February 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

SIP Overlap support?
by Dennis 06 Apr '10

06 Apr '10

hi there, i would like to ask, if fs has support for something like "SIP Overlap"? instead of receiving the phonenumber from our carrier in a block, we want to receive the phonenumber digit-by-digit and we want to tell fs when the number is complete. our carrier could send us the phonenumber digit-by-digit, but what about the fs-side? thanks and kind regards dennis

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ClueCon MMX - Call For Speakers!
by Michael Collins 26 Mar '10

26 Mar '10

Hello everyone! We are gearing up for ClueCon 2010 in August later this year. We are making the necessary arrangements for the conference facilities and rooming. Things are beginning to fall into place. Now we need to hear from you. We would like to put out a call for speakers for this year's event. Please contact us if you or your organization would like to give a presentation at ClueCon this year. We want to get the speakers scheduled as early as possible. Keep in mind that those organizations which sponsor ClueCon will be given the highest priority when it comes to scheduling. Please contact Brian West to discuss sponsorship opportunities for this year's event. We also would like to hear from the conference attendees: what would you like to see this year? Please give us your input. ClueCon is, of course, "By Developers, For Developers." However, developers come in all shapes and sizes and we would like have something for everyone. Please tell us what would make ClueCon MMX the best conference of the year! Stay tuned for more announcements. We look forward to hearing from you and seeing everyone this August in Chicago. -ClueCon team

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Freeswitch SPA3000 HUP Not Sent
by Ben Langfeld 21 Mar '10

21 Mar '10

Hey, I have a small 7 seat mostly softphone based PBX installation in place using freeswitch and a couple of SPA3000s for PSTN termination. Currently, aside from PSTN HUP detection, outbound calls are fine setup to directly dial the SPAs. When the internal VoIP side hangs up, freeswitch sends NORMAL_CLEARING to the ATA and the ATA drops the call. Lovely. Unfortunately inbound calls aren't so successful. Calls are sent using the SPA dialplan to an internal freeswitch extension, and the calls connect fine. This time, when the VoIP side hangs up, no NORMAL_CLEARING is sent to the ATA. Is there a reason for this? Can anyone give me any idea how I can get freeswitch to instruct the SPA to drop the line? Regards, Ben Langfeld Wave > Email

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Video pass problem
by Mikhail Krivushin 19 Mar '10

19 Mar '10

Hello! We have problem with pass video over FreeSWITCH. I tshark traf, and see that we have 1280 video packets input, and only 560 passed to B leg. Anyone can point me to right direction? I can send pcap file by request. Most time I have black screen, and then one frame can appear, and I see frozen picture pair minutes, and then other frozen picture. We have not network issues, we have good video, when bypass FS. We have not perfomance troubles to. We have ubuntu 9.10 x64, and powerfull server board. uname: Linux fs 2.6.31-17-generic #54-Ubuntu SMP Thu Dec 10 17:01:44 UTC 2009 x86_64 GNU/Linux config: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=BV32,BV16,speex@8000h @20i,speex@16000h@20i,speex@32000h @20i,PCMU,PCMA,GSM,H264,H263,H263-1998,H263-2000,H261"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=BV32,BV16,speex@8000h @20i,speex@16000h@20i,speex@32000h @20i,PCMU,PCMA,GSM,H264,H263,H263-1998,H263-2000,H261"/> -- С уважением, Кривушин Михаил г. Томск сот. +7 913 865 78 66 icq: 218 744 127 xmpp: KrivushinME(a)jabber.ru skype: mkrivushin

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conferences lead to high server load
by Christian Löschenkohl 14 Mar '10

14 Mar '10

hi we do experience a unusual high server load with the latest freeswitch versions. about 50 conference users lead to a server load of over 10 - reproducible by the way. this wans't the case until my latest trunk update. fs version: 16714 os: debian lenny x86_64 has something substantially changed in mod_conference recently? br -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl(a)xpirio.com

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internal/external profiles
by Brian May 14 Mar '10

14 Mar '10

Hello, Why is it recommended to use separate profiles for internal and external SIP? This page: <http://wiki.freeswitch.org/wiki/External_profile> suggests it is because of NAT. However this page recommends using separate profiles even if NAT is not an issue: <http://wiki.freeswitch.org/wiki/Auto_NAT>: "NOTE: It is still recommended that you use a second profile for your SIP providers. The default conf/sip_profiles/external.xml is set up specifically for use with providers." However I am still left uncertain what this means. Not trying to criticize here, just trying to learn. Thanks. -- Brian May <brian(a)microcomaustralia.com.au>

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Possible typo on Openzap.conf wiki page specifying fxs/fxo-channels
by Robert Hadley 14 Mar '10

14 Mar '10

On the http://wiki.freeswitch.org/wiki/Openzap.conf_Examples wiki page, is there a typo in the wanpipe /usr/local/freeswitch/conf/openzap.conf example concerning specifying the fxo-channel vs. fxs-channel? In the [span wanpipe FXS] section the channels are shown on wiki page as fxo-channels => 1:1 and 1:2 In the [span wanpipe FXO] section the channels are shown on wiki page as fxs-channels => 1:3 and 1:4 Are specifying the fxs-channels and fxo-channels shown in the wrong sections? I had to specify fxs-channels in the FXS span and fxo-channels in the FXO span to get it working on my hardware. Except from wiki page: Sangoma A200/A400 * A200, A200D, A400, A400D series and variants The configuration depends on whether wanpipe is configured to use Zaptel TDM Voice, or the Sangoma standalone TDM Voice API. This is determined in the installation and configuration of the Sangoma wanpipe software. If wanpipe is using Zaptel, you need to configure openzap.conf with [span zt] entries. For example, if ports 1 and 2 are FXS (e.g. configured to accept analog phone connections), and ports 3 and 4 are FXO (e.g. configured to accept PSTN analog lines) you will need: /usr/local/freeswitch/conf/openzap.conf: [span zt FXO] name => OpenZAP number => 3001 fxo-channel => 1 number => 3002 fxo-channel => 2 [span zt FXS] name => OpenZAP number => 4165551111 fxs-channel => 3 number => 4165552222 fxs-channel => 4 If wanpipe is standalone, you need to configure openzap.conf with [span wanpipe] entries. For the same example as above this would be: /usr/local/freeswitch/conf/openzap.conf: [span wanpipe FXS] # This is the value of the callerid_name variable that is raised in the dialplan name => Analog Phone 1 # This is the value of the callerid_number variable that is raised in the dialplan number => 3001 fxo-channel => 1:1 name => Analog Phone 2 number => 3002 fxo-channel => 1:2 [span wanpipe FXO] # the chan_name variable will raised as "OpenZAP/2:1/4165551111" in dialplan when an incoming call arrives on this port name => PSTN line 1 number => 4165551111 fxs-channel => 1:3 name => PSTN line 2 number => 4165552222 fxs-channel => 1:4

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Multiple gateways dial string and user busy
by Max Ivanov 12 Mar '10

12 Mar '10

Hi all! when I do test call from fs_cli: originate sofia/gateway/panas110/223|sofia/gateway/panas111/223 &playaback(local_stream://moh) If firest attempt returns USER_BUSY it tries to call via second one. Is it normal? How can I stop calling attempts after first USER_BUSY?

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Kernel timer warning with CentOS 5.4
by Bruce Hopkins 11 Mar '10

11 Mar '10

Hi, I wonder if anyone would be able to advise please: When I build FreeSWITCH on yum updated CentOS 5.4, I get a warning when I start FreeSWITCH that "Abnormally large timer gap detected" "Do you have your kernel timer set to greater than 1kHz? You may experience audio problems". I get no such warning if I build on CentOS 5.3, and the test timings it measures on starting FreeSWITCH do look lower. All I was doing to upgrade to Centos5.4 was a yum update on the 5.3 build. I guess the warning comes from here: http://fisheye.freeswitch.org/browse/~raw,r=16409/FreeSWITCH/src/switch_tim… This is all on pretty low spec hardware - a couple of different Dell optiplex p4's I use for testing. Does anyone happen to know if I should just stick to Cent)S 5.3, or use 5.4 and not worry about the warnings, or if there is something I can do to fix the problem it is warning about. Perhaps it is just that I shouldn't use such crummy hardware?! Many thanks in advance Bruce

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Re: [Freeswitch-users] smg_prid not bridging the call
by Tomasz Augustyn 11 Mar '10

11 Mar '10

Hello, I had similar problem and I think it is more a problem between Sangoma card and your E1 provider than with freeswitch. In my case it was necessary to set "origination_caller_id_number" to one of the telephone numbers linked to my E1 line. In other case the calls were rejected with "invalid information element" error. You can try Sangoma's support they are very helpful. Tomasz Augustyn ---------- Forwarded message ---------- From: lakshmanan ganapathy <lakindia89(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Date: Sat, 27 Feb 2010 10:32:08 +0530 Subject: Re: [Freeswitch-users] smg_prid not bridging the call In the Dchan log it is saying Invalid Information Elements. That might be a problem??? But I even don't know why it is saying Invalid Information Element?? Please guide me!!!

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