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Входящие звонки не верый Did

0

Через вебморду эластикса создаю транк и настраиваю инбоунд при звонке вместо номера вижу логин учетной записи!

  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [HellKlUsov1@from-trunk-sip-arctel:1] Set("SIP/arctel-000000ba", "GROUP()=OUT_4") in new stack
    -- Executing [HellKlUsov1@from-trunk-sip-arctel:2] Goto("SIP/arctel-000000ba", "from-trunk,HellKlUsov1,1") in new stack
    -- Goto (from-trunk,HellKlUsov1,1)
    -- Executing [HellKlUsov1@from-trunk:1] Dial("SIP/arctel-000000ba", "SIP/305") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Couldn't call 305
  == Everyone is busy/congested at this time (0:0/0/0)
    -- Executing [HellKlUsov1@from-trunk:2] NoOp("SIP/arctel-000000ba", "Received an unknown call with DID set to HellKlUsov1") in new stack
    -- Executing [HellKlUsov1@from-trunk:3] Goto("SIP/arctel-000000ba", "s,a2") in new stack
    -- Goto (from-trunk,s,2)
    -- Executing [s@from-trunk:2] Answer("SIP/arctel-000000ba", "") in new stack
    -- Executing [s@from-trunk:3] Wait("SIP/arctel-000000ba", "2") in new stack
    -- Executing [s@from-trunk:4] Playback("SIP/arctel-000000ba", "ss-noservice") in new stack
    -- Executing [s@from-trunk:5] SayAlpha("SIP/arctel-000000ba", "") in new stack
    -- Executing [s@from-trunk:6] Hangup("SIP/arctel-000000ba", "") in new stack
  == Spawn extension (from-trunk, s, 6) exited non-zero on 'SIP/arctel-000000ba'
    -- Executing [h@from-trunk:1] Hangup("SIP/arctel-000000ba", "") in new stack

Транк:

host=86.110.4.хх
fromdomain=86.110.4.хх
fromuser=HellKlUsov1
username=HellKlUsov1
secret=****
dtmfmode=rfc2833
type=peer
disallow=all
allow=alaw&ulaw&g729
nat=yes
canreinvite=no
insecure=invite

secret=****
type=friend
context=from-trunk

HellKlUsov1:****@86.110.4.хх/HellKlUsov1
удалить закрыть спам изменить тег редактировать

спросил 2011-09-01 16:05:50 +0400

DJs3000 Gravatar DJs3000 flag of Russian Federation
551 41 13 33
http://retrogamesbattle.c...

обновил 2011-09-01 16:47:40 +0400

Comments

почему did это логин а не номер? DJs3000 ( 2011-09-01 16:48:50 +0400 )редактировать

2 Ответа

1

как задано в транке, на то и приходит вызов

 register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry]

после косой черты должен быть прописан did

ссылка удалить спам редактировать

ответил 2011-09-01 17:26:14 +0400

zlat Gravatar zlat
471 3 12

Comments

вот спасибо тебе попробую) DJs3000 ( 2011-09-01 17:38:01 +0400 )редактировать
не выходит! примерно так должно быть? HellKlUsov1:******@86.110.4.148/4952253038 как-то странно) вроде активность при звонке есть == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 а диалплан не вызывается DJs3000 ( 2011-09-01 19:57:26 +0400 )редактировать
включайте дебаг по этому транку и смотрите пакеты во время звонка zlat ( 2011-09-01 20:51:28 +0400 )редактировать
0

ну вот дебага ток хз что тут понять можно

v=0
o=HuaweiSoftX3000 46524232 46524232 IN IP4 86.110.4.148
s=Sip Call
c=IN IP4 86.110.4.148
t=0 0
m=audio 46340 RTP/AVP 18 4 97
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=fmtp:18 annexb=yes

<------------->
--- (13 headers 11 lines) ---
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Sending to 86.110.4.148 : 5060 (NAT)
Using INVITE request as basis request - SBCbdeac770d39dcea08d78d2e05f0570da@172.21.0.2
Found peer 'arctel' for 'Anonymous' from 86.110.4.148:5060

<--- Reliably Transmitting (NAT) to 86.110.4.148:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 86.110.4.148:5060;branch=z9hG4bK80c761367163c159404141562;received=86.110.4.148
From: Anonymous <sip:Anonymous@86.110.4.148;user=phone>;tag=8013836e
To: <sip:4952253038@213.247.249.2;user=phone>;tag=as176b56c3
Call-ID: SBCbdeac770d39dcea08d78d2e05f0570da@172.21.0.2
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7258ac33"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'SBCbdeac770d39dcea08d78d2e05f0570da@172.21.0.2' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:86.110.4.148:5060 --->
ACK sip:HellKlUsov1@213.247.249.2:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 86.110.4.148:5060;branch=z9hG4bK80c761367163c159404141562
Call-ID: SBCbdeac770d39dcea08d78d2e05f0570da@172.21.0.2
From: Anonymous <sip:Anonymous@86.110.4.148;user=phone>;tag=8013836e
To: <sip:4952253038@213.247.249.2;user=phone>;tag=as176b56c3
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0


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

<--- SIP read from UDP:86.110.4.148:5060 --->
INVITE sip:HellKlUsov1@213.247.249.2:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 86.110.4.148:5060;branch=z9hG4bKa28d90ca664e5723cadcd2c29
Call-ID: SBC54877ac3dba80e99813bf2d8b2c6787f@172.21.0.2
From: Anonymous <sip:Anonymous@86.110.4.148;user=phone>;tag=c58eca08
To: <sip:4952253038@213.247.249.2;user=phone>
CSeq: 1 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER
Max-Forwards: 70
Supported: 100rel
User-Agent: Huawei SoftX3000 V300R006
Contact: <sip:Anonymous@86.110.4.148:5060;user=phone>
Content-Length: 252
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 46521956 46521956 IN IP4 86.110.4.148
s=Sip Call
c=IN IP4 86.110.4.148
t=0 0
m=audio 46354 RTP/AVP 18 4 97
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=fmtp:18 annexb=yes

<------------->
--- (13 headers 11 lines) ---
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Sending to 86.110.4.148 : 5060 (NAT)
Using INVITE request as basis request - SBC54877ac3dba80e99813bf2d8b2c6787f@172.21.0.2
Found peer 'arctel' for 'Anonymous' from 86.110.4.148:5060

<--- Reliably Transmitting (NAT) to 86.110.4.148:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 86.110.4.148:5060;branch=z9hG4bKa28d90ca664e5723cadcd2c29;received=86.110.4.148
From: Anonymous <sip:Anonymous@86.110.4.148;user=phone>;tag=c58eca08
To: <sip:4952253038@213.247.249.2;user=phone>;tag=as445945a6
Call-ID: SBC54877ac3dba80e99813bf2d8b2c6787f@172.21.0.2
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="030b4b68"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'SBC54877ac3dba80e99813bf2d8b2c6787f@172.21.0.2' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:86.110.4.148:5060 --->
ACK sip:HellKlUsov1@213.247.249.2:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 86.110.4.148:5060;branch=z9hG4bKa28d90ca664e5723cadcd2c29
Call-ID: SBC54877ac3dba80e99813bf2d8b2c6787f@172.21.0.2
From: Anonymous <sip:Anonymous@86.110.4.148;user=phone>;tag=c58eca08
To: <sip:4952253038@213.247.249.2;user=phone>;tag=as445945a6
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0


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

<--- SIP read from UDP:86.110.4.148:5060 --->
INVITE sip:HellKlUsov1@213.247.249.2:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 86.110.4.148:5060;branch=z9hG4bK9731a67203af000e185554203
Call-ID: SBC34a2ce634f96346e74b67164c5bb3d2f@172.21.0.2
From: Anonymous <sip:Anonymous@86.110.4.148;user=phone>;tag=f8677964
To: <sip:4952253038@213.247.249.2;user=phone>
CSeq: 1 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER
Max-Forwards: 70
Supported: 100rel
User-Agent: Huawei SoftX3000 V300R006
Contact: <sip:Anonymous@86.110.4.148:5060;user=phone>
Content-Length: 252
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 57252407 57252407 IN IP4 86.110.4.148
s=Sip Call
c=IN IP4 86.110.4.148
t=0 0
m=audio 46370 RTP/AVP 18 4 97
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=fmtp:18 annexb=yes

<------------->
--- (13 headers 11 lines) ---
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Sending to 86.110.4.148 : 5060 (NAT)
Using INVITE request as basis request - SBC34a2ce634f96346e74b67164c5bb3d2f@172.21.0.2
Found peer 'arctel' for 'Anonymous' from 86.110.4.148:5060

<--- Reliably Transmitting (NAT) to 86.110.4.148:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 86.110.4.148:5060;branch=z9hG4bK9731a67203af000e185554203;received=86.110.4.148
From: Anonymous <sip:Anonymous@86.110.4.148;user=phone>;tag=f8677964
To: <sip:4952253038@213.247.249.2;user=phone>;tag=as0dcf82f0
Call-ID: SBC34a2ce634f96346e74b67164c5bb3d2f@172.21.0.2
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="46fe7060"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'SBC34a2ce634f96346e74b67164c5bb3d2f@172.21.0.2' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:86.110.4.148:5060 --->
ACK sip:HellKlUsov1@213.247.249.2:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 86.110.4.148:5060;branch=z9hG4bK9731a67203af000e185554203
Call-ID: SBC34a2ce634f96346e74b67164c5bb3d2f@172.21.0.2
From: Anonymous <sip:Anonymous@86.110.4.148;user=phone>;tag=f8677964
To: <sip:4952253038@213.247.249.2;user=phone>;tag=as0dcf82f0
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0


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

<--- SIP read from UDP:86.110.4.148:5060 --->
ссылка удалить спам редактировать

ответил 2011-09-01 21:12:36 +0400

DJs3000 Gravatar DJs3000 flag of Russian Federation
551 41 13 33
http://retrogamesbattle.c...

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Задан: 2011-09-01 16:05:50 +0400

Просмотрен: 321 раз

Обновлен: Sep 01 '11

Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией GNU GPL.