[asterisk-users] no ringtone - just silence/bridging of external calls

[Jean-Michel Hiver]

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.mosburger at orange-ftgroup.com
<alex.mosburger at 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/06544564564 at sipcall.at&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
>
>
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-- 
Jean-Michel Hiver - Synapse co-founder & CTO
GSM +262 692 828 070