I am confused about a problem with pjsua. I built a sip server with Brekeke ondo sip server on my computer .Its IP is 218.9.121.185 .I can register to server successfully on my computer. Below is my way to register >>> +a Your SIP URL: (empty to cancel): sip:ljm at 218.9.121.185 URL of the registrar: (empty to cancel): sip:2189.121.185 Auth Realm: (empty to cancel): * Auth Username: (empty to cancel): ljm Auth Password: (empty to cancel): 123 Below is register information on the server: User Contact URL Detail ljm sip:ljm at 218.9.121.185:6000 Expires : 300 Priority : 1000 Accept Pattern : Requester : 218.9.121.185:6000 Time Update : Wed Mar 18 11:58:12 CST 2009 Below is another one who registers to server: >>> +a Your SIP URL: (empty to cancel): sip:zj at 218.9.124.236 URL of the registrar: (empty to cancel): sip:218.9.121.185 Auth Realm: (empty to cancel): * Auth Username: (empty to cancel): zj Auth Password: (empty to cancel): 123 He also can register to server successfully,below is his register information on the server: zj sip:zj at 218.9.124.236:6000 Expires : 300 Priority : 1000 Accept Pattern : Requester : 218.9.124.236:6000 Time Update : Wed Mar 18 12:40:06 CST 2009 He can call me successfully,and I can hear ringing,below is his way to call: >>> m (You currently have 0 calls) Buddy list: -none- Choices: 0 For current dialog. -1 All 0 buddies in buddy list [1 - 0] Select from buddy list URL An URL <Enter> Empty input (or 'q') to cancel Make call: sip:ljm at 218.9.121.185 Then I will get message and I will hear ringing: You have 1 active call Current call id=0 to <sip:zj at 218.9.124.236> [INCOMING] >>> 12:42:10.984 ec00AF6BE8 Underflow, buf_cnt=0, will generate 1 frame But I can't call him,he can't hear ringing: 12:49:42.187 pjsua_app.c Call 2 is DISCONNECTED [reason=408 (Request Timeou )] 12:49:42.203 pjsua_app.c [DISCONNCTD] To: sip:zj at 218.9.124.236 Call time: 00h:00m:00s, 1st res in 32719 ms, conn in 0ms SRTP status: Not active Crypto-suite: (null) I found a problem according to testing that if I register on my computer (it also is running sip server ),everyone who registered can call me successfully ,but I can't call back. A and B(no one is on my computer) can register to the server,but they can't call each other. I don't know the reason. what's wrong with it? thank you ! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20090318/dded74b6/attachment.html>
