Displaying 20 results from an estimated 114 matches for "vitelity".
2014 Dec 16
3
PJSIP configuration question
...[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...
2009 Oct 09
2
Incoming extension not working.
Hi, all. I'm probably doing Something Dumb(tm), so please feel free to
point out whatever I'm missing, no matter how stupid.
Anyway, I've got IAX set up to Vitelity. When I try to call my DID, I get:
Rejected connect attempt from 64.2.142.19, who was trying to reach
'6031234567@'
This leads me to my first question -- why doesn't it show a context?
(My second is, what's wrong with the snippets, below?):
iax.conf:
[vitelity]
context=vitelity...
2008 Oct 04
5
Vitelity Asterisk configuration help
I have a Asterisk server setup and I am able to connect to the server
using a soft client 'x-lite' and call and leave a message on my second
extension 102. I have setup a Vitelity account and add what I believe
to be the correct information to my sip.conf and extension.conf. I
would like to setup incoming and outgoing calls with voicemail
support. I've searched all over but many of the full configurations
that are available are a bit complex. Any tips or recommendations...
2008 Sep 30
3
Maybe OT - routing calls in PSTN
I have a Vitelity DID which generally works, but calls from a particular
caller do not reach it. Vitelity has thus far disavowed any
responsibility for working through this problem. I recognize that some
action might be required by another provider which is outside Vitelity's
control, but it seems that the...
2014 Dec 15
2
PJSIP configuration question
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:* aster...
2014 Dec 16
2
PJSIP configuration question
...etwork 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?
>
>
Yo...
2014 Dec 15
2
PJSIP configuration question
...at?
>
>
>
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
&g...
2014 Dec 10
4
PJSIP configuration question
Not sure why, but Vitelity changed the settings to IP based authentication on me. Here's the new sip.conf settings they sent me.
type=friend
dtmfmode=auto
host=64.2.142.93
allow=all
nat=yes
canreinvite=no
trustrpid=yes
sendrpid=yes
When I use these settings to originate calls using the sip.conf they sent me, everythin...
2014 Dec 16
4
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=64.2.14...
2008 Oct 15
1
Cisco 7960 not always receiving incoming calls
I've searched around and found a few similar situations where the
phone will call out when using a Asterisk server but not receive
inbound calls. My issue is a little stranger. If I call out from the
phone then the phone will receive the next inbound call. The phone
will not receive another inbound call until a call out again from it
first. Any ideas?
I am using SIP and am using the latest
2010 Sep 04
3
Vitelity offline?
Vitelity seems to be offline to both IP and voice traffic. Is there any
place to find out what their status is?
Roger Marquis
2014 Dec 16
1
PJSIP configuration question
...n update...
My network admin would not turn off the ALG because it would cause several other problems to other phone systems we have.
He looked at the sip trace. What he found is the PJSIP trace showed a different IP address than the older chan_sip so he had me change the aor contact to outbound.vitelity.net
At this point, it seems to be working (and this is going through a Cisco ALG).
I will run more tests, but here is the pjsip.conf I have.
[global]
type = global
debug = yes
[transport1]
type = transport
bind = 0.0.0.0
protocol = udp
[outbound.vitelity.net]
type = aor
remove_existing = yes...
2014 Dec 16
0
PJSIP configuration question
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 si...
2008 Oct 09
2
Menu for call forwarding or voicemail
I would like to create a simple menu that would allow a caller to
decide whether they want to leave a message or be forwarded to another
number (i.e cell phone). Thanks in advance for any insight.
Here's my current extension.conf
[general]
static=yes
writeprotect=yes
[globals]
[default]
exten => 101,1,Dial(SIP/101,20)
exten => 101,n,Voicemail(101 at default)
;This automatically
2014 Dec 16
0
PJSIP configuration question
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.
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?
Have a great day!
Da
From: asterisk-users-b...
2009 Jul 28
1
outbound calls not reaching vitelity
Any vitelity customers with pbxinaflash boxes? I'm able to call
in-house, but failing to make outbound calls. My assigned server at
vitelity is not reachable. I can ping to my ISP OK.
Any help appreciated. Such as actually how to make email contact with
support at vitelity. They're not respondin...
2014 Dec 15
0
PJSIP configuration question
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.
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of George Joseph
Sent: Monday,...
2014 Dec 16
0
PJSIP configuration question
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.c...
2016 Aug 08
2
Asterisk & Vitelity Invite issues
Hi All,
We have asterisk 11.23 running sip to vitelity and from there IAX trunks
split off to where they need to go. We are having a problem getting
chan_sip to quit ignoring re-invites from Vitelity. Our side ends up
sending a reinvite which their side & they do not support us sending a
reinvite. Ive tried:
canreinvite=no which was supposedly...
2016 Aug 10
2
Asterisk & Vitelity Invite issues
On 8/9/16 12:40 PM, Matt Fredrickson wrote:
> On Mon, Aug 8, 2016 at 9:25 AM, Tammy Firefly <tammy-lists at wiztech.biz> wrote:
>> Hi All,
>>
>> We have asterisk 11.23 running sip to vitelity and from there IAX trunks
>> split off to where they need to go. We are having a problem getting
>> chan_sip to quit ignoring re-invites from Vitelity. Our side ends up
>> sending a reinvite which their side & they do not support us sending a
>> reinvite. Ive tried:
&...