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

dbruce Sun, 24 Jul 2005 11:21:30 -0700

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: [email protected]

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:[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?

Angus

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

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