[asterisk-users] PJSIP configuration question
George Joseph
george.joseph at fairview5.com
Mon Dec 15 23:14:02 CST 2014
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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> --
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>
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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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