October 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Routing DIDs to Dynamically Registered SIP User
by David Allen 13 Oct '10

13 Oct '10

Hi, I'm trying to send multiple Direct Indial Numbers down to a dynamically registered SIP User. I need to ensure that both the To and Target URI contain the direct Indial number. I'm able to modify the SIP TO Header of a call that is sent to them like below: <action application="bridge" data="sofia/external/56778977%${domain}^61390009000"/> which sends the request as: ------------------------------------------------------------------------ INVITE sip:56778977@192.168.22.2:5061 SIP/2.0 Via: SIP/2.0/UDP 202.0.155.73;rport;branch=z9hG4bKXeND9cHtgKF2j Max-Forwards: 69 From: "0390001000" <sip:0390001000@192.168.2.200>;tag=K4HHaZ9v1H07Q To: <sip:61390009000@192.168.2.200> But in order to maintain compatability with a number of PBX's/VoIP devices on the market, I need to be able to send the invite to the dynamically registered SIP user, however have it set the Target URI and To as the same contact number like below: ------------------------------------------------------------------------ INVITE sip:61390009000@192.168.22.2:5061 SIP/2.0 Via: SIP/2.0/UDP 202.0.155.73;rport;branch=z9hG4bKXeND9cHtgKF2j Max-Forwards: 69 From: "0390001000" <sip:0390001000@192.168.2.200>;tag=K4HHaZ9v1H07Q To: <sip:61390009000@192.168.22.2:5061> Is there a way to do this setting via variables? I can't seem to find any details for it. Thanks David

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Routing DIDs to Dynamically Registered SIP User
by David Allen 12 Oct '10

12 Oct '10

Hi, I'm trying to send multiple Direct Indial Numbers down to a dynamically registered SIP User. I need to ensure that both the To and Target URI contain the direct Indial number. I'm able to modify the SIP TO Header of a call that is sent to them like below: <action application="bridge" data="sofia/external/56778977%${domain}^61390009000"/> which sends the request as: ------------------------------------------------------------------------ INVITE sip:56778977@192.168.22.2:5061 SIP/2.0 Via: SIP/2.0/UDP 202.0.155.73;rport;branch=z9hG4bKXeND9cHtgKF2j Max-Forwards: 69 From: "0390001000" <sip:0390001000@192.168.2.200>;tag=K4HHaZ9v1H07Q To: <sip:61390009000@192.168.2.200> But in order to maintain compatability with a number of PBX's/VoIP devices on the market, I need to be able to send the invite to the dynamically registered SIP user, however have it set the Target URI and To as the same contact number like below: ------------------------------------------------------------------------ INVITE sip:61390009000@192.168.22.2:5061 SIP/2.0 Via: SIP/2.0/UDP 202.0.155.73;rport;branch=z9hG4bKXeND9cHtgKF2j Max-Forwards: 69 From: "0390001000" <sip:0390001000@192.168.2.200>;tag=K4HHaZ9v1H07Q To: <sip:61390009000@192.168.22.2:5061> Is there a way to do this setting via variables? I can't seem to find any details for it. Thanks David

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64 bit CPU and 32 bit OS, is it good for freeswitch
by awais nazir 12 Oct '10

12 Oct '10

Hi I am newbie using and appreciating this remarkable software. I have heard that CENTOS 5.3 is best for freeswitch deployment but I am using 64 bit CPU and 32 bit OS. Can it give any problem? Earliest response will be appreciated. Thanks and Best Regards. Awais Nazeer

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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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