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@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@192.168.0.2>;tag=Zlq179E4Jf8KX2lB To: 13 <sip:13@192.168.0.2> Call-ID: 1e020TNnX5IvcvFu CSeq: 1 INVITE Contact: <sip:chinee@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@192.168.0.2>;tag=Zlq179E4Jf8KX2lB To: 13 <sip:13@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@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
What phone do you have? On Fri, 16 Jul 2004 11:59:39 +0500, atif <atif@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@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@192.168.0.2>;tag=Zlq179E4Jf8KX2lB > To: 13 <sip:13@192.168.0.2> > Call-ID: 1e020TNnX5IvcvFu > CSeq: 1 INVITE > Contact: <sip:chinee@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@192.168.0.2>;tag=Zlq179E4Jf8KX2lB > To: 13 <sip:13@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@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 > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
733 is a remote IP phone 608 is X-Lite on the internal LAN I see a: *CLI>... Registration from '"733" <sip:733@voip.elmit.com>' failed for '218.x.x.x' *CLI> sip show peers 733/733 (Unspecified ) D 255.255.255.255 0 Unknown 608/608 192.168.250.200 D 255.255.255.255 5060 OK (26 ms) *CLI> sip show users 733 password test733 No No 608 password default No No I can call 608, but 608 cannot call back 733 cannot register below is my sip.conf, ... Can anybody give me a hint? sip.conf [general] context=defaultport=5060 bindaddr=0.0.0.0 srvlookup=yes externip = 61.220.121.18 localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks [608] ; Note-Pen's X-lite type=friend disallow=all allow=ulaw allow=alaw type=friend username=608 secret=<password> host=dynamic dtmfmode=inband qualify=1000 mailbox=608 group=1 pickupgroup=1 [733] ; Test phone 733 context=unisen disallow=all allow=ulaw allow=alaw type=friend username=test733 secret=<password> host=dynamic dtmfmode=inband qualify=1000 mailbox=733 group=1 pickupgroup=1