Hello For the ringtone try progressinband=yes in sip.conf.
I don't think you can bridge & do a ringback at the same time, why not proxy the RTP and send the ringback yourself using the 'm' modifier? Cheers Jean-Michel. 2009/3/30, [email protected] <[email protected]>: > > Hi Community! > > If this issue was already topic, please excuse or delete my request... > > Topic 1 "no ringtone": > I configured a SIP registration with my SIP provider (SIPCall). > Everything works fine except the ring tone for the caller. The caller > hears silence until the called party takes up the phone. > > I used the DIAL command with the r and R option but no luck... :( > Has anybody the same problem than me and a resolution for it? > > --------- > > Topic 2 "external bridging": > The prior approach was to bridge to external calls. An external SIP > number terminates and will be re-routed back to a mobile phone number. > The session was first packet2packet switched, which did not work. After > setting reinvite=yes, the bridge works. Now I added 2 internal > extensions to the mobile phone number in the DIAL command --> did not > work (mobile phone rings but no communication possible; just silence). > > Topology: > SIP Provider --> Asterisk --> SIP Provider --> Mobile phone > /- ext 10 > /- ext 20 > > > The DIAL command was: > Dial(SIP/[email protected]&SIP/10&SIP/20,,r) > > The aim is that all extensions and the mobile rings and the first pick > up takes the call. During call setup music on hold would be good... > > It shows no errors in the debug of the CLI. > > I would appreciate if somebody could help me. > > Thanks, > Alex > > > ********************************* > This message and any attachments (the "message") are confidential and > intended solely for the addressees. > Any unauthorised use or dissemination is prohibited. > Messages are susceptible to alteration. > France Telecom Group shall not be liable for the message if altered, changed > or falsified. > If you are not the intended addressee of this message, please cancel it > immediately and inform the sender. > ******************************** > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- Jean-Michel Hiver - Synapse co-founder & CTO GSM +262 692 828 070 _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
