October 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

IVR menu failure/timeout action
by Raymond Chan 19 Nov '10

19 Nov '10

Hi all, I am experiencing an IVR problem. Except the top menu, all other sub menu will not hang up after reached max invalid input or timeout limit. It will go back up upper menu. I want it to load the exit-sound and then hang the call when it reached max invalid input or timeout limit. Do you tell how to configure? <menu name="m800_ivr_submenu-m_ch" greet-long="C:/FreeSWITCH/recordings/800-ivr-2nd-m.wav" greet-short="C:/FreeSWITCH/recordings/greet-short-m.wav" invalid-sound="ivr/ivr-that_was_an_invalid_entry-m.wav" exit-sound="voicemail/vm-goodbye-m.wav" timeout="5000" max-failures="2"> <entry action="menu-exec-app" digits="1" param="playback C:/FreeSWITCH/recordings/800-ivr-announcement-m_ch.wav"/> <entry action="menu-exec-app" digits="2" param="lua endcall-m.lua"/> </menu> Thanks Raymond

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Radius AAA
by Abid Saleem 16 Nov '10

16 Nov '10

Hi All, Can someone provide a clear guideline on how to setup Radius AAA to work with FS. Please send complete installation & configuration instructions. Thanks for help in advance. Regards--------------Abid SaleemSr. Product ManagerTerminus Technologies

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Video Support
by Samuel Macedo 15 Nov '10

15 Nov '10

Hi, I've already read that FS supports video, but I've never seen anyone using or talking about how to use it. Does FS support video conferences? Or only video calls? Is there any documentation about this subject? Thanks, -- Samuel Macedo

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FreeTDM - Screening Indicator Field
by Stephen Wilde 11 Nov '10

11 Nov '10

Hi, I have a server with Freeswitch (latest GIT revision) + FreeTDM with some Sangoma A108 boards. I have the necessity to set, for outgoing calls, the ISDN information element "Screening Indicator" of calling party number. I have see that some information element can be set in the "freetdm.conf.xml" file (i.e. calling and called typer of number or numbering plan) but I have not found any reference to the screening indicator. If there is no way to specify this information element for each trunk, it's possible to change its default value (that is 01=user provided, verified and passed)? Thanks, Stephen

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registration fails after several hours - FS problem?
by Mario 07 Nov '10

07 Nov '10

I really need help on this as I have weeks into this problem. I thought I had it nailed but I guess not. After 5.5 hours I get: 2010-10-25 15:05:43.407272 [WARNING] sofia_reg.c:387 mguuid Failed Registration, setting retry to 15 seconds. 2010-10-25 15:05:49.557478 [NOTICE] sofia_reg.c:342 Registering mvuuid 2010-10-25 15:05:59.206273 [NOTICE] sofia_reg.c:342 Registering mguuid 2010-10-25 15:06:04.923157 [WARNING] sofia_reg.c:387 mynum777 Failed Registration, setting retry to 30 seconds. 2010-10-25 15:06:05.358321 [WARNING] sofia_reg.c:387 mvuuid Failed Registration, setting retry to 15 seconds. 2010-10-25 15:06:16.125060 [WARNING] sofia_reg.c:387 mguuid Failed Registration, setting retry to 15 seconds. 2010-10-25 15:06:21.151240 [NOTICE] sofia_reg.c:342 Registering mvuuid 2010-10-25 15:06:33.060421 [NOTICE] sofia_reg.c:342 Registering mguuid 2010-10-25 15:06:35.392655 [NOTICE] sofia_reg.c:342 Registering mynum777 and no way to make/get calls until I restart FS. I did this: 1. log 7 2. sofia profile xxxx siptrace on for each profile/gateway 3. restarted router All three did not solve the problem. The trace and log produced no additional lines which is why I am wondering if FS has a problem since the trace shows no SIP activity. 3 gateways with 2 ITSPs 2 DSL/WAN lines, 1 static and 1 dynamic I am using autonat:1.2.3.4 in internal and external profiles. 1.2.3.4 is the external static ip. sofia status profile ... has the right ext ip nat_map status shows the dynamic (wrong) IP I tried starting with -nonat but that was worse the only way to fix is restart FS. I read the wiki on external nat, auto_nat and everything else many times. Thanks Mario

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ODBC and MSSQL
by Peter Hinman 07 Nov '10

07 Nov '10

We've got a MSSQL server that I was hoping to put to use as an ODBC data source for FS. I can connect via ODBS to the database from the FS box and run queries using tsql. FS also appears to be able to connect to the database, as it creates several tables when it starts up. After successfully creating tables, it appears to run a test query, receives an error from which it concludes that transactions are not supported and then disables ODBC. Is MSSQL expected to work with ODBC and FS? It does successfully create tables. Does this mean that FS disabling ODBC is a bug that should be reported in jira? Is there something I should be setting in the config other than <param name="core-db-dsn" value="DSN:USERID:PASSWORD" /> Thanks, Peter Log snippet below: 2010-10-05 15:37:33.169737 [INFO] switch_core_sqldb.c:1275 Opening DB 2010-10-05 15:37:33.234702 [ERR] switch_odbc.c:427 ERR: [begin;delete from channels where hostname='';delete from channels where hostname='';commit;] [STATE: 42000 CODE 102 ERROR: [FreeTDS][SQL Server]Incorrect syntax near ';'. ] 2010-10-05 15:37:33.234775 [ERR] switch_core_sqldb.c:404 SQL ERR [STATE: 42000 CODE 8180 ERROR: [FreeTDS][SQL Server]Statement(s) could not be prepared. ] begin;delete from channels where hostname='';delete from channels where hostname='';commit; 2010-10-05 15:37:33.234796 [ERR] switch_core_sqldb.c:1326 Transactions not supported on your DB, disabling ODBC 2010-10-05 15:37:33.235916 [INFO] switch_core_sqldb.c:1275 Opening DB Thanks, Peter

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PinDr0p: A GaTec System to Trace Call Paths Across Multiple Networks
by mazilo 05 Nov '10

05 Nov '10

Someone recently posted this http://www.gatech.edu/newsroom/release.html?nid=61428 linky on a http://www.dslreports.com/forum/r25000639-General-PinDr0p DSLR VoIP Forum . The project is called PinDr0p and it uses Single-Ended Audio Features to Determine Call Provenance through a series of algorithms to detect and analyze call artifacts. From an engineering point of view, the algorithms employed perhaps use some sort of orthogonal transformations which may require a lot of CPU power to transform/analyze segments of audio streams in a time domain to a frequency domain. Nevertheless, I posted this information here to see if it may interest FS developers to incorporate and implement such a scheme into the next releases of FS if the source codes are made available publicly. After all, the project is funded by NSF. ----- 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/PinDr0p-A-GaTec-System-to-Tra… Sent from the freeswitch-users mailing list archive at Nabble.com.

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no audio when originate to 2 PSTNs
by Tidiane Sy 04 Nov '10

04 Nov '10

Hi all, When I do an originate between two PSTNS, the two telephones ring. But when both answer, there is no audio. I have audio when the originate is between an sip client and a PSTN. My freeswitch is not natted my console logs here: http://pastebin.freeswitch.org/14351 my network capture here: https://rcpt.yousendit.com/978754413/751404552085a1a09ea574cb819a5f74 Your help will be really appreciated Tid

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problem with jingaling (may cause by TLS)
by joy this 04 Nov '10

04 Nov '10

Dear all: I am confused with dingaling for several days. I hope someone could help me please. Below are my config steps: 1. Download “gnutls-2.9.9.exe” from “ http://josefsson.org/gnutls4win/” and install. 2. Build “libgnutls-26.lib” (refer to “ http://wiki.freeswitch.org/wiki/Ixemel_MSVS_project_example”) I have rebuilt the FreeSwitch. 3. Copy the TLS dlls to “C:\FreeSWITCH\Debug”. 4. Set the conf in “\conf\jingle_profiles\client.xml” (refer to “ http://wiki.freeswitch.org/wiki/Mod_dingaling#TLS” and the letter “[Freeswitch-users] Problems making a receiveing calls with mod_jingling” ). 5. Start FreeSwitch. Below are my config in step 3: <include> <profile type="client"> <param name="name" value="thisjoy"/> <param name="login" value="account(a)gmail.com/gtalk"/> <param name="password" value="pwd"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="message" value="Jingle all the way"/> <param name="rtp-ip" value="$${bind_server_ip}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="auto-login" value="true"/> <param name="sasl" value="plain"/> <param name="server" value="talk.google.com"/> <param name="tls" value="true"/> <param name="use-rtp-timer" value="true"/> <param name="exten" value="888"/> <param name="vad" value="both"/> <param name="candidate-acl" value="wan.auto"/> <param name="local-network-acl" value="localnet.auto"/> </profile> </include> I input the command “dingaling status” and found the account was “UNCONNECTED”. Besides, I saw a message in “freeswitch.log”, it is: I. 2010-10-28 18:46:56.468750 [DEBUG] libdingaling.c:1237 TLS NOT SUPPORTED IN THIS BUILD! II. 2010-10-28 18:47:16.453125 [DEBUG] libdingaling.c:1607 io error 2 7 retry in 20 second(s) My OS is Windows XP and the FreeSwitch is 1.0.head. Thank you for helping. Sincerely yours, Thisjoy.

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mod_h323 h323plus debian shlibdeps dependancy problem
by Steven Ayre 03 Nov '10

03 Nov '10

I have install ptlib 2.8.2 and h323plus trunk as per the instructions on the mod_h323 wiki page. When I build the debian packages it finishes with this error: dpkg-shlibdeps: failure: no dependency information found for /usr/local/lib/libh323_linux_x86_64_.so.1.22.0 (used by debian/freeswitch/opt/freeswitch/mod/mod_h323.so) Does anyone know what causes this error, and how to fix it? -Steve

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