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<DIV><FONT face=Arial size=2>It appears from the debug that extension 200 is
trying to call 777, not 202. Your Asterisk server can't find an extension 777
and returns "404 not found". That will explain why you can't call extension 777
from extension 200. If you want to call extension 202, you will need to dial 202
on extension 200, not 777.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Regards,</FONT></DIV>
<DIV><FONT face=Arial size=2>Derek</FONT></DIV>
<DIV> </DIV>
<BLOCKQUOTE
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=angus@iteloffice.com href="mailto:angus@iteloffice.com">Angus
Comber</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=asterisk-users@lists.digium.com
href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Sunday, July 24, 2005 11:51
AM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> [Asterisk-Users] Why can't
sip/200 call sip/202</DIV>
<DIV><BR></DIV>
<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>
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