[asterisk-users] PJSIP configuration question

Tue Dec 16 10:10:18 CST 2014

On Tue, Dec 16, 2014 at 9:00 AM, Dan Cropp <dan at 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-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at 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 <dan at 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.joseph at fairview5.com>
> wrote:
>
>
>
>
>
> On Mon, Dec 15, 2014 at 7:34 PM, Dan Cropp <dan at 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-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at 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
>
>
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>
> --
> _____________________________________________________________________
> -- 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
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