September 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

files.freeswitch.org
by Brian West 22 Sep '10

22 Sep '10

YES we know files.freeswitch.org is down, hardware failure Dell will be on site to fix the motherboard the data is fine the nic on the MB failed some how. /b

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FS terminate when trying to load mod_h323
by gamalsa 21 Sep '10

21 Sep '10

Mod_h323 compiled and build successfully but when FS trying to load, it terminates with the following error: terminate called after throwing an instance of '__gnu_cxx::__concurrence_unlock_error' what(): __gnu_cxx::__concurrence_unlock_error -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-terminate-when-trying-to-l… Sent from the freeswitch-users mailing list archive at Nabble.com.

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all sip registrations intermittently failing
by covici＠ccs.covici.com 21 Sep '10

21 Sep '10

Hi. Suddenly all sip registrations to sip providers are failing -- it says its for ping, but manual pings always succeed and so I am quite confused. I get warnings about failed registrations, but sometimes they are up, so I have no idea as to what is happening. Not sure how to debug this one. Should I get rid of the ping parms in the gateways? This all has worked before, but I have done make current since, so maybe something broke? Thanks in advance for any ideas. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com

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Re: [Freeswitch-users] sip stack problem in fsw
by JRichey 21 Sep '10

21 Sep '10

It appears to be the lack of a continuous stream of RTP packets that is the issue, which I've incorrectly referred to as VAD. Is there a way to cause FreeSwitch to send a continuous stream of RTP packets even when there's really nothing to send? -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org]On Behalf Of Anthony Minessale Sent: Tuesday, September 21, 2010 9:14 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] sip stack problem in fsw are you in a conference running on FreeSWITCH or just bridging a phone out to a far end? There is nothing in FreeSWITCH that runs vad by default. There are vad params in the RTP stack but they are disabled by default      <!-- If these are not set then its the actual phone who is choosing to do vad not FreeSWITCH so disable it in your phone. any problems you are having in this regard are 100% to do with who you are calling being only partially implemented to carry voice over ip. On Mon, Sep 20, 2010 at 5:34 PM, JRichey <jrichey(a)itltd.net> wrote: > The gateway sending calls to the FreeSwitch box ends up dropping the calls. > There are also some pops and clicks that aren't there when playing a > continuous audio file. > > > > -----Original Message----- > From: freeswitch-users-bounces(a)lists.freeswitch.org > [mailto:freeswitch-users-bounces@lists.freeswitch.org]On Behalf Of Brian > West > Sent: Monday, September 20, 2010 12:23 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] sip stack problem in fsw > > > You shouldn't see RTP between files when sleeping... its 100% valid to NOT > send any media when their is NOTHING to be sent. Does your phone freak out > or something? > > /b > > On Sep 20, 2010, at 2:08 PM, JRichey wrote: > >> I'm in a similar situation where I need to disable VAD because the other > end >> will disconnect the call if the RTP stops. I've tried the setting Anthony >> suggested as well as several others I've found on the wiki, but I've not >> been able to get it to work. The variables I've been trying are listed >> below. >> >> suppress-cng >> suppress_cng >> rtp_disable_vad_out >> rtp_disable_vad_in >> bridge_generate_comfort_noise >> send_silence_when_idle >> >> >> For my test calls I'm just playing back sound files with sleep in-between >> them and I see no RTP when calling sleep. I'm using tcpdump and wireshark >> to analyze the calls. >> >> >> -Justin Richey >> jrichey(a)ITLtd.net > > > _______________________________________________ > 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 > > > > _______________________________________________ > 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 > -- 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 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 _______________________________________________ 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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Compile Error
by Nathaniel L Keeling 21 Sep '10

21 Sep '10

I am trying to compile FreeSwitch on a Solaris 10 sparc machine and I am getting this error: Compiling src/switch_core_sqldb.c ... Compiling src/switch_core_session.c ... src/switch_core_session.c: In function `switch_core_session_thread': src/switch_core_session.c:1193: warning: long int format, switch_size_t arg (arg 8) src/switch_core_session.c:1203: warning: long int format, switch_size_t arg (arg 8) src/switch_core_session.c:1221: warning: long int format, switch_size_t arg (arg 8) gmake[1]: *** [libfreeswitch_la-switch_core_session.lo] Error 1 gmake: *** [all] Error 2 I looked in the source directory and I was able to see this file; "libfreeswitch_la-switch_core_session.loT" Any help would be appreciated. Thanks Nathaniel

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Conference/intercom/paging
by Jon Young 21 Sep '10

21 Sep '10

How efficient is the conference module for FS? I'm trying to understand capacities, like how many call legs in a single conference? If all but one are muted (in the conference bridge) is it still a heavy burden if 25, 50 or 100 users are in the conference? Does it mix only active talkers or all? Is there a design document or something that will help me understand conferencing better? Thanks, Jon

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Re: [Freeswitch-users] G723-Wrong ptime
by Anthony Minessale 21 Sep '10

21 Sep '10

Turn off scrooge mode Define it as g723_1@30i in the config On Sep 18, 2010 8:27 AM, "gamalsa" <gamalsa(a)hotmail.com> wrote: I don't get any message from FS, i just hear a distorted voice when using g.723. and from the debug output i see that freeswitch compares 120 ms p-time to 20 ms p-time on the other end (and chooses 120ms), while the other end is using p-time 30 ms. It is strange because g.723 uses p-time 30 ms by default. Is there any way to force FS to use g723 with p-time 30 ms with this specific provider? This problem doesn't exist when using g729 codec. I'm using the last svn of freeswitch. thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/G723-Wrong-ptime-tp5543130p55… Sent from the freeswitch-users mailing list archive at Nabble.com. ________________________________...

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Usage of optional argument -bf ?
by Patrick Lists 21 Sep '10

21 Sep '10

Hi all, In the output of freeswitch -h I noticed there is a (for me) new argument -bf that FreeSWITCH can be started with. Is -bf supposed to work in combination with -nc where it will block until it has fully started and then goes into the background? When I start FreeSWITCH (git from yesterday) as follows: /opt/freeswitch/bin/freeswitch -u freeswitch -g freeswitch -core -bf -nc then FreeSWITCH does start, goes into the background but I can not connect to it with fs_cli on the same host. When I omit -bf from the line above then all is well and I can connect with fs_cli. I'm interested in the blocking part because when you do # service freeswitch start right now it will immediately return [ OK ] but I prefer to get an ok after FreeSWITCH actually has fully started. Did I misunderstood the -bf arg? Thanks! Patrick

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ITSP inbound ok but cannot complete outbound call
by Mario 21 Sep '10

21 Sep '10

For this ITSP, registration is ok and inbound calls work. But when making an outbound call the call is sent to the itsp but then gets a fast busy. From what I see in the trace the itsp uses asterisk. I sent a trace in a previous email but it shows nothing I could see missing or in error. The itsp sent the config info below but I have been using them without a STUN server fine on my old PBX. Is there anything that looks like it is out of sync with defaults (like port?). Also, the SIP and DP are below but since the call is sent the DP must be ok. Thanks for any help/suggestions! >From the ITSP:> Sip Proxy - sip.voipyourphone.com STN Server - stun.voipyourphone.com Registration Port - any one port between 5060 and 5069 DTMF port 5004 RTP ports 16384 to 16482 Audio Codecs G711u (account default), G711a, G729a and iLbc DTMF rfc2833 (default), inband, SIP-INFO SIP: <gateway name="uid1"> <param name="proxy" value="sip.voipyourphone.com"/> <param name="username" value="uid1"/> <param name="from-user" value="uid1"/> <param name="password" value="pwd1"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> <param name="retry-seconds" value="30"/> DP <extension name="local.uid1"> <condition field="${toll_allow}" expression="local"/> <condition field="destination_number" expression="^9(\d{7})$"> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="bridge" data="sofia/gateway/uid1/1${default_areacode}$1(a)sip.voipyourlife.com"/> </condition> </extension>

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Routing DID to a PSTN Number
by Aloysius Lloyd 21 Sep '10

21 Sep '10

Hello, I have the following Dial plan for Routing DID to a PSTN Number and it is working. <extension name="voipms_6471231234"> <condition field="destination_number" expression="^6471231234$"> <action application="answer"/> <action application="bridge" data="sofia/gateway/voipms/14161231234"/> <action application="hangup"/> </condition> </extension> Is there anyway to route the call without answer ( <action application="answer"/>) Any help is appreciated. Thanks Lloyd

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