Hello All,
Stable release of A2Billing has solved most of my problems and so far
everything is OK...
Right now the only problem I am facing with my SIP clients are: -
- Three-way Calling
Three-way calling works fine, but when SIP client hangs up the
call, the other two channels are s
In a2billing just change the 9 to what you need it is right in the conf
file.
Best regards,
Al Bochter
Bochter Services
--
Need to call me use our web phone at the link below
http://www.bochterservices.com/voip/iaxphone.php?cn=250
---
Thanks everyone for the input...
In real world we can not ask the customers to dial 9, if they want to
call another SIP user... and trust me its confusing for a customer
also... meaning when to dial 9 and when to not...
We have a custom proprietary system which does this part very well...
Befo
Not so.
The point of BUTTONS and LIGHTS is for users. Remember them?
Press a button to answer a call under a flashing light.
Press a button to grab a call on hold under a light flashing at a
different rate
Press a button to place an external call.
Too many more reasons to enumerate.
Also, NEV
Given that Asterisk is modeled on, in the telephone industry, an
obsolete PBX design, without many of the modern day hybrid features, and
only recently has any effort been made to provide buttons and lights for
"lines" ( Is that yet working in 1.4??) one would have to do some very
careful numbe
What is the point of line lights on the phone?
The lights are so you would know when the KSU is out of lines.
With Asterisk if the system is setup right it should never run out of
lines to use.
Best regards,
Al Bochter
Bochter Services
If you do not dial 9 then there will be a conflict between internal
extensions and external phone numbers.
How would Asterisk determine if you are dialing extension 458 or
458-1234? It cannot. Asterisk would have to wait for a timeout when
dialing the extensions. If you force users to always
Thanks man,
Is there any other way without dialing 9... it will be kinda pain for a
customer to dial 9 every time and plus they need to know also...
Is there any intelligent way to identify? if its a local SIP then don't
route to Trunk else route to Trunk.
Cheers,
Nitesh
Guillermo Salas M. w
On Tue, 2007-06-19 at 09:36 -0400, Nitesh Divecha wrote:
> Thanks man...
>
> So far everything worked as expected...
>
> How can I make internal calls stay within the PBX. For example, when
> one
> SIP-Friend tries to call another SIP-Friend without sending the call
> out
> on Trunk and receive
Thanks man...
So far everything worked as expected...
How can I make internal calls stay within the PBX. For example, when one
SIP-Friend tries to call another SIP-Friend without sending the call out
on Trunk and receive it back. Same like dialing from one extension
number to another extension
Strange...
Got it working now... I can receive incoming call...
Changed following parameters in additional_a2billing_sip.conf of the DID
to: -
qualify=yes
canreinvite=no
Cheers,
Nitesh
Guillermo Salas M. wrote:
> On Fri, 2007-06-15 at 17:42 -0400, Nitesh Divecha wrote:
>
>> When I call fr
Here is my "sip show peers"
hyperion*CLI> sip show peers
Name/username HostDyn Nat ACL Port Status
2486543210/2486543210 86.14.22.128 D N 61547LAGGED
(66 ms)
Now here is the catch, before it used to show the status OK but now its
showing LAGGE
On Fri, 2007-06-15 at 17:42 -0400, Nitesh Divecha wrote:
> When I call from my cell to the above DID, it hits on the Asterisk and
> I
> see A2Billing trying to call SIP/2486543210, but it fails because
> Asterisk says "Unable to create channel of type 'SIP' (cause 3 - No
> route to destination)
2486543210 is my SIP-Friend which I created manually and associated with
one of the card number.
My ATA is registered to Asterisk using the about DID Number.
So I want when I call the above number, it should ring on the ATA.
When I call from my cell to the above DID, it hits on the Asterisk and I
Thanks man... That really helped me to move couple of steps. Now I see the
incoming calls are going in proper direction... I know I am still missing a
small piece here... I did ADD the Destination as a SIP/2486543210, assigned
the card number, enabled VOIP_CALL, and enabled Active.
When I dial th
On Fri, 2007-06-15 at 17:01 -0400, Nitesh Divecha wrote:
> Thanks man... That really helped me to move couple of steps. Now I see
> the incoming calls are going in proper direction... I know I am still
> missing a small piece here... I did ADD the Destination as a
> SIP/2486543210, assigned the car
On Fri, 2007-06-15 at 14:20 -0400, Nitesh Divecha wrote:
> You said change the context for SIP Customers to
> "context=a2billing-did", do I have to create this context or
> A2Billing
> will generate by itself?
>
The a2billing package comes with some examples, you must have to create
the a2bill
Thanks Man...
Do I need to change my context in sip.conf to "context=a2billing" or
should I leave it to "context=default"?
You said change the context for SIP Customers to
"context=a2billing-did", do I have to create this context or A2Billing
will generate by itself?
Cheers,
Nitesh
Guiller
On Fri, 2007-06-15 at 12:19 -0400, Nitesh Divecha wrote:
> Thanks everyone,
>
> OK, I got everything working... I manage to create a SIP Customer with a
> real DID number and configured an ATA with the DID number. ATA can login
> and can make calls out without any issues.
>
> But incoming calls
Thanks everyone,
OK, I got everything working... I manage to create a SIP Customer with a
real DID number and configured an ATA with the DID number. ATA can login
and can make calls out without any issues.
But incoming calls are failing... As soon as the call hits Asterisk,
A2Billing script ru
That was easy... Thanks a million man...
Dunno what I was thinking and went too far writing custom scripts...
Cheers,
Nitesh
Guillermo Salas M. wrote:
On Thu, 2007-06-14 at 14:46 -0400, Nitesh Divecha wrote:
Hello All,
I got one quick question on A2Billing.
Specs: -
- A2Billing v1.3
- O
On Thu, 2007-06-14 at 14:46 -0400, Nitesh Divecha wrote:
> Hello All,
>
> I got one quick question on A2Billing.
>
> Specs: -
> - A2Billing v1.3
> - OS CentOS 4.5
> - Asterisk 1.2
> - Zaptel 1.2
>
> Did the installation and everything is working as it suppose to...
>
> Using the A2Billing docum
Hello All,
I got one quick question on A2Billing.
Specs: -
- A2Billing v1.3
- OS CentOS 4.5
- Asterisk 1.2
- Zaptel 1.2
Did the installation and everything is working as it suppose to...
Using the A2Billing documentation, I created the RateCard, SIP Trunks,
and SIP Customers. I was also able
23 matches
Mail list logo
