[asterisk-users] adding a second extension

[Stephen Reese]

On Sun, Oct 19, 2008 at 4:11 PM, Juan Rodríguez <jerdguez at gmail.com> wrote:
> First, I think is better to to have SIP/vitel-outbound/${EXTEN} than
> having SIP/${EXTEN}@vitel-outbound
> And try issuing SIP SET DEBUG on the cli to see what happens when making the
> call, post back what you see making calls from 101 to 102 and 102 to 101.
> Having the sip.conf sould help on getting whats going on.

Here are the relevant parts of the sip.conf

[general]
register => rsreese:test at inbound18.vitelity.net:5060/rsreese
context=default                 ; Default context for incoming calls
;allowguest=no                  ; Allow or reject guest calls (default is yes)
;match_auth_username=yes        ; if available, match user entry using the
                                ; 'username' field from the authentication line
                                ; instead of the From: field.
allowoverlap=no                 ; Disable overlap dialing support.
(Default is yes)
;allowtransfer=no               ; Disable all transfers (unless
enabled in peers or users)
                                ; Default is enabled
realm=ns1.neocipher.net         ; Realm for digest authentication
                                ; defaults to "asterisk". If you set a
system name in
                                ; asterisk.conf, it defaults to that system name
                                ; Realms MUST be globally unique
according to RFC 3261
                                ; Set this to your host name or domain name
bindport=5060                   ; UDP Port to bind to (SIP standard
port for unencrypted UDP
                                ; and TCP sessions is 5060)
                                ; bindport is the local UDP port that
Asterisk will listen on
bindaddr=0.0.0.0                ; IP address to bind UDP listen socket
to (0.0.0.0 binds to all)
                                ; You can specify port here too, like
123.123.123.123:5080
domain=neocipher.net

[101]
type=friend ; allows incoming and outgoing calls
username=101
secret=pass
mailbox=101
callerid=\"Stephen\" <101>
host=dynamic
nat=yes
dtmfmode=rfc2833
canreinvite=no
reinvite=no
qualify=yes

[102]
type=friend ; allows incoming and outgoing calls
username=102
secret=pass
mailbox=102
callerid=\"Stephen\" <102>
host=dynamic
nat=yes
dtmfmode=rfc2833
canreinvite=no
reinvite=no
qualify=yes


[vitel-inbound] ;(exact format/casing required)
type=friend
host=inbound18.vitelity.net
context=inbound ;(ext-did or from-trunk for A at H)
username=rsreese
secret=test
allow=all
;insecure=very
insecure = invite
canreinvite=no

[vitel-outbound] ;(exact format/casing required)
type=friend
host=outbound.vitelity.net
context=inbound ;(ext-did or from-trunk for A at H)
username=rsreese
fromuser=rsreese
trustrpid=yes
sendrpid=yes
secret=test
allow=all
canreinvite=no

Here is the sip debug error:

<------------>
    -- Executing [102 at default:1] Dial("SIP/102-08266f60",
"'SIP/102',20") in new stack
[Oct 19 16:21:21] WARNING[26690]: channel.c:3470 ast_request: No
channel type registered for ''SIP'
[Oct 19 16:21:21] WARNING[26690]: app_dial.c:1450 dial_exec_full:
Unable to create channel of type ''SIP' (cause 66 - Channel not
implemented)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [102 at default:2] Hangup("SIP/102-08266f60", "") in new stack
  == Spawn extension (default, 102, 2) exited non-zero on 'SIP/102-08266f60'
Scheduling destruction of SIP dialog
'NTQxOTRlZjI2MmEzMWYyOTliZmI2ZDJkMTVkOTYzZDQ.' in 32000 ms (Method:
INVITE)
ns1*CLI>
<--- Reliably Transmitting (NAT) to 68.156.63.118:56558 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP
68.156.63.118:56558;branch=z9hG4bK-d8754z-e0a3d830adb35401-1---d8754z-;received=68.156.63.118;rport=56558
From: <sip:102 at neocipher.net>;tag=7d39014c
To: "102"<sip:102 at neocipher.net>;tag=as4f32f2a7
Call-ID: NTQxOTRlZjI2MmEzMWYyOTliZmI2ZDJkMTVkOTYzZDQ.
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6.0.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:102 at 209.251.157.91>
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