Outbound Issue for PJSIP

Asterisk Asterisk SIP
1

Hello, Team,

I have two asterisk servers, each server has its own PJSIP files, I need to create PJSIP trunk between the two asterisk servers using PJSIP trunk.

For example, I have:
Server-A: 192.168.X.70
Server-B: 192.168.X.71

I created the PJSIP trunk, Inbound is coming via from 192.168.X.70 to 192.168.X.71.

Note: The E1/SIP connection is connected directly in to Server-A: 192.168.X.70, and I need to use calls from Server-B(192.168.X.70) to Server-A(192.168.X.71)

How Outbound will be the configuration please?.
When I try to make outbound call it says Busy/congested.

Here is the configuration for the Both sides,

Only Inboud is coming to server A, I cant make outbound.

[siptrunk]
type=endpoint
context=from-pstn
disallow=all
allow=ulaw
aors=siptrunk

[siptrunk]
type=aor
contact=sip:192.168.X.71:5060

[siptrunk]
type=identify
endpoint=siptrunk
match=192.168.X.71

Here is the outbound dialplan call in SERVER A:
exten => _07XXXXXXXX,1,Set(CALLERID(num)=+25xxxxxx)
exten => _07XXXXXXXX,n,Dial(PJSIP/${EXTEN}@siptrunk,30,T)
exten => _07XXXXXXXX,n,Hangup()

2

You should show the actual console output and SIP trace (pjsip set logger on).

3

Please how to make outgoing calls via between Pjsip servers sir.thanks

4

I can’t see anything obviously wrong with what you are doing, which is why you need to answer this question:

5

Hello, Every one, I need to map calls from ServerA pjsip to ServerB pjsip files.
Normally, Incoming is working, From ServerA to serverB with No issues.

When I try To make Outbound it doesn’t work.

Note: I configured two asterisk servers using asterisk PJSIP & chan_SIP, and Outbound is fine.
This is my Oubound dialplan working:
exten => _07XXXXXXXX,n,Set(CALLERID(number)=xxxxxxxxxx)
exten => _07XXXXXXXX,n,Dial(PJSIP/cor-siptrunk/sip:${EXTEN}@192.168.78.100:5070,30,T)
exten => _07XXXXXXXX,n,Hangup()

But, When I am configuring two asterisk servers using PJSIP, Outbound is not working.

Please someone can give Outboud call dialplan connecting two asterisk servers using PJSIP.

6

There is NOTHING obviously wrong with the dialplan you have.

As jcolp requested, we need to know what happens when the call is made, to be able to suggest a solution.

You open the Asterisk console, by logging in with SSH on the Asterisk server, then type asterisk -r as a user allowed to connect to Asterisk. The root user is always allowed, but you can configure the system to allow others to do it as well.

With the console open, type pjsip set logger on, there are other options to filter the output, but let’s keep it simple.

When that is done, clear your terminal, including the scrollback buffer. (In iTerm2 on a mac, CMD+K will do the trick)

Then make the call that’s not working.

Replace sensitive information with placeholders, but MAKE SURE to use unique placeholders for each part removed. Eg. <phonenumberA> everywhere the A number appears, <phonenumberB> where the B number is used. <userA> where username A appears etc. It’s important that when reading the information you know where things appear, even though they are censored.

Copy all the output into a text file.

Make a reply here, pasting all the text from the file in a code block. The </> icon at the top of the editor will make one for you.

Without that information, it’s very hard to guess what the problem might be.

7

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