VoIP Mailing List Archives

[ragb at mail.com]

Hi all, new to the list and this is probably a basic question and
couldn't find anything clear googling around but I don't know how to
handle calls to sip extensions not defined on sip.conf while using
pattern matching. On my example I have sip extensions 10, 11, 12, and 13
on sip.conf. On a basic extension.conf I set up a pattern starting with
"1" and a second digit should dial the sip extension entered by the user
and if the user don't pick up or is unavailable the call goes to the
user voicemail and then hangup. This basic setup can be seen next:

[default]
exten => _1X,1,Dial(SIP/${EXTEN},10)
exten => _1X,2,VoiceMail(${EXTEN}@default,u)
exten => _1X,3,HangUp()

Now, what happens if the user dials 15? Then the pattern is applied and
the asterisk tries to dial that sip extension that doesn't exist, the
next step that is the voicemail also fails as 15 is not defined on
voicemail.conf and finally reaches the last step where it hang ups. This
can be seen on the cli output copied below:

astbox*CLI>
-- Executing [15 at default:1] Dial("SIP/10-0820d8e0", "SIP/15|10") in new
stack
[Mar 21 19:57:48] WARNING[14321]: chan_sip.c:2860 create_addr: No such
host: 15
[Mar 21 19:57:48] WARNING[14321]: app_dial.c:1111 dial_exec_full: Unable
to create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [15 at default:2] VoiceMail("SIP/10-0820d8e0", "15 at default|u")
in new stack
[Mar 21 19:57:48] WARNING[14321]: app_voicemail.c:2808 leave_voicemail:
No entry in voicemail config file for '15'
-- Executing [15 at default:3] Hangup("SIP/10-0820d8e0", "") in new stack
== Spawn extension (default, 15, 3) exited non-zero on 'SIP/10-0820d8e0'
astbox*CLI>
What I am looking for is to play Playback(pbx-invalid) if a user enters a
sip extension not created. I've been testing a few options using
DIALSTATUS, AVAILSTATUS and their values but without luck as if the sip
phone 11 is not registered the pbx-invalid message.

Thansk for reading and any suggestion will be welcome.

Richard

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[asterisk.org at sedwar...]

On Fri, 21 Mar 2008, Ricardo B. wrote:

[ragb at mail.com]

Thanks Steve, your solution works but I am looking for is something more
general. The example I posted is a simplified one and on the real one I
am using extensions on the 5XXX scenario so I can have 5000 to 5999
range. You answer for just 10 extension is great but for a 1000 and when
not all extensions on the 5XXX are set up then it is not optimal to
define each extension manually or add exceptions manually and I was
looking to see if there a command or state that I could use like
DIALSTATUS, etc to see if the dialled extension doesn't exist and then
jump to the invalid prompt.

Regards
Richard

----- Original Message -----
From: "Steve Edwards"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Subject: Re: [asterisk-users] Calls to sip extensions not defined
Date: Fri, 21 Mar 2008 15:12:43 -0700 (PDT)
On Fri, 21 Mar 2008, Ricardo B. wrote:

[asterisk.org at sedwar...]

On Sat, 22 Mar 2008, R. B. wrote:

[sst at sil.at]

hello,

why do you build a realtime configuration loading the sip users and
extensions from a database?

if you want to use the username as extension it would be quit simple
looking like this:

exten => _X.,1,Select user from DB which has ${EXTEN}
exten => _X.n,GotoIf(Uservar = ""?nouser)
exten => _X.n,Dial(SIP/${EXTEN} or userval from db,60)
exten => _X.,n,Voicemail(${EXTEN} or userval from db)
exten => _X.,n,Hangup

exten => _X.,n(nouser),Playback(no_user_found)
exten => _X.,hangup

i am using this system on several servers serving more than 7000 active
users which more functionality than voicemail and sip dialing.

if you are interested in this solution give me a mail and i could send
you a copy of the config files and db structure you need.

best regards

steve smith

R. B. schrieb:

[eric at fnords.org]

exten => i is for IVRs.

You would need a wildcard catchall extension like the one below.
Unfortunately you are doing the classic newbie mistake of thinking you
can have a simple dialplan by making the SIP user/account ID be the same
as the extension. Eventually you will realize this is a bad idea.

If you think you have to use wildcards (you don't) then you would want
to check the value of DIALSTATUS or HANGUPCAUSE after the Dial, before
the Voicemail and handle the exception using a Goto.

[default]
exten = _1[1-3],1, dial(sip/${EXTEN},10)
exten = _1[1-3],n, voicemail(${EXTEN}@default,u)
exten = _1[1-3],n, hangup

exten = _XXX,1, playback(pbx-invalid)
exten = _XXX,n, hangup
Steve Edwards wrote:

[ragb at mail.com]

Thanks all, I will follow the recommendation of moving my sip extensions
to a db or use perl to check if the extension exists. I run tests with
dialstatus, hangupcase, chanisavail and the return codes are the same if
the extension doesn't exist on sip.conf or if the extension is not
registered.

Regards
Richard

----- Original Message -----
From: "Eric Wieling"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Subject: Re: [asterisk-users] Calls to sip extensions not defined
Date: Sun, 23 Mar 2008 07:50:21 -0500
exten => i is for IVRs.

You would need a wildcard catchall extension like the one below.
Unfortunately you are doing the classic newbie mistake of thinking
you
can have a simple dialplan by making the SIP user/account ID be the
same
as the extension. Eventually you will realize this is a bad idea.

If you think you have to use wildcards (you don't) then you would
want
to check the value of DIALSTATUS or HANGUPCAUSE after the Dial,
before
the Voicemail and handle the exception using a Goto.

[default]
exten = _1[1-3],1, dial(sip/${EXTEN},10)
exten = _1[1-3],n, voicemail(${EXTEN}@default,u)
exten = _1[1-3],n, hangup

exten = _XXX,1, playback(pbx-invalid)
exten = _XXX,n, hangup


Steve Edwards wrote:

[Guest]

Ricardo B. wrote: