Hi all, I've got something strange, that got me searching for quite awhile. Configuration as followed: Linphone on a laptop, that is connected via openvpn to a proxy. That proxy is connected with iax to another asterisk. On the second one i have several hard and softphones.
Behaviour at first glance: >From the softphone i can allways set up a connection, But the otherway round fails 9 out-of 10 times. However, if i stop-and-start linphone, the connections is allways succesful. First conclusion was, that if i got a diffrent (dynamic) ip-adress from openvpn, i got to restart linphone, to force a re-registration. Sounds reasonable, but why is linphone able to place calls, but not able to accept them? (guests are off) I mean, if the phone is registered with different values, also the outgoing call should fail. Not? To avoid this behaviour, should i drastically drop the registration duration at the softphone side? I still uses the default one (3600s). Or should i tweak the min/max/default expiry-timers at asterisk? Currently they are (also the default) 60/3600/120 seconds. Hans ps these are the lines from the console: -- Executing [00000277611@from_iax:1] noop("IAX2/kc3004-6511", ",00000277611") -- Executing [00000277611@from_iax:2] answer("IAX2/kc3004-6511", "") -- Executing [00000277611@from_iax:3] dial("IAX2/kc3004-6511", "SIP/00000277611 ") [Jun 6 19:03:32] WARNING[23015]: app_dial.c:2039 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [00000277611@from_iax:4] hangup("IAX2/kc3004-6511", "") == Spawn extension (from_iax, 00000277611, 4) exited non-zero on 'IAX2/kc3004-6511' -- Hungup 'IAX2/kc3004-6511' corresponding lines from the ARA-dialplan: | 118 | from_iax | 00000212676 |1 | noop | ${CALLERID},${EXTEN} | | 119 | from_iax | 00000212676 |2 | answer | | | 120 | from_iax | 00000212676 |3 | dial | SIP/00000212676 | | 121 | from_iax | 00000212676 |4 | hangup | | -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users