April 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSwitch SCXML
by jocke eriksson 10 Apr '11

10 Apr '11

Hello FreeSwitch user. I'm the owner of the freeswitch-scxml project, the project is housed at google-code address http://code.google.com/p/freeswitch-scxml/. My intention with this project is to make users able to write SCXML documents to handle there IVR. SCXML is small XML dialect that will manage your states and according to me it is well suited for IVR. The project is far from feature complete but pretty easy to extend. I will be adding more features so please feel free to drop in a feature request in the issue tracker. Regards Jocke Eriksson.

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how to pass arbitrary headers from A leg to B leg when bridging
by Dave Horton 10 Apr '11

10 Apr '11

Is there a way to pass any desired SIP header that I receive on an A leg through to the B leg? I have looked at sofia_glue.c, and I see either special cases for certain headers being handled (e.g., Alert-Info), or support for only X- or P- custom headers. I have a specific need to pass the Proxy-Authorization header that I receive on an A leg and have that same header and value contained on the B leg INVITE I generate (I have authentication turned off for this configuration, because I don't want FS generating challenges, I just want it to pass through the relevant headers). Is this possible without hacking the code? Dave

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problem setting request uri params when bridging a call
by Dave Horton 10 Apr '11

10 Apr '11

I'm relatively new to FS, so apologies if I'm missing something obvious.. I want to bridge an incoming call on the internal profile and have it go out to a configured gateway through the external profile, and I want any request-uri parameters that come in on the A leg to be also contained on the request-uri for the B leg. My first attempt at doing that was to create this dialplan in the public context: <extension name="epB" > <condition field="${sofia_profile_name}" expression="internal"> <action application="set" data="export_vars=sip_req_params"/> <action application="bridge" data="sofia/gateway/gwB/${destination_number}"/> </condition> </extension> This did not work -- the INVITE went out from the external sip profile without the parameters that came in on the received INVITE. The log messages did however indicate that the parameters were being exported: EXECUTE sofia/internal/5684935846(a)204.215.65.202 set(export_vars=sip_req_params) 2011-04-10 00:17:07.350172 [DEBUG] mod_dptools.c:1059 sofia/internal/5684935846(a)204.215.65.202 SET [export_vars]=[sip_req_params] EXECUTE sofia/internal/5684935846(a)204.215.65.202 bridge(sofia/gateway/gwB/15086160900) 2011-04-10 00:17:07.350172 [DEBUG] switch_channel.c:918 sofia/internal/5684935846(a)204.215.65.202 EXPORTING[export_vars] [sip_req_params]=[target=pcs_voip_originate] to event 2011-04-10 00:17:07.350172 [NOTICE] switch_channel.c:812 New Channel sofia/external/15086160900 [674ec775-befb-4b19-8e02-3b1f4159e1d6] However, if I put them in as dial string variables, it works: <extension name="epB" > <condition field="${sofia_profile_name}" expression="internal"> <action application="bridge" data="{sip_invite_params=${sip_req_params}}sofia/gateway/gwB/${destination_number}"/> </condition> </extension> Shouldn't I be able to use the 'set' application and 'export_vars' to get variables from the A leg to be passed to the B leg and to be incorporated appropriately into the B leg INVITE? (I am running a recent head build: FreeSWITCH-mod_sofia/1.0.head-git-244fd68 2011-03-21 14-27-57 -0400) Thanks in advance. Dave

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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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