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using FIFO to queue calls towards a gateway
by afshin afzali 14 Aug '10

14 Aug '10

Hi FreeSWITCH, Suppose there is a small voice gateway that just has been equipped with a few FXO ports which connected to PSTN's analog lines. I'm going to use FIFO to manage call requests which going through this voice gateway. As I've learned about mod_fifo service, I think that the queue should have statically defined members (one for each port) which in their dial strings target that gateway (parametrized with the PSTN phone number by a session variable) and nothing else! Am I wrong about this? BEST -- afshin

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FreeTDM and Digium analog cards
by Joseph L. Casale 14 Aug '10

14 Aug '10

Does FreeTDM support TDM cards, or just the digital variants? Reading through Sangoma's docs on FreeTDM, it's not clear? Thanks!

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Re: [Freeswitch-users] Strange early media SDP behavior
by Kristian Kielhofner 14 Aug '10

14 Aug '10

I missed the part about bypass_media but offer/answer stays the same. In this case the remote end is selecting PCMU from the advertised codec list. What is it configured to do? Do you have a full trace showing both legs? It's not clear from the few SDPs you've posted which endpoint is which, where fs is, etc. -- Kristian Kielhofner http://blog.krisk.org ----- Original Message ----- From: freeswitch-users-bounces(a)lists.freeswitch.org <freeswitch-users-bounces(a)lists.freeswitch.org> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Sent: Sat Aug 14 11:36:20 2010 Subject: Re: [Freeswitch-users] Strange early media SDP behavior I would disagree, in bypass media mode FS is supposed to forward SDP as is, let endpoints to negotiate the codec. On Saturday 14 August 2010, Kristian Kielhofner wrote: > It's standard SDP offer/answer. Fresswitch will not send three audio > streams with three codecs. You're advertising three codecs with PCMU > first. Freeswitch supports PCMU in your configuration and agrees on that > codec. > > There are several configuration options in freeswitch and the phone that > can influence this behavior. > > > -- > Kristian Kielhofner > http://blog.krisk.org > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Re: [Freeswitch-users] Strange early media SDP behavior
by Kristian Kielhofner 14 Aug '10

14 Aug '10

It's standard SDP offer/answer. Fresswitch will not send three audio streams with three codecs. You're advertising three codecs with PCMU first. Freeswitch supports PCMU in your configuration and agrees on that codec. There are several configuration options in freeswitch and the phone that can influence this behavior. -- Kristian Kielhofner http://blog.krisk.org ----- Original Message ----- From: freeswitch-users-bounces(a)lists.freeswitch.org <freeswitch-users-bounces(a)lists.freeswitch.org> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Sent: Sat Aug 14 11:19:07 2010 Subject: [Freeswitch-users] Strange early media SDP behavior Late negotiation is on, bypass_media=true. Leg B receives the following SDP in SIP 183 message: v=0 o=root 4913 4913 IN IP4 64.21.13.41 s=session c=IN IP4 64.21.13.41 t=0 0 m=audio 37650 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 3 codecs in SDP offer. Freeswitch send to leg A SIP 183 with the following SDP: v=0 o=root 4913 4913 IN IP4 64.21.13.41 s=session c=IN IP4 64.21.13.41 t=0 0 m=audio 37650 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 1 codec only. Why 2 other codecs are removed from SDP? Tested on today's git, older versions have the same behavior. Is it expected behavior or a bug? _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Limits on number of concurrent registered users?
by Fabio Pietrosanti (naif) 14 Aug '10

14 Aug '10

Hi all, i am preparing a setup where i would have a huge number of connected users via SIP/TLS . The amount of concurrent calls will not be such high. However he amount of concurrently registered users could be even of 15.000 . Are there some specific internal design/architectural limits of FreeSWITCH on the number of concurrent users? Fabio p.s. we'll try to do some stress testing next week

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speak in play_and_get_digits
by Seven Du 14 Aug '10

14 Aug '10

Hi, in uuid_broadcast, all the following forms work: xxxx.wav a wav file phrase:demo_ivr_main_menu phrase macro speak:hello in play_and_get_digits, speak doesn't work. I know that I could put the actual text into a macro, however, a speak version would be much more clear, especially for demo. EXECUTE portaudio/0000000000 play_and_get_digits(1 9 3 5000 # speak:'flite|kal|please input some digits, end with #' none x_cc_dtmf .*) 2010-08-14 20:00:05.680230 [ERR] mod_native_file.c:74 Error opening /usr/local/freeswitch/sounds/en/us/callie/speak:flite|kal|please input some digits, end with #.L16 Is there any other way to get it work? Or should I report a bug or request a new feature? Thanks. -- Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn

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Has anyone used Sangama USB FXO card with FreeSWITCH?
by Pankaj Chhabra 14 Aug '10

14 Aug '10

Hi, Has anyone used Sangoma USB FXO with FreeSWITCH?. If yes, could you please let me know if it works properly? I plan to use this to run FreeSWITCH on my laptop and connect to PSTN using this card. Here is the link to Sangoma USB FXO - http://www.sangoma.com/products/hardware_products/analog_telephony/usb_fxo.… Thanks, Pankaj

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FIFO queue callers ID: how to set to a real-life callers ID, not a static entry?
by Victor Chukalovskiy 14 Aug '10

14 Aug '10

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mod_skypopen (skype endpoint) changes, please test
by Giovanni Maruzzelli 13 Aug '10

13 Aug '10

Hi FreeSWITCHers, I've made some long due modifications to mod_skypopen, that maybe introduced bugs. Please test with the latest git and report any *new* problem (ok, old problems too), here in the mailing list, or - much better - in the Jira. ========= commit 45c6c4d3e42e3c114b47d52ca2e9fca6b1be8090 Author: Giovanni Maruzzelli <gmaruzz(a)gmail.com> Date: Fri Aug 13 16:19:20 2010 -0500 skypopen: now answer a call only when directed to do it (before was trying to answer any incoming call). Lot of changes to a messy part, so maybe some problem will come out... Signed-off-by: Giovanni Maruzzelli <gmaruzz(a)gmail.com> =========== Thank you to all, -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618

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Audio quesion
by Marshall Harrison 13 Aug '10

13 Aug '10

I have FreeSWITCH setup to answer calls on port 5080. If I have 2 simultaneous calls are both going over port 5080 or is that just listening port and the calls get routed to a different port? What port is the audio going over? 5080? A different port for each call? Thanks. Marshall Harrison, <https://mvp.support.microsoft.com/default.aspx/profile/marshall.harrison> MVP Microsoft Communications Server GotSpeech Consulting LLC <http://gotspeech.net/> www.GotSpeech.Net | <http://gotuc.net/> www.GotUC.Net * <mailto:marshall@gotspeech.net> marshall(a)gotspeech.net |* <mailto:marshall@gotuc.net> marshall(a)gotuc.net |( 904.342.6205 <http://www.microsoft.com/mvp> Description: Microsoft MVP Program

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