ded писал(а):а Вы сами то видели?
Научитесь делать SIP debug, смотреть и анализировать SIP диалоги. Иначе - решайте всё дальше с тех. спецами оператора.
Debug видел, но расшифровывать его скажу честно не научился еще.
Помогите его расшифровать.
Код: Выделить всё
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
-- Executing [83852604638@out_office:1] Set("SIP/705-000017b0", "VOLUME(TX)=3") in new stack
-- Executing [83852604638@out_office:2] Set("SIP/705-000017b0", "VOLUME(RX)=3") in new stack
-- Executing [83852604638@out_office:3] Set("SIP/705-000017b0", "fname=/var/spool/asterisk/monitor/office/2014.02.17.14:29_user7(705)--->83852604638") in new stack
-- Executing [83852604638@out_office:4] MixMonitor("SIP/705-000017b0", "/var/spool/asterisk/monitor/office/2014.02.17.14:29_user7(705)--->83852604638.wav") in new stack
-- Executing [83852604638@out_office:5] ChanIsAvail("SIP/705-000017b0", "SIP/ttk&SIP/73832331195&SIP/73832331196,s") in new stack
== Using SIP RTP CoS mark 5
Scheduling destruction of SIP dialog '23754dc760c829e74a978351268956a4@119.173.84.57:5060' in 32000 ms (Method: INVITE)
-- Executing [83852604638@out_office:6] NoOp("SIP/705-000017b0", "Availchan=SIP/ttk-000017b1 and Availstatus=1") in new stack
-- Executing [83852604638@out_office:7] Dial("SIP/705-000017b0", "SIP/ttk/73852604638,60,t,T") in new stack
== Using SIP RTP CoS mark 5
Audio is at 16352
Adding codec 0x100 (g729) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 80.237.13.4:5060:
INVITE sip:73852604638@80.237.13.4 SIP/2.0
Via: SIP/2.0/UDP 119.173.84.57:5060;branch=z9hG4bK0d96bffe;rport
Max-Forwards: 70
From: "user7" <sip:705@119.173.84.57>;tag=as164c32b0
To: <sip:73852604638@80.237.13.4>
Contact: <sip:705@119.173.84.57:5060>
Call-ID: 7b54a500517ccfae215c0ab21a80d249@119.173.84.57:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.23.1
Date: Mon, 17 Feb 2014 07:29:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 337
v=0
o=root 1543553555 1543553555 IN IP4 119.173.84.57
s=Asterisk PBX 1.8.23.1
c=IN IP4 119.173.84.57
t=0 0
m=audio 16352 RTP/AVP 18 8 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called SIP/ttk/73852604638
== Begin MixMonitor Recording SIP/705-000017b0
<--- SIP read from UDP:80.237.13.4:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 119.173.84.57:5060;received=119.173.84.57;branch=z9hG4bK0d96bffe;rport=5060
From: "user7" <sip:705@119.173.84.57>;tag=as164c32b0
To: <sip:73852604638@80.237.13.4>
Call-ID: 7b54a500517ccfae215c0ab21a80d249@119.173.84.57:5060
CSeq: 102 INVITE
<------------->
--- (6 headers 0 lines) ---
<--- SIP read from UDP:80.237.13.4:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 119.173.84.57:5060;received=119.173.84.57;branch=z9hG4bK0d96bffe;rport=5060
From: "user7" <sip:705@119.173.84.57>;tag=as164c32b0
To: <sip:73852604638@80.237.13.4>;tag=0004242100088798
Call-ID: 7b54a500517ccfae215c0ab21a80d249@119.173.84.57:5060
CSeq: 102 INVITE
Content-Length: 194
Content-Type: application/sdp
v=0
o=IWF 290475 535566 IN IP4 80.237.13.4
s=H323 Call
c=IN IP4 80.237.13.4
t=0 0
m=audio 26902 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
<------------->
--- (8 headers 9 lines) ---
list_route: no route
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 80.237.13.4:26902
-- SIP/ttk-000017b2 is making progress passing it to SIP/705-000017b0
<--- SIP read from UDP:80.237.13.4:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 119.173.84.57:5060;received=119.173.84.57;branch=z9hG4bK0d96bffe;rport=5060
From: "user7" <sip:705@119.173.84.57>;tag=as164c32b0
To: <sip:73852604638@80.237.13.4>;tag=0004242100088798
Call-ID: 7b54a500517ccfae215c0ab21a80d249@119.173.84.57:5060
CSeq: 102 INVITE
Content-Length: 194
Content-Type: application/sdp
Contact: <sip:73852604638@80.237.13.4:5060;transport=udp>
v=0
o=IWF 290502 535623 IN IP4 80.237.13.4
s=H323 Call
c=IN IP4 80.237.13.4
t=0 0
m=audio 26902 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
<------------->
--- (9 headers 9 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 80.237.13.4:26902
list_route: hop: <sip:73852604638@80.237.13.4:5060;transport=udp>
set_destination: Parsing <sip:73852604638@80.237.13.4:5060;transport=udp> for address/port to send to
set_destination: set destination to 80.237.13.4:5060
Transmitting (NAT) to 80.237.13.4:5060:
ACK sip:73852604638@80.237.13.4:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 119.173.84.57:5060;branch=z9hG4bK528ae9b7;rport
Max-Forwards: 70
From: "user7" <sip:705@119.173.84.57>;tag=as164c32b0
To: <sip:73852604638@80.237.13.4>;tag=0004242100088798
Contact: <sip:705@119.173.84.57:5060>
Call-ID: 7b54a500517ccfae215c0ab21a80d249@119.173.84.57:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.23.1
Content-Length: 0
---
-- SIP/ttk-000017b2 answered SIP/705-000017b0
set_destination: Parsing <sip:73852604638@80.237.13.4:5060;transport=udp> for address/port to send to
set_destination: set destination to 80.237.13.4:5060
Audio is at 16352
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 80.237.13.4:5060:
INVITE sip:73852604638@80.237.13.4:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 119.173.84.57:5060;branch=z9hG4bK2d18f72b;rport
Max-Forwards: 70
From: "user7" <sip:705@119.173.84.57>;tag=as164c32b0
To: <sip:73852604638@80.237.13.4>;tag=0004242100088798
Contact: <sip:705@119.173.84.57:5060>
Call-ID: 7b54a500517ccfae215c0ab21a80d249@119.173.84.57:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.23.1
Access-URL: <T>;mode=active
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 266
v=0
o=root 1543553555 1543553556 IN IP4 119.173.84.57
s=Asterisk PBX 1.8.23.1
c=IN IP4 119.173.84.57
t=0 0
m=audio 16352 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
<--- SIP read from UDP:80.237.13.4:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 119.173.84.57:5060;received=119.173.84.57;branch=z9hG4bK2d18f72b;rport=5060
From: "user7" <sip:705@119.173.84.57>;tag=as164c32b0
To: <sip:73852604638@80.237.13.4>;tag=0004242100088798
Call-ID: 7b54a500517ccfae215c0ab21a80d249@119.173.84.57:5060
CSeq: 103 INVITE
<------------->
--- (6 headers 0 lines) ---
Really destroying SIP dialog '23754dc760c829e74a978351268956a4@119.173.84.57:5060' Method: INVITE
== Spawn extension (out_office, 89232440904, 5) exited non-zero on 'SIP/711-000017ae'
== MixMonitor close filestream
== End MixMonitor Recording SIP/711-000017ae