[Freeswitch-users] FreeSwitch as SIP route server

Boris Kovalenko boris at tagnet.ru
Fri Jun 6 23:49:11 MSD 2014


Kristian, can't connect to your blog :(
Here is my siptrace

freeswitch at pbx> recv 532 bytes from udp/[127.0.0.1]:5062 at 01:43:22.418118:
------------------------------------------------------------------------
    INVITE sip:89826409893 at 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 at 127.0.0.1:5062;gwname=50999_MOA;user=phone>;tag=3399640017-3809587949-1143785123-2486801405
    To: <sip:89826409893 at 127.0.0.1:50601;user=phone>
    Call-ID: d163a324edb211e3a3c62c44fd933994 at 127.0.0.1
    CSeq: 1 INVITE
    Contact: "mtt" <sip:230069 at 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 at 127.0.0.1:5062;gwname=50999_MOA;user=phone>;tag=3399640017-3809587949-1143785123-2486801405
    To: <sip:89826409893 at 127.0.0.1:50601;user=phone>
    Call-ID: d163a324edb211e3a3c62c44fd933994 at 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 at 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 at 127.0.0.1:5062;gwname=50999_MOA;user=phone>;tag=3399640017-3809587949-1143785123-2486801405
    To: <sip:89826409893 at 127.0.0.1:50601;user=phone>;tag=cX8a9tcgQrD3r
    Call-ID: d163a324edb211e3a3c62c44fd933994 at 127.0.0.1
    CSeq: 1 INVITE
    Contact: <sip:mod_sofia at 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 at 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 at 127.0.0.1:5062;gwname=50999_MOA;user=phone>;tag=3399640017-3809587949-1143785123-2486801405
    To: <sip:89826409893 at 127.0.0.1:50601;user=phone>;tag=cX8a9tcgQrD3r
    Call-ID: d163a324edb211e3a3c62c44fd933994 at 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 at 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 at 127.0.0.1:50601;user=phone>;tag=cX8a9tcgQrD3r
    To: "mtt" 
<sip:230069 at 127.0.0.1:5062;gwname=50999_MOA;user=phone>;tag=3399640017-3809587949-1143785123-2486801405
    Call-ID: d163a324edb211e3a3c62c44fd933994 at 127.0.0.1
    CSeq: 60705837 BYE
    Contact: <sip:mod_sofia at 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 at 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 at 127.0.0.1:50601;user=phone>;tag=cX8a9tcgQrD3r
    To: "mtt" 
<sip:230069 at 127.0.0.1:5062;gwname=50999_MOA;user=phone>;tag=3399640017-3809587949-1143785123-2486801405
    Call-ID: d163a324edb211e3a3c62c44fd933994 at 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 at 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 at 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 at 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 at 127.0.0.1:5062;gwname=50999_MOA;user=phone>;tag=3399640017-3809587949-1143785123-2486801405
    To: <sip:89826409893 at 127.0.0.1:50601;user=phone>;tag=cX8a9tcgQrD3r
    Call-ID: d163a324edb211e3a3c62c44fd933994 at 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 at 127.0.0.1:5062;gwname=50999_MOA;user=phone>;tag=3399640017-3809587949-1143785123-2486801405
    To: <sip:89826409893 at 127.0.0.1:50601;user=phone>;tag=cX8a9tcgQrD3r
    Call-ID: d163a324edb211e3a3c62c44fd933994 at 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 at 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 at 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 at 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 at 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 at 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 at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
  




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