blueSIP Premium Asterisk only show 1 CLI number as incomming line ? Have 6 DID's CLI

voipdude

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I have a blueSIP Premium contract 6 DID Numbers. Problem is:

Only one CLI number shows up at my PBX Asterisk. Example:
My PBX Asterisk Register String:

usernumber/myusername:[email protected]/xxx-my-DID-Number
Set up like that only give --> xxx-my-DID-Number as incomming CLI

Asterisk Register String: If setup like this
usernumber/myusername:[email protected] --> No CLI indentification ??

Problem i have 6 DID they must show up in my PBX so i can route them.
So i need to see all if they arrive. Example like this:

("SIP/bluesip.net-b6b0fede", "__FROM_DID=xxx-my-DID-Number-001234") in new stack
("SIP/bluesip.net-b6b0fedf", "__FROM_DID=xxx-my-DID-Number-001235") in new stack
("SIP/bluesip.net-b6b0fedb", "__FROM_DID=xxx-my-DID-Number-001236") in new stack

I did see some options in there gui setup. But not clear of that are the right settings. Did contact blueSIP if they have a short how to or faq. Maybe somebody has experience with this ? I will post the solution if they reply. :cool:
 
R

rentier-s

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Try evaluating the "TO" SIP Header. For assistance search for Sipgate Trunking solutions, there's the same issue.
 

voipdude

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Try evaluating the "TO" SIP Header. For assistance search for Sipgate Trunking solutions, there's the same issue.
Thank you for the answer. Did try to do what you say. But problem is if bluesip.net is not sending the correct CLI identification then it will never work.

As a test i did use the same bluesip.net account with the same password but for each server i did change: (So used 2 Asterisk test servers)

usernumber/myusername:[email protected]/xxx-my-DID-Number01 Test server 1
usernumber/myusername:[email protected]/xxx-my-DID-Number02 Test server 2

Then what you see the number that is connected to the server 1 or 2 is receiving the correct CLI identification.

Conclusion is that at the end of the registration string the number xxx-my-DID-Number 02 or 01 is the parameter
That make bluesip.net sending the correct CLI identification. Problem is bluesip.

If bluesip would give for each DID number a account problem would also be solved. But now this is not funny. I think need to contact them again.
Still no reply for this issue.

Problem is this the parameter at the end of the registration string is used by bleusip to communicate CLI. Example:

When u use: (Registration string Asterisk to bluesip
usernumber/myusername:[email protected]/notfunny

The incoming CLI will be "notfunny"

Do you use:

usernumber/myusername:[email protected]/hello

The incoming CLI will be "hello"

Thats the problem ... try to talk to them again .. :(
 
R

rentier-s

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usernumber/myusername:[email protected]/notfunny

The incoming CLI will be "notfunny"
That's nothing special for your provider, but common behaviour of Asterisk. The part of the register string after the last slash will be sent as "Contact" Header.

Did you try an exten=>_X.,1,Noop(${SIP_HEADER(TO)}) ? Was the output really the same for all called numbers?

Btw, please do not refer to the called party id with CLI, because this is short for command line interface and meens the console you call with "asterisk -r".
 

voipdude

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That's nothing special for your provider, but common behaviour of Asterisk. The part of the register string after the last slash will be sent as "Contact" Header.

Did you try an exten=>_X.,1,Noop(${SIP_HEADER(TO)}) ? Was the output really the same for all called numbers?

Btw, please do not refer to the called party id with CLI, because this is short for command line interface and meens the console you call with "asterisk -r".
Did contact there support. And i think you are right. Has to do with behaviour of Asterisk. So i have some instruction from Bleusip and will add those parameters.
I understand what you mean with the "asterisk -r" Thats where i went wrong. I will post the outcome here. So then other Bleusip users can use those instructions. And they will be happy to continue using the service of Bleusip. :doktor: ;)
 

voipdude

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That's nothing special for your provider, but common behaviour of Asterisk. The part of the register string after the last slash will be sent as "Contact" Header.

Did you try an exten=>_X.,1,Noop(${SIP_HEADER(TO)}) ? Was the output really the same for all called numbers?

Btw, please do not refer to the called party id with CLI, because this is short for command line interface and meens the console you call with "asterisk -r".

The numbers are now coming in correct. :)But Bleusip gave me some parameters to set it up in Asterisk. Here some details. Still Asterisk is not knowing
where to route it to ?? Asterisk stops at line:

-- Goto (ext-did-direct-custom,+4924101234567,1) :confused:

Asterisk message: The number you have dialed is not in service

Must be just a simple setting .... also using


extensions_custom.conf


[from-pstn-custom]

; extract did from sip TO: header
exten => s,1,Set(DW=${SIP_HEADER(TO):5})
exten => s,2,Set(DW=${CUT(DW,@,1)})
exten => s,3,Goto(ext-did-direct-custom,${DW},1)

[ext-did-direct-custom]

; first expand DW to full E.164 number
exten => _00XXXXX.,1,Goto(from-pstn,+${EXTEN:2},1)
exten => _0XXXXX.,1,Goto(from-pstn,+49${EXTEN:1},1)
exten => _XXX.,1,Goto(from-pstn,+49241${EXTEN},1)

; static did inbound routing

exten => +4924101234567,101,Goto(ext-local,1004,1) ; routing zu Nst 1004

(Trunk setup incomming)

disallow=all
allow=alaw&ulaw
type=friend
host=bluesip.net
context=from-pstn-custom
fromdomain=bluesip.net
auth=md5,plaintext
qualify=yes
canreinvite=no
nat=no
insecure=very

------------------
CLI Output: " Not the real number but for example +4924101234567 "
------------------


Verbosity is at least 5

-- Executing [[email protected]:1] Set("SIP/bluesip.net-b6b01df8", "[email protected]>") in new stack
-- Executing [[email protected]:2] Set("SIP/bluesip.net-b6b01df8", "DW=+4924101234567") in new stack
-- Executing [[email protected]:3] Goto("SIP/bluesip.net-b6b01df8", "ext-did-direct-custom|+4924101234567|1") in new stack
-- Goto (ext-did-direct-custom,+4924101234567,1)

-- ast_get_srv: SRV lookup for '_sip._udp.bluesip.net' mapped to host bluesip.net, port 5060
-- ast_get_srv: SRV lookup for '_sip._udp.bluesip.net' mapped to host bluesip.net, port 5060
-- ast_get_srv: SRV lookup for '_sip._udp.bluesip.net' mapped to host bluesip.net, port 5060

myserver*CLI>
 

voipdude

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Problem i dont really understand the syntax. So if this: -- Goto (ext-did-direct-custom,+4924101234567,1) is
whats shows up in the CLI how to route it to a internal number or ring Queue ??

Do i understand this correct ? I need to set it up like this: exten => +4924101234567,1,Goto(ext-local,1004,1) ; routing zu Nst 1004
Then it should go to internal 1004 ? I did try this but dont get routed to 1004 ?? mmmh ??? :confused::confused:

Maybe this will fix it : http://www.freepbx.org/support/documentation/howtos/how-to-get-the-did-of-a-sip-trunk-when-the-provider-doesnt-send-it-and-
 

voipdude

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I did find the solution by using the next syntax: :cool:;)

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

[custom-Bleu-SIP-provider]
exten => s,1,Noop(Fixing DID to +491234567890 Example)
exten => s,n,Goto(from-trunk,+491234567890,1 Example)

Solution source: http://www.freepbx.org/support/documentation/howtos/how-to-get-the-did-of-a-sip-trunk-when-the-provider-doesnt-send-it-and-

Now all 6 lines get in and can be routed. Only strange when dialing the Bleusip number you can not here a dial ton ?? No ring ring ??
But if my PBX pics up the phone then a dial tone like ringe ring in the phone ??? Thats strange will mail support..
 
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