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Переадресация звонков в Asterisk не работает

Необходимо сделать перенаправление звонков.

Вот мой диалплан:

exten => 6004,1,Dial(SIP/6004,15,t)
exten => 6003,1,Dial(SIP/6003,15,t)
exten => 6001,1,Dial(SIP/6001,15,t)

Вот мой featuremap

[featuremap]
blindxfer => #1 ; Blind transfer (default is #) -- Make sure to set the T and/or t option in the Dial() or Queue() app call!
disconnect => *0 ; Disconnect (default is *) -- Make sure to set the H and/or h option in the Dial() or Queue() app call!
automon => *1 ; One Touch Record a.k.a. Touch Monitor -- Make sure to set the W and/or w option in the Dial() or Queue() app call!
atxfer => *2 ; Attended transfer -- Make sure to set the T and/or t option in the Dial() or Queue() app call!
parkcall => #72 ; Park call (one step parking) -- Make sure to set the K and/or k option in the Dial() app call!
automixmon => *3 ; One Touch Record a.k.a. Touch MixMonitor -- Make sure to set the X and/or x option in the Dial() or Queue() app call!

Номер 6001 звонит на 6003, 6003 отвечает набираю #16004, 6003 начинает звонить на 6004, 6003 свой звонок с 6001 ставит на удержание, в 6001 проигрывается музыка, 6004 поднимает трубку. Я так понимаю, что должно быть по-другому: должны разговаривать 6001 и 6004, а 6003 висеть на удержании. Что я делаю неправильно.

*2 почему то вообще не работает. Также не работает перенапраление, если установить blindxfer => #1.

В идеале, яхочу чтобы работало и #1 и *2. Помогите разобраться пожалуста.

Переадресация звонков в Asterisk не работает

Необходимо сделать перенаправление звонков.

Вот мой диалплан:

exten => 6004,1,Dial(SIP/6004,15,t)
exten => 6003,1,Dial(SIP/6003,15,t)
exten => 6001,1,Dial(SIP/6001,15,t)

Вот мой featuremap

[featuremap]
blindxfer => #1 ; Blind transfer (default is #) -- Make sure to set the T and/or t option in the Dial() or Queue() app call!
disconnect => *0 ; Disconnect (default is *) -- Make sure to set the H and/or h option in the Dial() or Queue() app call!
automon => *1 ; One Touch Record a.k.a. Touch Monitor -- Make sure to set the W and/or w option in the Dial() or Queue() app call!
atxfer => *2 ; Attended transfer -- Make sure to set the T and/or t option in the Dial() or Queue() app call!
parkcall => #72 ; Park call (one step parking) -- Make sure to set the K and/or k option in the Dial() app call!
automixmon => *3 ; One Touch Record a.k.a. Touch MixMonitor -- Make sure to set the X and/or x option in the Dial() or Queue() app call!

*2 почему то вообще не работает. Также не работает перенапраление, если установить blindxfer => ~~#1.~~#.

В идеале, яхочу чтобы работало и #1 и *2. Помогите разобраться пожалуста.

Переадресация звонков в Asterisk не работает

Необходимо сделать перенаправление звонков.

Вот мой диалплан:

exten => 6004,1,Dial(SIP/6004,15,t)
exten => 6003,1,Dial(SIP/6003,15,t)
exten => 6001,1,Dial(SIP/6001,15,t)

Вот мой featuremap

[featuremap]
blindxfer => #1 ; Blind transfer (default is #) -- Make sure to set the T and/or t option in the Dial() or Queue() app call!
disconnect => *0 ; Disconnect (default is *) -- Make sure to set the H and/or h option in the Dial() or Queue() app call!
automon => *1 ; One Touch Record a.k.a. Touch Monitor -- Make sure to set the W and/or w option in the Dial() or Queue() app call!
atxfer => *2 ; Attended transfer -- Make sure to set the T and/or t option in the Dial() or Queue() app call!
parkcall => #72 ; Park call (one step parking) -- Make sure to set the K and/or k option in the Dial() app call!
automixmon => *3 ; One Touch Record a.k.a. Touch MixMonitor -- Make sure to set the X and/or x option in the Dial() or Queue() app call!

*2 почему то вообще не работает. Также не работает перенапраление, если установить blindxfer => #.

В идеале, яхочу чтобы работало и #1 и *2. Помогите разобраться пожалуста.

Вот часть лога:

 == Using SIP RTP CoS mark 5
    -- Executing [6003@DLPN_offce:1] Dial("SIP/6001-00000004", "SIP/6003,15,t") in new stack
  == Using SIP RTP CoS mark 5
    -- Called 6003
    -- SIP/6003-00000005 is ringing
    -- SIP/6003-00000005 answered SIP/6001-00000004
[Jun  4 19:58:56] NOTICE[17174]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP codec 126 received from '192.168.10.198:65162'
[Jun  4 19:58:56] NOTICE[17174]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP codec 126 received from '192.168.10.198:65162'
[Jun  4 19:58:56] NOTICE[17174]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP codec 126 received from '192.168.10.198:65162'
[Jun  4 19:59:06] NOTICE[17174]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP codec 126 received from '192.168.10.198:65162'
  == Using SIP RTP CoS mark 5
    -- Executing [6004@DLPN_offce:1] Dial("SIP/6003-00000006", "SIP/6004,15,t") in new stack
  == Using SIP RTP CoS mark 5
    -- Called 6004
    -- Started music on hold, class 'default', on SIP/6001-00000004
    -- SIP/6004-00000007 is ringing
[Jun  4 19:59:16] NOTICE[17174]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP codec 126 received from '(null)'
    -- SIP/6004-00000007 answered SIP/6003-00000006
[Jun  4 19:59:20] NOTICE[17178]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP codec 126 received from '192.168.10.207:60596'
[Jun  4 19:59:20] NOTICE[17178]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP codec 126 received from '192.168.10.207:60596'
[Jun  4 19:59:20] NOTICE[17178]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP codec 126 received from '192.168.10.207:60596'
[Jun  4 19:59:20] NOTICE[17178]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP codec 126 received from '192.168.10.198:62242'
[Jun  4 19:59:20] NOTICE[17178]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP codec 126 received from '192.168.10.198:62242'
[Jun  4 19:59:20] NOTICE[17178]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP codec 126 received from '192.168.10.198:62242'
  == Spawn extension (DLPN_offce, 6004, 1) exited non-zero on 'SIP/6003-00000006'
[Jun  4 19:59:26] NOTICE[17174]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP codec 126 received from '(null)'
[Jun  4 19:59:37] NOTICE[17174]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP codec 126 received from '(null)'
[Jun  4 19:59:47] NOTICE[17174]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP codec 126 received from '(null)'

Переадресация звонков в Asterisk не работает

Необходимо сделать перенаправление звонков.

Вот мой диалплан:

exten => 6004,1,Dial(SIP/6004,15,t)
exten => 6003,1,Dial(SIP/6003,15,t)
exten => 6001,1,Dial(SIP/6001,15,t)

Вот мой featuremap

[featuremap]
blindxfer => #1 ; Blind transfer (default is #) -- Make sure to set the T and/or t option in the Dial() or Queue() app call!
disconnect => *0 ; Disconnect (default is *) -- Make sure to set the H and/or h option in the Dial() or Queue() app call!
automon => *1 ; One Touch Record a.k.a. Touch Monitor -- Make sure to set the W and/or w option in the Dial() or Queue() app call!
atxfer => *2 ; Attended transfer -- Make sure to set the T and/or t option in the Dial() or Queue() app call!
parkcall => #72 ; Park call (one step parking) -- Make sure to set the K and/or k option in the Dial() app call!
automixmon => *3 ; One Touch Record a.k.a. Touch MixMonitor -- Make sure to set the X and/or x option in the Dial() or Queue() app call!

*2 почему то вообще не работает. Также не работает перенапраление, если установить blindxfer => #.

В идеале, яхочу чтобы работало и #1 и *2. Помогите разобраться пожалуста.

Вот часть ~~лога:~~

лога:
Content-Length: 0
 Content-Length: 0> Content-Length: 0
  <------------->
  --- (10 headers 0 lines) ---
 <--- SIP read from
  UDP:192.168.10.198:46214 ---> INVITE
  sip:6004@192.168.10.159 SIP/2.0 Via:
  SIP/2.0/UDP
  192.168.10.198:46214;branch=z9hG4bK-d8754z-7f156d5d7b769dcb-1---d8754z-;rport
  Max-Forwards: 70 Contact:
  <sip:6003@192.168.10.198:46214> To:
  <sip:6004@192.168.10.159> From:
  "6003"<sip:6003@192.168.10.159>;tag=60f21add
  Call-ID:
  NTFlMGQ4OGU1MjY4YjExZjE0MTRlYWJhZGVkYzU4NTg.
  CSeq: 2 INVITE Allow: INVITE, ACK,
  CANCEL, OPTIONS, BYE, REFER, NOTIFY,
  MESSAGE, SUBSCRIBE, INFO Content-Type:
  application/sdp Supported: replaces
  User-Agent: X-Lite 4 release 4.1 stamp
  63214 Authorization: Digest
  username="6003",realm="asterisk",nonce="4e427875",uri="sip:6004@192.168.10.159",response="6b03ab1b19a09d4b7e42a9e106fd8463",algorithm=MD5
  Content-Length: 383
 v=0 o=- 12983303904982203 1 IN IP4
  192.168.10.198 s=CounterPath X-Lite 4.1 c=IN IP4 192.168.10.198 t=0 0 a=ice-ufrag:1f7bf3
  a=ice-pwd:43798db74704cb3c028ed1d5fe8b3fd3
  m=audio 53280 RTP/AVP 3 101
  a=rtpmap:101 telephone-event/8000
  a=fmtp:101 0-15 a=sendrecv
  a=candidate:1 1 UDP 659136
  192.168.10.198 53280 typ host a=candidate:1 2 UDP 659134
  192.168.10.198 53281 typ host <------------->
  --- (14 headers 13 lines) --- Sending to 192.168.10.198:46214 (no NAT) Using
  INVITE request as basis request -
  NTFlMGQ4OGU1MjY4YjExZjE0MTRlYWJhZGVkYzU4NTg.
  Found peer '6003' for '6003' from
  192.168.10.198:46214   == Using SIP RTP CoS mark 5
5 Found RTP audio format
  3 Found RTP audio format 101 Found
  audio description format
  telephone-event for ID 101
  Capabilities: us - 0x6 (gsm|ulaw),
  peer - audio=0x2 (gsm)/video=0x0
  (nothing)/text=0x0 (nothing), combined
  - 0x2 (gsm) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|),
  peer - 0x1 (telephone-event|),
  combined - 0x1 (telephone-event|) Peer
  audio RTP is at port
  192.168.10.198:53280 Looking for 6004 in DLPNoffce (domain 192.168.10.159)
  listroute: hop:
  <sip:6003@192.168.10.198:46214>
 <--- Transmitting (no NAT) to
  192.168.10.198:46214 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP
  192.168.10.198:46214;branch=z9hG4bK-d8754z-7f156d5d7b769dcb-1---d8754z-;received=192.168.10.198;rport=46214
  From:
  "6003"<sip:6003@192.168.10.159>;tag=60f21add
  To: <sip:6004@192.168.10.159> Call-ID:
  NTFlMGQ4OGU1MjY4YjExZjE0MTRlYWJhZGVkYzU4NTg.
  CSeq: 2 INVITE Server: Asterisk PBX
  1.8.4.4~dfsg-2ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
  SUBSCRIBE, NOTIFY, INFO, PUBLISH
  Supported: replaces, timer Contact:
  <sip:6004@192.168.10.159:5060>
  Content-Length: 0
 <------------>
      -- Executing [6003@DLPN_offce:1] Dial("SIP/6001-00000004", "SIP/6003,15,t") [6004@DLPN_offce:1] Dial("SIP/6003-0000000a",
  "SIP/6004,15,t") in new stack
  == stack   ==
  Using SIP RTP CoS mark 5
    5 Audio is at
  5060 Adding codec 0x2 (gsm) to SDP
  Adding codec 0x4 (ulaw) to SDP Adding
  non-codec 0x1 (telephone-event) to SDP
  Reliably Transmitting (no NAT) to
  192.168.10.207:34972: INVITE sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78 SIP/2.0 Via: SIP/2.0/UDP
  192.168.10.159:5060;branch=z9hG4bK7a84f118 Max-Forwards: 70 From: "Vasia"
  <sip:6003@192.168.10.159>;tag=as4933fb4e
  To:
  <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78>
  Contact:
  <sip:6003@192.168.10.159:5060>
  Call-ID:
  091fdd6d2ee3f203503ad55f67fc41be@192.168.10.159:5060
  CSeq: 102 INVITE User-Agent: Asterisk
  PBX 1.8.4.4~dfsg-2ubuntu1 Date: Mon,
  04 Jun 2012 17:18:39 GMT Allow:
  INVITE, ACK, CANCEL, OPTIONS, BYE,
  REFER, SUBSCRIBE, NOTIFY, INFO,
  PUBLISH Supported: replaces, timer
  Content-Type: application/sdp
  Content-Length: 275
 v=0 o=root 222967741 222967741 IN IP4
  192.168.10.159 s=Asterisk PBX 1.8.4.4~dfsg-2ubuntu1 c=IN IP4 192.168.10.159 t=0 0 m=audio 13748 RTP/AVP 3 0 101 a=rtpmap:3 GSM/8000
  a=rtpmap:0 PCMU/8000 a=rtpmap:101
  telephone-event/8000 a=fmtp:101 0-16
  a=ptime:20 a=sendrecv
 
 -- Called 6003
6004
 <--- SIP read from
  UDP:192.168.10.207:34972 ---> SIP/2.0
  100 Trying Via: SIP/2.0/UDP
  192.168.10.159:5060;branch=z9hG4bK7a84f118 To:
  <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78>
  From: "Vasia"
  <sip:6003@192.168.10.159>;tag=as4933fb4e
  Call-ID:
  091fdd6d2ee3f203503ad55f67fc41be@192.168.10.159:5060
  CSeq: 102 INVITE Content-Length: 0
 <------------->
  --- (7 headers 0 lines) ---
 <--- SIP read from
  UDP:192.168.10.207:34972 ---> SIP/2.0
  180 Ringing Via: SIP/2.0/UDP
  192.168.10.159:5060;branch=z9hG4bK7a84f118 Contact:
  <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78>
  To:
  <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78>;tag=cd963d9d
  From:
  "Vasia"<sip:6003@192.168.10.159>;tag=as4933fb4e
  Call-ID:
  091fdd6d2ee3f203503ad55f67fc41be@192.168.10.159:5060
  CSeq: 102 INVITE User-Agent: X-Lite 4
  release 4.1 stamp 63214
  Content-Length: 0
 <------------->
  --- (9 headers 0 lines) ---
      -- SIP/6003-00000005 SIP/6004-0000000b is ringing
    ringing
 <--- Transmitting (no NAT) to
  192.168.10.198:46214 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP
  192.168.10.198:46214;branch=z9hG4bK-d8754z-7f156d5d7b769dcb-1---d8754z-;received=192.168.10.198;rport=46214
  From:
  "6003"<sip:6003@192.168.10.159>;tag=60f21add
  To:
  <sip:6004@192.168.10.159>;tag=as7e4e4ed2
  Call-ID:
  NTFlMGQ4OGU1MjY4YjExZjE0MTRlYWJhZGVkYzU4NTg.
  CSeq: 2 INVITE Server: Asterisk PBX
  1.8.4.4~dfsg-2ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
  SUBSCRIBE, NOTIFY, INFO, PUBLISH
  Supported: replaces, timer Contact:
  <sip:6004@192.168.10.159:5060>
  Content-Length: 0
 <------------>
 <--- SIP read from
  UDP:192.168.10.174:36448 --->
 <------------->
 <--- SIP read from
  UDP:192.168.10.207:34972 ---> SIP/2.0
  200 OK Via: SIP/2.0/UDP
  192.168.10.159:5060;branch=z9hG4bK7a84f118 Contact:
  <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78>
  To:
  <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78>;tag=cd963d9d
  From:
  "Vasia"<sip:6003@192.168.10.159>;tag=as4933fb4e
  Call-ID:
  091fdd6d2ee3f203503ad55f67fc41be@192.168.10.159:5060
  CSeq: 102 INVITE Allow: INVITE, ACK,
  CANCEL, OPTIONS, BYE, REFER, NOTIFY,
  MESSAGE, SUBSCRIBE, INFO Content-Type:
  application/sdp Supported: replaces
  User-Agent: X-Lite 4 release 4.1 stamp
  63214 Content-Length: 383
 v=0 o=- 12983303902905112 1 IN IP4
  192.168.10.207 s=CounterPath X-Lite 4.1 c=IN IP4 192.168.10.207 t=0 0 a=ice-ufrag:600238
  a=ice-pwd:8f5433bfaf37837eda5c392d3c4751cd
  m=audio 63502 RTP/AVP 0 101
  a=rtpmap:101 telephone-event/8000
  a=fmtp:101 0-15 a=sendrecv
  a=candidate:1 1 UDP 659136
  192.168.10.207 63502 typ host a=candidate:1 2 UDP 659134
  192.168.10.207 63503 typ host <------------->
  --- (12 headers 13 lines) --- Found RTP audio format 0 Found RTP audio
  format 101 Found audio description
  format telephone-event for ID 101
  Capabilities: us - 0x6 (gsm|ulaw),
  peer - audio=0x4 (ulaw)/video=0x0
  (nothing)/text=0x0 (nothing), combined
  - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|),
  peer - 0x1 (telephone-event|),
  combined - 0x1 (telephone-event|) Peer
  audio RTP is at port
  192.168.10.207:63502 listroute: hop: <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78>
  setdestination: Parsing
  <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78>
  for address/port to send to
  set_destination: set destination to
  192.168.10.207:34972 Transmitting (no NAT) to 192.168.10.207:34972: ACK
  sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78 SIP/2.0 Via: SIP/2.0/UDP
  192.168.10.159:5060;branch=z9hG4bK7231f3c1 Max-Forwards: 70 From: "Vasia"
  <sip:6003@192.168.10.159>;tag=as4933fb4e
  To:
  <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78>;tag=cd963d9d
  Contact:
  <sip:6003@192.168.10.159:5060>
  Call-ID:
  091fdd6d2ee3f203503ad55f67fc41be@192.168.10.159:5060
  CSeq: 102 ACK User-Agent: Asterisk PBX
  1.8.4.4~dfsg-2ubuntu1 Content-Length: 0
 
 -- SIP/6003-00000005 SIP/6004-0000000b answered SIP/6001-00000004
SIP/6003-0000000a Audio is at 5060
 Adding codec 0x2 (gsm) to SDP Adding
  non-codec 0x1 (telephone-event) to SDP
 <--- Reliably Transmitting (no NAT) to
  192.168.10.198:46214 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP
  192.168.10.198:46214;branch=z9hG4bK-d8754z-7f156d5d7b769dcb-1---d8754z-;received=192.168.10.198;rport=46214
  From:
  "6003"<sip:6003@192.168.10.159>;tag=60f21add
  To:
  <sip:6004@192.168.10.159>;tag=as7e4e4ed2
  Call-ID:
  NTFlMGQ4OGU1MjY4YjExZjE0MTRlYWJhZGVkYzU4NTg.
  CSeq: 2 INVITE Server: Asterisk PBX
  1.8.4.4~dfsg-2ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
  SUBSCRIBE, NOTIFY, INFO, PUBLISH
  Supported: replaces, timer Contact:
  <sip:6004@192.168.10.159:5060>
  Content-Type: application/sdp
  Content-Length: 253
 v=0 o=root 1426165971 1426165971 IN
  IP4 192.168.10.159 s=Asterisk PBX
  1.8.4.4~dfsg-2ubuntu1 c=IN IP4 192.168.10.159 t=0 0 m=audio 14956 RTP/AVP 3 101 a=rtpmap:3 GSM/8000
  a=rtpmap:101 telephone-event/8000
  a=fmtp:101 0-16 a=ptime:20 a=sendrecv
 <------------>
 <--- SIP read from
  UDP:192.168.10.198:46214 ---> ACK
  sip:6004@192.168.10.159:5060 SIP/2.0
  Via: SIP/2.0/UDP
  192.168.10.198:46214;branch=z9hG4bK-d8754z-14e86b68c9eccc31-1---d8754z-;rport
  Max-Forwards: 70 Contact:
  <sip:6003@192.168.10.198:46214> To:
  <sip:6004@192.168.10.159>;tag=as7e4e4ed2
  From:
  "6003"<sip:6003@192.168.10.159>;tag=60f21add
  Call-ID:
  NTFlMGQ4OGU1MjY4YjExZjE0MTRlYWJhZGVkYzU4NTg.
  CSeq: 2 ACK User-Agent: X-Lite 4
  release 4.1 stamp 63214 Authorization:
  Digest
  username="6003",realm="asterisk",nonce="4e427875",uri="sip:6004@192.168.10.159",response="6b03ab1b19a09d4b7e42a9e106fd8463",algorithm=MD5
  Content-Length: 0
 <------------->
  --- (11 headers 0 lines) --- [Jun  4 19:58:56] NOTICE[17174]: res_rtp_asterisk.c:2190 ast_rtp_read: 20:18:42] NOTICE[17402]:
  resrtpasterisk.c:2190 astrtpread:
  Unknown RTP codec 126 received from '192.168.10.198:65162'
from
  '192.168.10.207:63502' [Jun  4 19:58:56] NOTICE[17174]: res_rtp_asterisk.c:2190 ast_rtp_read: 4
  20:18:42] NOTICE[17402]:
  resrtpasterisk.c:2190 astrtpread:
  Unknown RTP codec 126 received from '192.168.10.198:65162'
from
  '192.168.10.207:63502' [Jun  4 19:58:56] NOTICE[17174]: res_rtp_asterisk.c:2190 ast_rtp_read: 4
  20:18:42] NOTICE[17402]:
  resrtpasterisk.c:2190 astrtpread:
  Unknown RTP codec 126 received from '192.168.10.198:65162'
from
  '192.168.10.207:63502' [Jun  4 19:59:06] NOTICE[17174]: res_rtp_asterisk.c:2190 ast_rtp_read: 4
  20:18:42] NOTICE[17402]:
  resrtpasterisk.c:2190 astrtpread:
  Unknown RTP codec 126 received from '192.168.10.198:65162'
  == Using SIP RTP CoS mark 5
    -- Executing [6004@DLPN_offce:1] Dial("SIP/6003-00000006", "SIP/6004,15,t") in new stack
  == Using SIP RTP CoS mark 5
    -- Called 6004
    -- Started music on hold, class 'default', on SIP/6001-00000004
    -- SIP/6004-00000007 is ringing
from
  '192.168.10.198:53280' [Jun  4 19:59:16] NOTICE[17174]: res_rtp_asterisk.c:2190 ast_rtp_read: 4
  20:18:42] NOTICE[17402]:
  resrtpasterisk.c:2190 astrtpread:
  Unknown RTP codec 126 received from '(null)'
    -- SIP/6004-00000007 answered SIP/6003-00000006
from
  '192.168.10.198:53280' [Jun  4 19:59:20] NOTICE[17178]: res_rtp_asterisk.c:2190 ast_rtp_read: 4
  20:18:42] NOTICE[17402]:
  resrtpasterisk.c:2190 astrtpread:
  Unknown RTP codec 126 received from '192.168.10.207:60596'
from
  '192.168.10.198:53280' [Jun  4 19:59:20] NOTICE[17178]: res_rtp_asterisk.c:2190 ast_rtp_read: 4
  20:18:45] NOTICE[17397]:
  resrtpasterisk.c:2190 astrtpread:
  Unknown RTP codec 126 received from '192.168.10.207:60596'
[Jun  4 19:59:20] NOTICE[17178]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP codec 126 received from '192.168.10.207:60596'
[Jun  4 19:59:20] NOTICE[17178]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP codec 126 received from '192.168.10.198:62242'
[Jun  4 19:59:20] NOTICE[17178]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP codec 126 received from '192.168.10.198:62242'
[Jun  4 19:59:20] NOTICE[17178]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP codec 126 received from '192.168.10.198:62242'
from
  '(null)'
 <--- SIP read from
  UDP:192.168.10.198:46214 ---> BYE
  sip:6004@192.168.10.159:5060 SIP/2.0
  Via: SIP/2.0/UDP
  192.168.10.198:46214;branch=z9hG4bK-d8754z-7c4e33c7a54be0df-1---d8754z-;rport
  Max-Forwards: 70 Contact:
  <sip:6003@192.168.10.198:46214> To:
  <sip:6004@192.168.10.159>;tag=as7e4e4ed2
  From:
  "6003"<sip:6003@192.168.10.159>;tag=60f21add
  Call-ID:
  NTFlMGQ4OGU1MjY4YjExZjE0MTRlYWJhZGVkYzU4NTg.
  CSeq: 3 BYE User-Agent: X-Lite 4
  release 4.1 stamp 63214 Authorization:
  Digest
  username="6003",realm="asterisk",nonce="4e427875",uri="sip:6004@192.168.10.159:5060",response="c818868b00190add024dc8cce0db9b9f",algorithm=MD5
  Content-Length: 0
 <------------->
  --- (11 headers 0 lines) --- Sending to 192.168.10.198:46214 (no NAT)
  Scheduling destruction of SIP dialog
  'NTFlMGQ4OGU1MjY4YjExZjE0MTRlYWJhZGVkYzU4NTg.'
  in 32000 ms (Method: BYE)
 <--- Transmitting (no NAT) to
  192.168.10.198:46214 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP
  192.168.10.198:46214;branch=z9hG4bK-d8754z-7c4e33c7a54be0df-1---d8754z-;received=192.168.10.198;rport=46214
  From:
  "6003"<sip:6003@192.168.10.159>;tag=60f21add
  To:
  <sip:6004@192.168.10.159>;tag=as7e4e4ed2
  Call-ID:
  NTFlMGQ4OGU1MjY4YjExZjE0MTRlYWJhZGVkYzU4NTg.
  CSeq: 3 BYE Server: Asterisk PBX
  1.8.4.4~dfsg-2ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
  SUBSCRIBE, NOTIFY, INFO, PUBLISH
  Supported: replaces, timer
  Content-Length: 0
 <------------> Scheduling destruction
  of SIP dialog
  '091fdd6d2ee3f203503ad55f67fc41be@192.168.10.159:5060'
  in 32000 ms (Method: INVITE)
  setdestination: Parsing
  <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78>
  for address/port to send to
  setdestination: set destination to
  192.168.10.207:34972 Reliably Transmitting (no NAT) to
  192.168.10.207:34972: BYE sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78 SIP/2.0 Via: SIP/2.0/UDP
  192.168.10.159:5060;branch=z9hG4bK2b42cb2c Max-Forwards: 70 From: "Vasia"
  <sip:6003@192.168.10.159>;tag=as4933fb4e
  To:
  <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78>;tag=cd963d9d
  Call-ID:
  091fdd6d2ee3f203503ad55f67fc41be@192.168.10.159:5060
  CSeq: 103 BYE User-Agent: Asterisk PBX
  1.8.4.4~dfsg-2ubuntu1 X-Asterisk-HangupCause: Normal
  Clearing X-Asterisk-HangupCauseCode:
  16 Content-Length: 0
 ---   == Spawn extension (DLPN_offce, 6004, 1) exited non-zero on 'SIP/6003-00000006'
on
  'SIP/6003-0000000a'
 <--- SIP read from
  UDP:192.168.10.207:34972 ---> SIP/2.0
  200 OK Via: SIP/2.0/UDP
  192.168.10.159:5060;branch=z9hG4bK2b42cb2c Contact:
  <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78>
  To:
  <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78>;tag=cd963d9d
  From:
  "Vasia"<sip:6003@192.168.10.159>;tag=as4933fb4e
  Call-ID:
  091fdd6d2ee3f203503ad55f67fc41be@192.168.10.159:5060
  CSeq: 103 BYE User-Agent: X-Lite 4
  release 4.1 stamp 63214
  Content-Length: 0
 <------------->
  --- (9 headers 0 lines) --- Really destroying SIP dialog
  '091fdd6d2ee3f203503ad55f67fc41be@192.168.10.159:5060'
  Method: INVITE
 <--- SIP read from
  UDP:192.168.10.207:34972 --->
 <-------------> [Jun  4 19:59:26] NOTICE[17174]: res_rtp_asterisk.c:2190 ast_rtp_read: 20:18:55]
  NOTICE[17397]: resrtpasterisk.c:2190
  astrtpread: Unknown RTP codec 126 126
  received from '(null)'
[Jun  4 19:59:37] NOTICE[17174]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP codec 126 received from '(null)'
[Jun  4 19:59:47] NOTICE[17174]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP codec 126 received from '(null)'

'(null)'

<--- SIP read from UDP:192.168.10.198:46214 ---> BYE sip:6001@192.168.10.159:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.198:46214;branch=z9hG4bK-d8754z-868924b1ef488aec-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:6003@192.168.10.198:46214;rinstance=4bb76a0d9c1c8e4a> To: "Gena"<sip:6001@192.168.10.159>;tag=as118dd5df From: <sip:6003@192.168.10.198:46214;rinstance=4bb76a0d9c1c8e4a>;tag=d0859c1e Call-ID: 1cca2ca62dad19844165317a4333f6a6@192.168.10.159:5060 CSeq: 3 BYE User-Agent: X-Lite 4 release 4.1 stamp 63214 Content-Length: 0

<-------------> --- (10 headers 0 lines) --- Sending to 192.168.10.198:46214 (no NAT) Scheduling destruction of SIP dialog '1cca2ca62dad19844165317a4333f6a6@192.168.10.159:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 192.168.10.198:46214 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.198:46214;branch=z9hG4bK-d8754z-868924b1ef488aec-1---d8754z-;received=192.168.10.198;rport=46214 From: <sip:6003@192.168.10.198:46214;rinstance=4bb76a0d9c1c8e4a>;tag=d0859c1e To: "Gena"<sip:6001@192.168.10.159>;tag=as118dd5df Call-ID: 1cca2ca62dad19844165317a4333f6a6@192.168.10.159:5060 CSeq: 3 BYE Server: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0

<------------> == Spawn extension (DLPNoffce, 6003, 1) exited non-zero on 'SIP/6001-00000008' -- Stopped music on hold on SIP/6001-00000008 Scheduling destruction of SIP dialog 'NTQ5ZjFkNmRmYmMyMWQ5OWEwOTY0ZGQyMzMxOWIyYWU.' in 32000 ms (Method: ACK) setdestination: Parsing <sip:6001@192.168.10.174:36448> for address/port to send to set_destination: set destination to 192.168.10.174:36448 Reliably Transmitting (no NAT) to 192.168.10.174:36448: BYE sip:6001@192.168.10.174:36448 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.159:5060;branch=z9hG4bK5e15e25f;rport Max-Forwards: 70 From: <sip:6003@192.168.10.159>;tag=as51ebf86d To: "6001"<sip:6001@192.168.10.159>;tag=eebbd81c Call-ID: NTQ5ZjFkNmRmYmMyMWQ5OWEwOTY0ZGQyMzMxOWIyYWU. CSeq: 102 BYE User-Agent: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1 Proxy-Authorization: Digest username="6001", realm="asterisk", algorithm=MD5, uri="192.168.10.159", nonce="", response="17abd40c9e03a4315ceae5c5c945435b" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0

<--- SIP read from UDP:192.168.10.174:36448 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.159:5060;branch=z9hG4bK5e15e25f;rport=5060 Contact: <sip:6001@192.168.10.174:36448> To: "6001"<sip:6001@192.168.10.159>;tag=eebbd81c From: <sip:6003@192.168.10.159>;tag=as51ebf86d Call-ID: NTQ5ZjFkNmRmYmMyMWQ5OWEwOTY0ZGQyMzMxOWIyYWU. CSeq: 102 BYE User-Agent: X-Lite 4 release 4.1 stamp 63214 Content-Length: 0Content-Length: 0

Content-Length: 0

Переадресация звонков в Asterisk не работает

Необходимо сделать перенаправление звонков.

Вот мой диалплан:

exten => 6004,1,Dial(SIP/6004,15,t)
exten => 6003,1,Dial(SIP/6003,15,t)
exten => 6001,1,Dial(SIP/6001,15,t)

Вот мой featuremap

[featuremap]
blindxfer => #1 ; Blind transfer (default is #) -- Make sure to set the T and/or t option in the Dial() or Queue() app call!
disconnect => *0 ; Disconnect (default is *) -- Make sure to set the H and/or h option in the Dial() or Queue() app call!
automon => *1 ; One Touch Record a.k.a. Touch Monitor -- Make sure to set the W and/or w option in the Dial() or Queue() app call!
atxfer => *2 ; Attended transfer -- Make sure to set the T and/or t option in the Dial() or Queue() app call!
parkcall => #72 ; Park call (one step parking) -- Make sure to set the K and/or k option in the Dial() app call!
automixmon => *3 ; One Touch Record a.k.a. Touch MixMonitor -- Make sure to set the X and/or x option in the Dial() or Queue() app call!

*2 почему то вообще не работает. Также не работает перенапраление, если установить blindxfer => #.

В идеале, яхочу чтобы работало и #1 и *2. Помогите разобраться пожалуста.

root@ws-053:/etc/asterisk$ grep canreinvite *.*
extensions.conf:canreinvite=no
mgcp.conf:;canreinvite = 1
users.conf:canreinvite = no
users.conf:canreinvite = no
users.conf:canreinvite = no

Вот часть лога: Content-Length: 0

Content-Length: 0> Content-Length: 0

<-------------> --- (10 headers 0 lines) ---

<--- SIP read from UDP:192.168.10.198:46214 ---> INVITE sip:6004@192.168.10.159 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.198:46214;branch=z9hG4bK-d8754z-7f156d5d7b769dcb-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:6003@192.168.10.198:46214> To: <sip:6004@192.168.10.159> From: "6003"<sip:6003@192.168.10.159>;tag=60f21add Call-ID: NTFlMGQ4OGU1MjY4YjExZjE0MTRlYWJhZGVkYzU4NTg. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite 4 release 4.1 stamp 63214 Authorization: Digest username="6003",realm="asterisk",nonce="4e427875",uri="sip:6004@192.168.10.159",response="6b03ab1b19a09d4b7e42a9e106fd8463",algorithm=MD5 Content-Length: 383

v=0 o=- 12983303904982203 1 IN IP4 192.168.10.198 s=CounterPath X-Lite 4.1 c=IN IP4 192.168.10.198 t=0 0 a=ice-ufrag:1f7bf3 a=ice-pwd:43798db74704cb3c028ed1d5fe8b3fd3 m=audio 53280 RTP/AVP 3 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:1 1 UDP 659136 192.168.10.198 53280 typ host a=candidate:1 2 UDP 659134 192.168.10.198 53281 typ host <-------------> --- (14 headers 13 lines) --- Sending to 192.168.10.198:46214 (no NAT) Using INVITE request as basis request - NTFlMGQ4OGU1MjY4YjExZjE0MTRlYWJhZGVkYzU4NTg. Found peer '6003' for '6003' from 192.168.10.198:46214 == Using SIP RTP CoS mark 5 Found RTP audio format 3 Found RTP audio format 101 Found audio description format telephone-event for ID 101 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x2 (gsm)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x2 (gsm) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.10.198:53280 Looking for 6004 in DLPNoffce (domain 192.168.10.159) listroute: hop: <sip:6003@192.168.10.198:46214>

<--- Transmitting (no NAT) to 192.168.10.198:46214 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.198:46214;branch=z9hG4bK-d8754z-7f156d5d7b769dcb-1---d8754z-;received=192.168.10.198;rport=46214 From: "6003"<sip:6003@192.168.10.159>;tag=60f21add To: <sip:6004@192.168.10.159> Call-ID: NTFlMGQ4OGU1MjY4YjExZjE0MTRlYWJhZGVkYzU4NTg. CSeq: 2 INVITE Server: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:6004@192.168.10.159:5060> Content-Length: 0

<------------> -- Executing [6004@DLPN_offce:1] Dial("SIP/6003-0000000a", "SIP/6004,15,t") in new stack == Using SIP RTP CoS mark 5 Audio is at 5060 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.10.207:34972: INVITE sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.159:5060;branch=z9hG4bK7a84f118 Max-Forwards: 70 From: "Vasia" <sip:6003@192.168.10.159>;tag=as4933fb4e To: <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78> Contact: <sip:6003@192.168.10.159:5060> Call-ID: 091fdd6d2ee3f203503ad55f67fc41be@192.168.10.159:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1 Date: Mon, 04 Jun 2012 17:18:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 275

v=0 o=root 222967741 222967741 IN IP4 192.168.10.159 s=Asterisk PBX 1.8.4.4~dfsg-2ubuntu1 c=IN IP4 192.168.10.159 t=0 0 m=audio 13748 RTP/AVP 3 0 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv
-- Called 6004
<--- SIP read from UDP:192.168.10.207:34972 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.159:5060;branch=z9hG4bK7a84f118 To: <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78> From: "Vasia" <sip:6003@192.168.10.159>;tag=as4933fb4e Call-ID: 091fdd6d2ee3f203503ad55f67fc41be@192.168.10.159:5060 CSeq: 102 INVITE Content-Length: 0

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

<--- SIP read from UDP:192.168.10.207:34972 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.159:5060;branch=z9hG4bK7a84f118 Contact: <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78> To: <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78>;tag=cd963d9d From: "Vasia"<sip:6003@192.168.10.159>;tag=as4933fb4e Call-ID: 091fdd6d2ee3f203503ad55f67fc41be@192.168.10.159:5060 CSeq: 102 INVITE User-Agent: X-Lite 4 release 4.1 stamp 63214 Content-Length: 0

<-------------> --- (9 headers 0 lines) --- -- SIP/6004-0000000b is ringing

<--- Transmitting (no NAT) to 192.168.10.198:46214 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.198:46214;branch=z9hG4bK-d8754z-7f156d5d7b769dcb-1---d8754z-;received=192.168.10.198;rport=46214 From: "6003"<sip:6003@192.168.10.159>;tag=60f21add To: <sip:6004@192.168.10.159>;tag=as7e4e4ed2 Call-ID: NTFlMGQ4OGU1MjY4YjExZjE0MTRlYWJhZGVkYzU4NTg. CSeq: 2 INVITE Server: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:6004@192.168.10.159:5060> Content-Length: 0

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

<--- SIP read from UDP:192.168.10.174:36448 --->

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

<--- SIP read from UDP:192.168.10.207:34972 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.159:5060;branch=z9hG4bK7a84f118 Contact: <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78> To: <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78>;tag=cd963d9d From: "Vasia"<sip:6003@192.168.10.159>;tag=as4933fb4e Call-ID: 091fdd6d2ee3f203503ad55f67fc41be@192.168.10.159:5060 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite 4 release 4.1 stamp 63214 Content-Length: 383

v=0 o=- 12983303902905112 1 IN IP4 192.168.10.207 s=CounterPath X-Lite 4.1 c=IN IP4 192.168.10.207 t=0 0 a=ice-ufrag:600238 a=ice-pwd:8f5433bfaf37837eda5c392d3c4751cd m=audio 63502 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:1 1 UDP 659136 192.168.10.207 63502 typ host a=candidate:1 2 UDP 659134 192.168.10.207 63503 typ host <-------------> --- (12 headers 13 lines) --- Found RTP audio format 0 Found RTP audio format 101 Found audio description format telephone-event for ID 101 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.10.207:63502 listroute: hop: <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78> setdestination: Parsing <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78> for address/port to send to set_destination: set destination to 192.168.10.207:34972 Transmitting (no NAT) to 192.168.10.207:34972: ACK sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.159:5060;branch=z9hG4bK7231f3c1 Max-Forwards: 70 From: "Vasia" <sip:6003@192.168.10.159>;tag=as4933fb4e To: <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78>;tag=cd963d9d Contact: <sip:6003@192.168.10.159:5060> Call-ID: 091fdd6d2ee3f203503ad55f67fc41be@192.168.10.159:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1 Content-Length: 0
-- SIP/6004-0000000b answered SIP/6003-0000000a Audio is at 5060
Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.10.198:46214 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.198:46214;branch=z9hG4bK-d8754z-7f156d5d7b769dcb-1---d8754z-;received=192.168.10.198;rport=46214 From: "6003"<sip:6003@192.168.10.159>;tag=60f21add To: <sip:6004@192.168.10.159>;tag=as7e4e4ed2 Call-ID: NTFlMGQ4OGU1MjY4YjExZjE0MTRlYWJhZGVkYzU4NTg. CSeq: 2 INVITE Server: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:6004@192.168.10.159:5060> Content-Type: application/sdp Content-Length: 253

v=0 o=root 1426165971 1426165971 IN IP4 192.168.10.159 s=Asterisk PBX 1.8.4.4~dfsg-2ubuntu1 c=IN IP4 192.168.10.159 t=0 0 m=audio 14956 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv

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

<--- SIP read from UDP:192.168.10.198:46214 ---> ACK sip:6004@192.168.10.159:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.198:46214;branch=z9hG4bK-d8754z-14e86b68c9eccc31-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:6003@192.168.10.198:46214> To: <sip:6004@192.168.10.159>;tag=as7e4e4ed2 From: "6003"<sip:6003@192.168.10.159>;tag=60f21add Call-ID: NTFlMGQ4OGU1MjY4YjExZjE0MTRlYWJhZGVkYzU4NTg. CSeq: 2 ACK User-Agent: X-Lite 4 release 4.1 stamp 63214 Authorization: Digest username="6003",realm="asterisk",nonce="4e427875",uri="sip:6004@192.168.10.159",response="6b03ab1b19a09d4b7e42a9e106fd8463",algorithm=MD5 Content-Length: 0

<-------------> --- (11 headers 0 lines) --- [Jun 4 20:18:42] NOTICE[17402]: resrtpasterisk.c:2190 astrtpread: Unknown RTP codec 126 received from '192.168.10.207:63502' [Jun 4 20:18:42] NOTICE[17402]: resrtpasterisk.c:2190 astrtpread: Unknown RTP codec 126 received from '192.168.10.207:63502' [Jun 4 20:18:42] NOTICE[17402]: resrtpasterisk.c:2190 astrtpread: Unknown RTP codec 126 received from '192.168.10.207:63502' [Jun 4 20:18:42] NOTICE[17402]: resrtpasterisk.c:2190 astrtpread: Unknown RTP codec 126 received from '192.168.10.198:53280' [Jun 4 20:18:42] NOTICE[17402]: resrtpasterisk.c:2190 astrtpread: Unknown RTP codec 126 received from '192.168.10.198:53280' [Jun 4 20:18:42] NOTICE[17402]: resrtpasterisk.c:2190 astrtpread: Unknown RTP codec 126 received from '192.168.10.198:53280' [Jun 4 20:18:45] NOTICE[17397]: resrtpasterisk.c:2190 astrtpread: Unknown RTP codec 126 received from '(null)'

<--- SIP read from UDP:192.168.10.198:46214 ---> BYE sip:6004@192.168.10.159:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.198:46214;branch=z9hG4bK-d8754z-7c4e33c7a54be0df-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:6003@192.168.10.198:46214> To: <sip:6004@192.168.10.159>;tag=as7e4e4ed2 From: "6003"<sip:6003@192.168.10.159>;tag=60f21add Call-ID: NTFlMGQ4OGU1MjY4YjExZjE0MTRlYWJhZGVkYzU4NTg. CSeq: 3 BYE User-Agent: X-Lite 4 release 4.1 stamp 63214 Authorization: Digest username="6003",realm="asterisk",nonce="4e427875",uri="sip:6004@192.168.10.159:5060",response="c818868b00190add024dc8cce0db9b9f",algorithm=MD5 Content-Length: 0

<-------------> --- (11 headers 0 lines) --- Sending to 192.168.10.198:46214 (no NAT) Scheduling destruction of SIP dialog 'NTFlMGQ4OGU1MjY4YjExZjE0MTRlYWJhZGVkYzU4NTg.' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 192.168.10.198:46214 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.198:46214;branch=z9hG4bK-d8754z-7c4e33c7a54be0df-1---d8754z-;received=192.168.10.198;rport=46214 From: "6003"<sip:6003@192.168.10.159>;tag=60f21add To: <sip:6004@192.168.10.159>;tag=as7e4e4ed2 Call-ID: NTFlMGQ4OGU1MjY4YjExZjE0MTRlYWJhZGVkYzU4NTg. CSeq: 3 BYE Server: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0

<------------> Scheduling destruction of SIP dialog '091fdd6d2ee3f203503ad55f67fc41be@192.168.10.159:5060' in 32000 ms (Method: INVITE) setdestination: Parsing <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78> for address/port to send to setdestination: set destination to 192.168.10.207:34972 Reliably Transmitting (no NAT) to 192.168.10.207:34972: BYE sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.159:5060;branch=z9hG4bK2b42cb2c Max-Forwards: 70 From: "Vasia" <sip:6003@192.168.10.159>;tag=as4933fb4e To: <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78>;tag=cd963d9d Call-ID: 091fdd6d2ee3f203503ad55f67fc41be@192.168.10.159:5060 CSeq: 103 BYE User-Agent: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0

--- == Spawn extension (DLPN_offce, 6004, 1) exited non-zero on 'SIP/6003-0000000a'

<--- SIP read from UDP:192.168.10.207:34972 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.159:5060;branch=z9hG4bK2b42cb2c Contact: <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78> To: <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78>;tag=cd963d9d From: "Vasia"<sip:6003@192.168.10.159>;tag=as4933fb4e Call-ID: 091fdd6d2ee3f203503ad55f67fc41be@192.168.10.159:5060 CSeq: 103 BYE User-Agent: X-Lite 4 release 4.1 stamp 63214 Content-Length: 0

<-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '091fdd6d2ee3f203503ad55f67fc41be@192.168.10.159:5060' Method: INVITE

<--- SIP read from UDP:192.168.10.207:34972 --->

<-------------> [Jun 4 20:18:55] NOTICE[17397]: resrtpasterisk.c:2190 astrtpread: Unknown RTP codec 126 received from '(null)'

<--- SIP read from UDP:192.168.10.198:46214 ---> BYE sip:6001@192.168.10.159:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.198:46214;branch=z9hG4bK-d8754z-868924b1ef488aec-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:6003@192.168.10.198:46214;rinstance=4bb76a0d9c1c8e4a> To: "Gena"<sip:6001@192.168.10.159>;tag=as118dd5df From: <sip:6003@192.168.10.198:46214;rinstance=4bb76a0d9c1c8e4a>;tag=d0859c1e Call-ID: 1cca2ca62dad19844165317a4333f6a6@192.168.10.159:5060 CSeq: 3 BYE User-Agent: X-Lite 4 release 4.1 stamp 63214 Content-Length: 0

<-------------> --- (10 headers 0 lines) --- Sending to 192.168.10.198:46214 (no NAT) Scheduling destruction of SIP dialog '1cca2ca62dad19844165317a4333f6a6@192.168.10.159:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 192.168.10.198:46214 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.198:46214;branch=z9hG4bK-d8754z-868924b1ef488aec-1---d8754z-;received=192.168.10.198;rport=46214 From: <sip:6003@192.168.10.198:46214;rinstance=4bb76a0d9c1c8e4a>;tag=d0859c1e To: "Gena"<sip:6001@192.168.10.159>;tag=as118dd5df Call-ID: 1cca2ca62dad19844165317a4333f6a6@192.168.10.159:5060 CSeq: 3 BYE Server: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0

<------------> == Spawn extension (DLPNoffce, 6003, 1) exited non-zero on 'SIP/6001-00000008' -- Stopped music on hold on SIP/6001-00000008 Scheduling destruction of SIP dialog 'NTQ5ZjFkNmRmYmMyMWQ5OWEwOTY0ZGQyMzMxOWIyYWU.' in 32000 ms (Method: ACK) setdestination: Parsing <sip:6001@192.168.10.174:36448> for address/port to send to set_destination: set destination to 192.168.10.174:36448 Reliably Transmitting (no NAT) to 192.168.10.174:36448: BYE sip:6001@192.168.10.174:36448 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.159:5060;branch=z9hG4bK5e15e25f;rport Max-Forwards: 70 From: <sip:6003@192.168.10.159>;tag=as51ebf86d To: "6001"<sip:6001@192.168.10.159>;tag=eebbd81c Call-ID: NTQ5ZjFkNmRmYmMyMWQ5OWEwOTY0ZGQyMzMxOWIyYWU. CSeq: 102 BYE User-Agent: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1 Proxy-Authorization: Digest username="6001", realm="asterisk", algorithm=MD5, uri="192.168.10.159", nonce="", response="17abd40c9e03a4315ceae5c5c945435b" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0

<--- SIP read from UDP:192.168.10.174:36448 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.159:5060;branch=z9hG4bK5e15e25f;rport=5060 Contact: <sip:6001@192.168.10.174:36448> To: "6001"<sip:6001@192.168.10.159>;tag=eebbd81c From: <sip:6003@192.168.10.159>;tag=as51ebf86d Call-ID: NTQ5ZjFkNmRmYmMyMWQ5OWEwOTY0ZGQyMzMxOWIyYWU. CSeq: 102 BYE User-Agent: X-Lite 4 release 4.1 stamp 63214 Content-Length: 0Content-Length: 0

Content-Length: 0

Переадресация звонков в Asterisk не работает

Необходимо сделать перенаправление звонков.

Вот мой диалплан:

exten => 6004,1,Dial(SIP/6004,15,t)
exten => 6003,1,Dial(SIP/6003,15,t)
exten => 6001,1,Dial(SIP/6001,15,t)

Вот мой featuremap

[featuremap]
blindxfer => #1 ; Blind transfer (default is #) -- Make sure to set the T and/or t option in the Dial() or Queue() app call!
disconnect => *0 ; Disconnect (default is *) -- Make sure to set the H and/or h option in the Dial() or Queue() app call!
automon => *1 ; One Touch Record a.k.a. Touch Monitor -- Make sure to set the W and/or w option in the Dial() or Queue() app call!
atxfer => *2 ; Attended transfer -- Make sure to set the T and/or t option in the Dial() or Queue() app call!
parkcall => #72 ; Park call (one step parking) -- Make sure to set the K and/or k option in the Dial() app call!
automixmon => *3 ; One Touch Record a.k.a. Touch MixMonitor -- Make sure to set the X and/or x option in the Dial() or Queue() app call!

*2 почему то вообще не работает. Также не работает перенапраление, если установить blindxfer => #.

В идеале, яхочу чтобы работало и #1 и *2. Помогите разобраться пожалуста.

root@ws-053:/etc/asterisk$ grep canreinvite *.*
extensions.conf:canreinvite=no
mgcp.conf:;canreinvite = 1
users.conf:canreinvite = no
users.conf:canreinvite = no
users.conf:canreinvite = no

Вот часть лога: Content-Length: 0

Content-Length: 0> Content-Length: 0 0 > > <-------------> > --- (10 headers 0 lines) --- --- > > <--- SIP read from > UDP:192.168.10.198:46214 ---> INVITE > sip:6004@192.168.10.159 SIP/2.0 Via: > SIP/2.0/UDP > 192.168.10.198:46214;branch=z9hG4bK-d8754z-7f156d5d7b769dcb-1---d8754z-;rport > Max-Forwards: 70 Contact: > <sip:6003@192.168.10.198:46214> To: > <sip:6004@192.168.10.159> From: > "6003"<sip:6003@192.168.10.159>;tag=60f21add > Call-ID: > NTFlMGQ4OGU1MjY4YjExZjE0MTRlYWJhZGVkYzU4NTg. > CSeq: 2 INVITE Allow: INVITE, ACK, > CANCEL, OPTIONS, BYE, REFER, NOTIFY, > MESSAGE, SUBSCRIBE, INFO Content-Type: > application/sdp Supported: replaces > User-Agent: X-Lite 4 release 4.1 stamp > 63214 Authorization: Digest > username="6003",realm="asterisk",nonce="4e427875",uri="sip:6004@192.168.10.159",response="6b03ab1b19a09d4b7e42a9e106fd8463",algorithm=MD5 > Content-Length: 383 383 > > v=0 o=- 12983303904982203 1 IN IP4 > 192.168.10.198 s=CounterPath X-Lite 4.1 c=IN IP4 192.168.10.198 t=0 0 a=ice-ufrag:1f7bf3 > a=ice-pwd:43798db74704cb3c028ed1d5fe8b3fd3 > m=audio 53280 RTP/AVP 3 101 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 a=sendrecv > a=candidate:1 1 UDP 659136 > 192.168.10.198 53280 typ host a=candidate:1 2 UDP 659134 > 192.168.10.198 53281 typ host <-------------> > --- (14 headers 13 lines) --- Sending to 192.168.10.198:46214 (no NAT) Using > INVITE request as basis request - > NTFlMGQ4OGU1MjY4YjExZjE0MTRlYWJhZGVkYzU4NTg. > Found peer '6003' for '6003' from > 192.168.10.198:46214 == Using SIP RTP CoS mark 5 Found RTP audio format > 3 Found RTP audio format 101 Found > audio description format > telephone-event for ID 101 > Capabilities: us - 0x6 (gsm|ulaw), > peer - audio=0x2 (gsm)/video=0x0 > (nothing)/text=0x0 (nothing), combined > - 0x2 (gsm) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), > peer - 0x1 (telephone-event|), > combined - 0x1 (telephone-event|) Peer > audio RTP is at port > 192.168.10.198:53280 Looking for 6004 in DLPNoffce DLPN_offce (domain 192.168.10.159) listroute: > list_route: hop: <sip:6003@192.168.10.198:46214> > <sip:6003@192.168.10.198:46214> > > <--- Transmitting (no NAT) to > 192.168.10.198:46214 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP > 192.168.10.198:46214;branch=z9hG4bK-d8754z-7f156d5d7b769dcb-1---d8754z-;received=192.168.10.198;rport=46214 > From: > "6003"<sip:6003@192.168.10.159>;tag=60f21add > To: <sip:6004@192.168.10.159> Call-ID: > NTFlMGQ4OGU1MjY4YjExZjE0MTRlYWJhZGVkYzU4NTg. > CSeq: 2 INVITE Server: Asterisk PBX > 1.8.4.4~dfsg-2ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, > SUBSCRIBE, NOTIFY, INFO, PUBLISH > Supported: replaces, timer Contact: > <sip:6004@192.168.10.159:5060> > Content-Length: 0 0 > > > <------------> > -- Executing [6004@DLPN_offce:1] Dial("SIP/6003-0000000a", > "SIP/6004,15,t") in new stack == > Using SIP RTP CoS mark 5 Audio is at > 5060 Adding codec 0x2 (gsm) to SDP > Adding codec 0x4 (ulaw) to SDP Adding > non-codec 0x1 (telephone-event) to SDP > Reliably Transmitting (no NAT) to > 192.168.10.207:34972: INVITE sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78 SIP/2.0 Via: SIP/2.0/UDP > 192.168.10.159:5060;branch=z9hG4bK7a84f118 Max-Forwards: 70 From: "Vasia" > <sip:6003@192.168.10.159>;tag=as4933fb4e > To: > <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78> > Contact: > <sip:6003@192.168.10.159:5060> > Call-ID: > 091fdd6d2ee3f203503ad55f67fc41be@192.168.10.159:5060 > CSeq: 102 INVITE User-Agent: Asterisk > PBX 1.8.4.4~dfsg-2ubuntu1 Date: Mon, > 04 Jun 2012 17:18:39 GMT Allow: > INVITE, ACK, CANCEL, OPTIONS, BYE, > REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH Supported: replaces, timer > Content-Type: application/sdp > Content-Length: 275 275 > > v=0 o=root 222967741 222967741 IN IP4 > 192.168.10.159 s=Asterisk PBX 1.8.4.4~dfsg-2ubuntu1 c=IN IP4 192.168.10.159 t=0 0 m=audio 13748 RTP/AVP 3 0 101 a=rtpmap:3 GSM/8000 > a=rtpmap:0 PCMU/8000 a=rtpmap:101 > telephone-event/8000 a=fmtp:101 0-16 > a=ptime:20 a=sendrecv a=sendrecv > > --- > -- Called 6004 > > <--- SIP read from > UDP:192.168.10.207:34972 ---> SIP/2.0 > 100 Trying Via: SIP/2.0/UDP > 192.168.10.159:5060;branch=z9hG4bK7a84f118 To: > <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78> > From: "Vasia" > <sip:6003@192.168.10.159>;tag=as4933fb4e > Call-ID: > 091fdd6d2ee3f203503ad55f67fc41be@192.168.10.159:5060 > CSeq: 102 INVITE Content-Length: 0 0 > > <-------------> > --- (7 headers 0 lines) --- --- > > <--- SIP read from > UDP:192.168.10.207:34972 ---> SIP/2.0 > 180 Ringing Via: SIP/2.0/UDP > 192.168.10.159:5060;branch=z9hG4bK7a84f118 Contact: > <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78> > To: > <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78>;tag=cd963d9d > From: > "Vasia"<sip:6003@192.168.10.159>;tag=as4933fb4e > Call-ID: > 091fdd6d2ee3f203503ad55f67fc41be@192.168.10.159:5060 > CSeq: 102 INVITE User-Agent: X-Lite 4 > release 4.1 stamp 63214 > Content-Length: 0 0 > > <-------------> > --- (9 headers 0 lines) --- > -- SIP/6004-0000000b is ringing ringing > > <--- Transmitting (no NAT) to > 192.168.10.198:46214 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP > 192.168.10.198:46214;branch=z9hG4bK-d8754z-7f156d5d7b769dcb-1---d8754z-;received=192.168.10.198;rport=46214 > From: > "6003"<sip:6003@192.168.10.159>;tag=60f21add > To: > <sip:6004@192.168.10.159>;tag=as7e4e4ed2 > Call-ID: > NTFlMGQ4OGU1MjY4YjExZjE0MTRlYWJhZGVkYzU4NTg. > CSeq: 2 INVITE Server: Asterisk PBX > 1.8.4.4~dfsg-2ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, > SUBSCRIBE, NOTIFY, INFO, PUBLISH > Supported: replaces, timer Contact: > <sip:6004@192.168.10.159:5060> > Content-Length: 0 <------------> 0 > > > <------------> > > <--- SIP read from > UDP:192.168.10.174:36448 ---> <-------------> ---> > > > <-------------> > > <--- SIP read from > UDP:192.168.10.207:34972 ---> SIP/2.0 > 200 OK Via: SIP/2.0/UDP > 192.168.10.159:5060;branch=z9hG4bK7a84f118 Contact: > <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78> > To: > <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78>;tag=cd963d9d > From: > "Vasia"<sip:6003@192.168.10.159>;tag=as4933fb4e > Call-ID: > 091fdd6d2ee3f203503ad55f67fc41be@192.168.10.159:5060 > CSeq: 102 INVITE Allow: INVITE, ACK, > CANCEL, OPTIONS, BYE, REFER, NOTIFY, > MESSAGE, SUBSCRIBE, INFO Content-Type: > application/sdp Supported: replaces > User-Agent: X-Lite 4 release 4.1 stamp > 63214 Content-Length: 383 383 > > v=0 o=- 12983303902905112 1 IN IP4 > 192.168.10.207 s=CounterPath X-Lite 4.1 c=IN IP4 192.168.10.207 t=0 0 a=ice-ufrag:600238 > a=ice-pwd:8f5433bfaf37837eda5c392d3c4751cd > m=audio 63502 RTP/AVP 0 101 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 a=sendrecv > a=candidate:1 1 UDP 659136 > 192.168.10.207 63502 typ host a=candidate:1 2 UDP 659134 > 192.168.10.207 63503 typ host <-------------> > --- (12 headers 13 lines) --- Found RTP audio format 0 Found RTP audio > format 101 Found audio description > format telephone-event for ID 101 > Capabilities: us - 0x6 (gsm|ulaw), > peer - audio=0x4 (ulaw)/video=0x0 > (nothing)/text=0x0 (nothing), combined > - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), > peer - 0x1 (telephone-event|), > combined - 0x1 (telephone-event|) Peer > audio RTP is at port > 192.168.10.207:63502 listroute: list_route: hop: <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78> setdestination: > set_destination: Parsing > <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78> > for address/port to send to > set_destination: set destination to > 192.168.10.207:34972 Transmitting (no NAT) to 192.168.10.207:34972: ACK > sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78 SIP/2.0 Via: SIP/2.0/UDP > 192.168.10.159:5060;branch=z9hG4bK7231f3c1 Max-Forwards: 70 From: "Vasia" > <sip:6003@192.168.10.159>;tag=as4933fb4e > To: > <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78>;tag=cd963d9d > Contact: > <sip:6003@192.168.10.159:5060> > Call-ID: > 091fdd6d2ee3f203503ad55f67fc41be@192.168.10.159:5060 > CSeq: 102 ACK User-Agent: Asterisk PBX > 1.8.4.4~dfsg-2ubuntu1 Content-Length: 0 0 > > > --- > -- SIP/6004-0000000b answered SIP/6003-0000000a Audio is at 5060 > Adding codec 0x2 (gsm) to SDP Adding > non-codec 0x1 (telephone-event) to SDP SDP > > <--- Reliably Transmitting (no NAT) to > 192.168.10.198:46214 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP > 192.168.10.198:46214;branch=z9hG4bK-d8754z-7f156d5d7b769dcb-1---d8754z-;received=192.168.10.198;rport=46214 > From: > "6003"<sip:6003@192.168.10.159>;tag=60f21add > To: > <sip:6004@192.168.10.159>;tag=as7e4e4ed2 > Call-ID: > NTFlMGQ4OGU1MjY4YjExZjE0MTRlYWJhZGVkYzU4NTg. > CSeq: 2 INVITE Server: Asterisk PBX > 1.8.4.4~dfsg-2ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, > SUBSCRIBE, NOTIFY, INFO, PUBLISH > Supported: replaces, timer Contact: > <sip:6004@192.168.10.159:5060> > Content-Type: application/sdp > Content-Length: 253 253 > > v=0 o=root 1426165971 1426165971 IN > IP4 192.168.10.159 s=Asterisk PBX > 1.8.4.4~dfsg-2ubuntu1 c=IN IP4 192.168.10.159 t=0 0 m=audio 14956 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> a=sendrecv > > <------------> > > <--- SIP read from > UDP:192.168.10.198:46214 ---> ACK > sip:6004@192.168.10.159:5060 SIP/2.0 > Via: SIP/2.0/UDP > 192.168.10.198:46214;branch=z9hG4bK-d8754z-14e86b68c9eccc31-1---d8754z-;rport > Max-Forwards: 70 Contact: > <sip:6003@192.168.10.198:46214> To: > <sip:6004@192.168.10.159>;tag=as7e4e4ed2 > From: > "6003"<sip:6003@192.168.10.159>;tag=60f21add > Call-ID: > NTFlMGQ4OGU1MjY4YjExZjE0MTRlYWJhZGVkYzU4NTg. > CSeq: 2 ACK User-Agent: X-Lite 4 > release 4.1 stamp 63214 Authorization: > Digest > username="6003",realm="asterisk",nonce="4e427875",uri="sip:6004@192.168.10.159",response="6b03ab1b19a09d4b7e42a9e106fd8463",algorithm=MD5 > Content-Length: 0 0 > > <-------------> > --- (11 headers 0 lines) --- [Jun 4 20:18:42] NOTICE[17402]: resrtpasterisk.c:2190 astrtpread: > res_rtp_asterisk.c:2190 ast_rtp_read: > Unknown RTP codec 126 received from > '192.168.10.207:63502' [Jun 4 > 20:18:42] NOTICE[17402]: resrtpasterisk.c:2190 astrtpread: > res_rtp_asterisk.c:2190 ast_rtp_read: > Unknown RTP codec 126 received from > '192.168.10.207:63502' [Jun 4 > 20:18:42] NOTICE[17402]: resrtpasterisk.c:2190 astrtpread: > res_rtp_asterisk.c:2190 ast_rtp_read: > Unknown RTP codec 126 received from > '192.168.10.207:63502' [Jun 4 > 20:18:42] NOTICE[17402]: resrtpasterisk.c:2190 astrtpread: > res_rtp_asterisk.c:2190 ast_rtp_read: > Unknown RTP codec 126 received from > '192.168.10.198:53280' [Jun 4 > 20:18:42] NOTICE[17402]: resrtpasterisk.c:2190 astrtpread: > res_rtp_asterisk.c:2190 ast_rtp_read: > Unknown RTP codec 126 received from > '192.168.10.198:53280' [Jun 4 > 20:18:42] NOTICE[17402]: resrtpasterisk.c:2190 astrtpread: > res_rtp_asterisk.c:2190 ast_rtp_read: > Unknown RTP codec 126 received from > '192.168.10.198:53280' [Jun 4 > 20:18:45] NOTICE[17397]: resrtpasterisk.c:2190 astrtpread: > res_rtp_asterisk.c:2190 ast_rtp_read: > Unknown RTP codec 126 received from '(null)' > '(null)' > > <--- SIP read from > UDP:192.168.10.198:46214 ---> BYE > sip:6004@192.168.10.159:5060 SIP/2.0 > Via: SIP/2.0/UDP > 192.168.10.198:46214;branch=z9hG4bK-d8754z-7c4e33c7a54be0df-1---d8754z-;rport > Max-Forwards: 70 Contact: > <sip:6003@192.168.10.198:46214> To: > <sip:6004@192.168.10.159>;tag=as7e4e4ed2 > From: > "6003"<sip:6003@192.168.10.159>;tag=60f21add > Call-ID: > NTFlMGQ4OGU1MjY4YjExZjE0MTRlYWJhZGVkYzU4NTg. > CSeq: 3 BYE User-Agent: X-Lite 4 > release 4.1 stamp 63214 Authorization: > Digest > username="6003",realm="asterisk",nonce="4e427875",uri="sip:6004@192.168.10.159:5060",response="c818868b00190add024dc8cce0db9b9f",algorithm=MD5 > Content-Length: 0 0 > > <-------------> > --- (11 headers 0 lines) --- Sending to 192.168.10.198:46214 (no NAT) > Scheduling destruction of SIP dialog > 'NTFlMGQ4OGU1MjY4YjExZjE0MTRlYWJhZGVkYzU4NTg.' > in 32000 ms (Method: BYE) BYE) > > <--- Transmitting (no NAT) to > 192.168.10.198:46214 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP > 192.168.10.198:46214;branch=z9hG4bK-d8754z-7c4e33c7a54be0df-1---d8754z-;received=192.168.10.198;rport=46214 > From: > "6003"<sip:6003@192.168.10.159>;tag=60f21add > To: > <sip:6004@192.168.10.159>;tag=as7e4e4ed2 > Call-ID: > NTFlMGQ4OGU1MjY4YjExZjE0MTRlYWJhZGVkYzU4NTg. > CSeq: 3 BYE Server: Asterisk PBX > 1.8.4.4~dfsg-2ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, > SUBSCRIBE, NOTIFY, INFO, PUBLISH > Supported: replaces, timer > Content-Length: 0 0 > > > <------------> Scheduling destruction > of SIP dialog > '091fdd6d2ee3f203503ad55f67fc41be@192.168.10.159:5060' > in 32000 ms (Method: INVITE) setdestination: > set_destination: Parsing > <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78> > for address/port to send to setdestination: > set_destination: set destination to > 192.168.10.207:34972 Reliably Transmitting (no NAT) to > 192.168.10.207:34972: BYE sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78 SIP/2.0 Via: SIP/2.0/UDP > 192.168.10.159:5060;branch=z9hG4bK2b42cb2c Max-Forwards: 70 From: "Vasia" > <sip:6003@192.168.10.159>;tag=as4933fb4e > To: > <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78>;tag=cd963d9d > Call-ID: > 091fdd6d2ee3f203503ad55f67fc41be@192.168.10.159:5060 > CSeq: 103 BYE User-Agent: Asterisk PBX > 1.8.4.4~dfsg-2ubuntu1 X-Asterisk-HangupCause: Normal > Clearing X-Asterisk-HangupCauseCode: > 16 Content-Length: 0 0 > > > --- == Spawn extension (DLPN_offce, 6004, 1) exited non-zero on 'SIP/6003-0000000a' > 'SIP/6003-0000000a' > > <--- SIP read from > UDP:192.168.10.207:34972 ---> SIP/2.0 > 200 OK Via: SIP/2.0/UDP > 192.168.10.159:5060;branch=z9hG4bK2b42cb2c Contact: > <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78> > To: > <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78>;tag=cd963d9d > From: > "Vasia"<sip:6003@192.168.10.159>;tag=as4933fb4e > Call-ID: > 091fdd6d2ee3f203503ad55f67fc41be@192.168.10.159:5060 > CSeq: 103 BYE User-Agent: X-Lite 4 > release 4.1 stamp 63214 > Content-Length: 0 0 > > <-------------> > --- (9 headers 0 lines) --- Really destroying SIP dialog > '091fdd6d2ee3f203503ad55f67fc41be@192.168.10.159:5060' > Method: INVITE INVITE > > <--- SIP read from > UDP:192.168.10.207:34972 ---> ---> > > > <-------------> [Jun 4 20:18:55] > NOTICE[17397]: resrtpasterisk.c:2190 astrtpread: res_rtp_asterisk.c:2190 > ast_rtp_read: Unknown RTP codec 126 > received from '(null)' '(null)' > > <--- SIP read from > UDP:192.168.10.198:46214 ---> BYE > sip:6001@192.168.10.159:5060 SIP/2.0 > Via: SIP/2.0/UDP > 192.168.10.198:46214;branch=z9hG4bK-d8754z-868924b1ef488aec-1---d8754z-;rport > Max-Forwards: 70 Contact: > <sip:6003@192.168.10.198:46214;rinstance=4bb76a0d9c1c8e4a> > To: > "Gena"<sip:6001@192.168.10.159>;tag=as118dd5df > From: > <sip:6003@192.168.10.198:46214;rinstance=4bb76a0d9c1c8e4a>;tag=d0859c1e > Call-ID: > 1cca2ca62dad19844165317a4333f6a6@192.168.10.159:5060 > CSeq: 3 BYE User-Agent: X-Lite 4 > release 4.1 stamp 63214 > Content-Length: 0 0 > > <-------------> > --- (10 headers 0 lines) --- Sending to 192.168.10.198:46214 (no NAT) > Scheduling destruction of SIP dialog > '1cca2ca62dad19844165317a4333f6a6@192.168.10.159:5060' > in 32000 ms (Method: BYE) BYE) > > <--- Transmitting (no NAT) to > 192.168.10.198:46214 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP > 192.168.10.198:46214;branch=z9hG4bK-d8754z-868924b1ef488aec-1---d8754z-;received=192.168.10.198;rport=46214 > From: > <sip:6003@192.168.10.198:46214;rinstance=4bb76a0d9c1c8e4a>;tag=d0859c1e > To: > "Gena"<sip:6001@192.168.10.159>;tag=as118dd5df > Call-ID: > 1cca2ca62dad19844165317a4333f6a6@192.168.10.159:5060 > CSeq: 3 BYE Server: Asterisk PBX > 1.8.4.4~dfsg-2ubuntu1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, > SUBSCRIBE, NOTIFY, INFO, PUBLISH > Supported: replaces, timer > Content-Length: 0 0 > > > <------------> == Spawn extension (DLPNoffce, > (DLPN_offce, 6003, 1) exited non-zero > on 'SIP/6001-00000008' > -- Stopped music on hold on SIP/6001-00000008 Scheduling > destruction of SIP dialog > 'NTQ5ZjFkNmRmYmMyMWQ5OWEwOTY0ZGQyMzMxOWIyYWU.' > in 32000 ms (Method: ACK) setdestination: > set_destination: Parsing > <sip:6001@192.168.10.174:36448> for > address/port to send to > set_destination: set destination to > 192.168.10.174:36448 Reliably Transmitting (no NAT) to > 192.168.10.174:36448: BYE sip:6001@192.168.10.174:36448 SIP/2.0 > Via: SIP/2.0/UDP > 192.168.10.159:5060;branch=z9hG4bK5e15e25f;rport > Max-Forwards: 70 From: > <sip:6003@192.168.10.159>;tag=as51ebf86d > To: > "6001"<sip:6001@192.168.10.159>;tag=eebbd81c > Call-ID: > NTQ5ZjFkNmRmYmMyMWQ5OWEwOTY0ZGQyMzMxOWIyYWU. > CSeq: 102 BYE User-Agent: Asterisk PBX > 1.8.4.4~dfsg-2ubuntu1 Proxy-Authorization: Digest > username="6001", realm="asterisk", > algorithm=MD5, uri="192.168.10.159", > nonce="", > response="17abd40c9e03a4315ceae5c5c945435b" > X-Asterisk-HangupCause: Normal > Clearing X-Asterisk-HangupCauseCode: > 16 Content-Length: 0 0 > > > --- > > <--- SIP read from > UDP:192.168.10.174:36448 ---> SIP/2.0 > 200 OK Via: SIP/2.0/UDP > 192.168.10.159:5060;branch=z9hG4bK5e15e25f;rport=5060 > Contact: > <sip:6001@192.168.10.174:36448> To: > "6001"<sip:6001@192.168.10.159>;tag=eebbd81c > From: > <sip:6003@192.168.10.159>;tag=as51ebf86d > Call-ID: > NTQ5ZjFkNmRmYmMyMWQ5OWEwOTY0ZGQyMzMxOWIyYWU. > CSeq: 102 BYE User-Agent: X-Lite 4 > release 4.1 stamp 63214 > Content-Length: 0Content-Length: 0 0 Content-Length: 00