-- Executing [7021429@fxo:1] Goto("SIP/trunkPSTN-000002ad", "ivr,s,1") in new stack
-- Goto (ivr,s,1)
-- Executing [s@ivr:1] Set("SIP/trunkPSTN-000002ad", "home="/overlay"") in new stack
-- Executing [s@ivr:2] Wait("SIP/trunkPSTN-000002ad", "1") in new stack
-- Executing [s@ivr:3] NoOp("SIP/trunkPSTN-000002ad", ""asreceived" ") in new stack
-- Executing [s@ivr:4] BackGround("SIP/trunkPSTN-000002ad", ""/overlay"/ivr") in new stack
-- Playing '/overlay/ivr.slin' (language 'en')
-- Executing [s@ivr:5] WaitExten("SIP/trunkPSTN-000002ad", "1") in new stack
-- Timeout on SIP/trunkPSTN-000002ad, going to 't'
-- Executing [t@ivr:1] Dial("SIP/trunkPSTN-000002ad", "SIP/403,20") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/403
-- SIP/403-000002ae is ringing
-- SIP/403-000002ae is ringing
-- SIP/403-000002ae answered SIP/trunkPSTN-000002ad
-- Locally bridging SIP/trunkPSTN-000002ad and SIP/403-000002ae
== Spawn extension (ivr, t, 1) exited non-zero on 'SIP/trunkPSTN-000002ad'
Вот лог астера...сори за задержку, были кое какие траблы.
сделайте sip set debug on и покажите.
скорее всего астериск не передает полученный аон дальше
INVITE sip:7021429@172.16.0.149 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.17:5060;branch=z9hG4bK-204b336c
From: trunkPSTN ;tag=650705b2f60a294o1
To:
Remote-Party-ID: trunkPSTN ;screen=yes;party=calling
Call-ID: 97aa4d27-92d8c38@172.16.0.17
CSeq: 101 INVITE
Max-Forwards: 70
Contact: trunkPSTN
Expires: 240
User-Agent: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 446
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp
v=0
o=- 225582260 225582260 IN IP4 172.16.0.17
s=-
c=IN IP4 172.16.0.17
t=0 0
m=audio 16384 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
--- (15 headers 20 lines) ---
Sending to 172.16.0.17:5060 (NAT)
Using INVITE request as basis request - 97aa4d27-92d8c38@172.16.0.17
Found peer 'trunkPSTN' for 'trunkPSTN' from 172.16.0.17:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 100
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 2
Found audio description format G723 for ID 4
Found audio description format PCMA for ID 8
Found audio description format G729a for ID 18
Found unknown media description format G726-40 for ID 96
Found unknown media description format G726-24 for ID 97
Found unknown media description format G726-16 for ID 98
Found unknown media description format NSE for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x90d (g723|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.16.0.17:16384
Looking for 7021429 in fxo (domain 172.16.0.149)
list_route: hop:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.0.17:5060;branch=z9hG4bK-204b336c;received=172.16.0.17;rport=5060
From: trunkPSTN ;tag=650705b2f60a294o1
To:
Call-ID: 97aa4d27-92d8c38@172.16.0.17
CSeq: 101 INVITE
Server: Asterisk PBX 1.8.10.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Length: 0
-- Executing [7021429@fxo:1] Goto("SIP/trunkPSTN-0000033a", "ivr,s,1") in new stack
-- Goto (ivr,s,1)
-- Executing [s@ivr:1] Set("SIP/trunkPSTN-0000033a", "home="/overlay"") in new stack
-- Executing [s@ivr:2] Wait("SIP/trunkPSTN-0000033a", "1") in new stack
-- Executing [s@ivr:3] NoOp("SIP/trunkPSTN-0000033a", ""asreceived" ") in new stack
-- Executing [s@ivr:4] BackGround("SIP/trunkPSTN-0000033a", ""/overlay"/ivr") in new stack
Могу ещё дальше кинуть, но думаю основное видно: Call-ID: 97aa4d27-92d8c38@172.16.0.17 , значит он почему то не распознает cid?
У вас там trunkPSTN.
Покажите sip show peer trunkPSTN.
Зайдите на коробочку, Voice, PSTN Line в секции Subscriber Information что написано в поле Display Name:?
покажите
| Код: |
| sip show peer trunkPSTN |
И еще проверте на коробочке в той же вкладке с секции PSTN-To-VoIP Gateway Setup настройку PSTN CID For VoIP CID
* Name : trunkPSTN
Secret :
MD5Secret :
Remote Secret:
Context : fxo
Subscr.Cont. :
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
MOH Suggest :
Mailbox :
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit : 0
Max forwards : 0
Dynamic : Yes
Callerid : "asreceived"
MaxCallBR : 384 kbps
Expire : 3553
Insecure : no
Force rport : Yes
ACL : No
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 : No
Subscriptions: Yes
Overlap dial : No
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost :
Addr->IP : 172.16.0.17:5060
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: trunkPSTN
SIP Options : (none)
Codecs : 0x80000008000e (gsm|ulaw|alaw|h263|testlaw)
Codec Order : (none)
Auto-Framing : No
Status : Unmonitored
Useragent : Linksys/SPA3102-5.2.13(GW002)
Reg. Contact : sip:trunkPSTN@172.16.0.17:5060
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
На всякий случай сип и диалплан:
sip.conf:
[trunkPSTN]
context=fxo
type=peer
host=dynamic
canreinvite=no
dtmfmode=rfc2833
callerid=asreceived
extensions.conf
[fxo]
exten => 7021429,1,Goto(ivr,s,1)
exten => _3XXXXXX,1,Dial(SIP/trunkPSTN/${EXTEN})
exten => _7XXXXXX,1,Dial(SIP/trunkPSTN/${EXTEN})
exten => _5XXXXX,1,Dial(SIP/trunkPSTN/${EXTEN})
exten => _6XXXXX,1,Dial(SIP/trunkPSTN/${EXTEN})
exten => _9XXXXX,1,Dial(SIP/trunkPSTN/${EXTEN})
exten => 1508,1,Dial(SIP/trunkPSTN/${EXTEN})
Также:
PSTN CID For VoIP CID: yes
Caller ID Method: ETSI FSK
== Using SIP RTP CoS mark 5
-- Executing [7021429@fxo:1] Goto("SIP/trunkPSTN-0000001f", "ivr,s,1") in new stack
-- Goto (ivr,s,1)
-- Executing [s@ivr:1] Set("SIP/trunkPSTN-0000001f", "home="/overlay"") in new stack
-- Executing [s@ivr:2] Wait("SIP/trunkPSTN-0000001f", "1") in new stack
-- Executing [s@ivr:3] NoOp("SIP/trunkPSTN-0000001f", """ ") in new stack
-- Executing [s@ivr:4] BackGround("SIP/trunkPSTN-0000001f", ""/overlay"/ivr") in new stack
-- Playing '/overlay/ivr.slin' (language 'en')
== Spawn extension (ivr, s, 4) exited non-zero on 'SIP/trunkPSTN-0000001f'
Поменял уже в Caller ID FSK Standard: bell 202 на v.23 , не получилось по прежнему, в чем ещё может быть дело?