- FreeSWITCH-users - FreeSWITCH Mailing Lists

fs core dsn, agents, tiers and queues
by David Villasmil 16 Jul '14

16 Jul '14

Hello Guys, I configured core-dsn with mysql and it's working fine. I also configured mod_callcenter. When I restarted FS, it created the tables "agents" and "tiers", but did not create "queues"... is this by desing? Thanks!

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Stuck at branch v1.4 commit 9479729
by Victor Chukalovskiy 16 Jul '14

16 Jul '14

Good day, not sure what's wrong here. My FS installation appears stuck at this version: > FreeSWITCH Version 1.4.6+git~20140603T193516Z~9479729340~64bit (git > 9479729 2014-06-03 19:35:16Z 64bit) "make current" and/or "git pull" and recompiling don't help. However, if I look in fisheye there was lots of activity since 2014-06-03 in branch v1.4. Thanks!

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Git.freeswitch.org now points at stash!
by Ken Rice 16 Jul '14

16 Jul '14

Hey FreeSWITCHers, If you have not updated the remotes for you local git checkouts, you will no longer be able to git pull. The old git server has been retired and everything now points at the stash server. Visit https://stash.freeswitch.org/ for the repo that you are looking for and click the clone button to get the correct URL you need to update your remotes to! See you @ ClueCon! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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ClueCon Weekly! Today Bogdan-Andrei Iancu of OpenSIPS!
by Ken Rice 16 Jul '14

16 Jul '14

Join us today at 1PM Eastern, 10A Pacific for ClueCon Weekly! Today¹s special guest is Bogdan-Andrei Iancu, project leader of OpenSIPS! Drop in and find out the latest on OpenSIPS and get a preview of what Bogdan may be talking about at ClueCon 2014! For Access: Visit https://conference.freeswitch.org/ and use your headset for Verto and HD Audio (OPUS/48K/Stereo)!! Call sip:888@conference.freeswitch.org from your sip phones Or +1-919-386-9900 from the PSTN Want to know more about OpenSIPS? Come To ClueCon! Monday @ ClueCon is the OpenSIPS Summit visit http://www.opensips.org/Community/Summit-2014Chicago for more info! And use the discount code ³clucon2014² to get a 25% discount for OpenSIPS Summit! Then stay for ClueCon and get the FreeSWITCH Training on Friday! Visit ClueCon.com for more info! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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mod_httapi caching and sample rate
by Neil Patel 16 Jul '14

16 Jul '14

I have a Lua IVR app that is reading files from a remote server. mod_httapi is caching files to local storage. However, I see this error in the logs frequently: freeswitch.log.2014-07-14-20-00-06.1:2014-07-14 19:39:40.690935 [DEBUG] switch_core_file.c:216 File /usr/local/freeswitch/storage/http_file_cache/7b1120ea3a0ebce2718d6f5588846b9e.wav sample rate 44100 doesn't match requested rate 8000 Does this mean there is a cache miss? I am using session:playAndGetDigits to playback the file, how do I adjust my request to ensure a cache hit? Thanks, Neil

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hangup_time zero
by Aqs Younas 16 Jul '14

16 Jul '14

I am new to freeswitch.I am facing problem in cdr. json_cdr module is giving perfect call start time and duration but hangup_time is 0. Why i am getting hangup_time zero but correct duration of call? Any help would be appreciated. Thanks

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Out of band Signaling (RFC2833) getting lost between legs
by Örn Arnarson 16 Jul '14

16 Jul '14

Hello, I have an inbound call to FS with RFC2833 signaling. FreeSWITCH bridges the call to another profile. A pcap shows a certain DTMF sequence on the inbound leg, and another on the outbound leg. The sequence on the outbound leg is missing some digits from the sequence. Seemingly at random -- I can generate the same sequence twice and a different sequence will be put through. The interval between the digits does not seem to matter either. Firstly, has anyone ever heard of something similar? Secondly, I was informed by someone on #freeswitch on freenode that FS actually regenerates the signaling on the second leg. Can I make FS just blindly forward the RTP events instead of regenerating them to avoid this problem? Regards, Örn

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wss conection closed
by Javier Menendez 16 Jul '14

16 Jul '14

Hi, I am trying to get the wss connection work, tried everything and still doesn't work, I don't think it is a certificate problem because if I try this manually: curl -v https://myhost.com:10081/ -H "Upgrade: WebSocket" -H "Connection: Upgrade" -H "Sec-WebSocket-Protocol: sip" -H "Sec-WebSocket-Key: +LLGYSDSKelND6UVF9z71w==" -H "Sec-WebSocket-Version: 13" * About to connect() to myhost.com port 10081 (#0) * Trying X.X.X.X... connected * Connected to myhost.com (X.X.X.X) port 10081 (#0) * successfully set certificate verify locations: * CAfile: none CApath: /etc/ssl/certs * SSLv3, TLS handshake, Client hello (1): * SSLv3, TLS handshake, Server hello (2): * SSLv3, TLS handshake, CERT (11): * SSLv3, TLS handshake, Server finished (14): * SSLv3, TLS handshake, Client key exchange (16): * SSLv3, TLS change cipher, Client hello (1): * SSLv3, TLS handshake, Finished (20): * SSLv3, TLS change cipher, Client hello (1): * SSLv3, TLS handshake, Finished (20): * SSL connection using AES256-SHA * Server certificate: * subject: OU=Domain Control Validated; CN=myhost.com * start date: 2014-06-16 10:09:42 GMT * expire date: 2016-06-07 11:02:46 GMT * subjectAltName: myhost.com matched * issuer: C=US; ST=Arizona; L=Scottsdale; O=GoDaddy.com, Inc.; OU= http://certs.godaddy.com/repository/; CN=Go Daddy Secure Certificate Authority - G2 * SSL certificate verify ok. > GET / HTTP/1.1 > User-Agent: curl/7.21.0 (i486-pc-linux-gnu) libcurl/7.19.7 OpenSSL/0.9.8o zlib/1.2.3.3 libidn/1.15 > Host: myhost.com:10081 > Accept: */* > Upgrade: WebSocket > Connection: Upgrade > Sec-WebSocket-Protocol: sip > Sec-WebSocket-Key: +LLGYSDSKelND6UVF9z71w== > Sec-WebSocket-Version: 13 > < HTTP/1.1 101 Switching Protocols < Upgrade: websocket < Connection: Upgrade < Sec-WebSocket-Accept: CQsVOMdurBA so it seems to work, but if I try with jssip or sipml5 library I got this trace log and the socket gets disconnected within half second freeswitch@internal> tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0x7f2198004f20): events IN tport.c:862 tport_alloc_secondary() tport_alloc_secondary(0x7f2198004f20): new secondary tport 0x7f21980afb20 tport.c:2640 tport_accept() tport_accept(0x7f21980afb20): new connection from wss/130.117.88.33:62056/sips tport.c:2773 tport_wakeup() tport_wakeup(0x7f21980afb20): events IN tport.c:2864 tport_recv_event() tport_recv_event(0x7f21980afb20) tport.c:2296 tport_set_secondary_timer() tport(0x7f21980afb20): reset timer tport.c:2773 tport_wakeup() tport_wakeup(0x7f21980afb20): events IN HUP ERR nta.c:2719 agent_tp_error() nta_agent: tport: Conexión reinicializada por la máquina remota tport.c:2090 tport_close() tport_close(0x7f21980afb20): wss/ 130.117.88.33:62056/sips I am using last version from git and I also tried with 1.4 versions. I have accept-blind-auth and accept-blind-reg to true, any clue?

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confirmation for group_confirm
by Grant Bagdasarian 16 Jul '14

16 Jul '14

Hello, Is there a way to know in the incoming channel (A) if one of the destinations has pressed the group_confirm_key? Something like a "Answer Confirmation" confirmation. I'm bridging the incoming leg to multiple destinations simultaneously and on answer they have to press the confirm key in order to accept the call. When no one has answered or pressed the key, the incoming channel should be bridged to a final destination. I have no idea how to do the final bridge if the above condition is met. Simultaneous bridging and configuring the group_confirm is no problem. I'd like to this all of this in XML dialplan if possible. Other suggestions are welcome as well. Any ideas? Regards, Grant

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Freeswitch Week in Review(Master Branch) July 6th-12th
by Kathleen King 15 Jul '14

15 Jul '14

Hello, again. This week in the FreeSWITCH master branch we had 40 commits. We had quite a few new features this week such as work in mod_LDAP to allow for authenticating from LDAP instead of the XML directory, added a new option to request TEP to be used for an outbound fax call in spandsp, added additional warnings to remind users to change the default password in the default dialplan, support for hardware DTMF in ftmod_gsm, and added clear channel for mono participants while in a conference with 3D callers. These are just a few of the commits for this week and more miscellaneous commits can be found in the attached text document. The following bugs were squashed: ba067f3 Fix for failing rpm build in mod_spidermonkey. Jira: http://jira.freeswitch.org/browse/FS-6563 e1d814b Fix for CDR insert fail in mod_cdr_pg_csv when variable with quote="false" attribute is not defined in mod_cdr_pg_csv. Jira: http://jira.freeswitch.org/browse/FS-6565 8951d26 Fixed a strepoch daylight saving time bug in FS core. Jira: http://jira.freeswitch.org/browse/FS-6626 8ec7b25 Enabled fscore_pb to work on OS X. Jira: http://jira.freeswitch.org/browse/FS-4939 21ae587 Removed disabling of require timer for T.38 re-invites in mod_sofia. New features that were added: 9a4e29d Authenticate users from LDAP instead of XML directory. Jira: http://jira.freeswitch.org/browse/FS-4870 a79a9ea Set total detect time variable in mod_avmd. Jira: http://jira.freeswitch.org/browse/FS-6373 ec00823 FS-5516 allow get params in URL in mod_xml_cdr. Jira: http://jira.freeswitch.org/browse/FS-5516 28e90ad FS-5501 fire event on profile start in mod_sofia Jira: http://jira.freeswitch.org/browse/FS-5501 a03963f Fold livetables into verto obj and expose moderator mode in mod_verto. 0fb179a Add new option to request TEP to be used for an outbound fax call in mod_spandsp a99df48 FS-4861 improve logging when using c++ wrapper in FS core. Jira: http://jira.freeswitch.org/browse/FS-4861 e79f872 Additional fields can now be specified by adding LOG_FIELDS[name=string_value,@#name=number_value] to beginning of log. Added _microtimestamp additional field since graylog2 only has millisecond timestamp precision in mod_graylog2. 07f4c62 Added ORDER BY score to select statement for 'callcenter_config queue list members QUEUE' in mod_conference. Jira: http://jira.freeswitch.org/browse/FS-6646 ebe4238 Added additional alerting if default directory password is not changed in default dialplan configs. 4bc1b73 Allow hwdtmf parameter to accept detect or generate in ftdmod_gsm. eae1c44 Additional input checks on LOG_FIELDS[] in mod_graylog2. 4021482 Added additional checks that HDLC frames arriving at the T.38 gateway engine begin with the correct 2 bytes in spandsp. 0b6f10a Added support for hardware DTMF in ftmod_gsm. a0a8a66 Increased expires from integer to bigint in mod_sofia. Jira: http://jira.freeswitch.org/browse/FS-6420 64489c5 Added support for DTMF generation to be performed in the signaling modules (currently only used by the gsm signaling driver) in freetdm fd9d77f Add clear channel for mono participants while in a conference with 3d callers in mod_conference. 88940d3 Log websocket write failures in mod_verto. Improvements in cross platform build supports: 99d0f13 Attempt to fix build of mod_cdr_pg_csv Jira: http://jira.freeswitch.org/browse/FS-6566 In terms of stability these were the use cases that were fixed: e742522 Fix several leaks on module shutdown in mod_freetdm. Feedback is welcome at kathleen.king(a)quentustech.com and the commits for this week are in the attached file along with the corresponding Jira links. -- Kathleen King Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (253) 617-3210 Cell: (703) 859-3757 kathleen.king(a)quentustech.com

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