The Call-IDs are different on each INVITE. Not the same as you claim. Call-ID: 174ff657-48f7-1233-d586-080027f911ca Call-ID: 174f7870-48f7-1233-d586-080027f911ca The first part varies even though the others are the same. On 20 March 2015 at 09:46, Örn Arnarson <orn@arnarson.net> wrote:
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@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@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@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@192.168.10.101>;tag=eU3yr5rNjcvDa To: <sip:7712552@192.168.10.3> Call-ID: 174ff657-48f7-1233-d586-080027f911ca CSeq: 73055034 INVITE Contact: <sip:mod_sofia@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@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@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@192.168.10.101>;tag=Dja6pa8HN35te To: <sip:6595454@192.168.10.3> Call-ID: 174f7870-48f7-1233-d586-080027f911ca CSeq: 73055034 INVITE Contact: <sip:mod_sofia@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@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@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@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@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@172.25.200.121>;tag=33vB4BmmDtU0B To: <sip:5344446@172.25.200.101> Call-ID: 84b63791-4839-1233-639f-00215e2db0e0 CSeq: 73014324 INVITE Contact: <sip:mod_sofia@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@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@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@10.11.12.13>;tag=2BaZj0t076Q9B To: <sip:5344446@172.26.0.62:5060> Call-ID: a7c77ea5-4839-1233-73b9-00215e2c8c90 CSeq: 73014353 INVITE Contact: <sip:mod_sofia@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@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@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@172.26.0.62>;tag=as24a51ba6 To: <sip:7712552@10.11.12.13> Contact: <sip:4151502@172.26.0.62:5060> Call-ID: 135674a534fad0fd5bfff55c2fdc3280@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@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@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@172.26.0.62>;tag=as22f810b0 To: <sip:6595454@10.11.12.13> Contact: <sip:4151502@172.26.0.62:5060> Call-ID: 6979c3dd69c5f8e557131e485466ad57@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@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@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@172.25.200.111>;tag=Z6pSHe2eXSB2p To: <sip:7712552@172.25.200.101> Call-ID: a7d2b58b-4839-1233-73b9-00215e2c8c90 CSeq: 73014353 INVITE Contact: <sip:mod_sofia@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@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@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@172.25.200.111>;tag=0FgjK9jjt21mj To: <sip:6595454@172.25.200.101> Call-ID: a7d2dd62-4839-1233-73b9-00215e2c8c90 CSeq: 73014353 INVITE Contact: <sip:mod_sofia@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@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
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