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Problems with Mod_openMRCP
by mszlazak＠aol.com 02 Dec '08

02 Dec '08

I'm getting the following errors when trying to run the example in the wiki: http://wiki.freeswitch.org/wiki/Mod_openmrcp 2008-11-28 09:59:54 [DEBUG] switch_core_session.c:435 switch_core_session_receive_message() Send signal sofia/internal/1000(a)10.0.0.2 [BREAK] 2008-11-28 09:59:54 [DEBUG] sofia.c:2269 sofia_handle_sip_i_state() Channel sofia/internal/1000(a)10.0.0.2 entering state [completed] 2008-11-28 09:59:54 [NOTICE] mod_spidermonkey.c:2034 session_answer() Channel [sofia/internal/1000(a)10.0.0.2] has been answered 2008-11-28 09:59:54 [DEBUG] mod_spidermonkey.c:1851 init_speech_engine() Raw Codec Activation Success L16@8000hz 1 channel 20ms 2008-11-28 09:59:54 [DEBUG] mod_openmrcp.c:634 openmrcp_tts_open() Create Synthesizer Channel 2008-11-28 09:59:54 [DEBUG] sofia.c:2269 sofia_handle_sip_i_state() Channel sofia/internal/1000(a)10.0.0.2 entering state [ready] 2008-11-28 09:59:59 [ERR] mod_openmrcp.c:643 openmrcp_tts_open() No response from client stack 2008-11-28 09:59:59 [ERR] mod_openmrcp.c:647 openmrcp_tts_open() No synthesizer channel available 2008-11-28 09:59:59 [ERR] mod_spidermonkey.c:1859 init_speech_engine() Invalid TTS module! 2008-11-28 09:59:59 [ERR] inline:1 mod_spidermonkey()? Cannot allocate speech engine! 2008-11-28 09:59:59 [NOTICE] switch_core_state_machine.c:160 switch_core_standard_on_execute() Hangup sofia/internal/1000(a)10.0.0.2 [CS_EXECUTE] [NORMAL_CLEARING] 2008-11-28 09:59:59 [DEBUG] switch_channel.c:1449 switch_channel_perform_hangup() Send signal sofia/internal/1000(a)10.0.0.2 [KILL] 2008-11-28 09:59:59 [DEBUG] switch_core_session.c:727 switch_core_session_signal_state_change() Send signal sofia/internal/1000(a)10.0.0.2 [BREAK] 2008-11-28 09:59:59 [DEBUG] switch_core_state_machine.c:432 switch_core_session_run() (sofia/internal/1000(a)10.0.0.2) State EXECUTE going to sleep 2008-11-28 09:59:59 [DEBUG] switch_core_state_machine.c:367 switch_core_session_run() (sofia/internal/1000(a)10.0.0.2) Running State Change CS_HANGUP 2008-11-28 09:59:59 [DEBUG] switch_core_state_machine.c:395 switch_core_session_run() (sofia/internal/1000(a)10.0.0.2) State HANGUP 2008-11-28 09:59:59 [DEBUG] mod_sofia.c:276 sofia_on_hangup() Channel sofia/internal/1000(a)10.0.0.2 hanging up, cause: NORMAL_CLEARING 2008-11-28 09:59:59 [DEBUG] mod_sofia.c:333 sofia_on_hangup() Sending BYE to sofia/internal/1000(a)10.0.0.2 2008-11-28 09:59:59 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/internal/1000(a)10.0.0.2 Standard HANGUP, cause: NORMAL_CLEARING 2008-11-28 09:59:59 [DEBUG] switch_core_state_machine.c:395 switch_core_session_run() (sofia/internal/1000(a)10.0.0.2) State HANGUP going to sleep 2008-11-28 09:59:59 [DEBUG] switch_core_session.c:860 switch_core_session_thread() Session 1 (sofia/internal/1000(a)10.0.0.2) Locked, Waiting on external entities 2008-11-28 09:59:59 [NOTICE] switch_core_session.c:878 switch_core_session_thread() Session 1 (sofia/internal/1000(a)10.0.0.2) Ended 2008-11-28 09:59:59 [NOTICE] switch_core_session.c:880 switch_core_session_thread() Close Channel sofia/internal/1000(a)10.0.0.2 [CS_HANGUP] 2008-11-28 10:00:26 [DEBUG] mod_openmrcp.c:167 openmrcp_on_session_terminate() on_session_terminate called I believe I followed the instructions correctly but I can't get openmrcp to connect with Cepstrals TTS.

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Support for Junghanns duoBRI
by Sergey Kirillov 02 Dec '08

02 Dec '08

Greetings, Can somebody tell me, if it is possible to use duoBRI card (http://www.junghanns.net/en/duobri_express_produkt.html) from Junghanns.net together with Freeswitch? I've found that this card has Zaptel drivers, and Freeswitch has mod_openzap. On the other side, I saw somewhere in wiki that Freeswitch does not support BRI at all at the moment. Please confirm or allay my apprehensions.

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Listen to a file, while recording?
by Dennis 02 Dec '08

02 Dec '08

hi, i would like to be able to listen to conversations, while they are ongoing. this should not happen over a phone. i would like to be able to have a link or something in my admin-area, where i can click, if i want to listen to a conversation. i thought about to start a record with socket inbound on a specific uuid and while the recording is done, i would like to play the file. but i have the feeling, that this is not possible, while the file is not finished. is there a way to stream the audio over the web? thanks for your help.

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Re: [Freeswitch-users] Problems with Mod_openMRCP
by mszlazak＠aol.com 02 Dec '08

02 Dec '08

I need to barge in again and add to my last post with this email from Voxeo support. Here is their response to the port binding conflict and it brings up a possible problem if FreeSwitch will be looking for Prophecy at that port? I assumed it would if I set up the extension right but now I don't know and need your assistance with this issue ... as well. Thank you. MESSAGE: Hi Mark, You are correct in that having multiple applications binding to the same port can cause a bundle of problems. You can configure Prophecy to stay away from port 5060, but then the question is whether FreeSwitch will be looking for Prophecy at that port (if its assuming that it's residing on a different box). Port 5060 is the standard for SIP traffic. To get Prophecy off 5060 you will need to edit the config.xml and callrouting.xml files. You will need to search out all instances of "5060" and replace with, perhaps, port 5068. For instance: <item name="ListenOnIP1">0.0.0.0:5068</item> <item name="ListenOnIP2">0.0.0.0:5061</item> <item name="ListenOnIP3">0.0.0.0:5067</item> <item name="ListenOnIP4">0.0.0.0:5063</item> <item name="ListenOnIP5">0.0.0.0:5064</item> <item name="ListenOnIP6">0.0.0.0:5065</item> instead of this... <item name="ListenOnIP1">0.0.0.0:5060</item> <item name="ListenOnIP2">0.0.0.0:5061</item> <item name="ListenOnIP3">0.0.0.0:5062</item> <item name="ListenOnIP4">0.0.0.0:5063</item> <item name="ListenOnIP5">0.0.0.0:5064</item> <item name="ListenOnIP6">0.0.0.0:5065</item> Regards, Jeff Kustermann Voxeo Support ? -----Original Message----- From: mszlazak(a)aol.com To: freeswitch-users(a)lists.freeswitch.org Sent: Mon, 1 Dec 2008 10:40 pm Subject: Re: [Freeswitch-users] Problems with Mod_openMRCP Just to follow up. Moshe Yudkowsky has an article on "Routing calls from FreeSwitch to Prophecy":? http://www.prophecy2006.com/node/145 My problem is that Freeswitch and Prophecy need to be on the same machine BUT both need to bind to port 5060 so I'm getting errors from one or the other depending who's running first. So can I change what port(s) FS uses and that way avoid this conflict? Maybe, this might let me bridge the call via FreeSwitch to Prophecy similar to what Moshe's article discusses??? -----Original Message----- From: mszlazak(a)aol.com To: freeswitch-users(a)lists.freeswitch.org Sent: Mon, 1 Dec 2008 4:44 pm Subject: Re: [Freeswitch-users] Problems with Mod_openMRCP Does "bridging" a call from FS to Voxeo's Prophecy server require openMRCP? If not then the other issue I might have is a database look up that is part of the dialogue that maybe need as the person response to prompts from the asr. It's possible to run a php script for the database stuff that Prophecy might need or could that happen via Javascript in FS? Then after the dialogue has completed I go from Prophecy back to FS. -----Original Message----- From: Anthony Minessale <anthony.minessale(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Mon, 1 Dec 2008 11:17 am Subject: Re: [Freeswitch-users] Problems with Mod_openMRCP mod_openmrcp was a contribution to the community by a 3rd party individual. As i have clearly stated in 2 previous emails, the man has decided to discontinue the openmrcp project. So now we are left with the remains of the module and discontinued code.? This was not our decision it was his. Since the author of openmrcp has stated that he has a new unimrcp we are certainly going to work towards getting mod_unimrcp to replace mod_openmrcp.? He had already commented on that previous thread to state he is willing to consider making a new module. Some people use it without issue which may mean that the crash you reported is windows specific and I do not have a working lab of any mrcp capbable system to try it against in unix for that matter.? I have a list of work to do from here to the moon and back so on an issue like this, unless someone can hand me login credentials to some box and give me a phone number to dial to reporduce the issue, it will be a long time until we can deal with it.? And the question arises, should we bother working on it anymore if the lib has been abandoned and we cannot even get any support from it's author which is where the problem most likely lies. I try not to get too annoyed by these remarks about what we *ought to do* because I know people lose sight of how much of the work to support the project is done by a small group of 3 people and not the 2000 people it appears to be from the outside looking in. (I've been answering email for 4 hours now) My suggestion is to pool some cash and pay the guy to make mod_unimrcp for FS that we can maintain in tree knowing the development can be supported by the original author. On Mon, Dec 1, 2008 at 12:51 PM, David Knell <dave(a)3c.co.uk> wrote: Hi Mike, My experience is that it's somewhat broken - it took two trivial tweaks to get it to work with IBM's ASR and TTS, but there's a more intractable problem to do with memory getting overwritten (I assume that this is something to do with something being freed when it shouldn't be) which causes a segfault on the second or third session after the module being loaded. Without wishing to sound like a stuck record, one thing that you guys really ought to do is to decide what's supported and what isn't, and make this obvious - for example, move unsupported modules to a different place in the tree, don't have them built by default, etc.? MRCP is in the specsheet on the Wiki.? Otherwise folk like Mark and I spend time installing stuff, go round in circles a bit trying to make it work, and then find out (a) that it doesn't and (b) it's not going to be fixed because it's not supported. Cheers -- Dave I would not say it is totally broken, it is known to work in quite a few places, but we are unlikely to be doing any new fixes in it. Mike On Dec 1, 2008, at 1:19 PM, mszlazak(a)aol.com wrote: Hi Anthony, Oh! OK. So is this module "totally broken". I say this because I can't seem to get it to work at all with the example in that Mod_openmrcp wiki page but I thought it might because I'm not be using the right Cepstral software (freetrial download versus the paided for SDK) or that I'm not using the right port numbers or something else I didn't do. I used TcpView to look at local port associated with my Cepstral software and changed a few things but still nothing. I changed the loglevel setting to 7 in the wiki's example but I don't see the kind of output on the console that I would expect for debug mode. Thanks. Mark. _______________________________________________ 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 -- David Knell, Director, 3C Limited T: 020 8114 8901 F: 020 3002 7257 M: 001 415 630 3031 http://www.3c.co.uk _______________________________________________ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 _______________________________________________ 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 Tis the season to save your money! Get the new AOL Holiday Toolbar for money saving offers and gift ideas. _______________________________________________ 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 Tis the season to save your money! Get the new AOL Holiday Toolbar for money saving offers and gift ideas. _______________________________________________ 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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Question about wrapping libfreeswitch
by Woody Dickson 02 Dec '08

02 Dec '08

Hi, I am sorry again for sending another email to the group again. I am working on embedding libfreeswitch to provide better monitoring. The first thing I attempt to do is to run the sample code provided in the website: #include <switch.h> int main(int argc, char **argv) { switch_core_flag_t flags = SCF_USE_SQL; int nc=0; /* this is for 'no console' mode, FALSE console is there, TRUE it isnt */ const char **err = NULL; /* error value for return from freeswitch initialization */ #define LOGFILE "freeswitch.log" static char *lfile = LOGFILE; /* if NULL no logfile is generated */ switch_core_init_and_modload(*lfile,flags,err); switch_core_runtime_loop(nc); switch_core_destroy(); return (0); /* per C89 spec */ } But this code gives me segmentation fault when executing it. This piece of code is supposed to start up freeswitch and run it is a loop. Does anyone see what is wrong with it? Does anyone have any working example that I can refer to? Thanks, Woody

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Problem with Freeswitch capturing DTMF
by Keith Wood 02 Dec '08

02 Dec '08

Hi, I am wondering if I am the only one getting this problem or not. When sending in DTMF to freeswitch, freeswitch is not always capable of capturing all the DTMF being sent. For instance, sending 1000 to freeswitch may end up becoming 100 or 10003 becoming 1003. Am I the only one getting this strange issue? If anyone know how to fix this problem, I would greatly appreciate it. Regards, Keith

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libfreeswitch question
by Woody Dickson 02 Dec '08

02 Dec '08

Hi, I am just having a dumb question and hoping someone can help me. I am trying to run a c program with libfreeswitch embedded so I can use some external mechanism to keep track of freeswitch, but I am having problem while compiling: [root@localhost fs]# gcc switchnode.c -I/usr/local/freeswitch/include -L/usr/local/freeswitch/lib -lfreeswitch -lpthread switchnode.c: In function 'main': switchnode.c:11: warning: passing argument 1 of 'switch_core_init_and_modload' makes integer from pointer without a cast switchnode.c:11: warning: passing argument 3 of 'switch_core_init_and_modload' from incompatible pointer type /usr/local/freeswitch/lib/libfreeswitch.so: undefined reference to `clock_gettime' /usr/local/freeswitch/lib/libfreeswitch.so: undefined reference to `uuid_generate' /usr/local/freeswitch/lib/libfreeswitch.so: undefined reference to `crypt_r' collect2: ld returned 1 exit status [root@localhost fs]# Does anyone know which library is missing? Thanks, Woody

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Re: [Freeswitch-users] How to specify Path for sound files
by Jan Kubr 02 Dec '08

02 Dec '08

Hi Faisal, the path is either an absolute path or a path relative to the directory in the sound_prefix var in vars.xml. So this <action application="playback" data="/usr/local/freeswitch/sounds/en/us/callie/misc/8000/call_secured.wav"/> works fine on my box. You sure this one doesn't work for you? Jan > I tried to play a sound file using the dialplan given on the link > http://wiki.freeswitch.org/wiki/Playing_recording_external_media#Play_wav > > <action application="playback" data="/path/to/your.wav"/> > In place of /path/to/your.wave I used > "/en/us/callie/misc/8000/call_secured.wav" > "/usr/local/freeswitch/sounds/en/us/callie/misc/8000/call_secured.wav" > "/sounds/en/us/callie/misc/8000/call_secured.wav" > But none of these is useful bcoz when i call on 2009, which is > to b dialed to play the sound, same msg is > displayed "404 NOT FOUND" > Plz help me out.??? Faisal

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Re: [Freeswitch-users] Sound file as ringback
by Jan Kubr 01 Dec '08

01 Dec '08

Yes, setting the var to the full path works. Sorry, should have taken the "full path" in the wiki more seriously. MP3s are played only once, 8kHz WAVs work perfectly. Cheers, Jan > Can you try putting the full path to the file? Also what does the > console output look like? > > /b > > On Nov 30, 2008, at 12:30 PM, Jan Kubr wrote: > >> I have try different format of files (from 8KHz mono wavs to MP3s, all >> of which play fine via playback) and some caused the bridge to be >> finished immediately (with NO_USER_RESPONSE), some make it generate >> crazy beeping, but none is played while the phone is ringing on the >> other end.

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SIP INVITE timeout
by Gabriel Kuri 01 Dec '08

01 Dec '08

I have a phone that is registered to FS but is no longer available (Internet connection down, phone turned off, etc.). The registration still exists in the sip_registrations table (not expired yet), but the phone is not reachable on the network. According to my dialplan, if the bridge to the phone fails after 20 seconds, the call should be forwarded to a different box for handling (see dialplan below). <extension name="My Extension"> <condition field="destination_number" expression="1213XXXXXXX"> <action application="info"/> <action application="set" data="originate_timeout=20"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="${sofia_contact(default/1213XXXXXXX(a)mydomain.net"/> <action application="bridge" data="sofia/default/1213XXXXXXX(a)box.mydomain.net"/> </condition> </extension> If the phone is down and not responding to the INVITEs, it appears my carrier is canceling the SIP INVITE to FreeSWITCH after about 10 seconds. My timeout is 20 seconds. Is there anyway to deal with this situation, without going back to my carrier and asking them to increase their timeout on an INVITE? Call Progress: Carrier -> FS (INVITE) FS -> Carrier (100 Trying) <10 seconds pass while FS is attempting to contact the phone> Carrier -> FS (CANCEL) FS -> Carrier (200 OK) FS -> Carrier (487 Request Terminated) Carrier -> FS (ACK) Thanks ... Gabe

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