hi,Andrew Latham:
Thanks for your response! sorry my poor written english!
OK,I write my testing process......
* first step:*
(1): add the SIP trunk and try to register the remote SIP
softswitch(another asterisk system.the remote SIP port is 5060)
(2): add the dial plan, _1XX to dial the other system's sip phone.
(3): add a sip phone and make it to register the sip system(A)
(4): call 100.the two call can be connted successfully.
please see the attachment asterisk_5060.txt
* Second step:*
(1): Make the remote SIP system port 5080
(2): Modify the SIP trunk port 5080 and "Apply changes". (Yes, it can
register the remote sip sysyem and the remote port is 5080)
(3): call 100, I found the SIP system send "480" error messages
Please see the attachment asterisk_5080.txtJust modified the SIP register port not 5060, the call was failed...... In the asterisk_5080.txt file , I found the error messages, maybe it's the reason! but i do not know the reason..... [May 10 13:37:07] WARNING[14650]: chan_sip.c:3094 create_addr: No such host: trunk_1 [May 10 13:37:07] DEBUG[14650]: chan_sip.c:17346 sip_request_call: Cant create SIP call - target device not registered Thank you! 2011/5/8 Erin Spiceland <espicel...@digium.com> > > I use the GUI on over 60 servers. I am more than willing to help test. > > I have servers with up to 3000 users and use every feature. My > > question is how do I help, is there a place you want feedback? > > "Works for me" messages can go to the list. If you find bugs, please enter > them in the issue tracker at http://issues.asterisk.org. Thanks! > > Erin > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-gui mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-gui >
asterisk_5060_5080.rar
Description: application/rar
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