October 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 76, Issue 108
by Peter Olsson 17 Oct '12

17 Oct '12

As far as I know fs_ivrd listens on a specific port, and then forks another process (script) when something connects there, so only the forked process should be stopped when the script ends, but the main daemon should continue to run. /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Jeferson Rodrigo Almeida Skickat: den 17 oktober 2012 16:39 Till: freeswitch-users(a)lists.freeswitch.org Kopia: Daniel Romaniuk Ämne: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 76, Issue 108 Hi guys, I'm trying to use the fs_ivrd daemon, to execute a php script, that is called by the dialplan. It works like a charm, if not for the fact that the daemon (fs_ivrd) ends once the script (php) ends. In a system, I'll have to execute this script many times, and the daemon have to stay up. I think that it is wrong, right? I've already tryed to connect direct to a nc, simulating the deamon, and worked fine. I've already executed the script direct from de shell, and there's no sintax errors. Here is my dialplan extension: <extension name="outbound-socket"> <condition field="destination_number" expression="^55(522)$"> <action application="set" data="ivr_path=/usr/local/eox/programas/freeswitch/scripts/teste.php"/> <action application="socket" data="127.0.0.1:9099<http://127.0.0.1:9099> async full"/> </condition> </extension> Here is how I call the deamon: (from the freeswitch bin dir) ./fs_ivrd -h 127.0.0.1 -p 9099 Here is my PHP script: #!/usr/local/eox/programas/php/bin/php -q <?php // set a couple of things so we dont kill the system ob_implicit_flush(true); set_time_limit(30); // Open stdin so we can read the data in $in = fopen("php://stdin", "r"); // Connect echo "connect\n\n"; // Answer echo "sendmsg\n"; echo "call-command: execute\n"; echo "execute-app-name: answer\n\n"; // Play a prompt echo "sendmsg\n"; echo "call-command: execute\n"; echo "execute-app-name: log\n"; echo "execute-app-arg: CRIT Testing text...\n\n"; // Wait sleep(2); // Hangup echo "sendmsg\n"; echo "call-command: hangup\n\n"; fclose($in); ?> I'll be grateful for any help. Thanks, and sorry for any error in my english. -- Jeferson Rodrigo Almeida Engenheiro de Computação jefersonparanaense(a)gmail.com<mailto:jefersonparanaense@gmail.com> !DSPAM:507ec04a32769025299678!

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mod_skyopen question/clarification
by Luis Azedo 17 Oct '12

17 Oct '12

Hello, i am using mod_skyopen and i have a question about codecs. does mod_skypopen work with silk codec at 16khz ? i have a softswitch with several users using AMR-NB codec. i have a dialing plan to make calls to skype through feeswitch/mod_skypopen with G711U. the softswitch does transcoding from AMR-NB to G711U. does freeswitch do transcoding from G711U to SILK? the softswitch can also do transcoding from AMR-NB to SILK before going to frreeswitch. would this be better ? thank you very much for your attention. -- Luis Azedo

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Add Route header with extra parameters
by Yannick LE COENT 17 Oct '12

17 Oct '12

Hello, I would like to add a Route to the outgoing request. According to http://wiki.freeswitch.org/wiki/Variable_sip_network_destination, sip_route_uri should add a Route header, but it is not the case. The SIP request is routed as specified with sip_route_uri, but the Route header is not inserted. This is a copy of what I have done: <action application="bridge" data="{sip_route_uri='sip:127.0.0.1:24680;lr;param', execute_on_answer=record_session '$${base_dir}/recordings/${record_filename}', ignore_early_media=true}sofia/internal/echo(a)odin.nexcom.bzh"/> I have tried fs_path which inserts the Route header but without the extra parameters. Thanks for any help, Yannick

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Scripts for SNMP
by Luis Daniel Lucio Quiroz 16 Oct '12

16 Oct '12

Helo I-ve already read http://wiki.freeswitch.org/wiki/Snmp but it doesn answers my question, i need to know current inbound and outbound calls splited not in total numbers. I did this line but im not sure it is returning correect data. Can you give me yoru opinnion? /usr/local/freeswitch/bin/fs_cli -x "show calls " | grep inbound | wc -l|awk {'print $1'} /usr/local/freeswitch/bin/fs_cli -x "show calls " | grep outbound | wc -l|awk {'print $1'} Regards, LD

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Different DTMF behavior???
by Malay Thakershi 16 Oct '12

16 Oct '12

Hello, I have a toll-free number (877 area code) hosted @ WindStream and is simply getting forwarded to a local number hosted @ Vitelity (214 area code). Call 877 number from T-mobile or Vonage ==> digits get recognized fine (normal behavior) Call 877 number from Google voice/Verizon/ATT/certain area codes ==> digits are not recognized / garbled (almost 100% failure) Call 214 number form Google voice ==> digits get recognized fine (normal behavior) I had everything working fine when the 877 number was hosted with Vitelity. But we moved it to WindStream as we bought lines from them to our server. Something is going wrong in DTMF process. Can someone guide me regarding this? Where to look? I did try "dtmf-type=rfc.." setting and "liberal-dtmf=true" setting. But didn't work. Thanks for help. Malay

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Lua IVR example script issue
by Richard Weremiuk 16 Oct '12

16 Oct '12

I was looking to implement a multi-level IVR menu with Lua, that could be navigated forward and back. I came across a similar request from 2009 in this thread, which received a response from Michael with example script attached. http://lists.freeswitch.org/pipermail/freeswitch-users/2009-December/date.ht ml#50638 This issue I found with this script is that it does not collect DTMF digits. The menu plays normally but it is not possible to navigate the menu. Is there a corrected version of this script anywhere? Thanks Richard Weremiuk

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How to see which LUA processes are running?
by Yehavi Bourvine 16 Oct '12

16 Oct '12

Hello, Is there a wayto see whether there are LUA processes running which has been created with LUARUN api command? I am using these processes to do auto-redial for busy extensions, and would like to know whether the process is still running or died. Thanks! __Yehavi:

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SMS (chat) CDR logging
by Chad Engler 16 Oct '12

16 Oct '12

There are multiple ways to log CDRs in FreeSWITCH; is there a similar mechanism for chat and mod_sms? Incoming chat messages seem to not generate CDRs, is this truly the case? Currently to log chat messages I am using ESL to listen for "MESSAGE" events and logging the relevant information to MongoDB, however I was wondering if there was a more "standard" way I could accomplish this in the FreeSWITCH configs. Thanks, Chad Engler Web Programmer PATLive 1.800.775.7790 x746 Chad.Engler(a)patlive.com <mailto:Chad.Engler@patlive.com> Hosted Communications | Friendly Service www.patlive.com <http://www.patlive.com/>

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incoming DID, small talk with operator then transfer/forward call to extension
by Tomasz Szuster 16 Oct '12

16 Oct '12

Hello. I've searched wiki and other websites but unfortunatelly did not find info or pointers how to resolve this issue. I want to achieve below solution. When incoming DID call occur person assigned to this line can pick a call and speak with customer, then she can forward/transfer call to another local extension or to GSM number. I've found solutions when I can forward call without answering it by operator but this is not the case. I've sucessfullly implemented gsm_open but sadly didn't find any information about transfering active call to next extension. Would you be so kind and show me the way how to get this working ? Thank you. -- Regards Tom

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REMINDER
by Ken Rice 16 Oct '12

16 Oct '12

Please DO NOT reference a pastebin for information in Jira... Do Not assume the pastebing keeps things forever... It is subject to be reset at any time... If the pastebin is reset your ticket is now useless... Please upload traces, logs, backtraces whatever as a .txt file if its text to be read... Patches should be some_fix.patch or some_fix.patch.txt ... Easiest way to make a patch? git diff¹ committed it to your local tree already? git log p 1 ${HASH}¹ where ${HASH} is the has of the commit git log... Help us out guys.... -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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