[asterisk-users] Metaswitch help needed

2007-03-21 Thread 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:[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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Re: [asterisk-users] Metaswitch help needed

2007-03-21 Thread Steve Murphy
On Wed, 2007-03-21 at 06:15 -0700, Steve Edwards wrote:
 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 :(

Good news! I'm not the only guy to try to link his Asterisk box to a
Metaswitch!!

There is a wiki page on voip-info that might be helpful to you--

http://www.voip-info.org/wiki/view/Asterisk+How+to+connect+to+Metaswitch

If your scenario isn't included, please add it to the wiki when you
figure it out!

murf

-- 
Steve Murphy
Software Developer
Digium

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