I have a basic asterisk system with a grandstream GXW4108 8 port Analog IP gateway setup as a peer.
My overall expictation is to eventually have asterisk answer the line and ask which conference number they would like to join. Setin up a conference system for our office. Any help with the following will be appreciated, I have 2 total grandstream gateways that I am tring to set this up as a conference server for.
I keep getting this error,
[Dec 12 07:21:57] NOTICE[4790]: chan_sip.c:13774 handle_request_invite: Call from ‘6201’ to extension ‘6000’ rejected be
cause extension not found.
I have ext 6000 (main operator w/ voicemail) but still continue to get this error. 6201 is the sip gateway definition for the GXW4108 and is configured as such.
[6201]
type=peer
context=incoming
host=69.220.229.53
insecure=very
Here is the sip debug from the call attempt.
a=ptime:20
<------------->
— (13 headers 13 lines) —
Sending to 69.220.229.53 : 5062 (no NAT)
Using INVITE request as basis request - aae63ea4f861ba94@69.220.229.53
Found peer '6201’
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 3
Peer audio RTP is at port 69.220.229.53:5024
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G723 for ID 4
Found audio description format G729 for ID 18
Found audio description format GSM for ID 3
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combi
ned - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 69.220.229.53:5024
Looking for 6000 in incoming (domain 69.220.229.51)
<— Reliably Transmitting (no NAT) to 69.220.229.53:5062 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 69.220.229.53:5062;branch=z9hG4bKc4b4bf930b104514;received=69.220.229.53
From: "unknown"sip:unknown@69.220.229.51;tag=35134a2525914677
To: sip:6000@69.220.229.51;tag=as16a6b7fc
Call-ID: aae63ea4f861ba94@69.220.229.53
CSeq: 26229 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<— SIP read from 69.220.229.53:5062 —>
ACK sip:6000@69.220.229.51 SIP/2.0
Via: SIP/2.0/UDP 69.220.229.53:5062;branch=z9hG4bKc4b4bf930b104514
From: "unknown"sip:unknown@69.220.229.51;tag=35134a2525914677
To: sip:6000@69.220.229.51;tag=as16a6b7fc
Contact: sip:6201@69.220.229.53:5062
Call-ID: aae63ea4f861ba94@69.220.229.53
CSeq: 26229 ACK
User-Agent: Grandstream GXW4108 (HW 0001, Ch:5) 1.0.0.55
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0
<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog ‘aae63ea4f861ba94@69.220.229.53’ Method: ACK
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 69.220.229.53:5062;branch=z9hG4bKc4b4bf930b104514;received=69.220.229.53
From: "unknown"sip:unknown@69.220.229.51;tag=35134a2525914677
To: sip:6000@69.220.229.51;tag=as16a6b7fc
Call-ID: aae63ea4f861ba94@69.220.229.53
CSeq: 26229 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0