May 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Transfer calls from Server A Freeswitch box to Server B Freeswitch box.
by Sanath Prasanna 31 May '12

31 May '12

Hi, I transfer some calls from Server A Freeswitch box to Server B Freeswitch box. I need to get value of variable_sip_history_info parameter in Server B. It is coming to Server A. But not coming to Server B. Also I play few announcement from Server B. It is not hear to caller even though it is play from server B box. Pls help to sort out above 2 problems. Server A ip: 10.1.1.252 Server B ip: 10.1.1.253 Here is the dial plan of server A. <include> <extension name="op-inbound"> <condition field="destination_number" expression="^(\d{8}[1,2])$"> <action application="bridge" data="sofia/internal/$1(a)10.1.1.253"/> </condition> </extension> </include> Here is the acl.conf.xml server B <list name="domains" default="deny">  <node type="allow" cidr=" 10.1.1.252/32"/> </list> Server A: <list name="domains" default="deny">  <node type="allow" cidr=" 10.1.1.253/32"/> </list> Br, Sanath

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mod_callcenter making no attempt to contact agents
by Andrew Cassidy 31 May '12

31 May '12

Here's some pastebin links from earlier on IRC: Cli output and queue config: http://pastebin.freeswitch.org/19154 Call log output: http://pastebin.freeswitch.org/19156 So, here's the thing, queue configured, agents and tiers are in the database only (eventually going to be shared). Set up an extension, dial it, get put in queue, hear MoH. However, there's no sign of any attempt to contact the agent, even though it's registered and set as Available. No outgoing sip packets, nothing logged. I've literally just trid a make current, makes no difference. Attached a pcap from server side, showing no invites to the agent. Any ideas folks? -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T <info(a)cassidywebservices.co.uk> *03300 100 960 *F<info(a)cassidywebservices.co.uk> *03300 100 961 *E <info(a)cassidywebservices.co.uk> *andrew(a)cassidywebservices.co.uk *W <info(a)cassidywebservices.co.uk> *www.cassidywebservices.co.uk

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mod_sms MESSAGE request-uri
by John Rose 31 May '12

31 May '12

I'm trying to perform a SendEvent SMS::MESSAGE via ESL and have the MESSAGE go to a different URI than the "to" or "full_to" header that is set using AddHeader on the ESL event. Is it possible to set maybe a "sip_message_req_uri" akin to a "sip_invite_req_uri" variable or something similar for each MESSAGE? Looking at the FS source code it seems that "to" or "full_to" is always used. Thanks, John

1 0

Running multiple SIP profiles
by Adam Kelloway 31 May '12

31 May '12

Hi there, My understanding is that a SIP profile is single-threaded, and that if you wish to increase the amount of possible concurrent call setups/sessions, you should run more SIP profiles. Am I correct so far? Assuming so, is there a threshold you can reach with the number of profiles you are running whereby it starts to degrade performance? Is there an optimal number of profiles you could run based on the number of processors you have available? Out of curiosity, how many profiles are other people running on your typical high-call load system? Thanks, Adam

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random DTMF after bridge attempt timeout in Javascript
by Andy Ayers 31 May '12

31 May '12

Hi, I have the following script snippet (simplified for the purposes of this request): if (session.ready()) { // Play a welcome message, "please wait while we try to connect your call" session.sayPhrase('play', 'welcome.wav'); session.execute("set", "ringback=%(400,200,400,450);%(400,2000,400,450)"); session.execute("bridge", "{effective_caller_id_number=XXXX,origination_caller_id_number=XXXX,ignore_e arly_media=true,continue_on_fail=true,hangup_after_bridge=false,originate_ti meout=20}sofia/outbound/YYYY(a)sip.node4.co.uk"); if (session.getVariable("bridge_hangup_cause") != 'NORMAL_CLEARING') { // Play a sorry we could not connect your call message session.sayPhrase('play', 'sorry.wav'); // Ask if they would like a callback, "press 1 if you would like a callback" session.sayPhrase('play', 'callback.wav'); session.flushDigits(); var callback = session.getDigits(1, '', 5000); if (callback=='1') { //Do some more stuff } } } The problem I have is that if the bridge attempt times out, after playing the callback.wav request message and whilst waiting for input the session receives a random dtmf digit '6' without the caller pressing any keys on the keypad. Can anyone tell me where this digit comes from and how to prevent it from coming in on the original session? Or else tell me that I've got it all completely wrong and tell me how I can achieve the required call handling in a different way? Many thanks Andy

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DTMF Passthrough on different legs
by Michael Lutz 31 May '12

31 May '12

Hi Guys, I am stuck with a bit of a problem I cannot figure out where it is going wrong.. Here's my situation, I have an inbound call, comming from an external provider, they don't support RFC2833 so I turn on inband detection using 'start_dtmf' in the dialplan. This works ok, for the incomming channel and IVR provided (via Lua) to the caller. (Though sometimes it seems to misinterpret and returns wrong keys..) (Session:A) After that, this call is bridged (via Lua to a a new inbound session on the same switch) (without specifying start_dtmf) so RC2388 is used by default (?) (Session:B) This new sessions is playing soem audio (Lua) and is heard by the original caller (the two sessions are bridged). Seperatly I have an inbound ESL connection where I do an origanate using a originate {execute_on_answer=start_dtmf, .... etc) &park() to setup an new external call via the same gateway (they don't support RFC2388 so I activate the inband detection on the b-leg to receive the digits. Let's call this leg Session:C. After the orignated call is answered I bridge the parked call with the previous session (Session:B), so at the end Session:A and Session:B can talk to each other. Al this works perfectly. [Incommming call] Session:A ----- INBAND ------> Bridge(FS.Lua) ---> Session:B ---- RFC2388 ----> FS.Bridge ---> Session:C --- INBAND ------> ESL destination number My symptoms are... 1. I have a customer who had a IVR on his own side, (Session:C) and requires input from Session:A. The weird thing is that it seems the input is wrong. Keys entered do not correspond to what his IVR is receiving. As if newly dtmfs are generated somewhere. When I switch to another inbound provider, making Session:A using rfc2388 and not inband, but do dialout via the provider using inband detection. It seems to work fine. 2. Not being able to get the dtmf's from the ESL Destination number (Session:C). I'm reading this by subscribing to events via ESL for that particulair uuid. When I have a connection (inbound) without inband detection it seems to get the DTMF's from Session:C without any problems. Which makes it weird for me. note: I know the setup is a bit odd, but Session A cannot exit it's Lua script, so I needed to do a workaround to be able to keep te lua alive and being able to control the call (audio) via ESL. I realy need some help on this one, becuase we have a lot of customers complaining about not responding to dtmf's or receving invalid input in the (FS) IVR. Thanks for you help, Regards, Mike.

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Request for a little documentation help
by Ken Rice 31 May '12

31 May '12

Hey Guys, If you are currently maintaining a module in FreeSWITCH and you are the primary maintainer, can you please tag those in the header of the main C module below the Contributors Section and above the file description The only person exempted from this request is Tony. Everyone else please help us out... We want to make sure this is documented for a variety of reasons. See Example Below I snagged from mod_lcr.c /* * FreeSWITCH Modular Media Switching Software Library / Soft-Switch Application * Copyright (C) 2005-2011, Anthony Minessale II <anthm(a)freeswitch.org> * * Version: MPL 1.1 * * The contents of this file are subject to the Mozilla Public License Version * 1.1 (the "License"); you may not use this file except in compliance with * the License. You may obtain a copy of the License at * http://www.mozilla.org/MPL/ * * Software distributed under the License is distributed on an "AS IS" basis, * WITHOUT WARRANTY OF ANY KIND, either express or implied. See the License * for the specific language governing rights and limitations under the * License. * * The Original Code is FreeSWITCH Modular Media Switching Software Library / Soft-Switch Application * * The Initial Developer of the Original Code is * Anthony Minessale II <anthm(a)freeswitch.org> * Portions created by the Initial Developer are Copyright (C) * the Initial Developer. All Rights Reserved. * * Contributor(s): * * Raymond Chandler <intralanman(a)somedom.com> * Rupa Schomaker <rupa(a)somedom.com> * * Maintainer: Raymond Chandler <intralanman(a)somedome.com> * * mod_lcr.c -- Least Cost Routing Module * */

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how to take the result of running lua script on dialplan of freeswitch?
by Samira Mh 31 May '12

31 May '12

hi guys, <include> <extension name="welcome_ivr"> <condition field="destination_number" expression="^samira$"> <action application="lua" data="welcome.lua"/> </condition> </extension> </include> i have created lua script like this: /usr/local/freeswitch/script/welcome.lua #!/usr/bin/lua local sf = string.format function binary_op(a, b, callback) return callback(a, b); end function plus(a, b) return a + b end function minus(a, b) return a - b end function times(a, b) return a * b end print(sf("Plus : %d", binary_op(5, 5, plus))); print(sf("Minus: %d", binary_op(5, 5, minus))); print(sf("Times: %d", binary_op(5, 5, times))); the result of running that is 3 number, but how can i get the 3 number in dialplan of freeswitch so that use the result of this number on it ? thanks so much

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Run LUA script in different server
by Sanath Prasanna 31 May '12

31 May '12

Hi all, I have 2 servers. One server has SIP GW connection From Operator & IVR applications need to build in other server. How to call distributed LUA applications with Mysql Databases from the SIP GW server ? Pls advice. Main idea is, maintaining SIP connection in one server & all the IVR applications in other server. Br, Sanath

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Why I could not detect DTMF when using Originate command
by Chaiyawut Sookplang 30 May '12

30 May '12

I want to make an automatic IVR call using originate command. I issued command "originate {ignore_early_media=true}sofia/gateway/trunk_1/0860216060 8888" and then I found that DTMF from the receiver side couldn't be detected. On the other hand, if I call to extension 8888, IVR and DTMF work fine.

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