У меня следующую беда с астериском:
звоню с сип-фона через циску в ТфОП, кладу на сип-фоне трубку, а телефон (тот, который в ТфОП) продолжает звонить.
Помогите, плиз, люди добрые. Или может за денежку кто сможет помочь? Готов отдать многое, чтоб с работы не выгнали.

Конфиги:
=======================
sip.conf:
=======================
[root@SoftSwitch asterisk]# vi sip.conf
[general]
context=default
allowguest=yes
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
autocreatepeer=yes
sipdebug=yes
musicclass=default
; language=ru
canreinvite=no
disallow=all
allow=alaw
allow=g729

[200] ; Grandstream Budge Tone-100
type=friend
host=dynamic
username=200
secret=081268
; canreinvite=yes
nat=yes
context=kvant-4users
callerid="Grandstream Budge Tone-100" <96190>

[201] ; Planet VIP-153T
type=friend
host=dynamic
username=201
secret=081268
nat=yes
context=kvant-4users
callerid="Planet VIP-153T" <96191>

[96192] ; Planet VIP-320
type=friend
host=dynamic
username=96192
secret=081268
; canreinvite=yes
nat=yes
context=kvant-4users
callerid="Planet VIP-320" <96192>

[cisco]
type=friend
host=<cisco_ip>
context=default



===================
extentions.conf:
===================
[general]
static=yes
writeprotect=no
; clearglobalvars=yes
autofallthrough=yes
priorityjumping=no

[globals]
CONSOLE=Console/dsp ; Console interface for demo

[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten => s,1,Dial(${ARG2},20, t) ; Ring the interface, 20 seconds maximum
exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten => s-NOANSWER,1,Voicemail(${ARG1},u) ; If unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(default,s,1) ; If they press #, return to start

exten => s-BUSY,1,Voicemail(${ARG1},b) ; If busy, send to voicemail w/ busy announce
exten => s-BUSY,2,Goto(default,s,1) ; If they press #, return to start

exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer

exten => a,1,VoicemailMain(${ARG1}) ; If they press *, send the user into VoicemailMain

[default]
include => kvant-in

[kvant-in]
; exten => 96190,1, Macro(stdexten,200,SIP/200)
; exten => 96191,1, Macro(stdexten,201,SIP/201)
; exten => 96192,1, Macro(stdexten,202,SIP/202)
exten => 96190, 1, Dial(SIP/200, 30)
exten => 96190, 2, HangUp
exten => 96191, 1, Dial(SIP/201, 30)
exten => 96191, 2, HangUp

[kvant-4users]
exten => _8X.,1,Dial(SIP/cisco/777#${EXTEN},,r)
exten => _XXXXX,1,Dial(SIP/cisco/777#848439${EXTEN},,r)


Логи дальше...

Почему постоянно сбивается форматирование?.. Хм... дубль два:
У меня следующую беда с астериском:
звоню с сип-фона через циску в ТфОП, кладу на сип-фоне трубку, а телефон (тот, который в ТфОП) продолжает звонить.
Помогите, плиз, люди добрые. Или может за денежку кто сможет помочь? Готов отдать многое, чтоб с работы не выгнали.

Конфиги:
=======================
sip.conf:
=======================
[root@SoftSwitch asterisk]# vi sip.conf
[general]
context=default
allowguest=yes
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
autocreatepeer=yes
sipdebug=yes
musicclass=default
; language=ru
canreinvite=no
disallow=all
allow=alaw
allow=g729

[200] ; Grandstream Budge Tone-100
type=friend
host=dynamic
username=200
secret=081268
; canreinvite=yes
nat=yes
context=kvant-4users
callerid="Grandstream Budge Tone-100" <96190>

[201] ; Planet VIP-153T
type=friend
host=dynamic
username=201
secret=081268
nat=yes
context=kvant-4users
callerid="Planet VIP-153T" <96191>

[96192] ; Planet VIP-320
type=friend
host=dynamic
username=96192
secret=081268
; canreinvite=yes
nat=yes
context=kvant-4users
callerid="Planet VIP-320" <96192>

[cisco]
type=friend
host=<cisco_ip>
context=default



===================
extentions.conf:
===================
[general]
static=yes
writeprotect=no
; clearglobalvars=yes
autofallthrough=yes
priorityjumping=no

[globals]
CONSOLE=Console/dsp ; Console interface for demo

[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten => s,1,Dial(${ARG2},20, t) ; Ring the interface, 20 seconds maximum
exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten => s-NOANSWER,1,Voicemail(${ARG1},u) ; If unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(default,s,1) ; If they press #, return to start

exten => s-BUSY,1,Voicemail(${ARG1},b) ; If busy, send to voicemail w/ busy announce
exten => s-BUSY,2,Goto(default,s,1) ; If they press #, return to start

exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer

exten => a,1,VoicemailMain(${ARG1}) ; If they press *, send the user into VoicemailMain

[default]
include => kvant-in

[kvant-in]
; exten => 96190,1, Macro(stdexten,200,SIP/200)
; exten => 96191,1, Macro(stdexten,201,SIP/201)
; exten => 96192,1, Macro(stdexten,202,SIP/202)
exten => 96190, 1, Dial(SIP/200, 30)
exten => 96190, 2, HangUp
exten => 96191, 1, Dial(SIP/201, 30)
exten => 96191, 2, HangUp

[kvant-4users]
exten => _8X.,1,Dial(SIP/cisco/777#${EXTEN},,r)
exten => _XXXXX,1,Dial(SIP/cisco/777#848439${EXTEN},,r)

=====================
Лог звонка (с астериска) Часть 1:
=====================
<--- SIP read from <phone_ip>:10003 --->
INVITE sip:97198@asterisk SIP/2.0
Via: SIP/2.0/UDP <phone_ip_local>:5060;branch=z9hG4bK651f3a57b17ad14975343d191fee26df
From: 201 <sip:201@asterisk>;tag=66617e71
To: 97198 <sip:97198@asterisk>
Call-ID: 7afa752f003284a750a1023b5ff81b48@<phone_ip_local>
CSeq: 1 INVITE
Contact: <sip:201@<phone_ip_local>>
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 272

v=0
o=- 570 570 IN IP4 <phone_ip_local>
s=SIP Session
c=IN IP4 <phone_ip_local>
t=0 0
m=audio 10006 RTP/AVP 18 0 8 4 101
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<------------->
--- (10 headers 13 lines) ---
Sending to <phone_ip_local> : 5060 (no NAT)
Using INVITE request as basis request - 7afa752f003284a750a1023b5ff81b48@<phone_ip_local>
Found user '201' for '201'
SoftSwitch*CLI>
<--- Reliably Transmitting (NAT) to <phone_ip>:10003 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP <phone_ip_local>:5060;branch=z9hG4bK651f3a57b17ad14975343d191fee26df;received=<phone_ip>
From: 201 <sip:201@asterisk>;tag=66617e71
To: 97198 <sip:97198@asterisk>;tag=as0f49cdf7
Call-ID: 7afa752f003284a750a1023b5ff81b48@<phone_ip_local>
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2adbf374"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '7afa752f003284a750a1023b5ff81b48@<phone_ip_local>' in 32000 ms (Method: INVITE)
SoftSwitch*CLI>
<--- SIP read from <phone_ip>:10003 --->
ACK sip:97198@asterisk SIP/2.0
Via: SIP/2.0/UDP <phone_ip_local>:5060;branch=z9hG4bK651f3a57b17ad14975343d191fee26df
From: 201 <sip:201@asterisk>;tag=66617e71
To: 97198 <sip:97198@asterisk>;tag=as0f49cdf7
Call-ID: 7afa752f003284a750a1023b5ff81b48@<phone_ip_local>
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
SoftSwitch*CLI>
<--- SIP read from <phone_ip>:10003 --->
INVITE sip:97198@asterisk SIP/2.0
Via: SIP/2.0/UDP <phone_ip_local>:5060;branch=z9hG4bK1af9aefdc3e6454786132bf12be08938
From: 201 <sip:201@asterisk>;tag=66617e71
To: 97198 <sip:97198@asterisk>
Call-ID: 7afa752f003284a750a1023b5ff81b48@<phone_ip_local>
CSeq: 2 INVITE
Contact: <sip:201@<phone_ip_local>>
Max-Forwards: 70
Authorization: Digest algorithm=MD5,nonce="2adbf374",realm="asterisk",response="b22440f30afcfc57c9b7c54d5d011091",uri="sip:97198@<asterisk_ip>:5060",username="201"
Content-Type: application/sdp
Content-Length: 272

v=0
o=- 570 570 IN IP4 <phone_ip_local>
s=SIP Session
c=IN IP4 <phone_ip_local>
t=0 0
m=audio 10006 RTP/AVP 18 0 8 4 101
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<------------->
--- (11 headers 13 lines) ---
Sending to <phone_ip> : 10003 (NAT)
Using INVITE request as basis request - 7afa752f003284a750a1023b5ff81b48@<phone_ip_local>
Found user '201' for '201'
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 101
Peer audio RTP is at port <phone_ip_local>:10006
Found description format G729 for ID 18
Found description format PCMU for ID 0
Found description format PCMA for ID 8
Found description format G723 for ID 4
Found description format telephone-event for ID 101
Capabilities: us - 0x108 (alaw|g729), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x108 (alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port <phone_ip_local>:10006
Looking for 97198 in kvant-4users (domain asterisk)
list_route: hop: <sip:201@<phone_ip_local>>
SoftSwitch*CLI>
<--- Transmitting (NAT) to <phone_ip>:10003 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP <phone_ip_local>:5060;branch=z9hG4bK1af9aefdc3e6454786132bf12be08938;received=<phone_ip>
From: 201 <sip:201@asterisk>;tag=66617e71
To: 97198 <sip:97198@asterisk>
Call-ID: 7afa752f003284a750a1023b5ff81b48@<phone_ip_local>
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:97198@<asterisk_ip>>
Content-Length: 0


<------------>
Audio is at <asterisk_ip> port 10946
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to <cisco_ip>:5060:
INVITE sip:777#84843997198@<cisco_ip> SIP/2.0
Via: SIP/2.0/UDP <asterisk_ip>:5060;branch=z9hG4bK08598f27;rport
Max-Forwards: 70
From: "Planet VIP-153T" <sip:96191@<asterisk_ip>>;tag=as4a347237
To: <sip:777#84843997198@<cisco_ip>>
Contact: <sip:96191@<asterisk_ip>>
Call-ID: 577b23923a64856229e93f4e5cc00d26@<asterisk_ip>
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 01 Dec 2006 17:26:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 242

v=0
o=root 3100 3100 IN IP4 <asterisk_ip>
s=session
c=IN IP4 <asterisk_ip>
t=0 0
m=audio 10946 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

---
SoftSwitch*CLI>
<--- Transmitting (NAT) to <phone_ip>:10003 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP <phone_ip_local>:5060;branch=z9hG4bK1af9aefdc3e6454786132bf12be08938;received=<phone_ip>
From: 201 <sip:201@asterisk>;tag=66617e71
To: 97198 <sip:97198@asterisk>;tag=as4023aaef
Call-ID: 7afa752f003284a750a1023b5ff81b48@<phone_ip_local>
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:97198@<asterisk_ip>>
Content-Length: 0


<------------>
SoftSwitch*CLI>
<--- SIP read from <cisco_ip>:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP <asterisk_ip>:5060;branch=z9hG4bK08598f27;rport
From: "Planet VIP-153T" <sip:96191@<asterisk_ip>>;tag=as4a347237
To: <sip:777#84843997198@<cisco_ip>>;tag=A979D94-240B
Date: Wed, 03 Mar 1993 01:21:48 GMT
Call-ID: 577b23923a64856229e93f4e5cc00d26@<asterisk_ip>
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
SoftSwitch*CLI>
<--- SIP read from <cisco_ip>:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP <asterisk_ip>:5060;branch=z9hG4bK08598f27;rport
From: "Planet VIP-153T" <sip:96191@<asterisk_ip>>;tag=as4a347237
To: <sip:777#84843997198@<cisco_ip>>;tag=A979D94-240B
Date: Wed, 03 Mar 1993 01:21:48 GMT
Call-ID: 577b23923a64856229e93f4e5cc00d26@<asterisk_ip>
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Contact: <sip:777#84843997198@<cisco_ip>:5060>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Content-Length: 197

v=0
o=CiscoSystemsSIP-GW-UserAgent 4400 4176 IN IP4 <cisco_ip>
s=SIP Call
c=IN IP4 <cisco_ip>
t=0 0
m=audio 18568 RTP/AVP 8
c=IN IP4 <cisco_ip>
a=rtpmap:8 PCMA/8000
a=ptime:20

<------------->
--- (13 headers 9 lines) ---
Found RTP audio format 8
Peer audio RTP is at port <cisco_ip>:18568
Found description format PCMA for ID 8
Capabilities: us - 0x108 (alaw|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port <cisco_ip>:18568
SoftSwitch*CLI>

*********************************************
Дозвонился, в трубке сип-фона идут гудки, телефон в ТфОП звонит
*********************************************

=====================
Лог звонка (с астериска) Часть 2:
=====================

*********************************************
Нажал на сип-фоне на сброс, телефон в ТфОП продолжает звонить
*********************************************
SoftSwitch*CLI>
<--- SIP read from <phone_ip>:10003 --->
CANCEL sip:97198@asterisk SIP/2.0
Via: SIP/2.0/UDP <phone_ip_local>:5060;branch=z9hG4bK1af9aefdc3e6454786132bf12be08938
From: 201 <sip:201@asterisk>;tag=66617e71
To: 97198 <sip:97198@asterisk>
Call-ID: 7afa752f003284a750a1023b5ff81b48@<phone_ip_local>
CSeq: 2 CANCEL
Max-Forwards: 70
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Sending to <phone_ip> : 10003 (NAT)
SoftSwitch*CLI>
<--- Reliably Transmitting (NAT) to <phone_ip>:10003 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP <phone_ip_local>:5060;branch=z9hG4bK1af9aefdc3e6454786132bf12be08938;received=<phone_ip>
From: 201 <sip:201@asterisk>;tag=66617e71
To: 97198 <sip:97198@asterisk>;tag=as4023aaef
Call-ID: 7afa752f003284a750a1023b5ff81b48@<phone_ip_local>
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
SoftSwitch*CLI>
<--- Transmitting (NAT) to <phone_ip>:10003 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP <phone_ip_local>:5060;branch=z9hG4bK1af9aefdc3e6454786132bf12be08938;received=<phone_ip>
From: 201 <sip:201@asterisk>;tag=66617e71
To: 97198 <sip:97198@asterisk>;tag=as4023aaef
Call-ID: 7afa752f003284a750a1023b5ff81b48@<phone_ip_local>
CSeq: 2 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:97198@<asterisk_ip>>
Content-Length: 0


<------------>>
Scheduling destruction of SIP dialog '577b23923a64856229e93f4e5cc00d26@<asterisk_ip>' in 32000 ms (Method: INVITE)
SoftSwitch*CLI>
<--- SIP read from <phone_ip>:10003 --->
ACK sip:97198@<asterisk_ip>:5060 SIP/2.0
Via: SIP/2.0/UDP <phone_ip_local>:5060;branch=z9hG4bK1af9aefdc3e6454786132bf12be08938
From: 201 <sip:201@<asterisk_ip>>;tag=66617e71
To: 97198 <sip:97198@<asterisk_ip>>;tag=as4023aaef
Call-ID: 7afa752f003284a750a1023b5ff81b48@<phone_ip_local>
Contact: <sip:201@<phone_ip_local>:5060>
CSeq: 2 ACK
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '7afa752f003284a750a1023b5ff81b48@<phone_ip_local>' Method: ACK
SoftSwitch*CLI>
Really destroying SIP dialog '3366b30c123c189e22341d5b529fc33f@<phone_ip_local>' Method: REGISTER
SoftSwitch*CLI>

*********************************
Поднял трубку телефона в ТфОП, в нем короткие гудки
*********************************

SoftSwitch*CLI>
<--- SIP read from <cisco_ip>:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP <asterisk_ip>:5060;branch=z9hG4bK08598f27;rport
From: "Planet VIP-153T" <sip:96191@<asterisk_ip>>;tag=as4a347237
To: <sip:777#84843997198@<cisco_ip>>;tag=A979D94-240B
Date: Wed, 03 Mar 1993 01:21:48 GMT
Call-ID: 577b23923a64856229e93f4e5cc00d26@<asterisk_ip>
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
Allow-Events: telephone-event
Contact: <sip:777#84843997198@<cisco_ip>:5060>
Content-Type: application/sdp
Content-Length: 197

v=0
o=CiscoSystemsSIP-GW-UserAgent 4400 4176 IN IP4 <cisco_ip>
s=SIP Call
c=IN IP4 <cisco_ip>
t=0 0
m=audio 18568 RTP/AVP 8
c=IN IP4 <cisco_ip>
a=rtpmap:8 PCMA/8000
a=ptime:20

<------------->
--- (13 headers 9 lines) ---
Found RTP audio format 8
Peer audio RTP is at port <cisco_ip>:18568
Found description format PCMA for ID 8
Capabilities: us - 0x108 (alaw|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port <cisco_ip>:18568
--- set_address_from_contact host '<cisco_ip>'
list_route: hop: <sip:777#84843997198@<cisco_ip>:5060>
set_destination: Parsing <sip:777#84843997198@<cisco_ip>:5060> for address/port to send to
set_destination: set destination to <cisco_ip>, port 5060
Transmitting (no NAT) to <cisco_ip>:5060:
ACK sip:777#84843997198@<cisco_ip>:5060 SIP/2.0
Via: SIP/2.0/UDP <asterisk_ip>:5060;branch=z9hG4bK558a60cc;rport
Max-Forwards: 70
From: "Planet VIP-153T" <sip:96191@<asterisk_ip>>;tag=as4a347237
To: <sip:777#84843997198@<cisco_ip>>;tag=A979D94-240B
Contact: <sip:96191@<asterisk_ip>>
Call-ID: 577b23923a64856229e93f4e5cc00d26@<asterisk_ip>
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0


---
set_destination: Parsing <sip:777#84843997198@<cisco_ip>:5060> for address/port to send to
set_destination: set destination to <cisco_ip>, port 5060
Reliably Transmitting (no NAT) to <cisco_ip>:5060:
BYE sip:777#84843997198@<cisco_ip>:5060 SIP/2.0
Via: SIP/2.0/UDP <asterisk_ip>:5060;branch=z9hG4bK49cee1a4;rport
Max-Forwards: 70
From: "Planet VIP-153T" <sip:96191@<asterisk_ip>>;tag=as4a347237
To: <sip:777#84843997198@<cisco_ip>>;tag=A979D94-240B
Call-ID: 577b23923a64856229e93f4e5cc00d26@<asterisk_ip>
CSeq: 103 BYE
User-Agent: Asterisk PBX
Content-Length: 0


---
Scheduling destruction of SIP dialog '577b23923a64856229e93f4e5cc00d26@<asterisk_ip>' in 32000 ms (Method: INVITE)
SoftSwitch*CLI>
<--- SIP read from <cisco_ip>:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP <asterisk_ip>:5060;branch=z9hG4bK49cee1a4;rport
From: "Planet VIP-153T" <sip:96191@<asterisk_ip>>;tag=as4a347237
To: <sip:777#84843997198@<cisco_ip>>;tag=A979D94-240B
Date: Wed, 03 Mar 1993 01:21:57 GMT
Call-ID: 577b23923a64856229e93f4e5cc00d26@<asterisk_ip>
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
CSeq: 103 BYE


<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '577b23923a64856229e93f4e5cc00d26@<asterisk_ip>' Method: INVITE

=======================
Лог с циски (ccsip messages + isdn q931):
=======================
*Mar 3 01:28:28.227: Received:
INVITE sip:777#84843997198@<cisco_ip> SIP/2.0
Via: SIP/2.0/UDP <asterisk_ip>:5060;branch=z9hG4bK2d177bee;rport
Max-Forwards: 70
From: "Planet VIP-153T" <sip:96191@<asterisk_ip>>;tag=as3ba55f51
To: <sip:777#84843997198@<cisco_ip>>
Contact: <sip:96191@<asterisk_ip>>
Call-ID: 7e19fe9a3a713c1a04a5ae5255fa4ac6@<asterisk_ip>
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 01 Dec 2006 17:33:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 242

v=0
o=root 3100 3100 IN IP4 <asterisk_ip>
s=session
c=IN IP4 <asterisk_ip>
t=0 0
m=audio 10648 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

*Mar 3 01:28:28.235: ISDN Se3/0:15 Q931: TX -> SETUP pd = 8 callref = 0x00A8
Bearer Capability i = 0x8090A3
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA9839F
Exclusive, Channel 31
Calling Party Number i = 0x0080, '96191'
Plan:Unknown, Type:Unknown
Called Party Number i = 0x80, '84843997198'
Plan:Unknown, Type:Unknown
*Mar 3 01:28:28.235: Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP <asterisk_ip>:5060;branch=z9hG4bK2d177bee;rport
From: "Planet VIP-153T" <sip:96191@<asterisk_ip>>;tag=as3ba55f51
To: <sip:777#84843997198@<cisco_ip>>;tag=A9DB754-58F
Date: Wed, 03 Mar 1993 01:28:28 GMT
Call-ID: 7e19fe9a3a713c1a04a5ae5255fa4ac6@<asterisk_ip>
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0



*Mar 3 01:28:28.339: ISDN Se3/0:15 Q931: RX <- SETUP_ACK pd = 8 callref = 0x80A8
Channel ID i = 0xA9839F
Exclusive, Channel 31
*Mar 3 01:28:28.659: ISDN Se3/0:15 Q931: RX <- CALL_PROC pd = 8 callref = 0x80A8
*Mar 3 01:28:28.687: ISDN Se3/0:15 Q931: RX <- ALERTING pd = 8 callref = 0x80A8
Progress Ind i = 0x8182 - Destination address is non-ISDN
*Mar 3 01:28:28.691: Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP <asterisk_ip>:5060;branch=z9hG4bK2d177bee;rport
From: "Planet VIP-153T" <sip:96191@<asterisk_ip>>;tag=as3ba55f51
To: <sip:777#84843997198@<cisco_ip>>;tag=A9DB754-58F
Date: Wed, 03 Mar 1993 01:28:28 GMT
Call-ID: 7e19fe9a3a713c1a04a5ae5255fa4ac6@<asterisk_ip>
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Contact: <sip:777#84843997198@<cisco_ip>:5060>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Content-Length: 197

v=0
o=CiscoSystemsSIP-GW-UserAgent 3548 8774 IN IP4 <cisco_ip>
s=SIP Call
c=IN IP4 <cisco_ip>
t=0 0
m=audio 16520 RTP/AVP 8
c=IN IP4 <cisco_ip>
a=rtpmap:8 PCMA/8000
a=ptime:20

Router#
Router#
********************
Дозвонился, в сип-фоне гудки, телефон в ТфОП звонит
********************
Router#
Router#
*******************
Нажал отмену на сип-фоне, ниче не произошло.
*******************
Router#
Router#
Router#
*******************
Поднял трубку в ТфОП, короткие гудки в ней
*******************
Router#
Router#
*Mar 3 01:28:37.747: ISDN Se3/0:15 Q931: RX <- CONNECT pd = 8 callref = 0x80A8
Progress Ind i = 0x8182 - Destination address is non-ISDN
*Mar 3 01:28:37.751: %ISDN-6-CONNECT: Interface Serial3/0:30 is now connected to 84843997198 N/A
*Mar 3 01:28:37.751: ISDN Se3/0:15 Q931: TX -> CONNECT_ACK pd = 8 callref = 0x00A8
*Mar 3 01:28:37.751: Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP <asterisk_ip>:5060;branch=z9hG4bK2d177bee;rport
From: "Planet VIP-153T" <sip:96191@<asterisk_ip>>;tag=as3ba55f51
To: <sip:777#84843997198@<cisco_ip>>;tag=A9DB754-58F
Date: Wed, 03 Mar 1993 01:28:28 GMT
Call-ID: 7e19fe9a3a713c1a04a5ae5255fa4ac6@<asterisk_ip>
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
Allow-Events: telephone-event
Contact: <sip:777#84843997198@<cisco_ip>:5060>
Content-Type: application/sdp
Content-Length: 197

v=0
o=CiscoSystemsSIP-GW-UserAgent 3548 8774 IN IP4 <cisco_ip>
s=SIP Call
c=IN IP4 <cisco_ip>
t=0 0
m=audio 16520 RTP/AVP 8
c=IN IP4 <cisco_ip>
a=rtpmap:8 PCMA/8000
a=ptime:20

*Mar 3 01:28:37.755: Received:
ACK sip:777#84843997198@<cisco_ip>:5060 SIP/2.0
Via: SIP/2.0/UDP <asterisk_ip>:5060;branch=z9hG4bK13fda3e8;rport
Max-Forwards: 70
From: "Planet VIP-153T" <sip:96191@<asterisk_ip>>;tag=as3ba55f51
To: <sip:777#84843997198@<cisco_ip>>;tag=A9DB754-58F
Contact: <sip:96191@<asterisk_ip>>
Call-ID: 7e19fe9a3a713c1a04a5ae5255fa4ac6@<asterisk_ip>
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0



*Mar 3 01:28:37.755: Received:
BYE sip:777#84843997198@<cisco_ip>:5060 SIP/2.0
Via: SIP/2.0/UDP <asterisk_ip>:5060;branch=z9hG4bK35626ca7;rport
Max-Forwards: 70
From: "Planet VIP-153T" <sip:96191@<asterisk_ip>>;tag=as3ba55f51
To: <sip:777#84843997198@<cisco_ip>>;tag=A9DB754-58F
Call-ID: 7e19fe9a3a713c1a04a5ae5255fa4ac6@<asterisk_ip>
CSeq: 103 BYE
User-Agent: Asterisk PBX
Content-Length: 0



*Mar 3 01:28:37.759: Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP <asterisk_ip>:5060;branch=z9hG4bK35626ca7;rport
From: "Planet VIP-153T" <sip:96191@<asterisk_ip>>;tag=as3ba55f51
To: <sip:777#84843997198@<cisco_ip>>;tag=A9DB754-58F
Date: Wed, 03 Mar 1993 01:28:37 GMT
Call-ID: 7e19fe9a3a713c1a04a5ae5255fa4ac6@<asterisk_ip>
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
CSeq: 103 BYE



*Mar 3 01:28:37.763: ISDN Se3/0:15 Q931: TX -> DISCONNECT pd = 8 callref = 0x00A8
Cause i = 0x8090 - Normal call clearing
*Mar 3 01:28:37.807: ISDN Se3/0:15 Q931: RX <- RELEASE pd = 8 callref = 0x80A8
*Mar 3 01:28:37.811: ISDN Se3/0:15 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x00A8


Насколько я сумел понять эти два лога, то когда я нажимаю отмену на сип-фоне Астериск принимает от сип-фона CANCEL, а на циску ничего не отправляет. Только ответы сип-фону, что принял CANCEL.
От чего так?
Как его научить отправлять CANCEL (или ещё что) на циску после получения CANCEL?

Прошу... нет, не прошу, уже просто умоляю, помогите! Как угодно, хоть из идеологических соображений, хоть из финансовых.

С уважением, ...

1) Почему постоянно сбивается форматирование?.
Когда постишь включай emoticons. У этого форума забавная взаимосвязь между смайлами и разметкой текста :)

2)
[kvant-4users]
exten => _8X.,1,Dial(SIP/cisco/777#${EXTEN},,r)
exten => _8X.,2,Hangup
exten => _XXXXX,1,Dial(SIP/cisco/777#848439${EXTEN},,r)
exten => _XXXXX,2,Hangup

Попробуй так.

[kvant-4users]

exten => _8X.,1,Dial(SIP/cisco/777#${EXTEN},,r)

exten => _8X.,2,Hangup

exten => _XXXXX,1,Dial(SIP/cisco/777#848439${EXTEN},,r)

exten => _XXXXX,2,Hangup 

Уже пробовал :(. Ща попробовал еще раз - тоже самое.
Есть еще какие-нить варианты? Неужели ни у кого больше не было таких проблем?..

А за эмотиконс - спасибо! :).

Забыл описать что значит что в логе Астериска:
phone_ip --- внешний ИП телефона (телефон находится за НАТом).
phone_ip_local --- внутренний ИП телефона
asterisk_ip --- ИП Астериска
cisco_ipo --- ИП киски
В логе Астериска, который я кинул выше есть еще некоторые непонятные вещи:
1. Sending to <phone_ip_local> : 5060 (no NAT)
Откуда вообще берется локальный ИП телефона? Ведь на юзере висит: nat=yes
2. Peer audio RTP is at port <phone_ip_local>:10006
Снова тот же локальный ИП. С чего он тут?.. хмм...
Если поставить в настройках пользователя canreinvite=yes, то голос не идет. Насколько я понимаю именно эта строка и объясняет почему не идет голос. Но в общем-то вариант, когда голосовой трафик проходит через Астериск меня устраивает, так что это не критично. Но если вдруг кто что подскажет - буду благодарен.
3. Почему
<--- Transmitting (NAT) to <phone_ip>:10003 --->
SIP/2.0 180 Ringing
возникает раньше, чем с киски приходит:
<--- SIP read from <cisco_ip>:5060 --->
SIP/2.0 183 Session Progress

Ведь это же нелогично отправлять телефону Ringing раньше, чем пришел какой либо ответ от киски.

Вроде все, буду очень признателен за ответы.
Но все же основной вопрос остается тот же --- почему после получения от сип-фона команды CANCEL Астериск не передает ее киске?..

irc log:
< sergee> ruffer: есть проблема что если ты положишь трубку - до того как тебе ответят, то в некоторых случаях отвечающая
сторона будет звонить до бесконечности (до таймаута или до того как там не снимут трубку)
< sergee> Астериск передает CANCEL только в том случае если канал имеет состояние AST_STATE_RINGING
< sergee> в 1.2 он передвала CANCEL всегда
< sergee> короче суть в том что если астериск получает код 180 он устанавливает AST-STATE-Ringing и CANCEL работает нормально
< sergee> а если 183 - то не работает
< sergee> лечится это просто
< sergee> открывается chan_sip.c
< sergee> функция handle_response_invite()
< sergee> строка 11770
< sergee> и добавляется установка AST_STATE_RINGING
< sergee> и все работает отлично

патч - http://voipsolutions.ru/node/13

но:
< sergee> flot: я поговорил с oej он сказал что работает над этим и что предложенный мною метод не самый лучший, но
< sergee> мой метод работает здесь и сейчас :) а когда оеж пофиксит это - неизвестно

Спасибо, люди добрые!! Особенно тебе, dyer!
Сколько бы я еще бился с этим, если бы не ты.

Полцарства в благодарность готов --- еще немного и меня бы уволили. Ведь не объяснить же шефу, что это косяк в астериске - не поверит ни за что.

26420654 - ася моя. Напиши, постараюсь как-нить отблагодарить. К примеру wmr.