[Asterisk-Users] sip phone configuration problem

[gomer]

What phone do you have?

On Fri, 16 Jul 2004 11:59:39 +0500, atif <atif at convergence.com.pk> wrote:
> 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' fesponse 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
> 
> ________________________________________________________________
> Sent via the WebMail system at convergence.com.pk
> 
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