[asterisk-users] adding a second extension

[Stephen Reese]

I am now using a Cisco phone for the second extension (102). I am able
to contact 102 from 101 but not the other way around. The error seems
less severe now:

  == Using SIP RTP CoS mark 5
    -- Executing [101 at default:1] Dial("SIP/102-0825b118",
"SIP/101/20") in new stack
  == Using SIP RTP CoS mark 5
    -- Called 101/20
    -- SIP/101-08221a78 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [101 at default:2] Hangup("SIP/102-0825b118", "") in new stack
  == Spawn extension (default, 101, 2) exited non-zero on 'SIP/102-0825b118'

So maybe it's just a config issue now?

[general]
static=yes
writeprotect=no
clearglobalvars=no

[globals]
CONSOLE=Console/dsp                             ; Console interface for demo
IAXINFO=guest                                   ; IAXtel username/password
TRUNK=DAHDI/G2                                  ; Trunk interface
TRUNKMSD=1                                      ; MSD digits to strip
(usually 1 or 0)


[default]


exten => 101,1,Dial(SIP/101/20)
exten => 101,n,Hangup
exten => 101,n,Voicemail(101 at default)


exten => 102,1,Dial(SIP/102,20)
exten => 102,n,Hangup
exten => 102,n,Voicemail(102 at default)

exten=>*98,1,VoiceMailMain(${CALLERIDNUM}@${CONTEXT})

include => inbound
include => outgoing

[inbound]
exten => 9045622082,1,Goto(default,101,1)

[outgoing]

exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=9045622082)
exten => _1NXXNXXXXXX,n,Set(CALLERID(name)="Stephen Reese")
exten => _1NXXNXXXXXX,n,Dial(SIP/${EXTEN}@vitel-outbound)

exten => _NXXXXXX,1,Set(CALLERID(num)=9045622082)
exten => _NXXXXXX,n,Set(CALLERID(name)="Stephen Reese")
exten => _NXXXXXX,n,Dial(SIP/1904${EXTEN}@vitel-outbound)

exten => _NXXNXXXXXX,1,Set(CALLERID(num)=9045622082)
exten => _NXXNXXXXXX,n,Set(CALLERID(name)="Stephen Reese")
exten => _NXXNXXXXXX,n,Dial(SIP/1${EXTEN}@vitel-outbound)

exten => _011.,1,Set(CALLERID(num)=9045622082)
exten => _011.,n,Set(CALLERID(name)="Stephen Reese")
exten => _011.,n,Dial(SIP/${EXTEN}@vitel-outbound)

exten => _911,1,Set(CALLERID(num)=9045622082)
exten => _911,n,Set(CALLERID(name)="Stephen Reese")
exten => _911,n,Dial(SIP/911 at vitel-outbound)



On Mon, Oct 20, 2008 at 11:06 AM, Stephen Reese <rsreese at gmail.com> wrote:
> On Mon, Oct 20, 2008 at 10:37 AM, Juan Rodríguez <jerdguez at gmail.com> wrote:
>> I do not think NAT is the problem, NAT normally gives you problems like one
>> way audio or no registration.
>> Try calling the SIP/102 on other extension:
>> ;TEST
>> exten => 1002,1,Dial(SIP,102|20)
>> exten => 1002,n,Hangup()
>>  instead of:
>>
>> exten => 102,1,Dial...
>> But this is a very strange error... Check if there is no other definition of
>> default having 102 on it because Asterisk is going to merge the extensions.
>
> I get the following when trying to dial 1002 from 101. I've attached
> my extensions.conf file in-case there is something else that is
> conflicting as you mentioned.
>
>    -- Executing [1002 at default:1] Dial("SIP/101-082aca90",
> "SIP/102/20") in new stack
>  == Using SIP RTP CoS mark 5
>    -- Called 102/20
> [Oct 20 11:03:33] WARNING[20285]: chan_sip.c:2787 retrans_pkt: Maximum
> retries exceeded on transmission
> 60a5e13455b795df5a93427152cbfff1 at 209.251.157.91 for seqno 102
> (Critical Request) -- See doc/sip-retransmit.txt.
> [Oct 20 11:03:33] WARNING[20285]: chan_sip.c:2814 retrans_pkt: Hanging
> up call 60a5e13455b795df5a93427152cbfff1 at 209.251.157.91 - no reply to
> our critical packet (see doc/sip-retransmit.txt).
>  == Everyone is busy/congested at this time (1:0/0/1)
>    -- Executing [1002 at default:2] Hangup("SIP/101-082aca90", "") in new stack
>  == Spawn extension (default, 1002, 2) exited non-zero on 'SIP/101-082aca90'
>