On Fri, Jul 17, 2009 loganlogan...@gmail.com wrote:
Hi,
I'm an absolute newbie and wanted to know the following.
I want to have a setup where I have a PSTN line connected to my
Asterisk box and want to know if it is possible to make more than one
simultaneous outbound call through that VoIP
On 18/07/09 00:35, Gavin Henry wrote:
This has to be an Asterisk based appliance no?
http://www.truecall.co.uk/acatalog/trueCall_Features.html
I saw this on the TV the other night. Couldn't believe how the dragons
all thought it was such a cool idea.
I was shouting at the telly saying You
What is interesting is that there is no mention of the software used -
if it is asterisk, he would need to make the code available, no ?
Julian
2009/7/18 Alan Lord (News) alansli...@gmail.com:
On 18/07/09 00:35, Gavin Henry wrote:
This has to be an Asterisk based appliance no?
On Sat, Jul 18, 2009 at 4:36 AM, Alan Lord (News) alansli...@gmail.comwrote:
On 18/07/09 00:35, Gavin Henry wrote:
This has to be an Asterisk based appliance no?
http://www.truecall.co.uk/acatalog/trueCall_Features.html
I saw this on the TV the other night. Couldn't believe how the
Why would he mention what software is used?
My old school answering machine doesn't mention what OS it uses, nor do my
cordless phones.
Actually, none of my consumer goods say what OS they use except maybe a
WindowsCE machine or something.
Just because it can do alot of what Asterisk can do,
Hello,
I recently updated my asterisk-addons-1.6.2 to the last revision and I have
this problem that I don't know how to interpret, bug or not. I connected a
Nokia N80 phone to use chan_mobile and everything works great until the
phone starts getting disconnected after the call finished and
Hello,
I read on the wiki that chan_mobile supports one device per dongle. Is this
still the case?
From the official website, there is very little info but this line Channel
Groups for implementing ‘GSM Gateways’ which leads me to believe (or hope
at least) that more than one phone can be
On Sat, Jul 18, 2009 at 10:11:42AM +0100, Julian Lyndon-Smith wrote:
What is interesting is that there is no mention of the software used -
if it is asterisk, he would need to make the code available, no ?
Most likely.
--
Tzafrir Cohen
icq#16849755
yes, only one device per USB dongle.
On Sat, Jul 18, 2009 at 4:22 PM, Steve Totaro
stot...@totarotechnologies.com wrote:
Hello,
I read on the wiki that chan_mobile supports one device per dongle. Is
this still the case?
From the official website, there is very little info but this line
In general, I found it hard to get chan_mobile working straight out of the
box, and although there is a great effort to make it so, phone manufacturers
are not helping by making command sets and BT implementations different from
device to device, SW version to SW version. Elastix seems to have
Hi all,
Someone know how can I check for available members on a queue Before I
queue the call, so I can do something else with it? Note that is not the
case for joinempty
Thanks,
Gabriel Ortiz
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On Sat, Jul 18, 2009 at 11:11 AM, Gabriel Ortiz Lour
ortiz.ad...@gmail.comwrote:
Hi all,
Someone know how can I check for available members on a queue Before I
queue the call, so I can do something else with it? Note that is not the
case for joinempty
Thanks,
Gabriel Ortiz
Maybe too
Maybe this function will help
QUEUE_MEMBER_COUNTQUEUE_MEMBER_COUNT(queuename) Count number of
members answering a queue
This is in at least version 1.6.0.x.
There are a few other related functions that can be seen by using the CLI
command:
core show functions like QUEUE
Function
Gabriel Ortiz Lour wrote:
Someone know how can I check for available members on a queue Before I
queue the call, so I can do something else with it? Note that is not the
case for joinempty
It's going to take some sort of hack, since there appears to be no
dialplan app to do this
I am current running on a production system
zaptel 1.4.12.1
libxpri 1.4.1
asterisk 1.4.25
The above configuration works.
I tried to update to dahdi 2.2.0, libpri 1.4.7 and asterisk 1.4.25
This did not work. calls came in but not out. I dropped back to the
initial configuration.
Today, I dried
Gabriel Ortiz Lour wrote:
Someone know how can I check for available members on a queue Before
I queue the call, so I can do something else with it? Note that is not
the case for joinempty
On Sat, 18 Jul 2009, Alex Balashov wrote:
It's going to take some sort of hack, since there
On Sat, Jul 18, 2009 at 12:57 PM, Steve Edwards
asterisk@sedwards.comwrote:
Gabriel Ortiz Lour wrote:
Someone know how can I check for available members on a queue Before
I queue the call, so I can do something else with it? Note that is not
the case for joinempty
On Sat, 18 Jul
On Sat, Jul 18, 2009 at 12:05:51PM -0400, Jerry Geis wrote:
I am current running on a production system
zaptel 1.4.12.1
libxpri 1.4.1
asterisk 1.4.25
The above configuration works.
I tried to update to dahdi 2.2.0, libpri 1.4.7 and asterisk 1.4.25
This did not work. calls came in but
On Saturday 18 July 2009 11:57:32 Steve Edwards wrote:
Gabriel Ortiz Lour wrote:
Someone know how can I check for available members on a queue Before
I queue the call, so I can do something else with it? Note that is not
the case for joinempty
On Sat, 18 Jul 2009, Alex Balashov wrote:
On Sat, Jul 18, 2009 at 1:50 PM, Tilghman Lesher
tilgh...@mail.jeffandtilghman.com wrote:
On Saturday 18 July 2009 11:57:32 Steve Edwards wrote:
Gabriel Ortiz Lour wrote:
Someone know how can I check for available members on a queue Before
I queue the call, so I can do something else
Assuming that connecting to the socket and authenticating and getting
the data is really less latent and resource-intensive. But that's
just splitting hairs.
--
Sent from mobile device
On Jul 18, 2009, at 12:57 PM, Steve Edwards
asterisk@sedwards.com wrote:
Gabriel Ortiz Lour wrote:
I just now tried 1.4.10.1 and same thing.
Channel 0/18, span 1 got hangup, cause 99
Drop back to 1.4.7 and it works again.
Jerry
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Dear asterisk users,
We want setup TE121 digium board:
Model: Digium TE121: VoiceBus technology allows the TE121 to use an
industry standard bus-mastering PCI Express interface.
http://www.digium.com/en/products/digital/te121.php
My platform
Server: HP Proliant 150 G5
OS: UBUNTU x86_64
Assuming he just wants people routed somewhere else if a queue if full, then
my original fast queue timeout answer is the simplest. Just continue on in
the dialplan.
I hope OP posts again with his/her solution.
Thanks,
Steve T
On Sat, Jul 18, 2009 at 3:18 PM, Alex Balashov
Exactly. I was thinking that a similar service would be a good addon
as an option to an ITSP.
Gavin.
On 18/07/2009, Steve Totaro stot...@totarotechnologies.com wrote:
On Sat, Jul 18, 2009 at 4:36 AM, Alan Lord (News)
alansli...@gmail.comwrote:
On 18/07/09 00:35, Gavin Henry wrote:
This has
Yeah, and the fxs port too.
On 18/07/2009, Alan Lord (News) alansli...@gmail.com wrote:
On 18/07/09 00:35, Gavin Henry wrote:
This has to be an Asterisk based appliance no?
http://www.truecall.co.uk/acatalog/trueCall_Features.html
I saw this on the TV the other night. Couldn't believe how
Gtalk has similar stuff I have heard.
I haven't checked it out since it was Grand Central but a friend was telling
me about it. I think I got an email about keeping my account active and I
did, so time to revisit that.
He had me forward his PRI DID/EXTEN to his Gtalk number.
I just wish I
Call Digium.
On Sat, Jul 18, 2009 at 3:43 PM, Luis Morales faston...@gmail.com wrote:
Dear asterisk users,
We want setup TE121 digium board:
Model: Digium TE121: VoiceBus technology allows the TE121 to use an
industry standard bus-mastering PCI Express interface.
On Saturday 18 July 2009 13:09:27 Steve Totaro wrote:
On Sat, Jul 18, 2009 at 1:50 PM, Tilghman Lesher wrote:
On Saturday 18 July 2009 11:57:32 Steve Edwards wrote:
Gabriel Ortiz Lour wrote:
Someone know how can I check for available members on a queue
Before I queue the call, so
Thank for your time.
Do you used chan_mobile with Elastix distribution successfully? If so, I
will consider the switch. I can't jump to another distribution easily
because I have a working environment that will make really hard the
migration.
On Sat, Jul 18, 2009 at 10:57 AM, Sasa Bobek
Yes, chan_mobile works great on Elastix. If the migration is complicated,
you may consider installing/testing it on an old computer.
On Sun, Jul 19, 2009 at 2:21 AM, Carlos Ruiz Diaz carlos.ruizd...@gmail.com
wrote:
Thank for your time.
Do you used chan_mobile with Elastix distribution
2009/7/17 Alan Lord (News) alansli...@gmail.com
On 17/07/09 17:20, Danny Nicholas wrote:
Not that this will really help, but in my CDR, I get this find of format
Xxx incoming_number s context caller_id incoming_tech/line
target_tech/line function command time1 time2 time3. It
No need to migrate, just have a chan_mobile server to hand the calls over
via SIP.
It is your cell phone network gateway
I like to separate functions to different boxen. Database on one, Asterisk
on another, TDM - SIP gateway on another, GUI/CRM somewhere else. Why not
have a Cell - SIP
On Sat, Jul 18, 2009 at 9:42 PM, Tilghman Lesher
tilgh...@mail.jeffandtilghman.com wrote:
On Saturday 18 July 2009 13:09:27 Steve Totaro wrote:
On Sat, Jul 18, 2009 at 1:50 PM, Tilghman Lesher wrote:
On Saturday 18 July 2009 11:57:32 Steve Edwards wrote:
Gabriel Ortiz Lour wrote:
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