Help !!!!! SIP/2.0 400 Bad Request - 'Malformed/Missing URL'
*CLI>
*CLI> sip debug peer cisco
SIP Debugging Enabled for IP: 81.222.xxx.xxx:5060
*CLI> dial 3365050
*CLI> We're at 81.222.xxx.xxx port 19558
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 11 lines
Reliably Transmitting (no NAT) to 81.222.xxx.xxx:5060:
INVITE sip:81.222.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 81.222.xxx.xxx:5060;branch=z9hG4bK78a95a97;rport
From: "asterisk" <sip:asterisk@81.222.xxx.xxx>;tag=as40a0d0cd
To: <sip:81.222.x.x>
Contact: <sip:asterisk@81.222.x.x>
Call-ID: 11c60fc009e541ba24efeca81373b45c@81.222.x.x
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 07 Apr 2006 10:32:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 238
v=0
o=root 12410 12410 IN IP4 81.222.x.x
s=session
c=IN IP4 81.222.x.x
t=0 0
m=audio 19558 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
<-- SIP read from 81.222.x.x:5060:
SIP/2.0 400 Bad Request - 'Malformed/Missing URL'
Via: SIP/2.0/UDP 81.222.x.x:5060;branch=z9hG4bK78a95a97;rport
From: "asterisk" <sip:asterisk@81.222.x.x>;tag=as40a0d0cd
To: <sip:81.222.x.x>
Date: Fri, 07 Apr 2006 10:19:26 GMT
Call-ID: 11c60fc009e541ba24efeca81373b45c@81.222.x.x
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0
--- (10 headers 0 lines)---
Transmitting (no NAT) to 81.222.x.x:5060:
ACK sip:81.222.x.x SIP/2.0
Via: SIP/2.0/UDP 81.222.x.x:5060;branch=z9hG4bK78a95a97;rport
From: "asterisk" <sip:asterisk@81.222.x.x>;tag=as40a0d0cd
To: <sip:81.222.x.x>
Contact: <sip:asterisk@81.222.x.x>
Call-ID: 11c60fc009e541ba24efeca81373b45c@81.222.x.x
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
voice-port 3/0:D
cptone RU
bearer-cap Speech
!
voice-port 3/1:D
cptone RU
bearer-cap Speech
!
mgcp fax t38 ecm
!
dial-peer cor custom
!
!
!
dial-peer data 8 pots
incoming called-number 33651xx
!
dial-peer voice 100 pots
destination-pattern .T
port 3/0:D
forward-digits all
!
dial-peer voice 120 voip
destination-pattern 703....
voice-class codec 1
session protocol sipv2
session target ipv4:81.222.x.x
!
dial-peer voice 110 voip
voice-class codec 1
session protocol sipv2
session target ipv4:81.222.x.x
!
dial-peer voice 130 pots
incoming called-number .T
direct-inward-dial
forward-digits extra
!
!
sip-ua
sip-server ipv4:81.222.x.x
!
!
Люди , помогите , весь форум по десять раз перерыл , все перепроверил ... входящие с тфоп на астериск идут без проблем , а вот обратно циска не пускает !((( где грабли ?
8)
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