asterisk users - Apr 2019 - Incoming SIP call, outgoing SIP registration. PJSIP.

Hi,

Got problems with incoming SIP calls.

Scenario:

Server1: 3cx or any other server

Server2: Asterisk 16.2.1 . PJPROJECT 2.8

Server2 registers on Server1 with SIP ID 1121.

Registration is OK.

Server2 outgoing calls are OK.

INVITE, unauthorized, INVITE with password, OK, RINGING,...

Troubles with incoming calls / incoming INVITE's .

I can not identify endpoint by IP, I have multiple registrations on the 
same Server1.

As far as I understood, res_pjsip_endpoint_identifier_user match 
endpoint by "From" header, so it will not match also.

match_headers also seems useless (not able to match "INVITE" string, 
just headers like "TO:").

Is there any way to match incoming INVITE calls ? (also OPTIONS, NOTIFY, 
... packets)

It should be a typical scenario, but it does not work...

Is there any way to make it working ?


-------------- next part --------------
[0.0.0.0-udp]
type=transport
protocol=udp
bind=0.0.0.0:5060

[endpoint0](!)
type=endpoint
transport=0.0.0.0-udp
disallow=all
allow=alaw
allow=ulaw
t38_udptl=no
t38_udptl_ec=none
fax_detect=no
t38_udptl_nat=no
dtmf_mode=auto
direct_media=yes
from_domain=172.16.25.23
timers_sess_expires=1800
tone_zone=ru
language=ru
rewrite_contact=yes
rtp_symmetric=yes
force_rport=yes

[registration0](!)
type=registration
transport=0.0.0.0-udp
retry_interval=60
max_retries=10
expiration=3600
auth_rejection_permanent=yes
server_uri=sip:172.16.25.23


[fxs17](endpoint0)
context=from-sip-fxs
aors=fxs17
outbound_auth=fxs17
from_user=1121
set_var=DAHDICHAN=17

[fxs17]
type=aor
qualify_frequency=60
contact=sip:1121 at 172.16.25.23

[fxs17]
type=auth
auth_type=userpass
password=111111
username=1121

[fxs17](registration0)
outbound_auth=fxs17
client_uri=sip:1121 at 172.16.25.23
contact_user=fxs17

On Mon, Apr 22, 2019, at 1:43 PM, Pavel wrote:
> Hi,
> 
> Got problems with incoming SIP calls.
> 
> Scenario:
> 
> Server1: 3cx or any other server
> 
> Server2: Asterisk 16.2.1 . PJPROJECT 2.8
> 
> Server2 registers on Server1 with SIP ID 1121.
> 
> Registration is OK.
> 
> Server2 outgoing calls are OK.
> 
> INVITE, unauthorized, INVITE with password, OK, RINGING,...
> 
> Troubles with incoming calls / incoming INVITE's .
> 
> I can not identify endpoint by IP, I have multiple registrations on the 
> same Server1.
> 
> As far as I understood, res_pjsip_endpoint_identifier_user match 
> endpoint by "From" header, so it will not match also.
> 
> match_headers also seems useless (not able to match "INVITE"
string,
> just headers like "TO:").
> 
> Is there any way to match incoming INVITE calls ? (also OPTIONS, NOTIFY, 
> ... packets)
> 
> It should be a typical scenario, but it does not work...
> 
> Is there any way to make it working ?
Outbound registration provides the line option[1] which can be used to
differentiate traffic in regards to different outbound registrations. It
requires the remote server to adhere to the SIP RFC and report back some data we
give in our Contact, so you have to test it and see if it works.

[1]
https://blogs.asterisk.org/2016/01/27/the-pjsip-outbound-registration-line-option/

-- 
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

Hi,

Thank for your answer.

22.04.2019 23:47, Joshua C. Colp пишет:
> On Mon, Apr 22, 2019, at 1:43 PM, Pavel wrote:
>> Hi,
>>
>> Got problems with incoming SIP calls.
>>
>> Scenario:
>>
>> Server1: 3cx or any other server
>>
>> Server2: Asterisk 16.2.1 . PJPROJECT 2.8
>>
>> Server2 registers on Server1 with SIP ID 1121.
>>
>> Registration is OK.
>>
>> Server2 outgoing calls are OK.
>>
>> INVITE, unauthorized, INVITE with password, OK, RINGING,...
>>
>> Troubles with incoming calls / incoming INVITE's .
>>
>> I can not identify endpoint by IP, I have multiple registrations on the
>> same Server1.
>>
>> As far as I understood, res_pjsip_endpoint_identifier_user match
>> endpoint by "From" header, so it will not match also.
>>
>> match_headers also seems useless (not able to match "INVITE"
string,
>> just headers like "TO:").
>>
>> Is there any way to match incoming INVITE calls ? (also OPTIONS,
NOTIFY,
>> ... packets)
>>
>> It should be a typical scenario, but it does not work...
>>
>> Is there any way to make it working ?
> Outbound registration provides the line option[1] which can be used to
differentiate traffic in regards to different outbound registrations. It
requires the remote server to adhere to the SIP RFC and report back some data we
give in our Contact, so you have to test it and see if it works.
>
> [1]
https://blogs.asterisk.org/2016/01/27/the-pjsip-outbound-registration-line-option/
Tried already.

"line" is good, but not perfect.

Every time I restart asterisk, it will generate new random string for 
";line=".

So, every time I restart asterisk, registrar (Server1) will save one 
more contact in it's database.

Some will remove obsolete contacts, but some will not.

For example, FreePBX will not remove obsolete contacts, if max_contacts 
specified (FreePBX will set rewrite_contact=no in this case).

So, after a number of Asterisk restarts, FreePBX will reject new 
registrations, as max_contacts is reached.

Unfortunately, "line" does not save random between restarts.

It's also unable to specify "random" value in pjsip.conf.


I'm thinking to patch res_pjsip_outbound_registration to add this feature.

Am I wrong and  there is another way ?

It's also a security hole, as anybody can generate INVITE with 
";line=random" from any IP address !

res_pjsip_outbound_registration will only match "line", but will not 
take care about source IP, ...


Is there any more clear way to identify incoming INVITE/OPTIONS packets ?

Not very familliar with SIP, not sure, how should it be done.

-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20190423/f595b2ad/attachment.html>