Re: [asterisk-users] Que on A2Billing
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 still active and talking. - Call Forwarding I created the context for *72, *73, *90, *91, *52, and *53. SIP client can enable and disable but it never works because a2billing.php will time out and hang up the SIP channel. - Voice Mail I created the context for voice mail, but the calls will never go to voice mail because a2billing.php after 60 sec will hang up the channel. No doubt A2Billing is a great software, but the above features are also essential for home SIP users... Anyone can show or share their setup if they have implemented the above features with A2Billing Software. Cheers, Nitesh Al Bochter wrote: 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 -- Can you WIN gold today? Click on the link and see. http://www.bochterservices.com/?t=USbill_email -- Need cash we buy silver and gold -- Nitesh Divecha wrote: 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... Before it sends the call on a Trunk it will check the DID, if it exists within the local system. If it does then it will just use IP to IP call, else send the call to Trunk... I think its possible to do this by creating some basic dial plans... Same like creating local extensions. Cheers, Nitesh John Novack wrote: 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 number parsing to not use a trunk digit. If every phone in the system had buttons and lights representing external connections and internal connections on other button(s) ( intercom ) this wouldn't be an issue. Most legacy systems have been able to do this for the last 20 years or so. John Novack Nitesh Divecha wrote: 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. wrote: 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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 000750-2, 06/19/2007 - 6/19/2007 4:22:13 PM
Re: [asterisk-users] Que on A2Billing
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. Cheers, Nitesh ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Que on A2Billing
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 : [EMAIL PROTECTED] 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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Que on A2Billing
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. wrote: 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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Que on A2Billing
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 dial 11 digits (1-504-555-1212) for all calls, then you just have to make sure there are no extensions starting with 1 and you won't have problems. This is, of course, for NANPA. If your country uses a different numbering plan then the details will be different, but the idea is the same. Nitesh Divecha wrote: 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. wrote: 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. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Que on A2Billing
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 -- Can you WIN gold today? Click on the link and see. http://www.bochterservices.com/?t=USbill_email -- Need cash we buy silver and gold -- John Novack wrote: 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 number parsing to not use a trunk digit. If every phone in the system had buttons and lights representing external connections and internal connections on other button(s) ( intercom ) this wouldn't be an issue. Most legacy systems have been able to do this for the last 20 years or so. John Novack Nitesh Divecha wrote: 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. wrote: 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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 000750-2, 06/19/2007 - 6/19/2007 1:47:44 PM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Que on A2Billing
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 number parsing to not use a trunk digit. If every phone in the system had buttons and lights representing external connections and internal connections on other button(s) ( intercom ) this wouldn't be an issue. Most legacy systems have been able to do this for the last 20 years or so. John Novack Nitesh Divecha wrote: 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. wrote: 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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Que on A2Billing
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, NEVER say never Think out of the ( Asterisk ) box! John Novack Al Bochter wrote: 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 -- Can you WIN gold today? Click on the link and see. http://www.bochterservices.com/?t=USbill_email -- Need cash we buy silver and gold -- John Novack wrote: 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 number parsing to not use a trunk digit. If every phone in the system had buttons and lights representing external connections and internal connections on other button(s) ( intercom ) this wouldn't be an issue. Most legacy systems have been able to do this for the last 20 years or so. John Novack Nitesh Divecha wrote: 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. wrote: 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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 000750-2, 06/19/2007 - 6/19/2007 1:47:44 PM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Que on A2Billing
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... Before it sends the call on a Trunk it will check the DID, if it exists within the local system. If it does then it will just use IP to IP call, else send the call to Trunk... I think its possible to do this by creating some basic dial plans... Same like creating local extensions. Cheers, Nitesh John Novack wrote: 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 number parsing to not use a trunk digit. If every phone in the system had buttons and lights representing external connections and internal connections on other button(s) ( intercom ) this wouldn't be an issue. Most legacy systems have been able to do this for the last 20 years or so. John Novack Nitesh Divecha wrote: 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. wrote: 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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Que on A2Billing
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 -- Can you WIN gold today? Click on the link and see. http://www.bochterservices.com/?t=USbill_email -- Need cash we buy silver and gold -- Nitesh Divecha wrote: 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... Before it sends the call on a Trunk it will check the DID, if it exists within the local system. If it does then it will just use IP to IP call, else send the call to Trunk... I think its possible to do this by creating some basic dial plans... Same like creating local extensions. Cheers, Nitesh John Novack wrote: 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 number parsing to not use a trunk digit. If every phone in the system had buttons and lights representing external connections and internal connections on other button(s) ( intercom ) this wouldn't be an issue. Most legacy systems have been able to do this for the last 20 years or so. John Novack Nitesh Divecha wrote: 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. wrote: 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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 000750-2, 06/19/2007 - 6/19/2007 4:22:13 PM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Que on A2Billing
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. 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 - 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, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Que on A2Billing
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, -- 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 : [EMAIL PROTECTED] 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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Que on A2Billing
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, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Que on A2Billing
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 : [EMAIL PROTECTED]
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
___
--Bandwidth and Colocation provided by Easynews.com --
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Re: [asterisk-users] Que on A2Billing
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 card number, enabled VOIP_CALL, and enabled Active. 2486543210 is your card number? When I dial the DID number, on the *CLI it shows the following: - a2billing.php|1|did: bug -- AGI Script Executing Application: (DIAL) Options: (SIP/2486543210|60|HL(360:61000:3)) -- Limit Data for this call: -- - timelimit = 360 -- - play_warning = 61000 -- - play_to_caller= yes -- - play_to_callee= no -- - warning_freq = 3 -- - start_sound = UNDEF -- - warning_sound = timeleft -- - end_sound = UNDEF Destroying call '[EMAIL PROTECTED]' Jun 15 16:41:34 NOTICE[15346]: app_dial.c:1069 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) I think that 2486543210 is not a customer, card number or SIP/IAX2 friend, maybe is PSTN number. To redirect the call to any PSTN number you must need to set voip call to inactive and set the destination number to 2486543210. I bet I am missing something in extension.conf correct? I dont see any examples in my package. The context is fine don't worry about it. Any suggestion... Thanks once again... 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 : [EMAIL PROTECTED] 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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Que on A2Billing
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 the DID number, on the *CLI it shows the following: -
a2billing.php|1|did: bug
-- AGI Script Executing Application: (DIAL) Options:
(SIP/2486543210|60|HL(360:61000:3))
-- Limit Data for this call:
-- - timelimit = 360
-- - play_warning = 61000
-- - play_to_caller= yes
-- - play_to_callee= no
-- - warning_freq = 3
-- - start_sound = UNDEF
-- - warning_sound = timeleft
-- - end_sound = UNDEF
Destroying call '[EMAIL PROTECTED]'
Jun 15 16:41:34 NOTICE[15346]: app_dial.c:1069 dial_exec_full: Unable to
create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
I bet I am missing something in extension.conf correct? I dont see any
examples in my package.
Any suggestion... Thanks once again...
Cheers,
Nitesh
On 6/15/07, Guillermo Salas M. [EMAIL PROTECTED] wrote:
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 : [EMAIL PROTECTED]
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
___
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___
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Re: [asterisk-users] Que on A2Billing
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 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) . Any suggestion... Cheers, Nitesh Guillermo Salas M. wrote: 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 card number, enabled VOIP_CALL, and enabled Active. 2486543210 is your card number? When I dial the DID number, on the *CLI it shows the following: - a2billing.php|1|did: bug -- AGI Script Executing Application: (DIAL) Options: (SIP/2486543210|60|HL(360:61000:3)) -- Limit Data for this call: -- - timelimit = 360 -- - play_warning = 61000 -- - play_to_caller= yes -- - play_to_callee= no -- - warning_freq = 3 -- - start_sound = UNDEF -- - warning_sound = timeleft -- - end_sound = UNDEF Destroying call '[EMAIL PROTECTED]' Jun 15 16:41:34 NOTICE[15346]: app_dial.c:1069 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) I think that 2486543210 is not a customer, card number or SIP/IAX2 friend, maybe is PSTN number. To redirect the call to any PSTN number you must need to set voip call to inactive and set the destination number to 2486543210. I bet I am missing something in extension.conf correct? I dont see any examples in my package. The context is fine don't worry about it. Any suggestion... Thanks once again... Regards, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Que on A2Billing
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) . I know it, but the error is saying that you don't have one 2486543210 user registred. Show us the output of: sip show peers 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 : [EMAIL PROTECTED] 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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Que on A2Billing
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 LAGGED. Dunno what does that means... Any suggestions... Cheers, Nitesh Guillermo Salas M. wrote: 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) . I know it, but the error is saying that you don't have one 2486543210 user registred. Show us the output of: sip show peers Regards, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Que on A2Billing
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 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) . I know it, but the error is saying that you don't have one 2486543210 user registred. Show us the output of: sip show peers Regards, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Que on A2Billing
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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Que on A2Billing
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 : [EMAIL PROTECTED] 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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Que on A2Billing
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 - 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, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
