[asterisk-users] Asterisk Inbound calls, multiple SIP accounts, calledID

Wed Apr 8 10:34:06 CDT 2015

what about

exten => s,1,Goto(from-trunk,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1)

regards

2015-04-08 5:45 GMT+00:00 Dmitriy Serov <serov.d.p at gmail.com>:

>  Hi, Andrew.
>
> You are trying to solve two tasks: definition through what line the call
> came and a beautiful display of this information.
> 1. definition through what line the call came. If the username and
> password for inbound and outbound registration the same, then try the
> following:
> a) delete "register" lines.
> b) add option "callbackextension=Company1" to Company1 friend section..
> And in others with their names too.
> or you can change "/s" to "/Company1" in register line.
>
> 2. beautiful display of this information
> a) add option "setvar=fromCompany=Company1" to Company1 friend section..
> b) In dialplan add
> Set(CALLERID(name)=${fromCompany} ${CALLERID(name)})
>
> Maybe this will help?
>
> Dmitiy.
>
> 08.04.2015 2:48, Andrew Galdes пишет:
>
> Hi Dmitriy and others and thanks for your help so far.
>
>  The option "match_auth_username=yes" seems to have had no effect. From
> my reading, this option will try to match the username of the incoming SIP
> account to a section heading. If that is how it must work then i can see a
> big problem. I'm trying to present the receptionist with a nice display of
> which line the call came in on. For example, the receptionist answers calls
> for 8 different companies and would like the phone to display the company
> name that she should announce to the caller.
>
>  Here is a more complete output of an incoming call. I've changed the SIP
> numbers to "Company1', etc, to hide the numbers.
>
>  Connected to Asterisk 10.12.4 currently running on asterisk (pid = 32267)
>> Verbosity is at least 12
>> asterisk*CLI>
>> asterisk*CLI>
>> asterisk*CLI>
>>   == Using SIP RTP CoS mark 5
>>     -- Executing [s at incoming:1] *Set*("*SIP/Company1-00000797*", "*thedid=""NodePhone"<sip:Company2 at sip.internode.on.net
>> <sip%3ACompany2 at sip.internode.on.net>>"*") in new stack
>>     -- Executing [s at incoming:2] *Set*("*SIP/**Company1**-00000797*", "
>> *pseudodid="NodePhone"<sip:** sip:Company2**@sip.internode.on.net
>> <http://sip.internode.on.net>>*") in new stack
>>     -- Executing [s at incoming:3] *Set*("*SIP/**Company1**-00000797*", "
>> *pseudodid="NodePhone"<sip:** sip:Company2*") in new stack
>>     -- Executing [s at incoming:4] *Set*("*SIP/**Company1**-00000797*", "
>> *pseudodid=** sip:Company2*") in new stack
>>     -- Executing [s at incoming:5] *GotoIf*("*SIP/**Company1**-00000797*", "
>> *0?internal,33,1:6*") in new stack
>>     -- Goto (incoming,s,6)
>>     -- Executing [s at incoming:6] *GotoIf*("*SIP/**Company1**-00000797*", "
>> *0?internal,88,1:7*") in new stack
>>     -- Goto (incoming,s,7)
>>     -- Executing [s at incoming:7] *GotoIf*("*SIP/**Company1**-00000797*", "
>> *0?internal,36,1:8*") in new stack
>>     -- Goto (incoming,s,8)
>>     -- Executing [s at incoming:8] *GotoIf*("*SIP/**Company1**-00000797*", "
>> *1?internal,36,1:9*") in new stack
>>     -- Goto (internal,36,1)
>>     -- Executing [36 at internal:1] *Set*("*SIP/**Company1**-00000797*", "
>> *CALLERID(name)=SIP/**Company1**-00000797*") in new stack
>>     -- Executing [36 at internal:2] *Dial*("*SIP/**Company1**-00000797*", "
>> *SIP/36,20*") in new stack
>>   == Using SIP RTP CoS mark 5
>>     -- Called SIP/36
>>     -- SIP/36-00000798 is ringing
>>   == Spawn extension (internal, 36, 2) exited non-zero on
>> 'SIP/Company1-00000797'
>> asterisk*CLI> exit
>
>
>  And here is the "sip.conf":
>
>  [general]
>> match_auth_username=yes
>> register=081...:... at sip.internode.on.net/s
>> register=082...:... at sip.internode.on.net/s
>> register=083...:... at sip.internode.on.net:/s
>> register=084...:... at sip.internode.on.net:/s
>> register=085...:... at sip.internode.on.net/s
>> register=086...:... at sip.internode.on.net/s
>> register=087...:... at sip.internode.on.net/s
>> register=088...:... at sip.internode.on.net/s
>>
>> [Company1]
>> username=081...
>> fromuser=081...
>> secret=...
>> canreinvite=no
>> qualify=yes
>> context=incoming
>> type=friend
>> insecure=invite,port
>> fromdomain=sip.internode.on.net
>> host=sip.internode.on.net
>> dtmfmode=rfc2833
>> disallow=all
>> allow=alaw
>> allow=ulaw
>> allow=g729
>> bindport=5060
>> bindaddr=0.0.0.0
>> nat=yes
>> registertimeout=5
>> allowoverlap=no
>> srvlookup=no
>> ubscribecontext=from-sip
>> callcounter=yes
>
>
>
> [Company2]
>> ...
>> [Company3]
>> ...
>> [Company4]
>> ...
>
>       And here is some of the "extensions.conf" file:
>
>  [incoming]
>> ; Get the DID number from the TO header.
>> exten => s,1,Set(thedid="${SIP_HEADER(TO)}")
>> exten => s,2,Set(pseudodid=${SIP_HEADER(To)})
>> exten => s,3,Set(pseudodid=${CUT(pseudodid,@,1)})
>> exten => s,4,Set(pseudodid=${CUT(pseudodid,:,2)})
>
>
>> ; Direct the DID accordingly.
>> exten => s,5,GotoIf($["${pseudodid}" = "081"]?internal,33,1:6)
>> exten => s,6,GotoIf($["${pseudodid}" = "082"]?internal,88,1:7)
>> exten => s,7,GotoIf($["${pseudodid}" = "083"]?internal,36,1:8)
>> exten => s,8,GotoIf($["${pseudodid}" = "084"]?internal,36,1:9)
>> exten => s,9,GotoIf($["${pseudodid}" = "085"]?internal,36,1:10)
>> exten => s,10,GotoIf($["${pseudodid}" = "086"]?internal,89,1:11)
>> exten => s,11,GotoIf($["${pseudodid}" = "087"]?internal,36,1:12)
>> exten => s,12,GotoIf($["${pseudodid}" = "088"]?internal,13,1:13)
>
>
>
>  -Andrew Galdes
>
>
> On Thu, Apr 2, 2015 at 3:46 PM, Dmitriy Serov <serov.d.p at gmail.com> wrote:
>
>>
>> This is one of the chronic problems. Try this option in sip.conf:
>> match_auth_username=yes
>>
>> Carefully read the description, it is better to test in "after hours".
>>
>> 02.04.2015 2:50, Andrew Galdes пишет:
>>
>> Hello all,
>>
>>  I have an Asterisk server (Asterisk 10.12.4) with multiple sip accounts
>> with the same service provides. We have 8 phone numbers in total.
>>
>>  Incoming calls from the public are all correctly directed to
>> appropriate office handsets. However, the display on the reception phone
>> (the only one i care about) is always showing the same
>> "SIP/Account1_0843214321" rather than the account representing the number
>> dialed.
>>
>>  For-instance, if Sam on her mobile calls "*0811111111*", Asterisk will
>> show a log entry like the following:
>>
>>  -- Executing [s at incoming:1] Set("SIP/*Account1_0822222222*", "
>> thedid=""NodePhone"<sip:*0811111111*@sip.internode.on.net>"") in new
>> stack
>>  But "Account1_*0822222222*" (as the name suggests) has a phone number
>> of "*0822222222*" and not "*0811111111*".
>>
>>  So Sam's call will come through and be routed to the correct handset as
>> the business needs, but it seems that all incoming calls are being labeled
>> as though coming in on a different account. The effective problem is that
>> the calledID is now wrong.
>>
>>  I'm after some general advice on how to handle the problem.
>>
>> Ta,
>>
>>
>>   -Andrew
>>
>>
>>
>>
>> --
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>
>
>
>
>
> --
> _____________________________________________________________________
> -- 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
>
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> To UNSUBSCRIBE or update options visit:
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