April 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Kill, close or reset a channel that remains with "show channels count"
by mszlazak＠aol.com 11 Apr '09

11 Apr '09

Initially, "show" "channels count" gives "0 total." I then spawn a session from javascript thus: apiExecute("originate", "{id_name='" + call['Caller Name'] + "',id_number=" + call["Caller Number"] + "}sofia/gateway/spa3102PSTN/" + "1" + call["Caller Number"] + "@10.0.0.5:5061 '&javascript(reminder.js \'${id_name}\' ${id_number})'"); Next, I'd like to automatically spawn another session when my channels show "0 total" with apiExecute("show", "channels count"); I've set up a (loop with a msleep) to check when apiExecute("show", "channels count") becomes zero but it never does after the first call and stays at 1. This seems to "mess-up" making the next call. Thanks. Mark.

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announcing mod_cidlookup
by Rupa Schomaker 11 Apr '09

11 Apr '09

Another itch scratched. :) I just committed a new module mod_cidlookup. mod_cidlookup allows one to: * lookup number->name mapping in a local database * lookup number->name mapping from a URL * cache the results of the URL lookup in memcache The URL lookup is useful when using third party number to name lookup services. Read more about it at: http://wiki.freeswitch.org/wiki/Mod_cidlookup File a jira or email me if you have issues. -- -Rupa

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How to find/build FreeSWITCH.Managed.dll?
by Peter Olsson 10 Apr '09

10 Apr '09

Hello again! When trying to load the mod_managed module it get an error that it can't find FreeSWITCH.Managed.dll. So my question is simply - where do I find this file, or how do I build it? I'm using the VC++ Express edition when building, so I guess I also have to install the C# edition - will this solve my problem? Sorry for asking stupid questions here - but I've just been playing around with FreeSWITCH for a day or so :) Regards, Peter Olsson

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Some spidermonkey modules fails to load in Windows
by Peter Olsson 10 Apr '09

10 Apr '09

The spidermonkey modules core_db/odbc, curl, socket and teletone fails to load in Windows. They just return error 1271 (Sym Error). I'm not sure if this is a known issue, or if just doesn't work in Windows :) I've been using the latest SVN when trying this. Any ideas anyone? :) Regards, Peter Olsson

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Re: [Freeswitch-users] Two or more simultaneous calls not
by Traun Leyden 10 Apr '09

10 Apr '09

Hey you beat me to it. I was going to have a look this morning but had no internet because some asswipe cut a bunch of fiber optic cables and took out phone/internet for a big part of the bay area. I haven't tried your patch yet, but I see something that looks suspect: http://fisheye.freeswitch.org/browse/FreeSWITCH/src/switch_cpp.cpp?r=12958 In playAndGetDigits() there are now two calls to begin_allow_threads() (line 778 and 780) followed by only one call to end_allow_threads() (line 793) Also I guess it would have better to test against JS, since it should have had the same bug right? Lua just ignores the threadswapping stuff but IIRC javascript uses it in much the same way as python. Or did I miss something? > Message: 5 > Date: Thu, 9 Apr 2009 08:15:38 -0500 > From: Anthony Minessale <anthony.minessale(a)gmail.com> > Subject: Re: [Freeswitch-users] Two or more simultaneous calls not > To: freeswitch-users(a)lists.freeswitch.org > Message-ID: > <191c3a030904090615j2b736b7bp890c903c00a014f6(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > easier than you think, > > Just like Traun had suspected, the methods in the C++ wrapper for the read > app was missing the begin and end allow threads calls > that are only important to python to tell it to suspend the thread state > while FS specific code is being executed. > > 2 line fix in trunk rev 12958 > >

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Crash in mod_opal when hanging up call...
by Peter Olsson 10 Apr '09

10 Apr '09

Hello everyone. I've been following this project for quite some time now, but I never got the time to test it. But today I finally had a day off from work, so I could sit and play around with it for some time :) Everything wen't really smooth - even though I built everything from scratch on a Windows machine, including mod_opal (linked against the opal library). And with the docs I found I didn't even have to search Google to get it up and running :) So first of all - what a great job, guys - I'm really impressed, and the code seems really stable as well! My setup right now is FreeSWITCH in Windows XP, with both SIP and H323 enabled. And most of the stuff works just fine. However, I think I've found a bug in mod_opal - it sometimes causes FreeSWITCH to crash when hanging up a call. I think that mod_opal is considered to be in beta stage still, so I'm not all that surprised. :) Check the error found in the log below. Does anyone have any ideas? It's pretty easy to reproduce, just dial in to FreeSWITCH using H323 and hang up the call, for me it happens maybe 2 out of 5 times. I'm using latest SVN trunk versions (checked out today), for both FreeSWITCH and for opal/ptlib. If you need further information, or if I should file a jira case, please get back to me, and I'll try to help out as much as possible. 2009-04-09 14:37:05 [NOTICE] mod_opal.cpp:657 FSConnection::OnAlerting() Ring-Ready ! 2009-04-09 14:37:05 [NOTICE] switch_channel.c:597 switch_channel_set_name() New Channel opal/in:9999 [c7441e16-394c-d843-9ce4-760786dcecbf] 2009-04-09 14:37:05 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing Peter Olsson [172.18.96.100]->9999 in context default 2009-04-09 14:37:05 [INFO] mod_opal.cpp:1134 mod_opal() opal/in:9999 initialise opal/in:9999write audio codec G.711-uLaw-64k for connection FSMediaStream-Source-G.711-uLaw-64k 2009-04-09 14:37:05 [INFO] mod_opal.cpp:1134 mod_opal() opal/in:9999 initialise opal/in:9999read audio codec G.711-uLaw-64k for connection FSMediaStream-Sink-G.711-uLaw-64k 2009-04-09 14:37:05 [NOTICE] mod_opal.cpp:795 FSConnection::OnOpenMediaStream() Channel [opal/in:9999] has been answered Assertion failed: (*frame)->codec != ((void *)0), file ..\..\src\switch_core_io.c, line 202 2009-04-09 14:37:09 [NOTICE] mod_opal.cpp:650 FSConnection::OnReleased() Hangup opal/in:9999 [CS_EXECUTE] [NORMAL_CLEARING] 2009-04-09 14:37:09 [INFO] h323pdu.cxx:1005 H225() Read error (0):

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encryption gateway/proxy with freeswitch?
by excelsio＠gmx.net 10 Apr '09

10 Apr '09

Hi, the Alcatel OmniPCX Enterprise (OXE) of one of our customers seems to support the following encyption scenarios: - IP-phone on Alcatel OXE <=SRTP=> IP-phone on Alcatel OXE - IP-phone <=SIPS/SRTP=> Alcatel OXE <=> landline phone Unfortunately the Alcatel OXE doesn´t support SIPS/SRTP encryption betwenn itself and a SIP provider. So, for now it looks like: - Alcatel OXE <=SIP/RTP=> SIP provider The goal is, to encrypt that traffic, too. The SIP provider does support SRTP. So I´m asking myself whether to place a freeswitch between both which proxies and also encrypts the traffic? - Alcatel OXE <=SIP/STP> Freeswitch <=(SIPS)/SRTP=> SIP provider Has someone done something like this already? thanks in advance Michael

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Problem with originate in javascript.
by mszlazak＠aol.com 09 Apr '09

09 Apr '09

I want to run a script with a scheduler but I'm having a problem with how to set up the originate in Javascript. The originate would go something like: originate {id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/12223334444@10.0.0.5:5061 GINO_ANS I can get this to work: session = new Session("{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/14082031170@10.0.0.5:5061"); But I want to "drop" that into an extension that runs another script and can't get either of these to work: session = new Session("{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/14082031170@10.0.0.5:5061 GINO_ANS"); session = new Session("{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/14082031170@10.0.0.5:5061 GINO_ANS XML default"); Also, will I have problems running the second script from the first script? Thanks.

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Re: [Freeswitch-users] Polycom register problem in private address
by Chris Fowler 09 Apr '09

09 Apr '09

I'm running FS on Amazons' EC2 compute cloud (AWS) and have 30 Polycom phones working happily in this config. I modified the Internal profile in /usr/local/freeswitch/conf/sip_profiles/internal.xml to include: <param name="aggressive-nat-detection" value="true"/> <param name="NDLB-force-rport" value="true"/> The phones connect on port 5060 - nothing specical to config in the <mac>-phone.cfg file for the phone; just host, port, user/pass. Chris.

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need help getting ISDN talking to Cisco 3845
by John Wehle 09 Apr '09

09 Apr '09

> Okay, a few things. First off, the wanpipe2.conf file has a booboo. Don't think so. > This line is WRONG: > TDMV_DCHAN = 0 Not exactly. My understanding is you can use either: wanpipeX.conf: TDMV_DCHAN = 0 zaptel.conf: dchan = 24 (or in our case 48 since it's the second span) which means use zaptel to handle the d-channel hdlc or wanpipeX.conf: TDMV_DCHAN = 24 zaptel.conf: hardhdlc = 24 (or in our case 48 since it's the second span) which means use wanpipe to handle the d-channel hdlc assuming the wanpipe driver has the necessary support (wanpipe on my platform doesn't). > Also, I recommend changing this line: > wbg1 = wanpipe2, , TDM_VOICE, Comment > > To this: > wbg1 = wanpipe2, , TDM_VOICE_API, Comment The sangoma voice API interface isn't available on my platform and shouldn't be necessary when using zaptel. > assuming that this is what you want then you will need to use > ozmod_libpri because the default OpenZAP PRI stack does not > currently support being the network side. Are you sure? Openzap appears to contain implementations for both NT and TE. The configuration file supports specifying either user or network for the mode. Is the NT support currently nonfunctional? I had tried configuring the Cisco as the NT with similar results. > I don't see where timing is specified It's the same T1 which was being used for RBS between FreeSWITCH and the Cisco so that timing (etc) should be okay. No errors are showing up at the physical level and the Cisco reports Layer 1 as active. The trace on the Cisco seems to show Layer 2 coming up (timestamps 22:53:44.264 through 22:54:21.760), then there's a long pause during which no Receive Ready frames are received from FreeSWITCH. At this point the Cisco gets unhappy and marks Layer 2 as down. If nothing obvious comes to anyone's mind, then I'll simply need to trace through the FreeSWITCH ISDN code and see what's going on. -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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