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 to login using XLite Dialer and was able to call out to my SIP Trunk also. Now how can I remove the IVR Prompt... Meaning from my XLite dialer I want to dial directly and let A2Billing do the billing part. Right now is something like when I dial any number from XLite, A2Billing script is invoked and it will announce "You have XXX amount, please enter the number you wish to call followed by #". And then I have to enter the number again and then the call is initiated... Its kinda annoying to do that every time you want to call. Is there anyway to modify config some where, so it will do the billing in background when the phone call is hangup. Cheers, Nitesh
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 documentation, I created the RateCard, SIP Trunks, > and SIP Customers. I was also able to login using XLite Dialer and was > able to call out to my SIP Trunk also. > > Now how can I remove the IVR Prompt... Meaning from my XLite dialer I > want to dial directly and let A2Billing do the billing part. Right now > is something like when I dial any number from XLite, A2Billing script is > invoked and it will announce "You have XXX amount, please enter the > number you wish to call followed by #". And then I have to enter the > number again and then the call is initiated... Its kinda annoying to do > that every time you want to call. > > Is there anyway to modify config some where, so it will do the billing > in background when the phone call is hangup. >Yes, is possible using the a2billing.conf file in the right way. I don't have the v1.3 installed, but in the previous release 1.2.3 you must have to modify : use_dnid=YES number_try=1 say_balance_after_auth=NO say_balance_after_call=NO say_rateinitial=NO say_timetocall=NO Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : gsalas@manta.telconet.net www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting
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 Guillermo Salas M. wrote:> 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 are failing... As soon as the call hits Asterisk, >> A2Billing script runs and ask for PIN Number... I checked the context >> for my DID it shows "context=a2billing" and under sip.conf >> "context=a2billing". >> >> If I change the default context under sip.conf to "context=default", >> then the calls are failing... meaning I do not get any response back, >> but on *CLI debug show that its failing to look for the DID number. >> Well, I know this is due to my DID is in "context=a2billing". >> >> Anyone can suggest how can I fix this... I want to ring my incoming to >> that ATA which has DID assigned. >> > > You need to setup the DID on the DID section of a2billing. > > First create one SIP/IAX2 configuration for your DID provider and assign > the context a2billing-did. > > Later on the DID section, add the DID Provider, add the DID number and > asign one destination to the DID (your ata card number) or any SIP > extension enabling the "voip call" radius button. > > Try it. > > Regards, > > >
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 a2billing-did context : [a2billing-did] exten => _X.,1,NoOp,${CALLERID(all)} exten => _X.,2,DeadAGI(a2billing.php|1|did) exten => _X.,3,Hangup() This will be the context for your DID provider and not for your customers. Check this link for more information: http://forum.asterisk2billing.org/viewtopic.php?t=1784 Cheers! -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : gsalas at manta.telconet.net www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting
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 it back. Same like dialing from one extension > number to another extension. > > My SIP-Friends are using US DID numbers and I would like to keep the > local calls within the network. > > Right now when I try to call other SIP-Friend, I get a message saying > "The number you have dialer is currently not available"... while the > SIP-Friend is registered. >Try dialing the number 9 before the sip/iax2 friend number. Regards,> Cheers, > Nitesh-- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : gsalas at manta.telconet.net www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting