On Fri, Mar 2, 2012 at 1:52 PM, Михаил Родионов ma...@siplabs.ru wrote:
Updated (sipXecs (4.4.0- 2012-02-08EST08:58:30 ip-10-72-10-163)). Checked.
Now I confirm it fails when display name is in UTF8. Shall I fire an issue?
(I already owe you one :))
Yes, please open a JIRA
Thanks
George
Hi All,
After recently trialing an ITSP with a single DID and sipx I went ahead and
asked them to provide a SIP Trunk with a 150 DID range.
I was expecting 1 account number to register with as a gateway ITSP and that
all calls in/out of that DID range would go over the one trunk.. Instead I
On Thu, Mar 8, 2012 at 8:25 PM, Douglas Hubler dhub...@ezuce.com wrote:
On Thu, Mar 8, 2012 at 1:03 PM, Douglas Hubler dhub...@ezuce.com wrote:
Night Losers
* Option to save voicemail saving to MP3 working.
Nathaniel Watkins wrote
Douglas - can you clarify - did MP3 voicemail
On Fri, Mar 9, 2012 at 11:11 PM, Joe Micciche jmicc...@redhat.com wrote:
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On 03/09/2012 02:37 PM, Tony Graziano wrote:
i am not concerned with the endpoint since sipfoundry does not
produce an endpoint. i had this discussion with mp3licensing.com
On Tue, Mar 13, 2012 at 2:47 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
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
does it happen to be a netgear for the far end firewall (client side)?
i've also seen this not disabled properly with dd-wrt.
Mike
On Mon, Mar 12, 2012 at 9:32 PM, Roman Gelfand rgelfa...@gmail.com wrote:
I am at a loss, I thought I had mentioned in my second post of this trail
that I have
snicker... sorry... :-)
find an itsp that knows how to do SIP trunking... not just SIP phones.
mike
On Tue, Mar 13, 2012 at 4:47 AM, Dan Palmer danpal...@tcs.act.edu.auwrote:
Hi All,
After recently trialing an ITSP with a single DID and sipx I went ahead
and asked them to provide a SIP
The far end firewall is sonicwall. Also, I have previously setup sipx
behind NAT firewall and far end sonicwall clients worked just fine.
Thanks for your help.
On Tue, Mar 13, 2012 at 6:45 AM, Michael Picher mpic...@ezuce.com wrote:
does it happen to be a netgear for the far end firewall
Yea, sonicwalls usually work fine... out of ideas.
On Tue, Mar 13, 2012 at 8:37 AM, Roman Gelfand rgelfa...@gmail.com wrote:
The far end firewall is sonicwall. Also, I have previously setup sipx
behind NAT firewall and far end sonicwall clients worked just fine.
Thanks for your help.
On
Please see:
http://track.sipfoundry.org/browse/XX-8645
Look at #'s 7-9. Open a new JIRA and re-assert those would be a good idea.
On Tue, Mar 13, 2012 at 8:14 AM, Michael Picher mpic...@ezuce.com wrote:
track.sipfoundry.org
On Tue, Mar 13, 2012 at 8:08 AM, Becker, Jesse
Create issue
http://track.sipfoundry.org/browse/XX-10051http://track.sipfoundry.org/browse/XX-10051?focusedCommentId=56568#action_56568.
Tony, I re-asserted 7-9 in comments as I didn't notice your email until
after submitting.
Jes
On Tue, Mar 13, 2012 at 9:16 AM, Tony Graziano
I have been noticing an issue where shared lines are used. Randomly a phone
or two (Polycom 650) will instantly disconnect when you press a line.
Example: Press the first line key will give about a half second of dial
tone and then will disconnect. At first I thought it was a lost
registration,
I believe one of the products by PlantCML uses sipxecs for thier e911 products.
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael Picher
[mpic...@ezuce.com]
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@Mark,
Good post
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Found the issue with media anchoring, and
On Tue, Mar 13, 2012 at 2:04 AM, Emilio Panighetti emilio...@me.com wrote:
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I read through many of the threads and JIRAs about fax, but this
wasn't clear. In sipXecs 4.4, is fax via PCMU supposed to work? or is
it designed to work with only t.38?
I tested a fax: the machine dialed to sipXecs via a gateway @g711U and
the
It will not work without t.38 for sure.
On Mar 13, 2012 4:37 PM, Joe Micciche jmicc...@redhat.com wrote:
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I read through many of the threads and JIRAs about fax, but this
wasn't clear. In sipXecs 4.4, is fax via PCMU supposed to work? or is
it
Joe, I think if you enable t.38 in your gateway, things will begin to work.
I have customers receiving calls via PSTN connected fax machines, PRI
gateway has T.38 enabled, they are forwarded to customer voicemail account.
I did have to go into the gateway and turn on T.38 for it to start working.
It is not designed to work with t.30, only t.38. T.38 is a requirement.
On Mar 13, 2012 4:57 PM, Todd Hodgen thod...@frontier.com wrote:
Joe, I think if you enable t.38 in your gateway, things will begin to
work.
I have customers receiving calls via PSTN connected fax machines, PRI
gateway
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Ok,
That's
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Thanks Todd. I enabled t.38 and I'm still getting a 0 byte tiff, I'll
dig into it more tomorrow.
joe
On 03/13/2012 04:57 PM, Todd Hodgen wrote:
Joe, I think if you enable t.38 in your gateway, things will begin
to work. I have customers receiving
I believe that is what I was saying to Joe, he needs to enable the T.38
capability in his gateway for his fax to work.
Tony, are you saying that a call into a gateway with t.38 enabled, and
coming from a fax that is connected to the PSTN via an analog trunk does not
work?
From:
On Tue, Mar 13, 2012 at 5:17 AM, Emilio Panighetti emilio...@me.com wrote:
That's interesting. So if I read you correctly; sipXecs is a
stateless proxy only? Similar in objective to SER and its
derivative projects?
sipXecs includes a stateless proxy called sipXproxy (like SER) and
would be the
Is the gateway a PRI gateway? Which one is it? If it is a trunk provider,
they must support re-invite for t.38 from, uLaw.
If you are getting a 0 byte tiff, you may not have the latest code (pdf is
default in the latest update, I think). Besides that, 0 byte means there
was a communications
I would be interested in why the SBC tries to onvolve itself in an entirely
internal call. If you have an SBC, the question becomes what ytou are using
it for...
trunking
remote users
This is the most typical deployment, where the SBC passes the registrations
through to sipx and sipx is no aware
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Tony
Thanks
With other SBC's, it holds the refer locally and negotiates between the two
(UA and ITSP), and thats what the Acme needs to do in this case. The UA
(phone) knows how to handle REFER, and if sipxbridge is handling the
trunking or remote users, it also holds the refer and doesn't transmit it
to the
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