При попытке позвонить на FXO линию ловлю следующий отлуп.
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: log
-- Executing [2774141@phones:1] Dial("SIP/103-00000008", "SIP/2346739/2774141") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/2346739/2774141
-- SIP/2346739-00000009 answered SIP/103-00000008
-- Remotely bridging SIP/103-00000008 and SIP/2346739-00000009
[Feb 5 23:45:02] WARNING[7060]: chan_sip.c:20457 handle_response_invite: just did sched_add waitid(1288) for sip_reinvite_retry for dialog 3550750721@192_168_9_80 in handle_response_invite
[Feb 5 23:45:09] WARNING[7060]: chan_sip.c:3656 retrans_pkt: Retransmission timeout reached on transmission 3550750721@192_168_9_80 for seqno 103 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/ ... nsmissions
Packet timed out after 6400ms with no response
[Feb 5 23:45:09] WARNING[7060]: chan_sip.c:3685 retrans_pkt: Hanging up call 3550750721@192_168_9_80 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/ ... nsmissions).
== Spawn extension (phones, 2774141, 1) exited non-zero on 'SIP/103-00000008'
== Using SIP RTP CoS mark 5
-- Called SIP/2346739/2774141
-- SIP/2346739-00000009 answered SIP/103-00000008
-- Remotely bridging SIP/103-00000008 and SIP/2346739-00000009
[Feb 5 23:45:02] WARNING[7060]: chan_sip.c:20457 handle_response_invite: just did sched_add waitid(1288) for sip_reinvite_retry for dialog 3550750721@192_168_9_80 in handle_response_invite
[Feb 5 23:45:09] WARNING[7060]: chan_sip.c:3656 retrans_pkt: Retransmission timeout reached on transmission 3550750721@192_168_9_80 for seqno 103 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/ ... nsmissions
Packet timed out after 6400ms with no response
[Feb 5 23:45:09] WARNING[7060]: chan_sip.c:3685 retrans_pkt: Hanging up call 3550750721@192_168_9_80 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/ ... nsmissions).
== Spawn extension (phones, 2774141, 1) exited non-zero on 'SIP/103-00000008'
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: sip.conf
[general]
context=default
allowguest=no
udpbindaddr=0.0.0.0
tcpenable=no
transport=udp
srvlookup=yes
alwaysauthreject = yes
auth_options_requests = yes
qualify=yes
monitor=yes
[101]
type=friend
context=phones
secret=pass
host=dynamic
[103]
type=friend
context=phones
secret=pass
host=dynamic
context=default
allowguest=no
udpbindaddr=0.0.0.0
tcpenable=no
transport=udp
srvlookup=yes
alwaysauthreject = yes
auth_options_requests = yes
qualify=yes
monitor=yes
[101]
type=friend
context=phones
secret=pass
host=dynamic
[103]
type=friend
context=phones
secret=pass
host=dynamic
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: extensions.conf
[general]
autokill=yes
[default]
[phones]
exten => _XXXXXXX,1,Dial(SIP/2346739/${EXTEN})
exten => _XXXXXXX,n,Hangup()
include => internal-sip
include => zadarma-out
include => sipnet
[incoming]
exten => 500,1,Dial(SIP/101&SIP/103,,t)
exten => 500,n,Hangup()
[internal-sip]
exten => _1XX,1,Dial(SIP/${EXTEN},,t)
exten => _1XX,,n,Hangup()
autokill=yes
[default]
[phones]
exten => _XXXXXXX,1,Dial(SIP/2346739/${EXTEN})
exten => _XXXXXXX,n,Hangup()
include => internal-sip
include => zadarma-out
include => sipnet
[incoming]
exten => 500,1,Dial(SIP/101&SIP/103,,t)
exten => 500,n,Hangup()
[internal-sip]
exten => _1XX,1,Dial(SIP/${EXTEN},,t)
exten => _1XX,,n,Hangup()
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: Лог из "родной" подсети
-- Executing [2774141@phones:1] Dial("SIP/101-000000f7", "SIP/2346739/2774141") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/2346739/2774141
-- SIP/2346739-000000f8 answered SIP/101-000000f7
-- Remotely bridging SIP/101-000000f7 and SIP/2346739-000000f8
== Spawn extension (phones, 2774141, 1) exited non-zero on 'SIP/101-000000f7'
== Using SIP RTP CoS mark 5
-- Called SIP/2346739/2774141
-- SIP/2346739-000000f8 answered SIP/101-000000f7
-- Remotely bridging SIP/101-000000f7 and SIP/2346739-000000f8
== Spawn extension (phones, 2774141, 1) exited non-zero on 'SIP/101-000000f7'
На внутренние SIP номера звонит, на zadarma и sipnet тоже, а вот на FXO - нифига. Хотя и на sipnet он странно звонит вначале штук 5-6 коротких гудков, а уж потом соединение, у 470 - нормальные длинные гудки.
Прошивка на C610 - последняя 42.076 (420760000000 / V42.00). Пробовал его сбрасывать на заводские настройки и заново сконфигурировать - не помогло.
В роли FXO выступает Dlink 7111s.
Есть ли смысл копнуть еще куда, или тащить аппарат на обмен?