[Freeswitch-users] Google voice incoming call doesn't work

[Jian Ren]

mod_dingaling.c line 909:

if (!(tech_pvt->rtp_session
<http://docs.freeswitch.org/structprivate__object.html#d5829b7e4a473ccebac90e44afe586c3>
= switch_rtp_new(tech_pvt->profile
<http://docs.freeswitch.org/structprivate__object.html#6c15f3ddeb0b6b952e9a5c8a2e443426>->ip
<http://docs.freeswitch.org/structmdl__profile.html#38dfeac1fbf9e8d3784dd7eeb746cf6a>,00901

                          tech_pvt->local_port
<http://docs.freeswitch.org/structprivate__object.html#63221941aab4357700497a529d51f2ca>,00902

                          tech_pvt->remote_ip
<http://docs.freeswitch.org/structprivate__object.html#e1cbf100db6a0a4fd7eea10935fa2989>,00903

                          tech_pvt->remote_port
<http://docs.freeswitch.org/structprivate__object.html#ca66c963cc0f4abf206ee9681d91a5ec>,00904

                          tech_pvt->codec_num
<http://docs.freeswitch.org/structprivate__object.html#d1e23e3cec68cdff6135ef4c690aa8c4>,00905

                          tech_pvt->read_codec
<http://docs.freeswitch.org/structprivate__object.html#c696c2e7f6f43f4f24e14c6c2dc81670>.implementation
<http://docs.freeswitch.org/structswitch__codec.html#4e2eb0504df66a664f237ff4f2f9b2b5>->samples_per_packet
<http://docs.freeswitch.org/structswitch__codec__implementation.html#f6c2d7ee05d03910f7fb87cf3056f294>,00906

                          tech_pvt->read_codec
<http://docs.freeswitch.org/structprivate__object.html#c696c2e7f6f43f4f24e14c6c2dc81670>.implementation
<http://docs.freeswitch.org/structswitch__codec.html#4e2eb0504df66a664f237ff4f2f9b2b5>->microseconds_per_packet
<http://docs.freeswitch.org/structswitch__codec__implementation.html#f2bf6153bbb519fbb2523de02581c980>,00907

                          flags, tech_pvt->profile
<http://docs.freeswitch.org/structprivate__object.html#6c15f3ddeb0b6b952e9a5c8a2e443426>->timer_name
<http://docs.freeswitch.org/structmdl__profile.html#f34f270ada1c3c9c2203f4caba80eb8a>,
&err, switch_core_session_get_pool(tech_pvt->session
<http://docs.freeswitch.org/structprivate__object.html#ad69db0ca1011d01ada618b2b588a82d>))))
{00908                 switch_log_printf(SWITCH_CHANNEL_SESSION_LOG
<http://docs.freeswitch.org/switch__types_8h.html#bc899743e2a268974063fd82f654829f>(tech_pvt->session
<http://docs.freeswitch.org/structprivate__object.html#ad69db0ca1011d01ada618b2b588a82d>),
SWITCH_LOG_DEBUG
<http://docs.freeswitch.org/switch__types_8h.html#b74279dba7ee171b8b9d3decfcf13584a213e63b710064a30aa4524ddc30cbc4>,
"RTP ERROR %s\n", err);00909                 switch_channel_hangup
<http://docs.freeswitch.org/group__switch__channel.html#g2d4ca1c1e6bfea7ba35665397b27ec60>(channel
<http://docs.freeswitch.org/group__switch__channel.html#g991acd084cbac25bf6d2a325754ba221>,
SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER
<http://docs.freeswitch.org/switch__types_8h.html#1606cd34d8539e5da76ebdcb89cc36b5f85efe65333a82f93a5e2913707575ae>);00910
                return 0;


Something wrong with the rtp session, what's that?

Thanks!

On Thu, Nov 4, 2010 at 8:41 PM, Jian Ren <renjian at gmail.com> wrote:

> Outcall works but the incoming doesn't. It always said
> "DESTINATION_OUT_OF_ORDER"
>
> 2010-11-04 20:32:31.797390 [NOTICE] switch_channel.c:784 New Channel
> dingaling/1001 [2d48674e-e874-11df-ab9a-9787aa669b3a]
> 2010-11-04 20:32:31.797390 [NOTICE] switch_channel.c:782 Rename Channel
> dingaling/1001->DingaLing/new [2d48674e-e874-11df-ab9a-9787aa669b3a]
> 2010-11-04 20:32:31.797390 [NOTICE] mod_dingaling.c:1104 Ring-Ready
> DingaLing/new!
> 2010-11-04 20:32:32.164739 [NOTICE] mod_dingaling.c:909 Hangup
> DingaLing/new [CS_INIT] [DESTINATION_OUT_OF_ORDER]
> 2010-11-04 20:32:32.165726 [NOTICE] switch_core_session.c:1242 Session 1
> (DingaLing/new) Ended
> 2010-11-04 20:32:32.165726 [NOTICE] switch_core_session.c:1244 Close
> Channel DingaLing/new [CS_DESTROY]
>
> I had skypopen loaded but pointed to another extension, not sure whether
> it's related. Now I commented out mod_skypopen but still got the same
> problem.
> Please help!
>
> My jingling client.xml:
>
> <include>
>   <!-- Client Profile (Original mode) -->
>   <!-- to use this profile take the x- away from the open and close tags so
> its <profile> and$
>   <profile type="client">
>     <param name="name" value="$${xmpp_client_profile}"/>
>     <param name="login" value="xxxx at gmail.com/gtalk"/>
>     <param name="password" value="xxxx"/>
>     <param name="dialplan" value="XML"/>
>     <param name="context" value="public"/>
>     <param name="message" value="Jingle all the way"/>
>     <param name="rtp-ip" value="$${bind_server_ip}"/>
>     <!-- <param name="ext-rtp-ip" value="auto-nat"/> -->
>     <param name="auto-login" value="true"/>
>     <!-- SASL "plain" or "md5" -->
>     <param name="sasl" value="plain"/>
>     <!-- if the server where the jabber is hosted is not the same as the
> one in the jid -->
>     <param name="server" value="talk.google.com"/>
>     <!-- Enable TLS or not -->
>     <param name="tls" value="true"/>
>     <!-- disable to trade async for more calls -->
>     <param name="use-rtp-timer" value="true"/>
>     <!-- default extension (if one cannot be determined) -->
>     <param name="exten" value="1001"/>
>     <!-- VAD choose one -->
>     <!-- <param name="vad" value="in"/> -->
>     <!-- <param name="vad" value="out"/> -->
>     <param name="vad" value="both"/>
>     <!--<param name="avatar" value="/path/to/tiny.jpg"/>-->
>     <param name="candidate-acl" value="wan.auto"/>
>     <param name="local-network-acl" value="localnet.auto"/>
>   </profile>
> </include>
>
> Thanks!
> Jian
>
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