July 2019 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Problem with Dial Plan and Diversion Headers
by Rick Ewart 22 Jul '19

22 Jul '19

Hi all. I am having an issue with calls not being routed correctly in our freeswitch system and I am having trouble figuring out the real cause. We are using the following in our routing plan to determine the number being called on an incoming call, which seems pretty straight forward: <condition break="never"> <action inline="true" application="set" data="dest_num=${destination_number}"/> <action inline="true" application="set" data="did_num=0"/> </condition> Generally speaking it works fine - we have been in production for years now doing many millions of calls. But an issue has cropped up with a call that comes in and is not being routed properly. Below is the SIP INVITE and header we receive (sanitized to protect the guilty). You will note that the INVITE has 15615551234 which is the correct DID on our system being called. However the "To" header specifies 14145241234 which if I understand what I read, is apparently what the caller dialed and it was diverted to our DID. Since my system never heard of 14145241234, it routed it to our default "Not in Service" message. In discussing with my carrier, they are saying its being delivered correctly and my system must be setup improperly for this situation. Since I never heard of the diversion header before this, I am at a bit of a loss (even after reading some articles on it). I thought I had a pretty good handle on this stuff but this is throwing me for a loop as it seems to me that the "To" should have been rewritten on the far end B2BUA (or somewhere before it got to me) to be the same as the INVITE but apparently not? I am trying to figure out what would be the most failsafe variable to use if destination_number is not it (as its what I see in all the FS Doc examples and elsewhere). Seems the INVITE is the only place the real number exists in this situation. I am searching for what that variable would be correct as the Channel Variables docs don't contain a lot of descriptions. I am thinking perhaps it should be sip_req_user? Since it's a production box, I figured I should ask the experts rather than guess on my own and make people grumpy due to some other unintended consequence I don't realize. I appreciate any help/explanation you can/are willing to provide. Thanks! Rick Ewart Session Initiation Protocol (INVITE) Request-Line: INVITE sip:15615551234@123.123.41.36:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 68.232.9.155:5060;branch=z9hG4bK0cB87133736bc4b79bd From: "ABC COMPANY" <sip:15615554162@68.232.9.155>;tag=gK0c5693cd To: <sip:14145241234@123.123.41.36> Call-ID: 191660912_117119432(a)68.232.9.155 CSeq: 271179 INVITE Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,OPTIONS Accept: application/sdp Contact: "ABC COMPANY" <sip:15615554162@68.232.9.155:5060> Remote-Party-ID: "ABC COMPANY" <sip:15615554162@68.232.9.155:5060>;privacy=off Diversion: <sip:+1414524123@68.232.9.155:5060>;privacy=off;screen=no;reason=unknown;c ounter=1 Supported: replaces Content-Length: 281 Content-Disposition: session; handling=required Content-Type: application/sdp

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Re: [Freeswitch-users] freeswitch package on ARM (raspberry pi)
by Igor Olhovskiy 19 Jul '19

19 Jul '19

There is a repo for arm maintaned by FusionPBX. echo "deb https://repo.fusionpbx.com/armhf stretch 1.8.4" > /etc/apt/sources.list.d/freeswitch.list echo "deb https://repo.fusionpbx.com/armhf jessie 1.6.20" > /etc/apt/sources.list.d/freeswitch.list curl https://repo.fusionpbx.com/public.key | apt-key add - On Jun 26 2019, at 8:22 pm, Bipin Patel via FreeSWITCH-users <freeswitch-users(a)lists.freeswitch.org> wrote: > hi, > > i was wondering if FS packages will be available for ARM raspbian > because we have been running multiple SOHO PBX using rpi for clients for > quiet some time now, currently we build from master on raspbian stretch > as well as jessie and it works well. We did some stress test with 50 > calls and 100 channels using rpi 3b+ with g711u and it works flawless as > well as we have it on rpi 2 also easily able to handle 20+ calls and now > with the rpi 4 im sure it can handle a lot more easily so why not > officially create packages for ARM rather than have to compile from > master. We also developed a custom CDR portal we supply with it which > gives the user the ability to create extenions, gateways etc etc as xml > rather than use a DB or xml curl etc to keep the resource consumption > low. > > > Regards, > Bipin > > > _________________________________________________________________________ > 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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Mod_Python
by Caio Assis 19 Jul '19

19 Jul '19

Good afternoon. I use mod_python to execute scripts, and I notice that when there's a relative high call traffic, freeswitch's performance decreases significantly. And I also noticed that, the more interaction the script has with freeswitch's session object (getVariable, execute, etc) the more the performance decreases. I kept the interaction to the minimum, as in 1 session.execute('multiset'). Any sugestions?

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strange behavior when bridge fails.
by Caragea Silviu 18 Jul '19

18 Jul '19

Hello, I'm using last stable FS 1.8.7 and when I'm doing a bridge to a not existing destination I can see that even after bridge fails and the LEG a receives 480, FS is still re-transmitting the INVITE few times. Before bridge I'm setting the following vars: continue_on_fail=true, hangup_after_bridge=true, progress_timeout=5, bypass_media=true And then I'm bridging using: {origination_caller_id_name=400000000,origination_caller_id_number=400000000,ignore_early_media=consume}sofia/external/ 44444444@172.17.33.28:6060 Bridge fails with PROGRESS_TIMEOUT after 5 seconds which is correct but on PCAP I can still see lot of retransmission's for the INVITE. Please see the attached image as well. 172.17.33.27 is the caller, 172.17.33.111 is FS and 172.17.33.28 is the IP where we are doing the bridge and where no server listen. It's this expected behavior ? Even after 16 seconds after the bridge fails the reinvite is sent. [image: Screen Shot 2019-07-18 at 12.46.53 PM.png] Silviu

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Limit cps per source IP and port
by António Silva 18 Jul '19

18 Jul '19

Hi, is it possible to limit the number of calls per source ip and port before the dialplan? I know that this can be done using mod_limit in the dialplan, i was thinking of a limit like the sessions-per-second global limit, this way less resources are used if some ip/port is flooding our machine. Another possibility is using the firewall... but i like this option less because i wont know of the drop calls. Thanks, -- Saludos / Regards / Cumprimentos António Silva

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ANI Source
by Sean Devoy 17 Jul '19

17 Jul '19

My provider has responded to a question regarding missing caller id to some recipients and said: Per our records we are receiving a 'null' value in the ANI/Source portion But my log says: [SNIP] set(ani=4104440002) 0d29c542 - [SNIP] [DEBUG] mod_dptools.c:1598 SET sofia/internal/414(a)fs-esta2.bizfocused.com [ani]=[4104440002] Any ideas? Thanks, Sean

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Call_Center announce-sound
by Moshe Rosenberg 11 Jul '19

11 Jul '19

Hi this is the XML code that i have in order to play some announce-sound that says press 1 to leave a message <param name="announce-sound" value="/var/lib/freeswitch/recordings/dp/on_hold_msg.wav"/> <param name="announce-frequency" value="30"/> <action application="set" data="cc_exit_keys=1"/> This works great, the problem if an agent picks up the phone while the announce-sound is palying the system will connect the call however the caller will still hear the announce sound and will only connect to agent when the sound is done -- Moshe Rosenberg Tel. 718 633 1444 Cel. 347 678 3993 www.data <http://www.dataday.consulting/>phone.cloud Moshe(a)dataphone.cloud

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Trying to compile debian 10
by Jose Figueroa 11 Jul '19

11 Jul '19

Hello, Today I was trying to compile v1.8 branch using Debian 10 but I'm getting this error on the make process: >>> Log: /usr/src/freeswitch$ make config.status: creating Makefile config.status: creating include/apr.h config.status: creating build/apr_rules.mk config.status: creating build/pkg/pkginfo config.status: creating apr-1-config config.status: WARNING: 'apr-config.in' seems to ignore the --datarootdir setting config.status: creating apr.pc config.status: creating test/Makefile config.status: creating test/internal/Makefile config.status: creating include/arch/unix/apr_private.h config.status: executing libtool commands rm: cannot remove 'libtoolT': No such file or directory config.status: executing default commands config.status: include/apr.h is unchanged config.status: include/arch/unix/apr_private.h is unchanged make[1]: Entering directory '/usr/src/freeswitch/libs/apr' make[2]: Entering directory '/usr/src/freeswitch/libs/apr' make[2]: Nothing to be done for 'local-all'. make[2]: Leaving directory '/usr/src/freeswitch/libs/apr' make[1]: Leaving directory '/usr/src/freeswitch/libs/apr' make[1]: Entering directory '/usr/src/freeswitch/libs/apr-util' Making all in xml/expat make[2]: Entering directory '/usr/src/freeswitch/libs/apr-util/xml/expat' make[3]: Entering directory '/usr/src/freeswitch/libs/apr-util/xml/expat/lib' /bin/bash ../libtool --mode=compile gcc -DHAVE_CONFIG_H -DPACKAGE='"expat"' -DVERSION='"expat_1.95.2"' -I. -I.. -I/usr/src/freeswitch/libs/apr-util/xml/expat/lib -g -O2 -c xmlparse.c libtool: compile: gcc -DHAVE_CONFIG_H -DPACKAGE=\"expat\" -DVERSION=\"expat_1.95.2\" -I. -I.. -I/usr/src/freeswitch/libs/apr-util/xml/expat/lib -g -O2 -c xmlparse.c -fPIC -DPIC -o xmlparse.o /bin/bash ../libtool --mode=link gcc -version-info 1:0:1 -g -O2 -o libexpat.la -rpath /usr/local/freeswitch/lib xmlparse.lo xmltok.lo xmlrole.lo libtool: link: rm -fr .libs/libexpat.a .libs/libexpat.la .libs/libexpat.lai libtool: link: ar cru .libs/libexpat.a xmlparse.o xmltok.o xmlrole.o ar: `u' modifier ignored since `D' is the default (see `U') libtool: link: ranlib .libs/libexpat.a libtool: link: ( cd ".libs" && rm -f "libexpat.la" && ln -s "../libexpat.la" "libexpat.la" ) make[3]: Leaving directory '/usr/src/freeswitch/libs/apr-util/xml/expat/lib' make[2]: Leaving directory '/usr/src/freeswitch/libs/apr-util/xml/expat' make[2]: Entering directory '/usr/src/freeswitch/libs/apr-util' make[2]: Nothing to be done for 'local-all'. make[2]: Leaving directory '/usr/src/freeswitch/libs/apr-util' make[1]: Leaving directory '/usr/src/freeswitch/libs/apr-util' touch /usr/src/freeswitch/src/include/switch.h make[1]: Entering directory '/usr/src/freeswitch/libs/srtp' make[1]: Leaving directory '/usr/src/freeswitch/libs/srtp' make all-recursive make[1]: Entering directory '/usr/src/freeswitch' CC src/libfreeswitch_la-switch_apr.lo CC src/libfreeswitch_la-switch_buffer.lo CC src/libfreeswitch_la-switch_caller.lo CC src/libfreeswitch_la-switch_channel.lo CC src/libfreeswitch_la-switch_console.lo CC src/libfreeswitch_la-switch_mprintf.lo CC src/libfreeswitch_la-switch_core_media_bug.lo CC src/libfreeswitch_la-switch_core_timer.lo CC src/libfreeswitch_la-switch_core_asr.lo CC src/libfreeswitch_la-switch_core_event_hook.lo CC src/libfreeswitch_la-switch_core_speech.lo CC src/libfreeswitch_la-switch_core_memory.lo CC src/libfreeswitch_la-switch_core_codec.lo CC src/libfreeswitch_la-switch_core_file.lo CC src/libfreeswitch_la-switch_core_cert.lo CC src/libfreeswitch_la-switch_core_hash.lo CC src/libfreeswitch_la-switch_core_sqldb.lo src/switch_core_sqldb.c: In function ‘switch_cache_db_status’: src/switch_core_sqldb.c:3773:4: error: ‘strncpy’ output truncated before terminating nul copying as many bytes from a string as its length [-Werror=stringop-truncation] strncpy(cleankey_str, dbh->name, strlen(dbh->name)); ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ cc1: all warnings being treated as errors make[1]: *** [Makefile:2398: src/libfreeswitch_la-switch_core_sqldb.lo] Error 1 make[1]: Leaving directory '/usr/src/freeswitch' make: *** [Makefile:1413: all] Error 2 <<< log Ends GCC: $ gcc -v Using built-in specs. COLLECT_GCC=gcc COLLECT_LTO_WRAPPER=/usr/lib/gcc/x86_64-linux-gnu/8/lto-wrapper OFFLOAD_TARGET_NAMES=nvptx-none OFFLOAD_TARGET_DEFAULT=1 Target: x86_64-linux-gnu Configured with: ../src/configure -v --with-pkgversion='Debian 8.3.0-6' --with-bugurl=file:///usr/share/doc/gcc-8/README.Bugs --enable-languages=c,ada,c++,go,brig,d,fortran,objc,obj-c++ --prefix=/usr --with-gcc-major-version-only --program-suffix=-8 --program-prefix=x86_64-linux-gnu- --enable-shared --enable-linker-build-id --libexecdir=/usr/lib --without-included-gettext --enable-threads=posix --libdir=/usr/lib --enable-nls --enable-bootstrap --enable-clocale=gnu --enable-libstdcxx-debug --enable-libstdcxx-time=yes --with-default-libstdcxx-abi=new --enable-gnu-unique-object --disable-vtable-verify --enable-libmpx --enable-plugin --enable-default-pie --with-system-zlib --with-target-system-zlib --enable-objc-gc=auto --enable-multiarch --disable-werror --with-arch-32=i686 --with-abi=m64 --with-multilib-list=m32,m64,mx32 --enable-multilib --with-tune=generic --enable-offload-targets=nvptx-none --without-cuda-driver --enable-checking=release --build=x86_64-linux-gnu --host=x86_64-linux-gnu --target=x86_64-linux-gnu Thread model: posix gcc version 8.3.0 (Debian 8.3.0-6) CC: $ cc -v Using built-in specs. COLLECT_GCC=cc COLLECT_LTO_WRAPPER=/usr/lib/gcc/x86_64-linux-gnu/8/lto-wrapper OFFLOAD_TARGET_NAMES=nvptx-none OFFLOAD_TARGET_DEFAULT=1 Target: x86_64-linux-gnu Configured with: ../src/configure -v --with-pkgversion='Debian 8.3.0-6' --with-bugurl=file:///usr/share/doc/gcc-8/README.Bugs --enable-languages=c,ada,c++,go,brig,d,fortran,objc,obj-c++ --prefix=/usr --with-gcc-major-version-only --program-suffix=-8 --program-prefix=x86_64-linux-gnu- --enable-shared --enable-linker-build-id --libexecdir=/usr/lib --without-included-gettext --enable-threads=posix --libdir=/usr/lib --enable-nls --enable-bootstrap --enable-clocale=gnu --enable-libstdcxx-debug --enable-libstdcxx-time=yes --with-default-libstdcxx-abi=new --enable-gnu-unique-object --disable-vtable-verify --enable-libmpx --enable-plugin --enable-default-pie --with-system-zlib --with-target-system-zlib --enable-objc-gc=auto --enable-multiarch --disable-werror --with-arch-32=i686 --with-abi=m64 --with-multilib-list=m32,m64,mx32 --enable-multilib --with-tune=generic --enable-offload-targets=nvptx-none --without-cuda-driver --enable-checking=release --build=x86_64-linux-gnu --host=x86_64-linux-gnu --target=x86_64-linux-gnu Thread model: posix gcc version 8.3.0 (Debian 8.3.0-6) Any help, will be appreciated! Thanks Jose Figueroa

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About mod_soundtouch
by 王聡 11 Jul '19

11 Jul '19

Hi all, I’m trying to add some sound effect in my freeswitch server and testing mod_soundtouch. The test based on: Debian 8.11 Jessie FreeSWITCH Version 1.8.5+git~20190124T225359Z~31281a0bf1~64bit (git 31281a0 2019-01-24 22:53:59Z 64bit) libsoundtouch1/oldstable,now 1.7.1-13~1-jessie+1 amd64 I tried to load soundtouch in fs_cli and XML dialplan, but both of them turned human voice into noise and completely cannot hear any human voice. In fs_cli: soundtouch fa71575c-a392-11e9-becb-7701318508b0 start 0.7o In XML Dialplan: <action application=“soundtouch” data=“0.7o”/> The PCMU and SPEEX codec had been tested and got the same result. Any suggestions on this issue? Regards.

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endconf for the first member of the conference
by Ramesh Kandasamy 10 Jul '19

10 Jul '19

Hi all, Is there a way I can set the endconf flag for only the first participant of a conference room through dialplan? I know there is a way to do this via ESL. But I want this to be from the dialplan (may be through condition/action tags?). Thanks Ramesh

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