i m having a prob with asterisk...
my sip.conf port# 5060 ...clients (on lan) are xpro sip clients
when sip port#5060 clients connect fine and bidirectional communication is perfect
but when i try to bind asterisk from 5060 to 5061(or any other port than 5060) (in sip.conf)
xpro clients cant register into asterisk server..
all my clients and asterisk server are on real ips
can i change bindport from 5060 to 5061 and still let my xpro clients register to my asterisk server?(or is it not possible?)
i m using asterisk-1.2.11 (latest ver till now)
ok i found the solution.Instead of Xpro client i used X-lite 3 sip phone (not even xlite 2 works) .U can put in
domain=<asterisk server's ip>:port#
domain=x.x.x.x:5070 (if u r binding asterisk on sip port#5070)
it gets registered fine and everything works fine
I wonder y XPRO cant register if asterisk is being binded to any different port than 5060.
Even though my asterisk server and my sip phone both are on public ip addresses , i still hav one way audio prob (which is mostly seen in NAT situations).sip phone#2 connecting from another ISP can register in a flash to my asterisk server (having another WAN public ip) But after registering i can hear tht sip phone's voice but other party cant hear my voice .
My sip phone#1 (184.108.40.206)--Asterisk----Wan cloud---Sip phone #2(220.127.116.11)
i use IAX protocol instead of SIP which is not NAT compliant than SIP is. My asterisk server is working perfectly IAX-->PSTN and PSTN---> IAX is ok now. Seems like i m talking to myself here ,,,lol
I setup Asterisk successfully to get the call from voip and bridge it to Zaptel interface (Digium card analog or digital) so PC-->phone and Phone-->PC communication is working like a charm...
Thin Setting up asterisk requires configuring 3 or 4 configuration files under /etc/ and /etc/asterisk directory .
1) /etc/zaptel.conf (to define the zaptel channels if u have zaptel interface like digium pci cards or intel md3200 series modem .
2)/etc/asterisk/zapata.conf ( to set parameters for channels)
3) extensions.conf ( to configure dialplan for call routing )
3) sip.conf( defining sip clients if u r using sip )
4) iax.conf ( defining iax clients )
If u need further assistance do let me know.
Enjoy exploring this amazing freePBX solution tht turns ur PC into feature rich softswitch