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wait_for_silence not working through external gateway
by vidor 23 Aug '18

23 Aug '18

I'm attempting to detect when there is silence on the call in order to play audio file. If I dial to an internal extension the wait_for_silence and detect_silence commands are working fine. However if dial out through an external gateway to PSTN I don't get expected results. Silence is always detected, even if there is voice speaking. I can detect DTMF commands fine... Is there any known issue or parameter required on the gateway in order to detect the audio / silence. Thanks in advance! -- Sent from: http://freeswitch-users.2379917.n2.nabble.com/

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Execute HTTP request from LUA script not work
by Jose David Jurado Alonso 23 Aug '18

23 Aug '18

Hi, I have tried to make a HTTP POST request from LUA script but it does not work correctly in any way. This is an example of the errors that I have obtained: 09802b39-524f-4011-ada7-418e6a870248 2018-08-03 12:23:07.981581 [ERR] mod_lua.cpp:203 error loading module socket.core from file /usr/local/lib/lua/5.1/socket/core.so: 09802b39-524f-4011-ada7-418e6a870248 /usr/local/lib/lua/5.1/socket/core.so: undefined symbol: lua_pcall 09802b39-524f-4011-ada7-418e6a870248 stack traceback: 09802b39-524f-4011-ada7-418e6a870248 [C]: in ? 09802b39-524f-4011-ada7-418e6a870248 [C]: in function require 09802b39-524f-4011-ada7-418e6a870248 /usr/local/share/lua/5.1/socket.lua:12: in main chunk 09802b39-524f-4011-ada7-418e6a870248 [C]: in function require 09802b39-524f-4011-ada7-418e6a870248 /usr/local/share/lua/5.1/socket/http.lua:10: in main chunk 09802b39-524f-4011-ada7-418e6a870248 [C]: in function require 09802b39-524f-4011-ada7-418e6a870248 /usr/share/freeswitch/scripts/test.lua:3: in main chunk The lua script is: local http = require "socket.http"; local req_url = "http://localhost/" http.request{ url = req_url, method = "POST" } I tried several tutorials and several configurations: - https://freeswitch.org/confluence/display/FREESWITCH/Installing+LuaSocket - https://freeswitch.org/confluence/display/FREESWITCH/Third+Party+Libraries with https://luarocks.org/modules/jakeg/lua-requests <https://luarocks.org/modules/jakeg/lua-requests> And some other things that I do not remember but always fail for some "undefined symbol". Any ideas? Thanks! José D.

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variable to know who initiated call transfer
by Bipin Patel 23 Aug '18

23 Aug '18

hi, im stuck in one scenarion where A calls B and either of them can initiate a transfer using *3 and bind_meta_app in features.xml but i need a variable which can tell me which leg initiated the transfer, like was it A or B coz no matter who did i always see call going to features from A as A dialed B. the reason i need this is so i can add a hash value for the ext that dialed *3 i have been searching for such a variable like crazy but not able to find, i just want to know the ext number of the person initiating call transfer -- Regards, Bipin

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Open and closing channels continously
by Jose David Jurado Alonso 23 Aug '18

23 Aug '18

Hi, I'm just observing in the log that user 1001 (and other random users like '55340115419480247') is continuously creating and closing channels. That users doesn't exist (I delete the folder "directory" since I use database with different user names, no numbers). I've done a search some reference to user 1001 in all the configuration and haven't found anything (using "grep -r '1001' /etc/freeswitch/*") Any idea why this happens? A bot or spider maybe? Log trace example: 2018-08-21 09:34:50.448541 [WARNING] switch_core_state_machine.c:687 1f89de01-b82f-44fa-9b91-9d111c44e220 sofia/internal/1001(a)192.168.1.120 Abandoned 2018-08-21 09:34:50.448541 [NOTICE] switch_core_state_machine.c:690 Hangup sofia/internal/1001(a)192.168.1.120 [CS_NEW] [WRONG_CALL_STATE] 2018-08-21 09:34:50.448541 [NOTICE] switch_core_session.c:1731 Session 291 (sofia/internal/1001(a)192.168.1.120) Ended 2018-08-21 09:34:50.448541 [NOTICE] switch_core_session.c:1735 Close Channel sofia/internal/1001(a)192.168.1.120 [CS_DESTROY] 2018-08-21 09:34:51.288542 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/1001(a)192.168.1.120 [f8be20e3-1244-48bb-a89b-7b49382726af] 2018-08-21 09:34:52.568543 [WARNING] switch_core_state_machine.c:687 bae7044c-4915-4a9a-9265-6e48e969ffec sofia/internal/ 55340115419480247(a)192.168.1.120 Abandoned 2018-08-21 09:34:52.568543 [NOTICE] switch_core_state_machine.c:690 Hangup sofia/internal/55340115419480247(a)192.168.1.120 [CS_NEW] [WRONG_CALL_STATE] 2018-08-21 09:34:52.568543 [NOTICE] switch_core_session.c:1731 Session 292 (sofia/internal/55340115419480247(a)192.168.1.120) Ended 2018-08-21 09:34:52.568543 [NOTICE] switch_core_session.c:1735 Close Channel sofia/internal/55340115419480247(a)192.168.1.120 [CS_DESTROY] 2018-08-21 09:34:52.848563 [WARNING] switch_core_state_machine.c:687 a0a39ab8-f109-468b-8136-3ebf044c2ffd sofia/internal/1001(a)192.168.1.120 Abandoned 2018-08-21 09:34:52.848563 [NOTICE] switch_core_state_machine.c:690 Hangup sofia/internal/1001(a)192.168.1.120 [CS_NEW] [WRONG_CALL_STATE] 2018-08-21 09:34:52.848563 [NOTICE] switch_core_session.c:1731 Session 293 (sofia/internal/1001(a)192.168.1.120) Ended 2018-08-21 09:34:52.848563 [NOTICE] switch_core_session.c:1735 Close Channel sofia/internal/1001(a)192.168.1.120 [CS_DESTROY] 2018-08-21 09:34:54.248541 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/1001(a)192.168.1.120 [ef2d84f4-89e5-453d-89df-8280b129b42b] 2018-08-21 09:34:55.148543 [WARNING] switch_core_state_machine.c:687 9f1804f2-80d0-4fad-8e43-40f0cf2777e9 sofia/internal/1001(a)192.168.1.120 Abandoned José D. Jurado Alonso

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Re: [Freeswitch-users] SRTP crypto selection
by Tom Parrott 22 Aug '18

22 Aug '18

Hi FWIW worth I worked with the Zoiper support team to figure out a workaround in Freeswitch 1.6 for Zoiper 5. I wrote about it here: https://www.tomp.uk/blog/view/39 I haven't had chance to test whether it works out of the box on 1.8 yet though. Thanks Tom On 2018-08-21 21:29, freeswitch-users-request(a)lists.freeswitch.org wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > Today's Topics: > > 1. Re: SRTP crypto selection (Michael Jerris) > _______________________________________________ > 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 > http://www.freeswitch.org

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Zoiper5 broken with FS, outbound calls get dropped
by Rajil Saraswat 22 Aug '18

22 Aug '18

Hello, All my outbound calls made by Zoiper5 get dropped. I contacted Zoiper support and they mentioned that it is a FreeSwitch problem. How can this be fixed? Here is what Zoiper support said, There was a known issue with FreeSwitch related to the naming of the security suites sent from them, they were not completely matched to RFC 6188. On the left, the names are the standard ones - used in the RFC6188, and the right ones are the names used by FreeSwitch: AES_256_CM_HMAC_SHA1_80 == AES_CM_256_HMAC_SHA1_80 AES_256_CM_HMAC_SHA1_32 == AES_CM_256_HMAC_SHA1_32 AES_192_CM_HMAC_SHA1_80 == AES_CM_192_HMAC_SHA1_80 AES_192_CM_HMAC_SHA1_32 == AES_CM_192_HMAC_SHA1_32 In order to be robust we added the so called FS names but they are not different suits just one suite has two names (as the negotiation process requires a complete match of the names). In the previous Zoiper5 versions (e.g. 5.2.14) the additional FS names were not present. The offered by Zoiper suites were only the described in the RFC 6188 and 4568 in that priority order: AES_256_CM_HMAC_SHA1_80 AES_256_CM_HMAC_SHA1_32 AES_192_CM_HMAC_SHA1_80 AES_192_CM_HMAC_SHA1_32 AES_CM_128_HMAC_SHA1_80 AES_CM_128_HMAC_SHA1_32 After the addition of FS names (5.2.16 and newer) we now offer everything in the following order: AES_256_CM_HMAC_SHA1_80 // RFC 6188 AES_256_CM_HMAC_SHA1_32 // RFC 6188 AES_CM_256_HMAC_SHA1_80 // FS interpretation AES_CM_256_HMAC_SHA1_32 // FS interpretation AES_192_CM_HMAC_SHA1_80 // RFC 6188 AES_192_CM_HMAC_SHA1_32 // RFC 6188 AES_CM_192_HMAC_SHA1_80 // FS interpretation AES_CM_192_HMAC_SHA1_32 // FS interpretation AES_CM_128_HMAC_SHA1_80 // RFC 4568 AES_CM_128_HMAC_SHA1_32 // RFC 4568 It seems that FS also sends bad checksums with the AES_256_CM_HMAC_SHA1_80 and AES_CM_192_HMAC_SHA1_80, the only working suite is AES_CM_128_HMAC_SHA1. There is a workaround you can use until and if this issue is fixed by FS. You may try to rearrange the priority on your FS (put 128 on top instead of 256, or completely disable the non-working suites). Or you can simply keep using older version of Zoiper with 128 bit. Thanks

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Multiple Diversion headers
by Sjoerd Boomstra 22 Aug '18

22 Aug '18

Hi All, I'm struggling with multiple Diversion headers. In our case we have an incoming call with multiple diversion headers, we do some processing and want to add a diversion header ourselves to the outgoing call. What I've found so far is that for inbound calls one needs to set the profile param parse-all-invite-headers=true, this results in a sip_i_Diversion variable that uses the ARRAY:: syntax and has all the values I want. So far so good. Example: ARRAY::<sip:Carol@uas2.isp.com>;privacy=off;reason=unconditional;counter=2;screen=no|:<sip:Bert@uas2.isp.com>;privacy="full";reason=user-busy;counter=3;screen=no|:<sip:Bob@uas1.isp.com>;reason=do-not-disturb;counter=1 For the outbound call it's more difficult, I cannot get FreeSwitch to produce multiple Diversion headers. The only syntax I've found is using comma's in the value, like this: <action application="export" data="sip_h_Diversion=<sip:Carol@uas2.isp.com>;privacy=off;reason=unconditional;counter=2;screen=no,<sip:Bert@uas2.isp.com>;privacy="full";reason=user-busy;counter=3;screen=no,<sip:Bob@uas1.isp.com>;reason=do-not-disturb;counter=1,<sip:Bert@uas2.isp.com>;reason=user-busy"/> This results in a single diversion header with the contents I put in. From the RFC's I understand that this is technically equivalent to 4 (in my case) Diversion headers, is this true? Is there a way I can convince FS to generate multiple separate diversion headers? Best regards, Sjoerd. BTW I've tested with FS 1.6.20 and 1.4.26

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Two endpoints with same extension but only one is ringing
by Thorsten Göllner 22 Aug '18

22 Aug '18

Hi, I have running Freeswitch (Version 1.6.19~64bit) and 2 endpoints registered with the same extension in the same network First: Grandstream GXV3370 (Version 1.0.1.12) Second: Android GS Wave App (Version 1.0.3.10) Sometimes it works fine. But most time only one device is ringing when calling the extension. The logfile is fine - I can see invites to both devices. Registration-Status is fine. Where do I start to search for the problem? Local WLAN-Router? Best regards, Thorsten

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problem configuring freeswitch 1.8.1
by John Covici 22 Aug '18

22 Aug '18

I am trying to compile from source freeswitch 1.8.1 and I can't find anywhere a reference to v8fs_static. What package is this supposed to be in? I may need the source, as I use gentoo. Thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una covici(a)ccs.covici.com

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Reloading mod_fifo while it's in use
by Michael Avers 22 Aug '18

22 Aug '18

Hello, Trying to reload mod_fifo after a change frequently results in: 2018-08-02 21:51:17.385839 [WARNING] switch_loadable_module.c:2005 Module mod_fifo is in use, cannot unload. 2018-08-02 21:51:17.385839 [WARNING] switch_loadable_module.c:1592 Module mod_fifo Already Loaded! We then have to wait for the calls to end and hope to get a reload mod_fifo command in time before other calls come in. Is there a way to reload mod_fifo properly while a fifo call may be taking place? Thank you, Mike

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