January 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] How to Bridge To Local Extensions?
by George F. Phelps 22 Jan '15

22 Jan '15

Can someone help me with my question? Thanks, George From: George F. Phelps [mailto:GeorgePhelps@gfphelps.com] Sent: Saturday, January 10, 2015 12:02 PM To: freeswitch-users(a)lists.freeswitch.org Subject: How to Bridge To Local Extensions? The "transfer" statement, shown below, works (in my inbound dialplan): <action application="transfer" data="1001 XML default"/> What is the correct syntax for using "bridge" instead of "transfer"? The following statement does not work for me: <action application="bridge" data="sofia/internal/1001"/> My extensions are effectively default values and in the default directory location. For example: more /usr/local/freeswitch/conf/directory/default/1001.xml <include> <user id="1001"> <params> <param name="password" value="XXXX"/> <param name="vm-password" value="1001"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="1001"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 1001"/> <variable name="effective_caller_id_number" value="1001"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="techsupport"/> </variables> </user> </include> My goal is to configure simultaneous ringing for multiple extensions: <action application="bridge" data="{ignore_early_media=true}sofia/internal/1001,sofia/internal/1002"/> Thanks, George

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spandsp_start_dtmf detects # key many times
by Nagalenoj H. 22 Jan '15

22 Jan '15

Hi, When I use spandsp_start_dtmf along with play_and_get_digits, '#' is getting detected many times, though it is pressed once. Other DTMFs are getting detected properly, whereas the '#' is getting detected 2 or 3 times. I have checked the SIP phone's settings, I do not find anything wrong there. What could be the reason? Is this because of some SIP phone setting? Here is the freeswitch log, 2015-01-19 09:33:39.671681 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [1] 2015-01-19 09:33:39.671681 [DEBUG] switch_channel.c:488 RECV DTMF 1:2000 2015-01-19 09:33:39.851686 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [1], duration = 191 ms 2015-01-19 09:33:40.071688 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [4] 2015-01-19 09:33:40.071688 [DEBUG] switch_channel.c:488 RECV DTMF 4:2000 2015-01-19 09:33:40.251687 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [4], duration = 153 ms 2015-01-19 09:33:40.431699 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [7] 2015-01-19 09:33:40.431699 [DEBUG] switch_channel.c:488 RECV DTMF 7:2000 2015-01-19 09:33:40.631680 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [7], duration = 204 ms 2015-01-19 09:33:40.851688 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [4] 2015-01-19 09:33:40.851688 [DEBUG] switch_channel.c:488 RECV DTMF 4:2000 2015-01-19 09:33:40.971695 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [4], duration = 127 ms 2015-01-19 09:33:41.151691 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/sip:278@192.168.2.86:5060 [BREAK] 2015-01-19 09:33:41.171701 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [7] 2015-01-19 09:33:41.171701 [DEBUG] switch_channel.c:488 RECV DTMF 7:2000 2015-01-19 09:33:41.351680 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [7], duration = 191 ms 2015-01-19 09:33:41.371686 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16@8000hz 1 channels 20ms 2015-01-19 09:33:41.691686 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [#] 2015-01-19 09:33:41.691686 [DEBUG] switch_channel.c:488 RECV DTMF #:2000 EXECUTE sofia/internal/392(a)3427.vbiz.mundio.com flush_dtmf() 2015-01-19 09:33:41.691686 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16@8000hz 1 channels 30ms 2015-01-19 09:33:41.711684 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [#], duration = 38 ms 2015-01-19 09:33:41.771683 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [#] 2015-01-19 09:33:41.771683 [DEBUG] switch_channel.c:488 RECV DTMF #:2000 2015-01-19 09:33:41.771683 [DEBUG] switch_ivr_play_say.c:1747 done playing file file_string:///root/united_fone/voice_files/en//CR_enter_conf_pin.wav 2015-01-19 09:33:41.771683 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16@8000hz 1 channels 30ms 2015-01-19 09:33:41.811686 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [#], duration = 25 ms 2015-01-19 09:33:41.871691 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [#] 2015-01-19 09:33:41.871691 [DEBUG] switch_channel.c:488 RECV DTMF #:2000 2015-01-19 09:33:41.951681 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [#], duration = 76 ms -- Regards, Nagalenoj H.

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Install FusionPBX in minutes on Centos 6 & 7
by Luis Daniel Lucio Quiroz 22 Jan '15

22 Jan '15

Hello everybody Just to share you this: https://okay.com.mx/en/entrepreneurs/install-fusionpbx-in-a-moment.html With this alternate repo you will be able to install FusionPBX in minutes, just with a yum command. Please read the link for all details. Luis Daniel Lucio Quiroz CISSP, CISM, CISA Linux, VoIP and much more fun www.okay.com.mx Need LCR? Check out LCR for FusionPBX with FreeSWITCH Need Billing? Check out Billing for FusionPBX with FreeSWITCH

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media bypass on freeswitch
by Achintha 21 Jan '15

21 Jan '15

I configured freeswitch (FreeSWITCH Version 1.5.15b+git~20141117T202539Z~424df19083~64bit) on Ubuntu 12.04. It works properly. I’m using xml_curl module to generate the dialplan. Then I set the media bypass parameter on freeswitch dialplan. But the freeswitch crashed when following dialplan was given. <document type="freeswitch/xml"> <section name="dialplan" description="RE Dial Plan For FreeSwitch"> <context name="130_131_SANASA"> <extension name="2022"> <condition field="destination_number" expression="[^\s]*"> <action application="set" data="ringback=${us-ring}" /> <action application="export" data="OperationType=User" /> <action application="export" data="CustomCompanyStr=131_130" /> <action application="export" data="FromGuUserId=14111215533245504" /> <action application="export" data="ToGuUserId=14111215495317100" /> <action application="export" data="FromUser=sanasains2023_sanasa_lk" /> <action application="export" data="FromNumber=2023" /> <action application="export" data="nolocal:ToNumber=2022" /> <action application="export" data="nolocal:ToUser=sanasains2022_sanasa_lk" /> <action application="set" data="continue_on_fail=true" /> <action application="set" data="hangup_after_bridge=true" /> <action application="set" data="ignore_early_media=false" /> <action application="set" data="bypass_media=true" /> <action application="bind_meta_app" data="3 ab s execute_extension::att_xfer XML PBXFeatures" /> <action application="bind_meta_app" data="4 ab s execute_extension::att_xfer_group XML PBXFeatures" /> <action application="bind_meta_app" data="6 ab s execute_extension::att_xfer_outbound XML PBXFeatures" /> <action application="bind_meta_app" data="5 ab s execute_extension::att_xfer_conference XML PBXFeatures" /> <action application="bridge" data="[leg_timeout=60,origination_caller_id_name=sanasains2023_sanasa_lk,origination_caller_id_number=2023]user/ 2022(a)192.168.51.X <2022(a)192.168.51.86>X,pickup/3003" /> <action application="hangup" /> </condition> </extension> </context> </section> </document In here I ‘m using attendant transfer feature. Here I have attached the print screen of freeswitch console just after it crashed. Kindly advise me to solve this problem. -- Thanking You.. Achintha

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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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