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 > > -- > _____________________________________________________________________ > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20141215/a189ded5/attachment.html>
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...
[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<mailto:george.joseph at fairview5.com>> wrote:
On Mon, Dec 15, 2014 at 7:34 PM, Dan Cropp <dan at amtelco.com<mailto: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<http://192.168.0.1/255.255.255.0> then the
correct entry would be 192.168.0.0/24<http://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>
[mailto: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<http://outbound.vitelity.net>]
type = aor
remove_existing = yes
qualify_frequency = 60
contact = sip:64.2.142.93
[outbound.vitelity.net<http://outbound.vitelity.net>]
type = endpoint
context = TestApp
transport = transport1
aors = outbound.vitelity.net<http://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<http://outbound.vitelity.net>]
type = identify
endpoint = outbound.vitelity.net<http://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
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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 populatedwith 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 >> >> -- >> _____________________________________________________________________ >> -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20141215/992abdfd/attachment-0001.html>