July 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Break unaswered bridge
by Alexey Melnichuk 23 Jul '15

23 Jul '15

This is simple dialplan that I try to implement <extension> <condition field="destination_number" expression="^\d+$" break="on-true"> <action application="bind_digit_action" data="conf,*1,exec:execute_extension,xfer"/> <action application="digit_action_set_realm" data="conf"/> <action application="conference" data="100@default"/> </condition> <condition field="destination_number" expression="^xfer$" break="on-true"> <action application="bind_digit_action" data="xfer,*1,?????????"/> <action application="digit_action_set_realm" data="xfer"/> <action application="digit_action_set_realm" data="xfer"/> <action application="read" data="2 11 'tone_stream://%(10000,0,350,440)' num 30000 #"/> <action application="bridge" data="user/${num}@${domain_name}"/> </condition> </extension> I have two problem with this dialplan 1. in secon `bind_digit_action` I can not use `exec` only `api`. e.g. <action application="bind_digit_action" data="xfer,*1,exec:info"/> does not work. application executes only when I return from first `exec:execute_extension` 2. I can not cancel bridge before answer I can not use transfer because I do not whant leave conference (e.g. user has `end_conference=true` flag). I try use uuid_kill from Lua like "xfer,*1,lua:kill_legs.lua ${uuid}" -- kill_legs.lua local uuid = argv[1] local sql = ("select uuid from channels where call_uuid='%s' and uuid<>'%s'"):format(uuid, uuid) dbh:query(sql, function(row) local res = api:executeString("uuid_kill " .. row.uuid) end) But this is does not work because it kills only active channels, but for dial-string like `loopback/111/my.domain, [leg_delay_start=20]loopback/222/my.domain` second channels creates only after 20 sec. And because I may have more then one channels I can not use `originate_uuid` In documentation I found `break` application but as I say I can not use it. `uuid_break` api break only play audio.

1 0

Websocket creation logging
by Oleg Stolyar 22 Jul '15

22 Jul '15

Hi guys, I am experiencing what looks like a memory leak on my FS servers but I suspect that there is a bug in my WebRTC client code that keep establishing new websockets and not closing them properly which causes FS to require more and more memory. Is there a way in FS to log creation and closing of websockets? Thank you Oleg

2 7

Enterprise originate not honouring fs_path in 302 redirect
by Asgaroth 22 Jul '15

22 Jul '15

Hi All, First of all, I'm a relatively new user of FreeSWITCH so, please bear with me if I'm asking a silly question. I have users registering through multiple kamailio loadbalancer/registrars, so when we build a bridge command to multiple local users we send the initial invite(s) to the kamailio lookup server which then responds with a 302 redirect message specifying the contact details of the user. What I am trying to do is get FreeSWITCH to call a particular user via a particular loadbalancer based on information recieved in the 302 redirect. From what I have read in the archives, we can achieve this with the fs_path directive on the bridge command. However, how/where do I specify the fs_path in the contact header in the 302 redirect in such a manner that FreeSWITCH will honour the fs_path directive. I originally tried setting a fs_path parameter in the contact of the 302 redirect message, something like the following: Contact: <sip:subscriber@1.2.3.4:5678;rinstance=9af71926212a25cf;transport=UDP;received=sip:5.6.7.8:9012;fs_path=sip:1.2.3.5> However, FreeSWITCH ignores the fs_path directive here and attempts to send this call directly to 1.2.3.4, instead of sending it via 1.2.3.5. Is this at all possible using the enterprise originate (:_: seperator) feature of the bridge command? We're attempting this with FreeSWITCH v 1.4.20 ("official" rpm packages from the repo) Any info/tips/suggestions/beatings much appreciate :) Thanks

4 6

Disable 401 and 407 for internal profile
by Admin 22 Jul '15

22 Jul '15

Hi, I want to configure FS to not challenge calls that come in from an internal profile (internal network devices), meaning disable SIP '401 Unauthorized' and '407 Proxy Authentication Required' for calls that belong to internal profile. I changed 'auth-calls' parameter to 'false', in internal.xml file but FS still sends 401 method when a user attempts to register and it sends 407 when an invite comes in. It seems the parameter is for ACL though. Is there another parameter to change in order to disable SIP 401 and 407 methods? Thank you.

6 13

Disable 401 and 407 for internal profile
by Admin 22 Jul '15

22 Jul '15

I have had the dialplan configured for users 1000-1999 ("destination_number" expression="^(1[0-9][0-9][0-9])$"> for months. I have run thousands of calls using the same dialplan and users config. files. To disable auth., i made two changes in internal.xml. I set 'auth-calls' value to 'false' and removed the line that contains 'apply-inbound-acl'. I didn't make any changes to neither dialplan nor the users config. files.

2 1

Announcing the ClueCon Coder Games!
by Ken Rice 22 Jul '15

22 Jul '15

New Post on freeswitch.org from Kathleen King check it out at http://ift.tt/1CUbP1a Announcing the ClueCon Coder Games! So You Think You Can Code? You’ve seen the presentations, you’ve asked your questions, you have the resources, now it is your time to shine by using the sponsor APIs to create something exciting! We want to see what you can do! Bonus points for each API you can incorporate! Go check out the APIs now to get a head start on the competition and get those creative juices flowing! You have less than two weeks to prepare! Sponsor APIs: FreeSWITCH, Tropo, Kandy, Twilio, Plivo, and more… IPv6 Round Table IPv6 and why you should deploy it ASAP: John Brzozowski, Fellow and Chief Architect, IPv6 at Comcast, Bill Sandiford President of CNOC, Member of the board at ARIN. Flowroute – Jeopardy Think you know about SIP? Do you know enough to beat the competition? Flowroute is hosting a SIP themed game of Jeopardy! Put your brain to the test and come see how much you really know! DTMF-u Do you like all things games? Well, then this is the game for you! But there is a twist! Before you can win the game you must build it! Using your choice of language or API you must build a game! There will be three categories of DTMF-u; Tic-tac-toe, DTMF pattern recognition, or Freestyle. You could build something that plays a random DTMF sequence, receives player input, and then either continues or fails the player. Or maybe a WebRTC based game of tic-tac-toe? Or surprise us! Have fun with it! Creative ways to fail a player may give you bonus points. The top three games will be played by everyone and the winner of each will take home a prize! All gaming bots will be screened via a Turing test to ensure no unintended apocalyptic consequences. Show and Tell Alright, now is your chance! You have been playing with the code all day and this is your chance to show off! We want to know what you’ve done and how you did it. Use your creativity and skills as a programmer to impress the judges and win a prize! This is a no holds barred all out free for all! Any language doing anything! Knock our socks off and take home a fabulous prize and a year’s supply of bragging rights! Raffle Grand Prize! The grand prize is a laser engraved commemorative FreeSWITCH 1.6 Edition dual-core 13″ Retina MacBook Pro!

1 0

ClueCon Coder Games Monday Aug. 3rd 2015
by Brian West 21 Jul '15

21 Jul '15

[image: Description: HDD:Users:anthm:Downloads:ccxx.jpg] *August 3rd 2015* August 3rd – August 6th 2015 877-742-2583 • marketing(a)cluecon.com Register NOW! · $1299 Staying at the Hotel · $1599 Staying Elsewhere *Contact Us* https://cluecon.com marketing(a)cluecon.com *So You Think You Can Code?* You’ve seen the presentations, you’ve asked your questions, you have the resources, now it is your time to shine by using the sponsor APIs to create something exciting! We want to see what you can do! Bonus points for each API you can incorporate! Go check out the APIs now to get a head start on the competition and get those creative juices flowing! You have less than two weeks to prepare! Sponsor APIs: FreeSWITCH <http://www.freeswitch.org> - Corvisa <https://www.corvisa.com> - Tropo <https://www.tropo.com> - Kandy <https://www.kandy.io> - Twilio <https://www.twilio.com> - Plivo <https://www.plivo.com> - and more... IPv6 Round Table IPv6 and why you should deploy it ASAP: John Brzozowski, Fellow and Chief Architect, IPv6 at Comcast, Bill Sandiford President of CNOC, Member of the board at ARIN. Flowroute - Jeopardy Think you know about SIP? Do you know enough to beat the competition? Flowroute is hosting a SIP themed game of Jeopardy! Put your brain to the test and come see how much you really know! DTMF-u Do you like all things games? Well, then this is the game for you! But there is a twist! Before you can win the game you must build it! Using your choice of language or API you must build a game! There will be three categories of DTMF-u; Tic-tac-toe, DTMF pattern recognition, or Freestyle. You could build something that plays a random DTMF sequence, receives player input, and then either continues or fails the player. Or maybe a WebRTC based game of tic-tac-toe? Or surprise us! Have fun with it! Creative ways to fail a player may give you bonus points. The top three games will be played by everyone and the winner of each will take home a prize! All gaming bots will be screened via a Turing test to ensure no unintended apocalyptic consequences. Show and Tell Alright, now is your chance! You have been playing with the code all day and this is your chance to show off! We want to know what you've done and how you did it. Use your creativity and skills as a programmer to impress the judges and win a prize! This is a no holds barred all out free for all! Any language doing anything! Knock our socks off and take home a fabulous prize and a year’s supply of bragging rights! Grand Prize [image: Description: mac1]The grand prize is a laser engraved commemorative FreeSWITCH 1.6 Edition dual-core 13" Retina MacBook Pro! See the Important Dates Section for Registration details! Why I Think You MUST COME To ClueCon! [image: Description: kk]Hi, I’m Kathleen. I’m the FreeSWITCH and ClueCon Social Media Correspondent. I’ve been working hard all year keeping you all up to date on what’s going on with FreeSWITCH. Today I’m here to let you know more about the upcoming ClueCon 2015 Conference! This year we are adding an optional day on Monday with an all-day Hack-A-Thon with great coding contests, game shows and kick-off fun! If you are interested in WebRTC, Voice over IP or Open Source projects like FreeSWITCH, ClueCon is the greatest opportunity you have to gain exposure to the most knowledge and technology in one place. Also, it’s the most fun you can possibly have while still getting a ton of work done! I really look forward to seeing you all there and enjoying the amazing talks, the Epic Annual Kick-Off Pizza Party, The Gigabit Reception and so much more. Make sure you register today so you can reserve your place among the attendees! Be sure to follow us on Facebook and Twitter to get my latest updates in info! [image: Inline image 1] <https://facebook.com/cluecon> [image: Inline image 2] <https://twitter.com/cluecon> [image: Inline image 3] <https://www.youtube.com/watch?v=NLaDpGQuZDA> [image: Inline image 4] <https://www.youtube.com/watch?v=9XXgW34t40s>

1 0

Disable 401 and 407 for internal profile
by Admin 21 Jul '15

21 Jul '15

Thanks Brian. That seems to work partially and strangely too. All calls made to extensions above '1019' stopped working now. When a call is destined to any extension above '1019' comes back with '480 Temporarily Unavailable'. Below is a pasted copy of the SIP trace for a call from 1001 to 1019 (works), and 1001 to 1020 (doesn't work). Registration output is also pasted below. ========================================================================== recv 726 bytes from udp/[192.168.1.100]:5062 at 11:18:28.880410: ------------------------------------------------------------------------ INVITE sip:1020@192.168.1.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3263-1-0 From: 1001 <sip:1001@192.168.1.100>;tag=32631 To: 1020 <sip:1020@192.168.1.100> Call-ID: 1-3263(a)192.168.1.100 CSeq: 1 INVITE Contact: 1001 <sip:1001@192.168.1.100:5062> Max-Forwards: 70 Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE, SUBSCRIBE Supported: 100rel, replaces, from-change Timestamp: 1 Content-Type: application/sdp Content-Length: 238 v=0 o=1001 53655765 2353687637 IN IP4 192.168.1.100 s=Talk c=IN IP4 192.168.1.100 t=0 0 m=audio 6001 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 m=video 6001 RTP/AVP 31 a=rtpmap:31 H261/90000 ------------------------------------------------------------------------ send 333 bytes to udp/[192.168.1.100]:5062 at 11:18:28.880719: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3263-1-0 From: 1001 <sip:1001@192.168.1.100>;tag=32631 To: 1020 <sip:1020@192.168.1.100> Call-ID: 1-3263(a)192.168.1.100 CSeq: 1 INVITE Timestamp: 1 0.000152 User-Agent: FreeSWITCH-mod_sofia/1.4.20 +git~20150703T164215Z~b95362f965~64bit Content-Length: 0 ------------------------------------------------------------------------ 2015-07-21 11:18:28.863966 [NOTICE] switch_channel.c:1075 New Channel sofia/internal/1001(a)192.168.1.100 [d4084b55-8351-48fa-87ff-01d4b2355d4e] 2015-07-21 11:18:28.863966 [INFO] mod_dialplan_xml.c:635 Processing 1001 <1001>->1020 in context public 2015-07-21 11:18:28.863966 [NOTICE] switch_core_state_machine.c:315 sofia/internal/1001(a)192.168.1.100 has executed the last dialplan instruction, hanging up. 2015-07-21 11:18:28.863966 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/internal/1001(a)192.168.1.100 [CS_EXECUTE] [NORMAL_CLEARING] send 836 bytes to udp/[192.168.1.100]:5062 at 11:18:28.885154: ------------------------------------------------------------------------ SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3263-1-0 Max-Forwards: 70 From: 1001 <sip:1001@192.168.1.100>;tag=32631 To: 1020 <sip:1020@192.168.1.100>;tag=mrKveDD123K3K Call-ID: 1-3263(a)192.168.1.100 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.4.20 +git~20150703T164215Z~b95362f965~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 Remote-Party-ID: "1020" <sip:1020@192.168.1.100>;party=calling;privacy=off;screen=no ------------------------------------------------------------------------ 2015-07-21 11:18:28.883950 [NOTICE] switch_core_session.c:1641 Session 20 (sofia/internal/1001(a)192.168.1.100) Ended 2015-07-21 11:18:28.883950 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/1001(a)192.168.1.100 [CS_DESTROY] recv 361 bytes from udp/[192.168.1.100]:5062 at 11:18:28.887876: ------------------------------------------------------------------------ ACK sip:1020@192.168.1.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3263-1-0 From: 1001 <sip:1001@192.168.1.100>;tag=32631 To: 1020 <sip:1020@192.168.1.100>;tag=mrKveDD123K3K Call-ID: 1-3263(a)192.168.1.100 CSeq: 1 ACK Contact: <sip:sipp@192.168.1.100:5062;transport=UDP> Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ------------------------------------------------------------------------ >>>>>>>>>>> 1001 - 1019 - Working >>>>>>>>>>>>>>>> recv 726 bytes from udp/[192.168.1.100]:5062 at 11:20:10.977891: ------------------------------------------------------------------------ INVITE sip:1019@192.168.1.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3281-1-0 From: 1001 <sip:1001@192.168.1.100>;tag=32811 To: 1019 <sip:1019@192.168.1.100> Call-ID: 1-3281(a)192.168.1.100 CSeq: 1 INVITE Contact: 1001 <sip:1001@192.168.1.100:5062> Max-Forwards: 70 Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE, SUBSCRIBE Supported: 100rel, replaces, from-change Timestamp: 1 Content-Type: application/sdp Content-Length: 238 v=0 o=1001 53655765 2353687637 IN IP4 192.168.1.100 s=Talk c=IN IP4 192.168.1.100 t=0 0 m=audio 6001 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 m=video 6001 RTP/AVP 31 a=rtpmap:31 H261/90000 ------------------------------------------------------------------------ send 333 bytes to udp/[192.168.1.100]:5062 at 11:20:10.978324: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3281-1-0 From: 1001 <sip:1001@192.168.1.100>;tag=32811 To: 1019 <sip:1019@192.168.1.100> Call-ID: 1-3281(a)192.168.1.100 CSeq: 1 INVITE Timestamp: 1 0.000208 User-Agent: FreeSWITCH-mod_sofia/1.4.20 +git~20150703T164215Z~b95362f965~64bit Content-Length: 0 ------------------------------------------------------------------------ 2015-07-21 11:20:10.963951 [NOTICE] switch_channel.c:1075 New Channel sofia/internal/1001(a)192.168.1.100 [3ea757fa-c80f-471f-909c-21d6b35ccd94] 2015-07-21 11:20:10.963951 [INFO] mod_dialplan_xml.c:635 Processing 1001 <1001>->1019 in context public 2015-07-21 11:20:10.963951 [NOTICE] switch_ivr.c:1863 Transfer sofia/internal/1001(a)192.168.1.100 to XML[1019@default] 2015-07-21 11:20:10.963951 [INFO] mod_dialplan_xml.c:635 Processing 1001 <1001>->1019 in context default 2015-07-21 11:20:10.983953 [NOTICE] switch_channel.c:1075 New Channel sofia/internal/1019@192.168.1.100:5061 [2bb31faa-8cc2-400a-8997-3d5126acf148] send 1189 bytes to udp/[192.168.1.100]:5061 at 11:20:10.990948: ------------------------------------------------------------------------ INVITE sip:1019@192.168.1.100:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100;rport;branch=z9hG4bK8829733UK6ttr Max-Forwards: 68 From: "1001" <sip:1001@192.168.1.100>;tag=pa6Dj3e8vN08a To: <sip:1019@192.168.1.100:5061> Call-ID: 337749d6-aa67-1233-6ca8-705ab6872cf3 CSeq: 78411741 INVITE Contact: <sip:mod_sofia@192.168.1.100:5060> User-Agent: FreeSWITCH-mod_sofia/1.4.20 +git~20150703T164215Z~b95362f965~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 X-FS-Support: update_display,send_info Remote-Party-ID: "1001" <sip:1001@192.168.1.100>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1437471870 1437471871 IN IP4 192.168.1.100 s=FreeSWITCH c=IN IP4 192.168.1.100 t=0 0 m=audio 23740 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 325 bytes from udp/[192.168.1.100]:5061 at 11:20:10.994691: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.100;rport;branch=z9hG4bK8829733UK6ttr From: "1001" <sip:1001@192.168.1.100>;tag=pa6Dj3e8vN08a To: <sip:1019@192.168.1.100:5061>;tag=32791 Call-ID: 337749d6-aa67-1233-6ca8-705ab6872cf3 CSeq: 78411741 INVITE Contact: 1019 <sip:1019@192.168.1.100:5061> Content-Length: 0 ------------------------------------------------------------------------ 2015-07-21 11:20:10.983953 [NOTICE] sofia.c:6729 Ring-Ready sofia/internal/1019@192.168.1.100:5061! 2015-07-21 11:20:11.003950 [INFO] switch_ivr_originate.c:1193 Sending early media 2015-07-21 11:20:11.003950 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1001(a)192.168.1.100! send 1069 bytes to udp/[192.168.1.100]:5062 at 11:20:11.015428: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3281-1-0 From: 1001 <sip:1001@192.168.1.100>;tag=32811 To: 1019 <sip:1019@192.168.1.100>;tag=N1cNg8X4ZcapF Call-ID: 1-3281(a)192.168.1.100 CSeq: 1 INVITE Contact: <sip:1019@192.168.1.100:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.4.20 +git~20150703T164215Z~b95362f965~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 188 Remote-Party-ID: "1019" <sip:1019@192.168.1.100>;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1437471111 1437471112 IN IP4 192.168.1.100 s=FreeSWITCH c=IN IP4 192.168.1.100 t=0 0 m=audio 24500 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 m=video 0 RTP/AVP 19 ------------------------------------------------------------------------ recv 593 bytes from udp/[192.168.1.100]:5061 at 11:20:11.995950: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.100;rport;branch=z9hG4bK8829733UK6ttr From: "1001" <sip:1001@192.168.1.100>;tag=pa6Dj3e8vN08a To: <sip:1019@192.168.1.100:5061>;tag=32791 Call-ID: 337749d6-aa67-1233-6ca8-705ab6872cf3 CSeq: 78411741 INVITE Contact: 1019 <sip:1019@192.168.1.100:5061> Content-Type: application/sdp Content-Length: 238 v=0 o=5001 53655765 2353687637 IN IP4 192.168.1.100 s=Talk c=IN IP4 192.168.1.100 t=0 0 m=audio 6000 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 m=video 6000 RTP/AVP 31 a=rtpmap:31 H261/90000 ------------------------------------------------------------------------ send 360 bytes to udp/[192.168.1.100]:5061 at 11:20:11.998449: ------------------------------------------------------------------------ ACK sip:1019@192.168.1.100:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100;rport;branch=z9hG4bK9Hv29ymZgFHDm Max-Forwards: 70 From: "1001" <sip:1001@192.168.1.100>;tag=pa6Dj3e8vN08a To: <sip:1019@192.168.1.100:5061>;tag=32791 Call-ID: 337749d6-aa67-1233-6ca8-705ab6872cf3 CSeq: 78411741 ACK Contact: <sip:mod_sofia@192.168.1.100:5060> Content-Length: 0 ------------------------------------------------------------------------ 2015-07-21 11:20:11.983975 [NOTICE] sofia.c:7488 Channel [sofia/internal/1019@192.168.1.100:5061] has been answered send 1039 bytes to udp/[192.168.1.100]:5062 at 11:20:12.013969: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3281-1-0 From: 1001 <sip:1001@192.168.1.100>;tag=32811 To: 1019 <sip:1019@192.168.1.100>;tag=N1cNg8X4ZcapF Call-ID: 1-3281(a)192.168.1.100 CSeq: 1 INVITE Contact: <sip:1019@192.168.1.100:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.4.20 +git~20150703T164215Z~b95362f965~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 188 Remote-Party-ID: "Outbound Call" <sip:1019@192.168.1.100>;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1437471111 1437471112 IN IP4 192.168.1.100 s=FreeSWITCH c=IN IP4 192.168.1.100 t=0 0 m=audio 24500 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 m=video 0 RTP/AVP 19 2015-07-21 11:20:11.983975 [NOTICE] switch_ivr_originate.c:3523 Channel [sofia/internal/1001(a)192.168.1.100] has been answered ------------------------------------------------------------------------ recv 371 bytes from udp/[192.168.1.100]:5062 at 11:20:12.018300: ------------------------------------------------------------------------ ACK sip:1019@192.168.1.100:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3281-1-0 From: 1001 <sip:1001@192.168.1.100>;tag=32811 To: 1019 <sip:1019@192.168.1.100>;tag=N1cNg8X4ZcapF Call-ID: 1-3281(a)192.168.1.100 CSeq: 1 ACK Contact: 1001 <sip:1001@192.168.1.100:5062> Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ------------------------------------------------------------------------ recv 371 bytes from udp/[192.168.1.100]:5062 at 11:20:17.020325: ------------------------------------------------------------------------ BYE sip:1019@192.168.1.100:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3281-1-0 From: 1001 <sip:1001@192.168.1.100>;tag=32811 To: 1019 <sip:1019@192.168.1.100>;tag=N1cNg8X4ZcapF Call-ID: 1-3281(a)192.168.1.100 CSeq: 2 BYE Contact: 1001 <sip:1001@192.168.1.100:5062> Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ------------------------------------------------------------------------ 2015-07-21 11:20:17.023972 [NOTICE] sofia.c:952 Hangup sofia/internal/1001(a)192.168.1.100 [CS_EXECUTE] [NORMAL_CLEARING] send 465 bytes to udp/[192.168.1.100]:5062 at 11:20:17.035053: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3281-1-0 From: 1001 <sip:1001@192.168.1.100>;tag=32811 To: 1019 <sip:1019@192.168.1.100>;tag=N1cNg8X4ZcapF Call-ID: 1-3281(a)192.168.1.100 CSeq: 2 BYE User-Agent: FreeSWITCH-mod_sofia/1.4.20 +git~20150703T164215Z~b95362f965~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Length: 0 ------------------------------------------------------------------------ 2015-07-21 11:20:17.023972 [NOTICE] switch_ivr_bridge.c:758 Hangup sofia/internal/1019@192.168.1.100:5061 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] send 585 bytes to udp/[192.168.1.100]:5061 at 11:20:17.042705: ------------------------------------------------------------------------ BYE sip:1019@192.168.1.100:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100;rport;branch=z9hG4bKaUNUBt52Dr7ZF Max-Forwards: 70 From: "1001" <sip:1001@192.168.1.100>;tag=pa6Dj3e8vN08a To: <sip:1019@192.168.1.100:5061>;tag=32791 Call-ID: 337749d6-aa67-1233-6ca8-705ab6872cf3 CSeq: 78411742 BYE User-Agent: FreeSWITCH-mod_sofia/1.4.20 +git~20150703T164215Z~b95362f965~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 ------------------------------------------------------------------------ 2015-07-21 11:20:17.023972 [NOTICE] switch_core_session.c:1641 Session 21 (sofia/internal/1001(a)192.168.1.100) Ended 2015-07-21 11:20:17.023972 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/1001(a)192.168.1.100 [CS_DESTROY] 2015-07-21 11:20:17.023972 [NOTICE] switch_core_session.c:1641 Session 22 (sofia/internal/1019@192.168.1.100:5061) Ended 2015-07-21 11:20:17.023972 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/1019@192.168.1.100:5061 [CS_DESTROY] recv 317 bytes from udp/[192.168.1.100]:5061 at 11:20:17.051862: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.100;rport;branch=z9hG4bKaUNUBt52Dr7ZF From: "1001" <sip:1001@192.168.1.100>;tag=pa6Dj3e8vN08a To: <sip:1019@192.168.1.100:5061>;tag=32791 Call-ID: 337749d6-aa67-1233-6ca8-705ab6872cf3 CSeq: 78411742 BYE Contact: 1019 <sip:1019@192.168.1.100:5061> Content-Length: 0 ================================================================== Call-ID: 1-2629(a)192.168.1.100 User: 1001(a)192.168.1.100 Contact: "" <sip:1001@192.168.1.100:5061> Agent: SIPp Status: Registered(UDP)(unknown) EXP(2015-07-21 12:25:01) EXPSECS(3112) Ping-Status: Reachable Host: everest IP: 192.168.1.100 Port: 5061 Auth-User: unknown Auth-Realm: 192.168.1.100 MWI-Account: 1001(a)192.168.1.100 -------------------------------------------------------- Call-ID: 1-3203(a)192.168.1.100 User: 1020(a)192.168.1.100 Contact: "" <sip:1020@192.168.1.100:5061> Agent: SIPp Status: Registered(UDP)(unknown) EXP(2015-07-21 13:11:07) EXPSECS(5878) Ping-Status: Reachable Host: everest IP: 192.168.1.100 Port: 5061 Auth-User: unknown Auth-Realm: 192.168.1.100 MWI-Account: 1020(a)192.168.1.100 -------------------------------------------------------- Call-ID: 1-3216(a)192.168.1.100 User: 1019(a)192.168.1.100 Contact: "" <sip:1019@192.168.1.100:5061> Agent: SIPp Status: Registered(UDP)(unknown) EXP(2015-07-21 13:11:51) EXPSECS(5922) Ping-Status: Reachable Host: everest IP: 192.168.1.100 Port: 5061 Auth-User: unknown Auth-Realm: 192.168.1.100 MWI-Account: 1019(a)192.168.1.100

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mod_fifo in a database shared between multiple freswitches
by Mehdi Shirazi 21 Jul '15

21 Jul '15

HiIs it possible to use mode_fifo in a shared database to have a big call center on multi server freeswitch environment ? RegardsMehdi

2 1

problem with attended transfer
by Tomáš Boros 21 Jul '15

21 Jul '15

Hello all, I think I have found a bug in version 1.5.final+git~20150528T173517Z~6a2fc5e0f7~64bit My scenario is the following: A ----> calls ---> B B ----> calls ---> C B ---> sends REFER to join A and C (attended transfer via SIP) The two calls are connected A is able to talk to C, but after a while, the call is disconnected. When the REFER message arrives, FS sends BYE message to B with cause "NORMAL_CLEARING" and after a while (about 10 sec) FS sends a BYE message to C with cause "ATTENDED_TRANSFER" and the transferred call is hung up. I assume, that those causes should be swapped. BYE with attended transfer to B and no BYE at all for C until the call is ended by the endpoints. Do someone have this issue, how you fixed it? Should I upgrade my freeswitch to a newer version? -- Tomáš Boros

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