[Freeswitch-users] FreeSWITCH as SIP softswitch

Steven Ayre steveayre at gmail.com
Fri May 27 12:39:36 MSD 2011 

User directory = Inbound registration
Gateway = Outbound registration

-Steve


On 27 May 2011 07:30, Иван Чистяков <zetruger at gmail.com> wrote:

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