I get some really weird behavior on my system. This is my extensions.conf for
the incoming VoIp provider which comes into the "bypass"
When I call my home number (2812080888) from my cell, the phone rings in my
house but my cell goes straight to the voicemail (101). I have to remove the
Voicemail (101) so it can work. See the errors bellow. I don;t know if this is
Astlinux 1.1.1 related but it started to happen after the upgrade. Anyone can
see what's going on?
[bypass]
exten =>
2812080888,1,GotoIf(${DB_EXISTS(blacklist/${CALLERID(num)})}?blacklisted,s,1)
exten => 2812080888,2,DIAL(DAHDI/3)
;exten => 2812080888,3,Voicemail(101)
exten => 2812080888,102,Hangup()
HOME-PBX*CLI>
HOME-PBX*CLI>
== Using SIP RTP CoS mark 5
== Using SIP RTP CoS mark 5
-- Executing [2812080888@chila:1] GotoIf("SIP/vitel-inbound-00000008",
"0?blacklisted,2812080888,1") in new stack
-- Executing [2812080888@chila:2] Playback("SIP/vitel-inbound-00000008",
"wait-moment") in new stack
-- Executing [2812080888@chila:1] GotoIf("SIP/vitel-inbound-00000009",
"0?blacklisted,2812080888,1") in new stack
-- Executing [2812080888@chila:2] Playback("SIP/vitel-inbound-00000009",
"wait-moment") in new stack
-- <SIP/vitel-inbound-00000008> Playing 'wait-moment.ulaw' (language 'en')
-- <SIP/vitel-inbound-00000009> Playing 'wait-moment.ulaw' (language 'en')
-- Executing [2812080888@chila:3] WaitExten("SIP/vitel-inbound-00000008",
"1") in new stack
-- Executing [2812080888@chila:3] WaitExten("SIP/vitel-inbound-00000009",
"1") in new stack
-- Timeout on SIP/vitel-inbound-00000008, continuing...
-- Executing [2812080888@chila:4] Goto("SIP/vitel-inbound-00000008",
"bypass,2812080888,1") in new stack
-- Goto (bypass,2812080888,1)
-- Executing [2812080888@bypass:1] GotoIf("SIP/vitel-inbound-00000008",
"0?blacklisted,s,1") in new stack
-- Executing [2812080888@bypass:2] Dial("SIP/vitel-inbound-00000008",
"DAHDI/3") in new stack
-- Called DAHDI/3
-- DAHDI/3-1 is ringing
-- Timeout on SIP/vitel-inbound-00000009, continuing...
-- Executing [2812080888@chila:4] Goto("SIP/vitel-inbound-00000009",
"bypass,2812080888,1") in new stack
-- Goto (bypass,2812080888,1)
-- Executing [2812080888@bypass:1] GotoIf("SIP/vitel-inbound-00000009",
"0?blacklisted,s,1") in new stack
-- Executing [2812080888@bypass:2] Dial("SIP/vitel-inbound-00000009",
"DAHDI/3") in new stack
[May 10 22:08:10] WARNING[564][C-0000000b]: app_dial.c:2437 dial_exec_full:
Unable to create channel of type 'DAHDI' (cause 17 - User busy)
== Everyone is busy/congested at this time (1:1/0/0)
-- Auto fallthrough, channel 'SIP/vitel-inbound-00000009' status is 'BUSY'
-- DAHDI/3-1 is ringing
[May 10 22:08:14] WARNING[497]: chan_sip.c:4169 retrans_pkt: Retransmission
timeout reached on transmission 1225fc9e4d26ce2b258e71c5541cae06@66.241.96.221
for seqno 102 (Critical Response) -- See
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[May 10 22:08:14] WARNING[497]: chan_sip.c:4198 retrans_pkt: Hanging up call
1225fc9e4d26ce2b258e71c5541cae06@66.241.96.221 - no reply to our critical
packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
-- DAHDI/3-1 is ringing
-- DAHDI/3-1 is ringing
-- DAHDI/3-1 is ringing
-- DAHDI/3-1 answered SIP/vitel-inbound-00000008
-- Hanging up on 'DAHDI/3-1'
-- Hungup 'DAHDI/3-1'
== Spawn extension (bypass, 2812080888, 2) exited non-zero on
'SIP/vitel-inbound-00000008'
Regards
Ionel Chila------------------------------------------------------------------------------
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