Thank you for the fast reply! Your suggestion works well and I'm a bit
confused. I have tried several times in different combinations both _X.
and s as the pattern in Incoming Rules without success. There always was
the same 's extension' error. But now I manually add the

exten = s,1,Goto(default,6000,1) 

line under existent GUI-created line

exten = _X.,1,Goto(default,6000,1)

and it works. Does it mean that Asterisk GUI itself is unable to deal
with such configuration?.. Or I have to create two different Incoming
rules under the GUI, one for _X. and another for s?...

---
Best Regards,
Dmitry

On Thu, 2009-09-17 at 11:27 -0400, David Kerr wrote:
> I have seen this before with some VoIP carriers. You need to create an
> incoming rule to catch the 's' extension...
> 
> 
> [DID_trunk_1_default]
> exten = _X.,1,Goto(default,6000,1)
> exten = s,1,Goto(default,6000,1)
> 
> 
> You can do it in the Asterisk GUI, or manually edit extensions.conf
> (and the GUI will pick it up).
> 
> 
> Root cause is that your VoIP carrier is not providing the number that
> your caller called in the inbound SIP request. Normally the extension
> would be set to your phone number (DID) and this would match against
> the _X. extension in the dialplan.  However, if the VoIP carrier
> doesn't provide this then asterisk looks for the start 's' extension
> instead.
> 
> 
> This is nothing to do with Astlinux, Asterisk GUI or Asterisk itself.
>  It is simply dialplan setup required.
> 
> 
> David
> 
> 
> 
> On Thu, Sep 17, 2009 at 10:50 AM, Dmitry Komarov <d...@dmit.lv> wrote:
>         Hello,
>         
>         I found the Asterisk GUI included with 0.6.7 to be absolutely
>         unusable.
>         I tried to create a simple PBX setup with one external SIP
>         trunk to my
>         existing Asterisk PBX box. I did a clean install and
>         configured
>         everything Asterisk-related only by means of Asterisk GUI.
>         Local calls
>         between extensions work fine as well as outgoing calls to my
>         PBX. But
>         when incoming call arrives it always ends up with the
>         following error:
>         
>         [Sep 17 17:22:05] NOTICE[1551]: chan_sip.c:14847
>         handle_request_invite:
>         Call from '999' to extension 's' rejected because extension
>         not found.
>         
>         The strangest thing is that if I even later try to manually
>         create the
>         trunk with all related extensions etc, it does not work the
>         same way and
>         with the same error. Seems that when Asterisk GUI initializes
>         it somehow
>         messes up the config files to trash.
>         
>         Can anyone suggest a quick fix to this problem? I just need to
>         provide
>         my customer with a small PBX setup where Astlinux would fit
>         just fine
>         but simple GUI is important requirement :(
>         
>         Here is an extract from my GUI-generated config files related
>         to trunk
>         support (all the rest is default of Asterisk GUI):
>         
>         users.conf :
>         ------------------------------------------------
>         [trunk_1]
>         host = 10.10.10.5
>         username = 999
>         secret = 1234567890
>         trunkname = mypbx  ; GUI metadata
>         context = DID_trunk_1
>         group = null
>         hasexten = no
>         hasiax = no
>         hassip = yes
>         registeriax = no
>         registersip = yes
>         trunkstyle = voip
>         outboundproxy = 10.10.10.5
>         fromdomain = 10.10.10.5
>         fromuser = 999
>         authuser = 999
>         insecure = invite
>         disallow = all
>         allow = alaw,ulaw,gsm
>         
>         [6000]
>         username = 6000
>         transfer = yes
>         mailbox = 6000
>         call-limit = 100
>         type = peer
>         fullname = User1
>         registersip = no
>         host = dynamic
>         callgroup = 1
>         type = peer
>         context = DLPN_defaultDialPlan
>         cid_number = 6000
>         hasvoicemail = no
>         vmsecret =
>         email =
>         threewaycalling = no
>         hasdirectory = no
>         callwaiting = no
>         hasmanager = no
>         hasagent = no
>         hassip = yes
>         hasiax = yes
>         secret = 1234567890
>         nat = yes
>         canreinvite = no
>         dtmfmode = rfc2833
>         insecure = no
>         pickupgroup = 1
>         disallow = all
>         allow = alaw,gsm
>         autoprov = no
>         label =
>         macaddress =
>         linenumber = 1
>         LINEKEYS = 1
>         ================================================
>         
>         extensions.conf :
>         ------------------------------------------------
>         [DID_trunk_1]
>         include = DID_trunk_1_default
>         
>         [DID_trunk_1_default]
>         exten = _X.,1,Goto(default,6000,1)
>         
>         [CallingRule_outgoing]
>         exten = _XXXXXXX.,1,Macro(trunkdial-failover-0.3,
>         ${trunk_1}/${EXTEN:0},,trunk_1,)
>         
>         [DLPN_defaultDialPlan]
>         include = CallingRule_outgoing
>         include = default
>         include = parkedcalls
>         include = conferences
>         include = ringgroups
>         include = voicemenus
>         include = queues
>         include = voicemailgroups
>         include = directory
>         include = pagegroups
>         include = page_an_extension
>         
>         ---
>         Best Regards,
>         Dmitry
>         
>         
>         
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