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FreeSWITCH as SIP softswitch

Cyril Zlachevsky
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  *** $1 was
being Invited ***
 >>> All call occupied. <<<
No slot availabe for this call...
FindIPCall...All Slot is Busy
RvSipCallLegReject(486), hCallLeg: 57d334
--> Message Sent (Message type: 1) (call-leg 57d334)
SIP/2.0 486 Busy Here
From: "inboundtest"<sip:[hidden email];transport=udp>;tag=vgpp5vSBgcX6p
To: <sip:$[hidden email]>;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 [hidden email]
[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/[hidden email] [CS_EXECUTE] [USER_BUSY]
[NOTICE] switch_core_session.c:1304 Session 1 (sofia/internal/[hidden email]) Ended
[NOTICE] switch_core_session.c:1306 Close Channel sofia/internal/[hidden email] [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/[hidden email]
[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/[hidden email] [CS_EXECUTE]
[UNALLOCATED_NUMBER]
[NOTICE] switch_core_session.c:1304 Session 1 (sofia/internal/[hidden email]) Ended
[NOTICE] switch_core_session.c:1306 Close Channel sofia/internal/[hidden email] [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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Re: FreeSWITCH as SIP softswitch

Иван Чистяков
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/[hidden email]:p
&bridge(sofia/internal/[hidden email]:p)

2011/5/27 Cyril Zlachevsky <[hidden email]>:

> 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  *** $1 was
> being Invited ***
>  >>> All call occupied. <<<
> No slot availabe for this call...
> FindIPCall...All Slot is Busy
> RvSipCallLegReject(486), hCallLeg: 57d334
> --> Message Sent (Message type: 1) (call-leg 57d334)
> SIP/2.0 486 Busy Here
> From: "inboundtest"<sip:[hidden email];transport=udp>;tag=vgpp5vSBgcX6p
> To: <sip:$[hidden email]>;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 [hidden email]
> [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/[hidden email] [CS_EXECUTE] [USER_BUSY]
> [NOTICE] switch_core_session.c:1304 Session 1 (sofia/internal/[hidden email]) Ended
> [NOTICE] switch_core_session.c:1306 Close Channel sofia/internal/[hidden email] [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/[hidden email]
> [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/[hidden email] [CS_EXECUTE]
> [UNALLOCATED_NUMBER]
> [NOTICE] switch_core_session.c:1304 Session 1 (sofia/internal/[hidden email]) Ended
> [NOTICE] switch_core_session.c:1306 Close Channel sofia/internal/[hidden email] [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.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> [hidden email]
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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Re: FreeSWITCH as SIP softswitch

Steven Ayre
User directory = Inbound registration
Gateway = Outbound registration

-Steve


On 27 May 2011 07:30, Иван Чистяков <[hidden email]> 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/[hidden email]:p
&bridge(sofia/internal/[hidden email]:p)

2011/5/27 Cyril Zlachevsky <[hidden email]>:
> 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  *** $1 was
> being Invited ***
>  >>> All call occupied. <<<
> No slot availabe for this call...
> FindIPCall...All Slot is Busy
> RvSipCallLegReject(486), hCallLeg: 57d334
> --> Message Sent (Message type: 1) (call-leg 57d334)
> SIP/2.0 486 Busy Here
> From: "inboundtest"<sip:[hidden email];transport=udp>;tag=vgpp5vSBgcX6p
> To: <sip:$[hidden email]>;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 [hidden email]
> [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/[hidden email] [CS_EXECUTE] [USER_BUSY]
> [NOTICE] switch_core_session.c:1304 Session 1 (sofia/internal/[hidden email]) Ended
> [NOTICE] switch_core_session.c:1306 Close Channel sofia/internal/[hidden email] [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/[hidden email]
> [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/[hidden email] [CS_EXECUTE]
> [UNALLOCATED_NUMBER]
> [NOTICE] switch_core_session.c:1304 Session 1 (sofia/internal/[hidden email]) Ended
> [NOTICE] switch_core_session.c:1306 Close Channel sofia/internal/[hidden email] [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.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> [hidden email]
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

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Re: FreeSWITCH as SIP softswitch

Cyril Zlachevsky
In reply to this post by Cyril Zlachevsky
Can't understand you Answer, Ivan!
You mean I have to replace
<action application="bridge" data="sofia/gateway/Asotel/$1"/>
to
<action application="bridge" data="sofia/directory/Asotel/$1"/>
in conf/dialplan/public/test.xml file?


2011/5/27  <[hidden email]>:

> From: "Иван Чистяков" <[hidden email]>
> 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/[hidden email]:p
> &bridge(sofia/internal/[hidden email]:p)
>
> 2011/5/27 Cyril Zlachevsky <[hidden email]>:
>> 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.

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Re: FreeSWITCH as SIP softswitch

Иван Чистяков
<action application="bridge"
data="sofia/<SIP_PROFILE_NAME>/[hidden email]:p"/>
You must reconfigure FreeSWITCH. Create user directory xml files.

27 мая 2011 г. 14:46 пользователь Cyril Zlachevsky
<[hidden email]> написал:

> Can't understand you Answer, Ivan!
> You mean I have to replace
> <action application="bridge" data="sofia/gateway/Asotel/$1"/>
> to
> <action application="bridge" data="sofia/directory/Asotel/$1"/>
> in conf/dialplan/public/test.xml file?
>
>
> 2011/5/27  <[hidden email]>:
>> From: "Иван Чистяков" <[hidden email]>
>> 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/[hidden email]:p
>> &bridge(sofia/internal/[hidden email]:p)
>>
>> 2011/5/27 Cyril Zlachevsky <[hidden email]>:
>>> 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-users mailing list
> [hidden email]
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

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Re: FreeSWITCH as SIP softswitch

mazilo
In reply to this post by Cyril Zlachevsky
Cyril Zlachevsky wrote
Can't understand you Answer, Ivan!
There are several pre-configured extensions under the conf/directory/default directory on your FS machine. Its default password is set on conf/vars.xml file. You can configure your X-Lite softphone and/or any ATA device to register to your FS using one of the extensions.
FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
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