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Thanks everyone for replying. I'm going to investigate the
Polycom phones
I am at a loss, I thought I had mentioned in my second post of this trail
that I have done this.
Anyway, are there any other ideas? and yes, sip-helpers and alg are
disabled. Just to test that it is not firewall, I opened up all ports in
both directions.
Thanks
On Mon, Mar 12, 2012 at 8:39 PM
I think a lot of work is needed in this area. Currently the ivr does not
indicate you have a FAX, if the user portal stores them, it would also need
to indicate that.
There is also talk of being able to send faxes from the user portal.
I don't think, as a feature, you can address the user portal
I was just showing him where the Nat traversal (media relay) was...
On Mar 12, 2012 8:25 PM, "Michael Picher" wrote:
> I know, but the pic showed it clearly checked.
> On Mar 12, 2012 7:51 PM, "Tony Graziano"
> wrote:
>
>>
>>
>> On Mon, Mar 12, 2012 at 7:47 PM, Michael Picher wrote:
>>
>>> if yo
This may be the incorrect place for a feature suggestion...
I was wondering what others thought regarding the concept of having faxes
display in the user portal instead of transmitted via email. This way
confidential faxes have to be securely downloaded instead of being
transmitted unencrypted via
I know, but the pic showed it clearly checked.
On Mar 12, 2012 7:51 PM, "Tony Graziano"
wrote:
>
>
> On Mon, Mar 12, 2012 at 7:47 PM, Michael Picher wrote:
>
>> if your server is not behind nat, uncheck the 'server behind nat'
>
> (this is what I sayeth to him already)
>
>>
>>
>> On Mon, Mar 12,
On Mon, Mar 12, 2012 at 7:47 PM, Michael Picher wrote:
> if your server is not behind nat, uncheck the 'server behind nat'
(this is what I sayeth to him already)
>
>
> On Mon, Mar 12, 2012 at 7:38 PM, Tony Graziano <
> tgrazi...@myitdepartment.net> wrote:
>
>>
>>
>> On Mon, Mar 12, 2012 at 7:30
if your server is not behind nat, uncheck the 'server behind nat'
On Mon, Mar 12, 2012 at 7:38 PM, Tony Graziano wrote:
>
>
> On Mon, Mar 12, 2012 at 7:30 PM, Roman Gelfand wrote:
>
>> Actually, I neglected to mention that...
>>
>> 1. Snom are the local phones. I use x-lite for the remote phon
On Mon, Mar 12, 2012 at 7:30 PM, Roman Gelfand wrote:
> Actually, I neglected to mention that...
>
> 1. Snom are the local phones. I use x-lite for the remote phone.
2. When I call from x-lite, as remote workder, and get to voicemail,
> there is sound both ways as I am able to save voicemail.
Actually, I neglected to mention that...
1. Snom are the local phones. I use x-lite for the remote phone.
2. When I call from x-lite, as remote workder, and get to voicemail,
there is sound both ways as I am able to save voicemail. However,
when a snom phone picks up, the snom phone picks up t
system, nat traversal, internet calling, nat traversal.
i would ask other if snom supports moh and/or how it does.
On Mon, Mar 12, 2012 at 5:21 PM, Roman Gelfand wrote:
> I am not sure I understand. Does this mean, assuming everything is
> correctly configured, that the only problem I could h
I am not sure I understand. Does this mean, assuming everything is
correctly configured, that the only problem I could have is due to the
phone?
When you say, "should be set to use media relay (remote users=yes)",
where is it?
Thanks
On Mon, Mar 12, 2012 at 4:58 PM, Tony Graziano
wrote:
> the
the server has a public ip address, and hence is not behind nat.
the phones might also be behind a "different" network, and if so, should be
set to use media relay (remote users=yes).
the snom phones MAY NOT support the ietf draft for moh internally, but
might for sipx bridge if it is anchoring p
The topology is as follows.
1. The sipx server has public ip address in /29 network. It is behind
transparent mode firewall. The nat traversal is checked. In the
server NAT configuration, I specified this server ip address.
2. The phones (snom 370) are on a lan. The network traffic to and
from
a call trace with a description of the call flow would do lots to explain
why moh is not working.
describe the elements involved (sbc, gateway, UA), etc.
On Mon, Mar 12, 2012 at 11:38 AM, Roman Gelfand wrote:
> What should I be looking at if MOH is not working?
>
> Thanks for the help.
>
> On M
What should I be looking at if MOH is not working?
Thanks for the help.
On Mon, Mar 12, 2012 at 10:32 AM, Tony Graziano
wrote:
> Those are static messages that will always appear in the proxy
> unfortunately. I think they can be safely ignored. I think they appear
> whether moh works or not also
Those are static messages that will always appear in the proxy
unfortunately. I think they can be safely ignored. I think they appear
whether moh works or not also.
On Mar 12, 2012 9:44 AM, "Roman Gelfand" wrote:
> I am getting consistantly the following messages in sipXproxy.log.
>
> 1. NatTrave
I am getting consistantly the following messages in sipXproxy.log.
1. NatTraversalAgen[900_ntap]::handleOutputMessage failed to to
retrieve CallTracker to handle request
2. SipXProxy:"SipUserAgent::dispatch received response without transaction"
3. SipXProxy:"HttpMessage::get[4] Receiving failed o
I think you might be way better off with a Polycom handset. Aastra has some
known and discussed limitations with interoperability with sipx.
Not upset, just think you could more plainly describe your environment at
the outset instead of pieces at a time. Good luck.
On Sun, Mar 11, 2012 at 8:30 PM
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Last minute, say AudioCodes? I thought I explained that well
enough...
P
Still you have not indicated what the UA is. If you truly want help, you
could be more descriptive in your environment/use case. The UA is a key
piece you are leaving out here.
All of what you are trying to do is wholly dependent upon:
1. UA
2. Gateway
It is really not a sipxecs specific questio
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I should have been more clear on the CAMA trunks...CAMA
trunks are interf
Either sipx receives the did call and processes the FAX OR an gxs gateway
registers to sipx configured got t.38 and receives it.
The FAX machine can also send it BUT the provider MUST support t.38.
On Mar 12, 2012 3:17 AM, "Mark A. Smith" wrote:
>
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>
Its not a question of whether sipx does this, this is done by the UA.
The gateway can receive dtmf relay and hook flash if the UA sends it.
Refer to the manual for the UA.
On Mar 11, 2012 11:30 PM, "Johnathan Scott"
wrote:
>
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Hi,
Can you provide more applicatio
Good Call!
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Mark A. Smith
Sent: Sunday, March 11, 2012 12:07 PM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Sending Hook-Flash to FXO Gateway (possi
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