[Asterisk-Users] sip phone configuration problem

[atif]

I am configuring a sip-phone, receing calls, excellent voice quality. but it does not place calls, please, can some one sort out.

here is my debug output, and below that is sip-debug,

Jul 16 11:34:32 DEBUG[163850]: Setting NAT on RTP to 0
Jul 16 11:34:32 DEBUG[163850]: Stopping retransmission on 'iiasPlzFribMJMcW' of Response 1: Found
Jul 16 11:34:32 DEBUG[163850]: Setting NAT on RTP to 0
Jul 16 11:34:32 DEBUG[163850]: Stopping retransmission on 'UCWmUU1tF0s6roEx' of Response 2: Found
Jul 16 11:34:47 DEBUG[163850]: Auto destroying call 'iiasPlzFribMJMcW'
Jul 16 11:34:47 DEBUG[163850]: Auto destroying call 'UCWmUU1tF0s6roEx'


******SIP-DEBUG******
Sip read:
INVITE sip:13 at 192.168.0.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.187:5060;branch=z9hG4bKAtoCh7hGS
Max-Forwards: 70
From: chinee <sip:chinee at 192.168.0.2>;tag=Zlq179E4Jf8KX2lB
To: 13 <sip:13 at 192.168.0.2>
Call-ID: 1e020TNnX5IvcvFu
CSeq: 1 INVITE
Contact: <sip:chinee at 192.168.0.187:5060>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 221

v=0
o=- 0 0 IN IP4 192.168.0.187
s=-
c=IN IP4 192.168.0.187
t=0 0
m=audio 1400 RTP/AVP 0 8 4 18 0
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 telephone-event

12 headers, 11 lines
Using latest request as basis request
Sending to 192.168.0.187 : 5060 (non-NAT)
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 0
Peer RTP is at port 192.168.0.187:0
Found description format PCMU
Found description format PCMA
Found description format G723
Found description format G729
Found description format telephone-event
Capabilities: us - 0x10f(G723|GSM|ULAW|ALAW|G729A), peer - audio=0x109(G723|ALAW|G729A)/video=0x0(EMPTY), combined - 0x109(G723|ALAW|G729A)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723)
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.187:5060;branch=z9hG4bKAtoCh7hGS
From: chinee <sip:chinee at 192.168.0.2>;tag=Zlq179E4Jf8KX2lB
To: 13 <sip:13 at 192.168.0.2>;tag=as51de164a
Call-ID: 1e020TNnX5IvcvFu
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:13 at 192.168.0.2>
Proxy-Authenticate: Digest realm="asterisk", nonce="50b81cdd"
Content-Length: 0

 to 192.168.0.187:5060
Scheduling destruction of call '1e020TNnX5IvcvFu' in 15000 ms
Found user 'chinee'


Atif 




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