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Mono 2.8 released and need to update mod_managed
by Yitzchok 12 Oct '10

12 Oct '10

Mono 2.8 was released today with support for C# 4 and more http://www.mono-project.com/Release_Notes_Mono_2.8 <http://www.mono-project.com/Release_Notes_Mono_2.8>It seems like there was some changes in the way applications embed mono (which includes removing the reference to glib and replacing it with eglib from what I understand) in this link you can find information for the changes made with embedding mono http://www.mono-project.com/Embedding_Mono I am trying to see if I can make it work but since I am just a C# windows developer (not a c or c++ dev) what I am doing is just hacking around and don't know if I will even get it to work but I think a c++ linux developer might get it to work with only a little work (the info needed I think is in the link above). So I am wondering if anyone is interested to get it to work. <http://www.mono-project.com/Embedding_Mono> Yitzchok On Wed, Oct 6, 2010 at 11:40 AM, Yitzchok <adminjew(a)gmail.com> wrote: > I tried that and it seems to work. > > > ---diff from git head--- > > > @@ -385,9 +385,7 @@ SWITCH_STANDARD_API(managedrun_api_function) > } else { > stream->write_function(stream, "-ERR ExecuteBackground returned false > (unknown module or exception?).\n"); > } > -#ifndef _MANAGED > - mono_thread_detach(mono_thread_current()); > -#endif > + > return SWITCH_STATUS_SUCCESS; > } > > @@ -403,9 +401,7 @@ SWITCH_STANDARD_API(managed_api_function) > if (!(executeDelegate(cmd, stream, stream->param_event))) { > switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Execute failed > for %s (unknown module or exception).\n", cmd); > } > -#ifndef _MANAGED > - mono_thread_detach(mono_thread_current()); > -#endif > + > return SWITCH_STATUS_SUCCESS; > } > > @@ -421,9 +417,7 @@ SWITCH_STANDARD_APP(managed_app_function) > if (!(runDelegate(data, session))) { > switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Application run > failed for %s (unknown module or exception).\n", data); > } > -#ifndef _MANAGED > - mono_thread_detach(mono_thread_current()); > -#endif > + > } > > SWITCH_STANDARD_API(managedreload_api_function) > @@ -438,9 +432,7 @@ SWITCH_STANDARD_API(managedreload_api_function) > if (!(reloadDelegate(cmd))) { > switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Execute failed > for %s (unknown module or exception).\n", cmd); > } > -#ifndef _MANAGED > - mono_thread_detach(mono_thread_current()); > -#endif > + > return SWITCH_STATUS_SUCCESS; > } > > > > ----- > > Yitzchok > > > > On Wed, Oct 6, 2010 at 10:57 AM, Jeff Lenk <jeff(a)jefflenk.com> wrote: > >> >> Not sure how this makes sense yet. Yitzchok would you try this with only >> the >> mono_thread_detach calls removed leaving the mono_thread_attach in place >> and >> see what happens. >> >> Thanks >> Jeff >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/mod-managed-on-linux-centos-t… >> Sent from the freeswitch-users mailing list archive at Nabble.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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Click to Dial using REFER
by Neil Burgess 12 Oct '10

12 Oct '10

Hi, We are looking at how to implement some "Click to Dial" functionality, and have come across a "REFER based" method documented on several sites, e.g. http://www.tech-invite.com/Ti-sip-service-19.html, which shows the possibility of a Switch or some other agent sending a SIP REFER to a Client which causes that client to make an outbound call to the "referred to" URI. This is also documented elsewhere as:- "First, the SIP server sends an INVITE to one of the phones because phones usually do not accept REFER without prior invitation.The INVITE contains 0.0.0.0 as the IP address in SDP, because there is no remote phone (the message is sent by user agent within the SIP server which does not deal with media). After that, the server sends a REFER method which will ask the phone to send INVITE somewhere else.The URI of the called party is passed to the phone in a Refer-To header field of the REFER method." My question relates to whether or not it is possible to cause FS to issue this form of REFER via the Event Socket interface to implement this kind of click to dial functionality. Rgds, Neil

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bad audio of recorded session
by Piotr Sobolewski 12 Oct '10

12 Oct '10

Hi When conversation of the call is being saved to a file (record_session) audio tempo in such recording is like slowed down despite that call conversations lacks this issue. Above happens when call comes from asterisk. (Asterisk-->FS-->UA) WRONG If the call comes from IP-Phone, recording is normal. (UA-->FS-->Asterisk) OK I've tried multiple options for example: <action application="export" data="rtp_rewrite_timestamps=true"/> <action application="set" data="bridge_generate_comfort_noise=true"/> And setting <param name="rtp-timer-name" value="none"/> in the sofia profiles didn't solved the problem either. Global Codecs definition. <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU@20i,PCMA@20i,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU@20i,PCMA@20i,GSM"/> Is this known issue? -- Piotr Sobolewski sobolewski(a)gmail.com

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Newbie (sort of) questions
by Martin Joseph 12 Oct '10

12 Oct '10

Hi Again FreeSWITCHers, I am building a new setup for my "new" house (I have been remodeling for 5 years). I have build FS via GIT on my "new" xServe hardware (dual Xeon 2Ghz) which I was able to do pretty easily (Hurray!). In the past I have struggled with updating my older FS install on my anemic g3/500 ibook install which is still running by the way. So, now I am thinking about configuring my new installation, and I being no expert am not looking forward to figuring this all out again. My installation is overall pretty simple with a gateway to POTS (audiocodes MP114) and several Voip providors, one of whom is hopefully still in business (heh). Anyhow, my first request is for GUI based FS configuration systems? These could run on my OSX box (via X11 or Cocoa), or be some kind of web based thing. It doesn't need to be super customizable, but not having to find/edit the XML files would be lovely. I am a mac person after all. If there are no recommended GUI's available, then pointing me to the best sample pages for configuration would also be greatly appreciated... Thanks again for the great software and all of your efforts! Marty PS looking forward to killing me last Asterisk install...

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can't load python scripts
by Riccardo Magliocchetti 12 Oct '10

12 Oct '10

Hello, i'm trying to do an ivr in python but all i get is: 2010-10-12 12:39:56.053054 [DEBUG] mod_python.c:188 Call python script 2010-10-12 12:39:56.054128 [DEBUG] mod_python.c:191 Finished calling python script 2010-10-12 12:39:56.054128 [ERR] mod_python.c:200 Error calling python script This script fails, even without the import: from freeswitch import * def handler(uuid): return FreeSWITCH is 1.0.6. Any hints? thanks, riccardo

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fs_cli
by babak yakhchali 12 Oct '10

12 Oct '10

Hi is it possible to execute and get result of an api like this fs_cli.exe -H 192.168.11.30 -x "managed MY API" now I'm getting -ERR no reply but it works fine for show calls thanx

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administering a conference
by covici＠ccs.covici.com 12 Oct '10

12 Oct '10

Hi. I would like an alternate means of administering a conference other than the command line -- for me its OK, but other prefer a web interface and te one you use at conference.freeswitch.org would be just fine if I could find all its pieces -- event scripts database tables, whatever. For me, this would help spread fs and not have people say its unusable because of these trivialities. any assistance would be appreciated. -- 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] Setting origination_caller_id_number
by Esser, Holger 12 Oct '10

12 Oct '10

Many many thanks!!! That addressed my issue. Holger ________________________________ From: freeswitch-users-bounces(a)lists.freeswitch.org To: FreeSWITCH Users Help Sent: Mon Oct 11 19:56:48 2010 Subject: Re: [Freeswitch-users] Setting origination_caller_id_number Holger, try adding sip_cid_type=none in your {}. And in your gateway definition, you may add: <param name="caller-id-in-from" value="true"/> David Ponzone Direction Technique email: david.ponzone(a)ipeva.fr<mailto:david.ponzone@ipeva.fr> tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr<BLOCKED::http://www.ipeva.fr/> - www.ipeva-studio.com<BLOCKED::http://www.ipeva-studio.com/> Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 12/10/2010 à 02:30, Esser, Holger a écrit : I am looking in old postings to the list and it seems that in 2008 people ran across the same issue. In essence, I need a way to manipulate the FROM contact header to match my changing ANI. Currently it pulls it only from the gateway user name in the profile. I a looking in the sofia_glue and it seems that these params are set in switch_channel.c. Before I bang my head on the wall too many times in trying to change the code, has this already been addressed in a manner that I cannot find? It seems that origination_caller_id_number sets only RPID. Is that accurate? Teliax cannot set the ANI based on RPID but needs it in the FROM contact header. Any help would be greatly appreciated. Holger From: freeswitch-users-bounces(a)lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Holger Esser Sent: Friday, October 08, 2010 1:33 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Setting origination_caller_id_number Hi, Whenever I set the origination_caller_is_number in my dial request like this for a local or external context, perl http.pl<http://http.pl> originate {origination_caller_id_number=5xxxxxxxxxsofia/internal/1(a)148.181.145.148<mailto:1@148.181.145.148> 5000 the caller id is set. Whenever I use the gateway, it is no longer working and the username in the gateway becomes the ani number in the SIP trace. perl http.pl<http://http.pl> originate {origination_caller_id_number=972xxxxxxx}sofia/gateway/teliax/4695xxxxxxx 5000 Any ideas? Thx No virus found in this incoming message. Checked by AVG - www.avg.com<http://www.avg.com> Version: 9.0.862 / Virus Database: 271.1.1/3183 - Release Date: 10/08/10 13:34:00 ________________________________ This e-mail transmission may contain information that is proprietary, privileged and/or confidential and is intended exclusively for the person(s) to whom it is addressed. Any use, copying, retention or disclosure by any person other than the intended recipient or the intended recipient's designees is strictly prohibited. If you are the intended recipient, you must treat the information in confidence and in accordance with all laws related to the privacy and confidentiality of such information. If you are not the intended recipient or their designee, please notify the sender immediately by return e-mail and delete all copies of this email, including all attachments. Intervoice, Inc. 17811 Waterview Parkway Dallas, TX 75252 USA Intervoice Limited, Registered in England and Wales with number 2601740, 50 Park Road, Gatley, Cheshire, SK8 4HZ. VAT Number: 560421375 Intervoice GmbH, Hagenauer Straße 55, 65203 Wiesbaden, Sitz der Gesellschaft: Wiesbaden, Handelsregister: HRB 8186 (Amtsgericht Wiesbaden), Geschäftsführer: Wayne Barclay, Steffen Selbmann _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@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 virus found in this incoming message. Checked by AVG - www.avg.com Version: 9.0.862 / Virus Database: 271.1.1/3189 - Release Date: 10/11/10 13:34:00

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Setting origination_caller_id_number
by Holger Esser 12 Oct '10

12 Oct '10

Hi, Whenever I set the origination_caller_is_number in my dial request like this for a local or external context, perl http.pl originate {origination_caller_id_number=5xxxxxxxxxsofia/internal/1(a)148.181.145.1485000 the caller id is set. Whenever I use the gateway, it is no longer working and the username in the gateway becomes the ani number in the SIP trace. perl http.pl originate {origination_caller_id_number=972xxxxxxx}sofia/gateway/teliax/4695xxxxxxx 5000 Any ideas? Thx

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Dead lua scripts
by Tim St. Pierre 11 Oct '10

11 Oct '10

Hello, I was building some lua apps, and clearly went wrong somewhere. I now have several channels in the hangup state, all from the extension I was testing, to the script I was testing. There are lots of errors in the script at this point, so I'm sure it's something I did. However, these channels sent the CPU to 120% (according to top), and I'm not at a load average of 4. Fortunately, Freeswitch handles this rather gracefully and call quality seems to be okay in spite of it. Is there a way to kill these threads without killing the other calls? I tried using uuid_kill, but it came back as no such channel. I can see the channels in "show channels", but the uuids listed don't exist according to uuid_exists. Any suggestions? Thanks! -Tim

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