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Generic Question on switch_rtp.c and STFU
by Erwin Davis 25 Dec '09

25 Dec '09

Hi, I have a question on the time period when stfu_read_a_frame(), which reads a frame from a jiffer buffer within an interval T1, is called. Ideally the frames in the jitter buffer are transmitted within interval T1 to avoid the jitter. However, if I understand the code correctly, the interval is determined by the latency between two read_pollfd triggering instead of T1. The reason is that, at switch_rtp.c, Proc stfu_read_a_frame() is called by read_rtp_packet() which is called when read_pollfd is triggered. However, If I understand correctly, read_pollfd is triggeredd only when there is something on the sockets. For example, if the latency of two read_pollfd triggerings is 50 ms, the actual transmit of next new frame from the jitter buffer should be 50 ms delay instead of T1. Please correct me if I am missing something. Thanks, Regards, e

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FreeSWITCH Weekly Conference Call - Holiday Schedule
by Michael Collins 23 Dec '09

23 Dec '09

Hello all! Because the holidays fall on consecutive Fridays this year we decided to have a single conference call on Wednesday Dec 30th at the usual time of 11AM CST. The agenda is posted here: http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_30 Thanks for supporting the weekly calls. Don't forget that we will soon be having giveaways and fun stuff on the calls so be sure to plan on joining us! -Michael

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error with referencing channel->direction
by Juan Backson 23 Dec '09

23 Dec '09

Hi, The following lines give me some strange error: switch_channel_t *channel = switch_core_session_get_channel(session); if (channel->direction==SWITCH_CALL_DIRECTION_OUTBOUND) { error: dereferencing pointer to incomplete type

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Re: [Freeswitch-dev] Help ！ How to adjust the freeswitch speex Codec 's parameter for my sip phone
by sky tan 23 Dec '09

23 Dec '09

hi,Minessale thank for your aswer, I show my setting below: my previous setting in vars.xml is： <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,speex@8000h@20i,speex@16000h@20i,speex@32000h@20i,G7221@16000h,G722,PCMU,PCMA,iLBC@30i,GSM"/> then I change my setting as: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,speex@8000h,speex@16000h,speex@32000h,G7221@16000h,G722,PCMU,PCMA,iLBC@30i,GSM"/> the problem still exist. the appearance is : the x-lite (8000h) calling side sound good, and the x-lite called side's voice quality is very poor. also I test many sip softphone, and get my analyze as: the media consultation in SDP(Session Discription Protocol) is not work correctly in speex@8000. can you get me more suggestion? or any one who have such experience in it ? thanks a lot ! --Forwarded Message Attachment-- From: anthony.minessale(a)gmail.com To: freeswitch-dev(a)lists.freeswitch.org Date: Tue, 22 Dec 2009 09:20:19 -0600 Subject: Re: [Freeswitch-dev] Help ！ How to adjust the freeswitch speex Codec 's parameter for my sip phone add speex@16000h instead of just speex to the codec list. 2009/12/21 sky tan <sky.side(a)hotmail.com> So I explain my question as: if I use x-lite softphone in speex codec(it must be narrowband), the vioce quality is pool . but when I use X-lite in speex wideband codec , the voice quality is very nice. I do try to change the parameter of default_codec_settings in mod_speex.c , but it seen don't affect anything. so who can tell me how to ajust the freeswitch side to fit the X-lite softphone in speex narrowband? best regards . your freeswitch' fan. -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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:+19193869900 _________________________________________________________________ Windows Live Hotmail: Your friends can get your Facebook updates, right from Hotmail®. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/so…

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FreeSWITCH 1.0.5pre10 is now available
by Michael Collins 22 Dec '09

22 Dec '09

It's upgrade-and-test time! The new release announcement is on the main FreeSWITCH page: http://www.freeswitch.org/node/224 Please update, test, and report back bugs and questions. Thanks! -Michael

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"a1-has" param in gateway setting
by Babak Karvandi 22 Dec '09

22 Dec '09

Hi, Does any body know or has test the "a1-hash" parameter with gateway setting? I am not sure if it is even allowed. I have the following gateway setting but when the freeswitch starts up it simply ignores this provider without any error message or attempt to register in the log file. Thank you for your help in advance. <include> <gateway name="iptel"> <param name="username" value="MY-USERNAME"/> <param name="realm" value="iptel.org"/>    <param name="a1-hash" value="30f610a85e973f2b29b75ddc1ec3450e"/> <param name="proxy" value="sip.iptel.org"/> </gateway> </include>

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Re: [Freeswitch-dev] Help ！ How to adjust the freeswitch speex Codec 's parameter for my sip phone
by sky tan 22 Dec '09

22 Dec '09

So I explain my question as: if I use x-lite softphone in speex codec(it must be narrowband), the vioce quality is pool . but when I use X-lite in speex wideband codec , the voice quality is very nice. I do try to change the parameter of default_codec_settings in mod_speex.c , but it seen don't affect anything. so who can tell me how to ajust the freeswitch side to fit the X-lite softphone in speex narrowband? best regards . your freeswitch' fan. --Forwarded Message Attachment-- From: sky.side(a)hotmail.com To: freeswitch-dev(a)lists.freeswitch.org Date: Sun, 20 Dec 2009 10:36:41 +0000 Subject: [Freeswitch-dev] Help ！ How to adjust the speex Codec 's parameter for my sip phone hi, all I used my soft phone with codec in speex, 8000hz , my softphone ' codec is base on opal lib. but when I register it in freeswitch and call in a conference . I fine the voice quality is not good enough。 I did print the parameter in switch_speex_decode() function , I find that my softphone sent encoded_data_len=21. if I send frame as encoded_data_len=20, then the voice quality become nice. So ,who can tell me how to change the freeswitch's speex parameter, then I can operate the encoded_data_len=21 ? best regards . your freeswitch' fan. _________________________________________________________________ Windows Live: Friends get your Flickr, Yelp, and Digg updates when they e-mail you. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/so…

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Changes for files within freeswitch/libs/* compared with official version
by bossekr＠debian.org 21 Dec '09

21 Dec '09

Hi developers, I'm new to FreeSWITCH and would like to know if changes on the freeswitch/libs/* files are done ? I would like to know if it makes sense to use the original sources while prepare a FreeSWITCH packages for e.g. Debian distribution where most of these libraries exists (for some with an different version). From quality point of view it could be a bad idea to do so. Greetings, Raphael

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Re: [Freeswitch-dev] [Freeswitch-biz] Fwd: Implementation of Skinny Call Control Protocol
by Mathieu Parent 21 Dec '09

21 Dec '09

Hello, On Mon, Dec 21, 2009 at 3:56 PM, Itamar Reis Peixoto <itamar(a)ispbrasil.com.br> wrote: > On Mon, Dec 21, 2009 at 11:42 AM, Mathieu Parent <math.parent(a)gmail.com> wrote: >> Hello, >> >> I'm searching for a company or somebody to implement an SCCP (aka >> Skinny) endpoints module in FreeSWITCH (directly or via OPAL). The >> module should support Cisco IP Phones including Cisco IP Phone 7905, >> Cisco IP Phone 7920 and Cisco IP Phone 7960G. >> >> Regards >> >> Mathieu Parent >> > > why you don't flash your phones with the sip firmware ? because: - no french translation (english only) - no way to configure SpeedDial Buttons (remotely) - no way to configure SoftKeys (remotely) - according to <http://www.voip-info.org/wiki/view/Setup+SiP+on+7940+-+7960>: "the firmware runs into intermittent timeouts upon pulling URLs" Regards Mathieu Parent

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Implementation of Skinny Call Control Protocol
by Mathieu Parent 21 Dec '09

21 Dec '09

Hello, I'm searching for a company or somebody to implement an SCCP (aka Skinny) endpoints module in FreeSWITCH (directly or via OPAL). The module should support Cisco IP Phones including Cisco IP Phone 7905, Cisco IP Phone 7920 and Cisco IP Phone 7960G. NB: I am not on freeswitch-biz list. Regards Mathieu Parent

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