November 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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Internal profile startup fails if ssl/tls enabled
by Farhan Husain 11 Nov '10

11 Nov '10

Hello, I enabled ssl in my FS installation. Here are few lines from my vars.xml:  <X-PRE-PROCESS cmd="set" data="sip_tls_version=tlsv1"/>  <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/> <X-PRE-PROCESS cmd="set" data="internal_sip_port=5060"/> <X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/> <X-PRE-PROCESS cmd="set" data="*internal_ssl_enable=true*"/> <X-PRE-PROCESS cmd="set" data="internal_ssl_dir=$${base_dir}/conf/ssl"/> Then I restarted my server and found the following errors: 2010-11-10 20:22:37.555517 [ERR] sofia.c:1446 Error Creating SIP UA for profile: internal-ipv6 2010-11-10 20:22:37.576362 [ERR] sofia.c:1446 Error Creating SIP UA for profile: internal The "sofia status" command does not show the internal and internal-ipv6 profiles. This wiki page: http://wiki.freeswitch.org/wiki/SIP_TLS#If_your_SIP_profile_fails_to_start_… talks about exactly this problem and it suggest that libssl-dev is not installed. But when I checked my Ubuntu 10.04 I found it to be installed. I also installed "libcurl4-openssl-dev" and then did "make; make clean; make install" and reran the server. However, the problem persists. Can anyone tell me what else may be wrong with my server? Thanks, Farhan

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Wiki Questions
by Peder 11 Nov '10

11 Nov '10

Are there any guidelines as to what is acceptable as far as editing other people's work in the wiki? For example, on this page: http://wiki.freeswitch.org/wiki/Getting_Started_Guide If you go down to Command Line Parameters, it shows about 10 options. Right below it, it says "Recent options" and lists the 10 above, plus another 15 or so. So there is redundant information and it is confusing. It should all be put in one item. I don't mind editing when I see stuff like that, I just don't want to step on other people's toes. Also, under Starting in Unix-based Systems, it shows a block of code with a little computer symbol next to it. Then there is a block of code below it with no symbol. Then a block of code below it with a symbol again. Then way down at the bottom, there is a block of code with a pencil next to it. And a block of code with a light bulb next to it.. I'm sure the response will be "it's all yours, go ahead and fix it". I am just curious if there are guidelines, or if anybody looks over changes to make sure they look reasonable. Peder

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Extracting headers from REFER in dialplan?
by Marc Lewis 11 Nov '10

11 Nov '10

Is there any way to extract additional headers from the dialplan when FS receives a REFER message? The application I'm working on takes an incoming call, bridges it to another SIP server. The SIP server does some things with the call, then sends a REFER message back to FS with an extra header that I would like to use in the dialplan. The header shows in a trace, but not in any of the variables that I have access to. For example, the REFER message looks something like this (numbers and addresses changed to protect the innocent): REFER sip:gw+testserver@192.168.1.95:5060;tport=tcp;transport=tcp;gw=testserver SIP/2.0 FROM: <sip:4255551212@192.168.1.155>;epid=D4A38B23EF;tag=81a8713cf TO: <sip:testserver@192.168.1.155;transport=tcp>;tag=UNBQ363BgNvHa CSEQ: 1 REFER CALL-ID: b1aaff2e-6869-122e-53b9-001b216b53af MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 192.168.1.155:5060;branch=z9hG4bK606d7bc CONTACT: <sip:testing01:5060;transport=Tcp;maddr=192.168.1.155;ms-opaque=eaed293d6f65e314>;automata CONTENT-LENGTH: 0 REFER-TO: <sip:14255551213@192.168.1.95> REFERRED-BY: <sip:4255551212@192.168.1.155> X-Information: my dog has fleas I would then like to have something like this in the 4255551213 extension: <action application="eval" data="${chat(jingle|testing@fs.jabber.server|agent@another.jabber.server|${caller_id_number} to Queue - ${variable_sip_h_X-Information})}"/> The problem is the "X-Information" header is not exported by sofia as far as I can tell. Is there any way to get this variable via the XML dialplan? Thanks in advance. - Marc

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