[Freeswitch-users] FreeSWITCH using same Call-ID for forked calls
Brian West
brian at freeswitch.org
Thu Mar 19 22:11:46 MSK 2015
Looks like an invite that didnt' get a response. You sure dude?
On Thu, Mar 19, 2015 at 11:26 AM, Örn Arnarson <orn at arnarson.net> wrote:
> Hi,
>
> Thanks for your response.
>
> I installed FS from git on another server, and it still shows exactly the
> same behavior. See the two new INVITES for the forked call outbound from FS:
> INVITE sip:7712552 at 192.168.10.3 SIP/2.0
> Via: SIP/2.0/UDP 192.168.10.101:5080;rport;branch=z9hG4bK67cUpBr27p25r
> Max-Forwards: 69
> From: "..rn" <sip:5460000 at 192.168.10.101>;tag=eU3yr5rNjcvDa
> To: <sip:7712552 at 192.168.10.3>
> Call-ID: 174ff657-48f7-1233-d586-080027f911ca
> CSeq: 73055034 INVITE
> Contact: <sip:mod_sofia at 192.168.10.101:5080>
> User-Agent:
> FreeSWITCH-mod_sofia/1.5.15b+git~20150318T193012Z~21d1e6fc4b~32bit
> 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: 251
> Diversion: <sip:4151500 at 172.25.200.101>
> X-FS-Support: update_display,send_info
>
> v=0
> o=FreeSWITCH 1426761247 1426761248 IN IP4 192.168.10.101
> s=FreeSWITCH
> c=IN IP4 192.168.10.101
> t=0 0
> m=audio 20950 RTP/AVP 8 0 101 13
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
>
> INVITE sip:6595454 at 192.168.10.3 SIP/2.0
> Via: SIP/2.0/UDP 192.168.10.101:5080;rport;branch=z9hG4bK5yK2mg7yaecKD
> Max-Forwards: 69
> From: "..rn" <sip:5460000 at 192.168.10.101>;tag=Dja6pa8HN35te
> To: <sip:6595454 at 192.168.10.3>
> Call-ID: 174f7870-48f7-1233-d586-080027f911ca
> CSeq: 73055034 INVITE
> Contact: <sip:mod_sofia at 192.168.10.101:5080>
> User-Agent:
> FreeSWITCH-mod_sofia/1.5.15b+git~20150318T193012Z~21d1e6fc4b~32bit
> 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: 251
> Diversion: <sip:4151500 at 172.25.200.101>
> X-FS-Support: update_display,send_info
>
> v=0
> o=FreeSWITCH 1426750991 1426750992 IN IP4 192.168.10.101
> s=FreeSWITCH
> c=IN IP4 192.168.10.101
> t=0 0
> m=audio 31206 RTP/AVP 8 0 101 13
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
>
> Regards,
> Örn
>
> On Wed, Mar 18, 2015 at 6:57 PM, Brian West <brian at freeswitch.org> wrote:
>
>> You're using 1.2, I see 1.2.12 and 1.2.7 in your user agents above, I
>> would highly recommend you re-test with Master or at the very least 1.4.17
>> or 1.4.18 which should be out later today.
>>
>> 1.2 is not receiving patches, updates or support moving forward, our
>> release branch is 1.4.x
>>
>>
>>
>>
>> On Wed, Mar 18, 2015 at 1:26 PM, Örn Arnarson <orn at arnarson.net> wrote:
>>
>>> Hello,
>>>
>>> Not sure whether this belong in the users list or the dev list, but when
>>> in doubt; start with users :-)
>>>
>>> I am using FreeSWITCH as an SBC, talking to Kamailio on one and and
>>> Asterisk on the other, and am seeing some strange behavior when calls are
>>> being forked on the Asterisk.
>>>
>>> Call setup is like this:
>>> 1. FreeSWITCH receives INVITE from Kamailio
>>> 2. FreeSWITCH sends INVITE to Asterisk with new Call-ID
>>> 3. Asterisk forks call, sends out multiple INVITEs back to FreeSWITCH
>>> (each with a unique call-id)
>>> 4. FreeSWITCH sends multiple INVITEs to Kamailio, each with the new
>>> Call-ID from step 2.
>>>
>>> This is causing problems with one of the MGWs behind Kamailio, which is
>>> seeing multiple INVITEs to different destinations with the same Call-ID.
>>>
>>> So, firstly, why is FreeSWITCH reusing call-ids?
>>>
>>> Secondly, how is it matching up the calls? I can't find anything common
>>> in the INVITEs, other than the source number and obviously that the IP sent
>>> to and received from is the same.
>>>
>>> I'm not sure if this is intended behavior or not, but is there a way to
>>> have FreeSWITCH not do that?
>>>
>>> Regards,
>>> Örn
>>>
>>> P.S. Here is the sequence of INVITEs. I also have the console log (for a
>>> different call) if needed.
>>>
>>> *INVITE sent to FreeSWITCH by Kamailio:*
>>> INVITE sip:5344446 at 172.25.200.111:5080 SIP/2.0
>>> Record-Route: <sip:172.25.200.101;lr=on>
>>> Via: SIP/2.0/UDP 172.25.200.101;branch=z9hG4bK165.08b3b5c4.0
>>> Via: SIP/2.0/UDP 172.25.200.121:5080
>>> ;rport=5080;branch=z9hG4bK9aFD5m2KKerHN
>>> Max-Forwards: 16
>>> From: "4151502" <sip:4151502 at 172.25.200.121>;tag=33vB4BmmDtU0B
>>> To: <sip:5344446 at 172.25.200.101>
>>> Call-ID: 84b63791-4839-1233-639f-00215e2db0e0
>>> CSeq: 73014324 INVITE
>>> Contact: <sip:mod_sofia at 172.25.200.121:5080>
>>> User-Agent: FreeSWITCH-mod_sofia/1.2.7
>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>> REGISTER, REFER, NOTIFY
>>> Supported: timer, precondition, path, replaces
>>> Allow-Events: talk, hold, conference, refer
>>> Content-Type: application/sdp
>>> Content-Disposition: session
>>> Content-Length: 229
>>> X-FS-Support: update_display,send_info
>>> Remote-Party-ID: "4151502" <sip:4151502 at 172.25.200.121
>>> >;party=calling;screen=yes;privacy=off
>>>
>>> v=0
>>> o=FreeSWITCH 1426681750 1426681751 IN IP4 172.25.200.121
>>> s=FreeSWITCH
>>> c=IN IP4 172.25.200.121
>>> t=0 0
>>> m=audio 19026 RTP/AVP 8 101
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=silenceSupp:off - - - -
>>> a=ptime:20
>>>
>>>
>>> *INVITE sent to Asterisk by FreeSWITCH:*
>>> INVITE sip:5344446 at 172.26.0.62:5060 SIP/2.0
>>> Via: SIP/2.0/UDP 10.11.12.13;rport;branch=z9hG4bK0N733e7FHv4QF
>>> Max-Forwards: 15
>>> From: "4151502" <sip:4151502 at 10.11.12.13>;tag=2BaZj0t076Q9B
>>> To: <sip:5344446 at 172.26.0.62:5060>
>>> Call-ID: a7c77ea5-4839-1233-73b9-00215e2c8c90
>>> CSeq: 73014353 INVITE
>>> Contact: <sip:mod_sofia at 10.11.12.13:5060>
>>> User-Agent: FreeSWITCH-mod_sofia/1.2.12
>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>> Supported: timer, precondition, path, replaces
>>> Allow-Events: talk, hold, conference, presence, dialog, line-seize,
>>> call-info, sla, include-session-description, presence.winfo,
>>> message-summary, refer
>>> Content-Type: application/sdp
>>> Content-Disposition: session
>>> Content-Length: 223
>>> X-FS-Support: update_display,send_info
>>> Remote-Party-ID: "4151502" <sip:4151502 at 10.11.12.13
>>> >;party=calling;screen=yes;privacy=off
>>>
>>> v=0
>>> o=FreeSWITCH 1426677605 1426677606 IN IP4 10.11.12.13
>>> s=FreeSWITCH
>>> c=IN IP4 10.11.12.13
>>> t=0 0
>>> m=audio 23230 RTP/AVP 8 101
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=silenceSupp:off - - - -
>>> a=ptime:20
>>>
>>> *First INVITE sent to FreeSWITCH by Asterisk (forked call):*
>>> INVITE sip:7712552 at 10.11.12.13 SIP/2.0
>>> Via: SIP/2.0/UDP 172.26.0.62:5060;branch=z9hG4bK416db3f1;rport
>>> Max-Forwards: 70
>>> From: "4151502" <sip:4151502 at 172.26.0.62>;tag=as24a51ba6
>>> To: <sip:7712552 at 10.11.12.13>
>>> Contact: <sip:4151502 at 172.26.0.62:5060>
>>> Call-ID: 135674a534fad0fd5bfff55c2fdc3280 at 172.26.0.62:5060
>>> CSeq: 102 INVITE
>>> User-Agent: Asterisk PBX 1.8.15-cert2
>>> Date: Wed, 18 Mar 2015 17:47:11 GMT
>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>>> INFO, PUBLISH
>>> Supported: replaces, timer
>>> Diversion: <sip:5344446 at 172.26.0.62>
>>> Content-Type: application/sdp
>>> Content-Length: 312
>>>
>>> v=0
>>> o=root 693576967 693576967 IN IP4 172.26.0.62
>>> s=Asterisk PBX 1.8.15-cert2
>>> c=IN IP4 172.26.0.62
>>> t=0 0
>>> m=audio 30440 RTP/AVP 8 0 9 101
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:9 G722/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>>
>>> Second INVITE sent to FreeSWITCH by Asterisk (forked call):
>>> INVITE sip:6595454 at 10.11.12.13 SIP/2.0
>>> Via: SIP/2.0/UDP 172.26.0.62:5060;branch=z9hG4bK6796aff1;rport
>>> Max-Forwards: 70
>>> From: "4151502" <sip:4151502 at 172.26.0.62>;tag=as22f810b0
>>> To: <sip:6595454 at 10.11.12.13>
>>> Contact: <sip:4151502 at 172.26.0.62:5060>
>>> Call-ID: 6979c3dd69c5f8e557131e485466ad57 at 172.26.0.62:5060
>>> CSeq: 102 INVITE
>>> User-Agent: Asterisk PBX 1.8.15-cert2
>>> Date: Wed, 18 Mar 2015 17:47:11 GMT
>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>>> INFO, PUBLISH
>>> Supported: replaces, timer
>>> Diversion: <sip:5344446 at 172.26.0.62>
>>> Content-Type: application/sdp
>>> Content-Length: 310
>>>
>>> v=0
>>> o=root 89056081 89056081 IN IP4 172.26.0.62
>>> s=Asterisk PBX 1.8.15-cert2
>>> c=IN IP4 172.26.0.62
>>> t=0 0
>>> m=audio 30708 RTP/AVP 8 0 9 101
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:9 G722/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=silenceSupp:off - - - -
>>> a=ptime:20
>>> a=sendrecv
>>>
>>> *First INVITE sent by FreeSWITCH to Kamailio (call forked by Asterisk):*
>>> INVITE sip:7712552 at 172.25.200.101 SIP/2.0
>>> Via: SIP/2.0/UDP 172.25.200.111:5080;rport;branch=z9hG4bK3jvS9UyXU216H
>>> Max-Forwards: 69
>>> From: "4151502" <sip:4151502 at 172.25.200.111>;tag=Z6pSHe2eXSB2p
>>> To: <sip:7712552 at 172.25.200.101>
>>> Call-ID: a7d2b58b-4839-1233-73b9-00215e2c8c90
>>> CSeq: 73014353 INVITE
>>> Contact: <sip:mod_sofia at 172.25.200.111:5080>
>>> User-Agent: FreeSWITCH-mod_sofia/1.2.12
>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>> REGISTER, REFER, NOTIFY
>>> Supported: timer, precondition, path, replaces
>>> Allow-Events: talk, hold, conference, refer
>>> Content-Type: application/sdp
>>> Content-Disposition: session
>>> Content-Length: 209
>>> Diversion: <sip:5344446 at 172.25.200.101>
>>> X-FS-Support: update_display,send_info
>>>
>>> v=0
>>> o=FreeSWITCH 1426681031 1426681032 IN IP4 172.25.200.111
>>> s=FreeSWITCH
>>> c=IN IP4 172.25.200.111
>>> t=0 0
>>> m=audio 19804 RTP/AVP 8 0 9 101 13
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=ptime:20
>>>
>>> *Second INVITE sent by FreeSWITCH to Kamailio (call forked by Asterisk):*
>>> INVITE sip:6595454 at 172.25.200.101 SIP/2.0
>>> Via: SIP/2.0/UDP 172.25.200.111:5080;rport;branch=z9hG4bK4UNjBQF1rBrSD
>>> Max-Forwards: 69
>>> From: "4151502" <sip:4151502 at 172.25.200.111>;tag=0FgjK9jjt21mj
>>> To: <sip:6595454 at 172.25.200.101>
>>> Call-ID: a7d2dd62-4839-1233-73b9-00215e2c8c90
>>> CSeq: 73014353 INVITE
>>> Contact: <sip:mod_sofia at 172.25.200.111:5080>
>>> User-Agent: FreeSWITCH-mod_sofia/1.2.12
>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>> REGISTER, REFER, NOTIFY
>>> Supported: timer, precondition, path, replaces
>>> Allow-Events: talk, hold, conference, refer
>>> Content-Type: application/sdp
>>> Content-Disposition: session
>>> Content-Length: 209
>>> Diversion: <sip:5344446 at 172.25.200.101>
>>> X-FS-Support: update_display,send_info
>>>
>>> v=0
>>> o=FreeSWITCH 1426669459 1426669460 IN IP4 172.25.200.111
>>> s=FreeSWITCH
>>> c=IN IP4 172.25.200.111
>>> t=0 0
>>> m=audio 31376 RTP/AVP 8 0 9 101 13
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=ptime:20
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.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
>>>
>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>
--
*Brian West*
brian at freeswitch.org
*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com
ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY!
*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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