similar to: DNS load-balancing & SRV records

Given the following SRV records: _sip._udp.tollfree.sip-happens.com. 38400 IN SRV 10 0 5060 sometimes.sip-happens.com. _sip._udp.tollfree.sip-happens.com. 38400 IN SRV 20 0 5070 ares.sip-happens.com. Why is asterisk (1.4.17) not honouring the priority and not failing over to using other records when a connection fails? For a given call to tollfree.sip-happens.com ares.sip-happens.com was chosen

Hi all, I'm trying to provision my PAP2T's to use a SVR lookup to find the Asterisk server. I'm using a provisioning file that contains an element like: <Proxy_1_> _sip._udp.example.com </Proxy_1_> However, the PAP doesn't seem to be able to find my server with this hostname. The DNS records are in place because my Polycom and Grandstream servers work just fine.

I have added in my zone file; _sip._udp.elmit.com. IN SRV 20 0 5060 vpbx.elmit.com. As I understand it should mean that any sip connection to <anyname>@elmit.com should go to the udp port 5060 at the host vpb.elmit.com. In Asterisk's extensions.conf I have in the context [default] exten => ronald,1,Dial(${PHONE_615},60,tr) exten => ronald,2,Voicemail,u615@office exten =>

Hi, I would like to ask how exactly the "password server =" smb.conf option works. The man pages say that the option is to "restrict Samba to to do all its username/password validation using a specific remote server" I know that we should normally leave it empty, to have samba auto-discover the DCs. But my question is: Suppose it's defined it like: > password

Hello list. When using SIP DNS SRV to define a production Asterisk server with high priority and a backup Asterisk server with a lower priority on this DNS-server, will this work as follow : - production server is reachable, so registration of the IP-phone goes to this server - production server is unreachable, so registration goes to the backup Asterisk server - production server is

Hi, SRV records have been working fine until my hoster decided to upgrade BIND. working wrong syntax: _sip._udp SRV 10 10 5050 mydyndns. correct syntax: _sip._udp IN SRV 10 10 5050 mydyndns. That kicked of one of my domains completly caused by a syntax error that did no harm to the previous version After inserting the missing "IN" the zonefile loaded but now

Hi, i've ported some codecs to Arduino / Teensy. It's an Cortex-M4 with 64KB RAM from Kinetis Now I have discovered this wonderful codec. I would like to port it, but first I need to know what is the RAM usage ?. Is it constant ? I have previously ported FLAC and know that FLAC need more memory than MP3 or AAC (framesizes of 4096 bytes or more). Regards, Frank

Hey gang, I'm trying to setup the ability to dial a SIP user via their email address. I'm using SJPhone as my tester UA, but most clients will be using XTen Pro. I added an SRV DNS entry into our zone, and it returns: ; <<>> DiG 9.2.1 <<>> SRV _sip._udp.cytelcom.com ;; global options: printcmd ;; Got answer: ;; ->>HEADER<<- opcode: QUERY, status:

Having big problems and for months. Our service provider (via:talk) says they are Asterisk friendly but they are not. Here are the specifics (please read the bottom of the msg too) System: Dell SM Business server 2GB RAM, Core II Processor (should be plenty) OS: open SUSE 11 Asterisk Version: 1.4.2 Asterisk GUI Version: 2.0 The system was completely set up using the Asterisk GUI with a

Hi guys, I have this configuration: LAN -- LINUX BRIDGE -- LINUX BRIDGE -- TS The linux's bridges are for wifi purposes (2 centos machines) and is working fine. The problem I have is that the TS are 3 windows TS in load balancing configuration. There is 1 IP for general purposes and every TS has its own IP. For example, the configuration is: 192.168.0.1/24 TS1 192.168.0.2/24 TS2

What's the difference between user "123" and "devries"? Based on the output here, they seem the same..? tleilax*CLI> tleilax*CLI> sip show users Username Secret Accountcode Def.Context ACL Forcerport 201 password 201 default No Yes 123

Hello, I cannot call my server over the internet with SIP anymore. Even when I do a maximum logging on my firewall, I don't see packets coming from outside. I've tried it from an ekiga.net account and an sip2sip.info account. What could be wrong? I would expect incoming traffic on port 5060 UDP... The account is "paul at vandervlis.nl". This should connect trought DNS to the

Can asterisk use SRV record from DNS? In example - I've got: _sip._tcp.gda.pl. SRV 0 0 5060 welbot.task.gda.pl. _sip._udp.gda.pl. SRV 1 0 5060 welbot.task.gda.pl. And it works with sipd (from Columbia) and vocal. I see that with asterisk it does not want to work... :-/ -- pozdr. Pawe? Go?aszewski --------------------------------- worth to see:

Hi, I'm trying to do load balancing between 2 asterisk servers using SIP load balancer, provided by http://www.vovida.org I used the following options on lbproxy, but I get the below message continuously. ./lbProxy -name seneca -reqPort 5060 -respPort 5061 -proxy A1 -proxy A2 "No proxies are up - can not send message to anyone" Xlite is not able to register to the

Does anyone know when Asterisk is supposed to support prioritization of the SRV records returned. I think it's accepted that right now the Asterisk always just uses the first record returned regardless of priority. I also noticed, on register, that it queries for an A record first. If it gets a response, it will use that address to send the REGISTER. If it does not, then it will launch an SRV

Hi We've setup an Asterisk PBX recently and I encountered the following problem: When [mac address]-registration.cfg file includes the FQDN of the Asterisk PBX for the Polycom SoundPoint 501 phones it will not (even try to) register with the Asterisk PBX unless the DNS (it asks) successfully resolves the name: _sip._udp.[Asterisk FQDN]. Did this happen to anyone else? PS - The

Hi, I'm trying to use dovecot as a proxy as described on http://wiki.dovecot.org/PasswordDatabase/ExtraFields/Proxy but without much success. The problem is that when I send a IMAP login command dovecot is responding with "BYE Internal login failure. Refer to server log for more information." Can anyone help me shed light on this problem? Here's the log (Username,

On Tue, Nov 7, 2023 at 11:20 AM Marek Greško <marek.gresko at protonmail.com> wrote: > Hello, > > well I do not ask those who only guess, but those who know what is > asterisk expected to do when internet connectivity goes down. I did not had > a chance to make internet not to work yet, since it is needed. But > inspecting dns logs I found out that there started to be

Hello, well I do not ask those who only guess, but those who know what is asterisk expected to do when internet connectivity goes down. I did not had a chance to make internet not to work yet, since it is needed. But inspecting dns logs I found out that there started to be resolving for _sip._tcp and _sip._udp records for the provider's server. So apparently making hosts record make asterisk

Hello To cut down on the number of hackers trying to break into an Asterisk server, I'd like to simply move the SIP port from the standard UDP 5060 to something non-standard. Since this server must be able to receive INVITEs from any SIP UA (server or client), it appears that I must add an SRV record in the DNS so that they can locate the server and the port used to reach it. _sip._udp SRV