April 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

xml_curl dynamically register users problem
by JuanMa 17 Apr '09

17 Apr '09

Hi, I am using xml_curl, and what I'm trying to do is dynamically register the users to the platform, instead of bulking all the users on the FS boot time. So far I have successfully register the users and also make calls through gateways or to static users (those users that are in the physical files), however every time that I want to reach a User who has been registered and configured through this method, It seams that the FS could not find it (404). For what I have read in the mailings list (http://lists.freeswitch.org/pipermail/freeswitch-users/2009-January/009863.… ) I've seen that theoretically the users MUST be defined on boot time in order to be found, is this correct? For instance if I do have 10.000 users I will be forcing the FS to have all of them in its database when perhaps I only have 1000 online in that specific moment/switch. Perhaps this is something meaningless for the server(in terms of cost or performance), but my first impression is that is far from optimal. So far it seams to be a really cool approach to make the systems scale, how ever It's been very difficult to me to find accurate information and in general I've ended looking at the code, and make my own contribution to the docs. If anyone can give any tip or direction I would really appreciate. Thanks in advance. Juan Manuel

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Call bridge in free switch
by Prabhuram Mohan 17 Apr '09

17 Apr '09

Hello, > > I am trying to find a way to this through fs_cli > 1) call out to ClientA (1001(a)192.168.1.108) ClientB (1002(a)192.168.1.108) > & ClientC (1003(a)192.168.1.108) > 2) Bridge all the 3 legs together into one call > > Thanks > Prabhu >

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Re-2: Re-2: FreeSwitch Complex IVR System
by Guido Kuth 17 Apr '09

17 Apr '09

Hi Dave, didn't fully get your explanation. Maybe I should explain how I do it right now. I am using the http://www.codeplex.com/eventsocket library. I take my x-lite phone with number 1000 and call into FS with number 301234. Under this number I launch my app as async full. <extension name="MyApp"> <condition field="destination_number" expression="^30[0-9][0-9][0-9][0-9]$"> <action application="sleep data="2000"/> <action application="socket" data=192.168.49.237:8021 async full"/> </condition> </extension> I my app I initiate the FSEventManager with the same IP address and to subscribe to all Events and with correct set password from event_socket.conf.xml on port 8021. So now a call is coming in and I receive several Events. On Progress Event I check wether it is an inbound or outbound call direction and create a call Object in my app and set ANI and DNIS before I call a Method who takes care of the next steps. The call is then answered. After taht I get the next Event, who informs me about the call that has been answered. Then I play a greeting and when the greeting ends I get a new event informing me that the greeting has ended. Now comes the struggle I send a RecordSession Command, send a StartDTMF command and after that I send a Bridge Command to bridge the call from my extension 1000 to 1001. At thta point I hear moh on ext 1000 and the ext 1001 rings. I can pickup and I can even answer and hanup or hangup before the bridge was fully established that all works fine even if I have found out thta this is only working because there are some predefinition ins defaut.conf.xml for these internal sip extensions. Problem is that I need full control over the call and over the different call legs. Only the called party (1001) should be able to do something with DTMF input. If extension 1000 does dtmf it doesn't really matter. This is just testing environment. Normally my FS box is behind a PBX as gateway to the ISDN Network. So all the predifined things for internal SIP phones won't work if I try it with phones attached to my PBX I think. I am a little bit lost. I have much experience in phone apps but my programming skills are a little bit rusty, cause they are coming from the good old DOS times. Any help or hints will be appreciated....thanks...Guido -------- Original Message -------- Subject: Re: [Freeswitch-users] Re-2: FreeSwitch Complex IVR System (16-Apr-2009 19:02) From: David Knell <dave(a)3c.co.uk> To: gk(a)exram.de > Hi Guido, > > The event socket interface will give you DTMF events for bridged calls - > just tried it and it works fine. There's one mild snag, which is that > outbound sockets (which are easier for inbound call handling) will only > give you events relating to the specific call leg that's attached to > that socket - i.e. you can use an outbound socket app to bridge that leg > to an outbound call leg just fine, but you won't get events related to > that outbound call. > > So what we do is use an outbound socket app for call control and > scripting, and have a separate inbound socket app which listens for call > state changes and DTMF on all call legs, and a database table which > glues the two together. > > Cheers -- > > Dave > > > > Hi Dave, > > > > thanks for the answer. I am playing around with FS and Event Socket Library > > for .NET. I get pretty much to run with this, but the reason why I came > > from Asterisk to FS is that I cannot get DTMF in a bridged call. I thought > > that I get an Event as soon one dtmf digit is recognized. Unfortunately > > this isn't the case. > > > > If I use the default config files and map the keys with bind_meta_app the > > dtmf tones are recognized and the function behind the bound app is executed. > > Is this maybe a bug. > > > > I have read about mod_managed and that I should use it, but I haven't found > > anything about the usage of it. > > > > Any suggestions would help.... > > > > thanks...Guido > > > > -------- Original Message -------- > > Subject: Re: [Freeswitch-users] FreeSwitch Complex IVR System (16-Apr-2009 > > 17:35) > > From: David Knell <dave(a)3c.co.uk> > > To: gk(a)exram.de > > > > > Hi Guido, > > > > > > My preferred way is to talk to FS through its event socket > > > interface. This allows you fully to control FS, whilst giving > > > you the power to write the code in whatever language and on > > > whatever platform you choose. > > > > > > The documentation starts here: > > > http://wiki.freeswitch.org/wiki/Mod_event_socket > > > > > > Cheers -- > > > > > > Dave > > > > > > > Hi @all > > > > > > > > I have a question about a project I want to realize with FreeSwitch. I > > > > want to do a complex IVR System which takes a call, do many things in > > > > a MSSQL DB, send some Informations to one or many Middleware Servers > > > > via TCP/IP, call one or more mobile phones, the first is able to take > > > > the call, it can be that he must be able to hear a prompt before he is > > > > actually connected to the first caller, then the conversation must be > > > > recorded automatically and during the conversation it must be possible > > > > for the called party to redirect the call by dtmf. I know that this is > > > > all possible, but I want to know which way is the best to do all this? > > > > > > > > > > > > _______________________________________________ > > > > Freeswitch-users mailing list > > > > Freeswitch-users(a)lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users(a)lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org

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How does FS compare with OpenSIPs?
by Gilles 17 Apr '09

17 Apr '09

Hello There's an excellent article on FS vs. Asterisk, but unless I missed it, there's no equivalent to OpenSIPs (www.opensips.org) At this point, apart from the fact that OpenSIPs is not available for Windows, how does FreeSwitch compare with OpenSIPs, what are the strengths and weaknesses of each project? I need to know before recommending one or the other. Thank you for any feedback.

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[Remote SIP client] Couple of questions
by Gilles 17 Apr '09

17 Apr '09

Hello, I have a couple of questions related to having SIP users connecting from the Net to a Freeswitch server through NAT routers on both ends: 1. How must I configure routers on both ends? I understand that I need to route incoming TCP/UDP 5080 into the Freeswitch server, but what about the other router? I guess I also need to route this port to let the SIP phone ring, but what about data (RTP/RTCP)? 2. The Freeswitch server is connected to the POTS with either an OpenVox PCI card or a Linksys 3102 box: When a call is made between the POTS and a remote SIP phone (ie. out there on the Net, not on the same LAN as the Freeswitch server), is there a way for data to flow directly from the POTS to the remote SIP client instead of through the Freeswitch server? Thank you.

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leg_delay_start not working and hangup_after_bridge=true but not if MEDIA_TIMEOUT hangup cause
by Mikael Bjerkeland 17 Apr '09

17 Apr '09

Hi, I have two scenarios I'm having trouble figuring out and I'd be happy if someone could tell me what I'm doing wrong. 1. leg_delay_start=N not working I am trying to delay the origination of the second leg in a forked dial with the following: <action application="bridge" data="user/mikael-nokia(a)voip.domain.com ,[leg_delay_start=10]openzap/1/a/99355151"/> However the second leg is called at exactly the same time as the first one. I am away from my testing environment right now, so I'm sorry for not posting my logs. It appears to me that leg_delay_start is broken on at least rev 13013. 2. I'd like to stop processing the dialplan after a bridge, but not on specific hangup causes. If I get a MEDIA_TIMEOUT hangup cause in the call I'd like to continue in the dialplan. Currently I have the following: <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="user/mikael-nokia(a)voip.domain.com "/>  <action application="bridge" data="openzap/1/a/99355151"/> Any ideas on how to accomplish this? Thanks, Mikael

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FreeSwitch Complex IVR System
by Guido Kuth 16 Apr '09

16 Apr '09

Hi @all I have a question about a project I want to realize with FreeSwitch. I want to do a complex IVR System which takes a call, do many things in a MSSQL DB, send some Informations to one or many Middleware Servers via TCP/IP, call one or more mobile phones, the first is able to take the call, it can be that he must be able to hear a prompt before he is actually connected to the first caller, then the conversation must be recorded automatically and during the conversation it must be possible for the called party to redirect the call by dtmf. I know that this is all possible, but I want to know which way is the best to do all this?

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Optimum sound file format
by Nik Middleton 16 Apr '09

16 Apr '09

Hi Guys, I'm looking for the optimum audio format when using streamfile in a lua script. I've found CPU load increases rapidly with the number of threads playing a .wav file. Can anyone tell me the optimum when using g711a? Right now the the .wav files are Audio format: PCM Sample rate : 8 kHz Mono Sample Size: 16 bit Bit rate :128kbps Will it help CPU load if I resample to a bit rate of 64kbps and sample size of 8 bit? I have read that the sample size needs to be 13-14bit +1 for alaw/ulaw though Regards,

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Issues detecting DTMF tones
by Pete Mueller 16 Apr '09

16 Apr '09

Hey guys. Has anyone else experienced the inability to detect/receive DTMF tones? Just yesterday I had about 4-5 hours where One of my IVR scripts would not detect 1, 2 or 3, but detected the other digits perfectly. If I removed the sound file that was playing, and substituted silence it worked, add the sound file in, and it broke. I have a strong feeling that this is not an issue with FS, but with an upstream system. But wanted to know if anyone has seen this before, and how they went about identifying the culprit and/or fixing it. Some background: - Using FS trunk - Both legs of the call were via SIP gateway. - Setting loglevel to 9 (console and sofia) showed that the RTP packets were not received by FS for 1/2/3 but were received for other digits - Both legs of calls were to/from ATT cell phones - Was using session:setInputCallback() to receive tones, did not test with playAndGetDigits() Thanks for any help. -pete

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RTP errors
by Nik Middleton 16 Apr '09

16 Apr '09

Hi Guys, I'm getting a few of these errors below sofia.c:3247 sofia_handle_sip_i_state() Reinvite RTP Error! Are these caused by a fax machine? Or am I barking up the wrong tree? Regards,

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