February 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Recommend an IP Phone for use behind NAT
by Alan Schmitz 20 Feb '12

20 Feb '12

I am looking into changing my VOIP server to FreeSwitch. My current system uses Polycom SoundPoint IP 335 phones. It has a public IP and the FreeSwitch server will too, but several of the phones are behind a variety of DSL routers that do NAT. These remote users have reported intermittent problems with one way audio. The NAT configuration on the Polycoms allows for a static setting of the public IP. I've been told phones that support STUN should be more reliable. What IP desk phones work best with FreeSwitch for remote users behind a NAT router with a dynamic IP? Thanks, Alan

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Invitation from your Friend
by William 20 Feb '12

20 Feb '12

Hi, I have just joined xipcall, a free voip service that comes with webphone, mobile (android, blackberry, symbian and windows mobile) and desktop (windows, mac and linux) voip clients. Join me at xipcall by signing up at www.xipcall.com and we will be able to do chatting, make and receive phone and video calls for free. William

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Dropped calls on transfer
by Daniel Knaggs 20 Feb '12

20 Feb '12

Hello, We’ve recently switched from Asterisk to FreeSwitch and experiencing a minor problem. Currently we’ve got twenty LinkSys/Cisco phones (12x SPA942, 6x SPA962, and 2x SPA504) and experiencing dropped calls when transferring calls. This seems to happen more often when the caller has hit the xfer button for the second time before the callee has even answered the phone. But sometimes drops no more than one call a day if the caller waits until the callee has answered. · All phones are running the latest firmware (v6) and have been factory reset · FreeSwitch is running on version “1.0.head (git-b9b7266 2012-02-10 12-23-58 -0600)” · RTP packet size has been changed to 0.020 as per wiki for all phones Has anyone got any ideas? Thanks in advance. [cid:image0fa035.PNG@f9253b63.4a9a4d62] Daniel Knaggs Software Developer Reality Solutions Ltd, 1 Global Business Park, Hamburg Road, Kingston upon Hull, East Yorkshire, HU7 0AE Tel: 01482 828000 / Fax: 01482 373100 Daniel.Knaggs(a)realitysolutions.co.uk<mailto:Daniel.Knaggs@realitysolutions.co.uk> www.realitysolutions.co.uk<http://www.realitysolutions.co.uk/> ________________________________ Sage Accredited Business Partner serving businesses in Yorkshire & Lincolnshire [cid:image18c143.PNG@f04e1065.4da834c5]

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vm to email
by Aakash 20 Feb '12

20 Feb '12

Hi All, I am trying to send voicemail to email by following http://wiki.freeswitch.org/wiki/Mod_voicemail#Using_Outside_Programs_to_Sen… . But I am not able to receive the mail. I have also attached my configuration below through pastebin. • In /usr/local/freeswitch/scripts – ( I have uploaded the coding) http://pastebin.freeswitch.org/18414 • In /usr/local/freeswitch/conf/directory/default -- http://pastebin.freeswitch.org/18415 • In switch.conf.xml -- http://pastebin.freeswitch.org/18417 • After leaving voicemail – Logs shows http://pastebin.freeswitch.org/18418 Please advice where I am wrong ----- Regards, Aakash -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/vm-to-email-tp7292324p7292324… Sent from the freeswitch-users mailing list archive at Nabble.com.

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moh skip/previous
by Rodney 19 Feb '12

19 Feb '12

I have a moh directory with about 20 wav files . is there a method the caller can use to skip to next song or prevous? how would I implement this in a directory with known file names? with unknown file names ending in .wav?

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external inbound sip call questions
by Josh 19 Feb '12

19 Feb '12

A few questions to all knowledgeable heads out there: Everything I read so far about how to set up and configure my FS external sip profile tells me that I need to have IP address/interface and port (default being 5080) to bind to in order to receive inbound calls via the external sip profile. Is that really the case? Is this the only way I could receive such calls? The reason I am asking is this: my Internet-facing interface cannot receive calls (or any connections from outside) unless it initiates these connection first, simply because my ISP is blocking "server" connections (i.e. connections initiated from outside, destined to my IP address/interface). If my Internet-facing machine initiates the connection first (therefore punching a "hole" through the ISPs firewall), then that's allowed - no restrictions of any sort are applied by my ISP. So, I guess what I am asking is this: Is it possible to configure my external sip profile in FS in such a way so that it does not listen to a particular interface:port in order to receive external sip calls, but to use the control channel/connection established to my external SIP provider (on sip provider's IP address and port 5060) in order to find out which ports to use and initiate rtp connections on my machine, instead of waiting for such connections on a specific port? As I already pointed out above - such connections initiated from outside will be blocked by my ISP, but if these are started from my machine they will be allowed, no problem. Hope I am making myself clear with this. Many thanks!

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Sending Fax configuration
by Luis Daniel Lucio Quiroz 19 Feb '12

19 Feb '12

Helow :) Well, i'm new on freeswitch but not on linux or network, as my first tries i was able to configure a sip gw for inbound and outbound calls, and also inbound faxes, but I'm having problems with outbound faxes. Here is my log freeswitch@internal> originate {fax_verbose=true}sofia/gateway/sip.flowroute.com/16138007319 &txfax(/path_to_fax_file) -ERR INVALID_GATEWAY 2012-02-18 22:49:11.530545 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-02-18 22:49:11.530545 [DEBUG] switch_event.c:1522 Parsing variable [fax_verbose]=[true] 2012-02-18 22:49:11.530545 [ERR] mod_sofia.c:4382 Invalid Gateway 'sip.flowroute.com' 2012-02-18 22:49:11.530545 [NOTICE] mod_sofia.c:4768 Close Channel N/A [CS_NEW] 2012-02-18 22:49:11.530545 [DEBUG] switch_core_state_machine.c:494 () Running State Change CS_DESTROY 2012-02-18 22:49:11.530545 [DEBUG] switch_core_state_machine.c:504 (N/A) State DESTROY 2012-02-18 22:49:11.530545 [DEBUG] mod_sofia.c:374 N/A SOFIA DESTROY 2012-02-18 22:49:11.530545 [DEBUG] switch_core_state_machine.c:504 (N/A) State DESTROY going to sleep 2012-02-18 22:49:11.530545 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [sofia] cause: [INVALID_GATEWAY] 2012-02-18 22:49:11.530545 [DEBUG] switch_ivr_originate.c:3364 Originate Resulted in Error Cause: 608 [INVALID_GATEWAY] 2012-02-18 22:49:43.096288 [NOTICE] sofia_reg.c:415 Registering chiro.astraqom.ca-eico.thinktel.ca 2012-02-18 22:50:26.187940 [NOTICE] sofia_reg.c:415 Registering chiro.astraqom.ca-THINKTEL freeswitch@internal> sip.florroute.com gw works pretty good for outgoing calls but when tryng to do the fax thing it sends error. does anyone has an idea? LD

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FreeSWITCH Security Hardening Project
by Bob Smith 19 Feb '12

19 Feb '12

Definitley interested in a hardened FreeSwitch. Both in terms of seeing the software itself hardened (think OpenBSD ...), and also in terms of best practise configuration etc.

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HAPPY B-BDAY BRIAN!!@#!
by Ken Rice 19 Feb '12

19 Feb '12

Hey mr BKW... I know you are stuck in that hospital on your birthday today, but here's good happy birthday wishes anyway.... today your are 35... and in just a few more years we'll send you the 40 tombstone cake... I know the vampires are itching to get more blood and tissue from you... Just sit back relax and let them feast... it'll all be good in the end... To Everyone else, don't forget VoiceNetwork.ca has promised to match dollar for dollar up to $1000 to Brian to help with his hospital expenses etc... Lets see if we can get them to max out... see http://jira.freeswitch.org/browse/BKW-1 on how to get a donation or happy birthday gift to Brian! Ken

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building freeswitch with SSL/TL on Windows 64bit
by Tamer Higazi 18 Feb '12

18 Feb '12

Hi people! I have successfully built freeswitch on Windows with my Sangoma A100 board. I am happy so far, and figured out, that mod dingaling can't be loaded, because SSL / TLS is not supported (googletalk). I followed before the guide: http://wiki.sangoma.com/fs-windows-freeswitch-compile-isdn and at wiki.freeswitch.org: http://wiki.freeswitch.org/wiki/Installation_for_Windows#Compiling_the_Sour… I don't see any further informations, how to get FS compiled with complete SSL / TLS support. does anybody know how to accomplish this?! I got openSSL for windows64 bit installed. For any infos how to resolve that I would thank you. Tamer

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