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ip телефоны и кнопка hold

Newbies/FAQ Forum 4 сообщений -
#1
Возникла необходимость сделать удержание вызова (чтобы не было слышно говорящего и играла музыка).
При нажатии на кнопку hold на телефонах начинает играть MOH музыка у звонившего, однако, при повторно нажатии hold не происходит возврат вызова или происходит, но слышимость в одну сторону а то и просто не слышно ничего.
Куда копать?
В чем может быть проблема?
кодеки dtmf и прочее везде одинаковые.
Все телефоны находятся в одной сети /24.
#2
не плохо бы показывать логи при звонке - а то телепаты пока в отпуске
#3
Собственно лог. Но я в силу неграмотности ничего в нем не увидел.
SIP/2.0 200 OK
From: "Unknown" ;tag=as16b1ee3d
To: ;tag=zip2x2_399215311-43
Call-ID: 08e347f116cff37f667b42bf4e8e3104@192.168.117.10:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.117.10:5060;branch=z9hG4bK0c2fe8a0
Contact: 1007

Content-Type: application/sdp
Content-Length: 217

v=0
o=ZIP2x2-1007-1.0.8 0 0 IN IP4 192.168.117.138
s=zultys media
c=IN IP4 192.168.117.138
t=0 0
m=audio 10000 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

--- (10 headers 10 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 - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), 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 192.168.117.138:10000
list_route: hop:
set_destination: Parsing for address/port to send to
set_destination: set destination to 192.168.117.138:5060
Transmitting (no NAT) to 192.168.117.138:5060:
ACK sip:1007@192.168.117.138:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.117.10:5060;branch=z9hG4bK001befa8
Max-Forwards: 70
From: "Unknown" ;tag=as16b1ee3d
To: ;tag=zip2x2_399215311-43
Contact:
Call-ID: 08e347f116cff37f667b42bf4e8e3104@192.168.117.10:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Content-Length: 0


---
-- SIP/1007-00000054 answered SIP/1010-00000053
-- Remotely bridging SIP/1010-00000053 and SIP/1007-00000054
set_destination: Parsing for address/port to send to
set_destination: set destination to 192.168.117.138:5060
Audio is at 11748
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.117.138:5060:
INVITE sip:1007@192.168.117.138:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.117.10:5060;branch=z9hG4bK4606b516
Max-Forwards: 70
From: "Unknown" ;tag=as16b1ee3d
To: ;tag=zip2x2_399215311-43
Contact:
Call-ID: 08e347f116cff37f667b42bf4e8e3104@192.168.117.10:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
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: 256

v=0
o=root 1032170587 1032170588 IN IP4 192.168.117.118
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 192.168.117.118
t=0 0
m=audio 5090 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---


SIP/2.0 200 OK
From: "Unknown" ;tag=as16b1ee3d
To: ;tag=zip2x2_399215311-43
Call-ID: 08e347f116cff37f667b42bf4e8e3104@192.168.117.10:5060
CSeq: 103 INVITE
Via: SIP/2.0/UDP 192.168.117.10:5060;branch=z9hG4bK4606b516
Contact: 1007
Allow: INVITE,BYE,CANCEL,ACK,OPTIONS,REGISTER,NOTIFY,MESSAGE,REFER
Content-Type: application/sdp
Content-Length: 217

v=0
o=ZIP2x2-1007-1.0.8 0 0 IN IP4 192.168.117.138
s=zultys media
c=IN IP4 192.168.117.138
t=0 0
m=audio 10000 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

--- (10 headers 10 lines) ---
set_destination: Parsing for address/port to send to
set_destination: set destination to 192.168.117.138:5060
Transmitting (no NAT) to 192.168.117.138:5060:
ACK sip:1007@192.168.117.138:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.117.10:5060;branch=z9hG4bK640708bc
Max-Forwards: 70
From: "Unknown" ;tag=as16b1ee3d
To: ;tag=zip2x2_399215311-43
Contact:
Call-ID: 08e347f116cff37f667b42bf4e8e3104@192.168.117.10:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Content-Length: 0


---
== Using SIP RTP CoS mark 5
-- Executing [052@default:1] Goto("SIP/cisco-00000055", "menu,s,1") in new stack
-- Goto (menu,s,1)
-- Executing [s@menu:1] Answer("SIP/cisco-00000055", "") in new stack
-- Executing [s@menu:2] SetMusicOnHold("SIP/cisco-00000055", "none") in new stack
-- Executing [s@menu:3] Queue("SIP/cisco-00000055", "operators,R") in new stack
-- Started music on hold, class 'default', on SIP/cisco-00000055
== Using SIP RTP CoS mark 5
-- SIP/1004-00000056 is ringing
-- Stopped music on hold on SIP/cisco-00000055


INVITE sip:1010@192.168.117.10 SIP/2.0
From: ;tag=zip2x2_399215311-43
To: "Unknown" ;tag=as16b1ee3d
Call-ID: 08e347f116cff37f667b42bf4e8e3104@192.168.117.10:5060
CSeq: 2 INVITE
Via: SIP/2.0/UDP 192.168.117.138:5060
Contact: 1007
Content-Type: application/sdp
Content-Length: 209

v=0
o=ZIP2x2-1007-1.0.8 0 1 IN IP4 192.168.117.138
s=zultys media
c=IN IP4 0.0.0.0
t=0 0
m=audio 10000 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

--- (9 headers 10 lines) ---
Sending to 192.168.117.138:5060 (no NAT)
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 - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), 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 0.0.0.0:10000


SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.117.138:5060;received=192.168.117.138
From: ;tag=zip2x2_399215311-43
To: "Unknown" ;tag=as16b1ee3d
Call-ID: 08e347f116cff37f667b42bf4e8e3104@192.168.117.10:5060
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Length: 0



Audio is at 11748
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP


SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.117.138:5060;received=192.168.117.138
From: ;tag=zip2x2_399215311-43
To: "Unknown" ;tag=as16b1ee3d
Call-ID: 08e347f116cff37f667b42bf4e8e3104@192.168.117.10:5060
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 256

v=0
o=root 1032170587 1032170589 IN IP4 192.168.117.118
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 192.168.117.118
t=0 0
m=audio 5090 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


-- Started music on hold, class 'default', on SIP/1010-00000053


ACK sip:1010@192.168.117.10 SIP/2.0
From: ;tag=zip2x2_399215311-43
To: "Unknown" ;tag=as16b1ee3d
Call-ID: 08e347f116cff37f667b42bf4e8e3104@192.168.117.10:5060
CSeq: 2 ACK
Via: SIP/2.0/UDP 192.168.117.138:5060
Contact: 1007
Content-Length: 0


--- (8 headers 0 lines) ---
set_destination: Parsing for address/port to send to
set_destination: set destination to 192.168.117.138:5060
Audio is at 11748
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.117.138:5060:
INVITE sip:1007@192.168.117.138:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.117.10:5060;branch=z9hG4bK62fe08f0
Max-Forwards: 70
From: "Unknown" ;tag=as16b1ee3d
To: ;tag=zip2x2_399215311-43
Contact:
Call-ID: 08e347f116cff37f667b42bf4e8e3104@192.168.117.10:5060
CSeq: 104 INVITE
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
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: 256

v=0
o=root 1032170587 1032170590 IN IP4 192.168.117.118
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 192.168.117.118
t=0 0
m=audio 5090 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---


SIP/2.0 200 OK
From: "Unknown" ;tag=as16b1ee3d
To: ;tag=zip2x2_399215311-43
Call-ID: 08e347f116cff37f667b42bf4e8e3104@192.168.117.10:5060
CSeq: 104 INVITE
Via: SIP/2.0/UDP 192.168.117.10:5060;branch=z9hG4bK62fe08f0
Contact: 1007
Allow: INVITE,BYE,CANCEL,ACK,OPTIONS,REGISTER,NOTIFY,MESSAGE,REFER
Content-Type: application/sdp
Content-Length: 217

v=0
o=ZIP2x2-1007-1.0.8 0 0 IN IP4 192.168.117.138
s=zultys media
c=IN IP4 192.168.117.138
t=0 0
m=audio 10000 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

--- (10 headers 10 lines) ---
set_destination: Parsing for address/port to send to
set_destination: set destination to 192.168.117.138:5060
Transmitting (no NAT) to 192.168.117.138:5060:
ACK sip:1007@192.168.117.138:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.117.10:5060;branch=z9hG4bK54011e68
Max-Forwards: 70
From: "Unknown" ;tag=as16b1ee3d
To: ;tag=zip2x2_399215311-43
Contact:
Call-ID: 08e347f116cff37f667b42bf4e8e3104@192.168.117.10:5060
CSeq: 104 ACK
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Content-Length: 0


---
-- SIP/1004-00000056 answered SIP/cisco-00000055
== Begin MixMonitor Recording SIP/cisco-00000055


INVITE sip:1010@192.168.117.10 SIP/2.0
From: ;tag=zip2x2_399215311-43
To: "Unknown" ;tag=as16b1ee3d
Call-ID: 08e347f116cff37f667b42bf4e8e3104@192.168.117.10:5060
CSeq: 3 INVITE
Via: SIP/2.0/UDP 192.168.117.138:5060
Contact: 1007
Allow: INVITE,BYE,CANCEL,ACK,OPTIONS,REGISTER,NOTIFY,MESSAGE,REFER
User-Agent: Zultys ZIP2x2L - 1.0.8
Content-Type: application/sdp
Content-Length: 267

v=0
o=ZIP2x2-1007-1.0.8 0 0 IN IP4 192.168.117.138
s=zultys media
c=IN IP4 192.168.117.138
t=0 0
m=audio 10000 RTP/AVP 8 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

--- (11 headers 12 lines) ---
Sending to 192.168.117.138:5060 (no NAT)


SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.117.138:5060;received=192.168.117.138
From: ;tag=zip2x2_399215311-43
To: "Unknown" ;tag=as16b1ee3d
Call-ID: 08e347f116cff37f667b42bf4e8e3104@192.168.117.10:5060
CSeq: 3 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Length: 0



Audio is at 11748
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP


SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.117.138:5060;received=192.168.117.138
From: ;tag=zip2x2_399215311-43
To: "Unknown" ;tag=as16b1ee3d
Call-ID: 08e347f116cff37f667b42bf4e8e3104@192.168.117.10:5060
CSeq: 3 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 256

v=0
o=root 1032170587 1032170590 IN IP4 192.168.117.118
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 192.168.117.118
t=0 0
m=audio 5090 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv




ACK sip:1010@192.168.117.10 SIP/2.0
From: ;tag=zip2x2_399215311-43
To: "Unknown" ;tag=as16b1ee3d
Call-ID: 08e347f116cff37f667b42bf4e8e3104@192.168.117.10:5060
CSeq: 3 ACK
Via: SIP/2.0/UDP 192.168.117.138:5060
Contact: 1007
Content-Length: 0


--- (8 headers 0 lines) ---


INVITE sip:1010@192.168.117.10 SIP/2.0
From: ;tag=zip2x2_399215311-43
To: "Unknown" ;tag=as16b1ee3d
Call-ID: 08e347f116cff37f667b42bf4e8e3104@192.168.117.10:5060
CSeq: 4 INVITE
Via: SIP/2.0/UDP 192.168.117.138:5060
Contact: 1007
Content-Type: application/sdp
Content-Length: 209

v=0
o=ZIP2x2-1007-1.0.8 0 1 IN IP4 192.168.117.138
s=zultys media
c=IN IP4 0.0.0.0
t=0 0
m=audio 10000 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

--- (9 headers 10 lines) ---
Sending to 192.168.117.138:5060 (no NAT)


SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.117.138:5060;received=192.168.117.138
From: ;tag=zip2x2_399215311-43
To: "Unknown" ;tag=as16b1ee3d
Call-ID: 08e347f116cff37f667b42bf4e8e3104@192.168.117.10:5060
CSeq: 4 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Length: 0



Audio is at 11748
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP


SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.117.138:5060;received=192.168.117.138
From: ;tag=zip2x2_399215311-43
To: "Unknown" ;tag=as16b1ee3d
Call-ID: 08e347f116cff37f667b42bf4e8e3104@192.168.117.10:5060
CSeq: 4 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 256

v=0
o=root 1032170587 1032170590 IN IP4 192.168.117.118
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 192.168.117.118
t=0 0
m=audio 5090 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv




ACK sip:1010@192.168.117.10 SIP/2.0
From: ;tag=zip2x2_399215311-43
To: "Unknown" ;tag=as16b1ee3d
Call-ID: 08e347f116cff37f667b42bf4e8e3104@192.168.117.10:5060
CSeq: 4 ACK
Via: SIP/2.0/UDP 192.168.117.138:5060
Contact: 1007
Content-Length: 0


--- (8 headers 0 lines) ---


INVITE sip:1010@192.168.117.10 SIP/2.0
From: ;tag=zip2x2_399215311-43
To: "Unknown" ;tag=as16b1ee3d
Call-ID: 08e347f116cff37f667b42bf4e8e3104@192.168.117.10:5060
CSeq: 5 INVITE
Via: SIP/2.0/UDP 192.168.117.138:5060
Contact: 1007
Allow: INVITE,BYE,CANCEL,ACK,OPTIONS,REGISTER,NOTIFY,MESSAGE,REFER
User-Agent: Zultys ZIP2x2L - 1.0.8
Content-Type: application/sdp
Content-Length: 267

v=0
o=ZIP2x2-1007-1.0.8 0 0 IN IP4 192.168.117.138
s=zultys media
c=IN IP4 192.168.117.138
t=0 0
m=audio 10000 RTP/AVP 8 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

--- (11 headers 12 lines) ---
Sending to 192.168.117.138:5060 (no NAT)


SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.117.138:5060;received=192.168.117.138
From: ;tag=zip2x2_399215311-43
To: "Unknown" ;tag=as16b1ee3d
Call-ID: 08e347f116cff37f667b42bf4e8e3104@192.168.117.10:5060
CSeq: 5 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Length: 0



Audio is at 11748
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP


SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.117.138:5060;received=192.168.117.138
From: ;tag=zip2x2_399215311-43
To: "Unknown" ;tag=as16b1ee3d
Call-ID: 08e347f116cff37f667b42bf4e8e3104@192.168.117.10:5060
CSeq: 5 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 256

v=0
o=root 1032170587 1032170590 IN IP4 192.168.117.118
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 192.168.117.118
t=0 0
m=audio 5090 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv




ACK sip:1010@192.168.117.10 SIP/2.0
From: ;tag=zip2x2_399215311-43
To: "Unknown" ;tag=as16b1ee3d
Call-ID: 08e347f116cff37f667b42bf4e8e3104@192.168.117.10:5060
CSeq: 5 ACK
Via: SIP/2.0/UDP 192.168.117.138:5060
Contact: 1007
Content-Length: 0


--- (8 headers 0 lines) ---


INVITE sip:1010@192.168.117.10 SIP/2.0
From: ;tag=zip2x2_399215311-43
To: "Unknown" ;tag=as16b1ee3d
Call-ID: 08e347f116cff37f667b42bf4e8e3104@192.168.117.10:5060
CSeq: 6 INVITE
Via: SIP/2.0/UDP 192.168.117.138:5060
Contact: 1007
Content-Type: application/sdp
Content-Length: 209

v=0
o=ZIP2x2-1007-1.0.8 0 2 IN IP4 192.168.117.138
s=zultys media
c=IN IP4 0.0.0.0
t=0 0
m=audio 10000 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

--- (9 headers 10 lines) ---
Sending to 192.168.117.138:5060 (no NAT)
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 - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), 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 0.0.0.0:10000


SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.117.138:5060;received=192.168.117.138
From: ;tag=zip2x2_399215311-43
To: "Unknown" ;tag=as16b1ee3d
Call-ID: 08e347f116cff37f667b42bf4e8e3104@192.168.117.10:5060
CSeq: 6 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Length: 0



Audio is at 11748
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP


SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.117.138:5060;received=192.168.117.138
From: ;tag=zip2x2_399215311-43
To: "Unknown" ;tag=as16b1ee3d
Call-ID: 08e347f116cff37f667b42bf4e8e3104@192.168.117.10:5060
CSeq: 6 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 256

v=0
o=root 1032170587 1032170591 IN IP4 192.168.117.118
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 192.168.117.118
t=0 0
m=audio 5090 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


-- Stopped music on hold on SIP/1010-00000053
-- Started music on hold, class 'default', on SIP/1007-00000054


ACK sip:1010@192.168.117.10 SIP/2.0
From: ;tag=zip2x2_399215311-43
To: "Unknown" ;tag=as16b1ee3d
Call-ID: 08e347f116cff37f667b42bf4e8e3104@192.168.117.10:5060
CSeq: 6 ACK
Via: SIP/2.0/UDP 192.168.117.138:5060
Contact: 1007
Content-Length: 0


--- (8 headers 0 lines) ---
set_destination: Parsing for address/port to send to
set_destination: set destination to 192.168.117.138:5060
Audio is at 11748
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.117.138:5060:
INVITE sip:1007@192.168.117.138:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.117.10:5060;branch=z9hG4bK20ee1855
Max-Forwards: 70
From: "Unknown" ;tag=as16b1ee3d
To: ;tag=zip2x2_399215311-43
Contact:
Call-ID: 08e347f116cff37f667b42bf4e8e3104@192.168.117.10:5060
CSeq: 105 INVITE
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
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: 255

v=0
o=root 1032170587 1032170592 IN IP4 192.168.117.10
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 192.168.117.10
t=0 0
m=audio 11748 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---


SIP/2.0 200 OK
From: "Unknown" ;tag=as16b1ee3d
To: ;tag=zip2x2_399215311-43
Call-ID: 08e347f116cff37f667b42bf4e8e3104@192.168.117.10:5060
CSeq: 105 INVITE
Via: SIP/2.0/UDP 192.168.117.10:5060;branch=z9hG4bK20ee1855
Contact: 1007
Allow: INVITE,BYE,CANCEL,ACK,OPTIONS,REGISTER,NOTIFY,MESSAGE,REFER
Content-Type: application/sdp
Content-Length: 217

v=0
o=ZIP2x2-1007-1.0.8 0 0 IN IP4 192.168.117.138
s=zultys media
c=IN IP4 192.168.117.138
t=0 0
m=audio 10000 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

--- (10 headers 10 lines) ---
set_destination: Parsing for address/port to send to
set_destination: set destination to 192.168.117.138:5060
Transmitting (no NAT) to 192.168.117.138:5060:
ACK sip:1007@192.168.117.138:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.117.10:5060;branch=z9hG4bK3d6e7a42
Max-Forwards: 70
From: "Unknown" ;tag=as16b1ee3d
To: ;tag=zip2x2_399215311-43
Contact:
Call-ID: 08e347f116cff37f667b42bf4e8e3104@192.168.117.10:5060
CSeq: 105 ACK
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Content-Length: 0


---


INVITE sip:1010@192.168.117.10 SIP/2.0
From: ;tag=zip2x2_399215311-43
To: "Unknown" ;tag=as16b1ee3d
Call-ID: 08e347f116cff37f667b42bf4e8e3104@192.168.117.10:5060
CSeq: 7 INVITE
Via: SIP/2.0/UDP 192.168.117.138:5060
Contact: 1007
Allow: INVITE,BYE,CANCEL,ACK,OPTIONS,REGISTER,NOTIFY,MESSAGE,REFER
User-Agent: Zultys ZIP2x2L - 1.0.8
Content-Type: application/sdp
Content-Length: 267

v=0
o=ZIP2x2-1007-1.0.8 0 0 IN IP4 192.168.117.138
s=zultys media
c=IN IP4 192.168.117.138
t=0 0
m=audio 10000 RTP/AVP 8 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

--- (11 headers 12 lines) ---
Sending to 192.168.117.138:5060 (no NAT)


SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.117.138:5060;received=192.168.117.138
From: ;tag=zip2x2_399215311-43
To: "Unknown" ;tag=as16b1ee3d
Call-ID: 08e347f116cff37f667b42bf4e8e3104@192.168.117.10:5060
CSeq: 7 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Length: 0



Audio is at 11748
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP


SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.117.138:5060;received=192.168.117.138
From: ;tag=zip2x2_399215311-43
To: "Unknown" ;tag=as16b1ee3d
Call-ID: 08e347f116cff37f667b42bf4e8e3104@192.168.117.10:5060
CSeq: 7 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 255

v=0
o=root 1032170587 1032170592 IN IP4 192.168.117.10
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 192.168.117.10
t=0 0
m=audio 11748 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv




ACK sip:1010@192.168.117.10 SIP/2.0
From: ;tag=zip2x2_399215311-43
To: "Unknown" ;tag=as16b1ee3d
Call-ID: 08e347f116cff37f667b42bf4e8e3104@192.168.117.10:5060
CSeq: 7 ACK
Via: SIP/2.0/UDP 192.168.117.138:5060
Contact: 1007
Content-Length: 0


--- (8 headers 0 lines) ---
Really destroying SIP dialog '4fd19fc14b1a2f3f1760e34437d5165e@192.168.117.10:5060' Method: INVITE
Really destroying SIP dialog '2609627162-20' Method: REGISTER
voip*CLI> sip set debug off
SIP Debugging Disabled
-- Stopped music on hold on SIP/1007-00000054
== Spawn extension (taxi, 1007, 1) exited non-zero on 'SIP/1010-00000053'
== Using SIP RTP CoS mark 5
-- Executing [052@default:1] Goto("SIP/cisco-00000057", "menu,s,1") in new stack
-- Goto (menu,s,1)
-- Executing [s@menu:1] Answer("SIP/cisco-00000057", "") in new stack
-- Executing [s@menu:2] SetMusicOnHold("SIP/cisco-00000057", "none") in new stack
-- Executing [s@menu:3] Queue("SIP/cisco-00000057", "operators,R") in new stack
-- Started music on hold, class 'default', on SIP/cisco-00000057
== Using SIP RTP CoS mark 5
-- SIP/1008-00000058 is ringing
-- Stopped music on hold on SIP/cisco-00000057
#4
пока всё что вижу = это косячный клиент - который тупо не пишет номер порта в инвайтах

From: "Unknown" ;tag=as16b1ee3d -- видите ? номера порта нету ?
To: ;tag=zip2x2_399215311-43 - а вот тут прописан порт 5060

INVITE sip:1010@192.168.117.10 SIP/2.0 -- снова нет номера порта
From: ;tag=zip2x2_399215311-43

пока как вариант - меняйте клиента - User-Agent: Zultys ZIP2x2L - 1.0.8 - на другово