| Summary: | ASTERISK-12863: Asterisk SIP calls stop working having more than 300 calls (more than 600 channels) | ||
| Reporter: | Juan Rodriguez (jerdguez) | Labels: | |
| Date Opened: | 2008-10-09 23:16:51 | Date Closed: | 2008-10-10 07:12:33 |
| Priority: | Major | Regression? | No |
| Status: | Closed/Complete | Components: | Channels/chan_sip/General |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | After getting some ERRORS like this: [Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[2485] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[2489] rtp.c: No RTP ports remaining. Can't setup media stream for this call. I start getting: ERROR[14844] chan_sip.c: Unable to build sip pvt data for 'TRUNK/DESTINATION' (Out of memory or socket error) [Oct 9 22:26:45] ERROR[14832] chan_sip.c: Unable to build sip pvt data for 'TRUNK/DESTINATION' (Out of memory or socket error). I had installed Asterisk-1.4.21, but this version stop from receiving calls after these errors occured. Then I downgrade to version 1.4.19 (because I had have tested that version), but after getting these error it stop from creating the outbound call. ****** ADDITIONAL INFORMATION ****** The configuration of the * is an incomming call from the my SIP Provider and after internal manage it makes a second call to other destination--DID--. For AGI compatibility issues I could not use Version 1.4.22 (issues whith DeadAGI for billing purpuses). | ||
| Comments: | By: Russell Bryant (russell) 2008-10-10 07:12:30 This is a configuration error. Please use the asterisk-users mailing list or #asterisk on IRC to get help. | ||