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Freeswitch on VPS
by Zoltán Szabó 20 Jan '15

20 Jan '15

Hi, Does anyone have experience running Freeswitch at OVH? I have a VPS Classic 1 but the audio was really bad there. It was faltered. (tested with multiple clients from multiple locations) This is an OpenVZ virtualization if this counts. I tested it on an Alvotech VPS too, with more ram, like 2GB, CPU was also a little bit more, and it was a vServer virtualization, it worked perfectly. If you run your FS on a VPS without any problem, can you please recommend me some providers who has servers in the EU? I don't expect high call volume, just like 5-10 concurrent calls on 1-2 SIP trunks which is basically nothing. Many thanks, Zoltan

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Sound files from cache
by John Nash 20 Jan '15

20 Jan '15

I am using "playback" to play native sound file (<action application="playback" data="/tmp/hello"/>) My question is when it plays the file does it read file every time from the disk? or there is some way to "Load at the startup" and play directly from memory. I also saw the "mod_http_cache" but not sure if I should use it (I am unable to understand in what scenario it will be logical to use it)

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Performance test using Sipp
by Ahmed habiba 20 Jan '15

20 Jan '15

Dears, I'm looking how can I use sipp for freeswitch performance testing in term of number of concurrent calls. Your kind help will be appreciated. Thanks, Ahmed Habiba.

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Sending call to outbound proxy
by John Nash 20 Jan '15

20 Jan '15

I am trying to test a case where I am receiving call from opensips to freeswitch and need to route back to opensips again without changing Request URI. How can I do it?

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looking to pay for help in configuration
by Shai Perelman 19 Jan '15

19 Jan '15

Hi, Im new to freeswitch and the list, so hi everybody. I,m learning the subject of fusionpbx/freeswitch and have a server up and running, got a few problems. and need someone to assist me on a hourly rate. is there any one interested? thanks Shai

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Mod-Sofia ability to dial Alphanumeric characters in invite?
by Magnus Kelly 19 Jan '15

19 Jan '15

Hello Guys, Is it possible to use FS with alphanumeric characters in the SIP invite? As in I would like to send an invite addressed to +C27344xxxxxxxxxx as one of our upstream platforms uses a C prefix. What I observe with default is " [ERR] mod_sofia.c:4431 Invalid URL " If possible to support what would need to be changed? I am assuming that this is outside the SIP standards, but confirmation would equally be useful. Thanks Magnus

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Custom Columns into MySQL Channels table
by Jack Cortez 19 Jan '15

19 Jan '15

Hi, I would like to add a column to channel table (created by FS automatically using ODBC) and populate this new columns with my own custom value. Is this possible? In particular, as I'm creating a simple PHP page to show active calls, what I would like to add is a column with the gateway_name to identify the Outbound Provider Name. So, in my PHP page I can show where the call is going out. And also, I found that the application_data columns is showing the sofia dialstring (for ie. sofia/gateway/my_gateway/44798797978) but this columns is overwritten by last action I can't rely on this field. I'm looking for days into the wiki but I didn't found anything interesting. Thank you so much!

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Function - drop_dtmf
by Magnus Kelly 19 Jan '15

19 Jan '15

Hello Guys, Couple of questions on the dtmf masking function that was created/extended recently - is it possible to use half way through a call ? - in terms of invoking the DTMF masking by initiating dtmf as in the below dial plan, and associated features.xml entry - would the call be able to start as normal and then on demand mask dumf? <extension name="dtmftest"> <condition field="destination_number" expression=""^(\d+)$""> <action application="export" data="dialed_extension=$1"/> <action application="bind_meta_app" data="1 a s execute_extension::stopdtmf XML features"/> <action application="set" data="call_timeout=30"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="sofia/$1@10. <1(a)10.113.0.98>x.x.x"/> <action application="answer"/> <action application="sleep" data="1000"/> </condition> </extension> and an entry in features.xml as in - <include> <context name="features">  <extension name="stopdtmf"> <condition field="destination_number" expression="^stopdtmf$"> <action application="export" data="nolocal:drop_dtmf=true"/> <action application="export" data="nolocal:drop_dtmf_masking_digits=d"/> </condition> </extension> ------------snip I guess then I would need an alternative bind digit to turn the feature feature off again? Is this possible or is this only selectable at bridge time? Any thoughts? Thanks Magnus

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FreeSWITCH Week in Review (Master Branch) January 11th-16th
by Ken Rice 19 Jan '15

19 Jan '15

New Post on freeswitch.org from kathleen check it out at http://ift.tt/1Gl1ooB FreeSWITCH Week in Review (Master Branch) January 11th-16th Hello, again. This week in the FreeSWITCH master branch we had 15 commits. In an effort to improve the execution of the Week in Review, starting this week I will be using a Saturday-Friday time-line. This will give me more time to review the commits and get answers to questions. That being said, the features for this week are: completing the Linux portion of the update to flite-2.0.0-release and added param add-variables-to-events which will add channel variables to <offer>, <ringing>, <answered>, and <end> if set to true, while the default is false, in mod_rayo. New features that were added: 38d25f1 FS-7149 Linux portion of update to use flite-2.0.0-release complete [Jira: http://ift.tt/1Gl1nRp] 85fbddc FS-7150 Added param add-variables-to-events which will add channel variables to <offer>, <ringing>, <answered>, and <end> if set to true, while the default is false, in mod_rayo [Jira: http://ift.tt/1Gl1nRr] In terms of stability these were the use cases that were fixed: 59a9669 Prevent crash when calling mediaStats JSON function in certain circumstances in mod_conference Improvements to packaging: adb0de9 Update debian utils for flite 2.0.0 #resolve 5c5b9fc FS-7151 If available, use libjpeg-turbo in Debian build dependencies The following bugs were squashed: 3e6ffbc FS-7144 Cleaning up NAT’d RTP where ICE is involved [Jira: http://ift.tt/1CI1nVI] 403d32c FS-7148 Fixed a deadlock [Jira: http://ift.tt/1Gl1ooD] 378d350 FS-7144 More cleaning up NAT’d RTP where ICE is involved [Jira: http://ift.tt/1pgbVcN]

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Startup script is not respecting real time priority
by Babak Yakhchali 19 Jan '15

19 Jan '15

Hi I'm using startup script supplied in repo for centos 6.5 and setting the -rp like this: PROG_NAME=freeswitch PID_FILE=/usr/local/freeswitch/run/freeswitch.pid FS_USER=freeswitch FS_FILE=/usr/local/freeswitch/bin/freeswitch FS_HOME=/usr/local/freeswitch/run LOCK_FILE=/var/lock/subsys/freeswitch *FREESWITCH_ARGS="-nc -rp"* RETVAL=0 but freeswitch is always running with default priority as seen in top: 16263 freeswit 39 19 875m 30m 7416 S 1.0 13.0 0:17.44 freeswitch and in "ps aux|grep freeswitch" I see: 500 16263 0.8 12.9 897004 31120 ? SNl 15:38 0:18 /usr/local/freeswitch/bin/freeswitch -nc -rp so it seems freeswitch is starting with proper args but priority is not set. If I run freeswitch as root in bash it is working will: runing ./freeswitch -nc -rp results in: 16550 root -2 -10 865m 28m 6988 S 6.3 12.2 0:00.52 freeswitch Should I change anything else in the script? or add any special permission to user "freeswitch"? Thanks

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