June 2008 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] .NET Development
by Patrick Tran 31 Jul '08

31 Jul '08

I'm new to Freeswitch and trying to develop a .NET Application. I'm having a lot of trouble getting started though. There doesn't seem to be a lot support out there for this. My development has sort of come to a stand-still and I was wondering if anyone could help me get pointed in the right direction. Any help would be much appreciated. Thanks -Patrick

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[Freeswitch-dev] ARM cross compile
by Simon Capper 01 Jul '08

01 Jul '08

I'm trying to get freeswitch to cross compile on arm (linux 2.6.21), build o/s linux, using OpenWRT buildroot. I've gotten as far as getting the console running and loading all the modules (no calls attempted yet), the switch will be SIP only with a custom (yet to be developed) endpoint module for an analog line interface. I'm using these modules: mod_console mod_syslog mod_commands mod_conference mod_dptools mod_limit mod_expr mod_esf mod_fsv mod_ilbc mod_dialplan_asterisk mod_sofia mod_event_socket mod_sndfile mod_tone_stream mod_lua Is there a way to disable the c++ code in freeswitch without totally breaking it? I'm in an embedded environment and libstdc++ is large and takes up valuable flash space. Is there a way to specify a different install and runtime location for the modules (like you can do for log conf and db?) Could be runtime or build time, I found --with-modinstdir but this seems to affect the install location and the runtime search location. In switch_core_state_machine.c backtrace_symbols() fails, not supported in my arm environment. I hacked this to use the alternative version of print_trace. In pcre, libtool failed with the host build tools, had to add --tag=CC in LINK_FOR_BUILD. LINK_FOR_BUILD = $(LIBTOOL) --mode=link --tag=CC $(CC_FOR_BUILD) $(CFLAGS_FOR_BUILD) $(LDFLAGS_FOR_BUILD) -I. -I$(top_srcdir) Although I don't need it the voipcodecs lib it will not load, causes freeswitch to exit, lrintf & rintf are not available in my arm environment and the voipcodecs configure does not test for them (libsndfile handles this correctly). My configure command line: export config_TARGET_CC="$(TARGET_CC)"; \ export config_BUILD_CC="$(HOSTCC)"; \ export config_TARGET_CFLAGS="$(TARGET_CFLAGS)"; \ export config_TARGET_LIBS="-L$(STAGING_DIR)/usr/lib"; \ export CC_FOR_BUILD="$(HOSTCC)"; \ export ac_cv_file__dev_zero=no; \ export apr_cv_tcp_nodelay_with_cork=yes; \ export ac_cv_sizeof_ssize_t=4; \ export ac_cv_file_dbd_apr_dbd_mysql_c=yes; \ export ac_cv_path__libcurl_config=$(BUILD_DIR)/curl-7.18.2/install/bin/curl-config; \ export apr_cv_mutex_recursive=yes; \ (cd $(TOPDIR)/ooma/freeswitch; rm -rf config.{cache,status} ; \ $(TARGET_CONFIGURE_OPTS) \ CFLAGS='$(TARGET_CFLAGS)' \ CPPFLAGS="-I$(STAGING_DIR)/usr/include" \ LDFLAGS="-L$(STAGING_DIR)/usr/lib" \ ./configure \ --prefix=$(PKG_BUILD_DIR) \ --target=$(GNU_TARGET_NAME) \ --host=$(GNU_TARGET_NAME) \ --build=$(GNU_HOST_NAME) \ --with-libcurl=$(BUILD_DIR)/curl-7.18.2/install \ --with-devrandom=/dev/urandom \ --with-modinstdir=/mod \ Thanks, Simon

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[Freeswitch-dev] FW: [asterisk-users] Asterisk as an IVR
by Michael Collins 30 Jun '08

30 Jun '08

Question... Is this something that FreeSWITCH can do with the XML CDRs? Just curious. If it's possible to store the time of key DTMF received in the XML CDR then that would be awesome. The other thing I was thinking about was the list of apps that get stored in the XML. Is it possible/realistic to have not only the list of applications but also a timestamp for each one? That would allow one to calculate the amount of time spent in things like IVR menus, listening to message playbacks, etc. This kind of feature would allow us to snag all sorts of IVR-builders. Asterisk's handling of this kind of information is pathetic. FS is *way* better for gathering this kind of info. Thanks, MC -----Original Message----- From: asterisk-users-bounces(a)lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of research(a)businesstz.com Sent: Saturday, June 28, 2008 11:09 AM To: asterisk-users(a)lists.digium.com Subject: Re: [asterisk-users] Asterisk as an IVR Thanks Anselm Its true that is a lot of calls but i have a separate mysql database on different server (HP DL580G5 with 16cores). what am currently doing is capturing the information right after selection and insert that record into mySql. **** [macro-capture-input] ; ; ; Macro that feeds data into mysql through perl script: ; ${ARG1} - MSISDN ; ${ARG2} - src ; ${ARG3} - MainMenu Application ; ${ARG4} - Channel ; ${ARG5} - calldatetime ; ${ARG6} - Sub Menu Application ; ; exten => s,1,System(/var/lib/asterisk/agi-bin/capture.pl ${ARG1} ${ARG2} ${ARG3} ${ARG4} ${ARG5} ${ARG6}) [Data-Services-Options] ; This menu is aimed to provide user with info about data services offered by Vodacom, including ; 1 - SUBMenu 1 ; 2 - SUBMenu 2 ; 3 - SUBMenu 3 ; 4 - SUBMenu 4 ; ; ;SUBMENU 1 ; exten => 1,1,Macro(capture-input,"${MSISDN}","${OPT}","APPLICATION1","${CHANNEL}" ,"now()","SUBMENU1") exten => 1,n,Background(IVR/(1110) MENU 1) ;SUBMENU 2 exten => 2,1,Macro(sendsms,${MSISDN},1,${LANGUAGE}) exten => 2,n,Macro(ivrcdr,"${MSISDN}","${OPT}","APPLICATION2","${CHANNEL}","now() ","SUBMENU2") exten => 2,n,Background(IVR/(1120) MENU 2) . . . **** what i will also want to capture is how long a caller took to listen to say SUBMENU1 It should be noted that CDR doesnot capture such detailed info (Tzafrir) Regards Kili > > On Sat, Jun 28, 2008 at 03:37:56PM +0200, Anselm Martin Hoffmeister wrote: >> Am Samstag, den 28.06.2008, 08:15 -0500 schrieb research(a)businesstz.com: >> > Hi List >> > >> > I have build an IVR on Asterisk from 1.2 to now 1.4.18 and has already >> > processed more than 10million calls! >> > >> > I have one big challenge which is reporting... it is the requirement >> to >> > have a web reporting module which should the following info based on >> > selected time frame >> > - Number of calls on specific branch- Done >> > - Number of calls to branch 1 that came from branch 2 (this should be >> > flexible) >> > - talktime on specified branch (say how long caller listened to option >> 1 >> > before choosing option 2 or hangup) >> > >> > On IVR, it is so important to understand how many callers select a >> > specific branch and how long they spent on that branch. CDR stats can >> not >> > provide these type of information and on trying freepbx, still can not >> go >> > so detailed >> >> Dear Kili, >> >> in my opinion this is a good application for Database backends. You >> could, for example, write entries to a DB whenever someone presses a key >> (or is re-routed in the dialplan, which comes to a similar scheme). In >> data mining time some SQL logic can produce nearly any data you want, >> provided the input data is there. >> >> Millions of calls sounds a lot though, so be sure to have a reasonable >> database backend: The asterisk included one might be a bit on the small >> side here. >> >> This is just an idea, I did not implement anything the like (yet). > > Asterisk already has this separate database backend: CDR. > > -- > Tzafrir Cohen > icq#16849755 jabber:tzafrir.cohen@xorcom.com > +972-50-7952406 mailto:tzafrir.cohen@xorcom.com > http://www.xorcom.com iax:guest@local.xorcom.com/tzafrir > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

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Re: [Freeswitch-dev] [Freeswitch-users] YAML support as an alternative of XML for configuration
by Brian West 30 Jun '08

30 Jun '08

Diego, I can already tell you Its not going to happen. The choice was made early on and it was the correct one. It's not something that can be changed or modular. If XML is the single reason you won't be using FreeSWITCH then by all means stick with Asterisk, because we aren't going to beg you to use FreeSWITCH. Read this for more info http://www.freeswitch.org/node/123 /b On Jun 26, 2008, at 9:14 PM, Diego Viola wrote: > Hi everyone - > > I really love this project... I love the design you guys took with FS, > I love the stability, the features, the openness of it, it's the VoIP > platform I always dreamed of it... FreeSWITCH is what Asterisk should > have been. > > But I just don't like one thing: XML. > > I also don't intend to create a flame war or troll with this thread, > please take this as a suggestion of someone that really wants to use > FS... XML is the only reason why I still stay with Asterisk, I just > can't like XML, and I don't like Asterisk as well, I want FreeSWITCH > stability, features, etc, but I can't like XML, not even if I try to, > it's too slow to use/write, verbose and complex. > > As a sysadmin point of view, I prefer parameter=value but I realize > the advantages of XML (data serialization, etc), this is also why I > don't use mod_dialplan_asterisk, because I think it's time for > something better than INI files, but XML isn't for me, I'm a > Linux/UNIX guy and I don't want a GUI either, I love CLI, but I want > something simple to use, read and write, and I think YAML is the best > of the both worlds. > > I would like to have more than one option to configure FreeSWITCH, > also, not everyone likes to do the same thing the same way and it > would be really cool if FS gives this freedom to people... > > I would really like to see YAML as a configuration option in > FreeSWITCH, give me this any day and I swear I don't toch Asterisk > _ever again_, YAML can be used for data serialization and it's more > simple than XML, Ruby projects like Rails uses it. > > Take a look here: > > http://yaml.org/spec/1.2/ > http://yaml.org/ > > "YAML™ (rhymes with "camel") is a human-friendly, cross language, > Unicode based data serialization language designed around the common > native data types of agile programming languages. It is broadly useful > for programming needs ranging from configuration files to Internet > messaging to object persistence to data auditing. Together with the > Unicode standard for characters, this specification provides all the > information necessary to understand YAML Version 1.2 and to create > programs that process YAML information. " > > Please consider it, I'm also willing to pay for a mod_yaml, other than > this I really love FreeSWITCH and I think it's great, I will switch in > a heartbeat if you guys add support for it ;-) > > Best regards, > > Diego > > _______________________________________________ > 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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[Freeswitch-dev] DTMF issue on playback command with openzap
by Steve Laroche 29 Jun '08

29 Jun '08

I did run some test to reproduce the problem. I'm using Version 1.0.pre4 (8797:8820M). Test 1 ----------------------------------------------------------------- Enter this set of DTMF (123456789012345678901234567890*) while the record command is active. The call is originate through openzap. Event_Socket trace: http://pastebin.freeswitch.org/4677 Result: There's no problem. The DTMF handle correctly. I'm receiving all the DTMF in the right order without any duplication. Test 2 ----------------------------------------------------------------- Enter this set of DTMF (123456789012345678901234567890*) while the playback command is active. The call is originate through openzap. Event_Socket trace: http://pastebin.freeswitch.org/4678 Result: There's a problem. The DTMF don't seem to be handles correctly. I'm not receiving the DTMF in the right order, because there's some duplication in the DTMF and I even miss some DTMF. Test 3 ----------------------------------------------------------------- Enter this set of DTMF (123456789012345678901234567890*) while the record command is active. The call is originate through SIP. Event_Socket trace: http://pastebin.freeswitch.org/4679 Result: There's no problem. The DTMF handle correctly. I'm receiving all the DTMF in the right order without any duplication. Test 4 ----------------------------------------------------------------- Enter this set of DTMF (123456789012345678901234567890*) while the playback command is active. The call is originate through SIP. Event_Socket trace: http://pastebin.freeswitch.org/4680 Result: There's no problem. The DTMF handle correctly. I'm receiving all the DTMF in the right order without any duplication. SL

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[Freeswitch-dev] YAML support as an alternative of XML for configuration
by Diego Viola 27 Jun '08

27 Jun '08

Hi everyone - I really love this project... I love the design you guys took with FS, I love the stability, the features, the openness of it, it's the VoIP platform I always dreamed of it... FreeSWITCH is what Asterisk should have been. But I just don't like one thing: XML. I also don't intend to create a flame war or troll with this thread, please take this as a suggestion of someone that really wants to use FS... XML is the only reason why I still stay with Asterisk, I just can't like XML, and I don't like Asterisk as well, I want FreeSWITCH stability, features, etc, but I can't like XML, not even if I try to, it's too slow to use/write, verbose and complex. As a sysadmin point of view, I prefer parameter=value but I realize the advantages of XML (data serialization, etc), this is also why I don't use mod_dialplan_asterisk, because I think it's time for something better than INI files, but XML isn't for me, I'm a Linux/UNIX guy and I don't want a GUI either, I love CLI, but I want something simple to use, read and write, and I think YAML is the best of the both worlds. I would like to have more than one option to configure FreeSWITCH, also, not everyone likes to do the same thing the same way and it would be really cool if FS gives this freedom to people... I would really like to see YAML as a configuration option in FreeSWITCH, give me this any day and I swear I don't toch Asterisk _ever again_, YAML can be used for data serialization and it's more simple than XML, Ruby projects like Rails uses it. Take a look here: http://yaml.org/spec/1.2/ http://yaml.org/ "YAML™ (rhymes with "camel") is a human-friendly, cross language, Unicode based data serialization language designed around the common native data types of agile programming languages. It is broadly useful for programming needs ranging from configuration files to Internet messaging to object persistence to data auditing. Together with the Unicode standard for characters, this specification provides all the information necessary to understand YAML Version 1.2 and to create programs that process YAML information. " Please consider it, I'm also willing to pay for a mod_yaml, other than this I really love FreeSWITCH and I think it's great, I will switch in a heartbeat if you guys add support for it ;-) Best regards, Diego

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[Freeswitch-dev] Substutions in XML dial plan
by Simon Perreault 26 Jun '08

26 Jun '08

Hello, My dial plan contains something like the following: <action application="bridge" data="sofia/profile1/$1,sofia/profile2/$1"/> This causes a problem in mod_dialplan_xml.c, near line 150: len = (uint32_t) (strlen(data) + strlen(field_data) + 10); if (!(substituted = malloc(len))) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "memory error!\n"); proceed = 0; goto done; } memset(substituted, 0, len); switch_perform_substitution(re, proceed, data, field_data, substituted, len, ovector); app_data = substituted; The problem is simple: len is too small because the $1 pattern appears twice. Therefore adding strlen(field_data) once is not enough. It should be multiplied by the number of times the pattern appears in the string. It seems like someone experienced the same problem before: that would explain the "+ 10" magic number which I am very tempted to just increase to 100. ;-) Is this really a bug or is it my imagination? If it is a bug, how should we go about fixing it? Thanks, Simon -- Please try Numb, a STUN/TURN server implementation. Free access at http://numb.viagenie.ca/.

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[Freeswitch-dev] VC2008 Build Error Build 8839
by Robert Clayton 25 Jun '08

25 Jun '08

Problems with build 8839 mod_PortAudio error LNK2019: unresolved external symbol _PaWin_InitializeWaveFormatEx referenced in function _QueryFormatSupported portaudio.lib mod_PortAudio error LNK2019: unresolved external symbol _PaWin_InitializeWaveFormatExtensible referenced in function _QueryFormatSupported portaudio.lib mod_PortAudio error LNK2019: unresolved external symbol _PaWin_DefaultChannelMask referenced in function _OpenStream portaudio.lib mod_PortAudio fatal error LNK1120: 3 unresolved externals C:\Freeswitch\Debug/mod/mod_PortAudio.dll mod_PortAudio

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[Freeswitch-dev] Meeting time today. 5PM Central
by Brian West 25 Jun '08

25 Jun '08

Don't forget the wiki meeting is at 5pm today. Brian West sip:brian@freeswitch.org

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[Freeswitch-dev] OpenZAP Question re: Q931Rx23 (from Q931.c)
by Michael Collins 24 Jun '08

24 Jun '08

I have question about the two different buffers that are accessed in this function: /*********************************************************************** ****** Function: Q931Rx23 Description: Receive message from layer 2 (LAPD). Receiving a message is always done in 2 steps. First the message must be interpreted and translated to a static struct. Secondly the message is processed and responded to. The Q.931 message contains a static header that is interpreted in this function. The rest is interpreted in a sub function according to mestype. Parameters: pTrunk [IN] Ptr to trunk info. buf [IN] Ptr to buffer containing message. Size [IN] Size of message. Return Value: Error Code. 0 = No Error, < 0 :error, > 0 : Warning see q931errors.h for details. ************************************************************************ *****/ L3INT Q931Rx23(Q931_TrunkInfo_t *pTrunk, L3UCHAR * buf, L3INT Size) { L3UCHAR *Mes = &buf[Q931L2HeaderSpace]; L3INT RetCode = Q931E_NO_ERROR; Q931mes_Generic *m = (Q931mes_Generic *) pTrunk->L3Buf; L3INT ISize; L3INT IOff = 0; /* Protocol Discriminator */ m->ProtDisc = Mes[IOff++]; /* CRV */ m->CRVFlag = Mes[IOff + 1] & 0x80; m->CRV = Q931Uie_CRV(pTrunk, Mes, m->buf, &IOff, &ISize); /* Message Type */ m->MesType = Mes[IOff++]; /* Call table proc to unpack codec message */ /*debug */ printf("\n\nQ931Rx23- Dialect: %d, MsgType: %d\nCause: [%x]\n",pTrunk->Dialect,m->MesType,m->Cause); printf("Message: %s\n",Mes); RetCode = Q931Umes[pTrunk->Dialect][m->MesType](pTrunk, Mes, (Q931mes_Generic *)pTrunk->L3Buf, Q931L4HeaderSpace + IOff , Size - Q931L4HeaderSpace - IOff + 1); if(RetCode >= Q931E_NO_ERROR) { RetCode = Q931Proc[pTrunk->Dialect][m->MesType](pTrunk, pTrunk->L3Buf, 2); } return RetCode; } What's the difference between pTrunk->L3Buf and *buf in this function? I'm trying to understand where the q931 message is coming from as it is passed up the stack from layer 2 to layer 3. I may have found a clue about the phantom cause IE issue but I won't know for sure until I know what exactly I'm looking at. Also, can you tell me what is accomplished by creating *Mes as opposed to *m? I am familiar with the Q931mes_Generic data struct but the line: L3UCHAR *Mes = &buf[Q931L2HeaderSpace]; ... is confusing me. What does that do? Thanks for your time!! -MC

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