FreePBX and SIP trunk: Outbound calls always busy

@olilein wrote:

Hello there,

after trying for a few days to debug the problem for myself, I find myself completely at a loss. I tried to configure a FreePBX installation (based on raspbx, so Asterisk 13.26) to use a SIP trunk (from sipgate de).

Inbound calls work, outbound calling always fails. The log always looks something like this:

[2019-06-13 19:39:55] VERBOSE[24640][C-00000065] app_stack.c: Spawn extension (from-pstn, 08003301000, 1) exited non-zero on ‘PJSIP/sipgate-000000aa’
[2019-06-13 19:39:55] VERBOSE[24640][C-00000065] app_stack.c: PJSIP/sipgate-000000aa Internal Gosub(func-apply-sipheaders,s,1(1)) complete GOSUB_RETVAL=
[2019-06-13 19:39:55] VERBOSE[24640][C-00000065] app_dial.c: Called PJSIP/08003301000@sipgate
[2019-06-13 19:39:55] VERBOSE[24640][C-00000065] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)
[2019-06-13 19:39:55] VERBOSE[24640][C-00000065] pbx.c: Executing [s@macro-dialout-trunk:26] NoOp(“PJSIP/20-000000a9”, “Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 1”) in new stack
[2019-06-13 19:39:55] VERBOSE[24640][C-00000065] pbx.c: Executing [s@macro-dialout-trunk:27] GotoIf(“PJSIP/20-000000a9”, “0?continue,1:s-CHANUNAVAIL,1”) in new stack

(I can provide more lines but those seem the most relevant to me - as a novice Asterisk user)

I - of course - also turned on the pjsip logging on the console:

<--- Transmitting SIP request (1108 bytes) to UDP:217.10.68.150:5060 --->
INVITE sip:sipconnect.sipgate.de SIP/2.0
Via: SIP/2.0/UDP 172.20.0.250:5060;rport;branch=z9hG4bKPj0aded5b3-b217-452c-    9e06-2631d6d9388c
From: <sip:<accountname>@sipconnect.sipgate.de>;tag=1d5d535f-3668-4198-b3e5-54d2a7479da8
To: <sip:08003301000@sipconnect.sipgate.de>
Contact: <sip:<myaccount>@172.20.0.250:5060>
Call-ID: 32747040-e035-47ee-8644-9230169c6b22
CSeq: 15756 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: "me" <sip:<mynumber>@sipconnect.sipgate.de>
Route: <sip:08003301000@sipconnect.sipgate.de:5060>
Max-Forwards: 70
User-Agent: FPBX-14.0.11(13.26.0)
Content-Type: application/sdp
Content-Length:   292

v=0
o=- 1869075610 1869075610 IN IP4 172.20.0.250
s=Asterisk
c=IN IP4 172.20.0.250
t=0 0
m=audio 15934 RTP/AVP 0 8 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (418 bytes) from UDP:217.10.68.150:5060 --->
SIP/2.0 404 Not found (no match)
Via: SIP/2.0/UDP 172.20.0.250:5060;received=<my IP>;rport=65322;branch=z9hG4bKPj0aded5b3-    b217-452c-9e06-2631d6d9388c
From: <sip:<myaccount>@sipconnect.sipgate.de>;tag=1d5d535f-3668-4198-b3e5-54d2a7479da8
To: <sip:08003301000@sipconnect.sipgate.de>;tag=59090977d38d1697fd81fd97f073a25d.0c69
Call-ID: 32747040-e035-47ee-8644-9230169c6b22
CSeq: 15756 INVITE
Content-Length: 0

The phone number is a customer hotline from Deutsche Telekom which definitely is not busy and also reachable. I configured Linphone with the trunk account in parallel just to be sure the trunk account works - and it does. The SIP messages in the Linphone debug window are identical except for two things, as far as I can see (the following call works from Linphone):

INVITE sip:08003301000@sipconnect.sipgate.de SIP/2.0
Via: SIP/2.0/UDP 172.20.0.248:5060;branch=z9hG4bK.d2zIAW~GF;rport
From: <sip:<myaccount>@sipconnect.sipgate.de>;tag=VYvy0zFTp
To: sip:08003301000@sipconnect.sipgate.de
CSeq: 21 INVITE
Call-ID: Z-VHVSo9kQ
Max-Forwards: 70
Supported: replaces, outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 686
Contact: <sip:<myaccount>@<my IP>:65320;transport=udp>;+sip.instance="    <urn:uuid:98caef36-4e82-4f4c-9c6c-455a506dc76b>"
User-Agent: Linphone/3.12.0 (belle-sip/1.6.3)
Proxy-Authorization:  Digest realm="sipconnect.sipgate.de", nonce="<nonce>", username="<myaccount>",  uri="sip:08003301000@sipconnect.sipgate.de", response="313209faaf9c6b146ad6578561380960"

1.) The Contact header line. I tried playing around with the PJSIP settings to force it to reflect my public IP or sipconnect dot sipgate dot de, as some Google hits suggested. Didn’t change anything.

2.) The Proxy-Authorization header is not present when FreePBX makes the call (at this point, I’m not sure if this is relevant and how to change it).

Any ideas are greatly appreciated! Thank you!

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