asterisk users - Apr 2015 - Asterisk Inbound calls, multiple SIP accounts, calledID

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
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20150402/8be2b32b/attachment.html>

Can you show us the CDR record for that call?

And maybe what your s priority of your incoming context is?

It should be easy to get what number was dialed, Try:

${CUT(PASSTHRU(${SIP_HEADER(TO):5}),@,1)}

Normally I display the callers number on my phones, Not the number they
dialed?

On Wed, Apr 1, 2015 at 4:50 PM, Andrew Galdes <andrew.galdes at
agix.com.au>
wrote:
> 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
>
> --
> _____________________________________________________________________
> -- 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
>


-- 
A human being should be able to change a diaper, plan an invasion, butcher
a hog, conn a ship, design a building, write a sonnet, balance accounts,
build a wall, set a bone, comfort the dying, take orders, give orders,
cooperate, act alone, solve equations, analyze a new problem, pitch manure,
program a computer, cook a tasty meal, fight efficiently, die gallantly.
Specialization is for insects.
---Heinlein
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20150401/787ed141/attachment.html>

On 4/1/15 7:50 PM, Andrew Galdes wrote:
> 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
> <http://sip.internode.on.net>>"") in new stack
>
> But "Account1_*0822222222*" (as the name suggests) has a phone
number
> of "*0822222222*" and not "*0811111111*".
>
It looks like all incoming calls are all being matched against the same 
entry in sip.conf.   A 'set set debug on' should clearly indicate this.
Look for the line that says :  Found peer '<insert peer name here'
for
'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
>
>

-- 
Technical Support
http://www.cellroute.net

-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20150401/2bc2b535/attachment-0001.html>

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 ?????:
> 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
> <http://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
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20150402/c141fc58/attachment.html>

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 ?????:
>
> 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
>
>
>
>
> --
> _____________________________________________________________________
> -- 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20150408/a5fca90a/attachment.html>