Did you compare the siptraces on working vs. non-working calls? Any differences, like 180 vs. 183, etc.?<div><br></div><div>-MC<br><br><div class="gmail_quote">On Thu, Mar 17, 2011 at 10:29 AM, Achim Stamm <span dir="ltr"><<a href="mailto:stamm@lyth.de">stamm@lyth.de</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">No ringback tone, when calling<br>
<br>
Hello,<br>
<br>
i have the following problem:<br>
<br>
I use a phone on line port of LinkSys Spa 3102, which is registered with<br>
number 60 on freeswitch.<br>
I can hear a ringback tone, when I?m calling another registered phone<br>
for example 61 (see dialplan extension Call Voip).<br>
I cannot hear the ring tone on the phone connected to the Spa 3102, when<br>
I'm calling special number 70 leading to a self implemented freeswitch<br>
module<br>
(see dialplan Doing Something).<br>
This problem doesn't appear using X-Lite or an Siemens Voip telephone.<br>
My module does a preanswer as in the code shown below but something must<br>
be different - maybe I have to add somehting to the callhandling.<br>
Does anyone have an idea ?<br>
<br>
Greetings<br>
<br>
Achim Stamm<br>
<br>
<br>
Here is a snippet of my dialplan:<br>
-------------------------------------------------<br>
<extension name="Doing Something"><br>
<condition field="caller_id_number" expression="^6[0-9]$" /><br>
<condition field="destination_number" expression="^7([0-9])$" ><br>
<action application="set" data="call_timeout=120"/><br>
<action application="set" data="hangup_after_bridge=false"/><br>
<action application="set" data="ignore_early_media=true"/><br>
<action application="doingSomething"<br>
data="DoingSomething,5$1,${destination_number}"/><br>
</condition><br>
</extension><br>
<br>
<extension name="Call Voip"><br>
<condition field="destination_number" expression="^.*$" /><br>
<condition field="caller_id_number" expression="6([0-9])"><br>
<action application="bridge"<br>
data="sofia/internal/${destination_number}@${amtsleitung_5$1_ip_address}:5061"<br>
/><br>
<action application="answer"/><br>
</condition><br>
</extension><br>
-------------------------------------------------<br>
<br>
Here is a snippet of my code:<br>
---------------------------------------------------<br>
No ringback tone, when calling<br>
<br>
Hello,<br>
<br>
i have the following problem:<br>
<br>
I use a phone on line port of LinkSys Spa 3102, which is registered with<br>
number 60 on freeswitch.<br>
I can hear a ringback tone, when I?m calling another registered phone<br>
for example 61.<br>
I cannot hear the ring tone on the phone connected to the Spa 3102, when<br>
I'm calling special number 70 leading to a self implemented freeswitch<br>
module.<br>
This problem doesn't appear using X-Lite or an Siemens Voip telephone.<br>
My module does a preanswer as in the code shown below but something must<br>
be different - maybe I have to add somehting to the callhandling. Does<br>
anyone have an idea ?<br>
<br>
Greetings<br>
<br>
Achim Stamm<br>
<br>
<br>
Here is a snippet of my dialplan:<br>
-------------------------------------------------<br>
<extension name="Doing Something"><br>
<condition field="caller_id_number" expression="^6[0-9]$" /><br>
<condition field="destination_number" expression="^7([0-9])$" ><br>
<action application="set" data="call_timeout=120"/><br>
<action application="set" data="hangup_after_bridge=false"/><br>
<action application="set" data="ignore_early_media=true"/><br>
<action application="doingSomething"<br>
data="DoingSomething,5$1,${destination_number}"/><br>
</condition><br>
</extension><br>
<br>
<extension name="Call Voip"><br>
<condition field="destination_number" expression="^.*$" /><br>
<condition field="caller_id_number" expression="6([0-9])"><br>
<action application="bridge"<br>
data="sofia/internal/${destination_number}@${amtsleitung_5$1_ip_address}:5061"<br>
/><br>
<action application="answer"/><br>
</condition><br>
</extension><br>
-------------------------------------------------<br>
<br>
Here is a snippet of my code:<br>
---------------------------------------------------<br>
static switch_status_t doingSomething(switch_core_session_t *session,<br>
bool bWriteAudioLogfile, switch_input_args_t *args)<br>
{<br>
switch_codec_t codec = { 0 };<br>
switch_status_t status;<br>
switch_frame_t *read_frame;<br>
switch_channel_t *channel = switch_core_session_get_channel(session);<br>
<br>
if (switch_channel_pre_answer(channel) != SWITCH_STATUS_SUCCESS) {<br>
return SWITCH_STATUS_FALSE;<br>
}<br>
while (switch_channel_ready(channel) )<br>
{<br>
<br>
if (!process_read_write_frames(session,channel))<br>
break;<br>
}<br>
<br>
switch_channel_hangup(channel,SWITCH_CAUSE_NORMAL_CLEARING);<br>
switch_log_printf(SWITCH_CHANNEL_LOG,SWITCH_LOG_INFO,"doingSomething\n");<br>
return SWITCH_STATUS_SUCCESS;<br>
}<br>
---------------------------------------------------<br>
<br>
<br>
<br>
---------------------------------------------------<br>
<br>
<br>
--<br>
Achim Stamm, Dipl.-Inform. (FH)<br>
<br>
<br>
Lyncker & Theis GmbH<br>
Wilhelmstr. 16<br>
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Germany<br>
<br>
Fon +49 611/9006951<br>
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<br>
<br>
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</blockquote></div><br></div>