Colin Stefani
2005-Apr-13 09:11 UTC
[Asterisk-Users] SIP ACD system for station to station calls
[A bit long...sorry...] I have a unique application where we're handling *only* internal calls between extensions in a call center format. We are exploring moving this to a SIP based solution and I'm looking for recommendations/ideas/guidance on how to approach this since I'm new to VoIP in general. Before I get too far, here's the scenario: This system is used for handling truck traffic and processing truckers and their loads in and out of a container facility. We have N number of kiosks the truckers use, average is 20-30 extensions that are ringing in at once on the other side the call center is approximately 10-12 agents receiving these calls. Right now we have an Inter-Tel PBX with analog phones on one side (set up as house phones) and the call center uses digital phones as you'd see with any office phone system. The truckers pull up to a kiosk and press the call button, since these extensions are programmed as house phones to ring an ACD queue they then enter the queue they are designated to go to. There are several queues defined to handle different types of truck traffic. The call center is on the digital end and they login to the ACD queues and take calls from them, the agents may login to one or more queues at once. Thus this creates a closed system by which all calls are routed internally to the system and there are no outside lines. We have a software system that uses OAI to detect events from the PBX and thus provide call control through our client software (answer, hangup, DND, acd login/logout). Enter VoIP: We'd like to have all the end-points (agents and kiosks) be SIP clients (either pure software, a hybrid or simple ATA based). Then we'd like to put a highly available SIP server in the middle. Asterisk may fit this bill, but first off it's not really highly available and my dilemma is this; Asterisk is really good at acting as a proxy and there are also some good modules for advanced ACD/ICD functionality (http://www.voip-info.org/wiki-ICD). However, since we're attempting to design a pure SIP based system end to end, does it really make sense to put something like Asterisk in the middle (or any soft-pbx for that matter)? The goal would be to have the SIP clients setup the calls and then communicate directly with each other using whatever codec we find we like (as you'd expect in an end-to-end SIP system (think interoffice calls, ext to ext). We'll have total control over the entire system and can define and control the SIP clients, the router, the codec on down. So I don't have to worry about Joe User with some crazy codec or SIP client that's not compliant. My sense is no, it doesn't necessarily make sense (and might be overkill) to put a full blown soft PBX in the middle. It seems better to write our own SIP router or use one that's already out there. This would be lighter weight (gotta keep administration in mind too). Further there seems to be very little attention paid to high availability in the PBX world and I can't have the central server fail and lose all the calls at once, there has to be some transparency in failover or at least recovery of state. We have the server infrastructure in Java (www.jboss.org) to support JAIN and SIP servlets should we create our own. But I'd like to find an already existing solution as I hate reinventing the wheel. That said, I'm not sure what to do about the call routing ACD/ICD functionality. Does anyone know of a SIP router that does that or add-on software to perform that function with SIP clients? Where do I start in terms of looking for this? I'd love to hear your ideas or thoughts on this, any approaches that we should consider. I'm in research mode and have my eyes and ears open. We maybe going some place no one else has, but I would find that hard to believe as call centers are so common and it would surprise me if no one has attacked this from a pure VoIP point of view. Colin Stefani Tideworks Technology -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050413/d6454326/attachment.htm
dean collins
2005-Apr-13 11:16 UTC
[Asterisk-Users] SIP ACD system for station to station calls
Colin, Asterisk does exactly what you are after, as for your comment about high availability - it can be. Either research or hire in the expertise - your requirements are already being met in a number of very similar installations. Cheers, Dean ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Colin Stefani Sent: Wednesday, April 13, 2005 12:11 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SIP ACD system for station to station calls [A bit long...sorry...] I have a unique application where we're handling *only* internal calls between extensions in a call center format. We are exploring moving this to a SIP based solution and I'm looking for recommendations/ideas/guidance on how to approach this since I'm new to VoIP in general. Before I get too far, here's the scenario: This system is used for handling truck traffic and processing truckers and their loads in and out of a container facility. We have N number of kiosks the truckers use, average is 20-30 extensions that are ringing in at once on the other side the call center is approximately 10-12 agents receiving these calls. Right now we have an Inter-Tel PBX with analog phones on one side (set up as house phones) and the call center uses digital phones as you'd see with any office phone system. The truckers pull up to a kiosk and press the call button, since these extensions are programmed as house phones to ring an ACD queue they then enter the queue they are designated to go to. There are several queues defined to handle different types of truck traffic. The call center is on the digital end and they login to the ACD queues and take calls from them, the agents may login to one or more queues at once. Thus this creates a closed system by which all calls are routed internally to the system and there are no outside lines. We have a software system that uses OAI to detect events from the PBX and thus provide call control through our client software (answer, hangup, DND, acd login/logout). Enter VoIP: We'd like to have all the end-points (agents and kiosks) be SIP clients (either pure software, a hybrid or simple ATA based). Then we'd like to put a highly available SIP server in the middle. Asterisk may fit this bill, but first off it's not really highly available and my dilemma is this; Asterisk is really good at acting as a proxy and there are also some good modules for advanced ACD/ICD functionality (http://www.voip-info.org/wiki-ICD <http://www.voip-info.org/wiki-ICD> ). However, since we're attempting to design a pure SIP based system end to end, does it really make sense to put something like Asterisk in the middle (or any soft-pbx for that matter)? The goal would be to have the SIP clients setup the calls and then communicate directly with each other using whatever codec we find we like (as you'd expect in an end-to-end SIP system (think interoffice calls, ext to ext). We'll have total control over the entire system and can define and control the SIP clients, the router, the codec on down. So I don't have to worry about Joe User with some crazy codec or SIP client that's not compliant. My sense is no, it doesn't necessarily make sense (and might be overkill) to put a full blown soft PBX in the middle. It seems better to write our own SIP router or use one that's already out there. This would be lighter weight (gotta keep administration in mind too). Further there seems to be very little attention paid to high availability in the PBX world and I can't have the central server fail and lose all the calls at once, there has to be some transparency in failover or at least recovery of state. We have the server infrastructure in Java (www.jboss.org <http://www.jboss.org> ) to support JAIN and SIP servlets should we create our own. But I'd like to find an already existing solution as I hate reinventing the wheel. That said, I'm not sure what to do about the call routing ACD/ICD functionality. Does anyone know of a SIP router that does that or add-on software to perform that function with SIP clients? Where do I start in terms of looking for this? I'd love to hear your ideas or thoughts on this, any approaches that we should consider. I'm in research mode and have my eyes and ears open. We maybe going some place no one else has, but I would find that hard to believe as call centers are so common and it would surprise me if no one has attacked this from a pure VoIP point of view. Colin Stefani Tideworks Technology -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050413/84795f72/attachment.htm