[asterisk-users] adding a second extension

[Stephen Reese]

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'
-------------- next part --------------
A non-text attachment was scrubbed...
Name: extensions.conf
Type: application/octet-stream
Size: 27044 bytes
Desc: not available
Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20081020/68c07cf1/attachment.obj