Martin wrote:
> well maybe it doesn't work as it should ... anyways like the other
> poster said that's not the way you use it ...
>
> either call the sendfax app directly or use "Originate" / call file
> spooling...
>
> BTW there should be an Originate app executable from dialplan ...
> But sin
On Wednesday 23 September 2009 17:27:46 Administrator TOOTAI wrote:
> after having tested SFA in august, I didn't use it for some times and
> now I receive the subject error when calling through Skype channel.
>
> Has anyone an idea on what can be the problem?
Have you considered the possibility t
Hi,
after having tested SFA in august, I didn't use it for some times and
now I receive the subject error when calling through Skype channel.
Has anyone an idea on what can be the problem?
Thanks
--
Daniel
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I'm trying to implement Danny's suggestion but I'm blocked, at the moment,
dynamic features settings (I opened a dedicated thread to that purpose) : I
can't tie any DTMF string to my dialplan (I'm using AEL2).
Any suggestion ?
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-- Bandwidth and Coloca
Hello,
I'm using AEL2 (in Asterisk 1.6.1.6) and I can't find a way to successfully
come back into my dialplan.
I've tried things like this (in features.conf) :
toto => #9,peer,Goto,mylocal2,s,1
But typing #9 (from channel SIP/7275, in example bellow) I've got:
-- Feature Found: toto exten:
On Wed, 2009-09-23 at 09:39 -0700, mgra...@mstvp.com wrote:
> I had a good experience with that Polycom/Spectralink phone. Very rugged
> as you say. The experience did highlight the weaknesses in consumer
> Wifi AP, which reinforced my commitment to continue using DECT around my
> office.
>
> Mic
I SAID it was untested... I tried to look up this thread in my emails, but
that repository has about 8K messages.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A.
Sullivan III
Sent: Wednesday, September
Won't that hangup the call after 60 seconds? - John
On Wed, 2009-09-23 at 15:22 -0500, Danny Nicholas wrote:
> Here’s a snippet from a reply from Jared Smith (Digium, Huntsville AL)
> - untested
>
> exten => 11234,1,Set(TIMEOUT(absolute)=60)
>
> exten => 11234,n,MeetMe(11234,d1M)
>
>
>
> Th
Hi Juan, I didn't use the GoSub application, I put the name of the context in
the Originate and the variables and their values in the Variable field.
See http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate.
Good luck!
Anahi Ludueña
From: jcard...@tpmex.com
To: asterisk
Here's a snippet from a reply from Jared Smith (Digium, Huntsville AL) -
untested
exten => 11234,1,Set(TIMEOUT(absolute)=60)
exten => 11234,n,MeetMe(11234,d1M)
This should create a dynamic room 11234 and send the caller to it for 60
seconds.
_
From: asterisk-users-boun...@lists.
2009/9/23 Danny Nicholas
> This stands to be corrected, but for your purpose, a dynamic conference
> is preferable to a parking lot. The Park application is designed to
> sequentially use/reuse a series of “lots”. By transferring the caller to
> conference 11234, you would be able to have the
This stands to be corrected, but for your purpose, a dynamic conference is
preferable to a parking lot. The Park application is designed to
sequentially use/reuse a series of "lots". By transferring the caller to
conference 11234, you would be able to have the agent pick up the call by
going to c
Hi,
I'm having trouble to figure out how I could implement this feature :
"When on call with a contact, local operator would dial a sequence which
would park the remote party to a specific parking slot, among the hundred of
existing slots.
(to each extension, a single specific parking slot is atta
On Wed, Sep 23, 2009 at 09:39:09AM -0700, mgra...@mstvp.com wrote:
> I had a good experience with that Polycom/Spectralink phone. Very rugged
> as you say. The experience did highlight the weaknesses in consumer
> Wifi AP, which reinforced my commitment to continue using DECT around my
> office.
- Original Message -
From: "Martin"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, September 23, 2009 11:01:04 AM GMT -05:00 US/Canada Eastern
Subject: Re: [asterisk-users] Asterisk on a Beagleboard?
Even PCI has 133MB/s ... so what ? Also isn't USB
Hi All -
At Leif's suggestion, I'm soliciting testers for a patch to IMAP voicemail.
Currently, when asterisk checks for voicemails in an IMAP folder, it
only looks for messages in the same context and with the same
voicemail box number as the person dialing in to VoicemailMain(). I
believe this
Well 1.6.2 is not yet released - it's rc2 now of course the app
is somewhere ... since it's very easy to code ...
actually it should have been added at the time when originate was
added to CLI ... it's a pity someone who added cli originate
did not think about writing a few more lines for orig
- "Jason Baker" escreveu:
> Ken,
> I did lots of research on this for my VoIP deployment here where I
> work. We have a huge manufacturing floor and all the supervisors have
> wifi phones. We evetually settled on the Polycom Spectralink 8002. A
> nice rugged little phone with great sound qual
I had a good experience with that Polycom/Spectralink phone. Very rugged
as you say. The experience did highlight the weaknesses in consumer
Wifi AP, which reinforced my commitment to continue using DECT around my
office.
Michael
> Original Message
> Subject: Re: [asterisk-use
Tilghman Lesher wrote:
> On Wednesday 23 September 2009 05:49:54 stephen.hindma...@bt.com wrote:
>> I am using asterisk 1.6.1.6 and have been setting up a system to use a
>> Postgresql database as the realtime DB via the ODBC route. I have got
>> extensions and voicemail working but am having troub
Hi everyone,
Does someone know why the solution for bug 13115
(https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=13115)
was made only for trunk? Having that this bug went solved more than a
year ago, it means that all the 1.6.X.X branches have it applied
already? Can this be backpo
On Wednesday 23 September 2009 05:49:54 stephen.hindma...@bt.com wrote:
> I am using asterisk 1.6.1.6 and have been setting up a system to use a
> Postgresql database as the realtime DB via the ODBC route. I have got
> extensions and voicemail working but am having trouble with SIP
>
> The problem
On Wed, 2009-09-23 at 10:17 -0500, Martin wrote:
> BTW there should be an Originate app executable from dialplan ...
> But since there's none you can do
There is an Originate application, but it's only available in newer
versions of Asterisk. (I know I have it on the 1.6.2 branch, but I
don't rem
well maybe it doesn't work as it should ... anyways like the other
poster said that's not the way you use it ...
either call the sendfax app directly or use "Originate" / call file spooling...
BTW there should be an Originate app executable from dialplan ...
But since there's none you can do
ext
You can go to any context from AMI by using Context: context, priority: 1
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Juan Cardoza
Sent: Wednesday, September 23, 2009 10:10 AM
To: 'Asterisk Users Mailing List - Non
I need the same information, did you find that information Anahi???
Best regards
Juan Cardoza
De: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] En nombre de Anahi Ludueña
Enviado el: Miércoles, 16 de Septiembre de 2009 09:49 a.m.
Para: asterisk-users
Ken,
I did lots of research on this for my VoIP deployment here where I
work. We have a huge manufacturing floor and all the supervisors have
wifi phones. We evetually settled on the Polycom Spectralink 8002. A
nice rugged little phone with great sound quality and some good
features. We use a m
Even PCI has 133MB/s ... so what ? Also isn't USB only target ? It
doesn't do DMA ...
so it might be same as PCI Target chips that slow down the CPU
TDMoE has to have those frames on time all the time forever ...
these ethernet frames are sent both ways every 1ms
that might be (or not) too much lo
For the last 10 years we have had some Tandberg video conferencing units
spread across our WAN. To make a long story short we had to work with
another entity to allow access into those units with H.323. Calls from
public networks have never worked to those devices with private IP's.
Over the su
Anyone know of any *portable* SIP/WiFi handsets? Looking for a decent
price:quality ratio, of possible. Keep seeing handsets for Vonage, etc.,
in Best Buy and the like, but I imagine it's locked to Vonage, and can't
be re-appropriated.
Thanks!
-Ken
--
This message has been scanned for viruse
Thank you for answer. It was very informative, I put it in our wiki if you
don't mind.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
On Tue, Sep 22, 2009 at 05:32:13PM -1000, Julian Yap wrote:
> I have an issue where a particular dialplan works but another doesn't. I'm
> not sure why. To me they look identical and it has me stumped.
>
> This works:
> [to-test]
> exten => _X., 1, SetCallerPres(allowed)
> exten => _X., 2, Monit
On Wed, 23 Sep 2009, Tzafrir Cohen wrote:
> On Tue, Sep 22, 2009 at 07:43:51PM -0500, Martin wrote:
>> I do not know if fonebridge would work here since it sends/receives
>> the ~2 Mbps (for each circuit/port)
>> of data over ethernet ... constantly. That could choke the USB ...
>
> Ethernet has
On Wed, Sep 23, 2009 at 08:37:19AM +0200, Loic Didelot wrote:
> Hi Tzafrir,
> I just compiled the tarball, but now there seem to be some problems with
> the script lszaptel.
>
>
> Can't call method "is_twinstar" on unblessed reference
> at /usr/local/share/perl/5.8.8/Zaptel/Hardware/USB.pm line 1
On Wed, Sep 23, 2009 at 08:14:41AM +0200, Loic Didelot wrote:
> Thank you for the information. The tarball compiles fine. Except that it
> has no qozap module.
This is a snapshow of the upstream Zaptel tarball, and hence does not
include qozap. It should be simple to use hat tarball in the bristuf
On Tue, Sep 22, 2009 at 07:43:51PM -0500, Martin wrote:
> I do not know if fonebridge would work here since it sends/receives
> the ~2 Mbps (for each circuit/port)
> of data over ethernet ... constantly. That could choke the USB ...
Ethernet has frames. While I'm not exactly sure how ethernet over
I am using asterisk 1.6.1.6 and have been setting up a system to use a
Postgresql database as the realtime DB via the ODBC route. I have got
extensions and voicemail working but am having trouble with SIP
The problem seems to be with using a schema. If I put the table "sip" in
the schema "foo" the
dear all,
i have one issue in my DAHDI channel
while my incoming call connected it suddenly disconnected and got following
error
< Protocol Discriminator: Q.931 (8) len=9
< Call Ref: len= 2 (reference 357/0x165) (Originator)
< Message type: DISCONNECT (69)
< [08 02 82 90]
< Cause (len= 4) [ Ext:
On Wednesday 23 September 2009 01:44:31 sean darcy wrote:
> Does anyone use SendFax for analog faxing?
>
Yes. I have two contexts as follows:
[outbound]
exten => _X.,1,Dial(DAHDI/G2/${EXTEN})
[sendfax]
exten => s,1,SendFAX(${FAXFILE})
exten => h,n,Hangup()
When I want to send a fax, I initi
Dear All
Can you please do me favor and let me know which Asterisk codec you will
prefer when you want to offer your subscribers with dialup data connection ?
Let me thank you in advance
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