From rahman.duran at erzurum.edu.tr Wed Dec 1 10:46:01 2021 From: rahman.duran at erzurum.edu.tr (Rahman Duran) Date: Wed, 01 Dec 2021 10:46:01 -0000 Subject: [Freeswitch-users] Overriding hangup cause in CDR logs In-Reply-To: References: Message-ID: Martin, I was looking for a way to reduce cdr vars :) So I can search or generate reports on the logging server with the same fields. But know I have to consider two separate fields and it does not aggregate nicely on the log server. I also export some variables for "destination_number" as it changes when I do dialplan transfers, and I want to see the original destination number and the reason it changed with exporting lots of other variables. I hopped if I find a way in the cdr template I can combine many of them too. But I see there is no way to do what I want csv_cdr so I will focus on the log parsing side to generate accurate CDRs. Regards, Rahman Martin Paterson , 9 Kas 2021 Sal, 12:43 tarihinde şunu yazdı: > Rahman, > > CDRs don't have a mechanism like you describe, but variables do. You > can put any variable into the CDR and looking back at your original > post, you are doing exactly the right thing here by setting a variable > (cdr_hata anonsu) with the information you require and putting it in > the CDR. Your request was for a real solution - but I think you have > it already. > > Martin. > > Martin Paterson, Pattersong Music > Reduced orchestrations of G&S > > On Mon, 8 Nov 2021 at 23:20, David Villasmil > wrote: > > > > I’ve never tried actually manually setting the reason after hangup, you > may want to try that. > > > > On Mon, 8 Nov 2021 at 19:03, Rahman Duran > wrote: > >> > >> Hi David, > >> > >> You are right but I don't want or need to change any freeswitch > internals. All I need is to fiddle with cdr. So I wonder if the CSV CDR > template has any dynamic mechanism to use on variables. For example can I > say "if variable A is not empty use A, else use variable B" in the CDR > template? > >> > >> Regards, > >> > >> Rahman > >> > >> David Villasmil , 6 Kas 2021 Cmt, > 05:07 tarihinde şunu yazdı: > >>> > >>> The fact is A is hanging up the call. I don’t think you can actually > change this without changing FS source code to override it. > >>> > >>> > >>> On Sat, 6 Nov 2021 at 01:20, Rahman Duran > wrote: > >>>> > >>>> Hi, > >>>> > >>>> Bump. Any hints on this? > >>>> > >>>> Regards, > >>>> > >>>> Rahman > >>>> > >>>> Rahman Duran , 26 Eki 2021 Sal, 09:14 > tarihinde şunu yazdı: > >>>>> > >>>>> Hi, > >>>>> > >>>>> I am using announcements for fail hungup causes like busy, no_answer > etc. Here is my dial plan > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> expression="^true$|^TRUE$|^True$"/> > >>>>> > >>>>> data="call_pickup_group=${user_data(${destination_number}@${domain_name} > var call_pickup_group)}"/> > >>>>> data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"/> > >>>>> data="insert/${domain_name}-group_pickup_last_uuid/${call_pickup_group}/${uuid}"/> > >>>>> data="insert/${domain_name}-pickup_last_uuid/${destination_number}/${uuid}"/> > >>>>> > >>>>> > >>>>> data="nolocal:absolute_codec_string=${ep_codec_string}"/> > >>>>> data="callee_id_name=${user_data(${destination_number}@${domain_name} var > effective_caller_id_name)}"/> > >>>>> > >>>>> > >>>>> > >>>>> data="execute_on_answer=sched_hangup +21600 alloted_timeout" /> > >>>>> data="{origination_callee_id_name=${user_data(${destination_number}@${domain_name} > var effective_caller_id_name)}}user/${destination_number}@ > ${domain_name}"/> > >>>>> data="hata-${originate_disposition} XML hata_anonslari"/> > >>>>> > >>>>> > >>>>> > >>>>> And here is announcements context that I handle hangup causes: > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> expression="^hata-USER_BUSY$"> > >>>>> data="cdr_hata_anonsu=USER_BUSY"/> > >>>>> > >>>>> data="$${anons_dosya_yolu}/user_busy.wav"/> > >>>>> > >>>>> data="${originate_disposition}"/> > >>>>> > >>>>> > >>>>> > >>>>> expression="^hata-NO_ANSWER$"> > >>>>> data="cdr_hata_anonsu=NO_ANSWER"/> > >>>>> > >>>>> data="$${anons_dosya_yolu}/no_answer.wav"/> > >>>>> > >>>>> data="${originate_disposition}"/> > >>>>> > >>>>> > >>>>> > >>>>> expression="^hata-USER_NOT_REGISTERED$"> > >>>>> data="cdr_hata_anonsu=USER_NOT_REGISTERED"/> > >>>>> > >>>>> data="$${anons_dosya_yolu}/user_not_registered.wav"/> > >>>>> > >>>>> data="${originate_disposition}"/> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> data="cdr_hata_anonsu=${originate_disposition}"/> > >>>>> data="${originate_disposition}"/> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> The problem is if the calling leg does not listen the announcement > to the end and hangup, CDR logs shows "Originator Cancel" as hangup cause. > As I already know the real hangup cause, how can I override the CDR hangup > cause with the real one? For now I am setting another variable > (cdr_hata_anonsu) and added it to CDR logs, but if possible I want to fix > this with a real solution. > >>>>> > >>>>> Regards, > >>>>> > >>>>> Rahman Duran > >>>> > >>>> > _________________________________________________________________________ > >>>> > >>>> 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 at 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 at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> https://freeswitch.com > >>> > >>> -- > >>> Regards, > >>> > >>> David Villasmil > >>> email: david.villasmil.work at gmail.com > >>> phone: +34669448337 > >>> > _________________________________________________________________________ > >>> > >>> 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 at 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 at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> https://freeswitch.com > >> > >> > _________________________________________________________________________ > >> > >> 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 at 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 at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> https://freeswitch.com > > > > -- > > Regards, > > > > David Villasmil > > email: david.villasmil.work at gmail.com > > phone: +34669448337 > > _________________________________________________________________________ > > > > 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 at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > https://freeswitch.com > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From rahman.duran at erzurum.edu.tr Wed Dec 1 10:46:15 2021 From: rahman.duran at erzurum.edu.tr (Rahman Duran) Date: Wed, 01 Dec 2021 10:46:15 -0000 Subject: [Freeswitch-users] Overriding hangup cause in CDR logs In-Reply-To: References: Message-ID: I already tried to set hangup_cause in my "error anouncement" dialplan extension. But as A leg hangup before playback ends, Freeswitch does not execute remaining actions in the "error anouncement" dialplan extension so it does not work. In "freeradius" configuration, I can write dynamic templates for access and accounting logs like this "%{%{Aruba-Location-Id}:-%{%{Siemens-AP-Name}:-none}}" This will evaluates first "Aruba-Location-Id" and use its value. If it is empty or not exists, then it lookup for "Siemens-AP-Name". If it is empty than it uses string "none". I hoped for some dynamic markup language in Freeswitch csv_cdr module so I can write a template for my needs. I don't want to use xml_cdr because all I need is to key-value based cdr logs sent to syslog (graylog) and analysed there. If this is not possible I will try to use xml_cdr but this will add more complexity and I fear I will loose cdr records is http server is down and Freeswitch continue to operate. P.S. I already log both a and b legs so this is not about legs. Rahman Duran David Villasmil , 9 Kas 2021 Sal, 02:46 tarihinde şunu yazdı: > I’ve never tried actually manually setting the reason after hangup, you > may want to try that. > > On Mon, 8 Nov 2021 at 19:03, Rahman Duran > wrote: > >> Hi David, >> >> You are right but I don't want or need to change any freeswitch >> internals. All I need is to fiddle with cdr. So I wonder if the CSV CDR >> template has any dynamic mechanism to use on variables. For example can I >> say "if variable A is not empty use A, else use variable B" in the CDR >> template? >> >> Regards, >> >> Rahman >> >> David Villasmil , 6 Kas 2021 Cmt, 05:07 >> tarihinde şunu yazdı: >> >>> The fact is A is hanging up the call. I don’t think you can actually >>> change this without changing FS source code to override it. >>> >>> >>> On Sat, 6 Nov 2021 at 01:20, Rahman Duran >>> wrote: >>> >>>> Hi, >>>> >>>> Bump. Any hints on this? >>>> >>>> Regards, >>>> >>>> Rahman >>>> >>>> Rahman Duran , 26 Eki 2021 Sal, 09:14 >>>> tarihinde şunu yazdı: >>>> >>>>> Hi, >>>>> >>>>> I am using announcements for fail hungup causes like busy, no_answer >>>>> etc. Here is my dial plan >>>>> >>>>> >>>>> >>>>> >>>>> >>>> expression="^true$|^TRUE$|^True$"/> >>>>> >>>>> >>>> data="call_pickup_group=${user_data(${destination_number}@${domain_name} >>>>> var call_pickup_group)}"/> >>>>> >>>> data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"/> >>>>> >>>> data="insert/${domain_name}-group_pickup_last_uuid/${call_pickup_group}/${uuid}"/> >>>>> >>>> data="insert/${domain_name}-pickup_last_uuid/${destination_number}/${uuid}"/> >>>>> >>>>> >>>>> >>>> data="nolocal:absolute_codec_string=${ep_codec_string}"/> >>>>> >>>> data="callee_id_name=${user_data(${destination_number}@${domain_name} >>>>> var effective_caller_id_name)}"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> data="{origination_callee_id_name=${user_data(${destination_number}@${domain_name} >>>>> var effective_caller_id_name)}}user/${destination_number}@ >>>>> ${domain_name}"/> >>>>> >>>> data="hata-${originate_disposition} XML hata_anonslari"/> >>>>> >>>>> >>>>> >>>>> And here is announcements context that I handle hangup causes: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> expression="^hata-USER_BUSY$"> >>>>> >>>>> >>>>> >>>> data="$${anons_dosya_yolu}/user_busy.wav"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> expression="^hata-NO_ANSWER$"> >>>>> >>>>> >>>>> >>>> data="$${anons_dosya_yolu}/no_answer.wav"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> expression="^hata-USER_NOT_REGISTERED$"> >>>>> >>>> data="cdr_hata_anonsu=USER_NOT_REGISTERED"/> >>>>> >>>>> >>>> data="$${anons_dosya_yolu}/user_not_registered.wav"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> data="cdr_hata_anonsu=${originate_disposition}"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> The problem is if the calling leg does not listen the announcement to >>>>> the end and hangup, CDR logs shows "Originator Cancel" as hangup cause. As >>>>> I already know the real hangup cause, how can I override the CDR hangup >>>>> cause with the real one? For now I am setting another variable >>>>> (cdr_hata_anonsu) and added it to CDR logs, but if possible I want to fix >>>>> this with a real solution. >>>>> >>>>> Regards, >>>>> >>>>> Rahman Duran >>>>> >>>> >>>> _________________________________________________________________________ >>>> >>>> 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 at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> https://freeswitch.com >>> >>> -- >>> Regards, >>> >>> David Villasmil >>> email: david.villasmil.work at gmail.com >>> phone: +34669448337 >>> _________________________________________________________________________ >>> >>> 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 at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> _________________________________________________________________________ >> >> 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > -- > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com David Villasmil , 9 Kas 2021 Sal, 02:46 tarihinde şunu yazdı: > I’ve never tried actually manually setting the reason after hangup, you > may want to try that. > > On Mon, 8 Nov 2021 at 19:03, Rahman Duran > wrote: > >> Hi David, >> >> You are right but I don't want or need to change any freeswitch >> internals. All I need is to fiddle with cdr. So I wonder if the CSV CDR >> template has any dynamic mechanism to use on variables. For example can I >> say "if variable A is not empty use A, else use variable B" in the CDR >> template? >> >> Regards, >> >> Rahman >> >> David Villasmil , 6 Kas 2021 Cmt, 05:07 >> tarihinde şunu yazdı: >> >>> The fact is A is hanging up the call. I don’t think you can actually >>> change this without changing FS source code to override it. >>> >>> >>> On Sat, 6 Nov 2021 at 01:20, Rahman Duran >>> wrote: >>> >>>> Hi, >>>> >>>> Bump. Any hints on this? >>>> >>>> Regards, >>>> >>>> Rahman >>>> >>>> Rahman Duran , 26 Eki 2021 Sal, 09:14 >>>> tarihinde şunu yazdı: >>>> >>>>> Hi, >>>>> >>>>> I am using announcements for fail hungup causes like busy, no_answer >>>>> etc. Here is my dial plan >>>>> >>>>> >>>>> >>>>> >>>>> >>>> expression="^true$|^TRUE$|^True$"/> >>>>> >>>>> >>>> data="call_pickup_group=${user_data(${destination_number}@${domain_name} >>>>> var call_pickup_group)}"/> >>>>> >>>> data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"/> >>>>> >>>> data="insert/${domain_name}-group_pickup_last_uuid/${call_pickup_group}/${uuid}"/> >>>>> >>>> data="insert/${domain_name}-pickup_last_uuid/${destination_number}/${uuid}"/> >>>>> >>>>> >>>>> >>>> data="nolocal:absolute_codec_string=${ep_codec_string}"/> >>>>> >>>> data="callee_id_name=${user_data(${destination_number}@${domain_name} >>>>> var effective_caller_id_name)}"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> data="{origination_callee_id_name=${user_data(${destination_number}@${domain_name} >>>>> var effective_caller_id_name)}}user/${destination_number}@ >>>>> ${domain_name}"/> >>>>> >>>> data="hata-${originate_disposition} XML hata_anonslari"/> >>>>> >>>>> >>>>> >>>>> And here is announcements context that I handle hangup causes: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> expression="^hata-USER_BUSY$"> >>>>> >>>>> >>>>> >>>> data="$${anons_dosya_yolu}/user_busy.wav"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> expression="^hata-NO_ANSWER$"> >>>>> >>>>> >>>>> >>>> data="$${anons_dosya_yolu}/no_answer.wav"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> expression="^hata-USER_NOT_REGISTERED$"> >>>>> >>>> data="cdr_hata_anonsu=USER_NOT_REGISTERED"/> >>>>> >>>>> >>>> data="$${anons_dosya_yolu}/user_not_registered.wav"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> data="cdr_hata_anonsu=${originate_disposition}"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> The problem is if the calling leg does not listen the announcement to >>>>> the end and hangup, CDR logs shows "Originator Cancel" as hangup cause. As >>>>> I already know the real hangup cause, how can I override the CDR hangup >>>>> cause with the real one? For now I am setting another variable >>>>> (cdr_hata_anonsu) and added it to CDR logs, but if possible I want to fix >>>>> this with a real solution. >>>>> >>>>> Regards, >>>>> >>>>> Rahman Duran >>>>> >>>> >>>> _________________________________________________________________________ >>>> >>>> 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 at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> https://freeswitch.com >>> >>> -- >>> Regards, >>> >>> David Villasmil >>> email: david.villasmil.work at gmail.com >>> phone: +34669448337 >>> _________________________________________________________________________ >>> >>> 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 at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> _________________________________________________________________________ >> >> 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > -- > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From Antony.Stone at freeswitch.open.source.it Wed Dec 1 11:27:33 2021 From: Antony.Stone at freeswitch.open.source.it (Antony Stone) Date: Wed, 1 Dec 2021 12:27:33 +0100 Subject: [Freeswitch-users] Assigning a variable with a value from a system command? Message-ID: <202112011227.34223.Antony.Stone@freeswitch.open.source.it> Hi. I have a need to assign a variable in vars.xml with the value output from a system command (in this specific instance "hostname -s", but I'd like to have a generic solution for future use). How can I do that? Any suggestions are welcome, as is pointing at any documentation I've so far failed to find :) Thanks, Antony. -- Having been asked for a reference for this man, I can confirm that you will be very lucky indeed if you can get him to work for you. Please reply to the list; please *don't* CC me. From martin at pattersong.co.uk Wed Dec 1 16:03:59 2021 From: martin at pattersong.co.uk (Martin Paterson) Date: Wed, 1 Dec 2021 16:03:59 +0000 Subject: [Freeswitch-users] 302 handling with enterprise :_: dial. In-Reply-To: References: Message-ID: Thanks for your response, Brian. Actually I think I need the opposite of that option! I want FreeSWITCH to handle the 302s in the dialplan, but the problem is that when using :_: the redirect ends up in the public dialplan not the default dialplan (where the extensions are). When not using :_: the redirect works fine - the call goes through the default diaplan and finds the extension. However the option has led me to the FS code that handles this, so I shall take a look there. Thanks, Martin. Martin Paterson, Pattersong Music Reduced orchestrations of G&S On Tue, 30 Nov 2021 at 18:24, Brian West wrote: > You will want to set outbound_redirect_fatal=true, or the enterprise > originate will follow the 302, you can safely ignore those if needed. > > On Tue, Nov 30, 2021 at 12:09 PM Martin Paterson > wrote: > >> I have an issue that I’m struggling to resolve. I’ve tried this out in >> the vanilla config. >> >> If a bridged destination in the default context: >> > data="user/1001@${domain_name}:_:user/1002@${domain_name}"/> >> >> but it has the effect that if a destination returns 302 Moved >> Temporarily (1002 forwards to 1003 here), then it doesn’t run through >> the dialplan the same context, it goes to the public context and fails >> because the destination (1003) is an extension in the default context. >> The log looks like it’s handling the 302 as if it’s a new call: >> >> [INFO] sofia.c:10462 sofia/internal/1000 at 192.168.1.226 receiving >> invite from 192.168.1.226:5060 version: 1.10.7 -release-19-883d2cb662 >> 64bit call-id: ee6cd84a-cca5-123a-d6a1-080027d0c8d3 >> [INFO] mod_dialplan_xml.c:639 Processing Extension 1000 <1000>->1003 >> in context public >> EXECUTE [depth=0] sofia/internal/1000 at 192.168.1.226 deflect(1003) >> >> I must be missing something in my understanding here – I don’t >> understand why the behaviour is different, but more interestingly, is >> there a way of getting the enterprise dial to process the 302s in the >> same way as when dialling one destination? >> >> Best wishes, >> >> Martin. >> >> Martin Paterson, Pattersong Music >> Reduced orchestrations of G&S >> >> _________________________________________________________________________ >> >> 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > > -- > > Brian West | Co-founder and Developer > > Need Commercial support? email sales at freeswitch.com > > FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 > > > Email: brian at freeswitch.com > > Mobile: 918-424-9378 > > Website: https://www.FreeSWITCH.com > > [image: https://www.facebook.com/signalwireinc?src=email] > [image: > https://twitter.com/freeswitch] > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.com Wed Dec 1 17:24:57 2021 From: brian at freeswitch.com (Brian West) Date: Wed, 1 Dec 2021 11:24:57 -0600 Subject: [Freeswitch-users] 302 handling with enterprise :_: dial. In-Reply-To: References: Message-ID: You really can't use enterprise originate for that. It requires a lot more thought, you can't exit the loops since each one is in its own threads, the only option is to ignore. On Wed, Dec 1, 2021 at 10:51 AM Martin Paterson wrote: > Thanks for your response, Brian. Actually I think I need the opposite of > that option! I want FreeSWITCH to handle the 302s in the dialplan, but the > problem is that when using :_: the redirect ends up in the public dialplan > not the default dialplan (where the extensions are). When not using :_: the > redirect works fine - the call goes through the default diaplan and finds > the extension. However the option has led me to the FS code that handles > this, so I shall take a look there. > > Thanks, > > Martin. > > Martin Paterson, Pattersong Music > Reduced orchestrations of G&S > > > On Tue, 30 Nov 2021 at 18:24, Brian West wrote: > >> You will want to set outbound_redirect_fatal=true, or the enterprise >> originate will follow the 302, you can safely ignore those if needed. >> >> On Tue, Nov 30, 2021 at 12:09 PM Martin Paterson >> wrote: >> >>> I have an issue that I’m struggling to resolve. I’ve tried this out in >>> the vanilla config. >>> >>> If a bridged destination in the default context: >>> >> data="user/1001@${domain_name}:_:user/1002@${domain_name}"/> >>> >>> but it has the effect that if a destination returns 302 Moved >>> Temporarily (1002 forwards to 1003 here), then it doesn’t run through >>> the dialplan the same context, it goes to the public context and fails >>> because the destination (1003) is an extension in the default context. >>> The log looks like it’s handling the 302 as if it’s a new call: >>> >>> [INFO] sofia.c:10462 sofia/internal/1000 at 192.168.1.226 receiving >>> invite from 192.168.1.226:5060 version: 1.10.7 -release-19-883d2cb662 >>> 64bit call-id: ee6cd84a-cca5-123a-d6a1-080027d0c8d3 >>> [INFO] mod_dialplan_xml.c:639 Processing Extension 1000 <1000>->1003 >>> in context public >>> EXECUTE [depth=0] sofia/internal/1000 at 192.168.1.226 deflect(1003) >>> >>> I must be missing something in my understanding here – I don’t >>> understand why the behaviour is different, but more interestingly, is >>> there a way of getting the enterprise dial to process the 302s in the >>> same way as when dialling one destination? >>> >>> Best wishes, >>> >>> Martin. >>> >>> Martin Paterson, Pattersong Music >>> Reduced orchestrations of G&S >>> >>> _________________________________________________________________________ >>> >>> 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 at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> >> >> -- >> >> Brian West | Co-founder and Developer >> >> Need Commercial support? email sales at freeswitch.com >> >> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 >> >> >> Email: brian at freeswitch.com >> >> Mobile: 918-424-9378 >> >> Website: https://www.FreeSWITCH.com >> >> [image: https://www.facebook.com/signalwireinc?src=email] >> [image: >> https://twitter.com/freeswitch] >> _________________________________________________________________________ >> >> 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.com Thu Dec 2 01:42:20 2021 From: brian at freeswitch.com (Brian West) Date: Wed, 1 Dec 2021 19:42:20 -0600 Subject: [Freeswitch-users] 302 handling with enterprise :_: dial. In-Reply-To: References: Message-ID: You aren't understanding that when you use enterprise originate it's like doing a parallel dial, each one stands alone, one does a 302, it will END them all. Make sure manual redirect is setup, per confluence, and do a debug trace of it attempting this, I bet you can see why in that log. On Wed, Dec 1, 2021 at 10:51 AM Martin Paterson wrote: > Thanks for your response, Brian. Actually I think I need the opposite of > that option! I want FreeSWITCH to handle the 302s in the dialplan, but the > problem is that when using :_: the redirect ends up in the public dialplan > not the default dialplan (where the extensions are). When not using :_: the > redirect works fine - the call goes through the default diaplan and finds > the extension. However the option has led me to the FS code that handles > this, so I shall take a look there. > > Thanks, > > Martin. > > Martin Paterson, Pattersong Music > Reduced orchestrations of G&S > > > On Tue, 30 Nov 2021 at 18:24, Brian West wrote: > >> You will want to set outbound_redirect_fatal=true, or the enterprise >> originate will follow the 302, you can safely ignore those if needed. >> >> On Tue, Nov 30, 2021 at 12:09 PM Martin Paterson >> wrote: >> >>> I have an issue that I’m struggling to resolve. I’ve tried this out in >>> the vanilla config. >>> >>> If a bridged destination in the default context: >>> >> data="user/1001@${domain_name}:_:user/1002@${domain_name}"/> >>> >>> but it has the effect that if a destination returns 302 Moved >>> Temporarily (1002 forwards to 1003 here), then it doesn’t run through >>> the dialplan the same context, it goes to the public context and fails >>> because the destination (1003) is an extension in the default context. >>> The log looks like it’s handling the 302 as if it’s a new call: >>> >>> [INFO] sofia.c:10462 sofia/internal/1000 at 192.168.1.226 receiving >>> invite from 192.168.1.226:5060 version: 1.10.7 -release-19-883d2cb662 >>> 64bit call-id: ee6cd84a-cca5-123a-d6a1-080027d0c8d3 >>> [INFO] mod_dialplan_xml.c:639 Processing Extension 1000 <1000>->1003 >>> in context public >>> EXECUTE [depth=0] sofia/internal/1000 at 192.168.1.226 deflect(1003) >>> >>> I must be missing something in my understanding here – I don’t >>> understand why the behaviour is different, but more interestingly, is >>> there a way of getting the enterprise dial to process the 302s in the >>> same way as when dialling one destination? >>> >>> Best wishes, >>> >>> Martin. >>> >>> Martin Paterson, Pattersong Music >>> Reduced orchestrations of G&S >>> >>> _________________________________________________________________________ >>> >>> 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 at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> >> >> -- >> >> Brian West | Co-founder and Developer >> >> Need Commercial support? email sales at freeswitch.com >> >> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 >> >> >> Email: brian at freeswitch.com >> >> Mobile: 918-424-9378 >> >> Website: https://www.FreeSWITCH.com >> >> [image: https://www.facebook.com/signalwireinc?src=email] >> [image: >> https://twitter.com/freeswitch] >> _________________________________________________________________________ >> >> 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From dragos at freeswitch.org Thu Dec 2 10:04:12 2021 From: dragos at freeswitch.org (Dragos Oancea) Date: Thu, 2 Dec 2021 12:04:12 +0200 Subject: [Freeswitch-users] Assigning a variable with a value from a system command? In-Reply-To: <202112011227.34223.Antony.Stone@freeswitch.open.source.it> References: <202112011227.34223.Antony.Stone@freeswitch.open.source.it> Message-ID: Do it through environment. We also have "exec-set". https://freeswitch.org/confluence/display/FREESWITCH/Understanding+the+Configuration+Files On Wed, Dec 1, 2021 at 4:03 PM Antony Stone < Antony.Stone at freeswitch.open.source.it> wrote: > Hi. > > I have a need to assign a variable in vars.xml with the value output from > a > system command (in this specific instance "hostname -s", but I'd like to > have a > generic solution for future use). > > How can I do that? > > Any suggestions are welcome, as is pointing at any documentation I've so > far > failed to find :) > > > Thanks, > > > Antony. > > -- > Having been asked for a reference for this man, > I can confirm that you will be very lucky indeed > if you can get him to work for you. > > Please reply to the > list; > please *don't* CC > me. > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From martin at pattersong.co.uk Thu Dec 2 10:58:43 2021 From: martin at pattersong.co.uk (Martin Paterson) Date: Thu, 2 Dec 2021 10:58:43 +0000 Subject: [Freeswitch-users] Overriding hangup cause in CDR logs In-Reply-To: References: Message-ID: Rahman, If it's any consolation, I do exactly the same with the initial destination. As soon as the call arrives, I store the destination_number in a variable initial_destination. I also have a history variable that gets appended to each time something interesting happens in the dialplan. Both go in the CDR. Martin. Martin Paterson, Pattersong Music Reduced orchestrations of G&S On Tue, 9 Nov 2021 at 09:57, Rahman Duran wrote: > > Martin, > > I was looking for a way to reduce cdr vars :) So I can search or generate reports on the logging server with the same fields. But know I have to consider two separate fields and it does not aggregate nicely on the log server. I also export some variables for "destination_number" as it changes when I do dialplan transfers, and I want to see the original destination number and the reason it changed with exporting lots of other variables. I hopped if I find a way in the cdr template I can combine many of them too. > > But I see there is no way to do what I want csv_cdr so I will focus on the log parsing side to generate accurate CDRs. > > Regards, > > Rahman > > Martin Paterson , 9 Kas 2021 Sal, 12:43 tarihinde şunu yazdı: >> >> Rahman, >> >> CDRs don't have a mechanism like you describe, but variables do. You >> can put any variable into the CDR and looking back at your original >> post, you are doing exactly the right thing here by setting a variable >> (cdr_hata anonsu) with the information you require and putting it in >> the CDR. Your request was for a real solution - but I think you have >> it already. >> >> Martin. >> >> Martin Paterson, Pattersong Music >> Reduced orchestrations of G&S >> >> On Mon, 8 Nov 2021 at 23:20, David Villasmil >> wrote: >> > >> > I’ve never tried actually manually setting the reason after hangup, you may want to try that. >> > >> > On Mon, 8 Nov 2021 at 19:03, Rahman Duran wrote: >> >> >> >> Hi David, >> >> >> >> You are right but I don't want or need to change any freeswitch internals. All I need is to fiddle with cdr. So I wonder if the CSV CDR template has any dynamic mechanism to use on variables. For example can I say "if variable A is not empty use A, else use variable B" in the CDR template? >> >> >> >> Regards, >> >> >> >> Rahman >> >> >> >> David Villasmil , 6 Kas 2021 Cmt, 05:07 tarihinde şunu yazdı: >> >>> >> >>> The fact is A is hanging up the call. I don’t think you can actually change this without changing FS source code to override it. >> >>> >> >>> >> >>> On Sat, 6 Nov 2021 at 01:20, Rahman Duran wrote: >> >>>> >> >>>> Hi, >> >>>> >> >>>> Bump. Any hints on this? >> >>>> >> >>>> Regards, >> >>>> >> >>>> Rahman >> >>>> >> >>>> Rahman Duran , 26 Eki 2021 Sal, 09:14 tarihinde şunu yazdı: >> >>>>> >> >>>>> Hi, >> >>>>> >> >>>>> I am using announcements for fail hungup causes like busy, no_answer etc. Here is my dial plan >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> And here is announcements context that I handle hangup causes: >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> The problem is if the calling leg does not listen the announcement to the end and hangup, CDR logs shows "Originator Cancel" as hangup cause. As I already know the real hangup cause, how can I override the CDR hangup cause with the real one? For now I am setting another variable (cdr_hata_anonsu) and added it to CDR logs, but if possible I want to fix this with a real solution. >> >>>>> >> >>>>> Regards, >> >>>>> >> >>>>> Rahman Duran >> >>>> >> >>>> _________________________________________________________________________ >> >>>> >> >>>> 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 at 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 at lists.freeswitch.org >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>>> https://freeswitch.com >> >>> >> >>> -- >> >>> Regards, >> >>> >> >>> David Villasmil >> >>> email: david.villasmil.work at gmail.com >> >>> phone: +34669448337 >> >>> _________________________________________________________________________ >> >>> >> >>> 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 at 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 at lists.freeswitch.org >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>> https://freeswitch.com >> >> >> >> _________________________________________________________________________ >> >> >> >> 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 at 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 at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> https://freeswitch.com >> > >> > -- >> > Regards, >> > >> > David Villasmil >> > email: david.villasmil.work at gmail.com >> > phone: +34669448337 >> > _________________________________________________________________________ >> > >> > 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 at 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 at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > https://freeswitch.com >> >> _________________________________________________________________________ >> >> 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com From Antony.Stone at freeswitch.open.source.it Thu Dec 2 10:22:29 2021 From: Antony.Stone at freeswitch.open.source.it (Antony Stone) Date: Thu, 2 Dec 2021 11:22:29 +0100 Subject: [Freeswitch-users] Assigning a variable with a value from a system command? In-Reply-To: References: <202112011227.34223.Antony.Stone@freeswitch.open.source.it> Message-ID: <202112021122.29220.Antony.Stone@freeswitch.open.source.it> On Thursday 02 December 2021 at 11:04:12, Dragos Oancea wrote: > Do it through environment. > So, that allows a FreeSwitch variable to have the value of an environment variable? That's not quite the same as running a command and assigning the result to a variable, which is what I need. > We also have "exec-set". Ah, now *that* looks like what I'm looking for - thanks :) Antony -- Bill Gates has personally assured the Spanish Academy that he will never allow the upside-down question mark to disappear from Microsoft word-processing programs, which must be reassuring for millions of Spanish-speaking people, though just a piddling afterthought as far as he's concerned. - Lynne Truss, "Eats, Shoots and Leaves" Please reply to the list; please *don't* CC me. From alexanderhenryperkins at gmail.com Thu Dec 2 19:19:47 2021 From: alexanderhenryperkins at gmail.com (Alexander Perkins) Date: Thu, 2 Dec 2021 14:19:47 -0500 Subject: [Freeswitch-users] Getting the Media IP Message-ID: Hi All. Is it possible to get the media IP from an invite? We have a client that does not want to accept calls into his PBX if the media IP is not in the US. Is this even possible? If so, how? Thank you, Alex -------------- next part -------------- An HTML attachment was scrubbed... URL: From dragos at freeswitch.org Fri Dec 3 08:02:27 2021 From: dragos at freeswitch.org (Dragos Oancea) Date: Fri, 3 Dec 2021 10:02:27 +0200 Subject: [Freeswitch-users] Getting the Media IP In-Reply-To: References: Message-ID: Check value of channel variable "remote_media_ip". There are other channel variables that may help in cases like this: #define SWITCH_R_SDP_VARIABLE "switch_r_sdp" #define SWITCH_REMOTE_MEDIA_IP_VARIABLE "remote_media_ip" #define SWITCH_REMOTE_MEDIA_PORT_VARIABLE "remote_media_port" #define SWITCH_REMOTE_VIDEO_IP_VARIABLE "remote_video_ip" #define SWITCH_REMOTE_VIDEO_PORT_VARIABLE "remote_video_port See switch_types.h. On Thu, Dec 2, 2021 at 9:20 PM Alexander Perkins < alexanderhenryperkins at gmail.com> wrote: > Hi All. Is it possible to get the media IP from an invite? We have a > client that does not want to accept calls into his PBX if the media IP is > not in the US. Is this even possible? If so, how? > > Thank you, > Alex > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From k4kaleem at gmail.com Fri Dec 3 10:35:59 2021 From: k4kaleem at gmail.com (kaleem rehman) Date: Fri, 3 Dec 2021 10:35:59 +0000 Subject: [Freeswitch-users] WebRTC calls one way with custom sip messages UUI In-Reply-To: References: Message-ID: Hi All, any takes on this plz. On Fri, Nov 26, 2021 at 5:35 PM kaleem rehman wrote: > Hi Kaiduan, > > thanks for looking into it. > > Verto looks cool. we have no restriction as to what to use. main item is > to attach data to call so sip client at end user can strip and show data to > agent. > > no need for user to login to enter credentials, we want simple "*call us"* > type button which generates a call. > to make it safe from attacks as server will be on cloud, we would like > some sort of safety measure, either a login and pwd, which gets passed to > freeswitch to verify its genuine call from a webpage or app. or some hidden > message within the generate call command so freeswitch can verify and drop > any calls which arent from right source so answering party doesnt get too > many junk calls from random bots who discover port is open on Cloud FS. > > Regards, > K > ---------- Forwarded message ---------- > From: kaiduan xie > To: "freeswitch-users at lists.freeswitch.org" -users at lists.freeswitch.org> > Cc: > Bcc: > Date: Fri, 26 Nov 2021 02:57:01 +0000 (UTC) > Subject: Re: [Freeswitch-users] WebRTC calls one way with custom sip > messages UUI > You can use JSON based VERTO protocol instead of SIP to make things > easier. Does the user have to login in FS? > > /Kaiduan > > On Thursday, November 25, 2021, 03:30:52 p.m. EST, kaleem rehman < > k4kaleem at gmail.com> wrote: > > > Salaam Ehtasham, > > we are looking to use JSSIP or SIPJS, we are flexible and can look into > SIPML if for any reason we have to. > > Regards, > Kaleem > > ---------- Forwarded message ---------- > From: Ehtasham Ul-Haq > To: FreeSWITCH Users Help > Cc: Ahmed Hasan > Bcc: > Date: Thu, 25 Nov 2021 15:28:31 +0500 > Subject: Re: [Freeswitch-users] WebRTC calls one way with custom sip > messages UUI > Hi > Which Library you are using to start a call from Website ? > > Malik Ehtasham, CTI Product Manager / Technical Lead (Mr. ) > > WWW:[image: domain2.png].expertflow.com > FB: [image: FB-f-Logo__blue_29.png]/Expertflow > LinkedIn: [image: linkedIn.png] > /company/expertflow > Youtube: [image: YouTube-social-square_red_128px.png]/user/expertflow > Twitter: [image: twitter.JPG] > /Expertflow > 361 Model Town Lahore Pakistan ; Mobile +92 3347815664; email, Cisco > Spark and Google Talk: ehtasham.malik at expertflow.com > ; Skype: > *shani.awan3* > > > On Thu, Nov 25, 2021 at 4:16 AM kaleem rehman wrote: > > Hi All, > > our requirement is simple, we will have CALL US button on website > > when they click, we want a call generated to our FS Server via WebRTC (no > need for calls from FS to Users, it will be one way only from User to > Server. > > With call we want to send additional data like URL of page they on, login > if they are logged in. > we can get data like URL and userlogin but want to sent it with SIP call > as SIP Message (Probably as USER to USER Information) so we can pull at > other end. > > any ideas of achieving this > Thanks, > Kaleem > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > -------------- next part -------------- An HTML attachment was scrubbed... URL: From Antony.Stone at freeswitch.open.source.it Thu Dec 2 20:13:12 2021 From: Antony.Stone at freeswitch.open.source.it (Antony Stone) Date: Thu, 2 Dec 2021 21:13:12 +0100 Subject: [Freeswitch-users] Getting the Media IP In-Reply-To: References: Message-ID: <202112022113.12489.Antony.Stone@freeswitch.open.source.it> On Thursday 02 December 2021 at 20:19:47, Alexander Perkins wrote: > Hi All. Is it possible to get the media IP from an invite? We have a > client that does not want to accept calls into his PBX if the media IP is > not in the US. Is this even possible? If so, how? https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables variable_remote_media_ip ? Antony. -- The difference between theory and practice is that in theory there is no difference, whereas in practice there is. Please reply to the list; please *don't* CC me. From support at naud.io Fri Dec 3 19:16:34 2021 From: support at naud.io (Support from NetworkedAudio LLC) Date: Fri, 3 Dec 2021 19:16:34 +0000 Subject: [Freeswitch-users] WebRTC calls one way with custom sip messages UUI In-Reply-To: References: Message-ID: So the incoming request, Verto, WebRTC, SIPJS, whatever still gets authenticated with whatever credentials the web page supplies. So you could set up anonymous registration, and validate the credentials in the dial plan. You could dynamically validate the user and password and use those as tokens. You could also enforce only certain CODECs, for instance Opus, and anyone not using any of those would weed out most scripts. These measures, and Fail2Ban will prevent some unauthorized access but won’t help with DDoS or anyone actively looking to cause trouble (if an authentication token is provided by HTTPS its trivial to grab that if someone really wants to be malicious). Most other options would be expensive (hide behind CloudFlare) or onerous (use a CAPTCHA as authentication). It comes down to balancing requirements. If a client asked this from me we’d propose a one-time code provided on a Verto client that had a ten second timeout for login. The issue is ________________________________ From: FreeSWITCH-users on behalf of kaleem rehman Sent: Friday, December 3, 2021 5:08 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] WebRTC calls one way with custom sip messages UUI Hi All, any takes on this plz. On Fri, Nov 26, 2021 at 5:35 PM kaleem rehman > wrote: Hi Kaiduan, thanks for looking into it. Verto looks cool. we have no restriction as to what to use. main item is to attach data to call so sip client at end user can strip and show data to agent. no need for user to login to enter credentials, we want simple "call us" type button which generates a call. to make it safe from attacks as server will be on cloud, we would like some sort of safety measure, either a login and pwd, which gets passed to freeswitch to verify its genuine call from a webpage or app. or some hidden message within the generate call command so freeswitch can verify and drop any calls which arent from right source so answering party doesnt get too many junk calls from random bots who discover port is open on Cloud FS. Regards, K ---------- Forwarded message ---------- From: kaiduan xie > To: "freeswitch-users at lists.freeswitch.org" > Cc: Bcc: Date: Fri, 26 Nov 2021 02:57:01 +0000 (UTC) Subject: Re: [Freeswitch-users] WebRTC calls one way with custom sip messages UUI You can use JSON based VERTO protocol instead of SIP to make things easier. Does the user have to login in FS? /Kaiduan On Thursday, November 25, 2021, 03:30:52 p.m. EST, kaleem rehman > wrote: Salaam Ehtasham, we are looking to use JSSIP or SIPJS, we are flexible and can look into SIPML if for any reason we have to. Regards, Kaleem ---------- Forwarded message ---------- From: Ehtasham Ul-Haq > To: FreeSWITCH Users Help > Cc: Ahmed Hasan > Bcc: Date: Thu, 25 Nov 2021 15:28:31 +0500 Subject: Re: [Freeswitch-users] WebRTC calls one way with custom sip messages UUI Hi Which Library you are using to start a call from Website ? Malik Ehtasham, CTI Product Manager / Technical Lead (Mr. ) [https://lh6.googleusercontent.com/yE5sw1FNhk7A8YXwHvqJ_1eoUhx6Rly7EZQn8JVLClQgesUimaC5FBhoqcjL_0TiGnF8-z4kSqJywWlkYPcySl1rLcS18hzt1PbCqtGbvbtl6TVHQlFadPziihRZ17vCkCMArghS] WWW:[domain2.png].expertflow.com FB: [FB-f-Logo__blue_29.png] /Expertflow LinkedIn: [linkedIn.png] /company/expertflow Youtube: [YouTube-social-square_red_128px.png] /user/expertflow Twitter: [twitter.JPG] /Expertflow 361 Model Town Lahore Pakistan ; Mobile +92 3347815664; email, Cisco Spark and Google Talk: ehtasham.malik at expertflow.com; Skype: shani.awan3 On Thu, Nov 25, 2021 at 4:16 AM kaleem rehman > wrote: Hi All, our requirement is simple, we will have CALL US button on website when they click, we want a call generated to our FS Server via WebRTC (no need for calls from FS to Users, it will be one way only from User to Server. With call we want to send additional data like URL of page they on, login if they are logged in. we can get data like URL and userlogin but want to sent it with SIP call as SIP Message (Probably as USER to USER Information) so we can pull at other end. any ideas of achieving this Thanks, Kaleem _________________________________________________________________________ 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com _________________________________________________________________________ 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From k4kaleem at gmail.com Fri Dec 3 22:52:54 2021 From: k4kaleem at gmail.com (kaleem rehman) Date: Fri, 3 Dec 2021 22:52:54 +0000 Subject: [Freeswitch-users] WebRTC calls one way with custom sip messages UUI In-Reply-To: References: Message-ID: hi Thanks for ur msg, it was chopped towards the end says: ""the issue is" and nothing after. this gives me idea on solution. on verto demo, i can see it asks for login and then extension to make call. if i want all this hidden from user and in background. so end users sees a button only in app and it says "call us" once clicked everything is in background like logging softphone and starting call. all users hear is freeswitch answering and playing relative IVR script. what would be best way to achieve this. also, from Verto side, if i want to attach some data with call. what method i need to use to achieve. thanks, k On Fri, Dec 3, 2021 at 7:17 PM Support from NetworkedAudio LLC < support at naud.io> wrote: > So the incoming request, Verto, WebRTC, SIPJS, whatever still gets > authenticated with whatever credentials the web page supplies. > > So you could set up anonymous registration, and validate the credentials > in the dial plan. > > You could dynamically validate the user and password and use those as > tokens. > You could also enforce only certain CODECs, for instance Opus, and anyone > not using any of those would weed out most scripts. > > These measures, and Fail2Ban will prevent some unauthorized access but > won’t help with DDoS or anyone actively looking to cause trouble (if an > authentication token is provided by HTTPS its trivial to grab that if > someone really wants to be malicious). > > Most other options would be expensive (hide behind CloudFlare) or onerous > (use a CAPTCHA as authentication). It comes down to balancing requirements. > > If a client asked this from me we’d propose a one-time code provided on a > Verto client that had a ten second timeout for login. > > > > The issue is > ------------------------------ > *From:* FreeSWITCH-users > on behalf of kaleem rehman > *Sent:* Friday, December 3, 2021 5:08 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] WebRTC calls one way with custom sip > messages UUI > > Hi All, > > any takes on this plz. > > On Fri, Nov 26, 2021 at 5:35 PM kaleem rehman wrote: > >> Hi Kaiduan, >> >> thanks for looking into it. >> >> Verto looks cool. we have no restriction as to what to use. main item is >> to attach data to call so sip client at end user can strip and show data to >> agent. >> >> no need for user to login to enter credentials, we want simple "*call >> us"* type button which generates a call. >> to make it safe from attacks as server will be on cloud, we would like >> some sort of safety measure, either a login and pwd, which gets passed to >> freeswitch to verify its genuine call from a webpage or app. or some hidden >> message within the generate call command so freeswitch can verify and drop >> any calls which arent from right source so answering party doesnt get too >> many junk calls from random bots who discover port is open on Cloud FS. >> >> Regards, >> K >> ---------- Forwarded message ---------- >> From: kaiduan xie >> To: "freeswitch-users at lists.freeswitch.org" > -users at lists.freeswitch.org> >> Cc: >> Bcc: >> Date: Fri, 26 Nov 2021 02:57:01 +0000 (UTC) >> Subject: Re: [Freeswitch-users] WebRTC calls one way with custom sip >> messages UUI >> You can use JSON based VERTO protocol instead of SIP to make things >> easier. Does the user have to login in FS? >> >> /Kaiduan >> >> On Thursday, November 25, 2021, 03:30:52 p.m. EST, kaleem rehman < >> k4kaleem at gmail.com> wrote: >> >> >> Salaam Ehtasham, >> >> we are looking to use JSSIP or SIPJS, we are flexible and can look into >> SIPML if for any reason we have to. >> >> Regards, >> Kaleem >> >> ---------- Forwarded message ---------- >> From: Ehtasham Ul-Haq >> To: FreeSWITCH Users Help >> Cc: Ahmed Hasan >> Bcc: >> Date: Thu, 25 Nov 2021 15:28:31 +0500 >> Subject: Re: [Freeswitch-users] WebRTC calls one way with custom sip >> messages UUI >> Hi >> Which Library you are using to start a call from Website ? >> >> Malik Ehtasham, CTI Product Manager / Technical Lead (Mr. ) >> >> WWW:[image: domain2.png].expertflow.com >> FB: [image: FB-f-Logo__blue_29.png]/Expertflow >> LinkedIn: [image: linkedIn.png] >> /company/expertflow >> Youtube: [image: YouTube-social-square_red_128px.png]/user/expertflow >> Twitter: [image: twitter.JPG] >> /Expertflow >> 361 Model Town Lahore Pakistan ; Mobile +92 3347815664; email, Cisco >> Spark and Google Talk: ehtasham.malik at expertflow.com >> ; Skype: >> *shani.awan3* >> >> >> On Thu, Nov 25, 2021 at 4:16 AM kaleem rehman wrote: >> >> Hi All, >> >> our requirement is simple, we will have CALL US button on website >> >> when they click, we want a call generated to our FS Server via WebRTC (no >> need for calls from FS to Users, it will be one way only from User to >> Server. >> >> With call we want to send additional data like URL of page they on, login >> if they are logged in. >> we can get data like URL and userlogin but want to sent it with SIP call >> as SIP Message (Probably as USER to USER Information) so we can pull at >> other end. >> >> any ideas of achieving this >> Thanks, >> Kaleem >> _________________________________________________________________________ >> >> 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com >> >> _________________________________________________________________________ >> >> 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com >> > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From dujinfang at gmail.com Sun Dec 5 00:56:11 2021 From: dujinfang at gmail.com (Seven Du) Date: Sun, 5 Dec 2021 08:56:11 +0800 Subject: [Freeswitch-users] Serving sofia.conf via lua (updated) In-Reply-To: <202111301120.02543.Antony.Stone@freeswitch.open.source.it> References: <202111301120.02543.Antony.Stone@freeswitch.open.source.it> Message-ID: https://github.com/seven1240/xui it is outdated and stopped updating. but the lua scripts should still work. On Tue, Nov 30, 2021 at 10:08 PM Antony Stone < Antony.Stone at freeswitch.open.source.it> wrote: > Hi. > > Sending again with a bit of clarification - my original message hasn't hit > the > list yet, so I'm hoping people can reply to just this one... > > > I am trying to get Freeswitch to get its sofia configuation from lua > instead of > static XML files. > > https://freeswitch.org/confluence/display/FREESWITCH/mod_xml_curl tells > me for > example that "Section: configuration - This is used to send back > configuration > files such as sofia.conf." Yes, I know that that page refers to > mod_xml_curl > and not mod_xml_lua but I'm pretty certain that the principle is the same, > and > > https://freeswitch.org/confluence/display/FREESWITCH/Serving+Configuration+with+Lua > seems to confirm this, although is it much thinner on examples. I do not > that > this page also says "there is no Lua page yet, and Lua-related docs are > scattered all over the place..." :( > > So, anyway, I'm pretty sure what I want to do should be possible, but > nowhere > have I so far been able to find a fully working example telling me how to > do > it. > > My understanding so far is that: > > 1. I need to create a lua script which generates the *complete* XML as > would > be served from /etc/freeswitch/autoload_configs/sofia.conf.xml and all its > included sub-directories/files > > 2. This XML as output from the script should be enclosed within tags: > > >
> ... XML goes here ... >
>
> > 3. I should place this lua script in freeswitch's scripting directory, > which > on my (Debian) machine is /usr/share/freeswitch/scripts > > 4. I need to enable this script to be run by including in > /etc/freeswitch/autoload_configs/lua.conf.xml: > > > > > > The first thing which is not clear to me is what I should do about > /etc/freeswitch/autoload_configs/sofia.conf.xml > > - if I leave it as it is, it appears to be used in the normal way, and my > lua > script is ignored > - if I delete it (or rename it to sofia.conf.xml.noload for example) then > sofia > doesn't get loaded *at all* and when I go into fs_cli I do not even have a > "sofia" command > - if I "touch" the filename so that > /etc/reeswitch/autoload_configs/sofia.conf.xml exists but is empty, I get > the > same result egain - the "sofia" command does not even exist in the cli > > > So, can anyone point me at how to join these things up correctly so that I > can > serve sofia configuration settings from a lua script? > > > I've been told here that: > > On Friday 05 November 2021 at 12:28:55, David Villasmil wrote: > > > In FS there’s the embedded lua. You can do almost anything with it. > > I'm struggling to find the documentation telling me how, though :( > > > Thanks in advance, > > > Antony. > > -- > "It would appear we have reached the limits of what it is possible to > achieve > with computer technology, although one should be careful with such > statements; > they tend to sound pretty silly in five years." > > - John von Neumann (1949) > > Please reply to the > list; > please *don't* CC > me. > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn -------------- next part -------------- An HTML attachment was scrubbed... URL: From dujinfang at gmail.com Sun Dec 5 00:58:53 2021 From: dujinfang at gmail.com (Seven Du) Date: Sun, 5 Dec 2021 08:58:53 +0800 Subject: [Freeswitch-users] Assigning a variable with a value from a system command? In-Reply-To: <202112021122.29220.Antony.Stone@freeswitch.open.source.it> References: <202112011227.34223.Antony.Stone@freeswitch.open.source.it> <202112021122.29220.Antony.Stone@freeswitch.open.source.it> Message-ID: try exec-set On Thu, Dec 2, 2021 at 11:46 PM Antony Stone < Antony.Stone at freeswitch.open.source.it> wrote: > On Thursday 02 December 2021 at 11:04:12, Dragos Oancea wrote: > > > Do it through environment. > > > > So, that allows a FreeSwitch variable to have the value of an environment > variable? > > That's not quite the same as running a command and assigning the result to > a > variable, which is what I need. > > > We also have "exec-set". > > Ah, now *that* looks like what I'm looking for - thanks :) > > > Antony > > -- > Bill Gates has personally assured the Spanish Academy that he will never > allow > the upside-down question mark to disappear from Microsoft word-processing > programs, which must be reassuring for millions of Spanish-speaking > people, > though just a piddling afterthought as far as he's concerned. > > - Lynne Truss, "Eats, Shoots and Leaves" > > Please reply to the > list; > please *don't* CC > me. > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn -------------- next part -------------- An HTML attachment was scrubbed... URL: From dujinfang at gmail.com Sun Dec 5 01:00:07 2021 From: dujinfang at gmail.com (Seven Du) Date: Sun, 5 Dec 2021 09:00:07 +0800 Subject: [Freeswitch-users] Getting the Media IP In-Reply-To: References: Message-ID: you can get the entire sdp in channel var On Fri, Dec 3, 2021 at 4:27 AM Alexander Perkins < alexanderhenryperkins at gmail.com> wrote: > Hi All. Is it possible to get the media IP from an invite? We have a > client that does not want to accept calls into his PBX if the media IP is > not in the US. Is this even possible? If so, how? > > Thank you, > Alex > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn -------------- next part -------------- An HTML attachment was scrubbed... URL: From gregor at infomedia.si Mon Dec 6 08:55:35 2021 From: gregor at infomedia.si (Gregor Nanger) Date: Mon, 6 Dec 2021 09:55:35 +0100 Subject: [Freeswitch-users] Verto - Semantic Plan B Message-ID: Hi! We are getting this error on latest version of Chrome and Verto: There has been a problem retrieving the streams - did you allow access? Check Device Resolution DOMException: Failed to construct 'RTCPeerConnection': Plan B SDP semantics is a legacy version of the Session Description Protocol that has severe compatibility issues on modern browsers and is no longer supported. See https://www.chromestatus.com/feature/5823036655665152 for more details, including the possibility of registering for a Deprecation Trial in order to extend the Plan B deprecation deadline for a limited amount of time. at FSRTCPeerConnection (xxxxx/jquery.FSRTC.js?cdv=264:702:20 ) at Object.onSuccess [as onsuccess] (xxxxx/jquery.FSRTC.js?cdv=264:602:25 ) atxxxx/jquery.FSRTC.js?cdv=264:1015:25 Is there something to change in js or does verto need to support this? BR Gregor -------------- next part -------------- An HTML attachment was scrubbed... URL: From gregor at infomedia.si Mon Dec 6 09:07:57 2021 From: gregor at infomedia.si (Gregor Nanger) Date: Mon, 6 Dec 2021 10:07:57 +0100 Subject: [Freeswitch-users] Verto - Semantic Plan B In-Reply-To: References: Message-ID: I've changed in jquery.FSRTC.js to: config.sdpSemantics = "unified-plan"; And it works. Anyone else have same problem? BR, Gregor On Mon, 6 Dec 2021 at 09:55, Gregor Nanger wrote: > Hi! > > We are getting this error on latest version of Chrome and Verto: > > There has been a problem retrieving the streams - did you allow access? > Check Device Resolution DOMException: Failed to construct > 'RTCPeerConnection': Plan B SDP semantics is a legacy version of the > Session Description Protocol that has severe compatibility issues on modern > browsers and is no longer supported. See > https://www.chromestatus.com/feature/5823036655665152 for more details, > including the possibility of registering for a Deprecation Trial in order > to extend the Plan B deprecation deadline for a limited amount of time. > at FSRTCPeerConnection (xxxxx/jquery.FSRTC.js?cdv=264:702:20 > > ) > at Object.onSuccess [as onsuccess] (xxxxx > /jquery.FSRTC.js?cdv=264:602:25 > > ) > atxxxx/jquery.FSRTC.js?cdv=264:1015:25 > > > Is there something to change in js or does verto need to support this? > > BR Gregor > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From Antony.Stone at freeswitch.open.source.it Fri Dec 3 15:40:15 2021 From: Antony.Stone at freeswitch.open.source.it (Antony Stone) Date: Fri, 3 Dec 2021 16:40:15 +0100 Subject: [Freeswitch-users] Mailing list is very slow? Message-ID: <202112031640.15416.Antony.Stone@freeswitch.open.source.it> Hi. I wonder if there any list admins here who can either explain why this list is so slow, or preferably do something about it? For example, yesterday Alexander Perkins posted a question about Media IP. The question was timed at 2021-12-02 19:19:47 UTC I received the question from the list at 20:02:17 UTC and I replied to it at 20:13:12 UTC My own reply has just come back to me from the list at 2021-12-03 15:13:13 UTC - round-trip time = 19 hours! Looking at the headers of the question and my answer, I see the following timings (all UTC): 2021-12-02 19:19:47 Question is written and sent 2021-12-02 19:19:58 Question is received by a Google mail server 2021-12-02 19:20:29 Question is received by lists.freeswitch.org from Google 2021-12-02 19:20:31 Question is received by lists.freeswitch.org from itself 2021-12-02 20:02:17 Question is received by my inbound mail server from lists.freeswitch.org 2021-12-02 20:13:12 Answer is written and sent 2021-12-02 20:13:18 Answer is received by my outbound mail server 2021-12-02 20:13:19 Answer is received by lists.freeswitch.org from my server 2021-12-03 14:35:01 Answer is received by lists.freeswitch.org from itself 2021-12-03 15:13:13 Answer is received by my inbound mail server from lists.freeswitch.org So, that's approximately 40 minutes for the question to get from the original poster to me as a subscriber, and 19 hours for my answer to do the same round trip. Does anyone know what's causing things to be so slow, and can anyone fix it? Thanks, Antony. -- "I estimate there's a world market for about five computers." - Thomas J Watson, Chairman of IBM Please reply to the list; please *don't* CC me. From gmaruzz at gmail.com Mon Dec 6 12:16:46 2021 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Mon, 6 Dec 2021 13:16:46 +0100 Subject: [Freeswitch-users] Mailing list is very slow? In-Reply-To: <202112031640.15416.Antony.Stone@freeswitch.open.source.it> References: <202112031640.15416.Antony.Stone@freeswitch.open.source.it> Message-ID: On Mon, Dec 6, 2021 at 1:12 PM Antony Stone < Antony.Stone at freeswitch.open.source.it> wrote: > Hi. > > I wonder if there any list admins here who can either explain why this > list is > so slow, or preferably do something about it? > > IIRC posts are manually moderated to avoid spam Does anyone know what's causing things to be so slow, and can anyone fix it? > > > -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From dujinfang at gmail.com Mon Dec 6 13:17:19 2021 From: dujinfang at gmail.com (Seven Du) Date: Mon, 6 Dec 2021 21:17:19 +0800 Subject: [Freeswitch-users] Verto - Semantic Plan B In-Reply-To: References: Message-ID: yes, it works for me. can you make a pr? On Mon, Dec 6, 2021 at 5:41 PM Gregor Nanger wrote: > I've changed in jquery.FSRTC.js to: > config.sdpSemantics = "unified-plan"; > > And it works. > > Anyone else have same problem? > > BR, Gregor > > On Mon, 6 Dec 2021 at 09:55, Gregor Nanger wrote: > >> Hi! >> >> We are getting this error on latest version of Chrome and Verto: >> >> There has been a problem retrieving the streams - did you allow access? >> Check Device Resolution DOMException: Failed to construct >> 'RTCPeerConnection': Plan B SDP semantics is a legacy version of the >> Session Description Protocol that has severe compatibility issues on modern >> browsers and is no longer supported. See >> https://www.chromestatus.com/feature/5823036655665152 for more details, >> including the possibility of registering for a Deprecation Trial in order >> to extend the Plan B deprecation deadline for a limited amount of time. >> at FSRTCPeerConnection (xxxxx/jquery.FSRTC.js?cdv=264:702:20 >> >> ) >> at Object.onSuccess [as onsuccess] (xxxxx >> /jquery.FSRTC.js?cdv=264:602:25 >> >> ) >> atxxxx/jquery.FSRTC.js?cdv=264:1015:25 >> >> >> Is there something to change in js or does verto need to support this? >> >> BR Gregor >> >> > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn -------------- next part -------------- An HTML attachment was scrubbed... URL: From gregor at infomedia.si Mon Dec 6 14:26:38 2021 From: gregor at infomedia.si (Gregor Nanger) Date: Mon, 6 Dec 2021 15:26:38 +0100 Subject: [Freeswitch-users] Verto - Semantic Plan B In-Reply-To: References: Message-ID: I do not know which version of jquery.FSRTC.js I had. I had: config.sdpSemantics = "plan-b"; I copied new version which doesn't have this line and it is also working fine. On Mon, 6 Dec 2021 at 14:18, Seven Du wrote: > yes, it works for me. can you make a pr? > > > On Mon, Dec 6, 2021 at 5:41 PM Gregor Nanger wrote: > >> I've changed in jquery.FSRTC.js to: >> config.sdpSemantics = "unified-plan"; >> >> And it works. >> >> Anyone else have same problem? >> >> BR, Gregor >> >> On Mon, 6 Dec 2021 at 09:55, Gregor Nanger wrote: >> >>> Hi! >>> >>> We are getting this error on latest version of Chrome and Verto: >>> >>> There has been a problem retrieving the streams - did you allow access? >>> Check Device Resolution DOMException: Failed to construct >>> 'RTCPeerConnection': Plan B SDP semantics is a legacy version of the >>> Session Description Protocol that has severe compatibility issues on modern >>> browsers and is no longer supported. See >>> https://www.chromestatus.com/feature/5823036655665152 for more details, >>> including the possibility of registering for a Deprecation Trial in order >>> to extend the Plan B deprecation deadline for a limited amount of time. >>> at FSRTCPeerConnection (xxxxx/jquery.FSRTC.js?cdv=264:702:20 >>> >>> ) >>> at Object.onSuccess [as onsuccess] (xxxxx >>> /jquery.FSRTC.js?cdv=264:602:25 >>> >>> ) >>> atxxxx/jquery.FSRTC.js?cdv=264:1015:25 >>> >>> >>> Is there something to change in js or does verto need to support this? >>> >>> BR Gregor >>> >>> >> >> _________________________________________________________________________ >> >> 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > > -- > About: http://about.me/dujinfang > Blog: http://www.dujinfang.com > Proj: http://www.freeswitch.org.cn > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From lists at telium.io Mon Dec 6 15:00:30 2021 From: lists at telium.io (TTT) Date: Mon, 6 Dec 2021 15:00:30 +0000 Subject: [Freeswitch-users] Switch to MySQL db for config Message-ID: <0100017d9042b57d-0d6a19bd-5c8b-4f70-8408-fdfe648873c3-000000@email.amazonses.com> I would like to use MySQL to hold my FreeSWITCH config. I found instructions on how to setup the DSN, but how to I create the db/table structure? Is there a script to do this? -------------- next part -------------- An HTML attachment was scrubbed... URL: From nzaytsevc at gmail.com Mon Dec 6 15:35:01 2021 From: nzaytsevc at gmail.com (Nikolay Zaytsev) Date: Mon, 6 Dec 2021 17:35:01 +0200 Subject: [Freeswitch-users] Sereverless ESL and KSUUID support Message-ID: Hi Everyone, Could you please assist with those 2 questions: 1) Has anybody created an AWS serverless ESL controller for FreeSWITCH? If yes, could you please share the experience and the technology stack that was used. 2) Does FreeSWITCH support KSUUID instead of UUID? https://github.com/segmentio/ksuid Is there any roadmap for that? -Best Reagrds, Nikolay Zaytsev -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Mon Dec 6 17:31:56 2021 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 6 Dec 2021 17:31:56 +0000 Subject: [Freeswitch-users] Sereverless ESL and KSUUID support In-Reply-To: References: Message-ID: Hello, Last time I looked that lambda, there’s a running limit of 15 minutes, so it was not suited for esl. Things probably have changed, tho. On Mon, 6 Dec 2021 at 15:35, Nikolay Zaytsev wrote: > Hi Everyone, > > Could you please assist with those 2 questions: > > 1) Has anybody created an AWS serverless ESL controller for FreeSWITCH? > If yes, could you please share the experience and the technology stack > that was used. > > 2) Does FreeSWITCH support KSUUID instead of UUID? > https://github.com/segmentio/ksuid > Is there any roadmap for that? > > -Best Reagrds, > Nikolay Zaytsev > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.com Mon Dec 6 18:54:24 2021 From: brian at freeswitch.com (Brian West) Date: Mon, 6 Dec 2021 12:54:24 -0600 Subject: [Freeswitch-users] Sereverless ESL and KSUUID support In-Reply-To: References: Message-ID: > > 2) Does FreeSWITCH support KSUUID instead of UUID? > https://github.com/segmentio/ksuid > Is there any roadmap for that? > Unless there is a ksuuid library in C that is license compatible, it's not going to happen. /b -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From m.hald at outlook.com Tue Dec 7 01:43:52 2021 From: m.hald at outlook.com (Marcel Haldemann) Date: Tue, 7 Dec 2021 01:43:52 +0000 Subject: [Freeswitch-users] Sereverless ESL and KSUUID support In-Reply-To: References: Message-ID: Hi, Perhaps mod_AMQP would be an option, since AWS Lambda supports RabbitMQ?. I don't know anything about AWS and Lambda / Serverless tho. https://aws.amazon.com/about-aws/whats-new/2021/07/aws-lambda-now-supports-amazon-mq-for-rabbitmq-as-an-event-source/ Von: FreeSWITCH-users Im Auftrag von Nikolay Zaytsev Gesendet: Montag, 6. Dezember 2021 16:35 An: freeswitch-users at lists.freeswitch.org Betreff: [Freeswitch-users] Sereverless ESL and KSUUID support Hi Everyone, Could you please assist with those 2 questions: 1) Has anybody created an AWS serverless ESL controller for FreeSWITCH? If yes, could you please share the experience and the technology stack that was used. 2) Does FreeSWITCH support KSUUID instead of UUID? https://github.com/segmentio/ksuid Is there any roadmap for that? -Best Reagrds, Nikolay Zaytsev -------------- next part -------------- An HTML attachment was scrubbed... URL: From Antony.Stone at freeswitch.open.source.it Mon Dec 6 16:01:51 2021 From: Antony.Stone at freeswitch.open.source.it (Antony Stone) Date: Mon, 6 Dec 2021 17:01:51 +0100 Subject: [Freeswitch-users] Switch to MySQL db for config In-Reply-To: <0100017d9042b57d-0d6a19bd-5c8b-4f70-8408-fdfe648873c3-000000@email.amazonses.com> References: <0100017d9042b57d-0d6a19bd-5c8b-4f70-8408-fdfe648873c3-000000@email.amazonses.com> Message-ID: <202112061701.51558.Antony.Stone@freeswitch.open.source.it> On Monday 06 December 2021 at 16:00:30, TTT wrote: > I would like to use MySQL to hold my FreeSWITCH config. What do you mean by "config"? All of it, or just certain aspects (such as user directory, SIP registrations, dialplan...)? > I found instructions on how to setup the DSN, Point us at the URL so that we know what you're doing? > but how to I create the db/table structure? Is there a script to do this? What are you using as the "glue" between MySQL and FreeSwitch? Common options would be xml-curl or lua. In both cases you create your DB tables however you like, and you then write a lua script or create something that curl can interrogate, to return the XML which you would otherwise have placed into a flat file for FreeSwitch to parse. Regards, Antony. -- .evah I serutangis sseltniop tsom eht fo eno eb tsum sihT Please reply to the list; please *don't* CC me. From nzaytsevc at gmail.com Tue Dec 7 09:29:41 2021 From: nzaytsevc at gmail.com (Nikolay Zaytsev) Date: Tue, 7 Dec 2021 11:29:41 +0200 Subject: [Freeswitch-users] Sereverless ESL and KSUUID support In-Reply-To: References: Message-ID: Thank everyone for the reply. Regarding the KSUUID, understood. For the serverless part, lamda is limited to 15 minutes, that is right. But what do you think about the following design? 1. Have a bunch of lambda functions each corresponds to a certain event. 2. No data is stored in memory all go to Dynamo DB. 3. On the event triggered from FreeSWITCH Lamda queries the Dynamo DB for the config and related call data. 4. Lambda does processing and sends the commands to FreeSWITCH 5. Lambda updates the Dynamo DB. In case you agree that the above design might work, what do you think about the tech stack to use? I was thinking about python switchio ( https://github.com/friends-of-freeswitch/switchio), but 1) In the case of using serverless there is no need to use the async approach from the ESL engine 2) I am not sure it is developed anymore Maybe vanilla python ESL will work fine in this case. Please advise. Best regards, Nikolay Zaytsev -------------- next part -------------- An HTML attachment was scrubbed... URL: From martin at pattersong.co.uk Tue Dec 7 13:28:33 2021 From: martin at pattersong.co.uk (Martin Paterson) Date: Tue, 7 Dec 2021 13:28:33 +0000 Subject: [Freeswitch-users] 302 handling with enterprise :_: dial. In-Reply-To: References: Message-ID: Thanks for that Brian. Are you saying then that it is impossible to follow 302s when you do multiple outdials with enterprise originate, the only choices are ignore the 302 or follow but cancel the other outdials? But if that's how it is, then so be it. The 'Freeswitch IVR Originate' page implies the opposite though. Of enterprise originate it says: 'This can be helpful when dealing with call forwarding. Without it, FreeSWITCH would drop the simultaneous dial and transfer to the forwarded extension. For example, I'm bridging to two SIP phones, 101 and 102, with "," between them. 101 redirects to some other URL. 102 will now stop ringing. With :_: it wouldn't'. Anyhow my users really like the convenience of turning forwarding on and off by pressing a button on the (Snom) handset which chooses whether it returns 302 or not. So plan B is going to be to configure the Snoms to dial a code when hitting the button instead. The code then gets handled in the dialplan which remembers the setting. Then when a forwarded handset is subsequently dialed, logic in the dialplan can replace that with the forwarded number. Martin Paterson, Pattersong Music Reduced orchestrations of G&S On Thu, 2 Dec 2021 at 01:42, Brian West wrote: > You aren't understanding that when you use enterprise originate it's like > doing a parallel dial, each one stands alone, one does a 302, it will END > them all. Make sure manual redirect is setup, per confluence, and do a > debug trace of it attempting this, I bet you can see why in that log. > > On Wed, Dec 1, 2021 at 10:51 AM Martin Paterson > wrote: > >> Thanks for your response, Brian. Actually I think I need the opposite of >> that option! I want FreeSWITCH to handle the 302s in the dialplan, but the >> problem is that when using :_: the redirect ends up in the public dialplan >> not the default dialplan (where the extensions are). When not using :_: the >> redirect works fine - the call goes through the default diaplan and finds >> the extension. However the option has led me to the FS code that handles >> this, so I shall take a look there. >> >> Thanks, >> >> Martin. >> >> Martin Paterson, Pattersong Music >> Reduced orchestrations of G&S >> >> >> On Tue, 30 Nov 2021 at 18:24, Brian West wrote: >> >>> You will want to set outbound_redirect_fatal=true, or the enterprise >>> originate will follow the 302, you can safely ignore those if needed. >>> >>> On Tue, Nov 30, 2021 at 12:09 PM Martin Paterson < >>> martin at pattersong.co.uk> wrote: >>> >>>> I have an issue that I’m struggling to resolve. I’ve tried this out in >>>> the vanilla config. >>>> >>>> If a bridged destination in the default context: >>>> >>> data="user/1001@${domain_name}:_:user/1002@${domain_name}"/> >>>> >>>> but it has the effect that if a destination returns 302 Moved >>>> Temporarily (1002 forwards to 1003 here), then it doesn’t run through >>>> the dialplan the same context, it goes to the public context and fails >>>> because the destination (1003) is an extension in the default context. >>>> The log looks like it’s handling the 302 as if it’s a new call: >>>> >>>> [INFO] sofia.c:10462 sofia/internal/1000 at 192.168.1.226 receiving >>>> invite from 192.168.1.226:5060 version: 1.10.7 -release-19-883d2cb662 >>>> 64bit call-id: ee6cd84a-cca5-123a-d6a1-080027d0c8d3 >>>> [INFO] mod_dialplan_xml.c:639 Processing Extension 1000 <1000>->1003 >>>> in context public >>>> EXECUTE [depth=0] sofia/internal/1000 at 192.168.1.226 deflect(1003) >>>> >>>> I must be missing something in my understanding here – I don’t >>>> understand why the behaviour is different, but more interestingly, is >>>> there a way of getting the enterprise dial to process the 302s in the >>>> same way as when dialling one destination? >>>> >>>> Best wishes, >>>> >>>> Martin. >>>> >>>> Martin Paterson, Pattersong Music >>>> Reduced orchestrations of G&S >>>> >>>> >>>> _________________________________________________________________________ >>>> >>>> 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 at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> https://freeswitch.com >>> >>> >>> >>> -- >>> >>> Brian West | Co-founder and Developer >>> >>> Need Commercial support? email sales at freeswitch.com >>> >>> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 >>> >>> >>> Email: brian at freeswitch.com >>> >>> Mobile: 918-424-9378 >>> >>> Website: https://www.FreeSWITCH.com >>> >>> [image: https://www.facebook.com/signalwireinc?src=email] >>> [image: >>> https://twitter.com/freeswitch] >>> _________________________________________________________________________ >>> >>> 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 at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> _________________________________________________________________________ >> >> 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > > -- > > Brian West | Co-founder and Developer > > Need Commercial support? email sales at freeswitch.com > > FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 > > > Email: brian at freeswitch.com > > Mobile: 918-424-9378 > > Website: https://www.FreeSWITCH.com > > [image: https://www.facebook.com/signalwireinc?src=email] > [image: > https://twitter.com/freeswitch] > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From dujinfang at gmail.com Tue Dec 7 14:00:05 2021 From: dujinfang at gmail.com (Seven Du) Date: Tue, 7 Dec 2021 22:00:05 +0800 Subject: [Freeswitch-users] Sereverless ESL and KSUUID support In-Reply-To: References: Message-ID: I believe go can generate a .so could be called by C. Also there's a rust lib. someone will eventually make a C version ;) On Tue, Dec 7, 2021 at 3:20 AM Brian West wrote: > 2) Does FreeSWITCH support KSUUID instead of UUID? >> https://github.com/segmentio/ksuid >> Is there any roadmap for that? >> > > Unless there is a ksuuid library in C that is license compatible, it's not > going to happen. > > /b > > > -- > > Brian West | Co-founder and Developer > > Need Commercial support? email sales at freeswitch.com > > FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 > > > Email: brian at freeswitch.com > > Mobile: 918-424-9378 > > Website: https://www.FreeSWITCH.com > > [image: https://www.facebook.com/signalwireinc?src=email] > [image: > https://twitter.com/freeswitch] > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.com Tue Dec 7 14:11:23 2021 From: brian at freeswitch.com (Brian West) Date: Tue, 7 Dec 2021 08:11:23 -0600 Subject: [Freeswitch-users] 302 handling with enterprise :_: dial. In-Reply-To: References: Message-ID: You can also enable the Snom to do an HTTP request to enable/disable forwarding so its done server side too. On Tue, Dec 7, 2021 at 7:52 AM Martin Paterson wrote: > Thanks for that Brian. Are you saying then that it is impossible to follow > 302s when you do multiple outdials with enterprise originate, the only > choices are ignore the 302 or follow but cancel the other outdials? But if > that's how it is, then so be it. The 'Freeswitch IVR Originate' page > implies the opposite though. Of enterprise originate it says: > 'This can be helpful when dealing with call forwarding. Without it, > FreeSWITCH would drop the simultaneous dial and transfer to the forwarded > extension. For example, I'm bridging to two SIP phones, 101 and 102, with > "," between them. 101 redirects to some other URL. 102 will now stop > ringing. With :_: it wouldn't'. > > Anyhow my users really like the convenience of turning forwarding on and > off by pressing a button on the (Snom) handset which chooses whether it > returns 302 or not. So plan B is going to be to configure the Snoms to dial > a code when hitting the button instead. The code then gets handled in the > dialplan which remembers the setting. Then when a forwarded handset is > subsequently dialed, logic in the dialplan can replace that with the > forwarded number. > > Martin Paterson, Pattersong Music > Reduced orchestrations of G&S > > > On Thu, 2 Dec 2021 at 01:42, Brian West wrote: > >> You aren't understanding that when you use enterprise originate it's like >> doing a parallel dial, each one stands alone, one does a 302, it will END >> them all. Make sure manual redirect is setup, per confluence, and do a >> debug trace of it attempting this, I bet you can see why in that log. >> >> On Wed, Dec 1, 2021 at 10:51 AM Martin Paterson >> wrote: >> >>> Thanks for your response, Brian. Actually I think I need the opposite of >>> that option! I want FreeSWITCH to handle the 302s in the dialplan, but the >>> problem is that when using :_: the redirect ends up in the public dialplan >>> not the default dialplan (where the extensions are). When not using :_: the >>> redirect works fine - the call goes through the default diaplan and finds >>> the extension. However the option has led me to the FS code that handles >>> this, so I shall take a look there. >>> >>> Thanks, >>> >>> Martin. >>> >>> Martin Paterson, Pattersong Music >>> Reduced orchestrations of G&S >>> >>> >>> On Tue, 30 Nov 2021 at 18:24, Brian West wrote: >>> >>>> You will want to set outbound_redirect_fatal=true, or the enterprise >>>> originate will follow the 302, you can safely ignore those if needed. >>>> >>>> On Tue, Nov 30, 2021 at 12:09 PM Martin Paterson < >>>> martin at pattersong.co.uk> wrote: >>>> >>>>> I have an issue that I’m struggling to resolve. I’ve tried this out in >>>>> the vanilla config. >>>>> >>>>> If a bridged destination in the default context: >>>>> >>>> data="user/1001@${domain_name}:_:user/1002@${domain_name}"/> >>>>> >>>>> but it has the effect that if a destination returns 302 Moved >>>>> Temporarily (1002 forwards to 1003 here), then it doesn’t run through >>>>> the dialplan the same context, it goes to the public context and fails >>>>> because the destination (1003) is an extension in the default context. >>>>> The log looks like it’s handling the 302 as if it’s a new call: >>>>> >>>>> [INFO] sofia.c:10462 sofia/internal/1000 at 192.168.1.226 receiving >>>>> invite from 192.168.1.226:5060 version: 1.10.7 -release-19-883d2cb662 >>>>> 64bit call-id: ee6cd84a-cca5-123a-d6a1-080027d0c8d3 >>>>> [INFO] mod_dialplan_xml.c:639 Processing Extension 1000 <1000>->1003 >>>>> in context public >>>>> EXECUTE [depth=0] sofia/internal/1000 at 192.168.1.226 deflect(1003) >>>>> >>>>> I must be missing something in my understanding here – I don’t >>>>> understand why the behaviour is different, but more interestingly, is >>>>> there a way of getting the enterprise dial to process the 302s in the >>>>> same way as when dialling one destination? >>>>> >>>>> Best wishes, >>>>> >>>>> Martin. >>>>> >>>>> Martin Paterson, Pattersong Music >>>>> Reduced orchestrations of G&S >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> >>>>> 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 at 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 at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> https://freeswitch.com >>>> >>>> >>>> >>>> -- >>>> >>>> Brian West | Co-founder and Developer >>>> >>>> Need Commercial support? email sales at freeswitch.com >>>> >>>> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 >>>> >>>> >>>> Email: brian at freeswitch.com >>>> >>>> Mobile: 918-424-9378 >>>> >>>> Website: https://www.FreeSWITCH.com >>>> >>>> [image: https://www.facebook.com/signalwireinc?src=email] >>>> [image: >>>> https://twitter.com/freeswitch] >>>> >>>> _________________________________________________________________________ >>>> >>>> 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 at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> https://freeswitch.com >>> >>> _________________________________________________________________________ >>> >>> 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 at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> >> >> -- >> >> Brian West | Co-founder and Developer >> >> Need Commercial support? email sales at freeswitch.com >> >> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 >> >> >> Email: brian at freeswitch.com >> >> Mobile: 918-424-9378 >> >> Website: https://www.FreeSWITCH.com >> >> [image: https://www.facebook.com/signalwireinc?src=email] >> [image: >> https://twitter.com/freeswitch] >> _________________________________________________________________________ >> >> 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Tue Dec 7 17:38:38 2021 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 7 Dec 2021 17:38:38 +0000 Subject: [Freeswitch-users] Sereverless ESL and KSUUID support In-Reply-To: References: Message-ID: You’ll still hit the 15 limit limitation. On Tue, 7 Dec 2021 at 14:00, Seven Du wrote: > I believe go can generate a .so could be called by C. Also there's a rust > lib. someone will eventually make a C version ;) > > On Tue, Dec 7, 2021 at 3:20 AM Brian West wrote: > >> 2) Does FreeSWITCH support KSUUID instead of UUID? >>> https://github.com/segmentio/ksuid >>> Is there any roadmap for that? >>> >> >> Unless there is a ksuuid library in C that is license compatible, it's >> not going to happen. >> >> /b >> >> >> -- >> >> Brian West | Co-founder and Developer >> >> Need Commercial support? email sales at freeswitch.com >> >> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 >> >> >> Email: brian at freeswitch.com >> >> Mobile: 918-424-9378 >> >> Website: https://www.FreeSWITCH.com >> >> [image: https://www.facebook.com/signalwireinc?src=email] >> [image: >> https://twitter.com/freeswitch] >> _________________________________________________________________________ >> >> 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > > -- > About: http://about.me/dujinfang > Blog: http://www.dujinfang.com > Proj: http://www.freeswitch.org.cn > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From lists at telium.io Tue Dec 7 22:25:19 2021 From: lists at telium.io (TTT) Date: Tue, 7 Dec 2021 22:25:19 +0000 Subject: [Freeswitch-users] Switch to MySQL db for config In-Reply-To: <202112061701.51558.Antony.Stone@freeswitch.open.source.it> References: <0100017d9042b57d-0d6a19bd-5c8b-4f70-8408-fdfe648873c3-000000@email.amazonses.com> <202112061701.51558.Antony.Stone@freeswitch.open.source.it> Message-ID: <0100017d97004e0e-a8beaafc-538c-4a4b-9520-277b8619bdf6-000000@email.amazonses.com> I was actually referring to FreeSWITCH core. I'm reading a book on FreeSwitch (https://www.amazon.ca/FreeSWITCH-1-8-Anthony-Minessale-II-ebook/dp/B071ZZBH8G/ref=sr_1_3?keywords=freeswitch&qid=1638915802&s=books&sr=1-3) and it mentions that FreeSWITCH keeps its operating information in a database at runtime. I think it's PostgreSQL I was wondering if/how I could switch that to MySQL. I thought there might be a DSN setting in the XML registry that affects this. This books says you can setup ODBC providers but it's not clear how, or if this applies to "core". -----Original Message----- From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Antony Stone Sent: Monday, December 6, 2021 11:02 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Switch to MySQL db for config On Monday 06 December 2021 at 16:00:30, TTT wrote: > I would like to use MySQL to hold my FreeSWITCH config. What do you mean by "config"? All of it, or just certain aspects (such as user directory, SIP registrations, dialplan...)? > I found instructions on how to setup the DSN, Point us at the URL so that we know what you're doing? > but how to I create the db/table structure? Is there a script to do this? What are you using as the "glue" between MySQL and FreeSwitch? Common options would be xml-curl or lua. In both cases you create your DB tables however you like, and you then write a lua script or create something that curl can interrogate, to return the XML which you would otherwise have placed into a flat file for FreeSwitch to parse. Regards, Antony. -- .evah I serutangis sseltniop tsom eht fo eno eb tsum sihT Please reply to the list; please *don't* CC me. _________________________________________________________________________ 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com From martin at pattersong.co.uk Wed Dec 8 09:05:15 2021 From: martin at pattersong.co.uk (Martin Paterson) Date: Wed, 8 Dec 2021 09:05:15 +0000 Subject: [Freeswitch-users] Switch to MySQL db for config In-Reply-To: <0100017d97004e0e-a8beaafc-538c-4a4b-9520-277b8619bdf6-000000@email.amazonses.com> References: <0100017d9042b57d-0d6a19bd-5c8b-4f70-8408-fdfe648873c3-000000@email.amazonses.com> <202112061701.51558.Antony.Stone@freeswitch.open.source.it> <0100017d97004e0e-a8beaafc-538c-4a4b-9520-277b8619bdf6-000000@email.amazonses.com> Message-ID: This page talks about it: https://freeswitch.org/confluence/display/FREESWITCH/ODBC+DSN. Basically there's a parameter switch-core-dsn in switch.conf.xml. In the vanilla config there's a bunch of comments in that file explaining the options. No need to worry about creating the schema - FreeSWITCH does that for you. Martin. Martin Paterson, Pattersong Music Reduced orchestrations of G&S On Tue, 7 Dec 2021 at 22:25, TTT wrote: > > I was actually referring to FreeSWITCH core. I'm reading a book on FreeSwitch (https://www.amazon.ca/FreeSWITCH-1-8-Anthony-Minessale-II-ebook/dp/B071ZZBH8G/ref=sr_1_3?keywords=freeswitch&qid=1638915802&s=books&sr=1-3) and it mentions that FreeSWITCH keeps its operating information in a database at runtime. I think it's PostgreSQL > > I was wondering if/how I could switch that to MySQL. I thought there might be a DSN setting in the XML registry that affects this. This books says you can setup ODBC providers but it's not clear how, or if this applies to "core". > > -----Original Message----- > From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Antony Stone > Sent: Monday, December 6, 2021 11:02 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Switch to MySQL db for config > > On Monday 06 December 2021 at 16:00:30, TTT wrote: > > > I would like to use MySQL to hold my FreeSWITCH config. > > What do you mean by "config"? > > All of it, or just certain aspects (such as user directory, SIP registrations, dialplan...)? > > > I found instructions on how to setup the DSN, > > Point us at the URL so that we know what you're doing? > > > but how to I create the db/table structure? Is there a script to do this? > > What are you using as the "glue" between MySQL and FreeSwitch? Common options would be xml-curl or lua. In both cases you create your DB tables however you like, and you then write a lua script or create something that curl can interrogate, to return the XML which you would otherwise have placed into a flat file for FreeSwitch to parse. > > > Regards, > > > Antony. > > -- > .evah I serutangis sseltniop tsom eht fo eno eb tsum sihT > > Please reply to the list; > please *don't* CC me. > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com From rahman.duran at erzurum.edu.tr Wed Dec 8 05:27:42 2021 From: rahman.duran at erzurum.edu.tr (Rahman Duran) Date: Wed, 8 Dec 2021 08:27:42 +0300 Subject: [Freeswitch-users] Switch to MySQL db for config In-Reply-To: <0100017d97004e0e-a8beaafc-538c-4a4b-9520-277b8619bdf6-000000@email.amazonses.com> References: <0100017d9042b57d-0d6a19bd-5c8b-4f70-8408-fdfe648873c3-000000@email.amazonses.com> <202112061701.51558.Antony.Stone@freeswitch.open.source.it> <0100017d97004e0e-a8beaafc-538c-4a4b-9520-277b8619bdf6-000000@email.amazonses.com> Message-ID: Hi, Adding " " to "switch.conf.xml" did the trick for us. You can add a similar line for mariadb/mysql. Rahman TTT , 8 Ara 2021 Çar, 07:12 tarihinde şunu yazdı: > I was actually referring to FreeSWITCH core. I'm reading a book on > FreeSwitch ( > https://www.amazon.ca/FreeSWITCH-1-8-Anthony-Minessale-II-ebook/dp/B071ZZBH8G/ref=sr_1_3?keywords=freeswitch&qid=1638915802&s=books&sr=1-3) > and it mentions that FreeSWITCH keeps its operating information in a > database at runtime. I think it's PostgreSQL > > I was wondering if/how I could switch that to MySQL. I thought there > might be a DSN setting in the XML registry that affects this. This books > says you can setup ODBC providers but it's not clear how, or if this > applies to "core". > > -----Original Message----- > From: FreeSWITCH-users [mailto: > freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Antony Stone > Sent: Monday, December 6, 2021 11:02 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Switch to MySQL db for config > > On Monday 06 December 2021 at 16:00:30, TTT wrote: > > > I would like to use MySQL to hold my FreeSWITCH config. > > What do you mean by "config"? > > All of it, or just certain aspects (such as user directory, SIP > registrations, dialplan...)? > > > I found instructions on how to setup the DSN, > > Point us at the URL so that we know what you're doing? > > > but how to I create the db/table structure? Is there a script to do > this? > > What are you using as the "glue" between MySQL and FreeSwitch? Common > options would be xml-curl or lua. In both cases you create your DB tables > however you like, and you then write a lua script or create something that > curl can interrogate, to return the XML which you would otherwise have > placed into a flat file for FreeSwitch to parse. > > > Regards, > > > Antony. > > -- > .evah I serutangis sseltniop tsom eht fo eno eb tsum sihT > > Please reply to the > list; > please *don't* CC > me. > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From rahman.duran at erzurum.edu.tr Wed Dec 8 09:43:28 2021 From: rahman.duran at erzurum.edu.tr (Rahman Duran) Date: Wed, 8 Dec 2021 12:43:28 +0300 Subject: [Freeswitch-users] Sending Notify event to Phones inside fs_cli? Message-ID: Hi, Can I use sendevent in fs_cli to send Notify packets to Phones (to triger restart, provision etc). I can see how to do it in event socket documentation but I could not find any info about doing it in fs_cli. fs_cli says there is no command named sendevent. Regards, Rahman Duran -------------- next part -------------- An HTML attachment was scrubbed... URL: From franck.james at sewan.fr Wed Dec 8 18:19:20 2021 From: franck.james at sewan.fr (Franck James) Date: Wed, 8 Dec 2021 18:19:20 +0000 Subject: [Freeswitch-users] Max auth-acl set to 32 entries Message-ID: Hi, We're getting an issue on max allowed Ips in param name="auth-acl" ; it seems that actual max values is 32 entries and we are reaching this limit in some cases. After checking source code, we suspect the issue to come from : char *argv[32]; in switch_check_network_list_ip_port_token --- src/switch_core.c Is there any way to increase this max value ? Thanks in advance. Franck -------------- next part -------------- An HTML attachment was scrubbed... URL: From Antony.Stone at freeswitch.open.source.it Mon Dec 6 12:47:40 2021 From: Antony.Stone at freeswitch.open.source.it (Antony Stone) Date: Mon, 6 Dec 2021 13:47:40 +0100 Subject: [Freeswitch-users] Mailing list is very slow? In-Reply-To: References: <202112031640.15416.Antony.Stone@freeswitch.open.source.it> Message-ID: <202112061347.40277.Antony.Stone@freeswitch.open.source.it> On Monday 06 December 2021 at 13:16:46, Giovanni Maruzzelli wrote: > On Mon, Dec 6, 2021 at 1:12 PM Antony Stone wrote: > > Hi. > > > > I wonder if there any list admins here who can either explain why this > > list is so slow, or preferably do something about it? > > IIRC posts are manually moderated to avoid spam Hm, that seems strange for registered subscribers. I'm familiar with mailing lists moderating first posts by new members, but once someone has "joined the community", their posts are usually unmoderated to improve the response time of replies to questions, and maintain discussions. Perhaps a list admin could review this policy and see how often established members do send "spam" messages which do not get approved, and thereby consider reducing these delays? Antony. -- A few words to be cautious of between American and English: - momentarily - suspenders - chips - pants - jelly - pavement - vest - pint (and gallon) - pissed Please reply to the list; please *don't* CC me. From ericsgeek at gmail.com Wed Dec 8 20:27:12 2021 From: ericsgeek at gmail.com (Eric Schwertfeger) Date: Wed, 8 Dec 2021 12:27:12 -0800 Subject: [Freeswitch-users] Switch to MySQL db for config In-Reply-To: <0100017d97004e0e-a8beaafc-538c-4a4b-9520-277b8619bdf6-000000@email.amazonses.com> References: <0100017d9042b57d-0d6a19bd-5c8b-4f70-8408-fdfe648873c3-000000@email.amazonses.com> <202112061701.51558.Antony.Stone@freeswitch.open.source.it> <0100017d97004e0e-a8beaafc-538c-4a4b-9520-277b8619bdf6-000000@email.amazonses.com> Message-ID: On 12/7/2021 2:25 PM, TTT wrote: > I was actually referring to FreeSWITCH core. I'm reading a book on FreeSwitch (https://www.amazon.ca/FreeSWITCH-1-8-Anthony-Minessale-II-ebook/dp/B071ZZBH8G/ref=sr_1_3?keywords=freeswitch&qid=1638915802&s=books&sr=1-3) and it mentions that FreeSWITCH keeps its operating information in a database at runtime. I think it's PostgreSQL > > I was wondering if/how I could switch that to MySQL. I thought there might be a DSN setting in the XML registry that affects this. This books says you can setup ODBC providers but it's not clear how, or if this applies to "core". I believe you can.  The module in question is mod_mariadb, but that's because it uses the mariadb shared library rather than the mysql shared library, not because it only works with mariadb. Those two use compatible over-the-wire protocols, so using it to talk to mysql shouldn't be an issue. I haven't tested this with mysql itself, however.  At any rate, these are the steps it took for me using MariaDB.  Much simpler than ODBC in my opinion. You will want to make sure that mod_mariadb is installed and preloaded.  I'm not certain that preloaded is necessary, but it was specified in the writeup I found.  So this is the file autoload_configs/pre_load_modules.xml from my server.                 Then, you need to set the core-db-dsn param in switch.conf.xml as mentioned above.  In the case of a mariadb dsn, it would look like this. If you're using TCP instead of a unix socket, replace the "Socket=[socket name]" with "Host=[host name]" and a setting for Port if you're not using the default port.  Also, change the Database and uid parameter to suit your setup. Oh, and as an FYI, the default storage provider that FreeSWITCH uses is SQLLite, not PGSQL. From dragos at freeswitch.org Thu Dec 9 09:40:29 2021 From: dragos at freeswitch.org (Dragos Oancea) Date: Thu, 9 Dec 2021 11:40:29 +0200 Subject: [Freeswitch-users] Max auth-acl set to 32 entries In-Reply-To: References: Message-ID: try https://github.com/signalwire/freeswitch/pull/1480 . On Wed, Dec 8, 2021 at 9:34 PM Franck James wrote: > Hi, > > > > We’re getting an issue on max allowed Ips in param name="auth-acl" ; it > seems that actual max values is 32 entries and we are reaching this limit > in some cases. > > > > After checking source code, we suspect the issue to come from : > > > > char *argv[32]; in switch_check_network_list_ip_port_token --- > src/switch_core.c > > > > Is there any way to increase this max value ? > > > > Thanks in advance. > > > > Franck > > > > > > > > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From mayamatakeshi at gmail.com Fri Dec 10 07:15:58 2021 From: mayamatakeshi at gmail.com (mayamatakeshi) Date: Fri, 10 Dec 2021 16:15:58 +0900 Subject: [Freeswitch-users] Using mod_com_amd to process non-real-time audio Message-ID: Hi, I did some preliminary tests with mod_com_amd (Answering Machine Detection app voice_start) and got an accuracy of about 66% (well it was just for 27 calls so far but more will come). So I think I will need to adjust the parameters. So I am planning to either use brute force or a genetic algorithm to find adequate configuration values. I can automate a system to make such calls, play tagged call recording files over them and get the AMD result from freeswitch then check the fitness of the configuration. No problems here. However, setup and processing of calls take time and I was thinking if there would be some way to use mod_com_amd so that I could process hundreds of call recording files in a few seconds. It is a long shot but maybe someone has some hint. -------------- next part -------------- An HTML attachment was scrubbed... URL: From trle.fuad at gmail.com Fri Dec 10 08:50:06 2021 From: trle.fuad at gmail.com (Fuad Trle) Date: Fri, 10 Dec 2021 09:50:06 +0100 Subject: [Freeswitch-users] Problems with Hold, Dynamic Payload, Media Direction Message-ID: Hello, I have encountered several problems while configuring and testing FreeSWITCH. This is on a current master branch, but the same issues are present in older versions. The tests are done on a minimal config with these additions: vars: - remove auth - global_codec_prefs=OPUS,PCMU,PCMA,VP8 - rtp_pass_codecs_on_stream_change=true # To be able to toggle video stream profiles: - inbound-late-negotiation (with and w/o inherit codec) dialplan: - Simple dial plan. External to Internal flow. Condition for ext, then bridge call to the other side. Unfortunately, this Sofia options are removed from FS :( Are there alternatives for them (while using rtp_pass_codecs_on_stream_change)? renegotiate-codec-on-hold|true,false renegotiate-codec-on-reinvite|true,false (a) -- (FS) -- (b) *1st: Hold* - Break transcoding. -- INVITE (Opus, PCMA) --> | | -- INVITE (PCMA) --> | <---- OK (PCMA) ---- <------ OK (Opus) -------- | . . | <--- ReINVITE (Hold or video on/off) <---- ReINVITE (PCMA) ---- | x I tried with absolute codec string, inbound/outbound codec prefs... it seems that FS does not keep/honor previous channel state and does not differentiate stream change from stream parameters change (like media flow). *2nd: Dynamic Payload *- It will offer Opus to the other side, because both have it. But FS will renegotiate with new PT number on reinvite. This break stream for some endpoints. -- INVITE (Opus, 111) --> | | -- INVITE (Opus, 102) --> | <---- OK (Opus, 102) ---- <---- OK (Opus, 111) ---- | . . | <-- ReINVITE (Hold or new m=line) <- ReINVITE (Opus, 102) - | x Is there a way to originate call with a codec that has a dynamic payload number, but to choose PT number? Something like setting rfc2833-pt variable for outbound channel before originating call? *3rd Media direction* - FS does not propagate media direction parameter to inbound channel/leg. Inbound side expect media and some endpoints will drop call after awhile. I know that answering with recvonly does not have sense for audio, but for a video it has. -- INVITE (sendrecv) --> | | -- INVITE (sendrecv) --> | <---- OK (recvonly) ---- <---- OK (sendrecv) ---- | For late negotiation this is wrong. Even for early it should be possible to pass reinvite to the other side on media direction change. -------------- next part -------------- An HTML attachment was scrubbed... URL: From cedric at saooti.com Fri Dec 10 09:08:37 2021 From: cedric at saooti.com (=?utf-8?Q?C=C3=A9dric_Clavier?=) Date: Fri, 10 Dec 2021 10:08:37 +0100 Subject: [Freeswitch-users] 503 dns resolution failed Message-ID: <4976CE57-7F2A-4063-9A90-03EADF846DF3@saooti.com> Hello everyone, I recently migrate to Kubernetes and to the last release of freeswitch (1.10.7). Freeswitch is running in a docker container based on Debian bullseye. I’m having troubles with registering my Sip gateway. Most of the time everything is ok but sometimes, when register is in progress I have logs indicating that there is a dns problem 2021-12-09 13:13:25.635592 73.97% [ERR] sofia_reg.c:2677 studio Failed Registration with status DNS Error [503]. failure #75 2021-12-09 13:13:26.635697 73.93% [WARNING] sofia_reg.c:516 studio Failed Registration [503], setting retry to 5 seconds. When I scan dns call with tcpdump it seems stuck using dns request for srv and naptr server (none of them exists in my gateway) and does not require the A I tried to remove this from my external profile But it doesn’t change anything. Does anybody already have the same trouble or understand what append. Thank in advance Cédric Clavier -------------- next part -------------- An HTML attachment was scrubbed... URL: From aronp at guaranteedplus.com Fri Dec 10 05:54:23 2021 From: aronp at guaranteedplus.com (Podrigal, Aron) Date: Thu, 9 Dec 2021 23:54:23 -0600 Subject: [Freeswitch-users] Weird: Cloudflare 1.1.1.1 DNS server not working with sofia_dig Message-ID: Hi, I am experiencing weird behavior. I have my nameserver set to 1.1.1.1, doing sofia_dig returns no results, one request out of five succeeds. - When I do a dig / host lookup directly outside freeswitch using 1.1.1.1, it works. - sofia_dig with any other DNS server (8.8.8.8, 9.9.9.9 etc) works. PCAP shows DNS traffic being sent and received normally. In the non working scenario, the response from 1.1.1.1 simply has zero ips. See below. [image: image.png] -- - Aron Podrigal -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image.png Type: image/png Size: 753382 bytes Desc: not available URL: From david.villasmil.work at gmail.com Fri Dec 10 14:47:20 2021 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 10 Dec 2021 14:47:20 +0000 Subject: [Freeswitch-users] 503 dns resolution failed In-Reply-To: <4976CE57-7F2A-4063-9A90-03EADF846DF3@saooti.com> References: <4976CE57-7F2A-4063-9A90-03EADF846DF3@saooti.com> Message-ID: is it possible the other is indeed returning 503? do you have a trace? Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Fri, Dec 10, 2021 at 1:46 PM Cédric Clavier wrote: > Hello everyone, > > I recently migrate to Kubernetes and to the last release of freeswitch > (1.10.7). Freeswitch is running in a docker container based on Debian > bullseye. > I’m having troubles with registering my Sip gateway. > Most of the time everything is ok but sometimes, when register is in > progress I have logs indicating that there is a dns problem > > 2021-12-09 13:13:25.635592 73.97% [ERR] sofia_reg.c:2677 studio Failed > Registration with status DNS Error [503]. failure #75 > 2021-12-09 13:13:26.635697 73.93% [WARNING] sofia_reg.c:516 studio Failed > Registration [503], setting retry to 5 seconds. > > When I scan dns call with tcpdump it seems stuck using dns request for srv > and naptr server (none of them exists in my gateway) and does not require > the A > > I tried to remove this from my external profile > > > > > But it doesn’t change anything. > > Does anybody already have the same trouble or understand what append. > > Thank in advance > > Cédric Clavier > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From brians at iptel.co Fri Dec 10 16:56:43 2021 From: brians at iptel.co (Brian :) Date: Fri, 10 Dec 2021 16:56:43 +0000 Subject: [Freeswitch-users] Weird: Cloudflare 1.1.1.1 DNS server not working with sofia_dig In-Reply-To: References: Message-ID: We noticed something similar with 1.1.1.1 and srv records a while ago. Didn't spend much time on it just told customer to use a resolver that works. On Friday, December 10, 2021, Podrigal, Aron wrote: > Hi, > I am experiencing weird behavior. > I have my nameserver set to 1.1.1.1, doing sofia_dig returns no results, one request out of five succeeds. > - When I do a dig / host lookup directly outside freeswitch using 1.1.1.1, it works. > - sofia_dig with any other DNS server (8.8.8.8, 9.9.9.9 etc) works. > PCAP shows DNS traffic being sent and received normally. In the non working scenario, the response from 1.1.1.1 simply has zero ips. > See below. > > > > -- > > - > Aron Podrigal > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Sat Dec 11 13:12:14 2021 From: s.safarov at gmail.com (Sergey Safarov) Date: Sat, 11 Dec 2021 16:12:14 +0300 Subject: [Freeswitch-users] Weird: Cloudflare 1.1.1.1 DNS server not working with sofia_dig In-Reply-To: References: Message-ID: You can check DNS response. I think there may be a set flag that indicates "partial response." In this case, the client should switch transport used to send requests from UDP to TCP. As I know Freewithch (libsofia) do not support DNS request via TCP and this lead to failed DNS lookup. On Fri, Dec 10, 2021 at 8:10 PM Brian : wrote: > We noticed something similar with 1.1.1.1 and srv records a while ago. > Didn't spend much time on it just told customer to use a resolver that > works. > > On Friday, December 10, 2021, Podrigal, Aron > wrote: > > Hi, > > I am experiencing weird behavior. > > I have my nameserver set to 1.1.1.1, doing sofia_dig returns no results, > one request out of five succeeds. > > - When I do a dig / host lookup directly outside freeswitch using > 1.1.1.1, it works. > > - sofia_dig with any other DNS server (8.8.8.8, 9.9.9.9 etc) works. > > PCAP shows DNS traffic being sent and received normally. In the non > working scenario, the response from 1.1.1.1 simply has zero ips. > > See below. > > > > > > > > > -- > > > > - > > Aron Podrigal > > > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From Antony.Stone at freeswitch.open.source.it Sat Dec 11 14:19:56 2021 From: Antony.Stone at freeswitch.open.source.it (Antony Stone) Date: Sat, 11 Dec 2021 15:19:56 +0100 Subject: [Freeswitch-users] Weird: Cloudflare 1.1.1.1 DNS server not working with sofia_dig In-Reply-To: References: Message-ID: <202112111519.56777.Antony.Stone@freeswitch.open.source.it> On Saturday 11 December 2021 at 14:12:14, Sergey Safarov wrote: > You can check DNS response. > I think there may be a set flag that indicates "partial response." > In this case, the client should switch transport used to send requests from > UDP to TCP. > > As I know Freewithch (libsofia) do not support DNS request via TCP and this > lead to failed DNS lookup. That's not what the source code seems to indicate. src/mod/endpoints/mod_sofia/sip-dig.c contains: -------- The sip-dig utility accepts following command line options: -p protoname Use named transport protocol. The protoname can be either well-known, e.g., "udp", or it can specify NAPTR service and SRV identifier, e.g., "tls-udp/SIPS+D2U/_sips._udp.". --udp Use UDP transport protocol. --tcp Use TCP transport protocol. --tls Use TLS over TCP transport protocol. --sctp Use SCTP transport protocol. --tls-sctp Use TLS over SCTP transport protocol. --no-sctp Ignore SCTP or TLS-SCTP records in the list of default transports. This option has no effect if transport protocols has been explicitly listed. -4 Query IP4 addresses (A records) -6 Query IP6 addresses (AAAA records). -v Be verbatim. -------- I do wonder whether that last comment should be "Be verbose", but I can't say for certain :) However, this strongly suggests that sofia_dig / sip_dig does support TCP lookups. I have no idea whether that means it will automatically fail over to a TCP lookup if the UDP response is "too large". Antony. -- Normal people think "If it ain't broke, don't fix it". Engineers think "If it ain't broke, it doesn't have enough features yet". Please reply to the list; please *don't* CC me. From Antony.Stone at freeswitch.open.source.it Sat Dec 11 14:25:51 2021 From: Antony.Stone at freeswitch.open.source.it (Antony Stone) Date: Sat, 11 Dec 2021 15:25:51 +0100 Subject: [Freeswitch-users] Weird: Cloudflare 1.1.1.1 DNS server not working with sofia_dig In-Reply-To: References: Message-ID: <202112111525.51580.Antony.Stone@freeswitch.open.source.it> On Friday 10 December 2021 at 06:54:23, Podrigal, Aron wrote: > Hi, > > I am experiencing weird behavior. > > I have my nameserver set to 1.1.1.1, doing sofia_dig returns no results, > one request out of five succeeds. I cannot reproduce that (FreeSwitch 1.10.7 running under Debian 10 / Devuan 3): freeswitch> sofia_dig -v @1.1.1.1 google.com Preference Weight Transport Port Address ================================================================================ @1.1.1.1 1 1.000 udp 5060 1.1.1.1 2 1.000 tcp 5060 1.1.1.1 google.com 1 0.250 udp 5060 2a00:1450:4009:81e::200e 1 0.250 tcp 5060 2a00:1450:4009:81e::200e 1 0.250 udp 5060 142.250.187.206 1 0.250 tcp 5060 142.250.187.206 I get a result every single time. Antony. -- yes, but this is #lbw, we don't do normal Please reply to the list; please *don't* CC me. From avi at avimarcus.net Sun Dec 12 18:33:07 2021 From: avi at avimarcus.net (Avi Marcus) Date: Sun, 12 Dec 2021 18:33:07 +0000 Subject: [Freeswitch-users] Weird: Cloudflare 1.1.1.1 DNS server not working with sofia_dig In-Reply-To: <202112111525.51580.Antony.Stone@freeswitch.open.source.it> References: <202112111525.51580.Antony.Stone@freeswitch.open.source.it> Message-ID: <0100017dafeb82eb-e5a2186a-e0fe-408d-8b59-39f76e9211cf-000000@email.amazonses.com> Also FreeSWITCH Version 1.10.7-release-19-883d2cb662~64bit (-release-19-883d2cb662 64bit) On debian 9.13 `sofia_dig -v @1.1.1.1 google.com` works, with and without the -v flag. -Avi Marcus On Sun, Dec 12, 2021 at 8:22 PM Antony Stone < Antony.Stone at freeswitch.open.source.it> wrote: > On Friday 10 December 2021 at 06:54:23, Podrigal, Aron wrote: > > > Hi, > > > > I am experiencing weird behavior. > > > > I have my nameserver set to 1.1.1.1, doing sofia_dig returns no results, > > one request out of five succeeds. > > I cannot reproduce that (FreeSwitch 1.10.7 running under Debian 10 / > Devuan > 3): > > freeswitch> sofia_dig -v @1.1.1.1 google.com > Preference Weight Transport Port Address > > ================================================================================ > @1.1.1.1 1 1.000 udp 5060 1.1.1.1 > 2 1.000 tcp 5060 1.1.1.1 > google.com 1 0.250 udp 5060 > 2a00:1450:4009:81e::200e > 1 0.250 tcp 5060 > 2a00:1450:4009:81e::200e > 1 0.250 udp 5060 142.250.187.206 > 1 0.250 tcp 5060 142.250.187.206 > > I get a result every single time. > > > Antony. > > -- > yes, but this is #lbw, we don't do normal > > Please reply to the > list; > please *don't* CC > me. > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Sun Dec 12 19:37:41 2021 From: s.safarov at gmail.com (Sergey Safarov) Date: Sun, 12 Dec 2021 22:37:41 +0300 Subject: [Freeswitch-users] Weird: Cloudflare 1.1.1.1 DNS server not working with sofia_dig In-Reply-To: <0100017dafeb82eb-e5a2186a-e0fe-408d-8b59-39f76e9211cf-000000@email.amazonses.com> References: <202112111525.51580.Antony.Stone@freeswitch.open.source.it> <0100017dafeb82eb-e5a2186a-e0fe-408d-8b59-39f76e9211cf-000000@email.amazonses.com> Message-ID: To reproduce the issue you can create a NAPTR record with lot of data and then try to make a call to a domain that resolved via NAPTR/SRV In my case Example [safarov at safarov-dell ~]$ nslookup -type=NAPTR ok.nga911.com ;; Truncated, retrying in TCP mode. Server: 127.0.0.53 Address: 127.0.0.53#53 Non-authoritative answer: ok.nga911.com naptr = 100 10 "U" "LoST:findServiceByCivicAddress" "!.*! https://api.ok.nga911.com/api/v1/lost/find-service-by-civic-address!" . ok.nga911.com naptr = 100 10 "U" "LoST:listServicesByLocation" "!.*! https://api.ok.nga911.com/api/v1/lost/list-services-by-location!" . ok.nga911.com naptr = 100 10 "U" "LoST:listServices" "!.*! https://api.ok.nga911.com/api/v1/lost/list-services!" . ok.nga911.com naptr = 100 10 "U" "LIS:HELD" "!.*! https://psap.ok.nga911.com/LIS/!" . ok.nga911.com naptr = 50 500 "S" "SIPS+D2W" "" _sips._ws.ok.nga911.com. ok.nga911.com naptr = 30 300 "S" "SIP+D2U" "" _sip._udp.ok.nga911.com. ok.nga911.com naptr = 20 200 "S" "SIP+D2T" "" _sip._tcp.ok.nga911.com. ok.nga911.com naptr = 100 10 "U" "LoST:findServiceByLocation" "!.*! https://api.ok.nga911.com/api/v1/lost/find-service-by-location!" . ok.nga911.com naptr = 40 400 "S" "SIPS+D2T" "" _sip._tls.ok.nga911.com. Authoritative answers can be found from: NAPTR response UDP will not contain the whole responce. you can see ;; Truncated, retrying in TCP mode. If response via UDP do not contains ok.nga911.com naptr = 30 300 "S" "SIP+D2U" "" _sip._udp.ok.nga911.com. Then FS will fail place a call. On Sun, Dec 12, 2021 at 10:15 PM Avi Marcus wrote: > Also FreeSWITCH Version 1.10.7-release-19-883d2cb662~64bit > (-release-19-883d2cb662 64bit) > On debian 9.13 > > `sofia_dig -v @1.1.1.1 google.com` works, with and without the -v flag. > > > -Avi Marcus > > > > On Sun, Dec 12, 2021 at 8:22 PM Antony Stone < > Antony.Stone at freeswitch.open.source.it> wrote: > >> On Friday 10 December 2021 at 06:54:23, Podrigal, Aron wrote: >> >> > Hi, >> > >> > I am experiencing weird behavior. >> > >> > I have my nameserver set to 1.1.1.1, doing sofia_dig returns no results, >> > one request out of five succeeds. >> >> I cannot reproduce that (FreeSwitch 1.10.7 running under Debian 10 / >> Devuan >> 3): >> >> freeswitch> sofia_dig -v @1.1.1.1 google.com >> Preference Weight Transport Port Address >> >> ================================================================================ >> @1.1.1.1 1 1.000 udp 5060 1.1.1.1 >> 2 1.000 tcp 5060 1.1.1.1 >> google.com 1 0.250 udp 5060 >> 2a00:1450:4009:81e::200e >> 1 0.250 tcp 5060 >> 2a00:1450:4009:81e::200e >> 1 0.250 udp 5060 142.250.187.206 >> 1 0.250 tcp 5060 142.250.187.206 >> >> I get a result every single time. >> >> >> Antony. >> >> -- >> yes, but this is #lbw, we don't do normal >> >> Please reply to the >> list; >> please *don't* >> CC me. >> >> _________________________________________________________________________ >> >> 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From shaun at sysconfig.cloud Mon Dec 13 08:20:59 2021 From: shaun at sysconfig.cloud (Shaun Stokes) Date: Mon, 13 Dec 2021 08:20:59 +0000 Subject: [Freeswitch-users] Weird: Cloudflare 1.1.1.1 DNS server not working with sofia_dig In-Reply-To: References: <202112111525.51580.Antony.Stone@freeswitch.open.source.it> <0100017dafeb82eb-e5a2186a-e0fe-408d-8b59-39f76e9211cf-000000@email.amazonses.com> Message-ID: DNS works over UDP by default, that is unless the UDP response contains the Truncated flag which tells the client to use TCP instead. More information here: https://ns1.com/blog/when-dns-uses-udp-versus-tcp I've tested this myself, FreeSWITCH ignores the Truncated flag and continues to use UDP anyway. It seems that FreeSWITCH is not RFC 2671 compliant, this would need to be developed. There is a related open issue: https://github.com/signalwire/freeswitch/issues/907 ________________________________ From: FreeSWITCH-users on behalf of Sergey Safarov Sent: 12 December 2021 20:37 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Weird: Cloudflare 1.1.1.1 DNS server not working with sofia_dig To reproduce the issue you can create a NAPTR record with lot of data and then try to make a call to a domain that resolved via NAPTR/SRV In my case Example [safarov at safarov-dell ~]$ nslookup -type=NAPTR ok.nga911.com ;; Truncated, retrying in TCP mode. Server: 127.0.0.53 Address: 127.0.0.53#53 Non-authoritative answer: ok.nga911.com naptr = 100 10 "U" "LoST:findServiceByCivicAddress" "!.*!https://api.ok.nga911.com/api/v1/lost/find-service-by-civic-address!" . ok.nga911.com naptr = 100 10 "U" "LoST:listServicesByLocation" "!.*!https://api.ok.nga911.com/api/v1/lost/list-services-by-location!" . ok.nga911.com naptr = 100 10 "U" "LoST:listServices" "!.*!https://api.ok.nga911.com/api/v1/lost/list-services!" . ok.nga911.com naptr = 100 10 "U" "LIS:HELD" "!.*!https://psap.ok.nga911.com/LIS/!" . ok.nga911.com naptr = 50 500 "S" "SIPS+D2W" "" _sips._ws.ok.nga911.com. ok.nga911.com naptr = 30 300 "S" "SIP+D2U" "" _sip._udp.ok.nga911.com. ok.nga911.com naptr = 20 200 "S" "SIP+D2T" "" _sip._tcp.ok.nga911.com. ok.nga911.com naptr = 100 10 "U" "LoST:findServiceByLocation" "!.*!https://api.ok.nga911.com/api/v1/lost/find-service-by-location!" . ok.nga911.com naptr = 40 400 "S" "SIPS+D2T" "" _sip._tls.ok.nga911.com. Authoritative answers can be found from: NAPTR response UDP will not contain the whole responce. you can see ;; Truncated, retrying in TCP mode. If response via UDP do not contains ok.nga911.com naptr = 30 300 "S" "SIP+D2U" "" _sip._udp.ok.nga911.com. Then FS will fail place a call. On Sun, Dec 12, 2021 at 10:15 PM Avi Marcus > wrote: Also FreeSWITCH Version 1.10.7-release-19-883d2cb662~64bit (-release-19-883d2cb662 64bit) On debian 9.13 `sofia_dig -v @1.1.1.1 google.com` works, with and without the -v flag. -Avi Marcus On Sun, Dec 12, 2021 at 8:22 PM Antony Stone > wrote: On Friday 10 December 2021 at 06:54:23, Podrigal, Aron wrote: > Hi, > > I am experiencing weird behavior. > > I have my nameserver set to 1.1.1.1, doing sofia_dig returns no results, > one request out of five succeeds. I cannot reproduce that (FreeSwitch 1.10.7 running under Debian 10 / Devuan 3): freeswitch> sofia_dig -v @1.1.1.1 google.com Preference Weight Transport Port Address ================================================================================ @1.1.1.1 1 1.000 udp 5060 1.1.1.1 2 1.000 tcp 5060 1.1.1.1 google.com 1 0.250 udp 5060 2a00:1450:4009:81e::200e 1 0.250 tcp 5060 2a00:1450:4009:81e::200e 1 0.250 udp 5060 142.250.187.206 1 0.250 tcp 5060 142.250.187.206 I get a result every single time. Antony. -- yes, but this is #lbw, we don't do normal Please reply to the list; please *don't* CC me. _________________________________________________________________________ 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com _________________________________________________________________________ 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From sdevoy at bizfocused.com Mon Dec 13 17:38:17 2021 From: sdevoy at bizfocused.com (Sean Devoy) Date: Mon, 13 Dec 2021 17:38:17 +0000 Subject: params vs variable Message-ID: Happy Holidays everyone, I have a simple question about xml directory entries and the use of user_data. Is there any criteria for when I should use a param versus when to use a variable entry? I see I can use either. Regards, Sean Devoy VP Operations and Development Business Focused Internet Systems, Inc. -------------- next part -------------- An HTML attachment was scrubbed... URL: From brians at iptel.co Mon Dec 13 18:33:28 2021 From: brians at iptel.co (Brian :) Date: Mon, 13 Dec 2021 18:33:28 +0000 Subject: [Freeswitch-users] Weird: Cloudflare 1.1.1.1 DNS server not working with sofia_dig In-Reply-To: References: Message-ID: In our case this definitely wasn't the issue. 1.1.1.1 just didn't respond to some queries for SRV records at all or sent back an error response. It was a geo thing too. In emea it worked fine. In Africas it didn't. Like I said we had more things to be worrying about so we didn't spend a lot of time on it. On Saturday, December 11, 2021, Sergey Safarov wrote: > You can check DNS response. > I think there may be a set flag that indicates "partial response." > In this case, the client should switch transport used to send requests from UDP to TCP. > > As I know Freewithch (libsofia) do not support DNS request via TCP and this lead to failed DNS lookup. > > On Fri, Dec 10, 2021 at 8:10 PM Brian : wrote: >> >> We noticed something similar with 1.1.1.1 and srv records a while ago. Didn't spend much time on it just told customer to use a resolver that works. >> >> On Friday, December 10, 2021, Podrigal, Aron wrote: >> > Hi, >> > I am experiencing weird behavior. >> > I have my nameserver set to 1.1.1.1, doing sofia_dig returns no results, one request out of five succeeds. >> > - When I do a dig / host lookup directly outside freeswitch using 1.1.1.1, it works. >> > - sofia_dig with any other DNS server (8.8.8.8, 9.9.9.9 etc) works. >> > PCAP shows DNS traffic being sent and received normally. In the non working scenario, the response from 1.1.1.1 simply has zero ips. >> > See below. >> > >> > >> > >> > -- >> > >> > - >> > Aron Podrigal >> > >> > _________________________________________________________________________ >> >> 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From franck.james at sewan.fr Mon Dec 13 20:43:35 2021 From: franck.james at sewan.fr (Franck James) Date: Mon, 13 Dec 2021 20:43:35 +0000 Subject: [Freeswitch-users] Max auth-acl set to 32 entries In-Reply-To: References: Message-ID: Hi, Indeed, it’s working in our case. Thanks for your help ! De : FreeSWITCH-users De la part de Dragos Oancea Envoyé : jeudi 9 décembre 2021 10:40 À : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] Max auth-acl set to 32 entries try https://github.com/signalwire/freeswitch/pull/1480 . On Wed, Dec 8, 2021 at 9:34 PM Franck James > wrote: Hi, We’re getting an issue on max allowed Ips in param name="auth-acl" ; it seems that actual max values is 32 entries and we are reaching this limit in some cases. After checking source code, we suspect the issue to come from : char *argv[32]; in switch_check_network_list_ip_port_token --- src/switch_core.c Is there any way to increase this max value ? Thanks in advance. Franck _________________________________________________________________________ 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From igor.potjevlesch at gmail.com Tue Dec 14 15:03:53 2021 From: igor.potjevlesch at gmail.com (Igor Potjevlesh) Date: Tue, 14 Dec 2021 16:03:53 +0100 Subject: [Freeswitch-users] Remap cause code Message-ID: <001c01d7f0fb$d0869f20$7193dd60$@gmail.com> Hello there! I'd like to remap a specific cause code 501 receives on leg B to 503 on leg A. I see some documentation with proto_specific_hangup_cause into a specific extension but it's not clear how to write this for a specific map and keep the other as passthrough. Do you have any example? Thank you! Regards, Igor. -- L'absence de virus dans ce courrier électronique a été vérifiée par le logiciel antivirus Avast. https://www.avast.com/antivirus -------------- next part -------------- An HTML attachment was scrubbed... URL: From mrjoli021 at gmail.com Wed Dec 15 14:52:59 2021 From: mrjoli021 at gmail.com (Joli Martinez) Date: Wed, 15 Dec 2021 09:52:59 -0500 Subject: [Freeswitch-users] weird extra characters Message-ID: Hello, I am running freeswitch 1.10.3. I have several DID's on this system. Some work without any issues, but some somehow get some wierd characters added to the SIP message. I have already verified that there are no extra characters on the number. It appears to be a certain combination that it is happening to. If I change the last number from 1024 to 1023, nothing get added but on 1024 it does. Any suggestions on why this is happening and/or how to fix/patch this? INVITE sip:933 at 13.13.13.13 SIP/2.0 Via: SIP/2.0/UDP 12.12.12.12:5080;rport;branch=z9hG4bK3FX45c6Sj2Ftc Max-Forwards: 69 From: "122233331024" ;tag=7KmB96jZD2e8r To: Call-ID: 6d1bafb7-d855-123a-8dbc-129e3056e999 CSeq: 45235314 INVITE Contact: User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 297 X-accountcode: 12.12.12.12 X-FS-Support: update_display,send_info Remote-Party-ID: "122233331024" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1639556579 1639556580 IN IP4 12.12.12.12 s=FreeSWITCH c=IN IP4 12.12.12.12 t=0 0 m=audio 21634 RTP/AVP 9 0 8 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 -------------- next part -------------- An HTML attachment was scrubbed... URL: From sdevoy at bizfocused.com Wed Dec 15 16:14:57 2021 From: sdevoy at bizfocused.com (Sean Devoy) Date: Wed, 15 Dec 2021 16:14:57 +0000 Subject: Failed outbound call Message-ID: Hi, I have a very strange problem. My customer has one extension out of 16 that has an issue dialing outside numbers. She hears it ring 4 or 5 times, then gets a "busy" signal. She says the recipient's phone does not ring on their end. (I had another user test this and it is confirmed). The outbound dialing code (XML) is common to all extensions at that location. No one else has this issue. The phone is a CISCO SPA504g. I have compared every setup value with other SPA504Gs at that site. I have compared every value in the directory for this extension to others. I have created a pastebin of the FreeSwitch log file for this call: https://pastebin.com/caxdqdqg Any help would be great. Regards, Sean From sdevoy at bizfocused.com Wed Dec 15 16:19:09 2021 From: sdevoy at bizfocused.com (Sean Devoy) Date: Wed, 15 Dec 2021 16:19:09 +0000 Subject: [Freeswitch-users] Failed outbound call In-Reply-To: References: Message-ID: I failed to mention FS 1.10.2 From sdevoy at bizfocused.com Wed Dec 15 16:35:52 2021 From: sdevoy at bizfocused.com (Sean Devoy) Date: Wed, 15 Dec 2021 16:35:52 +0000 Subject: Outlook mail to FS List??? Message-ID: I am embarrassed that I can't seem to send plain text when submitting to this group. Any tips on Outlook settings? I have selected "Plain Text" as the format. It still appears as an attachment in the list. Regards, Sean Devoy From Antony.Stone at freeswitch.open.source.it Wed Dec 15 17:02:07 2021 From: Antony.Stone at freeswitch.open.source.it (Antony Stone) Date: Wed, 15 Dec 2021 18:02:07 +0100 Subject: [Freeswitch-users] Failed outbound call In-Reply-To: References: Message-ID: <202112151802.07386.Antony.Stone@freeswitch.open.source.it> On Wednesday 15 December 2021 at 17:54:32, Sean Devoy via FreeSWITCH-users wrote: > Hi, > > I have a very strange problem. My customer has one extension out of 16 > that has an issue dialing outside numbers. She hears it ring 4 or 5 > times, then gets a "busy" signal. She says the recipient's phone does not > ring on their end. (I had another user test this and it is confirmed). I suggest a packet capture of the SIP communications between FreeSwitch and both the extension, and the outbound trunk, would be helpful in this case. Antony. -- Some mistakes are too much fun to make only once. Please reply to the list; please *don't* CC me. From dragos at freeswitch.org Wed Dec 15 18:55:58 2021 From: dragos at freeswitch.org (Dragos Oancea) Date: Wed, 15 Dec 2021 20:55:58 +0200 Subject: [Freeswitch-users] Failed outbound call In-Reply-To: <202112151802.07386.Antony.Stone@freeswitch.open.source.it> References: <202112151802.07386.Antony.Stone@freeswitch.open.source.it> Message-ID: Param NDLB-allow-nondup-sdp on the sip profile. https://freeswitch.org/confluence/display/FREESWITCH/NDLB#:~:text=%22true%22%2F%3E-,NDLB%2Dallow%2Dnondup%2Dsdp,you%20want%20that%20broken%20behaviour . On Wed, Dec 15, 2021 at 7:02 PM Antony Stone < Antony.Stone at freeswitch.open.source.it> wrote: > On Wednesday 15 December 2021 at 17:54:32, Sean Devoy via FreeSWITCH-users > wrote: > > > Hi, > > > > I have a very strange problem. My customer has one extension out of 16 > > that has an issue dialing outside numbers. She hears it ring 4 or 5 > > times, then gets a "busy" signal. She says the recipient's phone does > not > > ring on their end. (I had another user test this and it is confirmed). > > I suggest a packet capture of the SIP communications between FreeSwitch > and > both the extension, and the outbound trunk, would be helpful in this case. > > > Antony. > > -- > Some mistakes are too much fun to make only once. > > Please reply to the > list; > please *don't* CC > me. > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From mrjoli021 at gmail.com Wed Dec 15 21:29:46 2021 From: mrjoli021 at gmail.com (Joli Martinez) Date: Wed, 15 Dec 2021 16:29:46 -0500 Subject: [Freeswitch-users] weird extra characters Message-ID: Hello, I am running freeswitch 1.10.3. I have several DID's on this system. Some work without any issues, but some somehow get some weird characters added to the SIP message. I have already verified that there are no extra characters on the number. It appears to be a certain combination that it is happening to. If I change the last number from 1024 to 1023, nothing get added but on 1024 it does. Any suggestions on why this is happening and/or how to fix/patch this? INVITE sip:933 at 13.13.13.13 SIP/2.0 Via: SIP/2.0/UDP 12.12.12.12:5080;rport;branch=z9hG4bK3FX45c6Sj2Ftc Max-Forwards: 69 From: "122233331024" ;tag=7KmB96jZD2e8r To: Call-ID: 6d1bafb7-d855-123a-8dbc-129e3056e999 CSeq: 45235314 INVITE Contact: User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 297 X-accountcode: 12.12.12.12 X-FS-Support: update_display,send_info Remote-Party-ID: "122233331024" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1639556579 1639556580 IN IP4 12.12.12.12 s=FreeSWITCH c=IN IP4 12.12.12.12 t=0 0 m=audio 21634 RTP/AVP 9 0 8 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 -------------- next part -------------- An HTML attachment was scrubbed... URL: From nhall at unixlan.com.ar Wed Dec 15 23:21:18 2021 From: nhall at unixlan.com.ar (Normando Hall) Date: Wed, 15 Dec 2021 20:21:18 -0300 Subject: [Freeswitch-users] Outlook mail to FS List??? In-Reply-To: References: Message-ID: <1c02df7c-b552-8db4-af0a-5b0ffaadd0cb@unixlan.com.ar> The same for me from thunderbird. El 15/12/2021 a las 13:41, Sean Devoy via FreeSWITCH-users escribió: > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com From ryharris at airmail.cc Wed Dec 15 23:55:39 2021 From: ryharris at airmail.cc (Ryan Harris) Date: Wed, 15 Dec 2021 18:55:39 -0500 Subject: [Freeswitch-users] weird extra characters In-Reply-To: References: Message-ID: <955c1d89-8ee2-1397-21d3-1784733b9e72@airmail.cc> Looks like a zero width char got in somewhere https://codepoints.net/U+202C?lang=en On 12/15/21 16:29, Joli Martinez wrote: > Hello, > > I am running freeswitch 1.10.3.  I have several DID's on this system.  > Some work without any issues, but some somehow get some weird > characters added to the SIP message.  I have already verified that > there are no extra characters on the number.  It appears to be a > certain combination that it is happening to.  If I change the last > number from 1024 to 1023, nothing get added but on 1024 it does.  Any > suggestions on why this is happening and/or how to fix/patch this? > > > > INVITEsip:933 at 13.13.13.13 SIP/2.0 > Via: SIP/2.0/UDP 12.12.12.12:5080;rport;branch=z9hG4bK3FX45c6Sj2Ftc > Max-Forwards: 69 > From: "122233331024" >;tag=7KmB96jZD2e8r > To: > > Call-ID: 6d1bafb7-d855-123a-8dbc-129e3056e999 > CSeq: 45235314 INVITE > Contact: > > User-Agent: FreeSWITCH > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY > Supported: timer, path, replaces > Allow-Events: talk, hold, conference, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 297 > X-accountcode: 12.12.12.12 > X-FS-Support: update_display,send_info > Remote-Party-ID: "122233331024" >;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 1639556579 1639556580 IN IP4 12.12.12.12 > s=FreeSWITCH > c=IN IP4 12.12.12.12 > t=0 0 > m=audio 21634 RTP/AVP 9 0 8 101 > a=rtpmap:9 G722/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWirehttps://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 timehttps://signalwire.community > > Professional FreeSWITCH Services > sales at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From lucas at toptive.co Thu Dec 16 13:59:08 2021 From: lucas at toptive.co (Lucas Ducculi) Date: Thu, 16 Dec 2021 10:59:08 -0300 Subject: [Freeswitch-users] Problem with socket connection Message-ID: Hi, I am using FreeSWITCH (version 1.10.2-release-14-f7bdd3845a ~ 64bit), the system handles 30-70 simultaneous calls. On the other hand, we have an IVR system that uses the node-esl library to connect to FreeSWITCH. The problem we have is that some calls are dropped at different stages of the IVR menu, and in the FS log we see the following. In this case, a connection with the IVR was not established: 2021-12-15 13:31:43.332652 [DEBUG] sofia.c:7301 Channel sofia/external/ XXXXXXXXXX at sip.telnyx.com entering state [early][183] 2021-12-15 13:31:43.332652 [DEBUG] switch_core_media_bug.c:970 Attaching BUG to sofia/external/XXXXXXXXXX at sip.telnyx.com EXECUTE [depth=0] sofia/external/XXXXXXXXXX at sip.telnyx.com set_audio_level(read 0) EXECUTE [depth=0] sofia/external/XXXXXXXXXX at sip.telnyx.com socket( 127.0.0.1:8085 async full) 2021-12-15 13:31:43.332652 [NOTICE] mod_event_socket.c:450 Trying host: 127.0.0.1:8085 2021-12-15 13:31:43.332652 [ERR] mod_event_socket.c:484 Socket Error: Connection refused 2021-12-15 13:31:43.332652 [ERR] mod_event_socket.c:488 Socket Error! 2021-12-15 13:31:43.332652 [NOTICE] switch_core_state_machine.c:386 sofia/external/XXXXXXXXXX at sip.telnyx.com has executed the last dialplan instruction, hanging up. 2021-12-15 13:31:43.332652 [NOTICE] switch_core_state_machine.c:388 Hangup sofia/external/XXXXXXXXXX at sip.telnyx.com [CS_EXECUTE] [NORMAL_CLEARING] In this other example I don't see an error with the socket in the log, but the call cuts out just after reproducing this silence: 2021-12-14 21:20:08.732663 [DEBUG] switch_ivr_play_say.c:1933 done playing file silence_stream://250 2021-12-14 21:20:08.732663 [NOTICE] switch_core_state_machine.c:386 sofia/external/XXXXXXXXXX at sip.telnyx.com has executed the last dialplan instruction, hanging up. 2021-12-14 21:20:08.732663 [NOTICE] switch_core_state_machine.c:388 Hangup sofia/external/XXXXXXXXXX at sip.telnyx.com [CS_EXECUTE] [NORMAL_CLEARING] Other example: 2021-12-15 13:08:10.452655 [DEBUG] switch_ivr.c:632 sofia/external/ XXXXXXXXXX at sip.telnyx.com Command Execute [depth=1] play_and_get_digits(0 1 1 0 1234567890 phrase:greeting silence_stream://250 dialed_digits) EXECUTE [depth=1] sofia/external/XXXXXXXXXX at sip.telnyx.com play_and_get_digits(0 1 1 0 1234567890 phrase:greeting silence_stream://250 dialed_digits) 2021-12-15 13:08:10.492654 [DEBUG] switch_ivr.c:632 sofia/external/ XXXXXXXXXX at sip.telnyx.com Command Execute [depth=2] sleep(3000) EXECUTE [depth=2] sofia/external/XXXXXXXXXX at sip.telnyx.com sleep(3000) 2021-12-15 13:08:10.532671 [DEBUG] sofia.c:7301 Channel sofia/external/ XXXXXXXXXX at sip.telnyx.com entering state [ready][200] 2021-12-15 13:08:30.092667 [NOTICE] switch_core_state_machine.c:386 sofia/external/XXXXXXXXXX at sip.telnyx.com has executed the last dialplan instruction, hanging up. 2021-12-15 13:08:30.092667 [NOTICE] switch_core_state_machine.c:388 Hangup sofia/external/XXXXXXXXXX at sip.telnyx.com [CS_EXECUTE] [NORMAL_CLEARING] My impression is that the problem is related to the socket connection between the IVR and FreeSWITCH. The two things are installed on the same server (Linux Debian 10), any idea what could cause this problem? -------------- next part -------------- An HTML attachment was scrubbed... URL: From Antony.Stone at freeswitch.open.source.it Thu Dec 16 14:28:02 2021 From: Antony.Stone at freeswitch.open.source.it (Antony Stone) Date: Thu, 16 Dec 2021 15:28:02 +0100 Subject: [Freeswitch-users] Problem with socket connection In-Reply-To: References: Message-ID: <202112161528.02822.Antony.Stone@freeswitch.open.source.it> On Thursday 16 December 2021 at 14:59:08, Lucas Ducculi wrote: > 2021-12-15 13:31:43.332652 [NOTICE] mod_event_socket.c:450 Trying host: > 127.0.0.1:8085 > > 2021-12-15 13:31:43.332652 [ERR] mod_event_socket.c:484 Socket Error: > Connection refused > > 2021-12-15 13:31:43.332652 [ERR] mod_event_socket.c:488 Socket Error! I would use tshark or tcpdump to perform a packet capture on the machine for connection requests to port 8085 and see what's happening at that level. Can you do any logging / debugging on the listener side to see whether the request to connecto to port 8085 is being seen and handled in some way? Antony. -- The Magic Words are Squeamish Ossifrage. Please reply to the list; please *don't* CC me. From gregor at infomedia.si Thu Dec 16 16:45:24 2021 From: gregor at infomedia.si (Gregor Nanger) Date: Thu, 16 Dec 2021 17:45:24 +0100 Subject: [Freeswitch-users] Originate with failover Message-ID: Guys, I need some advice. I would like to make an originate to provider and if call fails, proceed to second provider. For this, I can use pipe |. But will pipe obey what I set in continue_on_fail? For example: 1. continue_on_fail=NORMAL_TEMPORARY_FAILURE,NO_ROUTE_DESTINATION 2. originate sofia/gateway/provider1|sofia/gateway/provider2 19005551212 XML default Any advice would be appreciated. Br, Gregor -------------- next part -------------- An HTML attachment was scrubbed... URL: From freeswitch at jhcloos.com Thu Dec 16 19:25:36 2021 From: freeswitch at jhcloos.com (James Cloos) Date: Thu, 16 Dec 2021 14:25:36 -0500 Subject: [Freeswitch-users] weird extra characters In-Reply-To: (Joli Martinez's message of "Wed, 15 Dec 2021 16:29:46 -0500") References: Message-ID: >>>>> "JM" == Joli Martinez writes: JM> From: "122233331024" OpenPGP: 0x997A9F17ED7DAEA6 From Antony.Stone at freeswitch.open.source.it Thu Dec 16 20:21:27 2021 From: Antony.Stone at freeswitch.open.source.it (Antony Stone) Date: Thu, 16 Dec 2021 21:21:27 +0100 Subject: [Freeswitch-users] weird extra characters In-Reply-To: References: Message-ID: <202112162121.27632.Antony.Stone@freeswitch.open.source.it> On Thursday 16 December 2021 at 20:25:36, James Cloos wrote: > >>>>> "JM" == Joli Martinez writes: > JM> From: "122233331024" > 0xE2 0x80 0xAC is the utf-8 sequence for: > > U+202C POP DIRECTIONAL FORMATTING Does this help in any way to understand how it managed to get in there, though? I know that question sounds potentially confrontational, but it's not intended as such - it's a literal "does it give us a clue as to what might have happened so that this was the result?" Antony. -- Perfection in design is achieved not when there is nothing left to add, but rather when there is nothing left to take away. - Antoine de Saint-Exupery Please reply to the list; please *don't* CC me. From shaun at sysconfig.cloud Thu Dec 16 20:25:59 2021 From: shaun at sysconfig.cloud (Shaun Stokes) Date: Thu, 16 Dec 2021 20:25:59 +0000 Subject: [Freeswitch-users] Originate with failover In-Reply-To: References: Message-ID: Can't confirm if originate will work with pipe (never tried it), you can always test it, but I don't think continue_on_fail should be necessary. When using pipe with gateways, you probably want to add leg_timeout, we've had issues with calls getting stuck on a down gateway in the past, leg_timeout ensures the call fails over after the specified time if there's no response. Some mobiles take a while to respond with ringing but 15s should be ample. For example: {continue_on_fail=NORMAL_TEMPORARY_FAILURE,NO_ROUTE_DESTINATION}[leg_timeout=15]originate sofia/gateway/provider1|[leg_timeout=15]sofia/gateway/provider2 19005551212 XML default Shaun ________________________________ From: FreeSWITCH-users on behalf of Gregor Nanger Sent: 16 December 2021 17:45 To: FreeSWITCH Users Help Subject: [Freeswitch-users] Originate with failover Guys, I need some advice. I would like to make an originate to provider and if call fails, proceed to second provider. For this, I can use pipe |. But will pipe obey what I set in continue_on_fail? For example: 1. continue_on_fail=NORMAL_TEMPORARY_FAILURE,NO_ROUTE_DESTINATION 2. originate sofia/gateway/provider1|sofia/gateway/provider2 19005551212 XML default Any advice would be appreciated. Br, Gregor -------------- next part -------------- An HTML attachment was scrubbed... URL: From ryharris at airmail.cc Thu Dec 16 20:47:22 2021 From: ryharris at airmail.cc (Ryan Harris) Date: Thu, 16 Dec 2021 15:47:22 -0500 Subject: [Freeswitch-users] weird extra characters In-Reply-To: <202112162121.27632.Antony.Stone@freeswitch.open.source.it> References: <202112162121.27632.Antony.Stone@freeswitch.open.source.it> Message-ID: <233c2e5b-945e-ce47-91d7-b688355d47d4@airmail.cc> On 12/16/21 15:21, Antony Stone wrote: > On Thursday 16 December 2021 at 20:25:36, James Cloos wrote: > >>>>>>> "JM" == Joli Martinez writes: >> JM> From: "122233331024" > >> 0xE2 0x80 0xAC is the utf-8 sequence for: >> >> U+202C POP DIRECTIONAL FORMATTING > Does this help in any way to understand how it managed to get in there, > though? > > I know that question sounds potentially confrontational, but it's not intended > as such - it's a literal "does it give us a clue as to what might have > happened so that this was the result?" > > > Antony. > If it's a Windows machine they were editing with, it's very easy to accidentally insert unicode control characters with the context menu. From Antony.Stone at freeswitch.open.source.it Thu Dec 16 21:15:35 2021 From: Antony.Stone at freeswitch.open.source.it (Antony Stone) Date: Thu, 16 Dec 2021 22:15:35 +0100 Subject: [Freeswitch-users] weird extra characters In-Reply-To: <233c2e5b-945e-ce47-91d7-b688355d47d4@airmail.cc> References: <202112162121.27632.Antony.Stone@freeswitch.open.source.it> <233c2e5b-945e-ce47-91d7-b688355d47d4@airmail.cc> Message-ID: <202112162215.35261.Antony.Stone@freeswitch.open.source.it> On Thursday 16 December 2021 at 21:47:22, Ryan Harris wrote: > On 12/16/21 15:21, Antony Stone wrote: > > > > Does this help in any way to understand how it managed to get in there, > > though? > > If it's a Windows machine they were editing with, it's very easy to > accidentally insert unicode control characters with the context menu. Windows. Meh. Antony. -- RTFM may be the appropriate reply, but please specify exactly which FM to R. Please reply to the list; please *don't* CC me. From gregor at infomedia.si Thu Dec 16 21:21:35 2021 From: gregor at infomedia.si (Gregor Nanger) Date: Thu, 16 Dec 2021 22:21:35 +0100 Subject: [Freeswitch-users] Originate with failover In-Reply-To: References: Message-ID: Thank you Shaun. Guess if the first gateway will respond with NORMAL_CLEARING or USER_BUSY, call will not go to the next gateway? Br, Gregor On Thu, 16 Dec 2021 at 21:27, Shaun Stokes wrote: > Can't confirm if originate will work with pipe (never tried it), you can > always test it, but I don't think continue_on_fail should be necessary. > When using pipe with gateways, you probably want to add leg_timeout, we've > had issues with calls getting stuck on a down gateway in the past, > leg_timeout ensures the call fails over after the specified time if there's > no response. Some mobiles take a while to respond with ringing but 15s > should be ample. > > For example: > {continue_on_fail=NORMAL_TEMPORARY_FAILURE,NO_ROUTE_DESTINATION}[ > leg_timeout=15]originate sofia/gateway/provider1|[leg_timeout=15]sofia/gateway/provider2 > 19005551212 XML default > > Shaun > ------------------------------ > *From:* FreeSWITCH-users > on behalf of Gregor Nanger > *Sent:* 16 December 2021 17:45 > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] Originate with failover > > Guys, I need some advice. > > I would like to make an originate to provider and if call fails, proceed > to second provider. > > For this, I can use pipe |. But will pipe obey what I set > in continue_on_fail? > > For example: > > 1. continue_on_fail=NORMAL_TEMPORARY_FAILURE,NO_ROUTE_DESTINATION > 2. originate sofia/gateway/provider1|sofia/gateway/provider2 19005551212 > XML default > > Any advice would be appreciated. > > Br, Gregor > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From mrjoli021 at gmail.com Thu Dec 16 22:14:28 2021 From: mrjoli021 at gmail.com (Joli Martinez) Date: Thu, 16 Dec 2021 17:14:28 -0500 Subject: [Freeswitch-users] weird extra characters In-Reply-To: <202112162215.35261.Antony.Stone@freeswitch.open.source.it> References: <202112162121.27632.Antony.Stone@freeswitch.open.source.it> <233c2e5b-945e-ce47-91d7-b688355d47d4@airmail.cc> <202112162215.35261.Antony.Stone@freeswitch.open.source.it> Message-ID: Hello, This is a Debian 10 box running FusionPBX. There is no windows on this. Polycom phone makes a call and these characters are being added. It only add the characters to a certain combination of numbers. If I change the callerID to be 1023 instead of 1024 it works. Also the polycom does not have the caller id set on the registration. So something is adding it to the outbound leg. I am just trying to get more info on how fix it. Just googling " U+202C POP DIRECTIONAL FORMATTING" does not help much. it is adding it on the sip invite not before, so my application does not have any characters. I have made sure to strip anything before sending it to the Gateway. Could someone please point me in the right direction? On Thu, Dec 16, 2021 at 4:36 PM Antony Stone < Antony.Stone at freeswitch.open.source.it> wrote: > On Thursday 16 December 2021 at 21:47:22, Ryan Harris wrote: > > > On 12/16/21 15:21, Antony Stone wrote: > > > > > > Does this help in any way to understand how it managed to get in there, > > > though? > > > > If it's a Windows machine they were editing with, it's very easy to > > accidentally insert unicode control characters with the context menu. > > Windows. Meh. > > Antony. > > -- > RTFM may be the appropriate reply, but please specify exactly which FM to > R. > > Please reply to the > list; > please *don't* CC > me. > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From brians at iptel.co Fri Dec 17 06:02:24 2021 From: brians at iptel.co (Brian :) Date: Fri, 17 Dec 2021 06:02:24 +0000 Subject: [Freeswitch-users] weird extra characters In-Reply-To: References: <202112162121.27632.Antony.Stone@freeswitch.open.source.it> <233c2e5b-945e-ce47-91d7-b688355d47d4@airmail.cc> <202112162215.35261.Antony.Stone@freeswitch.open.source.it> Message-ID: Have you a trace from the leg from the poly? On Thursday, December 16, 2021, Joli Martinez wrote: > Hello, > > This is a Debian 10 box running FusionPBX. There is no windows on this. Polycom phone makes a call and these characters are being added. It only add the characters to a certain combination of numbers. If I change the callerID to be 1023 instead of 1024 it works. Also the polycom does not have the caller id set on the registration. So something is adding it to the outbound leg. I am just trying to get more info on how fix it. Just googling " U+202C POP DIRECTIONAL FORMATTING" does not help much. it is adding it on the sip invite not before, so my application does not have any characters. I have made sure to strip anything before sending it to the Gateway. Could someone please point me in the right direction? > On Thu, Dec 16, 2021 at 4:36 PM Antony Stone < Antony.Stone at freeswitch.open.source.it> wrote: >> >> On Thursday 16 December 2021 at 21:47:22, Ryan Harris wrote: >> >> > On 12/16/21 15:21, Antony Stone wrote: >> > > >> > > Does this help in any way to understand how it managed to get in there, >> > > though? >> > >> > If it's a Windows machine they were editing with, it's very easy to >> > accidentally insert unicode control characters with the context menu. >> >> Windows. Meh. >> >> Antony. >> >> -- >> RTFM may be the appropriate reply, but please specify exactly which FM to R. >> >> Please reply to the list; >> please *don't* CC me. >> >> _________________________________________________________________________ >> >> 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From dragos at freeswitch.org Fri Dec 17 07:16:08 2021 From: dragos at freeswitch.org (Dragos Oancea) Date: Fri, 17 Dec 2021 09:16:08 +0200 Subject: [Freeswitch-users] weird extra characters In-Reply-To: References: <202112162121.27632.Antony.Stone@freeswitch.open.source.it> <233c2e5b-945e-ce47-91d7-b688355d47d4@airmail.cc> <202112162215.35261.Antony.Stone@freeswitch.open.source.it> Message-ID: It could be a memcpy copying too much. Open a github issue with logs and pcaps. On Fri, Dec 17, 2021 at 8:03 AM Brian : wrote: > Have you a trace from the leg from the poly? > > On Thursday, December 16, 2021, Joli Martinez wrote: > > Hello, > > > > This is a Debian 10 box running FusionPBX. There is no windows on > this. Polycom phone makes a call and these characters are being added. It > only add the characters to a certain combination of numbers. If I change > the callerID to be 1023 instead of 1024 it works. Also the polycom does > not have the caller id set on the registration. So something is adding it > to the outbound leg. I am just trying to get more info on how fix it. > Just googling " U+202C POP DIRECTIONAL FORMATTING" does not help much. it > is adding it on the sip invite not before, so my application does not have > any characters. I have made sure to strip anything before sending it to > the Gateway. Could someone please point me in the right direction? > > On Thu, Dec 16, 2021 at 4:36 PM Antony Stone < > Antony.Stone at freeswitch.open.source.it> wrote: > >> > >> On Thursday 16 December 2021 at 21:47:22, Ryan Harris wrote: > >> > >> > On 12/16/21 15:21, Antony Stone wrote: > >> > > > >> > > Does this help in any way to understand how it managed to get in > there, > >> > > though? > >> > > >> > If it's a Windows machine they were editing with, it's very easy to > >> > accidentally insert unicode control characters with the context menu. > >> > >> Windows. Meh. > >> > >> Antony. > >> > >> -- > >> RTFM may be the appropriate reply, but please specify exactly which FM > to R. > >> > >> Please reply to the > list; > >> please *don't* > CC me. > >> > >> > _________________________________________________________________________ > >> > >> 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 at 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 at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> https://freeswitch.com > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From shaun at sysconfig.cloud Fri Dec 17 10:02:10 2021 From: shaun at sysconfig.cloud (Shaun Stokes) Date: Fri, 17 Dec 2021 10:02:10 +0000 Subject: [Freeswitch-users] Originate with failover In-Reply-To: References: Message-ID: You may want to specify fail_on_single_reject. For example: {continue_on_fail=^^:NORMAL_TEMPORARY_FAILURE:NO_ROUTE_DESTINATION,fail_on_single_reject=^^:CALL_REJECTED:NORMAL_CLEARING:USER_BUSY}[leg_timeout=15]originate sofia/gateway/provider1|[leg_timeout=15]sofia/gateway/provider2 19005551212 XML default Shaun ________________________________ From: FreeSWITCH-users on behalf of Gregor Nanger Sent: 16 December 2021 22:21 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Originate with failover Thank you Shaun. Guess if the first gateway will respond with NORMAL_CLEARING or USER_BUSY, call will not go to the next gateway? Br, Gregor On Thu, 16 Dec 2021 at 21:27, Shaun Stokes wrote: Can't confirm if originate will work with pipe (never tried it), you can always test it, but I don't think continue_on_fail should be necessary. When using pipe with gateways, you probably want to add leg_timeout, we've had issues with calls getting stuck on a down gateway in the past, leg_timeout ensures the call fails over after the specified time if there's no response. Some mobiles take a while to respond with ringing but 15s should be ample. For example: {continue_on_fail=NORMAL_TEMPORARY_FAILURE,NO_ROUTE_DESTINATION}[leg_timeout=15]originate sofia/gateway/provider1|[leg_timeout=15]sofia/gateway/provider2 19005551212 XML default Shaun ________________________________ From: FreeSWITCH-users > on behalf of Gregor Nanger > Sent: 16 December 2021 17:45 To: FreeSWITCH Users Help > Subject: [Freeswitch-users] Originate with failover Guys, I need some advice. I would like to make an originate to provider and if call fails, proceed to second provider. For this, I can use pipe |. But will pipe obey what I set in continue_on_fail? For example: 1. continue_on_fail=NORMAL_TEMPORARY_FAILURE,NO_ROUTE_DESTINATION 2. originate sofia/gateway/provider1|sofia/gateway/provider2 19005551212 XML default Any advice would be appreciated. Br, Gregor _________________________________________________________________________ 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From dragos at freeswitch.org Fri Dec 17 10:05:23 2021 From: dragos at freeswitch.org (Dragos Oancea) Date: Fri, 17 Dec 2021 12:05:23 +0200 Subject: [Freeswitch-users] Problem with socket connection In-Reply-To: <202112161528.02822.Antony.Stone@freeswitch.open.source.it> References: <202112161528.02822.Antony.Stone@freeswitch.open.source.it> Message-ID: It looks like you have outbound ESL so you have FS connecting to the IVR, not the other way around. IVR not running ? "mod_event_socket.c:484 Socket Error: Connection refused" On Thu, Dec 16, 2021 at 4:28 PM Antony Stone < Antony.Stone at freeswitch.open.source.it> wrote: > On Thursday 16 December 2021 at 14:59:08, Lucas Ducculi wrote: > > > 2021-12-15 13:31:43.332652 [NOTICE] mod_event_socket.c:450 Trying host: > > 127.0.0.1:8085 > > > > 2021-12-15 13:31:43.332652 [ERR] mod_event_socket.c:484 Socket Error: > > Connection refused > > > > 2021-12-15 13:31:43.332652 [ERR] mod_event_socket.c:488 Socket Error! > > I would use tshark or tcpdump to perform a packet capture on the machine > for > connection requests to port 8085 and see what's happening at that level. > > Can you do any logging / debugging on the listener side to see whether the > request to connecto to port 8085 is being seen and handled in some way? > > > Antony. > > -- > The Magic Words are Squeamish Ossifrage. > > Please reply to the > list; > please *don't* CC > me. > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From gregor at infomedia.si Fri Dec 17 11:45:34 2021 From: gregor at infomedia.si (Gregor Nanger) Date: Fri, 17 Dec 2021 12:45:34 +0100 Subject: [Freeswitch-users] Originate with failover In-Reply-To: References: Message-ID: It works! It looks that pipe | obey continue_on_fail. Nevertheless, in the end, I didn't specify this variable as it looks the pipe has all logic already implemented. 👍👍 On Fri, 17 Dec 2021 at 11:03, Shaun Stokes wrote: > You may want to specify fail_on_single_reject. > > For example: > {continue_on_fail=^^:NORMAL_TEMPORARY_FAILURE:NO_ROUTE_DESTINATION, > fail_on_single_reject=^^:CALL_REJECTED:NORMAL_CLEARING:USER_BUSY}[ > leg_timeout=15]originate sofia/gateway/provider1|[leg_timeout=15]sofia/gateway/provider2 > 19005551212 XML default > > Shaun > ------------------------------ > *From:* FreeSWITCH-users > on behalf of Gregor Nanger > *Sent:* 16 December 2021 22:21 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Originate with failover > > Thank you Shaun. > > Guess if the first gateway will respond with NORMAL_CLEARING or USER_BUSY, > call will not go to the next gateway? > > Br, Gregor > > On Thu, 16 Dec 2021 at 21:27, Shaun Stokes wrote: > > Can't confirm if originate will work with pipe (never tried it), you can > always test it, but I don't think continue_on_fail should be necessary. > When using pipe with gateways, you probably want to add leg_timeout, we've > had issues with calls getting stuck on a down gateway in the past, > leg_timeout ensures the call fails over after the specified time if there's > no response. Some mobiles take a while to respond with ringing but 15s > should be ample. > > For example: > {continue_on_fail=NORMAL_TEMPORARY_FAILURE,NO_ROUTE_DESTINATION}[ > leg_timeout=15]originate sofia/gateway/provider1|[leg_timeout=15]sofia/gateway/provider2 > 19005551212 XML default > > Shaun > ------------------------------ > *From:* FreeSWITCH-users > on behalf of Gregor Nanger > *Sent:* 16 December 2021 17:45 > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] Originate with failover > > Guys, I need some advice. > > I would like to make an originate to provider and if call fails, proceed > to second provider. > > For this, I can use pipe |. But will pipe obey what I set > in continue_on_fail? > > For example: > > 1. continue_on_fail=NORMAL_TEMPORARY_FAILURE,NO_ROUTE_DESTINATION > 2. originate sofia/gateway/provider1|sofia/gateway/provider2 19005551212 > XML default > > Any advice would be appreciated. > > Br, Gregor > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > > > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From lists at telium.io Thu Dec 16 22:54:33 2021 From: lists at telium.io (TTT) Date: Thu, 16 Dec 2021 22:54:33 +0000 Subject: [Freeswitch-users] weird extra characters In-Reply-To: References: <202112162121.27632.Antony.Stone@freeswitch.open.source.it> <233c2e5b-945e-ce47-91d7-b688355d47d4@airmail.cc> <202112162215.35261.Antony.Stone@freeswitch.open.source.it> Message-ID: <0100017dc5744c63-427a96e8-edc0-4082-ac2d-08378a545892-000000@email.amazonses.com> If you do a packet capture, are these characters really coming from the Polycom, or is freeswitch adding them on the incoming session? (Showing in the SDP) I missed first part of discussion – but interested From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Joli Martinez Sent: Thursday, December 16, 2021 5:14 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] weird extra characters Hello, This is a Debian 10 box running FusionPBX. There is no windows on this. Polycom phone makes a call and these characters are being added. It only add the characters to a certain combination of numbers. If I change the callerID to be 1023 instead of 1024 it works. Also the polycom does not have the caller id set on the registration. So something is adding it to the outbound leg. I am just trying to get more info on how fix it. Just googling " U+202C POP DIRECTIONAL FORMATTING" does not help much. it is adding it on the sip invite not before, so my application does not have any characters. I have made sure to strip anything before sending it to the Gateway. Could someone please point me in the right direction? On Thu, Dec 16, 2021 at 4:36 PM Antony Stone > wrote: On Thursday 16 December 2021 at 21:47:22, Ryan Harris wrote: > On 12/16/21 15:21, Antony Stone wrote: > > > > Does this help in any way to understand how it managed to get in there, > > though? > > If it's a Windows machine they were editing with, it's very easy to > accidentally insert unicode control characters with the context menu. Windows. Meh. Antony. -- RTFM may be the appropriate reply, but please specify exactly which FM to R. Please reply to the list; please *don't* CC me. _________________________________________________________________________ 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From sdevoy at bizfocused.com Fri Dec 17 17:08:15 2021 From: sdevoy at bizfocused.com (Sean Devoy) Date: Fri, 17 Dec 2021 17:08:15 +0000 Subject: [Freeswitch-users] Failed outbound call In-Reply-To: References: <202112151802.07386.Antony.Stone@freeswitch.open.source.it> Message-ID: THANK YOU. That fixed it! Any idea why it is only one extension out of the 16? Very odd. From: FreeSWITCH-users On Behalf Of Dragos Oancea Sent: Wednesday, December 15, 2021 1:56 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Failed outbound call Param NDLB-allow-nondup-sdp on the sip profile. https://freeswitch.org/confluence/display/FREESWITCH/NDLB#:~:text=%22true%22%2F%3E-,NDLB%2Dallow%2Dnondup%2Dsdp,you%20want%20that%20broken%20behaviour. On Wed, Dec 15, 2021 at 7:02 PM Antony Stone > wrote: On Wednesday 15 December 2021 at 17:54:32, Sean Devoy via FreeSWITCH-users wrote: > Hi, > > I have a very strange problem. My customer has one extension out of 16 > that has an issue dialing outside numbers. She hears it ring 4 or 5 > times, then gets a "busy" signal. She says the recipient's phone does not > ring on their end. (I had another user test this and it is confirmed). I suggest a packet capture of the SIP communications between FreeSwitch and both the extension, and the outbound trunk, would be helpful in this case. Antony. -- Some mistakes are too much fun to make only once. Please reply to the list; please *don't* CC me. _________________________________________________________________________ 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From lists at telium.io Fri Dec 17 16:57:43 2021 From: lists at telium.io (TTT) Date: Fri, 17 Dec 2021 16:57:43 +0000 Subject: [Freeswitch-users] weird extra characters In-Reply-To: <0100017dc5744c63-427a96e8-edc0-4082-ac2d-08378a545892-000000@email.amazonses.com> References: <202112162121.27632.Antony.Stone@freeswitch.open.source.it> <233c2e5b-945e-ce47-91d7-b688355d47d4@airmail.cc> <202112162215.35261.Antony.Stone@freeswitch.open.source.it> <0100017dc5744c63-427a96e8-edc0-4082-ac2d-08378a545892-000000@email.amazonses.com> Message-ID: <0100017dc953f98b-65991918-cada-4e1e-bb59-57a98216dcdb-000000@email.amazonses.com> It seems my helpful post was delayed by a day…so ignore it now. For some reason my posts are very slow to reach the list. From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of TTT Sent: Thursday, December 16, 2021 5:55 PM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] weird extra characters If you do a packet capture, are these characters really coming from the Polycom, or is freeswitch adding them on the incoming session? (Showing in the SDP) I missed first part of discussion – but interested From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Joli Martinez Sent: Thursday, December 16, 2021 5:14 PM To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] weird extra characters Hello, This is a Debian 10 box running FusionPBX. There is no windows on this. Polycom phone makes a call and these characters are being added. It only add the characters to a certain combination of numbers. If I change the callerID to be 1023 instead of 1024 it works. Also the polycom does not have the caller id set on the registration. So something is adding it to the outbound leg. I am just trying to get more info on how fix it. Just googling " U+202C POP DIRECTIONAL FORMATTING" does not help much. it is adding it on the sip invite not before, so my application does not have any characters. I have made sure to strip anything before sending it to the Gateway. Could someone please point me in the right direction? On Thu, Dec 16, 2021 at 4:36 PM Antony Stone > wrote: On Thursday 16 December 2021 at 21:47:22, Ryan Harris wrote: > On 12/16/21 15:21, Antony Stone wrote: > > > > Does this help in any way to understand how it managed to get in there, > > though? > > If it's a Windows machine they were editing with, it's very easy to > accidentally insert unicode control characters with the context menu. Windows. Meh. Antony. -- RTFM may be the appropriate reply, but please specify exactly which FM to R. Please reply to the list; please *don't* CC me. _________________________________________________________________________ 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From mrjoli021 at gmail.com Fri Dec 17 17:16:14 2021 From: mrjoli021 at gmail.com (Joli Martinez) Date: Fri, 17 Dec 2021 12:16:14 -0500 Subject: [Freeswitch-users] weird extra characters In-Reply-To: <0100017dc5744c63-427a96e8-edc0-4082-ac2d-08378a545892-000000@email.amazonses.com> References: <202112162121.27632.Antony.Stone@freeswitch.open.source.it> <233c2e5b-945e-ce47-91d7-b688355d47d4@airmail.cc> <202112162215.35261.Antony.Stone@freeswitch.open.source.it> <0100017dc5744c63-427a96e8-edc0-4082-ac2d-08378a545892-000000@email.amazonses.com> Message-ID: Hello, I am doing a pcap on the inbound and outbound leg. The pcaps shows that the characters are not being added until it sends the invite out it's Gateway. Before then the pcap does not show any anomalies. Somehow on the invite to the carrier the characters are being added not before. I again have made sure this is the case. I wrote a Lua script that would strip out any characters before, after and in between the called number. And this script is run right before sending the call out. I know how to control the call prior to it hitting the GW, what I don't know how to do is modify the invite this far along in the call. On Fri, Dec 17, 2021 at 11:48 AM TTT wrote: > If you do a packet capture, are these characters really coming from the > Polycom, or is freeswitch adding them on the incoming session? (Showing in > the SDP) > > > > I missed first part of discussion – but interested > > > > *From:* FreeSWITCH-users [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Joli > Martinez > *Sent:* Thursday, December 16, 2021 5:14 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] weird extra characters > > > > Hello, > > > > This is a Debian 10 box running FusionPBX. There is no windows on this. > Polycom phone makes a call and these characters are being added. It only > add the characters to a certain combination of numbers. If I change the > callerID to be 1023 instead of 1024 it works. Also the polycom does > not have the caller id set on the registration. So something is adding it > to the outbound leg. I am just trying to get more info on how fix it. > Just googling " U+202C POP DIRECTIONAL FORMATTING" does not help much. it > is adding it on the sip invite not before, so my application does not have > any characters. I have made sure to strip anything before sending it to > the Gateway. Could someone please point me in the right direction? > > > > On Thu, Dec 16, 2021 at 4:36 PM Antony Stone < > Antony.Stone at freeswitch.open.source.it> wrote: > > On Thursday 16 December 2021 at 21:47:22, Ryan Harris wrote: > > > On 12/16/21 15:21, Antony Stone wrote: > > > > > > Does this help in any way to understand how it managed to get in there, > > > though? > > > > If it's a Windows machine they were editing with, it's very easy to > > accidentally insert unicode control characters with the context menu. > > Windows. Meh. > > Antony. > > -- > RTFM may be the appropriate reply, but please specify exactly which FM to > R. > > Please reply to the > list; > please *don't* CC > me. > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From krice at freeswitch.org Fri Dec 17 18:36:41 2021 From: krice at freeswitch.org (Ken Rice) Date: Fri, 17 Dec 2021 12:36:41 -0600 Subject: [Freeswitch-users] weird extra characters In-Reply-To: References: Message-ID: turn on sofia debug and sofia global siptrace, and get a pcap and open a bug including the above data so someone can evaluate it Sent from my iPhone > On Dec 17, 2021, at 11:17, Joli Martinez wrote: > >  > Hello, > > I am doing a pcap on the inbound and outbound leg. The pcaps shows that the characters are not being added until it sends the invite out it's Gateway. Before then the pcap does not show any anomalies. Somehow on the invite to the carrier the characters are being added not before. I again have made sure this is the case. I wrote a Lua script that would strip out any characters before, after and in between the called number. And this script is run right before sending the call out. I know how to control the call prior to it hitting the GW, what I don't know how to do is modify the invite this far along in the call. > >> On Fri, Dec 17, 2021 at 11:48 AM TTT wrote: >> If you do a packet capture, are these characters really coming from the Polycom, or is freeswitch adding them on the incoming session? (Showing in the SDP) >> >> >> >> I missed first part of discussion – but interested >> >> >> >> From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Joli Martinez >> Sent: Thursday, December 16, 2021 5:14 PM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] weird extra characters >> >> >> >> Hello, >> >> >> >> This is a Debian 10 box running FusionPBX. There is no windows on this. Polycom phone makes a call and these characters are being added. It only add the characters to a certain combination of numbers. If I change the callerID to be 1023 instead of 1024 it works. Also the polycom does not have the caller id set on the registration. So something is adding it to the outbound leg. I am just trying to get more info on how fix it. Just googling " U+202C POP DIRECTIONAL FORMATTING" does not help much. it is adding it on the sip invite not before, so my application does not have any characters. I have made sure to strip anything before sending it to the Gateway. Could someone please point me in the right direction? >> >> >> >> On Thu, Dec 16, 2021 at 4:36 PM Antony Stone wrote: >> >> On Thursday 16 December 2021 at 21:47:22, Ryan Harris wrote: >> >> > On 12/16/21 15:21, Antony Stone wrote: >> > > >> > > Does this help in any way to understand how it managed to get in there, >> > > though? >> > >> > If it's a Windows machine they were editing with, it's very easy to >> > accidentally insert unicode control characters with the context menu. >> >> Windows. Meh. >> >> Antony. >> >> -- >> RTFM may be the appropriate reply, but please specify exactly which FM to R. >> >> Please reply to the list; >> please *don't* CC me. >> >> _________________________________________________________________________ >> >> 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com >> >> _________________________________________________________________________ >> >> 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Sat Dec 18 01:05:43 2021 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 17 Dec 2021 20:05:43 -0500 Subject: [Freeswitch-users] FEC and bitrates Message-ID: Hello all, i've been testing FEC and how low we can go with maxavgbiterate we only see FEC if we set maxavgbiterate to 18000, we see no FEC with 14400 anyone knows what i'm missing? Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From dragos at freeswitch.org Sat Dec 18 09:41:03 2021 From: dragos at freeswitch.org (Dragos Oancea) Date: Sat, 18 Dec 2021 11:41:03 +0200 Subject: [Freeswitch-users] FEC and bitrates In-Reply-To: References: Message-ID: David, Most likely your mod_opus is configured to respect the remote's fmtp and not the maxaveragebitrate and maxplaybackrate that you configure. You have bitrate-negotiation set to true. It looks it's going in WB mode if you are sure it starts sending FEC at over 16 kbit/s . Value 14400 was chosen as config default only because is one step above 14000 which is one of the LBRR thresholds (Opus's bitrate works in steps of 400 bits/s) and it also provides a good compromis loss/audio quality for NB. https://github.com/xiph/opus/blob/master/silk/define.h#L52 /* LBRR thresholds */ #define LBRR_NB_MIN_RATE_BPS 12000 #define LBRR_MB_MIN_RATE_BPS 14000 #define LBRR_WB_MIN_RATE_BPS 16000 FEC threshold in FS is this: #define SWITCH_OPUS_MIN_FEC_BITRATE 12400 - one step above codec's LBRR threshold for NB. LBRR means Low Bit Rate Redundancy which means FEC. Regards, Dragos On Sat, Dec 18, 2021 at 3:06 AM David Villasmil < david.villasmil.work at gmail.com> wrote: > Hello all, > > i've been testing FEC and how low we can go with maxavgbiterate > > > > > > > > > > > > > > > we only see FEC if we set maxavgbiterate to 18000, we see no FEC with 14400 > > anyone knows what i'm missing? > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Sat Dec 18 22:22:25 2021 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sat, 18 Dec 2021 17:22:25 -0500 Subject: [Freeswitch-users] FEC and bitrates In-Reply-To: References: Message-ID: Hey Dragos, Thanks for that, yeah i think the issue is on the client side somehow, i had already tried not negotiating... Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Sat, Dec 18, 2021 at 4:41 AM Dragos Oancea wrote: > David, > > Most likely your mod_opus is configured to respect the remote's fmtp and > not the maxaveragebitrate and maxplaybackrate that you configure. You have > bitrate-negotiation set to true. > It looks it's going in WB mode if you are sure it starts sending FEC at > over 16 kbit/s . > Value 14400 was chosen as config default only because is one step above > 14000 which is one of the LBRR thresholds (Opus's bitrate works in steps of > 400 bits/s) and it also provides a good compromis loss/audio quality for > NB. > https://github.com/xiph/opus/blob/master/silk/define.h#L52 > > /* LBRR thresholds */ > #define LBRR_NB_MIN_RATE_BPS 12000 > #define LBRR_MB_MIN_RATE_BPS 14000 > #define LBRR_WB_MIN_RATE_BPS 16000 > > > FEC threshold in FS is this: > #define SWITCH_OPUS_MIN_FEC_BITRATE 12400 - one step above codec's LBRR > threshold for NB. LBRR means Low Bit Rate Redundancy which means FEC. > > Regards, > Dragos > > > > > On Sat, Dec 18, 2021 at 3:06 AM David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> Hello all, >> >> i've been testing FEC and how low we can go with maxavgbiterate >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> we only see FEC if we set maxavgbiterate to 18000, we see no FEC with >> 14400 >> >> anyone knows what i'm missing? >> >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 >> _________________________________________________________________________ >> >> 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From tom at tomlynn.com Mon Dec 20 21:43:05 2021 From: tom at tomlynn.com (Tom Lynn) Date: Mon, 20 Dec 2021 13:43:05 -0800 Subject: [Freeswitch-users] License Documentation Message-ID: Is this page till current? It has not been modified since 2016. Is there a more current version? https://freeswitch.org/confluence/display/FREESWITCH/Licensing Thank you, Tom Lynn -------------- next part -------------- An HTML attachment was scrubbed... URL: From julf at julf.com Tue Dec 21 19:59:31 2021 From: julf at julf.com (Johan Helsingius) Date: Tue, 21 Dec 2021 20:59:31 +0100 Subject: [Freeswitch-users] debian repos for Raspberry Pi broken? Message-ID: Hi, Trying to install freeswitch on a RPi 4 (Debian 11 Bullseye), my /etc/apt/sources.list.d/freeswitch.list has: deb http://files.freeswitch.org/repo/deb/rpi/debian-release/ bullseye main deb-src http://files.freeswitch.org/repo/deb/rpi/debian-release/ bullseye main but when I try apt-get update && apt-get install -y freeswitch-meta-all I get Unable to locate package freeswitch-meta-all apt-cache search freeswitch only gives me: libsipwitch-dev - secure peer-to-peer SIP VoIP server - development files libsipwitch1 - secure peer-to-peer SIP VoIP server - shared libraries libsipwitch1-dbg - secure peer-to-peer SIP VoIP server - debug symbols sipwitch - secure peer-to-peer VoIP server for the SIP protocol sipwitch-cgi - secure peer-to-peer SIP VoIP server - CGI XML-RPC interface Any suggestions? Julf From andywolk at gmail.com Tue Dec 21 20:36:21 2021 From: andywolk at gmail.com (Andrey Volk) Date: Tue, 21 Dec 2021 23:36:21 +0300 Subject: [Freeswitch-users] debian repos for Raspberry Pi broken? In-Reply-To: References: Message-ID: Let me see what I can do. May take some time. вт, 21 дек. 2021 г. в 23:19, Johan Helsingius : > Hi, > > Trying to install freeswitch on a RPi 4 (Debian 11 Bullseye), > my /etc/apt/sources.list.d/freeswitch.list has: > > deb http://files.freeswitch.org/repo/deb/rpi/debian-release/ bullseye main > deb-src http://files.freeswitch.org/repo/deb/rpi/debian-release/ bullseye > main > > but when I try > > apt-get update && apt-get install -y freeswitch-meta-all > > I get > > Unable to locate package freeswitch-meta-all > > apt-cache search freeswitch only gives me: > libsipwitch-dev - secure peer-to-peer SIP VoIP server - development files > libsipwitch1 - secure peer-to-peer SIP VoIP server - shared libraries > libsipwitch1-dbg - secure peer-to-peer SIP VoIP server - debug symbols > sipwitch - secure peer-to-peer VoIP server for the SIP protocol > sipwitch-cgi - secure peer-to-peer SIP VoIP server - CGI XML-RPC interface > > Any suggestions? > > Julf > > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From julf at julf.com Tue Dec 21 21:11:37 2021 From: julf at julf.com (Johan Helsingius) Date: Tue, 21 Dec 2021 22:11:37 +0100 Subject: [Freeswitch-users] debian repos for Raspberry Pi broken? In-Reply-To: References: Message-ID: On 21-12-2021 21:36, Andrey Volk wrote: > Let me see what I can do. May take some time. Thanks! Julf From dragos at freeswitch.org Wed Dec 22 15:35:04 2021 From: dragos at freeswitch.org (Dragos Oancea) Date: Wed, 22 Dec 2021 17:35:04 +0200 Subject: [Freeswitch-users] Problems with Hold, Dynamic Payload, Media Direction In-Reply-To: References: Message-ID: Recent commit 6c87ed491597fb5e30935d8309aa7e0c3aa9e18f to master branch should fix at least the 1st and the 2nd described issues. On Fri, Dec 10, 2021 at 3:46 PM Fuad Trle wrote: > Hello, > > I have encountered several problems while configuring and testing > FreeSWITCH. This is on a current master branch, but the same issues are > present in older versions. The tests are done on a minimal config with > these additions: > > vars: > - remove auth > - global_codec_prefs=OPUS,PCMU,PCMA,VP8 > - rtp_pass_codecs_on_stream_change=true # To be able to toggle video stream > > profiles: > - inbound-late-negotiation (with and w/o inherit codec) > > dialplan: > - Simple dial plan. External to Internal flow. Condition for ext, then > bridge call to the other side. > > Unfortunately, this Sofia options are removed from FS :( Are there > alternatives for them (while using rtp_pass_codecs_on_stream_change)? > renegotiate-codec-on-hold|true,false > renegotiate-codec-on-reinvite|true,false > > (a) -- (FS) -- (b) > > *1st: Hold* - Break transcoding. > > -- INVITE (Opus, PCMA) --> | > | -- INVITE (PCMA) --> > | <---- OK (PCMA) ---- > <------ OK (Opus) -------- | > . > . > | <--- ReINVITE (Hold or video on/off) > <---- ReINVITE (PCMA) ---- | > x > > I tried with absolute codec string, inbound/outbound codec prefs... it > seems that FS does not keep/honor previous channel state and does not > differentiate stream change from stream parameters change (like media flow). > > > *2nd: Dynamic Payload *- It will offer Opus to the other side, because > both have it. But FS will renegotiate with new PT number on reinvite. This > break stream for some endpoints. > > -- INVITE (Opus, 111) --> | > | -- INVITE (Opus, 102) --> > | <---- OK (Opus, 102) ---- > <---- OK (Opus, 111) ---- | > . > . > | <-- ReINVITE (Hold or new m=line) > <- ReINVITE (Opus, 102) - | > x > > Is there a way to originate call with a codec that has a dynamic payload > number, but to choose PT number? Something like setting rfc2833-pt variable > for outbound channel before originating call? > > > *3rd Media direction* - FS does not propagate media direction parameter > to inbound channel/leg. Inbound side expect media and some endpoints will > drop call after awhile. I know that answering with recvonly does not have > sense for audio, but for a video it has. > > -- INVITE (sendrecv) --> | > | -- INVITE (sendrecv) --> > | <---- OK (recvonly) ---- > <---- OK (sendrecv) ---- | > > For late negotiation this is wrong. Even for early it should be possible > to pass reinvite to the other side on media direction change. > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From avi at avimarcus.net Wed Dec 22 18:07:29 2021 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 22 Dec 2021 18:07:29 +0000 Subject: [Freeswitch-users] SIP timer at 900 seconds? Message-ID: <0100017de353a254-af281522-5043-41f5-bd9d-8f9b2487f156-000000@email.amazonses.com> I see two recent calls from FreeSWITCH that end at 15 minutes: billsec: 904.1, hangup cause: MANDATORY_IE_MISSING billsec: 901.3, hangup cause: MANDATORY_IE_MISSING My system is hanging up the call. Is there some sort of SIP timer? I don't have a SIP trace to see what happened. The remote carrier said: >However I'm unsure of what the issue is you are reporting as the BYE simply came from your side? >This is not something we are familiar with or why your server did or requires this. > Reason: SIP;cause=488;text="No answer to offer". Any suggestions? -Avi Marcus -------------- next part -------------- An HTML attachment was scrubbed... URL: From Antony.Stone at freeswitch.open.source.it Wed Dec 22 18:46:30 2021 From: Antony.Stone at freeswitch.open.source.it (Antony Stone) Date: Wed, 22 Dec 2021 19:46:30 +0100 Subject: [Freeswitch-users] SIP timer at 900 seconds? In-Reply-To: <0100017de353a254-af281522-5043-41f5-bd9d-8f9b2487f156-000000@email.amazonses.com> References: <0100017de353a254-af281522-5043-41f5-bd9d-8f9b2487f156-000000@email.amazonses.com> Message-ID: <202112221946.30583.Antony.Stone@freeswitch.open.source.it> On Wednesday 22 December 2021 at 19:07:29, Avi Marcus wrote: > I see two recent calls from FreeSWITCH that end at 15 minutes: > > billsec: 904.1, hangup cause: MANDATORY_IE_MISSING > billsec: 901.3, hangup cause: MANDATORY_IE_MISSING https://freeswitch.org/confluence/display/FREESWITCH/Hangup+Cause+Code+Table "This cause indicates that the equipment sending this cause has received a message which is missing an information element which must be present in the message before that message can be processed." > My system is hanging up the call. > > Is there some sort of SIP timer? I don't have a SIP trace to see what > happened. I think you're going to have to get a SIP trace the next time this happens, so that you can see what message FreeSwitch might have received in order to produce the above complaint. Antony. -- This space intentionally has nothing but text explaining why this space has nothing but text explaining that this space would otherwise have been left blank, and would otherwise have been left blank. Please reply to the list; please *don't* CC me. From gregor at infomedia.si Mon Dec 27 15:11:23 2021 From: gregor at infomedia.si (Gregor Nanger) Date: Mon, 27 Dec 2021 16:11:23 +0100 Subject: [Freeswitch-users] Predictive dialer Message-ID: Hi! I want to build a predictive dialer and need some advice on architecture logic. There are a lot of studies on how to calculate predictive calls, but I need most from a programming point of view. When the campaign starts, should I create a schedule to run every n seconds and check current calls, logged agents, statistics and then decide to trigger new calls? Or should I subscribe to call events and on every hangup calculate new calls? Any thoughts? BR, Gregor -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Mon Dec 27 22:46:36 2021 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 27 Dec 2021 17:46:36 -0500 Subject: [Freeswitch-users] Predictive dialer In-Reply-To: References: Message-ID: Checkout https://github.com/davidcsi/mod_dialer Should help you On Mon, 27 Dec 2021 at 10:12, Gregor Nanger wrote: > Hi! > > I want to build a predictive dialer and need some advice on architecture > logic. > > There are a lot of studies on how to calculate predictive calls, but I > need most from a programming point of view. When the campaign starts, > should I create a schedule to run every n seconds and check current calls, > logged agents, statistics and then decide to trigger new calls? Or should I > subscribe to call events and on every hangup calculate new calls? > > Any thoughts? > > BR, Gregor > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From gregor at infomedia.si Tue Dec 28 09:02:27 2021 From: gregor at infomedia.si (Gregor Nanger) Date: Tue, 28 Dec 2021 10:02:27 +0100 Subject: [Freeswitch-users] Predictive dialer In-Reply-To: References: Message-ID: Thx for this :-) I am not a C++ expert, but what I grasped, it starts to loop on the campaign and waits until a batch of calls is finished, then starts a new batch. Am I right? This is what puzzled me. Thank you 👍 On Mon, 27 Dec 2021 at 23:48, David Villasmil < david.villasmil.work at gmail.com> wrote: > Checkout > https://github.com/davidcsi/mod_dialer > > Should help you > > On Mon, 27 Dec 2021 at 10:12, Gregor Nanger wrote: > >> Hi! >> >> I want to build a predictive dialer and need some advice on architecture >> logic. >> >> There are a lot of studies on how to calculate predictive calls, but I >> need most from a programming point of view. When the campaign starts, >> should I create a schedule to run every n seconds and check current calls, >> logged agents, statistics and then decide to trigger new calls? Or should I >> subscribe to call events and on every hangup calculate new calls? >> >> Any thoughts? >> >> BR, Gregor >> _________________________________________________________________________ >> >> 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > -- > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Tue Dec 28 18:34:38 2021 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 28 Dec 2021 13:34:38 -0500 Subject: [Freeswitch-users] Predictive dialer In-Reply-To: References: Message-ID: It yields (waits) the configured time-between-calls ms, counts how many calls are running, and generates as many as needed to increase the call count to the configured max_concurrent_calls. On Tue, 28 Dec 2021 at 04:03, Gregor Nanger wrote: > Thx for this :-) > > I am not a C++ expert, but what I grasped, it starts to loop on the > campaign and waits until a batch of calls is finished, then starts a new > batch. Am I right? > > This is what puzzled me. Thank you 👍 > > On Mon, 27 Dec 2021 at 23:48, David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> Checkout >> https://github.com/davidcsi/mod_dialer >> >> Should help you >> >> On Mon, 27 Dec 2021 at 10:12, Gregor Nanger wrote: >> >>> Hi! >>> >>> I want to build a predictive dialer and need some advice on architecture >>> logic. >>> >>> There are a lot of studies on how to calculate predictive calls, but I >>> need most from a programming point of view. When the campaign starts, >>> should I create a schedule to run every n seconds and check current calls, >>> logged agents, statistics and then decide to trigger new calls? Or should I >>> subscribe to call events and on every hangup calculate new calls? >>> >>> Any thoughts? >>> >>> BR, Gregor >>> _________________________________________________________________________ >>> >>> 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 at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> -- >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 >> _________________________________________________________________________ >> >> 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From dark.pearl007 at gmail.com Thu Dec 23 08:20:31 2021 From: dark.pearl007 at gmail.com (Jenish patel) Date: Thu, 23 Dec 2021 08:20:31 -0000 Subject: [Freeswitch-users] Adding optional SDP Header in the Invite Message-ID: Hi, I am looking for a way to add the optional sdp header for the bandwidth in the invite sdp (reference *rfc3890*). I see that the video has the option to modify it with the variables rtp_video_max_bandwidth_in rtp_video_max_bandwidth but for the audio I don't see any options or reference. Regards, Jenish -------------- next part -------------- An HTML attachment was scrubbed... URL: From tony.pemberton.cowroast at gmail.com Tue Dec 28 12:55:42 2021 From: tony.pemberton.cowroast at gmail.com (Tony Pemberton) Date: Tue, 28 Dec 2021 12:55:42 -0000 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 186, Issue 39 In-Reply-To: References: Message-ID: After many years of tinkering with Freeswitch at home, this is my first posting so I might get things a bit wrong! I also have been trying to install freeswitch 1.10.7 on a 4Gb Raspberry Pi and have come across the same installation problems. I thought I read that only 64 bit builds of Freeswitch were to be supported in Debian Bullseye so I also tried the development version of RaspbianOS 64bit but got just the same dependency problems. Looking at the Repo's, Freeswitch Bullseye binaries are only available for Intel/AMD architectures at present. As a result, I have decided to compile from source using armhf (32bit) version of Raspbian Bullseye. I did have to compile sofia-sip, spandsp and links from git sources to satisfy Freeswitch's bootstrap/configure dependencies. As my domestic system does not access the Signalwire cloud, I have, for the time being, disabled the mod_signalwire in modules.conf. A compilation is in progress as I write this so I'm not yet sure I will get a clean compile. But at least I have got past the dependency issue. Regards, Tony Pemberton On Wed, 22 Dec 2021 at 12:09, wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at 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 at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at 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. debian repos for Raspberry Pi broken? (Johan Helsingius) > 2. Re: debian repos for Raspberry Pi broken? (Andrey Volk) > 3. Re: debian repos for Raspberry Pi broken? (Johan Helsingius) > > > > ---------- Forwarded message ---------- > From: Johan Helsingius > To: FreeSWITCH Users Help > Cc: > Bcc: > Date: Tue, 21 Dec 2021 20:59:31 +0100 > Subject: [Freeswitch-users] debian repos for Raspberry Pi broken? > Hi, > > Trying to install freeswitch on a RPi 4 (Debian 11 Bullseye), > my /etc/apt/sources.list.d/freeswitch.list has: > > deb http://files.freeswitch.org/repo/deb/rpi/debian-release/ bullseye main > deb-src http://files.freeswitch.org/repo/deb/rpi/debian-release/ bullseye > main > > but when I try > > apt-get update && apt-get install -y freeswitch-meta-all > > I get > > Unable to locate package freeswitch-meta-all > > apt-cache search freeswitch only gives me: > libsipwitch-dev - secure peer-to-peer SIP VoIP server - development files > libsipwitch1 - secure peer-to-peer SIP VoIP server - shared libraries > libsipwitch1-dbg - secure peer-to-peer SIP VoIP server - debug symbols > sipwitch - secure peer-to-peer VoIP server for the SIP protocol > sipwitch-cgi - secure peer-to-peer SIP VoIP server - CGI XML-RPC interface > > Any suggestions? > > Julf > > > > > > > ---------- Forwarded message ---------- > From: Andrey Volk > To: FreeSWITCH Users Help > Cc: > Bcc: > Date: Tue, 21 Dec 2021 23:36:21 +0300 > Subject: Re: [Freeswitch-users] debian repos for Raspberry Pi broken? > Let me see what I can do. May take some time. > > вт, 21 дек. 2021 г. в 23:19, Johan Helsingius : > >> Hi, >> >> Trying to install freeswitch on a RPi 4 (Debian 11 Bullseye), >> my /etc/apt/sources.list.d/freeswitch.list has: >> >> deb http://files.freeswitch.org/repo/deb/rpi/debian-release/ bullseye >> main >> deb-src http://files.freeswitch.org/repo/deb/rpi/debian-release/ >> bullseye main >> >> but when I try >> >> apt-get update && apt-get install -y freeswitch-meta-all >> >> I get >> >> Unable to locate package freeswitch-meta-all >> >> apt-cache search freeswitch only gives me: >> libsipwitch-dev - secure peer-to-peer SIP VoIP server - development files >> libsipwitch1 - secure peer-to-peer SIP VoIP server - shared libraries >> libsipwitch1-dbg - secure peer-to-peer SIP VoIP server - debug symbols >> sipwitch - secure peer-to-peer VoIP server for the SIP protocol >> sipwitch-cgi - secure peer-to-peer SIP VoIP server - CGI XML-RPC interface >> >> Any suggestions? >> >> Julf >> >> >> _________________________________________________________________________ >> >> 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > > > ---------- Forwarded message ---------- > From: Johan Helsingius > To: freeswitch-users at lists.freeswitch.org > Cc: > Bcc: > Date: Tue, 21 Dec 2021 22:11:37 +0100 > Subject: Re: [Freeswitch-users] debian repos for Raspberry Pi broken? > On 21-12-2021 21:36, Andrey Volk wrote: > > Let me see what I can do. May take some time. > > Thanks! > > Julf > > > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: