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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Come build with us! The BlackBerry® Developer Conference in SF, CA
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developing skills, take BlackBerry mobile applications to market and stay
ahead of the curve. Join us from November 9-12, 2009. Register now!
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