On Tue, Dec 16, 2014 at 9:00 AM, Dan Cropp <d...@amtelco.com> wrote: > > I corrected my local_net setting (based on advice from network admin). > > > > I have tried several different values for the from_user and still have the > same problem. > > > > Asterisk receives the OK from Vitelity. > > Asterisk sends the ACK (without a Contact header). > > Vitelity doesn’t seem to process it, so they send an OK again. > > > > The OK receive, Transmit ACK occurs 4 times. > > A short while later, Vitelity hangs up on my cell phone. > > > > Asterisk is never told the call is gone. > > > > If I hangup the call from Asterisk side, > > Asterisk sends the BYE message. > > Vitelity responds with a “SIP/2.0 481 Call leg/transaction does not exist” > > > > Again, the trace indicates the ACK message is missing the Contact header. > > > > Additional note: the network admin is asking why the local_net, > external_media_address, and external_signalling_address are needed. He > wrote me…“You should NOT have to know your public IP. The firewall > should be doing fixup commands to change the values in the packet” > > First...
"The firewall should be doing fixup commands to change the values in the packet” *The firewall should NOT be changing values in the packet. If it is, all bets are off.* Second. Can you try making a call from a phone instead of from an AMI originate? > > > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *George Joseph > *Sent:* Monday, December 15, 2014 11:14 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] PJSIP configuration question > > > > On Mon, Dec 15, 2014 at 9:48 PM, Dan Cropp <d...@amtelco.com> wrote: > > Thanks George. > > > > I will correct my local_net in the morning. > > > > Vitelity chan_sip settings I have working, do not have a fromuser. > > sip.conf settings... > > > > I think you can actually specify anything, it just has to be populated > with something other than a sub-account username. > > > > > > [HVout] > > type=friend > > dtmfmode=auto > > host=64.2.142.93 > > disallow=all > > allow=ulaw > > canreinvite=no > > trustrpid=yes > > sendrpid=yes > > nat=yes > > context=TestApp > > > > > On Dec 15, 2014, at 9:32 PM, George Joseph <george.jos...@fairview5.com> > wrote: > > > > > > On Mon, Dec 15, 2014 at 7:34 PM, Dan Cropp <d...@amtelco.com> wrote: > > I am not sure if I entered the correct settings for the transport > information. > > For the local_net, I entered my local ip address, but no mask. I will > check with the network admin so he can verify the settings I entered. > > > > You need the network and mask. For example if the ip address and mask of > the test machine is 192.168.0.1/255.255.255.0 then the correct entry > would be 192.168.0.0/24. > > > > One minor detail, we are using ip authentication. When Vitelity changed > my account from user based authentication to IP based authentication, they > stopped including a user for the account. > > > > Should these settings work without the from_user (IP based authentication) > or do I need to get the account name from Vitelity? > > > > You definitely need the master account login username. If you has this > working with chan_sip, then try the 'fromuser' from sip.conf and user is > from_user. > > > > > > > > > > Have a great day! > > > > Da > > > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *George Joseph > *Sent:* Monday, December 15, 2014 7:27 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] PJSIP configuration question > > > > Ok Dan, try this... I was able to get this to work behind a NAT and with > ip address authentication. > > [global] > type = global > debug = yes > > [transport1] > type = transport > bind = 0.0.0.0 > protocol = udp > > > > *local_net=<yourlocalnet I.E. 10.10.10.10/24 > <http://10.10.10.10/24>>external_media_address=<your public ip > address>external_signaling_address=<your public address>* > [outbound.vitelity.net] > type = aor > remove_existing = yes > qualify_frequency = 60 > contact = sip:64.2.142.93 > > [outbound.vitelity.net] > type = endpoint > context = TestApp > transport = transport1 > aors = outbound.vitelity.net > dtmf_mode = rfc4733 > force_rport = yes > rtp_symmetric = yes > rewrite_contact = yes > send_rpid = yes > trust_id_inbound = yes > disallow = all > allow = ulaw > direct_media = no > > *from_user=<your main vitelity account name> ; Not subaccount* > > [outbound.vitelity.net] > type = identify > endpoint = outbound.vitelity.net > match = 64.2.142.93 > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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