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Доступен только один канал

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Доступен только один канал

Добавлено: 04 июн 2013, 16:09
sparrow
Добрый день!
Стоял asterisk10, digium te122p, всё было ок. После переезда в новый офис и смены провайдера стал работать только один канал, то есть в один момент времени атска обрабатывает только один входящий или один исходящий звонок (на профессиональном жаргоне высказаться не могу, к сожалению) :). При попытке дозвониться со второго аппарата идет отбой all circuits are busy, хотя подключен многоканальный поток. Позвонили провайдеру - нас попросили отключить питание, что-то протестировали и сказали что у них всё ок.
На asterisk я не специализируюсь, но вот ключевые конфиги и лог.

chan_dahdi.conf

Код: Выделить всё

[trunkgroups]
[channels]
relaxdtmf=yes
echotraining=yes
echocancelwhenbridged=yes
group=0,11
context=mycontext
switchtype=euroisdn
signalling=pri_cpe
echocancel = yes
channel => 1-15,17-31
pridialplan=national
resetinterval=never
dtmfmode = rfc2833
toneduration = 300

busydetect=yes
busycount=3
usecallerid=yes
callwaiting=yes
usecallingpres=yes
threewaycalling=yes
canpark=yes
cancallforward=yes
rxgain=2
txgain=2

;#include dahdi-channels.conf
;#include chan_dahdi_additional.con
dahdi-channels.conf

Код: Выделить всё

; Autogenerated by /usr/sbin/dahdi_genconf on Thu Oct 18 03:45:18 2012
; If you edit this file and execute /usr/sbin/dahdi_genconf again,
; your manual changes will be LOST.
; Dahdi Channels Configurations (chan_dahdi.conf)
;
; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended
; to be #include-d by /etc/chan_dahdi.conf that will include the global settings
;

; Span 1: WCT1/0 "Wildcard TE122 Card 0" (MASTER)

switchtype = euroisdn
context = from-pstn
signalling = pri_cpe
group = 0,11
channel => 1-15
channel => 17-31
context = default
group = 63
sip.conf

Код: Выделить всё

[general]
relaxdtmf = yes
tos_sip=cs3
tos_audio=ef
limitonpeers = yes
context = NewAsterisk
allowguest = no
alwaysauthreject = yes
bindport = 5060
binaddr = 0.0.0.0
tos_sip = cs3
tos_audio = ef
dtmfmode = rfc2833
disallow = all
allow = alaw
;allow = gsm
allow = ulaw
srvlookup = yes
language = ru
compactheaders = yes
canreinvite = no
sendrpid = yes
nat = never
subscribecontext = default

;allow = g729
;allow = wav
qualifyfreq = 60
;rpid_update = yes
;regextenonqualify = yes
rtptimeout = 120
rtpholdtimeout = 300
;rtpkeepalive = 30
;========================

[NewAsteriskTemplate](!)
type = friend
context = NewAsterisk
host = dynamic
qualify = yes
permit = 192.168.0.0/24
call-limit = 5

[101](NewAsteriskTemplate)
secret = 101
username = user1

и так далее...
включаем sip set debug peer 303, когда атска занята входящим звонком и пробуем осуществить исходящий звонок:

Код: Выделить всё

<--- SIP read from UDP:192.168.20.113:17954 --->


<------------->
[Jun  4 15:17:42] NOTICE[1964]: chan_sip.c:25841 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 203

<--- SIP read from UDP:192.168.20.113:17954 --->
SUBSCRIBE sip:303@192.168.20.137 SIP/2.0
Via: SIP/2.0/UDP 192.168.20.113:17954;branch=z9hG4bK-d8754z-042dab68db62aa14-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:303@192.168.20.113:17954>
To: "303"<sip:303@192.168.20.137>
From: "303"<sip:303@192.168.20.137>;tag=8a335616
Call-ID: Y2FjOGNjMTI4MjEwZTRhMWI5ODUxYzhlYTRmYzkxODI.
CSeq: 1 SUBSCRIBE
Expires: 300
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: eyeBeam release 1102u stamp 52345
Event: message-summary
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Creating new subscription
Sending to 192.168.20.113:17954 (NAT)
list_route: hop: <sip:303@192.168.20.113:17954>
Found peer '303' for '303' from 192.168.20.113:17954

<--- Transmitting (NAT) to 192.168.20.113:17954 --->
SIP/2.0 401 Unauthorized
v: SIP/2.0/UDP 192.168.20.113:17954;branch=z9hG4bK-d8754z-042dab68db62aa14-1---d8754z-;received=192.168.20.113;rport=17954
f: "303"<sip:303@192.168.20.137>;tag=8a335616
t: "303"<sip:303@192.168.20.137>;tag=as778073d8
i: Y2FjOGNjMTI4MjEwZTRhMWI5ODUxYzhlYTRmYzkxODI.
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX 10.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
k: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0ff3b8e3"
l: 0


<------------>
Scheduling destruction of SIP dialog 'Y2FjOGNjMTI4MjEwZTRhMWI5ODUxYzhlYTRmYzkxODI.' in 6848 ms (Method: SUBSCRIBE)

<--- SIP read from UDP:192.168.20.113:17954 --->
SUBSCRIBE sip:303@192.168.20.137 SIP/2.0
Via: SIP/2.0/UDP 192.168.20.113:17954;branch=z9hG4bK-d8754z-c962755d9f34ea4b-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:303@192.168.20.113:17954>
To: "303"<sip:303@192.168.20.137>
From: "303"<sip:303@192.168.20.137>;tag=8a335616
Call-ID: Y2FjOGNjMTI4MjEwZTRhMWI5ODUxYzhlYTRmYzkxODI.
CSeq: 2 SUBSCRIBE
Expires: 300
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: eyeBeam release 1102u stamp 52345
Authorization: Digest username="303",realm="asterisk",nonce="0ff3b8e3",uri="sip:303@192.168.20.137",response="29c9d689505231e3f96e432c2eeb3acf",algorithm=MD5
Event: message-summary
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Creating new subscription
Sending to 192.168.20.113:17954 (NAT)
Found peer '303' for '303' from 192.168.20.113:17954

<--- Transmitting (NAT) to 192.168.20.113:17954 --->
SIP/2.0 404 Not found (no mailbox)
v: SIP/2.0/UDP 192.168.20.113:17954;branch=z9hG4bK-d8754z-c962755d9f34ea4b-1---d8754z-;received=192.168.20.113;rport=17954
f: "303"<sip:303@192.168.20.137>;tag=8a335616
t: "303"<sip:303@192.168.20.137>;tag=as778073d8
i: Y2FjOGNjMTI4MjEwZTRhMWI5ODUxYzhlYTRmYzkxODI.
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX 10.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
k: replaces, timer
l: 0


<------------>
[Jun  4 15:17:42] NOTICE[1964]: chan_sip.c:25841 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 303
Really destroying SIP dialog 'Y2FjOGNjMTI4MjEwZTRhMWI5ODUxYzhlYTRmYzkxODI.' Method: SUBSCRIBE
    -- Remote UNIX connection
 Reloading SIP
    -- Remote UNIX connection disconnected
Reliably Transmitting (NAT) to 192.168.20.113:17954:
OPTIONS sip:303@192.168.20.113:17954;rinstance=c0f45aab2bd111ae SIP/2.0
v: SIP/2.0/UDP 192.168.20.137:5060;branch=z9hG4bK2cfbc1c5;rport
Max-Forwards: 70
f: "asterisk" <sip:asterisk@192.168.20.137>;tag=as1854ec78
t: <sip:303@192.168.20.113:17954;rinstance=c0f45aab2bd111ae>
m: <sip:asterisk@192.168.20.137:5060>
i: 066f19117c9ed4132490a32c65d5d60c@192.168.20.137:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.9.0
Date: Tue, 04 Jun 2013 11:18:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
k: replaces, timer
l: 0


---

<--- SIP read from UDP:192.168.20.113:17954 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.20.137:5060;branch=z9hG4bK2cfbc1c5;rport=5060
Contact: <sip:192.168.20.113:17954>
To: <sip:303@192.168.20.113:17954;rinstance=c0f45aab2bd111ae>;tag=c2553f4e
From: "asterisk"<sip:asterisk@192.168.20.137>;tag=as1854ec78
Call-ID: 066f19117c9ed4132490a32c65d5d60c@192.168.20.137:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: eyeBeam release 1102u stamp 52345
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '066f19117c9ed4132490a32c65d5d60c@192.168.20.137:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.20.113:17954 --->


<------------->
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [9263599354@NewAsterisk:1] Set("SIP/102-000000bb", "CALLERID(num)=тут_номер") in new stack
    -- Executing [9263599354@NewAsterisk:2] Set("SIP/102-000000bb", "CALLFILENAME=1370344707.304") in new stack
    -- Executing [9263599354@NewAsterisk:3] Set("SIP/102-000000bb", "MONITOR_EXEC=/usr/local/bin/lame -b 16 "/var/spool/asterisk/monitor/1370344707.304.wav" "/var/www/mp3/1370344707.304.mp3" && rm -f "/var/spool/asterisk/monitor/1370344707.304.wav"") in new stack
    -- Executing [9263599354@NewAsterisk:4] MixMonitor("SIP/102-000000bb", "/var/spool/asterisk/monitor/1370344707.304.wav,,/usr/local/bin/lame -b 16 "/var/spool/asterisk/monitor/1370344707.304.wav" "/var/www/mp3/1370344707.304.mp3" && rm -f "/var/spool/asterisk/monitor/1370344707.304.wav"") in new stack
    -- Executing [9263599354@NewAsterisk:5] Dial("SIP/102-000000bb", "Dahdi/g0/89263599354") in new stack
    -- Requested transfer capability: 0x00 - SPEECH
    -- Called Dahdi/g0/89263599354
  == Begin MixMonitor Recording SIP/102-000000bb
    -- DAHDI/i1/89263599354-76 is proceeding passing it to SIP/102-000000bb
    -- DAHDI/i1/89263599354-76 is ringing

<--- SIP read from UDP:192.168.20.113:17954 --->


<------------->
    -- DAHDI/i1/89263599354-76 answered SIP/102-000000bb
[Jun  4 15:18:50] NOTICE[1964]: chan_sip.c:25841 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 107

<--- SIP read from UDP:192.168.20.113:17954 --->
INVITE sip:набранный_номер@192.168.20.137 SIP/2.0
Via: SIP/2.0/UDP 192.168.20.113:17954;branch=z9hG4bK-d8754z-9d2d72036109b25a-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:303@192.168.20.113:17954>
To: "набранный_номер"<sip:набранный_номер@192.168.20.137>
From: "303"<sip:303@192.168.20.137>;tag=8e114e3f
Call-ID: ZTE4ZjMwYzdhNWQ2YzUwMWMyZGQyMzU5NDYxMTMxZDg.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1102u stamp 52345
Content-Length: 421

v=0
o=- 8 2 IN IP4 192.168.20.113
s=CounterPath eyeBeam 1.5
c=IN IP4 192.168.20.113
t=0 0
m=audio 10966 RTP/AVP 107 0 8 18 101
a=alt:1 3 : NsaLKYTQ WNs+xDXB 192.168.20.113 10966
a=alt:2 2 : qhFKt+he nSSlmNv8 192.168.226.1 10966
a=alt:3 1 : FZMFzDcl F31LeXaF 192.168.111.1 10966
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
--- (12 headers 15 lines) ---
Sending to 192.168.20.113:17954 (NAT)
Using INVITE request as basis request - ZTE4ZjMwYzdhNWQ2YzUwMWMyZGQyMzU5NDYxMTMxZDg.
Found peer '303' for '303' from 192.168.20.113:17954

<--- Reliably Transmitting (NAT) to 192.168.20.113:17954 --->
SIP/2.0 401 Unauthorized
v: SIP/2.0/UDP 192.168.20.113:17954;branch=z9hG4bK-d8754z-9d2d72036109b25a-1---d8754z-;received=192.168.20.113;rport=17954
f: "303"<sip:303@192.168.20.137>;tag=8e114e3f
t: "набранный_номер"<sip:набранный_номер@192.168.20.137>;tag=as24ee4f5a
i: ZTE4ZjMwYzdhNWQ2YzUwMWMyZGQyMzU5NDYxMTMxZDg.
CSeq: 1 INVITE
Server: Asterisk PBX 10.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
k: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5935e8f8"
l: 0


<------------>
Scheduling destruction of SIP dialog 'ZTE4ZjMwYzdhNWQ2YzUwMWMyZGQyMzU5NDYxMTMxZDg.' in 6784 ms (Method: INVITE)

<--- SIP read from UDP:192.168.20.113:17954 --->
ACK sip:набранный_номер@192.168.20.137 SIP/2.0
Via: SIP/2.0/UDP 192.168.20.113:17954;branch=z9hG4bK-d8754z-9d2d72036109b25a-1---d8754z-;rport
To: "набранный_номер"<sip:набранный_номер@192.168.20.137>;tag=as24ee4f5a
From: "303"<sip:303@192.168.20.137>;tag=8e114e3f
Call-ID: ZTE4ZjMwYzdhNWQ2YzUwMWMyZGQyMzU5NDYxMTMxZDg.
CSeq: 1 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:192.168.20.113:17954 --->
INVITE sip:набранный_номер@192.168.20.137 SIP/2.0
Via: SIP/2.0/UDP 192.168.20.113:17954;branch=z9hG4bK-d8754z-5f34e421993bda7a-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:303@192.168.20.113:17954>
To: "набранный_номер"<sip:набранный_номер@192.168.20.137>
From: "303"<sip:303@192.168.20.137>;tag=8e114e3f
Call-ID: ZTE4ZjMwYzdhNWQ2YzUwMWMyZGQyMzU5NDYxMTMxZDg.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1102u stamp 52345
Authorization: Digest username="303",realm="asterisk",nonce="5935e8f8",uri="sip:набранный_номер@192.168.20.137",response="1ef5383f23796d24ff7fcf653a9c063a",algorithm=MD5
Content-Length: 421

v=0
o=- 8 2 IN IP4 192.168.20.113
s=CounterPath eyeBeam 1.5
c=IN IP4 192.168.20.113
t=0 0
m=audio 10966 RTP/AVP 107 0 8 18 101
a=alt:1 3 : NsaLKYTQ WNs+xDXB 192.168.20.113 10966
a=alt:2 2 : qhFKt+he nSSlmNv8 192.168.226.1 10966
a=alt:3 1 : FZMFzDcl F31LeXaF 192.168.111.1 10966
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
--- (13 headers 15 lines) ---
Sending to 192.168.20.113:17954 (NAT)
Using INVITE request as basis request - ZTE4ZjMwYzdhNWQ2YzUwMWMyZGQyMzU5NDYxMTMxZDg.
Found peer '303' for '303' from 192.168.20.113:17954
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Found RTP audio format 107
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found unknown media description format BV32 for ID 107
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.20.113:10966
Looking for набранный_номер in NewAsterisk (domain 192.168.20.137)
list_route: hop: <sip:303@192.168.20.113:17954>

<--- Transmitting (NAT) to 192.168.20.113:17954 --->
SIP/2.0 100 Trying
v: SIP/2.0/UDP 192.168.20.113:17954;branch=z9hG4bK-d8754z-5f34e421993bda7a-1---d8754z-;received=192.168.20.113;rport=17954
f: "303"<sip:303@192.168.20.137>;tag=8e114e3f
t: "набранный_номер"<sip:набранный_номер@192.168.20.137>
i: ZTE4ZjMwYzdhNWQ2YzUwMWMyZGQyMzU5NDYxMTMxZDg.
CSeq: 2 INVITE
Server: Asterisk PBX 10.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
k: replaces, timer
m: <sip:набранный_номер@192.168.20.137:5060>
l: 0


<------------>
    -- Executing [набранный_номер@NewAsterisk:1] Set("SIP/303-000000bc", "CALLERID(num)=здесь_наш_номер") in new stack
    -- Executing [набранный_номер@NewAsterisk:2] Set("SIP/303-000000bc", "MONITOR_EXEC=/usr/local/bin/lame -b 16 "/var/spool/asterisk/monitor/.wav" "/var/www/mp3/.mp3" && rm -f "/var/spool/asterisk/monitor/.wav"") in new stack
    -- Executing [набранный_номер@NewAsterisk:3] MixMonitor("SIP/303-000000bc", "/var/spool/asterisk/monitor/.wav,,/usr/local/bin/lame -b 16 "/var/spool/asterisk/monitor/.wav" "/var/www/mp3/.mp3" && rm -f "/var/spool/asterisk/monitor/.wav"") in new stack
    -- Executing [набранный_номер@NewAsterisk:4] Dial("SIP/303-000000bc", "Dahdi/g0/набранный_номер") in new stack
    -- Requested transfer capability: 0x00 - SPEECH
    -- Called Dahdi/g0/набранный_номер
  == Begin MixMonitor Recording SIP/303-000000bc
    -- Span 1: Channel 0/2 got hangup, cause 34
    -- DAHDI/i1/набранный_номер-77 is circuit-busy
    -- Hungup 'DAHDI/i1/набранный_номер-77'
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [набранный_номер@NewAsterisk:5] Hangup("SIP/303-000000bc", "") in new stack
  == Spawn extension (NewAsterisk, набранный_номер, 5) exited non-zero on 'SIP/303-000000bc'
Scheduling destruction of SIP dialog 'ZTE4ZjMwYzdhNWQ2YzUwMWMyZGQyMzU5NDYxMTMxZDg.' in 6784 ms (Method: INVITE)

<--- Reliably Transmitting (NAT) to 192.168.20.113:17954 --->
SIP/2.0 503 Service Unavailable
v: SIP/2.0/UDP 192.168.20.113:17954;branch=z9hG4bK-d8754z-5f34e421993bda7a-1---d8754z-;received=192.168.20.113;rport=17954
f: "303"<sip:303@192.168.20.137>;tag=8e114e3f
t: "набранный_номер"<sip:набранный_номер@192.168.20.137>;tag=as498dca96
i: ZTE4ZjMwYzdhNWQ2YzUwMWMyZGQyMzU5NDYxMTMxZDg.
CSeq: 2 INVITE
Server: Asterisk PBX 10.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
k: replaces, timer
l: 0


<------------>

<--- SIP read from UDP:192.168.20.113:17954 --->
ACK sip:набранный_номер@192.168.20.137 SIP/2.0
Via: SIP/2.0/UDP 192.168.20.113:17954;branch=z9hG4bK-d8754z-5f34e421993bda7a-1---d8754z-;rport
To: "набранный_номер"<sip:набранный_номер@192.168.20.137>;tag=as498dca96
From: "303"<sip:303@192.168.20.137>;tag=8e114e3f
Call-ID: ZTE4ZjMwYzdhNWQ2YzUwMWMyZGQyMzU5NDYxMTMxZDg.
CSeq: 2 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
  == MixMonitor close filestream (mixed)
  == Executing [/usr/local/bin/lame -b 16 "/var/spool/asterisk/monitor/.wav" "/var/www/mp3/.mp3" && rm -f "/var/spool/asterisk/monitor/.wav"]
  == End MixMonitor Recording SIP/303-000000bc
> pri intense debug span 1 выдает что-то, похожее на это (при исходящем, когда атска уже обрабатывает один звонок):

Код: Выделить всё

Enabled debugging on span 1
PRI Span: 1
PRI Span: 1 < TEI: 0 State 7(Multi-frame established)
PRI Span: 1 < V(A)=95, V(S)=95, V(R)=26
PRI Span: 1 < K=7, RC=0, l3_initiated=0, reject_except=0, ack_pend=0
PRI Span: 1 < T200_id=0, N200=3, T203_id=8192
PRI Span: 1 < [ 02 01 01 bf ]
PRI Span: 1 < Supervisory frame:
PRI Span: 1 < SAPI: 00  C/R: 1 EA: 0
PRI Span: 1 <  TEI: 000        EA: 1
PRI Span: 1 < Zero: 0     S: 0 01: 1  [ RR (receive ready) ]
PRI Span: 1 < N(R): 095 P/F: 1
PRI Span: 1 < 0 bytes of data
PRI Span: 1
PRI Span: 1 > TEI: 0 State 7(Multi-frame established)
PRI Span: 1 > V(A)=95, V(S)=95, V(R)=26
PRI Span: 1 > K=7, RC=0, l3_initiated=0, reject_except=0, ack_pend=0
PRI Span: 1 > T200_id=0, N200=3, T203_id=8192
PRI Span: 1 > [ 02 01 01 35 ]
PRI Span: 1 > Supervisory frame:
PRI Span: 1 > SAPI: 00  C/R: 1 EA: 0
PRI Span: 1 >  TEI: 000        EA: 1
PRI Span: 1 > Zero: 0     S: 0 01: 1  [ RR (receive ready) ]
PRI Span: 1 > N(R): 026 P/F: 1
PRI Span: 1 > 0 bytes of data
PRI Span: 1 -- Got ACK for N(S)=95 to (but not including) N(S)=95
PRI Span: 1 -- T200 requested to stop when not started
PRI Span: 1 T203 requested to start without stopping first
PRI Span: 1 -- Starting T203 timer
PRI Span: 1 Done handling message for SAPI/TEI=0/0
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [набранный_номер@NewAsterisk:1] Set("SIP/303-000000c0", "CALLERID(num)=наш_номер") in new stack
    -- Executing [набранный_номер@NewAsterisk:2] Set("SIP/303-000000c0", "MONITOR_EXEC=/usr/local/bin/lame -b 16 "/var/spool/asterisk/monitor/.wav" "/var/www/mp3/.mp3" && rm -f "/var/spool/asterisk/monitor/.wav"") in new stack
    -- Executing [набранный_номер@NewAsterisk:3] MixMonitor("SIP/303-000000c0", "/var/spool/asterisk/monitor/.wav,,/usr/local/bin/lame -b 16 "/var/spool/asterisk/monitor/.wav" "/var/www/mp3/.mp3" && rm -f "/var/spool/asterisk/monitor/.wav"") in new stack
    -- Executing [набранный_номер@NewAsterisk:4] Dial("SIP/303-000000c0", "Dahdi/g0/набранный_номер") in new stack
PRI Span: 1 -- Making new call for cref 32854
    -- Requested transfer capability: 0x00 - SPEECH
PRI Span: 1
PRI Span: 1 > DL-DATA request
PRI Span: 1 > Protocol Discriminator: Q.931 (8)  len=45
PRI Span: 1 > TEI=0 Call Ref: len= 2 (reference 86/0x56) (Sent from originator)
PRI Span: 1 > Message Type: SETUP (5)
PRI Span: 1 TEI=0 Transmitting N(S)=95, window is open V(A)=95 K=7
PRI Span: 1
PRI Span: 1 > TEI: 0 State 7(Multi-frame established)
PRI Span: 1 > V(A)=95, V(S)=95, V(R)=26
PRI Span: 1 > K=7, RC=0, l3_initiated=0, reject_except=0, ack_pend=0
PRI Span: 1 > T200_id=0, N200=3, T203_id=8192
PRI Span: 1 > [ 00 01 be 34 08 02 00 56 05 04 03 80 90 a3 18 03 a1 83 82 6c 0d 21 81 38 34 39 35 37 38 39 32 30 35 32 70 0c 80 38 39 31 36 30 32 33 33 34 36 33 a1 ]
PRI Span: 1 > Informational frame:
PRI Span: 1 > SAPI: 00  C/R: 0 EA: 0
PRI Span: 1 >  TEI: 000        EA: 1
PRI Span: 1 > N(S): 095   0: 0
PRI Span: 1 > N(R): 026   P: 0
PRI Span: 1 > 45 bytes of data
PRI Span: 1 > Protocol Discriminator: Q.931 (8)  len=45
PRI Span: 1 > TEI=0 Call Ref: len= 2 (reference 86/0x56) (Sent from originator)
PRI Span: 1 > Message Type: SETUP (5)
PRI Span: 1 > [04 03 80 90 a3]
PRI Span: 1 > Bearer Capability (len= 5) [ Ext: 1  Coding-Std: 0  Info transfer capability: Speech (0)
PRI Span: 1 >                              Ext: 1  Trans mode/rate: 64kbps, circuit-mode (16)
PRI Span: 1 >                                User information layer 1: A-Law (35)
PRI Span: 1 > [18 03 a1 83 82]
PRI Span: 1 > Channel ID (len= 5) [ Ext: 1  IntID: Implicit  Other(PRI)  Spare: 0  Preferred  Dchan: 0
PRI Span: 1 >                       ChanSel: As indicated in following octets
PRI Span: 1 >                       Ext: 1  Coding: 0  Number Specified  Channel Type: 3
PRI Span: 1 >                       Ext: 1  Channel: 2 Type: CPE]
PRI Span: 1 > [6c 0d 21 81 38 34 39 35 37 38 39 32 30 35 32]
PRI Span: 1 > Calling Number (len=15) [ Ext: 0  TON: National Number (2)  NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)
PRI Span: 1 >                           Presentation: Presentation permitted, user number passed network screening (1)  'наш_номер' ]
PRI Span: 1 > [70 0c 80 38 39 31 36 30 32 33 33 34 36 33]
PRI Span: 1 > Called Number (len=14) [ Ext: 1  TON: Unknown Number Type (0)  NPI: Unknown Number Plan (0)  'набранный_номер' ]
PRI Span: 1 > [a1]
PRI Span: 1 > Sending Complete (len= 1)
PRI Span: 1 -- Stopping T203 timer
PRI Span: 1 -- Starting T200 timer
PRI Span: 1 q931.c:6036 q931_setup: Call 32854 enters state 1 (Call Initiated).  Hold state: Idle
    -- Called Dahdi/g0/набранный_номер
  == Begin MixMonitor Recording SIP/303-000000c0
PRI Span: 1
PRI Span: 1 < TEI: 0 State 7(Multi-frame established)
PRI Span: 1 < V(A)=95, V(S)=96, V(R)=26
PRI Span: 1 < K=7, RC=0, l3_initiated=0, reject_except=0, ack_pend=0
PRI Span: 1 < T200_id=8192, N200=3, T203_id=0
PRI Span: 1 < [ 00 01 01 c0 ]
PRI Span: 1 < Supervisory frame:
PRI Span: 1 < SAPI: 00  C/R: 0 EA: 0
PRI Span: 1 <  TEI: 000        EA: 1
PRI Span: 1 < Zero: 0     S: 0 01: 1  [ RR (receive ready) ]
PRI Span: 1 < N(R): 096 P/F: 0
PRI Span: 1 < 0 bytes of data
PRI Span: 1 -- Got ACK for N(S)=95 to (but not including) N(S)=96
PRI Span: 1 -- ACKing N(S)=95, tx_queue head is N(S)=-1 (-1 is empty, -2 is not transmitted)
PRI Span: 1 -- Stopping T200 timer
PRI Span: 1 -- Starting T203 timer
PRI Span: 1 Done handling message for SAPI/TEI=0/0
PRI Span: 1
PRI Span: 1 < TEI: 0 State 7(Multi-frame established)
PRI Span: 1 < V(A)=96, V(S)=96, V(R)=26
PRI Span: 1 < K=7, RC=0, l3_initiated=0, reject_except=0, ack_pend=0
PRI Span: 1 < T200_id=0, N200=3, T203_id=8192
PRI Span: 1 < [ 02 01 34 c0 08 02 80 56 5a 08 02 80 a2 ]
PRI Span: 1 < Informational frame:
PRI Span: 1 < SAPI: 00  C/R: 1 EA: 0
PRI Span: 1 <  TEI: 000        EA: 1
PRI Span: 1 < N(S): 026   0: 0
PRI Span: 1 < N(R): 096   P: 0
PRI Span: 1 < 9 bytes of data
PRI Span: 1 < Protocol Discriminator: Q.931 (8)  len=9
PRI Span: 1 < TEI=0 Call Ref: len= 2 (reference 86/0x56) (Sent to originator)
PRI Span: 1 < Message Type: RELEASE COMPLETE (90)
PRI Span: 1 < [08 02 80 a2]
PRI Span: 1 < Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0  Location: User (0)
PRI Span: 1 <                  Ext: 1  Cause: Circuit/channel congestion (34), class = Network Congestion (resource unavailable) (2) ]
PRI Span: 1 -- Got ACK for N(S)=96 to (but not including) N(S)=96
PRI Span: 1 -- T200 requested to stop when not started
PRI Span: 1 T203 requested to start without stopping first
PRI Span: 1 -- Starting T203 timer
PRI Span: 1 Received message for call 0x900c228 on link 0x8e7c0d4 TEI/SAPI 0/0
PRI Span: 1 -- Processing IE 8 (cs0, Cause)
PRI Span: 1 q931.c:8567 post_handle_q931_message: Call 32854 enters state 0 (Null).  Hold state: Idle
PRI Span: 1
PRI Span: 1 > TEI: 0 State 7(Multi-frame established)
PRI Span: 1 > V(A)=96, V(S)=96, V(R)=27
PRI Span: 1 > K=7, RC=0, l3_initiated=0, reject_except=0, ack_pend=0
PRI Span: 1 > T200_id=0, N200=3, T203_id=8192
PRI Span: 1 > [ 02 01 01 36 ]
PRI Span: 1 > Supervisory frame:
PRI Span: 1 > SAPI: 00  C/R: 1 EA: 0
PRI Span: 1 >  TEI: 000        EA: 1
PRI Span: 1 > Zero: 0     S: 0 01: 1  [ RR (receive ready) ]
PRI Span: 1 > N(R): 027 P/F: 0
PRI Span: 1 > 0 bytes of data
PRI Span: 1 Done handling message for SAPI/TEI=0/0
Span 1: Processing event PRI_EVENT_HANGUP(6)
    -- Span 1: Channel 0/2 got hangup, cause 34
    -- DAHDI/i1/набранный_номер-7b is circuit-busy
PRI Span: 1 q931.c:6837 q931_hangup: Hangup other cref:32854
PRI Span: 1 q931.c:6594 __q931_hangup: ourstate Null, peerstate Null, hold-state Idle
PRI Span: 1 Destroying call 0x900c228, ourstate Null, peerstate Null, hold-state Idle
    -- Hungup 'DAHDI/i1/набранный_номер-7b'
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [набранный_номер@NewAsterisk:5] Hangup("SIP/303-000000c0", "") in new stack
  == Spawn extension (NewAsterisk, набранный_номер, 5) exited non-zero on 'SIP/303-000000c0'
  == MixMonitor close filestream (mixed)
  == Executing [/usr/local/bin/lame -b 16 "/var/spool/asterisk/monitor/.wav" "/var/www/mp3/.mp3" && rm -f "/var/spool/asterisk/monitor/.wav"]
  == End MixMonitor Recording SIP/303-000000c0
Большая просьба, подскажите новичку, куда смотреть. Можно ли что-то сказать по этому логу? Для меня эти логи на данный момент - это как грамота хищника :) Настраивал по мануалам, ну и без форума не обошлось, естественно, давно только это было. Два дня сражаюсь. Есть ощущение, что достаточно подправить всего одну из директив конфига. Спасибо неравнодушным.

Re: Доступен только один канал

Добавлено: 04 июн 2013, 16:36
sparrow
Настраивал я, все стабильно работало в старом офисе. Под астериск у нас стоит отдельный системник с Debian, после переезда по сути только провайдер поменялся (сейчас это билайн), сервер тот же, плата та же, т.к. номер поменялся - в extensions.conf вписал только новый номер для обработки входящих, сменил CALLERID(num), после чего все и началось.. только один активный звонок, независимо, входящий или исходящий.

Re: Доступен только один канал

Добавлено: 04 июн 2013, 17:15
sparrow
вот что выдает. EvroIVR - контекст с голосовым меню, для входящих. NewAsterisk - диалплан для исходящих. В нем тоже нет ничего особенного

Код: Выделить всё

exten => _8XXXXXXXXXX,1,Set(CALLERID(num)=наш_новый_номер)
exten => _8XXXXXXXXXX,n,Set(CALLFILENAME=${UNIQUEID})
exten => _8XXXXXXXXXX,n,Set(MONITOR_EXEC=/usr/local/bin/lame -b 16 "/var/spool/asterisk/monitor/${CALLFILENAME}.wav" "/var/www/mp3/${CALLFILENAME}.mp3" && rm —f "/var/spool/asterisk/monitor/${CALLFILENAME}.wav")
exten => _8XXXXXXXXXX,n,MixMonitor(/var/spool/asterisk/monitor/${CALLFILENAME}.wav,,${MONITOR_EXEC})
exten => _8XXXXXXXXXX,n,Dial(Dahdi/g0/${EXTEN}) ; Звонок на все мобильные номера через 8 (g0)
exten => _8XXXXXXXXXX,n,Busy(5)
exten => _8XXXXXXXXXX,n,Hangup()

Код: Выделить всё

dahdi show channels
   Chan Extension  Context         Language   MOH Interpret        Blocked    State      Description
 pseudo            default                    default                         In Service
      1            EvroIVR                    default                          In Service
      2            EvroIVR                    default                         In Service
      3            EvroIVR                    default                         In Service
      4            EvroIVR                    default                         In Service
      5            EvroIVR                    default                         In Service
      6            EvroIVR                    default                         In Service
      7            EvroIVR                    default                         In Service
      8            EvroIVR                    default                         In Service
      9            EvroIVR                    default                         In Service
     10            EvroIVR                    default                         In Service
     11            EvroIVR                    default                         In Service
     12            EvroIVR                    default                         In Service
     13            EvroIVR                    default                         In Service
     14            EvroIVR                    default                         In Service
     15            EvroIVR                    default                         In Service
     17            EvroIVR                    default                         In Service
     18            EvroIVR                    default                         In Service
     19            EvroIVR                    default                         In Service
     20            EvroIVR                    default                         In Service
     21            EvroIVR                    default                         In Service
     22            EvroIVR                    default                         In Service
     23            EvroIVR                    default                         In Service
     24            EvroIVR                    default                         In Service
     25            EvroIVR                    default                         In Service
     26            EvroIVR                    default                         In Service
     27            EvroIVR                    default                         In Service
     28            EvroIVR                    default                         In Service
     29            EvroIVR                    default                         In Service
     30            EvroIVR                    default                         In Service
     31            EvroIVR                    default                         In Service

Код: Выделить всё

sip show peer 303


  * Name       : 303
  Description  :
  Secret       : <Set>
  MD5Secret    : <Not set>
  Remote Secret: <Not set>
  Context      : NewAsterisk
  Subscr.Cont. : default
  Language     : ru
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    :
  Pickupgroup  :
  MOH Suggest  :
  Mailbox      :
  VM Extension : asterisk
  LastMsgsSent : 0/0
  Call limit   : 5
  Max forwards : 0
  Dynamic      : Yes
  Callerid     : "" <>
  MaxCallBR    : 384 kbps
  Expire       : 882
  Insecure     : no
  Force rport  : Yes
  ACL          : Yes
  DirectMedACL : No
  T.38 support : No
  T.38 EC mode : Unknown
  T.38 MaxDtgrm: -1
  DirectMedia  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Text Support : No
  Ign SDP ver  : No
  Trust RPID   : No
  Send RPID    : Yes
  Subscriptions: Yes
  Overlap dial : Yes
  DTMFmode     : rfc2833
  Timer T1     : 500
  Timer B      : 32000
  ToHost       :
  Addr->IP     : 192.168.20.113:17954
  Defaddr->IP  : (null)
  Prim.Transp. : UDP
  Allowed.Trsp : UDP
  Def. Username: Max
  SIP Options  : (none)
  Codecs       : (ulaw|alaw)
  Codec Order  : (alaw:20,ulaw:20)
  Auto-Framing :  No
  Status       : OK (105 ms)
  Useragent    : eyeBeam release 1102u stamp 52345
  Reg. Contact : sip:303@192.168.20.113:17954;rinstance=c0f45aab2bd111ae
  Qualify Freq : 60000 ms
  Sess-Timers  : Accept
  Sess-Refresh : uas
  Sess-Expires : 1800 secs
  Min-Sess     : 90 secs
  RTP Engine   : asterisk
  Parkinglot   :
  Use Reason   : No
  Encryption   : No

Re: Доступен только один канал

Добавлено: 04 июн 2013, 18:05
sparrow
Имеется ввиду dahdi-channels.conf?
Возможно, Вы правы. Но думаю что это не критично, тем более эти конфиги я не менял с момента переезда.

Re: Доступен только один канал

Добавлено: 05 июн 2013, 10:03
sparrow
Out, большое спасибо за помощь! Вопрос снят. Проблема оказалась всё-таки на стороне провайдера.
Часовой пояс: UTC+03:00
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