Hi list,
This is comman issue when we use ODBC connection to get Database values. At
the 1st attempt connection is not connected but soon automatically connected
by aterisk. What is the real problem ? it this asterisk ODBC issue or else ?
-- Executing [_36899XX@incoming:1] Answer("SIP/100-
The output of that command is sane. I restarted Asterisk and things seem OK,
now. Not sure what happened, but I don't have time to ponder.
Thank you for your time, though.
Mike.
On Thursday 29 September 2011 11:17:24 pm Sam Govind wrote:
> Hey,
> Whats the output of command "features show" ?
Hey,
Whats the output of command "features show" ? on CLI ?
On Fri, Sep 30, 2011 at 1:51 AM, Mike Diehl wrote:
> Hi all.
>
> I could have sworn this working at one time...
>
> But it doesn't look like any of the functions provided by features.so is
> working for me. (one-touch monitoring, atte
I guess that was this variable like SPYGROUP which needs to be set for
specific extensions and then ask Chanspy to spy on that group. !!
On Thu, Sep 29, 2011 at 9:37 PM, A J Stiles
wrote:
> On Thursday 29 September 2011, Lyle McKarns wrote:
> > Hello Asterisk List!
> >
> > I have been asked to re
Hey Warren I thought that these are the complete CLI logs for one call. It
started like " == Using SIP RTP CoS mark 5" and from-internal priority-1
..So that seemed legit to me. Yeah I too suspect that dialing rules are not
being matched and thats why Gotoif's are failing.
On Thu, Sep 29, 2011 at
On Sun, Sep 25, 2011 at 8:26 AM, Mehmet Avcioglu wrote:
>
> Actually it doesn't say "AGI(async:script)" it says "AGI(async:agi)" and than
> continues further to setting up an AMI user so the script is executed through
> the manager interface?? Than it says "AGI(agi:async)".?? Well most
> import
Hello All,
I have a pstn line can have the local, STD and ISD
capabilities. My local number is 91471-2527XXX and the region is India. I
would like to use the number for all possible calls ( local, STD and ISD
call facilities to Land line and mobile phones) through an FXO card
con
Hello NaJIm,
Your zaptel.conf and zapata.conf files must match as to what channels
and signaling are in use.
See the examples at voip-info:
http://www.voip-info.org/wiki/view/Asterisk+config+zaptel.conf
On 9/29/2011 4:21 PM, NaJIm wrote:
> Hi Eric,
>
> This is the error messages I get I try to
Am I getting these error messages due to wrong configurations in my
zapata.conf. ??
I have got the following configurations in my zapata-channels.conf.
; Span 1: WCT1/0 "Wildcard TE122 Card 0" (MASTER) B8ZS/ESF RED
group=0,11
context=from-pstn
switchtype = national
signalling = pri_cpe
channel =
Hi Eric,
This is the error messages I get I try to load the module.
*CLI> module load chan_zap.so
[Sep 30 04:45:57] WARNING[5182]: pbx.c:2979 ast_register_application:
Already have an application 'ZapSendKeypadFacility'
== Parsing '/etc/asterisk/zapata.conf': Found
-- Registered channel 1,
Mike,
*What version of zaptel are you running?
*
My Zaptel version is - zaptel-1.4.12.1
*What zaptel commands have you tried?
*
None of the zaptel commands are working on my CLI. Its like on CLI, none of
the commands starting with zap are working. (When I give "zap+TAB key"
nothing shows up)
*
Try "module load chan_zap.so" in the CLI. You should see whatever errors are
generated.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of NaJIm
Sent: Thursday, September 29, 2011 5:52 PM
To: Asterisk Users Mail
On 9/29/2011 2:52 PM, NaJIm wrote:
> IRQ misses: 2
You are risking lots of audio problems if the card shares the IRQ with
any other device. Try and go in the BIOS and disable the other device or
change the IRQ it is using so that they do not conflict.
What version of zaptel are you running?
What
Hi,
We have got a new PRI card at one of our Office locations and now I need to
install the the device on a remote server. Is there any way to know if the
device is loaded already.
When I give " cat /proc/zaptel/* " it returns the following.
# cat /proc/zaptel/*
Span 1: WCT1/0 "Wildcard TE122
Hi all.
I could have sworn this working at one time...
But it doesn't look like any of the functions provided by features.so is
working for me. (one-touch monitoring, attended/blind transfer, etc)
I've (re)loaded features.so, as well as bridge_builtin_features.so.
The config file looks sane.
On 09/29/2011 12:22 PM, Tim Nelson wrote:
Greetings-
From time to time, I find myself working with (or customers working with) "dynamic
T1s". They are typically standard T1s that terminate to an Adtran device which
utilizes the channels for data (64kbps X 24) until a call is pushed inbound/ou
Greetings-
>From time to time, I find myself working with (or customers working with)
>"dynamic T1s". They are typically standard T1s that terminate to an Adtran
>device which utilizes the channels for data (64kbps X 24) until a call is
>pushed inbound/outbound on the circuit. One data channel
On Thursday 29 September 2011, Lyle McKarns wrote:
> Hello Asterisk List!
>
> I have been asked to record calls from specific agents, and I am having
> difficulty finding if this is possible, and if so, how exactly to do it.
Have a "recorded" context and an "unrecorded" context in your dialplan,
I'm digging this back up since the problem persists. I've been attempting to
figure out what's been going on and I'm at a stopping point again.
Even though I swore I checked it, it turns out the two cards and the ethernet
controller were all on the same IRQ. I moved the cards around so they ar
I would either use a gotoif to determine which queues get recorded or put
the recordable queues into a separate context (probably the simpler
solution).
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lyle McKarns
Sent: Thursday, Se
Hello Asterisk List!
I have been asked to record calls from specific agents, and I am having
difficulty finding if this is possible, and if so, how exactly to do it.
Some pertinent info:
We are using Asterisk 1.4.31 with T1/PRI/IAX/SIP calls coming inbound.
We have about 60 queues, but only a f
one thing i'm sure of? Honesty is a waste in this type of business.. all the
features youa re talking about .. have been offered and tested with customers..
the bottom like .. when a customer buys a 2$ calling card . he expects to make
a call and say his words and hangs up .. all those features
>>You will notice on the calling card shelves there are only a handful of
>>companies producing lots of different cards.
I have! That's what led me to CC for starters, then implementing a
more novel "startup" product. But. Regardless of all the corruption,
my goal is to offer something honest "TR
On Thu, 29 Sep 2011, Nick Khamis wrote:
Hello Jeff,
There will always be fierce competition, we are starting of with
prepaid for an obvious source of quick revenue, we will also be
rolling out a few more products in the next year.. It seems like they
LIE about their LD rates. A company in Aus
Hello Tarek,
Thanks again! I will look into TATA. Let's hope their comunication is
better than their cars ;).
Nick.
On Thu, Sep 29, 2011 at 11:38 AM, Nick Khamis wrote:
> Hello Jeff,
>
> There will always be fierce competition, we are starting of with
> prepaid for an obvious source of quick re
Hello Jeff,
There will always be fierce competition, we are starting of with
prepaid for an obvious source of quick revenue, we will also be
rolling out a few more products in the next year.. It seems like they
LIE about their LD rates. A company in Australia was
charged with this not too long ago
I have no knowledge of any commercial brand that operates in that region and
would offer DIDs in those countries.. AND your channel requirements are a bit
limited by technology in those regions... and VoIP termination Legislation in
those countries whether they allow Calling Cards business, all
Hello Tarek,
For channels, usually they charge per additional channels. I guess
being more explicit what it comes down to is:
* Reliable service
* Agressive Pricing
* For DIDs
- International Coverage
- Per Aditional Channel Pricing
* For SIP Termination
- Inte
On Thu, 29 Sep 2011, Nick Khamis wrote:
I should have mentioned we are interested in international long
distance. That will
be a big part of our business.
It sounds like you are intending to start a calling card company. Good
luck - the competition is fierce, and you will be competing aga
What does (international long) mean exactly? are you a calling cards company?
if so you should look for some company that will be charging you like 0.004
Cents per minute.. and you can find companies that will add more channels to
your DID.
Tarek Sawah
Information Technology Adviser
Inte
On Thu, Sep 29, 2011 at 7:51 AM, michael k wrote:
> Thanks for the update. but how do i resolve this issue ? can you help me
> please ?
>
You didn't provide a full CLI trace of the outgoing call, you only supplied
the hangup portion of the call. Please try again.
Also, what are the dialing rul
I should have mentioned we are interested in international long
distance. That will
be a big part of our business.
Cheers,
Nick.
On Thu, Sep 29, 2011 at 11:12 AM, Danny Nicholas wrote:
> They aren't everywhere, but we have had good experience with Voicepulse and
> their rate is typically less t
for some reason i don't think (unlimited incoming channels) fits with (dirt
cheap DIDs)
as you will be abusing their network .. they should start charging per minute
.. or you should pay for extra channels
several DID providers would offer you 20 channels per did at some rate of 9$ a
month pe
They aren't everywhere, but we have had good experience with Voicepulse and
their rate is typically less than $0.015 per minute.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis
Sent: Thursday, Septem
Very true... But there should be an equilibrium, the "relaiable
service, and aggressive pricing" comes to meet?
Guys please share your experiences.
Cheers,
Nick
On Thu, Sep 29, 2011 at 10:58 AM, C. Savinovich
wrote:
> In my professional opinion, the phrases "I don't want no Bull service" and
>
In my professional opinion, the phrases "I don't want no Bull service" and "I
want the cheapest service" are total contradictions. Down the road something is
not going to give.
C. Savinovich
On September 29, 2011 at 10:47 AM Danny Nicholas wrote:
> This belongs on the "commercial" list.
There is a commercial list!
Sorry about that
Nick.
On Thu, Sep 29, 2011 at 10:47 AM, Danny Nicholas wrote:
> This belongs on the "commercial" list.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
This belongs on the "commercial" list.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis
Sent: Thursday, September 29, 2011 9:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Hello Everyone,
We are looking for DID and SIP Termination service providers. Since
there are so many these days, can you
guy mention the BIG players that are supplying the rest of the little
guy? We are looking for the cheapest, and
scaleable infrastructure (i.e. unlimited channels for DID, and t
michael k wrote:
Thanks for the update. but how do i resolve this issue ? can you help me please
?
Can you PLEASE take this to the FreePBX support group?
It seems obvious to most that therein lies the problem
You are thinking you wish to dial out through the X100, but Asterisk is
attempti
Hi,
We are trying to get working Queue Stats, but it seems we get stucked.
Is anyone using this project for to track agent statistics ?
http://www.asteriskguru.com/tools/queue_stats.php
We are managed to install Queue Stats version 0.3 despite of the fact
that its using Zend framework, which i
Thanks for the update. but how do i resolve this issue ? can you help me
please ?
On Thu, Sep 29, 2011 at 1:00 PM, Sam Govind wrote:
> Actually its easier. I haven't worked on FreePBX lately so what I remember
> is here: You've created a Zap trunk. Focus on the Dial Rule - You can keep
> it em
ok thanks for your response i will try that and i will update you as soon as
i have any result
best regards
2011/9/29 A J Stiles
> (top-posting mess fixed the lazy man's way .)
>
> On Thursday 29 September 2011, salaheddine elharit wrote:
> > ok thanks it's work fine
> >
> > now i have one
(top-posting mess fixed the lazy man's way .)
On Thursday 29 September 2011, salaheddine elharit wrote:
> ok thanks it's work fine
>
> now i have one question please
>
> it's work fine when i call extension 222 but i want to call any number
> from my sip account 222 and the call hang up aft
Replace your phone number in place of ${EXTEN} and send it to your outgoing
provider.
with same dial argument.
On Thu, Sep 29, 2011 at 3:09 PM, salaheddine elharit <
salah.elharit...@gmail.com> wrote:
> ok thanks it's work fine
>
> now i have one question please
>
> it's work fine when i call e
ok thanks it's work fine
now i have one question please
it's work fine when i call extension 222 but i want to call any number from
my sip account 222 and the call hang up after 1 Min
for exemple i call my mobile phone 067XXX using my sip 222 (x-lite) and
the call hangup after 1 min
any he
This is a brilliant idea. How do I contribute my attackers to this
list?
Cheers
Andy
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert
Huddleston
Sent: 22 September 2011 16:11
To: 'Asterisk Use
Actually its easier. I haven't worked on FreePBX lately so what I remember
is here: You've created a Zap trunk. Focus on the Dial Rule - You can keep
it empty as well. Then you've created an outbound route its dial-rule is
important.
But the funny thing which I didn't mention before is that you've
Hi,
please use it then it will be helpfull for your application
exten => 66,1,Answer()
exten => 66,n,Set(CHANNEL(txgain)=20)
exten => 66,n,Set(CHANNEL(rxgain)=20)
exten => 66,n,Hangup()
On Thu, Aug 11, 2011 at 3:26 AM, Kevin P. Fleming wrote:
> On 08/03/2011 08:47 PM, Matt Riddell wrote:
>
>> On
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