February 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_xml_curl
by Londre 05 Feb '16

05 Feb '16

Hi, I have read the wiki several times and followed all the steps to the best of my knowledge. I have gotten mod_xml_curl build and making requests. I set xml_curl debug_on and I can see the request that freeswitch sends and I can take a look at the file it generates in the /tmp/ directory. However the file doesn't seem to be writing anything. I have my web service set up to hand a POST request but it looks like freeswitch is sending GET requests even though I set the method to POST in xml_curl.xml. Any ideal would could be going wrong? System Info: Debian Jessie Freeswitch 1.7 built from source. Xml_curl.xml: 1 <configuration name="xml_curl.conf" description="cURL XML Gateway"> 2 <bindings> 3 <binding name="directory"> 4 <param name="gateway-url" value="http://{domain}/api/directory" bindings="directory|dialplan"/> 5 <param name="method" value="POST"/> 6 <param name"gateway-credentials" value"user:pass"/> 7 </binding> 8 </bindings> 9 </configuration> It also doesn't seem to be sending the variables for the gateway-credentials. Its seems weird that its sending a GET request when the default should be a POST. Also it is a little confusing on the wiki some things show a query string and other show header variables. I'd prefer to use POST. Please offer some insight into what could be going wrong. Londre Blocker Developer iMerchant Systems Inc. T: 844.727.1998

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X-Auth-IP Variable?
by Tim King 05 Feb '16

05 Feb '16

I am using proxy authentication in my setup and it is working. To do this I have created an acl *autoload_configs/acl.conf.xml* <list name="opensips" default="deny"> <node type="allow" cidr="1.2.3.4/32"/> <node type="allow" cidr="1.2.3.5/32"/> <node type="allow" cidr="1.2.3.6/32"/> </list> *sip_profiles/external.xml* <param name="auth-calls" value="true"/> <param name="apply-proxy-acl" value="1.2.3.2/32"/> <param name="apply-inbound-acl" value="opensips"/> This is all working as desired. The problem is prior to adding the opensips I was using the network_addr variable in my dialplan. <extension name="tollfree_to_providerA"> <condition field="${acl(${network_addr} opensips)}" expression="true" break="on-false"/> <condition field="destination_number" expression="^(\+?1)?(8(00|44|55|66|77|88)[2-9]\d{6})$"> <action application = "set" data="dtmf_type=rfc2833"/> <action application="set" data="accountcode=customer1123"/> <action application="set" data="continue_on_fail=false"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="proxy_media=true"/> <action application="bridge" data="sofia/external/$2@8.7.6.5:5060"/> </condition> </extension> This of course no longer works because network_addr is always the address of my proxy server. How can I get the address from the X-authip into the dialplan?

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fs_cli under Windows and cygwin, no prompt
by Peter Steinbach 05 Feb '16

05 Feb '16

I have inherited a Freeswitch installation on a Windows server, which I remotely manage with cygwin and ssh. This worked more or less nicely so far. But today I cannot manage Freeswitch via ssh with fs_cli anymore. I restarted the hardware, but no change. I can run fs_cli.exe, and I see the Freeswitch messages, but I do __not__ get a prompt. When I try to enter characters, nothing happens. On the machine itself, when I open a console and start cygwin, fs_cli.exe works as expected with prompt. But not via ssh. But fs_cli.exe -x "command" works via ssh, so login to Freeswitch should be fine in general. What I understand is, that fs_cli connects to Freeswitch via port 8021. So when seeing FS messages, I expect that the connection is already fine, right? I also tried to connect via telnet to port 8021. This works in the local cygwin shell (get an auth request), but not via remote ssh shell (telnet terminates without message). I am puzzled. Anybody has a clue, why I do not have a prompt and where to search? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

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Time for the Annual FreeSWITCH Developers Summit
by Anthony Minessale 05 Feb '16

05 Feb '16

This weekend marks the beginning of the 2016 FreeSWITCH summit where the team meets up to work on ClueCon and code and the next release of FreeSWITCH. Now is your chance to virtually buy the guys a Beer or help fund part of their meal! paypal(a)freeswitch.org http://bit.ly/1L1Azn9 -- 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 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA

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Re: [Freeswitch-users] Best practices for determining if originated call is answered by human or voicemail system?
by Piotr Gregor 04 Feb '16

04 Feb '16

Hi, "I looked at the AVMD module documentation at https://freeswitch.org/confluence/display/FREESWITCH/mod_avmd , and it seems to indicate that this simply detects a beep, i.e. it does not use talking / silence heuristics into account to determine if the call is answered by a human or machine. Am I correct about that? " Yes. The AVMD module works by calculating the estimate of the frequency and amplitude of signal using DESA-2 algorithm. It fires amd::beep event when the variance for the frequency estimate is below the threshold. The event is of form: Event-Subclass: avmd%3A%3Abeep Event-Name: CUSTOM Core-UUID: 36a5b214-8d83-487e-8f8b-50af9a090486 FreeSWITCH-Hostname: home FreeSWITCH-Switchname: home FreeSWITCH-IPv4: 128.11.35.8 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2016-02-04%2016%3A56%3A55 Event-Date-GMT: Thu,%2004%20Feb%202016%2016%3A56%3A55%20GMT Event-Date-Timestamp: 1454605015915799 Event-Calling-File: mod_avmd.c Event-Calling-Function: avmd_process Event-Calling-Line-Number: 556 Event-Sequence: 950 Beep-Status: stop Unique-ID: 0a6f6df3-cc48-4b8a-93f3-b35ee1b8ef0a call-command: avmd The AVMD doesn't take into consideration segments of speech and segments of silence. It simply calculates it's estimate of frequency and amplitude. "How is this intended to be used, do I assume that the call is answered by a human until this module sends a avmd%3A%3Abeep event to me? How do others use this module or other techniques to determine human / machine answer to outbound calls with Freeswitch? " It's use is to detect beep only. You can test how it works by creating extension that will play a tone of given frequency, e.g: <extension name="10014"> <condition field="destination_number" expression="^10014$"> <action application="playback" data="tone_stream://L=3;%(500,6850,850)" /> <action application="hangup"/> </condition> </extension> This plays tone of 500Hz frequency and 6850 ms long, then silence by 850 ms. Start avmd on the call with fs_cli: avmd 8c6a6bb1-ec64-4b2c-9e48-93a6a2598b64 start and inspect events by regestering to avmd events with: events plain CUSTOM avmd::beep Analysing the audio for a presence of the tone is not enough for answering machine detection. You should also analyse at least the length of speech/silence segments. You can do this by subscription to TALK/NOTALK events. Hope this helps. cheers, Piotr On 4 February 2016 at 17:50, Piotr Gregor <peterg(a)sytelco.com> wrote: > Hi, > "I looked at the AVMD module documentation at > https://freeswitch.org/confluence/display/FREESWITCH/mod_avmd , and it > seems to indicate that this simply detects a beep, i.e. it does not use > talking / silence heuristics into account to determine if the call is > answered by a human or machine. > Am I correct about that? " > > Yes. The AVMD module works by calculating the estimate of the frequency > using DESA-2 algorithm. It fires amd::beep event when the variance for that > estimate is below threshold. > The event is of form: > > Event-Subclass: avmd%3A%3Abeep > Event-Name: CUSTOM > Core-UUID: 36a5b214-8d83-487e-8f8b-50af9a090486 > FreeSWITCH-Hostname: home > FreeSWITCH-Switchname: home > FreeSWITCH-IPv4: 128.11.35.8 > FreeSWITCH-IPv6: %3A%3A1 > Event-Date-Local: 2016-02-04%2016%3A56%3A55 > Event-Date-GMT: Thu,%2004%20Feb%202016%2016%3A56%3A55%20GMT > Event-Date-Timestamp: 1454605015915799 > Event-Calling-File: mod_avmd.c > Event-Calling-Function: avmd_process > Event-Calling-Line-Number: 556 > Event-Sequence: 950 > Beep-Status: stop > Unique-ID: 0a6f6df3-cc48-4b8a-93f3-b35ee1b8ef0a > call-command: avmd > > The AVMD doesn't take into consideration TALK/NOTALK events. > > > "How is this intended to be used, do I assume that the call is answered > by a human until this module sends a avmd%3A%3Abeep event to me? How do > others use this module or other techniques to determine human / machine > answer to outbound calls with Freeswitch? " > > You can test how it works by creating extension that will play a tone of > given frequency, e.g: > <extension name="10014"> > <condition field="destination_number" expression="^10014$"> > <action application="playback" > data="tone_stream://L=3;%(500,6850,850)" /> > <action application="hangup"/> > </condition> > </extension> > This plays tone of 500Hz frequency and 6850 ms long, then silence by 850 > ms. > > Start avmd on the call with fs_cli: > avmd 8c6a6bb1-ec64-4b2c-9e48-93a6a2598b64 start > > and inspect events by regestering to avmd events with: > events plain CUSTOM avmd::beep > > Analysing the audio for a presence of the tone is not enough for answering > machine detection. You should also analyse at least the length of > speech/silence segments. > Hope this helps. > > cheers, > Piotr > > > > > > > > On 3 February 2016 at 19:50, Christopher Rienzo <cmrienzo(a)gmail.com> > wrote: > >> I wouldn't use beep detection in a dialer application, but it could be >> useful in something like follow me to reduce the occurrence of voicemails >> being left on subscriber phones. >> >> On Wed, Feb 3, 2016 at 2:28 PM, Bob Hartwig <bobjectsfreeswitch(a)gmail.com >> > wrote: >> >>> I have a client that needs to reliably detect if their outbound calls >>> are answered by a human or voicemail system, so that they can take >>> different actions based on that determination. >>> >>> I looked at the AVMD module documentation at >>> https://freeswitch.org/confluence/display/FREESWITCH/mod_avmd , and it >>> seems to indicate that this simply detects a beep, i.e. it does not use >>> talking / silence heuristics into account to determine if the call is >>> answered by a human or machine. >>> >>> Am I correct about that? How is this intended to be used, do I assume >>> that the call is answered by a human until this module sends >>> a avmd%3A%3Abeep event to me? How do others use this module or other >>> techniques to determine human / machine answer to outbound calls with >>> Freeswitch? >>> >>> Thanks! >>> Bob >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting(a)freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users(a)lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > >

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CORE_PCM_MODULE doesn't seem to load correctly with DSN
by Cj B 04 Feb '16

04 Feb '16

Hi, I've been having a very strange issue lately with loading the core codecs when using core-db-dsn on 1.6.6 and 1.7 on Debian. As long as I am using sqlite, everything appears to work correctly but once I switch to using postgres the core codecs load but don't appear in the show codecs. uname -a Linux c.mydomain.com 3.16.0-4-amd64 #1 SMP Debian 3.16.7-ckt20-1+deb8u1 (2015-12-14) x86_64 GNU/Linux Freeswitch start: http://pastebin.com/jD1GLFAF It appears that they load Show codecs after start: http://pastebin.com/W1VfsQDr And the Core codecs are missing Reload the core_pcm_module and show codecs: http://pastebin.com/qNpD0Due And you can see that they are now there. Any help here would be greatly appreciated. Hopefully I'm just missing something small in my setups.

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loopback/XYZ doesn't contain variables from channel
by Michael Nielsen 04 Feb '16

04 Feb '16

I've got directory with subscribers containing a variable called "country_code". This variable can be used in regular dial plans and works perfectly. However, I've got a conference call dial plan where I use conference_set_auto_outcall. I'mm calling loopback/XYZ to get my other dial plans into play. Everything works, except my subscriber variable "country_code" doesn't get recognised after the loopback/. How can I fix this kind of issue?

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[Session timer too low][SIP 422]
by Bilal Abbasi 04 Feb '16

04 Feb '16

Hi users, I am integrating FreeSwitch with web-sockets, using http://tryit.jssip.net/. So my problem is the extension got registered, but after that when i dial a number i am getting "SIP FAILURE CODE" on JsSIP, i took a network trace, and its showing me SIP 422 responce to invite "Sesion interval too low", now i understand that i need to increase the value of Session Expires, but i cant increase that as it what is configured in tryjssip, so can i some how disable this check from freeswicth? As i guess 90 is the border value that RFC gave. Regards Abbasi

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BLF for Gateway
by Roman Dissauer 04 Feb '16

04 Feb '16

Hi All, is there a way to get gateway usage in freeswitch on my phones blf? I have multiple gateways registered and want to see which one is taken for a particular outbound call. Best Regards, Roman

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How do you check the settings a Freeswitch session was started with?
by vfclists . 04 Feb '16

04 Feb '16

My Freeswitch is configured to start at runtime, but doesn't seem to work until I stop it, switch to the /usr/local/freeswitch/conf directory, and start it using '/usr/local/freeswitch/bin/freeswitch -nc'. When I copy those commands to '/etc/rc.local', it still doesn't work ie cd /usr/local/freeswitch/conf /usr/local/freeswitch/bin/freeswitch -nc I think the service script running it is not configured correctly. Is there some fs_cli, or fsctl option that can be used to interrogate a running instance to see what settings it was started with, or what configuration files it is using? -- Frank Church ======================= http://devblog.brahmancreations.com

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