После обновления Ubuntu c 12.04 до 14.04 соответсвенно обновился Asterisk с 1.8 до 11.7 немогу дозвонииться до трубки Gigaset висящей за NAT'ом.
Но в пирах она есть как доступная:
webdc-sup/webdc-sup 92.63.108.130 D N A 5060 OK (159 ms)
Ловлю такую ошибку:
== Using SIP RTP CoS mark 5
-- Executing [402@office-internal:2] Gosub("SIP/stas-00000015", "personal,s,1(SIP/webdc-sup)") in new stack
== Begin MixMonitor Recording SIP/stas-00000015
-- Executing [s@personal:1] Dial("SIP/stas-00000015", "SIP/webdc-sup,,HtTr") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/webdc-sup
[Jan 21 15:40:39] WARNING[29489][C-0000000c]: translate.c:343 framein: no samples for alawtolin
[Jan 21 15:40:47] WARNING[25251]: chan_sip.c:4175 retrans_pkt: Retransmission timeout reached on transmission 46fd0a0b36cea7ab737bd6571c20c879@ispsystem.net for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/ ... nsmissions
Packet timed out after 8320ms with no response
[Jan 21 15:40:47] WARNING[25251]: chan_sip.c:4204 retrans_pkt: Hanging up call 46fd0a0b36cea7ab737bd6571c20c879@ispsystem.net - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/ ... nsmissions).
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@personal:2] Hangup("SIP/stas-00000015", "") in new stack
== Spawn extension (personal, s, 2) exited non-zero on 'SIP/stas-00000015'
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/stas-00000015
Гуглил, попробовал всякие способы, ничего не помогло.
[webdc-sup]
deny=0.0.0.0/0.0.0.0
permit=92.63.108.0/255.255.255.0
type=friend
host=dynamic
context=office-internal
defaultuser=webdc-sup
secret=X8nWrAnDx
callerid=402 Webdc
canreinvite=no
qualify=30000
call-limit=10
;nat=yes
directmedia=no
nat=force_rport,comedia
allow=all