> I have 2 sip accounts setup - 200 and 202. If I do sip show peers I get: > > sip show peers > Name/username Host Dyn Nat ACL Mask Port Status > 202/202 192.168.0.6 D 255.255.255.255 5060 > Unmonitored > 201/201 (Unspecified) D 255.255.255.255 5060 > Unmonitored > 200/200 192.168.0.3 D 255.255.255.255 5060 > Unmonitored > > 200 is a Grandstream GXP200 IP Phone and 202 is a Grandstream BT100 IP phone. > > relevant bit of sip.conf: > > [200] > username=200 > type=friend > secret=1234 > port=5060 > nat=never > dtmfmode=rfc2833 > context=default > callerid="Angus Comber" <200> > host=dynamic > disallow=all > allow=ulaw > allow=alaw > allow=g723.1 > allow=g729 > > [202] > username=202 > type=friend > secret=1234 > port=5060 > nat=never > dtmfmode=rfc2833 > context=default > callerid="Sam Comber" <202> > host=dynamic > disallow=all > allow=ulaw > allow=alaw > allow=g723.1 > allow=g729 > > > But whenever I try to dial between phones I get this: > > > Sip read: > > 0 headers, 0 lines > > > Sip read: > INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKa6cf8b6a7c7198a1 > From: "Angus Comber" <sip:[EMAIL PROTECTED];user=phone>;tag=a1afaf4fdb0ac845 > To: <sip:[EMAIL PROTECTED];user=phone> > Contact: <sip:[EMAIL PROTECTED];user=phone> > Supported: replaces, timer > Call-ID: [EMAIL PROTECTED] > CSeq: 45925 INVITE > User-Agent: Grandstream GXP2000 1.0.1.9 > Max-Forwards: 70 > Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK > Content-Type: application/sdp > Content-Length: 258 > > v=0 > o=200 8000 8000 IN IP4 192.168.0.3 > s=SIP Call > c=IN IP4 192.168.0.3 > t=0 0 > m=audio 5004 RTP/AVP 18 0 8 101 > a=sendrecv > a=rtpmap:18 G729/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=ptime:20 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-11 > > 13 headers, 13 lines > Using latest request as basis request > Sending to 192.168.0.3 : 5060 (non-NAT) > Reliably Transmitting (no NAT): > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKa6cf8b6a7c7198a1 > From: "Angus Comber" <sip:[EMAIL PROTECTED];user=phone>;tag=a1afaf4fdb0ac845 > To: <sip:[EMAIL PROTECTED];user=phone>;tag=as668982be > Call-ID: [EMAIL PROTECTED] > CSeq: 45925 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:[EMAIL PROTECTED]> > Proxy-Authenticate: Digest realm="asterisk", nonce="0c555366" > Content-Length: 0 > > > to 192.168.0.3:5060 > Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms > Found user '200' > > > Sip read: > ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKa6cf8b6a7c7198a1 > From: "Angus Comber" <sip:[EMAIL PROTECTED];user=phone>;tag=a1afaf4fdb0ac845 > To: <sip:[EMAIL PROTECTED];user=phone>;tag=as668982be > Contact: <sip:[EMAIL PROTECTED];user=phone> > Call-ID: [EMAIL PROTECTED] > CSeq: 45925 ACK > User-Agent: Grandstream GXP2000 1.0.1.9 > Max-Forwards: 70 > Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK > Content-Length: 0 > > > 11 headers, 0 lines > > > Sip read: > INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKe570dfe4b8441304 > From: "Angus Comber" <sip:[EMAIL PROTECTED];user=phone>;tag=a1afaf4fdb0ac845 > To: <sip:[EMAIL PROTECTED];user=phone> > Contact: <sip:[EMAIL PROTECTED];user=phone> > Supported: replaces, timer > Proxy-Authorization: Digest username="200", realm="asterisk", algorithm=MD5, uri="sip:[EMAIL PROTECTED];user=phone", nonce="0c555366", > response="ee6088fb4e50da5fe412913ae40dd45c" > Call-ID: [EMAIL PROTECTED] > CSeq: 45926 INVITE > User-Agent: Grandstream GXP2000 1.0.1.9 > Max-Forwards: 70 > Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK > Content-Type: application/sdp > Content-Length: 258 > > v=0 > o=200 8000 8001 IN IP4 192.168.0.3 > s=SIP Call > c=IN IP4 192.168.0.3 > t=0 0 > m=audio 5004 RTP/AVP 18 0 8 101 > a=sendrecv > a=rtpmap:18 G729/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=ptime:20 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-11 > > 14 headers, 13 lines > Using latest request as basis request > Sending to 192.168.0.3 : 5060 (non-NAT) > Found user '200' > Found RTP audio format 18 > Found RTP audio format 0 > Found RTP audio format 8 > Found RTP audio format 101 > Peer audio RTP is at port 192.168.0.3:5004 > Found description format G729 > Found description format PCMU > Found description format PCMA > Found description format telephone-event > Capabilities: us - 0x10d (g723|ulaw|alaw|g729), peer - audio=0x10c > (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729) > Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 > (g723) > Looking for 777 in default > Reliably Transmitting (no NAT): > SIP/2.0 404 Not Found > Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKe570dfe4b8441304 > From: "Angus Comber" <sip:[EMAIL PROTECTED];user=phone>;tag=a1afaf4fdb0ac845 > To: <sip:[EMAIL PROTECTED];user=phone>;tag=as668982be > Call-ID: [EMAIL PROTECTED] > CSeq: 45926 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:[EMAIL PROTECTED]> > Content-Length: 0 > > > to 192.168.0.3:5060 > > > Sip read: > ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKe570dfe4b8441304 > From: "Angus Comber" <sip:[EMAIL PROTECTED];user=phone>;tag=a1afaf4fdb0ac845 > To: <sip:[EMAIL PROTECTED];user=phone>;tag=as668982be > Contact: <sip:[EMAIL PROTECTED];user=phone> > Proxy-Authorization: Digest username="200", realm="asterisk", algorithm=MD5, uri="sip:[EMAIL PROTECTED];user=phone", nonce="0c555366", > response="7fcb1024a81b3ea3bcc56baeca4bac3e" > Call-ID: [EMAIL PROTECTED] > CSeq: 45926 ACK > User-Agent: Grandstream GXP2000 1.0.1.9 > Max-Forwards: 70 > Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK > Content-Length: 0 > > > 12 headers, 0 lines > Destroying call '[EMAIL PROTECTED]' > > > How can I troubleshoot? What should I be looking at?
In the debug trace shown above, I see: Looking for 777 in default Reliably Transmitting (no NAT): SIP/2.0 404 Not Found It would appear something is trying to dial "777" in the default context (in extensions.conf), and that extension isn't defined, therefor you are getting a "404 Not Found". Without looking at your extensions.conf contents, can't guess any closer on the problem. Also, until you get your arms around diagnosing problems, I'd suggest starting with a single codec, like: disallow=all allow=ulaw ; allow=alaw ; allow=g723.1 ; allow=g729 When the basics are well understood, then go back and experiment with various codecs. I'm also a strong believer in _not_ using a context name such as "default". Asterisk will frequently try to do something with a specified context and if it fails, fall back to the "default" context without you noticing. Unless you happen to see that in the CLI, you form an opinion that your configuration is working fine, but it really isn't. _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
