hmmm .. can you register using x-lite or some other softphone with the same credentials? <br><br>can you paste a siptrace of the failed registration?<br><br><br>BTW: Make sure nothing is already registered with this credentials when you try with FS<br>
<br>T.<br><br><div class="gmail_quote">On Tue, Sep 22, 2009 at 12:56 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;">
Hi Tihomir,<br>
<br>
Thanks for your help , I added the Asteriskparameters as you described<br>
below, but I still get the same timeout error:<br>
2009-09-22 12:50:52.261103 [WARNING] sofia_reg.c:364 asterisk Failed<br>
Registration, setting retry to 270 seconds.<br>
2009-09-22 12:50:54.324447 [ERR] sofia_reg.c:1460 asterisk Registration<br>
Failed with status Request Timeout [408]. failure #9<br>
<br>
Now, my gateway entry looks like the following :<br>
<div class="im"><br>
<include><br>
<gateway name="asterisk"><br>
<param name="username" value="28"/><br>
</div> <param name="realm" value="192.168.1.119"/><br>
<div class="im"> <param name="proxy" value="192.168.1.119"/><br>
</div><div class="im"> <param name="password" value="test"/><br>
</div><div class="im"> <param name="register" value="true"/><br>
<param name="caller-id-in-from" value="true"/><br>
</div><div class="im"> <param name="sip-port" value="5060"></param><br>
</div> </gateway><br>
</include><br>
<br>
<br>
What can be still wrong here?<br>
<br>
Regards,<br>
<br>
Filip<br>
<br>
<br>
<br>
Tihomir Culjaga schrieb:<br>
<div class="im">> hi Filip,<br>
><br>
><br>
> for calling a user... please read this first:<br>
> <a href="http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_A_Registered_User" target="_blank">http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_A_Registered_User</a><br>
> for making a GW register into e.g. asterisk please use this:<br>
><br>
><br>
> <include><br>
> <gateway name="gw01"><br>
> <param name="username" value="USERNAME_ON_ASTERISK"/><br>
> <param name="realm" value="ASTERISK_IP_ADDRESS"/><br>
> <param name="password" value="PASSWORD_ON_ASTERISK"/><br>
> <param name="register" value="true"/><br>
> <param name="caller-id-in-from" value="true"/><br>
> </gateway><br>
> </include><br>
><br>
> this should be enough to register the GW... after that please read<br>
> this:<br>
> <a href="http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_through_gateways" target="_blank">http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_through_gateways</a><br>
><br>
><br>
> in your case it will be something like this:<br>
><br>
> <extension name="dialGW"><br>
> <condition field="destination_number"<br>
> expression="^(NUMBER_TO_SEND_TO_ASTERISK)$"><br>
> <action application="bridge" data="sofia/gateway/gw01/$1"/><br>
> </condition><br>
> </extension><br>
><br>
><br>
><br>
><br>
><br>
><br>
><br>
><br>
><br>
> On Fri, Sep 18, 2009 at 4:22 PM, Filip Lyncker <<a href="mailto:lyncker@lyth.de">lyncker@lyth.de</a><br>
</div><div><div></div><div class="h5">> <mailto:<a href="mailto:lyncker@lyth.de">lyncker@lyth.de</a>>> wrote:<br>
><br>
> Hi List,<br>
><br>
> for the first experiments with freeswitch I downloaded the Windows<br>
> installation.<br>
> Now Im trying to get my 2 Sipphones get connected to. Later I want<br>
> connect the freeswitch to my asterisk gateway.<br>
><br>
> I find the examples pretty complex therfore Im trying to build up a<br>
> simple solution to understand the functions from the scratch ..<br>
><br>
> my current problem is , that I cant route my local sips to each<br>
> other (<br>
> registration seems to work now).<br>
> the next is , that freeshwitch is not able to connect to asterisk.<br>
> but I<br>
> will describe this later.<br>
><br>
> I installed in the Directory a xml file ( called 22.xml) with the<br>
> following content :<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"<br>
> 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"<br>
> value="24"></variable><br>
> </variables><br>
> </user><br>
> </domain><br>
> </include><br>
><br>
> This seems to be ok now. Now I want to dial from 22 to 24 ,<br>
> wherefore I<br>
> configured this dialplan :<br>
><br>
> <include><br>
> <context name="any"><br>
> <condition field="destination_number" expression="^(2[0-9])$"><br>
><br>
> <action application="bridge" data="user/${dialed_extension}"/><br>
><br>
> </condition><br>
> </include><br>
><br>
> wich doesnt work , mybe b/c the user/${dialed_extension} I dont<br>
> know...<br>
> Freeswitch says:<br>
> [INFO] switch_core_state_machine.c:136 No Route, Aborting<br>
> [NOTICE] switch_core_state_machine.c:137 Hangup<br>
</div></div>> sofia/internal/<a href="mailto:24@192.168.1.34">24@192.168.1.34</a> <mailto:<a href="mailto:24@192.168.1.34">24@192.168.1.34</a>><br>
<div class="im">> [CS_ROUTING] [NO_ROUTE_DESTINATION]<br>
> [NOTICE] switch_core_session.c:1086 Session 17<br>
</div>> (sofia/internal/<a href="mailto:24@192.168.1.34">24@192.168.1.34</a> <mailto:<a href="mailto:24@192.168.1.34">24@192.168.1.34</a>>) Ended<br>
<div class="im">> [NOTICE] switch_core_session.c:1088 Close Channel<br>
</div>> sofia/internal/<a href="mailto:24@192.168.1.34">24@192.168.1.34</a> <mailto:<a href="mailto:24@192.168.1.34">24@192.168.1.34</a>> [CS_DESTROY]<br>
<div><div></div><div class="h5">><br>
> Im sure , for you guys this cant be a big deal;)<br>
><br>
><br>
> Next Point is my Asterisk registration , mybe you can help me out here<br>
> to .. :<br>
><br>
> In the sip-profiles/external I installed the my_asterisk.xml with that<br>
> content :<br>
><br>
> <include><br>
> <gateway name="asterisk"><br>
> <param name="username" value="28"></param><br>
> <param name="password" value="test"></param><br>
> <param name="realm" value="28"></param><br>
> <param name="proxy" value="192.168.1.119"></param><br>
> <param name="register" value="true"></param><br>
> </gateway><br>
> </include><br>
><br>
> Freeswitch allways complains a timeout error :<br>
> [ERR] sofia_reg.c:1460 asterisk Registration Failed with status<br>
> Request<br>
> Timeout [408]. failure #17<br>
> [WARNING] sofia_reg.c:364 asterisk Failed Registration, setting retry<br>
> to 540 seconds.<br>
><br>
> it seems that It cant connect ( I also tried out to explicit set the<br>
> port to 5060 b/c I read something about 5080 .. : <param<br>
> name="sip-port"<br>
> value="5060"></param> but this didnt help)<br>
> In my Asterisk I set in the sip.conf the entry 28 with the pw test<br>
> ....<br>
><br>
><br>
> If someone could help me with my first steps I would be verrry<br>
> thankful ;))<br>
><br>
> cheers<br>
><br>
><br>
> Filip<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>
> _______________________________________________<br>
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> <a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
</div></div>> <mailto:<a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a>><br>
<div class="im">> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
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><br>
><br>
</div>> ------------------------------------------------------------------------<br>
<div class="im">><br>
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><br>
<br>
<br>
</div>--<br>
<div><div></div><div class="h5">_________________________________<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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</div></div></blockquote></div><br>