[Asterisk-Users] Why can't sip/200 call sip/202

Angus Comber angus at iteloffice.com
Sun Jul 24 10:51:26 MST 2005 

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:777 at 192.168.0.13;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKa6cf8b6a7c7198a1
From: "Angus Comber" <sip:200 at 192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845
To: <sip:777 at 192.168.0.13;user=phone>
Contact: <sip:200 at 192.168.0.3;user=phone>
Supported: replaces, timer
Call-ID: 11e4ca07b25c9335 at 192.168.0.3
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:200 at 192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845
To: <sip:777 at 192.168.0.13;user=phone>;tag=as668982be
Call-ID: 11e4ca07b25c9335 at 192.168.0.3
CSeq: 45925 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:777 at 192.168.0.13>
Proxy-Authenticate: Digest realm="asterisk", nonce="0c555366"
Content-Length: 0


 to 192.168.0.3:5060
Scheduling destruction of call '11e4ca07b25c9335 at 192.168.0.3' in 15000 ms
Found user '200'


Sip read:
ACK sip:777 at 192.168.0.13;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKa6cf8b6a7c7198a1
From: "Angus Comber" <sip:200 at 192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845
To: <sip:777 at 192.168.0.13;user=phone>;tag=as668982be
Contact: <sip:200 at 192.168.0.3;user=phone>
Call-ID: 11e4ca07b25c9335 at 192.168.0.3
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:777 at 192.168.0.13;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKe570dfe4b8441304
From: "Angus Comber" <sip:200 at 192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845
To: <sip:777 at 192.168.0.13;user=phone>
Contact: <sip:200 at 192.168.0.3;user=phone>
Supported: replaces, timer
Proxy-Authorization: Digest username="200", realm="asterisk", algorithm=MD5, uri="sip:777 at 192.168.0.13;user=phone", nonce="0c555366", response="ee6088fb4e50da5fe412913ae40dd45c"
Call-ID: 11e4ca07b25c9335 at 192.168.0.3
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:200 at 192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845
To: <sip:777 at 192.168.0.13;user=phone>;tag=as668982be
Call-ID: 11e4ca07b25c9335 at 192.168.0.3
CSeq: 45926 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:777 at 192.168.0.13>
Content-Length: 0


 to 192.168.0.3:5060


Sip read:
ACK sip:777 at 192.168.0.13;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKe570dfe4b8441304
From: "Angus Comber" <sip:200 at 192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845
To: <sip:777 at 192.168.0.13;user=phone>;tag=as668982be
Contact: <sip:200 at 192.168.0.3;user=phone>
Proxy-Authorization: Digest username="200", realm="asterisk", algorithm=MD5, uri="sip:777 at 192.168.0.13;user=phone", nonce="0c555366", response="7fcb1024a81b3ea3bcc56baeca4bac3e"
Call-ID: 11e4ca07b25c9335 at 192.168.0.3
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 '11e4ca07b25c9335 at 192.168.0.3'


How can I troubleshoot?  What should I be looking at?

Angus
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