[asterisk-users] no ringtone - just silence/bridging of external calls
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/06544564...@sipcall.atSIP/10SIP/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
Re: [asterisk-users] no ringtone - just silence/bridging of external calls
I had a similar situation a while ago and the fix was setting up indications.conf: http://www.voip-info.org/wiki-Asterisk+config+indications.conf -Dave snip 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? /snip ___ -- 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
Re: [asterisk-users] no ringtone - just silence/bridging of external calls
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, alex.mosbur...@orange-ftgroup.com alex.mosbur...@orange-ftgroup.com: 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/06544564...@sipcall.atSIP/10SIP/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
