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Starting dtmf_generation with mod_lcr
by Mimiko 04 Mar '21

04 Mar '21

Hello. I use lcr for outbound connection to providers based on phone call price. On one provider I need to use start_dtmf_generate to force recoding sip info from local extensions to inband dtmf. Setting variable from lcr execute_on_answer=start_dtmf_generate does not work. Also gateway profile have <variables> <variable direction="outbound" name="dtmf_type" value="none"/> </variables> <param name="dtmf-type" value="none"/> this also does not help. How can I accomplish per provider dtmf_generate for outbound calls?

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Don't deprecate this option - useful for buggy user agents
by Martin Paterson 04 Mar '21

04 Mar '21

I had a problem which I solved by turning on a FreeSWITCH option that logs a warning: "Enabling reg-deny-binding-fetch-and-no-lookup - this functionality is deprecated and will be removed - let FS devs know if you think it should stay". So here is me letting the FS devs know. The problem is to do with buggy user agents and their handling of multiple Contact headers in the 200 OK response to their REGISTER. My FreeSWITCH system uses mainly Snom phones and Zoiper soft phones. I allows multiple registrations and I have two sip profiles that the user agents connect to, one for Snoms and one for Zoiper. They share a registration database with the dbname option. When multiple user agents register with the same credentials, the correct behaviour in the 200 OK to a REGISTER is to list all the user agents in multiple Contact headers with their individual registration expiry times. This is exactly what FreeSWITCH does. However both clients (Snom and Zoiper) fail in a similar way: For their own reasons (https://www.zoiper.com/en/support/home/article/205/Zoiper%20Push%20Proxy) Zoiper clients register with a 1 month expiry time. If in the 200 OK to a Snom the Zoiper contact header is first, then the Snom assumes that the header is meant for it, reads the large expiry time and uses it (and complains in its log if it's over 14 days). The FreeSWITCH registration for the Snom then expires after 1 hour, which what the Snom asked for in the first place. The Snom then doesn't receive calls. Annoyingly a similar thing happens the other way round - if the Snom Contact header is first in the 200 OK sent to Zoiper, then the Zoiper client just looks at the first header, and complains that the expiry time is too short with a message "You are connecting to a server with a low re-registration time. Battery life will not be optimal. Please contact your VoIP provider or PBX server administrator." (An aside: it would be interesting to know why returning multiple Contact headers is the defined behaviour, and what a user agent is supposed to do with the others headers, and therefore what the consequences of not having them might be.) We are contacting both Snom and Zoiper about this, but for the moment this is a useful FreeSWITCH option to cope with these user agents' behaviours. Martin Martin Paterson Development Team Phone: 0207 953 8840 Email: martin.paterson(a)technologywithin.com Chevron Business Park, Limekiln Lane, Southampton, Hampshire, SO45 2QL Registered Office: CP House, Otterspool Way, Watford, WD25 8JJ, U.K Registered in England No: 5964349 | VAT Number: GB 902 5369 37

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mod lcr cid conditional replace.
by Mimiko 03 Mar '21

03 Mar '21

Hi. I use mod_lcr for dialout operators. On the freeswitch pbx we have extensions 500 to 899. When calling outside number the internal extension is converted ok to for external cid to be accepted by each operator using lcr_cid regular pattern: /(\d\d\d)/${operator_prefix}$1/ The problem lies with one operator where our extensions which start with 5 (500 to 599) must be converted to numbers which start with 9 (900 to 999) and then prefixed with operator's prefix. The rest of extensions does not have such a substitution of first number. How can I do this using regular expression in lcr_cid? Thank you.

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One Way Audio
by William Colls 03 Mar '21

03 Mar '21

I know I have seen this discussed before but can't find the information now. Freeswitch 1.10.5 compiled from source. When I try to make an outbound call the call connects, and the callee can hear me, but I can't hear him/her. Calls between phones registered on freeswitch work, Inbound calls work. The call appears to executed in the public context. Any suggestions as to how to proceed greatly appreciated. Thanks for you time. William

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B-leg ringing after A-leg hangs-up. LUA.
by Тимур Ситдиков 02 Mar '21

02 Mar '21

Hi all. Need help with a simple call script. I'm got this in dialplan <extension name="call_lua_home"> <condition field="destination_number" expression="^(1007)$">  For hangup hook try <action application="lua" data="fork.lua"/> </condition> </extension> And this in fork.lua if session:ready() then api = freeswitch.API() contact = api:execute("sofia_contact", "*/1007(a)compA.com"); caller = session:getVariable("caller_id_number"); session1 = freeswitch.Session("{origination_caller_id_name="..caller.. "}[leg_timeout=30]"..contact..""); -- session:setHangupHook("HangupHook"); if session1:ready() then freeswitch.bridge(session, session1); end freeswitch.consoleLog("WARNING","=====SCRIPT END\n"); session:hangup() end When Caller is hanging up before B-leg answers - Bleg is continuing to ring. There are still 2 channels in 'show channels' after A-leg hangs up. I want to drop B-leg when A-leg hangs up. Is it possible? UPD Tried to use hangup hook. It works with stramFile, but no luck with session. function HangupHook(s, status) freeswitch.consoleLog("WARNING","Event fired breaking out\n"); return exit; -- return die; end session:answer(); session:setHangupHook("HangupHook"); while (session:ready() == true) do session:streamFile( "/usr/share/freeswitch/sounds/en/us/callie/ivr/16000/ivr-on_hold_indefinitely.wav" ); -- session1 = freeswitch.Session("{origination_caller_id_name="..caller.."}[leg_timeout=30]"..contact..""); end freeswitch.consoleLog("WARNING","=====SCRIPT END\n"); session:hangup() end Can anyone help me with this? Thanks! Regards,Tim

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NOTIFYs seem unreliable under high number of SUBSCRIBEs
by Christian Berger 02 Mar '21

02 Mar '21

Hello, we are currently seeing some strange behavior under load conditions. Notifications for calls are sent to seemingly random subsets of subscribed users. This already happens with the default configuration. To recreate this we have created 10000 SUBSCRIBEs with sipp for 1002. sipp -sf sip.xml $ip_of_freeswitch -m 10000 -r 100 Meanwhile sngrep is running to record the dialogues, and we have 2 telephones registered to 1001 and 1002. When 1002 calls 1001, NOTIFYs will be sent out to the subscriptions. However when there are 10000 subscriptions only about 3000-4000 of them will get the NOTIFY. Trying it again will cause another subset of the SUBSCRIBEs to be answered with a NOTIFY. This test was conducted locally on an otherwise idle server, so network loss can be ruled out rather well. Has anybody else had a similar experience? I couldn't find any way sending out NOTIFYs could be limited and the number of NOTIFYs we get seems to be to random for such a limit. Thank you very much for your time Christian Berger foncloud.net Christian Berger foncloud GmbH & Co KG Hahlweg 2a 36093 Künzell Tel: / Fax: +49 661 968990-99 Email: Christian.Berger(a)foncloud.net Web: www.foncloud.net P.S.: Wussten Sie schon? Unter https://www.foncloud.net/wissen finden Sie zahlreiche Informationen und hilfreiche Artikel rund um unsere Produkte und Services. Registergericht: Amtsgericht Fulda, Persönlich haftende Gesellschafterin der foncloud GmbH&Co.KG: Global Brain Network GmbHGeschäftsführer der Global Brain Network GmbH: Peter Krug Sitz der Gesellschaft: Künzell. Diese E-Mail enthält vertrauliche und/oder rechtlich geschützte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorised copying, disclosure or distribution of the material in this e-mail is strictly forbidden.

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Freeswitch selecting the public ip
by David Villasmil 02 Mar '21

02 Mar '21

Hello Guys, Whenever freeswitch triggers an even, on the variable FreeSWITCH-IPv4: Freeswitch always selects the public IP. I made an implementation some time ago which grabs that variable and uses it for something. I need to have that variable as the private ip, not the public one, is this possible at all? Thanks Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Re: [Freeswitch-users] "Shadow calls" - duplicate calls executing in parallel
by David P 01 Mar '21

01 Mar '21

I think the last time we experienced that was with 10.4 (we're on 10.5 now). It happened infrequently as you note. I assumed it was an error in the way our WebRTC app invoked verto. (I didn't have any browser console log to confirm that, though.) On Sat, 27 Feb 2021, 9:07 am , < freeswitch-users-request(a)lists.freeswitch.org> wrote: > > > ---------- Forwarded message ---------- > From: Sam van Herwaarden <svanherwaarden(a)precisionag.org> > > Hi all, > > I was just wondering if others have experienced this: on a few of the > FreeSWITCH systems I've interacted with, I've observed situations where > incoming phone calls seem to be duplicated. So there seem to be two call > processes for the same phone number, also both receiving things like DTMF > events. The timestamps of events tend to be very close (sub-second) but not > necessarily identical. > > It's not very common, less than 1 per 1000 calls has this happening. It > does tend to throw off some of our tooling (both in call handling and > analysis). Curious to know if there could be an obvious cause/easy fix. > > Kind regards, > Sam >

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mod_sndfile error called from mod_tts_commandline - [WARNING] mod_sndfile.c:281 Error Opening File
by Lewis Bergman 27 Feb '21

27 Feb '21

Hello all. I am pretty new to Freeswitch and have an experimental container on Proxmox 6.3 Debian 10 to play with. The goal is to get mod_unimrcp working. To that end, I have mod_tts_commandline (installed from binary with FusionPBX as part of the default install) listed in modules.conf.xml /etc/freeswitch/autoload_configs/modules.conf.xml  <load module="mod_unimrcp"/> <load module="mod_tts_commandline"/> <load module="mod_flite"/> I have tried it with and without mod_flite enabled. Mod_tts_commandline refuses to pull a sound file and the debug log seems to blame mod_sndfile. This is using the example from the mod_tts_commandline page <https://freeswitch.org/confluence/display/FREESWITCH/mod_tts_commandline#mo…> as dialplan shows following the debug log output below: 2021-02-26 15:19:24.713040 [NOTICE] mod_dptools.c:1406 Channel [sofia/internal/102(a)66.151.243.45] has been answered 2021-02-26 15:19:24.713040 [DEBUG] switch_channel.c:3865 (sofia/internal/ 102(a)66.151.243.45) Callstate Change RINGING -> ACTIVE 2021-02-26 15:19:24.713040 [DEBUG] sofia.c:7326 Channel sofia/internal/ 102(a)66.151.243.45 entering state [completed][200] EXECUTE [depth=0] sofia/internal/102(a)66.151.243.45 speak(tts_commandline|pico|This is an example of using tts_commandline) 2021-02-26 15:19:24.713040 [DEBUG] switch_ivr_play_say.c:3023 OPEN TTS tts_commandline 2021-02-26 15:19:24.713040 [DEBUG] switch_ivr_play_say.c:3033 Raw Codec Activated 2021-02-26 15:19:24.713040 [DEBUG] mod_tts_commandline.c:160 Executing: echo 'This is an example of using tts_commandline' | text2wave -f 8000 > '/tmp/0665a30d-2bb8-40fe-b5a6-9625bd58306e.tmp.wav' 2021-02-26 15:19:25.433013 [WARNING] mod_sndfile.c:281 Error Opening File [/tmp/0665a30d-2bb8-40fe-b5a6-9625bd58306e.tmp.wav] [File contains data in an unknown format.] 2021-02-26 15:19:25.433013 [ERR] mod_tts_commandline.c:170 Failed to open file: /tmp/0665a30d-2bb8-40fe-b5a6-9625bd58306e.tmp.wav 2021-02-26 15:19:25.433013 [DEBUG] switch_ivr_play_say.c:2741 Speaking text: This is an example of using tts_commandline 2021-02-26 15:19:25.433013 [DEBUG] sofia.c:7326 Channel sofia/internal/ 102(a)66.151.243.45 entering state [ready][200] 2021-02-26 15:19:25.433013 [DEBUG] switch_ivr_play_say.c:2905 done speaking text 2021-02-26 15:19:25.433013 [NOTICE] switch_core_state_machine.c:386 sofia/internal/102(a)66.151.243.45 has executed the last dialplan instruction, hanging up. 2021-02-26 15:19:25.433013 [NOTICE] switch_core_state_machine.c:388 Hangup sofia/internal/102(a)66.151.243.45 [CS_EXECUTE] [NORMAL_CLEARING] The dialplan for extension I am testing looks like this: <extension name="tts_commandline_test"> <condition field="destination_number" expression="^561$"> <action application="speak" data="tts_commandline|pico|This is an example of using tts_commandline"/> </condition> </extension> The code lines mentioned in debug seem to be pretty mundane, not that I really know what I am looking for: >From mod_tts_commnandline line 160: switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Executing: %s\n", message); I tried setting the /tmp directory to 777 but that didn't help and the error output was identical. mod_tts_commandline never reports an error on creation and it looks like the next part of of mod_tts_commandline would print an error if it did like this: if (switch_system(message, SWITCH_TRUE) < 0) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Failed to execute command: %s\n", message); ret = SWITCH_STATUS_FALSE; goto done; } if (switch_core_file_open(info->fh, info->file, 0, //number_of_channels, info->rate, //samples_per_second, SWITCH_FILE_FLAG_READ | SWITCH_FILE_DATA_SHORT, NULL) != SWITCH_STATUS_SUCCESS) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Failed to open file: %s\n", info->file); ret = SWITCH_STATUS_FALSE; goto done; } Starting at line 279 in mod_sndfile (sorry for the space formatting): if (!context->handle) { if (sndfile_perform_open(context, path, mode, handle) != SWITCH_STATUS_SUCCESS) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "Error Opening File [%s] [%s]\n", path, sf_strerror(context->handle)); status = SWITCH_STATUS_GENERR; goto end; } } According to the dptools docs <https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+playbac…> and libsndfile wav has been supported forever and it would seem highly unlikely that is an issue. I check the /tmp directory and no files are present but I don't know if they are being deleted by some cleanup process or just not created. Not knowing C, it seems like mod_sndfile never received a file handle but that is just a guess. Other Freeswitch apps are writing to the /tmp directory (at least there are some files in there). Maybe someone sees an error from mod_tts_commandline I don't see or knows I am doing something stupid. Any help would be appreciated. I only found 2 references to the error in the mail history and neither had anything to do with a similar issue. Freeswitch version reports: FreeSWITCH Version 1.10.5-release-17-25569c1631~64bit (-release-17-25569c1631 64bit) FusionPBX version, in case it matters, is : 4.5.21 OS is Debian 10 Buster -- Lewis Bergman

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mod_sndfile error called from mod_tts_commandline
by Lewis Bergman 26 Feb '21

26 Feb '21

Hello all. I am pretty new to Freeswitch and have an experimental container on Proxmox 6.3 Debian 10 to play with. The goal is to get mod_unimrcp working. To that end, I have mod_tts_commandline (installed from binary with FusionPBX as part of the default install) listed in modules.conf.xml /etc/freeswitch/autoload_configs/modules.conf.xml  <load module="mod_unimrcp"/> <load module="mod_tts_commandline"/> <load module="mod_flite"/> I have tried it with and without mod_flite enabled. Mod_tts_commandline refuses to pull a sound file and the debug log seems to blame mod_sndfile. This is using the example from the mod_tts_commandline page <https://freeswitch.org/confluence/display/FREESWITCH/mod_tts_commandline#mo…> as dialplan shows following the debug log output below: 2021-02-26 15:19:24.713040 [NOTICE] mod_dptools.c:1406 Channel [sofia/internal/102(a)66.151.243.45] has been answered 2021-02-26 15:19:24.713040 [DEBUG] switch_channel.c:3865 (sofia/internal/ 102(a)66.151.243.45) Callstate Change RINGING -> ACTIVE 2021-02-26 15:19:24.713040 [DEBUG] sofia.c:7326 Channel sofia/internal/ 102(a)66.151.243.45 entering state [completed][200] EXECUTE [depth=0] sofia/internal/102(a)66.151.243.45 speak(tts_commandline|pico|This is an example of using tts_commandline) 2021-02-26 15:19:24.713040 [DEBUG] switch_ivr_play_say.c:3023 OPEN TTS tts_commandline 2021-02-26 15:19:24.713040 [DEBUG] switch_ivr_play_say.c:3033 Raw Codec Activated 2021-02-26 15:19:24.713040 [DEBUG] mod_tts_commandline.c:160 Executing: echo 'This is an example of using tts_commandline' | text2wave -f 8000 > '/tmp/0665a30d-2bb8-40fe-b5a6-9625bd58306e.tmp.wav' 2021-02-26 15:19:25.433013 [WARNING] mod_sndfile.c:281 Error Opening File [/tmp/0665a30d-2bb8-40fe-b5a6-9625bd58306e.tmp.wav] [File contains data in an unknown format.] 2021-02-26 15:19:25.433013 [ERR] mod_tts_commandline.c:170 Failed to open file: /tmp/0665a30d-2bb8-40fe-b5a6-9625bd58306e.tmp.wav 2021-02-26 15:19:25.433013 [DEBUG] switch_ivr_play_say.c:2741 Speaking text: This is an example of using tts_commandline 2021-02-26 15:19:25.433013 [DEBUG] sofia.c:7326 Channel sofia/internal/ 102(a)66.151.243.45 entering state [ready][200] 2021-02-26 15:19:25.433013 [DEBUG] switch_ivr_play_say.c:2905 done speaking text 2021-02-26 15:19:25.433013 [NOTICE] switch_core_state_machine.c:386 sofia/internal/102(a)66.151.243.45 has executed the last dialplan instruction, hanging up. 2021-02-26 15:19:25.433013 [NOTICE] switch_core_state_machine.c:388 Hangup sofia/internal/102(a)66.151.243.45 [CS_EXECUTE] [NORMAL_CLEARING] The dialplan for extension I am testing looks like this: <extension name="tts_commandline_test"> <condition field="destination_number" expression="^561$"> <action application="speak" data="tts_commandline|pico|This is an example of using tts_commandline"/> </condition> </extension> The code lines mentioned in debug seem to be pretty mundane, not that I really know what I am looking for: >From mod_tts_commnandline line 160: switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Executing: %s\n", message); I tried setting the /tmp directory to 777 but that didn't help and the error output was identical. mod_tts_commandline never reports an error on creation and it looks like the next part of of mod_tts_commandline would print an error if it did like this: if (switch_system(message, SWITCH_TRUE) < 0) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Failed to execute command: %s\n", message); ret = SWITCH_STATUS_FALSE; goto done; } if (switch_core_file_open(info->fh, info->file, 0, //number_of_channels, info->rate, //samples_per_second, SWITCH_FILE_FLAG_READ | SWITCH_FILE_DATA_SHORT, NULL) != SWITCH_STATUS_SUCCESS) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Failed to open file: %s\n", info->file); ret = SWITCH_STATUS_FALSE; goto done; } Starting at line 279 in mod_sndfile (sorry for the space formatting): if (!context->handle) { if (sndfile_perform_open(context, path, mode, handle) != SWITCH_STATUS_SUCCESS) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "Error Opening File [%s] [%s]\n", path, sf_strerror(context->handle)); status = SWITCH_STATUS_GENERR; goto end; } } According to the dptools docs <https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+playbac…> and libsndfile wav has been supported forever and it would seem highly unlikely that is an issue. I check the /tmp directory and no files are present but I don't know if they are being deleted by some cleanup process or just not created. Not knowing C, it seems like mod_sndfile never received a file handle but that is just a guess. Other Freeswitch apps are writing to the /tmp directory (at least there are some files in there). Maybe someone sees an error from mod_tts_commandline I don't see or knows I am doing something stupid. Any help would be appreciated. I only found 2 references to the error in the mail history and neither had anything to do with a similar issue. Freeswitch version reports: FreeSWITCH Version 1.10.5-release-17-25569c1631~64bit (-release-17-25569c1631 64bit) FusionPBX version, in case it matters, is : 4.5.21 OS is Debian 10 Buster -- Lewis Bergman

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