Found problem
xlite client Version 3 Build 34025 send wrong rtp port to asterisk. But I don't
know how to change this at xlite
venus*CLI
-- SIP read from 192.168.100.20:60726:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.32:5060;branch=z9hG4bK02d4cc64;rport=5060
Contact: sip:[EMAIL PROTECTED]:60726;rinstance=45385da6efafa3ea
To: sip:[EMAIL PROTECTED]:60726;rinstance=45385da6efafa3ea;tag=7b512144
From: Hans-Juergen Brandsip:[EMAIL PROTECTED];tag=as4530bf3b
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 179
v=0
o=- 5 2 IN IP4 127.0.0.1
s=CounterPath X-Lite 3.0
c=IN IP4 127.0.0.1
t=0 0
m=audio 59050 RTP/AVP 0 8 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
--- (11 headers 9 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 127.0.0.1:59050
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0
(nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
list_route: hop: sip:[EMAIL PROTECTED]:60726;rinstance=45385da6efafa3ea
set_destination: Parsing sip:[EMAIL
PROTECTED]:60726;rinstance=45385da6efafa3ea for address/port to send to
set_destination: set destination to 192.168.100.20, port 60726
Transmitting (no NAT) to 192.168.100.20:60726:
ACK sip:[EMAIL PROTECTED]:60726;rinstance=45385da6efafa3ea SIP/2.0
Via: SIP/2.0/UDP 192.168.100.32:5060;branch=z9hG4bK42575a4c;rport
From: Hans-Juergen Brand sip:[EMAIL PROTECTED];tag=as4530bf3b
To: sip:[EMAIL PROTECTED]:60726;rinstance=45385da6efafa3ea;tag=7b512144
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
Original-Nachricht
Datum: Thu, 28 Dec 2006 22:30:24 +0100
Von: Hans-Jürgen Brand [EMAIL PROTECTED]
An: asterisk-users@lists.digium.com
Betreff: [asterisk-users] one way rtp stream (Sent alwax to 127.0.0.1)
Asterisk version 1.2.14
I use snom190 and xliteV3 as sip phones.
asterisk send the rtp stream never to the xlite softphone.
Any hits for me?
*CLI rtp debug
RTP Debugging Enabled
-- Executing Dial(SIP/xlite-007918f0, SIP/snom) in new stack
-- Called snom
-- SIP/snom-00797110 is ringing
-- SIP/snom-00797110 is ringing
-- SIP/snom-00797110 answered SIP/xlite-007918f0
-- Attempting native bridge of SIP/xlite-007918f0 and
SIP/snom-00797110
Got RTP packet from 192.168.100.70:50002 (type 0, seq 6022, ts 32652224,
len 160)
Sent RTP packet to 127.0.0.1:17548 (type 0, seq 49874, ts 64, len 160)
Got RTP packet from 192.168.100.20:17548 (type 0, seq 6911, ts 1973300,
len 160)Sent RTP packet to 192.168.100.70:50002 (type 0, seq 28956, ts 16,
len 160)
Got RTP packet from 192.168.100.70:50002 (type 0, seq 6023, ts 32652544,
len 160)
Sent RTP packet to 127.0.0.1:17548 (type 0, seq 49875, ts 384, len 160)
*CLI sip show peers
Name/username HostDyn Nat ACL Port Status
snom/snom 192.168.100.70 D 2051
Unmonitored
xlite/xlite192.168.100.20 D 11420
Unmonitored
2 sip peers [2 online , 0 offline]
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