TIPS Not Able to Dial out using Trunk

burhanhd

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Dear PIAF
i have two pbx PIAF with Version 11.21

Server A - Primary Server

Server B - Site Server

Server A has Sip Trunk for DID

Ext range for Server A is - 4XXX

Server B has Ext range - 5XXX

Trunk Details
username=serverA
type=friend
qualify=yes
password=XXXXXX
insecure=port, invite
host=192.168.10.19
disallow=all
context=from-internal
canreinvite=yes
allow=ulaw



While calling internal from ext 4xxx to 5xxx or vise-versa works perfect.

but i am not able to dial out using DID from Server B...which should pass through Server A to dial out.

Can any one help me with this ..

regards
burhan
 

lbergey

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Your Dialplan on Server B needs to route to Server A with the Outbound Plan that Server A uses. Example:
Both Server A & Server B use 8 to Dial out.
In the Dial Plan of B, you need to make sure the 8 is passed to server A.
 

geopeterwc

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First, create "trunks" between your Server A and Server B. Helpful hint: use the IAX/SIP Trunk Excel Spreadsheets (http://pbxinaflash.com/community/resources/blanchaes-iax-sip-trunk-excel-spreadsheets.34/) to help you define the two-way trunks in both Server A and Server B.

It is not necessary to assign a digit (ie., "8") for outgoing calls if you've properly defined your dial plan in both PBXs. In the dial plan for Server A, you specify that dialed digits 5XXX are routed to Server B using the trunk you've defined above. Similarly, the dial plan for Server B will route calls with dialed digits 4XXX to Server A.

You must understand what your DID provider expects from your PBX to place an external call, and define your outbound route on Server A so that outbound calls are placed successfully to your chosen Telephony Service Provider DID.

At Server B, define the outbound route duplicating the OUTBOUND routes that are in Server A, BUT, specify that the between-the-pbx trunks you created above is the path that outbound calls will use.

You may want to change the incoming setting "from-trunk" to "from-internal" to keep the caller ID information working between PBXs.

/Pete./
 
Last edited:

burhanhd

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First, create "trunks" between your Server A and Server B. Helpful hint: use the IAX/SIP Trunk Excel Spreadsheets (http://pbxinaflash.com/community/resources/blanchaes-iax-sip-trunk-excel-spreadsheets.34/) to help you define the two-way trunks in both Server A and Server B.

It is not necessary to assign a digit (ie., "8") for outgoing calls if you've properly defined your dial plan in both PBXs. In the dial plan for Server A, you specify that dialed digits 5XXX are routed to Server B using the trunk you've defined above. Similarly, the dial plan for Server B will route calls with dialed digits 4XXX to Server A.

You must understand what your DID provider expects from your PBX to place an external call, and define your outbound route on Server A so that outbound calls are placed successfully to your chosen Telephony Service Provider DID.

At Server B, define the outbound route duplicating the OUTBOUND routes that are in Server A, BUT, specify that the between-the-pbx trunks you created above is the path that outbound calls will use.

You may want to change the incoming setting "from-trunk" to "from-internal" to keep the caller ID information working between PBXs.

/Pete./
Shoudl i Change to "from-trunk" from "from-internal"

becuase as of now its "from-internal"
 

burhanhd

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I changed to context=from-trunk

i get this error messge on Cli
Server B
localhost*CLI>
[2016-05-26 17:29:56] WARNING[2201][C-0000000b]: chan_sip.c:23273 handle_response_invite: Received response: "Forbidden" from '<sip:[email protected]>;tag=as0111e196'
localhost*CLI>

Server A

[2016-05-26 17:29:56] NOTICE[3994][C-0000015d]: chan_sip.c:25853 handle_request_invite: Failed to authenticate device <sip:[email protected]:5060>;tag=as0111e196
localhost*CLI>
would appropriate if you can help me to resolved this
 

geopeterwc

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@burhanhd ... I need to know a bit more of your situation before I can offer more assistance.
  1. Where are you located? If you're in the USA, the specifications in your outbound route aren't likely to get you "anywhere." What Area Code is your installation? If you're not in the USA, can you specify the dialing requirements for placing calls to the public network?
  2. Who is your Internet Telephony Service Provider (ITSP)?
  3. Who is your Internet Service Provider (ISP)?
  4. Are your servers configured on CentOS (version?)? or are you using RaspPi.
  5. What version of iPBX and Incredible GUI are you using?
  6. What IP phones (manufacturer, model) or ATA's are at your users desks?
Sometimes you need to back away from the situation a bit to get a clear view of the requirements, and then drill into each aspect of the problem in order. It seems that you've taken on a big bite, and that getting the various parts to play together nicely is the main issue, here.

/Pete./
 

burhanhd

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@burhanhd ... I need to know a bit more of your situation before I can offer more assistance.
  1. Where are you located? If you're in the USA, the specifications in your outbound route aren't likely to get you "anywhere." What Area Code is your installation? If you're not in the USA, can you specify the dialing requirements for placing calls to the public network?
  2. Who is your Internet Telephony Service Provider (ITSP)?
  3. Who is your Internet Service Provider (ISP)?
  4. Are your servers configured on CentOS (version?)? or are you using RaspPi.
  5. What version of iPBX and Incredible GUI are you using?
  6. What IP phones (manufacturer, model) or ATA's are at your users desks?
Sometimes you need to back away from the situation a bit to get a clear view of the requirements, and then drill into each aspect of the problem in order. It seems that you've taken on a big bite, and that getting the various parts to play together nicely is the main issue, here.

/Pete./

Located in Kuwait
ISP and TISP is Local ISP from whom We get SIP Trunk
Server is configured on CentOS
Version 11
Grandstream Phones and Softphones
 

geopeterwc

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OK ... please explain the dialed number scheme for placing calls within Kuwait, and how calls are made in the public phone system in Kuwait.

Please provide the information that appears when you log into Server A via an SSL connection; the "STATUS" information that is presented when you log in to the Command Line Interface - not an internet browser.

Please also explain what a typical dialed number "looks like." i.e., leading digit(s), city code(s), number of digits in a phone number. Are there "variable length" phone numbers?

Resolving these details is the best place to start in getting your Incredible PBX working.

/Pete./
 

burhanhd

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OK ... please explain the dialed number scheme for placing calls within Kuwait, and how calls are made in the public phone system in Kuwait.

Please provide the information that appears when you log into Server A via an SSL connection; the "STATUS" information that is presented when you log in to the Command Line Interface - not an internet browser.

Please also explain what a typical dialed number "looks like." i.e., leading digit(s), city code(s), number of digits in a phone number. Are there "variable length" phone numbers?

Resolving these details is the best place to start in getting your Incredible PBX working.

/Pete./
In Kuwait we have 8 Digit Number
start with
2XXXXXXX
5XXXXXXX
6XXXXXXX
9XXXXXXX

HotLine are 7 Digit
18XXXXX

I am able to dial an receive call from Server A.. only issue not able to Dial from Server B
 

lbergey

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You may want to do a quick IAX2 trunk to trunk

Server A
Type: IAX
Trunk Name: ServerB

Outgoing Settings:
Trunk Name: ServerB

Peer Details:
username=ServerA
[email protected]
host=192.168.90.254
type=friend
context=from-internal
qualify=yes
qualifyfreqok=25000
transfer=no
trunk=yes
forceencryption=yes
encryption=yes
auth=md5


Server B
Type: IAX
Trunk Name: ServerA

Outgoing Settings:
Trunk Name: ServerA

Peer Details:
username=ServerB
[email protected]
host=192.168.20.226
type=friend
context=from-internal
qualify=yes
qualifyfreqok=25000
transfer=no
trunk=yes
forceencryption=yes
encryption=yes
 
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