[Freeswitch-dev] No ringback tone, when calling

[Michael Collins]

Did you compare the siptraces on working vs. non-working calls? Any
differences, like 180 vs. 183, etc.?

-MC

On Thu, Mar 17, 2011 at 10:29 AM, Achim Stamm <stamm at lyth.de> wrote:

> No ringback tone, when calling
>
> Hello,
>
> i have the following problem:
>
> I use a phone on line port of LinkSys Spa 3102, which is registered with
> number 60 on freeswitch.
> I can hear a ringback tone, when I?m calling another registered phone
> for example 61 (see dialplan extension Call Voip).
> I cannot hear the ring tone on the phone connected to the Spa 3102, when
> I'm calling special number 70 leading to a self implemented freeswitch
> module
> (see dialplan Doing Something).
> This problem doesn't appear using X-Lite or an Siemens Voip telephone.
> My module does a preanswer as in the code shown below but something must
> be different - maybe I have to add somehting to the callhandling.
> Does anyone have an idea ?
>
> Greetings
>
> Achim Stamm
>
>
> Here is a snippet of my dialplan:
> -------------------------------------------------
> <extension name="Doing Something">
> <condition field="caller_id_number" expression="^6[0-9]$" />
> <condition field="destination_number" expression="^7([0-9])$" >
> <action application="set" data="call_timeout=120"/>
> <action application="set" data="hangup_after_bridge=false"/>
> <action application="set" data="ignore_early_media=true"/>
> <action application="doingSomething"
> data="DoingSomething,5$1,${destination_number}"/>
> </condition>
> </extension>
>
> <extension name="Call Voip">
> <condition field="destination_number" expression="^.*$" />
> <condition field="caller_id_number" expression="6([0-9])">
> <action application="bridge"
> data="sofia/internal/${destination_number}@
> ${amtsleitung_5$1_ip_address}:5061"
> />
> <action application="answer"/>
> </condition>
> </extension>
> -------------------------------------------------
>
> Here is a snippet of my code:
> ---------------------------------------------------
> No ringback tone, when calling
>
> Hello,
>
> i have the following problem:
>
> I use a phone on line port of LinkSys Spa 3102, which is registered with
> number 60 on freeswitch.
> I can hear a ringback tone, when I?m calling another registered phone
> for example 61.
> I cannot hear the ring tone on the phone connected to the Spa 3102, when
> I'm calling special number 70 leading to a self implemented freeswitch
> module.
> This problem doesn't appear using X-Lite or an Siemens Voip telephone.
> My module does a preanswer as in the code shown below but something must
> be different - maybe I have to add somehting to the callhandling. Does
> anyone have an idea ?
>
> Greetings
>
> Achim Stamm
>
>
> Here is a snippet of my dialplan:
> -------------------------------------------------
> <extension name="Doing Something">
> <condition field="caller_id_number" expression="^6[0-9]$" />
> <condition field="destination_number" expression="^7([0-9])$" >
> <action application="set" data="call_timeout=120"/>
> <action application="set" data="hangup_after_bridge=false"/>
> <action application="set" data="ignore_early_media=true"/>
> <action application="doingSomething"
> data="DoingSomething,5$1,${destination_number}"/>
> </condition>
> </extension>
>
> <extension name="Call Voip">
> <condition field="destination_number" expression="^.*$" />
> <condition field="caller_id_number" expression="6([0-9])">
> <action application="bridge"
> data="sofia/internal/${destination_number}@
> ${amtsleitung_5$1_ip_address}:5061"
> />
> <action application="answer"/>
> </condition>
> </extension>
> -------------------------------------------------
>
> Here is a snippet of my code:
> ---------------------------------------------------
> static switch_status_t doingSomething(switch_core_session_t *session,
> bool bWriteAudioLogfile, switch_input_args_t *args)
> {
> switch_codec_t codec = { 0 };
> switch_status_t status;
> switch_frame_t *read_frame;
> switch_channel_t *channel = switch_core_session_get_channel(session);
>
> if (switch_channel_pre_answer(channel) != SWITCH_STATUS_SUCCESS) {
> return SWITCH_STATUS_FALSE;
> }
> while (switch_channel_ready(channel) )
> {
>
> if (!process_read_write_frames(session,channel))
> break;
> }
>
> switch_channel_hangup(channel,SWITCH_CAUSE_NORMAL_CLEARING);
> switch_log_printf(SWITCH_CHANNEL_LOG,SWITCH_LOG_INFO,"doingSomething\n");
> return SWITCH_STATUS_SUCCESS;
> }
> ---------------------------------------------------
>
>
>
> ---------------------------------------------------
>
>
> --
> Achim Stamm, Dipl.-Inform. (FH)
>
>
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