3 jan 2011 kl. 00.26 skrev Bryan Field-Elliot:
Normally, no matter which Asterisk server an ATA connects to, we get our
database fields filled out correctly, such as regseconds, lastms,
ipadr, etc. However, with some ATA's we are experiencing a problem as
follows:
1. ATA reaches its
- Original Message -
--[ UxBoD ]-- wrote:
- Original Message -
Yes exactly that indeed. Though Asterisk appears to ignore which
context the user is in and selects default instead. Beginning to
think that it is a bug.
I got it figured out.
In your
Hello list,
how can I go from CALLINGout to just CALLING ?
I've tried :
exten = s,n,Set(newVAR=${CUT(CALLINGout,,3)})
or
exten = s,n,Set(newVAR=$[CUT(CALLINGout,,3)])
But no result :
[Jan 4 11:10:12] -- Executing [...@from-s:34] NoOp(SIP/s2-003b,
newVAR=) in new stack
Asterisk
On Sat, 01 Jan 2011 23:32:15 +, Sebastian s...@open-t.co.uk
wrote:
Anyway - there is a third option - which I have been using with some
success. I connected my softphone on my laptop to my Asterisk server at
home (through OpenVPN for extra security - but this is not compulsory). [...]
As a
Hi Siobhan,
Asterisk is all capacity to work-on but you need to find out some way of
handling conference system through WEB part , also one more thing on last
point for switching between conference
i am not much sure about it but i think it is possible if i will look into
code implementation.
Le 03/01/2011 18:28, Gilles a écrit :
On Mon, 03 Jan 2011 12:27:56 +0100, Administrator TOOTAI
ad...@tootai.net wrote:
As you are a Free Telecom customer, why not using your freephonie
account to forward incoming calls to your mobile?
Thanks for the tip, but experience shows that
On Tue, Jan 4, 2011 at 5:14 AM, Jonas Kellens jonas.kell...@telenet.be wrote:
Hello list,
how can I go from CALLINGout to just CALLING ?
I've tried :
exten = s,n,Set(newVAR=${CUT(CALLINGout,,3)})
or
exten = s,n,Set(newVAR=$[CUT(CALLINGout,,3)])
But no result :
[Jan 4 11:10:12] --
On 01/03/2011 07:53 PM, cjwstudios wrote:
Andy,
The 501 and 320 are EOL. I'd go for the IP335 and a 2626-PWR, since the
2626 can support vlans you can isolate data and voice. Make sure to
spec a UPS on the PoE switch.
CJW,
Awesome. Thanks for the input. For some reason or another I
On Tuesday 04 Jan 2011, Gilles wrote:
Thanks Sebastian for the tip. The goal is to 1) have clients call the
usual landline number instead of asking them to try a cellphone in
case no one's home, 2) get Asterisk to handle the call, 3) have the
cellphone ring with the CID of the original caller
On 01/03/2011 07:08 PM, Steve Underwood wrote:
On 01/04/2011 04:22 AM, Kevin P. Fleming wrote:
On 01/03/2011 11:26 AM, Tom Rymes wrote:
Hi folks,
I was hoping that someone might be able to help clarify some confusion I
have on DAHDI Fax detection after spending some time searching. My
On 01/03/2011 06:47 PM, Thomas Rymes wrote:
On Jan 3, 2011, at 3:22 PM, Kevin P. Fleming wrote:
On 01/03/2011 11:26 AM, Tom Rymes wrote:
[snip]
1.) Echo cancellation is automatically disabled upon recognition of a
CNG tone, regardless of the faxdetect setting. This can only be disabled
at
Le 04/01/2011 11:50, Gilles a écrit :
[...]
It looks like getting a 3G smartphone with SIP + OpenVPN + unlimited
Internet plan would solve the issue.
I Would avoid OpenVPN (tested an Android) as it drains quickly battery
[...]
2. what smartphone supports installing an SIP + OpenVPN
Thanks Olle. Do you suppose I am the first Asterisk user to discover this
behavior? I would find that hard to believe that I'm the first person to
notice...
Your idea for how to deal with sounds reasonable..
Thank you,
Bryan
On Jan 4, 2011, at 12:18 AM, Olle E. Johansson wrote:
3 jan
Hi All,
I have weird requirement for call forwarding. I have forward all call from A
to B extension because A is very busy and sometime not available so B is taking
care of all forwarding call from A. but in some case B need to transfer call to
A and in this case call coming back to B again
Hi,
Though this has no direct relation with Asterisk, I think Asterisk users in
Contact Centers might have an interesting answer.
Using Gmail, it's rather easy to label an incoming email so that any related
email (reply) inherits this label and are commonly displayed together in
threads.
Using
Anyone ever noticed that the reported holdtime is wrong when there are
different priorities? Also talktime is 0, but for the moment I don't
care.
queue show test reports:
test has 23 calls (max unlimited) in 'ringall' strategy (193s holdtime,
0s talktime)
[...]
Callers:
1.
Hi list,
I just installed Asterisk 1.4.38 (on an updated Centos 5.5 machine) and am
getting this error :
WARNING[6472]: res_musiconhold.c:856 moh_scan_files: getcwd() failed: No such
file or directory
with the default musiconhold.conf file. When I change musiconhold.conf to this:
[default]
Hi,
On 01/04/2011 10:50 AM, Gilles wrote:
On Sat, 01 Jan 2011 23:32:15 +, Sebastians...@open-t.co.uk
wrote:
Anyway - there is a third option - which I have been using with some
success. I connected my softphone on my laptop to my Asterisk server at
home (through OpenVPN for extra security
On 01/04/2011 8:55 AM, Kevin P. Fleming wrote:
On 01/03/2011 06:47 PM, Thomas Rymes wrote:
On Jan 3, 2011, at 3:22 PM, Kevin P. Fleming wrote:
On 01/03/2011 11:26 AM, Tom Rymes wrote:
[snip]
OK. Either way, though, the changes to echo cancellation are not
affected by the faxdetect setting,
On Tue, Jan 4, 2011 at 8:52 AM, Andy Graybeal
andy.grayb...@casanueva.com wrote:
On 01/03/2011 07:53 PM, cjwstudios wrote:
Andy,
The 501 and 320 are EOL. I'd go for the IP335 and a 2626-PWR, since the
2626 can support vlans you can isolate data and voice. Make sure to
spec a UPS on the PoE
Anyone else know about the holding concurrent conferences (and switching
back and forth) issue ? Is it possible?
And can you set up dynamic conferences that continue even when the initiator
leaves?
Thanks!
On Tue, Jan 4, 2011 at 7:11 AM, DHAVAL INDRODIYA
dhaval.it01...@gmail.comwrote:
Hi
On 01/04/2011 01:55 PM, A J Stiles wrote:
On Tuesday 04 Jan 2011, Gilles wrote:
Thanks Sebastian for the tip. The goal is to 1) have clients call the
usual landline number instead of asking them to try a cellphone in
case no one's home, 2) get Asterisk to handle the call, 3) have the
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel
Sent: Tuesday, January 04, 2011 9:36 AM
To: asterisk-users
Subject: [asterisk-users] Call forwrading but call transfer back
Hi All,
I have weird requirement for
This should probably go to the dev list but I think developers and
users both need to see the progress.
A year or two, I posted about ofono right at their launch. It seems
they have come a long way and Digium or other VoIP platforms may want
to make alliances with ofono.
It is not a joke
Hi,
On 01/04/2011 03:24 PM, Administrator TOOTAI wrote:
Le 04/01/2011 11:50, Gilles a écrit :
[...]
It looks like getting a 3G smartphone with SIP + OpenVPN + unlimited
Internet plan would solve the issue.
I Would avoid OpenVPN (tested an Android) as it drains quickly battery
Any chance
On Tuesday 04 January 2011 09:40:56 Bryan Field-Elliot wrote:
Thanks Olle. Do you suppose I am the first Asterisk user to discover
this behavior? I would find that hard to believe that I'm the first
person to notice...
It wasn't designed to do this. While you can have the same sippeers table
Hi
CLI module unload res_musiconhold.so
CLI module load res_musiconhold.so
or
service asterisk restart
Regards
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
On 01/04/2011 8:52 AM, Andy Graybeal wrote:
Is it possible that I can run one cable to the phone, then run a cable
from the phone to a computer or another device and have those the phone
and computer or other device be on separate networks?
I'm sorry if this sounds newbish; I'm still learning.
On 01/04/2011 12:31 PM, Earl Terwilliger wrote:
Hi list,
I just installed Asterisk 1.4.38 (on an updated Centos 5.5 machine) and am
getting this error :
WARNING[6472]: res_musiconhold.c:856 moh_scan_files: getcwd() failed: No such
file or directory
[snip]
Have you installed mpg123 or some
According to https://issues.asterisk.org/view.php?id=16339 , the default
value for the dialdebounce parameter of the wctdm module has been
changed to 32 and is now user configurable.
I have two questions:
1.) Am I correct in presuming that, if the default of 32 does not work
for me, I would
On 01/03/2011 9:46 PM, Matt Watson wrote:
I don't imagine this would be too complicated - don't have any
experience with AsteriskNOW - but on a 'vanilla' linux distro it would
just be a matter of making sure dahdi is loading the correct drivers and
doing a couple of minor config file updates.
The Polycom 321 has not been EOL'd and supports VLAN. It is, however,
lacking a 2nd ethernet port if you were to go that route.
-M
Thanks for the response Mark. I see the 331 has two ports and the same
features as the 321.
I'm wondering what phone would be best being used as an intercom in
Hi. I have a Debian Leni system with asterisk-1.8. I was trying to
get meetme to work and it depends on dahdi, so I compiled dahdi-trunk
and dahdi-tools-trunk, however, when trying to insert dahdi_dummy, it
complained about symbol crc_ccitt_table, although the module was
actually there in the
Hi all,
I am trying to set up DND in my asterisk, I am using the following context:
[app-naoperturbe]
exten = *11,1,Set(DND=${DB(ddisturbe/${CALLERIDNUM})})exten =
*11,2,GotoIf($[${DND} = YES]?*11,3:*11,101)exten =
*11,3,Set(DB(ddisturbe/${CALLERIDNUM})=NO)exten = *11,4,Playback(beep)exten =
On 01/05/2011 02:39 AM, Tom Rymes wrote:
On 01/04/2011 8:55 AM, Kevin P. Fleming wrote:
On 01/03/2011 06:47 PM, Thomas Rymes wrote:
On Jan 3, 2011, at 3:22 PM, Kevin P. Fleming wrote:
On 01/03/2011 11:26 AM, Tom Rymes wrote:
[snip]
OK. Either way, though, the changes to echo cancellation
On 01/04/2011 09:53 PM, Kevin P. Fleming wrote:
On 01/03/2011 07:08 PM, Steve Underwood wrote:
On 01/04/2011 04:22 AM, Kevin P. Fleming wrote:
No. CNG tone is never used to affect the state of an echo canceller.
All G.168 compliant echo cancellers will respond to the CED tone
(generated by
Flavio Miranda wrote:
Hi all,
I am trying to set up DND in my asterisk, I am using the following
context:
[app-naoperturbe]
exten = *11,1,Set(DND=${DB(ddisturbe/${CALLERIDNUM})})
This is mine:
[dnd]
;***
;* Do not disturb can be set via Asterisk
;*
On Tue, 4 Jan 2011, Andy Graybeal wrote:
The Polycom 321 has not been EOL'd and supports VLAN. It is, however,
lacking a 2nd ethernet port if you were to go that route.
-M
Thanks for the response Mark. I see the 331 has two ports and the same
features as the 321.
I'm wondering what
On 01/04/2011 05:09 PM, cov...@ccs.covici.com wrote:
Hi. I have a Debian Leni system with asterisk-1.8. I was trying to
get meetme to work and it depends on dahdi, so I compiled dahdi-trunk
and dahdi-tools-trunk, however, when trying to insert dahdi_dummy, it
complained about symbol
El 04/01/11 18:13, Flavio Miranda escribió:
Hi all,
I am trying to set up DND in my asterisk, I am using the following
context:
[app-naoperturbe]
exten = *11,1,Set(DND=${DB(ddisturbe/${CALLERIDNUM})})
exten = *11,2,GotoIf($[${DND} = YES]?*11,3:*11,101)
exten =
On Tuesday 04 January 2011 16:15:54 Andy Graybeal wrote:
The Polycom 321 has not been EOL'd and supports VLAN. It is, however,
lacking a 2nd ethernet port if you were to go that route.
-M
Thanks for the response Mark. I see the 331 has two ports and the same
features as the 321.
IMHO G.722 beats Clarity By Polycom every time.
I had an IP335 for review before they launched. The audio quality is the
same as the better models (IP450/550/650) only the user interface is
different. Very good speakerphone, too.
Review here:
On Tuesday, January 04, 2011 04:37:21 pm Tom Rymes wrote:
On 01/04/2011 12:31 PM, Earl Terwilliger wrote:
Hi list,
I just installed Asterisk 1.4.38 (on an updated Centos 5.5 machine) and
am getting this error :
WARNING[6472]: res_musiconhold.c:856 moh_scan_files: getcwd() failed: No
On Tuesday, January 04, 2011 04:29:49 pm bakko wrote:
Hi
CLI module unload res_musiconhold.so
CLI module load res_musiconhold.so
or
service asterisk restart
Regards
--
_
-- Bandwidth and Colocation Provided by
I really would like to understand why dont works!
should I to set up any other function? maybe on features?
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
Date: Tue, 4 Jan 2011 20:08:39 -0500
From: supp...@drdos.info
To:
Shaun Ruffell sruff...@digium.com wrote:
On 01/04/2011 05:09 PM, cov...@ccs.covici.com wrote:
Hi. I have a Debian Leni system with asterisk-1.8. I was trying to
get meetme to work and it depends on dahdi, so I compiled dahdi-trunk
and dahdi-tools-trunk, however, when trying to insert
On 1/4/11 9:26 PM, cov...@ccs.covici.com wrote:
Shaun Ruffellsruff...@digium.com wrote:
On 01/04/2011 05:09 PM, cov...@ccs.covici.com wrote:
Hi. I have a Debian Leni system with asterisk-1.8. I was trying to
get meetme to work and it depends on dahdi, so I compiled dahdi-trunk
and
On Jan 4, 2011, at 8:49 PM, Earl Terwilliger wrote:
On Tuesday, January 04, 2011 04:37:21 pm Tom Rymes wrote:
On 01/04/2011 12:31 PM, Earl Terwilliger wrote:
Hi list,
I just installed Asterisk 1.4.38 (on an updated Centos 5.5 machine) and
am getting this error :
WARNING[6472]:
On Jan 4, 2011, at 7:37 PM, Steve Underwood wrote:
It is very normal for many people to chat and then start their FAX machines,
especially domestic FAX users with a FAX machine attached to their home land
line. If you don't care about those your proposal is OK, otherwise.
Well, I was
Shaun Ruffell sruff...@digium.com wrote:
On 1/4/11 9:26 PM, cov...@ccs.covici.com wrote:
Shaun Ruffellsruff...@digium.com wrote:
On 01/04/2011 05:09 PM, cov...@ccs.covici.com wrote:
Hi. I have a Debian Leni system with asterisk-1.8. I was trying to
get meetme to work and it
On 1/5/11 12:46 AM, cov...@ccs.covici.com wrote:
Shaun Ruffellsruff...@digium.com wrote:
On 1/4/11 9:26 PM, cov...@ccs.covici.com wrote:
Shaun Ruffellsruff...@digium.com wrote:
On 01/04/2011 05:09 PM, cov...@ccs.covici.com wrote:
Hi. I have a Debian Leni system with asterisk-1.8. I
Hi Everyone,
1- Are the Siren7 and Siren14 the G.722 HD voice codecs?
2- Are these codecs only for Polycom units or are they universal across all
other SIP phones that advertise the HD voice codec like Aastra?
3- What is the main difference between the two and is it advisable to run
these over
Hi Guys,
What is out there other than OutCall that works with M$ Outlook and Asterisk
1.4/1.6 ? I prefer opensource and free (as in free in price) but can
consider low price - working - programs as well.
OutCall is giving issues with various versions of Outlook and it always
brings up NEW
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