Re: [asterisk-dev] SIP Presence using SIP SIMPLE: How?
Hi Olle, adding Kamailio is one option, but I must admit that I don't want to complicate the existing setup any more and Asterisk is working quite well. Actually I can live without the feature - it would just make things more nicer ;) About adding this feature - I can only offer some my work time (and one of my co-workers) to implement this feature. We could start this probably in June. But as I am not an experienced Asterisk-Developer it would be great, if you (or somebody else) could help me in finding the best starting point... Cheers, Dennis Kind regards Dennis Guse Quality and Usability Lab Telekom Innovation Laboratories TU Berlin Ernst-Reuter-Platz 7 D-10587 Berlin, Germany Tel: +49 30 8353 58874 Fax: +49 30 8353 58409 E-mail: dennis.g...@telekom.de Web: www.qu.tlabs.tu-berlin.de On Mon, Apr 28, 2014 at 10:26 AM, Olle E. Johansson wrote: > > On 28 Apr 2014, at 10:15, Dennis Guse wrote: > > Thanks Olle for the explanation. > > Is such a feature planned, so that the presence status of a hinted > extensions can be updated via SIP? > Is anybody interested in such a feature? > > I have an old branch that supports PUBLISH for this. If there's funding, I > can plan on working on this later this year. > > > PS: Switching to Kamailio is not an option as there are some required > features in Asterisk that I would really miss. > > You don't have to switch to Kamailio, you have to ADD kamailio to your > network and keep Asterisk. > > /O > > > --- > Dennis Guse > > Kind regards > > Dennis Guse > > Quality and Usability Lab > Telekom Innovation Laboratories > TU Berlin > Ernst-Reuter-Platz 7 > D-10587 Berlin, Germany > Tel: +49 30 8353 58874 > Fax: +49 30 8353 58409 > E-mail: dennis.g...@telekom.de > Web: www.qu.tlabs.tu-berlin.de > > > On Sun, Apr 27, 2014 at 8:50 PM, Olle E. Johansson wrote: > >> >> On 27 Apr 2014, at 20:01, Dennis Guse >> wrote: >> >> Hallo, >> >> I have successfully activated hints and those are working (NOTIFY is send >> by Asterisk on (un)-register to the subscribed clients). And the presence >> state can be set using CustomPresence, by calling the dialplan function >> PRESENCE_STATE [1]. >> >> However, I have some trouble, if clients are setting there presence state >> the sip way [2], but using Asterisk as proxy (no P2P presence). The clients >> do not send there presence updates to Asterisk, because is not subscribing >> on them (there is no SUBSCRIBE-message from Asterisk to a "hinted" client). >> >> How do I get Asterisk to subscribe on the clients, so Asterisk can the >> presence update and can relay it? Or is this not implemented? >> >> It is not implemented and Asterisk is not a proxy. >> >> Use Kamailio if you want full presence. >> >> /O >> >> >> Software: >> Asterisk is 11.7 on an Ubuntu 14.04 >> The clients we use are based upon PJSIP 2.1. >> >> [1] https://wiki.asterisk.org/wiki/display/AST/Presence+State >> [2] http://www.ietf.org/rfc/rfc3856.txt >> >> --- >> Dennis Guse >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-dev mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-dev >> >> >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-dev mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-dev >> > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-dev > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] SIP Presence using SIP SIMPLE: How?
On 28 Apr 2014, at 10:15, Dennis Guse wrote: > Thanks Olle for the explanation. > > Is such a feature planned, so that the presence status of a hinted extensions > can be updated via SIP? > Is anybody interested in such a feature? I have an old branch that supports PUBLISH for this. If there's funding, I can plan on working on this later this year. > > PS: Switching to Kamailio is not an option as there are some required > features in Asterisk that I would really miss. You don't have to switch to Kamailio, you have to ADD kamailio to your network and keep Asterisk. /O > > --- > Dennis Guse > > Kind regards > > Dennis Guse > > Quality and Usability Lab > Telekom Innovation Laboratories > TU Berlin > Ernst-Reuter-Platz 7 > D-10587 Berlin, Germany > Tel: +49 30 8353 58874 > Fax: +49 30 8353 58409 > E-mail: dennis.g...@telekom.de > Web: www.qu.tlabs.tu-berlin.de > > > On Sun, Apr 27, 2014 at 8:50 PM, Olle E. Johansson wrote: > > On 27 Apr 2014, at 20:01, Dennis Guse wrote: > >> Hallo, >> >> I have successfully activated hints and those are working (NOTIFY is send by >> Asterisk on (un)-register to the subscribed clients). And the presence state >> can be set using CustomPresence, by calling the dialplan function >> PRESENCE_STATE [1]. >> >> However, I have some trouble, if clients are setting there presence state >> the sip way [2], but using Asterisk as proxy (no P2P presence). The clients >> do not send there presence updates to Asterisk, because is not subscribing >> on them (there is no SUBSCRIBE-message from Asterisk to a "hinted" client). >> >> How do I get Asterisk to subscribe on the clients, so Asterisk can the >> presence update and can relay it? Or is this not implemented? > It is not implemented and Asterisk is not a proxy. > > Use Kamailio if you want full presence. > > /O >> >> Software: >> Asterisk is 11.7 on an Ubuntu 14.04 >> The clients we use are based upon PJSIP 2.1. >> >> [1] https://wiki.asterisk.org/wiki/display/AST/Presence+State >> [2] http://www.ietf.org/rfc/rfc3856.txt >> >> --- >> Dennis Guse >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-dev mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-dev > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-dev > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] SIP Presence using SIP SIMPLE: How?
Thanks Olle for the explanation. Is such a feature planned, so that the presence status of a hinted extensions can be updated via SIP? Is anybody interested in such a feature? PS: Switching to Kamailio is not an option as there are some required features in Asterisk that I would really miss. --- Dennis Guse Kind regards Dennis Guse Quality and Usability Lab Telekom Innovation Laboratories TU Berlin Ernst-Reuter-Platz 7 D-10587 Berlin, Germany Tel: +49 30 8353 58874 Fax: +49 30 8353 58409 E-mail: dennis.g...@telekom.de Web: www.qu.tlabs.tu-berlin.de On Sun, Apr 27, 2014 at 8:50 PM, Olle E. Johansson wrote: > > On 27 Apr 2014, at 20:01, Dennis Guse wrote: > > Hallo, > > I have successfully activated hints and those are working (NOTIFY is send > by Asterisk on (un)-register to the subscribed clients). And the presence > state can be set using CustomPresence, by calling the dialplan function > PRESENCE_STATE [1]. > > However, I have some trouble, if clients are setting there presence state > the sip way [2], but using Asterisk as proxy (no P2P presence). The clients > do not send there presence updates to Asterisk, because is not subscribing > on them (there is no SUBSCRIBE-message from Asterisk to a "hinted" client). > > How do I get Asterisk to subscribe on the clients, so Asterisk can the > presence update and can relay it? Or is this not implemented? > > It is not implemented and Asterisk is not a proxy. > > Use Kamailio if you want full presence. > > /O > > > Software: > Asterisk is 11.7 on an Ubuntu 14.04 > The clients we use are based upon PJSIP 2.1. > > [1] https://wiki.asterisk.org/wiki/display/AST/Presence+State > [2] http://www.ietf.org/rfc/rfc3856.txt > > --- > Dennis Guse > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-dev > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] SIP Presence using SIP SIMPLE: How?
On 27 Apr 2014, at 20:01, Dennis Guse wrote: > Hallo, > > I have successfully activated hints and those are working (NOTIFY is send by > Asterisk on (un)-register to the subscribed clients). And the presence state > can be set using CustomPresence, by calling the dialplan function > PRESENCE_STATE [1]. > > However, I have some trouble, if clients are setting there presence state the > sip way [2], but using Asterisk as proxy (no P2P presence). The clients do > not send there presence updates to Asterisk, because is not subscribing on > them (there is no SUBSCRIBE-message from Asterisk to a "hinted" client). > > How do I get Asterisk to subscribe on the clients, so Asterisk can the > presence update and can relay it? Or is this not implemented? It is not implemented and Asterisk is not a proxy. Use Kamailio if you want full presence. /O > > Software: > Asterisk is 11.7 on an Ubuntu 14.04 > The clients we use are based upon PJSIP 2.1. > > [1] https://wiki.asterisk.org/wiki/display/AST/Presence+State > [2] http://www.ietf.org/rfc/rfc3856.txt > > --- > Dennis Guse > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
