[asterisk-users] No register between Asterisk 15 and 13 running pjsip

Administrator TOOTAI admin at tootai.net
Thu Jul 5 12:24:38 CDT 2018


Hello,

we have 4 asteriks, 2 in office on one server (wazo and mobydick), and 2 
in DC (self compiled) each on his own server. All of them are VMs under 
Debian Stretch. We used OpenVPN to connect the machines together in TAP 
mode, everything was running well.

Setup is following: the 2 asterisk in office on the same server are 
Asterisk 15 (wazo) and Asterisk 11certified (mobydick). Each of them is 
connected to the 2 others Asterisk in DC, both being Asterisk 13, all 
using chan_sip except one in DC wich is pjsip. They are also 2 IP phones 
in the office which are connected to all servers. As stated above, in 
tap mode everything is running well.

Now we changed our VPNs to use tun. The setup was tested appart of 
asterisks, all connections are OK, all machines can speak to each others 
including Windows one.

Now the problem: all VOIP devices are connecting as before except the 
Asterisk 15 from Office who can't register to the Asterisk 13 in DC 
running pjsip. No problem with the Asterisk11 certified against the same 
pjsip, as well as no problem to the other Asterisk 13 in DC running 
chan_sip.

What we get:

<--- Received SIP request (394 bytes) from UDP:10.99.0.52:5060 --->
REGISTER sip:zone-s SIP/2.0
Via: SIP/2.0/UDP 192.168.12.250:5060;branch=z9hG4bK4d838884
Max-Forwards: 70
From: <sip:zwr-IPBX at zone-s>;tag=as17aa56c4
To: <sip:zwr-IPBX at zone-s>
Call-ID: 2efc5e2320a31ff1107505663a02397d at 127.0.1.1
CSeq: 102 REGISTER
Supported: replaces, timer
User-Agent: Office PBX
Expires: 3600
Contact: <sip:callbackextension at 192.168.12.250:5060>
Content-Length: 0


[2018-07-05 18:49:08] NOTICE[21317]: acl.c:750 ast_apply_acl: SIP ACL: 
Rejecting '10.99.0.52' due to a failure to pass ACL '(BASELINE)'
[2018-07-05 18:49:08] NOTICE[21317]: res_pjsip/pjsip_distributor.c:649 
log_failed_request: Request 'REGISTER' from '<sip:zwr-IPBX at zone-s>' 
failed for '10.99.0.52:5060' (call$
d: 2efc5e2320a31ff1107505663a02397d at 127.0.1.1) - Not match Endpoint ACL
<--- Transmitting SIP response (322 bytes) to UDP:10.99.0.52:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 
192.168.12.250:5060;rport=5060;received=10.99.0.52;branch=z9hG4bK4d838884
Call-ID: 2efc5e2320a31ff1107505663a02397d at 127.0.1.1
From: <sip:zwr-IPBX at zone-s>;tag=as17aa56c4
To: <sip:zwr-IPBX at zone-s>;tag=z9hG4bK4d838884
CSeq: 102 REGISTER
Server: TOOTAiAudio
Content-Length:  0

where 10.99.0.52 is the IP of the office tun VPN and 192.168.12.250 is 
the Asterisk 15 IP. zone-s is the hostname of the Asterisk pjsip server. 
The 10.99.0.52 is not in ACL (we tried by including it but no luck). 
zwr-IPBX is the username/auth_user. Remember, both VOIP phones as well 
as the Asterisk 11 server connect without problem. The Asterisk 11 an 
Asterisk 13 configuration is the same in pjsip.conf appart of 
username/auth_name.

If someone had any clue on this.

Regards

Daniel



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