February 2010 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] FSComm basic issue
by Suneel Papineni 09 Mar '10

09 Mar '10

Hi, I am trying to use FSComm with Freeswitch and facing following issues. 1. Using pre-build binary (windows), when the application is started FSComm is getting Registered properly. When I tried to make a call, UI displays Dialing... but unable to see any SIP (INVITE) messages in wireshark traces. After sometime UI displays with message "Call with (destination number) failed with reason DESTINATION_OUT_OF_ORDER though destination number is registered with another FSComm" 2. Also I am unable to see any logs generated in the log folder. Downloaded the latest source code (Freeswitch 1.0.5 latest updated as on 26/02/10 at 4am) and tried to build FSComm. Build was succeeded. Application (FSComm) also started and displayed with UI. When I try to change the preferences, it has thrown Porta Audio Error saying "Error Querying Audio Devices" even though proper audio devices are present. Also it doesn't create folders like "conf", "mod". Even after copying all the required dll's and mod files (as specified in FSComm wiki pages), application is throwing the same error. I am using Windows XP machine. Built a Debug & Release version with 32-bit option. If someone has built FSComm for windows environment and is working fine, could you please let me know if there are any additional things I need to do to make it work. Thanks & Regards Suneel ************************************************************************* Please consider the environment before printing this e-mail ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN *************************************************************************

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[Freeswitch-dev] When is 1.0.5 due out?
by Mark Sobkow 02 Mar '10

02 Mar '10

I've been advised that the Ubuntu/.deb packaging won't be getting updated until the 1.0.5 release (1.0.4 slipped through the cracks.) I need to know when 1.0.5 is scheduled for release, as we can't ship source builds to our customers, only .deb installations. (The boxes our customers run don't have space on them for doing source builds, and we don't have compilers and such installed on them by default.) Alternatively, if someone could point me to a repo with control files for 1.0.3 or 1.0.4, we could do our own source builds and packaging. I've no fear of tweaking and hacking control files; I just haven't done .deb builds before and starting off from scratch with a project as big as Freeswitch is a bit overwhelming (especially as a "proper" packaging needs to split out the mods, sound packagings, etc.)

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[Freeswitch-dev] core odbc vs in-module mysql
by Daniel-Constantin Mierla 26 Feb '10

26 Feb '10

Hello, is it something particular that recommends usage of core odbc api instead of connecting from a module directly to mysql database? I'm fine to keep using sqlite for the rest of FS needs, just need to do fast access to mysql, therefore I try to avoid the extra layer added by odbc. Thanks, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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[Freeswitch-dev] Call for help - adding information to the wiki: SIP ALG's
by Michael Collins 26 Feb '10

26 Feb '10

Hi all, I've just completed a new wiki page: http://wiki.freeswitch.org/wiki/ALG I would like all of you who have dealt with routers with SIP ALG's to submit your input. I would like to see this page have a list of how-to's for all of the popular routers. If we can make it easy for people to disable SIP ALG's then I think we can all save ourselves time and energy answering questions in IRC and the mailing lists. Please by all means add your knowledge here. I started with the 2wire 3800HGV that I got for my ATT Uverse service. If you have knowledge that you like to add to the wiki (on this subject or any other) but are not confident in your wiki editing skills then contact me off list and I will be happy to help you get up to speed. Editing your first wiki page is always the hardest... :) Thanks again for all of your help! By the way, today's community conference call was great. Please plan on attending next week and we'll talk about more great FreeSWITCH stuff. I will have the recording of Rupa discussing mod_limit up on line as soon as I can. Take care, Michael

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[Freeswitch-dev] New TTS engine via UniMRCP
by Liam Parker 25 Feb '10

25 Feb '10

Hi, I am integrating Cereproc's TTS into FreeSwitch and am having some trouble. I have the TTS engine working within the UniMRCP server and have built FreeSwitch with UniMRCP support. I 'think' I have setup the Dialplan XML correctly but I keep getting the following error when I place a call... 010-02-25 11:38:31.743074 [ERR] switch_core_speech.c:61 Invalid speech module [cereproc-mrcp1]! 2010-02-25 11:38:31.743074 [ERR] switch_ivr_play_say.c:2109 Invalid TTS module! I believe that I just need to setup the conf and XML files correctly. I don't believe that I should need to alter any of the code. Any help would be greatly appreciated and then we can get Cereproc's TTS verfied as working on FreeSwitch for other users. Thanks, Liam.

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Re: [Freeswitch-dev] FW: New TTS engine via UniMRCP
by Christopher Rienzo 25 Feb '10

25 Feb '10

Your dialplan is incorrect. The engine should be unimrcp and profile will be cereproc-mrcp1. Your log says FreeSWITCH thinks the engine is cereproc-mrcp1. Show me the part of your dialplan that is doing the TTS. *From:* freeswitch-dev-bounces(a)lists.freeswitch.org [mailto: > freeswitch-dev-bounces(a)lists.freeswitch.org] *On Behalf Of *Liam Parker > *Sent:* Thursday, February 25, 2010 7:34 AM > *To:* freeswitch-dev(a)lists.freeswitch.org > *Subject:* [Freeswitch-dev] New TTS engine via UniMRCP > > > > Hi, > > > > I am integrating Cereproc's TTS into FreeSwitch and am having some trouble. > > > > I have the TTS engine working within the UniMRCP server and have built > FreeSwitch > > with UniMRCP support. > > > > I 'think' I have setup the Dialplan XML correctly but I keep getting the > following error when > > I place a call... > > > > 010-02-25 11:38:31.743074 [ERR] switch_core_speech.c:61 Invalid speech > module [cereproc-mrcp1]! > > 2010-02-25 11:38:31.743074 [ERR] switch_ivr_play_say.c:2109 Invalid TTS > module! > > > > I believe that I just need to setup the conf and XML files correctly. I > don't believe that > > I should need to alter any of the code. > > > > Any help would be greatly appreciated and then we can get Cereproc's TTS > verfied > > as working on FreeSwitch for other users. > > > > Thanks, > > > > Liam. > > > > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > >

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Re: [Freeswitch-dev] Openzap and intermitent ringing
by Moises Silva 25 Feb '10

25 Feb '10

Hi again François, Please keep the conversation in the freeswitch-dev mailing list, unless you want private paid support ;-) See the answer to your question below ... On Wed, Jan 20, 2010 at 3:57 AM, François Legal <devel(a)thom.fr.eu.org>wrote: > Do you know if there is a way (except using an oscilloscope to sample the > line signal) to get a dump of the data sent to the card by openzap (is it > samples or something) and vice versa. > > > > What I would like to do is see what is returned to openzap when a call > arrives on an FXO port (CLIP is working in that direction) and compare it to > what is being sent to the card when sending CLIP to and FXS port. > Once the call gets into the computer we're talking about digital audio samples. In a typical configuration a Sangoma board will give you 160 samples of audio each 20ms (most likely encoded in ulaw, that can be verified in /etc/wanpipe/wanpipex.conf for the FXO/FXS board, look at the TDMV_LAW parameter). There is an openzap API command to record the digital audio read from the board, check ZAP_COMMAND_TRACE_INPUT and ZAP_COMMAND_TRACE_OUTPUT. What we currently lack is a FreeSWITCH command in mod_openzap to enable/disable the audio tracing. It should be trivial to hack mod_openzap.c and add that command, if you need help with that ping me on IRC (moy). -- Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy(a)sangoma.com

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[Freeswitch-dev] Doxygen help
by Yossi Neiman 24 Feb '10

24 Feb '10

I am not sure how many folks make use of the Doxygen documentation. I know I do all the time. I find that it provides a nice conceptual view of FreeSWITCH's API, which can make it very easy to find the functions and data types needed for writing modules for FreeSWITCH. A person by the name of Mohammad Shahzad (apologies if I misspelled it) started to do a lot of work on revamping the Doxygen configuration for FreeSWITCH a few months back. The problem with the work that he did was that he used Doxygen 1.6.x specific configuration parameters that are not understood by the Doxygen 1.4.x tree (which is what we currently use in FreeSWITCH) or even the 1.5.x tree which is what is in Slackware. In my discussions with Michael Collins we have decided that it might not hurt to go ahead and move the Doxygen version forward to match the 1.6.x tree. We are looking for somebody who has interest in working with the Doxygen configuration to continue the work that Mohammad had started and to help the project out. If you are such a person, please contact me off-list and I can provide further information. Thanks! Yossi Neiman

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[Freeswitch-dev] Issue with making calls through FSComm and Build issues with Freeswitch 1.0.5 latest updated on 24-Feb-2010 04:05
by Suneel Papineni 24 Feb '10

24 Feb '10

Hi, I am trying to make FSComm work with Freeswitch (1.0.5 latest), but failed to make any calls. FSComm is getting registered properly and I can see SIP messages at Freeswitch. When a call is tried to make from one FSComm to another, there is no INVITE message seen in Freeswitch and also in wireshark traces as well. (OS environment is Windows XP 32-bit ). Could someone let me know if there is any specific config changes to be made to FSComm to work. (I am trying initially with FSComm pre-build binary version. Results are same). As per previous suggestions I tried to work with the latest version, downloaded Freeswitch 1.0.5 Latest updated on 24th February 2010 at 4:00am from "http://latest.freeswitch.org/". When I tried to build this, it is giving 4 errors. So build is failing. Errors are as follows: Error 201 error C2491: 'spandsp_stop_inband_dtmf_session' : definition of dllimport function not allowed d:\FS\freeswitch-1.0.5-latest24022010\freeswitch-1.0.5-20100224-0400\src \mod\applications\mod_fax\mod_fax.c 812 mod_fax Error 202 error C2491: 'spandsp_inband_dtmf_session' : definition of dllimport function not allowed d:\FS\freeswitch-1.0.5-latest24022010\freeswitch-1.0.5-20100224-0400\src \mod\applications\mod_fax\mod_fax.c 825 mod_fax Error 213 error LNK2019: unresolved external symbol _sip_dig_function referenced in function _mod_sofia_load mod_sofia.obj mod_sofia Error 214 fatal error LNK1120: 1 unresolved externals D:\FS\freeswitch-1.0.5-latest24022010\freeswitch-1.0.5-20100224-0400\Deb ug\mod\mod_sofia.dll mod_sofia I am trying to build Freeswitch for a windows 32-bit system. Tried to build both Debug and Release versions but failed with the above errors. Could someone let me know from where I need to download the latest version (possibly without errors). Thanks & Regards Suneel ************************************************************************* Please consider the environment before printing this e-mail ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN *************************************************************************

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[Freeswitch-dev] Multicast stream listener module
by MohammedShehzad 24 Feb '10

24 Feb '10

Hello everybody, I am working on multicast listener format module, So that Freeswitch can listen on multicasted stream as example below: <action application="playback" data="rtp://224.168.168.168:34567"/> Please see on thread named 'stream a file multicast with mod_esf' on http://lists.freeswitch.org/pipermail/freeswitch-users/2010-February/026163… I am testing that module against the audio multicasted by mod_esf module. I got some success with help of existing mod_native_file module, and I able to receive and play the multicasted RAW PCMU RTP (modified the skel of format module provided by brian), so that caller can hear the multicast by other Freeswitch server, but when i change the caller's codec from PCMU to something else, it breaks. Then I looked into mod_sndfile for decoding PCMU to other codec, but it seems that module is using libsndfile, which reads sound file directly and decode them to L16. If something similar to libsndfile is available, which work on stream instead of reading file, then it would surely work. Please Let me know your ideas on this, as I don't know how exactly Freeswitch encoding-decoding works? -- -MohammedShehzad

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