balance reach for example 1 dollar!
Anyway?
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce
*Sent:* Monday, October 18, 2010 8:17 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re
Hi,
I have this on an Aastra phone:
Button 1:Login English Queue
Button 2:Login French Queue
Button 3:Logout both English and French
I am out of buttons and using only three buttons I want my third button to
be smarter. Currently the third button does a QueueRemoveMember to both
Turn on the voucher feature in System Settings and it will tell the user
right after the PIN authentication or CLID authentication that their balance
is below threshold and they should pay.
-Bruce
On Mon, Oct 18, 2010 at 2:35 PM, Baha @ SH i...@saudihome.com wrote:
Not sure if a2billing can be
...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce
*Sent:* Monday, October 18, 2010 12:46 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] a2billing
Turn on the voucher feature in System Settings
Jeff,
I suggest talking to your PSTN/VoIP provider. We had a large amount going
through TATA communications and have not accepted their word for payment
because they had a duty to not allow traffic if our credit went down to $1k
while the calls charged were actually more than that.
the
proper public IP of the device from the IP packet headers rather than the
SIP packets.
Thanks
On Sat, Oct 9, 2010 at 8:27 AM, Kevin P. Fleming kpflem...@digium.comwrote:
On 10/08/2010 10:16 PM, bruce bruce wrote:
I said previously, Asterisk receives packets like extens...@192.168.0.10
Glad to hear it helped you Dennison.
VPN is such a confusing beast to lots of people I think and hence the
responses to this thread were all sort of work around and sometimes
off-topic. It's also not well documented or maybe the feature is not widely
used within the Asterisk community. I think it
Kyle,
Got an empty response from you. Were you intending to give your feedback?
Regards,
Bruce
On Wed, Oct 6, 2010 at 8:10 PM, Kyle Kienapfel doctor.w...@gmail.comwrote:
On Wed, Oct 6, 2010 at 12:50 PM, bruce bruce bruceb...@gmail.com wrote:
Hi Guys,
This is such an annoying issue
8, 2010 at 3:32 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 10/06/2010 02:50 PM, bruce bruce wrote:
Hi Guys,
This is such an annoying issue whenever it comes up. The sender and
receive always receive the source public IP no matter what in the IP
packets but then SIP packets go
Hi Guys,
This is such an annoying issue whenever it comes up. The sender and receive
always receive the source public IP no matter what in the IP packets but
then SIP packets go out with something like 192.168.0.20.
In this instance, a Bell Canada DSL modem is installed and I saw the
SPA-2102
Thanks for the input guys.
So, the IP is resolved only when IPTABLES is loaded or reloaded. Therefore,
the best approach would be to ping the hostname every let's say 3 seconds
and see if the IP is still the same and if it is then move on, otherwise
update the iptables with the new IP address.
Hi Everyone
I think PAP2T supports DynDNS and other Dynamic DNS providers. I have a box
that needs to be secured at all times. Currently it's not connected to the
internet. If it were connected, I would have iptables block any and all
traffic from outside but I want a single device - Linksys
Hi Everyone,
Like always, here are IPs from China that try to hack an Asterisk server.
Can someone please explain what is happening or what the hacker is trying to
reach:
02/10/2010 11:10 SIP/113.105.152.51-00fb sip sip sip s ANSWERED 13
02/10/2010 11:10 SIP/113.105.152.51-00fe sip sip
, Oct 2, 2010 at 2:59 PM, jon pounder j...@inline.net wrote:
On 10/02/2010 02:56 PM, bruce bruce wrote:
Hi Everyone
I think PAP2T supports DynDNS and other Dynamic DNS providers. I have
a box that needs to be secured at all times. Currently it's not
connected to the internet. If it were
is not successful.
Or can I setup my own Dyndns server on the Asterisk server and have those
PAP2T units registered to it and then work it from there when their IPs
change?
Thanks
On Sat, Oct 2, 2010 at 3:32 PM, jon pounder j...@inline.net wrote:
On 10/02/2010 03:31 PM, bruce bruce wrote:
Hi,
Can
Can't I in my ip tables just accept the pap2t.dyndns.org if that is bind to
the PAP2T? do you think the devices comes in with it's external IP rather
than the dyndns domain?
Thanks
On Sat, Oct 2, 2010 at 3:43 PM, bruce bruce bruceb...@gmail.com wrote:
I was confusing the asterisk server side
Yeah, you are missing all :-)
Sorry, read the thread again.
On Sat, Oct 2, 2010 at 5:05 PM, sean darcy seandar...@gmail.com wrote:
On 10/02/2010 04:09 PM, bruce bruce wrote:
Can't I in my ip tables just accept the pap2t.dyndns.org
http://pap2t.dyndns.org if that is bind to the PAP2T? do
...@$523k4j98sd7fkjh324#@$832.dyndns.org
isn't that a security feature in itself?
Thanks
On Sat, Oct 2, 2010 at 4:32 PM, Roger Burton West ro...@firedrake.orgwrote:
On Sat, Oct 02, 2010 at 04:09:33PM -0400, bruce bruce wrote:
Can't I in my ip tables just accept the pap2t.dyndns.org
, 2010 at 10:57 AM, Benny Amorsen
benny+use...@amorsen.dkbenny%2buse...@amorsen.dk
wrote:
bruce bruce bruceb...@gmail.com writes:
Other than the price difference (2.5 is more expensive and can't find
many of the 1TB or so) is there any preference, advantage, or
disadvatage of chosing 2.5
Hi Everyone,
I am stack between two identical systems (2U Twin2, 4 nodes, SuperMicro)
servers that have the same exact specs except for HDDs. These nodes will all
either have Asterisk installed with CentOS or will have Asterisk install in
virtual environment.
Option 1: *12* x 3.5 HDD (3 HDDs per
Thanks for the detailed info. Problem was solved by including Server B
subnet as the localnet of the Server A (OpenVPN server) and setting each
extension NAT=NO.
Your points are good guides for future problem diagnoses.
Thanks again,
Bruce
On Thu, Sep 23, 2010 at 1:56 PM, Dave Platt
Thanks for the feedback. I thought about that but it's not an option for me
right now.
Any other ways folks?
Thanks
On Wed, Sep 22, 2010 at 4:06 AM, Roger Burton West ro...@firedrake.orgwrote:
On Wed, Sep 22, 2010 at 01:27:07AM -0400, bruce bruce wrote:
I have setup an OpenVPN tunnel between
I don't think it's an endpoint issue. I think the SIP packet headers get
over-written by the tunnel (openvpn) protocol.
Thanks,
Bruce
On Wed, Sep 22, 2010 at 9:49 AM, Paul Belanger paul.belan...@polybeacon.com
wrote:
On Wed, Sep 22, 2010 at 1:27 AM, bruce bruce bruceb...@gmail.com wrote
in the
Invite.
On Wed, 2010-09-22 at 01:27 -0400, bruce bruce wrote:
Hi Everyone,
I have setup an OpenVPN tunnel between Server A (running Asterisk) and
Server B suppling it's SIP Phones with DHCP pool of IPs.
So, the tunnel is established nicely and everyone can ping others.
sip
@sedwards.comwrote:
Un-top-posting...
On Wed, Sep 22, 2010 at 1:27 AM, bruce bruce bruceb...@gmail.com
wrote:
Any feed back is appreciated.
On Wed, Sep 22, 2010 at 9:49 AM, Paul Belanger
paul.belan...@polybeacon.com wrote:
Then configure you endpoints to use
...@polybeacon.com
wrote:
On Wed, Sep 22, 2010 at 1:46 PM, bruce bruce bruceb...@gmail.com wrote:
Thanks, but Carlos Chavez was right on point. This fixed the problem:
externip=123.123.123.123
localnet=192.168.100.0/255.255.255.0
nat=no in each extension.
So now I am confused, If you have
Hello,
This is what what I see after a Yum install asterisk16 asterisk16-config
freepbx:
Use of uninitialized value in string ne at
/var/www/html/panel/op_server.plline 4997.
Use of uninitialized value in substitution (s///) at /var/www/html/panel/
op_server.pl line 5439.
Use of uninitialized
Hi Everyone,
I have setup an OpenVPN tunnel between Server A (running Asterisk) and
Server B suppling it's SIP Phones with DHCP pool of IPs.
So, the tunnel is established nicely and everyone can ping others. sip show
peers shows the local subnet of the SIP Phones registered (192.168.100.0/24
).
Thanks guys. I wasn't able to collect enough SIP debug as the problem was
resolved as I was testing different configuration for the trunk. Probably a
change on the provider side.
John Novack: Unfortunately, it seems that this list has a non-stop list of
people who like to stir up things or try to
Hi Everyone,
Wondering if any of you folks ever had trouble using *Thomson
TG784http://www.w7forums.com/thomson-tg784-t1199.html
*DSL/Wireless Router/FXS ATA combo? I am looking to use this to connect
users from home to a hosted Asterisk PBX.
Any and all inputs are appreciated.
Thanks
--
Hi Everyone,
I see one long post on Cisco community forum where everyone including ISPs
are complaining about silence on FXS port, reboots, frozen state, etcof
WRP400. This is the a wireless router + 2 FXS combo box. I am looking to use
this for home user to connect to hosted Asterisk PBX.
I
Hi Everyone,
I have a provider whose DID used to come into the box just fine but recently
stopped working. Nothing has been changed on our end.
Here is what I get when doing sip set debug peer PROVIDER:
Sending to 123.123.123.123 : 5060 (no NAT)
That is ALL I am getting with sip debug
Hi Everyone,
My experience is only with the Canadian providers. What options/providers
are there in Dallas and Philadelphia other than Verizon when it comes to
internet? Something in the order of at least 10mbps down and up - I
understand that and higher bandwidths are easily available in USA due
1- I am interested in this as well. Looking into Proxmox as it provides a
nice interface (do you guys know of any other good one?)
2- Would the conference calls be fine as well? I understanding Asterisk
1.6.x uses a kernel timing source now a days so that ztdummy is not needed
anymore?
3- Would
Hi Everyone,
I have two servers as the following that are trunked with each other via
IAX2 trunk:
Server A:
Asterisk 1.4.21.2 (Elastix Flavor)
Server B (IP # 72.72.72.72):
Asterisk 1.6.2.0 (Vanilla)
Server B can place calls to Server A but when trying to place calls from
Server A to Server B
but there is in core set debug.
That's a petty.
-Bruce
On Thu, Sep 2, 2010 at 3:19 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Thu, Sep 02, 2010 at 02:21:11AM -0400, bruce bruce wrote:
Hi Everyone,
I have two servers as the following that are trunked with each other via
IAX2 trunk
Maybe dvossel can re-open issue # 16753 and fix the warning to show on iax2
debug as well along with core set debug like all other warnings. That way
it's straight forward. That ticket shouldn't have been closed without a fix.
On Thu, Sep 2, 2010 at 4:11 AM, bruce bruce bruceb...@gmail.com wrote
I am not interested in open source solutions. I want to know how much the
propriety systems cost in terms of licensing. Specially Avaya now a days per
extension. Exclusive or Inclusive of the hardware for 10 agents as noted.
Thanks
On Thu, Sep 2, 2010 at 8:18 AM, Muhammad Shomail Haider
Thanks Don for clarification.
There are lots of people on this list that hastily decide to answer without
even reading a post properly. I am sure they won't even read the follow-ups.
They just talk for the sake of talking. Sickens me!
Please note the subject line in my original post: To compete
:
bruce bruce wrote:
Hi Everyone,
I can connect the Jabra GN2124 + GN2100 (smart cord) to the Aastra 53i
receiver port and I get a tone. But when I connect it to the headset
port there is no tone. I am running firmware 2.4 and I can't seem to
find that DHSG, EHS or whatever the setting maybe
Hi Everyone,
There are a few things I like in OrderlyStats, specially some graph
presentations and the fact that if agent puts someone on HOLD or PAUSE it
shows fine.
1 -But I see a lot of similarities in pricing, descriptions, wording on both
sites. Were these same projects forked out? or is it
Hi Everyone,
I can connect the Jabra GN2124 + GN2100 (smart cord) to the Aastra 53i
receiver port and I get a tone. But when I connect it to the headset port
there is no tone. I am running firmware 2.4 and I can't seem to find that
DHSG, EHS or whatever the setting maybe called to enable to get
Bob,
Both ZanziIVR and Speechforge have similar look web pages. I guess you have
used one of those to get the speech going as this link:
http://scribblej.com/svn/ probably is not the full thing.
These seem like practical project. Thanks for pointing out. This is what I
was looking for.
Now
Thanks guys. A lot of info here :-)
I am wondering if anyone followed this and it was working for them:
http://scribblej.com/svn/
???
I am not looking for anything fancy. The basic yes, no, dialing a
number, asking for agent, etc...out of which probably the hardest is a 10
digit number to be
Hi Everyone,
Has anyone got any opensource speech recognition software to work with
Asterisk? Please only list WORKING ones. Not the theoretically should work
ones!
Thanks
--
_
-- Bandwidth and Colocation Provided by
=no in your peer configuration
Regards
On Wed, Aug 11, 2010 at 4:28 AM, bruce bruce bruceb...@gmail.com wrote:
Hello Everyone,
I am trying to diagnose issue with my IAX2 extension not working.
When I have iax2 set debug on all I see is this:
*Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000
Hi Everyone,
Can anyone share their experience with Motorola Canopy solution deployment
and Asterisk? Is this a good fit?
Thanks,
Bruce
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk?
Hi Everyone,
Just trying to connect the Zoiper Communicator to connect to Asterisk which
is behind Pfsense. Here is what I get at debug and it doesn't register.
Error code 16. Can someone please let me know their firewall, NAT, outbound
1-to-1 pfsense settings as it seems to me I am doing
*
On Wed, Aug 11, 2010 at 12:33 AM, Faisal Hanif fai...@vopium.com wrote:
read the value of var ${HANGUPCAUSE} next line to dial command.
Regards,
Faisal Hanif
*VoIP Manager
***Vopium A/S
On 8/10/2010 9:51 PM, bruce bruce wrote:
Hi Everyone
Asterisk 1.4.33 is running with Sangoma
Hi Everyone
Asterisk 1.4.33 is running with Sangoma/Dahdi for analogue lines to Bell
Canada.
User claims that call hangup without any interferance to the phone set.
Is there ANYWAY to find out which party hang-up the call or if the call was
cut-off due to other reasons?
I checked the
Hello Everyone,
I am trying to diagnose issue with my IAX2 extension not working.
When I have iax2 set debug on all I see is this:
*Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
REGREQ *
* Timestamp: 3ms SCall: 00130 DCall: 0 [64.229.229.111:64823]*
*
I agree but the mentioned software is not opensource.
My conditions clearly included opensource.
On Tue, Aug 3, 2010 at 12:35 AM, Nick Brown n...@ipera.com.au wrote:
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce
with Elastix 1.6
Regards
On Mon, Aug 2, 2010 at 11:26 PM, bruce bruce bruceb...@gmail.com wrote:
Hi Everyone,
Sorry, if it's not directly related to Asterisk. Some of people on this
list might have PBX deployed for their clients. What software do you use to
invoice them so the invoice looks like
Yep, I seen that. That is probably the closet thing but looking at he
interface it makes me not try to install it. Maybe too complicated. I
wouldn't want to send customer the whole CDRs but rather a nice Bill like
the telco sends out.
I am currently toying with NCH Invoicing. Those guys make a
...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce
*Sent:* Thursday, July 29, 2010 22:36
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Aastra phones occasionally show No Service -
Is there any network setting
Hi Everyone,
Sorry, if it's not directly related to Asterisk. Some of people on this list
might have PBX deployed for their clients. What software do you use to
invoice them so the invoice looks like a proper telecom invoice maybe?
Prefer:
-opensource with Windows binary available.
-able to
Maybe good but the first look brought me to a Pay version. Doesn't satisfy
the opensource condition.
thanks,
On Mon, Aug 2, 2010 at 2:39 PM, Jeff LaCoursiere j...@sunfone.com wrote:
On Mon, 2010-08-02 at 14:26 -0400, bruce bruce wrote:
Hi Everyone,
Sorry, if it's not directly related
Sorry, I am not familiar with them.
Wondering if any full package system out there does the job.
Thanks
On Mon, Aug 2, 2010 at 2:55 PM, Gordon Henderson
gordon+aster...@drogon.netgordon%2baster...@drogon.net
wrote:
On Mon, 2 Aug 2010, bruce bruce wrote:
Hi Everyone,
Sorry, if it's
. At best, I can think of
a cable or two jacked improperly into the patch panel and that's all which
MAYBE the cause for failing of DNS.
Thanks,
Bruce
On Fri, Jul 30, 2010 at 12:03 PM, bruce bruce bruceb...@gmail.com wrote:
DNSMasq has always been enabled. It's only one check box and if I didn't
2 users. So, it's probably never used as a free version as probably there
are no 2 seat call centers that can survive this economy. But, it should
great for testing.
On Sat, Jul 31, 2010 at 10:28 AM, Leif Madsen
leif.mad...@asteriskdocs.orgwrote:
On 7/30/2010 5:49 AM, Lenz Emilitri wrote:
, Jul 28, 2010 at 4:54 PM, bruce bruce bruceb...@gmail.com wrote:
Hmmwhat about call waiting?
You mean, when a call comes in on that specific line, it generate two
beep
tones and hence the system hangs up thinking it's end of the call?
Interesting!!!
If it is call-waiting do I have
for free.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-07-27 6:12 PM, bruce bruce bruceb...@gmail.com wrote:
:-) I knew someone would bring up FreePBX. I have FreePBX installed and
it's not good for Queues at all. It's using the reporting tool from Areski
and Areski has recently
Adria,
How can I build a dns cache into my lan? I am using a Linksys 48 port POE
switch and running a micro DD-WRT firmware on a linksys router.
Gareth,
I think the registration time is part of the reason. I have lowered it less
than 10 seconds.
Thanks
On Fri, Jul 30, 2010 at 8:21 AM, Adrià
DNSMasq has always been enabled. It's only one check box and if I didn't
have it enabled phones won't work. So, that is set. Any other suggestions?
including things regarding DNSMasq?
Thanks
On Fri, Jul 30, 2010 at 11:04 AM, Dave Cotton dcot...@linuxautrement.comwrote:
On 30/07/10 16:15, bruce
of the 26th at 9:22 the server was restarted because it was
un-reachable from outside and hence the restart log but where is the 24th,
and 25th?
Thanks,
Bruce
On Thu, Jul 29, 2010 at 9:10 AM, Paul Belanger paul.belan...@polybeacon.com
wrote:
On Wed, Jul 28, 2010 at 9:06 PM, bruce bruce bruceb
Hi Everyone,
I am running DD-WRT on a router that feeds about 30 Aastra 6753i. The phones
occasionally go into No Service mode. The POE switch doesn't seem to be
the problem as it's tested fine. I think the router sometimes gives up and
comes back quickly. Or something of that nature. However,
as there will be no
logs. Even MS Blue Screed Of Death does a better job of logging at instances
like this :-(
On Thu, Jul 29, 2010 at 10:13 PM, Lyle Giese l...@lcrcomputer.net wrote:
Lyle Giese wrote:
bruce bruce wrote:
I am not sure why it would be sleeping. I have never dealt with putting a
linux server
Hi Guys,
I am getting a complain that call on analogue lines (Sangoam A400D) drops
all of a sudden. Here is what I see in logs:
[Jul 28 15:49:08] DEBUG[21438] dsp.c: ast_dsp_busydetect detected busy,
avgtone: 75, avgsilence 135
[Jul 28 15:49:08] VERBOSE[21438] logger.c: -- Executing
:
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce
*Subject:* [asterisk-users] Why do Zaptel calls drop all of a sudden?
Couldbusy detect be the problem?
I am getting a complain that call on analogue lines (Sangoam A400D
Furthermore, these are lines in Hunt, so, I am not sure if Call-Waiting is
turned ON on these lines at all. But it's definitely an interesting idea.
On Wed, Jul 28, 2010 at 5:54 PM, bruce bruce bruceb...@gmail.com wrote:
Hmmwhat about call waiting?
You mean, when a call comes
Hi Everyone,
This is probably more related to Linux than to Asterisk. Analogue channels
on a system were un-responsive on Monday morning. Apparently something
happened over the weekend and the router went off or it lost it's DSL
connection.
[Jul 23 22:50:01] VERBOSE[12437] logger.c: --
that are neat in
interface or useful in features.
I guess no one else has any thoughts on this? Maybe there is none available?
Thanks,
Bruce
On Tue, Jul 27, 2010 at 11:41 AM, David Backeberg dbackeb...@gmail.comwrote:
On Mon, Jul 26, 2010 at 11:34 PM, bruce bruce bruceb...@gmail.com wrote:
I seem
Hi Guys,
I seem to not be able to find any good open source Asterisk Queue Analyzer
and Asterisk Log Analyzer on the web.
The Asterisk Queue Analyzer is to serve as the graphic tool for call center
or pbx admins. It will pull the info in queue.log and in MySQL asterisk CDR
to create a graphic
, bruce bruce bruceb...@gmail.com wrote:
Any help is appreciated.
Are you explicitly calling Hangup() within your dial-plans?
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com
You can also use Ethernet Over Power Lines solution or wireless :-)
On Fri, Jul 23, 2010 at 8:55 AM, David Backeberg dbackeb...@gmail.comwrote:
On Fri, Jul 23, 2010 at 8:46 AM, Matt mhop...@gmail.com wrote:
It's not necessarily this simple. There is an approximately 50-75foot
cable
run
Well, what about PRI? Why should this stay on? Isn't the native bridge just
a bridge channel that should go down automatically if the actually Dahdi/ZAP
channel is down and there are no SIP channels on either?
Thanks,
Bruce
On Fri, Jul 23, 2010 at 5:09 PM, Tzafrir Cohen
I am having this issue with PRI. But I do not use conference rooms. Our
system is a simple queue and extensions.
-Bruce
On Fri, Jul 23, 2010 at 6:13 PM, Maurizio Faccio adinet
mauf...@adinet.com.uy wrote:
You're right but it do not detect that I hungs on my side of the line.
I think that in
The Aastra 53i draws only 2 Watts from a Linksys 24 port POE switch. 25
phones is around 55 Watts.
-Bruce
On Thu, Jul 22, 2010 at 5:16 PM, Andrew Latham lath...@gmail.com wrote:
The Snom 360 phone in front of me draws 4w...
~
Andrew lathama Latham
lath...@gmail.com
* Learn more about
Hi Everyone,
Using a PRI with Sangoma A101D and Asterisk 1.4.2.x.
I notice that occasionally after a call is disconnected and both the phone
devices and the the channel is down but the bridge stays open for hours.
Channel Location State Application(Data)
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 supp...@drdos.info wrote:
bruce bruce wrote:
Hi Everyone,
If I
. :)
On Wed, Jul 14, 2010 at 12:39 AM, bruce bruce bruceb...@gmail.com wrote:
Thanks for the input guys. For other refrence, a CyberData Voip Amplifier
which supplies 10 Watt to each of the two bogen 30 Watt speakers did the
job
for a 35, square feet warehouse with environmental noise level
Hi Everyone,
If I receive a call on a ZAP line and pickup the call and drag and drop it
(by mouse) into a Parking Lot through FOP, it just hangs up. Is this feature
supported by FOP?
Thanks,
Bruce
--
_
-- Bandwidth and
I am stuck with the same problem but I have used asterisk yum repository and
it worked by itself without me worrying for kernel stuff.
However, I need to install speex codec and now I am stuck as it doesn't get
picked up by the yum asterisk install somehow. I have lib speex and speex
already
Hi Guys,
Running Asterisk v1.4.26 (Elastix flavour) and have Aastra 9480i, 6757i, and
6730i, but none of them indicate the voic-email. Where should I look for
trouble to find the root issue for MWI?
Thanks,
--
_
-- Bandwidth
Johnson stevej...@gmail.com wrote:
On Wed, Jul 14, 2010 at 10:04 AM, bruce bruce bruceb...@gmail.com wrote:
Hi Guys,
Running Asterisk v1.4.26 (Elastix flavour) and have Aastra 9480i, 6757i,
and
6730i, but none of them indicate the voic-email. Where should I look for
trouble to find the root
Hi Everyone,
Using Elastix (FreePBX + Asterisk 1.4.2x combination) with Aastra phones,
how can one receive distinctive ring tones for INTERNAL calls ONLY?
Even though FreePBX Inbound has an option for Alert_INFO but that doesn't
work when the call comes into an IVR or Queue. The calls has to go
Thanks for the input but that won't be good because people are not going to
remember two extensions for one person.
The sip header should be able to carry alert_info to internal extensions
really easily. Anyone else got a thought?
Thanks again,
On Wed, Jul 14, 2010 at 5:44 PM, Ira
Hi Everyone,
I have done yum install speex libspeex-devel speex-devel and it was
succesful on CentOS. I then tried yum install asterisk16 asterisk16-addons
asterisk16-configs but core show translation doesn't show speex loaded.
Is there a way to or an option that I can append to the asterisk
up 3 facing one way and 2 the other. You
can get double horn speakers which will face 2 sides. I wouldn't mount
them on the wall specifically not so low as fork lifts and what not
will damage them.
On Mon, Jul 12, 2010 at 2:17 PM, bruce bruce bruceb...@gmail.com wrote:
Well, these are horn
Hi Everyone,
I have done some php coding to come up with my own FollowME module for
FreePBX. The need for this has some security considerations behind it.
This is what my code does at core:
$sql=REPLACE INTO findmefollow(grpnum, strategy, grptime, grppre, grplist,
annmsg_id,postdest, dring,
5 ft
has never killed anyone, but this really depends on the power of the
speaker, I usually deal with 70v speakers tapped at 16 or 8 watts
depending on how many speakers I put on one amplifier and the output
wattage of that amplifier.
On Fri, Jul 9, 2010 at 2:29 AM, bruce bruce bruceb
tnel...@rockbochs.com wrote:
- bruce bruce bruceb...@gmail.com wrote:
Hi Everyone,
I have done some php coding to come up with my own FollowME module for
FreePBX. The need for this has some security considerations behind it.
This is what my code does at core:
$sql=REPLACE
and I'm guessing MySQL at
that.
Next, you seem to be accepting user input and not sanatizing it. DANGER
WILL ROBINSON!!!
This is bad, because it leaves you open to something known as a SQL
injection attack.
Now, as to syntax:
On Sat, Jul 10, 2010 at 12:07 AM, bruce bruce bruceb
:21 AM, bruce bruce bruceb...@gmail.com wrote:
Thank you for the amazing reply. First few lines of your e-mail was EXACTLY
getting me to where I made a mistake. I guess I didn't take the () and ' '
at their face value and was looking somewhere else for the problem.
For sanatizing you mean
,
Bruce
On Sat, Jul 10, 2010 at 1:41 PM, Gerald A geraldabli...@gmail.com wrote:
Hi Bruce,
On Sat, Jul 10, 2010 at 11:12 AM, bruce bruce bruceb...@gmail.com wrote:
Further to my last post, I added this to santize. I also created a new
mysql user with access to only findmefollow portion
You need read():
http://www.voip-info.org/wiki/view/Asterisk+cmd+Read
http://www.voip-info.org/wiki/view/Asterisk+cmd+ReadIt's as easy as:
exten = s,n,Read(variable,,11)
exten = s,n,NoOp(${variable})
Above will take up to 11 digits input by user and will display it back in
NoOP on Asterisk CLI.
) *
*
*
* {*
echo Number passed sanitization;
}
What do you think? :-)
-Bruce
On Sat, Jul 10, 2010 at 2:17 PM, bruce bruce bruceb...@gmail.com wrote:
Thanks again. Apparently all POST variables come through as strings. The
function you pointed out is I think already built
-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce
*Sent:* Saturday, July 10, 2010 9:30 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] How can get user inputs from called party
after dial?
You need read():
http://www.voip-info.org/wiki
For dial you do this:
[first-Dialplan]
exten = s,1,Answer
exten = s,n,Dial(SIP/provider/111222)
exten = s,n,Playback(Welcome)
exten = s,n,Read(numb,,10)
exten = s,n,NoOp(${numb})
-Bruce
On Sat, Jul 10, 2010 at 2:51 PM, bruce bruce bruceb...@gmail.com wrote:
You need to do some reading
I was under the impression that he is new to Asterisk. No need to fuss.
Hence the :-)
On Sat, Jul 10, 2010 at 3:35 PM, Steve Edwards asterisk@sedwards.comwrote:
On Sat, 10 Jul 2010, bruce bruce wrote:
You need to do some reading :-)
Now that is funny -- maybe you could take your own
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