The phone dial plan is almost ok. I've watched the SIP logs and found
out that the INVITE reachs the server but it gets NOT FOUND.
But I still can't find out why does it happen..
On Tue, 13 Sep 2011 10:14:31 -0400, Tony Graziano
wrote:
> look at the proxy and see if the call is hitting sip
Thank you for your answer Mike. But it doesn't seem to work with
linksys spa9xx phones... I've already tried everything ;[
When I dial *78+HGext or *78+ext I get just "Call ended".
Regards.
On Tue, 13 Sep 2011 09:43:18 -0400, Michael Picher
wrote:
> sort of...
>
> you can pickup a call th
I didn't understand exactly what you mean. But does sipX provide
something like "PickUP Group"? It's much more comfortable to use group
call interception (I mean press only ** for example) instead of direct
call interception.
On Tue, 13 Sep 2011 06:18:28 -0400, Tony Graziano
wrote:
> Ca
Thank you for the answer, but it doesn't work ;[
On Tue, 13 Sep 2011 11:19:53 +0200, pscheep...@epo.org wrote:
> *78
>
> See
>
> http://forum.sipfoundry.org/index.php?t=msg&th=12238&goto=42374&S=58bc77940ab3597c4c454ca47645680d#msg_42374
> [1]
>
> wrote on 13-09-2011 11:04:19:
>
> > Salu
Salut!
I've got a question about call interception in sipX 4.2. I need to
build some functionality using my sipX 4.2 server, and now I need
something like this:
For example I've got 2 extensions ( and ). Both of them are
included in a group called users12. Now, the phone is ri
> What model phone?
Linksys SPA922/21/42/41
> Do your phones have telnet or ssh access?
Nope.
> Would it be possible to
> have an end user go through the phone menu and enable web access?
I have about 4 phones on my table - so this is me who is the end user
:)
On Tue, 26 Jul 2011 10:17:35 +00
Thank you for a good answer, Tony.
But the point is that everything was ok about one week ago. I could
change whatever I want and load the configs via tftp to my SPA phones.
No problems!
And about 2-3 days ago I found out that when I change something and
press "Make config and Reset phones
TT
It starts working after recreating. But it does not pick up new
profile, it works with the old one.
The problem is I can't provision any phone in my system since last
several days. And a big trouble is that about 20 phones are in another
city, and web access of these phones is chekcked
No, I haven't. But I'll try tomorrow and post here the result.
On Mon, 25 Jul 2011 10:15:41 -0400, Tony Graziano
wrote:
> Have you considered deleting a phone in sipx that is having the
> issue,
> then recreating it to see if it does start working again?
>
> On Mon, Jul 25, 2011 at 9:47 AM,
1 more interesting fact:
If I take a new phone, make factory reset, create config via sipX
web-GUI for this new phone and fix it's profile rule - everything works.
So new phones are able to pick up profiles, but existing phones are not
able to change their profiles.
On Mon, 25 Jul 2011 08
Already checked. Running.
On Mon, 25 Jul 2011 08:54:36 -0400, Tony Graziano
wrote:
> ensure sipxprovisioning is running
>
> check from UI and then also from cli
>
> sipxproc --state
>
> On Mon, Jul 25, 2011 at 8:38 AM, wrote:
> Omg... Here is another phone provision problem. When I change
Omg... Here is another phone provision problem. When I change phone
config, and press Reload config and reboot the phone nothing happens..
3-4 days ago everything worked fine and there was no problem when
changing for example Background Text. How can I find out where the
problem is?
__
Hello Tony.
I always do factory reset before starting configuration. The phones
were configured via sipX config-server. And configs are still managed by
sipX.
> How are the phones powered?
Hm.. With a 5V adapter...
On Mon, 25 Jul 2011 06:29:01 -0400, Tony Graziano
wrote:
> Only three thi
Configurations files are ok, firmware files are ok, too. Nothing is
missing.
Hm.. I don't want to turn off provisioning, but the only way to solve
the problem is turning it off I guess. ;[
On Mon, 25 Jul 2011 10:45:36 +0200, "Henry Dogger"
wrote:
> Hm, I configure my Linksys phones with
> You could also try 0, which will disable this setting
After disabling Provisioning by trying "0" I won't be able to control
those phones (web-GUI is off).
> Are you on the latest firmware?
The firmware is the latest (6-1-5a).
> Are there any conditions on which the phone is rebooting or i
k fine, provisioning is ok,
>>> registration, calls - everythings seems to be fine. But! The only
>>> problem is Linksys reboot... I still can't find out where the
>>> reason
>>> of
>>> the problem can be hidden, but the
a922/942 can
>> randomly reboot anytime! Even during an active call! An interesting
>> fact
>> is that spa921/922 works fine, but 922/942 reboot very often.. it
>> seems
>> to be a provisioning issue, but I still can't find out what's going
>
. it seems
to be a provisioning issue, but I still can't find out what's going
wrong with it.
Thank you.
Regards, Gor.
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Are you sure that ITSP has port 5080 open and ready to accept your
REGISTER? You need to use 5080 to get the calls to SBC, but they don't.
They can use 5060 to get your SIP, but send to 5080.
Just post your sip-trace file here.
On Thu, 21 Jul 2011 23:48:18 +0100, William Salt
wrote:
> T
Hello Jeff.
Does this outside call pass sipX SBC? IF not, the problem is clear - it
seems to be REFER SIP message, but you need RE-INVITE I think. Try to
use port 5080 for outside calls and everyhing will be ok.
On Thu, 21 Jul 2011 17:06:03 -0400, Jeff Gilmore
wrote:
> I have an autoatte
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
from PSTN he can
hear music until I answer. Can I provide something like this for local
calls?
Thank you.
Regards, gor
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Thank you for the answer Michael, it's a pity that there is no way to
do it now.
On Wed, 13 Jul 2011 09:24:59 -0400, Michael Picher
wrote:
> No way to do this at present... only other way is through API.
>
> You'd REALLY need to have conf admin PINs if you wanted to do this
> (which we als
Is there any way to invite other users to my conference via my
IP-phone?
I mean that as I understand there are two ways of inviting another
person to the conference:
1) Via User WEB GUI (type a number and press "Invite user").
2) Ask another user to dial conference number to log in.
I thin
That' what I'm doing right now :)
And there are some differences..
The Siemens registering line doesn't have "rinstance="
But what is it (rinstance=)?
On Wed, 13 Jul 2011 06:51:32 -0400, Tony Graziano
wrote:
> view the registrations in sipx and compare the differences between
> the two.
>
m -> Internet calling?
>>
>> On Tue, 12 Jul 2011 11:33:53 -0400, Tony Graziano
>> wrote:
> >> His user needs to be remote for sipx to anchor the media. He can
> put
>>> sipx on its own network and connect to the ua thru a firewall and
> not
>>
I've found out a very interesting fact!
If I place a softphone (x-lite for example) under the same Router
everything works fine! SIP is ok, RTP is ok, too.
But Gigaset 470 still has no media during inbound calls. Does it mean
that the NAT router and my SIP server work correct and the only pr
and connect to the ua thru a firewall and not
> list that subnet as local.
> On Jul 12, 2011 11:21 AM, "Douglas Hubler" wrote:
>> I think gor actually wants to relay to media thru sipxecs to
> "control" it.
> >
>> gor, what end user feature are you lo
see all the RTP-streams of my users
via tcpdump..
Thank you.
On Tue, 12 Jul 2011 11:21:43 -0400, Douglas Hubler
wrote:
> I think gor actually wants to relay to media thru sipxecs to
> "control" it.
>
> gor, what end user feature are you looking to achieve?
>
> On T
Hello,
Are there any ways to make media go through my sipX server? If I'm
using a SIP-trunk all I need to do is to enable sipXbridge during trunk
creating, but what can I do if I need to control local calls (calls
between local users)?
Thank you.
___
Thank you for your replay Tony. But how can I use SBC with a single
phone? I thought that SBC can be "fixed" onlu using a trunk?
On Tue, 12 Jul 2011 04:09:00 -0400, Tony Graziano
wrote:
> because sip provider has a way handle your device randomizing portal.
> that's what a sbc does.. that d
I tried everything i could, and it still doesn't work.
An interesting thing is that there are 2 more connections via SIP
between this phone and other SIP-providers, and everything is fine!
>>
>> On Fri, Jul 8, 2011 at 4:48 AM, wrote:
>> Hey there!
>> This is my first post in sipx-users mai
I'm new to VoIP. This sipX server is my first VoIP system. But I've
read a lot about the importance of well-fixed DNS when using asterisk
and sipX.
On Fri, 8 Jul 2011 07:13:11 -0400, Michael Picher
wrote:
> have you been using asterisk and maybe is this your first sipXecs
> system?
>
> On
I don't use domain names in my VoIP system. Only IP's..
That's why I don't think that the reason of the problem is DNS, maybe
I'm wrong?
Regards, kga.
On Fri, 8 Jul 2011 07:02:03 -0400, Michael Picher
wrote:
> Have you done your DNS tests remotely (in the same manner a phone
> might)?
>
>
Hey there!
This is my first post in sipx-users mailing list, so I'm sorry if I do
something wrong.
Well.. the problem is strange enough, and I can't find out how to fix
it during 2 days, hm..
So, i've got a sipX station, and it has public IP. So "Server behind
NAT" is unchecked.
There ar
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