is it codec issue?
On 26/10/2011 04:07, Adrien Guillon wrote:
> Hi everyone,
>
> I have been working on incoming calls from a sip trunk, and debugging
> potential issues. Right now, calls are disconnected immediately after
> I dial an extension from the AA (when I call externally). I'm pre
I know that sipXecs 4.4 support t.38 now a day,
but may i know any existing fax software require to support this ?
Or anyway to let me understand more how to configure from PRI gateway to
SipXecs after go to my fax inbox with PDF file.
1) FaxVOIP?
2) Faxboom?
Hi all of you, may i know if i have mediant 1000 and i reinstall my
server to 4.3.
This is able to send and receive fax t.38 or t.30?
Mike, is possible i just configure my mediant and point back to my
extension number that i set from sipXecs server?
now i try to install on my VM by virtualbox. h
Does Sipxecs provide t38?
I try find out others solution to doing my FAX in Audiocodes gateway
mediant 1000.
Can comfirm this NET SatisFAXtion Server can support but it's expensive.
Any Idea?
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hen?
Coz I thought sipXconfig is supposed to do that automatically. hmmm...i
don't really know how to configure the manipulation table for
audiocodes but i guess i'll just google that if it's really necessary. .
On Wed, Mar 24, 2010 at 11:43 AM, Winson
(Elabram) <wins
route based on incoming fax but it doesn't seem to be able to. You route
based on incoming DID but you can have a pbx or fax server which detects
the fax and handles it. Mike On Mon, 08 Mar 2010 15:08:36 +0800,
Winson (Elabram) wrote: > Hi Mike, any news from you regar
-Original Message-
From: Picher, Michael [mailto:mpic...@cmctechgroup.com]
Sent: Monday, July 20, 2009 5:47 AM
To: winson (Elabram); sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] how to do Fax in SipXecs
No faxing capabilities in sipXecs.
You could look to a dedicated
Hi,
I am experiencing some call transfer issues when using the E1 gateway
m1000 pass a call to SipXecs
It can auto pass to my Extension DID (1303)
Example : outside person (0127788328) call this number (170089XXX)
when my gateway receive this number (170089XXX) , 1st i will Stripped
9 Dig
I found out the G-Tek Phone setting will come out in 4.2.
but now i have some issue is because my phone keep "hang",
i check in the log file (sipXproxy) it show me about:
"2010-02-08T08:33:36.550618Z":15968:SIP:WARNING:sip.xessb.com:SipRouter-11:B6C7DB90:SipXProxy:"SipUserAgent::send
SUBSCRIB
We are using sonicwall NSA 2400, it always keep said my External port
!= internal port,
How to check my port is be symmetric? isit any log will show?
Tony Graziano wrote:
On Wed, Feb 3, 2010 at 3:18 PM, Winson
(Elabram) <winson.k...@elabram.com>
wrote:
Dear
all Expert !
Hi i
Dear all Expert !
Hi i stuck in the firewall NAT mapping for my Sipbridge - ISTP .
I try direct use the modem connected to sipXecs and open the port
5060,5080, 3-31000,16384-16482(ISTP) is wan work.
after i plug in the firewall, the softphone show me is error 408,time out.
Then i check the s
Last 2 day my colleague bring one Sonicwall back.. i know my nightmare
come true..
i base on this http://sipx-wiki.calivia.com/index.php/Firewall_Configuration
will know more about the port setting
but i still have some issue can ask for advice?
basically my network diagram is :
internet -
May i know where to check my sip account is connect from Sipxecs?
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I looking for some sip trunk provider just purely make outbound call,
Because for inbound we already have our PSTN E1 PRI line to do the
receive call.
So i check for this pennyTel actually is cheap and near by our place (i
ping just 150++ms)
1) i register one account from PennyTel,
2) create
Thank all of you to reply,
Actually i already get some Idea and change it is can work properly now,
I check for this G-tek adapter actually just only 6watt.
I have 35 same model phone under deference Vlan., (base on department)
FD = Phone - 192.20.102.* , PC - 192.20.2.*
MG = Phone - 192.20
witch are you using? Does it have
enough power for all the phones you have on it?
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Winson
(Elabram)
Sent: Wednesday, January 27, 2010 7:42 PM
To: sipXecs users
Su
i implement 35 unit G-Tek IP-PHone in my company, however that keep
"hang" when i try hunt group or call out.
for example i dial 1300 is my FD department, i try dial on and off
shortly, but some of the phone still Ring, i have to turn off the power
and turn on again.
I just try call to local,
Base on you budget.
Basically can try with G-Tek AQ102 (audiocode), Linksys and Polycom
I try this all is ok,
but the G-Tek phone some time will "hang" , have to keep restart
Abdul Mayat wrote:
Hi
All,
I would like to test some hardphones on SipX (4.0).
Unless this has already be
I wish i can come true same like you
:-)
But i still want to thank you all of you when i face trouble.
Winson
Picher, Michael wrote:
Group hug... :-)
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Jos
Dear All,
Before i don have the DNS server, i point the (system
/ configure/"Hostname" & system /Domain Name) back to
my IP address is working fine.
I already get my DNS server since 2 days ago(i using OES =
"172.20.201.33")
>From sipxecs-system-setup
1) hostname = sip.example.com
2) IP-addr
?
Winson (Elabram) wrote:
Yes correct,
I try to add 1 Unmanaged SBC example (sipXbridge-voipstunt), after my
VOIPSTUNT this gateway ROUTE to this bridge.
IS cannot link that VOIPSTUNT already.
In version 4.2 there will be a parameter for each trunk to enforce a
limit on the
:
Telephone: 434.984.8426
Fax: 434.984.8427
Helpdesk Contract Customers:
http://www.myitdepartment.net/gethelp/
- Original Message -
From: Winson (Elabram)
To: Scott Lawrence
Cc: Josh Patten ; sipx-users@list.sipfoundry.org
; Tony Graziano
Sent: Thu Jan 21 21:09:23 2010
Subject: Re: [sipx
out,
the gateway will not turn to my 2ND gateway.
Attach file is my sipXbridge-1.xml configuration.
Scott Lawrence wrote:
On Fri, 2010-01-22 at 00:30 +0800, Winson (Elabram) wrote:
I'm so sorry about that, maybe my English no good so will make you all
angry.
It'
ining how to post queries for help,
and you send the same drivel again under a different subject name? How
rude.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 1/21/2010 7:40 AM, Winson (Elabram) wrote:
I
already apply 2 sip provider, and created in the
I already apply 2 sip provider, and created in the Sipx gateway account.
1. Voipstunt
2. Voipbuster
>From the Route
i point back my default sipXbridge-1 is work but problem is once 1st
gateway engage, the server does not auto change to backup gateway that
i
set...
(because same sipXbridge)
No
Sorry about that..
I means not the SipX can run..
I really hope have a solution..
Scott Lawrence wrote:
On Thu, 2010-01-21 at 20:59 +0800, Winson (Elabram) wrote:
I already apply 2 sip provider,
I already created 2 gateway account.
1. Voipstunt
2. Voipbuster
>From the Rout
I already apply 2 sip provider,
I already created 2 gateway account.
1. Voipstunt
2. Voipbuster
>From the Route
i poit back my default sipXbridge-1 is work but problem is onece 1st
gateway engage the server does not auto change to backup gateway that i
set...
(because same sipXbridge)
now i
May i know which Sip Provider is cheap and stable?
Below the Sip is what i checking in the the market is cheaper
1) PennyTel (Aus)
2) VoipStunt (EUP)
3) Voipbuster (EUP)
4)
FreeCall (EUP)
Problem is our site in Asia, it too far for me.
Any commend for you all?
I have a new model Audiocodes 310HD IP-Phone / G-Tek AQ102 but i cant
add any new model phone in the web base.
Thank you
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Is better update to 4.04?
because i just start configure in 4.02..
Any change?
Nikolay Kondratyev wrote:
Uups…
please ignore
my previous message.
I’m
sorry.
There
was wrong HTTP_PROXY
… fat fingers..
From:
sipx-users-boun...@list.sipfoundry.org
[ma
Thanks! i get it and connected already.
Andres Jaramillo wrote:
Try with the documentation for this Gateway, http://www.audiocodes.com/filehandler.ashx?fileid=36336
2009/11/19 Winson (Elabram) <winson.k...@elabram.com>
I
know that this not a topic for SIPX
But i really n
I know that this not a topic for SIPX
But i really no idea because no body can help me in audiocodes..
Actually i just bought 1 unit Median 1k from audiocodes starting to
lunch in sipX
But unfortunately i don know how to connect to my PC
May i know how just pure connect to the Median 1000?
1)M
I just want to purely receive FAX is it SipXecs possible can Do it?
--Winson
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Because our company just start planning use all POE ip-Phone to replace
our old normal phone (but very costly)
From the SipXecs Web base systems Add phone template have this Linksys
model (means has tested by sipXecs before )
This model IP-Phone is below USD100 so is normal price for me (but no
Before that my SipXecs server ip address is 192.168.1.xxx because got
something happen i need to change the ip to 192.168.2.xxx.
May i know how to change IP-address in commend line?
--Winson
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Li
May i know hows FOIP work in SipXecs?
Or any solution can able receive and send without any Fax machine
Example:
(Receive Fax)
PSTN --> M1000(T.38)--> Email account server
(Send Fax)
(FOIP solution)-->M1000(T.38)[if i not wrong]-->PSTN
--Winson
___
s
This is what i found in SipX wiki., mike said Moh
definitely is not supported on xLite.
http://list.sipfoundry.org/archive/sipx-users/msg16033.html
May i know which free softphone can support?
--winson
Winson (Elabram) wrote:
Thanks,
But why i still cannot listen anything when i press
Thanks,
But why i still cannot listen anything when i press hold in my x-lite.
I already try to use others softphone (mizu,Bria pro) but still is the
same.
Is it have to set something in Sipxecs?
--winson
Sathya Chandrakala wrote:
Winson (Elabram) wrote:
May i know how to use
May i know how to use music on hold in soft-phone,
Actually what is the different between call park and music on hold?
--Winson
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May i know beside Audiocodes m1000 gateway, still have any gateway can
running well in SipX?
-Cisco? can show the model?
-Others?
Because i need one stand alone gateway can support virtualization like
Xenserver.
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May i know who tried to
use the samgoma card in SipX before?
i am a end use for the VOIP system,
Before that i try to use Freeswitch in our company but i hear that SipX
has more GUI interface and easy to configure.
But i already purchase Sangoma card and the normal Server.
May i know the sa
a dedicated fax device like the FaxFinder from
> Multitech.
>
> Mike
>
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of winson
> (Elabram)
> Sent: Sunday, July 19, 2009 9:14 PM
>
Hi, How to receive fax form PSTN to Mail server in SipXecs?
--Winson
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