May 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Test
by FERNANDO VILLARROEL 21 May '15

21 May '15

This is a test

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Cisco SPA not responding problem with SIP TLS
by Evgeniy Sementsov 21 May '15

21 May '15

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Verto client SSL configuration
by Vik Killa 21 May '15

21 May '15

Hello, The confluence page indicates that both the Verto Client and the Verto Module on FS use the same SSL. We tested using a different a different SSL on each end and it still works. Is this a misconception I have regarding the SSL setup? Thanks, V

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mod_event_socket - run LUA application
by Dmitry Mordovin 21 May '15

21 May '15

Hello, Is it possible to use mod_event_socket to run lua application with some parameters? originate command can make a call to URL and bridge it to application. In my case - only run lua app without voice calls Thenks a lot!

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JsSIP -> OverSIP -> Freeswitch
by Victor Medina 20 May '15

20 May '15

Hi guys! Im having some problems while trying to connect to a Freswitch PBX using OverSip and JsSIP. Using Jssip i can recive calls but when calling from the OverSIP/JsSIp I always get a 422 error. Softphone -> WebRTC ext calls OK WebRTC -> Softphone fails WebRTC -> ECHO TEST call on freeswitch also fails Can somebody help find out what this could be? SIP/2.0 422 Session Interval Too Small Via: SIP/2.0/WSS vv3f75qcos24.invalid;branch=z9hG4bK1673051 From: "test" <sip:1002@conference.cibersys.com>;tag=5or80k3oc9 To: <sip:9196@conference.cibersys.com>;tag=Q3Xyt4X2vSX6Q Call-ID: 4norlmjdkqok6smt0vd4 CSeq: 9416 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150511T062448Z~1baaa24f9e~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Min-Se: 120 Content-Length: 0 BTW... already asked @oversip guys... But wanted to try this in here just in case! =) Thanks. -- Víctor E. Medina M. Platform Architect / Chief Infrastructure +58424 291 4561 BB #79A8AFA2 @VMCibersys

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NAT Settings with dynamic ip & port forward
by Chirag Ajmera 20 May '15

20 May '15

Kindly can anyone pls guide with how to achieve NAT settings in Freeswitch Scenario ------------ Local PC (Bria) <-------------> FS Server (192.168.1.199) (192.168.1.91) Remote Iphone (Bria) <-------------> Internet <-------------> Router (Port Fwd) <-------------> FS Server Router is on dynamic ip - Domain - myhome.ddns.net *ISSUE - cannot register remote extension* *Also, the same scenario works well with freepbx and asterisk (port forwards are ok)* [image: Inline image 1] Thank You Best Regards, Chirag A.

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MANDATORY_IE_MISSING when calling Linphone
by Ajith de Silva 20 May '15

20 May '15

Hi, I am trying to send a call to FreeBAPX from freeSWITCH. Allow the freeSWithc IP in freePABX. Followed the following sequence to do the calls Step 1 (staic user) 1. Register user using the xml file in the directory 2. Dial defined code "5151" 3. Bridge the call to FreePABX. 5. FreeBAPX extension rings & answer 4. Both parties can hear the voice. Step 2 (dynamic user) 1. Register user using the xml provided by the web server. fetch it using the xml_curl & I get <document type="freeswitch/xml"> <section name="directory"> <domain name="abc.com"> <params> <param name="dial-string" value=" {presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_us er}@${dialed_domain})}"/> </params> <groups> <group name="default"> <users> <user id="1000"> <params> <param name="password" value="123123"/> </params> <variables> <variable name="user_context" value="default" /> <variable name="accountcode" value="1000" /> </variables> </user> </users> </group> </groups> </domain> </section> </document> 2. Dial defined code "5151" 3. Try Bridge the call to FreePABX. 4. Call fail with following issue Cannot locate any authentication credentials to complete an authentication request for realm '"asterisk"' Originate Resulted in Error Cause: 96 [MANDATORY_IE_MISSING] 2015-05-20 07:29:59.782326 [INFO] mod_dptools.c:3244 Originate Failed. Cause: MANDATORY_IE_MISSING mod_sofia.c:413 Channel sofia/internal/1000(a)emrostel.com hanging up, cause: MANDATORY_IE_MISSING 2015-05-20 07:29:59.782326 [DEBUG] mod_sofia.c:549 Responding to INVITE with: 480 Can you please me guide what did I miss over there. Best Regards, Ajith de Silva

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Detecting external voicemail
by Ashwin Jain 20 May '15

20 May '15

Hi all, We are using freeswitch 1.4.18. Our setup is very basic, we have an incoming call, which we terminate on mobile phones (via gateways). Is there anyway to detect if the call is picked up by mobile phone provider voicemail? -- Thanks and Regards, Ashwin Jain

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Should I switch to Debian?
by Steven Schoch 20 May '15

20 May '15

When I set up our office PBX using FreeSwitch, I chose CentOS because I had used it before. However, it seems that FreeSwitch is better supported on Debian, at least for updated binary installations. It's not that big a deal to make a new VM and move my configuration. Would it make it easier on me in the long run to switch to a Debian machine? -- Steve

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newbie: originate dial problem
by Masakazu Nakano 20 May '15

20 May '15

Hi there. a softphone is registed called 5630 in public. show registered says following under like that. 5630,27.112.104.17,siGhOxWCoIW0n9JjzwxSTg..,sofia/external/sip:5630@49.97.4.168:61682 ;transport=UDP;rinstance=b1c4cf82f7d6e637,1432040746,49.97.4.168,61682,udp, b3am.com, and freeswitch@internal > originate user/5630 is successfull. but by dialplan/public/5630.xml <extension name="5630"> <condition field="destination_number" expression="5630"> <action application="bridge" data="user/5630"/> </condition> </extension> and make a call with another phone to 5630.to be "Cause: INVALID_PROFILE" 2015-05-19 22:08:53.724519 [NOTICE] switch_ivr_originate.c:2732 Cannot create outgoing channel of type [sofia] cause: [INVALID_PROFILE] 2015-05-19 22:08:53.724519 [DEBUG] switch_ivr_originate.c:3720 Originate Resulted in Error Cause: 611 [INVALID_PROFILE] 2015-05-19 22:08:53.724519 [INFO] mod_dptools.c:3246 Originate Failed. Cause: INVALID_PROFILE why? BR mack

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