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Baguk

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Dear All,

I'm just trying to replace my existing Asterisk PBX with the Kerio Operator 2.4 alpha. In gerneral it looks pretty nice, especially now I see features which prevented me to use it in the past, especially to send differnet Caller ID for different extensions in routes.

Unfortunately, I stuck now with call forwarding. I'm interesting in two simple features, which are already part of my existing dialplan:

1. How to preserve original caller ID when forwarding call to mobile? My provider supports "spoofing", however I can not see such options in 2.4, and the reference to "Advanced" tab mentioned in the forum seems to be obsolete.

2. Answer when forwarding. My current dialplan does not answer to a call before the destination mobile answered. Kerio Operator answers (hang on) immideately when call received by PBX and generates than ringtones (or plays MoH) while dialing destination mobile. This causes costs for the caller, as connection is established from the very beginning, even there is no answer of the destination. Can I configure this somehow that Kerio will not answer the call before the final destination answered?

Thank you!
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Filip Jenicek (Kerio)

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Hi,

1. You are right, the alpha version is missing this functionality. It will be re-added in beta 1. I can send you a download link for the beta version to test it once we have it ready. Feel free to contact me on <fjenicek<_at_>kerio.com>.

2. Calls are always answered before being forwarded somewhere else.

Best
Filip
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Baguk

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Hello Filip,

thanks a lot, good news! Yes, please send me a link to Beta Version, I appreciate it.

Regarding the auto answer - may I ask you which reasons are behind this behavior? I would tip that most of people will find it much more convinient (and as well cost friendly) when there will be an option to forward calls without answering it first by PBX. And it's also very simple to do in Asterisk.

Many thanks and best regards,
Baguk
  •  
Filip Jenicek (Kerio)

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To be honest, I don't recall all the reasons. I know that we had to explicitly answer calls coming from hardware cards and even add a slight delay so that the voice prompts played by Asterisk were not cut off. There were also some timeout issues where some SIP providers cancelled an inbound call before it could be connected to the forwarded number.

Can I suggest you to use the "Suggest idea" button. We can reconsider it for one of the future versions.
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Vladimir Toncar (Kerio)

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Calls are often forwarded to cell phones and it may take anything between 10-30 seconds before the mobile network "discovers" the phone and starts ringing it. Meanwhile, the original call may time out.
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Baguk

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Dear Filip and Vladimir,

thanks a lot for your answers. First of all I would like to confirm that the ORGINAL_NUMBER works properly according to my first tests, well done! There are some additional test scenarios I will check with Kerio compared to my existing Asterisk, but anyway it looks promising and very well now. Regarding to the question, I'm using few providers, but the major one is Sipgate Trunking (Germany).

Regarding to the second part of my question, I would accept (my IMHO) if it's limitation of any HW card, but my experience with SIP shows that this is very uncommon behavior, to answer a call before destination has been answered. If the original call has been dropped before destination answered - no worries, the destination will be dropped as well. To be honest the same will happen if you call somebody on mobile directly and than close the connection before his/her answer or initiation of the call. Otherwise what may happen if are forwarding to mobile destination and the destination has neither call waiting nor other function to be informed about missed calls: the caller will hear "free" signals from Kerio, but the destination will get no any info that somebody attempted to call. However, the caller will pay the established connection.

I think to the second topic the most fair, transparent and simple to implement (again IMHO) option would be to add under the "While forwarind a call" additionaly to the "Please Hold", "MoH" and "Ringing" a new option something like "Transfer blind" (or any other wording to avoid missunderstanding). This would prevent other people from the problems you mentioned, but same time will an alternative for people like me, who are using only "quick" SIP providers and when auto-answering of a forwarded call before destination's answer will cause to customers only costs and provide no any value.

PS: Appreciate your efforts and support.

Best Regards,
Baguk
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