bilal ghayyad
2011-Apr-17 18:46 UTC
[asterisk-users] Asterisk 1.8.3: Started but no SIP talking
Hi All; I installed Asterisk on a new Server, it is a Dell Server and has 4 Ethernet ports. I gave IP address 192.168.0.3 for one Ethernet port. I am able to login for asterisk using /usr/sbin/asterisk -rvvv and from there (in the command line) I can type a commands. I have an Polycom IP Phone that is able to register for other Asterisk boxes (and some of them is 1.8.3) but with this new server, I do not see any messages coming to the consol when I give the IP address of this new asterisk server !! What could be? Actually, in the sip.conf file, it is hearing for all IPs 0.0.0.0 and the IP phone sending on port 5060 UDP (every thing default). What could be I am missing? Even if username and password wrong, I should be able to see traffic but without registration ... By the way: on the same network, there is another Asterisk box running with IP address 192.168.0.2 .. does it effect? It should not. I am missing any thing? Should I do any thing? How can I know if my new asterisk is running sip well? Regards Bilal
Warren Selby
2011-Apr-17 21:08 UTC
[asterisk-users] Asterisk 1.8.3: Started but no SIP talking
On Sun, Apr 17, 2011 at 1:46 PM, bilal ghayyad <bilmar_gh at yahoo.com> wrote:> Hi All; ><snip>> I am missing any thing? Should I do any thing? How can I know if my new > asterisk is running sip well? > > Regards > Bilal >Check if you've got a software firewall running, check if SELinux is running, etc. Try running a packet capture using tcpdump to see if your asterisk box is getting any traffic from the phone, etc. Basic network troubleshooting at this point. Can you ping the box from your network, can you ping the phone from your box, etc? -- Thanks, --Warren Selby, dCAP http://www.selbytech.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110417/abfc925e/attachment.htm>
Apparently Analagous Threads
- Goto Queue, does not work, it should play message or any thing
- Cisco IP Phones and Skinny in asterisk 1.8.4.2 "tooooooooooooooooo"
- FreePBX: using context other than the default context and the generation for the configuration
- Cisco IP Phones and Skinny in asterisk
- ext-local and from-did-direct-ivr, how to change them?