and this is not enough for you?<br><br> <!--- The<span style="background-color: rgb(255, 255, 0);"> </span><b style="color: rgb(0, 0, 153); background-color: rgb(255, 255, 0);">%</b><span style="background-color: rgb(255, 255, 0);"> </span>behind the username tells FS to lookup the user in it's local sip_registration database --><br>
<action application="bridge" data="user/${dialed_extension}<span style="background-color: rgb(255, 255, 0);">@</span>${domain_name}"/><br> <!--- x.x.x.x in the line above is the IP address to the FreeSWITCH server/device --><br>
<!--- If you don't want to bridge a call to a local registered user, but to a SIP URI, use the <span style="background-color: rgb(255, 255, 0);">@</span> instead of <span style="background-color: rgb(255, 255, 0);">%</span>:<br>
<action application="bridge" data="sofia/profilename/500<span style="background-color: rgb(255, 255, 0);">@</span>x.x.x.x"/> --><br><br>T.<br><br><br><div class="gmail_quote">On Tue, Sep 22, 2009 at 1:52 PM, Filip Lyncker <span dir="ltr"><<a href="mailto:lyncker@lyth.de">lyncker@lyth.de</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Dear List,<br>
<br>
I read the documentation, but Im still confused about how to dial a<br>
internal registered sip user.<br>
<br>
I configured the both sip phones in the directory in my local.xml file :<br>
<br>
<include><br>
<domain name="$${domain}"><br>
<user id="22" mailbox="22"><br>
<params><br>
<param name="password" value="Xk21%"></param><br>
<param name="vm-password" value="22"></param><br>
<param name="sip-port" value="5060"></param><br>
<br>
</params><br>
<variables><br>
<variable name="accountcode" value="22"></variable><br>
<variable name="user_context" value="default"></variable><br>
<variable name="effective_caller_id_name" value="Extension<br>
22"></variable><br>
<variable name="effective_caller_id_number" value="22"></variable><br>
</variables><br>
</user><br>
<user id="24" mailbox="24"><br>
<params><br>
<param name="password" value="dudeldum"></param><br>
<param name="vm-password" value="24"></param><br>
<param name="sip-port" value="5060"></param><br>
<br>
</params><br>
<variables><br>
<variable name="accountcode" value="24"></variable><br>
<variable name="user_context" value="default"></variable><br>
<variable name="effective_caller_id_name" value="Extension<br>
24"></variable><br>
<variable name="effective_caller_id_number" value="24"></variable><br>
</variables><br>
</user><br>
</domain><br>
</include><br>
<br>
It seems, that they can connect to the freeswitch.<br>
<br>
I configured the dialplan like following :<br>
<br>
<include><br>
<context name="default"><br>
<extension name="diallocal"><br>
<condition field="destination_number" expression="^(2[0-9])$"><br>
<!--- The % behind the username tells FS to lookup the user in<br>
it's local sip_registration database --><br>
<action application="bridge"<br>
data="user/${dialed_extension}@${domain_name}"></action><br>
<!--- x.x.x.x in the line above is the IP address to the<br>
FreeSWITCH server/device --><br>
<!--- If you don't want to bridge a call to a local registered<br>
user, but to a SIP URI, use the @ instead of %:<br>
<action application="bridge"<br>
data="sofia/profilename/500@x.x.x.x"/> --><br>
</condition><br>
</extension><br>
...<br>
<br>
<br>
If I call from the sip user 24 to 22 , freeswitch logs the following and<br>
gives an busy tone immediately:<br>
<br>
freeswitch@Bigfish> 2009-09-22 13:50:29.367114 [NOTICE]<br>
switch_channel.c:602 New Channel sofia/internal/<a href="mailto:24@192.168.1.34">24@192.168.1.34</a><br>
[decc119c-a973-6b4c-bf11-ec251c653cda]<br>
2009-09-22 13:50:29.372973 [INFO] mod_dialplan_xml.c:315 Processing<br>
24->22 in context default<br>
2009-09-22 13:50:29.372973 [WARNING] mod_dptools.c:2365 Can't find user<br>
[@<a href="http://192.168.1.34" target="_blank">192.168.1.34</a>]<br>
2009-09-22 13:50:29.372973 [ERR] switch_ivr_originate.c:1510 Cannot<br>
create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT]<br>
2009-09-22 13:50:29.372973 [INFO] mod_dptools.c:2093 Originate Failed.<br>
Cause: SUBSCRIBER_ABSENT<br>
2009-09-22 13:50:29.372973 [NOTICE] mod_dptools.c:2125 Hangup<br>
sofia/internal/<a href="mailto:24@192.168.1.34">24@192.168.1.34</a> [CS_EXECUTE] [SUBSCRIBER_ABSENT]<br>
2009-09-22 13:50:29.390550 [NOTICE] switch_core_session.c:1086 Session<br>
13 (sofia/internal/<a href="mailto:24@192.168.1.34">24@192.168.1.34</a>) Ended<br>
2009-09-22 13:50:29.390550 [NOTICE] switch_core_session.c:1088 Close<br>
Channel sofia/internal/<a href="mailto:24@192.168.1.34">24@192.168.1.34</a> [CS_DESTROY]<br>
<br>
thanks again for your help ...<br>
<br>
<br>
regards,<br>
<br>
Filip<br>
<br>
<br>
--<br>
_________________________________<br>
Filip Lyncker, Dipl.-Inform. (FH)<br>
<br>
<br>
Lyncker & Theis GmbH<br>
Wilhelmstr. 16<br>
65185 Wiesbaden<br>
Germany<br>
<br>
Fon +49 611/9006951<br>
Fax +49 611/9406125<br>
<br>
<br>
Handelsregister: HRB 23156 Amtsgericht Wiesbaden<br>
Steuernummer: 4023897051<br>
USt-IdNr.: DE255806399<br>
<br>
Geschäftsführer:<br>
Filip Lyncker,<br>
Armin Theis<br>
<br>
<br>
<br>
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</blockquote></div><br>