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:[EMAIL PROTECTED]: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: [EMAIL PROTECTED]
From: <sip:[EMAIL PROTECTED]:5060;transport=udp>;tag=172.b.c.d+1+0+22022a3b
CSeq: 445762257 OPTIONS
Organization:
Supported: 100rel
Content-Length: 0
Contact: <sip:[EMAIL PROTECTED]:5060;transport=udp>
To: <sip:[EMAIL PROTECTED]>
--- (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:[EMAIL PROTECTED]:5060;transport=udp>;tag=172.b.c.d+1+0+22022a3b
To: <sip:[EMAIL PROTECTED]>;tag=as6a59273b
Call-ID: [EMAIL PROTECTED]
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 '[EMAIL PROTECTED]'
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:[EMAIL PROTECTED]: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: [EMAIL PROTECTED]: <sip:[EMAIL
PROTECTED]:5060;transport=udp>;tag=172.b.c.d
+1+0+85ece24c..CSeq: 528990954 OPTIONS..Organization: ..Supported: 100rel..Content-Length: 0..Contact: <sip:[EMAIL PROTECTED] .2:5060;transport=udp>..To: <sip:[EMAIL PROTECTED]>....
#
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:[EMAIL
PROTECTED]:5060;transport=udp>;tag=172.b.c.d+1+0+85ece24c..To: <sip:[EMAIL
PROTECTED]>;
tag=as26804e9e..Call-ID: [EMAIL PROTECTED]: 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 [EMAIL PROTECTED] Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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