.0.0.0/130.0.0.0 externhost=12.131.12.13

On 9/22/20 2:14 PM, Gregory Sloop via samba wrote > As an old sage (curmudgeon if you like) I'd encourage people to really consider if the Pi is really what you want. > For me, it's not - even though it's a totally cool device conceptually. They're a ton of fun to tinker with too. I am also an old guy and didn't want to use the Rpi for this purpose. However, many years

Warren, one slight correction on an other wise nicely written bit of info: The time transmitted from WWV is not Mountain Time. Even though the WWV transmitter farm is located in the Mountain time zone, the signals are transmitted as "Coordinated Universal time", UTC, or 'Zulu' time. Here, you can listen to a recording made at the transmitter site for the 5Mhz signal:

On Wed, May 24, 2017 10:45 am, Warren Young wrote: > On May 24, 2017, at 8:52 AM, Chris Adams <linux at cmadams.net> wrote: >> >> Once upon a time, Warren Young <warren at etr-usa.com> said: >>> a. It???s transmitting from a fixed location in a time zone you >>> probably aren???t in ??? US Mountain ??? being the least populous of >>> the lower

externhost is monitoring for ip changes with an interval of externrefresh, so far so good. Wouldnt it be handy if asterisk would do an sip reregister if it detects an ip change? My SIP provider has sometimes very high intervals of 1 hour and if ip changes, the registration doesnt work until it expires or asterisk is restarted or sip reload. Or just everyone uses fixed ip addresses? For now,

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

On Tuesday 10 January 2017 08:53:17 John R Pierce wrote: > On 1/9/2017 7:11 PM, fred roller wrote: > > On Mon, Jan 9, 2017 at 12:04 PM, Frank Cox<theatre at melvilletheatre.com> > > > > wrote: > >> That sounds like a weak signal from your wifi transmitter. > > > > Or signal interference. Where is the antennae located on the server? > > Ran

Good Day All I recently implemnetd asterisk outside our LAN (external network).It works well in a NAT settings. But on external network with PROXY setting ASTERISK DID NOT WORK. My question are 1 Can ASTERISK work in a PROXY setting . 2 If it can work how can i implement it . Expecting your reply Thanks Paul --------------------------------- Yahoo! Messenger

I've got a dilema. For months, I've been planning and implementing a large project involving vorbis, ices2, icecast2 and wifi networking. The plan is to stream live audio from 10 Austin nightclubs during the SXSW music festival this March. Well yesterday, it was made plain to me that I must also provide mp3 streams, since "it's an mp3 world." I'm upset about this

Dear list: I have the typical one way audio problem, as far as i know it's a nating problem, my hosts inside my lan can call to outside internet hosts, but can't listen a thing, i read a lot about sip and rtp and protocols and the problem it seems to be with NAT, this is the config i put on my sip.conf file about nat: externhost=sip.server.com.ar > my server name on the