I am trying to do a stress test on the asterisk server. I have created multiple accounts that call each other.
For example
1001 calls -----> 1002
1003 calls -----> 1004
1005 calls -----> 1006
Somehow when the server gets to 64 active calls it keeps saying that there is no active port. And I can't make any more calls.
The error log;
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: 07:40:28.584 pjsua_call.c !Making call with acc #1 to sip:[email protected]:25060
07:40:28.584 pjsua_aud.c .Set sound device: capture=-99, playback=-99
07:40:28.584 pjsua_aud.c ..Setting null sound device..
07:40:28.584 pjsua_app.c ...Turning sound device ON
07:40:28.584 pjsua_aud.c ...Opening null sound device..
07:40:28.584 pjsua_media.c .Call 0: initializing media..
07:40:28.584 pjsua_media.c ..Unable to find appropriate RTP/RTCP ports combination
07:40:28.584 pjsua_media.c ..Unable to create RTP/RTCP socket: Address already in use [status=120098]
07:40:28.584 pjsua_media.c ..Error creating media transport: Address already in use
07:40:28.584 pjsua_call.c .Error initializing media channel: Address already in use [status=120098]
07:40:28.584 pjsua_media.c .Call 0: deinitializing media..
>>> >>>>
Account list:
[ 0] <sip:172.31.31.91:30404>: does not register
Online status: Online
*[ 1] sip:[email protected]: 200/OK (expires=219)
Online status: Online
Buddy list:
-none-
Anybody a clue why this keeps happening. I have also created a script that registers 5000 users, all with different ports and this works perfectly without any problems. Would it be possible that there is somewhere a limit function?
Likly you have limited number of ports allowed in rtp.conf file
Please stop spam same questions, asterisk for sure can handle thousands of calls, check your setup(embeded?)/config.