|Subject:||Re: [osip-dev] RE-INVITE not supported?|
|Date:||Tue, 29 Nov 2016 22:25:15 +0100|
|User-agent:||Mozilla/5.0 (X11; Linux x86_64; rv:38.0) Gecko/20100101 Thunderbird/38.6.0|
how does libosip2 handle overlap dialling for incoming calls?
Is it necessary, that libosip sends "number incomplete" before the subsequent INVITE arrives (sending more digits)?
I do not know where it is specified (RFC?), but imho the SIP gateway should be able to accept the subsequent INVITE, even if it did not yet send the number incomplete message (e.g. because database lookup is still taking time). Maybe this behaviour is even not specified in RFC, but a common workaround in some gateways, which are used for gatewaying to PSTN in countries, where number length is variable.
(In asterisk this behaviour can be switched on by a conf file variable. Similar in Dialogic's IMG.)
I'm asking, because I'm working on a project, which will need to support overlap dialling, and I hope, I won't run into troubles. The project is based on libosip2, thus I will have to find a solution.
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