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 => #.
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 DLPN_offce (domain 192.168.10.159)
> list_route: 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 list_route: hop: <sip:6004@192.168.10.207:34972;rinstance=2a8ca938e4461e78>
> 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
>
>
> ---
> -- 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]:
> 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]:
> 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]:
> 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]:
> 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]:
> 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]:
> 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]:
> res_rtp_asterisk.c:2190 ast_rtp_read:
> 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)
> 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 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]: res_rtp_asterisk.c:2190
> ast_rtp_read: 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
> (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)
> 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
>
>
> ---
>
> <--- 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
