<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=Content-Type content="text/html; charset=iso-8859-1">
<META content="MSHTML 6.00.2900.2627" name=GENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=#ffffff>
<DIV><FONT face=Arial size=2>I have 2 sip accounts setup - 200 and 202. If
I do sip show peers I get:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>sip show peers<BR>Name/username
Host Dyn Nat
ACL Mask
Port
Status<BR>202/202
192.168.0.6
D 255.255.255.255
5060
Unmonitored<BR>201/201
(Unspecified)
D 255.255.255.255
5060
Unmonitored<BR>200/200
192.168.0.3
D 255.255.255.255
5060 Unmonitored</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>200 is a Grandstream GXP200 IP Phone and 202 is a
Grandstream BT100 IP phone.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>relevant bit of sip.conf:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial
size=2>[200]<BR>username=200<BR>type=friend<BR>secret=1234<BR>port=5060<BR>nat=never<BR>dtmfmode=rfc2833<BR>context=default<BR>callerid="Angus
Comber"
<200><BR>host=dynamic<BR>disallow=all<BR>allow=ulaw<BR>allow=alaw<BR>allow=g723.1<BR>allow=g729</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial
size=2>[202]<BR>username=202<BR>type=friend<BR>secret=1234<BR>port=5060<BR>nat=never<BR>dtmfmode=rfc2833<BR>context=default<BR>callerid="Sam
Comber"
<202><BR>host=dynamic<BR>disallow=all<BR>allow=ulaw<BR>allow=alaw<BR>allow=g723.1<BR>allow=g729</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>But whenever I try to dial between phones I get
this:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Sip read:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>0 headers, 0 lines</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV><FONT face=Arial size=2>
<DIV><BR>Sip read:<BR>INVITE sip:777@192.168.0.13;user=phone SIP/2.0<BR>Via:
SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKa6cf8b6a7c7198a1<BR>From: "Angus Comber"
<sip:200@192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845<BR>To:
<sip:777@192.168.0.13;user=phone><BR>Contact:
<sip:200@192.168.0.3;user=phone><BR>Supported: replaces, timer<BR>Call-ID:
<A
href="mailto:11e4ca07b25c9335@192.168.0.3">11e4ca07b25c9335@192.168.0.3</A><BR>CSeq:
45925 INVITE<BR>User-Agent: Grandstream GXP2000 1.0.1.9<BR>Max-Forwards:
70<BR>Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK<BR>Content-Type:
application/sdp<BR>Content-Length: 258</DIV>
<DIV> </DIV>
<DIV>v=0<BR>o=200 8000 8000 IN IP4 192.168.0.3<BR>s=SIP Call<BR>c=IN IP4
192.168.0.3<BR>t=0 0<BR>m=audio 5004 RTP/AVP 18 0 8
101<BR>a=sendrecv<BR>a=rtpmap:18 G729/8000<BR>a=rtpmap:0 PCMU/8000<BR>a=rtpmap:8
PCMA/8000<BR>a=ptime:20<BR>a=rtpmap:101 telephone-event/8000<BR>a=fmtp:101
0-11</DIV>
<DIV> </DIV>
<DIV>13 headers, 13 lines<BR>Using latest request as basis request<BR>Sending to
192.168.0.3 : 5060 (non-NAT)<BR>Reliably Transmitting (no NAT):<BR>SIP/2.0 407
Proxy Authentication Required<BR>Via: SIP/2.0/UDP
192.168.0.3;branch=z9hG4bKa6cf8b6a7c7198a1<BR>From: "Angus Comber"
<sip:200@192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845<BR>To:
<sip:777@192.168.0.13;user=phone>;tag=as668982be<BR>Call-ID: <A
href="mailto:11e4ca07b25c9335@192.168.0.3">11e4ca07b25c9335@192.168.0.3</A><BR>CSeq:
45925 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER<BR>Contact: <sip:777@192.168.0.13><BR>Proxy-Authenticate:
Digest realm="asterisk", nonce="0c555366"<BR>Content-Length: 0</DIV>
<DIV> </DIV>
<DIV><BR> to 192.168.0.3:5060<BR>Scheduling destruction of call <A
href="mailto:'11e4ca07b25c9335@192.168.0.3'">'11e4ca07b25c9335@192.168.0.3'</A>
in 15000 ms<BR>Found user '200'</DIV>
<DIV> </DIV>
<DIV><BR>Sip read:<BR>ACK sip:777@192.168.0.13;user=phone SIP/2.0<BR>Via:
SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKa6cf8b6a7c7198a1<BR>From: "Angus Comber"
<sip:200@192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845<BR>To:
<sip:777@192.168.0.13;user=phone>;tag=as668982be<BR>Contact:
<sip:200@192.168.0.3;user=phone><BR>Call-ID: <A
href="mailto:11e4ca07b25c9335@192.168.0.3">11e4ca07b25c9335@192.168.0.3</A><BR>CSeq:
45925 ACK<BR>User-Agent: Grandstream GXP2000 1.0.1.9<BR>Max-Forwards:
70<BR>Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK<BR>Content-Length:
0</DIV>
<DIV> </DIV>
<DIV><BR>11 headers, 0 lines</DIV>
<DIV> </DIV>
<DIV><BR>Sip read:<BR>INVITE sip:777@192.168.0.13;user=phone SIP/2.0<BR>Via:
SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKe570dfe4b8441304<BR>From: "Angus Comber"
<sip:200@192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845<BR>To:
<sip:777@192.168.0.13;user=phone><BR>Contact:
<sip:200@192.168.0.3;user=phone><BR>Supported: replaces,
timer<BR>Proxy-Authorization: Digest username="200", realm="asterisk",
algorithm=MD5, uri="sip:777@192.168.0.13;user=phone", nonce="0c555366",
response="ee6088fb4e50da5fe412913ae40dd45c"<BR>Call-ID: <A
href="mailto:11e4ca07b25c9335@192.168.0.3">11e4ca07b25c9335@192.168.0.3</A><BR>CSeq:
45926 INVITE<BR>User-Agent: Grandstream GXP2000 1.0.1.9<BR>Max-Forwards:
70<BR>Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK<BR>Content-Type:
application/sdp<BR>Content-Length: 258</DIV>
<DIV> </DIV>
<DIV>v=0<BR>o=200 8000 8001 IN IP4 192.168.0.3<BR>s=SIP Call<BR>c=IN IP4
192.168.0.3<BR>t=0 0<BR>m=audio 5004 RTP/AVP 18 0 8
101<BR>a=sendrecv<BR>a=rtpmap:18 G729/8000<BR>a=rtpmap:0 PCMU/8000<BR>a=rtpmap:8
PCMA/8000<BR>a=ptime:20<BR>a=rtpmap:101 telephone-event/8000<BR>a=fmtp:101
0-11</DIV>
<DIV> </DIV>
<DIV>14 headers, 13 lines<BR>Using latest request as basis request<BR>Sending to
192.168.0.3 : 5060 (non-NAT)<BR>Found user '200'<BR>Found RTP audio format
18<BR>Found RTP audio format 0<BR>Found RTP audio format 8<BR>Found RTP audio
format 101<BR>Peer audio RTP is at port 192.168.0.3:5004<BR>Found description
format G729<BR>Found description format PCMU<BR>Found description format
PCMA<BR>Found description format telephone-event<BR>Capabilities: us - 0x10d
(g723|ulaw|alaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing),
combined - 0x10c (ulaw|alaw|g729)<BR>Non-codec capabilities: us - 0x1 (g723),
peer - 0x1 (g723), combined - 0x1 (g723)<BR>Looking for 777 in
default<BR>Reliably Transmitting (no NAT):<BR>SIP/2.0 404 Not Found<BR>Via:
SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKe570dfe4b8441304<BR>From: "Angus Comber"
<sip:200@192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845<BR>To:
<sip:777@192.168.0.13;user=phone>;tag=as668982be<BR>Call-ID: <A
href="mailto:11e4ca07b25c9335@192.168.0.3">11e4ca07b25c9335@192.168.0.3</A><BR>CSeq:
45926 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER<BR>Contact: <sip:777@192.168.0.13><BR>Content-Length: 0</DIV>
<DIV> </DIV>
<DIV><BR> to 192.168.0.3:5060</DIV>
<DIV> </DIV>
<DIV><BR>Sip read:<BR>ACK sip:777@192.168.0.13;user=phone SIP/2.0<BR>Via:
SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKe570dfe4b8441304<BR>From: "Angus Comber"
<sip:200@192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845<BR>To:
<sip:777@192.168.0.13;user=phone>;tag=as668982be<BR>Contact:
<sip:200@192.168.0.3;user=phone><BR>Proxy-Authorization: Digest
username="200", realm="asterisk", algorithm=MD5,
uri="sip:777@192.168.0.13;user=phone", nonce="0c555366",
response="7fcb1024a81b3ea3bcc56baeca4bac3e"<BR>Call-ID: <A
href="mailto:11e4ca07b25c9335@192.168.0.3">11e4ca07b25c9335@192.168.0.3</A><BR>CSeq:
45926 ACK<BR>User-Agent: Grandstream GXP2000 1.0.1.9<BR>Max-Forwards:
70<BR>Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK<BR>Content-Length:
0</DIV>
<DIV> </DIV>
<DIV><BR>12 headers, 0 lines<BR>Destroying call <A
href="mailto:'11e4ca07b25c9335@192.168.0.3'">'11e4ca07b25c9335@192.168.0.3'</A></DIV>
<DIV> </DIV>
<DIV><BR>How can I troubleshoot? What should I be looking at?</DIV>
<DIV> </DIV>
<DIV>Angus</DIV>
<DIV></FONT> </DIV></BODY></HTML>