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[dujinfang at gmail.com]

Hi,

I'm on trunk 13174, and route a call to fifo, but two members ring at
the same time. I want it ring one by one in a round robin manner,
what's wrong with me?

here is fifo.xml

<fifo name="sales_fifo@$${domain}" importance="0">
<member timeout="60" simo="1"
lag="20">{fifo_member_wait=wait}user/1009@$${domain}</member>
<member timeout="60" simo="1"
lag="20">{fifo_member_wait=wait}user/1001@$${domain}</member>
</fifo>

We want to implement a call center where agents register to waiting
customers, when a customer calls in, it will drop in a queue and
search one available agent(in round robin manner). Most fifo functions
seems implemented for scenarios where agents dial in and waiting
callers, which is unnecessary on our condition.


Thanks.

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[anthony.minessale at g...]

learn to think 4th dimensionally =D

Add one <member> with a | sep list in the dial string.


On Mon, Apr 27, 2009 at 11:35 PM, seven <dujinfang@gmail.com (dujinfang@gmail.com)> wrote:

[dujinfang at gmail.com]

Ah, right, that works. I had thought the purpose of members is for sequential hunting. looks I was wrong.

However, add a | sep'ed dial string is hard to do round robin hunting, as we don't want the first agent always busy while others have nothing to do. It is possible to add/delete members using another script to set different dialstring to emulate a round robin hunt, but why not implement it in the queue logic?


questions:


1) What's the purpose for members? just for simultaneous ring?
2) What's the best use case of agents dial in a fifo to wait callers? They just listening to music and waiting if no caller? I guess that would be for very busy call centers.
3) In my test, other members keep ring after one answered, some times it even ring a long time after the caller hangup.



I'm currently using trixbox. when a call comes in, it just play a greeting and ring one free agent and fail over to other agents if no answer or playing moh if all agents are busy. I just want to implement the same logic in FS and replace it as * causing a lot of problems.


I know there are already rules of how to pull a call out from a fifo, and guess it would be possible to add some params to do sequential/round robin hunting for members, and by using dp tools to dynamically add/delete members it would me more powerful.


Thanks.

On Apr 28, 2009, at 8:33 PM, Anthony Minessale wrote:

[gallo at mctelefonia.com]

dujinfang ha scritto:

[anthony.minessale at g...]

keep in mind mod_fifo is not a call center app. it's a simple *fifo* queue hence the name.


On Tue, Apr 28, 2009 at 8:30 AM, dujinfang <dujinfang@gmail.com (dujinfang@gmail.com)> wrote:

[dujinfang at gmail.com]

Thank you for the detailed implementation, comments follows.


On Apr 28, 2009, at 10:24 PM, Anthony Minessale wrote:

[gallo at mctelefonia.com]

dujinfang ha scritto:

[dujinfang at gmail.com]

oh, thank you Antonio. I think it would be better to collect more
ideas before open a bounty. And I more interested in playing(including
patching the code) with that than use the function.

On Apr 28, 2009, at 11:39 PM, Antonio Gallo wrote:

[gallo at mctelefonia.com]

seven ha scritto:

[fdelawarde at wireless...]

Hi, It should be easy to modify mod_fifo to include this functionality. Correct me if I'm wrong: For "call back" agents at least, when X calls are in the the queue, Freeswitch tries to search for up to X agents in database. This algorithm is much more optimized than Asterisk, as Asterisk will take calls one by one and try to connect them to an agent, it should then stay as it is. The simplest idea to control the call distribution algorithm would be to modify the database query in the "find_consumers" function (right now, the algorithm is: "order by outbound_call_count"). A variable could control the "order by" of this query, and the problem would be solved at least for "call back" agents. I guess sqlite3 should allow very complex queries, but I don't know if there could be performance issues. Do you think it is a possible -trivial- solution? François. On Wed, 2009-04-29 at 08:46 +0200, Antonio Gallo wrote: seven ha scritto: > oh, thank you Antonio. I think it would be better to collect more > ideas before open a bounty. And I more interested in playing(including > patching the code) with that than use the function. > I was working on other stuff yesterday and just looked at the wiki: - it seems there is already a bounty for something like that; - there is a wiki page about how to implement it with Javascript, ofc you need to tailor it to your own needs; AgX _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org) http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

[dujinfang at gmail.com]

Actually, for the "call back" agents, because the fifo use originate to start a new session, the new session won't hang up unless one agent answered or timeout. Agents will hear nothing and wait(member_wait=wait) on the queue or hanup(nowait) if caller hang up before an agent answer the phone. '


And I also found out the the member timeout doesn't work but call_timeout works in a dial string. Is it a bug I should reported to jira?


<fifo name="sales_fifo@$${domain}" importance="0">
<member timeout="10" simo="1" lag="5">{call_timeout=6,fifo_member_wait=nowait}user/1009@$${domain}</member>
</fifo>


And even the timeout works, it's not ideal. It's better to bridge to an agent other than originate I think. Keep looking.


On Apr 29, 2009, at 4:27 PM, François Delawarde wrote:

[anthony.minessale at g...]

On Sun, May 3, 2009 at 11:01 PM, seven <dujinfang@gmail.com (dujinfang@gmail.com)> wrote:

[fdelawarde at wireless...]

Hello, Anthony, I would like to provide a patch allowing having different call distribution strategies, at least for "call back" agents. Do you think the simple approach of modifying the SQL query in find_consumers (given strategy that would be set from dialplan) would be enough? Thanks, François. On Mon, 2009-05-04 at 08:25 -0500, Anthony Minessale wrote: On Sun, May 3, 2009 at 11:01 PM, seven <dujinfang@gmail.com (dujinfang@gmail.com)> wrote: Actually, for the "call back" agents, because the fifo use originate to start a new session, the new session won't hang up unless one agent answered or timeout. Agents will hear nothing and wait(member_wait=wait) on the queue or hanup(nowait) if caller hang up before an agent answer the phone. ' When you are using on-hook agents, it's presumed to be under low call volume, you can just set the agents to get popped into the queue in nowait mode so if the caller changed his mind the agent will get a hangup. Remember, if there are X customers in the queue, mod_fifo generates X outbound calls to try to service them. And I also found out the the member timeout doesn't work but call_timeout works in a dial string. Is it a bug I should reported to jira? <fifo name="sales_fifo@$${domain}" importance="0"> <member timeout="10" simo="1" lag="5">{call_timeout=6,fifo_member_wait=nowait}user/1009@$${domain}</member> </fifo> call_timeout is only valid on inbound legs to set the timeout it's willing to wait for a caller to answer. You are confusing it with leg_timeout which is designed to go in the {} And even the timeout works, it's not ideal. It's better to bridge to an agent other than originate I think. Keep looking. I am not sure what you mean by that. bridge instead of originate? The process is to originate the call and then bridge the agent to the caller. All calls in FS start out as origiante???? If you want app_queue you are welcome to download and use it from Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org) Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org) Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org) MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email]) GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email]) IRC: sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email]) googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email]) pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org) http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

[dujinfang at gmail.com]

On May 4, 2009, at 9:25 PM, Anthony Minessale wrote:

[dujinfang at gmail.com]

On May 4, 2009, at 9:25 PM, Anthony Minessale wrote: