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Two gateways, second one dropping calls. How to change port ?
by David Robinson 18 Apr '09

18 Apr '09

I have two gateways setup like this: http://pastebin.com/m41092fa6 I have two dialplans setup as follows: http://pastebin.com/mde96a74 Everything works fine for the first provider, but the second provider drops calls after about 30 seconds. I've been told it's a NAT issue, but would only one provider be dropping calls ? If re-registration is interrupting calls I would have thought it would do it on both providers. It was suggested that I configure one gateway to use a different port and I've tried setting the port like this: <param name="realm" value="sip.pennytel.com:50601"/> But I can't get it to register. I've tried connecting my softphone (X- Lite) directly to either provider by specifying the hostname as sip.providername.com:5061 and it can't register either. Is this the correct way to specify a SIP port to use or is it just that my provider doesn't listen on any other ports than 5060 ? When I specify sip.pennytel.com:5060 it still registers fine. Is this the best way to solve my problem ? By using an alternate SIP port ? Or is there likely to be any other reason why calls are dropping out ?

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Re: [Freeswitch-users] Detecting DTMF during a bridged call
by pete＠privateconnect.com 18 Apr '09

18 Apr '09

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Detecting DTMF during a bridged call
by Pete Mueller 18 Apr '09

18 Apr '09

Quick question ya'll: Is there a way to detect DTMF during bridged conversation? Or do I have to use a conference for that? Scenario: Caller A dials in and is bridged with caller B (in my case via lua - session:execute("bridge", some_route)) If at any time during the conversation caller B presses "*" the bridge should break, Caller A transferred to extension 1234 and caller B transferred to extension 5678. I don't want to create a conference because I will only have 2 callers ever, and creating 200 2-caller conferences seemed like a waste of resources. However, if the actual resource usage for conference vs. bridge isn't really noticeable, then maybe conference is the way to go. I thought I figured out how to do this one day, but now I can't remember how J -pete

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Debug for a certain context?
by Ron McCarthy 18 Apr '09

18 Apr '09

Hi, We have several context setup with different users, all in the directory with IP based auth. Anyway we can turn on debug for just one context/user in the directory, when lots of people are making calls it's hard to see traffic to just for one user, and troubleshoot. Also, anyway you can show the status of a user in the directory? Far as like if the other side is reachable, I have not been able to find a way to! THanks

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Re: [Freeswitch-users] Issues detecting DTMF tones
by pete＠privateconnect.com 17 Apr '09

17 Apr '09

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Re: [Freeswitch-users] Issues detecting DTMF tones
by pete＠privateconnect.com 17 Apr '09

17 Apr '09

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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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