| Summary: | ASTERISK-12138: app_dial should set DIALSTATUS to CONGESTION if unable to open TCP connection to SIP peer? | ||
| Reporter: | Paul Belanger (pabelanger) | Labels: | |
| Date Opened: | 2008-06-04 09:05:26 | Date Closed: | 2011-06-07 14:07:55 |
| Priority: | Minor | Regression? | No |
| Status: | Closed/Complete | Components: | Applications/app_dial |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | Since asterisk is able to detect the TCP port of a SIP peer is not available (see below), perhaps we could set the DIALSTATUS to CONGESTION? Or am I out in left field? ;) Thanks again, PB ****** ADDITIONAL INFORMATION ****** *CLI> -- Starting simple switch on 'Zap/25-1' [Jun 4 09:50:42] NOTICE[30508]: chan_zap.c:7093 ss_thread: Got event 18 (Ring Begin)... [Jun 4 09:50:44] NOTICE[30508]: chan_zap.c:7093 ss_thread: Got event 2 (Ring/Answered)... -- Executing [s@from-zap:1] Dial("Zap/25-1", "SIP/sv0071iv-xxx_IVR") in new stack == Using SIP RTP CoS mark 5 [Jun 4 09:50:44] ERROR[30508]: tcptls.c:231 ast_tcptls_client_start: Unable to connect SIP socket to 10.0.3.71:5070: Connection refused -- Called sv0071iv-Veridian_IVR -- Starting simple switch on 'Zap/26-1' [Jun 4 09:51:04] ERROR[30508]: tcptls.c:231 ast_tcptls_client_start: Unable to connect SIP socket to 10.0.3.71:5070: Connection refused == Spawn extension (from-zap, s, 1) exited non-zero on 'Zap/25-1' -- Hungup 'Zap/25-1' | ||
| Comments: | By: Paul Belanger (pabelanger) 2008-06-04 09:08:21 Please ignore, delete, demote me! For got to add qualify=yes to my SIP.conf / I suck! By: Russell Bryant (russell) 2008-06-04 10:32:59 No problem. :) | ||