Grigory Puzankin
2004-Oct-05 00:16 UTC
[Asterisk-Users] H.323: Inbound calls, incorrect remoteIpAddress
Hello,
I'm running Asterisk 1.0.1 (the same was with 0.9, 1.0). When it
receives inbound H.323 call it makes connection and uses local
127.0.0.1 address to send audio stream:
remoteIpAddress: 127.0.0.1
When making outbound calls from Asterisk it makes correct connection
to send audio stream. Is it a bug in h.323? Is there some more
settings to make in .conf files?
See detailed debug below:
*CLI> == New H.323 Connection created.
-- Received SETUP message
-- Setting up Call
-- Call token: [ip$195.128.54.2:2689/1]
-- Calling party name: [Puzankin Grigoriy]
-- Calling party number: [5522]
-- Called party name: [822]
-- Called party number: [822]
Urgent handler
=-= In OnAnswerCall for call 1
Urgent handler
We're at 195.128.54.20 port 15500
Urgent handler
Answering/Requesting with root capability 4
Urgent handler
Answering with non-codec capability 0x1(G723)
Urgent handler
12 headers, 10 lines
Urgent handler
Reliably Transmitting:
INVITE sip:5522@195.128.54.35:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 195.128.54.20:5060;branch=z9hG4bK50f9d850
From: "Puzankin Grigoriy"
<sip:5522@195.128.54.20>;tag=as38dccf8e
To: <sip:5522@195.128.54.35:5060;user=phone;transport=udp>
Contact: <sip:5522@195.128.54.20>
Call-ID: 344872474b98be6b029d950d7394c409@195.128.54.20
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Tue, 05 Oct 2004 07:01:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 218
v=0
o=root 19545 19545 IN IP4 195.128.54.20
s=session
c=IN IP4 195.128.54.20
t=0 0
m=audio 15500 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
(no NAT) to 195.128.54.35:5060
Urgent handler
Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 195.128.54.20:5060;branch=z9hG4bK50f9d850
From: "Puzankin Grigoriy"
<sip:5522@195.128.54.20>;tag=as38dccf8e
To: <sip:5522@195.128.54.35:5060;user=phone;transport=udp>;tag=4027448251
Call-ID: 344872474b98be6b029d950d7394c409@195.128.54.20
CSeq: 102 INVITE
Server: Cisco-CP7912/1.02-040406A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Length: 0
9 headers, 0 lines
Urgent handler
Sip read:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 195.128.54.20:5060;branch=z9hG4bK50f9d850
From: "Puzankin Grigoriy"
<sip:5522@195.128.54.20>;tag=as38dccf8e
To: <sip:5522@195.128.54.35:5060;user=phone;transport=udp>;tag=4027448251
Call-ID: 344872474b98be6b029d950d7394c409@195.128.54.20
CSeq: 102 INVITE
Server: Cisco-CP7912/1.02-040406A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Length: 0
9 headers, 0 lines
Urgent handler
Sending alerting
Urgent handler
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.128.54.20:5060;branch=z9hG4bK50f9d850
From: "Puzankin Grigoriy"
<sip:5522@195.128.54.20>;tag=as38dccf8e
To: <sip:5522@195.128.54.35:5060;user=phone;transport=udp>;tag=4027448251
Call-ID: 344872474b98be6b029d950d7394c409@195.128.54.20
CSeq: 102 INVITE
Contact: Puzankin Grigoriy
<sip:5522@195.128.54.35:5060;user=phone;transport=udp>
Server: Cisco-CP7912/1.02-040406A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Length: 203
Content-Type: application/sdp
v=0
o=5522 7300 7300 IN IP4 195.128.54.35
s=Cisco 7905 SIP Call
c=IN IP4 195.128.54.35
t=0 0
m=audio 16384 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
11 headers, 9 lines
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 195.128.54.35:16384
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x4(ULAW), peer - audio=0x4(ULAW)/video=0x0(EMPTY), combined
- 0x4(ULAW)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723)
list_route: hop: <sip:5522@195.128.54.35:5060;user=phone;transport=udp>
set_destination: Parsing
<sip:5522@195.128.54.35:5060;user=phone;transport=udp> for address/port to
send to
set_destination: set destination to 195.128.54.35, port 5060
Transmitting:
ACK sip:5522@195.128.54.35:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 195.128.54.20:5060;branch=z9hG4bK37f40b93
From: "Puzankin Grigoriy"
<sip:5522@195.128.54.20>;tag=as38dccf8e
To: <sip:5522@195.128.54.35:5060;user=phone;transport=udp>;tag=4027448251
Contact: <sip:5522@195.128.54.20>
Call-ID: 344872474b98be6b029d950d7394c409@195.128.54.20
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 195.128.54.35:5060
Urgent handler
answering call
Urgent handler
=*= In CreateRealTimeLogicalChannel for call 1
-- externalIpAddress: 195.128.54.20
-- externalPort: 16322
-- SessionID: 1
-- Direction: IsTransmitter
-- Started logical channel: sending G.711-uLaw-64k{sw}
-- channelsOpen = 1
RTP channel id 1 parameters:
-- remoteIpAddress: 127.0.0.1
-- remotePort: 2069
-- ExternalIpAddress: 195.128.54.20
-- ExternalPort: 16322
=-= In OnConnectionEstablished for call 1
-- Connection Established with "Puzankin Grigoriy
[195.128.54.2]"
=*= In CreateRealTimeLogicalChannel for call 1
-- externalIpAddress: 195.128.54.20
-- externalPort: 16322
-- SessionID: 1
-- Direction: IsReceiver
-- Started logical channel: receiving G.711-uLaw-64k{sw}
-- channelsOpen = 2
RTP channel id 1 parameters:
-- remoteIpAddress: 195.128.54.2
-- remotePort: 4000
-- ExternalIpAddress: 195.128.54.20
-- ExternalPort: 16322
=-= In OnReceivedAckPDU for call 1
Sip read:
BYE sip:5522@195.128.54.20 SIP/2.0
Via: SIP/2.0/UDP 195.128.54.35:5060
From: <sip:5522@195.128.54.35;user=phone;transport=udp>;tag=4027448251
To: "Puzankin Grigoriy" <sip:5522@195.128.54.20>;tag=as38dccf8e
Call-ID: 344872474b98be6b029d950d7394c409@195.128.54.20
CSeq: 1 BYE
User-Agent: Cisco-CP7912/1.02-040406A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Length: 0
9 headers, 0 lines
Sending to 195.128.54.35 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.128.54.35:5060
From: <sip:5522@195.128.54.35;user=phone;transport=udp>;tag=4027448251
To: "Puzankin Grigoriy" <sip:5522@195.128.54.20>;tag=as38dccf8e
Call-ID: 344872474b98be6b029d950d7394c409@195.128.54.20
CSeq: 1 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:5522@195.128.54.20>
Content-Length: 0
to 195.128.54.35:5060
Urgent handler
-- ClearCall: Request to clear call with token ip$195.128.54.2:2689/1
-- Sending RELEASE COMPLETE
channelsOpen = 1
channelsOpen = 0
-- Call with Puzankin Grigoriy [195.128.54.2] completed (EndedByLocalUser)
== H.323 Connection deleted.
Destroying call '344872474b98be6b029d950d7394c409@195.128.54.20'
--
Grigoriy Puzankin
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