[Asterisk-Users] Why can't sip/200 call sip/202
dbruce
dbruce at bananatel.ca
Sun Jul 24 11:17:03 MST 2005
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.
Regards,
Derek
----- Original Message -----
From: Angus Comber
To: asterisk-users at lists.digium.com
Sent: Sunday, July 24, 2005 11:51 AM
Subject: [Asterisk-Users] Why can't sip/200 call sip/202
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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