its done by SIP UPDATE on polycom/aastra or sip INFO packets on snom when the call is bridged. X-lite does not update anything when it receives them. That's about it. 2010/4/4 Nguyễn Mạnh Hùng <hungngm@bkav.com.vn>
Hi Anthony,
Can you discuss some details in how polycom or snom can do this and x-lite not.
If can, I want to edit some open source soffphone like officeSIP to do this.
Best Regards.
Anthony Minessale <anthony.minessale@gmail.com> [ anthony.minessale@gmail.com]
We already do it. X-Lite does not support it. If you try it with a phone like snom or polycom you will see it works just like that.
2010/4/3 Nguyễn Mạnh Hùng <hungngm@bkav.com.vn>
Hi Seven Du.
Thanks to yours suggetion.
I have an ideal, it is: when the call between caller and agent is set, the caller_id is determined. So, i want to edit code to sent the agent information (the call_id and call_id_number) which will be displayed againt in the agent's softphone (as Xlite..) when the call is happening.
I read some documents but i still can't determine: It's maybe yes or maybe to do this and where to do this.
Can you give me some comments.
Best Regard.
Seven Du <dujinfang@gmail.com> [dujinfang@gmail.com]
As discussed in the list, it's not a freeswitch problem but a reality of life.
Think about customer A and B calls in one after another, then if FreeSWITCH call agent X with caller id A and Y with caller id B, and angent Y answers before X, then
1) if bridge Y with A with the FIFO rule, then the caller id is wrong 2) if bridge Y with B, the caller id is right but it breaks the rule of FIFO - A should be served before B!! And what even worse is that if X never answer A then A never can be served which is really unfair!!
Of course you don't want 1), and you don't need mod_fifo if you want behavior 2), you just need some dialplan trick or some simple Lua script I think. Also FreeSWITCH is designed to be easily extended with almost any languages so feel free to implement anything.
2010/3/31 Nguyễn Mạnh Hùng :
Hi Mike and Seven Du. Thanks to yours help. I known the mechanism of mod_fifo.
http://lists.freeswitch.org/pipermail/freeswitch-users/2009-November/050175.... . What a pity, It can't solve this problem. I can't use freeswitch for my call center. Hope new version can solve this !!!
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