FreeSWITCH as SIP softswitch
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@195.225.XXX.XXX;transport=udp>;tag=vgpp5vSBgcX6p To: <sip:$1@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@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@88.198.XXX.XXX [CS_EXECUTE] [USER_BUSY] [NOTICE] switch_core_session.c:1304 Session 1 (sofia/internal/inboundtest@88.198.XXX.XXX) Ended [NOTICE] switch_core_session.c:1306 Close Channel sofia/internal/inboundtest@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@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@88.198.XXX.XXX [CS_EXECUTE] [UNALLOCATED_NUMBER] [NOTICE] switch_core_session.c:1304 Session 1 (sofia/internal/inboundtest@88.198.XXX.XXX) Ended [NOTICE] switch_core_session.c:1306 Close Channel sofia/internal/inboundtest@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/$1 [CS_DESTROY] ---end--- What am I missing here? Thanks for your help.
Use directory for SIP clients (hardphone, softphone) instead gateways. Use gateway just for SIP providers (SIP trunks). Simple test (run from FS console): originate sofia/internal/outboundtest@x.x.x.x:p &bridge(sofia/internal/inboundtest@x.x.x.x:p) 2011/5/27 Cyril Zlachevsky <cyril.zlachevsky@gmail.com>:
Hello, I have success with compiling and installing latest (1.0.7) FreeSWITCH on my server. But all my today attempts to configure FreeSWITCH as softswitch failed and I really need help.
FreeSWITCH server (IP 88.198.XXX.XXX) have to receive SIP calls from the one IP address (X-Lite softphone with dynamic IP from subnet 92.112.0.0/16) and forward this calls to the hardware SIP phone (Asotel, IP 195.225.XXX.XXX). SIP mode on hardware softphone (Asotel) is peer-to-peer.
Direct calls from X-Lite to Asotel complete with success - I always hear incoming ring when dial 7777. But through FreeSWITCH calls always fail.
I created four configuration files - all other configs I left unchanged: 1) conf/sip-profiles/internal/X-Lite.xml: <include> <gateway name="X-Lite"> <param name="username" value="inboundtest"/> <param name="password" value="test"/> <param name="register" value="true"/> </gateway> </include>
2) conf/sip-profiles/external/Asotel.xml: <include> <gateway name="Asotel"> <param name="realm" value="195.225.XXX.XXX"/> <param name="username" value="outboundtest"/> <param name="password" value="test"/> <param name="register" value="true"/> </gateway> </include>
3) conf/dialplan/public/test.xml: <include> <extension name="test"> <condition field="destination_number" expression="^7777$"> <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@195.225.XXX.XXX;transport=udp>;tag=vgpp5vSBgcX6p To: <sip:$1@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@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@88.198.XXX.XXX [CS_EXECUTE] [USER_BUSY] [NOTICE] switch_core_session.c:1304 Session 1 (sofia/internal/inboundtest@88.198.XXX.XXX) Ended [NOTICE] switch_core_session.c:1306 Close Channel sofia/internal/inboundtest@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@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@88.198.XXX.XXX [CS_EXECUTE] [UNALLOCATED_NUMBER] [NOTICE] switch_core_session.c:1304 Session 1 (sofia/internal/inboundtest@88.198.XXX.XXX) Ended [NOTICE] switch_core_session.c:1306 Close Channel sofia/internal/inboundtest@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/$1 [CS_DESTROY] ---end---
What am I missing here?
Thanks for your help.
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User directory = Inbound registration Gateway = Outbound registration -Steve On 27 May 2011 07:30, Иван Чистяков <zetruger@gmail.com> wrote:
Use directory for SIP clients (hardphone, softphone) instead gateways. Use gateway just for SIP providers (SIP trunks).
Simple test (run from FS console): originate sofia/internal/outboundtest@x.x.x.x:p &bridge(sofia/internal/inboundtest@x.x.x.x:p)
2011/5/27 Cyril Zlachevsky <cyril.zlachevsky@gmail.com>:
Hello, I have success with compiling and installing latest (1.0.7) FreeSWITCH on my server. But all my today attempts to configure FreeSWITCH as softswitch failed and I really need help.
FreeSWITCH server (IP 88.198.XXX.XXX) have to receive SIP calls from the one IP address (X-Lite softphone with dynamic IP from subnet 92.112.0.0/16) and forward this calls to the hardware SIP phone (Asotel, IP 195.225.XXX.XXX). SIP mode on hardware softphone (Asotel) is peer-to-peer.
Direct calls from X-Lite to Asotel complete with success - I always hear incoming ring when dial 7777. But through FreeSWITCH calls always fail.
I created four configuration files - all other configs I left unchanged: 1) conf/sip-profiles/internal/X-Lite.xml: <include> <gateway name="X-Lite"> <param name="username" value="inboundtest"/> <param name="password" value="test"/> <param name="register" value="true"/> </gateway> </include>
2) conf/sip-profiles/external/Asotel.xml: <include> <gateway name="Asotel"> <param name="realm" value="195.225.XXX.XXX"/> <param name="username" value="outboundtest"/> <param name="password" value="test"/> <param name="register" value="true"/> </gateway> </include>
3) conf/dialplan/public/test.xml: <include> <extension name="test"> <condition field="destination_number" expression="^7777$"> <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@195.225.XXX.XXX ;transport=udp>;tag=vgpp5vSBgcX6p To: <sip:$1@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@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@88.198.XXX.XXX [CS_EXECUTE] [USER_BUSY] [NOTICE] switch_core_session.c:1304 Session 1 (sofia/internal/inboundtest@88.198.XXX.XXX) Ended [NOTICE] switch_core_session.c:1306 Close Channel sofia/internal/inboundtest@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@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@88.198.XXX.XXX [CS_EXECUTE] [UNALLOCATED_NUMBER] [NOTICE] switch_core_session.c:1304 Session 1 (sofia/internal/inboundtest@88.198.XXX.XXX) Ended [NOTICE] switch_core_session.c:1306 Close Channel sofia/internal/inboundtest@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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participants (3)
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Cyril Zlachevsky
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Steven Ayre
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Иван Чистяков