[asterisk-users] OUTBOUND CALLER ID
mahesh katta
maheshkatta at flexydial.com
Tue May 10 08:43:23 CDT 2011
Sir,
On Tue, May 10, 2011 at 6:15 PM, A J Stiles
<asterisk_list at earthshod.co.uk>wrote:
> (Reformatted *again*. The proper place to post your reply is *below* the
> message or section to which you are replying, so it reads like: question,
> answer, question, answer. This makes things much easier for anyone with a
> similar problem in future, trying to make sense of old messages in the
> archives.)
>
> sorry sir I am new to asterisk,
> On Tuesday 10 May 2011, mahesh katta wrote:
> > On Tue, May 10, 2011 at 4:49 PM, A J Stiles
> >
> > <asterisk_list at earthshod.co.uk>wrote:
> > > "Not working" can mean a lot of things.
> > >
> > > So, let's start at the beginning. Have you ever actually managed to
> get
> > > an outgoing call to work *at all* -- i.e., successfully placed a call
> > > onto a trunk, made an external phone ring and been able to speak to the
> > > other party,
> > > even if the caller ID that shew up was incorrect ?
> > >
> > > Assuming so, please show the dialplan section that worked for that.
> > >
> > >
> > > Otherwise, you need to go back to first principles and work out exactly
> > > what
> > > you are doing wrong. We need to start from a point where you are able
> to
> > > place external calls *before* we can discuss how to set the desired
> > > caller ID
> > > to appear on the receiving end.
> > >
> > > Obligatory car analogy: There's no point arguing over whether
> motorways
> > > are
> > > quicker than A-roads, if you can't even get the engine to start.
> > >
> > >
> > > Next, please describe how to determine, based on the internal
> extension,
> > > what
> > > caller ID number should be presented to the outside world.
> > >
> > A.j Sir,
> >
> > I am using vicidial server asterisk box. it has asterisk 1.27v ,
> >
> > In sip configuration is extensiosn like below
> >
> > [5001]
> > username=5001
> > secret=1234
> > mailbox=5001
> > type=friend^M
> > host=dynamic^M
> > canreinvite=no^M
> > qualify=yes^M
> > nat=yes^M
> > context=default
> >
> > 5001 to 5099
> >
> > and PRI pilot no. is 4457900 in dubai , telcom give a DID's 4457900 to
> > 4457999,
> >
> > when i am dialing from pstn on DID's(inbound) that will connecting (i was
> > configured with sip means when i dial the 4457901 from outside i will get
> a
> > call on 5001 extension)
>
> O.K. So what are you seeing in the console when you place an outgoing
> call?
> There should be two NoOp() lines per call; one with the original extension
> number, and another with what we are trying to set the caller ID to.
>
> If it helps, change the lines for steps 1 and 3 as follows:
>
> exten => _90XXXXXXXXX,1,NoOp(Int exten:${CALLERID(num)})
> exten => _90XXXXXXXXX,3,NoOp(Ext ident:${outgoing_ident})
>
> so it is more obvious what they are supposed to represent. Run
> asterisk -vvvvvvvvvr in a console, reload the dialplan, capture some output
> as you make calls from various extensions, and paste it here. The
> important
> question is: Do the "Ext ident" numbers look like what they are supposed
> to?
> And if not, then how do they differ from what they are supposed to look
> like?
>
> NoOp("SIP/5001-b792a1a8", "Int exten:044578999") in new
stack
-- Executing Set("SIP/5001-b792a1a8", "outgoing_ident=044578999") in new
stack
-- Executing NoOp("SIP/5001-b792a1a8", "Ext ident:044578999") in new
stack
-- Executing Set("SIP/5001-b792a1a8", "CALLERID(name)=044578999") in new
stack
-- Executing AGI("SIP/5001-b792a1a8", "agi://127.0.0.1:4577/call_log")
in new
stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning
0
-- Executing Set("SIP/5001-b792a1a8", "CALLERID(num)=044578999") in new
stack
-- Executing MixMonitor("SIP/5001-b792a1a8",
"/var/spool/asterisk/astrec/20110510-172343-044578999-90559566768-1305033823.129.gsm|av(0)V(0)")
in new stack
-- Executing Dial("SIP/5001-b792a1a8", "Zap/g0/0559566768||tTo") in new
stack
-- Requested transfer capability: 0x00 -
SPEECH
-- Called
g0/0559566768
== Begin MixMonitor Recording
SIP/5001-b792a1a8
-- Zap/1-1 is proceeding passing it to
SIP/5001-b792a1a8
-- Zap/1-1 is making progress passing it to
SIP/5001-b792a1a8
-- Hungup
'Zap/1-1'
== Spawn extension (default, 90559566768, 8) exited non-zero on
'SIP/5001-b792a1a8'
-- Executing DeadAGI("SIP/5001-b792a1a8", "agi://
127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL----------")
in new stack
-- AGI Script agi://
127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL----------completed,
returning 0
== End MixMonitor Recording SIP/5001-b792a1a8
>
>
> > here Problem when i dial (out bound) out of box from 5001 extensions the
> > caller id showing pilot no. (4457900) .
> > I need to display that DID's each extension.
>
> The usual cause of the caller ID not being set to what you want, is that
> you
> tried to set it to some number that is not allocated to you, and your telco
> changed it to some default. (Otherwise, you could fake your outgoing
> caller
> ID, which nobody wants.)
>
> sir i complaint to telco for this they are said that is set on
router.
>
> > exten => _90XXXXXXXXX,1,NoOp(${CALLERID(num)})
> > exten => _90XXXXXXXXX,2,Set(outgoing_ident=0445789${CALLERID(num):-2})
> > exten => _90XXXXXXXXX,3,NoOp(${outgoing_ident})
> > exten => _90XXXXXXXXX,4,Set(CALLERID(name)=${outgoing_ident})
> > exten => _90XXXXXXXXX,5,AGI(agi://127.0.0.1:4577/call_log)
> > exten => _90XXXXXXXXX,6,Set(CALLERID(num)=${outgoing_ident})
> > exten =>
> >
> _90XXXXXXXXX,7,MixMonitor(/var/spool/asterisk/astrec/${TIMESTAMP}-${CALLERI
> >DNUM}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0)) exten =>
> > _90XXXXXXXXX,8,Dial(${TRUNK}/${EXTEN:1},,tTo)
> > exten => _90XXXXXXXXX,9,Hangup
>
> Right, I'm confused now. This dialplan segment is expecting for you to
> dial
> 90, followed by nine digits. But the numbers as which you want to
> identify -- 4457901 to 4457999 -- are only seven digits long.
>
> sir we are dialing out of world when we dial out of world 4457901 to
4457999 will show them. these are DID's which is telco provided to us.
> I'm guessing there probably is an STD code consisting of 0 followed by two
> digits to indicate the town, and then a 7-digit local number within the
> town.
> In which case, I seem to have made a mistake at step 2, somehow letting an
> extraneous figure 8 get in. Try:
>
> exten => _90XXXXXXXXX,2,Set(outgoing_ident=44579${CALLERID(num):-2})
>
> Executing NoOp("SIP/5001-b790fc10", "Int exten:044578999") in new
stack
-- Executing Set("SIP/5001-b790fc10", "outgoing_ident=44578999") in new
stack
-- Executing NoOp("SIP/5001-b790fc10", "Ext ident:44578999") in new
stack
-- Executing Set("SIP/5001-b790fc10", "CALLERID(name)=44578999") in new
stack
-- Executing AGI("SIP/5001-b790fc10", "agi://127.0.0.1:4577/call_log")
in new
stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning
0
-- Executing Set("SIP/5001-b790fc10", "CALLERID(num)=44578999") in new
stack
-- Executing MixMonitor("SIP/5001-b790fc10",
"/var/spool/asterisk/astrec/20110510-174142-44578999-90559566768-1305034902.131.gsm|av(0)V(0)")
in new stack
-- Executing Dial("SIP/5001-b790fc10", "Zap/g0/0559566768||tTo") in new
stack
-- Requested transfer capability: 0x00 -
SPEECH
-- Called
g0/0559566768
== Begin MixMonitor Recording
SIP/5001-b790fc10
-- Zap/1-1 is proceeding passing it to
SIP/5001-b790fc10
-- Zap/1-1 is making progress passing it to
SIP/5001-b790fc10
-- Hungup
'Zap/1-1'
== Spawn extension (default, 90559566768, 8) exited non-zero on
'SIP/5001-b790fc10'
-- Executing DeadAGI("SIP/5001-b790fc10", "agi://
127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL----------")
in new stack
-- AGI Script agi://
127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL----------completed,
returning 0
== End MixMonitor Recording SIP/5001-b790fc10
> If that does not seem to work, try prefixing the "44579" with the STD code
> for
> your town, with and without the leading zero. And don't forget to reload
> the
> dialplan between edits :)
>
>
> --
> AJS
>
> Answers come *after* questions.
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110510/4d0616ed/attachment-0001.htm>
More information about the asterisk-users
mailing list