<div dir="ltr"><div><div><div>Hello Jimmy,<br><br></div>Can you let me now below things first.<br><br></div>1) Are you able to make calls with one extension to another extension?<br></div>2) if yes then please send me extensions and dialplan<br>
</div><div class="gmail_extra"><br clear="all"><div><b>Thanks<br>
=====================<br>
Manish Khilwani<br>
Skype: manish_khilwani<br>
Gmail: <a href="mailto:manish.khilwani@gmail.com" target="_blank">manish.khilwani@gmail.com</a><br>
MSN : <a href="mailto:manish.khilwani@hotmail.com" target="_blank">manish.khilwani@hotmail.com</a><br>
Phone : 919099923079</b><br></div>
<br><br><div class="gmail_quote">On Wed, Apr 24, 2013 at 5:09 PM, Jimmy Chang <span dir="ltr"><<a href="mailto:chang33.tw@gmail.com" target="_blank">chang33.tw@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF">
<div>Hi,<br>
<br>
I have set these params.<br>
It seems that the profile of internal and external are all the
same as before and still no audio in port 5080.<br>
Here are the profiles.<br>
<br>
freeswitch@internal> sofia status profile internal<br>
=================================================================================================<br>
Name internal<br>
Domain Name N/A<br>
Auto-NAT false<br>
DBName sofia_reg_internal<br>
Pres Hosts 10.0.110.33,10.0.110.33<br>
Dialplan XML<br>
Context public<br>
Challenge Realm auto_from<br>
RTP-IP 10.0.110.33<br>
Ext-RTP-IP 59.120.xx.xx<br>
SIP-IP 10.0.110.33<br>
URL <a>sip:mod_sofia@10.0.110.33:5060</a><br>
BIND-URL <a>sip:mod_sofia@10.0.110.33:5060</a><br>
HOLD-MUSIC local_stream://moh<br>
OUTBOUND-PROXY N/A<br>
CODECS IN G729,H264<br>
CODECS OUT G729,H264<br>
TEL-EVENT 101<br>
DTMF-MODE rfc2833<br>
CNG 13<br>
SESSION-TO 0<br>
MAX-DIALOG 0<br>
NOMEDIA false<br>
LATE-NEG true<br>
PROXY-MEDIA false<br>
ZRTP-PASSTHRU true<br>
AGGRESSIVENAT false<br>
STUN-ENABLED true<br>
STUN-AUTO-DISABLE false<br>
CALLS-IN 0<br>
FAILED-CALLS-IN 0<br>
CALLS-OUT 0<br>
FAILED-CALLS-OUT 0<br>
REGISTRATIONS 0<br>
<br>
<br>
freeswitch@internal> sofia status profile external<br>
=================================================================================================<br>
Name external<br>
Domain Name N/A<br>
Auto-NAT false<br>
DBName sofia_reg_external<br>
Pres Hosts<br>
Dialplan XML<br>
Context public<br>
Challenge Realm auto_to<br>
RTP-IP 10.0.110.33<br>
Ext-RTP-IP 59.120.xx.xx<br>
SIP-IP 10.0.110.33<br>
Ext-SIP-IP 59.120.xx.xx<br>
URL <a>sip:mod_sofia@59.120.xx.xx:5080</a><br>
BIND-URL
<a>sip:mod_sofia@59.120.xx.xx:5080;maddr=10.0.110.33</a><br>
HOLD-MUSIC local_stream://moh<br>
OUTBOUND-PROXY N/A<br>
CODECS IN G729,H264<br>
CODECS OUT G729,H264<br>
TEL-EVENT 101<br>
DTMF-MODE rfc2833<br>
CNG 13<br>
SESSION-TO 0<br>
MAX-DIALOG 0<br>
NOMEDIA false<br>
LATE-NEG true<br>
PROXY-MEDIA false<br>
ZRTP-PASSTHRU true<br>
AGGRESSIVENAT false<br>
STUN-ENABLED true<br>
STUN-AUTO-DISABLE false<br>
CALLS-IN 0<br>
FAILED-CALLS-IN 0<br>
CALLS-OUT 0<br>
FAILED-CALLS-OUT 0<br>
REGISTRATIONS 0<br>
<br>
<br>
Any advice?<br>
<br>
Thanks.<br>
Jimmy<br>
<br>
<br>
©ó 2013/4/24 ¤U¤È 07:08, pratham2013 ´£¨ì:<br>
</div>
<blockquote type="cite"><div class="im">
<div dir="ltr">
<div>Hello Jimmy,<br>
<br>
</div>
Put the bind-server-ip to auto and set your public ip to
ext-rtp-ip and ext-sip-ip in vars.xml file. Hope it will work.</div>
</div><div class="gmail_extra"><br clear="all">
<div><b><div class="im">Thanks<br>
=====================<br>
Manish Khilwani<br>
Skype: manish_khilwani<br></div><div class="im">
Gmail: <a href="http://user/SendEmail.jtp?type=node&node=7590072&i=0" rel="nofollow" link="external" target="_blank">[hidden
email]</a><br>
MSN : <a href="http://user/SendEmail.jtp?type=node&node=7590072&i=1" rel="nofollow" link="external" target="_blank">[hidden
email]</a><br></div>
Phone : 919099923079</b><br>
</div><div><div class="h5">
<br>
<br>
<div class="gmail_quote">On Wed, Apr 24, 2013 at 4:29 PM, Jimmy
Chang [via freeswitch-users] <span dir="ltr"><<a href="http://user/SendEmail.jtp?type=node&node=7590072&i=2" rel="nofollow" link="external" target="_blank">[hidden
email]</a>></span> wrote:<br>
<blockquote style="border-left:2px solid #cccccc;padding:0 1em" class="gmail_quote">
<div>
<div> Hi,<br>
<br>
In the default configuration, the internal profile
listen port 5060 and external profile listen port 5080.<br>
And follow the setting in <a href="http://wiki.freeswitch.org/wiki/NAT_Traversal" rel="nofollow" link="external" target="_blank">http://wiki.freeswitch.org/wiki/NAT_Traversal</a><br>
vars.xml<br>
<X-PRE-PROCESS cmd="set"
data="bind_server_ip=59.120.xx.xx"/><br>
<X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:<a href="http://stun.freeswitch.org" rel="nofollow" link="external" target="_blank">stun.freeswitch.org</a>"/><br>
<X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:<a href="http://stun.freeswitch.org" rel="nofollow" link="external" target="_blank">stun.freeswitch.org</a>"/><br>
<br>
internal.xml<br>
<param name="ext-rtp-ip"
value="$${external_rtp_ip}"/><br>
<br>
external.xml<br>
<param name="ext-sip-ip"
value="$${external_sip_ip}"/><br>
<param name="ext-rtp-ip"
value="$${external_rtp_ip}"/><br>
<br>
<br>
When user make a SIP call from public network using
external profile(5080), he encountered no audio.<br>
Here is the SDP of 200 OK from FS to user.<br>
<i>v=0</i><i><br>
</i><i>o=FreeSWITCH 1366781127 1366781128 IN IP4
10.0.110.33</i><i><br>
</i><i>s=FreeSWITCH</i><i><br>
</i><i>c=IN IP4 10.0.110.33</i><i><br>
</i><i>t=0 0</i><i><br>
</i><i>m=audio 10568 RTP/AVP 18 101</i><i><br>
</i><i>a=rtpmap:18 G729/8000</i><i><br>
</i><i>a=fmtp:18 annexb=no</i><i><br>
</i><i>a=rtpmap:101 telephone-event/8000</i><i><br>
</i><i>a=fmtp:101 0-16</i><i><br>
</i><i>a=silenceSupp:off - - - -</i><i><br>
</i><i>a=ptime:20</i><i><br>
</i><i>m=video 10530 RTP/AVP 116</i><i><br>
</i><i>a=rtpmap:116 H264/90000</i><i><br>
</i><i>a=fmtp:116 profile-level-id=428014</i><br>
<br>
<br>
When user make a SIP call from public network using
internal profile(5060), everything OK.<br>
Here is the SDP of 200 OK from FS to user.<br>
<i>v=0</i><i><br>
</i><i>o=FreeSWITCH 1366780920 1366780921 IN IP4 </i><i>59.120.xx.xx</i><i><br>
</i><i>s=FreeSWITCH</i><i><br>
</i><i>c=IN IP4 </i><i>59.120.xx.xx</i><i><br>
</i><i>t=0 0</i><i><br>
</i><i>m=audio 10892 RTP/AVP 18 101</i><i><br>
</i><i>a=rtpmap:18 G729/8000</i><i><br>
</i><i>a=fmtp:18 annexb=no</i><i><br>
</i><i>a=rtpmap:101 telephone-event/8000</i><i><br>
</i><i>a=fmtp:101 0-16</i><i><br>
</i><i>a=silenceSupp:off - - - -</i><i><br>
</i><i>a=ptime:20</i><i><br>
</i><i>m=video 10178 RTP/AVP 116</i><i><br>
</i><i>a=rtpmap:116 H264/90000</i><i><br>
</i><i>a=fmtp:116 profile-level-id=428014</i><i><br>
</i><br>
<br>
I can't figure out which parameters in sofia profile or
in dialplan make port 5060 set the public IP in SDP and
work fine.<br>
Any advice?<br>
<br>
Thanks in advance.<br>
Jimmy<br>
<br>
<br>
</div>
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