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@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 <[email protected]>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:[email protected]>' failed for '179.25.158.95:5060' (callid: > [email protected]: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 > <[email protected]>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@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@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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