September 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_odbc_query / mod_db
by Mujahid Ali 03 Sep '14

03 Sep '14

is mod_db module alternative of mod_odbc_query ? i want to make a condition in dialplan on behalf of database data and update it. is it possible with mod_db ?

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load module error switch_loadable_module.c
by Mujahid Ali 03 Sep '14

03 Sep '14

Hi when i install freeswitch by yum like this rpm -Uvh http://files.freeswitch.org/freeswitch-release-1-0.noarch.rpm then yum install --nogpgcheck freeswitch-config-vanilla everything looks fine and ok but when i go to freeswitch cli fs_cli and load any module like load module mod_db i got an error like 2014-09-02 08:52:12.950995 [CRIT] switch_loadable_module.c:1448 Error Loading module /usr/lib64/freeswitch/mod/module mod_db.so **/usr/lib64/freeswitch/mod/module mod_db.so: cannot open shared object file: No such file or directory** this error is coming on every module which i want to load. i don't know how to solve it.

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Double NAT LAN network address
by Nandy Dagondon 03 Sep '14

03 Sep '14

Hi Everybody, One of my client has 2 separate offices using the same LAN network address 192.168.1.0. With FS on one site and some phones on the other site, the double nat setup looks Freeswitch Phones 192.168.1.0 <NAT>Static Public IP<Internet>Dynamic Public IP<NAT>192.168.1.0 If feel that I should recommend changing one network address to eliminate ambiguity - which side is this 192.168.1.0? Am I right? Tks, Nandy

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[ERR] switch_loadable_module.c:769 Invalid chat application interface [log]!
by Snabel Kabiya 03 Sep '14

03 Sep '14

Hi, i'm trying to use mod_sms. i did the following steps 1 - used the instructions here to compile the module http://wiki.freeswitch.org/wiki/Mod_sms#Args https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_chat created two users on both servers x.x.x.1, x.x.x.2 <domain name="x.x.x.1"> <user id="2222"> <params> <param name="password" value="2222"/> </params> </user> </domain> <domain name="x.x.x.1"> <user id="1111"> <params> <param name="password" value="1111"/> </params> </user> </domain> created this chatplan: public.xml: <?xml version="1.0" encoding="utf-8"?> <include> <context name="public"> <extension name="public"> <condition field="to" expression="^(.*)$"> <action application="log" data="INFO to_chat remote"/> <action application="reply" data="Hello, you said: ${_body}"/> </condition> </extension> </context> </include> but now when i send this from server x.x.x.2 freeswitch@internal> chat sip|2222@x.x.x.2|external/sip:1111@x.x.x.1|hi from freeswitch snabel is here Sent i get the following on server x.x.x.1 2014-09-03 11:18:18.780102 [INFO] mod_sms.c:336 Processing text message 2222->1111 in context public Chatplan: 1111 parsing [public->public] continue=false Chatplan: 1111(a)x.x.x.1 Regex (PASS) [public] to(1111(a)x.x.x.1) =~ /^(.*)$/ break=on-false Chatplan: 1111(a)x.x.x.1 Action log(INFO to_chat remote) Chatplan: 1111(a)x.x.x.1 Action reply(Hello, you said: ${_body}) *2014-09-03 11:18:18.780102 [ERR] switch_loadable_module.c:769 Invalid chat application interface [log]!* what does it mean? what did i miss? Thanks, Snabel

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G729 License Server
by Colin Mason 03 Sep '14

03 Sep '14

Hello, Is the latest "fs-201405280931-installer" compatible with FreeSWITCH 1.2? If so, is it possible to have multiple FreeSWITCH processes on a single hardware box use a single license server? Thanks, Colin

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fax baud rate 9600
by Marlok Tamas 03 Sep '14

03 Sep '14

Hi! I found this thread: http://lists.freeswitch.org/pipermail/freeswitch-users/2013-March/094032.ht… and I have the same problem (all faxes with transfer rate: 14400 are dropped) It seems that nobody ha an answer back then for the question (is there any way of forcing freeswitch to use transfer rate: 9600) The result: 2014-08-29 15:21:39.187130 [DEBUG] mod_spandsp_fax.c:512 ============================================================================== 2014-08-29 15:21:39.187130 [DEBUG] mod_spandsp_fax.c:524 Fax processing not successful - result (48) Disconnected after permitted retries. 2014-08-29 15:21:39.187130 [DEBUG] mod_spandsp_fax.c:529 Remote station id: 2014-08-29 15:21:39.187130 [DEBUG] mod_spandsp_fax.c:530 Local station id: <local_number> 2014-08-29 15:21:39.187130 [DEBUG] mod_spandsp_fax.c:531 Pages transferred: 0 2014-08-29 15:21:39.187130 [DEBUG] mod_spandsp_fax.c:533 Total fax pages: 0 2014-08-29 15:21:39.187130 [DEBUG] mod_spandsp_fax.c:534 Image resolution: 0x0 2014-08-29 15:21:39.187130 [DEBUG] mod_spandsp_fax.c:535 Transfer Rate: 14400 2014-08-29 15:21:39.187130 [DEBUG] mod_spandsp_fax.c:537 ECM status off 2014-08-29 15:21:39.187130 [DEBUG] mod_spandsp_fax.c:538 remote country: 2014-08-29 15:21:39.187130 [DEBUG] mod_spandsp_fax.c:539 remote vendor: 2014-08-29 15:21:39.187130 [DEBUG] mod_spandsp_fax.c:540 remote model: 2014-08-29 15:21:39.187130 [DEBUG] mod_spandsp_fax.c:542 ============================================================================== Thanks in advance! Cheers, tmarlok

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FreeSWITCH to Implement Video Transcoding - Sponsors Needed
by Anthony Minessale 02 Sep '14

02 Sep '14

The FreeSWITCH project intends to devote attention in the upcoming months to work towards improving video functionality. This includes working to get fully function video transcoding working inside FreeSWITCH. Currently FreeSWITCH video is limited to pass-through and with the new features available with WebRTC we have decided to focus on trying to improve the overall video capabilities in FreeSWITCH. We currently have an issue open with chrome that helps underscore the reason we need to implement native video codec functionality especially in Conferencing. https://code.google.com/p/chromium/issues/detail?id=263867 We anticipate that this effort will take as long as 6 or more months to complete and have a large cost in developer resources. We already have some sponsors and volunteers helping with the effort but we can use all the help we can get. If you are interested in helping out either by financially sponsoring the effort or helping to test and bug report, please contact us at consulting(a)freeswitch.org and let us know. -- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900

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Freeswitch Week in Review(Master Branch) August 24th-August 30th
by Kathleen King 02 Sep '14

02 Sep '14

Hello, again. This week in the FreeSWITCH master branch we had 20 commits. We had two new features with improvements to recording in mod_conference and added support for the dtmf format dtmf-relay in mod_sofia, and a bug fix in mod_rayo related to nested calls. The following bugs were squashed: 323a3d6 FS-6764 #resolve Avoid output of junk text by fs_cli. Thanks-to: Nathan Neulinger <nneul(a)neulinger.org> 2db039b FS-6720 - fix issue with SLA - had to do with answering in wrong state in mod_skinny Jira: https://jira.freeswitch.org/browse/FS-6720 dc88617 Lower log level of limit increases and releases 89b4b0c Fix nested join to calls and conferences in mod_rayo New features that were added: 0f1494a Improvements to conference recording in mod_conference 1dc4406 Added support for the dtmf format dtmf-relay in mod_sofia Jira: https://jira.freeswitch.org/browse/FS-6770 Improvements in cross platform build supports: 1c058e0 Check gcc support for -Wno-unused-result 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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Sip phone lose communication appears registered
by Fernando Flórez 02 Sep '14

02 Sep '14

Hello, Just setup an FS test server on amazon ec2. Everything works flawlessly but after a while sip phone arent reacheable even as they seem registered on FS and are able to do outbound calling. Any idea of what could be happening? If i restart the sip phones every starts working again for a litrle while. Thanks, Sent from my iPhone

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Freeswitch-1.4.7 source rpm
by Mujahid Ali 02 Sep '14

02 Sep '14

hi everyone, where i can get freeswitch-1.4.7-src.rpm at the following link http://files.freeswitch.org/ there is only taz.bz2 file freeswitch-1.4.7.tar.bz2 <http://files.freeswitch.org/freeswitch-1.4.7.tar.bz2> but not source rpm. is there is any 1.4.7 src rpm ? if yes than please give me the link

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