Retrieving PJSIP_Header from a PJSIP registered trunk

1

Hello everybody ,

Since a few weeks , I have set up a FreePBX ( Asterisk based ) and I have registered a trunk with Chan_PJSIP trunk.

All is working fine for outbound calls , but for incoming calls , it works only if I don’t specify “DID inbound route”. After a lot of troubleshooting and time to research , I came to the conclusion that my provider is sending those DID Numbers in the SIP or PJSIP Headers. So I set up the context to “from-pstn-toheader” that contains :
[from-pstn-toheader]exten => .,1,Goto(from-pstn,${CUT(CUT(PJSIPHEADER(To),@,1),:,2)},1)
to recover the PJSIPHEADER , but I have the following error and I can’t figure it out :

ERROR[11578][C-00000000]: res_pjsip_header_funcs.c:465 func_read_header: This function requires a header name

Thanks for reading and have a nice day !

2

What is the complete console output and the actual dialplan logic? The dialplan function would be used as ${PJSIP_HEADER(To)} not what is above.

3

Hi , this is the complete output console when I use this function to retrieve the DID number in a dynamic way
This is the function :
exten => s,1,Goto(from-trunk,${CUT(CUT(PJSIP_HEADER(To),@,1),:,2)},1)

This is the output :

[2016-06-27 09:46:21] ERROR[25793][C-00000001]: res_pjsip_header_funcs.c:465 func_read_header: This function requires a header name.
– Executing [s@from-pstn-toheader:1] Goto(“PJSIP/PJSIP Trunk-00000001”, “from-trunk,1”) in new stack
– Goto (from-trunk,s,1)
– Executing [s@from-trunk:1] NoOp(“PJSIP/PJSIP Trunk-00000001”, “No DID or CID Match”) in new stack
– Executing [s@from-trunk:2] Answer(“PJSIP/PJSIP Trunk-00000001”, “”) in new stack
[2016-06-27 09:46:22] WARNING[25793][C-00000001]: chan_sip.c:22377 func_header_read: This function can only be used on SIP channels.
– Executing [s@from-trunk:3] Log(“PJSIP/PJSIP Trunk-00000001”, "WARNING,Friendly Scanner from ") in new stack
[2016-06-27 09:46:22] WARNING[25793][C-00000001]: Ext. s:3 @ from-trunk: Friendly Scanner from
– Executing [s@from-trunk:4] Wait(“PJSIP/PJSIP Trunk-00000001”, “2”) in new stack
> 0x2cf9de0 – Probation passed - setting RTP source address to X.X.X.X:30202
– Executing [s@from-trunk:5] Playback(“PJSIP/PJSIP Trunk-00000001”, “ss-noservice”) in new stack
– <PJSIP/PJSIP Trunk-00000001> Playing ‘ss-noservice.ulaw’ (language ‘en’)
– Executing [h@from-trunk:1] Macro(“PJSIP/PJSIP Trunk-00000001”, “hangupcall,”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“PJSIP/PJSIP Trunk-00000001”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [s@macro-hangupcall:3] ExecIf(“PJSIP/PJSIP Trunk-00000001”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [s@macro-hangupcall:4] Hangup(“PJSIP/PJSIP Trunk-00000001”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘PJSIP/PJSIP Trunk-00000001’ in macro ‘hangupcall’
== Spawn extension (from-trunk, h, 1) exited non-zero on ‘PJSIP/PJSIP Trunk-00000001’

However , I can make things work if I set up the extensions.conf file in a “hard way” :

exten => s,1,Goto(from-trunk,00334XXXXXXXX,1) --> where the X respresents my num

But ! Next problem , I have 4 nums , that when they are dialed , redirect to my trunk line so I set up :

exten => s,1,Goto(from-trunk,00334XXXXXXX0,1)
exten => s,n,Goto(from-trunk,00334XXXXXXX1,1)
exten => s,n,Goto(from-trunk,00334XXXXXXX2,1)
exten => s,n,Goto(from-trunk,00334XXXXXXX3,1)

It is working well for the first one , but when I dialed for example the second or third one , all the calls are redirected as if I dialed the first in the list ( calls are redirected to the ring group corresponding to the first line in the conf file to be more specific )

So is there a way to make things work in a hard or dynamic way ? No matter wich way lol

4

You have already bee told that the syntax of your original use of the function is wrong.

The best solution here is to have your provider provide a proper Direct In Dialing service, i.e. they pass routing information in the request URI.

An unconditional Goto, as the first step of a program will always be taken and the subsequent lines not executed. Your new code is behaving exactly as expected.

5

Yeah , I know :frowning:

Any way to put the same priority and that the inbound routes be considered as equal ?

I tried a pattern because my nums all start the same way so I could have only one line and then redirect to the good route but I didn’t make that work. I tried :

exten => s,1,Goto(from-trunk,_00334XXXXXXXX,1)
exten => s,1,Goto(from-trunk,_00334X.,1)

I think that I have to take this problem “in a dynamical” way to not have priority routes problem , but there is my question , if the function PJSIP_HEADER requires a name as it is told in the error , what I have to put ?
Or do I have to do a PJSIPAddHeader first ?

6

You are missing $ signs from the original expression and are using the wrong sort of brackets around variables and function calls.

It looks like you need a dialplan designing for you, rather than one debugging. Especially given that you seem to be unfamiliar with programming concepts, you need to pay a consultant to write it for you and to subsequently support it.

7

A provider won’t change his way for a client ^^

I know some stuff in programming but I have to say that I don’t really understand what this line really return ( this is provide by my sip provider for chan_sip :
exten => s,1,Goto(from-trunk,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1) --> provide by sip provider
exten => s,1,Goto(from-trunk,${CUT(CUT(PJSIP_HEADER(To),@,1),:,2)},1) --> to match my case

Because when I take my SIP packet :

Retransmitting #3 (NAT) to X.X.X.X:5060:
REGISTER sip:mytrunk SIP/2.0
Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK51c03678;rport
Max-Forwards: 70
From: sip:00334XXXXXXXX@mytrunk;tag=as61398322
**To: sip:00334XXXXXXXX@mytrunk **
Call-ID: 6681469103d0b6fc6484e92a5048406c@[::1]
CSeq: 102 REGISTER
Supported: replaces, timer
User-Agent: FPBX-13.0.119(13.7.1)
Expires: 3600
Contact: sip:334XXXXXXXX@X.X.X.X:5060
Content-Length: 0

-I understand that in that way : Take the “To” line
To: sip:00334XXXXXXXX@mytrunk
-Then cut with the “@” as the seperate char and take the first column so you get sip:00334XXXXXXXX , then recut with the “:” as the seperate char and take the 2nd colum so you retrieve the DID number, but the function in Chan_SIP and Chan_PJSIP don’t have the same comportement because Chan_PJSIP needs a name ?

I saw the syntax PJSIP_HEADER(action,name[,number]) and the source code of the error :

if (ast_strlen_zero(args.header_name)) {
ast_log(AST_LOG_ERROR, “This function requires a header name.\n”);
return -1;
}
But I can’t see what I’m missing ( a header name , but what ? …)

I think that the problem is all in the syntax of the extensions.conf line but I don’t still have the knowledge to correct myself.

8

PJSIP_HEADER(read,To) would be the correct format. PJSIP is different.

9

As well the wiki[1] has documentation for this sort of stuff[2].

[1] https://wiki.asterisk.org/
[2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_PJSIP_HEADER

10

Yep , I tried that and it worked ! Thanks all for your support !