[Users] Accounting Information and Call Transfers

Bogdan-Andrei Iancu bogdan at voice-system.ro
Wed Jul 6 11:16:54 CEST 2005


Hi Günther,

Indeed, accounting the transferred calls may be trick.
A short idea that might help you:
    - spy the REFER request and add to the "Refer-to" header's URI a 
parameter containing the billing user (usually is FROM URI of the REFER) 
- use avpops to do this -> avp_write() to get the from URI into an AVP 
and subst() to added the AVP as a Refer-to URI parameter (for example 
named "billing".
    - look for INVITEs which have in RURI, URI with the "billing" 
parameter; if so, move its value  to the X-Account header; For this you 
need the new avp_subst() function (only in OpenSER 0.10.x)-> the idea 
will be like this: move RURI in an AVP (avp_write), extract from the avp 
only the param (avp_subst) and push the avp as header (avp_pushto).

not sure if it's correct....just a thought.

regards,
bodgan

Günther Starnberger wrote:

>Hello,
>
>I'm currently working on a VOIP network where the phones register to an
>OpenSER server - we also have a PSTN Gateway (Asterisk).
>
>To be able to account the PSTN calls of a user we insert a custom header
>"X-Account" which stores information about the user account. We can't
>just use the From-Header for this as we allow users to e.g. create
>unconditional forwards which forward the call to a defined SIP uri when
>the user isn't available. If somebody creates a forward to the PSTN it
>is still his account which should be accounted and which is therefor put
>in the X-Account header.
>
>This part works fine, but I'm still not sure how to handle call
>transfers - e.g. when a user wants to transfer his conversational
>partner to the PSTN. An unattended transfer in SIP is implemented with a
>REFER request which causes the remote phone to send an INVITE to the SIP
>uri in question[1]. I'm not sure how to account these transfers. On a
>traditional PSTN network the person who caused the transfer would be the
>one who is paying for the call. But I see no solution how to implement
>this behaviour in SIP (because the old call isn't transfered, but the
>conversational partner starts a new call with the INVITE request).
>
>Is it somehow possible to detect if a call was caused by a transfer so
>that I can insert the right X-Account header? Do I need a B2BUA like
>Asterisk to implement the traditional behaviour? How do other VOIP
>providers solve this problem (IMO call transfers shouldn't be such an
>uncommon feature).
>
>tia for all suggestions on how to solve this problem
>/gst
>
>[1]
>http://www.ietf.org/internet-drafts/draft-ietf-sipping-service-examples-08.txt
>
>  
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