On Mon, Jul 19, 2010 at 09:19:55PM +0200, mattias wrote:
> Ok
> How to test on the cli
> As i say
> I running elastix and yes i know there a mailing list about elastix but the
> people there only point me to the book about elastix
What version of asteris is it? What is the output of:
ls /var/l
On Mon, Jul 19, 2010 at 05:06:37PM -0400, Jose P. Espinal wrote:
> Hello list,
>
>
> I'm facing a little issue with dahdi attempting to load the OSLEC echo
> canceller into my current kernel.
>
> After compiling dahdi 2.3.0.1 with OSLEC support, I get the following
> error when set 'oslec' as
I tried other fax machine and fax succesfully received.
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On Monday, July 19, 2010 01:03:57 am Peter Childs wrote:
> One of the problems with Distinctive Ring tones is that its not
> consistent, between different phones so if you have a mix of phone
> types you have a problem.
Agreed. I only mentioned what I did since I, along with the OP use Aastra
ph
Hello list,
I'm facing a little issue with dahdi attempting to load the OSLEC echo
canceller into my current kernel.
After compiling dahdi 2.3.0.1 with OSLEC support, I get the following
error when set 'oslec' as the echocanceller:
DAHDI_ATTACH_ECHOCAN failed on channel 1: Invalid argument (2
Flash Operator Panel (2?)
Is by best guess
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Ok
How to test on the cli
As i say
I running elastix and yes i know there a mailing list about elastix but the
people there only point me to the book about elastix
And i haven't adobe acrobat and have no plan to get it on this machine
-Ursprungligt meddelande-
Från: asterisk-users-boun...
One of two things is happening (In my opinion)
1. You aren't pointing to the right place
Or
2. The language variable isn't getting set , so Asterisk "gets lost".
The CLI output with verbose 5 would tell you this.
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Thanks for the example
But still english in the godbye message
-Ursprungligt meddelande-
Från: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] För Danny Nicholas
Skickat: den 19 juli 2010 20:24
Till: 'Asterisk Users Mailing List - Non-Commercial Dis
Trying my best here, don't want to start another TOP/Bottom flame war
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On 07/19/2010 01:23 PM, Danny Nicholas wrote:
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mattias
>> Sent: Monday, July 19, 2010 1:16 PM
>> To: asterisk-users@lists.digium.com
>> Subject: [asterisk-u
Hi,
I got the captured packet traces and we could see that it was coming from
our asterisk server. Is there any other things that I need to look into,
also we have updated from Zaptel to Dahdi 2.0.2.2, but no luck, still the
random redial dtmf tones are coming in between calls...Can anyone sha
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mattias
Sent: Monday, July 19, 2010 1:16 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Voice prompts
Have now installed a swedish prompt set
I
Have now installed a swedish prompt set
In /var/lib/asterisk/sounds/se
I run elastix
And set
Language=se in /etc/asterisk/sip.conf
But not work
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Thanks
Some companies here in swiden have a swedish female
And on the link only male voices
-Ursprungligt meddelande-
Från: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] För Danny Nicholas
Skickat: den 19 juli 2010 19:11
Till: 'Asterisk Users Mail
On 07/19/2010 11:28 AM, Tim Nelson wrote:
> - "Johann Steinwendtner" wrote:
>> Hello !
>>
>> I 'm using a TE405P with a HW echocanceller module attached on it.
>> dahdi version is dahdi-linux-complete-2.2.0.2+2.2.0.
>>
>> As far as I know, the fax tone detection is done on the FW board.
>> How
I am sure you can't achieve what you are trying to achieve here. Simply use
two different extensions instead of one.
Considering how SIP communication works, I believe SIP doesn't allow
multiple registrations like this. Maybe somebody can correct me here if I am
wrong.
Zeeshan A Zakaria
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-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mattias
Sent: Monday, July 19, 2010 11:31 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Swedish voiceprmpts
Exist it?
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Try this link
h
On Mon, Jul 19, 2010 at 12:22:32PM -0400, Zeeshan Zakaria wrote:
> Yes, you could do includes in sip.conference like:
> [general]
> ...
> ...
> ...
> #include sip1.conf
> #include sip2.conf
>
> Just make sure to do it AFTER the [general] section.
Actually, you can also use:
[general]
...
[some-
- "Johann Steinwendtner" wrote:
> Hello !
>
> I 'm using a TE405P with a HW echocanceller module attached on it.
> dahdi version is dahdi-linux-complete-2.2.0.2+2.2.0.
>
> As far as I know, the fax tone detection is done on the FW board.
> How can I verify that the echo canceller has been tu
Exist it?
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asterisk-users mailing list
thanks a lot zishan and philipp,
probably that is the problem that is occurring. I am gonna take some
wireshark or etherial trace to further investigate the problem.
i don't wanna stuck into port forwarding issue as it will waste lot of time
and also normal calling is working on my current port fo
Yes, you could do includes in sip.conference like:
[general]
...
...
...
#include sip1.conf
#include sip2.conf
Just make sure to do it AFTER the [general] section.
Zeeshan A Zakaria
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On 2010-07-19 12:00 PM, "Gareth Blades" wrote:
Ken D'Ambrosio wrote:
> Hey, all. I'm try
Hello !
I 'm using a TE405P with a HW echocanceller module attached on it.
dahdi version is dahdi-linux-complete-2.2.0.2+2.2.0.
As far as I know, the fax tone detection is done on the FW board.
How can I verify that the echo canceller has been turned off ?
When I do a cat /proc/dahdi/1 for span
Could you give me an example because I understand what you said, but not
sure what to put in my extensions.conf to accomplish that.
James Shigley
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P.
Flemi
On 07/19/2010 11:08 AM, James A. Shigley wrote:
>
>
> Let me rephrase this question.
>
>
>
> What context does a queue use for dialing out?
It doesn't, it dials the member directly. If you need it to dial out
through the dialplan, add a Local channel as a member, instead of the
actual chann
Let me rephrase this question.
What context does a queue use for dialing out?
James Shigley
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James A.
Shigley
Sent: Monday, July 19, 2010 7:42 AM
To: Asterisk Users Mailing L
When I had this problem I contacted digium who sent me instructions on
how to setup the span and make a loopback plug. I then left it running
for a while but no errors were reported.
The telco then started monitoring the line and after a couple of days
diagnosed a faulty card in the local excha
I did restart the zaptel after making changes.. but just to reconfirm I
restarted it again..
but the problem still persists.
On Mon, 19 Jul 2010 16:43:57 +0100, Gareth Blades
wrote:
> Have you restarted zaptel since making any changes?
> You are receiving FCS errors but you dont appear to ha
Ken D'Ambrosio wrote:
> Hey, all. I'm trying to do some fun with auto-provisioning of Polycom
> phones, and one thing that would make life easier for me would be if I
> could have a per-phone sip.conf file. If not, no biggie -- but if there's
> a way to do an include (as per extensions.conf) or s
Hey, all. I'm trying to do some fun with auto-provisioning of Polycom
phones, and one thing that would make life easier for me would be if I
could have a per-phone sip.conf file. If not, no biggie -- but if there's
a way to do an include (as per extensions.conf) or something, that would
be great.
Have you restarted zaptel since making any changes?
You are receiving FCS errors but you dont appear to have crc4 specified
in your span lines.
If you have removed the option but not restarted zaptel yet then do that
to see if it fixes the problem.
Chetan Meshram wrote:
> Hi All,
>
>
--- On Mon, 7/19/10, Kevin P. Fleming wrote:
> Usage of the standard Skype client is not "free"; it
> involves acting as
> part of the peer-to-peer Skype network
> The Skype
> business solutions (including Skype For Asterisk) don't
> participate in
> the peer-to-peer network
> Any solution t
Hi Nasir,
Please don't send me direct emails, unless you want to secure my paid
consultancy services or want to do some other business.
For setting up the RTP, you need to do it on your firewall, which is
receiving RTP traffic from these particular IP address. I can't guess how to
do it on your r
Hi All,
I am facing problem with E1 line. I have installed Asterisk
(1.4.20.1) on a system with Digium TE420 card (Zaptel- 1.4.10)
But every
now and then I face problem of down E1's. The log show lot of entries like
"pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of
s
We are a telco so when we receive calls via ISDN and the number is
witheld we see the correct presentation value but also still see the
actual callers number in the callerid(num) variable.
I am trying to forward some of these calls over to one of our other
boxes via SIP but I have found that if
It's doable with a work around. Create a misc extension with followme set to
##70# which point to your parking lots and failed destination to Misc
parking extension.
Regards,
Bruce
On Sun, Jul 18, 2010 at 3:38 PM, Doug Lytle wrote:
> bruce bruce wrote:
> > Hi Everyone,
> >
> > If I receive a ca
I have been using SiSky Enterprise Edition to integrate Skype with asterisk.
You can even call saved skype users from your asterisk system, by creating
speed dials in SiSky. Unfortunately it is not a free product but it is very
reasonable.
Thank you,
Brad Finberg
- Original Message -
Hi,
I am trying to write the regserver value into my database using ARA but the
field keeps empty.
Afaik all that needs to be done to make it work is having a db field called
regserver, the var systemname set in asterisk.conf and
rtsavesysname=yes in sip.conf.
But the regserver is not gettin
OK, now i added the column members in the table booking manually.
and disabled selinux to have this working.
Now i am struggling with recording option in webmeetme.
Not sure on how to enable it, though m checking the checkbox while creating
the conference. But where does this save and how to retr
Ok I have a queue that is working perfectly.
The problem is when one of the agents is outside the building on an
external phone line (say a cell phone). My telco hangs up on the call .
I think the telco is hanging up on these calls because there is no CID
attached. (I know my telco wo
Hi, the following configuration:
The number 0 will be forwarded to the Ring-Group 25 in which the numbers
are 71 and 73. If you call the 0 so the office is ringing at the 71 and
73 .
At the terminal stations are Snom 320.
In the evening the 71 to make call forwarding via web interface
On 07/18/2010 12:18 PM, Steve Kennedy wrote:
> On Sun, Jul 18, 2010 at 09:56:30AM -0700, Vieri wrote:
>
>>> As I said above, once you have purchased your SIP channel
>>> you can make
>>> free calls to your PBX using the special number but it's
>>> only INBOUND
>>> AFAIK.
> [lots snipped]
>
> With
On 07/18/2010 11:56 AM, Vieri wrote:
> I still don't see why one should pay for a channel when using a PBX but not
> when using a client such as Skype. OK, I know that the Skype network is
> proprietary and I have to accept whatever they say.
Usage of the standard Skype client is not "free"; it
Hi, the following configuration:
The number 0 will be forwarded to the Ring-Group 25 in which the numbers
are 71 and 73. If you call the 0 so the office is ringing at the 71 and
73 .
At the terminal stations are Snom 320.
In the evening the 71 to make call forwarding via web interface
You will need to contact the companies directly if you want a complete
price list.
We have a very high call volume so the price you get might be different
if you dont make many calls.
amit mehta wrote:
> Hi Gareth,
>
> Thanks for the swift reply.
>
> Kindly provide A-Z price list.
>
> Regards
Hi Gareth,
Thanks for the swift reply.
Kindly provide A-Z price list.
Regards,
Amit
On Mon, Jul 19, 2010 at 3:42 PM, Gareth Blades
wrote:
> amit mehta wrote:
> > Hello,
> >
> > I am looking for Voip providers for voip minutes to Mali(South Africa)
> >
> > Kindly provide the ratesheet for the s
amit mehta wrote:
> Hello,
>
> I am looking for Voip providers for voip minutes to Mali(South Africa)
>
> Kindly provide the ratesheet for the same.
>
> Regards,
> Amit Mehta
>
AQL - 0.1816 GBP/min
Magrathea high call volume rate - 0.126 GBP/min
They are a couple of UK providers. If it is only
Hello,
I am looking for Voip providers for voip minutes to Mali(South Africa)
Kindly provide the ratesheet for the same.
Regards,
Amit Mehta
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Rob Many thanks for the pointer - I was missing limitonpeers=yes in the
general section - Sorry I didn't say version (1.4.33.1) etc forgot with
frustration ;-)
Paddy
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