Re: [asterisk-users] Asterisk Inbound calls, multiple SIP accounts, calledID
Andrew,
Instead of your SET and GOTO blocks I'd recommend using the Asterisk DB to
make things easier to maintain.
You could make two database entries for each of your DID's
database put 4259981810 name JohnPersonal
database put 4259981810 target kiniston-extern,john-personal,1
Then you could do a single block that would do the lookup and call routing:
Set(DESTINATION=${CUT(PASSTHRU(${SIP_HEADER(TO):5}),@,1)})
Set(CALLERID(name)=${DB(${DESTINATION}/name)})
Goto(${DB(${DESTINATION}/target)})
On Tue, Apr 7, 2015 at 6:06 PM, Andrew Galdes
wrote:
> Solved it, kinda. It's not cute. I'm sure this is the way NOT to do it but
> it does work. For prosperity, the SIP service is through Internode.
>
> Here is my "extensions.conf" file:
>
> exten => s,5,Set(callersname=${IF($[ ${pseudodid} =
> 081...]?Company1:${callersname})})
> exten => s,6,Set(callersname=${IF($[ ${pseudodid}
> = 082...]?Company2:${callersname})})
>
> exten => s,13,GotoIf($["${callersname}" = "Company1"]?internal,36,1:14);
> to reception
> exten => s,14,GotoIf($["${callersname}" = "Company2"]?internal,88,1:15);
> to department1
>
> And later in same file:
>
> ; Phone 36 reception
>> *exten => 36,1,Set(CALLERID(name)=${callersname})*
>> exten => 36,n,Dial(SIP/36,20)
>> exten => 36,n,VoiceMail(36,u)
>> exten => 36,n,Hangup
>
>
>
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Inbound calls, multiple SIP accounts, calledID
what about
exten => s,1,Goto(from-trunk,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1)
regards
2015-04-08 5:45 GMT+00:00 Dmitriy Serov :
> 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@incoming:1] *Set*("*SIP/Company1-0797*",
>> "*thedid=""NodePhone"> >"*") in new stack
>> -- Executing [s@incoming:2] *Set*("*SIP/**Company1**-0797*", "
>> *pseudodid="NodePhone"> >*") in new stack
>> -- Executing [s@incoming:3] *Set*("*SIP/**Company1**-0797*", "
>> *pseudodid="NodePhone"> -- Executing [s@incoming:4] *Set*("*SIP/**Company1**-0797*", "
>> *pseudodid=** sip:Company2*") in new stack
>> -- Executing [s@incoming:5] *GotoIf*("*SIP/**Company1**-0797*", "
>> *0?internal,33,1:6*") in new stack
>> -- Goto (incoming,s,6)
>> -- Executing [s@incoming:6] *GotoIf*("*SIP/**Company1**-0797*", "
>> *0?internal,88,1:7*") in new stack
>> -- Goto (incoming,s,7)
>> -- Executing [s@incoming:7] *GotoIf*("*SIP/**Company1**-0797*", "
>> *0?internal,36,1:8*") in new stack
>> -- Goto (incoming,s,8)
>> -- Executing [s@incoming:8] *GotoIf*("*SIP/**Company1**-0797*", "
>> *1?internal,36,1:9*") in new stack
>> -- Goto (internal,36,1)
>> -- Executing [36@internal:1] *Set*("*SIP/**Company1**-0797*", "
>> *CALLERID(name)=SIP/**Company1**-0797*") in new stack
>> -- Executing [36@internal:2] *Dial*("*SIP/**Company1**-0797*", "
>> *SIP/36,20*") in new stack
>> == Using SIP RTP CoS mark 5
>> -- Called SIP/36
>> -- SIP/36-0798 is ringing
>> == Spawn extension (internal, 36, 2) exited non-zero on
>> 'SIP/Company1-0797'
>> asterisk*CLI> exit
>
>
> And here is the "sip.conf":
>
> [general]
>> match_auth_username=yes
>> register=081...:...@sip.internode.on.net/s
>> register=082...:...@sip.internode.on.net/s
>> register=083...:...@sip.internode.on.net:/s
>> register=084...:...@sip.internode.on.net:/s
>> register=085...:...@sip.internode.on.net/s
>> register=086...:...@sip.internode.on.net/s
>> register=087...:...@sip.internode.on.net/s
>> register=088...:...@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=${CUT(pseudodid,:,2)})
>
>
>> ; Direct the DID accordingly.
>> exten => s,5,GotoIf($["${pseudodid}" = "081"]?internal,33,1:6)
>> exten => s,6,GotoIf($["${pseudodid}" = "082"]?internal,88,1:7)
>> exten => s,7,GotoIf($["${pseudodid}" = "083"]?internal,36,1:8)
>> exten => s,8,GotoIf($["${pseudodid}" = "084"]?internal,36,1:9)
>> exten => s,9,GotoIf($["${pseudodid}" = "085"]?internal,36,1:10)
>> exten => s,10,GotoIf($["${pseudodid}"
Re: [asterisk-users] Asterisk Inbound calls, multiple SIP accounts, calledID
Hi, Andrew.
You are trying to solve two tasks: definition through what line the call
came and a beautiful display of this information.
1. definition through what line the call came. If the username and
password for inbound and outbound registration the same, then try the
following:
a) delete "register" lines.
b) add option "callbackextension=Company1" to Company1 friend section..
And in others with their names too.
or you can change "/s" to "/Company1" in register line.
2. beautiful display of this information
a) add option "setvar=fromCompany=Company1" to Company1 friend section..
b) In dialplan add
Set(CALLERID(name)=${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@incoming:1] *Set*("*SIP/Company1-0797*",
"*thedid=""NodePhone"mailto:sip%3acompa...@sip.internode.on.net>>"*") in new stack
-- Executing [s@incoming:2]
*Set*("*SIP/**Company1**-0797*",
"*pseudodid="NodePhone"http://sip.internode.on.net>>*") in new stack
-- Executing [s@incoming:3]
*Set*("*SIP/**Company1**-0797*",
"*pseudodid="NodePhone" exit
And here is the "sip.conf":
[general]
match_auth_username=yes
register=081...:...@sip.internode.on.net/s
register=082...:...@sip.internode.on.net/s
register=083...:...@sip.internode.on.net:/s
register=084...:...@sip.internode.on.net:/s
register=085...:...@sip.internode.on.net/s
register=086...:...@sip.internode.on.net/s
register=087...:...@sip.internode.on.net/s
register=088...:...@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 > 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" rath
Re: [asterisk-users] Asterisk Inbound calls, multiple SIP accounts, calledID
On 4/7/15 7:48 PM, Andrew Galdes wrote:
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@incoming:1] *Set*("*SIP/Company1-0797*",
"*thedid=""NodePhone"mailto:sip%3acompa...@sip.internode.on.net>>"*") in new stack
-- Executing [s@incoming:2]
*Set*("*SIP/**Company1**-0797*",
"*pseudodid="NodePhone"http://sip.internode.on.net>>*") in new stack
-- Executing [s@incoming:3]
*Set*("*SIP/**Company1**-0797*",
"*pseudodid="NodePhone" exit
And here is the "sip.conf":
[general]
match_auth_username=yes
register=081...:...@sip.internode.on.net/s
register=082...:...@sip.internode.on.net/s
register=083...:...@sip.internode.on.net:/s
register=084...:...@sip.internode.on.net:/s
register=085...:...@sip.internode.on.net/s
register=086...:...@sip.internode.on.net/s
register=087...:...@sip.internode.on.net/s
register=088...:...@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=${CUT(pseudodid,:,2)})
; Direct the DID accordingly.
exten => s,5,GotoIf($["${pseudodid}" = "081"]?internal,33,1:6)
exten => s,6,GotoIf($["${pseudodid}" = "082"]?internal,88,1:7)
exten => s,7,GotoIf($["${pseudodid}" = "083"]?internal,36,1:8)
exten => s,8,GotoIf($["${pseudodid}" = "084"]?internal,36,1:9)
exten => s,9,GotoIf($["${pseudodid}" = "085"]?internal,36,1:10)
exten => s,10,GotoIf($["${pseudodid}" = "086"]?internal,89,1:11)
exten => s,11,GotoIf($["${pseudodid}" = "087"]?internal,36,1:12)
exten => s,12,GotoIf($["${pseudodid}" = "088"]?internal,13,1:13)
Since your objective is to have the receptionist identify the company
she should be answering to then might I suggest a simple workaround to
your problem. Since right here you are already sending the call to the
expected internal context and extension, you could simply alter the
Caller Name and put in the Company Name so she could see it on the
screen. Something like:
[internal]
exten => 33,1,Set(CALLERID(name)=Company1:${CALLERID})
...
exten => 88,1,Set(CALLERID(name)=Company2:${CALLERID})
...
exten => 36,1,Set(CALLERID(name)=Company3:${CALLERID})
...
etc...
That will display the Company Name you want to see followed by the
caller ID #
-Andrew Galdes
On Thu, Apr 2, 2015 at 3:46 PM, Dmitriy Serov > 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 nu
Re: [asterisk-users] Asterisk Inbound calls, multiple SIP accounts, calledID
Solved it, kinda. It's not cute. I'm sure this is the way NOT to do it but
it does work. For prosperity, the SIP service is through Internode.
Here is my "extensions.conf" file:
exten => s,1,Set(thedid="${SIP_HEADER(TO)}"); ignore this one
exten => s,2,Set(pseudodid=${SIP_HEADER(To)})
exten => s,3,Set(pseudodid=${CUT(pseudodid,@,1)})
exten => s,4,Set(pseudodid=${CUT(pseudodid,:,2)})
exten => s,5,Set(callersname=${IF($[ ${pseudodid} =
081...]?Company1:${callersname})})
exten => s,6,Set(callersname=${IF($[ ${pseudodid}
= 082...]?Company2:${callersname})})
exten => s,7,Set(callersname=${IF($[ ${pseudodid}
= 083...]?Company3:${callersname})})
exten => s,8,Set(callersname=${IF($[ ${pseudodid}
= 084...]?Company4:${callersname})})
exten => s,9,Set(callersname=${IF($[ ${pseudodid}
= 085...]?Company5:${callersname})})
exten => s,10,Set(callersname=${IF($[ ${pseudodid}
= 086...]?Company6:${callersname})})
exten => s,11,Set(callersname=${IF($[ ${pseudodid}
= 087...]?Company7:${callersname})})
exten => s,12,Set(callersname=${IF($[ ${pseudodid}
= 088...]?Company8:${callersname})})
exten => s,13,GotoIf($["${callersname}" = "Company1"]?internal,36,1:14); to
reception
exten => s,14,GotoIf($["${callersname}" = "Company2"]?internal,88,1:15); to
department1
exten => s,15,GotoIf($["${callersname}" = "Company3"]?internal,36,1:16); to
reception
exten => s,16,GotoIf($["${callersname}" = "Company4"]?internal,36,1:17); to
reception
exten => s,17,GotoIf($["${callersname}" = "Company5"]?internal,36,1:18); to
reception
exten => s,18,GotoIf($["${callersname}" = "Company6"]?internal,89,1:19); to
department2
exten => s,19,GotoIf($["${callersname}" = "Company7"]?internal,36,1:20); to
reception
exten => s,20,GotoIf($["${callersname}" = "Company8"]?internal,13,1:21); to
department3
And later in same file:
; Phone 36 reception
> *exten => 36,1,Set(CALLERID(name)=${callersname})*
> exten => 36,n,Dial(SIP/36,20)
> exten => 36,n,VoiceMail(36,u)
> exten => 36,n,Hangup
Ta,
-Andrew Galdes
Managing Director
RHCE, LPI, CCENT
AGIX Linux
Ph: 08 7324 4429
Mb: 0422 927 598
Find us: Website | LinkedIn
| Blog |
YouTube | Google+
*Platform Architects for High Demand Web Applications.*
On Wed, Apr 8, 2015 at 9:18 AM, Andrew Galdes
wrote:
> 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@incoming:1] *Set*("*SIP/Company1-0797*",
>> "*thedid=""NodePhone"> >"*") in new stack
>> -- Executing [s@incoming:2] *Set*("*SIP/**Company1**-0797*", "
>> *pseudodid="NodePhone"> >*") in new stack
>> -- Executing [s@incoming:3] *Set*("*SIP/**Company1**-0797*", "
>> *pseudodid="NodePhone"> -- Executing [s@incoming:4] *Set*("*SIP/**Company1**-0797*", "
>> *pseudodid=** sip:Company2*") in new stack
>> -- Executing [s@incoming:5] *GotoIf*("*SIP/**Company1**-0797*", "
>> *0?internal,33,1:6*") in new stack
>> -- Goto (incoming,s,6)
>> -- Executing [s@incoming:6] *GotoIf*("*SIP/**Company1**-0797*", "
>> *0?internal,88,1:7*") in new stack
>> -- Goto (incoming,s,7)
>> -- Executing [s@incoming:7] *GotoIf*("*SIP/**Company1**-0797*", "
>> *0?internal,36,1:8*") in new stack
>> -- Goto (incoming,s,8)
>> -- Executing [s@incoming:8] *GotoIf*("*SIP/**Company1**-0797*", "
>> *1?internal,36,1:9*") in new stack
>> -- Goto (internal,36,1)
>> -- Executing [36@internal:1] *Set*("*SIP/**Company1**-0797*", "
>> *CALLERID(name)=SIP/**Company1**-0797*") in new stack
>> -- Executing [36@internal:2] *Dial*("*SIP/**Company1**-0797*", "
>> *SIP/36,20*") in new stack
>> == Using SIP RTP CoS mark 5
>> -- Called SIP/36
>> -- SIP/36-0798 is ringing
>> == Spawn extension (internal, 36, 2) exited non-zero on
>> 'SIP/Company1-0797'
>> asterisk*CLI> exit
>
>
> And here is the "sip.conf":
>
> [general]
>> match_auth_username=yes
>> register=081...:...@sip.internode.on.net/s
>> register=082...:...@sip.internode.on.net/s
>> register=083...:...@sip.in
Re: [asterisk-users] Asterisk Inbound calls, multiple SIP accounts, calledID
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@incoming:1] *Set*("*SIP/Company1-0797*",
> "*thedid=""NodePhone" >"*") in new stack
> -- Executing [s@incoming:2] *Set*("*SIP/**Company1**-0797*", "
> *pseudodid="NodePhone" >*") in new stack
> -- Executing [s@incoming:3] *Set*("*SIP/**Company1**-0797*", "
> *pseudodid="NodePhone" -- Executing [s@incoming:4] *Set*("*SIP/**Company1**-0797*", "
> *pseudodid=** sip:Company2*") in new stack
> -- Executing [s@incoming:5] *GotoIf*("*SIP/**Company1**-0797*", "
> *0?internal,33,1:6*") in new stack
> -- Goto (incoming,s,6)
> -- Executing [s@incoming:6] *GotoIf*("*SIP/**Company1**-0797*", "
> *0?internal,88,1:7*") in new stack
> -- Goto (incoming,s,7)
> -- Executing [s@incoming:7] *GotoIf*("*SIP/**Company1**-0797*", "
> *0?internal,36,1:8*") in new stack
> -- Goto (incoming,s,8)
> -- Executing [s@incoming:8] *GotoIf*("*SIP/**Company1**-0797*", "
> *1?internal,36,1:9*") in new stack
> -- Goto (internal,36,1)
> -- Executing [36@internal:1] *Set*("*SIP/**Company1**-0797*", "
> *CALLERID(name)=SIP/**Company1**-0797*") in new stack
> -- Executing [36@internal:2] *Dial*("*SIP/**Company1**-0797*", "
> *SIP/36,20*") in new stack
> == Using SIP RTP CoS mark 5
> -- Called SIP/36
> -- SIP/36-0798 is ringing
> == Spawn extension (internal, 36, 2) exited non-zero on
> 'SIP/Company1-0797'
> asterisk*CLI> exit
And here is the "sip.conf":
[general]
> match_auth_username=yes
> register=081...:...@sip.internode.on.net/s
> register=082...:...@sip.internode.on.net/s
> register=083...:...@sip.internode.on.net:/s
> register=084...:...@sip.internode.on.net:/s
> register=085...:...@sip.internode.on.net/s
> register=086...:...@sip.internode.on.net/s
> register=087...:...@sip.internode.on.net/s
> register=088...:...@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 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 "*08*", Asterisk will
> show a log entry like the following:
>
> -- Executing [s@incoming:1] Set("SIP/*Acc
Re: [asterisk-users] Asterisk Inbound calls, multiple SIP accounts, calledID
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 "*08*", Asterisk will
show a log entry like the following:
-- Executing [s@incoming:1] Set("SIP/*Account1_08*",
"thedid=""NodePhone">"") in new stack
But "Account1_*08*" (as the name suggests) has a phone number
of "*08*" and not "*08*".
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
Re: [asterisk-users] Asterisk Inbound calls, multiple SIP accounts, calledID
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 "*08*", Asterisk will
show a log entry like the following:
-- Executing [s@incoming:1] Set("SIP/*Account1_08*",
"thedid=""NodePhone">"") in new stack
But "Account1_*08*" (as the name suggests) has a phone number
of "*08*" and not "*08*".
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 ''08'
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
--
_
-- 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
Re: [asterisk-users] Asterisk Inbound calls, multiple SIP accounts, calledID
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
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 "*08*", Asterisk will
> show a log entry like the following:
>
> -- Executing [s@incoming:1] Set("SIP/*Account1_08*", "
> thedid=""NodePhone""") in new stack
> But "Account1_*08*" (as the name suggests) has a phone number of "
> *08*" and not "*08*".
>
> 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
--
_
-- 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
[asterisk-users] 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 "*08*", Asterisk will show
a log entry like the following:
-- Executing [s@incoming:1] Set("SIP/*Account1_08*", "
thedid=""NodePhone""") in new stack
But "Account1_*08*" (as the name suggests) has a phone number of "
*08*" and not "*08*".
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
