November 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

voice bbs using voicemail mod
by Rodney 18 Nov '11

18 Nov '11

windows server 2008 32 bit. I am trying to create a public voice bulletin board system that would let users on the ivr record messages and play the last 100 in a last in first out mode. I believe i can do this by using voicemail mod. is there a way to add "profiles" to voicemail mod like we do for conferences? if so then i would be able to have regular voicemail options and a separate voice bbs profile that would manage me creating a single or more voice bbs voicemail box that acts differently and doesn't allow public users to delete messages on the bbs box. if this isn't the case, then would it work to create a voice bbs mod based on voicemail mod and have it point to a voicebbs.conf.xml like voicemail.conf.xml does? if so, how do i create/edit this dll file in windows to do such a thing? how do i get ahold of the creators of voicemail mod for maybe an edit? thanks.

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Freeswitch sends BYE, Receives 200 OK, but keeps sending BYE's ?!
by Raimund Sacherer 18 Nov '11

18 Nov '11

Hello, I have a rather problematic provider with which I am stuck in Brasil. After wrestling around I am able to make calls, but when FreeSWITCH send's it's BYE, I do receive a 200 OK, but it seems that FreeSWITCH does not recognize what it receives and keeps sending BYE's with Answers from the remote party that the call no longer exists. Looking at the Sip trace and the debug log (attached as txt file) everything *seems* fine ... but I am not a FS developer either, so I would appreciate if someone with more experience can have a look at it. When connecting with a Eyebeam or X-Lite this problem does not exist, but I could not tell a real difference from the tcp-dumps other than X-Lite uses TCP instead of UPD. Trying to use TCP in FreeSWITCH times out after receiving a 100 Giving a try messages after the Invite, so not sure what to make of this either ..., but I would settle for a working BYE and move on ... Thank you, Best Ray IRC:Hatrix76, Hatrix, Hatrix|Away

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Re: [Freeswitch-users] Freeswitch sends BYE, Receives 200 OK, but keeps sending BYE's ?!
by Raimund Sacherer 18 Nov '11

18 Nov '11

Hello Brian, AFAICT the tags and the Call-ID and the Cseq method are correct in the Trace, therefore, if someone with more indepth knowledge could have a look on the (in the first email) atteched fs+log with siptrace I would be glad to hear what's exactly the problem, I did not open a JIRA because I do not think it's an FS bug. In any case here is a PB link to the same log+trace: http://pastebin.freeswitch.org/17816 best Ray ----- Original Message ----- From: "Brian West" <brian(a)freeswitch.org> To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Cc: "freeswitch-users" <freeswitch-users(a)lists.freeswitch.org> Sent: Friday, November 18, 2011 1:16:38 PM Subject: Re: [Freeswitch-users] Freeswitch sends BYE, Receives 200 OK, but keeps sending BYE's ?! That would indicate the dialog is not matching or the network path is not right! /b Sent from my iPhone On Nov 18, 2011, at 4:02 AM, Raimund Sacherer < raimund.sacherer(a)logitravel.com > wrote: Hello, I have a rather problematic provider with which I am stuck in Brasil. After wrestling around I am able to make calls, but when FreeSWITCH send's it's BYE, I do receive a 200 OK, but it seems that FreeSWITCH does not recognize what it receives and keeps sending BYE's with Answers from the remote party that the call no longer exists. Looking at the Sip trace and the debug log (attached as txt file) everything *seems* fine ... but I am not a FS developer either, so I would appreciate if someone with more experience can have a look at it. When connecting with a Eyebeam or X-Lite this problem does not exist, but I could not tell a real difference from the tcp-dumps other than X-Lite uses TCP instead of UPD. Trying to use TCP in FreeSWITCH times out after receiving a 100 Giving a try messages after the Invite, so not sure what to make of this either ..., but I would settle for a working BYE and move on ... Thank you, Best Ray IRC:Hatrix76, Hatrix, Hatrix|Away <blockquote> <surfix_trace_and_log.txt> </blockquote> <blockquote> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 </blockquote> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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] Freeswitch sends BYE, Receives 200 OK, but keeps sending BYE's ?!
by Raimund Sacherer 18 Nov '11

18 Nov '11

I read up on the RFC regarding that and it states that only the METHOD (BYE) has to be the same, only for REGISTERS the CSeq has to be the same, so If I interpret the RFC correctly (which does not have to be the case) this should not be the issue. best Ray ----- Original Message ----- From: "Joegen Baclor" <jbaclor(a)ezuce.com> To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Sent: Friday, November 18, 2011 12:46:28 PM Subject: Re: [Freeswitch-users] Freeswitch sends BYE, Receives 200 OK, but keeps sending BYE's ?! CSeq in the 200 Ok is wrong. It should match the CSeq of the bye request. Should be CSeq: 20466239 BYE On Fri, Nov 18, 2011 at 6:02 PM, Raimund Sacherer < raimund.sacherer(a)logitravel.com > wrote: Hello, I have a rather problematic provider with which I am stuck in Brasil. After wrestling around I am able to make calls, but when FreeSWITCH send's it's BYE, I do receive a 200 OK, but it seems that FreeSWITCH does not recognize what it receives and keeps sending BYE's with Answers from the remote party that the call no longer exists. Looking at the Sip trace and the debug log (attached as txt file) everything *seems* fine ... but I am not a FS developer either, so I would appreciate if someone with more experience can have a look at it. When connecting with a Eyebeam or X-Lite this problem does not exist, but I could not tell a real difference from the tcp-dumps other than X-Lite uses TCP instead of UPD. Trying to use TCP in FreeSWITCH times out after receiving a 100 Giving a try messages after the Invite, so not sure what to make of this either ..., but I would settle for a working BYE and move on ... Thank you, Best Ray IRC:Hatrix76, Hatrix, Hatrix|Away _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 -- Logitravel.com Raimund Sacherer Sistemas Agencia de Viajes Online www.logitravel.com Edificio Logitravel, Parcela 3B (Parc Bit) Ctra. Palma - Valldemossa km 7,4 | 07121 Palma de Mallorca Tel 902 366 847 | Fax 971 213 495 Síguenos en: Facebook de Logitravel Twitter de Logitravel Blog de Logitravel Logitravel en Youtube Logitravel en Foursquare Descarga nuestras aplicaciones para móvil Logitravel.com Este correo electrónico y, en su caso, cualquier fichero anexo, contiene información de carácter confidencial exclusivamente dirigida a su destinatario. Queda prohibida su divulgación, copia o distribución a terceros sin la previa autorización escrita de LOGITRAVEL S.L.. En caso de haber recibido este correo electrónico por error, se ruega notifíquese inmediatamente esta circunstancia mediante reenvío a la dirección electrónica del remitente. Al mismo tiempo LA EMPRESA le recuerda que sus datos forman o formarán parte de un fichero registrado como CLIENTES con número de inscripción 2070610043 en la Agencia General de Protección de Datos, propiedad de la empresa LOGITRAVEL, con domicilio en Edificio Logitravel, Ctra. Palma - Valldemosa km 7,4, Parc Bit, Palma de Mallorca. Usted tiene derecho de acceso, oposición, rectificación y cancelación a estos datos que deberá ejercer mediante escrito a la dirección anteriormente citada.

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iptables dropping SIP packets..?
by Avi Marcus 18 Nov '11

18 Nov '11

Service: sip (udp/5060) (nf_ct_sip: dropping packet) - 9 packets I see this every few days - it's via an IP with a Yealink T20p phone. I've tried looking this up in the past.. but I don't understand what rules would be dropping this, and what is being dropped. Any clues on how to investigate? -Avi

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dial plan help
by Elliott Vogel 18 Nov '11

18 Nov '11

Does anyone has an example on how to setup a normalization rule in the dial plan? basically I want to transform 7 digit number into to 10 digit number and then continue processing the dial plan

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Authenticate text messages
by Mario Karakanovski 18 Nov '11

18 Nov '11

Hi everyone, Does somebody know how to do authentication for text messages? I try to set auth-all-packets=true in the sofia profile, but without success. All help is appreciated!

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Polycom phone registration problem
by Elliott Vogel 18 Nov '11

18 Nov '11

Hello, I'm trying to setup freeswitch and I'm having a problem with a polycom phone registering. The phone is behind nat but the freeswitch is on a public network can anyone tell me if I don't have something configured correctly or where should look for the problem? <include> <domain name="zoogmedia.com"> <params> <param name="dial-string" value="{sip_invite_domain=${dialed_domain},presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> </params> <variables> <variable name="record_stereo" value="true"/> <variable name="default_gateway" value="$${default_provider}"/> <variable name="default_areacode" value="312"/> <variable name="transfer_fallback_extension" value="operator"/> </variables> <groups> <group name="default"> <users> <user id="14149823263"> <params> <param name="password" value="123456"/> </params> <variables> <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="1000"/> <variable name="user_context" value="default"/> <variable name="outbound_caller_id_name" value="Sales" /> <variable name="outbound_caller_id_number" value="+14149823263" /> </variables> </user> </users> </group> </groups> </domain> </include> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/>  <param name="debug-presence" value="0"/>  </global_settings> <profiles> <profile name="endpoints"> <gateways> <gateway name="TNG"> <param name="username" value="user"/> <param name="password" value="password"/> <param name="proxy" value="69.25.128.195:5060"/> <param name="from-domain" value="$${local_ip_v4}:5060"/> <param name="dtmf-type" value="rfc2833"/> <param name="extension-in-contact" value="true"/> <param name="caller-id-in-from" value="true"/> <param name="register" value="false"/> </gateway> </gateways> <aliases> </aliases> <domains> <domain name="all" alias="false" parse="true"/> </domains> <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="sip-capture" value="no"/> <param name="rtp-ip" value="$${external_rtp_ip}"/> <param name="sip-ip" value="$${external_sip_ip}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="sip-port" value="5070"/> <param name="tls" value="$${external_ssl_enable}"/> <param name="tls-bind-params" value="transport=tls"/> <param name="tls-sip-port" value="$${external_tls_port}"/> <param name="tls-cert-dir" value="$${external_ssl_dir}"/> <param name="tls-version" value="$${sip_tls_version}"/> <param name="rfc2833-pt" value="101"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="dtmf-duration" value="2000"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="rtp-timer-name" value="soft"/> <param name="local-network-acl" value="localnet.auto"/> <param name="manage-presence" value="false"/> <param name="inbound-codec-negotiation" value="greedy"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="rtp-rewrite-timestamps" value="true"/> <param name="track-calls" value="true"/> </settings> </profile> </profiles> </configuration> <?xml version="1.0" encoding="utf-8"?>  <include> <context name="default"> <extension name="unloop"> <condition field="${unroll_loops}" expression="^true$"/> <condition field="${sip_looped_call}" expression="^true$"> <action application="deflect" data="${destination_number}"/> </condition> </extension> <extension name="check_auth" continue="true"> <condition field="${sip_authorized}" expression="^true$" break="never"> <anti-action application="respond" data="407"/> </condition> </extension> <extension name="local"> <condition field="${toll_allow}" expression="local" /> <condition field="destination_number" expression="^([0-9]{7})$"> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}" /> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}" /> <action application="bridge" data="sofia/gateway/TNG/+1${default_areacode}$1"/> </condition> </extension> <extension name="domestic"> <condition field="${toll_allow}" expression="domestic" /> <condition field="destination_number" expression="^(1{0,1}\d{10})$"> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}" /> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}" /> <action application="bridge" data="sofia/gateway/TNG/+$1"/> </condition> </extension> <extension name="international"> <condition field="${toll_allow}" expression="international" /> <condition field="destination_number" expression="^011?(\d+)$"> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}" /> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}" /> <action application="bridge" data="sofia/gateway/TNG/+$1"/> </condition> </extension> </context> </include>

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Re: [Freeswitch-users] conference room announce
by Rodney 18 Nov '11

18 Nov '11

the conference name is "conference room1" in the dialplan. should i be naming it just a number? ________________________________ From: "freeswitch-users-request(a)lists.freeswitch.org" <freeswitch-users-request(a)lists.freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, November 17, 2011 5:36 PM Subject: FreeSWITCH-users Digest, Vol 65, Issue 133 ----- Forwarded Message ----- Send FreeSWITCH-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: 1. Re: call count bind digit on static conference (Michael Collins) 2. Re: a=fmtp missing on FS answer (Brian West) 3. Re: calling limit_usage via ESL (Avi Marcus) 4. Re: Timing issues in AWS? (Anthony Minessale) Conference conference room 1-127.0.0.1 (1 member rate:8000)13; sofia/external/2064200990@66.54.140.446;b11d94a6-94ff-4955-bc87-13e332a7443;unknown;2064200990;hear:speak:floor;0;0;0;50 > >I try to "conference conference room 1-127.0.0.1" > >I get response "conference not found" > > only one "conference" - like this: conference <conf_name> list or conference 1-127.0.0.1 list -MC {sip_append_audio_sdp=a=fmtp:18 annexb=no}sofia/internal/foo(a)bar.com Or some variation of that? /b On Nov 17, 2011, at 3:05 PM, Camila Troncoso wrote: This isnt the solution because it only appends rtpmap not fmtp, I tried it. > > > >*From:* freeswitch-users-bounces(a)lists.freeswitch.org [mailto: >freeswitch-users-bounces(a)lists.freeswitch.org] *On Behalf Of *Avi Marcus >*Sent:* jueves, 17 de noviembre de 2011 18:01 >*To:* FreeSWITCH Users Help >*Subject:* Re: [Freeswitch-users] a=fmtp missing on FS answer > > > >Anthony said you probably want this: > >http://wiki.freeswitch.org/wiki/Variable_verbose_sdp > > >-Avi > > > >On Thu, Nov 17, 2011 at 10:24 PM, Camila Troncoso <ctroncoso(a)redvoiss.net> >wrote: > >I´m trying to make a call trough FS with G729 annexb=no in one side and >G729 annexb=no ( the same) in the other. I have a sangoma transcoding card >installed and configured, all is working fine. I use this card because in >some cases I need transcoding . So I have late-negotiation=true and >disable-transcoding=false. The problem is that FS answer without a=fmtp as >show in the picture. Freeswitch version is 1.0 > > > >*Error! Filename not specified.**Error! Filename not specified.* > > > > > > > >*From:* freeswitch-users-bounces(a)lists.freeswitch.org [mailto: >freeswitch-users-bounces(a)lists.freeswitch.org] *On Behalf Of *Brian West >*Sent:* jueves, 17 de noviembre de 2011 16:18 >*To:* FreeSWITCH Users Help >*Subject:* Re: [Freeswitch-users] a=fmtp missing on FS answer > > > >what exactly are you doing and what are the sip traces? What firmware rev? > > > >/b > > > >On Nov 17, 2011, at 12:52 PM, Camila Troncoso wrote: > > > >sip_append_audio_sdp > > > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting(a)freeswitch.org >http://www.freeswitchsolutions.com > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server >http://www.cudatel.com > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >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 >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting(a)freeswitch.org >http://www.freeswitchsolutions.com > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server >http://www.cudatel.com > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >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 > no using "execute" it hangs. Can I craft/view ESL stuff raw somewhere other than telnet? I recall hitting ctrl-C and killing freeswitch in telnet.. -Avi On Thu, Nov 17, 2011 at 6:19 PM, Hector Geraldino <Hector.Geraldino(a)ip-soft.net> wrote: > What about: "execute" + command + " " + arguments > > From: freeswitch-users-bounces(a)lists.freeswitch.org > [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Avi > Marcus > Sent: Wednesday, November 16, 2011 6:20 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] calling limit_usage via ESL > > > > The wiki says it's an API also. I have frequently executed it directly in > fs_cli. > > When I try it without "api" at the start, it hangs. > > -Avi > > > > On Thu, Nov 17, 2011 at 1:00 AM, Hector Geraldino > <Hector.Geraldino(a)ip-soft.net> wrote: > > I think that, if it’s an application (i.e. can be executed directly from the > dialplan), you shouldn’t use the ‘api’ prefix. > > > > I’m not familiar with the php esl module, but in the Java ESL there’s a > difference when you try to call an api command vs and application: > > > > API: > > "api " + command + " " + arguments > > > > APP: > > SendMsg message = new SendMsg(uuid); > > message.addCallCommand("execute"); > > message.addExecuteAppName(appName); > > message.addExecuteAppArg(arguments); > > > > Also, you can always inspect the execution result if you’re executing the > command in sync mode. > > > > From: freeswitch-users-bounces(a)lists.freeswitch.org > [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Avi > Marcus > Sent: Wednesday, November 16, 2011 5:41 PM > To: FreeSWITCH Users Help > Subject: [Freeswitch-users] calling limit_usage via ESL > > > > I'm using the php esl implementation in fusionpbx.. I'm able to pass "api > show channels count" and "api show calls count" and the like, but when I do: > > "api limit_usage hash origination usa_pd" it's just blank. > > Pasting "limit_usage hash origination usa_pd" into fs_cli yields 0 (or > higher as the case may be). > > > > There DOES seem to be a \n returned, but nothing else. I've got no 'friggin > clue. > > Help, please? > > Thanks, > > Avi > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > the "amazon linux" works well with modern GIT revisions. older ones need a centos kernel to work well. On Thu, Nov 17, 2011 at 9:19 AM, Michael Picher <mpicher(a)gmail.com> wrote: Huh... maybe I need to get my own CentOS build up there instead of rely on Rightscale's 64 bit image... That's a whole other ball of wax I was trying to avoid having to figure out to do... > >Mike > > > >On Thu, Nov 17, 2011 at 8:49 AM, Stephen Dame <sdame(a)207me.com> wrote: > >I’m running freeswitch on about 40 different m1.small c1.medium’s in AWS regions, us-east, us-west, eu-west, and asia…. They are used in videoconf component of BigBlueButton.org . For most part they work great… There are occasional issues with voip quality but the app is 100% voip, with the BBB client all browser based. So the conference is subject to every ones local network connections and most issues “blamed” on the internet instead of freeswitch J We also tie in skype and DID direct to improve latency for some clients. But expectations are set so we meet them. >> >>You probably have tougher business conferencing clients that want perfect audio. But these are production deployed and generate revenue. All these are on Ubuntu 10.04 official amis with no mods to kernels. >> >>I’m sure there are tweaks that can be made, and bare metal solutions that would work a little better… Does anyone have any ideas how to optimize a Ubuntu instance. I would engage in a few hours of consulting is so. >> >>The tests below are interesting but above my paygrade to understand what they mean? >> >>I’m running them but don’t have a clue how to interpret. If you want a simple of real running data, would be glad to run sample tests on these distributed servers and provide back for analysis. >> >>Regards, >>Stephen >> >>From:freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Picher >>Sent: Thursday, November 17, 2011 8:03 AM >>To: FreeSWITCH Users Help >>Subject: Re: [Freeswitch-users] Timing issues in AWS? >> >>Not having anywhere near the same trouble in a Xen Server in my lab... >> >>Although test_time is skewed 180 degrees from what it was in AWS.... >> >>freeswitch@internal> timer_test 120 10 >>Avg: 119.945ms Total Time: 1199.727ms >> >>2011-11-17 07:57:22.721105 [CONSOLE] mod_commands.c:461 Timer Test: samplecount after init: 1 >>2011-11-17 07:57:22.751026 [CONSOLE] mod_commands.c:466 Timer Test: samplecount after first step: 2 >>2011-11-17 07:57:22.872047 [CONSOLE] mod_commands.c:475 Timer Test: 1 sleep 120 120993 >>2011-11-17 07:57:22.990983 [CONSOLE] mod_commands.c:475 Timer Test: 2 sleep 120 118914 >>2011-11-17 07:57:23.110953 [CONSOLE] mod_commands.c:475 Timer Test: 3 sleep 120 119919 >>2011-11-17 07:57:23.230928 [CONSOLE] mod_commands.c:475 Timer Test: 4 sleep 120 120032 >>2011-11-17 07:57:23.350831 [CONSOLE] mod_commands.c:475 Timer Test: 5 sleep 120 119802 >>2011-11-17 07:57:23.470886 [CONSOLE] mod_commands.c:475 Timer Test: 6 sleep 120 119999 >>2011-11-17 07:57:23.590873 [CONSOLE] mod_commands.c:475 Timer Test: 7 sleep 120 119975 >>2011-11-17 07:57:23.710893 [CONSOLE] mod_commands.c:475 Timer Test: 8 sleep 120 120003 >>2011-11-17 07:57:23.830774 [CONSOLE] mod_commands.c:475 Timer Test: 9 sleep 120 119849 >>2011-11-17 07:57:23.950808 [CONSOLE] mod_commands.c:475 Timer Test: 10 sleep 120 119971 >> >>freeswitch@internal> time_test 600 10 >>test 1 sleep 600 1 >>test 2 sleep 600 1 >>test 3 sleep 600 0 >>test 4 sleep 600 1 >>test 5 sleep 600 0 >>test 6 sleep 600 1 >>test 7 sleep 600 1 >>test 8 sleep 600 0 >>test 9 sleep 600 0 >>test 10 sleep 600 1 >>avg 0 >> >>[root@openuc bin]# uname -r >>2.6.18-274.7.1.el5 >> >>[root@openuc bin]# grep CONFIG_HZ /boot/config-* >>/boot/config-2.6.18-238.12.1.el5:# CONFIG_HZ_100 is not set >>/boot/config-2.6.18-238.12.1.el5:# CONFIG_HZ_250 is not set >>/boot/config-2.6.18-238.12.1.el5:CONFIG_HZ_1000=y >>/boot/config-2.6.18-238.12.1.el5:CONFIG_HZ=1000 >>/boot/config-2.6.18-238.19.1.el5:# CONFIG_HZ_100 is not set >>/boot/config-2.6.18-238.19.1.el5:# CONFIG_HZ_250 is not set >>/boot/config-2.6.18-238.19.1.el5:CONFIG_HZ_1000=y >>/boot/config-2.6.18-238.19.1.el5:CONFIG_HZ=1000 >>/boot/config-2.6.18-238.9.1.el5:# CONFIG_HZ_100 is not set >>/boot/config-2.6.18-238.9.1.el5:# CONFIG_HZ_250 is not set >>/boot/config-2.6.18-238.9.1.el5:CONFIG_HZ_1000=y >>/boot/config-2.6.18-238.9.1.el5:CONFIG_HZ=1000 >>/boot/config-2.6.18-274.3.1.el5:# CONFIG_HZ_100 is not set >>/boot/config-2.6.18-274.3.1.el5:# CONFIG_HZ_250 is not set >>/boot/config-2.6.18-274.3.1.el5:CONFIG_HZ_1000=y >>/boot/config-2.6.18-274.3.1.el5:CONFIG_HZ=1000 >>/boot/config-2.6.18-274.7.1.el5:# CONFIG_HZ_100 is not set >>/boot/config-2.6.18-274.7.1.el5:# CONFIG_HZ_250 is not set >>/boot/config-2.6.18-274.7.1.el5:CONFIG_HZ_1000=y >>/boot/config-2.6.18-274.7.1.el5:CONFIG_HZ=1000 >> >>Thoughts as to why AWS results are so different from XenServer? Other than not knowing who else is on the AWS box? >> >>Thanks, >> Mike >> >> >> >>On Wed, Nov 16, 2011 at 11:56 AM, Michael Picher <mpicher(a)gmail.com> wrote: >>m1.large >> >>I have a c1.xlarge queued up and ready to test... >> >>On Wed, Nov 16, 2011 at 11:52 AM, Chris Chen <chris.chen2004(a)gmail.com> wrote: >>Just a simple question, what kind of AWS instance are you running your FreeSWITCH? >>It makes huge difference. >>Thanks, >>Chris >>On Wed, Nov 16, 2011 at 10:43 AM, Michael Picher <mpicher(a)gmail.com> wrote: >>Hi guys, >>> >>>Trying to get to the bottom of some conference bridge issues I'm having with running the system in AWS. >>> >>>We're hearing a bunch of snap-crackle-pops in conference bridges and when I tcpdum on the server itself I see them in the RTP and see RTP timestamp problems. >>> >>>I've run the following: >>> >>> >>>timer_test >>>freeswitch@127.0.0.1@internal> timer_test 120 10 >>>Avg: 120.004ms Total Time: 1200.315ms >>> >>>2011-11-16 10:25:56.121131 [CONSOLE] mod_commands.c:461 Timer Test: samplecount after init: 1 >>>freeswitch@127.0.0.1@internal> 2011-11-16 10:25:56.219163 [CONSOLE] mod_commands.c:466 Timer Test: samplecount after first step: 2 >>>2011-11-16 10:25:56.339195 [CONSOLE] mod_commands.c:475 Timer Test: 1 sleep 120 120006 >>>2011-11-16 10:25:56.459259 [CONSOLE] mod_commands.c:475 Timer Test: 2 sleep 120 120040 >>>2011-11-16 10:25:56.579259 [CONSOLE] mod_commands.c:475 Timer Test: 3 sleep 120 119976 >>>2011-11-16 10:25:56.699291 [CONSOLE] mod_commands.c:475 Timer Test: 4 sleep 120 120007 >>>2011-11-16 10:25:56.819318 [CONSOLE] mod_commands.c:475 Timer Test: 5 sleep 120 120001 >>>2011-11-16 10:25:56.939375 [CONSOLE] mod_commands.c:475 Timer Test: 6 sleep 120 120031 >>>2011-11-16 10:25:57.059397 [CONSOLE] mod_commands.c:475 Timer Test: 7 sleep 120 119996 >>>2011-11-16 10:25:57.179422 [CONSOLE] mod_commands.c:475 Timer Test: 8 sleep 120 119994 >>>2011-11-16 10:25:57.299461 [CONSOLE] mod_commands.c:475 Timer Test: 9 sleep 120 120005 >>>2011-11-16 10:25:57.419478 [CONSOLE] mod_commands.c:475 Timer Test: 10 sleep 120 119991 >>> >>>test_time >>>freeswitch@127.0.0.1@internal> time_test 600 10 >>> >>>test 1 sleep 600 1592 >>>test 2 sleep 600 986 >>>test 3 sleep 600 1018 >>>test 4 sleep 600 980 >>>test 5 sleep 600 1005 >>>test 6 sleep 600 1000 >>>test 7 sleep 600 972 >>>test 8 sleep 600 990 >>>test 9 sleep 600 1006 >>>test 10 sleep 600 994 >>>avg 1054 >>> >>>For kernel: >>>[root@openuc bin]# uname -r >>> >>>2.6.21.7-2.fc8xen >>> >>>CONFIG_HZ: >>>[root@openuc bin]# grep CONFIG_HZ /boot/config-* >>> >>>/boot/config-2.6.16.33-xenU-x86_64:CONFIG_HZ_100=y >>>/boot/config-2.6.16.33-xenU-x86_64:# CONFIG_HZ_250 is not set >>>/boot/config-2.6.16.33-xenU-x86_64:# CONFIG_HZ_1000 is not set >>>/boot/config-2.6.16.33-xenU-x86_64:CONFIG_HZ=100 >>>/boot/config-2.6.18-164.15.1.el5.centos.plus:# CONFIG_HZ_100 is not set >>>/boot/config-2.6.18-164.15.1.el5.centos.plus:# CONFIG_HZ_250 is not set >>>/boot/config-2.6.18-164.15.1.el5.centos.plus:CONFIG_HZ_1000=y >>>/boot/config-2.6.18-164.15.1.el5.centos.plus:CONFIG_HZ=1000 >>>/boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_100 is not set >>>/boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_250 is not set >>>/boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_300 is not set >>>/boot/config-2.6.21-2952.fc8xen:CONFIG_HZ_1000=y >>>/boot/config-2.6.21-2952.fc8xen:CONFIG_HZ=1000 >>>are the xenU kernel settings screwing me here? >>> >>>Thanks, >>> Mike >>> >>> >>>-- >>>There are 10 kinds of people in this world, those who understand binary and those who don't. >>> >>>mpicher(a)gmail.com >>>blog: http://www.sipxecs.info >>>call: sip:mpicher@sipxecs.info >>>_________________________________________________________________________ >>>Professional FreeSWITCH Consulting Services: >>>consulting(a)freeswitch.org >>>http://www.freeswitchsolutions.com >>> >>>FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>http://www.cudatel.com >>> >>>Official FreeSWITCH Sites >>>http://www.freeswitch.org >>>http://wiki.freeswitch.org >>>http://www.cluecon.com >>> >>>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 >> >> >>_________________________________________________________________________ >>Professional FreeSWITCH Consulting Services: >>consulting(a)freeswitch.org >>http://www.freeswitchsolutions.com >> >>FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>http://www.cudatel.com >> >>Official FreeSWITCH Sites >>http://www.freeswitch.org >>http://wiki.freeswitch.org >>http://www.cluecon.com >> >>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 >> >> >> >>-- >>There are 10 kinds of people in this world, those who understand binary and those who don't. >> >>mpicher(a)gmail.com >>blog: http://www.sipxecs.info >>call: sip:mpicher@sipxecs.info >> >> >> >>-- >>There are 10 kinds of people in this world, those who understand binary and those who don't. >> >>mpicher(a)gmail.com >>blog: http://www.sipxecs.info >>call: sip:mpicher@sipxecs.info >>_________________________________________________________________________ >>Professional FreeSWITCH Consulting Services: >>consulting(a)freeswitch.org >>http://www.freeswitchsolutions.com >> >>FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>http://www.cudatel.com >> >>Official FreeSWITCH Sites >>http://www.freeswitch.org >>http://wiki.freeswitch.org >>http://www.cluecon.com >> >>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 >> >> > > >-- >There are 10 kinds of people in this world, those who understand binary and those who don't. > >mpicher(a)gmail.com >blog: http://www.sipxecs.info >call: sip:mpicher@sipxecs.info > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting(a)freeswitch.org >http://www.freeswitchsolutions.com > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server >http://www.cudatel.com > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >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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Comfort noise or silence rtp packets during recording
by Joegen Baclor 18 Nov '11

18 Nov '11

I tried asking in the freeswitch irc channel but got no response. I'm trying my luck here. I have recently received a packet capture from the sipx community that seems to indicate that when FreeSwitch is recording a voice mail message via our custom IVR, that it does not send either comfort noise or silence packets back to the caller. This results to the call getting dropped by the ITSP after 30 seconds because of RTP time out. Is there a configurable parameter to make the IVR send some voice activity back? Joegen

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