- FreeSWITCH-users - FreeSWITCH Mailing Lists

bypass_media=true & using continue_on_fail
by Matthew Fong 13 Nov '10

13 Nov '10

I am trying to write an application that bypasses media between 2 endpoints. If I set both legs to bypass_media=true then bypass media works GREAT, but if leg B (a SIP provider) returns an error code that is found inside continue_on_fail, then the leg A will immediately hangup instead of continuing on with the dial plan. ignore_early_media is set to true both on leg A (FS) and this program, which is suppose to do switching amongst various voip providers with a fail-over to the next provider if the first provider throws an error inside continue_on_fail I tried programming around this by instead setting bypass_media=false initially and listening to the to the CHANNEL_BRIDGE event and issuing a uuid_media off command but this causes the first half second or so of audio not to be transmitted between the endpoints. Thanks.

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Call a registered user
by Christian Hiller 13 Nov '10

13 Nov '10

How can i call a registered User in Freeswitch, but send any destination number, like i am sending calls to a gateway. It is working in asterisk without any problem, but i wasnt able to bridge or originate a call like this in FS Thanks in advance Chris

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FreeSWITCH Snapshot Information
by Michael Collins 13 Nov '10

13 Nov '10

Hello all, It was recently brought to my attention that the freeswitch-snapshot files were not getting updated properly. It turns out that the cron process was still using the SVN repo! Our bad. I have fixed the snapshot to use the git repo. I manually ran the script and it created new snapshot files which can be downloaded now. I will monitor the cron job this weekend to confirm if it is properly generating new files each night. Note: it can take up to 24 hours from the time the cron job runs to the time you see the files over at files.freeswitch.org so be patient. The new files are: freeswitch-snapshot.tar.bz2 freeswitch-snapshot.tar.bz2.md5 freeswitch-snapshot.tar.bz2.sha1 freeswitch-snapshot.tar.gz freeswitch-snapshot.tar.gz.md5 freeswitch-snapshot.tar.gz.sha1 The files should be dated 2010-Nov-12. Any files earlier than this should be considered unreliable. Feel free to try the snapshot on your system. Remember that the snapshots are pre-bootstrapped so you can go straight to the ./configure && make install process. Email me off list if you have any questions. Thanks, Michael

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mod_callcenter fails on load with the latest snapshot
by Frank Park 12 Nov '10

12 Nov '10

I am having problem loading mod_callcenter to freeswitch. No logs are generated at startup. I've narrowed it down to this mod, after doing so I've commented it out from auto_configs/modules.conf.xml and the fs functions normally (minus the callcenter). When i try to manually load mod_callcenter from fs_cli, the system immediate fails with "Socket interrupted, bye!" message. Anybody experiencing this? I wanted to revert to the previous version that I used to work (upto some level), but 1.0.6.tar.gz doesn't seem to include this module either.. Can anybody provide me with the source file that is currently working on your system? Thanks, Frank ----=======================---- Frank Park Telonium Communications, LLC frank(a)telonium.com http://www.telonium.com <http://www.doubleshotwebdev.com/>404-566-8888 x1001 Office 404-939-4242 Cell ----=======================----

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files.freeswitch.org
by Frank Park 12 Nov '10

12 Nov '10

I know this is pretty low priority, but can someone update the freeswitch-1.0.latest.* and freeswitch-1.0.snapshot.* to reflect the latest and the snapshot? I downloaded 1.0.latest.tar.gz, which was the version 1.0.1. Thanks, Frank ----=======================---- Frank Park Telonium Communications, LLC frank(a)telonium.com http://www.telonium.com <http://www.doubleshotwebdev.com/>404-566-8888 x1001 Office 404-939-4242 Cell ----=======================----

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mod opal installation
by budi wibowo 12 Nov '10

12 Nov '10

hi i have fs running on centos 5.5 and want to activate mod opal to support h323 endpoint. i flollow http://wiki.freeswitch.org/wiki/Mod_opal but failed to install opal itself with error [CC] spandsp_fax.cpp spandsp_fax.cpp: In member function âbool FaxT38::Open(t38_core_state_t*)â: spandsp_fax.cpp:498: error: ât38_core_get_logging_stateâ was not declared in this scope spandsp_fax.cpp: In function âstd::ostream& operator<<(std::ostream&, const MyStats&)â: spandsp_fax.cpp:630: error: âconst struct MyStatsâ has no member named âpages_txâ spandsp_fax.cpp:631: error: âconst struct MyStatsâ has no member named âpages_rxâ spandsp_fax.cpp:638: error: âconst struct MyStatsâ has no member named âerror_correcting_mode_retriesâ spandsp_fax.cpp: In member function âbool FaxTIFF::Open(t30_state_t*)â: spandsp_fax.cpp:685: error: ât30_get_logging_stateâ was not declared in this scope spandsp_fax.cpp: In destructor âvirtual T38_PCM::~T38_PCM()â: spandsp-0.0.5 is installed on system any help is highly appreciated

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: Got warning when using gizmo as the SIP client
by Jian Ren 12 Nov '10

12 Nov '10

Hi, I am trying to use my old Nokia 770 as a wifi SIP phone. Since gizmo is the only one available there, I setup the "second account" to connect my FS server. But every time when it got connected after dialing out, or accepting calls. I saw this warning in FS console. Could anyone tell me what's the meaning? Could I just ignore it? The last line seems like my device will be let die :-). 2010-11-11 19:07:47.067361 [WARNING] mod_sofia.c:1036 Asynchronous PTIME not supported, changing our end from 30 to 20 2010-11-11 19:07:50.276817 [WARNING] mod_sofia.c:1036 Asynchronous PTIME not supported, changing our end from 30 to 20 2010-11-11 19:07:57.596854 [WARNING] mod_sofia.c:1036 Asynchronous PTIME not supported, changing our end from 30 to 20 2010-11-11 19:07:57.596854 [WARNING] mod_sofia.c:1089 Ok, some devices *cough* X-lite *cough* seem to continue to lie over and over again so I guess we'll leave well-enough alone and let them lie Thanks! Jian

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TTS Related (Freeswitch, Asterisk, Solaris)
by RR 12 Nov '10

12 Nov '10

Hello All, I have a whole bunch of questions regarding the TTS functionality in FreeSWITCH and Just Generally about Freeswitch on a Sparc based Solaris Platform and then the combo of the two, whether FreeSWITCH can be built on a Sparc based Solaris platform when compiled with a "reasonably" good TTS module be it Cepstral, FLite OR NeoSpeech etc? 1) Can anyone reflect on any experience they might have had in terms of the performance difference between TTS on Asterisk (let's say Asterisk 1.8) and FreeSWITCH? Is one better than the other? 2) Can One or the other or both be compiled successfully on a Sparc based Solaris platform? 3) Would it benefit if I was to use a dedicated box for TTS and throw calls to that box for TTS processing and send back media the way it came after the sound files are generated from the TTS-process? 4) Please note that the quality of the Translation doesn't need to be impeccable or mimic human speech. It just needs to be understandable enough esp. over mobile phones. The more complex part about it is, that the user directory can/will have names from all sorts of ethnicities/nationalities which may mess up a lot of these TTS engines. So what is the best approach there in this mixed environment 5) Does the OpenMRCP play any part in any of this? Would I benefit with that? I think this is the crux of my concerns. Any ideas and/or suggestions to kick off the discussion would be very very much appreciated. Kind Regards, RR

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Broadsoft BLA on SPA525g
by jay binks 11 Nov '10

11 Nov '10

As per my post on the cisco forums ... https://supportforums.cisco.com/thread/2052011 <https://supportforums.cisco.com/thread/2052011>can anyone in the freeswitch community help me here. Im using a SPA525g ( with SPA500s ) registered to Freeswitch. Im trying to use BLF on the sidecart with this configuration <Server_Type group="Att_Console/General">Broadsoft</Server_Type> <Unit_1_Key_1 group="Att_Console/Unit_1">fnc=sd+cp+blf;sub= 99007(a)10.12.253.10</Unit_1_Key_1> <Unit_1_Key_2 group="Att_Console/Unit_1">fnc=cp+blf;sub=99007(a)10.12.253.10 </Unit_1_Key_2> <Unit_1_Key_3 group="Att_Console/Unit_1">fnc=blf;sub=99007(a)10.12.253.10 </Unit_1_Key_3> <Unit_1_Key_4 group="Att_Console/Unit_1">fnc=sd;sub=99007(a)10.12.253.10 </Unit_1_Key_4> <Unit_1_Key_5 group="Att_Console/Unit_1">fnc=blf+sd;sub=99007(a)10.12.253.10 </Unit_1_Key_5> <Unit_1_Key_6 group="Att_Console/Unit_1">fnc=blf+sd;sub=99007(a)10.12.253.10 ;ext=99007(a)10.12.253.10</Unit_1_Key_6> <Unit_1_Key_7 group="Att_Console/Unit_1">fnc=blf+sd+cp;sub= 99007(a)10.12.253.10</Unit_1_Key_7> <Unit_1_Key_8 group="Att_Console/Unit_1">fnc=blf+sd+cp;sub=99007@ $PROXY</Unit_1_Key_8> All keys flash orange except Key_4 ( which is simply a speed dial ) I can see freeswitch accepting my SIP SUBSCRIBE ( With - SIP/2.0 202 Accepted ) and then sending me SIP NOTIFY Messages ( to which the 525g responds SIP/2.0 200 OK ) however the lights continue to flash orange. in my SIP Profile I DO have <param name="manage-presence" value="true"/> <param name="manage-shared-appearance" value="true"/> and also <param name="dbname" value="my-shared-db_presence"/> <param name="presence-hosts" value="$${domain}"/> but this dosnt look to me like its a freeswitch config issue, since the pcap shows FS Sending NOTIFIES. ( please can someone confirm this looks ok from the freeswitch side ) please can someone point me in the right direction here, Id love to know whats wrong. PLEASE Find attached the gzipped pcap. -- Sincerely Jay

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can't add $us-ring at end of ringback string? FS Aborts
by Mario G 11 Nov '10

11 Nov '10

I want a custom ringback play 2 files then normal ringing. With ${us-ring} added it does not work and FS aborts. Without ${us-ring} the files play but no ring afterwards. I tried ${us-ring}, tone_stream://${us-ring}, local_stream:${us-ring}, etc. Thanks, Mario G <action application="set" data="ringback=file_string://${mvsounds}hi-long.wav!//${sayname}!tone_stream://${us-ring}"/> Freeswitch Aborts: 2010-11-11 12:26:08.413942 [DEBUG] switch_core_file.c:176 File ///usr/local/freeswitch/conf/mvsounds/mario.wav sample rate 16000 doesn't match requested rate 8000 Assertion failed: (fh->file_interface != ((void *)0)), function switch_core_file_read, file src/switch_core_file.c, line 210. Abort trap (core dumped)

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