June 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

att_xfer or other method for mod_callcenter queue to queue transfer
by Jaco Jacobs 24 Jun '17

24 Jun '17

I've been using mod_callcenter in what I can imagine is not the originally intended use case. It's worked so far but now I've run into a problem and being new to FS I've gotten stuck. I originate a call to an extension on another voice system (non FS), and then use Mod_callcenter uuid_standby to keep the agent ready in the queue. I then make an outbound call and put the call into the queue ( like a dialer or click to dial type setup). This all works great, the problem comes when I want to transfer. I've tried uuid-transfer, tried to park the call etc., as soon as I try this the outbound call drops. The closest I've gotten is to use att_xfer directly to an extension on the other non FS voice system, this works fine. The problem is if I want to transfer to another queue, which is a function I definitely want. Any assistance would be greatly appreciated.

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Question: Skype 4.3 support endet! alternate to Fs?
by Thomas Hoellriegel 24 Jun '17

24 Jun '17

Hi all, Microsoft remove all older skypeclients on 07.01.2017. The problem: skype is not working after this date. mod_skypopen support only skype 4.3.37. I don't found a manual to use fs on latest skype versions. Do your can help please? Do your have a way after on Jul 2017? thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste für blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc

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nat-options-ping algorithm
by Dmitriy Borisov 23 Jun '17

23 Jun '17

Hi, All! I have next topology: PSTN <--- FreeSWITCH <--- channel ---> SIP-users This channel is periodically broken (for 1-10 minutes). Dialplan invoked on incoming ring have 13 endpoints, 10sec to each of 12 first endpoint, and transfer to mobile terminal via PSTN. When channel working good all is fine, but when channel is broken, we have longterm timeout to each endpoint until each of endpoint do not gone to UNREGISTER state. Endpoint gone to UNREGISTER state when REGISTER is expired, but this term is too long for me. I have turn on nat-options-ping option in sip profile, but each endpoint after unsuccessful ping don't gone to UNREGISTER state. When channel is originated to such endpoint it have lifecycle 3sec long and it have ended with NORMAL_TEMPORARY_FAILURE hangup cause. 3 secs by 12 endpoints - too long waiting. When endpoint does not registered - near zero second and ended with USER_NOT_REGISTERED hangup cause. What and where I will must modify to achieve near zero secs term of life channel to inaccessible endpoints? Thanx!

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Verto specification
by Luca Pradovera 23 Jun '17

23 Jun '17

Hello, one of our clients has a requirement to use Verto in a mobile application (outside of a WebView). That means the JQuery dependency can't be met, and we need to figure out a way around it. I as thinking of just reimplementing the signaling plane without any JQuery dependency and taking it from there. Is the actual JSON protocol documented anywhere? I can also do a Verto Communicator install and sniff traffic, but I was looking if there was anything more structured. Alternatively, does anyone have better ideas to offer? Thanks! Best regards, Luca Pradovera

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POST additional channel variable in xml_curl - fs_curl
by Kashyap Dhamecha 23 Jun '17

23 Jun '17

Hello All, I have configured mod_xml_curl and did binding of the directory and configuration. To generate XML file I am using fs_curl and everything is working fine. Recently I was testing mod_directory module and found xml_curl will be used to lookup for names. When lookup event is called it is passing some predefined elements which can be accessed to generate XML file. Along with this predefined elements is there any way to pass any channel variable in this lookup and that variable can be used to add some logic in generating XML file. Initially I found "enable-post-var" param in xm_curl.conf.xml will do it. But it did not work. So is it possible to send any channel variable to access in this lookup? Currently, I am able to access following elements in fs_curl. [hostname] => Backend [section] => directory [tag_name] => domain [key_name] => name [key_value] => 192.168.3.47 [Event-Name] => REQUEST_PARAMS [Core-UUID] => a6470f48-ea06-409e-a4cb-1678e0c9a0d6 [FreeSWITCH-Hostname] => Backend [FreeSWITCH-Switchname] => Backend [FreeSWITCH-IPv4] => 192.168.3.47 [FreeSWITCH-IPv6] => ::1 [Event-Date-Local] => 2017-06-23 15:39:43 [Event-Date-GMT] => Fri, 23 Jun 2017 10:09:43 GMT [Event-Date-Timestamp] => 1498212583180796 [Event-Calling-File] => mod_directory.c [Event-Calling-Function] => populate_database [Event-Calling-Line-Number] => 633 [Event-Sequence] => 132072 Thank you very much in advance. -- With kind regards.... Kashyap Dhamecha

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Call limit on RTP-IP Address
by Nabeel Ahmad 22 Jun '17

22 Jun '17

Hello all, If we assign more then one RTP-IP parameter to a sip profile , they are used in round robin. Its perfect . I want to know is there a way to use some limit on that ip ? Say i've 5 ip address listening on box , and i want one concurrent call limit on each media ip. How can i set limit on profile level or set from dialplan. I tried to do but it didn't work . Any help or advice will be highly appreciated Thanking all Nabeel.

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Re- PocketSphinx
by Andrew Keil 22 Jun '17

22 Jun '17

To FreeSWITCH Users, My question is a simple one. Is there anyone that has recently used and got working PocketSphinx inside the latest production build of FreeSWITCH on Linux (CentOS 6 or 7)? I have currently managed to get it working to an extent, however the results are very mixed (eg. sometimes even the simple pizza_yesno grammar always returns yes at 100% confidence when I say "no"). I know it is "Free", however I am hoping that someone has a good grasp on how it works to enable me to improve the recognition and make it work more successfully. It could well be something I have done (or not done) inside my setup. My aim would be to update the confluence documentation to enable this information to be public to assist anyone else wishing to use mod_pocketsphinx. Cheers, Andrew Keil

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switch_core_sqldb.c:2987 invalied cdr data, call not recoverd
by Achintha 22 Jun '17

22 Jun '17

hi all, I configured two freeswitch servers (freeswitch 1.6.18 on debian 8 ) with call recovery feature. It is working properly on extension to extension , out bound and IVR Calls. Then i tried to land an incoming call, the call gets connected to the queue and then freeswitch generate a call to agent and bridge it with queued call and both sides can hear properly. Then i crashed the primary server, call got landed to the second freeswitch server but rtp is not functioning, and in the second freeswitch console, it printed "switch_core_sqldb.c:2987 invalied cdr data, call not recovered". But the call was not disconnected. I used the following configurations *switch.conf.xml : * switch name is same on both servers core-db-dsn and core-recovery-db-dsn configured with pgsql *both sip profiles:* odbc-dsn configured with pgsql track-calls elabled please provide me a solution to sort out this. -- Best Regards.. Achintha Udukumbura

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IVR menu-back after a timeout
by Steven Schoch 21 Jun '17

21 Jun '17

Is it possible for an IVR (in XML) to do a "menu-back" to go back to the top-level menu after a timeout? -- Steve

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Routing Calls to registered endpoints.
by Jospeh Waite 21 Jun '17

21 Jun '17

Hi Guys. Final question of the day. I promise!! I have 2 sofia profiles, one for IP authenticated calls and one for Sip Registrations. There both on port 5060 on 2 different IP addresses. This is basically a Wholesale/retail billing/switching platform. We have a JeraSoft VCS radius based billing and Sip Redirect server. Everything works ok, except for inbound calls from DID provider destined for a registered extension. Call comes in on the IP profile, which then sends a request to the JeraSoft SIP Redirect server, which replies with a 300 multiple choices as follows. SIP/2.0 300 Multiple Choices Via: SIP/2.0/UDP {ip of FS IP profile};rport;branch=z9hG4bKSKFBFy6K81U9H From: "07966677711" <sip:07966677711@{ip of did provider}>;tag=S6Xmmj7meS2Fr To: <sip:443307881011@{ip of jerasoft redirect}:5060> Contact: <sip:joehouse@{ip of FS IP profile}>;q=1.00 Call-ID: aed22c59-d082-1235-baa2-363165383663 CSeq: 108654212 INVITE Max-Forwards: 67 Content-Length: 0 Server: JeraSoft VCS SIP Redirect Server joehouse is the username of a registered user, registered to the reg profile. Now FS instead of connecting the call to the registered user it sends a new request to the Jerasoft redirect. How would I get it to route the call to the registered user? public/30_routing.xml <?xml version="1.0"?> <include> <extension name="JeraSoft VCS Routing">  <condition field="destination_number" expression="^(.+)$">  <action application="export" data="nolocal:h323-call-origin=originate"/> <action application="set" data="sip_h_X-accountcode=${accountcode}" /> <action application="set" data="call_direction=outbound" /> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="inherit_codec=true" /> <action application="set" data="call_timeout=20"/> <action application="set" data="fail_on_single_reject=USER_BUSY" /> <action application="set" data="origination_caller_id_name=${sip_req_user}"/> <action application="set" data="origination_caller_id_number=${sip_from_user}”/>  <action application="bridge" data="{sip_invite_from_uri=sip:${sip_from_user}@${sip_network_ip}}sofia/external/${destination_number}@{Jerasoft SIP Redirect IP}:5060" /> <action application="hangup" data="${bridge_hangup_cause}"/> </condition> </extension> </include> public.xml <profile name="external">   <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways> <aliases>  </aliases> <domains> <domain name="all" alias="false" parse="true"/> </domains> <settings> <param name="debug" value="0"/>   <param name="sip-trace" value="no"/> <param name="sip-capture" value="no"/> <param name="rfc2833-pt" value="101"/>   <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="2000"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="rtp-timer-name" value="soft"/>    <param name="local-network-acl" value="localnet.auto"/> <param name="manage-presence" value="passive"/>          <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="inbound-late-negotiation" value="true"/> <param name="inbound-zrtp-passthru" value="true"/>   <param name="rtp-ip" value=“{IP address}"/> <param name="sip-ip" value=“{ip address}"/> <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/>   <param name="tls" value="$${external_ssl_enable}"/>  <param name="tls-only" value="false"/>  <param name="tls-bind-params" value="transport=tls"/>  <param name="tls-sip-port" value="5061"/>  <param name="tls-cert-dir" value="$${external_ssl_dir}"/>  <param name="tls-passphrase" value=""/>  <param name="tls-verify-date" value="true"/>   <param name="tls-verify-policy" value="none"/>  <param name="tls-verify-depth" value="2"/>  <param name="tls-verify-in-subjects" value=""/>  <param name="tls-version" value="$${sip_tls_version}"/> </settings> </profile>

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