July 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Orphaned Calls
by Jerry Richards 26 Jul '10

26 Jul '10

Four days ago I pulled power on a phone during a call and I noticed the FS "show calls" display still shows the call is active (note I have bypass-media=true). I think SIP session timers would clear this (if I enable it). Is there any other way to clear orphaned calls besides SIP session timers? Thanks, Jerry

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BV32/G722 transcoding issue
by Robert Hadley 26 Jul '10

26 Jul '10

Hello, I might be having a transcoding issue. I am calling in from a Bria softphone that is configured to support BV32, 722, Speex wideband, and g711u to Voicemail (or IVR or Directory). My Freeswitch is configured to allow: global_codec_prefs=H263-1998,speex@8000h@20i,speex@16000h@20i,speex@32000h@2 0i,G7221@32000h,G7221@16000h,G722,PCMU,GSM,BV16,BV32 So naturally Freeswitch picks BV32. Then it would pick G722, SPEEX16, and PCMU as the codes are removed from the softphone. The voicemail phrases that Freeswitch plays are clipped when the BV32 and G722 codes are used to connect with the softphone. Speex16 and PCMU sound okay. The voicemail messages .wav file sounds okay when it plays regardless of the codec. By clipped I mean parts of words are missing. "Pressed" sounds like 'ess' and "Last name" sounds like 'ast name'. Here is a pastebin to the log file: http://pastebin.freeswitch.org/13533 As a workaround I forced Freeswitch to use Speex16 by setting codec_string in the dialplan for the Voicemail, IVR, and Directory. Is there something else I should try or should this transcoding issue be reported on JIRA and what information would be most useful? Thanks, Robert

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Setting up mod_lcr
by Randall Degges 26 Jul '10

26 Jul '10

Hi everyone, I've now got freeswitch installed and running (thanks to everyone who helped out with my previous email question), and now I'm working on configuring mod_lcr, for least cost routing. I've got the module compiled into freeswitch successfully, the problem I'm having is trying to configure the lcr.conf.xml file correctly. I've created a MySQL database on localhost (for simplicity purposes), and created a database called freeswitch, as well as a user 'freeswitch' with a password 'freeswitch' who has access to that database. DB: freeswitch User: freeswitch Pass: freeswitch In the <settings> xml section, the paramater value looks like: <param name="odbc-dsn" value="freeswitch-mysql:freeswitch:Fr33Sw1tch"/> Which, according to the documentation ( http://wiki.freeswitch.org/wiki/Mod_lcr#Installing) is the only thing I need to edit to get LCR to connect to my database, and prepare for using it. However, this value line doesn't contain any sort of IP address to the MySQL db, or database name. It looks like it only has options for a username and password. What do I need to do to correctly specify my database values for mod_lcr to use? Thanks! -Randall

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Ubuntu Server 10.04 64-bit Installation Woes
by Randall Degges 26 Jul '10

26 Jul '10

Hi everyone, I'm a new freeswitch user, and I'm basically trying to use it as a SIP proxy / media proxy (RTP) for handling some SIP phones. Basically, I'm trying to install freeswitch (latest) on a ubuntu-server 10.04 64-bit box, and get everything running smoothly. The requirements I'd like to meet are: - Freeswitch can be controlled via normal ubuntu init scripts (eg: service freeswitch start|stop|restart). - Freeswitch starts on system boot. - Freeswitch binaries are accessible in the default user PATH variable (so that I can simply type 'fs_cli' from the command line). - Freeswitch logs to the appropriate locations in /var/log/ (or wherever the suggested location is). I've found the install guide ( http://wiki.freeswitch.org/wiki/Installation_Guide) really confusing so far. Unless I'm crazy, the top of the install guide says to always install from the git repo (latest), which I also take to mean that one should not install from any OS packages, etc. So what I've tried is: 1. Cloned the git repo. 2. Installed all OS dependency packages. 3. Built from the git repo using the following commands: ./bootstrap.sh ./configure --enable-core-odbc-support sudo make all install sounds-install moh-install Up to this point, everything works great. Everything compiles, all files are located in /usr/local/freeswitch/ which seems good, and things look great. Here are the problems I'm having at this point, which I'm not sure how to resolve: 1. The freeswitch binaries are not in the default user PATH. Basically, if I type 'fs_cli' or 'freeswitch' form the command line, I get a command not found error. This is because the default PATH on ubuntu (obviously) doesn't contain /usr/local/freeswitch/bin. Shouldn't the installer add that? Or am I missing something? What is the best way to get this working? I could just modify my user's .bashrc file to append /usr/local/freeswitch/bin to the PATH variable, but that seems like a hack. 2. There are no system init scripts for freeswitch. I can't start, stop, or restart the freeswitch daemon. This makes managing it a real pain. What's the best way to get this working? I have a lot of prior experience with Asterisk, but very little with freeswitch. So I'm relatively competent working with these sorts of systems once I get the basics :) Thanks so much for any help / advice, I'm really looking forward to seeing what the ideal way to get this running is. -Randall

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SIP alaw to SIP T.38 gateway
by Weera Suriya 26 Jul '10

26 Jul '10

Hello list, I have a question about t.38 fax gateway feature in FS(mod_spandsp). here is my setup. PSTN <----SS7--->YATE(doesn't support T.38 fax gateway)<----IAX2--alaw---->CallWeaver-Faxgateway<------SIP--T.38------->OpenSIPs<------->ATAs(T.38)<---->fax machines I've 16 E1s connected with my Yate server and Yate server send faxes to Callweaver servers (6 servers for CallWeaver) via IAX2(alaw) . then CallWeaver servers send faxes to OpenSIPs server via SIP T.38 and OpenSIPs server send SIP calls to ATAs Now I want to expand this setup by adding 16 more E1 connections and I want to replace these 6 CallWeaver servers with one or two FreeSwitch :) I would like to know one thing. does FS able to convert SIP alaw fax (or IAX2 alaw fax) into SIP T.38 fax (I know TDM to SIP t.38 works with FS) Thank you. Weera Suriya

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Problem with Ribbit registration
by mazilo 26 Jul '10

26 Jul '10

When I used the following SIP profile configured my FS to register my Ribbit account to Ribbit SIP server, it ONLY worked the first hour. <include> <gateway name="u8712300000"> </gateway> </include> There after, FS keeps reporting with a [b]Registration Failed[/b] message as shown below. 2010-07-26 09:12:36.540607 [WARNING] sofia_reg.c:386 u8791600000 Failed Registration, setting retry to 30 seconds. 2010-07-26 09:12:37.524883 [ERR] sofia_reg.c:1496 u8791600000 Registration Failed with status Request Timeout [408]. failure #1 2010-07-26 09:12:37.615341 [WARNING] sofia_reg.c:386 u8791600000 Failed Registration, setting retry to 60 seconds. If I understood it correctly, Ribbit's SIP protocol isn't 100% compliant with SIP in the sense its user ID differs during the SIP registration and subsequent invites. If you have a Ribbit account configured on your FS and managed to get it working properly, I sure would appreciate some helps from you. ----- 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/Problem-with-Ribbit-registrat… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Looking for Roommate at cluecon
by Giovanni Maruzzelli 26 Jul '10

26 Jul '10

Dear all, unfortunately seems that my supposed roommate (Meftah - Delphiworld) cannot come to cluecon due to american consulate burocracy. So, if someone else wants to share a room for cluecon, please let me know. -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618

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Update fail on mod_spandsp
by Mircea HUH 26 Jul '10

26 Jul '10

Hello everybody, I tried to update from git and I got the following error: Compiling mod_spandsp.c ... In file included from /usr/src/freeswitch/freeswitch.git/libs/spandsp/src/spandsp/expose.h:81, from /usr/src/freeswitch/freeswitch.git/libs/spandsp/src/spandsp.h:127, from mod_spandsp.h:46, from mod_spandsp.c:37: /usr/src/freeswitch/freeswitch.git/libs/spandsp/src/spandsp/private/t4_tx.h:106: error: expected specifier-qualifier-list before 't4_t6_decode_state_t' make: *** [mod_spandsp_la-mod_spandsp.lo] Error 1 [root@cre-officetel freeswitch.git]# What can I do to make it working? Thank you. Best regards, Mircea

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Howler - Gone?
by David Allen 25 Jul '10

25 Jul '10

Hi All, I purchased a card from Howler Technologies a few months ago and I'm now only getting to setting it up. When I went to their website to download the drivers, I noticed that they have now ceased trading. Does anyone have any further information or how their products will be supported in the future with Freeswitch? Thanks David

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media gateway
by babak yakhchali 25 Jul '10

25 Jul '10

Hi is it possible to use freeswitch as media gateway in which freeswitch is used just for transcoding between g711u and g729? I need this to be able to use g729 codec available on linux because my main freeswitch box is based on windows(all registration info must be on this machine) and I've got a limited bandwidth to handle 10 concurrent calls so I need to have such thing call -> fs(windows)->g711->fs(linux)->g729->ip phones(7941) thanx

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