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detect_speech unimrcp ... ends with 'Aborted (core dumped)'
by Dmitriy Borisov 05 Jan '20

05 Jan '20

Hi, All! I have a strange problem... I have a big Lua script to initiate a call. It starts detect_speech with > execute_on_media=detect_speech unimrcp ... Script initiate Leg B on leg A answer. And when leg B answers FreeSWITCH halted with this message > 2019-12-25 00:00:55.903734 [INFO] switch_ivr_async.c:219 Digit parser > DPTOOLS: Setting realm to 'temp' > 2019-12-25 00:00:55.903734 [DEBUG] switch_ivr_async.c:344 Digit parser > DPTOOLS: binding *#/temp/0 callback: 0x7fb375cf9700 data: 0x7fb2fc119818 > 2019-12-25 00:00:55.903734 [DEBUG] switch_ivr_bridge.c:1672 > (sofia/internal/1-FROM_GEN@10.17.19.111:5060) State Change > CS_SOFT_EXECUTE -> CS_CONSUME_MEDIA > 2019-12-25 00:00:55.903734 [DEBUG] switch_core_state_machine.c:584 > (sofia/internal/1-FROM_GEN@10.17.19.111:5060) Running State Change > CS_CONSUME_MEDIA (Cur 2 Tot 2) > 2019-12-25 00:00:55.903734 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/1-FROM_GEN@10.17.19.111:5060) State CONSUME_MEDIA > 2019-12-25 00:00:55.903734 [DEBUG] switch_ivr_bridge.c:1063 sofia/internal/ > 1-FROM_GEN@10.17.19.111:5060 CUSTOM HOLD > 2019-12-25 00:00:55.903734 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/1-FROM_GEN@10.17.19.111:5060) State CONSUME_MEDIA going > to sleep > freeswitch: src/switch_buffer.c:348: switch_buffer_zero: Assertion > `buffer->data != ((void *)0)' failed. > Aborted (core dumped) > I need to fix it ASAP, but I don't know where to see at all! What is it can be? -- With best regards Dmitry Borisov

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calling sleep after answer results in no audio
by Babak Yakhchali 04 Jan '20

04 Jan '20

Hi trying below dialplan results in no audio: <extension name="DailyIVR" continue="false" uuid="d8037520-2a15-11ea-9a9f-9bf4580ab5b8"> <condition field="destination_number" expression="^110$"> <action application="answer" data=""/> <action application="sleep" data="1000"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="ringback=local_stream://default"/> <action application="set" data="transfer_ringback=local_stream://default"/> <action application="set" data="ivr_menu_uuid=d8049250-2a15-11ea-9f14-1b8cf3ef5a32"/> <action application="ivr" data="d8049250-2a15-11ea-9f14-1b8cf3ef5a32"/> <action application="" data=""/> </condition> </extension> but if I move *sleep before answer* everything works fine. Using tcpdump shows that when sleep is called after answer no rtp is established but if I move sleep before answer rtp is established I'm using freeswitch 1.8.7 on debian 8 thanks

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Re: [Freeswitch-users] No audio for conference call
by David P 02 Jan '20

02 Jan '20

You need a more detailed log, which on linux is usually at /var/log/freeswitch/freeswitch.log IIRC. Search your file system for this filename. You might also need to increase your logging level; do a web search for: freeswitch log level

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bypass_media_after_bridge
by Oleg Belousov 01 Jan '20

01 Jan '20

Happy festive season. Would you please guide me with usage of 'bypass_media_after_bridge' feature? Call flow is as following (running in lua script): 1. freeswitch preanswer call from UAC, then play some prompts 2, create second_session 3. when second session is ready - assign bypass_media_after_bridge=true and, see var is assigned. EXECUTE [depth=0] sofia/external/XXXXXXXXXX set(bypass_media_after_bridge=true) 2019-12-31 14:21:19.253279 [DEBUG] mod_dptools.c:1672 SET sofia/external/XXXXXXXXXX [bypass_media_after_bridge]=[true] 4. bridge both calls. At that time expect freeswtch fire re-invite to get out of rtp path, however do not see any re-invites. extract from script: ---- ConnectStr = "sofia/gateway/aed8dd2f-202f-4093-9851-436e21bafcf4/"..DN second_session = freeswitch.Session(ConnectStr) if (second_session:ready()) then second_session:execute("set","bypass_media_after_bridge=true") freeswitch.bridge(session, second_session) --- What could be wrong in my script? version: FreeSWITCH Version 1.10.1-release-12 -- Regards, Oleg

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Re: [Freeswitch-users] No audio for conference call
by David P 01 Jan '20

01 Jan '20

Search your log for 'Choose rtp'; the IP number following this shows what FS will use for a media stream. There is a line like this for each stream. For your scenario, you want these not to be private IPs. If they are private, you can try to auto-reject them by changing acl.conf.xml. But you'll have limited success due to this bug https://github.com/signalwire/freeswitch/issues/157

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FreeSWITCH v.1.10.2 Released
by Andrey Wolk 31 Dec '19

31 Dec '19

WooHooo! We are happy to present you the freshest FreeSWITCH v.1.10.2 Release! This release contains ~102 improvements. https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.10.x+Rele… <https://slack-redir.net/link?url=https%3A%2F%2Ffreeswitch.org%2Fconfluence%…> https://github.com/signalwire/freeswitch/releases/tag/v1.10.2 <https://slack-redir.net/link?url=https%3A%2F%2Fgithub.com%2Fsignalwire%2Ffr…> Happy New Year everyone!

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TLS1.2 ONLY
by Nathan Stratton 30 Dec '19

30 Dec '19

# vars.xml <X-PRE-PROCESS cmd="set" data="sip_tls_version=tlsv1.2"/> <X-PRE-PROCESS cmd="set" data="sip_tls_ciphers=ALL:!ADH:!LOW:!EXP:!RC4:!3DES:!MD5:@STRENGTH"/> # internal.xml <param name="wss-binding" value=":443"/> <param name="tls-version" value="$${sip_tls_version}"/> <param name="tls-ciphers" value="$${sip_tls_ciphers}"/> However when I check out my server with https://www.ssllabs.com/ssltest/analyze.html it shows TLS1.1 along with TLS1.2. Any way to JUST have the server answer TLS1.2? Also is it possible to set the order of cipher suites and get rid of: Cipher Suites # TLS 1.2 (server has no preference) TLS_RSA_WITH_AES_128_CBC_SHA (0x2f) *WEAK* 128 TLS_RSA_WITH_CAMELLIA_128_CBC_SHA (0x41) *WEAK* 128 TLS_RSA_WITH_AES_128_CBC_SHA256 (0x3c) *WEAK* 128 TLS_RSA_WITH_AES_128_GCM_SHA256 (0x9c) *WEAK* 128 TLS_RSA_WITH_AES_256_CBC_SHA (0x35) *WEAK* 256 TLS_RSA_WITH_CAMELLIA_256_CBC_SHA (0x84) *WEAK* 256 TLS_RSA_WITH_AES_256_CBC_SHA256 (0x3d) *WEAK* 256 TLS_RSA_WITH_AES_256_GCM_SHA384 (0x9d) *WEAK* 256 ><> nathan stratton

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Re: [Freeswitch-users] Upgrading DTLS
by Piotr Gregor 30 Dec '19

30 Dec '19

Hi Nathan, The best option is to upgrade system to one that ships with SSL above 1.1.0, like for instance new Debian 10 "Buster". peter@photon:~/$ openssl version OpenSSL 1.1.1d 10 Sep 2019 cheers, Piotr Gregor Software Engineer M: (+44) 07483 866 525 www: dataandsignal.com On Thu, Dec 12, 2019 at 4:03 PM kaiduan xie via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: kaiduan xie <kaiduanx(a)yahoo.ca> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Thu, 12 Dec 2019 15:31:40 +0000 (UTC) > Subject: Re: [Freeswitch-users] Upgrading DTLS > Looks like your SSL version is old. > > #if OPENSSL_VERSION_NUMBER >= 0x10100000 > > https://github.com/signalwire/freeswitch/blob/master/src/switch_rtp.c#L3757 > > The following is the SSL version on 16.0.4 Ubuntu. > > *~*:*/usr/include/openssl*$ grep -R VERSION_NUMBER . -n > > ./crypto.h:152:# define SSLEAY_*VERSION_NUMBER* OPENSSL_*VERSION_NUMBER* > > ./ssl.h:2868:# define SSL_R_BAD_PROTOCOL_*VERSION_NUMBER* > 116 > > ./ssl.h:3164:# define SSL_R_WRONG_*VERSION_NUMBER* > 267 > > ./pem.h:589:# define PEM_R_BAD_*VERSION_NUMBER* > 117 > > ./opensslv.h:33:# define OPENSSL_*VERSION_NUMBER* 0x1000207fL > > ./opensslv.h:83: * The current library version is stored in the macro > SHLIB_*VERSION_NUMBER*, > > ./opensslv.h:91:# define SHLIB_*VERSION_NUMBER* "1.0.0" > > :*/usr/include/openssl*$ lsb_release -a > > No LSB modules are available. > > Distributor ID: Ubuntu > > Description: Ubuntu 16.04.5 LTS > > Release: 16.04 > > Codename: xenial > > > > On Thursday, December 12, 2019, 09:00:42 a.m. CST, Mirko Brankovic < > mirkobrankovic(a)gmail.com> wrote: > > > VERSION="16.04.6 LTS (Xenial Xerus)" > ~# dpkg -l | grep openssl > ii libcurl4-openssl-dev:amd64 7.47.0-1ubuntu2.14 > amd64 development files and documentation for > libcurl (OpenSSL flavour) > ii libgnutls-openssl27:amd64 3.4.10-4ubuntu1.5 > amd64 GNU TLS library - OpenSSL wrapper > ii libxmlsec1-openssl 1.2.20-2ubuntu4 > amd64 Openssl engine for the XML security library > ii openssl 1.0.2g-1ubuntu4.15 > amd64 Secure Sockets Layer toolkit - > cryptographic utility > > > But the real problem appeared on another webrtc gateway (Janus) that > required TLS 1.2 minimum > > On Thu, Dec 12, 2019 at 3:48 PM kaiduan xie via FreeSWITCH-users < > freeswitch-users(a)lists.freeswitch.org> wrote: > > > > > ---------- Forwarded message ---------- > From: kaiduan xie <kaiduanx(a)yahoo.ca> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Thu, 12 Dec 2019 14:47:41 +0000 (UTC) > Subject: Re: [Freeswitch-users] Upgrading DTLS > What OS and version you run FS on? What is the openssl version on the box? > > /Kaiduan > > On Thursday, December 12, 2019, 03:29:32 a.m. CST, Mirko Brankovic < > mirkobrankovic(a)gmail.com> wrote: > > > I had a same problem, and I see you can set it in vars.conf: > > https://github.com/signalwire/freeswitch/blob/master/conf/vanilla/vars.xml#… > but since we have a custom module, it didn't work for me, so I replaced > OpenSSL with BorringSSL and fixed it that way :D > > On Wed, Dec 11, 2019 at 10:05 PM Nathan Stratton <nathan(a)robotics.net> > wrote: > > > Seeing this error on FreeSWITCH 1.10.1 > > 2019-12-11 00:19:34.288375 [ERR] switch_rtp.c:3266 video Handshake failure > 1. This may happen when you use legacy DTLS v1.0 (legacyDTLS channel var is > set) but endpoint requires DTLS v1.2. > > Any idea how to upgrade DTLS to 1.2? I could not find much with a google > search. > > ><> > nathan stratton > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales(a)freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > 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 > https://freeswitch.com > > > > -- > Regards, > Mirko > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales(a)freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > 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 > https://freeswitch.com > > > > ---------- Forwarded message ---------- > From: kaiduan xie via FreeSWITCH-users < > freeswitch-users(a)lists.freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Thu, 12 Dec 2019 06:48:21 -0800 (PST) > Subject: Re: [Freeswitch-users] Upgrading DTLS > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales(a)freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > 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 > https://freeswitch.com > > > > -- > Regards, > Mirko > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales(a)freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > 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 > https://freeswitch.com > > > > ---------- Forwarded message ---------- > From: kaiduan xie via FreeSWITCH-users < > freeswitch-users(a)lists.freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Thu, 12 Dec 2019 08:03:24 -0800 (PST) > Subject: Re: [Freeswitch-users] Upgrading DTLS > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales(a)freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > 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 > https://freeswitch.com

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PHP ESL Installation
by Vishal Pai 27 Dec '19

27 Dec '19

Hello everyone Season's Greetings. I am having php7.0 installed on my Debian 9 OS Using below command apt install php7.0 libapache2-mod-php7.0 php7.0-common php7.0-pdo php7.0-mbstring php7.0-xmlrpc php7.0-soap php7.0-gd php7.0-xml php7.0-intl php7.0-mysql php7.0-cli php7.0-mcrypt php7.0-ldap php7.0-zip php7.0-curl php7.0-dev Now when i compile esl under /usr/src/freeswitch-1.10.1.-release/libs/esl make phpmod-install i am getting following errors make MYLIB=".././.libs/libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-I/usr/src/freeswitch-1.10.1.-release/libs/esl/src/include -I/usr/include/uuid -I/usr/src/freeswitch-1.10.1.-release/src/include -I/usr/src/freeswitch-1.10.1.-release/src/include -I/usr/src/freeswitch-1.10.1.-release/libs/libteletone/src -fPIC -ffast-math -Werror -Wno-unused-result -Wno-misleading-indentation -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DCJSON_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL" CXXFLAGS="-I/usr/src/freeswitch-1.10.1.-release/libs/esl/src/include -I/usr/src/freeswitch-1.10.1.-release/src/include -I/usr/src/freeswitch-1.10.1.-release/src/include -I/usr/src/freeswitch-1.10.1.-release/libs/libteletone/src -fPIC -ffast-math -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DCJSON_API_VISIBILITY=1 -DHAVE_VISIBILITY=1" CXX_CFLAGS="" -C php make[1]: Entering directory '/usr/src/freeswitch-1.10.1.-release/libs/esl/php' g++ -I/usr/src/freeswitch-1.10.1.-release/libs/esl/src/include -I/usr/src/freeswitch-1.10.1.-release/src/include -I/usr/src/freeswitch-1.10.1.-release/src/include -I/usr/src/freeswitch-1.10.1.-release/libs/libteletone/src -fPIC -ffast-math -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DCJSON_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I/usr/src/freeswitch-1.10.1.-release/libs/esl/src/include -I/usr/src/freeswitch-1.10.1.-release/src/include -I/usr/src/freeswitch-1.10.1.-release/src/include -I/usr/src/freeswitch-1.10.1.-release/libs/libteletone/src -fPIC -ffast-math -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DCJSON_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I/usr/include/php/20151012 -I/usr/include/php/20151012/main -I/usr/include/php/20151012/TSRM -I/usr/include/php/20151012/Zend -I/usr/include/php/20151012/ext -I/usr/include/php/20151012/ext/date/lib -Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o esl_wrap.cpp:876:105: error: macro "zend_hash_update" passed 6 arguments, but takes just 3 zend_hash_update(HASH_OF(z), (char*)"_cPtr", sizeof("_cPtr"), (void*)&resource, sizeof(zval), NULL); esl_wrap.cpp:1246:50: error: macro "ZVAL_STRING" passed 3 arguments, but takes just 2 ZVAL_STRING(return_value, (char *)result, 1); ^ esl_wrap.cpp:1521:50: error: macro "ZVAL_STRING" passed 3 arguments, but takes just 2 ZVAL_STRING(return_value, (char *)result, 1); ^ esl_wrap.cpp:1610:50: error: macro "ZVAL_STRING" passed 3 arguments, but takes just 2 ZVAL_STRING(return_value, (char *)result, 1); ^ esl_wrap.cpp:1640:50: error: macro "ZVAL_STRING" passed 3 arguments, but takes just 2 ZVAL_STRING(return_value, (char *)result, 1); ^ esl_wrap.cpp:1670:50: error: macro "ZVAL_STRING" passed 3 arguments, but takes just 2 ZVAL_STRING(return_value, (char *)result, 1); ^ esl_wrap.cpp:1918:50: error: macro "ZVAL_STRING" passed 3 arguments, but takes just 2 ZVAL_STRING(return_value, (char *)result, 1); ^ esl_wrap.cpp:1948:50: error: macro "ZVAL_STRING" passed 3 arguments, but takes just 2 ZVAL_STRING(return_value, (char *)result, 1); ^ esl_wrap.cpp: In function ‘void SWIG_landfill(zend_resource*)’: esl_wrap.cpp:813:51: error: ‘rsrc’ was not declared in this scope static ZEND_RSRC_DTOR_FUNC(SWIG_landfill) { (void)rsrc; } ^~~~ esl_wrap.cpp: In function ‘void SWIG_ZTS_SetPointerZval(zval*, void*, swig_type_info*, int)’: esl_wrap.cpp:836:66: error: ‘ZEND_REGISTER_RESOURCE’ was not declared in this scope ZEND_REGISTER_RESOURCE(z, value, *(int *)(type->clientdata)); ^ esl_wrap.cpp:857:29: error: ‘MAKE_STD_ZVAL’ was not declared in this scope MAKE_STD_ZVAL(resource); ^ esl_wrap.cpp:858:73: error: ‘ZEND_REGISTER_RESOURCE’ was not declared in this scope ZEND_REGISTER_RESOURCE(resource, value, *(int *)(type->clientdata)); ^ esl_wrap.cpp:863:93: error: cannot convert ‘char*’ to ‘zend_string* {aka _zend_string*}’ for argument ‘1’ to ‘zend_class_entry* zend_lookup_class(zend_string*)’ result = zend_lookup_class(classname, SWIG_PREFIX_LEN + type_name_len, &ce TSRMLS_CC); ^ esl_wrap.cpp:866:83: error: cannot convert ‘char*’ to ‘zend_string* {aka _zend_string*}’ for argument ‘1’ to ‘zend_class_entry* zend_lookup_class(zend_string*)’ result = zend_lookup_class((char *)type_name, type_name_len, &ce TSRMLS_CC); Which doesn't get compiled successfully. Can anyone let me know what i am doing wrong here.

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SIP trunk Cisco-Freeswitch: Cannot return from HOLD
by Yehavi Bourvine 25 Dec '19

25 Dec '19

Hello, I am connecting FS 1.8 with Cisco CUCM. A call is initiated correctly, and there is two way audio. When the Cisco phone places the call on hold, cisco sends SDP with "sendonly". Upon return from HOLD, Cisco sends INVITE without SDP and FS replies with SDP with "sendonly"... As a result, there is a one way audio from FS to Cisco. I can roll this problem to Cisco as it does not cancels the "sendonly" from its side, but why FS sends also "sendonly"? This probably causes Cisco to not send audio anymore. Anyone noticed this problem and has a solution? Thanks, __yehavi:

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