- FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] how to configure pin for use in gsmopen ?
by Luis Azedo 12 Jul '13

12 Jul '13

Hi Giovanni, it works great! tried in windows and linux (linux guest on windows virtualbox host with usb passthru) in windows is very to configure since you can check which ports are used in device manager. in linux is more try and error , when we connect the usb it creates 3 ttyUSB* devices. thanks > > ---------- Forwarded message ---------- > From: Giovanni Maruzzelli <gmaruzz(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Fri, 5 Jul 2013 11:49:21 +0200 > Subject: Re: [Freeswitch-users] how to configure pin for use in gsmopen ? > Hello Luis, > > seems your mod_gsmopen cannot open the ttyUSB ports. > > This can be because those ports are not accessible from the user > FreeSWITCH is running as. Try to run FreeSWITCH as root, and see if it > works, or chmod the ttyUSB* to be rw for FreeSWITCH's user. > > Or, it can be because those ports are not in a usable state, because > were not initialized by Linux. Dongles have a complex machinery > bringing them from a USB-disk state to a USB-modem state. > > I believe you must at least install usb_modeswitch, and all the udev > machinery. I'm not sure those are installed in a minimal centos > installation. > > So, install the usb_modeswitch package and all related stuff (eg: > udev, etc). Or, you can try to compile usb_modeswitch and use it at > boot in an /etc/rc.local. usb_modeswitch very useful site is: > http://www.draisberghof.de/usb_modeswitch/ > > Good luck, and let us know. > > -giovanni > > >

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does FS support any kind of MMS?
by Vincent Xia 12 Jul '13

12 Jul '13

does FS support any kind of MMS SMS is ok, but what about MMS, multimedia messaging service?

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issues on mod_sms to confirm
by Vincent Xia 12 Jul '13

12 Jul '13

issues on mod_sms to confirm there are some issues on mod_sms i need to confirm: 1. mod_sms supports sending message in pager/message mode, but not session mode, neither MSRP is supported, right? 2. mod_sms does not provide any record for sms sent/replied between user agents as it does(CDR) for audio calls, right?

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Nagios Monitoring
by John Kennedy 12 Jul '13

12 Jul '13

Hello all, Are there any Nagios modules that you all can recommend for Freeswitch? I have not found too many and want to get peoples opinions about the few that do exist. What do people use to monitor their Freeswitch installations? Thanks, John John Kennedy If you always do what you've always done you'll always be where you've always been - Unknown (if you can show me who said this, please do) It is impossible to defeat an ignorant man in argument - William G McAdoo I am not an advocate for frequent changes in laws and Constitutions. *But laws and institutions must go hand in hand with the progress of the human mind.* As that becomes more developed, more enlightened, as new discoveries are made, new truths discovered and manners and opinions change, with the change of circumstances, institutions must advance also to keep pace with the times. We might as well require a man to wear still the coat which fitted him when a boy as civilized society to remain ever under the regimen of their barbarous ancestors. -- Thomas Jefferson

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Realtime Call stats
by adahary 11 Jul '13

11 Jul '13

Hi, I'm bridging calls between two FS boxes (different region locations) where both are setup with proxy-media (RTP passthru). I would like to measure calls performence in real time. How to collect FS realtime RTP call stats of: - Packet loss - Jitter - Round trip delay. thanks Assaf -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Realtime-Call-stats-tp7592766… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Limiting available codecs in bypass media mode
by Phil Quesinberry 11 Jul '13

11 Jul '13

I'm wondering what the best way is to limit the available codecs when making outbound bypass-media calls to the PSTN. The scenario is as follows: Extension phones have HD codecs including G722 enabled for the best audio quality between extensions for internal calls. When an outbound call is made to the PSTN, the first-choice wholesale provider we use offers (and seems to prefer) G722. We'd rather ensure that G711 is used instead. Despite the fact that G722 is not enabled for any of the codec prefs (I grep'ed the entire conf dir path to be sure), FS still passes along the offer of G722 from the endpoint, presumably since it's operating in bypass-media mode. Is there a setting to limit the codecs despite having bypass media set? Could we work around this by getting rid of the global bypass media setting and then setting bypass-media before bridging the call? Thanks, Phil Quesinberry Q Systems Engineering, Inc. Embedded Hardware/Software Development and VoIP Business Telephone Hosting Improve your business telephone services and save money (410) 969-8002 http://www.qsystemsengineering.com <http://www.qsystemsengineering.com/>

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fs_cli and button TAB
by Fernando Villarroel 11 Jul '13

11 Jul '13

Dear All. The commands running well but my problem is the follow: When i press TAB on fs_cli console this not show list of command or when i try to do reloadxml fs_cli not autocomplete when i press the buttonTAB? Any idea? Regards. Enviado desde mi iPhone

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using the originate command
by Christopher Hall 11 Jul '13

11 Jul '13

I have written code to interact with FreeSWITCH using the event socket interface. I am originating calls from my code but my understanding of what is required is very limited. I would like to be able to originate a call and have the call I originate processed through the dial plan and I don't appear to be able to make this happen. An example of my problem: Using a default configuration on FreeSWITCH: If I originate a call as below with a SIP phone registered on extension 1000, my sockets server is called as expected. api originate {originate_timeout=30}sofia/internal/1000%192.168.1.5 &socket(127.0.0.1:10000 async full) if I then modify the configuration of my server such that the public dial plan contains the following <include> <extension name="Test"> <condition field="destination_number" expression="^10000$"> <action application="set" data="domain_name=$${domain}"/> <action application="socket" data="127.0.0.1:10000 async full"/> </condition> </extension> </include> And call 10000 from the SIP phone registered on 1000, again my sockets server is called. If however, I attempt to originate a call to 10000 from my code using api originate {originate_timeout=30}sofia/internal/10000%192.168.XX.XXX &socket(127.0.0.1:10000 async full) I get the response. -ERR USER_NOT_REGISTERED Is there a way in which I can originate a call from my code and have it processed in the same way as a call originated from a SIP phone? I would like to be able to do this so that my code can simply originate calls to numbers in the knowledge that FreeSWITCH will then route the call accordingly and that doesn't appear to be the case at the moment as the originate does not appear to be processed by the dial plan. Thanks Chris.

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FreeTDM originate status
by Ashish gautam 11 Jul '13

11 Jul '13

I am generating outgoing calls to PSTN network through event socket originate action. I want to get the status that the call was generated successfully or not i.e. if the call was failed, what was the reason for that (Network failure, user busy, out of reach etc). How can I get that. Please help. -- Ashish

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ClueCon room sharing
by João Mesquita 11 Jul '13

11 Jul '13

Folks, Anyone willing to share a room for ClueCon? I get there on the 5th and leave on the 9th. João Mesquita

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