Thanks Tony! I will open a ticket with the ITSP. When we signed up with
Voxitas, they said T.38 was supported... Voxitas was previously Netlogic, hence
the netlogic switch.
However, Voxitas was acquired by Appia Communications a few months ago.
(appiaservices.com). They have been doing a lot
I dunno what a netlogic switch is. It doesn't matter.
They send the invite in g711, we send a reinvite in t.38.
Look at frame 26, reinvite t.38.
Now look at frame 30. They don't accept the codec. 488 not acceptable simply
means a codec we are asking for (only t.38 in the re-invite) is not in the
Let's understand perfectly why this sort of behavior happens.
If the call is sent to sipx on port 5060, it is considered a sip call (media
relay), and so call transfers using refer are not handled by sipxbridge, and
the call drops. The call is not anchored and sipxbridge is not involved in
the cal
Matthew,
The issue as I understand it is in the way that the system handles SIP
messages. The REAL number depends on multiple things... Server speed,
network card speed and the speed at which responses start coming back to the
sipXecs server. The server will prioritize processing inbound SIP me
I think this acually sums it all up:
http://blog.myitdepartment.net/?p=127
On Wed, Jul 20, 2011 at 3:50 PM, Tony Graziano wrote:
> Pointedly, you need to ensure that sipxroxd package is not installed (this
> is a default in pfsense 1.2.3, if so simply remove the package).
>
> Secondly, "most" i
Hello Will.
When I had the same problem (ITSP was sending SIP messages on 5060, not
5080) I asked my ITSP to send SIP on 5080 and everything was ok.
But you must be sure about the reason off call failing. But it really
seems to be so, because when ITSP sends SIP on 5060, media can't go via
Pointedly, you need to ensure that sipxroxd package is not installed (this
is a default in pfsense 1.2.3, if so simply remove the package).
Secondly, "most" itsp's will allow you to register on a port other than
5060. The default for a lot of itsp's is to send you a call on port 5080 if
you regist
Hi William,
I also had this problem with aql and would love to see it solved. I'm currently
using gradwell and it works well.
Adam
Sent from my BlackBerry® smartphone from du
-Original Message-
From: William Salt
Sender: sipx-users-boun...@list.sipfoundry.org
Date: Wed, 20 Jul 2011 20:
I have now set symmetric nat internally and externally for ports 5060-5061
5080-5081 and 3-31000.
I still get the same problem of the inbound call failing on pickup.
I have tcp dumped the ITSP registration process and i see the pbx asking for
a reply on 5080, but instead i see the ITSP (aql.co
not that I know of. usually for sipx that done with a vlan and wish doe that
vlan.
On Jul 20, 2011 1:47 PM, "McIlvin, Don"
wrote:
> Is there a way on SipX to strictly define the UDP port range that is
> used for RTP packets coming off the SipX server, such as for; MOH,
> Voicemail audio, auto-atte
Is there a way on SipX to strictly define the UDP port range that is
used for RTP packets coming off the SipX server, such as for; MOH,
Voicemail audio, auto-attendant audio, or conferencing audio etc.?
I am looking to use the UDP port range detail for an ingress ACL on the
Servers own data swi
make sure manual outbound nat is checked. make sure static port nat is used
on all nat assignments for sipx. make sure sipxroxd is u installed in
pfsense.
On Jul 20, 2011 12:56 PM, "Tony Graziano"
wrote:
> and you nature outbound is not right. refer to my blog.
> On Jul 20, 2011 12:55 PM, "Tony Gr
and you nature outbound is not right. refer to my blog.
On Jul 20, 2011 12:55 PM, "Tony Graziano"
wrote:
> destroy your state table and restart siptrunking.
> On Jul 20, 2011 12:18 PM, "William Salt" wrote:
>> I was under the impression i could use nat to NAT port 5080 externally to
>> port 5060
destroy your state table and restart siptrunking.
On Jul 20, 2011 12:18 PM, "William Salt" wrote:
> I was under the impression i could use nat to NAT port 5080 externally to
> port 5060 internally to the SBC?
> I have just changed the ITSP to register on port 5080, and removed
> my natting on my p
Your registration will go out over Port 5060, and it requests it be returned
on 5080. If your request comes back on 5060, it will go to the proxy, and
not to the sipxbridge. Ensure it returns back to you on 5080.
On the Wiki, get familiar with sipviewer, and the use of the merge-logs, and
si
I was under the impression i could use nat to NAT port 5080 externally to
port 5060 internally to the SBC?
I have just changed the ITSP to register on port 5080, and removed
my natting on my pfsense box, so it is natting those ports to eachother
internally and externally...
It wont register, so i w
I can confirm that I have 12 users in a huntgroup that all receive a call
when someone rings a doorphone. It has worked flawlessly for over a year.
That doorphone is used at least multiple times each day. I would say Mike's
answer is spot on.
From: sipx-users-boun...@list.sipfoundry.org
[mail
the itsp won;t work unless you can register to them on port 5080 and they
send you calls on port 5080.
your firewall is not configured properly either and is not using symmetric
nat. what firewall is it?
consider another firewall (perhaps pfsense)
http://blog.myitdepartment.net/?p=52
On Wed,
Hi Tony,
Thanks for the help.
1. Im registering a static IP to the ITSP (our external one)
2. they are sending calls to 5060, and im natting them to 6080 internally
3. Im not sure looks like from a tcpdump i am registering on UDP port
5060 to them, and 62941 locally.
>Fr
I don't know about improving it unless someone creates a JIRA and such. IMO
large hunt groups really ought to use an acd (OpenACD is being fervently
worked on).
201-212 in one ring instance, then if no answer 2013-2024 in a second, etc.
is how I would accomplish it now using hunt group.
On Wed, J
On the 12 for option 2 method, are there any plans to "improve" that? I
frequently have to do 15 to 18, and I think I may run into some occasional
gremlins.
-Original Message-
From: Michael Picher
Sender: sipx-users-boun...@list.sipfoundry.org
Date: Wed, 20 Jul 2011 08:42:05
To: Discus
>Did you expect calls w/mybuddy to go to voicemail when in DND? Can
>you explain terms "I can't use" and "this is not working" in more
>detail? Are you getting errors? silent ignores to your request...
Normally I use mybuddy with function like 'listen', 'pickup', 'call xxx
from work' and so on.
Thanks!
On Jul 20, 2011 9:56 AM, "Tony Graziano"
wrote:
> http://wiki.sipfoundry.org/display/sipXecs/Hunt+Groups
>
> On Wed, Jul 20, 2011 at 9:06 AM, Tony Graziano <
tgrazi...@myitdepartment.net
>> wrote:
>
>>
>>
>> On Wed, Jul 20, 2011 at 8:42 AM, Michael Picher
wrote:
>>
>>> 6 to 8 for option 1
http://wiki.sipfoundry.org/display/sipXecs/Hunt+Groups
On Wed, Jul 20, 2011 at 9:06 AM, Tony Graziano wrote:
>
>
> On Wed, Jul 20, 2011 at 8:42 AM, Michael Picher wrote:
>
>> 6 to 8 for option 1 and 12 for option 2.
>>
>
> I would ask if the "12" is "probably" a setting per ringdown or the max
Are you registering or using a static IP to the ITSP? Are they sending the
calls to port 5080 or 5060? If you register does the provider allow you to
confirm ip and port they see you registering on?
It would appear (and you can confirm) that the call is coming in to port
5060 from the ITSP in the
Hi All,
I am new to SipXecs, but I have worked with a couple of voip pbx's
before. Ive got a few days to set up a PBX in our office, and so far i am
struggling to find a solution to the problem i am experiencing. So id
really appreciate it if someone would reach out and help me.
I have a
12 per ringdown... You can have many in a hunt group... You just can't
ring them all at the same time.
Bryan, check the wiki :-)
On Jul 20, 2011 9:07 AM, "Tony Graziano"
wrote:
> On Wed, Jul 20, 2011 at 8:42 AM, Michael Picher wrote:
>
>> 6 to 8 for option 1 and 12 for option 2.
>>
>
> I would
On Mon, Jul 18, 2011 at 12:44 PM, Henry Dogger wrote:
> When my status is set to DoNotDisturb (I have checked the setting, send to
> voicemail on DND) I can’t use the function pickup or listen on the mybuddy.
> In fact every function concerning the MyBuddy calling my phone is not
> possible, I wou
On Wed, Jul 20, 2011 at 8:42 AM, Michael Picher wrote:
> 6 to 8 for option 1 and 12 for option 2.
>
I would ask if the "12" is "probably" a setting per ringdown or the max
number of ua's in a HG?
call 200, 201, 202 first for xx seconds, if no answer ring 203-14 for xx
seconds. Are you saying 12
Is that published anywhere
Sent from my iPhone
On Jul 20, 2011, at 7:42 AM, "Michael Picher" wrote:
> 6 to 8 for option 1 and 12 for option 2.
>
> On Jul 20, 2011 3:57 AM, "John Pi" wrote:
> >
> > Content-Type: text/plain;
> > charset="utf-8"
> > Content-Transfer-Encoding: 8bit
> > Organizat
6 to 8 for option 1 and 12 for option 2.
On Jul 20, 2011 3:57 AM, "John Pi" wrote:
>
> Content-Type: text/plain;
> charset="utf-8"
> Content-Transfer-Encoding: 8bit
> Organization: SipXecs Forum
> X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <61535>
> Message-ID:
>
>
>
> What is the better way to
it will depend on how many phones you have in the huntgroup. I would not use
a common line for huntgrohp purposes on more than 5 phones. I typically
create huntgrouplines unique and individual for each phone, disable
voicemail for these, and make sure they are noticed the first line on any
phone.
none exists.
On Jul 20, 2011 3:44 AM, "John Pi" wrote:
>
> Content-Type: text/plain;
> charset="utf-8"
> Content-Transfer-Encoding: 8bit
> Organization: SipXecs Forum
> In-Reply-To:
> X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <61534>
> Message-ID:
>
>
>
> Thank you all for your responses!
>
>
Anyone?
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Henry
Dogger
Sent: maandag 18 juli 2011 18:45
To: Discussion list for users of sipXecs software
Subject: [sipx-users] IM - MyBuddy
Hi all,
I noticed two strange issues/choic
Content-Type: text/plain;
charset="utf-8"
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <61535>
Message-ID:
What is the better way to configure a group of phones to
ring simultaneously?
1. Create a common sip user and insert a line
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charset="utf-8"
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
In-Reply-To:
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <61534>
Message-ID:
Thank you all for your responses!
Can you suggest me an analog media gateway that supports
this feature (a SIP
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