User directory = Inbound registration<br>Gateway = Outbound registration<br><br>-Steve<br><br><br><div class="gmail_quote">On 27 May 2011 07:30, Иван Чистяков <span dir="ltr"><<a href="mailto:zetruger@gmail.com">zetruger@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">Use directory for SIP clients (hardphone, softphone) instead gateways.<br>
Use gateway just for SIP providers (SIP trunks).<br>
<br>
Simple test (run from FS console):<br>
originate sofia/internal/outboundtest@x.x.x.x:p<br>
&bridge(sofia/internal/inboundtest@x.x.x.x:p)<br>
<br>
2011/5/27 Cyril Zlachevsky <<a href="mailto:cyril.zlachevsky@gmail.com">cyril.zlachevsky@gmail.com</a>>:<br>
<div><div></div><div class="h5">> Hello,<br>
> I have success with compiling and installing latest (1.0.7) FreeSWITCH on my server.<br>
> But all my today attempts to configure FreeSWITCH as softswitch failed and I really need help.<br>
><br>
> FreeSWITCH server (IP 88.198.XXX.XXX) have to receive SIP calls from the one IP address (X-Lite<br>
> softphone with dynamic IP from subnet <a href="http://92.112.0.0/16" target="_blank">92.112.0.0/16</a>) and forward this calls to the hardware SIP<br>
> phone (Asotel, IP 195.225.XXX.XXX). SIP mode on hardware softphone (Asotel) is peer-to-peer.<br>
><br>
> Direct calls from X-Lite to Asotel complete with success - I always hear incoming ring when dial<br>
> 7777. But through FreeSWITCH calls always fail.<br>
><br>
> I created four configuration files - all other configs I left unchanged:<br>
> 1) conf/sip-profiles/internal/X-Lite.xml:<br>
> <include><br>
> <gateway name="X-Lite"><br>
> <param name="username" value="inboundtest"/><br>
> <param name="password" value="test"/><br>
> <param name="register" value="true"/><br>
> </gateway><br>
> </include><br>
><br>
> 2) conf/sip-profiles/external/Asotel.xml:<br>
> <include><br>
> <gateway name="Asotel"><br>
> <param name="realm" value="195.225.XXX.XXX"/><br>
> <param name="username" value="outboundtest"/><br>
> <param name="password" value="test"/><br>
> <param name="register" value="true"/><br>
> </gateway><br>
> </include><br>
><br>
> 3) conf/dialplan/public/test.xml:<br>
> <include><br>
> <extension name="test"><br>
> <condition field="destination_number" expression="^7777$"><br>
> <action application="bridge" data="sofia/gateway/Asotel/$1"/><br>
> </condition><br>
> </extension><br>
> </include><br>
><br>
> 4) conf/directory/default/inboundtest.xml:<br>
> <include><br>
> <user id="inboundtest" cidr="<a href="http://92.112.0.0/16" target="_blank">92.112.0.0/16</a>"><br>
> <params><br>
> <param name="from-domain" value="88.198.XXX.XXX"/><br>
> <param name="password" value="test"/><br>
> </params><br>
> </user><br>
> </include><br>
><br>
><br>
> This debug from Asotel ip-phone:<br>
> ---begin---<br>
> Incoming CallLeg at callleg created 0x57d334 Incoming CallLeg at MsgReceived 0x57d334 *** $1 was<br>
> being Invited ***<br>
> >>> All call occupied. <<<<br>
> No slot availabe for this call...<br>
> FindIPCall...All Slot is Busy<br>
> RvSipCallLegReject(486), hCallLeg: 57d334<br>
> --> Message Sent (Message type: 1) (call-leg 57d334)<br>
> SIP/2.0 486 Busy Here<br>
> From: "inboundtest"<sip:outboundtest@195.225.XXX.XXX;transport=udp>;tag=vgpp5vSBgcX6p<br>
> To: <sip:$1@195.225.XXX.XXX>;tag=c3e19fb6-13c4-4ddecb5a-1f577-5f2a<br>
> Call-ID: 232b19fe-0285-122f-b5b5-1b5bdf4f9807<br>
> CSeq: 12887277 INVITE<br>
> Via: SIP/2.0/UDP 88.198.XXX.XXX:5080;rport=5080;branch=z9hG4bKv0XUcNtvK3c2K<br>
> Supported: replaces<br>
> User-Agent: FXS_GW (1asipfxs.109)<br>
> Content-Length: 0<br>
> ---end---<br>
><br>
> In freeswitch log I can see this:<br>
> ---begin---<br>
> [NOTICE] switch_channel.c:816 New Channel sofia/internal inboundtest@88.198.XXX.XXX<br>
> [bf8e8081-eaf1-453e-a643-ee03df36ba0f]<br>
> [INFO] mod_dialplan_xml.c:336 Processing inboundtest <inboundtest>->7777 in context public<br>
> [NOTICE] switch_channel.c:816 New Channel sofia/external/$1 [4beaba1f-c9c6-4ed7-94c5-efec453e895a]<br>
> [NOTICE] sofia.c:5416 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [USER_BUSY]<br>
> [INFO] mod_dptools.c:2685 Originate Failed. Cause: USER_BUSY<br>
> [NOTICE] mod_dptools.c:2799 Hangup sofia/internal/inboundtest@88.198.XXX.XXX [CS_EXECUTE] [USER_BUSY]<br>
> [NOTICE] switch_core_session.c:1304 Session 1 (sofia/internal/inboundtest@88.198.XXX.XXX) Ended<br>
> [NOTICE] switch_core_session.c:1306 Close Channel sofia/internal/inboundtest@88.198.XXX.XXX [CS_DESTROY]<br>
> [NOTICE] switch_core_session.c:1304 Session 2 (sofia/external/$1) Ended<br>
> [NOTICE] switch_core_session.c:1306 Close Channel sofia/external/$1 [CS_DESTROY]<br>
> ---end---<br>
><br>
> In my last tests I called to the voip-provider test number and got UNALLOCATED_NUMBER disconnect cause:<br>
> ---begin---<br>
> [NOTICE] switch_channel.c:816 New Channel sofia/internal/inboundtest@88.198.XXX.XXX<br>
> [ee9e4e33-0676-4c9c-9952-ff97c4d8db18]<br>
> [INFO] mod_dialplan_xml.c:336 Processing inboundtest <inboundtest>->555 in context public<br>
> [NOTICE] switch_channel.c:816 New Channel sofia/external/$1 [b8278ac0-8e90-4d48-bdbf-8d0f608ec35a]<br>
> [NOTICE] sofia.c:5416 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [UNALLOCATED_NUMBER]<br>
> [INFO] mod_dptools.c:2685 Originate Failed. Cause: UNALLOCATED_NUMBER<br>
> [NOTICE] mod_dptools.c:2799 Hangup sofia/internal/inboundtest@88.198.XXX.XXX [CS_EXECUTE]<br>
> [UNALLOCATED_NUMBER]<br>
> [NOTICE] switch_core_session.c:1304 Session 1 (sofia/internal/inboundtest@88.198.XXX.XXX) Ended<br>
> [NOTICE] switch_core_session.c:1306 Close Channel sofia/internal/inboundtest@88.198.XXX.XXX [CS_DESTROY]<br>
> [NOTICE] switch_core_session.c:1304 Session 2 (sofia/external/$1) Ended<br>
> [NOTICE] switch_core_session.c:1306 Close Channel sofia/external/$1 [CS_DESTROY]<br>
> ---end---<br>
><br>
><br>
> What am I missing here?<br>
><br>
> Thanks for your help.<br>
><br>
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