Author: ITshef
Subject: Next…
Posted: Thu Jan 12, 2012 17:59 (GMT 1)
sip.conf
=============================
[100]
type = friend
host = dynamic
username = 100
secret = 1234
canreinvite = no
nat = yes
context = out1
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
[101]
type = friend
host = dynamic
username = 101
secret = 1234
canreinvite = no
nat = yes
context = out1
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
[102]
type = friend
host = dynamic
username = 102
secret = 1234
canreinvite = no
nat = yes
context = out1
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
[103]
type = friend
host = dynamic
username = 103
secret = 1234
nat = yes
context = out1
canreinvite = no
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
extension.conf
===================
[out1]
exten => _8XXXXXXXXXX,1,Set(fname=${STRFTIME(${EPOCH},,%Y/%m/%d/external/%H-%M-%S)} ${CALLERID(number)}-${EXTEN})
exten => _8XXXXXXXXXX,n,MixMonitor(/home/admin/${CALLERID(number)}/${fname}.wav)
exten => _8XXXXXXXXXX,n,Dial(SIP/7${EXTEN:1}@extsip1)
exten => _8XXXXXXXXXX,n,hangup
exten => _2XXXX,1,Dial(SIP/784463${EXTEN:0}@extsip1)
exten => _2XXXX,n,hangup
exten => _4XXXX,1,Dial(SIP/784463${EXTEN:0}@extsip1)
exten => _5XXXX,1,Dial(SIP/784463${EXTEN:0}@extsip1)
exten => _6XXXX,1,Dial(SIP/784463${EXTEN:0}@extsip1)
;exten => _0.,1,Set(fname=${STRFTIME(${EPOCH},,%Y/%m/%d/external/%H-%M-%S)} ${CALLERID(number)}-${EXTEN})
;exten => _0.,n,MixMonitor(/home/admin/${CALLERID(number)}/${fname}.wav)
;exten => _0.,n,Dial(SIP/${EXTEN:0}@extsip1)
exten => _1XX,1,Set(fname=${STRFTIME(${EPOCH},,%Y/%m/%d/internal/%H-%M-%S)} ${CALLERID(number)}-${EXTEN})
exten => _1XX,2,MixMonitor(/home/admin/${CALLERID(number)}/${fname}.wav)
exten => _1XX,3,Dial(SIP/${EXTEN})
exten => 000,1,gosub(time)
exten => 001,1,gosub(ani)
exten => 002,1,Meetme(1234,rm)
Added after 29 minutes:
— Time to scan old dialplan and merge leftovers back into the new: 0.005357 sec
— Time to restore hints and swap in new dialplan: 0.000002 sec
— Time to delete the old dialplan: 0.000530 sec
— Total time merge_contexts_delete: 0.005889 sec
[Jan 12 20:47:29] NOTICE[5182]: pbx_ael.c:150 pbx_load_module: AEL load process: merged config file name ‘/etc/asterisk/extensions.ael’.
[Jan 12 20:47:29] NOTICE[5182]: pbx_ael.c:153 pbx_load_module: AEL load process: verified config file name ‘/etc/asterisk/extensions.ael’.
— Reloading module ‘app_voicemail.so’ (Comedian Mail (Voicemail System))
— Reloading module ‘app_meetme.so’ (MeetMe conference bridge)
== Parsing ‘/etc/asterisk/meetme.conf’: == Found
== Parsing ‘/etc/asterisk/sla.conf’: == Found
— Reloading module ‘app_playback.so’ (Sound File Playback Application)
— Reloading module ‘codec_g726.so’ (ITU G.726-32kbps G726 Transcoder)
— Reloading module ‘pbx_dundi.so’ (Distributed Universal Number Discovery (DUNDi))
== Parsing ‘/etc/asterisk/dundi.conf’: == Found
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
— Executing [23404@out1:1] Dial("SIP/103-00000042", "SIP/78446323404@extsip1") in new stack
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
— Called 78446323404@extsip1
— SIP/extsip1-00000043 is ringing
— SIP/extsip1-00000043 is making progress passing it to SIP/103-00000042
== Spawn extension (out1, 23404, 1) exited non-zero on ‘SIP/103-00000042′
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
— Executing [100@out1:1] Set("SIP/103-00000044", "fname=2012/01/12/internal/20-48-29 103-100") in new stack
— Executing [100@out1:2] MixMonitor("SIP/103-00000044", "/home/admin/103/2012/01/12/internal/20-48-29 103-100.wav") in new stack
— Executing [100@out1:3] Dial("SIP/103-00000044", "SIP/100") in new stack
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
— Called 100
== Begin MixMonitor Recording SIP/103-00000044
— SIP/100-00000045 is ringing
— SIP/100-00000045 answered SIP/103-00000044
[Jan 12 20:48:56] WARNING[3073]: chan_sip.c:3778 retrans_pkt: Maximum retries exceeded on transmission Y2U1ZGE1MDQ5NWUzY2E4MjI0YmMzNDQ0Nzc1ZGQxNDA. for seqno 2 (Critical Response) — See doc/sip-retransmit.txt.
[Jan 12 20:48:56] WARNING[3073]: chan_sip.c:3805 retrans_pkt: Hanging up call Y2U1ZGE1MDQ5NWUzY2E4MjI0YmMzNDQ0Nzc1ZGQxNDA. – no reply to our critical packet (see doc/sip-retransmit.txt).
== Spawn extension (out1, 100, 3) exited non-zero on ‘SIP/103-00000044′
== MixMonitor close filestream
== End MixMonitor Recording SIP/103-00000044
— Registered SIP ’103′ at 178.35.161.52 port 28062
localhost*CLI>
Это после звонка удаленного клиента. Когда с этого же номера звонить локально, то все норм
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