September 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Problem making call using itsp
by covici＠ccs.covici.com 06 Sep '10

06 Sep '10

Hi. I am getting a strange (to me) error message when I try to make an outgoing call using flowroute -- which is one of the itsp's I use. Here is the messages I am getting: 2010-09-03 10:59:47.525764 [CRIT] switch_channel.c:915 Invalid data (${local_media_ip} contains a variable) 2010-09-03 10:59:47.525764 [CRIT] switch_channel.c:915 Invalid data (${sip_local_sdp_str} contains a variable) What the heck do these mean and what can I do to fix? Thanks in advance for any ideas. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com

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getting into fs_cli
by covici＠ccs.covici.com 06 Sep '10

06 Sep '10

Hi. How do I prevent any user logging onto my system from getting into fs_cli -- they could shut down the whole system if they wanted to. Can I set a password, only allow root to get in, or what? Thanks in advance for any ideas. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com

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mod_skypopen autoconfiguration
by François Delawarde 06 Sep '10

06 Sep '10

Anyone knows a way to automatically generate skype configurations (basically the encrypted password) *without* needing to manually access the skype client? Thanks, François.

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ODBC not installed
by mazilo 06 Sep '10

06 Sep '10

I excerpted the freeswitch.log file below that shows ODBC isn't installed in the system and suggests to reconfigure freeswitch with --enable-core-odbc-support option. I ran ./configure --help and couldn't find such an option. Can anyone please confirm this? 2010-09-05 13:33:03.980480 [CRIT] mod_nibblebill.c:233 ODBC does not appear to be installed in the core or your dsn is empty. You need to run ./configure --enable-core-odbc-support ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/ODBC-not-installed-tp5501204p… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Advice on a tool for SIP callflow testing that asserts that a number of proxies are correctly used
by tom strickland 06 Sep '10

06 Sep '10

I have a setup where a SIP proxy will route through a number of applications. An application may in turn proxy the call on, or it may become a UAS. I'd like to write test code for the original proxy that allows me to take a call from Alice to Bob and string it along several proxies to an eventual endpoint and I'd like to be able to monitor the progress of the SIP traffic at each point. Here's my question: I am mostly used to test software that pretends to be an endpoint - the UAC or UAS - but not a proxy. Does anyone have any pointers on ways of asserting that SIP traffic is flowing throuh a given proxy? I'd like to be able to test that, when Alice calls Bob, the proxy being tested correctly routes the call through proxyAppA, thence back to the proxy, then proxyAppB and finally to Bob (the UAS). Any thoughts? Thanks, Tom

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find out terminators for play_and_get_digits
by Tony Tin 06 Sep '10

06 Sep '10

Hi, Is there a way to find out the terminator which is used during "play_and_get_digits", thanks. Regards, Tony

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Re: [Freeswitch-users] Troubleshooting in high usage environments
by Bernhard Suttner 06 Sep '10

06 Sep '10

Hi, nice. Maybe a good solution, but I thought about a "freeswitch service". A client connects to a number of freeswitch servers and tell the freeswitch servers to "search for any activity" of user XYZ. If such a request comes in on mod_sofia and will be handeld in the dialplan and in every other module, the logging output is sent to the client. Something like a service interface. Best regards, Bernhard ----- Original Message ----- From: Michael Scheidell [mailto:michael.scheidell@secnap.com] To: freeswitch-users(a)lists.freeswitch.org Sent: Sat, 04 Sep 2010 00:04:43 +0200 Subject: Re: [Freeswitch-users] Troubleshooting in high usage environments > On 9/3/10 5:32 PM, Bernhard Suttner wrote: > > Hi, > > > > what is the best approach to debug SIP registration or the call dialplan > processing on systems under very high usage? Its very difficult to view the > complete log file for all users in debug mode because there are to much call > attempts / sip registrations. > > > > Are there tools where I can turn on debugging only for a certain user or > IP? It would be really cool to have an debugging interface which can be used > to turn on debugging for a user on freeswitch. Maybe also in combination > with not only one server but e.g. a "server farm" :-) > I know in sipx there is a program that looks at logs, called sipviewer. > you can drill down to a certain sender/recipient if you want. > might work here. has X windows and MS windows versions. > > -- > Michael Scheidell, CTO > o: 561-999-5000 > d: 561-948-2259 > ISN: 1259*1300 > > *| *SECNAP Network Security Corporation > > * Certified SNORT Integrator > * 2008-9 Hot Company Award Winner, World Executive Alliance > * Five-Star Partner Program 2009, VARBusiness > * Best in Email Security,2010: Network Products Guide > * King of Spam Filters, SC Magazine 2008 > > > ______________________________________________________________________ > This email has been scanned and certified safe by SpammerTrap(r). > For Information please see http://www.secnap.com/products/spammertrap/ > ______________________________________________________________________

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2 endpoints behing NATs
by Stas Khirman 06 Sep '10

06 Sep '10

Hi, Sorry for novice question: Successfully setup FreeSWITCH on publicly accessible IP using 1.0.6 standard configuration. No changes in configuration, dial plan, etc. Basic tests (music,echo,etc) are working ok. However, the problem observed with call between two softphones (X-lite) behind NATs - call signal are coming through but no media. Both NATs are kind of typical household WiFi routers. Everything ok if one of softphones is connected without NAT. Does two NATs configuration supported by stock FreeSWITCH ? Can I make it work by changing some configuration parameters or I need something like RTP proxy or similar? Regards Stas

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mod_managed API interface question
by Malay Thakershi 05 Sep '10

05 Sep '10

Hello, Please let me know if this is feasible / possible. I need to make outgoing calls based on a scheduler using FS (SIP). I am planning to write a windows service that runs at particular time interval (say 10 minutes), checks who needs to be called, fires commands to originate these calls. I have a single FS command line running (where FS console is running). I am planning to fire FS commands such as: ---------------------- >managed apidemo a1 123 API CALL [managed(apidemo a1 123)] output: ApiDemo executed with args 'a1 123' and event type API. >managedrun apidemo a1 123 API CALL [managedrun(apidemo a1 123)] output: +OK 2008-10-08 17:44:25 [DEBUG] switch_cpp.cpp:1059 console_log() ApiDemo on a background thread #(3), with args 'a1 123'. 2008-10-08 17:44:25 [DEBUG] switch_cpp.cpp:1059 console_log() ExecuteBackground in apidemo completed. ---------------------- So I will pass some arguments to the mod_managed script and it will fire an API to originate call. My main question is: How can I communicate windows service to fire FreeSwitch commands? Thank you for any help or guidance.

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Video calls
by Boris Kovalenko 05 Sep '10

05 Sep '10

Hello! I can't get video calls working with current FS trunk and two eyeBeam 1.5. Is this even possible? May be I haven't enabled importrant module or something else? May someone provide a good example or point me to documentation page. -- With respect, Boris

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