August 2019 - FreeSWITCH-users - FreeSWITCH Mailing Lists

[unable to compile freeswitch on debian 10]
by Bilal Abbasi 21 Aug '19

21 Aug '19

Hi Users, I am unable to build freeswich on debain 10(buster) freeswitch version 1.10, here are the error i get. making all mod_http_cache make[4]: Entering directory '/usr/src/freeswitch/src/mod/applications/mod_http_cache' Making all in . make[5]: Entering directory '/usr/src/freeswitch/src/mod/applications/mod_http_cache' CC mod_http_cache_la-mod_http_cache.lo mod_http_cache.c: In function ‘cached_url_filename_create.isra.4’: mod_http_cache.c:988:2: error: ‘strncpy’ output may be truncated copying 2 bytes from a string of length 256 [-Werror=stringop-truncation] strncpy(uuid_dir, uuid_str, 2); ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ mod_http_cache.c:988:2: error: ‘strncpy’ output may be truncated copying 2 bytes from a string of length 256 [-Werror=stringop-truncation] In function ‘parse_domain.constprop’, inlined from ‘url_cache_http_profile_find_by_fqdn.part.2.constprop’ at mod_http_cache.c:929:3: mod_http_cache.c:276:2: error: ‘strncpy’ specified bound 255 equals destination size [-Werror=stringop-truncation] strncpy(domain_buf, start, domain_buf_len); ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ cc1: all warnings being treated as errors make[5]: *** [Makefile:761: mod_http_cache_la-mod_http_cache.lo] Error 1 make[5]: Leaving directory '/usr/src/freeswitch/src/mod/applications/mod_http_cache' make[4]: *** [Makefile:801: all-recursive] Error 1 make[4]: Leaving directory '/usr/src/freeswitch/src/mod/applications/mod_http_cache' make[3]: *** [Makefile:704: mod_http_cache-all] Error 1 make[3]: Leaving directory '/usr/src/freeswitch/src/mod' make[2]: *** [Makefile:607: all-recursive] Error 1 make[2]: Leaving directory '/usr/src/freeswitch/src' make[1]: *** [Makefile:3736: all-recursive] Error 1 make[1]: Leaving directory '/usr/src/freeswitch' make: *** [Makefile:1445: all] Error 2 Regards Abbasi

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Freeswitch Slack channel
by Petarr Jones 21 Aug '19

21 Aug '19

Hi Is there an active FS Slack channel? If so, can someone please send me the invite link. Thanks in advance Petarr

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FS-9102 - core modules not loaded
by Bipin Patel 20 Aug '19

20 Aug '19

hi, what was the fix to FS-9102 because im seeing same behavior again on 1.10.1-dev running on a few raspberry pis, its happening to some of them only. core_pcm_module doesnt show in show codecs or modules but it still works, rebooting the machine loads all of the modules but restarting the service always causes them to not show loaded Regards, Bipin

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Silence detection in voicemail greeting file
by shivali chalke 20 Aug '19

20 Aug '19

Hi Guys, I am trying to record greeting for voicemai. After done recording, Listening to the playback there is 2sec of silence then actual recording start playing which also not complete. I have found there to be a longer time in the beginning before the record function actually starts. Can anyone help me on this? Thanks in advance.

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FYI: fs 1.10 incompatible config change PGSQL
by Giovanni Maruzzelli 20 Aug '19

20 Aug '19

Ye People: because now Fs1.10 packages are installed also if you still use the 1.8 repo, you better aware of an incompatible change: pgsql is no more in core, but in aptly named mod_pgsql. You MUST NOT load mod_pgsql in modules.conf.xml (do not work), but in a special additional xml file, in same "autoload_configs" directory. File name: pre_load_modules.conf.xml File content: <configuration name="pre_load_modules.conf" description="Modules"> <modules>   <load module="mod_pgsql"/> </modules> </configuration> Hope this help you keep your hair :) -giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

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Params in verto from fs_cli and backwards
by Jose Figueroa 17 Aug '19

17 Aug '19

Hello, I need to pass some variables from originate command on fs_cli or lua script to verto client I have already and backwards from verto to the fs_cli. I have this from fs_cli -> originate {*foo=val1,bar=val2*,ignore_early_media=true,origination_caller_id_number=1003}user/1002 &echo() What I want is to get the *foo* and *bar* variables on verto client. In addition, I see the documentation https://evoluxbr.github.io/verto-docs/tut/making-a-call.html that if you want to make a call and pass a userVariables they will be visible to dialplan as verto_dvar_{variable} which is correct I can catch them. But not from the fs_cli to the verto, or even if I can do it from lua script to verto might work for my need. Any help would be appreciated. Thanks! Jose Figueroa

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hangup cause / sip cause ?
by morbidus＠rx900.org 17 Aug '19

17 Aug '19

Hi i've wrote some dialplans for billing with this great engine which is freeswitch and i'm also using limits and failover ('manually', without the lcr module). Everything works ok and i'm impressed by its stability and performance ! But i'm unable to get a 'final' termination cause for the calls as i'm used to get with asterisk: i'd like to have something like 'answered', 'cancel', 'busy', 'congestion', 'chanunavail', and so on so i'm writing last_bridge_hangup_cause, bridge_hangup_cause and hangup_cause in my cdr and in particular when the call goes thru the second or third trunk (when using failover) i get 'NORMAL_TEMPORARY_FAILURE', 'NORMAL_TEMPORARY_FAILURE', 'NORMAL_CLEARING' and i don't know which one of the three is the actual termination cause :) what are you guys using to get 'consistent' tc ?

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Freeswitch security and BLF
by Petarr Jones 17 Aug '19

17 Aug '19

Hi The following question is based on the fact that I am trying configure Freeswitch so that users can simply add an extension number in the BLF field of the phone but "user id" is alphanumeric. I am making the assumption that having a "user id" of freeswitchuser1001(a)example.com is better than 1001 since the fraudsters would automatically try extensions but are less likely to know a complicated username? Would that logic only hold true if I am using TLS? If this is correct, and I use the "number-alias" to link the "user id" and user's extension, what would I have to use in the BLF field of the phone. Also, in the documentation there is a warning about using "number-alias" with mod_xml_curl. When will this be an issue and should I consider a different approach? (no sure if I can ask this last question here - should I be asking the Kamailio crowd) I have considered using Kamailio to do the authentication and then re-write the SIP request to only include extension when it gets passed to Freeswitch but don't understand the BLF mechanism and how that would then work. Would it be better for Kamailio to manage the BLF functionality? All help greatly appreciated. Petarr

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adding modules
by Joli Martinez 16 Aug '19

16 Aug '19

I know that I can add modules on compiled FreeSWITCH and there are instructions for that. What I don't find is how to add a module when installed from binary. I am running Debian 9 and installed FreeSWITCH with apt. how do I install a module? I am looking primarily for sms_flowroute module to enable sms messaging. thanks,

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FS gateway over TLS - force ephemeral port in contact header
by Victor Chukalovskiy 15 Aug '19

15 Aug '19

Hi Folks, long time away from this list but I do run into another FS questions I can't answer myself. General scenario is latest FS V1.8 registering to Oracle SBC via gateway with register-transport TLS, and then trying to send / receive some calls. Issue I observe is as following: Registration goes well, as well as outbound call is established (from FS to Oracle). So far FS is always an SSL "Client" and SBC is always an SSL "Server". So TCP connection is between an ephemeral port e.g 57647 on FS side and 5061 on SBC side. However, when SBC sends a BYE to FS, it does so in a brand new TCP socket going from a high range port on SBC side e.g. 27696 and 5061 on FS side. In this new connection SBC tries to be an SSL client and expect FS to act as SSL server. FS ignores it completely (nothing in debug log or in sofia sip trace). My presumption is that it's incorrect, and that everything should go back to the port that REGISTER / INVITE came from. I thinks SBC does so because of FS contact header not matching the actual TCP connection. I'm trying to figure out FS guilt in this, as I suspect it happens because of the way FS sets Contact header port on the outbound REGISTER and INVITE: Contact: <sip:gw+test@<freeswitch_ip>:5061;tport=tcp;transport=tls;gw=test> I suspect that 5061 is not the right port given FS is actually using ephemeral port for this registration. Is this a bug I should report, or a configuration I can change, or normal? How can I force FS to put it's actual ephemeral port? So that it looks something like this: Contact: <sip:gw+test@<freeswitch_ip>:57647;tport=tcp;transport=tls;gw=test> Below is just an illustration of two TCP connections that end-up being created on a single outbound call. FreeSWITCH - initiated REGISTER, INVITE, 100 Trying, 200 OK etc: 11:05:53.303006 IP <freeswitch_ip>.57647 > <sbc_ip>.5061 11:05:53.408646 IP <sbc_ip>.5061 > <freeswitch_ip>.57647 11:05:53.408745 IP <freeswitch_ip>.57647 > <sbc_ip>.5061 11:05:54.242105 IP <freeswitch_ip>.57647 > <sbc_ip>.5061 11:05:54.242307 IP <freeswitch_ip>.57647 > <sbc_ip>.5061 11:05:54.309673 IP <sbc_ip>.5061 > <freeswitch_ip>.57647 11:05:54.309767 IP <freeswitch_ip>.57647 > <sbc_ip>.5061 11:05:54.393547 IP <sbc_ip>.5061 > <freeswitch_ip>.57647 11:05:54.393584 IP <freeswitch_ip>.57647 > <sbc_ip>.5061 11:05:54.394273 IP <freeswitch_ip>.57647 > <sbc_ip>.5061 11:05:54.395283 IP <freeswitch_ip>.57647 > <sbc_ip>.5061 11:05:54.395430 IP <freeswitch_ip>.57647 > <sbc_ip>.5061 11:05:54.395500 IP <freeswitch_ip>.57647 > <sbc_ip>.5061 11:05:54.463197 IP <sbc_ip>.5061 > <freeswitch_ip>.57647 11:05:54.511325 IP <freeswitch_ip>.57647 > <sbc_ip>.5061 11:05:57.283340 IP <sbc_ip>.5061 > <freeswitch_ip>.57647 11:05:57.283498 IP <freeswitch_ip>.57647 > <sbc_ip>.5061 11:06:02.904345 IP <sbc_ip>.5061 > <freeswitch_ip>.57647 11:06:02.904391 IP <freeswitch_ip>.57647 > <sbc_ip>.5061 11:06:02.918036 IP <freeswitch_ip>.57647 > <sbc_ip>.5061 11:06:03.085751 IP <sbc_ip>.5061 > <freeswitch_ip>.57647 Far-end (SBC) initiated "BYE". Note that ports have changed to a new set, and ssl server / client roles reversed: 11:06:11.285869 IP <sbc_ip>.27696 > <freeswitch_ip>.5061 11:06:11.285960 IP <freeswitch_ip>.5061 > <sbc_ip>.27696 11:06:11.354400 IP <sbc_ip>.27696 > <freeswitch_ip>.5061 11:06:11.357416 IP <sbc_ip>.27696 > <freeswitch_ip>.5061 11:06:11.357464 IP <freeswitch_ip>.5061 > <sbc_ip>.27696 11:06:11.358337 IP <freeswitch_ip>.5061 > <sbc_ip>.27696 11:06:11.358342 IP <freeswitch_ip>.5061 > <sbc_ip>.27696 11:06:11.426766 IP <sbc_ip>.27696 > <freeswitch_ip>.5061 11:06:11.427018 IP <freeswitch_ip>.5061 > <sbc_ip>.27696 11:06:11.492787 IP <sbc_ip>.27696 > <freeswitch_ip>.5061 Thanks a lot, -Vic

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