September 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Leg stuck in a conference
by Oleg Stolyar 24 Sep '14

24 Sep '14

Guys, I wonder if anyone else ran into this situation: Very rarely I have a conference participant stuck in a conference. When I try to run kick command on that conference, it returns "OK kicked <member ID>" but nothing happens. In the logs there is only this: db1790fe-fa75-42a6-9054-75c600b9539c 2014-09-23 15:43:07.066947 [DEBUG] mod_conference.c:5619 Send signal sofia/<profile>/anonymous(a)anonymous.invalid [BREAK] As an additional piece of information if I use uuid_kill on the leg, it does disconnect the leg and ends the conference. Any suggestions?

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is there a group_confirm_timeout setting?
by Phillip Jones 24 Sep '14

24 Sep '14

Hi there, I have question about group_confirm and I am hoping someone can help: I am bridging a call and prompting the called party to press 1 to answer. Works great: <action application="answer"/> <action application="set" data="group_confirm_file=phrase:ppn_to_accept_call_from"/> <action application="set" data="group_confirm_key=1"/> <action application="bridge" data="{leg_timeout=10}user2/1001 <action application="say" data="en number pronounced 54321"/> You will notice that leg_timeout is 10 (set low to test). So that gives the caller just a couple of seconds to answer. So I add: * <action application="set" data="group_confirm_cancel_timeout=true"/>* This cancels that leg_timeout - the call now will stay up until the called party presses 1 or either party hangs up. If a call goes to a cell phone voicemail - this can result in a long message of repeated "press 1 to accept call". And I cannot move the calling party on to the next dialplan action. I tried originate_timeout and call_timeout and all seemed to be cancelled by group_confirm_cancel_timeout, as you might expect. Is there a way to set a timeout in that group_confirm stage? For example a *group_confirm_timeout *setting. That would set the seconds the called party has to confirm? Thanks in anticipation. Phil

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Re: [Freeswitch-users] session:playAndGetDigits() play two files When digits are fetched
by Snabel Kabiya 24 Sep '14

24 Sep '14

Hi, I'm calling the method from lua script local digits = session:playAndGetDigits(minLength, maxLength, maxAttempts, 5000, termChar, menuPrompt, noMatchPrompt, grammarRegex, "digits", 1500); Now when i tried using file_string: menuPrompt_part1=/usr/audio/phrase1.wav menuPrompt_part2=/usr/audio/phrase2.wav local digits = session:playAndGetDigits(minLength, maxLength, maxAttempts, 5000, termChar, file_string://menuPrompt_part1!menuPrompt_part2, noMatchPrompt, grammarRegex, "digits", 1500); i got the following error: *2014-09-24 07:58:25.677222 [ERR] mod_lua.cpp:203 /usr/local/freeswitch/scripts/....: '<name>' expected near '/'* what did i miss? Thanks, Snabel ---------- Forwarded message ---------- > From: Michael Collins <msc(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Thu, 18 Sep 2014 10:08:38 -0700 > Subject: Re: [Freeswitch-users] session:playAndGetDigits() play two files > When digits are fetched > I think this might be what you're looking for: > https://wiki.freeswitch.org/wiki/Mod_file_string > > -MC > > On Thu, Sep 18, 2014 at 5:56 AM, Snabel Kabiya <snabel(a)lexifone.com> > wrote: > >> Hi, >> >> I'm using this method >> https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits >> I've this parameter >> >> - file = Sound file to play while digits are fetched >> >> i wanted to ask if i can play two files? if yes what's the syntax? >> >> Thanks, >> Snabel >> >>

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Dialplan Tools pickup - Bug/Memory Leak
by Lloyd Aloysius 24 Sep '14

24 Sep '14

Hello Switch crashing all the time when using the Pickup not immediately, but after couple times. Back trace show the pickup but do know where this is happening. https://jira.freeswitch.org/browse/FS-6851 Any help is appreciated. Thank you LLoyd

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ClueCon Weekly! Sep 24th 1PM Eastern! Darren Schreiber of 2600hz!
by Ken Rice 24 Sep '14

24 Sep '14

Greetings! Don¹t forget to join us today at 1PM Eastern (that¹s 10AM Pacific) for ClueCon Weekly! This week we¹ll have Darren Schreiber of 2600hz, co-author of the FreeSWITCH Cookbook, and FreeSWITCH Trainer! This week he¹ll be talking 2600hz and where they are headed plus an announcement! Join us on https://conference.freeswitch.org/ with your headset and click the Call 888 button! And Don¹t forget Thursdays @ 2PM Central (Noon Pacific) is the Weekly FreeSWITCH BugHunt! Join us to get your Jira Tickets, Pull Requests and Patches reviewed! K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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Transferring an outbound call
by Chirag Desai 24 Sep '14

24 Sep '14

Hi all, I'm currently using kamailio, and freeswitch. My set up is almost identical to the one here http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc The only difference is I've tweaked to allow for multiple domains and I use ASTPP. Also, users register with an alphanumeric name but have a 4 digit alias for easy dialling. This is configured within Kamailio. My sip phones (snom and jitsi) register fine and can call each other / transfer calls. Using ASTPP I am able to make calls via a SIP trunk to PSTN. When I make an outbound call (via ASTPP) to my landline for example, I'm unable to transfer (blind or attended) the call to another internal phone. FS just hangs up. Lookings at the FS CLI, it seems as though FS tries to process the transfer in the PUBLIC context rather than my default.xml (which is modified as per the above link). I've tried numerous things, but can't seem to get this working. I would really appreciate your thoughts. Kind regards, Chirag

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Freeswitch Week in Review(Master Branch) September 7th- 13th
by Kathleen King 23 Sep '14

23 Sep '14

Hello, again. This week in the FreeSWITCH master branch we had 83 commits. There was a lot of new work this week with quite a few updates to the packaging of RPMs for 1.0.50 sound files, the addition of French Canadian RPMs, updates to the sounds to streamline the sound release process, and the addition of a new audio analysis application module mod_bert! The following bugs were squashed: 9215c93 FS-6785 - pass and use sample rate in switch_core_session_get_payload_code() Jira: https://jira.freeswitch.org/browse/FS-6785 a73583b FS-6806 Fix for invalid Cseq value greater than 2^31 when sending NOTIFY in mod_sofia Jira: https://jira.freeswitch.org/browse/FS-6806 590382f Fix URL parsing in core 574d19e Fix behavior of transfer when target extension falls through to voicemail - keep bridge from dropping out during that operation in mod_skinny 66302ca FS-6811 Fixed Debian Wheezy build failure with mod_java Jira: https://jira.freeswitch.org/browse/FS-6811 908dd26 FS-6815 #comment force 1600hz for native g722 Jira: https://jira.freeswitch.org/browse/FS-6815 e2b1ee2 Fix libpq location detection in event_handlers ce5d211 FS-6761 Fix to properly set CURLOPT_NOSIGNAL in mod_httapi and mod_xml_curl Jira: https://jira.freeswitch.org/browse/FS-6761 6953881 Fix valet parking sample in vanilla configs 622e0e1 FS-6775 FS-6783 Check for null hash on increment of mod_hash limit efe0ebd Fix for multicast in mod_verto Jira: https://jira.freeswitch.org/browse/FS-6818 50c62cd Added missing host to network conversion highest_sequence_number_received New features that were added: 2eed5ec Clean the uri to protect us from uri vulnerability attack like request /../../../../etc/passwd 3c7e7c7 Added more logging during transfer operations in mod_skinny fa3a1a0 Added mod_bert module 66a9e12 Improvements to mod_bert a60876b Use a triangle wave in mod_bert d3b214a Improvements to mod_bert a84ce1d Improvements to mod_bert ef69451 Improvements to mod_bert 3248e06 Add timer to mod_bert to avoid relying on remote RTP for timing cf2fd15 Add BERT events for timeout and out of sync in mod_bert 58e4774 Only use a timer if bert_timer_name is specified, otherwise time based on input 557a4a8 Treat CNG as a silent frame in mod_bert 25c6276 Added cng counter to the bert test app 89e8e14 Improvements to mod_bert c3d65e9 Merge in Moises Silva's mod_bert 8973ffc Improvements to error handling/detection in mod_skinny 37d7fb7 Calculate jitter percentage in jitterbuffer to factor into conditions for reducing the size when in adaptave mode d06ac75 Clarify tls ciphers and update num to allow for ipv6 and tls options in vanilla configs d1e00c2 Updated the spec file for better dependency tracking and added verto.conf.xml 80542e2 Patch to exclude MIME header and MIME types if the file argument is set to NULL and the body argument doesn't include "content-type" Jira: https://jira.freeswitch.org/browse/FS-5142 7144b25 Obey sip_copy_custom_headers on bye in mod_sofia 5bd3547 FS-6817 Add var to suppress `Privacy: none` header in mod_sofia. Thanks Stéphane Alnet <stephane(a)shimaore.net> 688f225 Update ru/RU sounds for 1.0.50 0b7863a FS-6623 Implement RTCP report generation Jira: https://jira.freeswitch.org/browse/FS-6623 3afc086 Updating the Russian RPMs for 1.0.50 sound files and added French Canadian RPMs 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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Freeswitch Week in Review(Master Branch) September 14th- 20th
by Kathleen King 23 Sep '14

23 Sep '14

Hello, again. This week in the FreeSWITCH master branch we had 31 commits. We had a variety of work done, but this week I would like to highlight the addition of documentation. The documentation changes include: guidelines for creating commits (http://goo.gl/PiyrsM) the dev team's deviations from commit guidelines (http://goo.gl/j8LqZd) fixes to straggling references to old commit hooks (http://goo.gl/6WK2hJ) the policy against merges in a feature branch (http://goo.gl/wwajWo) references to the contributor guidelines on git setup (http://goo.gl/yYc1Jn) a pre-formatted text for example commit messages (http://goo.gl/zyQGXs) and an update to the contributor guidelines for pull requests (http://goo.gl/op6B10) Check out the changes and see the guidelines for submitting pull requests, reporting bugs, and formatting commits to ensure your submissions will meet the requirements. The following bugs were squashed: f262dbc FS-6821 Fix interface name for logs emitted from mod_gsmopen.cpp during startup Jira: https://jira.freeswitch.org/browse/FS-6821 f780077 Fix for avoiding reset when the video floor is locked in mod_conference 1f5bb34 Avoid truncation of non-null-terminated strings in protocol in mod_skinny New features that were added: b738775 FS-6623 Add RTCP report generation Jira: https://jira.freeswitch.org/browse/FS-6623 f924684 FS-6623 #resolve Fix init and logging for rtcp Jira: https://jira.freeswitch.org/browse/FS-6623 04269fd Added additional logging in mod_skinny b2917e0 Improve ssl errors in mod_verto 295fcce Add buffer_seconds param to shout filehandles to override the original default of 1 and remove previous code to attempt to buffer several seconds of audio in the open routine. 1379a2c Document some guidelines for creating commits bb2427e Add state change to allow call to stay up when network conditions change in mod_verto 718e3c6 Document our deviations from commit guidelines 527016c Fix straggling reference to old commit hooks 58110da Document policy against merges in a feature branch 2ef2146 Refer to our contributor guidelines on git setup c166afb Use preformatted text for example commit message b7b01d0 Make example commit look like git log output 393c7c3 Update our contributor guidelines for Prs 4dc7f92 Update document to HTTPS URLs In terms of stability these were the use cases that were fixed: 372455c Added an if statement to prevent a crash caused by a NULL in switch_core_media Jira: https://jira.freeswitch.org/browse/FS-6829 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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mod_skypopen fresh install problem
by ghernandez 23 Sep '14

23 Sep '14

Hi! I recently installed new mod_skypopen in a fresh freeswitch installation using instructions (https://confluence.freeswitch.org/display/FREESWITCH/mod_skypopen) step by step. skype for linux starts and logged into account but mod_skypopen cannot link to it. Centos 6.5 freeswitch.log error: 2014-09-23 16:15:11.526625 [DEBUG] skypopen_protocol.c:1767 [1.5.14b+git~20140923T160125Z~e8d6866899~32bit ] [DEBUG_SKYPE 1767 ][skype101 ][IDLE,IDLE] SENDING: |||PROTOCOL 999|||| 2014-09-23 16:15:11.526625 [ERR] mod_skypopen.c:1838 [1.5.14b+git~20140923T160125Z~e8d6866899~32bit ] [ERRORA 1838 ][skype101 ][IDLE,IDLE] The Skype client to which we are connected FAILED to gave us CURRENTUSERHAN DLE=camundanet, interface_id=1 FAILED to start. No Skype client logged in as 'camundanet' has been found. Please (re)launch a Skype client logged in as 'camundanet'. Skypopen exiting now 2014-09-23 16:15:11.546625 [DEBUG] mod_skypopen.c:1506 [1.5.14b+git~20140923T160125Z~e8d6866899~32bit ] [DEBUG_SKYPE 1506 ][skype101 ][IDLE,IDLE] EXITING 2014-09-23 16:15:11.626626 [DEBUG] skypopen_protocol.c:2122 [1.5.14b+git~20140923T160125Z~e8d6866899~32bit ] [DEBUG_SKYPE 2122 ][skype101 ][IDLE,IDLE] EXITING 2014-09-23 16:15:11.626626 [DEBUG] mod_skypopen.c:3328 [1.5.14b+git~20140923T160125Z~e8d6866899~32bit ] [DEBUG_SKYPE 3328 ][none ][N/A,N/A] BEGIN REMOVE 2014-09-23 16:15:11.626626 [DEBUG] mod_skypopen.c:3357 [1.5.14b+git~20140923T160125Z~e8d6866899~32bit ] [DEBUG_SKYPE 3357 ][none ][N/A,N/A] EXIT REMOVE but in process table skype exists and this user appears online when clients start (discarding auth or network problems): 973 pts/0 S 0:00 /usr/bin/Xvfb :101 -ac -nolisten tcp -screen 0 640x480x8 979 pts/0 Sl 0:16 /usr/local/freeswitch/skypopen/skype-clients-symlinks-dir/skype101 --dbpath=/usr/local/freeswitch/skypopen/skype-clients-configuration-dir/skype101 --pipelogin 1016 pts/0 S 0:00 /usr/bin/Xvfb :102 -ac -nolisten tcp -screen 0 640x480x8 1022 pts/0 Sl 0:17 /usr/local/freeswitch/skypopen/skype-clients-symlinks-dir/skype102 --dbpath=/usr/local/freeswitch/skypopen/skype-clients-configuration-dir/skype102 --pipelogin 1061 pts/0 S 0:00 /usr/bin/Xvfb :103 -ac -nolisten tcp -screen 0 640x480x8 1070 pts/0 Sl 0:16 /usr/local/freeswitch/skypopen/skype-clients-symlinks-dir/skype103 --dbpath=/usr/local/freeswitch/skypopen/skype-clients-configuration-dir/skype103 --pipelogin 1106 pts/0 S 0:00 /usr/bin/Xvfb :104 -ac -nolisten tcp -screen 0 640x480x8 1116 pts/0 Sl 0:13 /usr/local/freeswitch/skypopen/skype-clients-symlinks-dir/skype104 --dbpath=/usr/local/freeswitch/skypopen/skype-clients-configuration-dir/skype104 --pipelogin 1153 pts/0 S 0:00 /usr/bin/Xvfb :105 -ac -nolisten tcp -screen 0 640x480x8 1162 pts/0 Sl 0:12 /usr/local/freeswitch/skypopen/skype-clients-symlinks-dir/skype105 --dbpath=/usr/local/freeswitch/skypopen/skype-clients-configuration-dir/skype105 --pipelogin 1200 pts/0 S 0:00 /usr/bin/Xvfb :106 -ac -nolisten tcp -screen 0 640x480x8 1209 pts/0 Sl 0:14 /usr/local/freeswitch/skypopen/skype-clients-symlinks-dir/skype106 --dbpath=/usr/local/freeswitch/skypopen/skype-clients-configuration-dir/skype106 --pipelogin 1249 pts/0 S 0:00 /usr/bin/Xvfb :107 -ac -nolisten tcp -screen 0 640x480x8 1270 pts/0 Sl 0:12 /usr/local/freeswitch/skypopen/skype-clients-symlinks-dir/skype107 --dbpath=/usr/local/freeswitch/skypopen/skype-clients-configuration-dir/skype107 --pipelogin 1311 pts/0 S 0:00 /usr/bin/Xvfb :108 -ac -nolisten tcp -screen 0 640x480x8 1321 pts/0 Sl 0:12 /usr/local/freeswitch/skypopen/skype-clients-symlinks-dir/skype108 --dbpath=/usr/local/freeswitch/skypopen/skype-clients-configuration-dir/skype108 --pipelogin 1359 pts/0 S 0:00 /usr/bin/Xvfb :109 -ac -nolisten tcp -screen 0 640x480x8 1368 pts/0 Sl 0:12 /usr/local/freeswitch/skypopen/skype-clients-symlinks-dir/skype109 --dbpath=/usr/local/freeswitch/skypopen/skype-clients-configuration-dir/skype109 --pipelogin 1408 pts/0 S 0:00 /usr/bin/Xvfb :110 -ac -nolisten tcp -screen 0 640x480x8 1417 pts/0 Sl 0:15 /usr/local/freeswitch/skypopen/skype-clients-symlinks-dir/skype110 --dbpath=/usr/local/freeswitch/skypopen/skype-clients-configuration-dir/skype110 --pipelogin Someone can help? Thanks a lot! -- Gustavo Hernández Baratta CAMUNDANET INC. http://www.camundanet.com

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mod_python error everytime session is invoked for the first time right after a freeswitch start
by Nuno Reis 23 Sep '14

23 Sep '14

Hi guys. I've reported this before and i can tell that it is still happening in the latest v1.4. How reproduce: 1. just create a dialplan that calls a python script 2. dial that dialplan entry Example code (works with anything that invokes *session*): #!/usr/bin/env python # -*- coding: utf-8 -*- from __future__ import absolute_import # http://docs.python.org/whatsnew/pep-328.html def handler(session, args): freeswitch.consoleLog("notice", "Testing123...") freeswitch.consoleLog("notice", "Dumping current CDR INFO") freeswitch.consoleLog("info", str(session.getXMLCDR())) freeswitch.consoleLog("notice", "The END!") mod_pyhton crashes everytime the script is called right after a freeswitch start (the very first time session is invoked inside a mod_python python script). After the first time (error), everything starts to work just fine. This doesn't happen in e6b8b3e4f2deac73547bc9b9a2cb3e1d3147d101 from v1.2.stable Should i open a jira on this? -- *Nuno Miguel Reis* | *Unified Communication** Systems* M. +351 913907481 | nreis(a)wavecom.pt WAVECOM-Soluções Rádio, S.A. Cacia Park | Rua do Progresso, Lote 15 3800-639 AVEIRO | Portugal T. +351 309 700 225 | F. +351 234 919 191 *GPS <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…> | www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>* [image: Description: Description: WavecomSignature] <http://www.wavecom.pt/pt/wavecom/premios.php> [image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php>

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