Hello! Is it possible to use freeswitch as SIP route server? I have commercial SIP server which does not has LCR with it. Is uses external routing instead. It sends INVITE packet to external SIP server and waits for 3XX response. I tried to get FS working as RS. My DP is very simple, I just want to see I get a call. <extension name="del-group"> <condition field="destination_number" expression="^.*$"> <action application="log" data="ALERT Here I am"/> </condition> </extension> Console output: 2014-06-07 00:41:18.475135 [NOTICE] switch_channel.c:1053 New Channel sofia/lcr_standard/230069@127.0.0.1:5062 [bc7fe4e8-7496-48b6-9381-a5debd7c2423] 2014-06-07 00:41:18.475135 [NOTICE] sofia.c:7112 Hangup sofia/lcr_standard/230069@127.0.0.1:5062 [CS_HIBERNATE] [MANDATORY_IE_MISSING] 2014-06-07 00:41:18.495131 [NOTICE] switch_core_session.c:1632 Session 1 (sofia/lcr_standard/230069@127.0.0.1:5062) Ended 2014-06-07 00:41:18.495131 [NOTICE] switch_core_session.c:1636 Close Channel sofia/lcr_standard/230069@127.0.0.1:5062 [CS_DESTROY] So, if the FS can not be used as SIP route server may be You can suggest another software? -- Regards, Boris
Boris, Yes, it is possible. I demonstrated using FreeSWITCH as a high scale (300cps) LCR server using mod_lcr and 302s four years ago at ClueCon: http://blog.krisk.org/2010/05/cluecon-preview.html http://blog.krisk.org/2010/05/ive-said-it-before-but-ill-say-it-again.html To debug this issue we would need to see a full SIP trace with FreeSWITCH console log output. On Fri, Jun 6, 2014 at 2:50 PM, Boris Kovalenko <boris@tagnet.ru> wrote:
Hello!
Is it possible to use freeswitch as SIP route server? I have commercial SIP server which does not has LCR with it. Is uses external routing instead. It sends INVITE packet to external SIP server and waits for 3XX response. I tried to get FS working as RS. My DP is very simple, I just want to see I get a call.
<extension name="del-group"> <condition field="destination_number" expression="^.*$"> <action application="log" data="ALERT Here I am"/> </condition> </extension>
Console output: 2014-06-07 00:41:18.475135 [NOTICE] switch_channel.c:1053 New Channel sofia/lcr_standard/230069@127.0.0.1:5062 [bc7fe4e8-7496-48b6-9381-a5debd7c2423] 2014-06-07 00:41:18.475135 [NOTICE] sofia.c:7112 Hangup sofia/lcr_standard/230069@127.0.0.1:5062 [CS_HIBERNATE] [MANDATORY_IE_MISSING] 2014-06-07 00:41:18.495131 [NOTICE] switch_core_session.c:1632 Session 1 (sofia/lcr_standard/230069@127.0.0.1:5062) Ended 2014-06-07 00:41:18.495131 [NOTICE] switch_core_session.c:1636 Close Channel sofia/lcr_standard/230069@127.0.0.1:5062 [CS_DESTROY]
So, if the FS can not be used as SIP route server may be You can suggest another software?
-- Regards, Boris
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Kristian Kielhofner
Kristian, can't connect to your blog :( Here is my siptrace freeswitch@pbx> recv 532 bytes from udp/[127.0.0.1]:5062 at 01:43:22.418118: ------------------------------------------------------------------------ INVITE sip:89826409893@127.0.0.1:50601;user=phone SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5062;rport;branch=z9hG4bK-2527683537-3809587949-1143785123-2486801405 From: "mtt" <sip:230069@127.0.0.1:5062;gwname=50999_MOA;user=phone>;tag=3399640017-3809587949-1143785123-2486801405 To: <sip:89826409893@127.0.0.1:50601;user=phone> Call-ID: d163a324edb211e3a3c62c44fd933994@127.0.0.1 CSeq: 1 INVITE Contact: "mtt" <sip:230069@127.0.0.1:5062;gwname=50999_MOA;user=phone> Max-Forwards: 70 User-Agent: TS-v4.5.1-16aW Content-Length: 0 ------------------------------------------------------------------------ send 426 bytes to udp/[127.0.0.1]:5062 at 01:43:22.418689: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 127.0.0.1:5062;rport=5062;branch=z9hG4bK-2527683537-3809587949-1143785123-2486801405 From: "mtt" <sip:230069@127.0.0.1:5062;gwname=50999_MOA;user=phone>;tag=3399640017-3809587949-1143785123-2486801405 To: <sip:89826409893@127.0.0.1:50601;user=phone> Call-ID: d163a324edb211e3a3c62c44fd933994@127.0.0.1 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.4.6~32bit Content-Length: 0 ------------------------------------------------------------------------ 2014-06-07 01:43:22.415127 [NOTICE] switch_channel.c:1053 New Channel sofia/lcr_standard/230069@127.0.0.1:5062 [a5044611-4da8-4e49-8a74-7b0c362019f6] send 983 bytes to udp/[127.0.0.1]:5062 at 01:43:22.421298: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5062;rport=5062;branch=z9hG4bK-2527683537-3809587949-1143785123-2486801405 From: "mtt" <sip:230069@127.0.0.1:5062;gwname=50999_MOA;user=phone>;tag=3399640017-3809587949-1143785123-2486801405 To: <sip:89826409893@127.0.0.1:50601;user=phone>;tag=cX8a9tcgQrD3r Call-ID: d163a324edb211e3a3c62c44fd933994@127.0.0.1 CSeq: 1 INVITE Contact: <sip:mod_sofia@127.0.0.1:50601> User-Agent: FreeSWITCH-mod_sofia/1.4.6~32bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 244 v=0 o=FreeSWITCH 1402056178 1402056179 IN IP4 X.X.21.251 s=FreeSWITCH c=IN IP4 127.0.0.1 t=0 0 m=audio 27624 RTP/AVP 0 8 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ recv 472 bytes from udp/[127.0.0.1]:5062 at 01:43:22.422169: ------------------------------------------------------------------------ ACK sip:89826409893@127.0.0.1:50601;user=phone SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5062;rport;branch=z9hG4bK-3225576657-3809587949-1143785123-2486801405 From: "mtt" <sip:230069@127.0.0.1:5062;gwname=50999_MOA;user=phone>;tag=3399640017-3809587949-1143785123-2486801405 To: <sip:89826409893@127.0.0.1:50601;user=phone>;tag=cX8a9tcgQrD3r Call-ID: d163a324edb211e3a3c62c44fd933994@127.0.0.1 CSeq: 1 ACK Max-Forwards: 70 User-Agent: TS-v4.5.1-16aW Content-Length: 0 ------------------------------------------------------------------------ send 693 bytes to udp/[127.0.0.1]:5062 at 01:43:22.422541: ------------------------------------------------------------------------ BYE sip:230069@127.0.0.1:5062;gwname=50999_MOA;user=phone SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:50601;rport;branch=z9hG4bKcm6XFe1pva6Xg Max-Forwards: 70 From: <sip:89826409893@127.0.0.1:50601;user=phone>;tag=cX8a9tcgQrD3r To: "mtt" <sip:230069@127.0.0.1:5062;gwname=50999_MOA;user=phone>;tag=3399640017-3809587949-1143785123-2486801405 Call-ID: d163a324edb211e3a3c62c44fd933994@127.0.0.1 CSeq: 60705837 BYE Contact: <sip:mod_sofia@127.0.0.1:50601> User-Agent: FreeSWITCH-mod_sofia/1.4.6~32bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Reason: SIP;cause=488;text="No answer to offer" Content-Length: 0 ------------------------------------------------------------------------ 2014-06-07 01:43:22.415127 [NOTICE] sofia.c:7112 Hangup sofia/lcr_standard/230069@127.0.0.1:5062 [CS_HIBERNATE] [MANDATORY_IE_MISSING] recv 492 bytes from udp/[127.0.0.1]:5062 at 01:43:22.423261: ------------------------------------------------------------------------ SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 127.0.0.1:50601;rport=50601;branch=z9hG4bKcm6XFe1pva6Xg;received=127.0.0.1 From: <sip:89826409893@127.0.0.1:50601;user=phone>;tag=cX8a9tcgQrD3r To: "mtt" <sip:230069@127.0.0.1:5062;gwname=50999_MOA;user=phone>;tag=3399640017-3809587949-1143785123-2486801405 Call-ID: d163a324edb211e3a3c62c44fd933994@127.0.0.1 CSeq: 60705837 BYE Server: TS-v4.5.1-16aW Reason: Q.850;cause=0;text="Normal call clearing" Content-Length: 0 ------------------------------------------------------------------------ 2014-06-07 01:43:22.415127 [NOTICE] switch_core_session.c:1632 Session 1 (sofia/lcr_standard/230069@127.0.0.1:5062) Ended 2014-06-07 01:43:22.415127 [NOTICE] switch_core_session.c:1636 Close Channel sofia/lcr_standard/230069@127.0.0.1:5062 [CS_DESTROY] recv 471 bytes from udp/[127.0.0.1]:5062 at 01:43:22.425628: ------------------------------------------------------------------------ BYE sip:89826409893@127.0.0.1:50601;user=phone SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5062;rport;branch=z9hG4bK-474375377-3809587949-1143785123-2486801405 From: "mtt" <sip:230069@127.0.0.1:5062;gwname=50999_MOA;user=phone>;tag=3399640017-3809587949-1143785123-2486801405 To: <sip:89826409893@127.0.0.1:50601;user=phone>;tag=cX8a9tcgQrD3r Call-ID: d163a324edb211e3a3c62c44fd933994@127.0.0.1 CSeq: 3 BYE Max-Forwards: 70 User-Agent: TS-v4.5.1-16aW Content-Length: 0 ------------------------------------------------------------------------ send 577 bytes to udp/[127.0.0.1]:5062 at 01:43:22.425851: ------------------------------------------------------------------------ SIP/2.0 481 Call Does Not Exist Via: SIP/2.0/UDP 127.0.0.1:5062;rport=5062;branch=z9hG4bK-474375377-3809587949-1143785123-2486801405 From: "mtt" <sip:230069@127.0.0.1:5062;gwname=50999_MOA;user=phone>;tag=3399640017-3809587949-1143785123-2486801405 To: <sip:89826409893@127.0.0.1:50601;user=phone>;tag=cX8a9tcgQrD3r Call-ID: d163a324edb211e3a3c62c44fd933994@127.0.0.1 CSeq: 3 BYE User-Agent: FreeSWITCH-mod_sofia/1.4.6~32bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Content-Length: 0 ------------------------------------------------------------------------
Boris,
Yes, it is possible. I demonstrated using FreeSWITCH as a high scale (300cps) LCR server using mod_lcr and 302s four years ago at ClueCon:
http://blog.krisk.org/2010/05/cluecon-preview.html http://blog.krisk.org/2010/05/ive-said-it-before-but-ill-say-it-again.html
To debug this issue we would need to see a full SIP trace with FreeSWITCH console log output.
On Fri, Jun 6, 2014 at 2:50 PM, Boris Kovalenko <boris@tagnet.ru> wrote:
Hello!
Is it possible to use freeswitch as SIP route server? I have commercial SIP server which does not has LCR with it. Is uses external routing instead. It sends INVITE packet to external SIP server and waits for 3XX response. I tried to get FS working as RS. My DP is very simple, I just want to see I get a call.
<extension name="del-group"> <condition field="destination_number" expression="^.*$"> <action application="log" data="ALERT Here I am"/> </condition> </extension>
Console output: 2014-06-07 00:41:18.475135 [NOTICE] switch_channel.c:1053 New Channel sofia/lcr_standard/230069@127.0.0.1:5062 [bc7fe4e8-7496-48b6-9381-a5debd7c2423] 2014-06-07 00:41:18.475135 [NOTICE] sofia.c:7112 Hangup sofia/lcr_standard/230069@127.0.0.1:5062 [CS_HIBERNATE] [MANDATORY_IE_MISSING] 2014-06-07 00:41:18.495131 [NOTICE] switch_core_session.c:1632 Session 1 (sofia/lcr_standard/230069@127.0.0.1:5062) Ended 2014-06-07 00:41:18.495131 [NOTICE] switch_core_session.c:1636 Close Channel sofia/lcr_standard/230069@127.0.0.1:5062 [CS_DESTROY]
So, if the FS can not be used as SIP route server may be You can suggest another software?
-- Regards, Boris
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Regards, Boris
participants (2)
-
Boris Kovalenko
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Kristian Kielhofner