Dmitry, To elaborate on this a little. Suppose you have a trunk from a
VoIP provider, but subscribe to several DIDs with the same provider... one
in New York, another in California, maybe a toll free number. And you want
to direct the incoming call to different places based on the number that the
caller dialed.
exten = _800X.,1,GoTo(wherever I'll deal with inbound toll free)
exten = _212X.,1,GoTo(calls to my New York number)
exten = _415X.,1,GoTo(calls to my San Francisco number)
exten = _X.,1,GoTo(calls to any other DID)
exten = s,1,GoTo(handle case where no DID is provided)
This would allow a business to direct calls to the appropriate customer
service for example depending upon where the caller is located... or could
prioritize calls to the 800 number in a queue (because you're paying for
them, not the caller). etc etc.
David
On Thu, Sep 17, 2009 at 12:52 PM, David Kerr <da...@kerr.net> wrote:
> You can do it in the gui... incoming rules tab. You can add multiple
> incoming rules for each trunk. just enter 's' for the pattern. When you
> have multiple rules for the same trunk, the gui also lets you reorder them
> (up/down arrows appear). The tooltip on the gui incoming rules create
> dialog suggests use 's' or '_X.' pattern to define a rule for
> all incoming calls. I think it would be more accurate to state use 's' *and
> *'_X.' because both are necessary to be bullet proof.
>
> David
>
>
> On Thu, Sep 17, 2009 at 12:23 PM, Dmitry Komarov <d...@dmit.lv> wrote:
>
>> 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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