Абонент PSTN - АТС – по (PRI EDSS) cisco 3745 – Asterisk 1.4 – SIP абонент
Со стороны Абонент PSTN до SIP абонента звонок проходит.
Звоним от SIP абонента на номер 440000 в сторону абонента PSTN и звонка нет.
Делаем трассировку EDSS, видим что сообщение SETUP от циски приходит с номером вызываемого 440000 и номером вызывающего 100. но соединение не устанавливается.
sip debug peer cisco
--- (10 headers 0 lines) ---
-- Got SIP response 503 "Service Unavailable" back from 192.168.1.1
Transmitting (no NAT) to 192.168.1.1:5060:
ACK sip:440000@192.168.1.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.42:5060;branch=z9hG4bK396558ac;rport
From: "Test" ;tag=as2f8dffe0
To: ;tag=256CFE80-C75
Contact:
Call-ID: 699438941ee06287109ad81c6e3f4986@192.168.1.42
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
Может кто подскажет, будем рады!
говорит о том, что на Cisco 3745 (192.168.1.1) не настроены диал пиры.
sh dial-peer voice 103
VoiceEncapPeer103
peer type = voice, information type = voice,
description = `',
tag = 103, destination-pattern = `440000',
answer-address = `', preference=0,
CLID Restriction = None
CLID Network Number = `'
CLID Second Number sent
source carrier-id = `', target carrier-id = `',
source trunk-group-label = `', target trunk-group-label = `',
numbering Type = `unknown'
group = 103, Admin state is up, Operation state is up,
incoming called-number = `', connections/maximum = 0/unlimited,
DTMF Relay = disabled,
huntstop = disabled,
in bound application associated: 'DEFAULT'
out bound application associated: ''
dnis-map =
permission :both
incoming COR list:maximum capability
outgoing COR list:minimum requirement
Translation profile (Incoming):
Translation profile (Outgoing):
incoming call blocking:
translation-profile = `'
disconnect-cause = `no-service'
advertise 0x40 capacity_update_timer 25 addrFamily 4 oldAddrFamily 4
type = pots, prefix = `',
forward-digits all
session-target = `', voice-port = `3/1:15',
direct-inward-dial = enabled,
digit_strip = enabled,
register E.164 number with GK = TRUE
fax rate = system, payload size = 20 bytes
supported-language = ''
Time elapsed since last clearing of voice call statistics never
Connect Time = 2963, Charged Units = 0,
Successful Calls = 0, Failed Calls = 192, Incomplete Calls = 0
Accepted Calls = 23, Refused Calls = 9,
Last Disconnect Cause is "22 ",
Last Disconnect Text is "no circuit (34)",
Last Setup Time = 63345102.
-----------------
dial-peer voice 102 voip
destination-pattern 1T
progress_ind setup enable 3
voice-class codec 1
session protocol sipv2
session target sip-server
dtmf-relay cisco-rtp
!
dial-peer voice 103 pots
destination-pattern 440000
direct-inward-dial
port 3/1:15
forward-digits all
--------------------
sh dial-peer voice 102
VoiceOverIpPeer102
peer type = voice, information type = voice,
description = `',
tag = 102, destination-pattern = `1T',
answer-address = `', preference=0,
CLID Restriction = None
CLID Network Number = `'
CLID Second Number sent
source carrier-id = `', target carrier-id = `',
source trunk-group-label = `', target trunk-group-label = `',
numbering Type = `unknown'
group = 102, Admin state is up, Operation state is up,
incoming called-number = `', connections/maximum = 0/unlimited,
DTMF Relay = enabled,
modem transport = system,
huntstop = disabled,
in bound application associated: 'DEFAULT'
out bound application associated: ''
dnis-map =
permission :both
incoming COR list:maximum capability
outgoing COR list:minimum requirement
Translation profile (Incoming):
Translation profile (Outgoing):
incoming call blocking:
translation-profile = `'
disconnect-cause = `no-service'
advertise 0x40 capacity_update_timer 25 addrFamily 4 oldAddrFamily 4
type = voip, session-target = `sip-server',
technology prefix:
settle-call = disabled
ip media DSCP = ef, ip signaling DSCP = af31, UDP checksum = disabled,
session-protocol = sipv2, session-transport = system, req-qos = best-effort,
acc-qos = best-effort,
dtmf-relay = cisco-rtp,
RTP dynamic payload type values: NTE = 101
Cisco: NSE=100, fax=96, fax-ack=97, dtmf=121, fax-relay=122
CAS=123, ClearChan=125, PCM switch over u-law=0,A-law=8
RTP comfort noise payload type = 19
fax rate = voice, payload size = 20 bytes
fax protocol = system
fax-relay ecm enable
fax NSF = 0xAD0051 (default)
codec = g729r8, payload size = 20 bytes,
Media Setting = flow-through (global)
Expect factor = 10, Icpif = 20,
Playout Mode is set to adaptive,
Initial 60 ms, Max 250 ms
Playout-delay Minimum mode is set to default, value 40 ms
Fax nominal 300 ms
Max Redirects = 1, signaling-type = cas,
VAD = enabled, Poor QOV Trap = disabled,
Source Interface = NONE
voice class sip url = system,
voice class sip rel1xx = system,
voice class perm tag = `'
Time elapsed since last clearing of voice call statistics never
Connect Time = 27102, Charged Units = 0,
Successful Calls = 26, Failed Calls = 40, Incomplete Calls = 0
Accepted Calls = 0, Refused Calls = 205,
Last Disconnect Cause is "22 ",
Last Disconnect Text is "no circuit (34)",
Last Setup Time = 63345096.
Added after 2 hours 10 minutes:
теперь так:
sip debug peer cisco
SIP Debugging Enabled for IP: 192.168.1.1:5060
The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead.
*CLI> -- Executing [440000@local-phones] Dial("SIP/100-086c4000", "SIP/440000@cisco|20") in new stack
Audio is at 192.168.1.42 port 12618
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.1:5060:
INVITE sip:440000@192.168.1.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.42:5060;branch=z9hG4bK3ba4112d;rport
From: "" ;tag=as063906cc
To:
Contact:
Call-ID: 4b1395490ca6842651882ac448cfd838@192.168.1.42
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 12 Feb 2009 22:17:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 287
v=0
o=root 53259 53259 IN IP4 192.168.1.42
s=session
c=IN IP4 192.168.1.42
t=0 0
m=audio 12618 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called 440000@cisco
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.42:5060;branch=z9hG4bK3ba4112d;rport
From: "" ;tag=as063906cc
To: ;tag=29A1380C-15B3
Date: Tue, 09 Mar 1993 02:00:31 GMT
Call-ID: 4b1395490ca6842651882ac448cfd838@192.168.1.42
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0
--- (10 headers 0 lines) ---
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.42:5060;branch=z9hG4bK3ba4112d;rport
From: "" ;tag=as063906cc
To: ;tag=29A1380C-15B3
Date: Tue, 09 Mar 1993 02:00:31 GMT
Call-ID: 4b1395490ca6842651882ac448cfd838@192.168.1.42
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0
--- (10 headers 0 lines) ---
Transmitting (no NAT) to 192.168.1.1:5060:
ACK sip:440000@192.168.1.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.42:5060;branch=z9hG4bK3ba4112d;rport
From: "" ;tag=as063906cc
To: ;tag=29A1380C-15B3
Contact:
Call-ID: 4b1395490ca6842651882ac448cfd838@192.168.1.42
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/cisco-086cd000 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [440000@local-phones] Congestion("SIP/100-086c4000", "") in new stack
== Spawn extension (local-phones, 440000, 2) exited non-zero on 'SIP/100-086c4000'
Really destroying SIP dialog '4b1395490ca6842651882ac448cfd838@192.168.1.42' Method: INVITE
Added after 45 minutes:
все заработало!!!
ошибка была на АТС. неправильно выставлены параметры сигнализации