[solved]Asterisk to SIP softswitch problem


#1

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


#2

sunspec,

This is a little off your topic but I was wondering how you like your Metaswitch? We’ve been looking at them and CopperCom to update our Nortel DMS10. We’ve pretty much settled on Metaswtich at this point though.

Are yo using SIP trunks between Asterisk and your Metaswitch? If so, how are they working?

Thanks,
Charlie