June 2020 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Teams direct routing
by Flavio Goncalves 02 Jul '20

02 Jul '20

Hi, I've managed to connect teams using FS, now I´m trying to implement call hold and call transfer. Teams use an unusual way for hold, it transfers the call. In the refer I can see the refer-to pointing to an url without a request uri user REFER-TO: <sip:sip.pstnhub.microsoft.com:5061 ;transport=tls;x-m=8:orgid:e83c0390-43cb-4c40-8d09-8a972709235b;x-t=9c36cf22-e6e8-4792-b823-873069b82489> How can I bridge a call to an URI without a request uri user? I tried sofia/gateway/ms1/ (without passing a user). I tried <action application="export" data="nolocal:sip_invite_req_uri=$1;$2"/> where $1 is the domain and $2 are the invite parameters but no success. Did anyone have success with TEAMS and call hold, call transfer? Thanks Flavio E Goncalves

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Mod_sms and script to send to Twilio
by drodrig 02 Jul '20

02 Jul '20

Hi all. I wrote a small Python script that is run from a chatplan and takes a chat message from a sip extension and sends the message to the appropriate phone number via Twilio (they provide an Internet-based API to do this). The script works great, but I still see an error in on fs_cli console: I've tried mucking with the "to" and "to_user" headers (and other headers as well). I've also tried adding the following line in the chatplan XML file: Finally, I've tried different return values in the Python script. Nothing I've tried removes the error messages. In truth, the texting works fine -- I just don't like seeing the error. Here's my chatplan XML file: And the python script: I'm new to the world of SIP and Freeswitch, so I am hoping I am missing something simple. Any help is appreciated. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Mod-sms-and-script-to-send-to… Sent from the freeswitch-users mailing list archive at Nabble.com.

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freeswitch.Dbh 'fetch' functions?
by Chad Phillips 02 Jul '20

02 Jul '20

Curious if https://freeswitch.org/confluence/display/FREESWITCH/Lua+with+Database#Luaw… is missing any support functions that freeswitch.Dbh has under the hood? It seems some convenience functions for the following would be quite nice: - Fetch query results into a table of tables - Fetch one result row - Fetch one row I hacked up those in some quick helpers here: https://gist.github.com/thehunmonkgroup/f92ca495c4c366d74fa9269320806cec Just wondering if I'm missing anything.

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missing bgapi?
by David Villasmil 30 Jun '20

30 Jun '20

Hello all, I have mod_commands loaded but I'm getting: Unknown Command: bgapi originate I'm on 1.10 on Buster Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Segmentation Faults on fs_cli when trying to rescan sofia profile
by Keith Laaks 30 Jun '20

30 Jun '20

Hi, I have also been experiencing this same problem. Tested and confirmed the issue also exists in the development branch (FreeSWITCH Version 1.10.4-dev+git~20200624T182031Z~ffe0a1b629~64bit) also. In file “sofia.c”, function “static void parse_gateways(sofia_profile_t *profile, switch_xml_t gateways_tag, const char *gwname)”, line 4008 has: gateway->options_user_agent = options_user_agent; Which should instead be: gateway->options_user_agent = switch_core_strdup(gateway->pool, options_user_agent); —————————————————————————————————————————————————————————————————————————————————————————————————— diff --git a/src/mod/endpoints/mod_sofia/sofia.c b/src/mod/endpoints/mod_sofia/sofia.c index 32f27462e5..da57072cca 100644 --- a/src/mod/endpoints/mod_sofia/sofia.c +++ b/src/mod/endpoints/mod_sofia/sofia.c @@ -4005,7 +4005,7 @@ static void parse_gateways(sofia_profile_t *profile, switch_xml_t gateways_tag, gateway->auth_username = switch_core_strdup(gateway->pool, auth_username); gateway->register_password = switch_core_strdup(gateway->pool, password); gateway->distinct_to = distinct_to; - gateway->options_user_agent = options_user_agent; + gateway->options_user_agent = switch_core_strdup(gateway->pool, options_user_agent); if (switch_true(caller_id_in_from)) { sofia_set_flag(gateway, REG_FLAG_CALLERID); —————————————————————————————————————————————————————————————————————————————————————————————————— Not sure of the process to get this patch approved and applied to the git repo….. Can someone in the know take care of this? Best Regards Keith -------------------------------------------------------------------------------------------------------------------- On Thu, 11 Jun 2020 at 00:58, Mahmood Alkhalil <mahmood.alkhalil at outlook.com <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> wrote: > > Hi Everyone, > > I'm always getting segmentation fault on fs_cli whenever I'm trying to rescan sofia profile and fs_cli exists, below is an example: > > freeswitch at freeswitch <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> sofia profile AtcomSipProfile rescan > Reload XML [Success] > +OK scan complete > > 2020-06-10 11:30:57.688648 [DEBUG] sofia.c:4628 debug [7] > 2020-06-10 11:30:57.688648 [DEBUG] sofia.c:4628 sip-trace [true] > 2020-06-10 11:30:57.688648 [DEBUG] sofia.c:4628 sip-capture [true] > 2020-06-10 11:30:57.688648 [INFO] mod_enum.c:884 ENUM Reloaded > nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params > 2020-06-10 11:30:57.688648 [DEBUG] sofia.c:4628 watchdog-enabled [no] > 2020-06-10 11:30:57.688648 [DEBUG] sofia.c:4628 watchdog-step-timeout [30000] > 2020-06-10 11:30:57.688648 [DEBUG] sofia.c:4628 watchdog-event-timeout [30000] > 2020-06-10 11:30:57.688648 [DEBUG] sofia.c:4628 log-auth-failures [false] > 2020-06-10 11:30:57.688648 [DEBUG] sofia.c:4628 forward-unsolicited-mwi-notify [false] > 2020-06-10 11:30:57.688648 [DEBUG] sofia.c:4628 context [atcomDialplan] > 2020-06-10 11:30:57.688648 [DEBUG] sofia.c:4628 rfc2833-pt [101] > 2020-06-10 11:30:57.688648 [DEBUG] sofia.c:4628 sip-port [5080] > 2020-06-10 11:30:57.688648 [DEBUG] sofia.c:4628 dialplan [XML] > 2020-06-10 11:30:57.688648 [DEBUG] sofia.c:4628 dtmf-duration [2000] > 2020-06-10 11:30:57.688648 [DEBUG] sofia.c:4628 inbound-codec-prefs [OPUS,G722,PCMU,PCMA,H264,VP8] > freeswitch at freeswitch <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> Segmentation fault > > Please let me know if this is expected or if i can share more details to further investigate this. > > Mahmood Alkhalil.

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mod_mariadb install problem
by Mehdi Shirazi 30 Jun '20

30 Jun '20

Hi I tried to install mod_mariadb in debian 9 Freeswitch 1.10.3 and mariadb 10.4 via package. When I want to load start module I see this error: 2020-06-20 12:02:15.274711 [CRIT] switch_loadable_module.c:1785 Error Loading module /usr/lib/freeswitch/mod/mod_mariadb.so **/usr/lib/freeswitch/mod/mod_mariadb.so: symbol mysql_get_timeout_value, version libmysqlclient_18 not defined in file libmariadb.so.3 with link time reference** please help me solve the problem. Regards M.Shirazi

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Incoming call audio issue
by Roberto Scarpone 30 Jun '20

30 Jun '20

Hi, I have a problem when phones receive a call, for a time varying between 1-3 seconds, they hear nothing (at the beginning of the call). The problem does not always occur, but with a frequency of 2/3 time every 10 calls. After a few seconds the conversation presents no further problems. Listening to the captured audio the call doesn't seems to have any problem, the audio is complete, but when they answer the call the problem is there and they can't hear the first 1-3 seconds of the conversation. I have a FreeSWITCH 1.6.20 on Debian 8, installed from repo. Has anyone already encountered this problem or would you know how to help me solve it? Best regards, Roberto

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UniMRCP Setting variables
by David Villasmil 29 Jun '20

29 Jun '20

Hello, I'm testing unimrcp server with freeswitch using mod_unimrcp. Is there any way to set channel variables on the unimrcp channel? i.e.: I need to set some variables on the recognition channel, like all of the incoming channel variables. Thanks! David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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PBX Call-Transfer Issue - timing of RTP?
by Andy Newlands 29 Jun '20

29 Jun '20

We have Freeswitch acting as a SIP platform, with internects with various UK carriers (including BT) on one side, with customer PBX connected from the other side. We have an intermittent problem where a call comes into FS from the PSTN and is routed to the customer PBX (and Avaya), with the call then transferred back, via Freeswitch, then the PSTN to another phone number. Sometimes the original caller continues to hear ringtone after the transfer-destination has answered (and no conversation is possible). I have attached Wireshark SIP flows for the scenario where this works and the one where transfer does not complete properly (labelled in red, above each ladder diagram, "BROKEN", on the left and "WORKING" on the right). In the working example, we see RTP BEFORE the REFER. In the non-working example, RTP is sent AFTER the BYE. 1. We don't understand why the Avaya sends messages in a different order, on different occasions, for the exact same operation? 2. Should Freeswitch be able to handle both the scenarios shown in the attached image? 3. Are there any config or dialplan settings which can be used to help fix this? NOTE: We also have other issues with call transfers in general, whereby the transfer succeeds, but there is no audio heard by the transferee. Thanks

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Crypto negotiated, but FS thinks otherwise and drops outbound calls.
by Keith Laaks 29 Jun '20

29 Jun '20

> Hi All, > > I am running: FreeSWITCH Version 1.10.3-release-15-129de34d84~64bit (-release-15-129de34d84 64bit) on Debian GNU/Linux 10 (buster) > > I am bridging an incoming call to a destination over SIP TLS, using SRTP and ICE-lite. rtcp_mux is disabled. > > I have to use ‘mandatory’ in the export below (before the bridge), as this specific destination fails the call if the SDP includes the unencrypted optioN (i.e. m=audio 2354 RTP/AVP 18 8 98) > <action application="export" data=“rtp_secure_media=mandatory:AES_CM_128_HMAC_SHA1_32"/> > > I am having a problem in that FS drops the call as soon as the destination answers. > > In the debug traces I can see that, although Secure RTP is indeed being negotiated (Activating audio Secure RTP RECV) FS somehow thinks otherwise (switch_core_media.c:5665 Crypto not negotiated but required.) > > 2020-06-26 18:20:40.208635 [INFO] switch_core_media.c:8843 Activating Audio ICE > 2020-06-26 18:20:40.208635 [NOTICE] switch_rtp.c:4915 Activating RTP audio ICE: ZyVg:nzPxIzWUEUYj6GjG 11.22.33.44:50004 > 2020-06-26 18:20:40.208635 [DEBUG] switch_core_media.c:8975 sofia/teams/27101234567 Set 2833 dtmf send payload to 101 > 2020-06-26 18:20:40.208635 [DEBUG] switch_core_media.c:8982 sofia/teams/27101234567 Set 2833 dtmf receive payload to 101 > 2020-06-26 18:20:40.208635 [DEBUG] switch_core_media.c:9005 sofia/teams/27101234567 Set rtp dtmf delay to 40 > 2020-06-26 18:20:40.208635 [DEBUG] switch_core_media.c:1501 LIFETIME found in |2^31, base 2 exp 31 > 2020-06-26 18:20:40.208635 [NOTICE] switch_core_media.c:1524 Skipping MKI due to empty index > 2020-06-26 18:20:40.208635 [INFO] switch_rtp.c:4217 Activating audio Secure RTP SEND > 2020-06-26 18:20:40.208635 [INFO] switch_rtp.c:4195 Activating audio Secure RTP RECV > 2020-06-26 18:20:40.208635 [DEBUG] switch_core_sqldb.c:2836 Secure Type: srtp:sdes:AES_CM_128_HMAC_SHA1_32 > 2020-06-26 18:20:40.208635 [DEBUG] switch_core_sqldb.c:2836 Secure Type: srtp:sdes:AES_CM_128_HMAC_SHA1_32 > 2020-06-26 18:20:40.208635 [NOTICE] sofia.c:8503 Channel [sofia/teams/27101234567] has been answered > 2020-06-26 18:20:40.208635 [DEBUG] switch_channel.c:3865 (sofia/teams/27101234567) Callstate Change RINGING -> ACTIVE > 2020-06-26 18:20:40.228625 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [G729:18:8000:20:8000:1]/[opus:116:48000:20:0:1] > 2020-06-26 18:20:40.228625 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [G729:18:8000:20:8000:1]/[PCMA:8:8000:20:64000:1] > 2020-06-26 18:20:40.228625 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [G729:18:8000:20:8000:1]/[PCMU:0:8000:20:64000:1] > 2020-06-26 18:20:40.228625 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[opus:116:48000:20:0:1] > 2020-06-26 18:20:40.228625 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] > 2020-06-26 18:20:40.228625 [DEBUG] switch_core_media.c:5647 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match > 2020-06-26 18:20:40.228625 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] > 2020-06-26 18:20:40.228625 [DEBUG] switch_core_media.c:5508 Set telephone-event payload to 98@8000 > 2020-06-26 18:20:40.228625 [WARNING] switch_core_media.c:5665 Crypto not negotiated but required. > 2020-06-26 18:20:40.228625 [DEBUG] switch_core_media.c:5909 sofia/vox/0831233210(a)10.10.1.1 <mailto:sofia/vox/0831233210@10.10.1.1> Set 2833 dtmf send payload to 98 recv payload to 98 > 2020-06-26 18:20:40.228625 [NOTICE] switch_channel.c:3908 Hangup sofia/vox/0831233210(a)10.10.1.1 <mailto:sofia/vox/0831233210@10.10.1.1> [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] > > Am I missing something here? > > Thanks > > Keith >

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