Hi.
I’m running Asterisk SVN-branch-1.8-r331578M. I’m getting not found error mostly i sending calls to asterisk. my configration is as below
extenstion.usr
[code]context NIC-NIKE-505-01 {
_50X. => {
Dial(SIP/${EXTEN:2}@192.168.0.13:5060,120);
Hangup();
}
_51X. => {
Dial(SIP/${EXTEN:2}@192.168.0.16:5060,120);
Hangup();
}
}
[/code]
extensions.ael
[code]context 198276790 {
_50X. => {
Dial(IAX2/NIC-NIKE-505-01/${EXTEN},60);
Congestion();
}
_51X. => {
Dial(IAX2/NIC-NIKE-505-01/${EXTEN},60);
Congestion();
}
}[/code]
sip.conf
[code][198.27.67.90]
type=friend
host=198.27.67.90
context=198276790
call-limit=90
[/code]
SIP DEBUG
[code]<— SIP read from UDP:198.27.67.90:5061 —>
INVITE sip:5050582266270@188.166.42.31;user=phone SIP/2.0
Via: SIP/2.0/UDP 198.27.67.90:5061;rport;branch=z9hG4bK-1452200248-3843144968-1416655780-3635550674
From: sip:5@198.27.67.90:5061;user=phone;tag=4234858808-3843144968-1416655780-3635550674
To: sip:5050582266270@188.166.42.31;user=phone
Call-ID: 38d16b2408bd11e5a4737054d219b2d8@198.27.67.90
CSeq: 1 INVITE
Contact: sip:5@198.27.67.90:5061;user=phone
Content-Type: application/sdp
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, REFER, REGISTER, UPDATE
Max-Forwards: 70
User-Agent: MERA MVTS3G v.4.3.0-34
Cisco-Guid: 3949742632-146674149-2956292180-3532626974
Content-Length: 196
v=0
o=- 1433204449 1433204449 IN IP4 198.27.67.90
s=-
c=IN IP4 198.27.67.90
t=0 0
m=audio 57730 RTP/AVP 18
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=sendrecv
a=silenceSupp:off - - - -
<------------->
— (13 headers 10 lines) —
Sending to 198.27.67.90:5061 (no NAT)
Using INVITE request as basis request - 38d16b2408bd11e5a4737054d219b2d8@198.27.67.90
No matching peer for ‘5’ from ‘198.27.67.90:5061’
== Using SIP RTP CoS mark 5
Found RTP audio format 18
Found audio description format G729 for ID 18
Capabilities: us - 0x101 (g723|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 198.27.67.90:57730
Looking for 5050582266270 in default (domain 188.166.42.31)
<— Reliably Transmitting (no NAT) to 198.27.67.90:5061 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 198.27.67.90:5061;branch=z9hG4bK-1452200248-3843144968-1416655780-3635550674;received=198.27.67.90;rport=5061
From: sip:5@198.27.67.90:5061;user=phone;tag=4234858808-3843144968-1416655780-3635550674
To: sip:5050582266270@188.166.42.31;user=phone;tag=as328acc1d
Call-ID: 38d16b2408bd11e5a4737054d219b2d8@198.27.67.90
CSeq: 1 INVITE
Server: MS Thin Technologies Termination Solution
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘38d16b2408bd11e5a4737054d219b2d8@198.27.67.90’ in 32000 ms (Method: INVITE)
<— SIP read from UDP:198.27.67.90:5061 —>
ACK sip:5050582266270@188.166.42.31;user=phone SIP/2.0
Via: SIP/2.0/UDP 198.27.67.90:5061;rport;branch=z9hG4bK-1452200248-3843144968-1416655780-3635550674
From: sip:5@198.27.67.90:5061;user=phone;tag=4234858808-3843144968-1416655780-3635550674
To: sip:5050582266270@188.166.42.31;user=phone;tag=as328acc1d
Call-ID: 38d16b2408bd11e5a4737054d219b2d8@198.27.67.90
CSeq: 1 ACK
Max-Forwards: 70
User-Agent: MERA MVTS3G v.4.3.0-34
Content-Length: 0
[/code]