Outbound calls terminated from ITSP with status 484

FreePBX Providers
1

Hello everyone

FreePBX 17 here, with Asterisk 21.

I’m not able to understand why ITSP is rejecting INVITE with status 484.

I’ve checked the chan_pjsip:Outbound proxy field, that is:

sip:<ITSP_domain>;lr

Here are logs from Asterisk. I guess I must ask ITSP why but, unfortunately, they aren’t reachable simply.

INVITE sip: SIP/2.0
Via: SIP/2.0/UDP <my_public_ip>:5060;rport;branch=z9hG4bKPjda1504f9-d727-4757-be27-9c860885e979
From: <sip:<caller_id>@>;tag=f0b2d1eb-2373-4819-ac84-7b1b16f39241
To: <sip:<called_party_number>@<ITSP_domain>
Contact: <sip:<caller_id>@<my_public_ip>:5060>
Call-ID: a4c9eaff-2182-4d0e-8003-f5f79bc7696d
CSeq: 11190 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: FPBX-17.0.25(22.8.0)
Proxy-Authorization: Digest username=“<caller_id>”, realm=“<ITSP_domain>”, nonce=“6991d1f300001cb1527cd6d9b0ac05d94f21a9ff5b3a1b66”, uri=“sip:<ITSP_domain>”, response=“5bccabaea32c97b81f268fbceb42f27e”
Route: <sip:<called_party_number>@:5060>
Content-Type: application/sdp
Content-Length: 384

v=0
o=- 2115908347 2115908347 IN IP4 <my_public_ip>
s=Asterisk
c=IN IP4 <my_public_ip>
t=0 0
m=audio 19064 RTP/AVP 9 18 0 8 3 111 101
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<— Received SIP response (442 bytes) from UDP:<ITSP_domain>:5060 —>
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP <my_public_ip>:5060;received=<my_public_ip>;rport=29381;branch=z9hG4bKPjda1504f9-d727-4757-be27-9c860885e979
From: <sip:<caller_id>@>;tag=f0b2d1eb-2373-4819-ac84-7b1b16f39241
To: <sip:<called_party_number>@>;tag=190cb40f7d50535d8671786d15223e97.67c7
Call-ID: a4c9eaff-2182-4d0e-8003-f5f79bc7696d
CSeq: 11190 INVITE
Server: OpenSIPS (1.9.0-notls (x86_64/linux))
Content-Length: 0

Someone can gently point me to the possible root cause(s) of that ?

Here is Wireshark capture of INVITE:

Frame 59: Packet, 1362 bytes on wire (10896 bits), 1362 bytes captured (10896 bits)
Ethernet II, Src: RaspberryPiT_xx:xx:xx (xx:xx:xx:xx:xx:xx), Dst: ChangwangTec_xx:xx:xx (xx:xx:xx:xx:xx:xx)
Internet Protocol Version 4, Src: <Asterisk_Internal_IP>, Dst: <ITSP_domain>
User Datagram Protocol, Src Port: 5060, Dst Port: 5060
Session Initiation Protocol (INVITE)
Request-Line: INVITE <ITSP_domain> SIP/2.0
Method: INVITE
Request-URI: sip:<ITSP_domain>
Request-URI Host Part: <ITSP_domain>
[Resent Packet: False]
Message Header
Message Body

2

Outbound Proxy should be

sip:1.2.3.4\;lr\;hide

(replace 1.2.3.4 with the proper address or domain name supplied by your provider)

Note backslash semicolon in two places.

3

I’ve updated original post, trying to anonimyze logs.

It is already like described, with only exception of \;hide

4

Your logs are very garbled. Correct markup for the forum (pre-formatted text) would help, although discord have broken that a bit.

Nonetheless, it is fairly clear that Asterisk didn’t see the ;lr, which strongly suggests you didn’t include the back slash.

5

As @david55 said. But also, if the value of ITSP_domain in your Outbound Proxy is the same as ITSP_domain in SIP Server, then you shouldn’t need Outbound Proxy at all; try leaving it blank.

6

Hi

Sorry to have posted garbled logs. I have solved the issue, removing all settings from Outbound Proxy (without taking care of FreePBX warning about field format. I had also to increase maximum channels, even if this trunk is able to handle only 1 conversation across the trunk itself.

Now I’m facing with mono-directional audio, but this could be related to firewall & nat.

Thanks for you’re time.