Kostyantyn Ahafontsev
2003-Aug-01 05:37 UTC
[Asterisk-Users] Asterisk SIP bug with Net2Phone
When I try call to net2pohe sip service in my debug I
look next:
----------------------------------------------------
We're at 192.0.0.0 port 27916
Answering with preferred capability 1
Answering with preferred capability 2
Answering with preferred capability 256
Answering with capability 4
Answering with capability 8
Answering with capability 16
Answering with capability 32
Answering with capability 64
Answering with capability 128
Answering with capability 512
Answering with capability 1024
Answering with capability 2048
Answering with capability 4096
Answering with capability 8192
Answering with capability 16384
Answering with capability 32768
10 headers, 17 lines
Reliably Transmitting:
INVITE sip:1800XXXXXXX@sip.net2phone.com SIP/2.0
Via: SIP/2.0/UDP 192.0.0.0:5060;branch=z9hG4bK6afc45f2
From: "111111111111@net2phone.com"
<sip:111111111111@net2phone.com>;tag=as26712c28
To: <sip:1800XXXXXXX@sip.net2phone.com>
Contact: <sip:111111111111@192.0.0.0>
Call-ID: 265fdf1e0f61bd2a3422940c4ec90131@192.0.0.0
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Content-Type: application/sdp
Content-Length: 384
v=0
o=root 21604 21604 IN IP4 192.0.0.0
s=session
c=IN IP4 192.0.0.0
t=0 0
m=audio 27916 RTP/AVP 4 3 18 0 8 14 5 10 7 110 97
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:14 MPA/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:110 SPEEX/8000
a=rtpmap:97 iLBC/8000
(no NAT) to 66.33.146.12:5060
-- Called 1800XXXXXXX@net2phone
Sip read:
SIP/2.0 407 Unauthorized
Via: SIP/2.0/UDP 192.0.0.0:5060;branch=z9hG4bK6afc45f2
From: "111111111111@net2phone.com"
<sip:111111111111@net2phone.com>;tag=as26712c28
To:
<sip:1800XXXXXXX@sip.net2phone.com>;tag=3f2a5b8b-12006
Call-ID: 265fdf1e0f61bd2a3422940c4ec90131@192.0.0.0
CSeq: 102 INVITE
Contact: "net2phone" <sip:66.33.146.12:5060>
User-Agent: Asterisk PBX
Proxy-Authenticate: Digest
realm="net2phone",nonce="55895A5A2566A49758E30C701D17BD49"
Content-Length: 0
10 headers, 0 lines
Transmitting:
ACK sip:1800XXXXXXX@sip.net2phone.com SIP/2.0
Via: SIP/2.0/UDP 192.0.0.0:5060;branch=z9hG4bK6afc45f2
From: "111111111111@net2phone.com"
<sip:111111111111@net2phone.com>;tag=as26712c28
To:
<sip:1800XXXXXXX@sip.net2phone.com>;tag=3f2a5b8b-12006
Contact: <sip:111111111111@192.0.0.0>
Call-ID: 265fdf1e0f61bd2a3422940c4ec90131@192.0.0.0
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 66.33.146.12:5060
We're at 192.0.0.0 port 27916
Answering with preferred capability 1
Answering with preferred capability 2
Answering with preferred capability 256
Answering with capability 4
Answering with capability 8
Answering with capability 16
Answering with capability 32
Answering with capability 64
Answering with capability 128
Answering with capability 512
Answering with capability 1024
Answering with capability 2048
Answering with capability 4096
Answering with capability 8192
Answering with capability 16384
Answering with capability 32768
Reliably Transmitting:
INVITE sip:1800XXXXXXX@sip.net2phone.com SIP/2.0
Via: SIP/2.0/UDP 192.0.0.0:5060;branch=z9hG4bK6afc45f2
From: "111111111111@net2phone.com"
<sip:111111111111@net2phone.com>;tag=as26712c28
To: <sip:1800XXXXXXX@sip.net2phone.com>
Contact: <sip:111111111111@192.0.0.0>
Call-ID: 265fdf1e0f61bd2a3422940c4ec90131@192.0.0.0
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Proxy-Authorization: Digest username="111111111111",
realm="net2phone", algorithm="MD5",
uri="sip:1800XXXXXXX@sip.net2phone.com",
nonce="55895A5A2566A49758E30C701D17BD49",
response="bd4841816ed727ed12c3bc4d1b19e7a5"
Content-Type: application/sdp
Content-Length: 384
v=0
o=root 21586 21586 IN IP4 192.0.0.0
s=session
c=IN IP4 192.0.0.0
t=0 0
m=audio 27916 RTP/AVP 4 3 18 0 8 14 5 10 7 110 97
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:14 MPA/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:110 SPEEX/8000
a=rtpmap:97 iLBC/8000
(no NAT) to 66.33.146.12:5060
Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.0.0.0:5060;branch=z9hG4bK6afc45f2
From: "111111111111@net2phone.com"
<sip:111111111111@net2phone.com>;tag=as26712c28
To:
<sip:1800XXXXXXX@sip.net2phone.com>;tag=3f2a5b8c-12006
Call-ID: 265fdf1e0f61bd2a3422940c4ec90131@192.0.0.0
CSeq: 103 INVITE
Contact: "net2phone" <sip:66.33.146.12:5060>
User-Agent: Asterisk PBX
Content-Length: 0
9 headers, 0 lines
Sip read:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.0.0.0:5060;branch=z9hG4bK6afc45f2
From: "111111111111@net2phone.com"
<sip:111111111111@net2phone.com>;tag=as26712c28
To:
<sip:1800XXXXXXX@sip.net2phone.com>;tag=3f2a5b8c-12006
Call-ID: 265fdf1e0f61bd2a3422940c4ec90131@192.0.0.0
CSeq: 103 INVITE
Contact: "net2phone" <sip:66.33.146.12:5060>
User-Agent: Asterisk PBX
Content-Length: 240
Content-Type: application/sdp
v=0
o=Net2Phone 562767273 562767273 IN IP4 66.33.136.130
s=Net2Phone
c=IN IP4 66.33.136.130
t=0 0
m=audio 20182 RTP/AVP 4 101
a=ptime:90
a=rtpmap:4 G723/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
10 headers, 11 lines
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.0.0.0:5060;branch=z9hG4bK6afc45f2
From: "111111111111@net2phone.com"
<sip:111111111111@net2phone.com>;tag=as26712c28
To:
<sip:1800XXXXXXX@sip.net2phone.com>;tag=3f2a5b8c-12006
Call-ID: 265fdf1e0f61bd2a3422940c4ec90131@192.0.0.0
CSeq: 103 INVITE
Contact: "net2phone" <sip:66.33.146.12:5060>
User-Agent: Asterisk PBX
Content-Length: 240
Content-Type: application/sdp
v=0
o=Net2Phone 562767273 562767273 IN IP4 66.33.136.130
s=Net2Phone
c=IN IP4 66.33.136.130
t=0 0
m=audio 20182 RTP/AVP 4 101
a=ptime:90
a=rtpmap:4 G723/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
10 headers, 11 lines
-----------------------------------------------------
And in the debug log file i see next:
------------------------------------------------------
Aug 1 15:25:52 DEBUG[49159]: File chan_sip.c, Line
4628 (handle_request): That's odd... Got a response
on a call we dont know about.
Aug 1 15:25:52 DEBUG[49159]: File chan_sip.c, Line
830 (__sip_destroy): Destorying call
'265fdf1e0f61bd2a3422940c4ec90131@192.0.0.0'
------------------------------------------------------
Why asterisk not see the last messeges from Net2Phone
SIP :
SIP/2.0 183 Session Progress
SIP/2.0 200 OK
and not reply. After recieved messages - SIP/2.0 200
OK asterisk must send ACK, but it do not do it.
Konstantin
This may help: I had to edit chan_sip.c and change User-Agent: "Asterisk PBX" to "Cisco ATA 186" to get n2p sip to work -----Original Message----- From: Kostyantyn Ahafontsev [mailto:kostik_sa@yahoo.com] Sent: 01 August 2003 13:38 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk SIP bug with Net2Phone When I try call to net2pohe sip service in my debug I look next: