October 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

setting up basic call center using mod_fifo
by Ken Kelly-Turner 22 Oct '12

22 Oct '12

Hi all, I'm trying to set up a very basic call center using mod_fifo and I was wondering if there was a way to control which agent the next call on the queue gets sent to as well as a way to monitor the behaviour of the each agent (tell when they've logged in/out, number of calls rejected, record calls etc.) Thanks, Ken

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TCPdump of curl_xml connection
by Yehavi Bourvine 22 Oct '12

22 Oct '12

Hello, I am trying to write a simple server for xcurl_xml which will do the minimal work needed and with maximum security. Hence, I do not use any existing server, but write my own one on C. Can someone please send me a TCPdump of one such transaction, so I can see what needs to be included in the reply? Thanks! __Yehavi:

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call centre agent statistics report
by Jason Holden 22 Oct '12

22 Oct '12

Can anyone provide advice on being able to provide a client with a call centre statistics report?

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session timers
by Darcy Primrose 22 Oct '12

22 Oct '12

I am trying to extend my session timers. If I modify SOFIA_NAT_SESSION_TIMEOUT it does extend the timeout. However, the Min-SE: stays at 120. What is the relationship between the two timers as related to the freeswitch, Session-Exipires and Min-SE re calculation of time delay for the freeswitch to generate the re invite. Darcy -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/session-timers-tp7583825.html Sent from the freeswitch-users mailing list archive at Nabble.com.

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Wrong presented number in FROM
by Grygoriy Dobrovolskyy 22 Oct '12

22 Oct '12

Good day dear list, i would like to get you help, i am facing very simple issue i am not able to resolve: I receive incoming call from my provider which i forward to URI with the following Dialplan <extension name="FACTEO"> <condition field="${sip_to_user}" expression="(12345678).*$" break="on-false"> <action application="set" data="sip_recovery_break_rfc=1"/> <action application="set" data="effective_caller_id_name=${sip_to_user}"/> <action application="set" data="effective_caller_id_number=${sip_to_user}"/> <action application="bridge" data="{absolute_codec_string=PCMA}sofia/gateway/opensips/999999(a)1.2.2.5"/> For whatever reason FS replace the FROM field when he does bridge, the one i receive from the provider is FROM "+33637" <sip:+33637@1.1.1.1> when FS do the invite he does: FROM "+33637 <sip:Freeswith@IP_OF_FREESWITCH> I have tried to do this: <action application="set" data="origination_caller_id_number=${sip_to_user}"/> Same result, when opensips send the invite, the end user sees the number un Caller_id_name, but still FREESWITCH in caller_id_number Please help me replace the FROM "+33637 <sip:Freeswith@IP_OF_FREESWITCH> By FROM "+33637 <sip:+33637@IP_OF_FREESWITCH> !! Thank you dear list

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FSClient
by Scelo Mbokazi 22 Oct '12

22 Oct '12

Hi All. I am new to Freeswitch and would like to replicate the FSClient functionality to our CRM dialler dev. on (Windows VS2010/C#) however I am battling to find the source code. I followed the instructions on the website but I'm still un able to attain the source as I get the following error. [Description: cid:image001.png@01CDADE1.4DC55FC0] Below is the FSClient that I am trying to download. Your help will be much appreciated. [cid:image002.png@01CDB02C.F0D198D0] Regards Scelo Please consider your environmental responsibility before printing this e-mail This e-mail and any file attachments transmitted with it are intended solelyfor the addressee(s) and may contain confidential proprietary information. Access by any other party without the express written permission of the sender is unauthorized. If you have received this e-mail in error you may not copy, distribute or use the contents, attachments or information in any way. Please delete and contact the sender. Velociti is a licensed FSP provider. FSP No 24456

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mod_skypopen not generating inbound A-leg CDR
by DP . 22 Oct '12

22 Oct '12

Hi, Does mod_skypopen generate inbound A-leg CDRs? Simple scenario, inbound call from Skype to anywhere in Freeswitch (currently a Lua script, but also tried direct to extensions), does not generate any CDRs for that A-Leg. I tried transferring to another context, bridging, etc to no avail... only the b-leg, if present, generates a CDR. To eliminate my xml_cdr, I purposely set the post url to an invalid link, which throws a red crit error in console upon call hangup of regular mod_sofia calls. The mod_skypopen throws no error whatsoever, possibly indicating its not generating the CDR? Im running FreeSWITCH Version 1.0.head (git-5e4a514 2012-03-10 22-56-29 -0500). (Tried updating to the latest 1.2 branch last week but got random restarts and crashes of freeswitch, also kept getting "unclean" in the git version header after several make currents', had to quickly revert as this is a production box). Will clone the install again and try and test the latest git at a later date). Thanks, -Damian

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Fridays My Favorite Day... Oh and Friday Free For All
by Ken Rice 22 Oct '12

22 Oct '12

Hey Guys, Don¹t forget Friday Free For All.... Join as http://conference.freeswitch.org GTALK: conf+888(a)conference.freeswitch.org SIP:888@conference.freeswitch.org H323:888@conference.freeswitch.org PSTN: +1.919.386.9900 http://www.youtube.com/watch?v=kfVsfOSbJY0 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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mod db - hostname
by Anders Sonderkjaer 22 Oct '12

22 Oct '12

Hi, I cannot get fetch the data in db_data from the dialplan. The ODBC is running and has been tested. I have not found any documentation on what the 'hostname' column in the db_data table should be - so I just put what the server has right now ('freeswitch01') - is that ok? This is what I am doing in the dialplan: <extension name="incoming_did"> <condition field="${sip_received_ip}" expression="^1.2.3.\d+$"/> <condition field="destination_number" expression="^(\d+)$"> <action application="info"/> <action application="set" data="sip_to_user=${db(select/forward/${sip_to_user})}"/> <action application="bridge" data="sofia/gateway/main_trunk/9999$1"/> </condition> </extension> 1.2.3 are the first four digits of the IP address I get the call from - this works, have tested it with a hardcoded forward. The call sent to FS has the To User set to 2000000. I use this to identify the user sending the call, and then want to look in db_data to find out which number the call should be forwarded to. So 2000000 would be the data_key and the value/data is the number to forward to. What I want to get out of it, is that a bridge is made to sofia/gateway/main_trunk/9999forward_number, where the forward_number is the value/data retrieved from db_data, and 9999 is the prefix for the termination partner (this works, has been tested). Input would really be appreciated! Thanks, A.

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I could really use help with my ITSP outage, translate asterisk to freeswitch
by Mario G 21 Oct '12

21 Oct '12

This is a critical request for help.... All 6 lines are out again today. I am forwarding inbound calls to a land lines but no one can make any outbound calls. Thanks for any suggestions. DDos Attacks have targeted Callcentric again, their instructions to me are for Asterisk not FreeSwitch (their info page for FreeSwitch does not help). I already researched the web for a week with no success. It may be buried in the wiki but I have not been able to find it. The parms they sent are below, can someone tell me what FreeSwitch equivalents are so I can get let them know it's right and possibly get back up? Mario G We have discovered new information for Asterisk users. In order to get DNS SRV working properly you will need to set the following in your /etc/asterisk/dnsmgr.conf file: [general] enable=yes refreshinterval=3600 And for your Callcentric trunk: [callcentric] srvlookup = yes domain = callcentric.com host = callcentric.com outboundproxy = callcentric.com register => 1234MYCCID:SUPERSECRET@callcentric.com Besides user and password all I have is: <param name="realm" value="callcentric.com"/> <param name="proxy" value="callcentric.com"/> <param name="outbound-proxy" value="callcentric.com"/> They suggested changing 2 lines but then I get the error below: <param name="proxy" value="srv.callcentric.com"/> <param name="outbound-proxy" value="srv.callcentric.com"/> ERR] sofia_reg.c:2039 userid Registration Failed with status DNS Error [503]. failure #82

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