Thanks for your reply. this case happens sometimes,may be one twentieth.
yes，the ztdummy is using the RTC, and i make sure the USE_RTC had defined in
ztdummy.c.
and i will use the method in patch 13930.
Thanks.
Best wishs to you!
jordan
2010/7/12 Shaun Ruffell
> On 7/11/10 9:47 PM, jordan
Hi All,
Can anyone clerify that HUD3 is a fonality product, tied into the
various trixbox systems? Is there a HUD3 client/server standalone
project that can be installed and used with other Asterisk projects?
Any comments on using the hudlite client/server package that came out
a few years ago?
I think you need to ask your SIP provider about Redirecting Header, ask what
they support and how-to.
I work more with Cisco CallManager and SIP Rediversion Header is new in
CallManager 8x. Not sure about Asterisk. We have this same problem with Cisco
Mobility/Single Number Reach, providers usu
At 08:52 AM 7/12/2010, you wrote:
>All the Aastra equipment I have so far all has a 00:08:5d prefix.
As do my 3 Aastra phones.
Ira
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Jo
Hi Bruce,
On Sat, Jul 10, 2010 at 2:17 PM, bruce bruce wrote:
>
> I have my html/php file set so that the input field only takes 3 digit 3
> digit 4 digit (NPA, NXX, Block) so your purposal of: *'201,0); drop
> database YOUR_DATABASE'; *would fail due to big length and also I tested
> with input
Hello all,
I have a project which requires me to rout calls from ten blocks of
sequential numbers i.e. 02081000100 - 02081000200 (each block - 100 numbers)
coming in from a telco gateway via Dahdi-SS7 to 10 specific numbers outside
the box through two to three SIP trunks (trunk 2 and 3 will be spa
Sanjay,
http://www.callingcircles.com
Check out the Browser Screen Pop feature.
Regards,
Elliot
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai
Sent: Monday, July 12, 2010 3:59 PM
To: Asterisk Users Mailing List - Non-Com
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan
Zakaria
Sent: Monday, July 12, 2010 3:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Anybody with experience with Acu
I am using zaptel. And this is the issue that I can't let zaptel see it. But
I was able to see it few days ago and it was all related to the settings on
aculab, which are very complex.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-07-12 3:54 PM, "Danny Nicholas" wrote:
-
Hi,
I am looking for a Windows Desktop based application which will open a web
browser with the below url upon CALL RING on the softphone.
*http://192.168.1.4:3100/popup.php?did=DNID* (where DNID is the called DID
number)
Let me know for any help!!
Thank you.
Best regards,
Sanjay
--
_
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan
Zakaria
Sent: Monday, July 12, 2010 2:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Anybody with experience with Acu
I didn't get any reply to this email, so sending it once again. I have read
that Aculab supports commercial version of Asterisk. But isn't E1 ISDN just
and E1 ISDN, and if it is configured on Aculab, should it not sync with
Asterisk configured as E1 ISDN. In my current setup, Asterisk is ISDN E1 PR
Manmohan wrote:
> Unfortunately m not able to get rid of the below mentioned errors. not sure
> on where i am missing now.
> On Sat, Jul 10, 2010 at 9:41 AM, Manmohan Singh Jandu
> wrote:
> Ahh here is the catch i was still using app_cbmysql for this.
> now i had removed and just fol
More googling got me this page - http://www.freepbx.org/v2/wiki/DevicesTakeTwo
Very useful
Thanks
On 12 July 2010 16:41, Frank Church wrote:
> Is there a database of MAC address prefixes used the common VoIP
> devices. I see the Linksys Sipura devices state with 00:0E.
>
> Does the same apply t
Thanks for that Tim. *Wondering how I can trigger that reload?* I have tried
dialplan reload and reload but that doesn't work. Obviously amportal reload
wouldn't be doable in this case even if it works because the system will go
down.
Thanks,
Bruce
On Mon, Jul 12, 2010 at 2:13 PM, Tim Nelson wro
Well, these are horn speakers with 30 Watt which will receive 10 Watt only
from Amplifer. I am not connecting them to ceiling so maybe 10 feet off the
ground. I guess my coverage would be better???
Based on your calculations for for 40k sqfeet that would be 33 speakers. I
think that's way too much
On Mon, 2010-07-12 at 10:48 -0400, unsero...@aol.com wrote:
> No, the receiving side shows name and number as it should.
> But as calling person I only see the number of the called person
> instead of name and number.
> So we seem to struggle with the same issue.
This is something that is
- "bruce bruce" wrote:
> Hi Everyone,
>
I have done some php coding to come up with my own FollowME module for FreePBX.
The need for this has some security considerations behind it.
>
This is what my code does at core:
>
$sql="REPLACE INTO findmefollow (grpnum, strategy, grptime, g
Hello Group,
I found a solution for my problem.
I use the CallerID Variable with the Orginate Action to send a value to another
System
Action: Originate
Channel: IAX2/user1:passw...@192.168.1.2/6...@default
Application: Meetme
Data: 111,q
CallerID: 111
and on the other Side:
exten => ,n,Meet
Hi Everyone,
I have done some php coding to come up with my own FollowME module for
FreePBX. The need for this has some security considerations behind it.
This is what my code does at core:
$sql="REPLACE INTO findmefollow(grpnum, strategy, grptime, grppre, grplist,
annmsg_id,postdest, dring, need
On Mon, 2010-07-12 at 10:48 -0400, unsero...@aol.com wrote:
> No, the receiving side shows name and number as it should.
> But as calling person I only see the number of the called person
> instead of name and number.
> So we seem to struggle with the same issue.
This is something that is not supp
> Is there a database of MAC address prefixes used the common VoIP
> devices. I see the Linksys Sipura devices state with 00:0E.
>
> Does the same apply to other Linksys VoIP equipment?
The Ethernet prefixes ("OUIs") are three octets long.
Linksys / Cisco has been assigned a number of OUIs,
one
The only thing I read about changes in trunk is :
/ * The sendrpid parameter has been expanded to include the options
'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
header to be sent (equivalent to setting sendrpid=yes) and setting
sendrpid to 'pai' will cause P
Hi all,
is it possible to send a Variable to another System via IAX Protocoll by using
AMI / Orginate
Like this:
Action: Originate
Channel: IAX2/user1:passw...@192.168.1.2/6...@default
Application: Meetme
Data: 111,q
Variable: var1=111
and the Remote System knows the Variable "var1" ?
In my Tes
Hi all,
I'm running into a easily replicated problem at the moment, with Asterisk
1.6.0.28 (built from source, no special configure parameters, other than a
path) running on top of a fully up-to-date CentOS 5.5, and I'm looking for
suggestions as to why this is occurring. I've spent some time
Another tool, to search by company.
http://standards.ieee.org/regauth/oui/index.shtml
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan C.
Bailey
Sent: Monday, July 12, 2010 11:53 AM
To: Asterisk Users
http://www.coffer.com/mac_find/
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Church
Sent: Monday, July 12, 2010 11:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-
Hi Frank.
The MAC got two parts:
00:00:00:11:11:11
The 00:00:00 part identify the vendor, and the last part is unique
inside that Vendor ID.
Linksys got more than one "code", so it may be different bettew Linksys
devices.
You can check them here, for example:
http://www.coffer.com/mac_find/?
On Mon, Jul 12, 2010 at 04:41:16PM +0100, Frank Church wrote:
> Is there a database of MAC address prefixes used the common VoIP
> devices. I see the Linksys Sipura devices state with 00:0E.
See http://standards.ieee.org/regauth/oui/oui.txt
--
Barry
--
_
All the Aastra equipment I have so far all has a 00:08:5d prefix.
-Jon
- Original Message -
From: "Frank Church"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Monday, July 12, 2010 10:41:16 AM GMT -06:00 US/Canada Central
Subject: [asterisk-users] MAC Address pref
The first 6 digits of a mac address are the vendor ID and the 2nd 6 are the
unique device ID. Some vendors use more than 6 digits of device IDs, so
they have multiple vendor IDs. So 00:0E isn't Linksys, it is 00:0E:xx that
is Linksys. Some devices use CDP or LLDP to request voice vlan informatio
On 12 Jul 2010, at 16:35, Jonas Kellens wrote:
> On 07/12/2010 05:01 PM, Steve Howes wrote:
>> On 12 Jul 2010, at 14:09, Jonas Kellens wrote:
>>
>>> I want to set the SIP-header Remote-Party-ID to display the name of the
>>> calling party on my phone in stead of the number.
>>>
>> Am I missing s
Is there a database of MAC address prefixes used the common VoIP
devices. I see the Linksys Sipura devices state with 00:0E.
Does the same apply to other Linksys VoIP equipment?
Is there some way VoIP equipment allow themselves to be identified by
requesting data from some ports?
--
___
Hi guys,
I've got a question about chanspy and meetme.
I'd like to transfer all the persons involved in a chanspy (the guy
spying, the guy that is spied and the guy that is speaking to the spied
one -> total: 3) in a conference room.
Is there a way to do it quickly without especially knowing e
On 07/12/2010 05:01 PM, Steve Howes wrote:
> On 12 Jul 2010, at 14:09, Jonas Kellens wrote:
>
>> I want to set the SIP-header Remote-Party-ID to display the name of the
>> calling party on my phone in stead of the number.
>>
> Am I missing something or is this waht CALLERID(name) and sen
On 12 Jul 2010, at 14:09, Jonas Kellens wrote:
> I want to set the SIP-header Remote-Party-ID to display the name of the
> calling party on my phone in stead of the number.
Am I missing something or is this waht CALLERID(name) and sendrpid is for?..
S
--
___
No, the receiving side shows name and number as it should.
But as calling person I only see the number of the called person instead of
name and number.
So we seem to struggle with the same issue.
-Original Message-
From: Jonas Kellens
To: Asterisk Users Mailing List - Non-Commercia
In my case, it shows the name "eric" and number 20 on the receiving
phone. As if the From-header is overwritten...
That's off course not what I'm trying to accomplish. Therefore I can use
the P-Asserted-Identity (which works well if I may add).
Jonas.
On 07/12/2010 04:05 PM, unsero...@aol.
It is not a very straight forward procedure. First of all you need to decide
how you would make the two PBXs communicate with each other. I am sure the
old PBX doesn't support VoIP ptotocols. Can you use T1 or E1 to make them
communicate with Asterisk? Once the calls are successfully sent from the
Hello,
the SIP header now should be sent. What the remote device
is doing with this header, or whether the syntax of the
header is as the remote device expects it, is another
question.
You can check with
sip set debug on
whether the header is now sent as you expect!
If it does, I cannot tell you
It doesn't work for me too..
exten => 1400,1,SIPAddHeader(Remote-Party-ID: "Test"
\;party=called)
exten => 1400,n,Dial(SIP/${EXTEN},15)
leads to
-- Executing [1...@default:1] SIPAddHeader("SIP/1401-0159",
"Remote-Party-ID: "Test" \;party=called") in new
stack
-- Executing [1...@d
Roger,
your answer did resolve something :
/[Jul 12 15:51:24] -- Executing [...@from-test:2]
SIPAddHeader("SIP/test6-009a", "Remote-Party-ID: "eric"
;party=called ") in new stack/
However this SIP-header is never send as a SIP-message to the phone from
where I'm placing the call. Th
On 14/06/10 18:11, Gordon Henderson wrote:
> On Mon, 14 Jun 2010, Chris Bagnall wrote:
>
>> Actually, the Atom seems to be surprisingly powerful. We have a couple of
>> Atom boxes with transcoding and conferences enabled without issue. I
>> wouldn't pretend it'll cope with hundreds of conference pa
-Original Message-
From: Jonas Kellens
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Mon, Jul 12, 2010 3:09 pm
Subject: [asterisk-users] Remote-Party-ID party=called
Hello list,
using Asterisk 1.4.30.
I want to set the SIP-header Remote-Party-ID to display t
Hello,
escape the semicolons with a backslash! At least in astersik-1.6.X
this works fine.
I.e. replace in the SIP-Header-command all ; by \;
Regards,
Roger.
Jonas Kellens schrieb:
> Hello list,
>
> using Asterisk 1.4.30.
>
> I want to set the SIP-header Remote-Party-ID to display the name o
Hello list,
using Asterisk 1.4.30.
I want to set the SIP-header Remote-Party-ID to display the name of the
calling party on my phone in stead of the number.
This is the dialplan :
exten => 10,1,NoOp()
exten => 10,n,SIPAddHeader(Remote-Party-ID: "eric"
;party=called )
exten => 10,n,Dial(SIP
That sounds like buggy software on the Cisco. My guess is that the issue is
that you are getting a red alarm BECAUSE the Cisco crashed, not that the red
alarm is causing the Cisco to crash. I've never had that happen in 10 years
of using Cisco gateways.
-Original Message-
From: asterisk-
Hi list,
I'm having trouble with DTFM tones detection. Usually, some tones are
being received duplicated in Asterisk, some not. As you can imagine,
that's a very big problem involving IVR menu options, Meetme conferences
protected with passwords, and so on.
We are currently using DAHDi 2.2.0.2,
Hi there,
Thank you for your response. So I can use the ModemSetOriginCmd command to
assign the outbound number on the iaxmodem, but how do I choose which modem
to use for my specific sip client (GXW-4004), as I have 2 faxes connected to
my GXW box?
Thank you very much.
Regards,
- Marta
On
Title: DAVID MATÍAS HERNÁNDEZ
Hi list,
I'm having trouble with DTFM tones detection. Usually, some tones are
being received duplicated in Asterisk, some not. As you can imagine,
that's a very big problem involving IVR menu options, Meetme
conferences protected with passwords, and so on.
We ar
Hi Peder,
thanks for the advice, I'll send it to the technician managing the Cisco
device. Moreover he told me that when I get the red alarm the Cisco 2800
crashes.
Did it happened to you, too?
Thank you
Giorgio
Peder wrote:
> If you do back to back, then one end needs to clock. To set it on
Hello, Asterisk party,
If block the call before dialing (Hangup()), CDR's don't write to
MySQL or CSV. Otherwise, if Hangup() is after Dial(), CDRs write
normally.
Here is the dialplan:
; we skipped dial, because the number is "blocked"
exten => _X.,n(Finish),Hangup()
exten => h,1,NoOP("hangup")
e
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