- FreeSWITCH-dev - FreeSWITCH Mailing Lists

How to enable Call Waiting on Freeswitch UA???
by Irshad Ahamed 26 Dec '07

26 Dec '07

Hi, I couldn't hear any beeps when there is a second incoming call. I was of the notion that Call Waiting is enabled by default. If not, how should I enable it?? Any pointers buddies..... Thanks in advance... Best Regards, Irshad.

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srtp, nat, speed
by Ray 25 Dec '07

25 Dec '07

Hi Folks, Its my first post here and Merry Christmas. I've just spend the last few days installing freeswitch + wikipbx with the latest svn sources (as of a couple of days ago). It is working. The server is an amd64 vserver on a gentoo. Machine is a dual EM64T 3GHZ with 8GB of RAM with a public IP. Twinkle able to call in from behind a nat and hear the echo test. I then got the speex codec working .It required sofiaconfig.py to be modified on wikipbx: "codec-prefs":"speex,speex@16000k,iLBC@30i,GSM,G722,PCMU@20i,PCMA@20i", I am able to call the sip provider (vitelity) and get an 800 number IVR audio. I am unable to get an incoming DID call answered via twinkle. Firstly it takes a long time to decide to call the extension and the provider hangs up, and then I'm not sure the NAT configuration is right. Is there some dns lookup stalling the extension calling? On my Grandstream BT200 I can get it registered but it does not do the echo test at all - for some reason this works on twinkle. I would like to see SRTP/ZRTP working. I've grepped around the configs and googled but all I can find that looks relevant is in: libs/sofia-sip/libsofia-sip-ua/soa/soa.c where there is an srtp_enable check. How is this enabled in the xml configs? Opportunistic zrtp encryption would be great. Should I bother hooking up an openser + mediaproxy solution for seamless NAT? I was hoping that freeswitch was a complete solution for an internet wide pbx where there would be a mix of NAT'ed and normal clients, and secure conferences would be possible. Cheers, Ray. -- http://www.rayservers.com/

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localized dialplans
by trixter aka Bret McDanel 21 Dec '07

21 Dec '07

I have redone the localized dialplan stuff, there is even a readme file. The scope of this is limited, its only to provide a replacement dialplan for what users would be used to in a given country. That way the way they used the PSTN telephone provider they can with freeswitch. Currently there is: NANPA (the 18 or so countries that are country code 1) Netherlands Spain - thanks to David Villasmil Venezuela - thanks to David Villasmil If you want to create a new one for your local area the readme gives full instructions on how. The rumor is that this is going to go into trunk soon so getting this filled out (it only takes 2 minutes) would be nice. look for it in SRC/scripts/contrib/trixter/dialplan in an svn repository near you. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast +44 28 9099 6461 US +1 516 687 5200 http://www.trxtel.com the phone company that pays you!

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Conference tones
by Brian West 20 Dec '07

20 Dec '07

I need everyone to help pickout a default tone set for various tones outlined in phrase_en.xml for mod_conference. I'm going to be placing an order for more sound files with gmvoices to fill in any gaps so please try to go over voicemail or anything else that might be missing and let me know what we need to order. Thanks, /b

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Call for Help
by Brian West 15 Dec '07

15 Dec '07

I need someone to do some work on this one: At the cli type help... verify each item listed has the proper decription and correct args. For the most part they should be correct on args but the description might not be enough for someone to know what it does. Anyone up for this task? I'm still testing and tweaking the default configuration and have a few more things to add to it. /b

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Interop list
by Brian West 15 Dec '07

15 Dec '07

http://wiki.freeswitch.org/wiki/Interop_List Please visit that page and add anything you have done any interop testing with.. Please list dates and such if possible. If you have any hardware you wish to donate to on going interop testing please contact me off list. Thanks, Brian West

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Note on using sipphone.com with FreeSWITCH
by Brian West 13 Dec '07

13 Dec '07

Attention everyone: I have added some examples for setting up various sip providers into SVN under directory/default/ that you can use as templates to setup your connections to those providers. One note that I must inform you all on is the fact that SIPPhone.com advertises _sip._tcp.proxy01.sipphone.com that maps to uap.sipphone.com port 443 which is 100% invalid and doesn't work. So when you send an invite to sipphone it'll take 30 seconds for TCP to timeout and then fall back to UDP. You have three options here: 1. Email SIPPhone and ask them to fix it. Which I have done and yet to get a response about. 2. Disable TCP and only use UDP in oubound.xml profile that you register from for sipphone. 3. Do not use SIPPhone at all. The bottom line is they shouldn't advertise an SRV record for TCP if they do not officially support it. Thanks, Brian West

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performance tuning
by trixter aka Bret McDanel 08 Dec '07

08 Dec '07

I am wanting to do some performance tuning and wanted to enlist the help of others but would request that we do this in a coordinated way so that we dont duplicate effort and end up basically spending time on all the same sections of the code. I have been fighting gprof today which does not do multithreaded apps (I have tried some hacks to make it work, that failed). I gave up when I discovered that intel has vtune free for non-commercial work. Read the license carefully technically if you are paid by your employer to work on FS then you dont qualify (but can still get an eval copy). Vtune can visually display code paths and other goodies. http://www.intel.com/cd/software/products/asmo-na/eng/vtune/239144.htm I am going to be working on the core, sofia, curl, odbc (only as it relates to sofia). I have a plan for how to do this in segments so that it should end up with a fairly good result if anything can be tuned. What I would really like though is people to try to deal with applications, mod_conference and mod_voicemail come to mind fairly quickly, both of which are 'native modules' meaning all the functionality is actually in the core. So that would be core stuff too. Codecs are another big area that should be looked at since they are things very commonly used (who only does l16?) Any takers? If you dont know, a profiler generally only tells you 'this is called a bunch look there'. So generally it wont tell you how to change stuff, although it claims "Easy access to compiler generated tuning advice" which implies that it will spell out code (but its language independent?!) that should optimize better. vtune claims that its compiler and language independent, so it *should* work with either suns compiler or gcc. You dont have to use vtune, that is just what I am going to use in linux. I havent fully downloaded it yet so I really cant say if it even works (you may want to ./configure --enable-debug if -g isnt enabled by default). Vtune is 500M and their flash demo is a 20M downloadable .exe (why if its flash is it a .exe? why not just a flv? artards). -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast +44 28 9099 6461 US +1 516 687 5200 http://www.trxtel.com the phone company that pays you!

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Re: [Freeswitch-dev] Update: Problem with sending Instant Messages
by Anthony Minessale 05 Dec '07

05 Dec '07

SIP is domain based like HTTP (Web), SMTP (E-Mail) and Jabber (IM/presence). You need the phones to register with the same domain that the sofia profile is configured to use. The most painless way to do this is: 1) setup DNS for A and/or SRV for SIP to point at your box eg: sip.my.com 2) set the domain variable in the conf/vars.xml to this domain -<X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/> +<X-PRE-PROCESS cmd="set" data="domain=sip.my.com"/> 3) When you configure the phone/soft phone, use sip.my.com as the domain you are registering to so for ext 1000 you become sip:1000@sip.my.com. Now you have a fully qualified unique sip url and all your phones can chat etc. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Alois Komenda <alois.komenda(a)esk.fraunhofer.de> To: freeswitch-dev(a)lists.freeswitch.org Sent: Tuesday, December 4, 2007 8:47:34 AM Subject: [Freeswitch-dev] Update: Problem with sending Instant Messages Yesterday, 12-03-2007, I wrote: Hello, sending IM (SIP MESSAGE) from one user to another through Freeswitch does not work for me. I have traced the problem to this point now: the call of switch_core_hash_find() in sofia_glue_find_profile__() returns NULL. Finally "Invalid Profile <IP of Freeswitch>" is diplayed on the console (among other data about the message). What is the problem here? After a small change to sofia_presence_chat_send(), sending messages worked. The parameter given to sofia_glue_find_profile() was the variable host which contains an IP address. I changed it to the hardcoded value "default". Now sofia_glue_find_profile() returns a valid profile and messages can be sent. What is the correct way to choose the profile name for an IM? Best regards, Alois ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

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Update: Problem with sending Instant Messages
by Alois Komenda 04 Dec '07

04 Dec '07

Yesterday, 12-03-2007, I wrote: Hello, sending IM (SIP MESSAGE) from one user to another through Freeswitch does not work for me. I have traced the problem to this point now: the call of switch_core_hash_find() in sofia_glue_find_profile__() returns NULL. Finally "Invalid Profile <IP of Freeswitch>" is diplayed on the console (among other data about the message). What is the problem here? After a small change to sofia_presence_chat_send(), sending messages worked. The parameter given to sofia_glue_find_profile() was the variable host which contains an IP address. I changed it to the hardcoded value "default". Now sofia_glue_find_profile() returns a valid profile and messages can be sent. What is the correct way to choose the profile name for an IM? Best regards, Alois

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