Сообщений: 22

Sipnet.ru и звонки через него.

Всем здрасте.
Такое дело.
Ось CentOC 4.2
Asterisk 1.2.6.
Железо Cisco ATA 186 и Софтфоны X-Lite.
------------------------------------------------
Все поставил без проблем. внутри сети звонки без проблемм. также без проблем звонки наружу но только софтфон по набору SIP ID тоже X-lite зереген в томже Sipnete. как звонить куданить еще нифига либо тишина либо короткие гудки сразу после решетки.
---------------------------------------------------------

2006-04-09 17:31

ermakov

Сообщений: 22

Re: Sipnet.ru и звонки через него.

*sip.conf

[general]
context=default
;allowguest=no
;realm=mydomain.tld
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes

;domain=mydomain.tld
;domain=mydomain.tld,mydomain-incoming

;domain=1.2.3.4
;allowexternalinvites=no
;autodomain=yes
;pedantic=yes
;tos=184
;tos=lowdelay
;maxexpiry=3600
;defaultexpiry=120
;notifymimetype=text/plain
;checkmwi=10
;vmexten=voicemail

;videosupport=yes
;recordhistory=yes

;disallow=all
;allow=ulaw
;allow=ilbc
;musicclass=default

language=en
;relaxdtmf=yes
;rtptimeout=60

;rtpholdtimeout=300

;trustrpid = no
;sendrpid = yes
;progressinband=never
;useragent=Asterisk PBX
;promiscredir = no

;usereqphone = no

;dtmfmode = rfc2833

;compactheaders = yes
;sipdebug = yes
;subscribecontext = default

;notifyringing = yes

regcontext=sipregistrations

; register => user[:secret[:authuser]]@host[:port][/extension]
;register => 1234:password@mysipprovider.com
;register => 2345:password@sip_proxy/1234

;registertimeout=20
;registerattempts=10
;
;callevents=no
;----------------------------------------- NAT SUPPORT ------------------------

externip = 84.54.230.30
;externhost=foo.dyndns.net
;externrefresh=10
;nat=no

;rtcachefriends=yes
;rtupdate=yes
;rtautoclear=yes

;ignoreregexpire=yes

[authentication]
;register => xxxxxxxx:aaaaaaaa@sipnet.ru
register => xxxxxxx:aaaaaaaa@sipnet:5060
;------------------------------------------------------------------------------
[100]
host=dynamic
context=default
type=friend
username=100
nat=no
secret=100
callerid=X-lite Asterisk <100>

[101]
host=dynamic
context=default
type=friend
username=101
nat=no
secret=101
callerid=X lite Win <101>

[102]
host=dynamic
context=default
type=friend
username=102
nat=no
secret=102
callerid=Cab1 <102>

[103]
host=dynamic
context=default
type=friend
username=103
nat=no
secret=103
callerid=Cab2 <103>

[104]
host=dynamic
context=default
type=friend
username=104
nat=no
secret=104
callerid=Cab3 <104>

[105]
host=dynamic
context=default
type=friend
username=105
nat=no
secret=105
callerid=Cab4 <105>

[106]
host=dynamic
context=default
type=friend
username=106
nat=no
secret=106
callerid=Cisco 1 <106>

[107]
host=dynamic
context=default
type=friend
username=107
nat=no
secret=107
callerid=Cisco 2 <107>

[108]
host=dynamic
context=default
type=friend
username=108
nat=no
secret=108
callerid=phone9 <108>

[sipnet]
type=peer
username=xxxxxx
secret=xxxxxx
fromuser=xxxxxxx
fromdomain=sipnet.ru
host=sipnet.ru
port=5060
canreinvite=no
;nat=yes
;dtmfmode=rfc2833
insecure=very
reinvite=no

;context=hg
disallow=all
;allow=ulaw
;allow=alaw
allow=g729
allow=g723
;allow=gsm

2006-04-09 17:32

ermakov

Сообщений: 22

Re: Sipnet.ru и звонки через него.

*sip.conf

[general]
context=default
;allowguest=no
;realm=mydomain.tld
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes

;domain=mydomain.tld
;domain=mydomain.tld,mydomain-incoming

;domain=1.2.3.4
;allowexternalinvites=no
;autodomain=yes
;pedantic=yes
;tos=184
;tos=lowdelay
;maxexpiry=3600
;defaultexpiry=120
;notifymimetype=text/plain
;checkmwi=10
;vmexten=voicemail

;videosupport=yes
;recordhistory=yes

;disallow=all
;allow=ulaw
;allow=ilbc
;musicclass=default

language=en
;relaxdtmf=yes
;rtptimeout=60

;rtpholdtimeout=300

;trustrpid = no
;sendrpid = yes
;progressinband=never
;useragent=Asterisk PBX
;promiscredir = no

;usereqphone = no

;dtmfmode = rfc2833

;compactheaders = yes
;sipdebug = yes
;subscribecontext = default

;notifyringing = yes

regcontext=sipregistrations

; register => user[:secret[:authuser]]@host[:port][/extension]
;register => 1234:password@mysipprovider.com
;register => 2345:password@sip_proxy/1234

;registertimeout=20
;registerattempts=10
;
;callevents=no
;----------------------------------------- NAT SUPPORT ------------------------

externip = 84.54.230.30
;externhost=foo.dyndns.net
;externrefresh=10
;nat=no

;rtcachefriends=yes
;rtupdate=yes
;rtautoclear=yes

;ignoreregexpire=yes

[authentication]
;register => xxxxxxxx:aaaaaaaa@sipnet.ru
register => xxxxxxx:aaaaaaaa@sipnet:5060
;------------------------------------------------------------------------------
[100]
host=dynamic
context=default
type=friend
username=100
nat=no
secret=100
callerid=X-lite Asterisk <100>

[101]
host=dynamic
context=default
type=friend
username=101
nat=no
secret=101
callerid=X lite Win <101>

[102]
host=dynamic
context=default
type=friend
username=102
nat=no
secret=102
callerid=Cab1 <102>

[103]
host=dynamic
context=default
type=friend
username=103
nat=no
secret=103
callerid=Cab2 <103>

[104]
host=dynamic
context=default
type=friend
username=104
nat=no
secret=104
callerid=Cab3 <104>

[105]
host=dynamic
context=default
type=friend
username=105
nat=no
secret=105
callerid=Cab4 <105>

[106]
host=dynamic
context=default
type=friend
username=106
nat=no
secret=106
callerid=Cisco 1 <106>

[107]
host=dynamic
context=default
type=friend
username=107
nat=no
secret=107
callerid=Cisco 2 <107>

[108]
host=dynamic
context=default
type=friend
username=108
nat=no
secret=108
callerid=phone9 <108>

[sipnet]
type=peer
username=xxxxxx
secret=xxxxxx
fromuser=xxxxxxx
fromdomain=sipnet.ru
host=sipnet.ru
port=5060
canreinvite=no
;nat=yes
;dtmfmode=rfc2833
insecure=very
reinvite=no

;context=hg
disallow=all
;allow=ulaw
;allow=alaw
allow=g729
allow=g723
;allow=gsm

2006-04-09 17:32

ermakov

Сообщений: 22

Re: Sipnet.ru и звонки через него.

*extensions.conf

[general]

static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no

[globals]
CONSOLE=Console/dsp
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest
;IAXINFO=myuser:mypass
TRUNK=Zap/g2
TRUNKMSD=1
;TRUNK=IAX2/user:pass@provider

[demo]
exten => s,1,Wait,1
exten => s,n,Answer
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n(restart),BackGround(demo-congrats)
exten => s,n(instruct),BackGround(demo-instruct)
exten => s,n,WaitExten

exten => 2,1,BackGround(demo-moreinfo)
exten => 2,n,Goto(s,instruct)

exten => 3,1,Set(LANGUAGE()=ru)
exten => 3,n,Goto(s,restart)

exten => 1000,1,Goto(default,s,1)

exten => 1234,1,Playback(transfer,skip) ; "Please hold while..."
; (but skip if channel is not up)
exten => 1234,n,Macro(stdexten,1234,${CONSOLE})

exten => 1235,1,Voicemail(u1234) ; Right to voicemail

exten => 1236,1,Dial(Console/dsp) ; Ring forever
exten => 1236,n,Voicemail(u1234) ; Unless busy

exten => #,1,Playback(demo-thanks) ; "Thanks for trying the demo"
exten => #,n,Hangup ; Hang them up.

exten => t,1,Goto(#,1) ; If they take too long, give up
exten => i,1,Playback(invalid) ; "That's not valid, try again"

exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
exten => 500,n,Dial(IAX2/guest@misery.digium.com/s@default) ; Call the Asterisk demo
exten => 500,n,Playback(demo-nogo) ; Couldn't connect to the demo site
exten => 500,n,Goto(s,6) ; Return to the start over message.

exten => 600,1,Playback(demo-echotest) ; Let them know what's going on
exten => 600,n,Echo ; Do the echo test
exten => 600,n,Playback(demo-echodone) ; Let them know it's over
exten => 600,n,Goto(s,6) ; Start over

exten => 8500,1,VoicemailMain
exten => 8500,n,Goto(s,6)
;exten => 1265,1,Dial(Phone/phone0,15)
;exten => 1265,n,Goto(s,5)

;[mainmenu]
;
; Example "main menu" context with submenu
;
;exten => s,1,Answer
;exten => s,n,Background(thanks)
;exten => s,n,WaitExten
;exten => 1,1,Goto(submenu,s,1)
;exten => 2,1,Hangup
;include => default
;
;[submenu]
;exten => s,1,Ringing
;exten => s,n,Wait,2
;exten => s,n,Background(submenuopts)
;exten => s,n,WaitExten
;exten => 1,1,Goto(default,steve,1)
;exten => 2,1,Goto(default,mark,2)

[default]
exten => 100,1, Macro(stdexten,100,SIP/100)
exten => user1, 1, Goto(100|1)

exten => 101,1, Macro(stdexten,101,SIP/101)
exten => user2, 1, Goto(101|1)

exten => 102,1, Macro(stdexten,102,SIP/102)
exten => user3, 1, Goto(102|1)

exten => 103,1, Macro(stdexten,103,SIP/103)
exten => user4, 1, Goto(103|1)

exten => 104,1, Macro(stdexten,104,SIP/104)
exten => user5, 1, Goto(104|1)

exten => 105,1, Macro(stdexten,105,SIP/105)
exten => user6, 1, Goto(105|1)

exten => 106,1, Macro(stdexten,106,SIP/106)
exten => user7, 1, Goto(106|1)

exten => 107,1, Macro(stdexten,107,SIP/107)
exten => user8, 1, Goto(107|1)

exten => 108,1, Macro(stdexten,108,SIP/108)
exten => user9, 1, Goto(108|1)

exten => _810.,1,Answer
exten => _810.,2,Wait,1
exten => _810.,3,Dial(SIP/810${EXTEN:3}@sipnet.ru,60,r)
exten => _8.,1,Answer
exten => _8.,2,Wait,1
exten => _8.,3,Dial(SIP/8${EXTEN:1}@sipnet.ru:5060,60,r)
exten => _1.,1,Answer
exten => _1.,2,Wait,1
exten => _1.,3,Dial(SIP/${EXTEN}@sipnet.ru,60,r)
include => demo

[demo1]

exten => s,1, Wait,1
exten => s,n, Answer
exten => s,n, SetVar(TIMEOUT(digit)=5)
exten => s,n, SetVar(TIMEOUT(response)=10)
exten => s,n(restart), BackGround(demo-congrats)
exten => s,n(instruct), BackGround(demo-instruct)
exten => s,n, WaitExten
exten => 2,1, BackGround(demo-moreinfo)
exten => 2,n,Goto(s,instruct)
exten => 3,1, SetVar(Language()=en)
exten => 3,2, Goto(s, restart)
exten => 8500, 1, VoiceMailMain
exten => 8500, n, Goto,s
exten => 1000, 1, Goto(default,s,1)

[macro-stdexten]

exten => s, 1, Dial (${ARG2},20,t)
exten => s, 2, Goto(s-$(DIALSTATUS),1)
exten => s-NOANSWER, 1, Voicemail(u${ARG1})
exten => s-NOANSWER, 2, (Goto(default,s,1))
exten => s-BUSY, 1, Voicemail(b,${ARG1})
exten => s-BUSY, 2, (Goto(default,s,1))
exten => _s-., 1, Goto(s-NOANSWER)
;exten => a, 1, VoiceMailMain(${ARG1})

[sipnet_inc]

exten => s,1,Wait, 1
exten => s, 2, Answer
exten => s,3, BackGround(local-welcomе)
exten => s,4, WaitExten
exten => 100,1, Macro(stdexten,100,SIP/100)
exten => 101,1, Macro(stdexten,101,SIP/101)
exten => 102,1, Macro(stdexten,102,SIP/102)
exten => 103,1, Macro(stdexten,103,SIP/103)
exten => 104,1, Macro(stdexten,104,SIP/104)
exten => 105,1, Macro(stdexten,105,SIP/105)
exten => 106,1, Macro(stdexten,106,SIP/106)
exten => 107,1, Macro(stdexten,107,SIP/107)
exten => 108,1, Macro(stdexten,108,SIP/108)
exten => 8500,1, VoiceMailMain
exten => 8500,n, Hangup

2006-04-09 17:34

ermakov

Сообщений: 22

Re: Sipnet.ru и звонки через него.

*extensions.conf

[general]

static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no

[globals]
CONSOLE=Console/dsp
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest
;IAXINFO=myuser:mypass
TRUNK=Zap/g2
TRUNKMSD=1
;TRUNK=IAX2/user:pass@provider

[demo]
exten => s,1,Wait,1
exten => s,n,Answer
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n(restart),BackGround(demo-congrats)
exten => s,n(instruct),BackGround(demo-instruct)
exten => s,n,WaitExten

exten => 2,1,BackGround(demo-moreinfo)
exten => 2,n,Goto(s,instruct)

exten => 3,1,Set(LANGUAGE()=ru)
exten => 3,n,Goto(s,restart)

exten => 1000,1,Goto(default,s,1)

exten => 1234,1,Playback(transfer,skip) ; "Please hold while..."
; (but skip if channel is not up)
exten => 1234,n,Macro(stdexten,1234,${CONSOLE})

exten => 1235,1,Voicemail(u1234) ; Right to voicemail

exten => 1236,1,Dial(Console/dsp) ; Ring forever
exten => 1236,n,Voicemail(u1234) ; Unless busy

exten => #,1,Playback(demo-thanks) ; "Thanks for trying the demo"
exten => #,n,Hangup ; Hang them up.

exten => t,1,Goto(#,1) ; If they take too long, give up
exten => i,1,Playback(invalid) ; "That's not valid, try again"

exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
exten => 500,n,Dial(IAX2/guest@misery.digium.com/s@default) ; Call the Asterisk demo
exten => 500,n,Playback(demo-nogo) ; Couldn't connect to the demo site
exten => 500,n,Goto(s,6) ; Return to the start over message.

exten => 600,1,Playback(demo-echotest) ; Let them know what's going on
exten => 600,n,Echo ; Do the echo test
exten => 600,n,Playback(demo-echodone) ; Let them know it's over
exten => 600,n,Goto(s,6) ; Start over

exten => 8500,1,VoicemailMain
exten => 8500,n,Goto(s,6)
;exten => 1265,1,Dial(Phone/phone0,15)
;exten => 1265,n,Goto(s,5)

;[mainmenu]
;
; Example "main menu" context with submenu
;
;exten => s,1,Answer
;exten => s,n,Background(thanks)
;exten => s,n,WaitExten
;exten => 1,1,Goto(submenu,s,1)
;exten => 2,1,Hangup
;include => default
;
;[submenu]
;exten => s,1,Ringing
;exten => s,n,Wait,2
;exten => s,n,Background(submenuopts)
;exten => s,n,WaitExten
;exten => 1,1,Goto(default,steve,1)
;exten => 2,1,Goto(default,mark,2)

[default]
exten => 100,1, Macro(stdexten,100,SIP/100)
exten => user1, 1, Goto(100|1)

exten => 101,1, Macro(stdexten,101,SIP/101)
exten => user2, 1, Goto(101|1)

exten => 102,1, Macro(stdexten,102,SIP/102)
exten => user3, 1, Goto(102|1)

exten => 103,1, Macro(stdexten,103,SIP/103)
exten => user4, 1, Goto(103|1)

exten => 104,1, Macro(stdexten,104,SIP/104)
exten => user5, 1, Goto(104|1)

exten => 105,1, Macro(stdexten,105,SIP/105)
exten => user6, 1, Goto(105|1)

exten => 106,1, Macro(stdexten,106,SIP/106)
exten => user7, 1, Goto(106|1)

exten => 107,1, Macro(stdexten,107,SIP/107)
exten => user8, 1, Goto(107|1)

exten => 108,1, Macro(stdexten,108,SIP/108)
exten => user9, 1, Goto(108|1)

exten => _810.,1,Answer
exten => _810.,2,Wait,1
exten => _810.,3,Dial(SIP/810${EXTEN:3}@sipnet.ru,60,r)
exten => _8.,1,Answer
exten => _8.,2,Wait,1
exten => _8.,3,Dial(SIP/8${EXTEN:1}@sipnet.ru:5060,60,r)
exten => _1.,1,Answer
exten => _1.,2,Wait,1
exten => _1.,3,Dial(SIP/${EXTEN}@sipnet.ru,60,r)
include => demo

[demo1]

exten => s,1, Wait,1
exten => s,n, Answer
exten => s,n, SetVar(TIMEOUT(digit)=5)
exten => s,n, SetVar(TIMEOUT(response)=10)
exten => s,n(restart), BackGround(demo-congrats)
exten => s,n(instruct), BackGround(demo-instruct)
exten => s,n, WaitExten
exten => 2,1, BackGround(demo-moreinfo)
exten => 2,n,Goto(s,instruct)
exten => 3,1, SetVar(Language()=en)
exten => 3,2, Goto(s, restart)
exten => 8500, 1, VoiceMailMain
exten => 8500, n, Goto,s
exten => 1000, 1, Goto(default,s,1)

[macro-stdexten]

exten => s, 1, Dial (${ARG2},20,t)
exten => s, 2, Goto(s-$(DIALSTATUS),1)
exten => s-NOANSWER, 1, Voicemail(u${ARG1})
exten => s-NOANSWER, 2, (Goto(default,s,1))
exten => s-BUSY, 1, Voicemail(b,${ARG1})
exten => s-BUSY, 2, (Goto(default,s,1))
exten => _s-., 1, Goto(s-NOANSWER)
;exten => a, 1, VoiceMailMain(${ARG1})

[sipnet_inc]

exten => s,1,Wait, 1
exten => s, 2, Answer
exten => s,3, BackGround(local-welcomе)
exten => s,4, WaitExten
exten => 100,1, Macro(stdexten,100,SIP/100)
exten => 101,1, Macro(stdexten,101,SIP/101)
exten => 102,1, Macro(stdexten,102,SIP/102)
exten => 103,1, Macro(stdexten,103,SIP/103)
exten => 104,1, Macro(stdexten,104,SIP/104)
exten => 105,1, Macro(stdexten,105,SIP/105)
exten => 106,1, Macro(stdexten,106,SIP/106)
exten => 107,1, Macro(stdexten,107,SIP/107)
exten => 108,1, Macro(stdexten,108,SIP/108)
exten => 8500,1, VoiceMailMain
exten => 8500,n, Hangup

2006-04-09 17:34

ermakov

Сообщений: 22

Re: Sipnet.ru и звонки через него.

--------------------------------------------

при звонке в * пишет

-----------------------------

-- Executing Macro("SIP/101-9605", "stdexten|106|SIP/106") in new stack
-- Executing Dial("SIP/101-9605", "SIP/106|20") in new stack
-- Called 106
-- SIP/106-da63 is ringing
== Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/101-9605' in macro 'stdexten'
== Spawn extension (default, 106, 1) exited non-zero on 'SIP/101-9605'
Apr 9 20:22:10 WARNING[21894]: chan_sip.c:1210 retrans_pkt: Maximum retries exceeded on transmission 5D009D7F-E7EF-9D 2F-7934-C46F60DD48B5@192.168.1.200 for seqno 45309 (Critical Response)
-- Registered SIP '101' at 192.168.1.200 port 5063 expires 1800
-- Added extension '101' priority 1 to sipregistrations
-- Executing Answer("SIP/101-ad53", "") in new stack
-- Executing Wait("SIP/101-ad53", "1") in new stack
-- Executing Dial("SIP/101-ad53", "SIP/84959818818@sipnet.ru:5060|60|r") in new stack
-- parse_srv: SRV mapped to host mail.tario.ru, port 5060
-- Called 84959818818@sipnet.ru:5060
Apr 9 20:24:23 NOTICE[21894]: chan_sip.c:9521 handle_response_invite: Failed to authenticate on INVITE to '"X lite Wi n" <sip:101@192.168.1.200>;tag=as60b1e9f5'
-- SIP/sipnet.ru:5060-1230 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'SIP/101-ad53' status is 'CONGESTION'
Apr 9 20:24:53 WARNING[21894]: chan_sip.c:1210 retrans_pkt: Maximum retries exceeded on transmission 273DD19A-9628-A0 B8-EBA2-66D437CA686D@192.168.1.200 for seqno 13945 (Critical Response)

2006-04-09 17:36

ermakov

Сообщений: 22

Re: Sipnet.ru и звонки через него.

------------------------------------------------------------------------------------

в x-lite

---------------------------------------------------------------

© 2004 Xten Networks, Inc. All rights reserved.

X-Lite release 1105d build stamp 99999

License key: 5AE28EFFA089086960CF2F2F38A0686D

Attempting SIP protocol listen on: 192.168.1.200:5061

Attempting SIP protocol listen on: 192.168.1.200:5062

Established SIP protocol listen on: 192.168.1.200:5063

Discovered Single Mapped Port Symmetric NAT Firewall

SIP: 192.168.1.200:5063

RTP: 192.168.1.200:8000

NAT: 84.54.230.30

Discovering external SIP port on symmetric firewall...

Discovered external SIP port on symmetric firewall: 1195

RECEIVE TIME: 2138539547

RECEIVE << 64.69.76.23:3478

PROXY#0: 192.168.1.200:5060

OUTBOUND-PROXY#0: 192.168.1.200:5060

SEND TIME: 2138539700

SEND >> 192.168.1.200:5060

REGISTER sip:192.168.1.200 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.200:5063;rport;branch=z9hG4bK2C058EFA85DAE32916DF96FC08FA0081

From: 101 <sip:101@192.168.1.200>;tag=1341242745

To: 101 <sip:101@192.168.1.200>

Contact: "101" <sip:101@192.168.1.200:5063>

Call-ID: 3CC82A887F73DA5AB88F1B29500A578C@192.168.1.200

CSeq: 59909 REGISTER

Expires: 1800

Max-Forwards: 70

User-Agent: X-Lite release 1105d

Content-Length: 0

RECEIVE TIME: 2138539707

RECEIVE << 192.168.1.200:5060

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.1.200:5063;rport;branch=z9hG4bK2C058EFA85DAE32916DF96FC08FA0081;received=192.168.1.200

From: 101 <sip:101@192.168.1.200>;tag=1341242745

To: 101 <sip:101@192.168.1.200>

Call-ID: 3CC82A887F73DA5AB88F1B29500A578C@192.168.1.200

CSeq: 59909 REGISTER

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Contact: <sip:101@192.168.1.200>

Content-Length: 0

RECEIVE TIME: 2138539708

RECEIVE << 192.168.1.200:5060

SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 192.168.1.200:5063;rport;branch=z9hG4bK2C058EFA85DAE32916DF96FC08FA0081;received=192.168.1.200

From: 101 <sip:101@192.168.1.200>;tag=1341242745

To: 101 <sip:101@192.168.1.200>;tag=as342c9001

Call-ID: 3CC82A887F73DA5AB88F1B29500A578C@192.168.1.200

CSeq: 59909 REGISTER

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Contact: <sip:101@192.168.1.200>

WWW-Authenticate: Digest realm="asterisk", nonce="0edbc15f"

Content-Length: 0

SEND TIME: 2138539711

SEND >> 192.168.1.200:5060

REGISTER sip:192.168.1.200 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.200:5063;rport;branch=z9hG4bK138B7E4227ACAB374E73DE56417E5E91

From: 101 <sip:101@192.168.1.200>;tag=1341242745

To: 101 <sip:101@192.168.1.200>

Contact: "101" <sip:101@192.168.1.200:5063>

Call-ID: 3CC82A887F73DA5AB88F1B29500A578C@192.168.1.200

CSeq: 59910 REGISTER

Expires: 1800

Authorization: Digest username="101",realm="asterisk",nonce="0edbc15f",response="5e8e065b69865f52ad8f5341632911d3",uri="sip:192.168.1.200"

Max-Forwards: 70

User-Agent: X-Lite release 1105d

Content-Length: 0

RECEIVE TIME: 2138539713

RECEIVE << 192.168.1.200:5060

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.1.200:5063;rport;branch=z9hG4bK138B7E4227ACAB374E73DE56417E5E91;received=192.168.1.200

From: 101 <sip:101@192.168.1.200>;tag=1341242745

To: 101 <sip:101@192.168.1.200>

Call-ID: 3CC82A887F73DA5AB88F1B29500A578C@192.168.1.200

CSeq: 59910 REGISTER

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Contact: <sip:101@192.168.1.200>

Content-Length: 0

RECEIVE TIME: 2138539754

RECEIVE << 192.168.1.200:5060

SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.200:5063;rport;branch=z9hG4bK138B7E4227ACAB374E73DE56417E5E91;received=192.168.1.200

From: 101 <sip:101@192.168.1.200>;tag=1341242745

To: 101 <sip:101@192.168.1.200>;tag=as342c9001

Call-ID: 3CC82A887F73DA5AB88F1B29500A578C@192.168.1.200

CSeq: 59910 REGISTER

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Expires: 1800

Contact: <sip:101@192.168.1.200:5063>;expires=1800

Date: Sun, 09 Apr 2006 16:24:00 GMT

Content-Length: 0

RECEIVE TIME: 2138539770

RECEIVE << 64.69.76.23:3478

SEND TIME: 2138561198

SEND >> 192.168.1.200:5060

INVITE sip:84959818818@192.168.1.200 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.200:5063;rport;branch=z9hG4bK309E01B7F371B21F32F969B7CB417FC8

From: 101 <sip:101@192.168.1.200:5063>;tag=390476301

To: <sip:84959818818@192.168.1.200>

Contact: <sip:101@192.168.1.200:5063>

Call-ID: 273DD19A-9628-A0B8-EBA2-66D437CA686D@192.168.1.200

CSeq: 13944 INVITE

Max-Forwards: 70

Content-Type: application/sdp

User-Agent: X-Lite release 1105d

Content-Length: 310

v=0

o=101 2138560590 2138561197 IN IP4 192.168.1.200

s=X-Lite

c=IN IP4 192.168.1.200

t=0 0

m=audio 8004 RTP/AVP 0 8 3 98 97 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:3 gsm/8000

a=rtpmap:98 iLBC/8000

a=rtpmap:97 speex/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

RECEIVE TIME: 2138561200

RECEIVE << 192.168.1.200:5060

SIP/2.0 407 Proxy Authentication Required

Via: SIP/2.0/UDP 192.168.1.200:5063;rport;branch=z9hG4bK309E01B7F371B21F32F969B7CB417FC8;received=192.168.1.200

From: 101 <sip:101@192.168.1.200:5063>;tag=390476301

To: <sip:84959818818@192.168.1.200>;tag=as77f053ef

Call-ID: 273DD19A-9628-A0B8-EBA2-66D437CA686D@192.168.1.200

CSeq: 13944 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Contact: <sip:84959818818@192.168.1.200>

Proxy-Authenticate: Digest realm="asterisk", nonce="23ffaf9b"

Content-Length: 0

SEND TIME: 2138561204

SEND >> 192.168.1.200:5060

ACK sip:84959818818@192.168.1.200 SIP/2.0

2006-04-09 17:38

ermakov

Сообщений: 22

Re: Sipnet.ru и звонки через него.

Via: SIP/2.0/UDP 192.168.1.200:5063;rport;branch=z9hG4bK309E01B7F371B21F32F969B7CB417FC8

From: 101 <sip:101@192.168.1.200:5063>;tag=390476301

To: <sip:84959818818@192.168.1.200>;tag=as77f053ef

Contact: <sip:101@192.168.1.200:5063>

Call-ID: 273DD19A-9628-A0B8-EBA2-66D437CA686D@192.168.1.200

CSeq: 13944 ACK

Max-Forwards: 70

Content-Length: 0

SEND TIME: 2138561248

SEND >> 192.168.1.200:5060

INVITE sip:84959818818@192.168.1.200 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.200:5063;rport;branch=z9hG4bK6C1D2BEB9A1F9B6AD2FBEA3DBD5A1241

From: 101 <sip:101@192.168.1.200:5063>;tag=390476301

To: <sip:84959818818@192.168.1.200>

Contact: <sip:101@192.168.1.200:5063>

Call-ID: 273DD19A-9628-A0B8-EBA2-66D437CA686D@192.168.1.200

CSeq: 13945 INVITE

Proxy-Authorization: Digest username="101",realm="asterisk",nonce="23ffaf9b",response="c30ca415a56f34bba0a966d0892fe6d9",uri="sip:84959818818@192.168.1.200"

Max-Forwards: 70

Content-Type: application/sdp

User-Agent: X-Lite release 1105d

Content-Length: 310

v=0

o=101 2138560590 2138561197 IN IP4 192.168.1.200

s=X-Lite

c=IN IP4 192.168.1.200

t=0 0

m=audio 8004 RTP/AVP 0 8 3 98 97 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:3 gsm/8000

a=rtpmap:98 iLBC/8000

a=rtpmap:97 speex/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

RECEIVE TIME: 2138561250

RECEIVE << 192.168.1.200:5060

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.1.200:5063;rport;branch=z9hG4bK6C1D2BEB9A1F9B6AD2FBEA3DBD5A1241;received=192.168.1.200

From: 101 <sip:101@192.168.1.200:5063>;tag=390476301

To: <sip:84959818818@192.168.1.200>

Call-ID: 273DD19A-9628-A0B8-EBA2-66D437CA686D@192.168.1.200

CSeq: 13945 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Contact: <sip:84959818818@192.168.1.200>

Content-Length: 0

RECEIVE TIME: 2138561269

RECEIVE << 192.168.1.200:5060

SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.200:5063;rport;branch=z9hG4bK6C1D2BEB9A1F9B6AD2FBEA3DBD5A1241;received=192.168.1.200

From: 101 <sip:101@192.168.1.200:5063>;tag=390476301

To: <sip:84959818818@192.168.1.200>;tag=as4e6e0fe2

Call-ID: 273DD19A-9628-A0B8-EBA2-66D437CA686D@192.168.1.200

CSeq: 13945 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Contact: <sip:84959818818@192.168.1.200>

Content-Type: application/sdp

Content-Length: 265

v=0

o=root 21879 21879 IN IP4 192.168.1.200

s=session

c=IN IP4 192.168.1.200

t=0 0

m=audio 14454 RTP/AVP 3 0 8 101

a=rtpmap:3 GSM/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

SEND TIME: 2138561350

SEND >> 192.168.1.200:5060

ACK sip:84959818818@192.168.1.200 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.200:5063;rport;branch=z9hG4bK2A4764BB9C0BCF828E066DCBC3C7DD05

From: 101 <sip:101@192.168.1.200:5063>;tag=390476301

To: <sip:84959818818@192.168.1.200>;tag=as4e6e0fe2

Contact: <sip:101@192.168.1.200:5063>

Call-ID: 273DD19A-9628-A0B8-EBA2-66D437CA686D@192.168.1.200

CSeq: 13945 ACK

Max-Forwards: 70

Content-Length: 0

RECEIVE TIME: 2138572452

RECEIVE << 192.168.1.200:5060

SIP/2.0 603 Declined

Via: SIP/2.0/UDP 192.168.1.200:5063;rport;branch=z9hG4bK6C1D2BEB9A1F9B6AD2FBEA3DBD5A1241;received=192.168.1.200

From: 101 <sip:101@192.168.1.200:5063>;tag=390476301

To: <sip:84959818818@192.168.1.200>;tag=as4e6e0fe2

Call-ID: 273DD19A-9628-A0B8-EBA2-66D437CA686D@192.168.1.200

CSeq: 13945 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Contact: <sip:84959818818@192.168.1.200>

Content-Length: 0

SEND TIME: 2138572454

SEND >> 192.168.1.200:5060

ACK sip:84959818818@192.168.1.200 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.200:5063;rport;branch=z9hG4bK2A4764BB9C0BCF828E066DCBC3C7DD05

From: 101 <sip:101@192.168.1.200:5063>;tag=390476301

To: <sip:84959818818@192.168.1.200>;tag=as4e6e0fe2

Contact: <sip:101@192.168.1.200:5063>

Call-ID: 273DD19A-9628-A0B8-EBA2-66D437CA686D@192.168.1.200

CSeq: 13945 ACK

Max-Forwards: 70

Content-Length: 0

RECEIVE TIME: 2138573454

RECEIVE << 192.168.1.200:5060

SIP/2.0 603 Declined

Via: SIP/2.0/UDP 192.168.1.200:5063;rport;branch=z9hG4bK6C1D2BEB9A1F9B6AD2FBEA3DBD5A1241;received=192.168.1.200

From: 101 <sip:101@192.168.1.200:5063>;tag=390476301

To: <sip:84959818818@192.168.1.200>;tag=as4e6e0fe2

Call-ID: 273DD19A-9628-A0B8-EBA2-66D437CA686D@192.168.1.200

CSeq: 13945 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Contact: <sip:84959818818@192.168.1.200>

Content-Length: 0

SEND TIME: 2138573455

SEND >> 192.168.1.200:5060

ACK sip:84959818818@192.168.1.200 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.200:5063;rport;branch=z9hG4bK2A4764BB9C0BCF828E066DCBC3C7DD05

From: 101 <sip:101@192.168.1.200:5063>;tag=390476301

To: <sip:84959818818@192.168.1.200>;tag=as4e6e0fe2

Contact: <sip:101@192.168.1.200:5063>

Call-ID: 273DD19A-9628-A0B8-EBA2-66D437CA686D@192.168.1.200

CSeq: 13945 ACK

Max-Forwards: 70

Content-Length: 0

RECEIVE TIME: 2138574453

RECEIVE << 192.168.1.200:5060

SIP/2.0 603 Declined

Via: SIP/2.0/UDP 192.168.1.200:5063;rport;branch=z9hG4bK6C1D2BEB9A1F9B6AD2FBEA3DBD5A1241;received=192.168.1.200

From: 101 <sip:101@192.168.1.200:5063>;tag=390476301

To: <sip:84959818818@192.168.1.200>;tag=as4e6e0fe2

Call-ID: 273DD19A-9628-A0B8-EBA2-66D437CA686D@192.168.1.200

CSeq: 13945 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Contact: <sip:84959818818@192.168.1.200>

Content-Length: 0

SEND TIME: 2138574459

SEND >> 192.168.1.200:5060

ACK sip:84959818818@192.168.1.200 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.200:5063;rport;branch=z9hG4bK2A4764BB9C0BCF828E066DCBC3C7DD05

From: 101 <sip:101@192.168.1.200:5063>;tag=390476301

To: <sip:84959818818@192.168.1.200>;tag=as4e6e0fe2

Contact: <sip:101@192.168.1.200:5063>

Call-ID: 273DD19A-9628-A0B8-EBA2-66D437CA686D@192.168.1.200

CSeq: 13945 ACK

Max-Forwards: 70

Content-Length: 0

RECEIVE TIME: 2138576453

RECEIVE << 192.168.1.200:5060

SIP/2.0 603 Declined

Via: SIP/2.0/UDP 192.168.1.200:5063;rport;branch=z9hG4bK6C1D2BEB9A1F9B6AD2FBEA3DBD5A1241;received=192.168.1.200

From: 101 <sip:101@192.168.1.200:5063>;tag=390476301

To: <sip:84959818818@192.168.1.200>;tag=as4e6e0fe2

Call-ID: 273DD19A-9628-A0B8-EBA2-66D437CA686D@192.168.1.200

CSeq: 13945 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Contact: <sip:84959818818@192.168.1.200>

Content-Length: 0

SEND TIME: 2138576457

SEND >> 192.168.1.200:5060

ACK sip:84959818818@192.168.1.200 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.200:5063;rport;branch=z9hG4bK2A4764BB9C0BCF828E066DCBC3C7DD05

From: 101 <sip:101@192.168.1.200:5063>;tag=390476301

To: <sip:84959818818@192.168.1.200>;tag=as4e6e0fe2

Contact: <sip:101@192.168.1.200:5063>

Call-ID: 273DD19A-9628-A0B8-EBA2-66D437CA686D@192.168.1.200

CSeq: 13945 ACK

Max-Forwards: 70

Content-Length: 0

RECEIVE TIME: 2138580455

RECEIVE << 192.168.1.200:5060

SIP/2.0 603 Declined

Via: SIP/2.0/UDP 192.168.1.200:5063;rport;branch=z9hG4bK6C1D2BEB9A1F9B6AD2FBEA3DBD5A1241;received=192.168.1.200

From: 101 <sip:101@192.168.1.200:5063>;tag=390476301

To: <sip:84959818818@192.168.1.200>;tag=as4e6e0fe2

Call-ID: 273DD19A-9628-A0B8-EBA2-66D437CA686D@192.168.1.200

CSeq: 13945 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Contact: <sip:84959818818@192.168.1.200>

Content-Length: 0

SEND TIME: 2138580457

SEND >> 192.168.1.200:5060

ACK sip:84959818818@192.168.1.200 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.200:5063;rport;branch=z9hG4bK2A4764BB9C0BCF828E066DCBC3C7DD05

From: 101 <sip:101@192.168.1.200:5063>;tag=390476301

To: <sip:84959818818@192.168.1.200>;tag=as4e6e0fe2

Contact: <sip:101@192.168.1.200:5063>

Call-ID: 273DD19A-9628-A0B8-EBA2-66D437CA686D@192.168.1.200

CSeq: 13945 ACK

Max-Forwards: 70

Content-Length: 0

RECEIVE TIME: 2138584453

RECEIVE << 192.168.1.200:5060

SIP/2.0 603 Declined

Via: SIP/2.0/UDP 192.168.1.200:5063;rport;branch=z9hG4bK6C1D2BEB9A1F9B6AD2FBEA3DBD5A1241;received=192.168.1.200

From: 101 <sip:101@192.168.1.200:5063>;tag=390476301

To: <sip:84959818818@192.168.1.200>;tag=as4e6e0fe2

2006-04-09 17:39

ermakov

Сообщений: 22

Re: Sipnet.ru и звонки через него.

Call-ID: 273DD19A-9628-A0B8-EBA2-66D437CA686D@192.168.1.200

CSeq: 13945 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Contact: <sip:84959818818@192.168.1.200>

Content-Length: 0

SEND TIME: 2138584456

SEND >> 192.168.1.200:5060

ACK sip:84959818818@192.168.1.200 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.200:5063;rport;branch=z9hG4bK2A4764BB9C0BCF828E066DCBC3C7DD05

From: 101 <sip:101@192.168.1.200:5063>;tag=390476301

To: <sip:84959818818@192.168.1.200>;tag=as4e6e0fe2

Contact: <sip:101@192.168.1.200:5063>

Call-ID: 273DD19A-9628-A0B8-EBA2-66D437CA686D@192.168.1.200

CSeq: 13945 ACK

Max-Forwards: 70

Content-Length: 0

RECEIVE TIME: 2138588470

RECEIVE << 192.168.1.200:5060

SIP/2.0 603 Declined

Via: SIP/2.0/UDP 192.168.1.200:5063;rport;branch=z9hG4bK6C1D2BEB9A1F9B6AD2FBEA3DBD5A1241;received=192.168.1.200

From: 101 <sip:101@192.168.1.200:5063>;tag=390476301

To: <sip:84959818818@192.168.1.200>;tag=as4e6e0fe2

Call-ID: 273DD19A-9628-A0B8-EBA2-66D437CA686D@192.168.1.200

CSeq: 13945 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Contact: <sip:84959818818@192.168.1.200>

Content-Length: 0

SEND TIME: 2138588471

SEND >> 192.168.1.200:5060

ACK sip:84959818818@192.168.1.200 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.200:5063;rport;branch=z9hG4bK2A4764BB9C0BCF828E066DCBC3C7DD05

From: 101 <sip:101@192.168.1.200:5063>;tag=390476301

To: <sip:84959818818@192.168.1.200>;tag=as4e6e0fe2

Contact: <sip:101@192.168.1.200:5063>

Call-ID: 273DD19A-9628-A0B8-EBA2-66D437CA686D@192.168.1.200

CSeq: 13945 ACK

Max-Forwards: 70

Content-Length: 0

SEND TIME: 2138592152

SEND >> 192.168.1.200:5060

BYE sip:84959818818@192.168.1.200 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.200:5063;rport;branch=z9hG4bK0B6F37FF16C70D5F87191C08733B9EA6

From: 101 <sip:101@192.168.1.200:5063>;tag=390476301

To: <sip:84959818818@192.168.1.200>;tag=as4e6e0fe2

Contact: <sip:101@192.168.1.200:5063>

Call-ID: 273DD19A-9628-A0B8-EBA2-66D437CA686D@192.168.1.200

CSeq: 13946 BYE

Max-Forwards: 70

User-Agent: X-Lite release 1105d

Content-Length: 0

RECEIVE TIME: 2138592154

RECEIVE << 192.168.1.200:5060

SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.200:5063;branch=z9hG4bK0B6F37FF16C70D5F87191C08733B9EA6;received=192.168.1.200;rport=5063

From: 101 <sip:101@192.168.1.200:5063>;tag=390476301

To: <sip:84959818818@192.168.1.200>;tag=as4e6e0fe2

Call-ID: 273DD19A-9628-A0B8-EBA2-66D437CA686D@192.168.1.200

CSeq: 13946 BYE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Contact: <sip:84959818818@192.168.1.200>

Content-Length: 0

SEND TIME: 2140329785

SEND >> 192.168.1.200:5060

REGISTER sip:192.168.1.200 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.200:5063;rport;branch=z9hG4bK1BAC2EE81F1F31BE981EAF85A182E1DB

From: 101 <sip:101@192.168.1.200>;tag=1341242745

To: 101 <sip:101@192.168.1.200>

Contact: "101" <sip:101@192.168.1.200:5063>

Call-ID: 3CC82A887F73DA5AB88F1B29500A578C@192.168.1.200

CSeq: 59911 REGISTER

Expires: 1800

Max-Forwards: 70

User-Agent: X-Lite release 1105d

Content-Length: 0

RECEIVE TIME: 2140329872

RECEIVE << 192.168.1.200:5060

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.1.200:5063;rport;branch=z9hG4bK1BAC2EE81F1F31BE981EAF85A182E1DB;received=192.168.1.200

From: 101 <sip:101@192.168.1.200>;tag=1341242745

To: 101 <sip:101@192.168.1.200>

Call-ID: 3CC82A887F73DA5AB88F1B29500A578C@192.168.1.200

CSeq: 59911 REGISTER

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Contact: <sip:101@192.168.1.200>

Content-Length: 0

RECEIVE TIME: 2140329874

RECEIVE << 192.168.1.200:5060

SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 192.168.1.200:5063;rport;branch=z9hG4bK1BAC2EE81F1F31BE981EAF85A182E1DB;received=192.168.1.200

From: 101 <sip:101@192.168.1.200>;tag=1341242745

To: 101 <sip:101@192.168.1.200>;tag=as7de5b9ed

Call-ID: 3CC82A887F73DA5AB88F1B29500A578C@192.168.1.200

CSeq: 59911 REGISTER

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Contact: <sip:101@192.168.1.200>

WWW-Authenticate: Digest realm="asterisk", nonce="6188c191"

Content-Length: 0

SEND TIME: 2140329879

SEND >> 192.168.1.200:5060

REGISTER sip:192.168.1.200 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.200:5063;rport;branch=z9hG4bK2CE2CAA1E96F16617D2866DE82EC20B8

From: 101 <sip:101@192.168.1.200>;tag=1341242745

To: 101 <sip:101@192.168.1.200>

Contact: "101" <sip:101@192.168.1.200:5063>

Call-ID: 3CC82A887F73DA5AB88F1B29500A578C@192.168.1.200

CSeq: 59912 REGISTER

Expires: 1800

Authorization: Digest username="101",realm="asterisk",nonce="6188c191",response="a18d1c0e3af01b5639f4dfeae3e5c658",uri="sip:192.168.1.200"

Max-Forwards: 70

User-Agent: X-Lite release 1105d

Content-Length: 0

RECEIVE TIME: 2140329951

RECEIVE << 192.168.1.200:5060

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.1.200:5063;rport;branch=z9hG4bK2CE2CAA1E96F16617D2866DE82EC20B8;received=192.168.1.200

From: 101 <sip:101@192.168.1.200>;tag=1341242745

To: 101 <sip:101@192.168.1.200>

Call-ID: 3CC82A887F73DA5AB88F1B29500A578C@192.168.1.200

CSeq: 59912 REGISTER

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Contact: <sip:101@192.168.1.200>

Content-Length: 0

RECEIVE TIME: 2140330013

RECEIVE << 192.168.1.200:5060

SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.200:5063;rport;branch=z9hG4bK2CE2CAA1E96F16617D2866DE82EC20B8;received=192.168.1.200

From: 101 <sip:101@192.168.1.200>;tag=1341242745

To: 101 <sip:101@192.168.1.200>;tag=as7de5b9ed

Call-ID: 3CC82A887F73DA5AB88F1B29500A578C@192.168.1.200

CSeq: 59912 REGISTER

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Expires: 1800

Contact: <sip:101@192.168.1.200:5063>;expires=1800

Date: Sun, 09 Apr 2006 16:53:50 GMT

Content-Length: 0
/?????????????????????????????????????????????????????????????????????????????????????????

вобщем прошу помощи голова уже раскалывается.

2006-04-09 17:40

bsg

Откуда: Москва

Сообщений: 135

Re: Sipnet.ru и звонки через него.

рабочий конфиг sip.conf
ip внешний, ната нет

register => USER:PASSWORD@sipnet.ru/SIPID

[sipnet]
type=friend
secret=PASSWORD
username=USER
fromuser=USER
fromdomain=sipnet.ru
host=sipnet.ru
port=5060
disallow=all
allow=g729
dtmfmode=auto
canreinvite=no
insecure=very

поставьте
type=friend
ну и register подправьте - но это для входящих

2006-04-09 22:59