asterisk users - Mar 2009 - 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/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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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>

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
>
>
>  *********************************
>  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.
>  ********************************
>
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-- 
Jean-Michel Hiver - Synapse co-founder & CTO
GSM +262 692 828 070