- FreeSWITCH-users - FreeSWITCH Mailing Lists

Sending SMS to SIPtoGSM gateway
by Imthiyaz Ahmed 15 Jan '09

15 Jan '09

Hi I have a IP to GSM gateway which supports SIP. How I can send SMS to the GSM phone using FreeSwitch + SIP GSM GW? Thanks Imthiyaz

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Announcing the FreeSWITCH Technology Preview VMWare Appliance.
by Ken Rice 15 Jan '09

15 Jan '09

FreeSWITCH + Centos 5.2 rolled into a VMWare appliance that's compatible with VMWare for Windows, VMWare Player, and VMWare Fusion on the Mac. Couple of things you need to know. 1) This is a Centos 5.2 X86_64 that was updated before we started testing it. 2) FreeSWITCH SVN Trunk is installed and operational. See /etc/motd on the running image for all the good information. We'll be unvailing a wiki page for this shortly. For now you can get the head start by downloading this at http://files.freeswitch.org/FreeSWITCH-VM1.0.0.zip Have fun guys! Ken krice(a)freeswitch.org krice(a)rmktek.com

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SQLExecute catches not all errors
by Helmut Kuper 15 Jan '09

15 Jan '09

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, today I moved the voicemail database from sqlite to mysql via odbc. FS started up and connected successfully to the empty database. Normally voicemail adds the neccessary database tables automaticly during startup. In this case I forgot to add permissions for creating, dropping and altering tables to the database user in mysql. So no SM table was createt in database. Unfortunately FS can not detect this. FS thinks everything is ok. After adding the permissions one table was created but the voicemail_prefs wasn't. This was, because I extented the create statement for this table - and the statement was wrong, so mysql couldn't execute it. This case wasn't detected by FS as well and FS resp. voicemail modul thought everything is fine Is there a way to detect those errors, because bug hunting can be last quite long without error messages especially when you have no way to access mysql.log to see the sql statements from FS. regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAklvFuYACgkQ4tZeNddg3dwT5ACfd/ArMsLfsLrnc5peY1qxaDWu kbsAn00gvNJjXwtFYIX41lbbgGWF+m1P =GO2m -----END PGP SIGNATURE-----

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FS 1.2 Windows Error
by Lito Manansala 15 Jan '09

15 Jan '09

Hi, Im getting error on startup when executing freeSWITCH.exe , "The procedure entry point_apr_md5@12 could not be located in the dynamic library libaprutil.dll" -- /Lito

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OpenZAP hardware timers
by Peter P GMX 15 Jan '09

15 Jan '09

Is there a way to use the hardware timers e.g. of a PRI card in fresswitch? Or other question: Is it recommended to use those if they are available? I have installed a dual PRI card, and "show timer" shows one soft timer. Best regards Peter

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Failed bridge in dial plan still answers an OpenZap inbound call
by Scott Ellis 14 Jan '09

14 Jan '09

I have an inbound call via OpenZap, when I attempt to bridge to a SIP extension, I get the ring tone (inbound line) up until the bridge fails (for timeout or do not disturb). At this point the call is answered and then my dial plan moves on to attempt another bridge to different extensions. So I no longer have the ring tone for the person dialing in. The call can still be answered and everything works ok, but I would rather not answer the call until someone actually picks up. Failing that simulating a ring tone would be good enough. Have searched around, but at a bit of a loss on how to dothis. Any suggestions greatly appreciated. Scott From my dialplan <extension name="LandLine IN"> <condition field="source" expression="mod_openzap"/> <condition field="caller_id_number" expression="^[1-8]$">   <action application="set" data="continue_on_fail=true"/>  <action application="bridge" data="{leg_timeout=30}sofia/$${domain}/500"/>   <action application="set" data="call_timeout=15"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="ring_ready"/> <action application="bridge" data="${group_call(ringgroup2@${domain_name})"/>  <action application="set" data="call_timeout=15"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="${group_call(everyone@${domain_name})"/> <action application="hangup" data="NO_ANSWER"/> </condition> </extension>

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ilbc & alaw transcoding not working correctly?
by Hostinsky Miroslav 14 Jan '09

14 Jan '09

Hello all!, I am using freeswitch with Nokia E65 phone. When phone uses iLBC codec and I am trying "monkey demo" in the example IVR sound is choppy. When using alaw/ulaw there is no problem, it works OK. IVR prompts work OK with G711 and iLBC, there is no problem. So probably freeswitch is not able properly transcode between iLBC and G711 (when calling to external SIP peer?). Machine is powerful but it runs inside Xen (but load is minimal, I think this is not problem). So, what do you think? Where is problem? Thank you! -- Miro

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Registrar performance
by kokoska.rokoska 14 Jan '09

14 Jan '09

Hi all, could someone be so kind and share some data about concurrently registered users at FreeSWITCH from production usage? I have done few sipp testing and fall in doubt... BTW: I use xml_curl for directory and the data are served through: 1. apache, php, mysql 2. lighttpd, "c" binary, mysql and in both cases the performance is near the same - well, let say, not differs dramatically. When run kamailio with usrloc db mode 3 (no cache, all in db) on the same machine, performance is very good, so I hope the uderlaying system is setted-up properly. Thanks a lot! Best regards, kokoska.rokoska

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SVN Commits now tweet!
by Brian West 14 Jan '09

14 Jan '09

http://twitter.com/freeswitchsvn Check it out. ;) /b

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Internal vs. External Call, How to detect answered call?
by Klaus Teller 14 Jan '09

14 Jan '09

Hi, I'm unable to detect whether a call was answered or not. Using a custom Java API to connect to Freeswitch Socket Interface, i observed that: 1) session.originate("sofia/internal/1003%192.168.50.94") does block until the callee picks up while 2) session.originate("sofia/gateway/sip.gafachi.com/14156782222") does not block and returns +OK (and the channel identifier) even when the callee doesn't pick up. I was thinking that the channel Id is returned only when the callee picks up. Since this is apparently not the case, what is the definitive way to know if the remote user picked up? Thanks, Klaus. -- Sensationsangebot verlängert: GMX FreeDSL - Telefonanschluss + DSL für nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T4569a

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