Public feed #604089

Mine

public text v1 · immutable
Raw Edit
·published 2007-07-05 09:02 UTC
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Схема такая: [analog phone] <---> {m200} <--e1--> {cisco} <---> {*} <---> [ip phone]
Настройки из sip.conf
 для cisco 
 ----------------------
 [cisco]
 context=pstn-incoming
 type=friend
 host=192.168.11.120
 insecure=very
 dtmfmode=rfc2833
 allow=all
 ----------------------
 для ip phone. Номер 214059
 -------------------------
 [214059]
 context=local-phones
 type=friend
 username=214059
 secret=214214
 host=dynamic
 -------------------------
настройки из extensions.conf
 -------------------------
 [bogon-calls]
 include => lan-phones
 exten => _9.,2,Congestion

 [pstn-incoming]
 include => lan-phones

 [local-phones]
 include => pstn-outbound
 include => lan-phones

 [pstn-outbound]
 exten => _9.,1,Dial(SIP/${EXTEN:1}@cisco,20)
 exten => _9.,2,Congestion

 [lan-phones]
 exten => 214059,1,Dial(SIP/214059,20)
 exten => 214059,2,Hangup
 -------------------------

 Звонок с analog phone (214020) на ip phone (214059) работает.

 Пробуем позвонить с ip phone (214059) на analog phone (214020)
 при это в логах вот что:
 на циску пакеты даже не уходят (на машине с * запущен tcpdump)
***************************************************************
<--- SIP read from 192.168.11.123:5060 --->
REGISTER sip:192.168.11.124 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK443738D0
CSeq: 3361 REGISTER
To: <sip:214059@192.168.11.124>
Expires: 900
From: <sip:214059@192.168.11.124>
Call-ID: 1193929882@192.168.11.123
Content-Length: 0
User-Agent: kphone/4.2
Event: registration
Allow-Events: presence
Contact: <sip:214059@192.168.11.123;transport=udp>;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER"


<------------->
--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.11.123 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.11.123:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK443738D0;received=192.168.11.123
From: <sip:214059@192.168.11.124>
To: <sip:214059@192.168.11.124>
Call-ID: 1193929882@192.168.11.123
CSeq: 3361 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:214059@192.168.11.124>
Content-Length: 0


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

<--- Transmitting (no NAT) to 192.168.11.123:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK443738D0;received=192.168.11.123
From: <sip:214059@192.168.11.124>
To: <sip:214059@192.168.11.124>;tag=as0a53f2fa
Call-ID: 1193929882@192.168.11.123
CSeq: 3361 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="16552637"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1193929882@192.168.11.123' in 32000 ms (Method: REGISTER)

<--- SIP read from 192.168.11.123:5060 --->
REGISTER sip:192.168.11.124 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK7C8CBE93
CSeq: 3362 REGISTER
To: <sip:214059@192.168.11.124>
Authorization: Digest username="214059", realm="asterisk", nonce="16552637", uri="sip:192.168.11.124", cnonce="abcdefghi", nc=00000001, response="2b0aa4d4f308cae79e183370891fd896", opaque="", algorithm="MD5"
Expires: 900
From: <sip:214059@192.168.11.124>
Call-ID: 1193929882@192.168.11.123
Content-Length: 0
User-Agent: kphone/4.2
Event: registration
Allow-Events: presence
Contact: <sip:214059@192.168.11.123;transport=udp>;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER"


<------------->
--- (13 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.11.123 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.11.123:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK7C8CBE93;received=192.168.11.123
From: <sip:214059@192.168.11.124>
To: <sip:214059@192.168.11.124>
Call-ID: 1193929882@192.168.11.123
CSeq: 3362 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:214059@192.168.11.124>
Content-Length: 0


<------------>
    -- Registered SIP '214059' at 192.168.11.123 port 5060 expires 900
    -- Saved useragent "kphone/4.2" for peer 214059

<--- Transmitting (no NAT) to 192.168.11.123:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.11.123;branch=z9hG4bK7C8CBE93;received=192.168.11.123
From: <sip:214059@192.168.11.124>
To: <sip:214059@192.168.11.124>;tag=as0a53f2fa
Call-ID: 1193929882@192.168.11.123
CSeq: 3362 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 900
Contact: <sip:214059@192.168.11.123;transport=udp>;expires=900
Date: Thu, 05 Jul 2007 08:58:52 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1193929882@192.168.11.123' in 32000 ms (Method: REGISTER)

<--- 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=z9hG4bK71DAC439
CSeq: 1580 INVITE
To: <sip:9214020@192.168.11.124>
Content-Type: application/sdp
From: <sip:214059@192.168.11.124>;tag=549EF8FC
Call-ID: 659162755@192.168.11.123
Subject: sip:214059@192.168.11.124
Content-Length: 232
User-Agent: kphone/4.2
Contact: <sip:214059@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 32864 RTP/AVP 0 97 8 3
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 11 lines) ---
Sending to 192.168.11.123 : 5060 (no NAT)
Using INVITE request as basis request - 659162755@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=z9hG4bK71DAC439;received=192.168.11.123
From: <sip:214059@192.168.11.124>;tag=549EF8FC
To: <sip:9214020@192.168.11.124>;tag=as219d6edd
Call-ID: 659162755@192.168.11.123
CSeq: 1580 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="545a29e9"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '659162755@192.168.11.123' in 32000 ms (Method: INVITE)
Found user '214059'

<--- 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=z9hG4bK71DAC439
CSeq: 1580 ACK
To: <sip:9214020@192.168.11.124>;tag=as219d6edd
From: <sip:214059@192.168.11.124>;tag=549EF8FC
Call-ID: 659162755@192.168.11.123
Content-Length: 0
User-Agent: kphone/4.2
Contact: <sip:214059@192.168.11.123;transport=udp>


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

<--- 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=z9hG4bK57360059
CSeq: 1581 INVITE
To: <sip:9214020@192.168.11.124>
Proxy-Authorization: Digest username="214059", realm="asterisk", nonce="545a29e9", uri="sip:9214020@192.168.11.124", cnonce="abcdefghi", nc=00000001, response="fd2f2499de32b7693801f473604d3ef1", opaque="", algorithm="MD5"
Content-Type: application/sdp
From: <sip:214059@192.168.11.124>;tag=549EF8FC
Call-ID: 659162755@192.168.11.123
Subject: sip:214059@192.168.11.124
Content-Length: 232
User-Agent: kphone/4.2
Contact: <sip:214059@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 32864 RTP/AVP 0 97 8 3
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 11 lines) ---
Sending to 192.168.11.123 : 5060 (no NAT)
Using INVITE request as basis request - 659162755@192.168.11.123
Found user '214059'
Found RTP audio format 0
Found RTP audio format 97
Found RTP audio format 8
Found RTP audio format 3
Peer audio RTP is at port 192.168.11.123:32864
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 - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.11.123:32864
Looking for 9214020 in local-phones (domain 192.168.11.124)
list_route: hop: <sip:214059@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=z9hG4bK57360059;received=192.168.11.123
From: <sip:214059@192.168.11.124>;tag=549EF8FC
To: <sip:9214020@192.168.11.124>
Call-ID: 659162755@192.168.11.123
CSeq: 1581 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


<------------>
    -- Executing [9214020@local-phones:1] Dial("SIP/214059-0820dab8", "SIP/214020@cisco|20") in new stack
Audio is at 192.168.11.124 port 10868
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x800 (g726) 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=z9hG4bK0b6b4987;rport
From: "214059" <sip:214059@192.168.11.124>;tag=as59330d19
To: <sip:214020@192.168.11.120>
Contact: <sip:214059@192.168.11.124>
Call-ID: 2559adbd47880cf343a170534a4c68ab@192.168.11.124
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 05 Jul 2007 08:58:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 473

v=0
o=root 4851 4851 IN IP4 192.168.11.124
s=session
c=IN IP4 192.168.11.124
t=0 0
m=audio 10868 RTP/AVP 0 3 8 112 5 10 7 97 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 214020@cisco

<--- SIP read from 192.168.11.120:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK0b6b4987;rport
From: "214059" <sip:214059@192.168.11.124>;tag=as59330d19
To: <sip:214020@192.168.11.120>;tag=4C96F2D1-127A
Date: Thu, 05 Jul 2007 08:58:47 GMT
Call-ID: 2559adbd47880cf343a170534a4c68ab@192.168.11.124
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0


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

<--- SIP read from 192.168.11.120:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK0b6b4987;rport
From: "214059" <sip:214059@192.168.11.124>;tag=as59330d19
To: <sip:214020@192.168.11.120>;tag=4C96F2D1-127A
Date: Thu, 05 Jul 2007 08:58:47 GMT
Call-ID: 2559adbd47880cf343a170534a4c68ab@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=z9hG4bK0b6b4987;rport
From: "214059" <sip:214059@192.168.11.124>;tag=as59330d19
To: <sip:214020@192.168.11.120>;tag=4C96F2D1-127A
Contact: <sip:214059@192.168.11.124>
Call-ID: 2559adbd47880cf343a170534a4c68ab@192.168.11.124
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
    -- SIP/cisco-08212870 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [9214020@local-phones:2] Congestion("SIP/214059-0820dab8", "") in new stack

<--- 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=z9hG4bK57360059;received=192.168.11.123
From: <sip:214059@192.168.11.124>;tag=549EF8FC
To: <sip:9214020@192.168.11.124>;tag=as38a39c3c
Call-ID: 659162755@192.168.11.123
CSeq: 1581 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: Unallocated (unassigned) number
X-Asterisk-HangupCauseCode: 1


<------------>
  == Spawn extension (local-phones, 9214020, 2) exited non-zero on 'SIP/214059-0820dab8'

<--- 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=z9hG4bK57360059
CSeq: 1581 ACK
To: <sip:9214020@192.168.11.124>;tag=as38a39c3c
From: <sip:214059@192.168.11.124>;tag=549EF8FC
Call-ID: 659162755@192.168.11.123
Content-Length: 0
User-Agent: kphone/4.2
Contact: <sip:214059@192.168.11.123;transport=udp>


<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '2559adbd47880cf343a170534a4c68ab@192.168.11.124' Method: INVITE
Really destroying SIP dialog '659162755@192.168.11.123' Method: ACK
*********************************************************************************************