[asterisk-users] Metaswitch help needed

[Steve Edwards]

I'm attempting to connect to a Metaswitch, inbound only (at this time). 
The Metaswitch is the only "connection" (at this time).

All I'm getting so far is a bunch of "OPTION" messages which my Asterisk 
box replies to but I don't get inbound calls.

Here's my sip.conf. As you can see I've been trying a bunch of different 
options without success :(

(206.b.c.d is the address of my Asterisk box. 172.b.c.d is the address of 
the Metaswitch)

[general]
          disallow                       = all
         allguest                        = yes
         allow                           = all
         allowguest                      = yes
         autocreatepeer                  = yes
         autodomain                      = yes
         bindaddr                        = 206.b.c.d
         bindport                        = 5060
         callerid                        = "metaswitch" <>
         canreinvite                     = no
         context                         = test
         dtmfmode                        = rfc2833
         host                            = 172.b.c.d
;       insecure                        = invite
         insecure                        = very
         nat                             = never
;       nat                             = yes
         port                            = 5060
         qualify                         = yes
         qualifysmoothing                = yes
         realm                           = 206.b.c.d
;       realm                           = metaswitch
         regcontext                      = test
         secret                          = metaswitch
         sipdebug                        = yes
         type                            = friend
;       type                            = peer
;       type                            = user
         username                        = metaswitch

Here's the console SIP debug messages:

<-- SIP read from 172.b.c.d:5060: 
OPTIONS sip:metaswitch at 206.b.c.d:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.b.c.d:5060;rport;branch=z9hG4bK-17eb587208b656d9c2fbd516b5e5401e-172.b.c.d-1
Allow-Events: message-summary
Allow-Events: refer
Allow-Events: dialog
Allow-Events: line-seize
Max-Forwards: 70
Call-ID: BECE8AC6 at 172.b.c.d
From: <sip:metaswitch at 172.b.c.d:5060;transport=udp>;tag=172.b.c.d+1+0+22022a3b
CSeq: 445762257 OPTIONS
Organization: 
Supported: 100rel
Content-Length: 0
Contact: <sip:metaswitch at 172.b.c.d:5060;transport=udp>
To: <sip:metaswitch at 206.b.c.d>


--- (15 headers 0 lines) ---
Looking for metaswitch in test (domain 206.b.c.d)
Transmitting (no NAT) to 172.b.c.d:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.b.c.d:5060;rport;branch=z9hG4bK-17eb587208b656d9c2fbd516b5e5401e-172.b.c.d-1;received=172.b.c.d
From: <sip:metaswitch at 172.b.c.d:5060;transport=udp>;tag=172.b.c.d+1+0+22022a3b
To: <sip:metaswitch at 206.b.c.d>;tag=as6a59273b
Call-ID: BECE8AC6 at 172.b.c.d
CSeq: 445762257 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:206.b.c.d>
Accept: application/sdp
Content-Length: 0


---
Destroying call 'BECE8AC6 at 172.b.c.d'

And this is what I get from "sudo ngrep -s 2048 port 5060":

U 172.b.c.d:5060 -> 206.b.c.d:5060
   OPTIONS sip:metaswitch at 206.b.c.d:5060;transport=udp SIP/2.0..Via: SIP/2.0/UDP 172.b.c.d:5060;rport;branch=z9hG4bK-815d5107
   ec165bef012bcfebc6e214fd-172.b.c.d-1..Allow-Events: message-summary..Allow-Events: refer..Allow-Events: dialog..Allow-Events:
   line-seize..Max-Forwards: 70..Call-ID: 7DF6F0D6 at 172.b.c.d..From: <sip:metaswitch at 172.b.c.d:5060;transport=udp>;tag=172.b.c.d
   +1+0+85ece24c..CSeq: 528990954 OPTIONS..Organization: ..Supported: 100rel..Content-Length: 0..Contact: <sip:metaswitch at 172.b.c.d .2:5060;transport=udp>..To: <sip:metaswitch at 206.b.c.d>.... 
#
U 206.b.c.d:5060 -> 172.b.c.d:5060
   SIP/2.0 200 OK..Via: SIP/2.0/UDP 172.b.c.d:5060;rport;branch=z9hG4bK-815d5107ec165bef012bcfebc6e214fd-172.b.c.d-1;received=17
   2.16.1.2..From: <sip:metaswitch at 172.b.c.d:5060;transport=udp>;tag=172.b.c.d+1+0+85ece24c..To: <sip:metaswitch at 206.b.c.d>;
   tag=as26804e9e..Call-ID: 7DF6F0D6 at 172.b.c.d..CSeq: 528990954 OPTIONS..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OP
   TIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Contact: <sip:206.b.c.d>..Accept: application/sdp..Content-Length: 0.... 
#

Any clues will be appreciated :)

Thanks in advance,
------------------------------------------------------------------------
Steve Edwards      sedwards at sedwards.com      Voice: +1-760-468-3867 PST
Newline                                             Fax: +1-760-731-3000