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FreeSwitch make ping remote site (gateway) by SIP OPTIONS pakets.
If remote side nothing answers to freeswitch, a gateway state
changes to DOWN. After it calls (INVITES) from dialplan won't be
sent on this gateway.<br>
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29.11.2012 17:12, Mimiko пишет:<br>
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<blockquote cite="mid:50B75F40.30000@gmail.com" type="cite">
<pre wrap="">On 29.11.2012 14:56, "Гришин В.С." wrote:
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<pre wrap="">*From: <a class="moz-txt-link-rfc2396E" href="sip:172.30.0.2"><sip:172.30.0.2></a>;tag=BZN3H5ZBpm8gQ <----in this string no
username
To: <a class="moz-txt-link-rfc2396E" href="sip:172.30.0.2"><sip:172.30.0.2></a> <---- and in this string no username*
it is necessary to make as
From: "USERNAME_STRING" <a class="moz-txt-link-rfc2396E" href="sip:USERNAME_STRING@172.30.0.1"><sip:USERNAME_STRING@172.30.0.1></a>;tag=9DcB7gSNZZ41j
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You have to use
<action application="set" data="effective_caller_id_name=USERNAME_STRING"/>
<action application="set"
data="effective_caller_id_number=USERNAME_STRING"/>
in dialplan before bridging, or in users directory profile.
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<font size="-1"><font face="Arial"><font size="-1">Vladislav</font></font></font><br>
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