[Freeswitch-users] FreeSWITCH using same Call-ID for forked calls

Fri Mar 20 12:46:53 MSK 2015

Look at the URI -- it's different in each case. These were received within
1 ms from each other as well. This is not a retransmission.
Do you think this is a bug? Should I mail this to the dev list?

On Thu, Mar 19, 2015 at 7:11 PM, Brian West <brian at freeswitch.org> wrote:

> 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:
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>>>>
>>>
>>>
>>>
>>> --
>>>
>>> *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
>>>
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>>>
>>
>>
>> _________________________________________________________________________
>> 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
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> <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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
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>
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