INVITE method has no number set

pjsip
trunk
Tags: #<Tag:0x00007f7029300ea8> #<Tag:0x00007f7029300930>

(virtexultra) #1

Hello,

when I try to make an outbound call via a pjsip trunk with provider sipgate, the capture of wireshark shows that in the request line there is no number transmitted.

Session Initiation Protocol (INVITE)
Request-Line: INVITE sip:sipconnect.sipgate.de SIP/2.0
Should be: INVITE sip:CalledNumber@sipconnect.sipgate.de SIP/2.0
    From: <sip:USERNAME_SIP@sipgate.de>
    To: <sip:CalledNumber@sipconnect.sipgate.de>
    Contact: <sip:USERNAME_SIP@ExternalIPAddr:5062>
    P-Preferred-Identity: <sip:OWN_Number@sipconnect.sipgate.de>
    User-Agent: FPBX-14.0.13.40(13.36.0)
    Proxy-Authorization:  Auth data with nonce and so on
    Route: <sip:CalledNumber@sipconnect.sipgate.de>
    Content-Type: application/sdp
    Content-Length:   264
Message Body

As a result of this the provider sipgate is responding with 404 not found.

On another trunk the number is transferred, I’ve found no real differences between the two trunks - only that the other trunk has an Client-URI for registration, but this is not necessary for sipgate.

Incoming calls are working.

To set the P-Preferred-Identity as required by the provider I use the following:

[macro-dialout-trunk-predial-hook]
s,n,ExecIF($["${OUT_${DIAL_TRUNK}_SUFFIX}"!=""]?Set(trunk_name=${OUT_${DIAL_TRUNK}_SUFFIX}):Set(trunk_name=${OUT_${DIAL_TRUNK}}))
exten => s,n,GoSubIf($["${trunk_name}"="@Sipgate_Trunk"]?func-set-sipheader,s,1(P-Preferred-Identity,<sip:${CALLERID(number)}@sipconnect.sipgate.de>))
exten => s,n,MacroExit

Same problem also described here:


Maybe somebody can give me a hint how to specify the format of the request line in pjsip.

Thanks in advance.


(virtexultra) #2

Hello again,

I’ve found the solution. For my understanding, the use of an proxy makes the difference. You have to add the loose routing option to the outbound_proxy parameter.

Like “outbound_proxy = sip:192.168.0.1” to “outbound_proxy = sip:192.168.0.1;lr”. This is all documented in the wiki:

https://wiki.asterisk.org/wiki/display/AST/PJSIP+with+Proxies

Hope this helps someone else.


(system) closed #3

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