| Summary: | ASTERISK-23048: Simple chan_pjsip sip to sip call, call dies about 30 seconds in | ||
| Reporter: | Rusty Newton (rnewton) | Labels: | |
| Date Opened: | 2013-12-20 10:45:05.000-0600 | Date Closed: | 2013-12-20 10:51:27.000-0600 |
| Priority: | Critical | Regression? | |
| Status: | Closed/Complete | Components: | |
| Versions: | SVN | Frequency of Occurrence | |
| Related Issues: | |||
| Environment: | Attachments: | ( 0) calldies.pcap ( 1) full.txt ( 2) messages.txt | |
| Description: | Asterisk SVN-branch-12-r404375M
Running patch from https://reviewboard.asterisk.org/r/3043/diff/6/, otherwise no modifications Dialplan: {noformat} exten => _6XXX,1,Dial(PJSIP/${EXTEN},15) {noformat} | ||
| Comments: | By: Rusty Newton (rnewton) 2013-12-20 10:51:27.033-0600 Not a bug. I still had some NAT related settings enabled while in a different non-natted environment. | ||