rendered paste bodyзвонок с 214058 на (9)214020 через кошку
Вывод с астериска sip set debug
bee*CLI>
<--- SIP read from 192.168.11.123:5060 --->
INVITE sip:9214020@192.168.11.124 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK25FE1777
CSeq: 1604 INVITE
To: <sip:9214020@192.168.11.124>
Content-Type: application/sdp
From: <sip:214058@192.168.11.124>;tag=59297E9A
Call-ID: 767094243@192.168.11.123
Subject: sip:214058@192.168.11.124
Content-Length: 262
User-Agent: kphone/4.2
Contact: <sip:214058@192.168.11.123;transport=udp>
v=0
o=username 0 0 IN IP4 192.168.11.123
s=The Funky Flow
c=IN IP4 192.168.11.123
t=0 0
m=audio 32916 RTP/AVP 8 97 3 0
m=audio 32916 RTP/AVP 97 3 0
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
<------------->
--- (11 headers 12 lines) ---
Sending to 192.168.11.123 : 5060 (no NAT)
Using INVITE request as basis request - 767094243@192.168.11.123
<--- Reliably Transmitting (no NAT) to 192.168.11.123:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK25FE1777;received=192.168.11.123
From: <sip:214058@192.168.11.124>;tag=59297E9A
To: <sip:9214020@192.168.11.124>;tag=as6dc44a40
Call-ID: 767094243@192.168.11.123
CSeq: 1604 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="15f9d22c"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '767094243@192.168.11.123' in 32000 ms (Method: INVITE)
Found user '214058'
bee*CLI>
<--- SIP read from 192.168.11.123:5060 --->
ACK sip:9214020@192.168.11.124 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK25FE1777
CSeq: 1604 ACK
To: <sip:9214020@192.168.11.124>;tag=as6dc44a40
From: <sip:214058@192.168.11.124>;tag=59297E9A
Call-ID: 767094243@192.168.11.123
Content-Length: 0
User-Agent: kphone/4.2
Contact: <sip:214058@192.168.11.123;transport=udp>
<------------->
--- (9 headers 0 lines) ---
bee*CLI>
<--- SIP read from 192.168.11.123:5060 --->
INVITE sip:9214020@192.168.11.124 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK5624B9F
CSeq: 1605 INVITE
To: <sip:9214020@192.168.11.124>
Proxy-Authorization: Digest username="214058", realm="asterisk", nonce="15f9d22c", uri="sip:9214020@192.168.11.124", cnonce="abcdefghi", nc=00000001, response="0b64017e9c7cb3b3275bd7f39f14d339", opaque="", algorithm="MD5"
Content-Type: application/sdp
From: <sip:214058@192.168.11.124>;tag=59297E9A
Call-ID: 767094243@192.168.11.123
Subject: sip:214058@192.168.11.124
Content-Length: 262
User-Agent: kphone/4.2
Contact: <sip:214058@192.168.11.123;transport=udp>
v=0
o=username 0 0 IN IP4 192.168.11.123
s=The Funky Flow
c=IN IP4 192.168.11.123
t=0 0
m=audio 32916 RTP/AVP 8 97 3 0
m=audio 32916 RTP/AVP 97 3 0
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
<------------->
--- (12 headers 12 lines) ---
Sending to 192.168.11.123 : 5060 (no NAT)
Using INVITE request as basis request - 767094243@192.168.11.123
Found user '214058'
Found RTP audio format 8
Found RTP audio format 97
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 97
Found RTP audio format 3
Found RTP audio format 0
Peer audio RTP is at port 192.168.11.123:32916
Found description format PCMU for ID 0
Found description format GSM for ID 3
Found description format PCMA for ID 8
Found description format iLBC for ID 97
Capabilities: us - 0x90c (ulaw|alaw|g726|g729), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.11.123:32916
Looking for 9214020 in local-phones (domain 192.168.11.124)
list_route: hop: <sip:214058@192.168.11.123;transport=udp>
<--- Transmitting (no NAT) to 192.168.11.123:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK5624B9F;received=192.168.11.123
From: <sip:214058@192.168.11.124>;tag=59297E9A
To: <sip:9214020@192.168.11.124>
Call-ID: 767094243@192.168.11.123
CSeq: 1605 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:9214020@192.168.11.124>
Content-Length: 0
<------------>
Audio is at 192.168.11.124 port 14596
Adding codec 0x8 (alaw) to SDP
Adding codec 0x800 (g726) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.11.120:5060:
INVITE sip:214020@192.168.11.120 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK460d39aa;rport
From: "214058" <sip:214058@192.168.11.124>;tag=as5ca83e36
To: <sip:214020@192.168.11.120>
Contact: <sip:214058@192.168.11.124>
Call-ID: 2b4510122f2f403b052800453841a143@192.168.11.124
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 10 Jul 2007 17:09:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 356
v=0
o=root 8154 8154 IN IP4 192.168.11.124
s=session
c=IN IP4 192.168.11.124
t=0 0
m=audio 14596 RTP/AVP 8 111 112 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[Jul 10 21:09:28] WARNING[9033]: channel.c:3147 ast_channel_make_compatible: No path to translate from SIP/cisco-081e8ea0(256) to SIP/214058-081e3600(8)
bee*CLI>
<--- SIP read from 192.168.11.120:5060 --->
SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK460d39aa;rport
From: "214058" <sip:214058@192.168.11.124>;tag=as5ca83e36
To: <sip:214020@192.168.11.120>;tag=A070E75-1A98
Date: Tue, 10 Jul 2007 17:09:26 GMT
Call-ID: 2b4510122f2f403b052800453841a143@192.168.11.124
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Transmitting (no NAT) to 192.168.11.120:5060:
ACK sip:214020@192.168.11.120 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK460d39aa;rport
From: "214058" <sip:214058@192.168.11.124>;tag=as5ca83e36
To: <sip:214020@192.168.11.120>;tag=A070E75-1A98
Contact: <sip:214058@192.168.11.124>
Call-ID: 2b4510122f2f403b052800453841a143@192.168.11.124
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Scheduling destruction of SIP dialog '2b4510122f2f403b052800453841a143@192.168.11.124' in 32000 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 192.168.11.120:5060:
BYE sip:214020@192.168.11.120 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK5b83d3ec;rport
From: "214058" <sip:214058@192.168.11.124>;tag=as5ca83e36
To: <sip:214020@192.168.11.120>;tag=A070E75-1A98
Call-ID: 2b4510122f2f403b052800453841a143@192.168.11.124
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
<--- Transmitting (no NAT) to 192.168.11.123:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK5624B9F;received=192.168.11.123
From: <sip:214058@192.168.11.124>;tag=59297E9A
To: <sip:9214020@192.168.11.124>;tag=as576dd9e2
Call-ID: 767094243@192.168.11.123
CSeq: 1605 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:9214020@192.168.11.124>
Content-Length: 0
X-Asterisk-HangupCause: Bearer capability not available
X-Asterisk-HangupCauseCode: 58
<------------>
bee*CLI>
<--- SIP read from 192.168.11.123:5060 --->
ACK sip:9214020@192.168.11.124 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK5624B9F
CSeq: 1605 ACK
To: <sip:9214020@192.168.11.124>;tag=as576dd9e2
From: <sip:214058@192.168.11.124>;tag=59297E9A
Call-ID: 767094243@192.168.11.123
Content-Length: 0
User-Agent: kphone/4.2
Contact: <sip:214058@192.168.11.123;transport=udp>
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '767094243@192.168.11.123' Method: ACK
bee*CLI>
<--- SIP read from 192.168.11.120:5060 --->
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK5b83d3ec;rport
From: "214058" <sip:214058@192.168.11.124>;tag=as5ca83e36
To: <sip:214020@192.168.11.120>;tag=A070E75-1A98
Call-ID: 2b4510122f2f403b052800453841a143@192.168.11.124
CSeq: 103 BYE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
[Jul 10 21:09:28] WARNING[8183]: chan_sip.c:12311 handle_response: Remote host can't match request BYE to call '2b4510122f2f403b052800453841a143@192.168.11.124'. Giving up.
Really destroying SIP dialog '2b4510122f2f403b052800453841a143@192.168.11.124' Method: INVITE
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
sh run
voice class codec 1
codec preference 1 g729r8
codec preference 2 g726r16
codec preference 3 g726r24
codec preference 4 g726r32
codec preference 5 g711alaw
codec preference 6 g711ulaw
dial-peer voice 1 pots
tone ringback alert-no-PI
incoming called-number .T
destination-pattern .
direct-inward-dial
port 1:15
forward-digits all
!
dial-peer voice 2 voip
incoming called-number .
destination-pattern 2140[5-9][5-9]
session protocol sipv2
session target ipv4:192.168.11.124
(как только добавляем сюда )
(codec g711alaw - все прекрасно работает)
debug c кошки пустой :(
sip.conf [cisco]
context=pstn-incoming
type=friend
host=192.168.11.120
insecure=very
dtmfmode=rfc2833
allow=g729
allow=g726
allow=g726aal2
allow=alaw
allow=ulaw
[214058]
context=local-phones
type=friend
username=214058
secret=214058
callerid=214058
host=dynamic
disallow=all
allow=g729
allow=g726
allow=alaw
allow=ulaw
вот.