similar to: externhost and reregister

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

My Asterisk server is behind a NAT and I have set: ---------------------------------------------------------------------------- externhost="my.server.address" externrefresh=180 localnet=192.168.0.0/255.255.0.0 localnet=10.0.0.0/255.0.0.0 localnet=172.16.0.0/12 nat=yes --------------------------------------------------------------------------- in [general] section of sip.conf. I can

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

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, 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

Have you tried NAT=force_rport ? 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 11:44 To: Asterisk Users Subject: [asterisk-users] Curious problem with NAT Hi list! Since the internal calls work as expected and I can register my Asterisk on an external

hi, making may way through all this...internal sip registration works,(cant call yet but anyhow)... the asterisk box is obvisoulsy behind a router. im not 100% sure if i should go with port forwarding or NAT and if a or b, what additional setup is actually correct? sip_nat.conf # this is when i got the NAT -route, right? #gets all the dyndns-stuff #externip = home.mydomain.com (Enter your

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>

I am having trouble setting this system up and wonder if some one help me. Does anyone know what is missing if anything to get 2 phones on my asterisk home server to be able to call each other. I have a WRT54GS running OpenWRT/asterisk connected to a PAP2 with 2 extensions 5060/5061, this is on the lan side of my gateway/router WRT54G 192.168.1.1 BusyBox v1.00 (2006.11.07-01:40+0000)

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

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

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

Hi again, list! I know, I'm really annoying the list... :) Well, maybe I got my Asterisk at home ("wrt" on the previous E-Mails) to accept my mobile phone from Internet. It was a problem with the network and the firewall. Now I can log my mobile phone in my Asterisk in and the phone is REACHABLE. Wow! Got it! If I call a phone at home using my cellphone it works and the

Hi! The script/generator scaffold for models doesnt respect (source in) this config/environment.rb setup, right? Because I can put in there ActiveRecord::Base.pluralize_table_names = false but when I run a command like ruby script/generate scaffold model Something then it throws an error saying that my database scheme doesnt have a Something. However there is a table called something. If I

Please forgive any stupidity in the following question. The admin who maintained samba left recently and we (a bunch of programmers) are trying to fix it. We had a Solaris box called svr2, it provided samba and nfs services. We bought a new computer called originally newsvr2. We ported the samba settings across to newsvr2 - everything worked. We swapped the hostname and IP addresses of svr2 and

stimpak wrote: > exactly the file hierarchy i was thinking about! > > put the core files somewhere reachable for all users,but instead of making syslinks, wouldnt be best or possible to create separate *.reg for each account, so you wont risk a registry corruption? > > still that script doesnt make WINE multiuser (as in many users as the same time - so dont try that if you're

Hello all, I have installed the .deb packages of the Asterisk v1.8.3.3 from the upstream project on my Debian GNU/Linux Squeeze server and bought the Comodo's PossitiveSSL SSL certificate to be used for my SIP/TLS exercise. After setting up everything and trying to fix this problem, I am still getting a 401 Unauthorized SIP message. So as of this writing, I still cannot successfully REGISTER

if CONFIG[''load_rails''] ActiveRecord::Base.allow_concurrency = true ActiveRecord::Base.establish_connection(YAML.load(ERB.new(IO.read ("#{RAILS_ROOT}/#{CONFIG[''database_yml'']}")).result)[CONFIG[''environment'']]) end if i read the code right, this option doesnt really load rails. wouldnt it be better (for those who rely on the