[Freeswitch-dev] No ringback tone, when calling
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) Lyncker & Theis GmbH Wilhelmstr. 16 65185 Wiesbaden Germany Fon +49 611/9006951 Fax +49 611/9406125 Handelsregister: HRB 23156 Amtsgericht Wiesbaden Steuernummer: 04323897052 USt-IdNr.: DE255806399 Geschäftsführer: Filip Lyncker, Armin Theis, Patrick Schmidt
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@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)
Lyncker & Theis GmbH Wilhelmstr. 16 65185 Wiesbaden Germany
Fon +49 611/9006951 Fax +49 611/9406125
Handelsregister: HRB 23156 Amtsgericht Wiesbaden Steuernummer: 04323897052 USt-IdNr.: DE255806399
Geschäftsführer: Filip Lyncker, Armin Theis, Patrick Schmidt
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participants (2)
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Achim Stamm
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Michael Collins