[asterisk-users] multi tenant
Hi all, I need to configure DIDs for different companies and they should reach on different extension with different context. Cant we have same extension in different context? This is what we we want Company A: Context Company_A IVR Company A Extensions: 101,102,103,104 etc. Company B: Context Company_B IVR Company B Extensions: 101,102,103,104 etc. Company C: Context Company_C IVR Company Extensions: 101,102,103,104 etc. Company D: Context Company_D IVR Company D Extensions: 101,102,103,104 etc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multi tenant
Not possible to have same sip usernames. However you can create custA_user1 == 101 custB_user1 == 101 In the dialplan context. Mitul On Oct 30, 2012 12:47 PM, Darin Iv adari...@gmail.com wrote: Hi all, I need to configure DIDs for different companies and they should reach on different extension with different context. Cant we have same extension in different context? This is what we we want Company A: Context Company_A IVR Company A Extensions: 101,102,103,104 etc. Company B: Context Company_B IVR Company B Extensions: 101,102,103,104 etc. Company C: Context Company_C IVR Company Extensions: 101,102,103,104 etc. Company D: Context Company_D IVR Company D Extensions: 101,102,103,104 etc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multi tenant
Yes, you can do this. You should point the trunks to the right context and done. Op 30-10-2012 8:15, Darin Iv schreef: Hi all, I need to configure DIDs for different companies and they should reach on different extension with different context. Cant we have same extension in different context? This is what we we want Company A: Context Company_A IVR Company A Extensions: 101,102,103,104 etc. Company B: Context Company_B IVR Company B Extensions: 101,102,103,104 etc. Company C: Context Company_C IVR Company Extensions: 101,102,103,104 etc. Company D: Context Company_D IVR Company D Extensions: 101,102,103,104 etc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multi tenant
Its depends on your incoming trunks. You can define different context to different trunks and your DID/extension will be called as per dialplan in that parituclar context of trunk. On Tue, Oct 30, 2012 at 12:57 PM, Henk Dick h...@osocoms.com wrote: Yes, you can do this. You should point the trunks to the right context and done. Op 30-10-2012 8:15, Darin Iv schreef: Hi all, I need to configure DIDs for different companies and they should reach on different extension with different context. Cant we have same extension in different context? This is what we we want Company A: Context Company_A IVR Company A Extensions: 101,102,103,104 etc. Company B: Context Company_B IVR Company B Extensions: 101,102,103,104 etc. Company C: Context Company_C IVR Company Extensions: 101,102,103,104 etc. Company D: Context Company_D IVR Company D Extensions: 101,102,103,104 etc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bharat Lalcheta -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multi tenant
On Tue, Oct 30, 2012 at 12:15 AM, Darin Iv adari...@gmail.com wrote:
Hi all,
I need to configure DIDs for different companies and they should reach on
different extension with different context. Cant we have same extension
in different context?
This is what we we want
Company A:
Context Company_A
IVR Company A
Extensions: 101,102,103,104 etc.
Company B:
Context Company_B
IVR Company B
Extensions: 101,102,103,104 etc.
There are multiple ways to do this. One way is the Local dial. We have
done this for companies who are different entities but want to do 3-digit
dial.
Dial(Local/101@company_a#extensions,25)
Where we assume you have a context like:
[company_a#extensions]
exten = 101,1,Dial(SIP/company_a.${EXTEN},25)
Another way is to simply do an include for the other company's extension
context. However that requires that you not duplicate the extension
numbers between the contexts/companies.
--
Carlos Alvarez
TelEvolve
602-889-3003
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multi tenant
I am attempting to send this again. The mail processor is interpreting the
Asterisk commands in my message as mail processor command and bouncing the
message. That's why where is junk before many of the lines below.
On Tue, Oct 30, 2012 at 12:15 AM, Darin Iv adari...@gmail.com wrote:
Hi all,
I need to configure DIDs for different companies and they should reach on
different extension with different context. Cant we have same extension
in different context?
This is what we we want
Company A:
Context Company_A
IVR Company A
Extensions: 101,102,103,104 etc.
Company B:
Context Company_B
IVR Company B
Extensions: 101,102,103,104 etc.
There are multiple ways to do this. One way is the Local dial. We have
done this for companies who are different entities but want to do 3-digit
dial.
... Dial(Local/101@company_a#extensions,25)
Where we assume you have a context like:
... [company_a#extensions]
... exten = 101,1,Dial(SIP/company_a.${EXTEN},25)
Another way is to simply do an include for the other company's extension
context. However that requires that you not duplicate the extension
numbers between the contexts/companies.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-tenant IVR
Hi Carlos, The idea is this. We are planning to offer customized version of Asterisk for specialized purposes. When we offer hosted PBX, using multi-tenancy support, it is just going to be PBX, as opposed to a fully blown IVR. It will have automated attendant feature, but not IVR. In contrast, hosted IVR will have only one number dedicated to a business, and the business can maintain the call flow and sound files. The system will integrate with their CRM and offer personalized services to the customers of the business. And, of course, the system will have the support to connect to the PBX of the business, should the customer of the business selects to talk to the customer care agent of the business. That is our system won’t be used for the communication between the extensions of the business. Do you have any reservations on this? Regards, Kannan. On Thu, Aug 9, 2012 at 11:38 PM, Carlos Alvarez car...@televolve.comwrote: On Thu, Aug 9, 2012 at 10:59 AM, Kannan vasdevelo...@gmail.com wrote: Is should be possible to CONFGURE Asterisk as a multi-tenant IVR server right? Like tweaking configuration to configure a multi-tenant PBX with Asterisk. I don't know why you make a distinction between a multi-tenant IVR and a multi-tenant PBX. The IVR would just be in tenant contexts just like all other features. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-tenant IVR
What you want can be done by OpenVBX, why dont you try exploring that model ? Regards, Mitul Limbani, Chief Architech Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-71967121 Cell: +91-9820332422 On Fri, Aug 10, 2012 at 3:19 PM, Kannan vasdevelo...@gmail.com wrote: Hi Carlos, The idea is this. We are planning to offer customized version of Asterisk for specialized purposes. When we offer hosted PBX, using multi-tenancy support, it is just going to be PBX, as opposed to a fully blown IVR. It will have automated attendant feature, but not IVR. In contrast, hosted IVR will have only one number dedicated to a business, and the business can maintain the call flow and sound files. The system will integrate with their CRM and offer personalized services to the customers of the business. And, of course, the system will have the support to connect to the PBX of the business, should the customer of the business selects to talk to the customer care agent of the business. That is our system won’t be used for the communication between the extensions of the business. Do you have any reservations on this? Regards, Kannan. On Thu, Aug 9, 2012 at 11:38 PM, Carlos Alvarez car...@televolve.comwrote: On Thu, Aug 9, 2012 at 10:59 AM, Kannan vasdevelo...@gmail.com wrote: Is should be possible to CONFGURE Asterisk as a multi-tenant IVR server right? Like tweaking configuration to configure a multi-tenant PBX with Asterisk. I don't know why you make a distinction between a multi-tenant IVR and a multi-tenant PBX. The IVR would just be in tenant contexts just like all other features. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-tenant IVR
On Fri, Aug 10, 2012 at 2:49 AM, Kannan vasdevelo...@gmail.com wrote: In contrast, hosted IVR will have only one number dedicated to a business, and the business can maintain the call flow and sound files. The system will integrate with their CRM and offer personalized services to the customers of the business. And, of course, the system will have the support to connect to the PBX of the business, should the customer of the business selects to talk to the customer care agent of the business. That is our system won’t be used for the communication between the extensions of the business. In order to do CRM or other client-side application integration, you'll need to create your own connectivity into Asterisk. The security in Asterisk's remote interfaces isn't great, and I'd say you need to develop some middleware that handles security and also makes it more robust. Letting the customers manage their changes would also require some interface you develop, and that part can get very complex because of things like dialplan reloading. We do not allow client access to our hosted PBX/IVR systems, so I can't advise you on that. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multi-tenant IVR
Hi There, Is should be possible to CONFGURE Asterisk as a multi-tenant IVR server right? Like tweaking configuration to configure a multi-tenant PBX with Asterisk. Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-tenant IVR
On Thu, Aug 9, 2012 at 10:59 AM, Kannan vasdevelo...@gmail.com wrote: Is should be possible to CONFGURE Asterisk as a multi-tenant IVR server right? Like tweaking configuration to configure a multi-tenant PBX with Asterisk. I don't know why you make a distinction between a multi-tenant IVR and a multi-tenant PBX. The IVR would just be in tenant contexts just like all other features. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-Tenant PBX with Asterisk
On Mon, 2012-07-30 at 15:06 +0530, Kannan wrote: Hi I came across couple of pointers on the Internet regarding solutions available for providing hosted PBX service. 1. Multiple PBXs: Using separate hardware to host each PBX. Pretty straightforward, but no hosting company wants to use it. 2. Multi-tenant PBX: Configuring multiple PBXs within the same instance of Asterisk. I.e. partitioning a single instance of Asterisk into multiple PBXs by way of configurations, using unique landing context for each tenant. 3. Virtual PBX: Multiple virtual machines within the same hardware, each host an instance of Asterisk. Which one of the method above is generally used by hosted PBX service providers? Isn't the second option with ARA a good choice for dynamic creation of multiple small PBX tenants? Is the last option alone or combination of options 2 and 3 good for cloud based hosted PBX service offering? We use 2 and I'd have to agree with most of what the previous replies have said. You really need to nail down your conventions and stick to them. We did this by creating our own custom front end so our conventions are built in to the front end code. ARA is really useful for this type of thing. If you're expanding to the point that you need to add new servers for extra capacity, ARA enables you to retain all your config on a single (pair of) machine(s). It also means that, if you have the framework to allow it, your customers can make changes to their own account themselves. -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-Tenant PBX with Asterisk
Kannan I have to disagree with Leanrod. We are a hosted (cloud) PBX company we successfully run our Multi-tenant systems in Virtual machines and have no issues with them. It comes down to designing your virtual environment for your target loads and then not exceeding them. This allows for fail over of hardware and scalability. We have moved our virtual phone switches live with full call loads and have no call drops. We do not usually dedicate a single Virtual Machine to each customer either. We have built our own Multi-tenant PBX on top of asterisk. We achieve many of the features available in freepbx/trixbox (not all). This method allows us to cost effectively service our customers with a presence of scale in mind. It is not uncommon to have 5000 + extensions per virtual switch. This method does require highly skilled engineering to achieve stability. Bryant From: Kannan vasdevelo...@gmail.com Sent: Tuesday, July 31, 2012 12:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Multi-Tenant PBX with Asterisk Thanks Leandro for your comments. On Mon, Jul 30, 2012 at 6:35 PM, Leandro Dardini ldard...@gmail.com wrote: 2012/7/30 Kannan vasdevelo...@gmail.com Hi I came across couple of pointers on the Internet regarding solutions available for providing hosted PBX service. 1. Multiple PBXs: Using separate hardware to host each PBX. Pretty straightforward, but no hosting company wants to use it. 2. Multi-tenant PBX: Configuring multiple PBXs within the same instance of Asterisk. I.e. partitioning a single instance of Asterisk into multiple PBXs by way of configurations, using unique landing context for each tenant. 3. Virtual PBX: Multiple virtual machines within the same hardware, each host an instance of Asterisk. Which one of the method above is generally used by hosted PBX service providers? Isn't the second option with ARA a good choice for dynamic creation of multiple small PBX tenants? Is the last option alone or combination of options 2 and 3 good for cloud based hosted PBX service offering? Thanks, Kannan. Working in the voip field from a lots of years, I have found all three type of business. The first is maybe the easier and most common. Hardware is cheap and it is easier to sell a service like the PBX if it is sold together with a piece of iron. Usually the hardware is placed on client's network, using the bandwidth of the client. Usually together with the PBX is sold also a router/firewall/traffic shaper/vpn endpoint to try to optimize the traffic on the client's DSL. The major pros about this solution is you can use a normal PBX like freepbx/trixbox, the client can mess the config how he likes, without disrupting other services, you can install VoIP card to connect landlines,. The major cons is the cost of the hardware, the cost of the g.729 licenses (if any) and the maintenance cost of replacing hardware failures and the need to be physically near each client. The second is the holy grail of the VoIP providers. The major pros is the cost. Having a single hardware is cheap and it is still cheap also if you decide to get two to be ready in case of an hardware failure. The major cons is the software. You cannot use the award winning freepbx/trixbox family and you need to deal with sometime limited or incomplete developed interfaces. The client always asks for the missing feature. One other major cons is the reload. If the PBX software is not made using ARA, then every time you add a new peer or a new DID, you need to reload the entire PBX and that is a resource killer. Again, if the pbx interface is not made using ARA, then you cannot let your clients to change the configuration or they will trigger continuous reload (and delaying reload for example every 10 minutes is not a solution) The last one is sometime the chosen compromise, but from my point of view, pbxes are not good software to virtualize. They are too sensible to delays and your voice quality can go down if the real server is overloaded. The same for the cloud based solutions (I have yet to found). I suspect the cloud is good for services like http, not for real time applications. Leandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello
Re: [asterisk-users] Multi-Tenant PBX with Asterisk
Hello Bryant, it is nice to hear someone with different experience, so I am happy to know the cloud is indeed a feasible environment even for VoIP. Can you share with us some of your configuration magic? Like the cloud service you are using, the power of each node and the load you are experiencing on them in regards to the number of channels active and phone registered? Leandro 2012/7/31 Bryant Zimmerman brya...@zktech.com Kannan I have to disagree with Leanrod. We are a hosted (cloud) PBX company we successfully run our Multi-tenant systems in Virtual machines and have no issues with them. It comes down to designing your virtual environment for your target loads and then not exceeding them. This allows for fail over of hardware and scalability. We have moved our virtual phone switches live with full call loads and have no call drops. We do not usually dedicate a single Virtual Machine to each customer either. We have built our own Multi-tenant PBX on top of asterisk. We achieve many of the features available in freepbx/trixbox (not all). This method allows us to cost effectively service our customers with a presence of scale in mind. It is not uncommon to have 5000 + extensions per virtual switch. This method does require highly skilled engineering to achieve stability. Bryant -- *From*: Kannan vasdevelo...@gmail.com *Sent*: Tuesday, July 31, 2012 12:37 AM *To*: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Subject*: Re: [asterisk-users] Multi-Tenant PBX with Asterisk Thanks Leandro for your comments. On Mon, Jul 30, 2012 at 6:35 PM, Leandro Dardini ldard...@gmail.comwrote: 2012/7/30 Kannan vasdevelo...@gmail.com Hi I came across couple of pointers on the Internet regarding solutions available for providing hosted PBX service. 1. Multiple PBXs: Using separate hardware to host each PBX. Pretty straightforward, but no hosting company wants to use it. 2. Multi-tenant PBX: Configuring multiple PBXs within the same instance of Asterisk. I.e. partitioning a single instance of Asterisk into multiple PBXs by way of configurations, using unique landing context for each tenant. 3. Virtual PBX: Multiple virtual machines within the same hardware, each host an instance of Asterisk. Which one of the method above is generally used by hosted PBX service providers? Isn't the second option with ARA a good choice for dynamic creation of multiple small PBX tenants? Is the last option alone or combination of options 2 and 3 good for cloud based hosted PBX service offering? Thanks, Kannan. Working in the voip field from a lots of years, I have found all three type of business. The first is maybe the easier and most common. Hardware is cheap and it is easier to sell a service like the PBX if it is sold together with a piece of iron. Usually the hardware is placed on client's network, using the bandwidth of the client. Usually together with the PBX is sold also a router/firewall/traffic shaper/vpn endpoint to try to optimize the traffic on the client's DSL. The major pros about this solution is you can use a normal PBX like freepbx/trixbox, the client can mess the config how he likes, without disrupting other services, you can install VoIP card to connect landlines,. The major cons is the cost of the hardware, the cost of the g.729 licenses (if any) and the maintenance cost of replacing hardware failures and the need to be physically near each client. The second is the holy grail of the VoIP providers. The major pros is the cost. Having a single hardware is cheap and it is still cheap also if you decide to get two to be ready in case of an hardware failure. The major cons is the software. You cannot use the award winning freepbx/trixbox family and you need to deal with sometime limited or incomplete developed interfaces. The client always asks for the missing feature. One other major cons is the reload. If the PBX software is not made using ARA, then every time you add a new peer or a new DID, you need to reload the entire PBX and that is a resource killer. Again, if the pbx interface is not made using ARA, then you cannot let your clients to change the configuration or they will trigger continuous reload (and delaying reload for example every 10 minutes is not a solution) The last one is sometime the chosen compromise, but from my point of view, pbxes are not good software to virtualize. They are too sensible to delays and your voice quality can go down if the real server is overloaded. The same for the cloud based solutions (I have yet to found). I suspect the cloud is good for services like http, not for real time applications. Leandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory
Re: [asterisk-users] Multi-Tenant PBX with Asterisk
Particularly, what virtualization software are you using? On Tue, Jul 31, 2012 at 8:19 AM, Leandro Dardini ldard...@gmail.com wrote: Hello Bryant, it is nice to hear someone with different experience, so I am happy to know the cloud is indeed a feasible environment even for VoIP. Can you share with us some of your configuration magic? Like the cloud service you are using, the power of each node and the load you are experiencing on them in regards to the number of channels active and phone registered? Leandro 2012/7/31 Bryant Zimmerman brya...@zktech.com Kannan I have to disagree with Leanrod. We are a hosted (cloud) PBX company we successfully run our Multi-tenant systems in Virtual machines and have no issues with them. It comes down to designing your virtual environment for your target loads and then not exceeding them. This allows for fail over of hardware and scalability. We have moved our virtual phone switches live with full call loads and have no call drops. We do not usually dedicate a single Virtual Machine to each customer either. We have built our own Multi-tenant PBX on top of asterisk. We achieve many of the features available in freepbx/trixbox (not all). This method allows us to cost effectively service our customers with a presence of scale in mind. It is not uncommon to have 5000 + extensions per virtual switch. This method does require highly skilled engineering to achieve stability. Bryant -- *From*: Kannan vasdevelo...@gmail.com *Sent*: Tuesday, July 31, 2012 12:37 AM *To*: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Subject*: Re: [asterisk-users] Multi-Tenant PBX with Asterisk Thanks Leandro for your comments. On Mon, Jul 30, 2012 at 6:35 PM, Leandro Dardini ldard...@gmail.comwrote: 2012/7/30 Kannan vasdevelo...@gmail.com Hi I came across couple of pointers on the Internet regarding solutions available for providing hosted PBX service. 1. Multiple PBXs: Using separate hardware to host each PBX. Pretty straightforward, but no hosting company wants to use it. 2. Multi-tenant PBX: Configuring multiple PBXs within the same instance of Asterisk. I.e. partitioning a single instance of Asterisk into multiple PBXs by way of configurations, using unique landing context for each tenant. 3. Virtual PBX: Multiple virtual machines within the same hardware, each host an instance of Asterisk. Which one of the method above is generally used by hosted PBX service providers? Isn't the second option with ARA a good choice for dynamic creation of multiple small PBX tenants? Is the last option alone or combination of options 2 and 3 good for cloud based hosted PBX service offering? Thanks, Kannan. Working in the voip field from a lots of years, I have found all three type of business. The first is maybe the easier and most common. Hardware is cheap and it is easier to sell a service like the PBX if it is sold together with a piece of iron. Usually the hardware is placed on client's network, using the bandwidth of the client. Usually together with the PBX is sold also a router/firewall/traffic shaper/vpn endpoint to try to optimize the traffic on the client's DSL. The major pros about this solution is you can use a normal PBX like freepbx/trixbox, the client can mess the config how he likes, without disrupting other services, you can install VoIP card to connect landlines,. The major cons is the cost of the hardware, the cost of the g.729 licenses (if any) and the maintenance cost of replacing hardware failures and the need to be physically near each client. The second is the holy grail of the VoIP providers. The major pros is the cost. Having a single hardware is cheap and it is still cheap also if you decide to get two to be ready in case of an hardware failure. The major cons is the software. You cannot use the award winning freepbx/trixbox family and you need to deal with sometime limited or incomplete developed interfaces. The client always asks for the missing feature. One other major cons is the reload. If the PBX software is not made using ARA, then every time you add a new peer or a new DID, you need to reload the entire PBX and that is a resource killer. Again, if the pbx interface is not made using ARA, then you cannot let your clients to change the configuration or they will trigger continuous reload (and delaying reload for example every 10 minutes is not a solution) The last one is sometime the chosen compromise, but from my point of view, pbxes are not good software to virtualize. They are too sensible to delays and your voice quality can go down if the real server is overloaded. The same for the cloud based solutions (I have yet to found). I suspect the cloud is good for services like http, not for real time applications. Leandro
[asterisk-users] Multi-Tenant PBX with Asterisk
Hi I came across couple of pointers on the Internet regarding solutions available for providing hosted PBX service. 1. Multiple PBXs: Using separate hardware to host each PBX. Pretty straightforward, but no hosting company wants to use it. 2. Multi-tenant PBX: Configuring multiple PBXs within the same instance of Asterisk. I.e. partitioning a single instance of Asterisk into multiple PBXs by way of configurations, using unique landing context for each tenant. 3. Virtual PBX: Multiple virtual machines within the same hardware, each host an instance of Asterisk. Which one of the method above is generally used by hosted PBX service providers? Isn't the second option with ARA a good choice for dynamic creation of multiple small PBX tenants? Is the last option alone or combination of options 2 and 3 good for cloud based hosted PBX service offering? Thanks, Kannan. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-Tenant PBX with Asterisk
2012/7/30 Kannan vasdevelo...@gmail.com Hi I came across couple of pointers on the Internet regarding solutions available for providing hosted PBX service. 1. Multiple PBXs: Using separate hardware to host each PBX. Pretty straightforward, but no hosting company wants to use it. 2. Multi-tenant PBX: Configuring multiple PBXs within the same instance of Asterisk. I.e. partitioning a single instance of Asterisk into multiple PBXs by way of configurations, using unique landing context for each tenant. 3. Virtual PBX: Multiple virtual machines within the same hardware, each host an instance of Asterisk. Which one of the method above is generally used by hosted PBX service providers? Isn't the second option with ARA a good choice for dynamic creation of multiple small PBX tenants? Is the last option alone or combination of options 2 and 3 good for cloud based hosted PBX service offering? Thanks, Kannan. Working in the voip field from a lots of years, I have found all three type of business. The first is maybe the easier and most common. Hardware is cheap and it is easier to sell a service like the PBX if it is sold together with a piece of iron. Usually the hardware is placed on client's network, using the bandwidth of the client. Usually together with the PBX is sold also a router/firewall/traffic shaper/vpn endpoint to try to optimize the traffic on the client's DSL. The major pros about this solution is you can use a normal PBX like freepbx/trixbox, the client can mess the config how he likes, without disrupting other services, you can install VoIP card to connect landlines,. The major cons is the cost of the hardware, the cost of the g.729 licenses (if any) and the maintenance cost of replacing hardware failures and the need to be physically near each client. The second is the holy grail of the VoIP providers. The major pros is the cost. Having a single hardware is cheap and it is still cheap also if you decide to get two to be ready in case of an hardware failure. The major cons is the software. You cannot use the award winning freepbx/trixbox family and you need to deal with sometime limited or incomplete developed interfaces. The client always asks for the missing feature. One other major cons is the reload. If the PBX software is not made using ARA, then every time you add a new peer or a new DID, you need to reload the entire PBX and that is a resource killer. Again, if the pbx interface is not made using ARA, then you cannot let your clients to change the configuration or they will trigger continuous reload (and delaying reload for example every 10 minutes is not a solution) The last one is sometime the chosen compromise, but from my point of view, pbxes are not good software to virtualize. They are too sensible to delays and your voice quality can go down if the real server is overloaded. The same for the cloud based solutions (I have yet to found). I suspect the cloud is good for services like http, not for real time applications. Leandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-Tenant PBX with Asterisk
On Mon, Jul 30, 2012 at 2:36 AM, Kannan vasdevelo...@gmail.com wrote: 2. Multi-tenant PBX: Configuring multiple PBXs within the same instance of Asterisk. I.e. partitioning a single instance of Asterisk into multiple PBXs by way of configurations, using unique landing context for each tenant. 3. Virtual PBX: Multiple virtual machines within the same hardware, each host an instance of Asterisk. We use number two. We dabbled with number three but didn't like the results for a lot of different reasons. As others have mentioned, there is a certain level of danger when you mix companies so closely. We have in the past made a mistake and brought down the whole system, but it's been many years since we've done that. Part is improved skill and part is that Asterisk has improved and no longer commits suicide for certain minor errors. To do this, you need to plan out a good naming convention for everything that will be unique to customers accounts. SIP accounts, macros, contexts, etc etc. We use the accountcode feature and prepend the accountcode through the dial plan and accounts. accountcode.301 would be a SIP account accountcode#function would be a context name We do deploy custom hardware for specific functions or customers who are particularly large in some cases. We just need a good reason to. Like they want to self-manage, or they make a lot of changes, need custom integration with databases, etc. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-Tenant PBX with Asterisk
Thanks Leandro for your comments. On Mon, Jul 30, 2012 at 6:35 PM, Leandro Dardini ldard...@gmail.com wrote: 2012/7/30 Kannan vasdevelo...@gmail.com Hi I came across couple of pointers on the Internet regarding solutions available for providing hosted PBX service. 1. Multiple PBXs: Using separate hardware to host each PBX. Pretty straightforward, but no hosting company wants to use it. 2. Multi-tenant PBX: Configuring multiple PBXs within the same instance of Asterisk. I.e. partitioning a single instance of Asterisk into multiple PBXs by way of configurations, using unique landing context for each tenant. 3. Virtual PBX: Multiple virtual machines within the same hardware, each host an instance of Asterisk. Which one of the method above is generally used by hosted PBX service providers? Isn't the second option with ARA a good choice for dynamic creation of multiple small PBX tenants? Is the last option alone or combination of options 2 and 3 good for cloud based hosted PBX service offering? Thanks, Kannan. Working in the voip field from a lots of years, I have found all three type of business. The first is maybe the easier and most common. Hardware is cheap and it is easier to sell a service like the PBX if it is sold together with a piece of iron. Usually the hardware is placed on client's network, using the bandwidth of the client. Usually together with the PBX is sold also a router/firewall/traffic shaper/vpn endpoint to try to optimize the traffic on the client's DSL. The major pros about this solution is you can use a normal PBX like freepbx/trixbox, the client can mess the config how he likes, without disrupting other services, you can install VoIP card to connect landlines,. The major cons is the cost of the hardware, the cost of the g.729 licenses (if any) and the maintenance cost of replacing hardware failures and the need to be physically near each client. The second is the holy grail of the VoIP providers. The major pros is the cost. Having a single hardware is cheap and it is still cheap also if you decide to get two to be ready in case of an hardware failure. The major cons is the software. You cannot use the award winning freepbx/trixbox family and you need to deal with sometime limited or incomplete developed interfaces. The client always asks for the missing feature. One other major cons is the reload. If the PBX software is not made using ARA, then every time you add a new peer or a new DID, you need to reload the entire PBX and that is a resource killer. Again, if the pbx interface is not made using ARA, then you cannot let your clients to change the configuration or they will trigger continuous reload (and delaying reload for example every 10 minutes is not a solution) The last one is sometime the chosen compromise, but from my point of view, pbxes are not good software to virtualize. They are too sensible to delays and your voice quality can go down if the real server is overloaded. The same for the cloud based solutions (I have yet to found). I suspect the cloud is good for services like http, not for real time applications. Leandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-Tenant PBX with Asterisk
Thanks Carlos, it is good to hear from one who is in a similar business. Are you getting use of ARA too in similar hosted PBX offerings? On Mon, Jul 30, 2012 at 10:00 PM, Carlos Alvarez car...@televolve.comwrote: On Mon, Jul 30, 2012 at 2:36 AM, Kannan vasdevelo...@gmail.com wrote: 2. Multi-tenant PBX: Configuring multiple PBXs within the same instance of Asterisk. I.e. partitioning a single instance of Asterisk into multiple PBXs by way of configurations, using unique landing context for each tenant. 3. Virtual PBX: Multiple virtual machines within the same hardware, each host an instance of Asterisk. We use number two. We dabbled with number three but didn't like the results for a lot of different reasons. As others have mentioned, there is a certain level of danger when you mix companies so closely. We have in the past made a mistake and brought down the whole system, but it's been many years since we've done that. Part is improved skill and part is that Asterisk has improved and no longer commits suicide for certain minor errors. To do this, you need to plan out a good naming convention for everything that will be unique to customers accounts. SIP accounts, macros, contexts, etc etc. We use the accountcode feature and prepend the accountcode through the dial plan and accounts. accountcode.301 would be a SIP account accountcode#function would be a context name We do deploy custom hardware for specific functions or customers who are particularly large in some cases. We just need a good reason to. Like they want to self-manage, or they make a lot of changes, need custom integration with databases, etc. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-Tenant PBX with Asterisk
I don't know what ARA is. We use just bare Asterisk, no GUI, and from the context it seems that's related to a GUI. We have no problem doing a config reload during production hours. We never do a full reload, just the relevant module (SIP, dialplan, voicemail, etc). I don't believe there is any freeware PBX software that is good for hosted services unless they are kept tiny and limited. Switchvox is excellent as a hosted platform, but extremely expensive and totally closed so you can't customize as needed. And at least 50% of our customers have customization that wouldn't fit into any of the GUI-based systems. You'll need to decide what your market is and your value proposition as well as your ability to learn Asterisk (which I don't think anyone would argue is easy or fast). On Mon, Jul 30, 2012 at 9:41 PM, Kannan vasdevelo...@gmail.com wrote: Thanks Carlos, it is good to hear from one who is in a similar business. Are you getting use of ARA too in similar hosted PBX offerings? On Mon, Jul 30, 2012 at 10:00 PM, Carlos Alvarez car...@televolve.comwrote: On Mon, Jul 30, 2012 at 2:36 AM, Kannan vasdevelo...@gmail.com wrote: 2. Multi-tenant PBX: Configuring multiple PBXs within the same instance of Asterisk. I.e. partitioning a single instance of Asterisk into multiple PBXs by way of configurations, using unique landing context for each tenant. 3. Virtual PBX: Multiple virtual machines within the same hardware, each host an instance of Asterisk. We use number two. We dabbled with number three but didn't like the results for a lot of different reasons. As others have mentioned, there is a certain level of danger when you mix companies so closely. We have in the past made a mistake and brought down the whole system, but it's been many years since we've done that. Part is improved skill and part is that Asterisk has improved and no longer commits suicide for certain minor errors. To do this, you need to plan out a good naming convention for everything that will be unique to customers accounts. SIP accounts, macros, contexts, etc etc. We use the accountcode feature and prepend the accountcode through the dial plan and accounts. accountcode.301 would be a SIP account accountcode#function would be a context name We do deploy custom hardware for specific functions or customers who are particularly large in some cases. We just need a good reason to. Like they want to self-manage, or they make a lot of changes, need custom integration with databases, etc. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-Tenant PBX with Asterisk
ARA is an acronym for Asterisk Realtime Architecture and is a different way to keep configuration files in asterisk. Instead of reading configuration from plain files at startup, asterisk read them from database, in realtime. This mean, if you need to add a peer, you drop a new line in the sippeers table and you are fine. You start defining an ODBC source in res_odbc.conf and then configure the ARA source for each plain configuration files in extconfig.conf About the config reload, reloading only the module changed is a good idea, but the commercial GUI I have meet so far doesn't support it. I have clients with very simple dialplan, able to reload it even if more than 130.000 rows long, others, with more complicated dialplan cannot reload it during work hours even if only 30.000 rows long. You are right about freeware PBX for hosted services. Independent from the fact a GUI is free or needs a payment, I think it is important to have the source for it to be able to customize it and also it is important to have a clean dialplan, so you can debug and customize it as well. I am a developer selling software. I never protect my code obfuscating or compiling it and my clients enjoy it and never steal my work (so far). Leandro 2012/7/31 Carlos Alvarez car...@televolve.com I don't know what ARA is. We use just bare Asterisk, no GUI, and from the context it seems that's related to a GUI. We have no problem doing a config reload during production hours. We never do a full reload, just the relevant module (SIP, dialplan, voicemail, etc). I don't believe there is any freeware PBX software that is good for hosted services unless they are kept tiny and limited. Switchvox is excellent as a hosted platform, but extremely expensive and totally closed so you can't customize as needed. And at least 50% of our customers have customization that wouldn't fit into any of the GUI-based systems. You'll need to decide what your market is and your value proposition as well as your ability to learn Asterisk (which I don't think anyone would argue is easy or fast). On Mon, Jul 30, 2012 at 9:41 PM, Kannan vasdevelo...@gmail.com wrote: Thanks Carlos, it is good to hear from one who is in a similar business. Are you getting use of ARA too in similar hosted PBX offerings? On Mon, Jul 30, 2012 at 10:00 PM, Carlos Alvarez car...@televolve.comwrote: On Mon, Jul 30, 2012 at 2:36 AM, Kannan vasdevelo...@gmail.com wrote: 2. Multi-tenant PBX: Configuring multiple PBXs within the same instance of Asterisk. I.e. partitioning a single instance of Asterisk into multiple PBXs by way of configurations, using unique landing context for each tenant. 3. Virtual PBX: Multiple virtual machines within the same hardware, each host an instance of Asterisk. We use number two. We dabbled with number three but didn't like the results for a lot of different reasons. As others have mentioned, there is a certain level of danger when you mix companies so closely. We have in the past made a mistake and brought down the whole system, but it's been many years since we've done that. Part is improved skill and part is that Asterisk has improved and no longer commits suicide for certain minor errors. To do this, you need to plan out a good naming convention for everything that will be unique to customers accounts. SIP accounts, macros, contexts, etc etc. We use the accountcode feature and prepend the accountcode through the dial plan and accounts. accountcode.301 would be a SIP account accountcode#function would be a context name We do deploy custom hardware for specific functions or customers who are particularly large in some cases. We just need a good reason to. Like they want to self-manage, or they make a lot of changes, need custom integration with databases, etc. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:
Re: [asterisk-users] Multi-Tenant PBX with Asterisk
We tried realtime and decided it wasn't for us. Never got it into production, so I can't say much about it. -- Carlos Alvarez Sent from my Nexus 7 On Jul 30, 2012 10:25 PM, Leandro Dardini ldard...@gmail.com wrote: ARA is an acronym for Asterisk Realtime Architecture and is a different way to keep configuration files in asterisk. Instead of reading configuration from plain files at startup, asterisk read them from database, in realtime. This mean, if you need to add a peer, you drop a new line in the sippeers table and you are fine. You start defining an ODBC source in res_odbc.conf and then configure the ARA source for each plain configuration files in extconfig.conf About the config reload, reloading only the module changed is a good idea, but the commercial GUI I have meet so far doesn't support it. I have clients with very simple dialplan, able to reload it even if more than 130.000 rows long, others, with more complicated dialplan cannot reload it during work hours even if only 30.000 rows long. You are right about freeware PBX for hosted services. Independent from the fact a GUI is free or needs a payment, I think it is important to have the source for it to be able to customize it and also it is important to have a clean dialplan, so you can debug and customize it as well. I am a developer selling software. I never protect my code obfuscating or compiling it and my clients enjoy it and never steal my work (so far). Leandro 2012/7/31 Carlos Alvarez car...@televolve.com I don't know what ARA is. We use just bare Asterisk, no GUI, and from the context it seems that's related to a GUI. We have no problem doing a config reload during production hours. We never do a full reload, just the relevant module (SIP, dialplan, voicemail, etc). I don't believe there is any freeware PBX software that is good for hosted services unless they are kept tiny and limited. Switchvox is excellent as a hosted platform, but extremely expensive and totally closed so you can't customize as needed. And at least 50% of our customers have customization that wouldn't fit into any of the GUI-based systems. You'll need to decide what your market is and your value proposition as well as your ability to learn Asterisk (which I don't think anyone would argue is easy or fast). On Mon, Jul 30, 2012 at 9:41 PM, Kannan vasdevelo...@gmail.com wrote: Thanks Carlos, it is good to hear from one who is in a similar business. Are you getting use of ARA too in similar hosted PBX offerings? On Mon, Jul 30, 2012 at 10:00 PM, Carlos Alvarez car...@televolve.comwrote: On Mon, Jul 30, 2012 at 2:36 AM, Kannan vasdevelo...@gmail.com wrote: 2. Multi-tenant PBX: Configuring multiple PBXs within the same instance of Asterisk. I.e. partitioning a single instance of Asterisk into multiple PBXs by way of configurations, using unique landing context for each tenant. 3. Virtual PBX: Multiple virtual machines within the same hardware, each host an instance of Asterisk. We use number two. We dabbled with number three but didn't like the results for a lot of different reasons. As others have mentioned, there is a certain level of danger when you mix companies so closely. We have in the past made a mistake and brought down the whole system, but it's been many years since we've done that. Part is improved skill and part is that Asterisk has improved and no longer commits suicide for certain minor errors. To do this, you need to plan out a good naming convention for everything that will be unique to customers accounts. SIP accounts, macros, contexts, etc etc. We use the accountcode feature and prepend the accountcode through the dial plan and accounts. accountcode.301 would be a SIP account accountcode#function would be a context name We do deploy custom hardware for specific functions or customers who are particularly large in some cases. We just need a good reason to. Like they want to self-manage, or they make a lot of changes, need custom integration with databases, etc. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 --
[asterisk-users] Multi tenant Parking issue
Hi, I am working on call packing feature of asterisk. Call packing is working fine but I want to make this feature as multi tenant. exp:- *for A client* packing extension are parkext = 700 parkpos = 701-720 context = parkedcalls_A parkingtime = 45 *for B client *packing extension are parkext = 800 parkpos = 801-820 context = parkedcalls_B parkingtime = 45 Is it possible or not ? - Thanks and regards Virendra Bhati +91-9172341457 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-Tenant Hosted PBX system with Reseller functionality
On Mon, 2011-03-21 at 21:45 -0300, Juan wrote: damn, advertisements everywhere, also in non commercial mailing lists... ITSPTEC.COM seems don't understand what a NON-COMMERCIAL DISCUSSION is about I will never buy anything from people like you who don't seems to understand so basic things @itsptec.com should be blacklisted... Or atleast kicked off this list -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-Tenant Hosted PBX system with Reseller functionality
On 22 Mar 2011, at 01:09, Outback Dingo wrote: Even worse... now it smells of a scam At least their website isn't hideous... Oh..wait.. ;) S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multi-Tenant Hosted PBX system with Reseller functionality
We are glad to announce that ITSPtec now offers a complete ITSP system for Asterisk with powerful routing engine, billing System- including invoicing, configuration, phone auto-provisioning and tones of other features. For Asterisk, we offer a Multi-Tenant Hosted PBX system with Reseller functionality that can be used by ITSP’s to provide HostedPBX services with reseller capability. For more information, please visit us at http://www.itsptec.com Thank You -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-Tenant Hosted PBX system with Reseller functionality
damn, advertisements everywhere, also in non commercial mailing lists... ITSPTEC.COM seems don't understand what a NON-COMMERCIAL DISCUSSION is about I will never buy anything from people like you who don't seems to understand so basic things @itsptec.com should be blacklisted... On Mon, 21 Mar 2011 20:38:10 -0300, john.bo...@itsptec.com wrote: We are glad to announce that ITSPtec now offers a complete ITSP system for Asterisk with powerful routing engine, billing System- including invoicing, configuration, phone auto-provisioning and tones of other features. For Asterisk, we offer a Multi-Tenant Hosted PBX system with Reseller functionality that can be used by ITSP’s to provide HostedPBX services with reseller capability. For more information, please visit us at http://www.itsptec.com Thank You -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-Tenant Hosted PBX system with Reseller functionality
Thanks John Bower and ITSPTEC.COM, you've made it easy for me to not feel bad about never using your products... On Mon, Mar 21, 2011 at 7:45 PM, Juan hardwareven...@gmail.com wrote: damn, advertisements everywhere, also in non commercial mailing lists... ITSPTEC.COM seems don't understand what a NON-COMMERCIAL DISCUSSION is about I will never buy anything from people like you who don't seems to understand so basic things @itsptec.com should be blacklisted... On Mon, 21 Mar 2011 20:38:10 -0300, john.bo...@itsptec.com wrote: We are glad to announce that ITSPtec now offers a complete ITSP system for Asterisk with powerful routing engine, billing System- including invoicing, configuration, phone auto-provisioning and tones of other features. For Asterisk, we offer a Multi-Tenant Hosted PBX system with Reseller functionality that can be used by ITSP’s to provide HostedPBX services with reseller capability. For more information, please visit us at http://www.itsptec.com Thank You -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-Tenant Hosted PBX system with Reseller functionality
great way to kill sales for your company idiot.! On Mon, Mar 21, 2011 at 7:38 PM, john.bo...@itsptec.com wrote: We are glad to announce that ITSPtec now offers a complete ITSP system for Asterisk with powerful routing engine, billing System- including invoicing, configuration, phone auto-provisioning and tones of other features. For Asterisk, we offer a Multi-Tenant Hosted PBX system with Reseller functionality that can be used by ITSP’s to provide HostedPBX services with reseller capability. For more information, please visit us at http://www.itsptec.com Thank You -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-Tenant Hosted PBX system with Reseller functionality
Even worse... now it smells of a scam Registered through: GoDaddy.com, Inc. (http://www.godaddy.com) Domain Name: ITSPTEC.COM Created on: 27-Jan-11 Expires on: 27-Jan-12 Last Updated on: 27-Jan-11 On Mon, Mar 21, 2011 at 9:06 PM, Outback Dingo outbackdi...@gmail.comwrote: great way to kill sales for your company idiot.! On Mon, Mar 21, 2011 at 7:38 PM, john.bo...@itsptec.com wrote: We are glad to announce that ITSPtec now offers a complete ITSP system for Asterisk with powerful routing engine, billing System- including invoicing, configuration, phone auto-provisioning and tones of other features. For Asterisk, we offer a Multi-Tenant Hosted PBX system with Reseller functionality that can be used by ITSP’s to provide HostedPBX services with reseller capability. For more information, please visit us at http://www.itsptec.com Thank You -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-Tenant Hosted PBX system with Reseller functionality
Oh my... On Mon, Mar 21, 2011 at 8:09 PM, Outback Dingo outbackdi...@gmail.comwrote: Even worse... now it smells of a scam Registered through: GoDaddy.com, Inc. (http://www.godaddy.com) Domain Name: ITSPTEC.COM Created on: 27-Jan-11 Expires on: 27-Jan-12 Last Updated on: 27-Jan-11 On Mon, Mar 21, 2011 at 9:06 PM, Outback Dingo outbackdi...@gmail.comwrote: great way to kill sales for your company idiot.! On Mon, Mar 21, 2011 at 7:38 PM, john.bo...@itsptec.com wrote: We are glad to announce that ITSPtec now offers a complete ITSP system for Asterisk with powerful routing engine, billing System- including invoicing, configuration, phone auto-provisioning and tones of other features. For Asterisk, we offer a Multi-Tenant Hosted PBX system with Reseller functionality that can be used by ITSP’s to provide HostedPBX services with reseller capability. For more information, please visit us at http://www.itsptec.com Thank You -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-Tenant Hosted PBX system with Resellerfunctionality
I listened to your email using DriveCarefully and will respond as soon as I can. Download DriveCarefully for free at www.drivecarefully.com -- Sent from my BlackBerry® Senior Support Engineer US Numbers: 561-886-0664 Nicaragua Mobile: +505.8488.6876 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multi-Tenant
HI , Please give idea for Multi tenant with Trixbox or elastix. Thanks Amardeep Rana -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-Tenant
On 11-01-27 11:41 AM, Amardeep Rana wrote: Please give idea for Multi tenant with Trixbox or elastix. http://astbook.asteriskdocs.org -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-Tenant
Oh man, I'm sorry, but I laughed so hard at that response, I think I peed a little :P To the original poster, Mr Belanger is most definitely being VERY kind compared to what some people might have responded with A little effort (and showing that you have put in that effort) goes a long way in an OSS users' mailing list On Thu, Jan 27, 2011 at 11:59 AM, Paul Belanger pabelan...@digium.com wrote: On 11-01-27 11:41 AM, Amardeep Rana wrote: Please give idea for Multi tenant with Trixbox or elastix. http://astbook.asteriskdocs.org -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-Tenant Parking
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of UxBoD Sent: Tuesday, 12 January 2010 17:16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Multi-Tenant Parking Should that not say parkinglot and not parkinglog in features.conf? It should – but that’s not a cut and paste, as the asterisk setup is on a separate, non-connected network, and I just retyped it out – not cut/paste. It’s spelt correctly in the real system (typo on here!) IMPORTANT NOTICE TO RECIPIENT Computer viruses - It is your responsibility to scan this email and any attachments for viruses and defects and rely on those scans as Communications Design Management Pty Limited (CDM) does not accept any liability for loss or damage arising from receipt or use of this email or any attachments. Confidentiality - This email and any attachments are intended for the named recipient only and may contain personal information, be it confidential or subject to privilege, none of which are lost or waived because this email may have been sent to you in error. If you are not the named addressee please let CDM know by return email, permanently delete it from your system and destroy all copies and do not use or disclose the contents. Copyright - This email is subject to copyright and no part of it maybe reproduced in any manner without the written permission of the copyright owner. Privacy - Within the jurisdiction of Australian law, personal information in this email must be dealt with in compliance with the Australian Federal Privacy Act 1988. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-Tenant Parking
Have you looked at this? http://www.google.com/#q=app_valetparking I have - but would rather use the inbuilt functionality if possible before resorting to third-party code... IMPORTANT NOTICE TO RECIPIENT Computer viruses - It is your responsibility to scan this email and any attachments for viruses and defects and rely on those scans as Communications Design Management Pty Limited (CDM) does not accept any liability for loss or damage arising from receipt or use of this email or any attachments. Confidentiality - This email and any attachments are intended for the named recipient only and may contain personal information, be it confidential or subject to privilege, none of which are lost or waived because this email may have been sent to you in error. If you are not the named addressee please let CDM know by return email, permanently delete it from your system and destroy all copies and do not use or disclose the contents. Copyright - This email is subject to copyright and no part of it maybe reproduced in any manner without the written permission of the copyright owner. Privacy - Within the jurisdiction of Australian law, personal information in this email must be dealt with in compliance with the Australian Federal Privacy Act 1988. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-Tenant Parking (HALF SOLVED)
I have found that this seems to be a functional difference between the Park() and the ParkAndAnnounce() functions. Park() respects the parking lot specification, yet ParkAndAnnounce() does not respect the fact that you’ve tried to arbitrarily set the parking lot. The code below “works” as designed when the Park() function is used instead. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of UxBoD Sent: Tuesday, 12 January 2010 17:16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Multi-Tenant Parking Should that not say parkinglot and not parkinglog in features.conf? It should – but that’s not a cut and paste, as the asterisk setup is on a separate, non-connected network, and I just retyped it out – not cut/paste. It’s spelt correctly in the real system (typo on here!) IMPORTANT NOTICE TO RECIPIENT Computer viruses - It is your responsibility to scan this email and any attachments for viruses and defects and rely on those scans as Communications Design Management Pty Limited (CDM) does not accept any liability for loss or damage arising from receipt or use of this email or any attachments. Confidentiality - This email and any attachments are intended for the named recipient only and may contain personal information, be it confidential or subject to privilege, none of which are lost or waived because this email may have been sent to you in error. If you are not the named addressee please let CDM know by return email, permanently delete it from your system and destroy all copies and do not use or disclose the contents. Copyright - This email is subject to copyright and no part of it maybe reproduced in any manner without the written permission of the copyright owner. Privacy - Within the jurisdiction of Australian law, personal information in this email must be dealt with in compliance with the Australian Federal Privacy Act 1988. IMPORTANT NOTICE TO RECIPIENT Computer viruses - It is your responsibility to scan this email and any attachments for viruses and defects and rely on those scans as Communications Design Management Pty Limited (CDM) does not accept any liability for loss or damage arising from receipt or use of this email or any attachments. Confidentiality - This email and any attachments are intended for the named recipient only and may contain personal information, be it confidential or subject to privilege, none of which are lost or waived because this email may have been sent to you in error. If you are not the named addressee please let CDM know by return email, permanently delete it from your system and destroy all copies and do not use or disclose the contents. Copyright - This email is subject to copyright and no part of it maybe reproduced in any manner without the written permission of the copyright owner. Privacy - Within the jurisdiction of Australian law, personal information in this email must be dealt with in compliance with the Australian Federal Privacy Act 1988. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multi-Tenant Parking
Has anyone managed to get multi-parking lot call parking working correctly?
I've had several attempts at it, and never seem to be able to get it to go
properly - (actually, at all):
I've most recently done this with 1.6.1.x, and now 1.6.2.x, with no luck in
either case. What I've been trying is the following:
features.conf
[general]
parkext = 100
[featuremap]
[applicationmap]
[parkinglog_customer1-park]
parkext = 100
parkpos = 101-199
findslot = next
context = customer1-park
[parkinglog_customer2-park]
parkext = 100
parkpos = 101-199
findslot = next
context = customer2-park
extensions.conf
[customer1-call-park]
exten = _X.,1,NoOp(The user ${EXTEN} is seeking place a call into the parking
lot...)
exten = _X.,2,Set(PARKINGLOT=customer1-park)
exten =
_X.,3,ParkAndAnnounce(PARKED:call-waiting,60,Local/${ext...@customer1-callback)
exten = _X.,4,Hangup()
[customer2-call-park]
exten = _X.,1,NoOp(The user ${EXTEN} is seeking place a call into the parking
lot...)
exten = _X.,2,Set(PARKINGLOT=customer2-park)
exten =
_X.,3,ParkAndAnnounce(PARKED:call-waiting,60,Local/${ext...@customer2-callback)
exten = _X.,4,Hangup()
Calls are passed to the contexts in extensions by the number of the user trying
to place the call on park. The calls park fine, can be retrieved fine, and the
callbacks work fine (via the customerX-callback) contexts which are not shown
here.
However, it simply does not seem to be putting calls into the parking lots
defined for each customer. It seems to place them all into the default parking
lot regardless of the lot you are trying to put them into. I see a lot of
people having similar issues, and I see some people claiming to have overcome
it, but no actual examples of how it was overcome.
Love anyone's input here! I'm already thinning on top - don't want to lose any
more hair on this one!
Michael Wyres
Technical Specialist
Communications Design Management
Level 1 / 99 King St
Melbourne Victoria 3000
P + 61 3 9601 6600
F + 61 3 9601 6601
mwy...@cdm.com.aublocked::mailto:sbro...@cdm.com.au
[cid:image001.jpg@01CA93A6.2B669DC0]
IMPORTANT NOTICE TO RECIPIENT
Computer viruses - It is your responsibility to scan this email and any
attachments for viruses and defects and rely on those scans as Communications
Design Management Pty Limited (CDM) does not accept any liability for loss or
damage arising from receipt or use of this email or any attachments.
Confidentiality - This email and any attachments are intended for the named
recipient only and may contain personal information, be it confidential or
subject to privilege, none of which are lost or waived because this email may
have been sent to you in error. If you are not the named addressee please let
CDM know by return email, permanently delete it from your system and destroy
all copies and do not use or disclose the contents.
Copyright - This email is subject to copyright and no part of it maybe
reproduced in any manner without the written permission of the copyright owner.
Privacy - Within the jurisdiction of Australian law, personal information in
this email must be dealt with in compliance with the Australian Federal Privacy
Act 1988.
inline: image001.jpg--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-Tenant Parking
At 23:57 1/11/2010, Michael Wyres wrote:
Content-Language: en-US
Content-Type: multipart/related;
boundary=_004_11FDCFCDD2B4B0439630AEC725D1635D1BAC56FDC4ssyd10exinter_;
type=multipart/alternative
Has anyone managed to get multi-parking lot call
parking working correctly? Ive had several
attempts at it, and never seem to be able to get
it to go properly (actually, at all):
Ive most recently done this with 1.6.1.x, and
now 1.6.2.x, with no luck in either case. What
Ive been trying is the following:
features.conf
[general]
parkext = 100
[featuremap]
[applicationmap]
[parkinglog_customer1-park]
parkext = 100
parkpos = 101-199
findslot = next
context = customer1-park
[parkinglog_customer2-park]
parkext = 100
parkpos = 101-199
findslot = next
context = customer2-park
extensions.conf
[customer1-call-park]
exten = _X.,1,NoOp(The user ${EXTEN} is seeking
place a call into the parking lot
)
exten = _X.,2,Set(PARKINGLOT=customer1-park)
exten =
_X.,3,ParkAndAnnounce(PARKED:call-waiting,60,Local/${ext...@customer1-callback)
exten = _X.,4,Hangup()
[customer2-call-park]
exten = _X.,1,NoOp(The user ${EXTEN} is seeking
place a call into the parking lot
)
exten = _X.,2,Set(PARKINGLOT=customer2-park)
exten =
_X.,3,ParkAndAnnounce(PARKED:call-waiting,60,Local/${ext...@customer2-callback)
exten = _X.,4,Hangup()
Calls are passed to the contexts in extensions
by the number of the user trying to place the
call on park. The calls park fine, can be
retrieved fine, and the callbacks work fine (via
the customerX-callback) contexts which are not shown here.
However, it simply does not seem to be putting
calls into the parking lots defined for each
customer. It seems to place them all into the
default parking lot regardless of the lot you
are trying to put them into. I see a lot of
people having similar issues, and I see some
people claiming to have overcome it, but no
actual examples of how it was overcome.
Love anyones input here! Im already thinning
on top dont want to lose any more hair on this one!
Have you looked at this?
http://www.google.com/#q=app_valetparking
Michael Wyres
Technical Specialist
Communications Design Management
Level 1 / 99 King St
Melbourne Victoria 3000
P + 61 3 9601 6600
F + 61 3 9601 6601
blocked::mailto:sbro...@cdm.com.aumwy...@cdm.com.au
cid:image001.jpg@01CA1A98.C7E957F0
IMPORTANT NOTICE TO RECIPIENT
Computer viruses - It is your responsibility to
scan this email and any attachments for viruses
and defects and rely on those scans as
Communications Design Management Pty Limited
(CDM) does not accept any liability for loss or
damage arising from receipt or use of this email or any attachments.
Confidentiality - This email and any attachments
are intended for the named recipient only and
may contain personal information, be it
confidential or subject to privilege, none of
which are lost or waived because this email may
have been sent to you in error. If you are not
the named addressee please let CDM know by
return email, permanently delete it from your
system and destroy all copies and do not use or disclose the contents.
Copyright - This email is subject to copyright
and no part of it maybe reproduced in any manner
without the written permission of the copyright owner.
Privacy - Within the jurisdiction of Australian
law, personal information in this email must be
dealt with in compliance with the Australian Federal Privacy Act 1988.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
inline: 1b794ca8.jpg--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-Tenant Parking
Should that not say parkinglot and not parkinglog in features.conf?
-
Sent from Zimbra and my iPhone!
SplatNIX IT Services :: Innovation through collaboration
On 12 Jan 2010, at 06:03, Michael Wyres mwy...@cdm.com.au wrote:
Has anyone managed to get multi-parking lot call parking working
correctly? I’ve had several attempts at it, and never seem to be ab
le to get it to go properly – (actually, at all):
I’ve most recently done this with 1.6.1.x, and now 1.6.2.x, with no
luck in either case. What I’ve been “trying” is the following:
features.conf
[general]
parkext = 100
[featuremap]
[applicationmap]
[parkinglog_customer1-park]
parkext = 100
parkpos = 101-199
findslot = next
context = customer1-park
[parkinglog_customer2-park]
parkext = 100
parkpos = 101-199
findslot = next
context = customer2-park
extensions.conf
[customer1-call-park]
exten = _X.,1,NoOp(The user ${EXTEN} is seeking place a call into
the parking lot…)
exten = _X.,2,Set(PARKINGLOT=customer1-park)
exten = _X.,3,ParkAndAnnounce(PARKED:call-waiting,60,Local/${EXTEN}
@customer1-callback)
exten = _X.,4,Hangup()
[customer2-call-park]
exten = _X.,1,NoOp(The user ${EXTEN} is seeking place a call into
the parking lot…)
exten = _X.,2,Set(PARKINGLOT=customer2-park)
exten = _X.,3,ParkAndAnnounce(PARKED:call-waiting,60,Local/${EXTEN}
@customer2-callback)
exten = _X.,4,Hangup()
Calls are passed to the contexts in extensions by the number of the
user trying to place the call on park. The calls park fine, can be
retrieved fine, and the callbacks work fine (via the “customerX-call
back”) contexts which are not shown here.
However, it simply does not seem to be putting calls into the
parking lots defined for each customer. It seems to place them all
into the default parking lot regardless of the lot you are trying to
put them into. I see a lot of people having similar issues, and I
see some people claiming to have overcome it, but no actual examples
of how it was “overcome”.
Love anyone’s input here! I’m already thinning on top –
don’t want to lose any more hair on this one!
Michael Wyres
Technical Specialist
Communications Design Management
Level 1 / 99 King St
Melbourne Victoria 3000
P + 61 3 9601 6600
F + 61 3 9601 6601
mwy...@cdm.com.au
image001.jpg
IMPORTANT NOTICE TO RECIPIENT
Computer viruses - It is your responsibility to scan this email and
any attachments for viruses and defects and rely on those scans as
Communications Design Management Pty Limited (CDM) does not accept
any liability for loss or damage arising from receipt or use of this
email or any attachments.
Confidentiality - This email and any attachments are intended for
the named recipient only and may contain personal information, be it
confidential or subject to privilege, none of which are lost or
waived because this email may have been sent to you in error. If you
are not the named addressee please let CDM know by return email,
permanently delete it from your system and destroy all copies and do
not use or disclose the contents.
Copyright - This email is subject to copyright and no part of it
maybe reproduced in any manner without the written permission of the
copyright owner.
Privacy - Within the jurisdiction of Australian law, personal
information in this email must be dealt with in compliance with the
Australian Federal Privacy Act 1988.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multi Tenant Asterisk Server ?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi List, What I hope is a simple question... As the subject states, I would like to know if anyone has setup a Multi Tenant Asterisk Server ? If so, what would I need to do to get to a Multi Tenant setup (preferably an Open Source solution) ? Any suggestions/comments/pointers/URLs ? - -- Gavin Spurgeon. AKA Da Geek - -- The happiest of people don't necessarily have the best of everything, they just make the most of everything that comes along their way.. -BEGIN PGP SIGNATURE- Version: GnuPG/MacGPG2 v2.0.12 (Darwin) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAkr9ucAACgkQvp6arS3vDioyfgCgimKexiFzTRnajuZmljDgHWEQ t9UAnidkNJd8r9hKsiEU4no9jglG7uNF =YHUR -END PGP SIGNATURE- -- This message was scanned by DaGeek Spam Filter and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi Tenant Asterisk Server ?
I added some examples a while back to the extensions.conf.sample and the voicemail.conf.sample code to show how to support distinct domains for voice mail contexts... which was a big obstacle to multi-tenancy... otherwise, you couldn't have individual greetings, etc. For places (like Montreal and Bruxelles) where you need to further tailor context on a per-language basis, that's not been fully exercised... or for states like Indiana and Idaho that exist in two timezones... Actually, that's not entirely true. I tested having a default timezone and then overriding it on a per-SIP context basis and it seemed to work: https://issues.asterisk.org/view.php?id=16090 For POTS or ISDN this would be a little more work, but not impossible. See the [acme] stuff in extensions.conf.sample and voicemail.conf.sample and reply back (on list) if you have questions. -Philip On 11/13/2009 11:55 AM, Gavin Spurgeon wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi List, What I hope is a simple question... As the subject states, I would like to know if anyone has setup a Multi Tenant Asterisk Server ? If so, what would I need to do to get to a Multi Tenant setup (preferably an Open Source solution) ? Any suggestions/comments/pointers/URLs ? - -- Gavin Spurgeon. AKA Da Geek - -- The happiest of people don't necessarily have the best of everything, they just make the most of everything that comes along their way.. -BEGIN PGP SIGNATURE- Version: GnuPG/MacGPG2 v2.0.12 (Darwin) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAkr9ucAACgkQvp6arS3vDioyfgCgimKexiFzTRnajuZmljDgHWEQ t9UAnidkNJd8r9hKsiEU4no9jglG7uNF =YHUR -END PGP SIGNATURE- -- This message was scanned by DaGeek Spam Filter and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi Tenant Asterisk Server ?
On Fri, 2009-11-13 at 19:55 +, Gavin Spurgeon wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi List, What I hope is a simple question... As the subject states, I would like to know if anyone has setup a Multi Tenant Asterisk Server ? If so, what would I need to do to get to a Multi Tenant setup (preferably an Open Source solution) ? Any suggestions/comments/pointers/URLs ? snip Entirely doable and reasonably well documented in the literature. Pay particular attention to the use of contexts. If I recall correctly, the followme and meetme applications do not support contexts. I believe you also have to be careful with SIP ids even in different contexts (someone correct me on that if I'm wrong as Asterisk is only a small part of my job and so the details are not always fresh in my mind). For those, we rely upon some other globally unique attribute, e.g., in our environment, all tenants have a unique posix uid and username. We use that username for the SIP ID and the uid for the meetme and followme identifiers. Hope this helps - John PS - Ah - multi-tenant parking - it is broken as recently as 1.6.1.7. There is a patch which works perfectly. I do not know if that patch was included in 1.6.1.8. In fact, if someone knows, please respond as we need to do that upgrade for security purposes and are concerned about breaking multi-tenant parking. -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi Tenant Asterisk Server ?
On 17:54, Fri 13 Nov 09, John A. Sullivan III wrote: On Fri, 2009-11-13 at 19:55 +, Gavin Spurgeon wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi List, What I hope is a simple question... As the subject states, I would like to know if anyone has setup a Multi Tenant Asterisk Server ? If so, what would I need to do to get to a Multi Tenant setup (preferably an Open Source solution) ? Any suggestions/comments/pointers/URLs ? snip Entirely doable and reasonably well documented in the literature. Pay particular attention to the use of contexts. If I recall correctly, the followme and meetme applications do not support contexts. I believe you also have to be careful with SIP ids even in different contexts (someone correct me on that if I'm wrong as Asterisk is only a small part of my job and so the details are not always fresh in my mind). For those, we rely upon some other globally unique attribute, e.g., in our environment, all tenants have a unique posix uid and username. We use that username for the SIP ID and the uid for the meetme and followme identifiers. Hope this helps - John PS - Ah - multi-tenant parking - it is broken as recently as 1.6.1.7. There is a patch which works perfectly. I do not know if that patch was included in 1.6.1.8. In fact, if someone knows, please respond as we need to do that upgrade for security purposes and are concerned about breaking multi-tenant parking. That patch is not yet in. I'm planning to get it in this weekend. -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi Tenant Asterisk Server ?
On Sat, 2009-11-14 at 00:30 +0100, Michiel van Baak wrote: On 17:54, Fri 13 Nov 09, John A. Sullivan III wrote: On Fri, 2009-11-13 at 19:55 +, Gavin Spurgeon wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi List, What I hope is a simple question... As the subject states, I would like to know if anyone has setup a Multi Tenant Asterisk Server ? If so, what would I need to do to get to a Multi Tenant setup (preferably an Open Source solution) ? Any suggestions/comments/pointers/URLs ? snip Entirely doable and reasonably well documented in the literature. Pay particular attention to the use of contexts. If I recall correctly, the followme and meetme applications do not support contexts. I believe you also have to be careful with SIP ids even in different contexts (someone correct me on that if I'm wrong as Asterisk is only a small part of my job and so the details are not always fresh in my mind). For those, we rely upon some other globally unique attribute, e.g., in our environment, all tenants have a unique posix uid and username. We use that username for the SIP ID and the uid for the meetme and followme identifiers. Hope this helps - John PS - Ah - multi-tenant parking - it is broken as recently as 1.6.1.7. There is a patch which works perfectly. I do not know if that patch was included in 1.6.1.8. In fact, if someone knows, please respond as we need to do that upgrade for security purposes and are concerned about breaking multi-tenant parking. That patch is not yet in. I'm planning to get it in this weekend. snip Thanks for the update. How will it be available at that point? Will there be an immediate 1.6.1.9 release or will it only be via SVN? - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi Tenant Asterisk Server ?
On 18:55, Fri 13 Nov 09, John A. Sullivan III wrote: On Sat, 2009-11-14 at 00:30 +0100, Michiel van Baak wrote: On 17:54, Fri 13 Nov 09, John A. Sullivan III wrote: On Fri, 2009-11-13 at 19:55 +, Gavin Spurgeon wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi List, What I hope is a simple question... As the subject states, I would like to know if anyone has setup a Multi Tenant Asterisk Server ? If so, what would I need to do to get to a Multi Tenant setup (preferably an Open Source solution) ? Any suggestions/comments/pointers/URLs ? snip Entirely doable and reasonably well documented in the literature. Pay particular attention to the use of contexts. If I recall correctly, the followme and meetme applications do not support contexts. I believe you also have to be careful with SIP ids even in different contexts (someone correct me on that if I'm wrong as Asterisk is only a small part of my job and so the details are not always fresh in my mind). For those, we rely upon some other globally unique attribute, e.g., in our environment, all tenants have a unique posix uid and username. We use that username for the SIP ID and the uid for the meetme and followme identifiers. Hope this helps - John PS - Ah - multi-tenant parking - it is broken as recently as 1.6.1.7. There is a patch which works perfectly. I do not know if that patch was included in 1.6.1.8. In fact, if someone knows, please respond as we need to do that upgrade for security purposes and are concerned about breaking multi-tenant parking. That patch is not yet in. I'm planning to get it in this weekend. snip Thanks for the update. How will it be available at that point? Will there be an immediate 1.6.1.9 release or will it only be via SVN? - John not sure yet. Will have a look at it tomorrow and get back to you here ok ? -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multi-tenant parking broken in 1.6.1.1?
Hello, all. My apologies for troubling the developer list as an end
user but we were not able to resolve this issue on the user list and it
is smelling like a possible bug when using multi-tenant call parking.
There seem to be two problems:
1. Parking assigns parking spaces from the default group no matter
what we do.
2. When the parked call timer expires, the callback to the original
callee fails because a | delimiter is used in the Dial()
function.
The second was fixed by backporting a patch from SVN but we still have
the first problem.
Perhaps we have configured it incorrectly. Here is the pertinent
section from features.conf:
[parkinglot_a10] ; EBC
context = a10parking
parkpos = 101-110
;parkext = 100
findslot = next
[parkinglot_a100] ; SSI
context = a100parking
;parkext = 1000
parkpos = 1001-1020
findslot = next
If I understand this correctly, the parkinglog_a100 would be the channel
variable and a100parking the context into which parking extensions are
placed.
We set the channel parameter in sip.conf:
[a100](!,common)
context=a100
vmext=999
parkinglot=parkinglot_a100
subscribecontext=a100
accountcode=a-0100
fromdomain=ssiservices.biz
[userx](a100)
mailbox=...@a100,x...@a100
secret=something
callerid=John A. Sullivan III xxx
fromuser=userid
and we included the context in extensions.conf:
[a100] ; SSI
exten = 911,1,Macro(emergency-US,xx)
exten = 9911,1,Macro(emergency-US,xx)
exten = ,1,VoiceMailMain(${CALLERID(num)}...@a100) ; Direct mail
retrieval
include = a100pub
include = a100conf
include = a100parking
include = US-international
include = dial-uri
We also tried Set(CHANNEL(parkinglot)=parkinglot_a100). We also tried
creating our own parking which yielded interesting data but not
solution.
Here is the console output using the regular setup described:
Call comes in and is answered:
-- SIP/gss-cc01c918 answered SIP/localhost-cc002cf8
-- Native bridging SIP/localhost-cc002cf8 and SIP/gss-cc01c918
-- Started music on hold, class 'default', on SIP/localhost-cc002cf8
== Using SIP RTP TOS bits 176
== Using SIP RTP CoS mark 5
Call is parked:
-- Executing [...@a100:1] Park(SIP/gss-cc05ceb8, ) in new stack
== Parked SIP/gss-cc05ceb8 on 701 (lot default). Will timeout back to
extension [a100] s, 1 in 60 seconds
-- Added extension '701' priority 1 to parkedcalls (0x2cca3f70)
-- SIP/gss-cc05ceb8 Playing 'digits/7.ulaw' (language 'en')
-- SIP/gss-cc05ceb8 Playing 'digits/0.ulaw' (language 'en')
-- SIP/gss-cc05ceb8 Playing 'digits/1.ulaw' (language 'en')
-- Started music on hold, class 'default', on SIP/gss-cc05ceb8
I'm not sure what is happening here but I think this is the original
callee releasing the call. I don't know what the ZOMBIE extension is
about:
== Spawn extension (a100, s, 1) exited non-zero on
'Parked/SIP/gss-cc05ceb8ZOMBIE'
-- Auto fallthrough, channel 'Parked/SIP/gss-cc05ceb8ZOMBIE' status is
'UNKNOWN'
-- Executing [...@a100:1] Answer(Parked/SIP/gss-cc05ceb8ZOMBIE, 0.5)
in new stack
== Spawn extension (a100, h, 1) exited non-zero on
'Parked/SIP/gss-cc05ceb8ZOMBIE'
-- Stopped music on hold on SIP/gss-cc05ceb8
-- Stopped music on hold on SIP/localhost-cc002cf8
-- Started music on hold, class 'default', on SIP/localhost-cc002cf8
== Spawn extension (macro-common, s, 1) exited non-zero on
'SIP/gss-cc05ceb8ZOMBIE' in macro 'common'
== Spawn extension (a100pub, 314, 2) exited non-zero on
'SIP/gss-cc05ceb8ZOMBIE'
== Using SIP RTP TOS bits 176
== Using SIP RTP CoS mark 5
Then we see the destination callee attempting to pick up the call and is
the output of our routine to catch misdialed/unknown extensions:
-- Executing [...@a100:1] GotoIf(SIP/jasiii-cc05ceb8, 0?:_.,1) in new
stack
-- Goto (a100,_.,1)
-- Executing [...@a100:1] Answer(SIP/jasiii-cc05ceb8, 0.5) in new stack
-- Executing [...@a100:2] Playback(SIP/jasiii-cc05ceb8, im-sorry) in
new stack
-- SIP/jasiii-cc05ceb8 Playing 'im-sorry.ulaw' (language 'en')
-- Executing [...@a100:3] Wait(SIP/jasiii-cc05ceb8, 0.0.5) in new stack
-- Executing [...@a100:4] Playback(SIP/jasiii-cc05ceb8,
you-dialed-wrong-number) in new stack
-- SIP/jasiii-cc05ceb8 Playing 'you-dialed-wrong-number.ulaw' (language
'en')
-- Executing [...@a100:5] Wait(SIP/jasiii-cc05ceb8, 0.4) in new stack
-- Executing [...@a100:6] Playback(SIP/jasiii-cc05ceb8, vm-goodbye) in
new stack
-- SIP/jasiii-cc05ceb8 Playing 'vm-goodbye.ulaw' (language 'en')
-- Executing [...@a100:7] Hangup(SIP/jasiii-cc05ceb8, ) in new stack
== Spawn extension (a100, _., 7) exited non-zero on 'SIP/jasiii-cc05ceb8'
-- Executing [...@a100:1] Answer(SIP/jasiii-cc05ceb8, 0.5) in new stack
== Spawn extension (a100, h, 1) exited non-zero on 'SIP/jasiii-cc05ceb8'
We then see the park timeout and fail to return
Re: [asterisk-users] Multi-tenant parking broken in 1.6.1.1?
Oops! Thought I had changed to address! My apologies - John
On Fri, 2009-07-17 at 13:29 -0400, John A. Sullivan III wrote:
Hello, all. My apologies for troubling the developer list as an end
user but we were not able to resolve this issue on the user list and it
is smelling like a possible bug when using multi-tenant call parking.
There seem to be two problems:
1. Parking assigns parking spaces from the default group no matter
what we do.
2. When the parked call timer expires, the callback to the original
callee fails because a | delimiter is used in the Dial()
function.
The second was fixed by backporting a patch from SVN but we still have
the first problem.
Perhaps we have configured it incorrectly. Here is the pertinent
section from features.conf:
[parkinglot_a10] ; EBC
context = a10parking
parkpos = 101-110
;parkext = 100
findslot = next
[parkinglot_a100] ; SSI
context = a100parking
;parkext = 1000
parkpos = 1001-1020
findslot = next
If I understand this correctly, the parkinglog_a100 would be the channel
variable and a100parking the context into which parking extensions are
placed.
We set the channel parameter in sip.conf:
[a100](!,common)
context=a100
vmext=999
parkinglot=parkinglot_a100
subscribecontext=a100
accountcode=a-0100
fromdomain=ssiservices.biz
[userx](a100)
mailbox=...@a100,x...@a100
secret=something
callerid=John A. Sullivan III xxx
fromuser=userid
and we included the context in extensions.conf:
[a100] ; SSI
exten = 911,1,Macro(emergency-US,xx)
exten = 9911,1,Macro(emergency-US,xx)
exten = ,1,VoiceMailMain(${CALLERID(num)}...@a100) ; Direct mail
retrieval
include = a100pub
include = a100conf
include = a100parking
include = US-international
include = dial-uri
We also tried Set(CHANNEL(parkinglot)=parkinglot_a100). We also tried
creating our own parking which yielded interesting data but not
solution.
Here is the console output using the regular setup described:
Call comes in and is answered:
-- SIP/gss-cc01c918 answered SIP/localhost-cc002cf8
-- Native bridging SIP/localhost-cc002cf8 and SIP/gss-cc01c918
-- Started music on hold, class 'default', on SIP/localhost-cc002cf8
== Using SIP RTP TOS bits 176
== Using SIP RTP CoS mark 5
Call is parked:
-- Executing [...@a100:1] Park(SIP/gss-cc05ceb8, ) in new stack
== Parked SIP/gss-cc05ceb8 on 701 (lot default). Will timeout back to
extension [a100] s, 1 in 60 seconds
-- Added extension '701' priority 1 to parkedcalls (0x2cca3f70)
-- SIP/gss-cc05ceb8 Playing 'digits/7.ulaw' (language 'en')
-- SIP/gss-cc05ceb8 Playing 'digits/0.ulaw' (language 'en')
-- SIP/gss-cc05ceb8 Playing 'digits/1.ulaw' (language 'en')
-- Started music on hold, class 'default', on SIP/gss-cc05ceb8
I'm not sure what is happening here but I think this is the original
callee releasing the call. I don't know what the ZOMBIE extension is
about:
== Spawn extension (a100, s, 1) exited non-zero on
'Parked/SIP/gss-cc05ceb8ZOMBIE'
-- Auto fallthrough, channel 'Parked/SIP/gss-cc05ceb8ZOMBIE' status is
'UNKNOWN'
-- Executing [...@a100:1] Answer(Parked/SIP/gss-cc05ceb8ZOMBIE,
0.5) in new stack
== Spawn extension (a100, h, 1) exited non-zero on
'Parked/SIP/gss-cc05ceb8ZOMBIE'
-- Stopped music on hold on SIP/gss-cc05ceb8
-- Stopped music on hold on SIP/localhost-cc002cf8
-- Started music on hold, class 'default', on SIP/localhost-cc002cf8
== Spawn extension (macro-common, s, 1) exited non-zero on
'SIP/gss-cc05ceb8ZOMBIE' in macro 'common'
== Spawn extension (a100pub, 314, 2) exited non-zero on
'SIP/gss-cc05ceb8ZOMBIE'
== Using SIP RTP TOS bits 176
== Using SIP RTP CoS mark 5
Then we see the destination callee attempting to pick up the call and is
the output of our routine to catch misdialed/unknown extensions:
-- Executing [...@a100:1] GotoIf(SIP/jasiii-cc05ceb8, 0?:_.,1) in new
stack
-- Goto (a100,_.,1)
-- Executing [...@a100:1] Answer(SIP/jasiii-cc05ceb8, 0.5) in new
stack
-- Executing [...@a100:2] Playback(SIP/jasiii-cc05ceb8, im-sorry) in
new stack
-- SIP/jasiii-cc05ceb8 Playing 'im-sorry.ulaw' (language 'en')
-- Executing [...@a100:3] Wait(SIP/jasiii-cc05ceb8, 0.0.5) in new
stack
-- Executing [...@a100:4] Playback(SIP/jasiii-cc05ceb8,
you-dialed-wrong-number) in new stack
-- SIP/jasiii-cc05ceb8 Playing 'you-dialed-wrong-number.ulaw' (language
'en')
-- Executing [...@a100:5] Wait(SIP/jasiii-cc05ceb8, 0.4) in new stack
-- Executing [...@a100:6] Playback(SIP/jasiii-cc05ceb8, vm-goodbye)
in new stack
-- SIP/jasiii-cc05ceb8 Playing 'vm-goodbye.ulaw' (language 'en')
-- Executing [...@a100:7] Hangup(SIP/jasiii-cc05ceb8, ) in new stack
== Spawn
Re: [asterisk-users] Multi-tenant parking broken in 1.6.1.1?
Jonathan Thurman jthurma...@gmail.com writes: Sorry, I am relatively new to the Asterisk project and probably don't fully understand how the release cycle for this project works. Are you saying that the minor releases are only for security bugs? Minor releases aren't only for security bugs, in general. This particular one was rushed because a security bug needed fixing, and so there wasn't enough time to properly test the other bug fixes waiting in the queue. Therefore it only contains the security fix. You'll see something similar again when new security holes are found. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-tenant parking broken in 1.6.1.1?
On Wed, 2009-07-01 at 07:17 -0700, Jonathan Thurman wrote:
On Tue, Jun 30, 2009 at 11:53 PM, John A. Sullivan III
jsulli...@opensourcedevel.com wrote:
On Wed, 2009-07-01 at 02:17 -0400, John A. Sullivan III wrote:
Hello, all. With the assistance of very helpful folks, our
brand new
multi-tenant setup seems to be working smoothly from start
to finish
with just a bump or two. The biggest is parking. Now that
we got most
kinks worked out, I'm a little more comfortable in trying to
resolve
this.
There seem to be two problems:
1. Parking assigns parking spaces from the default
group no matter
what we do.
I haven't tested this.
2. When the parked call timer expires, the callback to
the original
callee fails because a | delimiter is used in the
Dial()
function.
This has been fixed in the 1.6.1 SVN, and you will have to back port a
patch until these changes are rolled into another release. I was
disappointed that more bug fixes were not included in 1.6.1.1.
snip
Hello, all. I applied the patch as graciously supplied by Jonathan. It
solves the callback problem of the | delimited Dial parameters but the
basic problem of pulling parking places from the default parking lot
still exists. Same results as last time:
Here is the pertinent
section from features.conf:
[parkinglot_a10] ; EBC
context = a10parking
parkpos = 101-110
;parkext = 100
findslot = next
[parkinglot_a100] ; SSI
context = a100parking
;parkext = 1000
parkpos = 1001-1020
findslot = next
We set the channel parameter in sip.conf:
[a100](!,common)
context=a100
vmext=999
parkinglot=parkinglot_a100
subscribecontext=a100
accountcode=a-0100
fromdomain=ssiservices.biz
[userx](a100)
mailbox=...@a100,x...@a100
secret=something
callerid=John A. Sullivan III xxx
fromuser=userid
and we included the context in extensions.conf:
[a100] ; SSI
exten = 911,1,Macro(emergency-US,xx)
exten = 9911,1,Macro(emergency-US,xx)
exten = ,1,VoiceMailMain(${CALLERID(num)}...@a100) ; Direct mail
retrieval
include = a100pub
include = a100conf
include = a100parking
include = US-international
include = dial-uri
What are we doing wrong? Thanks - John
--
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com
http://www.spiritualoutreach.com
Making Christianity intelligible to secular society
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-tenant parking broken in 1.6.1.1?
On Thu, 2009-07-02 at 17:42 -0400, John A. Sullivan III wrote:
On Wed, 2009-07-01 at 07:17 -0700, Jonathan Thurman wrote:
On Tue, Jun 30, 2009 at 11:53 PM, John A. Sullivan III
jsulli...@opensourcedevel.com wrote:
On Wed, 2009-07-01 at 02:17 -0400, John A. Sullivan III wrote:
Hello, all. With the assistance of very helpful folks, our
brand new
multi-tenant setup seems to be working smoothly from start
to finish
with just a bump or two. The biggest is parking. Now that
we got most
kinks worked out, I'm a little more comfortable in trying to
resolve
this.
There seem to be two problems:
1. Parking assigns parking spaces from the default
group no matter
what we do.
I haven't tested this.
2. When the parked call timer expires, the callback to
the original
callee fails because a | delimiter is used in the
Dial()
function.
This has been fixed in the 1.6.1 SVN, and you will have to back port a
patch until these changes are rolled into another release. I was
disappointed that more bug fixes were not included in 1.6.1.1.
snip
Hello, all. I applied the patch as graciously supplied by Jonathan. It
solves the callback problem of the | delimited Dial parameters but the
basic problem of pulling parking places from the default parking lot
still exists. Same results as last time:
Here is the pertinent
section from features.conf:
[parkinglot_a10] ; EBC
context = a10parking
parkpos = 101-110
;parkext = 100
findslot = next
[parkinglot_a100] ; SSI
context = a100parking
;parkext = 1000
parkpos = 1001-1020
findslot = next
We set the channel parameter in sip.conf:
[a100](!,common)
context=a100
vmext=999
parkinglot=parkinglot_a100
subscribecontext=a100
accountcode=a-0100
fromdomain=ssiservices.biz
[userx](a100)
mailbox=...@a100,x...@a100
secret=something
callerid=John A. Sullivan III xxx
fromuser=userid
and we included the context in extensions.conf:
[a100] ; SSI
exten = 911,1,Macro(emergency-US,xx)
exten = 9911,1,Macro(emergency-US,xx)
exten = ,1,VoiceMailMain(${CALLERID(num)}...@a100) ; Direct mail
retrieval
include = a100pub
include = a100conf
include = a100parking
include = US-international
include = dial-uri
What are we doing wrong? Thanks - John
By the way, I did try it both ways - creating the lot from features.conf
using 700 and creating my own 700 extension for parking using CHANNEL.
Neither worked. Thanks - John
--
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com
http://www.spiritualoutreach.com
Making Christianity intelligible to secular society
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multi-tenant parking broken in 1.6.1.1?
Hello, all. With the assistance of very helpful folks, our brand new
multi-tenant setup seems to be working smoothly from start to finish
with just a bump or two. The biggest is parking. Now that we got most
kinks worked out, I'm a little more comfortable in trying to resolve
this.
There seem to be two problems:
1. Parking assigns parking spaces from the default group no matter
what we do.
2. When the parked call timer expires, the callback to the original
callee fails because a | delimiter is used in the Dial()
function.
Perhaps we have configured it incorrectly. Here is the pertinent
section from features.conf:
[parkinglot_a10] ; EBC
context = a10parking
parkpos = 101-110
;parkext = 100
findslot = next
[parkinglot_a100] ; SSI
context = a100parking
;parkext = 1000
parkpos = 1001-1020
findslot = next
If I understand this correctly, the parkinglog_a100 would be the channel
variable and a100parking the context into which parking extensions are
placed.
We set the channel parameter in sip.conf:
[a100](!,common)
context=a100
vmext=999
parkinglot=parkinglot_a100
subscribecontext=a100
accountcode=a-0100
fromdomain=ssiservices.biz
[userx](a100)
mailbox=...@a100,x...@a100
secret=something
callerid=John A. Sullivan III xxx
fromuser=userid
and we included the context in extensions.conf:
[a100] ; SSI
exten = 911,1,Macro(emergency-US,xx)
exten = 9911,1,Macro(emergency-US,xx)
exten = ,1,VoiceMailMain(${CALLERID(num)}...@a100) ; Direct mail
retrieval
include = a100pub
include = a100conf
include = a100parking
include = US-international
include = dial-uri
We also tried Set(CHANNEL(parkinglot)=parkinglot_a100). We also tried
creating our own parking which yielded interesting data but not
solution.
Here is the console output using the regular setup described:
Call comes in and is answered:
-- SIP/gss-cc01c918 answered SIP/localhost-cc002cf8
-- Native bridging SIP/localhost-cc002cf8 and SIP/gss-cc01c918
-- Started music on hold, class 'default', on SIP/localhost-cc002cf8
== Using SIP RTP TOS bits 176
== Using SIP RTP CoS mark 5
Call is parked:
-- Executing [...@a100:1] Park(SIP/gss-cc05ceb8, ) in new stack
== Parked SIP/gss-cc05ceb8 on 701 (lot default). Will timeout back to
extension [a100] s, 1 in 60 seconds
-- Added extension '701' priority 1 to parkedcalls (0x2cca3f70)
-- SIP/gss-cc05ceb8 Playing 'digits/7.ulaw' (language 'en')
-- SIP/gss-cc05ceb8 Playing 'digits/0.ulaw' (language 'en')
-- SIP/gss-cc05ceb8 Playing 'digits/1.ulaw' (language 'en')
-- Started music on hold, class 'default', on SIP/gss-cc05ceb8
I'm not sure what is happening here but I think this is the original
callee releasing the call. I don't know what the ZOMBIE extension is
about:
== Spawn extension (a100, s, 1) exited non-zero on
'Parked/SIP/gss-cc05ceb8ZOMBIE'
-- Auto fallthrough, channel 'Parked/SIP/gss-cc05ceb8ZOMBIE' status is
'UNKNOWN'
-- Executing [...@a100:1] Answer(Parked/SIP/gss-cc05ceb8ZOMBIE, 0.5)
in new stack
== Spawn extension (a100, h, 1) exited non-zero on
'Parked/SIP/gss-cc05ceb8ZOMBIE'
-- Stopped music on hold on SIP/gss-cc05ceb8
-- Stopped music on hold on SIP/localhost-cc002cf8
-- Started music on hold, class 'default', on SIP/localhost-cc002cf8
== Spawn extension (macro-common, s, 1) exited non-zero on
'SIP/gss-cc05ceb8ZOMBIE' in macro 'common'
== Spawn extension (a100pub, 314, 2) exited non-zero on
'SIP/gss-cc05ceb8ZOMBIE'
== Using SIP RTP TOS bits 176
== Using SIP RTP CoS mark 5
Then we see the destination callee attempting to pick up the call and is
the output of our routine to catch misdialed/unknown extensions:
-- Executing [...@a100:1] GotoIf(SIP/jasiii-cc05ceb8, 0?:_.,1) in new
stack
-- Goto (a100,_.,1)
-- Executing [...@a100:1] Answer(SIP/jasiii-cc05ceb8, 0.5) in new stack
-- Executing [...@a100:2] Playback(SIP/jasiii-cc05ceb8, im-sorry) in
new stack
-- SIP/jasiii-cc05ceb8 Playing 'im-sorry.ulaw' (language 'en')
-- Executing [...@a100:3] Wait(SIP/jasiii-cc05ceb8, 0.0.5) in new stack
-- Executing [...@a100:4] Playback(SIP/jasiii-cc05ceb8,
you-dialed-wrong-number) in new stack
-- SIP/jasiii-cc05ceb8 Playing 'you-dialed-wrong-number.ulaw' (language
'en')
-- Executing [...@a100:5] Wait(SIP/jasiii-cc05ceb8, 0.4) in new stack
-- Executing [...@a100:6] Playback(SIP/jasiii-cc05ceb8, vm-goodbye) in
new stack
-- SIP/jasiii-cc05ceb8 Playing 'vm-goodbye.ulaw' (language 'en')
-- Executing [...@a100:7] Hangup(SIP/jasiii-cc05ceb8, ) in new stack
== Spawn extension (a100, _., 7) exited non-zero on 'SIP/jasiii-cc05ceb8'
-- Executing [...@a100:1] Answer(SIP/jasiii-cc05ceb8, 0.5) in new stack
== Spawn extension (a100, h, 1) exited non-zero on 'SIP/jasiii-cc05ceb8'
We then see the park timeout and fail to return to the original
Re: [asterisk-users] Multi-tenant parking broken in 1.6.1.1?
On Wed, 2009-07-01 at 02:17 -0400, John A. Sullivan III wrote:
Hello, all. With the assistance of very helpful folks, our brand new
multi-tenant setup seems to be working smoothly from start to finish
with just a bump or two. The biggest is parking. Now that we got most
kinks worked out, I'm a little more comfortable in trying to resolve
this.
There seem to be two problems:
1. Parking assigns parking spaces from the default group no matter
what we do.
2. When the parked call timer expires, the callback to the original
callee fails because a | delimiter is used in the Dial()
function.
Perhaps we have configured it incorrectly. Here is the pertinent
section from features.conf:
[parkinglot_a10] ; EBC
context = a10parking
parkpos = 101-110
;parkext = 100
findslot = next
[parkinglot_a100] ; SSI
context = a100parking
;parkext = 1000
parkpos = 1001-1020
findslot = next
If I understand this correctly, the parkinglog_a100 would be the channel
variable and a100parking the context into which parking extensions are
placed.
We set the channel parameter in sip.conf:
[a100](!,common)
context=a100
vmext=999
parkinglot=parkinglot_a100
subscribecontext=a100
accountcode=a-0100
fromdomain=ssiservices.biz
[userx](a100)
mailbox=...@a100,x...@a100
secret=something
callerid=John A. Sullivan III xxx
fromuser=userid
and we included the context in extensions.conf:
[a100] ; SSI
exten = 911,1,Macro(emergency-US,xx)
exten = 9911,1,Macro(emergency-US,xx)
exten = ,1,VoiceMailMain(${CALLERID(num)}...@a100) ; Direct mail
retrieval
include = a100pub
include = a100conf
include = a100parking
include = US-international
include = dial-uri
We also tried Set(CHANNEL(parkinglot)=parkinglot_a100). We also tried
creating our own parking which yielded interesting data but not
solution.
Here is the console output using the regular setup described:
Call comes in and is answered:
-- SIP/gss-cc01c918 answered SIP/localhost-cc002cf8
-- Native bridging SIP/localhost-cc002cf8 and SIP/gss-cc01c918
-- Started music on hold, class 'default', on SIP/localhost-cc002cf8
== Using SIP RTP TOS bits 176
== Using SIP RTP CoS mark 5
Call is parked:
-- Executing [...@a100:1] Park(SIP/gss-cc05ceb8, ) in new stack
== Parked SIP/gss-cc05ceb8 on 701 (lot default). Will timeout back to
extension [a100] s, 1 in 60 seconds
-- Added extension '701' priority 1 to parkedcalls (0x2cca3f70)
-- SIP/gss-cc05ceb8 Playing 'digits/7.ulaw' (language 'en')
-- SIP/gss-cc05ceb8 Playing 'digits/0.ulaw' (language 'en')
-- SIP/gss-cc05ceb8 Playing 'digits/1.ulaw' (language 'en')
-- Started music on hold, class 'default', on SIP/gss-cc05ceb8
I'm not sure what is happening here but I think this is the original
callee releasing the call. I don't know what the ZOMBIE extension is
about:
== Spawn extension (a100, s, 1) exited non-zero on
'Parked/SIP/gss-cc05ceb8ZOMBIE'
-- Auto fallthrough, channel 'Parked/SIP/gss-cc05ceb8ZOMBIE' status is
'UNKNOWN'
-- Executing [...@a100:1] Answer(Parked/SIP/gss-cc05ceb8ZOMBIE,
0.5) in new stack
== Spawn extension (a100, h, 1) exited non-zero on
'Parked/SIP/gss-cc05ceb8ZOMBIE'
-- Stopped music on hold on SIP/gss-cc05ceb8
-- Stopped music on hold on SIP/localhost-cc002cf8
-- Started music on hold, class 'default', on SIP/localhost-cc002cf8
== Spawn extension (macro-common, s, 1) exited non-zero on
'SIP/gss-cc05ceb8ZOMBIE' in macro 'common'
== Spawn extension (a100pub, 314, 2) exited non-zero on
'SIP/gss-cc05ceb8ZOMBIE'
== Using SIP RTP TOS bits 176
== Using SIP RTP CoS mark 5
Then we see the destination callee attempting to pick up the call and is
the output of our routine to catch misdialed/unknown extensions:
-- Executing [...@a100:1] GotoIf(SIP/jasiii-cc05ceb8, 0?:_.,1) in new
stack
-- Goto (a100,_.,1)
-- Executing [...@a100:1] Answer(SIP/jasiii-cc05ceb8, 0.5) in new
stack
-- Executing [...@a100:2] Playback(SIP/jasiii-cc05ceb8, im-sorry) in
new stack
-- SIP/jasiii-cc05ceb8 Playing 'im-sorry.ulaw' (language 'en')
-- Executing [...@a100:3] Wait(SIP/jasiii-cc05ceb8, 0.0.5) in new
stack
-- Executing [...@a100:4] Playback(SIP/jasiii-cc05ceb8,
you-dialed-wrong-number) in new stack
-- SIP/jasiii-cc05ceb8 Playing 'you-dialed-wrong-number.ulaw' (language
'en')
-- Executing [...@a100:5] Wait(SIP/jasiii-cc05ceb8, 0.4) in new stack
-- Executing [...@a100:6] Playback(SIP/jasiii-cc05ceb8, vm-goodbye)
in new stack
-- SIP/jasiii-cc05ceb8 Playing 'vm-goodbye.ulaw' (language 'en')
-- Executing [...@a100:7] Hangup(SIP/jasiii-cc05ceb8, ) in new stack
== Spawn extension (a100, _., 7) exited non-zero on 'SIP/jasiii-cc05ceb8'
-- Executing
Re: [asterisk-users] Multi-tenant parking broken in 1.6.1.1?
On Tue, Jun 30, 2009 at 11:53 PM, John A. Sullivan III jsulli...@opensourcedevel.com wrote: On Wed, 2009-07-01 at 02:17 -0400, John A. Sullivan III wrote: Hello, all. With the assistance of very helpful folks, our brand new multi-tenant setup seems to be working smoothly from start to finish with just a bump or two. The biggest is parking. Now that we got most kinks worked out, I'm a little more comfortable in trying to resolve this. There seem to be two problems: 1. Parking assigns parking spaces from the default group no matter what we do. I haven't tested this. 2. When the parked call timer expires, the callback to the original callee fails because a | delimiter is used in the Dial() function. This has been fixed in the 1.6.1 SVN, and you will have to back port a patch until these changes are rolled into another release. I was disappointed that more bug fixes were not included in 1.6.1.1. -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-tenant parking broken in 1.6.1.1?
This has been fixed in the 1.6.1 SVN, and you will have to back port a patch until these changes are rolled into another release. I was disappointed that more bug fixes were not included in 1.6.1.1. -Jonathan Asterisk 1.6.1.1 was released for a security issue, AST-2009-001. Why would you think that more bug fixes would be in it? Security release are only supposed to have the fix for the issue that caused the release to take place. - Brad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-tenant parking broken in 1.6.1.1?
On Wed, 2009-07-01 at 07:17 -0700, Jonathan Thurman wrote: On Tue, Jun 30, 2009 at 11:53 PM, John A. Sullivan III jsulli...@opensourcedevel.com wrote: On Wed, 2009-07-01 at 02:17 -0400, John A. Sullivan III wrote: Hello, all. With the assistance of very helpful folks, our brand new multi-tenant setup seems to be working smoothly from start to finish with just a bump or two. The biggest is parking. Now that we got most kinks worked out, I'm a little more comfortable in trying to resolve this. There seem to be two problems: 1. Parking assigns parking spaces from the default group no matter what we do. I haven't tested this. 2. When the parked call timer expires, the callback to the original callee fails because a | delimiter is used in the Dial() function. This has been fixed in the 1.6.1 SVN, and you will have to back port a patch until these changes are rolled into another release. I was disappointed that more bug fixes were not included in 1.6.1.1. snip Phew! At least I know I'm not out of my mind! Being fairly new to the Asterisk community, which patch shall I look for and in what section of the SVN? Can I apply it to the release tarball (hopefully) or must I compile out of SVN (which I hate to do in a production environment)? Thanks very much - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-tenant parking broken in 1.6.1.1?
This has been fixed in the 1.6.1 SVN, and you will have to back port a patch until these changes are rolled into another release. I was disappointed that more bug fixes were not included in 1.6.1.1. -Jonathan Asterisk 1.6.1.1 was released for a security issue, AST-2009-001. Why would you think that more bug fixes would be in it? Security release are only supposed to have the fix for the issue that caused the release to take place. - Brad Sorry, I am relatively new to the Asterisk project and probably don't fully understand how the release cycle for this project works. Are you saying that the minor releases are only for security bugs? I haven't seen anything in the on-line documentation that states this. I would think that major usability issues (like parked calls getting dropped if you don't pick them up) would be addressed in a release, not only in SVN. To me the point of a minor release is to fix bugs. It is sometimes quite a headache to download the latest release, have an issue, dig through the issue tracker to find that it was fixed a month ago, then update to SVN or back port a patch. This is especially difficult for those that are new to the project/community. -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-tenant with receptionist features for managed service
On Mon, 16 Mar 2009, Gavin Henry wrote: Dear all, I'm currently researching options for a MT asterisk gui/system for a small business centre that will have 12 units in it. Each unit will be configured for one extension. The system there will have a max of 12 concurrent calls to PSTN provided via an ADSL/SDSL link to our VoIP provider in the UK, using g.711, maybe g.729 dependant on networking costs. Fallback will be to 4 analogue lines should this go down. Gavin, You won't get 12 concurent G711 calls over a standard ADSL line in the UK. If you're on an ADSL2+ service you may get up to 1.1Mb/sec upload speed, but even then, 12 * 80 = 960Kb/sec which is really pushing it, so use G729, or get that 2Mb SDSL line in. Make sure it's a decent ISP too. Using IAX will give you a few extra channels though as the IP overhead is less. What is key is billing information and the ability for a receptionist to see all active calls and do transfers etc. Much like the Flash Operator Panel. Desktop Software may also be needed for this purpose or can be done via a traditional bank of lines on an IP phone accessory module. Have a look at: http://www.astassistant.com/ rather than FOP. Even has a Linux client which is nice... If anyone has any ideas on the best way to put this together, I'm all ears ;-) The consultant in me says Pay someone to do it for you :) However it's not that hard to do and setup if youve done something similar in the past - and your budget is tight. If you know you're going to get more of these, then go for it - spend your time on the software and front-end for the the first one, then the rest are clones... I was going to use an OpenVOX card and Dell T100 box, with 12 Aastra 53i phones. There's a £4k budget for this (still waiting for more into)which will include the networking connection and equipment. If I can afford it I normally go Sangoma with Echo cancellation, but as it's a fallback service, so I'm not bothered. When budgets tight - I've deployed a lot of Grandstream phones - might give you a bit more breathing space if you use (eg) GXP280's for the client phones and a GXP2000 + button box for the receptionist. You can save money by building your own hardware too. Atom mobo, 1GB of RAM and an OpenVox card running oslec is still overkill for this. I mostly use 1GHz VIA boards for these sort of projects with up to 60 extensions. Billings a PITA and other than what I've written myself, have never found anything that works the way I'm happy with... Good luck! I think I've covered everything. There will be many more business centres to come as this first project will be the blueprint one. The end goal is to also move this to a data centre and not have it on site with the pstn fallback options, but use redundant links to our DC. Like a mini-ITSP for our area. I haven't figured the receptionist part for that bit yet though ;-) Personally I'd stick the box on-site and have a central peering server or 2 in the DC - well that's how I do it ;-) You'll struggle to get properly redundant links in that budget range too - one JCB can ruin everyones day! Cheers, Gordon -- www.drogon.net___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-tenant with receptionist features for managed service
We can put about 9/10 calls using SIP/gsm through our BT Business Network ADSL package connection (832kbit upstream, £65/month) before you notice the quality starting to drop, but you could always get two connections and bond them together into one using openvpn or some other method if you wanted to. 2009/3/17 Gordon Henderson gordon+aster...@drogon.netgordon%2baster...@drogon.net On Mon, 16 Mar 2009, Gavin Henry wrote: Dear all, I'm currently researching options for a MT asterisk gui/system for a small business centre that will have 12 units in it. Each unit will be configured for one extension. The system there will have a max of 12 concurrent calls to PSTN provided via an ADSL/SDSL link to our VoIP provider in the UK, using g.711, maybe g.729 dependant on networking costs. Fallback will be to 4 analogue lines should this go down. Gavin, You won't get 12 concurent G711 calls over a standard ADSL line in the UK. If you're on an ADSL2+ service you may get up to 1.1Mb/sec upload speed, but even then, 12 * 80 = 960Kb/sec which is really pushing it, so use G729, or get that 2Mb SDSL line in. Make sure it's a decent ISP too. Using IAX will give you a few extra channels though as the IP overhead is less. What is key is billing information and the ability for a receptionist to see all active calls and do transfers etc. Much like the Flash Operator Panel. Desktop Software may also be needed for this purpose or can be done via a traditional bank of lines on an IP phone accessory module. Have a look at: http://www.astassistant.com/ rather than FOP. Even has a Linux client which is nice... If anyone has any ideas on the best way to put this together, I'm all ears ;-) The consultant in me says Pay someone to do it for you :) However it's not that hard to do and setup if youve done something similar in the past - and your budget is tight. If you know you're going to get more of these, then go for it - spend your time on the software and front-end for the the first one, then the rest are clones... I was going to use an OpenVOX card and Dell T100 box, with 12 Aastra 53i phones. There's a £4k budget for this (still waiting for more into)which will include the networking connection and equipment. If I can afford it I normally go Sangoma with Echo cancellation, but as it's a fallback service, so I'm not bothered. When budgets tight - I've deployed a lot of Grandstream phones - might give you a bit more breathing space if you use (eg) GXP280's for the client phones and a GXP2000 + button box for the receptionist. You can save money by building your own hardware too. Atom mobo, 1GB of RAM and an OpenVox card running oslec is still overkill for this. I mostly use 1GHz VIA boards for these sort of projects with up to 60 extensions. Billings a PITA and other than what I've written myself, have never found anything that works the way I'm happy with... Good luck! I think I've covered everything. There will be many more business centres to come as this first project will be the blueprint one. The end goal is to also move this to a data centre and not have it on site with the pstn fallback options, but use redundant links to our DC. Like a mini-ITSP for our area. I haven't figured the receptionist part for that bit yet though ;-) Personally I'd stick the box on-site and have a central peering server or 2 in the DC - well that's how I do it ;-) You'll struggle to get properly redundant links in that budget range too - one JCB can ruin everyones day! Cheers, Gordon -- www.drogon.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-tenant with receptionist features for managed service
On Tue, 17 Mar 2009, Geraint Lee wrote: We can put about 9/10 calls using SIP/gsm through our BT Business Network ADSL package connection (832kbit upstream, £65/month) before you notice the quality starting to drop, but you could always get two connections and bond them together into one using openvpn or some other method if you wanted to. Ugh. GSM )-: I've never really had much luck with BT as an Internet provider either - their wholesale network - good, retail broadband, bad... In theory, you should be able to get 10 G711 SIP calls over a business quality 830Kb/sec upload ADSL line. I get 9 on my test setup before any packet loss. I managed 11 calls using IAX over the same line before loss. (Entanet ADSL and a Draytek router - £25 a month) Intersting idea re. using openvpn or similar.. I have sites with 3 ADSL connections - one for incoming calls, one for outgoing and one for general office use.. That works when the call numbers in/out is relatively balanced though. I know of a local company who're regularly putting 20 concurrent calls over the same broadband setup using G729... Gordon 2009/3/17 Gordon Henderson gordon+aster...@drogon.netgordon%2baster...@drogon.net On Mon, 16 Mar 2009, Gavin Henry wrote: Dear all, I'm currently researching options for a MT asterisk gui/system for a small business centre that will have 12 units in it. Each unit will be configured for one extension. The system there will have a max of 12 concurrent calls to PSTN provided via an ADSL/SDSL link to our VoIP provider in the UK, using g.711, maybe g.729 dependant on networking costs. Fallback will be to 4 analogue lines should this go down. Gavin, You won't get 12 concurent G711 calls over a standard ADSL line in the UK. If you're on an ADSL2+ service you may get up to 1.1Mb/sec upload speed, but even then, 12 * 80 = 960Kb/sec which is really pushing it, so use G729, or get that 2Mb SDSL line in. Make sure it's a decent ISP too. Using IAX will give you a few extra channels though as the IP overhead is less. What is key is billing information and the ability for a receptionist to see all active calls and do transfers etc. Much like the Flash Operator Panel. Desktop Software may also be needed for this purpose or can be done via a traditional bank of lines on an IP phone accessory module. Have a look at: http://www.astassistant.com/ rather than FOP. Even has a Linux client which is nice... If anyone has any ideas on the best way to put this together, I'm all ears ;-) The consultant in me says Pay someone to do it for you :) However it's not that hard to do and setup if youve done something similar in the past - and your budget is tight. If you know you're going to get more of these, then go for it - spend your time on the software and front-end for the the first one, then the rest are clones... I was going to use an OpenVOX card and Dell T100 box, with 12 Aastra 53i phones. There's a £4k budget for this (still waiting for more into)which will include the networking connection and equipment. If I can afford it I normally go Sangoma with Echo cancellation, but as it's a fallback service, so I'm not bothered. When budgets tight - I've deployed a lot of Grandstream phones - might give you a bit more breathing space if you use (eg) GXP280's for the client phones and a GXP2000 + button box for the receptionist. You can save money by building your own hardware too. Atom mobo, 1GB of RAM and an OpenVox card running oslec is still overkill for this. I mostly use 1GHz VIA boards for these sort of projects with up to 60 extensions. Billings a PITA and other than what I've written myself, have never found anything that works the way I'm happy with... Good luck! I think I've covered everything. There will be many more business centres to come as this first project will be the blueprint one. The end goal is to also move this to a data centre and not have it on site with the pstn fallback options, but use redundant links to our DC. Like a mini-ITSP for our area. I haven't figured the receptionist part for that bit yet though ;-) Personally I'd stick the box on-site and have a central peering server or 2 in the DC - well that's how I do it ;-) You'll struggle to get properly redundant links in that budget range too - one JCB can ruin everyones day! Cheers, Gordon -- www.drogon.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-tenant with receptionist features for managed service
2009/3/17 Gordon Henderson gordon+aster...@drogon.net: On Mon, 16 Mar 2009, Gavin Henry wrote: Dear all, I'm currently researching options for a MT asterisk gui/system for a small business centre that will have 12 units in it. Each unit will be configured for one extension. The system there will have a max of 12 concurrent calls to PSTN provided via an ADSL/SDSL link to our VoIP provider in the UK, using g.711, maybe g.729 dependant on networking costs. Fallback will be to 4 analogue lines should this go down. Gavin, You won't get 12 concurent G711 calls over a standard ADSL line in the UK. If you're on an ADSL2+ service you may get up to 1.1Mb/sec upload speed, but even then, 12 * 80 = 960Kb/sec which is really pushing it, so use G729, or get that 2Mb SDSL line in. Make sure it's a decent ISP too. Using IAX will give you a few extra channels though as the IP overhead is less. Thanks. We're waiting to hear abou twhat we can provide. We use Gradwell for termination and their ADSL. DSL Premium M does 2.5 up, but I'll limit this to 10 calls to be safe. What is key is billing information and the ability for a receptionist to see all active calls and do transfers etc. Much like the Flash Operator Panel. Desktop Software may also be needed for this purpose or can be done via a traditional bank of lines on an IP phone accessory module. Have a look at: http://www.astassistant.com/ rather than FOP. Even has a Linux client which is nice... Looks good. Just tested it on VirtualBox for box. If anyone has any ideas on the best way to put this together, I'm all ears ;-) The consultant in me says Pay someone to do it for you :) However it's not that hard to do and setup if youve done something similar in the past - and your budget is tight. If you know you're going to get more of these, then go for it - spend your time on the software and front-end for the the first one, then the rest are clones... Yeah. I normal use PBXinAFlash for this. Just the receptionist part that was missing and maybe add on a2billing. I was going to use an OpenVOX card and Dell T100 box, with 12 Aastra 53i phones. There's a £4k budget for this (still waiting for more into)which will include the networking connection and equipment. If I can afford it I normally go Sangoma with Echo cancellation, but as it's a fallback service, so I'm not bothered. When budgets tight - I've deployed a lot of Grandstream phones - might give you a bit more breathing space if you use (eg) GXP280's for the client phones and a GXP2000 + button box for the receptionist. Yeah, don't really like them though. I could go down to a 51i for £67 ex VAT. You can save money by building your own hardware too. Atom mobo, 1GB of RAM and an OpenVox card running oslec is still overkill for this. I mostly use 1GHz VIA boards for these sort of projects with up to 60 extensions. What would that come in at? A Dell T100 is £300 ex VAT for 160GB, 1GB RAM and a Dual Core Intel® Pentium® E2220; 2.4GHz with 3yrs nxt bday. A 4 port FXO card is £126.95 ex vat. Billings a PITA and other than what I've written myself, have never found anything that works the way I'm happy with... Good luck! Thanks. I think I've covered everything. There will be many more business centres to come as this first project will be the blueprint one. The end goal is to also move this to a data centre and not have it on site with the pstn fallback options, but use redundant links to our DC. Like a mini-ITSP for our area. I haven't figured the receptionist part for that bit yet though ;-) Personally I'd stick the box on-site and have a central peering server or 2 in the DC - well that's how I do it ;-) You'll struggle to get properly redundant links in that budget range too - one JCB can ruin everyones day! Yeah, as I planned, but not for this project. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-tenant with receptionist features for managed service
2009/3/17 Gordon Henderson gordon+aster...@drogon.net: On Tue, 17 Mar 2009, Geraint Lee wrote: We can put about 9/10 calls using SIP/gsm through our BT Business Network ADSL package connection (832kbit upstream, £65/month) before you notice the quality starting to drop, but you could always get two connections and bond them together into one using openvpn or some other method if you wanted to. Ugh. GSM )-: I've never really had much luck with BT as an Internet provider either - their wholesale network - good, retail broadband, bad... In theory, you should be able to get 10 G711 SIP calls over a business quality 830Kb/sec upload ADSL line. I get 9 on my test setup before any packet loss. I managed 11 calls using IAX over the same line before loss. (Entanet ADSL and a Draytek router - £25 a month) Intersting idea re. using openvpn or similar.. I have sites with 3 ADSL connections - one for incoming calls, one for outgoing and one for general office use.. That works when the call numbers in/out is relatively balanced though. I know of a local company who're regularly putting 20 concurrent calls over the same broadband setup using G729... Yeah, we use g.729 ourselves too. Gavin. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-tenant with receptionist features for managed service
On Tue, 17 Mar 2009, Gavin Henry wrote: 2009/3/17 Gordon Henderson gordon+aster...@drogon.net: On Mon, 16 Mar 2009, Gavin Henry wrote: When budgets tight - I've deployed a lot of Grandstream phones - might give you a bit more breathing space if you use (eg) GXP280's for the client phones and a GXP2000 + button box for the receptionist. Yeah, don't really like them though. I could go down to a 51i for £67 ex VAT. Grandstreams aren't to everyones liking, this is true... You can save money by building your own hardware too. Atom mobo, 1GB of RAM and an OpenVox card running oslec is still overkill for this. I mostly use 1GHz VIA boards for these sort of projects with up to 60 extensions. What would that come in at? A Dell T100 is £300 ex VAT for 160GB, 1GB RAM and a Dual Core Intel® Pentium® E2220; 2.4GHz with 3yrs nxt bday. Under £200 from someone like http://linitx.com/ I don't put disk drives in my boxes though - they boot out of flash. I guess with the Dell, you have on-site or next day replacement if you take that deal though. A 4 port FXO card is £126.95 ex vat. (From voipon by the looks of that price ;-) Billings a PITA and other than what I've written myself, have never found anything that works the way I'm happy with... Good luck! Thanks. I've been approcached by a client who wants a sort of hotel billing system though - tailored to their needs - it's for a retirement home sort of thing. I suggested they just did a fixed-price deal with the inmates, but that didn't go down well. They want to account for everything to the last penny )-: I think I've covered everything. There will be many more business centres to come as this first project will be the blueprint one. The end goal is to also move this to a data centre and not have it on site with the pstn fallback options, but use redundant links to our DC. Like a mini-ITSP for our area. I haven't figured the receptionist part for that bit yet though ;-) Personally I'd stick the box on-site and have a central peering server or 2 in the DC - well that's how I do it ;-) You'll struggle to get properly redundant links in that budget range too - one JCB can ruin everyones day! Yeah, as I planned, but not for this project. Good luck! Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-tenant with receptionist features for managed service
On Tue, 17 Mar 2009, Gavin Henry wrote: 2009/3/17 Gordon Henderson gordon+aster...@drogon.net: On Tue, 17 Mar 2009, Geraint Lee wrote: I know of a local company who're regularly putting 20 concurrent calls over the same broadband setup using G729... Yeah, we use g.729 ourselves too. The issues I've had have been when theres transcoding going on that you can't control - ie. outside your network, so I can go point to point from end-user phone to the people I peer with, but if they then transcode to G711 to go to the PSTN, it's OK, but if it then gets transcoded to GSM for a mobile, or back to G729 to go to an expensive overseas location, then quality does suffer )-: Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-tenant with receptionist features for managed service
A2billing is a good fit for that then. Yeah, voipon. Thanks for the input Gordon. Maybe worth hooking up offline if we're doing similar stuff. Gavin. On 17/03/2009, Gordon Henderson gordon+aster...@drogon.net wrote: On Tue, 17 Mar 2009, Gavin Henry wrote: 2009/3/17 Gordon Henderson gordon+aster...@drogon.net: On Mon, 16 Mar 2009, Gavin Henry wrote: When budgets tight - I've deployed a lot of Grandstream phones - might give you a bit more breathing space if you use (eg) GXP280's for the client phones and a GXP2000 + button box for the receptionist. Yeah, don't really like them though. I could go down to a 51i for £67 ex VAT. Grandstreams aren't to everyones liking, this is true... You can save money by building your own hardware too. Atom mobo, 1GB of RAM and an OpenVox card running oslec is still overkill for this. I mostly use 1GHz VIA boards for these sort of projects with up to 60 extensions. What would that come in at? A Dell T100 is £300 ex VAT for 160GB, 1GB RAM and a Dual Core Intel® Pentium® E2220; 2.4GHz with 3yrs nxt bday. Under £200 from someone like http://linitx.com/ I don't put disk drives in my boxes though - they boot out of flash. I guess with the Dell, you have on-site or next day replacement if you take that deal though. A 4 port FXO card is £126.95 ex vat. (From voipon by the looks of that price ;-) Billings a PITA and other than what I've written myself, have never found anything that works the way I'm happy with... Good luck! Thanks. I've been approcached by a client who wants a sort of hotel billing system though - tailored to their needs - it's for a retirement home sort of thing. I suggested they just did a fixed-price deal with the inmates, but that didn't go down well. They want to account for everything to the last penny )-: I think I've covered everything. There will be many more business centres to come as this first project will be the blueprint one. The end goal is to also move this to a data centre and not have it on site with the pstn fallback options, but use redundant links to our DC. Like a mini-ITSP for our area. I haven't figured the receptionist part for that bit yet though ;-) Personally I'd stick the box on-site and have a central peering server or 2 in the DC - well that's how I do it ;-) You'll struggle to get properly redundant links in that budget range too - one JCB can ruin everyones day! Yeah, as I planned, but not for this project. Good luck! Gordon -- Sent from my mobile device http://www.suretecsystems.com/services/openldap/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-tenant with receptionist features for managed service
Yeah, I've experienced that. But what can you do other than stick woth a fat codec. On 17/03/2009, Gordon Henderson gordon+aster...@drogon.net wrote: On Tue, 17 Mar 2009, Gavin Henry wrote: 2009/3/17 Gordon Henderson gordon+aster...@drogon.net: On Tue, 17 Mar 2009, Geraint Lee wrote: I know of a local company who're regularly putting 20 concurrent calls over the same broadband setup using G729... Yeah, we use g.729 ourselves too. The issues I've had have been when theres transcoding going on that you can't control - ie. outside your network, so I can go point to point from end-user phone to the people I peer with, but if they then transcode to G711 to go to the PSTN, it's OK, but if it then gets transcoded to GSM for a mobile, or back to G729 to go to an expensive overseas location, then quality does suffer )-: Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Sent from my mobile device http://www.suretecsystems.com/services/openldap/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-tenant with receptionist features for managed service
For MT check out Thirdlane's MT PBX: http://www.thirdlane.com/products/thirdlane-pbx-mte I use the PBX Manager which it's based on and it works very well. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-tenant with receptionist features for managed service
On Tue, 17 Mar 2009, Gavin Henry wrote: Yeah, I've experienced that. But what can you do other than stick woth a fat codec. It's tricky. I've been experimenting looking at the possibilitys of using different codecs based on destination, so UK landlines stick to g729 as teh transcode to alaw is OK, but to offshore destiantions look at taking the call in G711... Tricky to get it right without transcoding yourself which you always wnt to avoice (well I do!) Gordon On 17/03/2009, Gordon Henderson gordon+aster...@drogon.net wrote: On Tue, 17 Mar 2009, Gavin Henry wrote: 2009/3/17 Gordon Henderson gordon+aster...@drogon.net: On Tue, 17 Mar 2009, Geraint Lee wrote: I know of a local company who're regularly putting 20 concurrent calls over the same broadband setup using G729... Yeah, we use g.729 ourselves too. The issues I've had have been when theres transcoding going on that you can't control - ie. outside your network, so I can go point to point from end-user phone to the people I peer with, but if they then transcode to G711 to go to the PSTN, it's OK, but if it then gets transcoded to GSM for a mobile, or back to G729 to go to an expensive overseas location, then quality does suffer )-: Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Sent from my mobile device http://www.suretecsystems.com/services/openldap/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multi-tenant with receptionist features for managed service
Dear all, I'm currently researching options for a MT asterisk gui/system for a small business centre that will have 12 units in it. Each unit will be configured for one extension. The system there will have a max of 12 concurrent calls to PSTN provided via an ADSL/SDSL link to our VoIP provider in the UK, using g.711, maybe g.729 dependant on networking costs. Fallback will be to 4 analogue lines should this go down. What is key is billing information and the ability for a receptionist to see all active calls and do transfers etc. Much like the Flash Operator Panel. Desktop Software may also be needed for this purpose or can be done via a traditional bank of lines on an IP phone accessory module. If anyone has any ideas on the best way to put this together, I'm all ears ;-) I was going to use an OpenVOX card and Dell T100 box, with 12 Aastra 53i phones. There's a £4k budget for this (still waiting for more into)which will include the networking connection and equipment. If I can afford it I normally go Sangoma with Echo cancellation, but as it's a fallback service, so I'm not bothered. I think I've covered everything. There will be many more business centres to come as this first project will be the blueprint one. The end goal is to also move this to a data centre and not have it on site with the pstn fallback options, but use redundant links to our DC. Like a mini-ITSP for our area. I haven't figured the receptionist part for that bit yet though ;-) Thanks, Gavin. P.S. I have thought about pbxinaflash and a2billing, but I'm not sure if it would not be clunky for a novice to handle (receptionist). I may go down that route and hire the FreePBX team to fill in the mixing pieces of Multi-tenant if they are interested. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multi tenant
Hi all, i just wanted to know if any one has done any multi-tenant version of the asterisk. thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi tenant
Hi Mujtaba, We have a multi-tenant version of our Asterisk based management and end-user software called Thirdlane PBX Manager. You can see a demo of a single-tenant version on our web site http://www.thirdlane.com/pbxmanager.htm the multi-tenant adds tenant and DID management, and allows to partition Asterisk to manage independent tenants with their own administrators, extensions, routes, queues, etc Please contact me off list for more information. Best regards, Alex Alex Epshteyn Third Lane Technologies, LLC http://www.thirdlane.com _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mujtaba Mahmood Sent: Thursday, October 04, 2007 2:14 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Multi tenant Hi all, i just wanted to know if any one has done any multi-tenant version of the asterisk. thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] multi tenant with queues
You could use a prefix-based agent numbering scheme, like Agent/XXYYY where XX is your customer code and YYY their own agent number. When showing activity to a customer, you strip the XX part or you may leave it alone, as it makes no big confusion to the client. Yours, l. On Fri, 18 Nov 2005 01:13:09 +0100, snacktime [EMAIL PROTECTED] wrote: I'd like some feedback on my solution so far for using queues in a multi tenant configuration. For most of the configuration files I've been able to use a naming scheme for the context names, which works nicely for making multi tenant fairly transparent. However that won't work for everything and queues is one of them. In queues.conf the naming scheme will work for defining a queue. It won't work for the agents though as they all have to have unique names. My thought is to create a pool of available agent numbers, and the web gui for the tenants will let the tenant pick the agent numbers they want to assign out of the pool. As numbers are used they are taken out of the pool, and as they become available they go back into the pool. The downside to this is that a tenant won't get to pick the exact numbers they want, but that doesn't seem like too much of a compromise for a multi tenant system. Anyone have any better ideas? Chris -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] multi tenant with queues
I'd like some feedback on my solution so far for using queues in a multi tenant configuration. For most of the configuration files I've been able to use a naming scheme for the context names, which works nicely for making multi tenant fairly transparent. However that won't work for everything and queues is one of them. In queues.conf the naming scheme will work for defining a queue. It won't work for the agents though as they all have to have unique names. My thought is to create a pool of available agent numbers, and the web gui for the tenants will let the tenant pick the agent numbers they want to assign out of the pool. As numbers are used they are taken out of the pool, and as they become available they go back into the pool. The downside to this is that a tenant won't get to pick the exact numbers they want, but that doesn't seem like too much of a compromise for a multi tenant system. Anyone have any better ideas? Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] multi tenant with queues
I had the exact same dilemma and switched to using AddQueueMember/ RemoveQueueMember instead of using agents. This solved my problem. - Waldo On Nov 17, 2005, at 7:13 PM, snacktime wrote: I'd like some feedback on my solution so far for using queues in a multi tenant configuration. For most of the configuration files I've been able to use a naming scheme for the context names, which works nicely for making multi tenant fairly transparent. However that won't work for everything and queues is one of them. In queues.conf the naming scheme will work for defining a queue. It won't work for the agents though as they all have to have unique names. My thought is to create a pool of available agent numbers, and the web gui for the tenants will let the tenant pick the agent numbers they want to assign out of the pool. As numbers are used they are taken out of the pool, and as they become available they go back into the pool. The downside to this is that a tenant won't get to pick the exact numbers they want, but that doesn't seem like too much of a compromise for a multi tenant system. Anyone have any better ideas? Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] multi tenant with queues
On 11/17/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: I had the exact same dilemma and switched to using AddQueueMember/RemoveQueueMember instead of using agents. This solved my problem. Thanks!! That looks like a better solution all the way around. Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multi-tenant Setup
Why cant you use operator1 and operator2 On 5/3/05, Daniel Salama [EMAIL PROTECTED] wrote: I'm trying to setup a multi-tenant configuration of * and have the following question: In extensions.conf, there is a [global] section that I would normally use to define global variables for my single tenant setups. Now, is there a way to have something like global variables on a per tenant basis, so that I could define something like operator = SIP/123 for tenant A and operator = SIP/456 for tenant B? I read about SetGlobalVar, what I think that would make the variable available to all contexts (in my case tenants). Thanks, Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multi-tenant Setup
I'm trying to setup a multi-tenant configuration of * and have the following question: In extensions.conf, there is a [global] section that I would normally use to define global variables for my single tenant setups. Now, is there a way to have something like global variables on a per tenant basis, so that I could define something like operator = SIP/123 for tenant A and operator = SIP/456 for tenant B? I read about SetGlobalVar, what I think that would make the variable available to all contexts (in my case tenants). Thanks, Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
