[asterisk-users] Connecting 2 asterisks, one with PJSIP and other SIP returning 401

[Gervasio Marchand Cassataro]

I'm not exactly nailing it in participation on my thread, so I swear this
is my last message if no one replies ;)

I went ahead and enabled full debugging, and got some interesting results
(available at http://pastebin.com/KiY6DMHi)

What I see is:

   1. The registration works fine
   2. When the client tries to establish a call to the server, the server
   looks into either the From or Contact headers... but the client sends the
   caller id in that place and that's when the INVITE gets rejected

I guess a proper question would be "Is there any way on the sip.conf to
specify the contact (I think that s at ip would work, as that's registered on
the client) or on the pjsip.conf to whitelist the ip of a registered
contact? something like insecure=invite"

Thanks!


On Wed, Apr 16, 2014 at 8:02 PM, Gervasio Marchand Cassataro <gmc at gmc.uy>wrote:

> Just a heads up... Enabled NOTICEs on the server and I see this every 10
> seconds or so
>
> [Apr 16 18:58:28] NOTICE[2138]: res_pjsip/pjsip_distributor.c:246 log_unidentified_request: Request from '"asterisk" <sip:asterisk at 179.25.158.95>' failed for '179.25.158.95:5060' (callid: 477ca2fd0db3a5542dcf2afd50673b89 at 179.25.158.95:5060) - No matching endpoint found
>
> Thanks in advance for any help / ideas / clues or something! I'm
> scratching my head around this and at this point
>
>
> On Wed, Apr 16, 2014 at 6:26 PM, Gervasio Marchand Cassataro <gmc at gmc.uy>wrote:
>
>> It's my first post here, so I'll cut to the chase
>>
>> I have 2 Asterisk servers and want to connect them using sip on one and
>> pjsip on the other one. One is running at home and another at a VPS. The
>> first one will be the client (with dynamic ip) and the 2nd the server.
>>
>> The client uses sip and the server pjsip.
>>
>> This is the client's sip.conf
>>
>> [general]
>> context = default
>> allowguest = no
>> realm = myrealm.com
>> udpbindaddr = 0.0.0.0
>> qualify = yes
>> subscribecontext = default
>> localnet=192.168.1.0/255.255.255.0
>> externhost=myhost.com <http://192.168.1.0/255.255.255.0externhost=myhost.com>
>> externrefresh=30
>> dtmfmode = auto
>> canreinvite = no
>> jbenable = no
>> sendrpid = yes
>> trustrpid = no
>> disallow=all
>> allow=ulaw
>> allow=alaw
>> register => myuser:mypass at vpsserver
>>
>> [vpsserver]
>> type=friend
>> secret=myuser
>> defaultuser=mypass
>> host=vpsserver.domain.com
>> context=inbound
>> canreinvite=no
>> insecure=port,invite
>>
>> And this is the server's pjsip.conf
>>
>> [transport-udp]
>> type=transport
>> protocol=udp
>> bind=0.0.0.0
>>
>> [home]
>> type=endpoint
>> context=trusted
>> disallow=all
>> allow=ulaw
>> allow=alaw
>> transport=transport-udp
>> auth=home
>> aors=home
>>
>> [home]
>> type=auth
>> auth_type=userpass
>> password=mypass
>> username=myuser
>>
>> [home]
>> type=aor
>> max_contacts=10
>>
>> When I check on the client, executing sip show registry I get
>>
>> Host                                    dnsmgr Username       Refresh State                Reg.Time
>> vpsserver:5060                              N      myuser       104 Registered           Tue, 15 Apr 2014 22:57:34
>>
>> which I guess means everything is ok... on the client side, I have on my
>> extensions.conf
>>
>> exten => 66,1,Dial(SIP/1 at vpsserver)
>>
>> and on the server's extensions.conf (in the trusted context) I have
>>
>> exten => 1,1,Playback(hello-world)
>>
>> So far so good... but when I dial 66 on my client Asterisk, I see the
>> following SIP dialog on the server... the only weird thing is that I see
>> some From: 192.168.1.112 (that's my home Asterisk's internal IP... the
>> externhost works fine for all the providers I'm using, so I doubt that's an
>> issue)
>>
>> http://pastebin.com/hkFezB8j
>>
>> Thanks in advance!
>>
>
>
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