- FreeSWITCH-users - FreeSWITCH Mailing Lists

time_test on Centos 5.5
by Louis Huppenbauer 09 Apr '11

09 Apr '11

Hi, I am testing FreeSWITCH on a Centos 5.5. The hardware is a 32 bit AMD Opteron 275. On the system I recognize a avg of 2000 if I run the time_test 1000 application. Currently I have no audio problems but I want to make sure that the system works well. There are 4 possible options: - Try to update the kernel to a newer one - Try to use debian. - Try to use the kernel-rt (realtime kernel from Centos) - Set some nice kernel options. Which? What do you think is the better solution? Thanks in advance. Kind regards, Louis

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Esl outbound connection with java
by alex pappas 09 Apr '11

09 Apr '11

Dear all, I'm trying to start and learn - use the esl interface for creating applications. I try this in java. >From the esl example I see that i need to invoke the SocketClient in order to start a socket and accept connections. I don't understand the AbstractOutboundPipelineFactory class at all.. Does anyone can provide an example on how to start a socket deamon for this use ? Like a kick start example in which I will be able to start and listen for FreeSwitch connections. Also I would like to ask if there is any limitation in using java for this. Thank you in advance Alex Pappas

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SIP to H323 on FS
by FERNANDO VILLARROEL 09 Apr '11

09 Apr '11

Dear all. I have FreeSWITCH Version 1.0.trunk (16526) I need receive traffic SIP from a Gateway A and forward h323 traffic to another Gateway B like this: Gateway A----SIP---> My FS ---H323----> Gateway B It's possible and how i can do? I am seeing the Wiki http://wiki.freeswitch.org/wiki/FreeSwitch_H323 What i need install to my FS Box? How i can do in the Dialplan for send h323 call? <action application="bridge" data="opal/h323:${dialed_extension}@${ip_gateway_B}"/> It's Fine? I hope you can help me Fernando

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Failed to startup as daemon on CentOS 5.5 with latest GIT head
by fieldpeak 09 Apr '11

09 Apr '11

I'm a newbie, and trying use scritp for FS to auto startup when OS start, i follow below link, http://wiki.freeswitch.org/wiki/Installation_Guide#Linux_and_Unix http://wiki.freeswitch.org/wiki/Freeswitch_init#Fedora However, after the OS started for some time, the FS still not startup (by register from eyebeam failure and use './freeswitch status' to check), if i manually excute the scritp (./freeswitch start), it works well. Could anyone advise any clue to to resolve it? Attached is the startup script, FS running on CentOS 5.5 with latest GIT head. Thanks Regards, Charles

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How to stop and replay an audio from the beginning?
by Frankie Yiu 09 Apr '11

09 Apr '11

Hi there, I would like to know what is the preferred way to do the following. My application is this: Make a phone call to a person, after the person picks up the phone an message would play. If he presses the * key, the message would stop and start from the beginning again. I am using C# to start the call and play an audio using PlayAndGetDigits() while in my C++ code would check the DTMF event. If it finds a * key pressed, it will call the uuid_displace with the same file (but for testing purpose, I am using different file). This is not working right because I can hear that the new audio file is playing on top of the original audio instead of stopping the original audio and play the new audio. Anyone has an idea how I should do this? Thanks in advance. Frankie

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defunct processes
by Javier Aristizábal 09 Apr '11

09 Apr '11

Hi folks, I have FS running on a CentOS 5.3 (64 bits) and the last git source. And i'm using ps -ef to look at the process running on my system and i notice that i have more than 20 [freeswitch] <defunct> processes. Is this normal? What exactly do that processes? Thanks in advance -- Javier Aristizábal

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New FreeSWITCH Graphical IVR Released...Testers Needed!
by Zac Wolfe 08 Apr '11

08 Apr '11

Hi all, We (Safi Systems) have just released version Safi Communication Suite (SCS) 1.5.5.Beta, now with FreeSWITCH support! You can check out the blog posting here <http://blog.safisystems.com/2011/04/08/safi-communications-suite-1-5-5-beta…> The product is actually fairly mature and has been around now for over 2 years. Up until now, however, Asterisk has been the only (non-commercial) telephony platform supported. The FreeSWITCH support should be considered Alpha at this point and I'd like to appeal to you in the FreeSWITCH community for feedback and suggestions to help up improve the product. SCS is free to download and use in commercial and non-commercial capacities. Downloads are available HERE <http://www.safisystems.com/downloads> You may find our screencasts<http://www.safisystems.com/screencasts/?pagemode=screencasts>and wiki <http://wiki.safisystems.com/display/DOCS/Home> useful in getting started. Although some screencasts are still Asterisk-specific, most concepts will apply to FreeSWITCH as well. If you still have questions or issues, our forums <http://www.safisystems.com/forums> are active and are an excellent resource. Thanks, Zac Wolfe Safi Systems LLC www.safisystems.com

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Cannot compile freetdm!
by Valery Kalinin 08 Apr '11

08 Apr '11

# cd /usr/local/freeswitch/libs/freetdm # ./configure --with-libisdn # make bla-bla-bla cc1: warnings being treated as errors src/ftmod/ftmod_isdn/ftmod_isdn.c: In function 'ftdm_isdn_931_34': src/ftmod/ftmod_isdn/ftmod_isdn.c:982: warning: unused variable 'cplen' src/ftmod/ftmod_isdn/ftmod_isdn.c: In function 'isdn_configure_span': src/ftmod/ftmod_isdn/ftmod_isdn.c:2794: warning: passing argument 2 of 'Q931SetLogCB' from incompatible pointer type make: *** [ftmod_isdn_la-ftmod_isdn.lo] Error 1 Why? libisdn-0.0.1 installed

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Re: [Freeswitch-users] Failed to startup as daemon on CentOS 5.5 with latest GIT head
by JRichey 08 Apr '11

08 Apr '11

What do you see if you run "chkconfig --list | grep freeswitch"? # chkconfig --list | grep freeswitch freeswitch 0:off 1:off 2:on 3:on 4:on 5:on 6:off If you don't get anything you'll need to add it with "chkconfig --add freeswitch" and "chkconfig freeswitch on". -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org]On Behalf Of fieldpeak Sent: Friday, April 08, 2011 9:21 AM To: FreeSWITCH-users Cc: 13910936628(a)139.com Subject: [Freeswitch-users] Failed to startup as daemon on CentOS 5.5 with latest GIT head I'm a newbie, and trying use scritp for FS to auto startup when OS start, i follow below link, http://wiki.freeswitch.org/wiki/Installation_Guide#Linux_and_Unix <http://wiki.freeswitch.org/wiki/Installation_Guide#Linux_and_Unix> http://wiki.freeswitch.org/wiki/Freeswitch_init#Fedora <http://wiki.freeswitch.org/wiki/Freeswitch_init#Fedora> However, after the OS started for some time, the FS still not startup (by register from eyebeam failure and use './freeswitch status' to check), if i manually excute the scritp (./freeswitch start), it works well. Could anyone advise any clue to to resolve it? Attached is the startup script, FS running on CentOS 5.5 with latest GIT head. Thanks Regards, Charles

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New FreeSWITCH IVR coming, but need HELP!
by Zac Wolfe 08 Apr '11

08 Apr '11

Hi guys, First some good news: we're finally close to releasing our free IVR Development platform SafiServer/SafiWorkshop (www.safisystems.com) with FreeSWITCH support! It's happening much later than we originally anticipated as we've been unexpectedly busy with contracting opportunities but I think it will be worth the wait. Currently everything is working fine with one minor exception: if the user-created script (we call them Saflets) doesn't explicitly hang up the call, the call will remain parked until the caller hangs up. Some details: In Asterisk we invoke our server-side scripting applications via the extensions.conf using the following syntax: exten = 1111,1,Agi(agi:// 192.168.0.10:3573/safletEngine.agi?saflet=project1/callflow1) Here '192.168.0.10' is the IP address of the SafiServer and project1/callflow1 identifies the Saflet to be executed. Asterisk creates a socket connection to the SafiServer and, once the socket is disconnected, the call proceeds to the next entry in the dialplan (typically 'hangup'). For FreeSWITCH, the process is slightly different. Currently, rather than create a separate socket connection for each incoming call, we're invoking an event that informs the SafiServer that there is a new incoming call. The event contains the contextual information including the Saflet name. For example: <extension name="9194"> <condition field="destination_number" expression="9194$"> <action inline="true" application="set" data="saficall=true"/> <action application="event" data="Event-Subclass=saficall::incoming,Event-Name=CUSTOM,saflet_project=test,saflet=flow1,new_saficall=true"/>  <action application="park" /> </condition> </extension> So once the event is fired, the call is parked to prevent further execution within the dialplan. From there on, SafiServer is controlling the call via Inbound Mod event socket. So this works perfectly, except that if the invoked Saflet doesn't explicitly hang-up the call it will remain parked until the caller hangs up. My question is, is there a better way to do this? Is there some better alternative to park in this case? Ideally I'd like to initiate a 'session' of some kind when the SafiServer is "controlling" the call and then exit that session as soon as the Saflet is complete, at which point the call would continue on to the next entry in the dialplan. I understand I could use Outbound sockets to achieve this but, as I mentioned, I'd like to avoid the overhead of a separate socket connection for each incoming call. I actually have a mod_saficall.c app that does basically the the same thing as I described in the dialplan entry. Perhaps there's something more I could do in code that would allow me to be notified when the session is complete. Here's the relevant code I have so far: switch_channel_t *channel = NULL; switch_event_t *event; const char *safiCallFlag = NULL; channel = switch_core_session_get_channel(session); safiCallFlag = switch_channel_get_variable(channel, "saficall"); if (!safiCallFlag) switch_channel_set_variable(channel, "saficall", "true"); if (switch_event_create_subclass(&event, SWITCH_EVENT_CUSTOM, "saficall::incoming") == SWITCH_STATUS_SUCCESS) { switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "new_saficall", safiCallFlag ? "false" : "true"); switch_channel_event_set_data(channel, event); switch_event_fire(&event); switch_ivr_park(session, NULL); } Any ideas you might have on this are welcome. Thanks, Zac Wolfe Safi Systems LLC www.safisystems.com

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