I am trying to get Asterisk setup as a voicemail server by sending calls directly to a vm box. Everything works great as long as the calls are local to the Asterisk box. My upstream connection is to a Metaswitch Softswitch which I help manage. When the Metaswitch transfers a call to Asterisk I see all the SIP information but when the Asterisk replies I am getting 404 not found. Could someone please look at this and tell me what might be wrong?
Also, Asterisk is erroring out saying ‘Insufficient information for SDP (m = ‘’, c = ‘’)’
However, if you look at the following sip debug you will see the info the Metaswitch sent. Any help would be greatly appreciated.
Thaniks,
<-- SIP read from 192.168.2.10:5060:
INVITE sip:6014500666@192.168.2.4:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.2.10;rport;branch=z9hG4bK-8f5644d17b74153fa37d4a220d0c825a-192.168.2.10-1
Allow-Events: message-summary
Allow-Events: refer
Max-Forwards: 70
Call-ID: 82E0E719@192.168.2.10
From: sip:6014504236@192.168.2.10;transport=udp;tag=192.168.2.10+1+1fdc09+903d96da;isup-oli=00
To: sip:6014500666@192.168.2.10
CSeq: 536294906 INVITE
Expires: 180
Organization:
Supported: 100rel
Content-Length: 124
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REGISTER, OPTIONS, PRACK, UPDATE, SUBSCRIBE, NOTIFY, REFER
Referred-By: sip:6014500759@192.168.2.10
Contact: sip:6014504236@192.168.2.10;transport=udp;isup-oli=00
Diversion: sip:6014500759@192.168.2.10;reason=no-answer
P-Charging-Vector: icid-value=0@192.168.2.10
v=0
o=- 4293225616 4293225616 IN IP4 192.168.2.14
s=-
c=IN IP4 192.168.2.14
t=0 0
m=audio 32782 RTP/AVP 0
a=ptime:20
— (19 headers 7 lines)—
Using INVITE request as basis request - 82E0E719@192.168.2.10
Sending to 192.168.2.10 : 5060 (non-NAT)
Found peer 'Metaswitch’
Found RTP audio format 0
Peer audio RTP is at port 192.168.2.14:32782
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Looking for 6014500666 in from-pstn (domain 192.168.2.4)
RDNIS is 6014500759
Reliably Transmitting (no NAT) to 192.168.2.10:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.2.10;rport;branch=z9hG4bK-8f5644d17b74153fa37d4a220d0c825a-192.168.2.10-1;received=192.168.2.10
From: sip:6014504236@192.168.2.10;transport=udp;tag=192.168.2.10+1+1fdc09+903d96da;isup-oli=00
To: sip:6014500666@192.168.2.10;tag=as7e647f87
Call-ID: 82E0E719@192.168.2.10
CSeq: 536294906 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: sip:6014500666@192.168.2.4
Content-Length: 0
If more information is needed I can supply. BTW, here is my SIP config:
[Metaswitch]
type=friend
host=dynamic
fromuser=6014500666
context=from-pstn