Alexander,
By the way, 4.3.x - is development release.
All releases, where the second digit is odd are not stable.
Latest stable is 4.2.1. If you are not going to test the very latest features -
use 4.2.1.
Even if you are going to test the very latest features, it may worth starting
with the st
You build a custom dial plan that uses the first digits dialed as a filter,
and then strip what you don't want to go out over a specified gateway.
As an example, if you wanted all calls to a 415 area code to go over Gateway
NoCali, you would set the dial plan to look for anything that starts with
I did create dial plan. But I am getting "SIP/2.0 482 Loop Detected
with 2 hops ago"
Here is what I changed from single number ITSP trunk configuration...
1. Removed ITSP caller id info
2. Removed SBC SIP Incoming call destination
3. Added custom dial plan for the ITSP gateway where dialed nu
(System>Dialplans).
Choose the call type, add the gateway to use, and save.
(or create custom dialplans)
On Sun, Oct 10, 2010 at 7:01 PM, Tony Graziano wrote:
> You specify the gateway. The wiki has this well documented.
>
> Tony Graziano, Manager
> Telephone: 434.
You specify the gateway. The wiki has this well documented.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 434.984.8427
Helpdesk Co
I think the problem is simply connecting an attended transfer through the
media server.
There is a tracker item and it is being worked on.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and
I am using version 4.2.1 and am having issues with external call forwarding
through an auto attendant.
I am using an auto attendant whereby callers upon pressing a certain dialpad
number have the call transferred to a local external seven digit number. I am
using pots lines only with a patton
Where do you specify the line to use to call out.
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List Archive: http://list.sipfoundry.org/archive/sipx-users/
It is being rejected by the proxy. The question is why. Tail the
sipxproxy.log to make sure your alias matches the invite...
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control System
I compared the call with the field "Incoming calls destination" = user
ID (or another number in the system: AA, hunt group) and empty field
"Incoming calls destination".
In the first case sipXbridge sends Invite to SipXproxy and the call
goes.
In the second case sipXbridge don't send an invite any
Except google411 also has a sip uri to call it with...
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 434.984.8427
Helpdesk Contrac
Dial plan
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 434.984.8427
Helpdesk Contract Customers:
http://www.myitdepartment.net/ge
I am sorry, I meant where do you setup call routing based on
destination phone number.
On Sun, Oct 10, 2010 at 4:13 PM, Roman Gelfand wrote:
> Where do you configure routing based on caller id?
>
> Thanks for all the help
>
> On Sun, Oct 10, 2010 at 4:06 PM, Todd Hodgen wrote:
>> Sorry, meant ca
On Sun, 10 Oct 2010, Marcello Manzardo wrote:
> Anybody any great ideas on how to handle 411 and 555-1212
> calls besides from just blocking???
It there really a need for '411' when prevasive access to
search engines exists?
We did not block prohibited outbound numbers (out of permited
area c
Where do you configure routing based on caller id?
Thanks for all the help
On Sun, Oct 10, 2010 at 4:06 PM, Todd Hodgen wrote:
> Sorry, meant caller id, not did - should have read - With regards to caller
> ID, you can set caller ID on a per user basis, or a group basis, or ITSP
> trunk basis, y
Sorry, meant caller id, not did - should have read - With regards to caller
ID, you can set caller ID on a per user basis, or a group basis, or ITSP
trunk basis, your choice. Provided of course that the ITSP supports it.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mai
With regards to sipXecs your impression is correct. If you have an ITSP
that supports the SIP standards, you will be able to multiple DID numbers
(many many) that termintate at an auto attendant, or multiple auto
attendants, your choice, or as DID to specific stations, or specific hunt
groups, yo
Also, if you setup a trunk (gateway) to callcentric and do not use a
template you WILL have a "Route by To:" header option at the bottom to get
around the "all calls route to an account number in the INVITE...
I think I stated that in an earlier thread to you though...
First, callcentric is a BAD example for too many reasons.
ANY commercial oriented ITSP can allow you multiple DID numbers on a single
trunking account. They will also let you specify your outbound callerid at
the user level.
For example: the "templated" list of ITSP's in sipx leads you in a certa
I was under the impression that multple phone number sip trunking is
allowed with sipx. If I am not mistaken, I saw someplace that
multiple phone numbres could receive incoming phone calls. Whereas,
outgoing, you could have only one. If this is so, sipx doesn't allow
me to add another gateway fo
Ugh, callcentric...
"Callcentric's SIP config specifically states its Register Expiry should be
1800 secs. In addition, Callcentric uses its own Keep Alive scheme to keep a
device connected. I've found that it works well. "
Not my quote, but they evidently provide this information somewhere and
o
I found the following messages identify the registration cycle. It
appears that ITSP is the one that expires the registrations after
roughly a minute.
"2010-10-10T15:24:08.818000Z":258:OUTGOING:INFO:sipx.comms.com:Timer-0::sipXbridge:"Sent
SIP Message :\nRemote Host:204.11.192.36
This hasn't affected me, but this seems to list at least one that may work.
http://blog.tmcnet.com/blog/tom-keating/voip/free-411-directory-assistance.asp
-Original Message-
From: Tony Graziano
Sender: sipx-users-boun...@list.sipfoundry.org
Date: Sun, 10 Oct 2010 13:22:35
To: ;
Reply-To
It depends on what you want to do... You can create a dialplan entry to send
them somewhere of you choosing (AA, etc.).
I am not aware of any other free sip 411 service because I have never had to
look (yet).
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984
Take a look at the Expires header or expires parameter on the contact
header that is returned from the 200 OK of the REGISTER.
On Sun, Oct 10, 2010 at 12:14 PM, Tony Graziano
wrote:
> The initial part of the log, where it starts up up and registers is the
> piece that needs to be inspected.
> ===
OK. I see that. Is your user behind NAT somewhere else (remote) or local to
the sipx server? Can you test with a hardphone or something like xlite? I've
never had great success with (sipcommunicator myself). From the trace it
"appears" the user is not registered before the invite comes in, what doe
I know of a couple of people with them, I'll ask for a wiki contribution
tomorrow.
On Sun, Oct 10, 2010 at 11:35 AM, Matthew Kitchin (Public) <
mkitchin.pub...@gmail.com> wrote:
> Thanks. I made a tiny bit of progress. I got the phone converted to SIP at
> least. I'm hoping it gets most of what i
The initial part of the log, where it starts up up and registers is the
piece that needs to be inspected.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephon
Thanks. I made a tiny bit of progress. I got the phone converted to SIP at
least. I'm hoping it gets most of what it needs by DHCP. The only SIP
properties I can find are username and password. I setup a test user and put in
the correct SIP password. That is about as far as I can get. It only st
Unless I misunderstood you, I deleted sipxbridge log and restarted
sipx trunk service and still I am seeing registration every minute.
On Sun, Oct 10, 2010 at 10:42 AM, Tony Graziano
wrote:
> I think I pointed that out to you on your other post.
>
> I would get a fresh log or pcap from your fire
I think I pointed that out to you on your other post.
I would get a fresh log or pcap from your firewall for a restart of
sipxbridge.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Contr
Matthew, unsupported phones may be configured manually... no need to put
them into the list of phones.
Mike
On Thu, Oct 7, 2010 at 3:10 PM, Matthew Kitchin (public/usenet) <
mkitchin.pub...@gmail.com> wrote:
> Can anyone send me some pointers for getting a Polycom Spectralink
> 8020 with 4.2.1
On the ITSP gateway, I specified registration interval of 600 ( I
assume seconds as it states so on the page). However, the sipx is
attempting to register with ITSP every minute. Why?
Thanks in advance
___
sipx-users mailing list
sipx-users@list.sipfou
Uncheck that.
The alias (DID number)goes in the user alias field.
812xxx
Either you want ALL CALLS from the trunk to go to the destination you set in
the SBC field OR you want them to go to unique users based on the DID in the
INVITE.
812xxx
Put the DID number in the user alias field.
If you have snom 370, could you send me configuration file from it?
Thanks,.
2010/10/10 Rene Pankratz :
> We have several Snom's working with SipX (Best results with FW 7.3.14).
> René
> 2010/10/10 Roman Gelfand
>>
>> Would anyone have a snom 370 configuration that works with 4.2.1?
>> Please, s
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