Just setup fanvil x3sp phone as local extension
FreePBX is local on same lan etc
Shows as registered on phone
Can dial out
However FreePBX shows as unavailable and unregistered
Any ideas
Works as extension with another voip service (external)
Not without logs, and the configuration would also help.
Is it registering to the correct channel driver?
I’m assuming it’s a pjsip channel compatible device ?
I can dial out without any issues
Line 2 is fine
Just wondering if there was anything odd for this and FreePBX
I have another local extension configured on Zoiper exactly the same that works ok
Thanks for time, appreciated
Config below
– endpoint conf —
[1111]
type=endpoint
aors=1111
auth=1111-auth
tos_audio=ef
tos_video=af41
cos_audio=5
cos_video=4
allow=ulaw,alaw,gsm,g726,g722
context=from-internal
callerid=1111 <1111>
dtmf_mode=rfc4733
direct_media=yes
mailboxes=1111@device
mwi_subscribe_replaces_unsolicited=yes
aggregate_mwi=no
use_avpf=no
rtcp_mux=no
max_audio_streams=1
max_video_streams=1
bundle=no
ice_support=no
media_use_received_transport=no
trust_id_inbound=yes
user_eq_phone=no
media_encryption=no
timers=yes
timers_min_se=90
media_encryption_optimistic=no
refer_blind_progress=yes
refer_blind_progress=yes
rtp_timeout=30
rtp_timeout_hold=300
rtp_keepalive=0
send_pai=yes
rtp_symmetric=yes
rewrite_contact=yes
force_rport=yes
language=en_GB
one_touch_recording=on
record_on_feature=apprecord
record_off_feature=apprecord
Endpoint: 1111/1111 Unavailable 0 of inf
InAuth: 1111-auth/1111
Aor: 1111 1
Contact: 1111/sip:[email protected]:5060 781a22956d Unavail nan
Here’s an odd thing in the cap trace
Contact: < sip:[email protected]:5060**>**
I have no idea what that IP address is as its obviously not a local IP
11346 [2024-04-21 20:46:11] VERBOSE[18162] res_pjsip_registrar.c: Added contact ‘sip:[email protected]:5126’ to AOR ‘1111’ with expiration of 3600 seconds
11347 [2024-04-21 20:46:14] VERBOSE[4720] res_pjsip/pjsip_options.c: Contact 1111/sip:[email protected]:5126 is now Unreachable. RTT: 0.000 msec
According to Whois Lookup Captcha , it’s assigned to a Virgin Media cable customer in the UK. If that address is not yours, possibly you accidentally connected to a neighbor’s Wi-Fi, or if you system is open to the internet and extension 1111 has a very weak password, an attacker may have registered it.
1 Like
Thanks for suggestions, appreciated…
I kinda thought similar, however everything here is hardwired cat6 poe Fanvil x3sp phone and hardwired pbx etc, system isn’t cloud based or remote in any way and I don’t even use Virgin
I don’t know where it would get that from…
Also from asterisk logs it shows Fanvil phone immediately becomes unreachable
11346 [2024-04-21 20:46:11] VERBOSE[18162] res_pjsip_registrar.c: Added contact ‘sip:[email protected]:5126’ to AOR ‘1111’ with expiration of 3600 seconds
11347 [2024-04-21 20:46:14] VERBOSE[4720] res_pjsip/pjsip_options.c: Contact 1111/sip:[email protected]:5126 is now Unreachable. RTT: 0.000 msec
At the Asterisk command prompt, type
pjsip set logger on
wait for the Fanvil to attempt another registration (or reboot it), paste the relevant section of the Asterisk log at pastebin.com and post the link here.
I can’t find where pjsip command lives on the box?
I can capture and create a cap file
At the Asterisk command prompt, not a shell prompt.
From a root shell prompt, type
asterisk -r
to get an Asterisk command prompt.
Great. Put it in a .tgz file (or just rename it with extension .tgz) and attach it to your post.
In Asterisk SIP Settings, confirm that Local Networks and External Address are correctly set.
On the pjsip tab, the settings for 0.0.0.0 (udp), other than Port to Listen On, should be left blank.
After Submit and Apply Config, you must restart Asterisk.
Based on what you have told us so far, Local Networks should be 192.168.1.0 / 24
If you believe otherwise, please explain.
Indeed it is… Just checking network settings
restarted gracefully
20121 [2024-04-21 21:45:38] VERBOSE[2312] asterisk.c: Asterisk Ready.
20122 [2024-04-21 21:46:02] VERBOSE[13933] res_pjsip/pjsip_options.c: Contact 1111/sip:[email protected]:5126 is now Unreachable. RTT: 0.000 msec
But your SIP trace showed
11930 [2024-04-21 21:25:31] VERBOSE[4720] res_pjsip_logger.c: <--- Transmitting SIP request (423 bytes) to UDP:192.168.1.111:5126 --->
11931 OPTIONS sip:[email protected]:5126 SIP/2.0
11932 Via: SIP/2.0/UDP 82.45.17.35:5060;rport;branch=z9hG4bKPja95e6aef-61eb-49e7-8e9a-d6993f372bac
11933 From: <sip:[email protected]>;tag=1a53c4e2-1801-4caf-bf0f-2e041371df46
11934 To: <sip:[email protected]>
11935 Contact: <sip:[email protected]:5060>
11936 Call-ID: a13899d8-6f81-465a-9ac9-b57c2be12405
11937 CSeq: 43562 OPTIONS
11938 Max-Forwards: 70
11939 User-Agent: FPBX-16.0.40.7(19.8.0)
11940 Content-Length: 0
So Asterisk thought that 192.168.1.111 was not local and substituted the incorrect 82.45.17.35 as its public IP. Can you post screenshots of the Asterisk SIP settings pages?
Hi
I cannot thank you enough, it was indeed a setting for the NAT that freepbx had picked up from our alternate service provider!
I’m guessing it grabbed it when it coincidentally rebooted while the main service provider was down and it had switched over…
Re-detected network and forced a full reboot and YEH 
One to remember going forward and not to repeat any time soon…
NB for those watching this thread, the settings exposed have been changed… **