ISABEL ORDAS ARNAL
2011-Oct-21 07:46 UTC
[asterisk-users] how to know RTP por of a SIP client in the dialplan
Hi all, How can I get the RTP port one SIP client is using for sending/receiving RTP flow? Can I obtain it in from SIP_HEADER of something like that in the dialplan? Thank you! Isabel ________________________________ Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra pol?tica de env?o y recepci?n de correo electr?nico en el enlace situado m?s abajo. This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at. http://www.tid.es/ES/PAGINAS/disclaimer.aspx -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111021/cf32ad98/attachment.htm>
Bruce B
2011-Oct-21 13:41 UTC
[asterisk-users] how to know RTP por of a SIP client in the dialplan
Do you need to know to get it in dialplan? If I not, from shell (not Asterisk CLI) I usually use: netstata -a | grep asterisk By default Asterisk settings it should be something between 10k-20k -Bruce On Fri, Oct 21, 2011 at 3:46 AM, ISABEL ORDAS ARNAL <ioa at tid.es> wrote:> Hi all, **** > > How can I get the RTP port one SIP client is using for sending/receiving > RTP flow? Can I obtain it in from SIP_HEADER of something like that in the > dialplan?**** > > Thank you!**** > > ** ** > > Isabel**** > > ------------------------------ > Este mensaje se dirige exclusivamente a su destinatario. Puede consultar > nuestra pol?tica de env?o y recepci?n de correo electr?nico en el enlace > situado m?s abajo. > This message is intended exclusively for its addressee. We only send and > receive email on the basis of the terms set out at. > http://www.tid.es/ES/PAGINAS/disclaimer.aspx > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111021/f7faf373/attachment-0001.htm>
Warren Selby
2011-Oct-21 17:13 UTC
[asterisk-users] how to know RTP por of a SIP client in the dialplan
On Fri, Oct 21, 2011 at 2:46 AM, ISABEL ORDAS ARNAL <ioa at tid.es> wrote:> Hi all, **** > > How can I get the RTP port one SIP client is using for sending/receiving > RTP flow? Can I obtain it in from SIP_HEADER of something like that in the > dialplan?**** > > Thank you!**** > > ** ** > >I don't think you can pull this information from a dialplan native application, but you could probably write an AGI that pulls this information for you. The AGI Environment data includes things like the current channel in use, which should be able to start you off in the right direction. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.com> -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111021/9ddf4cb0/attachment.htm>