On 10.12.2014 11:42, Frederic Van Espen wrote:
Hi,
- Could you share the details of the SDP in each INVITE and OK packet?
- How are your SIP endpoints configured in asterisk sip.conf? (the SIP
trunk provider and the local endpoint)
- What type is the local endpoint?
Cheers,
Frederic
On 14.12.2014 22:10, Larry Moore wrote:
Asterisk 1.8 with the T.38 Gateway patch (not the one by Niccolò Belli) sends
a T.38 invite to the ITSP when the fax tones are detected from the callee,
the T.38 Gateway implementation in Asterisk 10 and Niccolò's back-port for
Asterisk 1.8.11 does
On Mon, Dec 15, 2014 at 3:34 AM, Recursive li...@binarus.de wrote:
snip
For asterisk 1.6 1.8 you would need to set 'canreinvite=no', I don't know
what Asterisk 13 will do with this setting.
I suspect Asterisk 13 will just ignore it. To make things worse, there seems
to be a
The Asterisk Development Team has announced the release of Asterisk 11.15.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 11.15.0 resolves several issues reported by the
community and would have not been possible
The Asterisk Development Team has announced the release of Asterisk 12.8.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 12.8.0 resolves several issues reported by the
community and would have not been possible
The Asterisk Development Team has announced the release of Asterisk 13.1.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.1.0 resolves several issues reported by the
community and would have not been possible
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.
On Mon, Dec 15, 2014 at 2:08 PM, Dan Cropp d...@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
Hi George,
Thank you for looking into this.
This is behind a 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]
On Mon, Dec 15, 2014 at 3:33 PM, Dan Cropp d...@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 =
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:
On Mon, Dec 15, 2014 at 3:54 PM, Dan Cropp d...@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
Yes, outbound calls are the only ones I’m trying.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@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:
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://10.10.10.10/24external_media_address=your
Thanks George.
I will remote into and give this a try.
Have a great evening!
Dan
On Dec 15, 2014, at 7:27 PM, George Joseph
george.jos...@fairview5.commailto:george.jos...@fairview5.com wrote:
Ok Dan, try this... I was able to get this to work behind a NAT and with ip
address
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
On Mon, Dec 15, 2014 at 7:34 PM, Dan Cropp d...@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.
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
On Mon, Dec 15, 2014 at 9:48 PM, Dan Cropp d...@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
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