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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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Polycom register problem in private address
by zhaoxxqq 09 Apr '09

09 Apr '09

hi, I use FS server at public Address. I use polycom's IP550 at private address(192.168.0.120), Now there is a problem that the IP550 can not register to FS. But when I use account to eyebeam, the registering is OK. the attachment is my IP 550's config file, I think it must be NAT problem. Can anyone can help me solve it? 2009-04-09 zhaoxxqq

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Status of Sangoma support in Windows?
by Peter Olsson 09 Apr '09

09 Apr '09

>From what I've found in the docs and lists, the support for Sangoma (PRI) cards is still not avaiable in the Windows port. Is this planned to be implemented in the future, or will it never be included in Win32? Just a curious thought, since I might need to use some PRI stuff in the future... Regards, Peter Olsson

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Re: [Freeswitch-users] Two or more simultaneous calls not
by Adam Wilt 09 Apr '09

09 Apr '09

Traun, thanks again for your help. I followed your advice and I made some progress! I tested with the latest trunk version and also with 1.0.2, and both exhibited the same behavior. I then tried writing a test script in Lua, and it worked fine. So this meant the problem was in the Python module (I was sure it was some FS config issue). So I started playing with a small test Python script, and I narrowed the problem down to when I'm using the "read" function. Here is my test script: from freeswitch import * def handler(session, args): #answer the call session.answer(); #play a file session.streamFile("long_prompt.mp3") # Test 1 - FAILED! digits = session.read(5, 10, "long_prompt.mp3", 3000, "#") # Test 2 - WORKED OK! #session.getDigits(1,"#",7000) #session.streamFile("long_prompt.mp3") # TEST 3 - WORKED OK! #digits = session.playAndGetDigits(5, 10, 1, 60, "#","long_prompt.mp3", "", "") #hangup session.hangup(); When I uncomment the code under test 1 and I make two simultaneous calls, the initial prompt plays for both calls just fine. But then the second prompt only plays on one of the channels and the other one just has dead air. When the first channel finishes playing the prompt, then the second channel starts playing it. Then I re-comment test 1 and uncomment either test 2 or test 3, Both prompts play just fine for both channels. So I think there may be a bug in the read() function somewhere. I took a look at it, but it's way over my head. Thanks again, Adam >Message: 8 >Date: Thu, 9 Apr 2009 02:36:08 +0430 >From: Traun Leyden <tleyden(a)branchcut.com> >Subject: Re: [Freeswitch-users] Two or more simultaneous calls not >To: freeswitch-users(a)lists.freeswitch.org >Message-ID: > <f9f80d540904081506p3f0647fei4cf843d0b1904bc0(a)mail.gmail.com> >Content-Type: text/plain; charset="iso-8859-1" >Hi Adam, >I'm stumped .. I guess you could try the following: >* Try with the trunk version of freeswitch. I don't think it will matter, >but just in case >* Try to simulate the same test with a Lua script. Do you see the same >problem? >If those don't turn up anything, then the next logical step would be >to start adding printf() statements in the mod_python code and >find out where it is getting stuck. In particular around the parts where >it swaps the threadstate in and out. I might be able to create a patch >for you, but try those other tests first. >HTH, >Traun > >> >> Thanks for the response Traun. The version of Python is 2.4.3, and I >> didn't >> build it myself, I installed it with yum. >> The version of Red Hat is 4.1.2-41. >> "import threading" works fine, so I don't think it's a Python threading >> issue. >> The FreeSWITCH version I installed is the >> freeswitch-1.0.3.tar.gz< >> http://files.freeswitch.org/freeswitch-1.0.3.tar.gz> >> located >> at files.freeswitch.org. >> I didn't make any major changes to the configuration; I enabled Python and >> set-up the SIP profile, directory and dialplan. No other changes. >> Any other help would be appreciated, since I really don't know where to >> look. >> >> Thanks, >> Adam

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How to design my project ?
by Michele M 09 Apr '09

09 Apr '09

Hi there, I'm quite a newbie about freeswitch. I have an application (IVR) that needs to have endpoints SIP to register,answer the calls and transfer them to the right phones.(I( have my own SIP server).Moreover it needs also a ASR/TTS API' set to communicate with my ASR/TTS engine ( just for example let's assume it is Cepstral). I'd wouldn't want to have freeswitch running and communicate with it to accomplish that but just to use the libfreeswitch library embedded. As I don't know that much about freeswitch can it be done? or just I need to have freeswitch running as a must? Can somebody point me to the right place where to find example of using library embedded (best examples for what I'm trying to do) as I have not found that many? Thanks in advance Miki

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