On Mon, Dec 15, 2014 at 3:54 PM, Dan Cropp <dan at amtelco.com> wrote:> > Yes, everything is behind the same NAT. > > > > For the application I?m working on, the only endpoint is the endpoint to > Vitelity. > > We use AMI to Originate calls from Asterisk endpoint through Vitelity to > phones. > > After that, we control the call through AMI to perform the work we need. > > > >And it's outbound calls that aren't working right?> > > > > *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 4:42 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] PJSIP configuration question > > > > > > > > On Mon, Dec 15, 2014 at 3:33 PM, Dan Cropp <dan at amtelco.com> wrote: > > Hi George, > > > > Thank you for looking into this. > > This is behind a nat? > > > > > > Just to be clear...both the pbx and local endpoints are behind the same > NAT? > > > > > > [global] > > type = global > > debug = yes > > > > [transport1] > > type = transport > > bind = 0.0.0.0 > > protocol = udp > > > > [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 > > > > [outbound.vitelity.net] > > type = identify > > endpoint = outbound.vitelity.net > > match = 64.2.142.93 > > > > Have a great day! > > > > Dan > > > > *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 3:40 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] PJSIP configuration question > > > > On Mon, Dec 15, 2014 at 2:08 PM, Dan Cropp <dan at amtelco.com> wrote: > > Today, I tried the same behavior on Asterisk 13.1.0 and Asterisk 12.2.0. > > > > Same problem is happening with both of them. > > > > Could this be caused by PJPROJECT 2.3? > > > > Anyone have any suggestions for what I can try? > > > > My boss is giving me until tomorrow to get the PJSIP support working with > Vitelity. Otherwise, he?s told me to go back to using chan_sip and wait a > year or two for PJSIP to be in the field more. > > > > I have a Vitelity account I can try. Re-post your pjsip config and I'll > try it now. > > > > > > > > Have a great day! > > > > Dan > > > -- > _____________________________________________________________________ > -- 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/afae2bd0/attachment-0001.html>
Yes, outbound calls are the only ones I?m trying.
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 4:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP configuration question
On Mon, Dec 15, 2014 at 3:54 PM, Dan Cropp <dan at amtelco.com<mailto:dan
at amtelco.com>> wrote:
Yes, everything is behind the same NAT.
For the application I?m working on, the only endpoint is the endpoint to
Vitelity.
We use AMI to Originate calls from Asterisk endpoint through Vitelity to phones.
After that, we control the call through AMI to perform the work we need.
And it's outbound calls that aren't working right?
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 4:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP configuration question
On Mon, Dec 15, 2014 at 3:33 PM, Dan Cropp <dan at amtelco.com<mailto:dan
at amtelco.com>> wrote:
Hi George,
Thank you for looking into this.
This is behind a nat?
Just to be clear...both the pbx and local endpoints are behind the same NAT?
[global]
type = global
debug = yes
[transport1]
type = transport
bind = 0.0.0.0
protocol = udp
[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
[outbound.vitelity.net<http://outbound.vitelity.net>]
type = identify
endpoint = outbound.vitelity.net<http://outbound.vitelity.net>
match = 64.2.142.93
Have a great day!
Dan
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 3:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP configuration question
On Mon, Dec 15, 2014 at 2:08 PM, Dan Cropp <dan at amtelco.com<mailto:dan
at amtelco.com>> wrote:
Today, I tried the same behavior on Asterisk 13.1.0 and Asterisk 12.2.0.
Same problem is happening with both of them.
Could this be caused by PJPROJECT 2.3?
Anyone have any suggestions for what I can try?
My boss is giving me until tomorrow to get the PJSIP support working with
Vitelity. Otherwise, he?s told me to go back to using chan_sip and wait a year
or two for PJSIP to be in the field more.
I have a Vitelity account I can try. Re-post your pjsip config and I'll try
it now.
Have a great day!
Dan
--
_____________________________________________________________________
-- 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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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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20141215/15754797/attachment.html>