From alex at freeswitch.com Thu Mar 1 00:20:40 2018 From: alex at freeswitch.com (Alexey Sibyakin) Date: Thu, 1 Mar 2018 09:20:40 +0900 Subject: [Freeswitch-users] Fax Transmit Rate - force to 9600? In-Reply-To: References: Message-ID: Hi, You can try fax_transfer_rate var https://freeswitch.org/confluence/display/FREESWITCH/Variables or configure spandsp https://freeswitch.org/confluence/display/FREESWITCH/mod_spandsp On Thu, Mar 1, 2018 at 8:39 AM, Andrew wrote: > Thanks Vallimamod, that gives me some additional info we weren't aware of > (re: preamble and t38 max bitrate). > > > I noticed you mentioned disabling v17; which would allow the bitrate to > increase to 14400, which is where we're seeing the problem. We want to > force/limit the max bitrate under all circumstances (t38 or t30 over g711) > to 9600, which we know works. > > Some additional info; > > Doing some further digging; the problem doesn't appear to specifically lie > with t38; from what we can see some fax calls come in via t30 over g711 > which also fail with a bit rate of 14400. > > Given the preamble info you mentioned, could it be we're sending the > sending fax our capabilities (i.e. max bitrate = 9600) and it's ignoring > this and sends it across at 14400? I would assume that most, if not all > faxes have the capability to send at 9600 if requested? > > Given our issue relates to incoming (i.e we're the receiving end), is > there anything else we can do within FS to force the lower bit rate? > > cheers, > A > > > > ------------------------------ > *From:* FreeSWITCH-users > on behalf of Vallimamod Abdullah > *Sent:* Thursday, 1 March 2018 4:34 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Fax Transmit Rate - force to 9600? > > Hi, > > From the error message, it looks like the remote fax drops the call when > it sees no match between capabilities (it's the receiving fax who sends the > preamble with its capabilities first.) > According to the source code, with v17 disabled, the max bitrate over t38 > is 9600, else it's 14400. So I guess you need to enable v17 to communicate > with faxes with min bitrates greater than 9600. > > Best Regards, > -- > Vallimamod Abdullah > SIP Solutions > vma at sipsolutions.fr > . > > On 28 Feb 2018, at 02:32, Andrew wrote: > > Hi all, > > We've been trying to get our fax implementation as reliable as possible > using FS; we've had some great results with inbound faxes with the > following items set in the incoming dial-plan: > > action application ="answer"/> > action application="playback" data="silence_stream://2000"/> > action set ignore_early_media=true > action set absolute_codec_string='PCMU,PCMA' > action set fax_enable_t38=true > action set fax_verbose=true > action set fax_use_ecm=true > action set disable-v17=true > action set fax_v17_disabled=true > action set fax_disable_v17=true > action set fax_enable_t38_request=true > application rxfax /tmp/FAX-IN-${uuid}.tif > > application hangup > > The one thing that we have identified with this configuration is that 99% > of inbound faxes come through fine when the fax transfer rate is negotiated > at 9600 bps (as a result of thefax_v17_disabled=true setting). > > However, we still see quite a few incoming faxes reported as coming in > with a fax transfer rate of 14400 bps, which have a 100% fail rate. > > I understand that the two ends need to negotiate a rate; but how do we > force (or only offer) the rate of 9600 or lower? My understanding is that the > fax_v17_disabled=true setting should cover that? My first thought was that > the remote end was forcing the rate at 14400, regardless of what we were > sending back in terms of fax capabilities. > > Not sure if it helps, but the two error messages we receive when faxes > fail due to the 14400 rate are: > > - Unexpected DCN while waiting for DCS or DIS, and > - The call dropped prematurely > > My thought was, If we could find a way to get all faxes coming in at 9600, > we'd be able to improve the reliability. > > Looking through the wiki and previous list messages, the > fax_disable_v17=true setting seems to be the documented approach, but I'm > not sure why we're still seeing faxes come in at a higher rate. > > Any advice on this or are we missing something with regards to locking in > the lower speed? > > cheers, > A > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > > FreeSWITCH > www.freeswitch.org > FreeSWITCH is an open-source media application designed to support popular > protools such as SIP and WebRTC and provides a platform to develop voice > and video applications. > > http://confluence.freeswitch.org > http://www.cluecon.com > > ClueCon Telephony and WebRTC Developer's conference > > www.cluecon.com > ClueCon is an annual event bringing together all of the open source > Telephony and WebRTC developers to collaberate on the latest technology in > the communications industry. > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > FreeSWITCH-users Info Page > > lists.freeswitch.org > To see the collection of prior postings to the list, visit the > FreeSWITCH-users Archives. Using FreeSWITCH-users: To post a message to all > the list ... > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > FreeSWITCH-users list: member options login page > > lists.freeswitch.org > In order to change your membership option, you must first log in by giving > your email address and membership password in the section below. > > http://www.freeswitch.org > > FreeSWITCH > www.freeswitch.org > FreeSWITCH is an open-source media application designed to support popular > protools such as SIP and WebRTC and provides a platform to develop voice > and video applications. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > -- Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: alex at freeswitch.com Website: https://www.FreeSWITCH.com Need commercial support? Contact sales at freeswitch.com for details. -------------- next part -------------- An HTML attachment was scrubbed... URL: From asims1979 at hotmail.com Thu Mar 1 07:26:29 2018 From: asims1979 at hotmail.com (Andrew) Date: Thu, 1 Mar 2018 07:26:29 +0000 Subject: [Freeswitch-users] Fax Transmit Rate - force to 9600? In-Reply-To: <48681EF6-1692-415D-B525-4CF366820DA7@jerris.com> References: , <48681EF6-1692-415D-B525-4CF366820DA7@jerris.com> Message-ID: Thanks Michael, that's what we observed with t38 too, however here's a t38 curve ball I've come across; I looked at a few recent failed incoming faxes that used t38. As an example, all showed the following characteristics. The SDP info suggests MaxBitRate should be 9600 via t38, which is in-line with Vallimamod's comments earlier: t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T 38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:tr ansferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC :t38UDPRedundancy However, the fax_transfer_rate variable shows 14400 as the speed. I'm just not sure how that could be possible if t38 sets the max rate at 9600? ________________________________ From: FreeSWITCH-users on behalf of Michael Jerris Sent: Thursday, 1 March 2018 10:53 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Fax Transmit Rate - force to 9600? its not uncommon for 14400 faxes to fail over g711. should have better luck with t38. On Feb 28, 2018, at 6:39 PM, Andrew > wrote: Thanks Vallimamod, that gives me some additional info we weren't aware of (re: preamble and t38 max bitrate). I noticed you mentioned disabling v17; which would allow the bitrate to increase to 14400, which is where we're seeing the problem. We want to force/limit the max bitrate under all circumstances (t38 or t30 over g711) to 9600, which we know works. Some additional info; Doing some further digging; the problem doesn't appear to specifically lie with t38; from what we can see some fax calls come in via t30 over g711 which also fail with a bit rate of 14400. Given the preamble info you mentioned, could it be we're sending the sending fax our capabilities (i.e. max bitrate = 9600) and it's ignoring this and sends it across at 14400? I would assume that most, if not all faxes have the capability to send at 9600 if requested? Given our issue relates to incoming (i.e we're the receiving end), is there anything else we can do within FS to force the lower bit rate? cheers, A ________________________________ From: FreeSWITCH-users > on behalf of Vallimamod Abdullah > Sent: Thursday, 1 March 2018 4:34 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Fax Transmit Rate - force to 9600? Hi, >From the error message, it looks like the remote fax drops the call when it sees no match between capabilities (it's the receiving fax who sends the preamble with its capabilities first.) According to the source code, with v17 disabled, the max bitrate over t38 is 9600, else it's 14400. So I guess you need to enable v17 to communicate with faxes with min bitrates greater than 9600. Best Regards, -- Vallimamod Abdullah SIP Solutions vma at sipsolutions.fr . On 28 Feb 2018, at 02:32, Andrew > wrote: Hi all, We've been trying to get our fax implementation as reliable as possible using FS; we've had some great results with inbound faxes with the following items set in the incoming dial-plan: action application ="answer"/> action application="playback" data="silence_stream://2000"/> action set ignore_early_media=true action set absolute_codec_string='PCMU,PCMA' action set fax_enable_t38=true action set fax_verbose=true action set fax_use_ecm=true action set disable-v17=true action set fax_v17_disabled=true action set fax_disable_v17=true action set fax_enable_t38_request=true application rxfax /tmp/FAX-IN-${uuid}.tif application hangup The one thing that we have identified with this configuration is that 99% of inbound faxes come through fine when the fax transfer rate is negotiated at 9600 bps (as a result of thefax_v17_disabled=true setting). However, we still see quite a few incoming faxes reported as coming in with a fax transfer rate of 14400 bps, which have a 100% fail rate. I understand that the two ends need to negotiate a rate; but how do we force (or only offer) the rate of 9600 or lower? My understanding is that the fax_v17_disabled=true setting should cover that? My first thought was that the remote end was forcing the rate at 14400, regardless of what we were sending back in terms of fax capabilities. Not sure if it helps, but the two error messages we receive when faxes fail due to the 14400 rate are: - Unexpected DCN while waiting for DCS or DIS, and - The call dropped prematurely My thought was, If we could find a way to get all faxes coming in at 9600, we'd be able to improve the reliability. Looking through the wiki and previous list messages, the fax_disable_v17=true setting seems to be the documented approach, but I'm not sure why we're still seeing faxes come in at a higher rate. Any advice on this or are we missing something with regards to locking in the lower speed? cheers, A _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org [https://freeswitch.org/theme/img/freeswitch-social.png] FreeSWITCH www.freeswitch.org [https://freeswitch.org/theme/img/freeswitch-social.png] FreeSWITCH www.freeswitch.org FreeSWITCH is an open-source media application designed to support popular protools such as SIP and WebRTC and provides a platform to develop voice and video applications. FreeSWITCH is an open-source media application designed to support popular protools such as SIP and WebRTC and provides a platform to develop voice and video applications. http://confluence.freeswitch.org http://www.cluecon.com [https://cluecon.com/theme/img/cluecon/hook1.jpg] ClueCon Telephony and WebRTC Developer's conference www.cluecon.com [https://cluecon.com/theme/img/cluecon/hook1.jpg] ClueCon Telephony and WebRTC Developer's conference www.cluecon.com ClueCon is an annual event bringing together all of the open source Telephony and WebRTC developers to collaberate on the latest technology in the communications industry. ClueCon is an annual event bringing together all of the open source Telephony and WebRTC developers to collaberate on the latest technology in the communications industry. FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users FreeSWITCH-users Info Page lists.freeswitch.org lists.freeswitch.org Mailing Lists lists.freeswitch.org lists.freeswitch.org Mailing Lists: Welcome! Below is a listing of all the public mailing lists on lists.freeswitch.org. Click on a list name to get more ... To see the collection of prior postings to the list, visit the FreeSWITCH-users Archives. Using FreeSWITCH-users: To post a message to all the list ... UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users FreeSWITCH-users list: member options login page lists.freeswitch.org lists.freeswitch.org Mailing Lists lists.freeswitch.org lists.freeswitch.org Mailing Lists: Welcome! Below is a listing of all the public mailing lists on lists.freeswitch.org. Click on a list name to get more ... In order to change your membership option, you must first log in by giving your email address and membership password in the section below. http://www.freeswitch.org [https://freeswitch.org/theme/img/freeswitch-social.png] FreeSWITCH www.freeswitch.org [https://freeswitch.org/theme/img/freeswitch-social.png] FreeSWITCH www.freeswitch.org FreeSWITCH is an open-source media application designed to support popular protools such as SIP and WebRTC and provides a platform to develop voice and video applications. FreeSWITCH is an open-source media application designed to support popular protools such as SIP and WebRTC and provides a platform to develop voice and video applications. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org [https://freeswitch.org/theme/img/freeswitch-social.png] FreeSWITCH www.freeswitch.org FreeSWITCH is an open-source media application designed to support popular protools such as SIP and WebRTC and provides a platform to develop voice and video applications. http://confluence.freeswitch.org http://www.cluecon.com [https://cluecon.com/theme/img/cluecon/hook1.jpg] ClueCon Telephony and WebRTC Developer's conference www.cluecon.com ClueCon is an annual event bringing together all of the open source Telephony and WebRTC developers to collaberate on the latest technology in the communications industry. FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users FreeSWITCH-users Info Page lists.freeswitch.org To see the collection of prior postings to the list, visit the FreeSWITCH-users Archives. Using FreeSWITCH-users: To post a message to all the list ... UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users FreeSWITCH-users list: member options login page lists.freeswitch.org In order to change your membership option, you must first log in by giving your email address and membership password in the section below. http://www.freeswitch.org [https://freeswitch.org/theme/img/freeswitch-social.png] FreeSWITCH www.freeswitch.org FreeSWITCH is an open-source media application designed to support popular protools such as SIP and WebRTC and provides a platform to develop voice and video applications. -------------- next part -------------- An HTML attachment was scrubbed... URL: From asims1979 at hotmail.com Thu Mar 1 07:37:25 2018 From: asims1979 at hotmail.com (Andrew) Date: Thu, 1 Mar 2018 07:37:25 +0000 Subject: [Freeswitch-users] Fax Transmit Rate - force to 9600? In-Reply-To: References: , Message-ID: Hi Alexy, When setting: fax_transfer_rate=9600 The result is: 2018-03-01 18:31:24.986610 [WARNING] mod_spandsp.c:667 Unknown parameter fax_transfer_rate That was set in the dialplan, not spandsp config - does it make a difference where its set? I didn't see it in the list of settings for mod_spandsp; only as a result variable. cheers, Andrew ________________________________ From: FreeSWITCH-users on behalf of Alexey Sibyakin Sent: Thursday, 1 March 2018 11:20 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Fax Transmit Rate - force to 9600? Hi, You can try fax_transfer_rate var https://freeswitch.org/confluence/display/FREESWITCH/Variables Variables - FreeSWITCH - Confluence freeswitch.org Variable References. You will see references to variables in the dialplan of the form ${variable} as well as $${variable}. $${variable} is evaluated once and becomes ... or configure spandsp https://freeswitch.org/confluence/display/FREESWITCH/mod_spandsp mod_spandsp - FreeSWITCH - Confluence freeswitch.org mod_spandsp is enabled by default in modules.conf and therefore compiled automatically. Its also enabled by default in modules.conf.xml. Ensure that you have libtiff ... On Thu, Mar 1, 2018 at 8:39 AM, Andrew > wrote: Thanks Vallimamod, that gives me some additional info we weren't aware of (re: preamble and t38 max bitrate). I noticed you mentioned disabling v17; which would allow the bitrate to increase to 14400, which is where we're seeing the problem. We want to force/limit the max bitrate under all circumstances (t38 or t30 over g711) to 9600, which we know works. Some additional info; Doing some further digging; the problem doesn't appear to specifically lie with t38; from what we can see some fax calls come in via t30 over g711 which also fail with a bit rate of 14400. Given the preamble info you mentioned, could it be we're sending the sending fax our capabilities (i.e. max bitrate = 9600) and it's ignoring this and sends it across at 14400? I would assume that most, if not all faxes have the capability to send at 9600 if requested? Given our issue relates to incoming (i.e we're the receiving end), is there anything else we can do within FS to force the lower bit rate? cheers, A ________________________________ From: FreeSWITCH-users > on behalf of Vallimamod Abdullah > Sent: Thursday, 1 March 2018 4:34 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Fax Transmit Rate - force to 9600? Hi, >From the error message, it looks like the remote fax drops the call when it sees no match between capabilities (it's the receiving fax who sends the preamble with its capabilities first.) According to the source code, with v17 disabled, the max bitrate over t38 is 9600, else it's 14400. So I guess you need to enable v17 to communicate with faxes with min bitrates greater than 9600. Best Regards, -- Vallimamod Abdullah SIP Solutions vma at sipsolutions.fr . On 28 Feb 2018, at 02:32, Andrew > wrote: Hi all, We've been trying to get our fax implementation as reliable as possible using FS; we've had some great results with inbound faxes with the following items set in the incoming dial-plan: action application ="answer"/> action application="playback" data="silence_stream://2000"/> action set ignore_early_media=true action set absolute_codec_string='PCMU,PCMA' action set fax_enable_t38=true action set fax_verbose=true action set fax_use_ecm=true action set disable-v17=true action set fax_v17_disabled=true action set fax_disable_v17=true action set fax_enable_t38_request=true application rxfax /tmp/FAX-IN-${uuid}.tif application hangup The one thing that we have identified with this configuration is that 99% of inbound faxes come through fine when the fax transfer rate is negotiated at 9600 bps (as a result of thefax_v17_disabled=true setting). However, we still see quite a few incoming faxes reported as coming in with a fax transfer rate of 14400 bps, which have a 100% fail rate. I understand that the two ends need to negotiate a rate; but how do we force (or only offer) the rate of 9600 or lower? My understanding is that the fax_v17_disabled=true setting should cover that? My first thought was that the remote end was forcing the rate at 14400, regardless of what we were sending back in terms of fax capabilities. Not sure if it helps, but the two error messages we receive when faxes fail due to the 14400 rate are: - Unexpected DCN while waiting for DCS or DIS, and - The call dropped prematurely My thought was, If we could find a way to get all faxes coming in at 9600, we'd be able to improve the reliability. Looking through the wiki and previous list messages, the fax_disable_v17=true setting seems to be the documented approach, but I'm not sure why we're still seeing faxes come in at a higher rate. Any advice on this or are we missing something with regards to locking in the lower speed? cheers, A _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org [https://freeswitch.org/theme/img/freeswitch-social.png] FreeSWITCH www.freeswitch.org [https://freeswitch.org/theme/img/freeswitch-social.png] FreeSWITCH www.freeswitch.org FreeSWITCH is an open-source media application designed to support popular protools such as SIP and WebRTC and provides a platform to develop voice and video applications. FreeSWITCH is an open-source media application designed to support popular protools such as SIP and WebRTC and provides a platform to develop voice and video applications. http://confluence.freeswitch.org http://www.cluecon.com [https://cluecon.com/theme/img/cluecon/hook1.jpg] ClueCon Telephony and WebRTC Developer's conference www.cluecon.com ClueCon is an annual event bringing together all of the open source Telephony and WebRTC developers to collaberate on the latest technology in the communications industry. FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users FreeSWITCH-users Info Page lists.freeswitch.org To see the collection of prior postings to the list, visit the FreeSWITCH-users Archives. Using FreeSWITCH-users: To post a message to all the list ... UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users FreeSWITCH-users list: member options login page lists.freeswitch.org In order to change your membership option, you must first log in by giving your email address and membership password in the section below. http://www.freeswitch.org [https://freeswitch.org/theme/img/freeswitch-social.png] FreeSWITCH www.freeswitch.org FreeSWITCH is an open-source media application designed to support popular protools such as SIP and WebRTC and provides a platform to develop voice and video applications. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.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 http://www.freeswitch.org -- [https://docs.google.com/uc?export=download&id=1xswZRZyVDo0WQhaemK47pU266yzDRmi0&revid=0B2xnT7i45ngrMTVKM1dpSHZIN28zU0QzbW9xeVF6RXFyRHhBPQ] Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: alex at freeswitch.com Website: https://www.FreeSWITCH.com Need commercial support? Contact sales at freeswitch.com for details. -------------- next part -------------- An HTML attachment was scrubbed... URL: From alex at freeswitch.com Thu Mar 1 08:20:20 2018 From: alex at freeswitch.com (Alexey Sibyakin) Date: Thu, 1 Mar 2018 17:20:20 +0900 Subject: [Freeswitch-users] DTMF negotiation In-Reply-To: <510252213.6154862.1519830130424@mail.yahoo.com> References: <510252213.6154862.1519830130424.ref@mail.yahoo.com> <510252213.6154862.1519830130424@mail.yahoo.com> Message-ID: Hi, You need in dialplan and probably in sofia profile Alex On Thu, Mar 1, 2018 at 12:02 AM, kaiduan xie wrote: > Hi, > > We are encountering the following DTMF issue, > > FreeSwitch adds telephone-event support in INVITE to SBC, SBC replies back > without telephone-event support in 200. > > However FreeSwitch still sends out DTMF in RFC 2833 event. > > I think FreeSwitch should send out DTMF inband instead of 2833 event, what > is the configuration for this? > > Many thanks for help, > > /Kaiduan > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > -- Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: alex at freeswitch.com Website: https://www.FreeSWITCH.com Need commercial support? Contact sales at freeswitch.com for details. -------------- next part -------------- An HTML attachment was scrubbed... URL: From alexandr.popov at iqoption.com Thu Mar 1 09:21:07 2018 From: alexandr.popov at iqoption.com (Alexandr Popov) Date: Thu, 1 Mar 2018 11:21:07 +0200 Subject: [Freeswitch-users] javascript scheduling multiple events excecute_on_answer In-Reply-To: <01bc01d3b070$0f3b0bf0$2db123d0$@delagarda.com> References: <01bc01d3b070$0f3b0bf0$2db123d0$@delagarda.com> Message-ID: try execute_on_answer_1 execute_on_answer_2 or set leg_timeout=120 and remove first execute_on_answer 2018-02-28 10:42 GMT+02:00 Francesco Facco de Lagarda < francesco at delagarda.com>: > I am developing a calling platform using javascript. > > > > I have code that calculates how long THAT user is allowed to call THAT > number for. > > > > I am trying to schedule 2 events, > > 1. 1 min before time ends, that plays a message “you have 1 minute > left ..” > 2. The actual hangup when the time expires.. > > > > Despite a zillion tests I have been unable to schedule BOTH events.. > > > > This is my code: for simplicity’s sake I have set call time to 120 secs, > with warning at 60: > > > > if (session.ready()) { > > > > /*** > > Get user, number, etc… code omitted for simplicity > > **/ > > > > var sessOut = new Session("sofia/gateway/ht503/" + dialedNum + "@ > 192.168.0.201:5062"); > > var totTime = 60; > > sessOut.execute("set", "execute_on_answer=sched_hangup +120 > alloted_timeout") > > sessOut.execute("set", "execute_on_answer=sched_broadcast +60 > playback::" + soundDir + "one_min_left.wav both"); > > > > if (sessOut.ready()) { > > bridge(session, sessOut); > > } > > sessOut.hangup(); > > session.hangup(); > > } > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From tahir at ictinnovations.com Thu Mar 1 11:11:12 2018 From: tahir at ictinnovations.com (Tahir Almas) Date: Thu, 1 Mar 2018 16:11:12 +0500 Subject: [Freeswitch-users] ICTDialer , A freeswitch based unified autodialer software featuring sms, fax and voice broadcasting Message-ID: We are pleased to annouse release of ICTDialer Version 3.0 with complete new design based on open source communication development framework ICTCore http://www.ictcore.org and freeswitch communication engine http://www.freeswitch.org regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT -------------- next part -------------- An HTML attachment was scrubbed... URL: From tahir at ictinnovations.com Thu Mar 1 11:13:56 2018 From: tahir at ictinnovations.com (Tahir Almas) Date: Thu, 1 Mar 2018 16:13:56 +0500 Subject: [Freeswitch-users] ICTDialer , A freeswitch based unified autodialer software featuring sms, fax and voice broadcasting In-Reply-To: References: Message-ID: Here is link to ICTDialer website http://www.ictdialer.org *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT * On Thu, Mar 1, 2018 at 4:11 PM, Tahir Almas wrote: > We are pleased to annouse release of ICTDialer Version 3.0 with > complete new design based on open source communication development > framework ICTCore http://www.ictcore.org and freeswitch communication > engine http://www.freeswitch.org > > regards > *Tahir Almas* > > Managing Partner > ICT Innovations > http://www.ictinnovations.com > Leveraging open source in ICT > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From kaiduanx at yahoo.ca Thu Mar 1 15:39:49 2018 From: kaiduanx at yahoo.ca (kaiduan xie) Date: Thu, 1 Mar 2018 15:39:49 +0000 (UTC) Subject: [Freeswitch-users] DTMF negotiation In-Reply-To: References: <510252213.6154862.1519830130424.ref@mail.yahoo.com> <510252213.6154862.1519830130424@mail.yahoo.com> Message-ID: <188910919.6866630.1519918789572@mail.yahoo.com> Alex, Thanks for the help. With in sofia profile, the offer from FreeSwitch does not add telephone event support. With , FreeSwitch always sends DTMF inband even the offer from FreeSwitch and answer both supports telephone event. What I want is DTMF negotiation as below, FreeSwitch always adds telephone event support in offer, 1. If answer supports telephone event, then FreeSwitch sends DTMF in rfc2833.2. If answer does not support telephone event, then FreeSwitch sends DTMF inband. Is it possible? Thanks again. /Kaiduan On Thursday, March 1, 2018 3:22 AM, Alexey Sibyakin wrote: Hi, You need in dialplan and probably in sofia profile Alex On Thu, Mar 1, 2018 at 12:02 AM, kaiduan xie wrote: Hi, We are encountering the following DTMF issue, FreeSwitch adds telephone-event support in INVITE to SBC, SBC replies back without telephone-event support in 200.  However FreeSwitch still sends out DTMF in RFC 2833 event. I think FreeSwitch should send out DTMF inband instead of 2833 event, what is the configuration for this? Many thanks for help, /Kaiduan ______________________________ ______________________________ _____________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www. freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch. org http://www.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 http://www.freeswitch.org -- Alex Sibyakin | Support EngineerFreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045Email: alex at freeswitch.comWebsite: https://www.FreeSWITCH.comNeed commercial support? Contact sales at freeswitch.com for details. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.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 http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From francesco at delagarda.com Thu Mar 1 10:14:30 2018 From: francesco at delagarda.com (Francesco Facco de Lagarda) Date: Thu, 1 Mar 2018 11:14:30 +0100 Subject: [Freeswitch-users] javascript scheduling multiple events excecute_on_answer In-Reply-To: References: <01bc01d3b070$0f3b0bf0$2db123d0$@delagarda.com> Message-ID: <9942B9BD-938B-40B6-9FB7-C719CC419172@delagarda.com> Alexandr sounds interesting but I’m very new to this! Can you please be more explicit? Do you mean: >> sessOut.execute("set", "execute_on_answer_1=sched_hangup +120 alloted_timeout") >> >> sessOut.execute("set", "execute_on_answer_2=sched_broadcast +60 playback::" + soundDir + "one_min_left.wav both"); >> >> Ie just adding _1 and _2 tor this two lines? >> > Francesco Facco de Lagarda > On 1 Mar 2018, at 10:21, Alexandr Popov wrote: > > try > execute_on_answer_1 > execute_on_answer_2 > or set > leg_timeout=120 > and remove first execute_on_answer > > 2018-02-28 10:42 GMT+02:00 Francesco Facco de Lagarda : >> I am developing a calling platform using javascript. >> >> >> >> I have code that calculates how long THAT user is allowed to call THAT number for. >> >> >> >> I am trying to schedule 2 events, >> >> 1 min before time ends, that plays a message “you have 1 minute left ..” >> The actual hangup when the time expires.. >> >> >> Despite a zillion tests I have been unable to schedule BOTH events.. >> >> >> >> This is my code: for simplicity’s sake I have set call time to 120 secs, with warning at 60: >> >> >> >> if (session.ready()) { >> >> >> >> /*** >> >> Get user, number, etc… code omitted for simplicity >> >> **/ >> >> >> >> var sessOut = new Session("sofia/gateway/ht503/" + dialedNum + "@192.168.0.201:5062"); >> >> var totTime = 60; >> >> if (sessOut.ready()) { >> >> bridge(session, sessOut); >> >> } >> >> sessOut.hangup(); >> >> session.hangup(); >> >> } >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.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 >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From stefano.favaro at edistar.com Thu Mar 1 16:24:47 2018 From: stefano.favaro at edistar.com (Stefano Favaro) Date: Thu, 1 Mar 2018 17:24:47 +0100 (CET) Subject: [Freeswitch-users] Send RTP to external server In-Reply-To: Message-ID: <1760115799.197.1519921479661.JavaMail.sfa@EDISTAR-SFA> Hello, I'm currently using: Freeswitch with mod_unimrcp and the plugin from UniMRCP ( http://unimrcp.org/gsr) It connects to Google Speech API for audio transcription in realtime using mrcp. Stefano. ----- Messaggio originale ----- Da: "Robert Mundkowsky" A: "FreeSWITCH Users Help" Inviato: Mercoledì, 28 febbraio 2018 20:17:01 Oggetto: Re: [Freeswitch-users] Send RTP to external server It is actually fairly common to want to stream audio/video to somewhere to process the data in real time, rather than waiting until recording is finished. For example, if want to convert audio to text (ASR) then you do not want to wait until the conference is over before you start the ASR. Imagine, your ASR takes 1 minute to convert 1 minute of audio then if a conference is 30 minutes long, you would have to wait another 30 minutes for the ASR to finish before you could do something with the text. I believe you can use gstreamer ( https://gstreamer.freedesktop.org/ ) to handle receiving the RTP from a FreeSWITCH conference. I think it allows you to sample frames if you want to get frames from video granted I think you are only interested in audio, but I think you still have to develop a daemon that understand SIP/RTP in order to talk to FreeSWITCH. Another approach might be to create a plugin for the Unimrcp (MRCPv2 server http://www.unimrcp.org/ ) and talk to the FreeSWITCH using their mrcp module and tight that to the conference call. You might be able to hack the Kaldi plugin for Unimrcp to do what you want. Robert From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tihomir Culjaga Sent: Wednesday, February 28, 2018 1:47 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Send RTP to external server why do you want to stream... isn't it enough just to rsync the file once the call is finished ? On 28 February 2018 at 16:16, Mickael Hubert < mickael at winlux.fr > wrote: Hi thanks a lot for your answer. But I want to send stream to external server, not record in audio file. i can use record to "capture" the voice, but to stream it, it's more complicated ;) thanks in advance 2018-02-21 0:50 GMT+01:00 Brian West < brian at freeswitch.com >:
you can already do this without SIPREC in freeswitch. By setting the RECORD_READ_ONLY or RECORD_WRITE_ONLY variables. /b On Fri, Feb 16, 2018 at 10:33 AM, Mickael Hubert < mickael at winlux.fr > wrote:
Hi list, I want to record each call through freeswitch. But i want record only caller (SSRC 1) OR callee (SSRC 2) voice (not both). I read about SIPREC, Jack, etc ... not interesting Do you have a idea for me please ? Thanks in advance _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.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 http://www.freeswitch.org -- 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 color-facebook-96.pngcolor-twitter-96.png _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.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 http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.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 http://www.freeswitch.org
This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.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 http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From domrumsey at hotmail.com Thu Mar 1 21:02:53 2018 From: domrumsey at hotmail.com (Dom Rumsey) Date: Thu, 1 Mar 2018 21:02:53 +0000 Subject: [Freeswitch-users] Conference mux mode video flickering In-Reply-To: <35A8243B-2258-4ADE-9CF2-F360F72BCFDA@jerris.com> References: <095A5D88-7540-4675-A9C2-F9E0B7028AEB@jerris.com> , <35A8243B-2258-4ADE-9CF2-F360F72BCFDA@jerris.com> Message-ID: So I've tested again and this time it looks like individual cores are maxing out (it's showing 332%), which it wasn't doing previously. I only have two users and since switching to MUX I have a decent processor. [X] ________________________________ From: FreeSWITCH-users on behalf of Michael Jerris Sent: Wednesday, February 28, 2018 5:57 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Conference mux mode video flickering look at individual cores, are any maxing out? On Feb 28, 2018, at 5:07 AM, Dom Rumsey > wrote: No it doesn’t. So in a 3 way chat, for the person speaking, it shows their own video feed back to themself. The other two people (who are listening) see a flicker of the video feeds from both the speaker and the other listener. It's as if there's a paradox, I just can't work out what it is. Looked at CPU and we still have +70% left. Network is also fine, got loads of bandwidth left. Thanks On Feb 27, 2018, at 5:01 PM, Michael Jerris > wrote: only other thing i can think of would be some sort of resource starvation, cpu or network. Does it do the same without personal canvas? On Feb 27, 2018, at 11:53 AM, Dom Rumsey < domrumsey at hotmail.com> wrote: Thanks Mike. I'm using 1.6.20. ________________________________ From: FreeSWITCH-users > on behalf of Michael Jerris > Sent: Tuesday, February 27, 2018 4:24 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Conference mux mode video flickering The flickering sounds like an old fixed bug we had with overlap and zoom layouts. Are you using old code? On Feb 26, 2018, at 3:57 PM, Dom Rumsey < domrumsey at hotmail.com> wrote: Hi guys Thanks for your help previously. I'm now using MUX mode for video and I'm getting a problem whereby a conference with 3 or more participants keeps flickering video streams on the floor (we are using video-muxing-personal-canvas parameter). It's as if it can't decide who to show. Below is my conference profile: < profile name = "cp" > < param name = "domain" value = "$${domain}" /> < param name = "rate" value = "8000" /> < param name = "video-mode" value = "mux" /> < param name = "video-layout-name" value = "1x1" /> < param name = "interval" value = "20" /> < param name = "caller-controls" value = "default" /> < param name = "energy-level" value = "0" /> < param name = "video-auto-floor-msec" value = "3000" /> < param name = "video_no_video_avatar_png" value = "/usr/local/freeswitch/conf/images/video-muted.png" /> < param name = "video_mute_png" value = "/usr/local/freeswitch/conf/images/video-muted.png" /> < param name = "conference-flags" value = "audio-always|livearray-sync|livearray-json-status|video-mute-exit-canvas|video-muxing-personal-canvas" /> < param name = "max-members" value = "25" /> < param name = "sound-prefix" value = "/usr/local/freeswitch/conf/sounds/" /> < param name = "enter-sound" value = "tone_stream://%(200,0,500,600,700)" /> < param name = "exit-sound" value = "tone_stream://%(500,0,300,200,100,50,25)" /> < param name = "inbound-late-negotiation" value = "false" /> Any pointers on where I'm going wrong would be really appreciated. Thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: From mickael at winlux.fr Fri Mar 2 10:09:31 2018 From: mickael at winlux.fr (Mickael Hubert) Date: Fri, 2 Mar 2018 11:09:31 +0100 Subject: [Freeswitch-users] Send RTP to external server In-Reply-To: <1760115799.197.1519921479661.JavaMail.sfa@EDISTAR-SFA> References: <1760115799.197.1519921479661.JavaMail.sfa@EDISTAR-SFA> Message-ID: Hi thanks a lot for your answers. Do you split media stream ? I want to stream only caller voice for exemple. My ASR works better if it recognizes only one stream ;) thanks in advance 2018-03-01 17:24 GMT+01:00 Stefano Favaro : > > Hello, I'm currently using: > > > Freeswitch with mod_unimrcp and the plugin from UniMRCP ( > http://unimrcp.org/gsr) > > > It connects to Google Speech API for audio transcription in realtime using > mrcp. > > > Stefano. > > > > > > ------------------------------ > *Da: *"Robert Mundkowsky" > *A: *"FreeSWITCH Users Help" > *Inviato: *Mercoledì, 28 febbraio 2018 20:17:01 > *Oggetto: *Re: [Freeswitch-users] Send RTP to external server > > > It is actually fairly common to want to stream audio/video to somewhere to > process the data in real time, rather than waiting until recording is > finished. > > > > For example, if want to convert audio to text (ASR) then you do not want > to wait until the conference is over before you start the ASR. Imagine, > your ASR takes 1 minute to convert 1 minute of audio then if a conference > is 30 minutes long, you would have to wait another 30 minutes for the ASR > to finish before you could do something with the text. > > > > I believe you can use gstreamer (https://gstreamer.freedesktop.org/) to > handle receiving the RTP from a FreeSWITCH conference. I think it allows > you to sample frames if you want to get frames from video granted I think > you are only interested in audio, but I think you still have to develop a > daemon that understand SIP/RTP in order to talk to FreeSWITCH. Another > approach might be to create a plugin for the Unimrcp (MRCPv2 server > http://www.unimrcp.org/ ) and talk to the FreeSWITCH using their mrcp > module and tight that to the conference call. You might be able to hack > the Kaldi plugin for Unimrcp to do what you want. > > > > Robert > > > > *From:* FreeSWITCH-users [mailto:freeswitch-users- > bounces at lists.freeswitch.org] *On Behalf Of *Tihomir Culjaga > *Sent:* Wednesday, February 28, 2018 1:47 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Send RTP to external server > > > > why do you want to stream... isn't it enough just to rsync the file once > the call is finished ? > > > > On 28 February 2018 at 16:16, Mickael Hubert wrote: > > Hi > thanks a lot for your answer. > > But I want to send stream to external server, not record in audio file. i > can use record to "capture" the voice, but to stream it, it's more > complicated ;) > > thanks in advance > > > > 2018-02-21 0:50 GMT+01:00 Brian West : > > you can already do this without SIPREC in freeswitch. By setting the > RECORD_READ_ONLY or RECORD_WRITE_ONLY variables. > > > > /b > > > > > > On Fri, Feb 16, 2018 at 10:33 AM, Mickael Hubert > wrote: > > Hi list, > I want to record each call through freeswitch. But i want record only > caller (SSRC 1) OR callee (SSRC 2) voice (not both). > > I read about SIPREC, Jack, etc ... not interesting > > Do you have a idea for me please ? > > Thanks in advance > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > Official FreeSWITCH Sites > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.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 > > http://www.freeswitch.org > > > > > > > -- > > 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 <(918)%20424-9378> > > Website: https://www.FreeSWITCH.com > > > [image: color-facebook-96.png] > [image: > color-twitter-96.png] > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > Official FreeSWITCH Sites > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.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 > > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > Official FreeSWITCH Sites > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.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 > > http://www.freeswitch.org > > > > > ------------------------------ > > This e-mail and any files transmitted with it may contain privileged or > confidential information. It is solely for use by the individual for whom > it is intended, even if addressed incorrectly. If you received this e-mail > in error, please notify the sender; do not disclose, copy, distribute, or > take any action in reliance on the contents of this information; and delete > it from your system. Any other use of this e-mail is prohibited. > > Thank you for your compliance. > ------------------------------ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From umair at tezrosolutions.com Thu Mar 1 21:12:48 2018 From: umair at tezrosolutions.com (Muhammad Umair) Date: Thu, 1 Mar 2018 16:12:48 -0500 Subject: [Freeswitch-users] Conference mux mode video flickering In-Reply-To: References: <095A5D88-7540-4675-A9C2-F9E0B7028AEB@jerris.com> <35A8243B-2258-4ADE-9CF2-F360F72BCFDA@jerris.com> Message-ID: <20CA97BC-F2C2-40CE-BE60-FB0D1ECB59FE@tezrosolutions.com> Also tell him that we have 8 CPU cores. Umair Khokhar Chief Technology Officer Tezro Solutions Website Email > On Mar 1, 2018, at 4:02 PM, Dom Rumsey wrote: > > So I've tested again and this time it looks like individual cores are maxing out (it's showing 332%), which it wasn't doing previously. I only have two users and since switching to MUX I have a decent processor. > > > > > From: FreeSWITCH-users on behalf of Michael Jerris > Sent: Wednesday, February 28, 2018 5:57 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Conference mux mode video flickering > > look at individual cores, are any maxing out? > >> On Feb 28, 2018, at 5:07 AM, Dom Rumsey > wrote: >> >> No it doesn’t. So in a 3 way chat, for the person speaking, it shows their own video feed back to themself. The other two people (who are listening) see a flicker of the video feeds from both the speaker and the other listener. It's as if there's a paradox, I just can't work out what it is. >> >> Looked at CPU and we still have +70% left. Network is also fine, got loads of bandwidth left. >> >> Thanks >> On Feb 27, 2018, at 5:01 PM, Michael Jerris > wrote: >> only other thing i can think of would be some sort of resource starvation, cpu or network. Does it do the same without personal canvas? >> >>> On Feb 27, 2018, at 11:53 AM, Dom Rumsey < domrumsey at hotmail.com > wrote: >>> >>> >>> Thanks Mike. I'm using 1.6.20. >>> >>> From: FreeSWITCH-users > on behalf of Michael Jerris > >>> Sent: Tuesday, February 27, 2018 4:24 PM >>> To: FreeSWITCH Users Help >>> Subject: Re: [Freeswitch-users] Conference mux mode video flickering >>> >>> The flickering sounds like an old fixed bug we had with overlap and zoom layouts. Are you using old code? >>> >>>> On Feb 26, 2018, at 3:57 PM, Dom Rumsey < domrumsey at hotmail.com > wrote: >>>> >>>> Hi guys >>>> >>>> Thanks for your help previously. >>>> >>>> I'm now using MUX mode for video and I'm getting a problem whereby a conference with 3 or more participants keeps flickering video streams on the floor (we are using video-muxing-personal-canvas parameter). It's as if it can't decide who to show. Below is my conference profile: >>>> >>>> < profile name = "cp" > >>>> < param name = "domain" value = "$${domain}" /> >>>> < param name = "rate" value = "8000" /> >>>> < param name = "video-mode" value = "mux" /> >>>> < param name = "video-layout-name" value = "1x1" /> >>>> < param name = "interval" value = "20" /> >>>> < param name = "caller-controls" value = "default" /> >>>> < param name = "energy-level" value = "0" /> >>>> < param name = "video-auto-floor-msec" value = "3000" /> >>>> < param name = "video_no_video_avatar_png" value = "/usr/local/freeswitch/conf/images/video-muted.png" /> >>>> < param name = "video_mute_png" value = "/usr/local/freeswitch/conf/images/video-muted.png" /> >>>> < param name = "conference-flags" value = "audio-always|livearray-sync|livearray-json-status|video-mute-exit-canvas|video-muxing-personal-canvas" /> >>>> < param name = "max-members" value = "25" /> >>>> < param name = "sound-prefix" value = "/usr/local/freeswitch/conf/sounds/" /> >>>> < param name = "enter-sound" value = "tone_stream://%(200,0,500,600,700)" /> >>>> < param name = "exit-sound" value = "tone_stream://%(500,0,300,200,100,50,25)" /> >>>> < param name = "inbound-late-negotiation" value = "false" /> >>>> >>>> >>>> Any pointers on where I'm going wrong would be really appreciated. >>>> >>>> Thank you >>>> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From melekoktay at gmail.com Fri Mar 2 08:01:19 2018 From: melekoktay at gmail.com (Melek Oktay) Date: Fri, 2 Mar 2018 11:01:19 +0300 Subject: [Freeswitch-users] Media storage for Freeswitch Message-ID: Hi, I need a solution to store and read media files from Freeswitch servers (I have multiple Freeswitch). Media files are recorded voicemails, recorded calls and user files like welcome sounds, IVR sounds, waiting music, etc. Currently this is solved by using a mount to a SMB share (NFS) on local storage. This works good but I think it has a drawbacks like it is not scalable.. I wonder, what kind of solution exist in the literature ? What are the best practices that you face? Best Regards, Melek P.S. At first a solution with a distributed file system was tried with GlusterFS. This had some drawbacks, especially with high latency synchronization between nodes. This approach was dropped. -------------- next part -------------- An HTML attachment was scrubbed... URL: From emrahciftcibasi at gmail.com Fri Mar 2 14:23:12 2018 From: emrahciftcibasi at gmail.com (Emrah Ciftcibasi) Date: Fri, 2 Mar 2018 17:23:12 +0300 Subject: [Freeswitch-users] Best way for storing Media for Freeswitch Message-ID: Hi, I need a solution to store and read media files from Freeswitch servers (I have multiple Freeswitch). Media files are recorded voicemails, recorded calls and user files like welcome sounds, IVR sounds, waiting music, etc. Currently this is solved by using a mount to a SMB share (NFS) on local storage. This works good but I think it has a drawbacks like it is not scalable.. I wonder, what kind of solution exist in the literature ? What are the best practices that you face? Best Regards, Melek P.S. At first a solution with a distributed file system was tried with GlusterFS. This had some drawbacks, especially with high latency synchronization between nodes. This approach was dropped. -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Fri Mar 2 18:46:50 2018 From: s.safarov at gmail.com (Sergey Safarov) Date: Fri, 02 Mar 2018 18:46:50 +0000 Subject: [Freeswitch-users] Best way for storing Media for Freeswitch In-Reply-To: References: Message-ID: You can play media files from http server пт, 2 мар. 2018 г. в 19:57, Emrah Ciftcibasi : > Hi, > > I need a solution to store and read media files from Freeswitch servers (I > have multiple Freeswitch). > Media files are recorded voicemails, recorded calls and user files like > welcome sounds, > IVR sounds, waiting music, etc. > > Currently this is solved by using a mount to a SMB share (NFS) on local > storage. This works good but I think it has a drawbacks like it is not > scalable.. > > I wonder, what kind of solution exist in the literature ? What are the > best practices that you face? > > Best Regards, > Melek > > P.S. At first a solution with a distributed file system was tried with > GlusterFS. This had some drawbacks, especially with high latency > synchronization between nodes. This approach was dropped. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From domrumsey at hotmail.com Sat Mar 3 08:48:42 2018 From: domrumsey at hotmail.com (Dom Rumsey) Date: Sat, 3 Mar 2018 08:48:42 +0000 Subject: [Freeswitch-users] Conference mux mode video flickering In-Reply-To: References: <095A5D88-7540-4675-A9C2-F9E0B7028AEB@jerris.com> , <35A8243B-2258-4ADE-9CF2-F360F72BCFDA@jerris.com>, Message-ID: Hi All, I've sorted it out. Anyone reading this thread in the future who is experiencing the same problem will have the peram video-muxing-personal-canvas set in conference flags while layout is set to 1x1. video-muxing-personal-canvas is designed for layouts showing several people at the same time - when implementing this the concept of the floor doesn't really apply as this peram does it for you. Setting the layout to 1x1 means it places all of these canvases on top of each other, hence the flicker and CPU usage. Resolution: if you want 1x1 then remove video-muxing-personal-canvas, otherwise change the layout to show multiple people. Feel free to contact me directly for any further help. Thanks, Dom ________________________________ From: FreeSWITCH-users on behalf of Dom Rumsey Sent: Thursday, March 1, 2018 9:02 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Conference mux mode video flickering So I've tested again and this time it looks like individual cores are maxing out (it's showing 332%), which it wasn't doing previously. I only have two users and since switching to MUX I have a decent processor. [X] ________________________________ From: FreeSWITCH-users on behalf of Michael Jerris Sent: Wednesday, February 28, 2018 5:57 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Conference mux mode video flickering look at individual cores, are any maxing out? On Feb 28, 2018, at 5:07 AM, Dom Rumsey > wrote: No it doesn’t. So in a 3 way chat, for the person speaking, it shows their own video feed back to themself. The other two people (who are listening) see a flicker of the video feeds from both the speaker and the other listener. It's as if there's a paradox, I just can't work out what it is. Looked at CPU and we still have +70% left. Network is also fine, got loads of bandwidth left. Thanks On Feb 27, 2018, at 5:01 PM, Michael Jerris > wrote: only other thing i can think of would be some sort of resource starvation, cpu or network. Does it do the same without personal canvas? On Feb 27, 2018, at 11:53 AM, Dom Rumsey < domrumsey at hotmail.com> wrote: Thanks Mike. I'm using 1.6.20. ________________________________ From: FreeSWITCH-users > on behalf of Michael Jerris > Sent: Tuesday, February 27, 2018 4:24 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Conference mux mode video flickering The flickering sounds like an old fixed bug we had with overlap and zoom layouts. Are you using old code? On Feb 26, 2018, at 3:57 PM, Dom Rumsey < domrumsey at hotmail.com> wrote: Hi guys Thanks for your help previously. I'm now using MUX mode for video and I'm getting a problem whereby a conference with 3 or more participants keeps flickering video streams on the floor (we are using video-muxing-personal-canvas parameter). It's as if it can't decide who to show. Below is my conference profile: < profile name = "cp" > < param name = "domain" value = "$${domain}" /> < param name = "rate" value = "8000" /> < param name = "video-mode" value = "mux" /> < param name = "video-layout-name" value = "1x1" /> < param name = "interval" value = "20" /> < param name = "caller-controls" value = "default" /> < param name = "energy-level" value = "0" /> < param name = "video-auto-floor-msec" value = "3000" /> < param name = "video_no_video_avatar_png" value = "/usr/local/freeswitch/conf/images/video-muted.png" /> < param name = "video_mute_png" value = "/usr/local/freeswitch/conf/images/video-muted.png" /> < param name = "conference-flags" value = "audio-always|livearray-sync|livearray-json-status|video-mute-exit-canvas|video-muxing-personal-canvas" /> < param name = "max-members" value = "25" /> < param name = "sound-prefix" value = "/usr/local/freeswitch/conf/sounds/" /> < param name = "enter-sound" value = "tone_stream://%(200,0,500,600,700)" /> < param name = "exit-sound" value = "tone_stream://%(500,0,300,200,100,50,25)" /> < param name = "inbound-late-negotiation" value = "false" /> Any pointers on where I'm going wrong would be really appreciated. Thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: From bilaln018 at gmail.com Sat Mar 3 08:56:58 2018 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Sat, 03 Mar 2018 08:56:58 +0000 Subject: [Freeswitch-users] Best way for storing Media for Freeswitch In-Reply-To: References: Message-ID: Hi, I am using amazon S3 and mod_http_cache, works good for me. Regards Abbasi On Fri, 2 Mar 2018 at 11:48 PM, Sergey Safarov wrote: > You can play media files from http server > > > > > пт, 2 мар. 2018 г. в 19:57, Emrah Ciftcibasi : > >> Hi, >> >> I need a solution to store and read media files from Freeswitch servers >> (I have multiple Freeswitch). >> Media files are recorded voicemails, recorded calls and user files like >> welcome sounds, >> IVR sounds, waiting music, etc. >> >> Currently this is solved by using a mount to a SMB share (NFS) on local >> storage. This works good but I think it has a drawbacks like it is not >> scalable.. >> >> I wonder, what kind of solution exist in the literature ? What are the >> best practices that you face? >> >> Best Regards, >> Melek >> >> P.S. At first a solution with a distributed file system was tried with >> GlusterFS. This had some drawbacks, especially with high latency >> synchronization between nodes. This approach was dropped. >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.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 >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.com Sat Mar 3 12:40:40 2018 From: brian at freeswitch.com (Brian West) Date: Sat, 03 Mar 2018 12:40:40 +0000 Subject: [Freeswitch-users] Conference mux mode video flickering In-Reply-To: References: <095A5D88-7540-4675-A9C2-F9E0B7028AEB@jerris.com> <35A8243B-2258-4ADE-9CF2-F360F72BCFDA@jerris.com> Message-ID: JIRA is where this belongs! On Sat, Mar 3, 2018 at 2:56 AM Dom Rumsey wrote: > Hi All, > > > I've sorted it out. Anyone reading this thread in the future who is > experiencing the same problem will have the peram video-muxing-personal-canvas > set in conference flags while layout is set to 1x1. video-muxing-personal-canvas > is designed for layouts showing several people at the same time - when > implementing this the concept of the floor doesn't really apply as this > peram does it for you. Setting the layout to 1x1 means it places all of > these canvases on top of each other, hence the flicker and CPU usage. > > Resolution: if you want 1x1 then remove video-muxing-personal-canvas, > otherwise change the layout to show multiple people. Feel free to contact > me directly for any further help. > > Thanks, > Dom > > *From:* FreeSWITCH-users > on behalf of Dom Rumsey > *Sent:* Thursday, March 1, 2018 9:02 PM > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Conference mux mode video flickering > > So I've tested again and this time it looks like individual cores are > maxing out (it's showing 332%), which it wasn't doing previously. I only > have two users and since switching to MUX I have a decent processor. > > > > > ------------------------------ > *From:* FreeSWITCH-users > on behalf of Michael Jerris > *Sent:* Wednesday, February 28, 2018 5:57 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Conference mux mode video flickering > > look at individual cores, are any maxing out? > > On Feb 28, 2018, at 5:07 AM, Dom Rumsey wrote: > > No it doesn’t. So in a 3 way chat, for the person speaking, it shows their > own video feed back to themself. The other two people (who are listening) > see a flicker of the video feeds from both the speaker and the other > listener. It's as if there's a paradox, I just can't work out what it is. > > Looked at CPU and we still have +70% left. Network is also fine, got loads > of bandwidth left. > > Thanks > On Feb 27, 2018, at 5:01 PM, Michael Jerris wrote: > > only other thing i can think of would be some sort of resource starvation, > cpu or network. Does it do the same without personal canvas? > > On Feb 27, 2018, at 11:53 AM, Dom Rumsey < domrumsey at hotmail.com> wrote: > > > Thanks Mike. I'm using 1.6.20. > > ------------------------------ > *From:* FreeSWITCH-users > on behalf of Michael Jerris > *Sent:* Tuesday, February 27, 2018 4:24 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Conference mux mode video flickering > > The flickering sounds like an old fixed bug we had with overlap and zoom > layouts. Are you using old code? > > On Feb 26, 2018, at 3:57 PM, Dom Rumsey < domrumsey at hotmail.com> wrote: > > Hi guys > > Thanks for your help previously. > > I'm now using MUX mode for video and I'm getting a problem whereby a > conference with 3 or more participants keeps flickering video streams on > the floor (we are using video-muxing-personal-canvas parameter). It's as > if it can't decide who to show. Below is my conference profile: > > > < profile name = "cp" > > < param name = "domain" value = "$${domain}" /> > < param name = "rate" value = "8000" /> > < param name = "video-mode" value = "mux" /> > < param name = "video-layout-name" value = "1x1" /> > < param name = "interval" value = "20" /> > < param name = "caller-controls" value = "default" /> > < param name = "energy-level" value = "0" /> > < param name = "video-auto-floor-msec" value = "3000" /> > < param name = "video_no_video_avatar_png" value = > "/usr/local/freeswitch/conf/images/video-muted.png" /> > < param name = "video_mute_png" value = > "/usr/local/freeswitch/conf/images/video-muted.png" /> > < param name = "conference-flags" value = > "audio-always|livearray-sync|livearray-json-status|video-mute-exit-canvas|video-muxing-personal-canvas" > /> > < param name = "max-members" value = "25" /> > < param name = "sound-prefix" value = > "/usr/local/freeswitch/conf/sounds/" /> > < param name = "enter-sound" value = > "tone_stream://%(200,0,500,600,700)" /> > < param name = "exit-sound" value = > "tone_stream://%(500,0,300,200,100,50,25)" /> > < param name = "inbound-late-negotiation" value = "false" /> > > > > Any pointers on where I'm going wrong would be really appreciated. > > Thank you > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org -- 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: color-facebook-96.png] [image: color-twitter-96.png] -------------- next part -------------- An HTML attachment was scrubbed... URL: From magnus.kelly at gmail.com Sat Mar 3 15:17:40 2018 From: magnus.kelly at gmail.com (Magnus Kelly) Date: Sat, 3 Mar 2018 15:17:40 +0000 Subject: [Freeswitch-users] News of 1.8 Release ? Message-ID: Hello all, Due to mail bounces temporally removing me from the mailing list, I may have missed news on the potential 1.8 release. I see the development branch now moved to 1.9, but can't see any info on next stable release to match the Packt Publishing book I'm just reading and enjoying. Tip of the hat to to Anthony, Giovanni and the Freeswitch team. Thanks Magnus -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Sat Mar 3 21:48:20 2018 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Sat, 03 Mar 2018 21:48:20 +0000 Subject: [Freeswitch-users] News of 1.8 Release ? In-Reply-To: References: Message-ID: The book is completely 1.6 compatible, I swear! :) -giovanni Sent via mobile, please forgive typos and brevity. cell: +39 347 266 56 18 Giovanni Maruzzelli OpenTelecom.IT On Sat, Mar 3, 2018, 16:18 Magnus Kelly wrote: > Hello all, > > Due to mail bounces temporally removing me from the mailing list, I may > have missed news on the potential 1.8 release. I see the development branch > now moved to 1.9, but can't see any info on next stable release to match > the Packt Publishing book I'm just reading and enjoying. Tip of the hat to > to Anthony, Giovanni and the Freeswitch team. > > Thanks > Magnus > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From umair at tezrosolutions.com Mon Mar 5 00:04:44 2018 From: umair at tezrosolutions.com (Muhammad Umair) Date: Sun, 4 Mar 2018 19:04:44 -0500 Subject: [Freeswitch-users] Uninstalling Freeswitch 1.6 (Compiled using source code) Message-ID: Hi, I need to install Freeswitch 1.8 using code compilation method. I already have a Freeswitch 1.6 installed on that server, which was also installed using code compilation method. The server OS is Debian 8 (Jessie). How do I uninstall the Freeswitch 1.6? Thanks in advance. - Umair From social at bohboh.info Sun Mar 4 21:34:18 2018 From: social at bohboh.info (Social Boh) Date: Sun, 4 Mar 2018 16:34:18 -0500 Subject: [Freeswitch-users] Internal Sip profile error Message-ID: <5c2536ff-b889-8a80-9460-04f12fa9733a@bohboh.info> Hello list, I'm experimenting a "strange" issue with FreeSWITCH. If I Start Freeswitch from command line: *freeswitch -nonat -nf* IPv4 internal profile working fine: * sofia status*                      Name          Type                                       Data      State =================================================================================================             external-ipv6       profile   sip:mod_sofia@[2001:19f0:9002:11f6:5400:1ff:fe64:1503]:5080   RUNNING (0)              45.77.161.19         alias                                   internal      ALIASED                  external       profile            sip:mod_sofia at 45.77.161.19:5080      RUNNING (0)     external::example.com       gateway                    sip:joeuser at example.com      NOREG             internal-ipv6       profile   sip:mod_sofia@[2001:19f0:9002:11f6:5400:1ff:fe64:1503]:5060   RUNNING (0)                  internal       profile            sip:mod_sofia at 45.77.161.19:5060      RUNNING (0) ================================================================================================= If I use script daemon *systemctl start freeswitch* with this parameters: *-nonat -nf -nc* I have this error: Error Creating SIP UA for profile: internal (sip:mod_sofia at 45.77.161.19:5060;transport=udp,tcp) The likely causes for this are: 1) Another application is already listening on the specified address. 2) The IP the profile is attempting to bind to is not local to this system  sofia status                      Name          Type                                       Data      State =================================================================================================             external-ipv6       profile   sip:mod_sofia@[2001:19f0:9002:11f6:5400:1ff:fe64:1503]:5080   RUNNING (0)                  external       profile            sip:mod_sofia at 45.77.161.19:5080      RUNNING (0)     external::example.com       gateway                    sip:joeuser at example.com      NOREG             internal-ipv6       profile   sip:mod_sofia@[2001:19f0:9002:11f6:5400:1ff:fe64:1503]:5060   RUNNING (0) ================================================================================================= I compiled 1.6 version from source (git) on CentOS 7.4. Remote VPS with Public and Private IPv4 Any Hint? Thank you Regards -- --- I'm SoCIaL, MayBe -------------- next part -------------- An HTML attachment was scrubbed... URL: From stefan.mititelu92 at gmail.com Mon Mar 5 12:51:02 2018 From: stefan.mititelu92 at gmail.com (Mititelu Stefan) Date: Mon, 5 Mar 2018 14:51:02 +0200 Subject: [Freeswitch-users] [mod_conference] playback .wav after x seconds of silence Message-ID: Hi, Is there a way to playback a random .wav message after silence threshold has been kept for more than X seconds, in a conference? Thank you, Stefan -------------- next part -------------- An HTML attachment was scrubbed... URL: From vishalmpai at gmail.com Mon Mar 5 13:42:49 2018 From: vishalmpai at gmail.com (Vishal Pai) Date: Mon, 5 Mar 2018 19:12:49 +0530 Subject: [Freeswitch-users] IVR in Freeswitch using LUA Message-ID: Hi All I have built a IVR system in Freeswitch using LUA. In my IVR i am doing basic operation like select, insert etc. with remote MSSQL database. I am using FreeSWITCH version: 1.6.19+git~20171120T163416Z~b1b21d0695~64bit (git b1b21d0 2017-11-20 16:34:16Z 64bit) My concern is with the CPU usage it get more than 100% in just 3 calls. Please suggest what should i do to get normal usage of CPU. -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Mon Mar 5 16:47:54 2018 From: mike at jerris.com (Michael Jerris) Date: Mon, 5 Mar 2018 11:47:54 -0500 Subject: [Freeswitch-users] Uninstalling Freeswitch 1.6 (Compiled using source code) In-Reply-To: References: Message-ID: you can do make uninstall from the old code directory but really shouldn’t be necessary, if you build to the same locations it should install on top fine. > On Mar 4, 2018, at 7:04 PM, Muhammad Umair wrote: > > Hi, > > I need to install Freeswitch 1.8 using code compilation method. I already have a Freeswitch 1.6 installed on that server, which was also installed using code compilation method. The server OS is Debian 8 (Jessie). > > How do I uninstall the Freeswitch 1.6? > > Thanks in advance. > > - Umair > > From abaci64 at gmail.com Mon Mar 5 16:50:45 2018 From: abaci64 at gmail.com (Abaci B) Date: Mon, 5 Mar 2018 11:50:45 -0500 Subject: [Freeswitch-users] IVR in Freeswitch using LUA In-Reply-To: References: Message-ID: how do you connect to the database? On Mon, Mar 5, 2018 at 8:42 AM, Vishal Pai wrote: > Hi All > > I have built a IVR system in Freeswitch using LUA. In my IVR i am doing > basic operation like select, insert etc. with remote MSSQL database. > > I am using FreeSWITCH version: 1.6.19+git~20171120T163416Z~b1b21d0695~64bit > (git b1b21d0 2017-11-20 16:34:16Z 64bit) > > My concern is with the CPU usage it get more than 100% in just 3 calls. > > Please suggest what should i do to get normal usage of CPU. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From abaci64 at gmail.com Mon Mar 5 16:52:14 2018 From: abaci64 at gmail.com (Abaci B) Date: Mon, 5 Mar 2018 11:52:14 -0500 Subject: [Freeswitch-users] [mod_conference] playback .wav after x seconds of silence In-Reply-To: References: Message-ID: you can probably do it using an external application listening for conference events, if there is silence for a specific time you can execute an api to play the file On Mon, Mar 5, 2018 at 7:51 AM, Mititelu Stefan wrote: > Hi, > > Is there a way to playback a random .wav message after silence threshold > has been kept for more than X seconds, in a conference? > > Thank you, > Stefan > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From alihaider.4189 at gmail.com Mon Mar 5 16:52:37 2018 From: alihaider.4189 at gmail.com (Ali Haider) Date: Mon, 5 Mar 2018 21:52:37 +0500 Subject: [Freeswitch-users] =?utf-8?q?=28no_subject=29?= Message-ID: <70570982-5FD5-4245-8BF4-BE4C483713EB@gmail.com> [ERROR] fs_cli.c:1659 main() Error Connecting [Socket Connection Error] Usage: fs_cli [-H ] [-P ] [-p References: <70570982-5FD5-4245-8BF4-BE4C483713EB@gmail.com> Message-ID: <485B908B-34AB-4E34-8ED7-FB50366BAF04@jerris.com> this is the error you get when fs_cli tries to connect to freeswitch but can not. It could be a number of things such as freeswitch not running, or mod_event_socket not listening on the socket you are trying to connect to. > On Mar 5, 2018, at 11:52 AM, Ali Haider wrote: > > > [ERROR] fs_cli.c:1659 main() Error Connecting [Socket Connection Error] > Usage: fs_cli [-H ] [-P ] [-p how you can ressolve this error kindly tell me please > Sent from my iPhone From domrumsey at hotmail.com Mon Mar 5 17:37:43 2018 From: domrumsey at hotmail.com (Dom Rumsey) Date: Mon, 5 Mar 2018 17:37:43 +0000 Subject: [Freeswitch-users] MUX mode, don't want to see my own video Message-ID: Hi Guys, I have compiled FS using the master branch (on a Debian 8 Jessie) and I'm using MUX mode for conferences with layout set to 1x1, so whoever speaks is seen full screen. Can anyone tell me how I can get it so I don't see my own video feed when I talk? In a multiway conversation, it would be useful for me to see the video feed of the last person that spoke while everyone else sees my video feed. I wasn't sure if this was easily achievable with the correct settings. Thanks in advance for any help. Dom -------------- next part -------------- An HTML attachment was scrubbed... URL: From vishalmpai at gmail.com Mon Mar 5 17:42:32 2018 From: vishalmpai at gmail.com (Vishal Pai) Date: Mon, 5 Mar 2018 23:12:32 +0530 Subject: [Freeswitch-users] IVR in Freeswitch using LUA In-Reply-To: References: Message-ID: I connect the database using FreeTDS ODBC driver. On Mon, Mar 5, 2018 at 10:20 PM, Abaci B wrote: > how do you connect to the database? > > On Mon, Mar 5, 2018 at 8:42 AM, Vishal Pai wrote: > >> Hi All >> >> I have built a IVR system in Freeswitch using LUA. In my IVR i am doing >> basic operation like select, insert etc. with remote MSSQL database. >> >> I am using FreeSWITCH version: 1.6.19+git~20171120T163416Z~b1b21d0695~64bit >> (git b1b21d0 2017-11-20 16:34:16Z 64bit) >> >> My concern is with the CPU usage it get more than 100% in just 3 calls. >> >> Please suggest what should i do to get normal usage of CPU. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.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 >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From abaci64 at gmail.com Mon Mar 5 17:48:07 2018 From: abaci64 at gmail.com (Abaci B) Date: Mon, 5 Mar 2018 12:48:07 -0500 Subject: [Freeswitch-users] IVR in Freeswitch using LUA In-Reply-To: References: Message-ID: do you maybe have an infinite loop in your code? is the code you use something that you can share so that others can take a look if there is something obvious? On Mon, Mar 5, 2018 at 12:42 PM, Vishal Pai wrote: > I connect the database using FreeTDS ODBC driver. > > On Mon, Mar 5, 2018 at 10:20 PM, Abaci B wrote: > >> how do you connect to the database? >> >> On Mon, Mar 5, 2018 at 8:42 AM, Vishal Pai wrote: >> >>> Hi All >>> >>> I have built a IVR system in Freeswitch using LUA. In my IVR i am doing >>> basic operation like select, insert etc. with remote MSSQL database. >>> >>> I am using FreeSWITCH version: 1.6.19+git~20171120T163416Z~b1b21d0695~64bit >>> (git b1b21d0 2017-11-20 16:34:16Z 64bit) >>> >>> My concern is with the CPU usage it get more than 100% in just 3 calls. >>> >>> Please suggest what should i do to get normal usage of CPU. >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.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 >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.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 >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Mon Mar 5 17:48:12 2018 From: mike at jerris.com (Michael Jerris) Date: Mon, 5 Mar 2018 12:48:12 -0500 Subject: [Freeswitch-users] IVR in Freeswitch using LUA In-Reply-To: References: Message-ID: Unless you are on a raspberry pi, thats extremely strange. Something is running away with the cpu. I suggest you stub out pieces of your solution until you find the problem piece. > On Mar 5, 2018, at 12:42 PM, Vishal Pai wrote: > > I connect the database using FreeTDS ODBC driver. > > On Mon, Mar 5, 2018 at 10:20 PM, Abaci B > wrote: > how do you connect to the database? > > On Mon, Mar 5, 2018 at 8:42 AM, Vishal Pai > wrote: > Hi All > > I have built a IVR system in Freeswitch using LUA. In my IVR i am doing basic operation like select, insert etc. with remote MSSQL database. > > I am using FreeSWITCH version: 1.6.19+git~20171120T163416Z~b1b21d0695~64bit (git b1b21d0 2017-11-20 16:34:16Z 64bit) > > My concern is with the CPU usage it get more than 100% in just 3 calls. > > Please suggest what should i do to get normal usage of CPU. > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Mon Mar 5 18:06:01 2018 From: mike at jerris.com (Michael Jerris) Date: Mon, 5 Mar 2018 13:06:01 -0500 Subject: [Freeswitch-users] Internal Sip profile error In-Reply-To: <5c2536ff-b889-8a80-9460-04f12fa9733a@bohboh.info> References: <5c2536ff-b889-8a80-9460-04f12fa9733a@bohboh.info> Message-ID: <45D63B1A-ACDF-40A4-BBF2-31A1FB7A4C5A@jerris.com> maybe something with permissions > On Mar 4, 2018, at 4:34 PM, Social Boh wrote: > > Hello list, > > I'm experimenting a "strange" issue with FreeSWITCH. > > If I Start Freeswitch from command line: > > freeswitch -nonat -nf > > IPv4 internal profile working fine: > > sofia status > > Name Type Data State > ================================================================================================= > external-ipv6 profile sip:mod_sofia@ [2001:19f0:9002:11f6:5400:1ff:fe64:1503]:5080 RUNNING (0) > 45.77.161.19 alias internal ALIASED > external profile sip:mod_sofia at 45.77.161.19:5080 RUNNING (0) > external::example.com gateway sip:joeuser at example.com NOREG > internal-ipv6 profile sip:mod_sofia@ [2001:19f0:9002:11f6:5400:1ff:fe64:1503]:5060 RUNNING (0) > internal profile sip:mod_sofia at 45.77.161.19:5060 RUNNING (0) > ================================================================================================= > If I use script daemon > systemctl start freeswitch > with this parameters: > > -nonat -nf -nc > > I have this error: > > Error Creating SIP UA for profile: internal (sip:mod_sofia at 45.77.161.19:5060;transport=udp,tcp ) > The likely causes for this are: > 1) Another application is already listening on the specified address. > 2) The IP the profile is attempting to bind to is not local to this system > > sofia status > Name Type Data State > ================================================================================================= > external-ipv6 profile sip:mod_sofia@ [2001:19f0:9002:11f6:5400:1ff:fe64:1503]:5080 RUNNING (0) > external profile sip:mod_sofia at 45.77.161.19:5080 RUNNING (0) > external::example.com gateway sip:joeuser at example.com NOREG > internal-ipv6 profile sip:mod_sofia@ [2001:19f0:9002:11f6:5400:1ff:fe64:1503]:5060 RUNNING (0) > ================================================================================================= > I compiled 1.6 version from source (git) on CentOS 7.4. Remote VPS with Public and Private IPv4 > Any Hint? > > Thank you > > Regards > -- > --- > I'm SoCIaL, MayBe -------------- next part -------------- An HTML attachment was scrubbed... URL: From kathleen at freeswitch.com Mon Mar 5 18:59:44 2018 From: kathleen at freeswitch.com (Kathleen King) Date: Mon, 5 Mar 2018 10:59:44 -0800 Subject: [Freeswitch-users] ClueCon Weekly with Matrix! Message-ID: Hello, Don't miss the call this Wednesday with Matthew Hodgson from Matrix! He will be talking about everything new they are working on and some details of the exciting new release. You can join the call by dialing 888 at https://conference.freeswitch.org/vc/ or watch it live at https://youtu.be/PIDKy2JtiJc Kathleen King | Public Relations / Administrative Assistant FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: Kathleen at freeswitch.com Mobile: 703-859-3757 Website: https://www.FreeSWITCH.com [image: color-facebook-96.png] [image: color-twitter-96.png] -------------- next part -------------- An HTML attachment was scrubbed... URL: From raimundo.perez.cuba at gmail.com Mon Mar 5 19:10:52 2018 From: raimundo.perez.cuba at gmail.com (=?utf-8?Q?Raimundo_P=C3=A9rez_Nieves?=) Date: Mon, 5 Mar 2018 20:10:52 +0100 Subject: [Freeswitch-users] (no subject) Message-ID: Try this: Go to path /usr/bin and execute ./freeswitch It starts FS in secure mode, and you can debug the error. In my case, it was bad close tag in html and I just shutdown FS with that error, so It did not open. Doing this gave the idea of the problem. Hope solves Enviado desde mi iPhone El 05/03/2018, a la(s) 17:52, Ali Haider > escribió: > > [ERROR] fs_cli.c:1659 main() Error Connecting [Socket Connection Error] > Usage: fs_cli [-H ] [-P ] [-p how you can ressolve this error kindly tell me please > Sent from my iPhone > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From garrett.allen at teotech.com Mon Mar 5 18:23:49 2018 From: garrett.allen at teotech.com (Garrett Allen) Date: Mon, 5 Mar 2018 18:23:49 +0000 Subject: [Freeswitch-users] Internal Sip profile error In-Reply-To: <45D63B1A-ACDF-40A4-BBF2-31A1FB7A4C5A@jerris.com> References: <5c2536ff-b889-8a80-9460-04f12fa9733a@bohboh.info> <45D63B1A-ACDF-40A4-BBF2-31A1FB7A4C5A@jerris.com> Message-ID: Make sure to check permissions/ownership on your certificates if you have TLS enabled, that can cause this error if you run the daemon as a different user than when you manually start from the terminal. From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Monday, March 05, 2018 10:06 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Internal Sip profile error maybe something with permissions On Mar 4, 2018, at 4:34 PM, Social Boh > wrote: Hello list, I'm experimenting a "strange" issue with FreeSWITCH. If I Start Freeswitch from command line: freeswitch -nonat -nf IPv4 internal profile working fine: sofia status Name Type Data State ================================================================================================= external-ipv6 profile sip:mod_sofia@[2001:19f0:9002:11f6:5400:1ff:fe64:1503]:5080 RUNNING (0) 45.77.161.19 alias internal ALIASED external profile sip:mod_sofia at 45.77.161.19:5080 RUNNING (0) external::example.com gateway sip:joeuser at example.com NOREG internal-ipv6 profile sip:mod_sofia@[2001:19f0:9002:11f6:5400:1ff:fe64:1503]:5060 RUNNING (0) internal profile sip:mod_sofia at 45.77.161.19:5060 RUNNING (0) ================================================================================================= If I use script daemon systemctl start freeswitch with this parameters: -nonat -nf -nc I have this error: Error Creating SIP UA for profile: internal (sip:mod_sofia at 45.77.161.19:5060;transport=udp,tcp) The likely causes for this are: 1) Another application is already listening on the specified address. 2) The IP the profile is attempting to bind to is not local to this system sofia status Name Type Data State ================================================================================================= external-ipv6 profile sip:mod_sofia@[2001:19f0:9002:11f6:5400:1ff:fe64:1503]:5080 RUNNING (0) external profile sip:mod_sofia at 45.77.161.19:5080 RUNNING (0) external::example.com gateway sip:joeuser at example.com NOREG internal-ipv6 profile sip:mod_sofia@[2001:19f0:9002:11f6:5400:1ff:fe64:1503]:5060 RUNNING (0) ================================================================================================= I compiled 1.6 version from source (git) on CentOS 7.4. Remote VPS with Public and Private IPv4 Any Hint? Thank you Regards -- --- I'm SoCIaL, MayBe -------------- next part -------------- An HTML attachment was scrubbed... URL: From domrumsey at hotmail.com Mon Mar 5 19:47:17 2018 From: domrumsey at hotmail.com (Dom Rumsey) Date: Mon, 5 Mar 2018 19:47:17 +0000 Subject: [Freeswitch-users] (no subject) In-Reply-To: <70570982-5FD5-4245-8BF4-BE4C483713EB@gmail.com> References: <70570982-5FD5-4245-8BF4-BE4C483713EB@gmail.com> Message-ID: Hi Ali Few things. a) First of all, did you compile using code or a package manager? What is your OS? b) If you compiled using code, did you enable mod_event_socket in you modules.conf file? c) Check if you can find event_socket.conf.xml in you autoload_configs? See what is set as the password and try connecting using following syntax fs_cli -p d) What is the value of apply-inbound-acl in your event_socket.conf.xml file? Dom ________________________________ From: FreeSWITCH-users on behalf of Ali Haider Sent: Monday, March 5, 2018 4:52 PM To: FreeSWITCH-users at lists.freeswitch.org Subject: [Freeswitch-users] (no subject) [ERROR] fs_cli.c:1659 main() Error Connecting [Socket Connection Error] Usage: fs_cli [-H ] [-P ] [-p FreeSWITCH www.freeswitch.org FreeSWITCH is an open-source media application designed to support popular protools such as SIP and WebRTC and provides a platform to develop voice and video applications. http://confluence.freeswitch.org http://www.cluecon.com [https://cluecon.com/theme/img/cluecon/hook1.jpg] ClueCon Telephony and WebRTC Developer's conference www.cluecon.com ClueCon is an annual event bringing together all of the open source Telephony and WebRTC developers to collaberate on the latest technology in the communications industry. FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users FreeSWITCH-users Info Page lists.freeswitch.org To see the collection of prior postings to the list, visit the FreeSWITCH-users Archives. Using FreeSWITCH-users: To post a message to all the list ... UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org [https://freeswitch.org/theme/img/freeswitch-social.png] FreeSWITCH www.freeswitch.org FreeSWITCH is an open-source media application designed to support popular protools such as SIP and WebRTC and provides a platform to develop voice and video applications. -------------- next part -------------- An HTML attachment was scrubbed... URL: From wsimon at stratusvideo.com Mon Mar 5 19:50:59 2018 From: wsimon at stratusvideo.com (William Simon) Date: Mon, 5 Mar 2018 19:50:59 +0000 Subject: [Freeswitch-users] Internal Sip profile error In-Reply-To: References: <5c2536ff-b889-8a80-9460-04f12fa9733a@bohboh.info> <45D63B1A-ACDF-40A4-BBF2-31A1FB7A4C5A@jerris.com> Message-ID: <406B2CD9-E3FE-4F97-92DF-F82371639836@stratusvideo.com> I just encountered this error today. Running from systemd caused the same sofia profile loading error. Running freeswitch manually in console mode loaded everything fine. But then when I started up from systemd again, it was fine. I have no explanation for this. On Mar 5, 2018, at 1:23 PM, Garrett Allen > wrote: Make sure to check permissions/ownership on your certificates if you have TLS enabled, that can cause this error if you run the daemon as a different user than when you manually start from the terminal. From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Monday, March 05, 2018 10:06 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Internal Sip profile error maybe something with permissions On Mar 4, 2018, at 4:34 PM, Social Boh > wrote: Hello list, I'm experimenting a "strange" issue with FreeSWITCH. If I Start Freeswitch from command line: freeswitch -nonat -nf IPv4 internal profile working fine: sofia status Name Type Data State ================================================================================================= external-ipv6 profile sip:mod_sofia@[2001:19f0:9002:11f6:5400:1ff:fe64:1503]:5080 RUNNING (0) 45.77.161.19 alias internal ALIASED external profile sip:mod_sofia at 45.77.161.19:5080 RUNNING (0) external::example.com gateway sip:joeuser at example.com NOREG internal-ipv6 profile sip:mod_sofia@[2001:19f0:9002:11f6:5400:1ff:fe64:1503]:5060 RUNNING (0) internal profile sip:mod_sofia at 45.77.161.19:5060 RUNNING (0) ================================================================================================= If I use script daemon systemctl start freeswitch with this parameters: -nonat -nf -nc I have this error: Error Creating SIP UA for profile: internal (sip:mod_sofia at 45.77.161.19:5060;transport=udp,tcp) The likely causes for this are: 1) Another application is already listening on the specified address. 2) The IP the profile is attempting to bind to is not local to this system sofia status Name Type Data State ================================================================================================= external-ipv6 profile sip:mod_sofia@[2001:19f0:9002:11f6:5400:1ff:fe64:1503]:5080 RUNNING (0) external profile sip:mod_sofia at 45.77.161.19:5080 RUNNING (0) external::example.com gateway sip:joeuser at example.com NOREG internal-ipv6 profile sip:mod_sofia@[2001:19f0:9002:11f6:5400:1ff:fe64:1503]:5060 RUNNING (0) ================================================================================================= I compiled 1.6 version from source (git) on CentOS 7.4. Remote VPS with Public and Private IPv4 Any Hint? Thank you Regards -- --- I'm SoCIaL, MayBe _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.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 http://www.freeswitch.org "The information transmitted is intended only for the person or entity to which it is addressed and may contain proprietary, business-confidential and/or privileged material. If you are not the intended recipient of this message you are hereby notified that any use, review, retransmission, dissemination, distribution, reproduction or any action taken in reliance upon this message is prohibited. If you received this in error, please contact the sender and delete the material from any computer." -------------- next part -------------- An HTML attachment was scrubbed... URL: From asilva at wirelessmundi.com Mon Mar 5 23:53:56 2018 From: asilva at wirelessmundi.com (antonio) Date: Tue, 6 Mar 2018 00:53:56 +0100 Subject: [Freeswitch-users] Internal Sip profile error In-Reply-To: <406B2CD9-E3FE-4F97-92DF-F82371639836@stratusvideo.com> References: <5c2536ff-b889-8a80-9460-04f12fa9733a@bohboh.info> <45D63B1A-ACDF-40A4-BBF2-31A1FB7A4C5A@jerris.com> <406B2CD9-E3FE-4F97-92DF-F82371639836@stratusvideo.com> Message-ID: <891ccfad-ffc5-e95c-19e0-21e3c28f7806@wirelessmundi.com> it was on boot time? probably it start before the network target so you didn't have interfaces to bind sofia... On 03/05/2018 08:50 PM, William Simon wrote: > I just encountered this error today. Running from systemd caused the > same sofia profile loading error. Running freeswitch manually in > console mode loaded everything fine. But then when I started up from > systemd again, it was fine. I have no explanation for this. > > >> On Mar 5, 2018, at 1:23 PM, Garrett Allen > > wrote: >> >> Make sure to check permissions/ownership on your certificates if you >> have TLS enabled, that can cause this error if you run the daemon as >> a different user than when you manually start from the terminal. >> >> *From:*FreeSWITCH-users >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of >> *Michael Jerris >> *Sent:* Monday, March 05, 2018 10:06 >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Internal Sip profile error >> >> maybe something with permissions >> >> On Mar 4, 2018, at 4:34 PM, Social Boh > > wrote: >> >> Hello list, >> >> I'm experimenting a "strange" issue with FreeSWITCH. >> >> If I Start Freeswitch from command line: >> >> *freeswitch -nonat -nf* >> >> IPv4 internal profile working fine: >> >> * sofia status* >> >>                      Name Type Data      State >> ================================================================================================= >>             external-ipv6 profile >> sip:mod_sofia@[2001:19f0:9002:11f6:5400:1ff:fe64:1503]:5080 >> RUNNING (0) >>              45.77.161.19 alias internal      ALIASED >>                  external profile sip:mod_sofia at 45.77.161.19:5080 >> RUNNING (0) >>     external::example.com gateway >> sip:joeuser at example.com NOREG >>             internal-ipv6 profile >> sip:mod_sofia@[2001:19f0:9002:11f6:5400:1ff:fe64:1503]:5060 >> RUNNING (0) >>                  internal profile sip:mod_sofia at 45.77.161.19:5060 >> RUNNING (0) >> ================================================================================================= >> >> If I use script daemon >> >> *systemctl start freeswitch* >> >> with this parameters: >> >> *-nonat -nf -nc* >> >> I have this error: >> >> Error Creating SIP UA for profile: internal >> (sip:mod_sofia at 45.77.161.19:5060;transport=udp,tcp) >> The likely causes for this are: >> 1) Another application is already listening on the specified address. >> 2) The IP the profile is attempting to bind to is not local to >> this system >> >>  sofia status >>                      Name Type Data      State >> ================================================================================================= >>             external-ipv6 profile >> sip:mod_sofia@[2001:19f0:9002:11f6:5400:1ff:fe64:1503]:5080 >> RUNNING (0) >>                  external profile sip:mod_sofia at 45.77.161.19:5080 >> RUNNING (0) >>     external::example.com gateway >> sip:joeuser at example.com NOREG >>             internal-ipv6 profile >> sip:mod_sofia@[2001:19f0:9002:11f6:5400:1ff:fe64:1503]:5060 >> RUNNING (0) >> ================================================================================================= >> >> I compiled 1.6 version from source (git) on CentOS 7.4. Remote >> VPS with Public and Private IPv4 >> >> Any Hint? >> >> Thank you >> >> Regards >> >> -- >> >> --- >> >> I'm SoCIaL, MayBe >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.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 >> http://www.freeswitch.org > > > > “The information transmitted is intended only for the person or entity > to which it is addressed and may contain proprietary, > business-confidential and/or privileged material. If you are not the > intended recipient of this message you are hereby notified that any > use, review, retransmission, dissemination, distribution, reproduction > or any action taken in reliance upon this message is prohibited. If > you received this in error, please contact the sender and delete the > material from any computer.” > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org -- Saludos / Regards / Cumprimentos Anónio Silva -------------- next part -------------- An HTML attachment was scrubbed... URL: From stefan.mititelu92 at gmail.com Tue Mar 6 09:00:54 2018 From: stefan.mititelu92 at gmail.com (Mititelu Stefan) Date: Tue, 6 Mar 2018 11:00:54 +0200 Subject: [Freeswitch-users] [mod_conference] playback .wav after x seconds of silence In-Reply-To: References: Message-ID: Hi Abaci, You are talking about using mod_event_socket, right? Thank you, Stefan On Mon, Mar 5, 2018 at 6:52 PM, Abaci B wrote: > you can probably do it using an external application listening for > conference events, if there is silence for a specific time you can execute > an api to play the file > > On Mon, Mar 5, 2018 at 7:51 AM, Mititelu Stefan < > stefan.mititelu92 at gmail.com> wrote: > >> Hi, >> >> Is there a way to playback a random .wav message after silence threshold >> has been kept for more than X seconds, in a conference? >> >> Thank you, >> Stefan >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.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 >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From infos at madovsky.org Tue Mar 6 14:03:49 2018 From: infos at madovsky.org (Madovsky) Date: Tue, 6 Mar 2018 06:03:49 -0800 Subject: [Freeswitch-users] pgsql core and odbc Message-ID: Hi, I noticed that if I use pgsql core protocol in switch.conf.xml (pgsql://) to a remote postgresql through VPN ps shows an idle connection taking between 15 and 50% of cpu. with odbc there is no such kind of issue. jira or a special setting needed? thanks From abaci64 at gmail.com Tue Mar 6 14:32:15 2018 From: abaci64 at gmail.com (Abaci B) Date: Tue, 6 Mar 2018 09:32:15 -0500 Subject: [Freeswitch-users] [mod_conference] playback .wav after x seconds of silence In-Reply-To: References: Message-ID: yes On Tue, Mar 6, 2018 at 4:00 AM, Mititelu Stefan wrote: > Hi Abaci, > > You are talking about using mod_event_socket, right? > > Thank you, > Stefan > > On Mon, Mar 5, 2018 at 6:52 PM, Abaci B wrote: > >> you can probably do it using an external application listening for >> conference events, if there is silence for a specific time you can execute >> an api to play the file >> >> On Mon, Mar 5, 2018 at 7:51 AM, Mititelu Stefan < >> stefan.mititelu92 at gmail.com> wrote: >> >>> Hi, >>> >>> Is there a way to playback a random .wav message after silence threshold >>> has been kept for more than X seconds, in a conference? >>> >>> Thank you, >>> Stefan >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.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 >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.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 >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From asonesh at gmail.com Mon Mar 5 23:34:55 2018 From: asonesh at gmail.com (Ari Sonesh) Date: Mon, 5 Mar 2018 18:34:55 -0500 Subject: [Freeswitch-users] Custom Headers with mod_callcenter Message-ID: Can anyone advise when using the mod callcenter ... how to pass custom sip headers from the calling party call leg to the agent call leg? thanks! *Ari Sonesh * -------------- next part -------------- An HTML attachment was scrubbed... URL: From tculjaga at gmail.com Tue Mar 6 17:22:32 2018 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 6 Mar 2018 18:22:32 +0100 Subject: [Freeswitch-users] NAT / UDP hole punching issue In-Reply-To: References: <018801d3b06c$aac462f0$004d28d0$@delagarda.com> <01da01d3b077$5fe24ee0$1fa6eca0$@delagarda.com> Message-ID: <8DD8259F-BA2F-4CF9-A58B-B6A3CA4EDF68@gmail.com> So, does anyone know why FS doesnt initiate rtp stream towards the client. So if its not a bug, i guess its the guy in between the keyboard and chair ;) What m’I missing? Sent from my iPhone > On 28 Feb 2018, at 11:11, Tihomir Culjaga wrote: > > Nope, this is a real life scenario and just the fact that it works when i port forward the RTP range on the FS side proves everything network related is working just fine. > > My question here goes to FS team ....why FS is not sending any RTP packets to a remote client even when the client advertises the public IP:PORT in SDP.. > Both client and FS uses STUN ( in network capture i see them communicate with their stun servers respectively ). The client sends RTP toward FS while FS does not. > > how do i debug rtp forwarding on FS itself ... is there any debug i can turn on ? > > > > > >> On 28 February 2018 at 10:34, Francesco Facco de Lagarda wrote: >> Sorry to hear Tihomir. >> >> >> >> Is there ANYWAY you can test everything locally on same network (i.e. without stun, nat, etc..) maybe the issue isn’t with stun! >> >> >> >> >> >> From: FreeSWITCH-users On Behalf Of Tihomir Culjaga >> Sent: mercoledì 28 febbraio 2018 09:57 >> >> >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] NAT / UDP hole punching issue >> >> >> RTP ports are defined. When i do a port forward for my RTP range i get my RTP audio working i guess due to rtp-auto-adjust feature on FS... but it should work without port forwarding. >> >> >> >> here simply FS is not starting to send RTP traffic to the client even if it notified its public IP:PORT in SDP on 200 OK. >> >> >> >> i see FS contacting a STUN server, getting the public IP:PORT and than ... doesn't send any RTP traffic towards the client... this is what its bugging me. >> >> >> >> T. >> >> >> >> On 28 February 2018 at 09:18, Francesco Facco de Lagarda wrote: >> >> Check your RTP ports .. in the fs config and the port forwarding on firewalls. >> >> Also, (two cent’s worth), I had a lot of problems with rtp (video and audio) using VErto.. in the end I read that if you don’t specify a stun server, by default it uses google’s.. I don’t know if its applicable in this case, but you never know! >> >> >> >> Good luck! >> >> F >> >> >> >> From: FreeSWITCH-users On Behalf Of Tihomir Culjaga >> Sent: mercoledì 28 febbraio 2018 09:01 >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] NAT / UDP hole punching issue >> >> >> >> does anyone have a clue ? :=) >> >> >> >> On 27 February 2018 at 11:50, Tihomir Culjaga wrote: >> >> hi, >> >> >> >> >> >> I have "no audio" issue with TLS and i hope someone could help as Im getting crazy ... literally :( >> >> >> >> my setup is like this: >> >> >> >> Phone <> NAT <> INTERNET <> NAT >> >> >> >> FreeSWITCH version: 1.6.12~64bit ( 64bit) >> >> >> >> I have a separate profile configured for TLS: >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> ================================================================================================= >> >> Name tls-public >> >> Domain Name N/A >> >> Auto-NAT false >> >> DBName sofia_reg_tls-public >> >> Pres Hosts 192.168.100.60,192.168.100.60 >> >> Dialplan XML >> >> Context public >> >> Challenge Realm auto_from >> >> RTP-IP 192.168.100.60 >> >> Ext-RTP-IP stun:stun.freeswitch.org >> >> SIP-IP 192.168.100.60 >> >> Ext-SIP-IP 85.114.41.180 >> >> TLS-URL sip:mod_sofia at 85.114.41.180:15061 >> >> TLS-BIND-URL sips:mod_sofia at 85.114.41.180:15061;maddr=192.168.100.60;transport=tls >> >> WS-BIND-URL sip:mod_sofia at 192.168.100.60:5066;transport=ws >> >> WSS-BIND-URL sips:mod_sofia at 192.168.100.60:7443;transport=wss >> >> HOLD-MUSIC local_stream://moh >> >> OUTBOUND-PROXY N/A >> >> CODECS IN PCMA >> >> CODECS OUT PCMA >> >> TEL-EVENT 101 >> >> DTMF-MODE rfc2833 >> >> CNG 13 >> >> SESSION-TO 0 >> >> MAX-DIALOG 0 >> >> NOMEDIA false >> >> LATE-NEG true >> >> PROXY-MEDIA false >> >> ZRTP-PASSTHRU false >> >> AGGRESSIVENAT false >> >> CALLS-IN 0 >> >> FAILED-CALLS-IN 0 >> >> CALLS-OUT 2 >> >> FAILED-CALLS-OUT 2 >> >> REGISTRATIONS 0 >> >> >> >> >> >> >> i manage to register the phone with no problems but when i call the phone i get no audio; >> >> >> >> bgapi expand originate ${sofia_contact(tls-profile/agent2/nexios at 192.168.100.60)} &echo() >> >> >> >> >> >> >> >> FS sends the invite as: >> >> >> >> >> >> SDP in INVITE message from FS >> >> >> >> v=0 >> >> o=FreeSWITCH 1519708899 1519708900 IN IP4 85.114.41.180 >> >> s=FreeSWITCH >> >> c=IN IP4 85.114.41.180 >> >> t=0 0 >> >> m=audio 17480 RTP/AVP 8 101 >> >> a=rtpmap:8 PCMA/8000 >> >> a=rtpmap:101 telephone-event/8000 >> >> a=fmtp:101 0-16 >> >> a=ptime:20 >> >> >> >> >> >> SIP Client responds with: >> >> >> >> SDP in 200 OK from the client >> >> >> >> >> >> v=0 >> >> o=- 3728718779 3728718780 IN IP4 213.147.96.240 >> >> s=pjmedia >> >> b=AS:84 >> >> t=0 0 >> >> a=X-nat:0 >> >> m=audio 4002 RTP/AVP 8 101 >> >> c=IN IP4 213.147.96.240 >> >> b=TIAS:64000 >> >> a=rtcp:4003 IN IP4 213.147.96.240 >> >> a=sendrecv >> >> a=rtpmap:8 PCMA/8000 >> >> a=rtpmap:101 telephone-event/8000 >> >> a=fmtp:101 0-16 >> >> >> >> >> >> >> >> So the UDP stream is: client( 4002 ) <> ( 17480 )FS >> >> >> >> when i sniff the traffic (on both sides client/FS) using wireshark, i see RTP packets ( src:4002, dst:17480 ) leaving the client towards FS, but I don't see RTP packets ( src:17480, dst:4002 ) from FS side leaving towards the client. >> >> >> >> >> >> so my question, of course, is why FS is not sending RTP packets to the IP:PORT notified in 200 OK SDP from the client ? Did i miss some needed configuration ? >> >> >> >> >> >> in FS logs i see 192.168.100.60 port 17480 -> 213.147.96.240 port 4002 but nothing is actually being sent out from FS >> >> >> >> 2018-02-27 11:13:02.733376 [DEBUG] sofia.c:6965 Channel sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 entering state [ready][200] >> >> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4311 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] >> >> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4366 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match >> >> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4227 Set telephone-event payload to 101 at 8000 >> >> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:3021 Set Codec sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 PCMA/8000 20 ms 160 samples 64000 bits 1 channels >> >> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_codec.c:111 sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Original read codec set to PCMA:8 >> >> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4572 Set telephone-event payload to 101 at 8000 >> >> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4631 sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Set 2833 dtmf send payload to 101 recv payload to 101 >> >> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6588 AUDIO RTP [sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551] 192.168.100.60 port 17480 -> 213.147.96.240 port 4002 codec: 8 ms: 20 >> >> 2018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:3875 Starting timer [soft] 160 bytes per 20ms >> >> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6806 Activating RTCP PORT 4003 >> >> 2018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:4261 RTCP send rate is: 5000 and packet rate is: 20000 Remote Port: 4003 >> >> 2018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:2534 Setting RTCP remote addr to 213.147.96.240:4003 2 >> >> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6887 sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Set 2833 dtmf send payload to 101 >> >> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6894 sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Set 2833 dtmf receive payload to 101 >> >> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6917 sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Set rtp dtmf delay to 40 >> >> 2018-02-27 11:13:02.733376 [NOTICE] sofia.c:8023 Channel [sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551] has been answered >> >> 2018-02-27 11:13:02.733376 [DEBUG] switch_channel.c:3770 (sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551) Callstate Change RINGING -> ACTIVE >> >> 2018-02-27 11:13:02.753388 [DEBUG] switch_ivr_originate.c:3686 Originate Resulted in Success: [sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551] >> >> 2018-02-27 11:13:02.753388 [INFO] switch_channel.c:3127 sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Flipping CID from "" <0000000000> to "Outbound Call" >> >> 2018-02-27 11:13:02.753388 [DEBUG] mod_commands.c:4788 (sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551) State Change CS_CONSUME_MEDIA -> CS_EXECUTE >> >> 2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:584 (sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551) Running State Change CS_EXECUTE >> >> 2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:650 (sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551) State EXECUTE >> >> 2018-02-27 11:13:02.753388 [DEBUG] mod_sofia.c:198 sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 SOFIA EXECUTE >> >> 2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:328 sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Standard EXECUTE >> >> EXECUTE sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 echo() >> >> >> >> >> >> >> >> Regards, >> >> Tihomir. >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.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 >> http://www.freeswitch.org >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.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 >> http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From asilva at wirelessmundi.com Tue Mar 6 18:03:26 2018 From: asilva at wirelessmundi.com (antonio) Date: Tue, 6 Mar 2018 19:03:26 +0100 Subject: [Freeswitch-users] Custom Headers with mod_callcenter In-Reply-To: References: Message-ID: <67d27eba-e6ed-174b-2606-1f14c92b988e@wirelessmundi.com> you can try dialplan variable: cc_export_vars src: https://freeswitch.org/confluence/display/FREESWITCH/mod_callcenter#mod_callcenter-cc_export_vars On 03/06/2018 12:34 AM, Ari Sonesh wrote: > Can anyone advise when using the mod callcenter ... how to pass custom > sip headers from the calling party call leg to the agent call leg? > thanks! > > *Ari Sonesh* > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org -- Saludos / Regards / Cumprimentos Anónio Silva -------------- next part -------------- An HTML attachment was scrubbed... URL: From wsimon at stratusvideo.com Tue Mar 6 20:24:36 2018 From: wsimon at stratusvideo.com (William Simon) Date: Tue, 6 Mar 2018 20:24:36 +0000 Subject: [Freeswitch-users] Internal Sip profile error In-Reply-To: <891ccfad-ffc5-e95c-19e0-21e3c28f7806@wirelessmundi.com> References: <5c2536ff-b889-8a80-9460-04f12fa9733a@bohboh.info> <45D63B1A-ACDF-40A4-BBF2-31A1FB7A4C5A@jerris.com> <406B2CD9-E3FE-4F97-92DF-F82371639836@stratusvideo.com> <891ccfad-ffc5-e95c-19e0-21e3c28f7806@wirelessmundi.com> Message-ID: <61CAB14C-A2D5-4BC1-8017-5A5B6AE82B65@stratusvideo.com> No, after the system was already up for some time. On Mar 5, 2018, at 6:53 PM, antonio > wrote: it was on boot time? probably it start before the network target so you didn't have interfaces to bind sofia... On 03/05/2018 08:50 PM, William Simon wrote: I just encountered this error today. Running from systemd caused the same sofia profile loading error. Running freeswitch manually in console mode loaded everything fine. But then when I started up from systemd again, it was fine. I have no explanation for this. On Mar 5, 2018, at 1:23 PM, Garrett Allen > wrote: Make sure to check permissions/ownership on your certificates if you have TLS enabled, that can cause this error if you run the daemon as a different user than when you manually start from the terminal. From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Monday, March 05, 2018 10:06 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Internal Sip profile error maybe something with permissions On Mar 4, 2018, at 4:34 PM, Social Boh > wrote: Hello list, I'm experimenting a "strange" issue with FreeSWITCH. If I Start Freeswitch from command line: freeswitch -nonat -nf IPv4 internal profile working fine: sofia status Name Type Data State ================================================================================================= external-ipv6 profile sip:mod_sofia@[2001:19f0:9002:11f6:5400:1ff:fe64:1503]:5080 RUNNING (0) 45.77.161.19 alias internal ALIASED external profile sip:mod_sofia at 45.77.161.19:5080 RUNNING (0) external::example.com gateway sip:joeuser at example.com NOREG internal-ipv6 profile sip:mod_sofia@[2001:19f0:9002:11f6:5400:1ff:fe64:1503]:5060 RUNNING (0) internal profile sip:mod_sofia at 45.77.161.19:5060 RUNNING (0) ================================================================================================= If I use script daemon systemctl start freeswitch with this parameters: -nonat -nf -nc I have this error: Error Creating SIP UA for profile: internal (sip:mod_sofia at 45.77.161.19:5060;transport=udp,tcp) The likely causes for this are: 1) Another application is already listening on the specified address. 2) The IP the profile is attempting to bind to is not local to this system sofia status Name Type Data State ================================================================================================= external-ipv6 profile sip:mod_sofia@[2001:19f0:9002:11f6:5400:1ff:fe64:1503]:5080 RUNNING (0) external profile sip:mod_sofia at 45.77.161.19:5080 RUNNING (0) external::example.com gateway sip:joeuser at example.com NOREG internal-ipv6 profile sip:mod_sofia@[2001:19f0:9002:11f6:5400:1ff:fe64:1503]:5060 RUNNING (0) ================================================================================================= I compiled 1.6 version from source (git) on CentOS 7.4. Remote VPS with Public and Private IPv4 Any Hint? Thank you Regards -- --- I'm SoCIaL, MayBe _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.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 http://www.freeswitch.org “The information transmitted is intended only for the person or entity to which it is addressed and may contain proprietary, business-confidential and/or privileged material. If you are not the intended recipient of this message you are hereby notified that any use, review, retransmission, dissemination, distribution, reproduction or any action taken in reliance upon this message is prohibited. If you received this in error, please contact the sender and delete the material from any computer.” _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.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 http://www.freeswitch.org -- Saludos / Regards / Cumprimentos Anónio Silva _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.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 http://www.freeswitch.org “The information transmitted is intended only for the person or entity to which it is addressed and may contain proprietary, business-confidential and/or privileged material. If you are not the intended recipient of this message you are hereby notified that any use, review, retransmission, dissemination, distribution, reproduction or any action taken in reliance upon this message is prohibited. If you received this in error, please contact the sender and delete the material from any computer.” -------------- next part -------------- An HTML attachment was scrubbed... URL: From steveayre at gmail.com Tue Mar 6 22:00:46 2018 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 6 Mar 2018 22:00:46 +0000 Subject: [Freeswitch-users] Internal Sip profile error In-Reply-To: <61CAB14C-A2D5-4BC1-8017-5A5B6AE82B65@stratusvideo.com> References: <5c2536ff-b889-8a80-9460-04f12fa9733a@bohboh.info> <45D63B1A-ACDF-40A4-BBF2-31A1FB7A4C5A@jerris.com> <406B2CD9-E3FE-4F97-92DF-F82371639836@stratusvideo.com> <891ccfad-ffc5-e95c-19e0-21e3c28f7806@wirelessmundi.com> <61CAB14C-A2D5-4BC1-8017-5A5B6AE82B65@stratusvideo.com> Message-ID: Try searching the logs for 'IP change detected' On 6 March 2018 at 20:24, William Simon wrote: > No, after the system was already up for some time. > > On Mar 5, 2018, at 6:53 PM, antonio wrote: > > it was on boot time? probably it start before the network target so you > didn't have interfaces to bind sofia... > > > On 03/05/2018 08:50 PM, William Simon wrote: > > I just encountered this error today. Running from systemd caused the same > sofia profile loading error. Running freeswitch manually in console mode > loaded everything fine. But then when I started up from systemd again, it > was fine. I have no explanation for this. > > > On Mar 5, 2018, at 1:23 PM, Garrett Allen > wrote: > > Make sure to check permissions/ownership on your certificates if you have > TLS enabled, that can cause this error if you run the daemon as a different > user than when you manually start from the terminal. > > *From:* FreeSWITCH-users [mailto:freeswitch-users- > bounces at lists.freeswitch.org > ] *On Behalf Of *Michael > Jerris > *Sent:* Monday, March 05, 2018 10:06 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Internal Sip profile error > > maybe something with permissions > > > On Mar 4, 2018, at 4:34 PM, Social Boh wrote: > > Hello list, > I'm experimenting a "strange" issue with FreeSWITCH. > If I Start Freeswitch from command line: > *freeswitch -nonat -nf* > IPv4 internal profile working fine: > * sofia status* > > Name Type > Data State > ============================================================ > ===================================== > external-ipv6 profile sip:mod_sofia@[2001: > 19f0:9002:11f6:5400:1ff:fe64:1503]:5080 RUNNING (0) > 45.77.161.19 alias > internal ALIASED > external profile sip:mod_ > sofia at 45.77.161.19:5080 RUNNING (0) > external::example.com gateway sip > :joeuser at example.com NOREG > internal-ipv6 profile sip:mod_sofia@[2001: > 19f0:9002:11f6:5400:1ff:fe64:1503]:5060 RUNNING (0) > internal profile sip:mod_ > sofia at 45.77.161.19:5060 RUNNING (0) > ============================================================ > ===================================== > If I use script daemon > *systemctl start freeswitch* > with this parameters: > *-nonat -nf -nc* > I have this error: > Error Creating SIP UA for profile: internal (sip:mod_sofia at 45.77.161.19: > 5060;transport=udp,tcp) > The likely causes for this are: > 1) Another application is already listening on the specified address. > 2) The IP the profile is attempting to bind to is not local to this system > sofia status > Name Type > Data State > ============================================================ > ===================================== > external-ipv6 profile sip:mod_sofia@[2001: > 19f0:9002:11f6:5400:1ff:fe64:1503]:5080 RUNNING (0) > external profile sip:mod_ > sofia at 45.77.161.19:5080 RUNNING (0) > external::example.com gateway sip > :joeuser at example.com NOREG > internal-ipv6 profile sip:mod_sofia@[2001: > 19f0:9002:11f6:5400:1ff:fe64:1503]:5060 RUNNING (0) > ============================================================ > ===================================== > I compiled 1.6 version from source (git) on CentOS 7.4. Remote VPS with > Public and Private IPv4 > Any Hint? > Thank you > Regards > > -- > > --- > > I'm SoCIaL, MayBe > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > > > > > “The information transmitted is intended only for the person or entity to > which it is addressed and may contain proprietary, business-confidential > and/or privileged material. If you are not the intended recipient of this > message you are hereby notified that any use, review, retransmission, > dissemination, distribution, reproduction or any action taken in reliance > upon this message is prohibited. If you received this in error, please > contact the sender and delete the material from any computer.” > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > -- > Saludos / Regards / Cumprimentos > Anónio Silva > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > > > > > “The information transmitted is intended only for the person or entity to > which it is addressed and may contain proprietary, business-confidential > and/or privileged material. If you are not the intended recipient of this > message you are hereby notified that any use, review, retransmission, > dissemination, distribution, reproduction or any action taken in reliance > upon this message is prohibited. If you received this in error, please > contact the sender and delete the material from any computer.” > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From alex at freeswitch.com Tue Mar 6 23:26:27 2018 From: alex at freeswitch.com (Alexey Sibyakin) Date: Wed, 7 Mar 2018 08:26:27 +0900 Subject: [Freeswitch-users] DTMF negotiation In-Reply-To: <188910919.6866630.1519918789572@mail.yahoo.com> References: <510252213.6154862.1519830130424.ref@mail.yahoo.com> <510252213.6154862.1519830130424@mail.yahoo.com> <188910919.6866630.1519918789572@mail.yahoo.com> Message-ID: Yes, it's possible. Vanilla config has example. Alex On Fri, Mar 2, 2018 at 12:39 AM, kaiduan xie wrote: > Alex, > > Thanks for the help. > > With in sofia profile, the offer > from FreeSwitch does not add telephone event support. > > With , > FreeSwitch always sends DTMF inband even the offer from FreeSwitch and > answer both supports telephone event. > > What I want is DTMF negotiation as below, > > FreeSwitch always adds telephone event support in offer, > > 1. If answer supports telephone event, then FreeSwitch sends DTMF in > rfc2833. > 2. If answer does not support telephone event, then FreeSwitch sends DTMF > inband. > > Is it possible? > > Thanks again. > > /Kaiduan > > On Thursday, March 1, 2018 3:22 AM, Alexey Sibyakin > wrote: > > > Hi, > > You need > > in dialplan and probably > > in sofia profile > > Alex > > On Thu, Mar 1, 2018 at 12:02 AM, kaiduan xie wrote: > > Hi, > > We are encountering the following DTMF issue, > > FreeSwitch adds telephone-event support in INVITE to SBC, SBC replies back > without telephone-event support in 200. > > However FreeSwitch still sends out DTMF in RFC 2833 event. > > I think FreeSwitch should send out DTMF inband instead of 2833 event, what > is the configuration for this? > > Many thanks for help, > > /Kaiduan > > ______________________________ ______________________________ _____________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www. freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch. org > http://www.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 > > http://www.freeswitch.org > > > > > -- > > Alex Sibyakin | Support Engineer > FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 > > Email: alex at freeswitch.com > Website: https://www.FreeSWITCH.com > Need commercial support? Contact sales at freeswitch.com for details. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > -- Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: alex at freeswitch.com Website: https://www.FreeSWITCH.com Need commercial support? Contact sales at freeswitch.com for details. -------------- next part -------------- An HTML attachment was scrubbed... URL: From alex at freeswitch.com Tue Mar 6 23:29:49 2018 From: alex at freeswitch.com (Alexey Sibyakin) Date: Wed, 7 Mar 2018 08:29:49 +0900 Subject: [Freeswitch-users] Media storage for Freeswitch In-Reply-To: References: Message-ID: Hi, Try to look at Ceph. Alex On Fri, Mar 2, 2018 at 5:01 PM, Melek Oktay wrote: > Hi, > > I need a solution to store and read media files from Freeswitch servers (I > have multiple Freeswitch). > Media files are recorded voicemails, recorded calls and user files like > welcome sounds, > IVR sounds, waiting music, etc. > > Currently this is solved by using a mount to a SMB share (NFS) on local > storage. This works good but I think it has a drawbacks like it is not > scalable.. > > I wonder, what kind of solution exist in the literature ? What are the > best practices that you face? > > Best Regards, > Melek > > P.S. At first a solution with a distributed file system was tried with > GlusterFS. This had some drawbacks, especially with high latency > synchronization between nodes. This approach was dropped. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > -- Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: alex at freeswitch.com Website: https://www.FreeSWITCH.com Need commercial support? Contact sales at freeswitch.com for details. -------------- next part -------------- An HTML attachment was scrubbed... URL: From jungleboogie0 at gmail.com Wed Mar 7 02:36:28 2018 From: jungleboogie0 at gmail.com (jungle boogie) Date: Tue, 6 Mar 2018 18:36:28 -0800 Subject: [Freeswitch-users] NAT / UDP hole punching issue In-Reply-To: <8DD8259F-BA2F-4CF9-A58B-B6A3CA4EDF68@gmail.com> References: <018801d3b06c$aac462f0$004d28d0$@delagarda.com> <01da01d3b077$5fe24ee0$1fa6eca0$@delagarda.com> <8DD8259F-BA2F-4CF9-A58B-B6A3CA4EDF68@gmail.com> Message-ID: <214628a7-b2bd-f748-7778-f272e293ef4a@gmail.com> Thus said Tihomir Culjaga on Tue, 6 Mar 2018 18:22:32 +0100 > So, does anyone know why FS doesnt initiate rtp stream towards the client. > > So if its not a bug, i guess its the guy in between the keyboard and chair ;) > > What m’I missing? > You could upgrade your freeswitch server and try again. Also, use something sngrep to watch the sip and rtp packets travel through the network. From alexandr.popov at iqoption.com Thu Mar 8 08:00:36 2018 From: alexandr.popov at iqoption.com (Alexandr Popov) Date: Thu, 8 Mar 2018 10:00:36 +0200 Subject: [Freeswitch-users] Custom Headers with mod_callcenter In-Reply-To: References: Message-ID: cc_export_vars 2018-03-06 1:34 GMT+02:00 Ari Sonesh : > Can anyone advise when using the mod callcenter ... how to pass custom sip > headers from the calling party call leg to the agent call leg? > thanks! > > *Ari Sonesh * > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From francesco at delagarda.com Thu Mar 8 10:12:20 2018 From: francesco at delagarda.com (Francesco Facco de Lagarda) Date: Thu, 8 Mar 2018 11:12:20 +0100 Subject: [Freeswitch-users] Javascript Message-ID: <201b01d3b6c5$f2fff6b0$d8ffe410$@delagarda.com> Good morning all.. Could someone please tell me how to initiate a call recording in javascript? Simplyfied code I have: if (session.ready()) { var sessOut = new Session("sofia/gateway/realtoneFXO/" + dialedNum + "@192.168.0.216:5060") if (sessOut.ready()) { bridge(session, sessOut); } } Any help appreciated.. have a happy and productive day and Happy Women's day for any ladies there! Francesco -------------- next part -------------- An HTML attachment was scrubbed... URL: From tculjaga at gmail.com Thu Mar 8 11:15:42 2018 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Thu, 8 Mar 2018 12:15:42 +0100 Subject: [Freeswitch-users] Javascript In-Reply-To: <201b01d3b6c5$f2fff6b0$d8ffe410$@delagarda.com> References: <201b01d3b6c5$f2fff6b0$d8ffe410$@delagarda.com> Message-ID: <49D32671-0CE1-42AC-9568-DCBEF47935E9@gmail.com> Just set bridge_pre_execute_bleg/aleg_app/data for record application before you execute bridge . Sent from my iPhone > On 8 Mar 2018, at 11:12, Francesco Facco de Lagarda wrote: > > Good morning all.. > Could someone please tell me how to initiate a call recording in javascript? > > Simplyfied code I have: > > if (session.ready()) { > var sessOut = new Session("sofia/gateway/realtoneFXO/" + dialedNum + "@192.168.0.216:5060") > if (sessOut.ready()) { > bridge(session, sessOut); > } > } > > Any help appreciated.. have a happy and productive day and Happy Women’s day for any ladies there! > > > Francesco > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From francesco at delagarda.com Thu Mar 8 14:26:41 2018 From: francesco at delagarda.com (Francesco Facco de Lagarda) Date: Thu, 8 Mar 2018 15:26:41 +0100 Subject: [Freeswitch-users] Javascript In-Reply-To: <49D32671-0CE1-42AC-9568-DCBEF47935E9@gmail.com> References: <201b01d3b6c5$f2fff6b0$d8ffe410$@delagarda.com> <49D32671-0CE1-42AC-9568-DCBEF47935E9@gmail.com> Message-ID: <206f01d3b6e9$7af8b070$70ea1150$@delagarda.com> Thank you Tihomir, I am VERY new to this.. can you please elaborate just a little more? Thanks in advance! From: FreeSWITCH-users On Behalf Of Tihomir Culjaga Sent: giovedì 8 marzo 2018 12:16 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Javascript Just set bridge_pre_execute_bleg/aleg_app/data for record application before you execute bridge . Sent from my iPhone On 8 Mar 2018, at 11:12, Francesco Facco de Lagarda > wrote: Good morning all.. Could someone please tell me how to initiate a call recording in javascript? Simplyfied code I have: if (session.ready()) { var sessOut = new Session("sofia/gateway/realtoneFXO/" + dialedNum + "@192.168.0.216:5060") if (sessOut.ready()) { bridge(session, sessOut); } } Any help appreciated.. have a happy and productive day and Happy Women’s day for any ladies there! Francesco _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.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 http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From francesco at delagarda.com Thu Mar 8 18:16:30 2018 From: francesco at delagarda.com (Francesco Facco de Lagarda) Date: Thu, 8 Mar 2018 19:16:30 +0100 Subject: [Freeswitch-users] Javascript call recording revisited Message-ID: <20d901d3b709$9616c660$c2445320$@delagarda.com> 'Evening one and all. Im going crazy trying to record calls from javascript scripting This is what I have done: I have installed mod_shout $ apt-get install freeswitch-mod-shout enable in /autoload ../modules.xml reloaded freeswitch $ service freeswitch restart Checked that module was loaded fs_cli -x "show modules" | grep shout And set it up before bridge: set "bridge_pre_execute_aleg_app record_session /tmp/test.mp3" Simplyfied code I have: if (session.ready()) { var sessOut = new Session("sofia/gateway/realtoneFXO/" + dialedNum + "@192.168.0.216:5060") if (sessOut.ready()) { sessOut.execute("set", "bridge_pre_execute_aleg_app record_session /tmp/test.mp3"); bridge(session, sessOut); } } But nothing appears in "/tmp" !!!! Where am I going wrong? -------------- next part -------------- An HTML attachment was scrubbed... URL: From melekoktay at gmail.com Fri Mar 9 06:27:42 2018 From: melekoktay at gmail.com (Melek Oktay) Date: Fri, 9 Mar 2018 07:27:42 +0100 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 141, Issue 12 In-Reply-To: References: Message-ID: Hi again, I am planing to use mod_http_cache to request (there will be posting recording file also) music/mp3/vaw files nginx&php web server that stores these files to Cassandra ! Proof of concept study has already done, ~3.5 MB file is requested by FreeSwitch, and I log file say that [INFO] mod_http_cache.c: X file downloaded in 265 ms. Cassandra store these files in 1 MB of chunks.. My question is about idea!! Cassandra is the new topic for me.. Am I pushing forward these technologies, in order to achieve my goal.. Is there any other better/easy way to do it ? My goal is providing redundancy to my media server, multi FreeSwitch could read/write file to theses media server, no single point of failure (VRRP will be used for my new Webserver), Regards, Melek On Sat, Mar 3, 2018 at 1:00 PM, < freeswitch-users-request at lists.freeswitch.org> 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. Re: Best way for storing Media for Freeswitch (Bilal Abbasi) > > > ---------- Forwarded message ---------- > From: Bilal Abbasi > To: FreeSWITCH Users Help > Cc: > Bcc: > Date: Sat, 03 Mar 2018 08:56:58 +0000 > Subject: Re: [Freeswitch-users] Best way for storing Media for Freeswitch > Hi, > I am using amazon S3 and mod_http_cache, works good for me. > > Regards > Abbasi > > On Fri, 2 Mar 2018 at 11:48 PM, Sergey Safarov > wrote: > >> You can play media files from http server >> >> >> >> >> пт, 2 мар. 2018 г. в 19:57, Emrah Ciftcibasi : >> >>> Hi, >>> >>> I need a solution to store and read media files from Freeswitch servers >>> (I have multiple Freeswitch). >>> Media files are recorded voicemails, recorded calls and user files like >>> welcome sounds, >>> IVR sounds, waiting music, etc. >>> >>> Currently this is solved by using a mount to a SMB share (NFS) on local >>> storage. This works good but I think it has a drawbacks like it is not >>> scalable.. >>> >>> I wonder, what kind of solution exist in the literature ? What are the >>> best practices that you face? >>> >>> Best Regards, >>> Melek >>> >>> P.S. At first a solution with a distributed file system was tried with >>> GlusterFS. This had some drawbacks, especially with high latency >>> synchronization between nodes. This approach was dropped. >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.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 >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.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 >> http://www.freeswitch.org > > > _______________________________________________ > 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 > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From tculjaga at gmail.com Fri Mar 9 19:59:46 2018 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Fri, 9 Mar 2018 20:59:46 +0100 Subject: [Freeswitch-users] Javascript In-Reply-To: <206f01d3b6e9$7af8b070$70ea1150$@delagarda.com> References: <201b01d3b6c5$f2fff6b0$d8ffe410$@delagarda.com> <49D32671-0CE1-42AC-9568-DCBEF47935E9@gmail.com> <206f01d3b6e9$7af8b070$70ea1150$@delagarda.com> Message-ID: i use ESL instead of JavaScript, but the logic is the same :=) try this :=) if (session.ready()) { var sessOut = new Session("sofia/gateway/realtoneFXO/" + dialedNum + "@ 192.168.0.216:5060") if (sessOut.ready()) { session.setVariable("bridge_pre_execute_aleg_app", "record_session"); session.setVariable("bridge_pre_execute_aleg_data" , "/path/to/recordingFile.wav"); bridge(session, sessOut); } } T. On 8 March 2018 at 15:26, Francesco Facco de Lagarda < francesco at delagarda.com> wrote: > Thank you Tihomir, I am VERY new to this.. can you please elaborate just a > little more? > > Thanks in advance! > > > > *From:* FreeSWITCH-users *On > Behalf Of *Tihomir Culjaga > *Sent:* giovedì 8 marzo 2018 12:16 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Javascript > > > > Just set bridge_pre_execute_bleg/aleg_app/data for record application > before you execute bridge . > > Sent from my iPhone > > > On 8 Mar 2018, at 11:12, Francesco Facco de Lagarda < > francesco at delagarda.com> wrote: > > Good morning all.. > > Could someone please tell me how to initiate a call recording in > javascript? > > > > Simplyfied code I have: > > > > if (session.ready()) { > > var sessOut = new Session("sofia/gateway/realtoneFXO/" + dialedNum + "@ > 192.168.0.216:5060") > > if (sessOut.ready()) { > > bridge(session, sessOut); > > } > > } > > > > Any help appreciated.. have a happy and productive day and Happy Women’s > day for any ladies there! > > > > > > Francesco > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From tculjaga at gmail.com Fri Mar 9 20:07:36 2018 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Fri, 9 Mar 2018 21:07:36 +0100 Subject: [Freeswitch-users] Javascript call recording revisited In-Reply-To: <20d901d3b709$9616c660$c2445320$@delagarda.com> References: <20d901d3b709$9616c660$c2445320$@delagarda.com> Message-ID: bridge_pre_execute_aleg_app pointso to application bridge_pre_execute_aleg_data points to application argument you need to set them both. also, you should set them on A-leg. and use: session.setVariable("varName","varValue") On 8 March 2018 at 19:16, Francesco Facco de Lagarda < francesco at delagarda.com> wrote: > ‘Evening one and all. > > Im going crazy trying to record calls from javascript scripting > > This is what I have done: > > > > I have installed mod_shout > > $ apt-get install freeswitch-mod-shout > > > > enable in /autoload ../modules.xml > > > > reloaded freeswitch > > $ service freeswitch restart > > > > Checked that module was loaded > > fs_cli -x "show modules" | grep shout > > > > And set it up before bridge: > > set "bridge_pre_execute_aleg_app record_session /tmp/test.mp3" > > > > > > Simplyfied code I have: > > > > if (session.ready()) { > > var sessOut = new Session("sofia/gateway/realtoneFXO/" + dialedNum + "@ > 192.168.0.216:5060") > > if (sessOut.ready()) { > > sessOut.execute("set", "bridge_pre_execute_aleg_app record_session > /tmp/test.mp3"); > > bridge(session, sessOut); > > } > > } > > > > But nothing appears in “/tmp” !!!! > > Where am I going wrong? > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From tculjaga at gmail.com Fri Mar 9 20:09:21 2018 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Fri, 9 Mar 2018 21:09:21 +0100 Subject: [Freeswitch-users] Javascript call recording revisited In-Reply-To: References: <20d901d3b709$9616c660$c2445320$@delagarda.com> Message-ID: another thing, its better not to record directly into mp3 ... takes a lot of cpu. better dump down into native or wav than to an async conversion to mp3 if needed. On 9 March 2018 at 21:07, Tihomir Culjaga wrote: > bridge_pre_execute_aleg_app pointso to application > bridge_pre_execute_aleg_data points to application argument > > you need to set them both. > > > also, you should set them on A-leg. > > and use: > > session.setVariable("varName","varValue") > > > > > > On 8 March 2018 at 19:16, Francesco Facco de Lagarda < > francesco at delagarda.com> wrote: > >> ‘Evening one and all. >> >> Im going crazy trying to record calls from javascript scripting >> >> This is what I have done: >> >> >> >> I have installed mod_shout >> >> $ apt-get install freeswitch-mod-shout >> >> >> >> enable in /autoload ../modules.xml >> >> >> >> reloaded freeswitch >> >> $ service freeswitch restart >> >> >> >> Checked that module was loaded >> >> fs_cli -x "show modules" | grep shout >> >> >> >> And set it up before bridge: >> >> set "bridge_pre_execute_aleg_app record_session /tmp/test.mp3" >> >> >> >> >> >> Simplyfied code I have: >> >> >> >> if (session.ready()) { >> >> var sessOut = new Session("sofia/gateway/realtoneFXO/" + dialedNum + "@ >> 192.168.0.216:5060") >> >> if (sessOut.ready()) { >> >> sessOut.execute("set", "bridge_pre_execute_aleg_app record_session >> /tmp/test.mp3"); >> >> bridge(session, sessOut); >> >> } >> >> } >> >> >> >> But nothing appears in “/tmp” !!!! >> >> Where am I going wrong? >> >> >> >> >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.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 >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From dujinfang at gmail.com Sat Mar 10 06:30:53 2018 From: dujinfang at gmail.com (Seven Du) Date: Sat, 10 Mar 2018 14:30:53 +0800 Subject: [Freeswitch-users] MUX mode, don't want to see my own video In-Reply-To: References: Message-ID: there's multi-canvas and personal-canvas feature, but I guess you need some external control scripts to get you to your place. On Tue, Mar 6, 2018 at 1:37 AM, Dom Rumsey wrote: > Hi Guys, > > > I have compiled FS using the master branch (on a Debian 8 Jessie) and I'm > using MUX mode for conferences with layout set to 1x1, so whoever speaks is > seen full screen. Can anyone tell me how I can get it so I don't see my own > video feed when I talk? In a multiway conversation, it would be useful for > me to see the video feed of the last person that spoke while everyone else > sees my video feed. I wasn't sure if this was easily achievable with the > correct settings. > > > Thanks in advance for any help. > > > Dom > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > -- 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 francesco at delagarda.com Sat Mar 10 11:45:38 2018 From: francesco at delagarda.com (Francesco Facco de Lagarda) Date: Sat, 10 Mar 2018 12:45:38 +0100 Subject: [Freeswitch-users] Javascript call recording revisited In-Reply-To: References: <20d901d3b709$9616c660$c2445320$@delagarda.com> Message-ID: <002401d3b865$510228a0$f30679e0$@delagarda.com> Thanks a million Tihomir, Will try immediately! Have a nice weekend wherever you are! From: FreeSWITCH-users On Behalf Of Tihomir Culjaga Sent: venerdì 9 marzo 2018 21:09 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Javascript call recording revisited another thing, its better not to record directly into mp3 ... takes a lot of cpu. better dump down into native or wav than to an async conversion to mp3 if needed. On 9 March 2018 at 21:07, Tihomir Culjaga > wrote: bridge_pre_execute_aleg_app pointso to application bridge_pre_execute_aleg_data points to application argument you need to set them both. also, you should set them on A-leg. and use: session.setVariable("varName","varValue") On 8 March 2018 at 19:16, Francesco Facco de Lagarda > wrote: ‘Evening one and all. Im going crazy trying to record calls from javascript scripting This is what I have done: I have installed mod_shout $ apt-get install freeswitch-mod-shout enable in /autoload ../modules.xml reloaded freeswitch $ service freeswitch restart Checked that module was loaded fs_cli -x "show modules" | grep shout And set it up before bridge: set "bridge_pre_execute_aleg_app record_session /tmp/test.mp3" Simplyfied code I have: if (session.ready()) { var sessOut = new Session("sofia/gateway/realtoneFXO/" + dialedNum + "@192.168.0.216:5060 ") if (sessOut.ready()) { sessOut.execute("set", "bridge_pre_execute_aleg_app record_session /tmp/test.mp3"); bridge(session, sessOut); } } But nothing appears in “/tmp” !!!! Where am I going wrong? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.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 http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From francesco at delagarda.com Sat Mar 10 11:50:22 2018 From: francesco at delagarda.com (Francesco Facco de Lagarda) Date: Sat, 10 Mar 2018 12:50:22 +0100 Subject: [Freeswitch-users] Javascript In-Reply-To: References: <201b01d3b6c5$f2fff6b0$d8ffe410$@delagarda.com> <49D32671-0CE1-42AC-9568-DCBEF47935E9@gmail.com> <206f01d3b6e9$7af8b070$70ea1150$@delagarda.com> Message-ID: <003101d3b865$f9fb4cc0$edf1e640$@delagarda.com> Yes Tihomir, It works fine! Thanks again From: FreeSWITCH-users On Behalf Of Tihomir Culjaga Sent: venerdì 9 marzo 2018 21:00 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Javascript i use ESL instead of JavaScript, but the logic is the same :=) try this :=) if (session.ready()) { var sessOut = new Session("sofia/gateway/realtoneFXO/" + dialedNum + "@ 192.168.0.216:5060") if (sessOut.ready()) { session.setVariable("bridge_pre_execute_aleg_app", "record_session"); session.setVariable("bridge_pre_execute_aleg_data" , "/path/to/recordingFile.wav"); bridge(session, sessOut); } } T. On 8 March 2018 at 15:26, Francesco Facco de Lagarda > wrote: Thank you Tihomir, I am VERY new to this.. can you please elaborate just a little more? Thanks in advance! From: FreeSWITCH-users > On Behalf Of Tihomir Culjaga Sent: giovedì 8 marzo 2018 12:16 To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Javascript Just set bridge_pre_execute_bleg/aleg_app/data for record application before you execute bridge . Sent from my iPhone On 8 Mar 2018, at 11:12, Francesco Facco de Lagarda > wrote: Good morning all.. Could someone please tell me how to initiate a call recording in javascript? Simplyfied code I have: if (session.ready()) { var sessOut = new Session("sofia/gateway/realtoneFXO/" + dialedNum + "@192.168.0.216:5060 ") if (sessOut.ready()) { bridge(session, sessOut); } } Any help appreciated.. have a happy and productive day and Happy Women’s day for any ladies there! Francesco _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.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 http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.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 http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From francesco at delagarda.com Sat Mar 10 13:20:04 2018 From: francesco at delagarda.com (Francesco Facco de Lagarda) Date: Sat, 10 Mar 2018 14:20:04 +0100 Subject: [Freeswitch-users] javascript get session extension number Message-ID: <006e01d3b872$817f4320$847dc960$@delagarda.com> Goodmorning to all. Can anyone please tell me how I can get the extension number that originated the session in javascript? -------------- next part -------------- An HTML attachment was scrubbed... URL: From francesco at delagarda.com Sat Mar 10 13:30:50 2018 From: francesco at delagarda.com (Francesco Facco de Lagarda) Date: Sat, 10 Mar 2018 14:30:50 +0100 Subject: [Freeswitch-users] javascript get session extension number In-Reply-To: <006e01d3b872$817f4320$847dc960$@delagarda.com> References: <006e01d3b872$817f4320$847dc960$@delagarda.com> Message-ID: <008001d3b874$02839010$078ab030$@delagarda.com> Solved this one myself: function dumpSessVars(session) { var svars = ["context","destination_number","caller_id_name","caller_id_number","network _addr","ani","aniii","rdnis","source","chan_name","uuid"]; svars.forEach(function (v) { cLog(1,"**************** SESSION VAR " + v + " : " + session.getVariable(v)); }); } From: FreeSWITCH-users On Behalf Of Francesco Facco de Lagarda Sent: sabato 10 marzo 2018 14:20 To: 'FreeSWITCH Users Help' Subject: [Freeswitch-users] javascript get session extension number Goodmorning to all. Can anyone please tell me how I can get the extension number that originated the session in javascript? -------------- next part -------------- An HTML attachment was scrubbed... URL: From domrumsey at hotmail.com Sun Mar 11 19:26:22 2018 From: domrumsey at hotmail.com (Dom Rumsey) Date: Sun, 11 Mar 2018 19:26:22 +0000 Subject: [Freeswitch-users] MUX mode, don't want to see my own video In-Reply-To: References: , Message-ID: Hi Seven Having video-muxing-personal-canvas in conference flags and layout=1x1 should give me the desired effect. At present it causes flickering with 3 or more people, but I've created a JIRA ticket to have the overlap fixed (https://freeswitch.org/jira/browse/FS-11010) Dom ________________________________ From: FreeSWITCH-users on behalf of Seven Du Sent: Saturday, March 10, 2018 6:30 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] MUX mode, don't want to see my own video there's multi-canvas and personal-canvas feature, but I guess you need some external control scripts to get you to your place. On Tue, Mar 6, 2018 at 1:37 AM, Dom Rumsey > wrote: Hi Guys, I have compiled FS using the master branch (on a Debian 8 Jessie) and I'm using MUX mode for conferences with layout set to 1x1, so whoever speaks is seen full screen. Can anyone tell me how I can get it so I don't see my own video feed when I talk? In a multiway conversation, it would be useful for me to see the video feed of the last person that spoke while everyone else sees my video feed. I wasn't sure if this was easily achievable with the correct settings. Thanks in advance for any help. Dom _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org [https://freeswitch.org/theme/img/freeswitch-social.png] FreeSWITCH www.freeswitch.org FreeSWITCH is an open-source media application designed to support popular protools such as SIP and WebRTC and provides a platform to develop voice and video applications. http://confluence.freeswitch.org http://www.cluecon.com [https://cluecon.com/theme/img/cluecon/hook1.jpg] ClueCon Telephony and WebRTC Developer's conference www.cluecon.com ClueCon is an annual event bringing together all of the open source Telephony and WebRTC developers to collaberate on the latest technology in the communications industry. FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users FreeSWITCH-users Info Page lists.freeswitch.org To see the collection of prior postings to the list, visit the FreeSWITCH-users Archives. Using FreeSWITCH-users: To post a message to all the list ... UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org [https://freeswitch.org/theme/img/freeswitch-social.png] FreeSWITCH www.freeswitch.org FreeSWITCH is an open-source media application designed to support popular protools such as SIP and WebRTC and provides a platform to develop voice and video applications. -- About: http://about.me/dujinfang [https://aboutme.imgix.net/background/users/d/u/j/dujinfang_gmail.com_1470281041_21.jpg?q=80&dpr=1&auto=format&fit=max&w=250&h=140&rect=0,57,240,126] Seven Du on about.me about.me I am a consultant, writer, and software engineer in Yantai China. Read my blog. Blog: http://www.dujinfang.com Seven's Blog - ^dujinfang www.dujinfang.com Seven's Blog. FreeSWITCH中文网站创始人 Proj: http://www.freeswitch.org.cn 首页 - FreeSWITCH-CN中文社区 www.freeswitch.org.cn 最新招聘. 北京联信志诚信息技术有限公司; 烟台小樱桃网络科技有限公司; 中移在线服务有限公司; 上海航动科技有限公司 -------------- next part -------------- An HTML attachment was scrubbed... URL: From rick at magicmail.mooo.com Mon Mar 12 13:17:25 2018 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Mon, 12 Mar 2018 13:17:25 +0000 Subject: [Freeswitch-users] Verto - audio only between FS and client Message-ID: <49A1D091-26B2-4DD8-86F4-E9FF63099606@magicmail.mooo.com> What is the best way to remove all call control abilities from the web client, and have just audio between FS and browser? Let me explain what I mean and why: My app server (running on Node) controls all call handling by ESL to FreeSWITCH. As the service is multi-tenant I need to consider security, but all the multi-tenant stuff is on the app server, and I’m trying to avoid the complexities of having multiple domains etc on FS as I would then need to build a way of syncing this with the app server. So, ideally, the browser would connect to FS (via Verto, as I think it is Verto that handles all the clever re-connection stuff?), and would share its call uuid with the app server, which would then handle all the call transfer / conferencing etc using ESL to FS. The sticking point is that I want to make sure that a user can’t instruct Verto to transfer the call to a guessed conference name. At the moment, I have a rather elaborate system of using a randomly generated auth code, which the app server keeps in Redis, and adds to the name of the conference. But the complexities of this are proving problematic in my code. I hope this makes sense, I definitely consider myself to still be very new to webrtc/verto etc, despite having been playing with it for many months now! Hoping for some ideas or guidance. Thanks R From mohammad.mowhabuth at umail.uom.ac.mu Mon Mar 12 11:27:38 2018 From: mohammad.mowhabuth at umail.uom.ac.mu (MOWHABUTH MOHAMMAD SHAFDAAR ALLY) Date: Mon, 12 Mar 2018 15:27:38 +0400 Subject: [Freeswitch-users] Multi-Tenant in Freeswitch Message-ID: Dear all, I have been using freeswitch in the default context for the past few months. Everything works fine in the default context. Now i want to use multi-tenancy and i have followed all the steps at https://wiki.freeswitch.org/wiki/Multiple_Companies . However, i am unable to register my softphone, with user 1000 in the company-a.org directory, to the server. I am trying on CSipSimple(softphone) the credentials as follows: Account Name: 1000 User:1000 Server: company-a.org Password: 1234 Bad Gateway is the response. Can anyone please help me with this? Mowhabuth Shafdaar BSc (Hons.) Applied Computing Faculty of Informations, Communications and Digital Technologies University of Mauritius Virus-free. www.avast.com <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2> -- -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Mon Mar 12 18:38:58 2018 From: mike at jerris.com (Michael Jerris) Date: Mon, 12 Mar 2018 14:38:58 -0400 Subject: [Freeswitch-users] Verto - audio only between FS and client In-Reply-To: <49A1D091-26B2-4DD8-86F4-E9FF63099606@magicmail.mooo.com> References: <49A1D091-26B2-4DD8-86F4-E9FF63099606@magicmail.mooo.com> Message-ID: you can just remove the features you don’t want from the client > On Mar 12, 2018, at 9:17 AM, Rick Jarvis wrote: > > What is the best way to remove all call control abilities from the web client, and have just audio between FS and browser? Let me explain what I mean and why: > > My app server (running on Node) controls all call handling by ESL to FreeSWITCH. As the service is multi-tenant I need to consider security, but all the multi-tenant stuff is on the app server, and I’m trying to avoid the complexities of having multiple domains etc on FS as I would then need to build a way of syncing this with the app server. > > So, ideally, the browser would connect to FS (via Verto, as I think it is Verto that handles all the clever re-connection stuff?), and would share its call uuid with the app server, which would then handle all the call transfer / conferencing etc using ESL to FS. > > The sticking point is that I want to make sure that a user can’t instruct Verto to transfer the call to a guessed conference name. At the moment, I have a rather elaborate system of using a randomly generated auth code, which the app server keeps in Redis, and adds to the name of the conference. But the complexities of this are proving problematic in my code. > > I hope this makes sense, I definitely consider myself to still be very new to webrtc/verto etc, despite having been playing with it for many months now! > > Hoping for some ideas or guidance. > > Thanks > R From rick at magicmail.mooo.com Mon Mar 12 18:53:34 2018 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Mon, 12 Mar 2018 18:53:34 +0000 Subject: [Freeswitch-users] Verto - audio only between FS and client In-Reply-To: References: <49A1D091-26B2-4DD8-86F4-E9FF63099606@magicmail.mooo.com> Message-ID: <5820AD15-5F71-4221-ACA6-20999299A393@magicmail.mooo.com> How would I do that, in the verto config? > On 12 Mar 2018, at 18:38, Michael Jerris wrote: > > you can just remove the features you don’t want from the client > >> On Mar 12, 2018, at 9:17 AM, Rick Jarvis wrote: >> >> What is the best way to remove all call control abilities from the web client, and have just audio between FS and browser? Let me explain what I mean and why: >> >> My app server (running on Node) controls all call handling by ESL to FreeSWITCH. As the service is multi-tenant I need to consider security, but all the multi-tenant stuff is on the app server, and I’m trying to avoid the complexities of having multiple domains etc on FS as I would then need to build a way of syncing this with the app server. >> >> So, ideally, the browser would connect to FS (via Verto, as I think it is Verto that handles all the clever re-connection stuff?), and would share its call uuid with the app server, which would then handle all the call transfer / conferencing etc using ESL to FS. >> >> The sticking point is that I want to make sure that a user can’t instruct Verto to transfer the call to a guessed conference name. At the moment, I have a rather elaborate system of using a randomly generated auth code, which the app server keeps in Redis, and adds to the name of the conference. But the complexities of this are proving problematic in my code. >> >> I hope this makes sense, I definitely consider myself to still be very new to webrtc/verto etc, despite having been playing with it for many months now! >> >> Hoping for some ideas or guidance. >> >> Thanks >> R > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org From mike at jerris.com Mon Mar 12 18:59:19 2018 From: mike at jerris.com (Michael Jerris) Date: Mon, 12 Mar 2018 14:59:19 -0400 Subject: [Freeswitch-users] Verto - audio only between FS and client In-Reply-To: <5820AD15-5F71-4221-ACA6-20999299A393@magicmail.mooo.com> References: <49A1D091-26B2-4DD8-86F4-E9FF63099606@magicmail.mooo.com> <5820AD15-5F71-4221-ACA6-20999299A393@magicmail.mooo.com> Message-ID: <832C4C52-F642-4E15-AD51-B56FA6A48BC0@jerris.com> in the source code for the client. > On Mar 12, 2018, at 2:53 PM, Rick Jarvis wrote: > > How would I do that, in the verto config? > >> On 12 Mar 2018, at 18:38, Michael Jerris wrote: >> >> you can just remove the features you don’t want from the client >> >>> On Mar 12, 2018, at 9:17 AM, Rick Jarvis wrote: >>> >>> What is the best way to remove all call control abilities from the web client, and have just audio between FS and browser? Let me explain what I mean and why: >>> >>> My app server (running on Node) controls all call handling by ESL to FreeSWITCH. As the service is multi-tenant I need to consider security, but all the multi-tenant stuff is on the app server, and I’m trying to avoid the complexities of having multiple domains etc on FS as I would then need to build a way of syncing this with the app server. >>> >>> So, ideally, the browser would connect to FS (via Verto, as I think it is Verto that handles all the clever re-connection stuff?), and would share its call uuid with the app server, which would then handle all the call transfer / conferencing etc using ESL to FS. >>> >>> The sticking point is that I want to make sure that a user can’t instruct Verto to transfer the call to a guessed conference name. At the moment, I have a rather elaborate system of using a randomly generated auth code, which the app server keeps in Redis, and adds to the name of the conference. But the complexities of this are proving problematic in my code. >>> >>> I hope this makes sense, I definitely consider myself to still be very new to webrtc/verto etc, despite having been playing with it for many months now! >>> >>> Hoping for some ideas or guidance. >>> >>> Thanks >>> R >> From rick at magicmail.mooo.com Mon Mar 12 19:27:43 2018 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Mon, 12 Mar 2018 19:27:43 +0000 Subject: [Freeswitch-users] Verto - audio only between FS and client In-Reply-To: <832C4C52-F642-4E15-AD51-B56FA6A48BC0@jerris.com> References: <49A1D091-26B2-4DD8-86F4-E9FF63099606@magicmail.mooo.com> <5820AD15-5F71-4221-ACA6-20999299A393@magicmail.mooo.com> <832C4C52-F642-4E15-AD51-B56FA6A48BC0@jerris.com> Message-ID: That would be open to someone changing the code and gaining access though > On 12 Mar 2018, at 18:59, Michael Jerris wrote: > > in the source code for the client. > >> On Mar 12, 2018, at 2:53 PM, Rick Jarvis wrote: >> >> How would I do that, in the verto config? >> >>> On 12 Mar 2018, at 18:38, Michael Jerris wrote: >>> >>> you can just remove the features you don’t want from the client >>> >>>> On Mar 12, 2018, at 9:17 AM, Rick Jarvis wrote: >>>> >>>> What is the best way to remove all call control abilities from the web client, and have just audio between FS and browser? Let me explain what I mean and why: >>>> >>>> My app server (running on Node) controls all call handling by ESL to FreeSWITCH. As the service is multi-tenant I need to consider security, but all the multi-tenant stuff is on the app server, and I’m trying to avoid the complexities of having multiple domains etc on FS as I would then need to build a way of syncing this with the app server. >>>> >>>> So, ideally, the browser would connect to FS (via Verto, as I think it is Verto that handles all the clever re-connection stuff?), and would share its call uuid with the app server, which would then handle all the call transfer / conferencing etc using ESL to FS. >>>> >>>> The sticking point is that I want to make sure that a user can’t instruct Verto to transfer the call to a guessed conference name. At the moment, I have a rather elaborate system of using a randomly generated auth code, which the app server keeps in Redis, and adds to the name of the conference. But the complexities of this are proving problematic in my code. >>>> >>>> I hope this makes sense, I definitely consider myself to still be very new to webrtc/verto etc, despite having been playing with it for many months now! >>>> >>>> Hoping for some ideas or guidance. >>>> >>>> Thanks >>>> R >>> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org From domrumsey at hotmail.com Mon Mar 12 19:35:48 2018 From: domrumsey at hotmail.com (Dom Rumsey) Date: Mon, 12 Mar 2018 19:35:48 +0000 Subject: [Freeswitch-users] Verto - audio only between FS and client In-Reply-To: <832C4C52-F642-4E15-AD51-B56FA6A48BC0@jerris.com> References: <49A1D091-26B2-4DD8-86F4-E9FF63099606@magicmail.mooo.com> <5820AD15-5F71-4221-ACA6-20999299A393@magicmail.mooo.com>, <832C4C52-F642-4E15-AD51-B56FA6A48BC0@jerris.com> Message-ID: Rick, I take it you've been through this https://evoluxbr.github.io/verto-docs/tut/transferring-a-call.html I'm fairly new to this too - we used the concept of tokens and REST API to get the users in the right session. Set useVideo to false to keep it audio only obviously. ________________________________ From: FreeSWITCH-users on behalf of Michael Jerris Sent: Monday, March 12, 2018 6:59 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Verto - audio only between FS and client in the source code for the client. > On Mar 12, 2018, at 2:53 PM, Rick Jarvis wrote: > > How would I do that, in the verto config? > >> On 12 Mar 2018, at 18:38, Michael Jerris wrote: >> >> you can just remove the features you don’t want from the client >> >>> On Mar 12, 2018, at 9:17 AM, Rick Jarvis wrote: >>> >>> What is the best way to remove all call control abilities from the web client, and have just audio between FS and browser? Let me explain what I mean and why: >>> >>> My app server (running on Node) controls all call handling by ESL to FreeSWITCH. As the service is multi-tenant I need to consider security, but all the multi-tenant stuff is on the app server, and I’m trying to avoid the complexities of having multiple domains etc on FS as I would then need to build a way of syncing this with the app server. >>> >>> So, ideally, the browser would connect to FS (via Verto, as I think it is Verto that handles all the clever re-connection stuff?), and would share its call uuid with the app server, which would then handle all the call transfer / conferencing etc using ESL to FS. >>> >>> The sticking point is that I want to make sure that a user can’t instruct Verto to transfer the call to a guessed conference name. At the moment, I have a rather elaborate system of using a randomly generated auth code, which the app server keeps in Redis, and adds to the name of the conference. But the complexities of this are proving problematic in my code. >>> >>> I hope this makes sense, I definitely consider myself to still be very new to webrtc/verto etc, despite having been playing with it for many months now! >>> >>> Hoping for some ideas or guidance. >>> >>> Thanks >>> R >> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org [https://freeswitch.org/theme/img/freeswitch-social.png] FreeSWITCH www.freeswitch.org FreeSWITCH is an open-source media application designed to support popular protools such as SIP and WebRTC and provides a platform to develop voice and video applications. http://confluence.freeswitch.org http://www.cluecon.com [https://cluecon.com/theme/img/cluecon/hook1.jpg] ClueCon Telephony and WebRTC Developer's conference www.cluecon.com ClueCon is an annual event bringing together all of the open source Telephony and WebRTC developers to collaberate on the latest technology in the communications industry. FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users FreeSWITCH-users Info Page lists.freeswitch.org To see the collection of prior postings to the list, visit the FreeSWITCH-users Archives. Using FreeSWITCH-users: To post a message to all the list ... UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org [https://freeswitch.org/theme/img/freeswitch-social.png] FreeSWITCH www.freeswitch.org FreeSWITCH is an open-source media application designed to support popular protools such as SIP and WebRTC and provides a platform to develop voice and video applications. -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Mon Mar 12 20:04:42 2018 From: mike at jerris.com (Michael Jerris) Date: Mon, 12 Mar 2018 16:04:42 -0400 Subject: [Freeswitch-users] Verto - audio only between FS and client In-Reply-To: References: <49A1D091-26B2-4DD8-86F4-E9FF63099606@magicmail.mooo.com> <5820AD15-5F71-4221-ACA6-20999299A393@magicmail.mooo.com> <832C4C52-F642-4E15-AD51-B56FA6A48BC0@jerris.com> Message-ID: There are granular security functions available on the server side. Exploring exactly what you want available and not available is a bit outside of the scope of the support we can provide here, but we could help you out with a solution as part of a professional services offering. > On Mar 12, 2018, at 3:27 PM, Rick Jarvis wrote: > > That would be open to someone changing the code and gaining access though > >> On 12 Mar 2018, at 18:59, Michael Jerris wrote: >> >> in the source code for the client. >> >>> On Mar 12, 2018, at 2:53 PM, Rick Jarvis wrote: >>> >>> How would I do that, in the verto config? >>> >>>> On 12 Mar 2018, at 18:38, Michael Jerris wrote: >>>> >>>> you can just remove the features you don’t want from the client >>>> >>>>> On Mar 12, 2018, at 9:17 AM, Rick Jarvis wrote: >>>>> >>>>> What is the best way to remove all call control abilities from the web client, and have just audio between FS and browser? Let me explain what I mean and why: >>>>> >>>>> My app server (running on Node) controls all call handling by ESL to FreeSWITCH. As the service is multi-tenant I need to consider security, but all the multi-tenant stuff is on the app server, and I’m trying to avoid the complexities of having multiple domains etc on FS as I would then need to build a way of syncing this with the app server. >>>>> >>>>> So, ideally, the browser would connect to FS (via Verto, as I think it is Verto that handles all the clever re-connection stuff?), and would share its call uuid with the app server, which would then handle all the call transfer / conferencing etc using ESL to FS. >>>>> >>>>> The sticking point is that I want to make sure that a user can’t instruct Verto to transfer the call to a guessed conference name. At the moment, I have a rather elaborate system of using a randomly generated auth code, which the app server keeps in Redis, and adds to the name of the conference. But the complexities of this are proving problematic in my code. >>>>> >>>>> I hope this makes sense, I definitely consider myself to still be very new to webrtc/verto etc, despite having been playing with it for many months now! >>>>> >>>>> Hoping for some ideas or guidance. >>>>> >>>>> Thanks >>>>> R >>>> >> From rick at magicmail.mooo.com Mon Mar 12 20:29:03 2018 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Mon, 12 Mar 2018 20:29:03 +0000 Subject: [Freeswitch-users] Verto - audio only between FS and client In-Reply-To: References: <49A1D091-26B2-4DD8-86F4-E9FF63099606@magicmail.mooo.com> <5820AD15-5F71-4221-ACA6-20999299A393@magicmail.mooo.com> <832C4C52-F642-4E15-AD51-B56FA6A48BC0@jerris.com> Message-ID: <216A9C04-5F18-400F-9396-39FF094E745B@magicmail.mooo.com> Yes, that’s excellent documentation, thanks for reminding me of that. How exactly did you use the tokens to ensure that people couldn’t invade other sessions, or was that not an issue in your context? ATM I have users connecting to conferences in the format of as the conference name, but it’s proving quite tricky to keep track of, especially if someone logs on from two browsers. Ideally I’d like to just have users connecting to ‘nothing’, with the app transferring the channel to (and no ability for the client to write their own code to force a connection to ). > On 12 Mar 2018, at 19:35, Dom Rumsey wrote: > > Rick, I take it you've been through this https://evoluxbr.github.io/verto-docs/tut/transferring-a-call.html > > I'm fairly new to this too - we used the concept of tokens and REST API to get the users in the right session. Set useVideo to false to keep it audio only obviously. > > > From: FreeSWITCH-users > on behalf of Michael Jerris > > Sent: Monday, March 12, 2018 6:59 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Verto - audio only between FS and client > > in the source code for the client. > > > On Mar 12, 2018, at 2:53 PM, Rick Jarvis > wrote: > > > > How would I do that, in the verto config? > > > >> On 12 Mar 2018, at 18:38, Michael Jerris > wrote: > >> > >> you can just remove the features you don’t want from the client > >> > >>> On Mar 12, 2018, at 9:17 AM, Rick Jarvis > wrote: > >>> > >>> What is the best way to remove all call control abilities from the web client, and have just audio between FS and browser? Let me explain what I mean and why: > >>> > >>> My app server (running on Node) controls all call handling by ESL to FreeSWITCH. As the service is multi-tenant I need to consider security, but all the multi-tenant stuff is on the app server, and I’m trying to avoid the complexities of having multiple domains etc on FS as I would then need to build a way of syncing this with the app server. > >>> > >>> So, ideally, the browser would connect to FS (via Verto, as I think it is Verto that handles all the clever re-connection stuff?), and would share its call uuid with the app server, which would then handle all the call transfer / conferencing etc using ESL to FS. > >>> > >>> The sticking point is that I want to make sure that a user can’t instruct Verto to transfer the call to a guessed conference name. At the moment, I have a rather elaborate system of using a randomly generated auth code, which the app server keeps in Redis, and adds to the name of the conference. But the complexities of this are proving problematic in my code. > >>> > >>> I hope this makes sense, I definitely consider myself to still be very new to webrtc/verto etc, despite having been playing with it for many months now! > >>> > >>> Hoping for some ideas or guidance. > >>> > >>> Thanks > >>> R > >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > > FreeSWITCH > www.freeswitch.org > FreeSWITCH is an open-source media application designed to support popular protools such as SIP and WebRTC and provides a platform to develop voice and video applications. > > > http://confluence.freeswitch.org > http://www.cluecon.com > > ClueCon Telephony and WebRTC Developer's conference > www.cluecon.com > ClueCon is an annual event bringing together all of the open source Telephony and WebRTC developers to collaberate on the latest technology in the communications industry. > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > FreeSWITCH-users Info Page > lists.freeswitch.org > To see the collection of prior postings to the list, visit the FreeSWITCH-users Archives. Using FreeSWITCH-users: To post a message to all the list ... > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > FreeSWITCH > www.freeswitch.org > FreeSWITCH is an open-source media application designed to support popular protools such as SIP and WebRTC and provides a platform to develop voice and video applications. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From domrumsey at hotmail.com Mon Mar 12 22:11:32 2018 From: domrumsey at hotmail.com (Dom Rumsey) Date: Mon, 12 Mar 2018 22:11:32 +0000 Subject: [Freeswitch-users] Verto - audio only between FS and client In-Reply-To: <216A9C04-5F18-400F-9396-39FF094E745B@magicmail.mooo.com> References: <49A1D091-26B2-4DD8-86F4-E9FF63099606@magicmail.mooo.com> <5820AD15-5F71-4221-ACA6-20999299A393@magicmail.mooo.com> <832C4C52-F642-4E15-AD51-B56FA6A48BC0@jerris.com> , <216A9C04-5F18-400F-9396-39FF094E745B@magicmail.mooo.com> Message-ID: We generate a token every time a user joins a session (which is developed using a server side scripting language) - it can only be validated once. The token and the conference name is forwarded to a VertoJS client, which validates them using a REST API. If either of them is not validated the session fails. ________________________________ From: FreeSWITCH-users on behalf of Rick Jarvis Sent: Monday, March 12, 2018 8:29 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Verto - audio only between FS and client Yes, that’s excellent documentation, thanks for reminding me of that. How exactly did you use the tokens to ensure that people couldn’t invade other sessions, or was that not an issue in your context? ATM I have users connecting to conferences in the format of as the conference name, but it’s proving quite tricky to keep track of, especially if someone logs on from two browsers. Ideally I’d like to just have users connecting to ‘nothing’, with the app transferring the channel to (and no ability for the client to write their own code to force a connection to ). On 12 Mar 2018, at 19:35, Dom Rumsey > wrote: Rick, I take it you've been through this https://evoluxbr.github.io/verto-docs/tut/transferring-a-call.html I'm fairly new to this too - we used the concept of tokens and REST API to get the users in the right session. Set useVideo to false to keep it audio only obviously. ________________________________ From: FreeSWITCH-users > on behalf of Michael Jerris > Sent: Monday, March 12, 2018 6:59 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Verto - audio only between FS and client in the source code for the client. > On Mar 12, 2018, at 2:53 PM, Rick Jarvis > wrote: > > How would I do that, in the verto config? > >> On 12 Mar 2018, at 18:38, Michael Jerris > wrote: >> >> you can just remove the features you don’t want from the client >> >>> On Mar 12, 2018, at 9:17 AM, Rick Jarvis > wrote: >>> >>> What is the best way to remove all call control abilities from the web client, and have just audio between FS and browser? Let me explain what I mean and why: >>> >>> My app server (running on Node) controls all call handling by ESL to FreeSWITCH. As the service is multi-tenant I need to consider security, but all the multi-tenant stuff is on the app server, and I’m trying to avoid the complexities of having multiple domains etc on FS as I would then need to build a way of syncing this with the app server. >>> >>> So, ideally, the browser would connect to FS (via Verto, as I think it is Verto that handles all the clever re-connection stuff?), and would share its call uuid with the app server, which would then handle all the call transfer / conferencing etc using ESL to FS. >>> >>> The sticking point is that I want to make sure that a user can’t instruct Verto to transfer the call to a guessed conference name. At the moment, I have a rather elaborate system of using a randomly generated auth code, which the app server keeps in Redis, and adds to the name of the conference. But the complexities of this are proving problematic in my code. >>> >>> I hope this makes sense, I definitely consider myself to still be very new to webrtc/verto etc, despite having been playing with it for many months now! >>> >>> Hoping for some ideas or guidance. >>> >>> Thanks >>> R >> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org [https://freeswitch.org/theme/img/freeswitch-social.png] FreeSWITCH www.freeswitch.org [https://freeswitch.org/theme/img/freeswitch-social.png] FreeSWITCH www.freeswitch.org FreeSWITCH is an open-source media application designed to support popular protools such as SIP and WebRTC and provides a platform to develop voice and video applications. FreeSWITCH is an open-source media application designed to support popular protools such as SIP and WebRTC and provides a platform to develop voice and video applications. http://confluence.freeswitch.org http://www.cluecon.com [https://cluecon.com/theme/img/cluecon/hook1.jpg] ClueCon Telephony and WebRTC Developer's conference www.cluecon.com [https://cluecon.com/theme/img/cluecon/hook1.jpg] ClueCon Telephony and WebRTC Developer's conference www.cluecon.com ClueCon is an annual event bringing together all of the open source Telephony and WebRTC developers to collaberate on the latest technology in the communications industry. ClueCon is an annual event bringing together all of the open source Telephony and WebRTC developers to collaberate on the latest technology in the communications industry. FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users FreeSWITCH-users Info Page lists.freeswitch.org lists.freeswitch.org Mailing Lists lists.freeswitch.org lists.freeswitch.org Mailing Lists: Welcome! Below is a listing of all the public mailing lists on lists.freeswitch.org. Click on a list name to get more ... To see the collection of prior postings to the list, visit the FreeSWITCH-users Archives. Using FreeSWITCH-users: To post a message to all the list ... UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org [https://freeswitch.org/theme/img/freeswitch-social.png] FreeSWITCH www.freeswitch.org FreeSWITCH is an open-source media application designed to support popular protools such as SIP and WebRTC and provides a platform to develop voice and video applications. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.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 http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From bipin at xbipin.com Tue Mar 13 14:57:29 2018 From: bipin at xbipin.com (Bipin Patel) Date: Tue, 13 Mar 2018 18:57:29 +0400 Subject: [Freeswitch-users] Sofia stops responding after a few days In-Reply-To: <0287837f-8ff5-185f-c0ec-8878a84e3f7a@xbipin.com> References: <5c53627a-9596-840e-efa0-f978f784fc7f@xbipin.com> <008c01d37b01$f96b9f40$ec42ddc0$@smartic.es> <1607ec61170.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com> <8a89ffbd-db08-1931-db1e-78b9e6b392e5@airmail.cc> <801bc750-2c8b-2dc3-698b-43265e0eb484@xbipin.com> <0287837f-8ff5-185f-c0ec-8878a84e3f7a@xbipin.com> Message-ID: hi, its been quiet some time and cant seem to get over this issue Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: Re: [Freeswitch-users] Sofia stops responding after a few days From: Bipin Patel To: FreeSWITCH Users Help Date: 1/28/2018, 9:51:23 AM > hi, > > any pointers on how to solve this? > > > Regards, > Bipin > > > ------------------------------------------------------------------------ > -------- Original Message -------- > Subject: Re: [Freeswitch-users] Sofia stops responding after a few days > From: Bipin Patel > To: FreeSWITCH Users Help > Date: 1/26/2018, 7:02:49 PM >> hi, >> >> im on the latest master but this has been happening since few months >> now and i almost checked all the modules and disabled everything >> other than xml_curl, sofia on tls which i use and its happening only >> on the sofia TLS profile >> Happens every few days at random times >> >> FreeSWITCH Version 1.9.0+git~20180119T195505Z~3f8585f636~64bit (git >> 3f8585f 2018-01-19 19:55:05Z 64bit) >> >> >> Regards, >> Bipin >> >> >> ------------------------------------------------------------------------ >> -------- Original Message -------- >> Subject: Re: [Freeswitch-users] Sofia stops responding after a few days >> From: Brian West >> To: FreeSWITCH Users Help >> Date: 1/26/2018, 6:53:56 PM >>> Critical information needed, What FreeSWITCH Revision? >>> >>> /b >>> >>> >>> On Fri, Jan 26, 2018 at 8:49 AM, Bipin Patel >> > wrote: >>> >>> hi, >>> >>> after trying a lot i still suffer the issue and when this >>> happens i see the below constantly flooding the cli >>> >>> ..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_tls.c:434 >>> tport_tls_recv() tport_type_tls(00000006A25BB0C0): tls_read() >>> returned -1 >>> ..\..\sofia-sip\libsofia-sip-ua\tport\tport.c:2864 >>> tport_recv_event() tport_recv_event(00000006A25BB0C0) >>> ..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_tls.c:434 >>> tport_tls_recv() tport_type_tls(00000006A25BB0C0): tls_read() >>> returned -1 >>> ..\..\sofia-sip\libsofia-sip-ua\tport\tport.c:2864 >>> tport_recv_event() tport_recv_event(00000006A25BB0C0) >>> ..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_tls.c:434 >>> tport_tls_recv() tport_type_tls(00000006A25BB0C0): tls_read() >>> returned -1 >>> ..\..\sofia-sip\libsofia-sip-ua\tport\tport.c:2864 >>> tport_recv_event() tport_recv_event(00000006A25BB0C0) >>> ..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_tls.c:434 >>> tport_tls_recv() tport_type_tls(00000006A25BB0C0): tls_read() >>> returned -1 >>> ..\..\sofia-sip\libsofia-sip-ua\tport\tport.c:2864 >>> tport_recv_event() tport_recv_event(00000006A25BB0C0) >>> ..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_tls.c:434 >>> tport_tls_recv() tport_type_tls(00000006A25BB0C0): tls_read() >>> returned -1 >>> ..\..\sofia-sip\libsofia-sip-ua\tport\tport.c:2864 >>> tport_recv_event() tport_recv_event(00000006A25BB0C0) >>> ..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_tls.c:434 >>> tport_tls_recv() tport_type_tls(00000006A25BB0C0): tls_read() >>> returned -1 >>> >>> >>> sorry for double posting, sent it to the wrong thread earlier >>> >>> Regards, >>> Bipin >>> >>> >>> ------------------------------------------------------------------------ >>> -------- Original Message -------- >>> Subject: Re: [Freeswitch-users] Sofia stops responding after a >>> few days >>> From: Ryan Harris >>> To: freeswitch-users at lists.freeswitch.org >>> >>> Date: 12/22/2017, 8:55:42 PM >>>> >>>> Hello, >>>> >>>> This comment in the default sip_profiles/internal.xml has >>>> caught my eye in the past: >>>> https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/conf/vanilla/sip_profiles/internal.xml#63-82 >>>> >>>> >>>> Maybe you can enable the watchdog and see if you can get a >>>> useful core dump. >>>> >>>> >>>> On 12/22/2017 10:09 AM, Bipin Patel wrote: >>>>> >>>>> Hi, >>>>> >>>>> Thanks for the reply but in my case the server already has 8gb >>>>> of ram and it's not at all busy and plus this instance of fs >>>>> hardly has less than 10 registrations at any given time. >>>>> >>>>> On December 22, 2017 12:53:04 PM Miguel Jesús López Valverde >>>>> wrote: >>>>> >>>>>> I had similar problems with FS installed under an Amazon EFS >>>>>> instance. When FS did not attend registration requests and >>>>>> executed the "sofia profile internal restart" command, it did >>>>>> not load the profile and it no longer appeared before the >>>>>> "sofia status" query. >>>>>> >>>>>> I checked that this instance was short of ram memory and I >>>>>> changed the instance to a higher one with more memory. Since >>>>>> then I have not appreciated this problem again. >>>>>> >>>>>> Receive a greeting. >>>>>> >>>>>> *De:*FreeSWITCH-users >>>>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org >>>>>> ] *En >>>>>> nombre de *Bipin Patel >>>>>> *Enviado el:* viernes, 22 de diciembre de 2017 7:46 >>>>>> *Para:* FreeSWITCH Users Help >>>>>> >>>>>> >>>>>> *Asunto:* [Freeswitch-users] Sofia stops responding after a >>>>>> few days >>>>>> >>>>>> hi, >>>>>> >>>>>> I have 2 instances of FS running on a single windows box, >>>>>> first instance uses normal sip UDP profiles mainly used for >>>>>> routing calls to carriers and its running as a service since >>>>>> a few months without any issues. The second instance runs a >>>>>> sip TLS profile and accepts inbound registrations and >>>>>> forwards calls in sip UDP ahead, it uses xml_curl for >>>>>> directory users but the problem is every 2-3 days the inbound >>>>>> TLS profile stops responding to registrations, and when that >>>>>> happens i cant even stop and restart the service, have to >>>>>> kill it and start again. FS_CLI works but doesnt show any >>>>>> error, at first i though it could be the xml_curl causing the >>>>>> issue but later realized it never sends any requests when >>>>>> sofia stops responding. >>>>>> >>>>>> i have been banging my head from the past week or so but not >>>>>> able to find the cause, could any1 help in guiding me what to >>>>>> check when this happens so can find the root cause, im using >>>>>> commercial certs for TLS and when its running there r no >>>>>> issues other than sofia stops responding every few days, it >>>>>> happens at random times. >>>>>> >>>>>> -- >>>>>> Regards, >>>>>> Bipin >>>>>> >>>>>> ------------------------------------------------------------------------ >>>>>> >>>>>> >>>>>> Libre de virus. www.avast.com >>>>>> >>>>>> >>>>>> >>>>>> <#m_7793210244807471277_DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> >>>>>> http://www.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 >>>>>> >>>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> >>>>> http://www.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 >>>>> >>>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.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 >>>> >>>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.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 >>> >>> http://www.freeswitch.org >>> >>> >>> >>> >>> -- >>> >>> 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 >>> >>> color-facebook-96.png >>> color-twitter-96.png >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.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 >>> http://www.freeswitch.org >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From internet at hernandez.eu Tue Mar 13 09:30:40 2018 From: internet at hernandez.eu (internet at hernandez.eu) Date: Tue, 13 Mar 2018 10:30:40 +0100 Subject: [Freeswitch-users] Embedded Freeswitch on ARM platform In-Reply-To: References: <49A1D091-26B2-4DD8-86F4-E9FF63099606@magicmail.mooo.com> Message-ID: Hello, We are looking to implement Freeswitch on an embedded ARM platform. This platform has 3 interfaces: - PSTN - USB with a 3g dongle - IP This platform will manage only one simultaneous call. From your experience would 256M of RAM be enough even though it’s slightly outside the Freeswitch recommendations? Many thanks in advance, Regards, Bruno From ronybeck at themenz.biz Tue Mar 13 15:03:39 2018 From: ronybeck at themenz.biz (Ronnie Beck) Date: Tue, 13 Mar 2018 16:03:39 +0100 Subject: [Freeswitch-users] Sending SIP Messages from LUA In-Reply-To: References: <4757e45e0bfeb5156b2a0d149ccaf6b5@themenz.biz> Message-ID: <3bb9f14310dc3c83d826f0dfbbba332f@themenz.biz> Hi All, I am trying to write a LUA script which will slot into a chatplan whose goal is to simple take a simple text message (SIMPLE MESSAGE) and then distribute it to multiple recipients. I wrote a script which takes some arguments (sender, recipieant list and the body) then would generate new SIP Messages to be sent to each of the recipient. The code I use to generate the message (which is more or less the example given in mod_sms for sending an SMS, not exactly what I want to do): local event = freeswitch.Event("CUSTOM", "SMS::SEND_MESSAGE"); event:addHeader("proto", "global"); event:addHeader("dest_proto", "sip"); event:addHeader("from", source); event:addHeader("to", domain.."/"..recipient); event:addHeader("type", "text/plain"); event:addHeader("skip_global_process", "true"); event:addBody( body ); event:fire(); This gives the error: [WARNING] sofia_presence.c:221 Not sending to local box for XXXX at 10.11.12.13 [ERR] sofia_presence.c:272 Chat proto [sip] from [;tag=uUQudJKEl1SKjq8wdjqwi68768] to [XXXX at 10.11.12.13] Some message text Nobody to send to: Profile internal But the message actually arrives successfully. This is fine but I would prefer not to have lots of error messages. So I tried to find a more correct way to do this. I tried the following (based on what I could find in freeswitch 1.6.20 source code): local event = freeswitch.Event("SWITCH_EVENT_SEND_MESSAGE"); event:addHeader("user", recipient ); event:addHeader("host", domain ); event:addHeader("profile", domain ); event:addHeader("subject", "SIMPLE MESSAGE"); event:addHeader("content-type", "text/plain"); event:addHeader("from", source); event:addBody( body ); event:fire(); But this gives the same error. How can I send a simple SIP Message with LUA to a locally registered user? Many thanks, Aaron -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Tue Mar 13 15:55:16 2018 From: mike at jerris.com (Michael Jerris) Date: Tue, 13 Mar 2018 11:55:16 -0400 Subject: [Freeswitch-users] Embedded Freeswitch on ARM platform In-Reply-To: References: <49A1D091-26B2-4DD8-86F4-E9FF63099606@magicmail.mooo.com> Message-ID: <26BD5348-E1C7-4C45-8E6B-05FC91E4064D@jerris.com> Freeswitch recommendations are hard to nail down due to the hugely varied uses. The best policy is to properly test all your use cases on the platform and see how it performs. It could very much depend on the hardware/software requirements for the 3g and pstn interfaces. > On Mar 13, 2018, at 5:30 AM, internet at hernandez.eu wrote: > > Hello, > > We are looking to implement Freeswitch on an embedded ARM platform. > > This platform has 3 interfaces: > - PSTN > - USB with a 3g dongle > - IP > > This platform will manage only one simultaneous call. > > From your experience would 256M of RAM be enough even though it’s slightly outside the Freeswitch recommendations? > > Many thanks in advance, > > Regards, > > Bruno > From igorolhovskiy at gmail.com Tue Mar 13 15:22:49 2018 From: igorolhovskiy at gmail.com (Igor Olhovskiy) Date: Tue, 13 Mar 2018 17:22:49 +0200 Subject: [Freeswitch-users] Embedded Freeswitch on ARM platform In-Reply-To: References: <49A1D091-26B2-4DD8-86F4-E9FF63099606@magicmail.mooo.com> Message-ID: <4cc41b16-33bb-4006-92c4-66ec5976ccd6@Spark> Would be enough. Regards, Igor On Mar 13, 2018, 5:14 PM +0200, internet at hernandez.eu, wrote: > Hello, > > We are looking to implement Freeswitch on an embedded ARM platform. > > This platform has 3 interfaces: > - PSTN > - USB with a 3g dongle > - IP > > This platform will manage only one simultaneous call. > > From your experience would 256M of RAM be enough even though it’s > slightly outside the Freeswitch recommendations? > > Many thanks in advance, > > Regards, > > Bruno > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From bipin at xbipin.com Tue Mar 13 16:01:45 2018 From: bipin at xbipin.com (Bipin Patel) Date: Tue, 13 Mar 2018 20:01:45 +0400 Subject: [Freeswitch-users] Embedded Freeswitch on ARM platform In-Reply-To: References: <49A1D091-26B2-4DD8-86F4-E9FF63099606@magicmail.mooo.com> Message-ID: hi, works flawless for me on many customer raspberry pi devices Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: [Freeswitch-users] Embedded Freeswitch on ARM platform From: internet at hernandez.eu To: FreeSWITCH Users Help Date: 3/13/2018, 1:30:40 PM > Hello, > > We are looking to implement Freeswitch on an embedded ARM platform. > > This platform has 3 interfaces: > - PSTN > - USB with a 3g dongle > - IP > > This platform will manage only one simultaneous call. > > From your experience would 256M of RAM be enough even though it’s > slightly outside the Freeswitch recommendations? > > Many thanks in advance, > > Regards, > > Bruno > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org From paul.mateer at outlook.com Tue Mar 13 16:34:18 2018 From: paul.mateer at outlook.com (Paul Mateer) Date: Tue, 13 Mar 2018 16:34:18 +0000 Subject: [Freeswitch-users] Server driven calls Message-ID: I've been using FreeSWITCH for a while and I now have a query about whether something is possible. I have clients that can place calls to one another, but it's a user driven action – one user has to deliberately place a call to the other. I have a server component which monitors certain activities and when specific events occur, I'd like the server to initiate a call between two parties. Ideally, the server would initiate the call to each party and then when the call is established (both parties have at least reached the “ringing” stage) it would plug the two calls together and step back. I think it might be possible to do this by constructing an appropriate conference call and then having the call initiator hang up, but I haven't tried this to confirm if it's possible and I'm not sure if there's a more appropriate way to achieve this. Does anyone have any thoughts on this? Sent from my Windows 10 phone -------------- next part -------------- An HTML attachment was scrubbed... URL: From grcamauer at gmail.com Tue Mar 13 16:47:45 2018 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Tue, 13 Mar 2018 13:47:45 -0300 Subject: [Freeswitch-users] Server driven calls In-Reply-To: References: Message-ID: I would use an incoming ESL connection from the server component. Look into the commands 'originate' and 'bridge'. Regards, Guillermo On Tue, Mar 13, 2018 at 1:34 PM, Paul Mateer wrote: > I've been using FreeSWITCH for a while and I now have a query about > whether something is possible. > > > > I have clients that can place calls to one another, but it's a user driven > action – one user has to deliberately place a call to the other. > > > > I have a server component which monitors certain activities and when > specific events occur, I'd like the server to initiate a call between two > parties. Ideally, the server would initiate the call to each party and then > when the call is established (both parties have at least reached the > “ringing” stage) it would plug the two calls together and step back. > > > > I think it might be possible to do this by constructing an appropriate > conference call and then having the call initiator hang up, but I haven't > tried this to confirm if it's possible and I'm not sure if there's a more > appropriate way to achieve this. > > > > Does anyone have any thoughts on this? > > > > Sent from my Windows 10 phone > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: From paul.mateer at outlook.com Tue Mar 13 17:02:44 2018 From: paul.mateer at outlook.com (Paul Mateer) Date: Tue, 13 Mar 2018 17:02:44 +0000 Subject: [Freeswitch-users] Server driven calls In-Reply-To: References: , Message-ID: Ok. Thanks Guillermo. I'll look into that. Paul Sent from my Windows 10 phone ________________________________ From: FreeSWITCH-users on behalf of Guillermo Ruiz Camauer Sent: Tuesday, March 13, 2018 4:47:45 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Server driven calls I would use an incoming ESL connection from the server component. Look into the commands 'originate' and 'bridge'. Regards, Guillermo On Tue, Mar 13, 2018 at 1:34 PM, Paul Mateer > wrote: I've been using FreeSWITCH for a while and I now have a query about whether something is possible. I have clients that can place calls to one another, but it's a user driven action – one user has to deliberately place a call to the other. I have a server component which monitors certain activities and when specific events occur, I'd like the server to initiate a call between two parties. Ideally, the server would initiate the call to each party and then when the call is established (both parties have at least reached the “ringing” stage) it would plug the two calls together and step back. I think it might be possible to do this by constructing an appropriate conference call and then having the call initiator hang up, but I haven't tried this to confirm if it's possible and I'm not sure if there's a more appropriate way to achieve this. Does anyone have any thoughts on this? Sent from my Windows 10 phone _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.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 http://www.freeswitch.org -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: From tculjaga at gmail.com Tue Mar 13 21:47:52 2018 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 13 Mar 2018 22:47:52 +0100 Subject: [Freeswitch-users] exessive bounces issue Message-ID: Hi all, this is the only list that keeps complaining about "excessive bounces" I'm subscribed to .. currently I'm subscribed to 7 lists with no bounce issues whatsoever. Im using gmail and i cannot see if any bounces are really happening... so i mailed reeswitch-users-owner at lists.freeswitch.org to come to the root of the problem but seems the mail went to /dev/null So, does anyone experiences a similar problem with freeswitch-users list ? Regards, Tihomir. -------------- next part -------------- An HTML attachment was scrubbed... URL: From juanito1982 at gmail.com Tue Mar 13 22:16:11 2018 From: juanito1982 at gmail.com (=?UTF-8?Q?Juan_Antonio_Iba=C3=B1ez_Santorum?=) Date: Tue, 13 Mar 2018 23:16:11 +0100 Subject: [Freeswitch-users] How to bridge a call to an extension without the display updating? In-Reply-To: <1512719565.3344027.1198209576.1A343915@webmail.messagingengine.com> References: <1512719565.3344027.1198209576.1A343915@webmail.messagingengine.com> Message-ID: Any idea about how to get that? Libre de virus. www.avast.com <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2> 2017-12-08 8:52 GMT+01:00 Michael Avers : > Hello, > > I'm trying to create a random dialplan extension (say 5050) that when > dialed into by a registered extension would bridge the call to another > registered user (say 3000), but not change the display > or P-Asserted-Identity. > > Right now on the bridge it sends to the calling extension > > *P-Asserted-Identity: "5050" .* > > I tried setting all the possible values for sip_cid_type and > setting ignore_display_updates to false, but it still sends the P-Asserted > update as shown above. > > The bridge itself I tried both with user/3000 at sip.domain > and ${sofia_contact(566055 at rsi.solidphone.net)} - same result. > > This is a problem because A. the display on the phone changes to "3000" > and more importantly, B. if the calling user later hits redial it will dial > 3000 instead of 5050. > > Any way to achieve what I'm trying to do? > > Thanks > Mike > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From grcamauer at gmail.com Tue Mar 13 22:35:12 2018 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Tue, 13 Mar 2018 19:35:12 -0300 Subject: [Freeswitch-users] exessive bounces issue In-Reply-To: References: Message-ID: Yes, I got the "excessive bounces" email. I am pasting it below. I just confirmed once again. Guillermo Your membership in the mailing list FreeSWITCH-users has been disabled due to excessive bounces The last bounce received from you was dated 12-Mar-2018. You will not get any more messages from this list until you re-enable your membership. You will receive 3 more reminders like this before your membership in the list is deleted. To re-enable your membership, you can simply respond to this message (leaving the Subject: line intact), or visit the confirmation page at http://lists.freeswitch.org/mailman/confirm/freeswitch-users/ f0c17d631d70fa1d55766 XXXXXXXXXXXXXXXXXXX You can also visit your membership page at http://lists.freeswitch.org/mailman/options/freeswitch- users/XXXXXXXXXXXX%40gmail.com On your membership page, you can change various delivery options such as your email address and whether you get digests or not. As a reminder, your membership password is XXXXXXXXXXXXXXX If you have any questions or problems, you can contact the list owner at freeswitch-users-owner at lists.freeswitch.org On Tue, Mar 13, 2018 at 6:47 PM, Tihomir Culjaga wrote: > Hi all, > > this is the only list that keeps complaining about "excessive bounces" > I'm subscribed to .. currently I'm subscribed to 7 lists with no bounce > issues whatsoever. > > Im using gmail and i cannot see if any bounces are really happening... so > i mailed reeswitch-users-owner at lists.freeswitch.org > to come to the root of the > problem but seems the mail went to /dev/null > > > So, does anyone experiences a similar problem with freeswitch-users list ? > > Regards, > Tihomir. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: From krice at tollfreegateway.com Tue Mar 13 23:01:20 2018 From: krice at tollfreegateway.com (Ken Rice) Date: Tue, 13 Mar 2018 19:01:20 -0400 Subject: [Freeswitch-users] exessive bounces issue In-Reply-To: References: Message-ID: <53BEF45B-7DD5-428E-8F11-DD13B040A73A@tollfreegateway.com> this has been addressed on this list already. Mailman delivers email in bulk. this means it looks up the MX for the different domains and does some network traffic optimization to deliver the emails to 1 MX (mail exchanger) at one time. here’s where the problem comes in. Google (via gmail) hosts emails for many domains all on the same MX, however, even tho the RFC’s allow multiple destinations on multiple domains for the exact same email, google does not. this issue has been around forever, however until recently those of you on a google domain never noticed it. simple reply to the email and it will clear the counters. Sent from my iPhone > On Mar 13, 2018, at 18:35, Guillermo Ruiz Camauer wrote: > > Yes, I got the "excessive bounces" email. I am pasting it below. I just confirmed once again. > > Guillermo > > > Your membership in the mailing list FreeSWITCH-users has been disabled > due to excessive bounces The last bounce received from you was dated > 12-Mar-2018. You will not get any more messages from this list until > you re-enable your membership. You will receive 3 more reminders like > this before your membership in the list is deleted. > > To re-enable your membership, you can simply respond to this message > (leaving the Subject: line intact), or visit the confirmation page at > > http://lists.freeswitch.org/mailman/confirm/freeswitch-users/f0c17d631d70fa1d55766XXXXXXXXXXXXXXXXXXX > > > You can also visit your membership page at > > http://lists.freeswitch.org/mailman/options/freeswitch-users/XXXXXXXXXXXX%40gmail.com > > > On your membership page, you can change various delivery options such > as your email address and whether you get digests or not. As a > reminder, your membership password is > > XXXXXXXXXXXXXXX > > If you have any questions or problems, you can contact the list owner > at > > freeswitch-users-owner at lists.freeswitch.org > > > >> On Tue, Mar 13, 2018 at 6:47 PM, Tihomir Culjaga wrote: >> Hi all, >> >> this is the only list that keeps complaining about "excessive bounces" I'm subscribed to .. currently I'm subscribed to 7 lists with no bounce issues whatsoever. >> >> Im using gmail and i cannot see if any bounces are really happening... so i mailed reeswitch-users-owner at lists.freeswitch.org to come to the root of the problem but seems the mail went to /dev/null >> >> >> So, does anyone experiences a similar problem with freeswitch-users list ? >> >> Regards, >> Tihomir. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.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 >> http://www.freeswitch.org > > > > -- > Guillermo Ruiz Camauer > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.com Tue Mar 13 23:19:02 2018 From: brian at freeswitch.com (Brian West) Date: Tue, 13 Mar 2018 18:19:02 -0500 Subject: [Freeswitch-users] exessive bounces issue In-Reply-To: References: Message-ID: I seen your email but was busy and forgot to reply, According to the logs, We're hitting a couple of things, One is the volume of emails we send, we hit this currently: https://support.google.com/mail/?p=ReceivingRatePerm On Tue, Mar 13, 2018 at 4:47 PM, Tihomir Culjaga wrote: > Hi all, > > this is the only list that keeps complaining about "excessive bounces" > I'm subscribed to .. currently I'm subscribed to 7 lists with no bounce > issues whatsoever. > > Im using gmail and i cannot see if any bounces are really happening... so > i mailed reeswitch-users-owner at lists.freeswitch.org > to come to the root of the > problem but seems the mail went to /dev/null > > > So, does anyone experiences a similar problem with freeswitch-users list ? > > Regards, > Tihomir. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > -- 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: color-facebook-96.png] [image: color-twitter-96.png] -------------- next part -------------- An HTML attachment was scrubbed... URL: From alex at freeswitch.com Tue Mar 13 23:36:06 2018 From: alex at freeswitch.com (Alexey Sibyakin) Date: Wed, 14 Mar 2018 08:36:06 +0900 Subject: [Freeswitch-users] Multi-Tenant in Freeswitch In-Reply-To: References: Message-ID: Hi Probably you need to add domain to your username (1000->1000 at company-a.com) Alex On Mon, Mar 12, 2018 at 8:27 PM, MOWHABUTH MOHAMMAD SHAFDAAR ALLY < mohammad.mowhabuth at umail.uom.ac.mu> wrote: > Dear all, > > I have been using freeswitch in the default context for the past few > months. Everything works fine in the default context. Now i want to use > multi-tenancy and i have followed all the steps at > https://wiki.freeswitch.org/wiki/Multiple_Companies . However, i am > unable to register my softphone, with user 1000 in the company-a.org > directory, to the server. I am trying on CSipSimple(softphone) the > credentials as follows: > Account Name: 1000 > User:1000 > Server: company-a.org > Password: 1234 > > Bad Gateway is the response. Can anyone please help me with this? > > Mowhabuth Shafdaar > BSc (Hons.) Applied Computing > Faculty of Informations, Communications and Digital Technologies > University of Mauritius > > > > Virus-free. > www.avast.com > > <#m_-3865193698141521469_DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > -- Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: alex at freeswitch.com Website: https://www.FreeSWITCH.com Need commercial support? Contact sales at freeswitch.com for details. -------------- next part -------------- An HTML attachment was scrubbed... URL: From alex at freeswitch.com Tue Mar 13 23:42:41 2018 From: alex at freeswitch.com (Alexey Sibyakin) Date: Wed, 14 Mar 2018 08:42:41 +0900 Subject: [Freeswitch-users] How to bridge a call to an extension without the display updating? In-Reply-To: References: <1512719565.3344027.1198209576.1A343915@webmail.messagingengine.com> Message-ID: Hi I used profile with these params for this: also adding something like this to dialplan may help Alex On Wed, Mar 14, 2018 at 7:16 AM, Juan Antonio Ibañez Santorum < juanito1982 at gmail.com> wrote: > Any idea about how to get that? > > > Libre > de virus. www.avast.com > > <#m_3613607242244324482_DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2> > > 2017-12-08 8:52 GMT+01:00 Michael Avers : > >> Hello, >> >> I'm trying to create a random dialplan extension (say 5050) that when >> dialed into by a registered extension would bridge the call to another >> registered user (say 3000), but not change the display >> or P-Asserted-Identity. >> >> Right now on the bridge it sends to the calling extension >> >> *P-Asserted-Identity: "5050" .* >> >> I tried setting all the possible values for sip_cid_type and >> setting ignore_display_updates to false, but it still sends the P-Asserted >> update as shown above. >> >> The bridge itself I tried both with user/3000 at sip.domain >> and ${sofia_contact(566055 at rsi.solidphone.net)} - same result. >> >> This is a problem because A. the display on the phone changes to "3000" >> and more importantly, B. if the calling user later hits redial it will dial >> 3000 instead of 5050. >> >> Any way to achieve what I'm trying to do? >> >> Thanks >> Mike >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.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 >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > -- Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: alex at freeswitch.com Website: https://www.FreeSWITCH.com Need commercial support? Contact sales at freeswitch.com for details. -------------- next part -------------- An HTML attachment was scrubbed... URL: From roman at dissauer.net Wed Mar 14 10:55:10 2018 From: roman at dissauer.net (Roman Dissauer) Date: Wed, 14 Mar 2018 11:55:10 +0100 Subject: [Freeswitch-users] start_dtmf false positives In-Reply-To: <674E93B3-D4D8-4E98-90A4-A002A8ECD26B@dissauer.net> References: <57951C1A-1F3E-487E-8CFA-EBF633CCF1FC@dissauer.net> <8a660d39-4b3b-195a-cdd4-c47ce80e3209@gmail.com> <674E93B3-D4D8-4E98-90A4-A002A8ECD26B@dissauer.net> Message-ID: <733B5C2F-E581-4A1E-9EB4-84DD1ACE9561@dissauer.net> Hi, more than one month later and no more issues with dtmf detection! Thank you for you help! Best Regards, Roman > Am 30.01.2018 um 15:05 schrieb Roman Dissauer : > > Hi, > > In a quick check I recognized that spandsp_start_dtmf and start_dtmf use same function pointer dtmf_session_function. Seems I didn’t dig deep enough, just found out that spandsp_start_dtmf uses different algorithm. > > Thanks for the hint Brian! > I’ll try out spandsp_start_dtmf and report back if issues are gone. > > Kind Regards, > Roman > > >> Am 30.01.2018 um 13:08 schrieb Brian West : >> >> spandsp_start_dtmf absolutely does NOT use libteletone, where exactly did you get that idea? >> >> On Tue, Jan 30, 2018 at 4:51 AM, Roman Dissauer wrote: >> Hi, >> >> thanks Alexey, but spandsp_start_dtmf uses exactly same libteletone under the hood. >> >> Kind Regards, >> Roman >> >> RDI SOLUTIONS e.U. >> Rosasgasse 13/24 >> 1120 Wien, Österreich >> T: +43 1 3530349 - 10 >> F: +43 1 3530349 - 99 >> roman.dissauer at rdi.at >> www.rdi.at >>> Am 30.01.2018 um 00:06 schrieb Alexey Sibyakin : >>> >>> Have you tried spandsp_start_dtmf? It has its quirks but after some tuning It worked for me much better than start_dtmf. >>> >>> On Tue, Jan 30, 2018 at 1:15 AM, Steve Underwood wrote: >>> The false positive rate for the DTMF decoder in spandsp is extremely low. Tests show it has one of the lowest false positive rates in the industry. If you are seeing significant false positives you might actually have some DTMF in the audio. Can you capture some troublesome audio to a file for analysis? >>> >>> Regards, >>> >>> Steve >>> On 01/29/2018 02:35 PM, Roman Dissauer wrote: >>>> Hi Guys, >>>> >>>> we have troubles reliably recognizing DTMF tones by using dialplan application start_dtmf. >>>> From our carriers we get some calls with inband DTMF and some with RFC2833. To our customers we need to always do RFC2833. Therefore we check SDP information and enable start_dtmf when RFC2833 is not available on the incoming call. When start_dtmf is executed we have a lot of false positives mostly at the start of audio but also during the call. >>>> >>>> Is there a way to prevent false DTMF detection? Does anyone have same issues and a solution for it? >>>> >>>> Kind regards >>>> Roman Dissauer >>>> >>>> RDI SOLUTIONS e.U. >>>> Rosasgasse 13/24 >>>> 1120 Wien, Österreich >>>> T: +43 1 3530349 - 10 >>>> F: +43 1 3530349 - 99 >>>> roman.dissauer at rdi.at >>>> www.rdi.at >>>> >>>> >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> Official FreeSWITCH Sites >>>> >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.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 >>>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.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 >>> http://www.freeswitch.org >>> >>> >>> >>> -- >>> >>> Alex Sibyakin | Support Engineer >>> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 >>> Email: alex at freeswitch.com >>> Website: https://www.FreeSWITCH.com >>> Need commercial support? Contact sales at freeswitch.com for details. >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.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 >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.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 >> http://www.freeswitch.org >> >> >> >> -- >> >> 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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.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 >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org From tg-maillistings at level5.de Wed Mar 14 11:42:09 2018 From: tg-maillistings at level5.de (=?UTF-8?Q?Thorsten_G=c3=b6llner?=) Date: Wed, 14 Mar 2018 12:42:09 +0100 Subject: [Freeswitch-users] Multi tenant / user 1 in domain A can not reach user 2 in domain B Message-ID: <7ecfb223-6e01-0869-fb4d-eb66175449bf@level5.de> Hi, I setup freeswitch with fusionpbx in a multi tenant evironment. Works fine. But currently I can not call "cross domain". So user 1 in domain A can not reach user 2 in domain B. When calling my log show the following: 2018-03-14 12:37:39.580028 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/1 at domain-A [19bbf41c-277c-11e8-81fa-4dba4b7d93a7] 2018-03-14 12:37:39.580028 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/1 at domain-A) Running State Change CS_NEW (Cur 1 Tot 84) 2018-03-14 12:37:39.580028 [DEBUG] sofia.c:9873 sofia/internal/1 at domain-A receiving invite from 1.2.3.4:54948 version: 1.6.19  64bit 2018-03-14 12:37:39.580028 [DEBUG] sofia.c:10044 IP 1.2.3.4 Rejected by acl "domains". Falling back to Digest auth. 2018-03-14 12:37:39.580028 [WARNING] sofia_reg.c:1792 SIP auth challenge (INVITE) on sofia profile 'internal' for [2 at domain-B] from ip 1.2.3.4 2018-03-14 12:37:39.580028 [DEBUG] switch_core_state_machine.c:603 (sofia/internal/1 at domain-A) State NEW 2018-03-14 12:37:39.580028 [DEBUG] sofia.c:2334 detaching session 19bbf41c-277c-11e8-81fa-4dba4b7d93a7 2018-03-14 12:37:39.619980 [DEBUG] sofia.c:2442 Re-attaching to session 19bbf41c-277c-11e8-81fa-4dba4b7d93a7 2018-03-14 12:37:39.619980 [DEBUG] sofia.c:9873 sofia/internal/1 at domain-A receiving invite from 1.2.3.4:54948 version: 1.6.19  64bit 2018-03-14 12:37:39.619980 [DEBUG] sofia.c:10044 IP 1.2.3.4 Rejected by acl "domains". Falling back to Digest auth. 2018-03-14 12:37:39.639961 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7fdb641160c0 Connected. 2018-03-14 12:37:39.639961 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7fdb641160c0 released. 2018-03-14 12:37:39.639961 [WARNING] sofia_reg.c:2906 Can't find user [12 at telequest.qcall.cloud] from 1.2.3.4 You must define a domain called 'telequest.qcall.cloud' in your directory and add a user with the id="12" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2018-03-14 12:37:39.639961 [WARNING] sofia_reg.c:1737 SIP auth failure (INVITE) on sofia profile 'internal' for [2 at domain-B] from ip 1.2.3.4 Is there a special setting or configuration I missed? Is it a ACL issue? Thanks in advance, Thorsten From brian at freeswitch.com Wed Mar 14 15:42:49 2018 From: brian at freeswitch.com (Brian West) Date: Wed, 14 Mar 2018 10:42:49 -0500 Subject: [Freeswitch-users] exessive bounces issue In-Reply-To: References: Message-ID: I think I have fixed the issue now, There were a multitude of issues at play, Spammers were also nailing the subscribe link with gmail addresses at random, I've now started blocking those, And lowered the amount of emails per message that are sent into the queue. /b On Tue, Mar 13, 2018 at 6:19 PM, Brian West wrote: > I seen your email but was busy and forgot to reply, According to the logs, > We're hitting a couple of things, One is the volume of emails we send, we > hit this currently: > > https://support.google.com/mail/?p=ReceivingRatePerm > > > > On Tue, Mar 13, 2018 at 4:47 PM, Tihomir Culjaga > wrote: > >> Hi all, >> >> this is the only list that keeps complaining about "excessive bounces" >> I'm subscribed to .. currently I'm subscribed to 7 lists with no bounce >> issues whatsoever. >> >> Im using gmail and i cannot see if any bounces are really happening... >> so i mailed reeswitch-users-owner at lists.freeswitch.org >> to come to the root of the >> problem but seems the mail went to /dev/null >> >> >> So, does anyone experiences a similar problem with freeswitch-users list >> ? >> >> Regards, >> Tihomir. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.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 >> http://www.freeswitch.org >> > > > > -- > > 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 <(918)%20424-9378> > > Website: https://www.FreeSWITCH.com > > [image: color-facebook-96.png] [image: > color-twitter-96.png] > > -- 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: color-facebook-96.png] [image: color-twitter-96.png] -------------- next part -------------- An HTML attachment was scrubbed... URL: From john at industromatic.com Tue Mar 13 20:10:10 2018 From: john at industromatic.com (John Griessen) Date: Tue, 13 Mar 2018 15:10:10 -0500 Subject: [Freeswitch-users] Embedded Freeswitch on ARM platform In-Reply-To: References: <49A1D091-26B2-4DD8-86F4-E9FF63099606@magicmail.mooo.com> Message-ID: <3f1b0993-db93-3fe1-08cb-ee1558ad8cdb@industromatic.com> On 03/13/2018 11:01 AM, Bipin Patel wrote: > hi, > > works flawless for me on many customer raspberry pi devices What are other favorite ARM platforms for FS? Perhaps with more memory? From robcapo at gmail.com Tue Mar 13 21:00:29 2018 From: robcapo at gmail.com (Rob Capo) Date: Tue, 13 Mar 2018 17:00:29 -0400 Subject: [Freeswitch-users] Setting a c= line underneath audio m= SDP Message-ID: Hi, I'm using an MRCPv2 server for TTS in a particular call environment. I have written the MRCPv2 client in a Go application that is talking to FreeSWITCH via ESL. I've set this application to directly send the audio to remote_media_ip:remote_media_port. However, some VoIP clients seem to filter the UDP data because the c= line specifies the address of the FS server instead of the MRCP server. SDP supports adding a c= line in a media section if it's different than the c= line at the top of the SDP. I'm wondering if FreeSWITCH supports doing this via setting some variable or executing some command. Thanks, Rob -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Wed Mar 14 17:09:57 2018 From: mike at jerris.com (Michael Jerris) Date: Wed, 14 Mar 2018 13:09:57 -0400 Subject: [Freeswitch-users] Setting a c= line underneath audio m= SDP In-Reply-To: References: Message-ID: <1D3F7A3A-A0D4-48FC-9D5C-7C18278D1B1B@jerris.com> the way the mrcp integrates into freeswitch, freeswitch would receive the media and relay that to the endpoint. It wouldn’t be reliable to do it directly to the endpoint without a re-invite, which wouldn’t be practical due to short duration, and needing to re-invite the media back to freeswitch. It also would remove the ability to do things such as secure media, webrtc, or non rtp endpoints to work with mrcp. > On Mar 13, 2018, at 5:00 PM, Rob Capo wrote: > > Hi, > > I'm using an MRCPv2 server for TTS in a particular call environment. I have written the MRCPv2 client in a Go application that is talking to FreeSWITCH via ESL. > > I've set this application to directly send the audio to remote_media_ip:remote_media_port. However, some VoIP clients seem to filter the UDP data because the c= line specifies the address of the FS server instead of the MRCP server. SDP supports adding a c= line in a media section if it's different than the c= line at the top of the SDP. I'm wondering if FreeSWITCH supports doing this via setting some variable or executing some command. > > Thanks, > Rob -------------- next part -------------- An HTML attachment was scrubbed... URL: From tculjaga at gmail.com Wed Mar 14 18:00:54 2018 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Wed, 14 Mar 2018 19:00:54 +0100 Subject: [Freeswitch-users] exessive bounces issue In-Reply-To: References: Message-ID: Hi Brian, thanks for your prompt action. I been busy lately and sometimes i didn't see the bounce alert message sent so i was getting "disconnected" from the mailing list. Hope its okay now and we are back to the business as usual :=) Im still facing that NAT issue but im overflowed with internal tasks and cannot dedicate more than 1 hour straight to track down that issue properly. Next week i bring up my development platform to the latest git and will see whats gonna happen... Unfortunately production is still still on 1.6.12 for obvious reasons. Tihomir. On 14 March 2018 at 16:42, Brian West wrote: > I think I have fixed the issue now, There were a multitude of issues at > play, Spammers were also nailing the subscribe link with gmail addresses at > random, I've now started blocking those, And lowered the amount of emails > per message that are sent into the queue. > > /b > > > On Tue, Mar 13, 2018 at 6:19 PM, Brian West wrote: > >> I seen your email but was busy and forgot to reply, According to the >> logs, We're hitting a couple of things, One is the volume of emails we >> send, we hit this currently: >> >> https://support.google.com/mail/?p=ReceivingRatePerm >> >> >> >> On Tue, Mar 13, 2018 at 4:47 PM, Tihomir Culjaga >> wrote: >> >>> Hi all, >>> >>> this is the only list that keeps complaining about "excessive bounces" >>> I'm subscribed to .. currently I'm subscribed to 7 lists with no bounce >>> issues whatsoever. >>> >>> Im using gmail and i cannot see if any bounces are really happening... >>> so i mailed reeswitch-users-owner at lists.freeswitch.org >>> to come to the root of >>> the problem but seems the mail went to /dev/null >>> >>> >>> So, does anyone experiences a similar problem with freeswitch-users list >>> ? >>> >>> Regards, >>> Tihomir. >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.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 >>> http://www.freeswitch.org >>> >> >> >> >> -- >> >> 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 <(918)%20424-9378> >> >> Website: https://www.FreeSWITCH.com >> >> [image: color-facebook-96.png] [image: >> color-twitter-96.png] >> >> > > > > -- > > 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: color-facebook-96.png] [image: > color-twitter-96.png] > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From tculjaga at gmail.com Wed Mar 14 18:10:07 2018 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Wed, 14 Mar 2018 19:10:07 +0100 Subject: [Freeswitch-users] Embedded Freeswitch on ARM platform In-Reply-To: <3f1b0993-db93-3fe1-08cb-ee1558ad8cdb@industromatic.com> References: <49A1D091-26B2-4DD8-86F4-E9FF63099606@magicmail.mooo.com> <3f1b0993-db93-3fe1-08cb-ee1558ad8cdb@industromatic.com> Message-ID: i like this one: https://www.pine64.org/?page_id=1194 T. On 13 March 2018 at 21:10, John Griessen wrote: > On 03/13/2018 11:01 AM, Bipin Patel wrote: > >> hi, >> >> works flawless for me on many customer raspberry pi devices >> > > What are other favorite ARM platforms for FS? Perhaps with more memory? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From abaci64 at gmail.com Wed Mar 14 20:24:08 2018 From: abaci64 at gmail.com (Abaci B) Date: Wed, 14 Mar 2018 16:24:08 -0400 Subject: [Freeswitch-users] Embedded Freeswitch on ARM platform In-Reply-To: References: <49A1D091-26B2-4DD8-86F4-E9FF63099606@magicmail.mooo.com> <3f1b0993-db93-3fe1-08cb-ee1558ad8cdb@industromatic.com> Message-ID: any one knows if there are any embedded platform that has FXO/FXS interfaces? On Wed, Mar 14, 2018 at 2:10 PM, Tihomir Culjaga wrote: > i like this one: https://www.pine64.org/?page_id=1194 > > > T. > > On 13 March 2018 at 21:10, John Griessen wrote: > >> On 03/13/2018 11:01 AM, Bipin Patel wrote: >> >>> hi, >>> >>> works flawless for me on many customer raspberry pi devices >>> >> >> What are other favorite ARM platforms for FS? Perhaps with more memory? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.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 >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From alex at freeswitch.com Thu Mar 15 04:44:02 2018 From: alex at freeswitch.com (Alexey Sibyakin) Date: Thu, 15 Mar 2018 13:44:02 +0900 Subject: [Freeswitch-users] Multi tenant / user 1 in domain A can not reach user 2 in domain B In-Reply-To: <7ecfb223-6e01-0869-fb4d-eb66175449bf@level5.de> References: <7ecfb223-6e01-0869-fb4d-eb66175449bf@level5.de> Message-ID: Hi, > Is there a special setting or configuration I missed? Your call goes to "12 at telequest.qcall.cloud" but you don't have such user. Probably your dialplan is broken. > Is it a ACL issue? Probably not. FreeSWITCH is looking into inbound acl and if call can't be authorized by ip digest auth will be performed. Alex On Wed, Mar 14, 2018 at 8:42 PM, Thorsten Göllner wrote: > Hi, > > I setup freeswitch with fusionpbx in a multi tenant evironment. Works > fine. But currently I can not call "cross domain". So user 1 in domain A > can not reach user 2 in domain B. > > When calling my log show the following: > > 2018-03-14 12:37:39.580028 [NOTICE] switch_channel.c:1104 New Channel > sofia/internal/1 at domain-A [19bbf41c-277c-11e8-81fa-4dba4b7d93a7] > 2018-03-14 12:37:39.580028 [DEBUG] switch_core_state_machine.c:584 > (sofia/internal/1 at domain-A) Running State Change CS_NEW (Cur 1 Tot 84) > 2018-03-14 12:37:39.580028 [DEBUG] sofia.c:9873 > sofia/internal/1 at domain-A receiving invite from 1.2.3.4:54948 version: > 1.6.19 64bit > 2018-03-14 12:37:39.580028 [DEBUG] sofia.c:10044 IP 1.2.3.4 Rejected by > acl "domains". Falling back to Digest auth. > 2018-03-14 12:37:39.580028 [WARNING] sofia_reg.c:1792 SIP auth challenge > (INVITE) on sofia profile 'internal' for [2 at domain-B] from ip 1.2.3.4 > 2018-03-14 12:37:39.580028 [DEBUG] switch_core_state_machine.c:603 > (sofia/internal/1 at domain-A) State NEW > 2018-03-14 12:37:39.580028 [DEBUG] sofia.c:2334 detaching session > 19bbf41c-277c-11e8-81fa-4dba4b7d93a7 > 2018-03-14 12:37:39.619980 [DEBUG] sofia.c:2442 Re-attaching to session > 19bbf41c-277c-11e8-81fa-4dba4b7d93a7 > 2018-03-14 12:37:39.619980 [DEBUG] sofia.c:9873 > sofia/internal/1 at domain-A receiving invite from 1.2.3.4:54948 version: > 1.6.19 64bit > 2018-03-14 12:37:39.619980 [DEBUG] sofia.c:10044 IP 1.2.3.4 Rejected by > acl "domains". Falling back to Digest auth. > 2018-03-14 12:37:39.639961 [DEBUG] freeswitch_lua.cpp:365 DBH handle > 0x7fdb641160c0 Connected. > 2018-03-14 12:37:39.639961 [DEBUG] freeswitch_lua.cpp:382 DBH handle > 0x7fdb641160c0 released. > 2018-03-14 12:37:39.639961 [WARNING] sofia_reg.c:2906 Can't find user > [12 at telequest.qcall.cloud] from 1.2.3.4 > You must define a domain called 'telequest.qcall.cloud' in your > directory and add a user with the id="12" attribute > and you must configure your device to use the proper domain in it's > authentication credentials. > 2018-03-14 12:37:39.639961 [WARNING] sofia_reg.c:1737 SIP auth failure > (INVITE) on sofia profile 'internal' for [2 at domain-B] from ip 1.2.3.4 > > Is there a special setting or configuration I missed? > > Is it a ACL issue? > > Thanks in advance, > Thorsten > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org -- Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: alex at freeswitch.com Website: https://www.FreeSWITCH.com Need commercial support? Contact sales at freeswitch.com for details. -------------- next part -------------- An HTML attachment was scrubbed... URL: From Alexander.Haugg at c4b.de Thu Mar 15 06:39:34 2018 From: Alexander.Haugg at c4b.de (Alexander Haugg) Date: Thu, 15 Mar 2018 06:39:34 +0000 Subject: [Freeswitch-users] Transcoding Performance OPUS <-> G722 Message-ID: <02f6100bf1bb4e70b0af3cbd861cb8a1@c4b.de> Hi, at the moment we are testing the Freewitch performance. The Freeswitch run on Windows Server 2012 R2 on a Hyper V The virtual maschine have an Intel Xeon 2,2 GHz CPU wiith 8 cores ( = max CPU performance 800%) Here some results: - 5000 SIP Registrations (reregister every 2 minutes) - 500 parallel calls (1000 channels) SRTP Opus without transcoding - Freeswitch CPU performance – 67% of possible 800% - That is a very good result! Now the transcoding part SRTP OPUS <-> RTP G711 - 500 SIP Registrations - 25 parallel calls (50 channels) SRTP OPUS <-> RTP G711 - Freeswitch CPU performance – 126% of possible 800% - 50 parallel calls (100 channels) SRTP OPUS <-> RTP G711 - Freeswitch CPU performance – 280% of possible 800% What is your experience? Are the transconding results ok, or to much? -------------- next part -------------- An HTML attachment was scrubbed... URL: From bipin at xbipin.com Thu Mar 15 12:03:40 2018 From: bipin at xbipin.com (Bipin Patel) Date: Thu, 15 Mar 2018 16:03:40 +0400 Subject: [Freeswitch-users] SRTP crypto selection Message-ID: <9579490d-150d-f42f-f99e-328cc1c75262@xbipin.com> hi, is it possible to force a selected crypto for SRTP as i cant seem to find a way to force a crypto to the user such that is phone uses that for media, secondly i see zoiper sends the below in SDP a=crypto:5 AES_256_CM_HMAC_SHA1_80 inline:854nJNIKSyhr0sJs+Ol+Qv2eSqC2gnydK8iVHUAXb4Xx7oW8iMM2jVi7R9xV6A== a=crypto:6 AES_256_CM_HMAC_SHA1_32 inline:854nJNIKSyhr0sJs+Ol+Qv2eSqC2gnydK8iVHUAXb4Xx7oW8iMM2jVi7R9xV6A== a=crypto:3 AES_192_CM_HMAC_SHA1_80 inline:854nJNIKSyhr0sJs+Ol+Qv2eSqC2gnydK8iVHUAXb4Xx7oW8iMM= a=crypto:4 AES_192_CM_HMAC_SHA1_32 inline:854nJNIKSyhr0sJs+Ol+Qv2eSqC2gnydK8iVHUAXb4Xx7oW8iMM= a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:854nJNIKSyhr0sJs+Ol+Qv2eSqC2gnydK8iVHUAX a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:854nJNIKSyhr0sJs+Ol+Qv2eSqC2gnydK8iVHUAX where as FS always uses the below only switch_core_media.c:1484 Found suite AES_CM_128_HMAC_SHA1_80 and also i noticed FS and zoiper have some variations in crypto names such as switch_core_media.c:1479 looking for crypto suite [AES_CM_192_HMAC_SHA1_32] in [4 AES_192_CM_HMAC_SHA1_32 and this never matches and never used at all (notice the placement of CM_192 in FS whereas zoiper has 192_CM) -- Regards, Bipin ------------------------------------------------------------------------ -------------- next part -------------- An HTML attachment was scrubbed... URL: From jurijs.ivolga at gmail.com Thu Mar 15 12:13:48 2018 From: jurijs.ivolga at gmail.com (Jurijs Ivolga) Date: Thu, 15 Mar 2018 14:13:48 +0200 Subject: [Freeswitch-users] SRTP crypto selection In-Reply-To: <9579490d-150d-f42f-f99e-328cc1c75262@xbipin.com> References: <9579490d-150d-f42f-f99e-328cc1c75262@xbipin.com> Message-ID: Hi, Have you checked this: https://freeswitch.org/confluence/display/FREESWITCH/rtp_secure_media With kind regards, Jurijs On Thu, Mar 15, 2018 at 2:03 PM, Bipin Patel wrote: > hi, > > is it possible to force a selected crypto for SRTP as i cant seem to find > a way to force a crypto to the user such that is phone uses that for media, > secondly i see zoiper sends the below in SDP > > a=crypto:5 AES_256_CM_HMAC_SHA1_80 inline:854nJNIKSyhr0sJs+Ol+ > Qv2eSqC2gnydK8iVHUAXb4Xx7oW8iMM2jVi7R9xV6A== > a=crypto:6 AES_256_CM_HMAC_SHA1_32 inline:854nJNIKSyhr0sJs+Ol+ > Qv2eSqC2gnydK8iVHUAXb4Xx7oW8iMM2jVi7R9xV6A== > a=crypto:3 AES_192_CM_HMAC_SHA1_80 inline:854nJNIKSyhr0sJs+Ol+ > Qv2eSqC2gnydK8iVHUAXb4Xx7oW8iMM= > a=crypto:4 AES_192_CM_HMAC_SHA1_32 inline:854nJNIKSyhr0sJs+Ol+ > Qv2eSqC2gnydK8iVHUAXb4Xx7oW8iMM= > a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:854nJNIKSyhr0sJs+Ol+ > Qv2eSqC2gnydK8iVHUAX > a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:854nJNIKSyhr0sJs+Ol+ > Qv2eSqC2gnydK8iVHUAX > > where as FS always uses the below only > > switch_core_media.c:1484 Found suite AES_CM_128_HMAC_SHA1_80 > > and also i noticed FS and zoiper have some variations in crypto names such > as > > switch_core_media.c:1479 looking for crypto suite > [AES_CM_192_HMAC_SHA1_32] in [4 AES_192_CM_HMAC_SHA1_32 > > and this never matches and never used at all (notice the placement of > CM_192 in FS whereas zoiper has 192_CM) > > > -- > Regards, > Bipin > > > ------------------------------ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bipin at xbipin.com Thu Mar 15 20:22:37 2018 From: bipin at xbipin.com (Bipin Patel) Date: Fri, 16 Mar 2018 00:22:37 +0400 Subject: [Freeswitch-users] SRTP crypto selection In-Reply-To: References: <9579490d-150d-f42f-f99e-328cc1c75262@xbipin.com> Message-ID: <1622b541ee0.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com> Hi, Thanks for that link but the other thing is the below line where zoiper sends a crypto that looks alike but the placement of characters is different so doesn't match so is zoiper doing it wrongly or it's a total different crypto that FS doesn't support switch_core_media.c:1479 looking for crypto suite [AES_CM_192_HMAC_SHA1_32] in [4 AES_192_CM_HMAC_SHA1_32 On March 15, 2018 4:15:59 PM Jurijs Ivolga wrote: Hi, Have you checked this: https://freeswitch.org/confluence/display/FREESWITCH/rtp_secure_media With kind regards, Jurijs On Thu, Mar 15, 2018 at 2:03 PM, Bipin Patel wrote: hi, is it possible to force a selected crypto for SRTP as i cant seem to find a way to force a crypto to the user such that is phone uses that for media, secondly i see zoiper sends the below in SDP a=crypto:5 AES_256_CM_HMAC_SHA1_80 inline:854nJNIKSyhr0sJs+Ol+Qv2eSqC2gnydK8iVHUAXb4Xx7oW8iMM2jVi7R9xV6A== a=crypto:6 AES_256_CM_HMAC_SHA1_32 inline:854nJNIKSyhr0sJs+Ol+Qv2eSqC2gnydK8iVHUAXb4Xx7oW8iMM2jVi7R9xV6A== a=crypto:3 AES_192_CM_HMAC_SHA1_80 inline:854nJNIKSyhr0sJs+Ol+Qv2eSqC2gnydK8iVHUAXb4Xx7oW8iMM= a=crypto:4 AES_192_CM_HMAC_SHA1_32 inline:854nJNIKSyhr0sJs+Ol+Qv2eSqC2gnydK8iVHUAXb4Xx7oW8iMM= a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:854nJNIKSyhr0sJs+Ol+Qv2eSqC2gnydK8iVHUAX a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:854nJNIKSyhr0sJs+Ol+Qv2eSqC2gnydK8iVHUAX where as FS always uses the below only switch_core_media.c:1484 Found suite AES_CM_128_HMAC_SHA1_80 and also i noticed FS and zoiper have some variations in crypto names such as switch_core_media.c:1479 looking for crypto suite [AES_CM_192_HMAC_SHA1_32] in [4 AES_192_CM_HMAC_SHA1_32 and this never matches and never used at all (notice the placement of CM_192 in FS whereas zoiper has 192_CM) -- Regards, Bipin _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.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 http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.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 http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From bipin at xbipin.com Fri Mar 16 15:56:08 2018 From: bipin at xbipin.com (Bipin Patel) Date: Fri, 16 Mar 2018 19:56:08 +0400 Subject: [Freeswitch-users] SRTP crypto selection In-Reply-To: <1622b541ee0.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com> References: <9579490d-150d-f42f-f99e-328cc1c75262@xbipin.com> <1622b541ee0.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com> Message-ID: HI, it seems many other dialers also send the crypto in a similar way so all cant be wrong i guess, should i file a bug request in FS? Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: Re: [Freeswitch-users] SRTP crypto selection From: Bipin Patel To: FreeSWITCH Users Help Date: 3/16/2018, 12:22:37 AM > Hi, > > Thanks for that link but the other thing is the below line where > zoiper sends a crypto that looks alike but the placement of characters > is different so doesn't match so is zoiper doing it wrongly or it's a > total different crypto that FS doesn't support > > switch_core_media.c:1479 looking for crypto suite > [AES_CM_192_HMAC_SHA1_32] in [4 AES_192_CM_HMAC_SHA1_32 > > > > On March 15, 2018 4:15:59 PM Jurijs Ivolga > wrote: > >> Hi, >> >> Have you checked this: >> >> https://freeswitch.org/confluence/display/FREESWITCH/rtp_secure_media >> >> With kind regards, >> >> Jurijs >> >> On Thu, Mar 15, 2018 at 2:03 PM, Bipin Patel > > wrote: >> >> hi, >> >> is it possible to force a selected crypto for SRTP as i cant seem >> to find a way to force a crypto to the user such that is phone >> uses that for media, secondly i see zoiper sends the below in SDP >> >> a=crypto:5 AES_256_CM_HMAC_SHA1_80 >> inline:854nJNIKSyhr0sJs+Ol+Qv2eSqC2gnydK8iVHUAXb4Xx7oW8iMM2jVi7R9xV6A== >> a=crypto:6 AES_256_CM_HMAC_SHA1_32 >> inline:854nJNIKSyhr0sJs+Ol+Qv2eSqC2gnydK8iVHUAXb4Xx7oW8iMM2jVi7R9xV6A== >> a=crypto:3 AES_192_CM_HMAC_SHA1_80 >> inline:854nJNIKSyhr0sJs+Ol+Qv2eSqC2gnydK8iVHUAXb4Xx7oW8iMM= >> a=crypto:4 AES_192_CM_HMAC_SHA1_32 >> inline:854nJNIKSyhr0sJs+Ol+Qv2eSqC2gnydK8iVHUAXb4Xx7oW8iMM= >> a=crypto:1 AES_CM_128_HMAC_SHA1_80 >> inline:854nJNIKSyhr0sJs+Ol+Qv2eSqC2gnydK8iVHUAX >> a=crypto:2 AES_CM_128_HMAC_SHA1_32 >> inline:854nJNIKSyhr0sJs+Ol+Qv2eSqC2gnydK8iVHUAX >> >> where as FS always uses the below only >> >> switch_core_media.c:1484 Found suite AES_CM_128_HMAC_SHA1_80 >> >> and also i noticed FS and zoiper have some variations in crypto >> names such as >> >> switch_core_media.c:1479 looking for crypto suite >> [AES_CM_192_HMAC_SHA1_32] in [4 AES_192_CM_HMAC_SHA1_32 >> >> and this never matches and never used at all (notice the >> placement of CM_192 in FS whereas zoiper has 192_CM) >> >> >> -- >> Regards, >> Bipin >> >> >> ------------------------------------------------------------------------ >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.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 >> >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.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 >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From michael at mailworks.org Mon Mar 19 08:03:28 2018 From: michael at mailworks.org (Michael Avers) Date: Mon, 19 Mar 2018 01:03:28 -0700 Subject: [Freeswitch-users] Multiple concurrent Verto calls in browser Message-ID: <1521446608.822385.1307854600.6BAD2C6E@webmail.messagingengine.com> Hello, I'm using Verto Communicator as a base for building a simple app to make/receive calls - and it works great. I'm trying to add the ability to manage multiple concurrent calls. My first thought is to create a new Verto service (in Angular) that maintains call state and reference to currently active call and otherwise keeps all calls in an array. Then to switch calls it'd place the currently active one on hold and then unhold any of the other calls in the array. Does this make sense? Any tips on handling multiple calls with Verto? Do I need a separate websocket connection for each? (they're all using the same proxy and credentials). Thanks! Mike From gregor at infomedia.si Mon Mar 19 08:56:02 2018 From: gregor at infomedia.si (Gregor Nanger) Date: Mon, 19 Mar 2018 09:56:02 +0100 Subject: [Freeswitch-users] Multiple concurrent Verto calls in browser In-Reply-To: <1521446608.822385.1307854600.6BAD2C6E@webmail.messagingengine.com> References: <1521446608.822385.1307854600.6BAD2C6E@webmail.messagingengine.com> Message-ID: Hi, we are using such scenario, but it is not based on Verto Communicator. We took core libraries (javascript) and implement GUI on our own. Everything what happens on verto triggers event and you can implement own logic and callstate for each call in array. For example, when user picks up a call, put active on hold via verto command. And for info, you can issue as many calls you want through one verto connection. 2018-03-19 9:03 GMT+01:00 Michael Avers : > Hello, > > I'm using Verto Communicator as a base for building a simple app to > make/receive calls - and it works great. > > I'm trying to add the ability to manage multiple concurrent calls. > > My first thought is to create a new Verto service (in Angular) that > maintains call state and reference to currently active call and otherwise > keeps all calls in an array. Then to switch calls it'd place the currently > active one on hold and then unhold any of the other calls in the array. > Does this make sense? > > Any tips on handling multiple calls with Verto? Do I need a separate > websocket connection for each? (they're all using the same proxy and > credentials). > > Thanks! > Mike > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org -- Gregor Nanger *CTO* t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia • www.infomedia.si -------------- next part -------------- An HTML attachment was scrubbed... URL: From mohammad.mowhabuth at umail.uom.ac.mu Wed Mar 14 19:12:01 2018 From: mohammad.mowhabuth at umail.uom.ac.mu (MOWHABUTH MOHAMMAD SHAFDAAR ALLY) Date: Wed, 14 Mar 2018 12:12:01 -0700 Subject: [Freeswitch-users] Multi-Tenant in Freeswitch In-Reply-To: References: Message-ID: Hi alex, Thank you for your response. I have been able to do it! . However, in csipsimple you need to user only the domain name (e.g ns.company-a.org) Coming to the domain, i have used bind to create a DNS which accomodates the nameserver for company-a.org. That is when i have been able to register at ns.company-a.org. Tutorian on bind at: https://help.ubuntu.com/community/BIND9ServerHowto Grateful to you for replying. Mowhabuth Shafdaar BSc (Hons.) Applied Computing Year 3 F/T Faculty of Informations, Communications and Digital Technologies University of Mauritius On Tue, Mar 13, 2018 at 4:36 PM, Alexey Sibyakin wrote: > Hi > > Probably you need to add domain to your username (1000->1000 at company-a.com > ) > > Alex > > On Mon, Mar 12, 2018 at 8:27 PM, MOWHABUTH MOHAMMAD SHAFDAAR ALLY < > mohammad.mowhabuth at umail.uom.ac.mu> wrote: > >> Dear all, >> >> I have been using freeswitch in the default context for the past few >> months. Everything works fine in the default context. Now i want to use >> multi-tenancy and i have followed all the steps at >> https://wiki.freeswitch.org/wiki/Multiple_Companies . However, i am >> unable to register my softphone, with user 1000 in the company-a.org >> directory, to the server. I am trying on CSipSimple(softphone) the >> credentials as follows: >> Account Name: 1000 >> User:1000 >> Server: company-a.org >> Password: 1234 >> >> Bad Gateway is the response. Can anyone please help me with this? >> >> Mowhabuth Shafdaar >> BSc (Hons.) Applied Computing >> Faculty of Informations, Communications and Digital Technologies >> University of Mauritius >> >> >> >> Virus-free. >> www.avast.com >> >> <#m_7696044620796561286_m_-3865193698141521469_DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.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 >> http://www.freeswitch.org >> > > > > -- > > Alex Sibyakin | Support Engineer > FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 > > Email: alex at freeswitch.com > Website: https://www.FreeSWITCH.com > Need commercial support? Contact sales at freeswitch.com for details. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > -- -------------- next part -------------- An HTML attachment was scrubbed... URL: From caioebassis at hotmail.com Fri Mar 16 17:20:07 2018 From: caioebassis at hotmail.com (Caio Assis) Date: Fri, 16 Mar 2018 17:20:07 +0000 Subject: [Freeswitch-users] Change caller ID Message-ID: Hello. I'm trying to set a different caller id, but it's not working. It show '00000' no matter what parameter or variable I change. I've tried to change effective_caller_id_name/number directly on the sip account parameters, I've tried to change it in the dialplan, but nothing seems to work. Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: From ovoshlook at gmail.com Mon Mar 19 04:20:39 2018 From: ovoshlook at gmail.com (Yuriy Gorlichenko) Date: Mon, 19 Mar 2018 07:20:39 +0300 Subject: [Freeswitch-users] sched_api conference call with flags Message-ID: Hi all I trying to call conference via sched_api but can't find any way to send flags to the conference ${sched_api +1 none conference $1-${domain} play file_string://${namefile} } so ${sched_api +1 none conference $1-${domain}+${flags} play file_string://${namefile} } cant find conference with searched name ${sched_api +1 none conference $1-${domain} ${flags} play file_string://${namefile}} Says that ${flags} is can not be used here So is here any way to call conference with flags? -------------- next part -------------- An HTML attachment was scrubbed... URL: From jungleboogie0 at gmail.com Mon Mar 19 17:08:19 2018 From: jungleboogie0 at gmail.com (jungle Boogie) Date: Mon, 19 Mar 2018 10:08:19 -0700 Subject: [Freeswitch-users] Change caller ID In-Reply-To: References: Message-ID: On 16 March 2018 at 10:20, Caio Assis wrote: > Hello. > > > > I'm trying to set a different caller id, but it's not working. It show > '00000' no matter what parameter or variable I change. > If you're editing this in vars.xml, I don't think a simple reloadxml will work. Stop freeswitch and start it back up for it to read vars.xml From hunterj91 at hotmail.com Mon Mar 19 19:41:21 2018 From: hunterj91 at hotmail.com (Jonathan Hunter) Date: Mon, 19 Mar 2018 19:41:21 +0000 Subject: [Freeswitch-users] 183 early media" followed by 180 Ringing From Mobile Carriers Message-ID: Hi Guys, Sorry to bring this up again but I see it when performing interopt testing with mobile carriers. A number of them due to ISUP interworking, have gateways that generate a 183 with SDP, then a 180ringing. I understand from ready for example this jira case FreeSWITCHFS-3859 SOFIA "183 early media" followed by "180 Ringing" where this is done by design. I just wondered if this position had changed and if there is a configuration parameter/option that can allow any early media message to be sent back to A leg by freeswitch, in particular a 183 with SDP, then a subsequent 180 ringing, as carriers tend to want to see both. Many thanks Jon -------------- next part -------------- An HTML attachment was scrubbed... URL: From mario_fs at mgtech.com Tue Mar 20 16:51:48 2018 From: mario_fs at mgtech.com (Mario) Date: Tue, 20 Mar 2018 09:51:48 -0700 Subject: [Freeswitch-users] Change caller ID In-Reply-To: References: Message-ID: Not sure I am clear on what you want, but this may help: I dynamically change the caller ID for outgoing calls, a LUA sample is below. But…. Here is the big one: your ITSP MUST support and allow this. My ITSP requires me to verify my numbers since this can be misused (spoofing is illegal). I use this to forward calls to my cell phones using the inbound caller ID to FreeSwitch so we know who is calling on the cell phone, rather than seeing the FreeSwitch line number. session:setVariable("effective_caller_id_name",call_name) > On Mar 19, 2018, at 10:08 AM, jungle Boogie wrote: > > On 16 March 2018 at 10:20, Caio Assis wrote: >> Hello. >> >> >> >> I'm trying to set a different caller id, but it's not working. It show >> '00000' no matter what parameter or variable I change. >> > > If you're editing this in vars.xml, I don't think a simple reloadxml > will work. Stop freeswitch and start it back up for it to read > vars.xml > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org From jose.lopes at itcenter.com.pt Tue Mar 20 18:22:39 2018 From: jose.lopes at itcenter.com.pt (=?UTF-8?Q?Jos=C3=A9_Lopes?=) Date: Tue, 20 Mar 2018 18:22:39 +0000 Subject: [Freeswitch-users] Ghost Ringing from Verto Client when connection interrupted Message-ID: Hello, I am using Freeswitch as WebRTC Gateway with mod_verto and mod_sofia. When some verto client suspend its laptop, the verto contact was keept on verto status (fs_cli -x 'verto status') . When verto client suspend its laptop, the connection is interrupted, the verto client didn't close the connection. When some SIP User make a call to that verto client, the call is ringing (It receives SIP Ringing). But the call never arrive to verto client. There is any way to avoid the SIP Ringing to SIP User, since this cause the feeling that it is ringing on verto client? Thanks in advance. Best Regards, Jose Lopes -------------- next part -------------- An HTML attachment was scrubbed... URL: From joelists at tm.net.uk Tue Mar 20 22:22:14 2018 From: joelists at tm.net.uk (Joseph Waite) Date: Tue, 20 Mar 2018 22:22:14 +0000 Subject: [Freeswitch-users] Extract variables from 300 redirect Message-ID: Hi Guys I am wondering if its possible to extract a custom variable from a 300 Redirect response from a billing system? Effectively I am trying to add a custom variable to the 300 message to give a maximum call length to then use in the dial plan for a Scheduled Hangup. Any pointers of how to do this or the magic words to search confluence/google with? Regards From tculjaga at gmail.com Wed Mar 21 06:19:47 2018 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Wed, 21 Mar 2018 07:19:47 +0100 Subject: [Freeswitch-users] Multiple concurrent Verto calls in browser In-Reply-To: References: <1521446608.822385.1307854600.6BAD2C6E@webmail.messagingengine.com> Message-ID: <7B8D444E-7087-450C-96B8-AC7755BAB74D@gmail.com> @Michael, We did the same thing as Gregor, implemented verto api in our own web interface... a single wss is enough :) @Gregor, sorry to hijack the mail ;) But did you happen to be able to set googDscp in RTCPeerConnection and chrome actually to mark rtp packets? My chrome (latest) on win10 keeps ignoring it :(. Tihomir. Sent from my iPhone > On 19 Mar 2018, at 09:56, Gregor Nanger wrote: > > Hi, > > we are using such scenario, but it is not based on Verto Communicator. We took core libraries (javascript) and implement GUI on our own. Everything what happens on verto triggers event and you can implement own logic and callstate for each call in array. For example, when user picks up a call, put active on hold via verto command. > > And for info, you can issue as many calls you want through one verto connection. > > > > > 2018-03-19 9:03 GMT+01:00 Michael Avers : >> Hello, >> >> I'm using Verto Communicator as a base for building a simple app to make/receive calls - and it works great. >> >> I'm trying to add the ability to manage multiple concurrent calls. >> >> My first thought is to create a new Verto service (in Angular) that maintains call state and reference to currently active call and otherwise keeps all calls in an array. Then to switch calls it'd place the currently active one on hold and then unhold any of the other calls in the array. Does this make sense? >> >> Any tips on handling multiple calls with Verto? Do I need a separate websocket connection for each? (they're all using the same proxy and credentials). >> >> Thanks! >> Mike >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.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 >> http://www.freeswitch.org > > > > -- > Gregor Nanger > > CTO > t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 > • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia > • www.infomedia.si > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From tculjaga at gmail.com Wed Mar 21 07:01:37 2018 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Wed, 21 Mar 2018 08:01:37 +0100 Subject: [Freeswitch-users] 183 early media" followed by 180 Ringing From Mobile Carriers In-Reply-To: References: Message-ID: <444A2AFF-2D39-4122-B57C-2BEC12AB8DEC@gmail.com> Hi, im not sure about the callflow you have so i suppose its an incoming call from a carrier to freeswitch and you want to generate an inband ringback to the calling user. In that case you use pre_answer followed by ring_ready. Of course you set ringback var to a tone or wav file so freeswitch can actually generate that inband. Sent from my iPhone > On 19 Mar 2018, at 20:41, Jonathan Hunter wrote: > > Hi Guys, > > Sorry to bring this up again but I see it when performing interopt testing with mobile carriers. > > A number of them due to ISUP interworking, have gateways that generate a 183 with SDP, then a 180ringing. > > I understand from ready for example this jira case FreeSWITCHFS-3859 SOFIA "183 early media" followed by "180 Ringing" where this is done by design. > > I just wondered if this position had changed and if there is a configuration parameter/option that can allow any early media message to be sent back to A leg by freeswitch, in particular a 183 with SDP, then a subsequent 180 ringing, as carriers tend to want to see both. > > Many thanks > > Jon > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From hunterj91 at hotmail.com Wed Mar 21 08:14:29 2018 From: hunterj91 at hotmail.com (Jonathan Hunter) Date: Wed, 21 Mar 2018 08:14:29 +0000 Subject: [Freeswitch-users] 183 early media" followed by 180 Ringing From Mobile Carriers In-Reply-To: <444A2AFF-2D39-4122-B57C-2BEC12AB8DEC@gmail.com> References: , <444A2AFF-2D39-4122-B57C-2BEC12AB8DEC@gmail.com> Message-ID: Hi Tihomir, Thankyou very much for the reply. I have tried pre_answer (which sends the 183 with sdp) and the ring_ready (which as you know generates a 180 ringing ) but I only ever see the 183 sent back to the A leg, not both as I thought this was as design in Sofia? Does this work for you or am I missing something? Thanks! Jon Get Outlook for iOS _____________________________ From: Tihomir Culjaga > Sent: Wednesday, March 21, 2018 7:02 am Subject: Re: [Freeswitch-users] 183 early media" followed by 180 Ringing From Mobile Carriers To: FreeSWITCH Users Help > Hi, im not sure about the callflow you have so i suppose its an incoming call from a carrier to freeswitch and you want to generate an inband ringback to the calling user. In that case you use pre_answer followed by ring_ready. Of course you set ringback var to a tone or wav file so freeswitch can actually generate that inband. Sent from my iPhone On 19 Mar 2018, at 20:41, Jonathan Hunter > wrote: Hi Guys, Sorry to bring this up again but I see it when performing interopt testing with mobile carriers. A number of them due to ISUP interworking, have gateways that generate a 183 with SDP, then a 180ringing. I understand from ready for example this jira case FreeSWITCHFS-3859 SOFIA "183 early media" followed by "180 Ringing" where this is done by design. I just wondered if this position had changed and if there is a configuration parameter/option that can allow any early media message to be sent back to A leg by freeswitch, in particular a 183 with SDP, then a subsequent 180 ringing, as carriers tend to want to see both. Many thanks Jon _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.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 http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From gregor at infomedia.si Wed Mar 21 10:36:52 2018 From: gregor at infomedia.si (Gregor Nanger) Date: Wed, 21 Mar 2018 11:36:52 +0100 Subject: [Freeswitch-users] Multiple concurrent Verto calls in browser In-Reply-To: <7B8D444E-7087-450C-96B8-AC7755BAB74D@gmail.com> References: <1521446608.822385.1307854600.6BAD2C6E@webmail.messagingengine.com> <7B8D444E-7087-450C-96B8-AC7755BAB74D@gmail.com> Message-ID: @Tihomir Sorry, to advanced question for me :-) We just insert new video canvas for each call and hide, since we only use audio and keep own array list with call guid and states as we want. Gregor 2018-03-21 7:19 GMT+01:00 Tihomir Culjaga : > @Michael, > We did the same thing as Gregor, implemented verto api in our own web > interface... a single wss is enough :) > > @Gregor, sorry to hijack the mail ;) > > But did you happen to be able to set googDscp in RTCPeerConnection and > chrome actually to mark rtp packets? > My chrome (latest) on win10 keeps ignoring it :(. > > Tihomir. > > > > Sent from my iPhone > > On 19 Mar 2018, at 09:56, Gregor Nanger wrote: > > Hi, > > we are using such scenario, but it is not based on Verto Communicator. We > took core libraries (javascript) and implement GUI on our own. Everything > what happens on verto triggers event and you can implement own logic and > callstate for each call in array. For example, when user picks up a call, > put active on hold via verto command. > > And for info, you can issue as many calls you want through one verto > connection. > > > > 2018-03-19 9:03 GMT+01:00 Michael Avers : > >> Hello, >> >> I'm using Verto Communicator as a base for building a simple app to >> make/receive calls - and it works great. >> >> I'm trying to add the ability to manage multiple concurrent calls. >> >> My first thought is to create a new Verto service (in Angular) that >> maintains call state and reference to currently active call and otherwise >> keeps all calls in an array. Then to switch calls it'd place the currently >> active one on hold and then unhold any of the other calls in the array. >> Does this make sense? >> >> Any tips on handling multiple calls with Verto? Do I need a separate >> websocket connection for each? (they're all using the same proxy and >> credentials). >> >> Thanks! >> Mike >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.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 >> http://www.freeswitch.org > > > > > -- > Gregor Nanger > > *CTO* > t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 > • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia > • www.infomedia.si > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > -- Gregor Nanger *CTO* t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia • www.infomedia.si -------------- next part -------------- An HTML attachment was scrubbed... URL: From tculjaga at gmail.com Wed Mar 21 12:31:11 2018 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Wed, 21 Mar 2018 13:31:11 +0100 Subject: [Freeswitch-users] Change caller ID In-Reply-To: References: Message-ID: <56840B3A-E6FE-4C8F-9174-D5DCF645EEAE@gmail.com> Pls provide the dialplan and pastebin of fs logs. This is trivial. Sent from my iPhone > On 20 Mar 2018, at 17:51, Mario wrote: > > Not sure I am clear on what you want, but this may help: I dynamically change the caller ID for outgoing calls, a LUA sample is below. But…. Here is the big one: your ITSP MUST support and allow this. My ITSP requires me to verify my numbers since this can be misused (spoofing is illegal). I use this to forward calls to my cell phones using the inbound caller ID to FreeSwitch so we know who is calling on the cell phone, rather than seeing the FreeSwitch line number. > > session:setVariable("effective_caller_id_name",call_name) > >> On Mar 19, 2018, at 10:08 AM, jungle Boogie wrote: >> >> On 16 March 2018 at 10:20, Caio Assis wrote: >>> Hello. >>> >>> >>> >>> I'm trying to set a different caller id, but it's not working. It show >>> '00000' no matter what parameter or variable I change. >>> >> >> If you're editing this in vars.xml, I don't think a simple reloadxml >> will work. Stop freeswitch and start it back up for it to read >> vars.xml >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.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 >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org From giggsey at gmail.com Wed Mar 21 17:27:49 2018 From: giggsey at gmail.com (Joshua Gigg) Date: Wed, 21 Mar 2018 17:27:49 +0000 Subject: [Freeswitch-users] Subscribing to ESL Events 'dynamically' per channel Message-ID: Hi, We're using the inbound ESL socket to subscribe to certain events. However, we receive these events for every call. Is it possible to dynamically subscribe to events per call via the API? We're hoping to add/remove certain events at certain points during our call flows to avoid unnecessary noise in the events received. -------------- next part -------------- An HTML attachment was scrubbed... URL: From grcamauer at gmail.com Wed Mar 21 22:09:06 2018 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Wed, 21 Mar 2018 19:09:06 -0300 Subject: [Freeswitch-users] Subscribing to ESL Events 'dynamically' per channel In-Reply-To: References: Message-ID: Joshua, Maybe this can help (from https://wiki.freeswitch.org/wiki/Event_Socket#event): filter Specify event types to listen for. Note, this is not a filter out but rather a "filter in," that is, when a filter is applied only the filtered values are received. Multiple filters on a socket connection are allowed. Usage: filter Example: The following example will subscribe to all events and then create two filters, one to listen for HEARTBEATS and one to listen for CHANNEL_EXECUTE events. events plain all Content-Type: command/reply Reply-Text: +OK event listener enabled plain filter Event-Name CHANNEL_EXECUTE Content-Type: command/reply Reply-Text: +OK filter added. [filter]=[Event-Name CHANNEL_EXECUTE] filter Event-Name HEARTBEAT Content-Type: command/reply Reply-Text: +OK filter added. [Event-Name]=[HEARTBEAT] Now only HEARTBEAT and CHANNEL_EXECUTE events will be received. You can filter on any of the event headers. To filter for a specific channel you will need to use the uuid: filter Unique-ID d29a070f-40ff-43d8-8b9d-d369b2389dfe This method is an alternative to the myevents event type. If you need *only* the events for a specific channel then use *myevents*, otherwise use a combination of filters to narrow down the events you wish to receive on the socket. To filter multiple unique IDs, you can just add another filter for events for each UUID. This can be useful for example if you want to receive start/stop-talking events for multiple users on a particular conference. filter plain all filter plain CUSTOM conference::maintenance filter Unique-ID $participantB filter Unique-ID $participantA filter Unique-ID $participantC This will give you events for Participant A,B and C on any conference. To receive events for all users on a conference you can use something like: filter Conference-Unique-ID $ConfUUID You can filter on any of the parameters you get in a freeSWITCH event: filter plain all filter call-direction Inbound filter Event-Calling-File mod_conference.c filter Conference-Unique-ID $ConfUUID You can use them individually or compound them depending on whatever end result you desire for the type of events you want to receive filter delete Specify the events which you want to revoke the filter. filter delete can be used when some filters are applied wrongly or when there is no use of the filter. Usage: filter delete Example: filter delete Event-Name HEARTBEAT Now, you will no longer receive HEARTBEAT events. You can delete any filter that is applied by this way. filter delete Unique-ID d29a070f-40ff-43d8-8b9d-d369b2389dfe This is to delete the filter which is applied for the given unique-id. After this, you won't receive any events for this unique-id. filter delete Unique-ID This deletes all the filters which are applied based on the unique-id. Regards, Guillermo On Wed, Mar 21, 2018 at 2:27 PM, Joshua Gigg wrote: > Hi, > > We're using the inbound ESL socket to subscribe to certain events. > However, we receive these events for every call. > > Is it possible to dynamically subscribe to events per call via the API? > > We're hoping to add/remove certain events at certain points during our > call flows to avoid unnecessary noise in the events received. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: From ssinyagin at gmail.com Fri Mar 23 13:17:04 2018 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Fri, 23 Mar 2018 14:17:04 +0100 Subject: [Freeswitch-users] Server driven calls In-Reply-To: References: Message-ID: see two_step_dial here: https://github.com/voxserv/freeswitch-helper-scripts/tree/master/esl it's very likely what you want. On Tue, Mar 13, 2018 at 5:34 PM, Paul Mateer wrote: > I've been using FreeSWITCH for a while and I now have a query about whether > something is possible. > > > > I have clients that can place calls to one another, but it's a user driven > action – one user has to deliberately place a call to the other. > > > > I have a server component which monitors certain activities and when > specific events occur, I'd like the server to initiate a call between two > parties. Ideally, the server would initiate the call to each party and then > when the call is established (both parties have at least reached the > “ringing” stage) it would plug the two calls together and step back. > > > > I think it might be possible to do this by constructing an appropriate > conference call and then having the call initiator hang up, but I haven't > tried this to confirm if it's possible and I'm not sure if there's a more > appropriate way to achieve this. > > > > Does anyone have any thoughts on this? > > > > Sent from my Windows 10 phone > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org From mike at jerris.com Fri Mar 23 15:33:41 2018 From: mike at jerris.com (Michael Jerris) Date: Fri, 23 Mar 2018 11:33:41 -0400 Subject: [Freeswitch-users] Extract variables from 300 redirect In-Reply-To: References: Message-ID: <77D358F6-9D88-45D4-8B7C-59A58AFF93EC@jerris.com> We don’t currently capture those, but it could be done with code modifications. > On Mar 20, 2018, at 6:22 PM, Joseph Waite wrote: > > Hi Guys > > I am wondering if its possible to extract a custom variable from a 300 Redirect response from a billing system? > > Effectively I am trying to add a custom variable to the 300 message to give a maximum call length to then use in the dial plan for a Scheduled Hangup. > > Any pointers of how to do this or the magic words to search confluence/google with? > > Regards From bipin at xbipin.com Sun Mar 25 06:17:57 2018 From: bipin at xbipin.com (Bipin Patel) Date: Sun, 25 Mar 2018 10:17:57 +0400 Subject: [Freeswitch-users] SRTP crypto selection In-Reply-To: References: <9579490d-150d-f42f-f99e-328cc1c75262@xbipin.com> <1622b541ee0.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com> Message-ID: <4543c486-848b-11f5-1ede-6f3b1bc6cb23@xbipin.com> hi, i had reported this to zoiper team and they said they have talked to FS team about this and its related to crypto naming convention in the RFC standards and it should be fixed in FS so eagerly waiting for the patch Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: Re: [Freeswitch-users] SRTP crypto selection From: Bipin Patel To: FreeSWITCH Users Help Date: 3/16/2018, 7:56:08 PM > HI, > > it seems many other dialers also send the crypto in a similar way so > all cant be wrong i guess, should i file a bug request in FS? > > > Regards, > Bipin > > > ------------------------------------------------------------------------ > -------- Original Message -------- > Subject: Re: [Freeswitch-users] SRTP crypto selection > From: Bipin Patel > To: FreeSWITCH Users Help > Date: 3/16/2018, 12:22:37 AM >> Hi, >> >> Thanks for that link but the other thing is the below line where >> zoiper sends a crypto that looks alike but the placement of >> characters is different so doesn't match so is zoiper doing it >> wrongly or it's a total different crypto that FS doesn't support >> >> switch_core_media.c:1479 looking for crypto suite >> [AES_CM_192_HMAC_SHA1_32] in [4 AES_192_CM_HMAC_SHA1_32 >> >> >> >> On March 15, 2018 4:15:59 PM Jurijs Ivolga >> wrote: >> >>> Hi, >>> >>> Have you checked this: >>> >>> https://freeswitch.org/confluence/display/FREESWITCH/rtp_secure_media >>> >>> With kind regards, >>> >>> Jurijs >>> >>> On Thu, Mar 15, 2018 at 2:03 PM, Bipin Patel >> > wrote: >>> >>> hi, >>> >>> is it possible to force a selected crypto for SRTP as i cant >>> seem to find a way to force a crypto to the user such that is >>> phone uses that for media, secondly i see zoiper sends the below >>> in SDP >>> >>> a=crypto:5 AES_256_CM_HMAC_SHA1_80 >>> inline:854nJNIKSyhr0sJs+Ol+Qv2eSqC2gnydK8iVHUAXb4Xx7oW8iMM2jVi7R9xV6A== >>> a=crypto:6 AES_256_CM_HMAC_SHA1_32 >>> inline:854nJNIKSyhr0sJs+Ol+Qv2eSqC2gnydK8iVHUAXb4Xx7oW8iMM2jVi7R9xV6A== >>> a=crypto:3 AES_192_CM_HMAC_SHA1_80 >>> inline:854nJNIKSyhr0sJs+Ol+Qv2eSqC2gnydK8iVHUAXb4Xx7oW8iMM= >>> a=crypto:4 AES_192_CM_HMAC_SHA1_32 >>> inline:854nJNIKSyhr0sJs+Ol+Qv2eSqC2gnydK8iVHUAXb4Xx7oW8iMM= >>> a=crypto:1 AES_CM_128_HMAC_SHA1_80 >>> inline:854nJNIKSyhr0sJs+Ol+Qv2eSqC2gnydK8iVHUAX >>> a=crypto:2 AES_CM_128_HMAC_SHA1_32 >>> inline:854nJNIKSyhr0sJs+Ol+Qv2eSqC2gnydK8iVHUAX >>> >>> where as FS always uses the below only >>> >>> switch_core_media.c:1484 Found suite AES_CM_128_HMAC_SHA1_80 >>> >>> and also i noticed FS and zoiper have some variations in crypto >>> names such as >>> >>> switch_core_media.c:1479 looking for crypto suite >>> [AES_CM_192_HMAC_SHA1_32] in [4 AES_192_CM_HMAC_SHA1_32 >>> >>> and this never matches and never used at all (notice the >>> placement of CM_192 in FS whereas zoiper has 192_CM) >>> >>> >>> -- >>> Regards, >>> Bipin >>> >>> >>> ------------------------------------------------------------------------ >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.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 >>> >>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.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 >>> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.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 >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Sun Mar 25 15:55:53 2018 From: s.safarov at gmail.com (Sergey Safarov) Date: Sun, 25 Mar 2018 15:55:53 +0000 Subject: [Freeswitch-users] Presense, FS as client Message-ID: Hello I want subscribe to BLF events of several extension on remote FS server. I configured gateway with subscription as it described "Presence " page. My FS servers is subscribed to extension same as gateway username. Is possible to define "To" header in subscription other than gateway username? -------------- next part -------------- An HTML attachment was scrubbed... URL: From joel at gogii.net Sun Mar 25 16:07:36 2018 From: joel at gogii.net (Joel Serrano) Date: Sun, 25 Mar 2018 09:07:36 -0700 Subject: [Freeswitch-users] SRTP crypto selection In-Reply-To: <4543c486-848b-11f5-1ede-6f3b1bc6cb23@xbipin.com> References: <9579490d-150d-f42f-f99e-328cc1c75262@xbipin.com> <1622b541ee0.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com> <4543c486-848b-11f5-1ede-6f3b1bc6cb23@xbipin.com> Message-ID: Not sure if this is related: https://freeswitch.org/jira/browse/FS-11052 On Sat, Mar 24, 2018 at 11:17 PM, Bipin Patel wrote: > hi, > > i had reported this to zoiper team and they said they have talked to FS > team about this and its related to crypto naming convention in the RFC > standards and it should be fixed in FS so eagerly waiting for the patch > > > Regards, > Bipin > > > ------------------------------ > -------- Original Message -------- > Subject: Re: [Freeswitch-users] SRTP crypto selection > From: Bipin Patel > To: FreeSWITCH Users Help > > Date: 3/16/2018, 7:56:08 PM > > HI, > > it seems many other dialers also send the crypto in a similar way so all > cant be wrong i guess, should i file a bug request in FS? > > > Regards, > Bipin > > > ------------------------------ > -------- Original Message -------- > Subject: Re: [Freeswitch-users] SRTP crypto selection > From: Bipin Patel > To: FreeSWITCH Users Help > > Date: 3/16/2018, 12:22:37 AM > > Hi, > > Thanks for that link but the other thing is the below line where zoiper > sends a crypto that looks alike but the placement of characters is > different so doesn't match so is zoiper doing it wrongly or it's a total > different crypto that FS doesn't support > > switch_core_media.c:1479 looking for crypto suite > [AES_CM_192_HMAC_SHA1_32] in [4 AES_192_CM_HMAC_SHA1_32 > > > > On March 15, 2018 4:15:59 PM Jurijs Ivolga > wrote: > >> Hi, >> >> Have you checked this: >> >> https://freeswitch.org/confluence/display/FREESWITCH/rtp_secure_media >> >> With kind regards, >> >> Jurijs >> >> On Thu, Mar 15, 2018 at 2:03 PM, Bipin Patel wrote: >> >>> hi, >>> >>> is it possible to force a selected crypto for SRTP as i cant seem to >>> find a way to force a crypto to the user such that is phone uses that for >>> media, secondly i see zoiper sends the below in SDP >>> >>> a=crypto:5 AES_256_CM_HMAC_SHA1_80 inline:854nJNIKSyhr0sJs+Ol+Qv2 >>> eSqC2gnydK8iVHUAXb4Xx7oW8iMM2jVi7R9xV6A== >>> a=crypto:6 AES_256_CM_HMAC_SHA1_32 inline:854nJNIKSyhr0sJs+Ol+Qv2 >>> eSqC2gnydK8iVHUAXb4Xx7oW8iMM2jVi7R9xV6A== >>> a=crypto:3 AES_192_CM_HMAC_SHA1_80 inline:854nJNIKSyhr0sJs+Ol+Qv2 >>> eSqC2gnydK8iVHUAXb4Xx7oW8iMM= >>> a=crypto:4 AES_192_CM_HMAC_SHA1_32 inline:854nJNIKSyhr0sJs+Ol+Qv2 >>> eSqC2gnydK8iVHUAXb4Xx7oW8iMM= >>> a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:854nJNIKSyhr0sJs+Ol+Qv2 >>> eSqC2gnydK8iVHUAX >>> a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:854nJNIKSyhr0sJs+Ol+Qv2 >>> eSqC2gnydK8iVHUAX >>> >>> where as FS always uses the below only >>> >>> switch_core_media.c:1484 Found suite AES_CM_128_HMAC_SHA1_80 >>> >>> and also i noticed FS and zoiper have some variations in crypto names >>> such as >>> >>> switch_core_media.c:1479 looking for crypto suite >>> [AES_CM_192_HMAC_SHA1_32] in [4 AES_192_CM_HMAC_SHA1_32 >>> >>> and this never matches and never used at all (notice the placement of >>> CM_192 in FS whereas zoiper has 192_CM) >>> >>> >>> -- >>> Regards, >>> Bipin >>> >>> >>> ------------------------------ >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.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 >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.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 >> http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bipin at xbipin.com Sun Mar 25 19:36:43 2018 From: bipin at xbipin.com (Bipin Patel) Date: Sun, 25 Mar 2018 23:36:43 +0400 Subject: [Freeswitch-users] SRTP crypto selection In-Reply-To: References: <9579490d-150d-f42f-f99e-328cc1c75262@xbipin.com> <1622b541ee0.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com> <4543c486-848b-11f5-1ede-6f3b1bc6cb23@xbipin.com> Message-ID: <1625ea9b4f8.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com> Hi, Seems like it's zoiper guys only who might have created that bug request coz I was going to do the same but this makes things easier for me and yes it's the same thing I'm talking about so far. On March 25, 2018 8:22:34 PM Joel Serrano wrote: Not sure if this is related: https://freeswitch.org/jira/browse/FS-11052 On Sat, Mar 24, 2018 at 11:17 PM, Bipin Patel wrote: hi, i had reported this to zoiper team and they said they have talked to FS team about this and its related to crypto naming convention in the RFC standards and it should be fixed in FS so eagerly waiting for the patch Regards, Bipin -------- Original Message -------- Subject: Re: [Freeswitch-users] SRTP crypto selection From: Bipin Patel To: FreeSWITCH Users Help Date: 3/16/2018, 7:56:08 PM HI, it seems many other dialers also send the crypto in a similar way so all cant be wrong i guess, should i file a bug request in FS? Regards, Bipin -------- Original Message -------- Subject: Re: [Freeswitch-users] SRTP crypto selection From: Bipin Patel To: FreeSWITCH Users Help Date: 3/16/2018, 12:22:37 AM Hi, Thanks for that link but the other thing is the below line where zoiper sends a crypto that looks alike but the placement of characters is different so doesn't match so is zoiper doing it wrongly or it's a total different crypto that FS doesn't support switch_core_media.c:1479 looking for crypto suite [AES_CM_192_HMAC_SHA1_32] in [4 AES_192_CM_HMAC_SHA1_32 On March 15, 2018 4:15:59 PM Jurijs Ivolga wrote: Hi, Have you checked this: https://freeswitch.org/confluence/display/FREESWITCH/rtp_secure_media With kind regards, Jurijs On Thu, Mar 15, 2018 at 2:03 PM, Bipin Patel wrote: hi, is it possible to force a selected crypto for SRTP as i cant seem to find a way to force a crypto to the user such that is phone uses that for media, secondly i see zoiper sends the below in SDP a=crypto:5 AES_256_CM_HMAC_SHA1_80 inline:854nJNIKSyhr0sJs+Ol+Qv2eSqC2gnydK8iVHUAXb4Xx7oW8iMM2jVi7R9xV6A== a=crypto:6 AES_256_CM_HMAC_SHA1_32 inline:854nJNIKSyhr0sJs+Ol+Qv2eSqC2gnydK8iVHUAXb4Xx7oW8iMM2jVi7R9xV6A== a=crypto:3 AES_192_CM_HMAC_SHA1_80 inline:854nJNIKSyhr0sJs+Ol+Qv2eSqC2gnydK8iVHUAXb4Xx7oW8iMM= a=crypto:4 AES_192_CM_HMAC_SHA1_32 inline:854nJNIKSyhr0sJs+Ol+Qv2eSqC2gnydK8iVHUAXb4Xx7oW8iMM= a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:854nJNIKSyhr0sJs+Ol+Qv2eSqC2gnydK8iVHUAX a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:854nJNIKSyhr0sJs+Ol+Qv2eSqC2gnydK8iVHUAX where as FS always uses the below only switch_core_media.c:1484 Found suite AES_CM_128_HMAC_SHA1_80 and also i noticed FS and zoiper have some variations in crypto names such as switch_core_media.c:1479 looking for crypto suite [AES_CM_192_HMAC_SHA1_32] in [4 AES_192_CM_HMAC_SHA1_32 and this never matches and never used at all (notice the placement of CM_192 in FS whereas zoiper has 192_CM) -- Regards, Bipin _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.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 http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.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 http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.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 http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.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 http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.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 http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.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 http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From gregor at infomedia.si Mon Mar 26 13:11:07 2018 From: gregor at infomedia.si (Gregor Nanger) Date: Mon, 26 Mar 2018 15:11:07 +0200 Subject: [Freeswitch-users] sip_hangup_disposition Message-ID: Hi! I am bridging incoming call to provider. In CDR of b leg I see send_cancel. But how do I know which side initiated cancel (originator or provider)? Thank you, Gregor -------------- next part -------------- An HTML attachment was scrubbed... URL: From gdoermann at perfectpitchtech.com Fri Mar 23 22:53:51 2018 From: gdoermann at perfectpitchtech.com (Greg Doermann) Date: Fri, 23 Mar 2018 16:53:51 -0600 Subject: [Freeswitch-users] mod_com_amd Message-ID: We have been using waitforresult to try and wait for AMD to come back with something but we are having issues with it hanging (never returning from that) and locking up a license). We have also tried using the amd_execute_on_person command as it doesn't seem to be working. We originally had the script setup to run amd_execute_on_person but that was never hit and we would end up just waiting the full timeout each time before checking the result: (amd.lua) dst_number = session:getVariable("sip_to_user") > initial_wait = session:getVariable("ppamd_initial_wait") > max_wait = session:getVariable("ppamd_max_wait") > if not max_wait then > max_wait = 3000 > end > while (session:ready() and not session:answered()) do > -- Waiting for answer. > session:sleep(500) > -- freeswitch.msleep(500) > end > function on_finished(reason) > if reason == nil then reason = "unspecified" end > session:setVariable("ppamd_reason", reason) > session:execute("transfer", "process_amd XML default") > end > if session:ready() and session:answered() then > freeswitch.consoleLog("INFO", string.format("AMD Enable on %s.\n", > dst_number)) > if initial_wait then > session:sleep(initial_wait) > end > local use_amd = api:executeString("amd_available") > if use_amd == "true" then > -- Be sure to set these variables. They will stop the wait and > immediatly transfer the call if it gets a hit > session:setVariable("amd_execute_on_machine", "transfer > amd_machine_detected XML default") > session:setVariable("amd_execute_on_person", "transfer > amd_person_detected XML default") > session:setVariable("amd_execute_on_unsure", "transfer > amd_unsure_detected XML default") > session:execute("voice_start") > -- Giving some time to AMD to work on the call. > -- If a voicemail is detected, it will auto-transfer from the > variables that are set. > session:sleep(max_wait) > local amd_finished_called = > session:getVariable("amd_finished_called") > if amd_finished_called == 'true' then > freeswitch.consoleLog("INFO", string.format("AMD Finish script > already called!\n")) > else > -- NOTE: Hopefully this never gets called as the module should > auto transfer this > freeswitch.consoleLog("INFO", string.format("AMD Timeout > reached for vmd: %s\n", max_wait)) > session:execute("voice_stop") > local amd_detect = session:getVariable("amd_status") > freeswitch.consoleLog("INFO", string.format("AMD amd_status: > %s\n", amd_detect)) > on_finished(amd_detect) > end > else > freeswitch.consoleLog("WARNING", string.format("AMD not available > for %s\n", dst_number)) > on_finished("unavailable") > end > else > freeswitch.consoleLog("WARNING", string.format("AMD Failed to run > %s\n", dst_number)) > on_finished("failure") > return > end the amd_execute_on_XXX does not seem to do anything... The sleep would always max out and we would always hit the Timeout reached part of the script instead of getting transferred. After that didn't work because we would always wait the sleep time (3 seconds) we scrapped transfers and tried handling the whole thing through a single lua script using waitforresult: DEFAULT_MAX_WAIT = 3000 -- in (ms) > DEFAULT_INITIAL_WAIT = 100 -- in (ms) > HUMAN_EXTENSION = "amd_queue" > UNSURE_EXTENSION = "amd_queue" > api = freeswitch.API() > > dst_number = session:getVariable("sip_to_user") > initial_wait = session:getVariable("ppamd_initial_wait") or > DEFAULT_INITIAL_WAIT > max_wait = session:getVariable("ppamd_max_wait") or DEFAULT_MAX_WAIT > > while (session:ready() and not session:answered()) do > session:sleep(500) > end > > freeswitch.consoleLog("INFO", string.format("AMD Attempting on %s. \n", > dst_number)) > if initial_wait then > session:sleep(initial_wait) > end > > if session:ready() and session:answered() then > local use_amd = api:executeString("amd_available") > if use_amd == "true" then > freeswitch.consoleLog("INFO", string.format("AMD Available Max > Wait: %s, Initial Wait %s. \n", max_wait, initial_wait)) > -- Allow AMD to run and analyze. > freeswitch.consoleLog("INFO", "AMD Voice start. \n") > session:execute("voice_start") > freeswitch.consoleLog("INFO", "AMD Wait for result. \n") > session:execute("waitforresult", string.format("%s", max_wait)) > freeswitch.consoleLog("INFO", "AMD Voice stop. \n") > session:execute("voice_stop") > amd_detect = session:getVariable("amd_status") > freeswitch.consoleLog("INFO", string.format("AMD Result: %s. \n", > amd_detect)) > if amd_detect == "machine" then > freeswitch.consoleLog("INFO", "AMD Detected machine. \n") > session:hangup() > return > elseif amd_detect == "person" then > freeswitch.consoleLog("INFO", "AMD Detected human, transfering > call. \n") > session:execute("transfer", string.format("%s XML default", > HUMAN_EXTENSION)) > return > else > freeswitch.consoleLog("INFO", string.format("AMD Unknown: %s. > \n", amd_detect)) > session:execute("transfer", string.format("%s XML default", > UNSURE_EXTENSION)) > return > end > end > else > freeswitch.consoleLog("INFO", "AMD not enabled. \n") > session:execute("transfer", string.format("%s XML default", > UNSURE_EXTENSION)) > return > end I prefer not having to do all that transferring around but to just run with waitforresult but it never comes back... The dialplan extensions just connect the call to an agent. Once we do the transfer it all works. The only problem is we have to replace the: session:execute("waitforresult", string.format("%s", max_wait)) with a sleep so no matter if the mod_com_amd detects its a person or not we ALWAYS have to wait the full sleep time. This is not really legal as we would be essentially abandoning every single call... Any thoughts on what we may be missing or how we could improve this? -------------- next part -------------- An HTML attachment was scrubbed... URL: From morteza.bashsiz at gmail.com Sun Mar 25 11:39:04 2018 From: morteza.bashsiz at gmail.com (Morteza Bashsiz) Date: Sun, 25 Mar 2018 16:09:04 +0430 Subject: [Freeswitch-users] using ssh connection in module python in dialplan Message-ID: Hi everybody I have an asterisk(172.16.1.33) and freeswitch(172.16.1.30) My scenario : 1. Calls come to freeswitch ivr menu 2. In ivr calls transfer to asterisk after running an python script freeswitch start recording voice 3. In asterisk calls push to an queue my python script is : ############################################################ import paramiko hostname = '172.16.1.33' port = 3022 username = 'root' pkey_file = '/root/.ssh/id_rsa' def get_agent_id(num): command="asterisk -rx \"pjsip show channels\" | grep -a1 \"CLCID: \\\"%s\\\"\" | grep Channel | cut -d \/ -f2 | cut -d\- -f1 " % (num) key = paramiko.RSAKey.from_private_key_file(pkey_file) s = paramiko.SSHClient() s.load_system_host_keys() s.connect(hostname, port, pkey=key) stdin, stdout, stderr = s.exec_command(command) result=str(stdout.read()) s.close() return result #if __name__ == "__main__": # agent_id=get_agent_id('1000') # print (agent_id) import freeswitch as freeswitch def handler(session, args): caller_num = session.getVariable("caller") agent_num = get_agent_id(caller_num) session.setVariable("AGENT_NUM", agent_num); session.setVariable("CALLER_NUM", caller_num); ############################################################ my dial plan is : ############################################################ ############################################################ *MY PROBLEM IS:* when i want to use my variable AGENT_NUM it is null BUT when i use the python manually it returns true value please help me what is my mistake ?????? -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.com Mon Mar 26 18:15:18 2018 From: brian at freeswitch.com (Brian West) Date: Mon, 26 Mar 2018 13:15:18 -0500 Subject: [Freeswitch-users] mod_com_amd In-Reply-To: References: Message-ID: please send your request to support at freeswitch.com /b On Fri, Mar 23, 2018 at 5:53 PM, Greg Doermann < gdoermann at perfectpitchtech.com> wrote: > We have been using waitforresult to try and wait for AMD to come back > with something but we are having issues with it hanging (never returning > from that) and locking up a license). We have also tried using > the amd_execute_on_person command as it doesn't seem to be working. > > We originally had the script setup to run amd_execute_on_person but that > was never hit and we would end up just waiting the full timeout each time > before checking the result: > > (amd.lua) > > dst_number = session:getVariable("sip_to_user") >> initial_wait = session:getVariable("ppamd_initial_wait") >> max_wait = session:getVariable("ppamd_max_wait") >> if not max_wait then >> max_wait = 3000 >> end >> while (session:ready() and not session:answered()) do >> -- Waiting for answer. >> session:sleep(500) >> -- freeswitch.msleep(500) >> end >> function on_finished(reason) >> if reason == nil then reason = "unspecified" end >> session:setVariable("ppamd_reason", reason) >> session:execute("transfer", "process_amd XML default") >> end >> if session:ready() and session:answered() then >> freeswitch.consoleLog("INFO", string.format("AMD Enable on %s.\n", >> dst_number)) >> if initial_wait then >> session:sleep(initial_wait) >> end >> local use_amd = api:executeString("amd_available") >> if use_amd == "true" then >> -- Be sure to set these variables. They will stop the wait and >> immediatly transfer the call if it gets a hit >> session:setVariable("amd_execute_on_machine", "transfer >> amd_machine_detected XML default") >> session:setVariable("amd_execute_on_person", "transfer >> amd_person_detected XML default") >> session:setVariable("amd_execute_on_unsure", "transfer >> amd_unsure_detected XML default") >> session:execute("voice_start") >> -- Giving some time to AMD to work on the call. >> -- If a voicemail is detected, it will auto-transfer from the >> variables that are set. >> session:sleep(max_wait) >> local amd_finished_called = session:getVariable("amd_ >> finished_called") >> if amd_finished_called == 'true' then >> freeswitch.consoleLog("INFO", string.format("AMD Finish >> script already called!\n")) >> else >> -- NOTE: Hopefully this never gets called as the module >> should auto transfer this >> freeswitch.consoleLog("INFO", string.format("AMD Timeout >> reached for vmd: %s\n", max_wait)) >> session:execute("voice_stop") >> local amd_detect = session:getVariable("amd_status") >> freeswitch.consoleLog("INFO", string.format("AMD amd_status: >> %s\n", amd_detect)) >> on_finished(amd_detect) >> end >> else >> freeswitch.consoleLog("WARNING", string.format("AMD not >> available for %s\n", dst_number)) >> on_finished("unavailable") >> end >> else >> freeswitch.consoleLog("WARNING", string.format("AMD Failed to run >> %s\n", dst_number)) >> on_finished("failure") >> return >> end > > > > the amd_execute_on_XXX does not seem to do anything... The sleep would > always max out and we would always hit the Timeout reached part of the > script instead of getting transferred. > > > After that didn't work because we would always wait the sleep time (3 > seconds) we scrapped transfers and tried handling the whole thing through a > single lua script using waitforresult: > > > DEFAULT_MAX_WAIT = 3000 -- in (ms) >> DEFAULT_INITIAL_WAIT = 100 -- in (ms) >> HUMAN_EXTENSION = "amd_queue" >> UNSURE_EXTENSION = "amd_queue" >> api = freeswitch.API() >> >> dst_number = session:getVariable("sip_to_user") >> initial_wait = session:getVariable("ppamd_initial_wait") or >> DEFAULT_INITIAL_WAIT >> max_wait = session:getVariable("ppamd_max_wait") or DEFAULT_MAX_WAIT >> >> while (session:ready() and not session:answered()) do >> session:sleep(500) >> end >> >> freeswitch.consoleLog("INFO", string.format("AMD Attempting on %s. \n", >> dst_number)) >> if initial_wait then >> session:sleep(initial_wait) >> end >> >> if session:ready() and session:answered() then >> local use_amd = api:executeString("amd_available") >> if use_amd == "true" then >> freeswitch.consoleLog("INFO", string.format("AMD Available Max >> Wait: %s, Initial Wait %s. \n", max_wait, initial_wait)) >> -- Allow AMD to run and analyze. >> freeswitch.consoleLog("INFO", "AMD Voice start. \n") >> session:execute("voice_start") >> freeswitch.consoleLog("INFO", "AMD Wait for result. \n") >> session:execute("waitforresult", string.format("%s", max_wait)) >> freeswitch.consoleLog("INFO", "AMD Voice stop. \n") >> session:execute("voice_stop") >> amd_detect = session:getVariable("amd_status") >> freeswitch.consoleLog("INFO", string.format("AMD Result: %s. \n", >> amd_detect)) >> if amd_detect == "machine" then >> freeswitch.consoleLog("INFO", "AMD Detected machine. \n") >> session:hangup() >> return >> elseif amd_detect == "person" then >> freeswitch.consoleLog("INFO", "AMD Detected human, >> transfering call. \n") >> session:execute("transfer", string.format("%s XML default", >> HUMAN_EXTENSION)) >> return >> else >> freeswitch.consoleLog("INFO", string.format("AMD Unknown: %s. >> \n", amd_detect)) >> session:execute("transfer", string.format("%s XML default", >> UNSURE_EXTENSION)) >> return >> end >> end >> else >> freeswitch.consoleLog("INFO", "AMD not enabled. \n") >> session:execute("transfer", string.format("%s XML default", >> UNSURE_EXTENSION)) >> return >> end > > > > I prefer not having to do all that transferring around but to just run > with waitforresult but it never comes back... > > The dialplan extensions just connect the call to an agent. Once we do the > transfer it all works. The only problem is we have to replace the: > > session:execute("waitforresult", string.format("%s", max_wait)) > > > with a sleep so no matter if the mod_com_amd detects its a person or not > we ALWAYS have to wait the full sleep time. This is not really legal as we > would be essentially abandoning every single call... > > Any thoughts on what we may be missing or how we could improve this? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > -- 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: color-facebook-96.png] [image: color-twitter-96.png] -------------- next part -------------- An HTML attachment was scrubbed... URL: From joel at gogii.net Mon Mar 26 19:11:08 2018 From: joel at gogii.net (Joel Serrano) Date: Mon, 26 Mar 2018 12:11:08 -0700 Subject: [Freeswitch-users] using ssh connection in module python in dialplan In-Reply-To: References: Message-ID: Seems to me you might be running freeswitch as a non-root user, and this user doesn't have access to root's private key referenced in your python script? Just a guess though. On Sun, Mar 25, 2018 at 4:39 AM, Morteza Bashsiz wrote: > Hi everybody > I have an asterisk(172.16.1.33) and freeswitch(172.16.1.30) > My scenario : > > 1. Calls come to freeswitch ivr menu > 2. In ivr calls transfer to asterisk after running an python script > freeswitch start recording voice > 3. In asterisk calls push to an queue > > my python script is : > > ############################################################ > import paramiko > > hostname = '172.16.1.33' > port = 3022 > username = 'root' > pkey_file = '/root/.ssh/id_rsa' > > def get_agent_id(num): > command="asterisk -rx \"pjsip show channels\" | grep -a1 \"CLCID: > \\\"%s\\\"\" | grep Channel | cut -d \/ -f2 | cut -d\- -f1 " % (num) > key = paramiko.RSAKey.from_private_key_file(pkey_file) > s = paramiko.SSHClient() > s.load_system_host_keys() > s.connect(hostname, port, pkey=key) > stdin, stdout, stderr = s.exec_command(command) > result=str(stdout.read()) > s.close() > return result > > #if __name__ == "__main__": > # agent_id=get_agent_id('1000') > # print (agent_id) > import freeswitch as freeswitch > def handler(session, args): > caller_num = session.getVariable("caller") > agent_num = get_agent_id(caller_num) > session.setVariable("AGENT_NUM", agent_num); > session.setVariable("CALLER_NUM", caller_num); > ############################################################ > > > my dial plan is : > > ############################################################ > > > > > > > > > > > > > > > ############################################################ > > > *MY PROBLEM IS:* > > when i want to use my variable AGENT_NUM it is null > BUT > when i use the python manually it returns true value > > please help me what is my mistake ?????? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From me at nevian.org Mon Mar 26 23:59:09 2018 From: me at nevian.org (Serge Yuriev) Date: Tue, 27 Mar 2018 02:59:09 +0300 Subject: [Freeswitch-users] sip_hangup_disposition In-Reply-To: References: Message-ID: Hi, You SEND cancel so it’s initiated on your side. %) btw leb B just can't send you CANCEL. > On 26 Mar 2018, at 16:11, Gregor Nanger wrote: > > Hi! > > I am bridging incoming call to provider. In CDR of b leg I see send_cancel. But how do I know which side initiated cancel (originator or provider)? > > Thank you, Gregor > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org -- Serge S. Yuriev -------------- next part -------------- An HTML attachment was scrubbed... URL: From gregor at infomedia.si Tue Mar 27 07:52:51 2018 From: gregor at infomedia.si (Gregor Nanger) Date: Tue, 27 Mar 2018 09:52:51 +0200 Subject: [Freeswitch-users] sip_hangup_disposition In-Reply-To: References: Message-ID: Thank u Serge for answer. What if callee cancel incoming call on phone, would I see in CDR as send_cancel? 2018-03-27 1:59 GMT+02:00 Serge Yuriev : > Hi, > > You SEND cancel so it’s initiated on your side. %) > btw leb B just can't send you CANCEL. > > On 26 Mar 2018, at 16:11, Gregor Nanger wrote: > > Hi! > > I am bridging incoming call to provider. In CDR of b leg I see > send_cancel. But how do I know which side initiated cancel (originator or > provider)? > > Thank you, Gregor > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > > > -- > Serge S. Yuriev > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > -- Gregor Nanger *CTO* t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia • www.infomedia.si -------------- next part -------------- An HTML attachment was scrubbed... URL: From paul.mateer at outlook.com Tue Mar 27 09:00:04 2018 From: paul.mateer at outlook.com (Paul Mateer) Date: Tue, 27 Mar 2018 09:00:04 +0000 Subject: [Freeswitch-users] Server driven calls In-Reply-To: References: , Message-ID: Thanks Stanislav. I'll take a look at when I have some free time. Sent from my Windows 10 phone ________________________________ From: FreeSWITCH-users on behalf of Stanislav Sinyagin Sent: Friday, March 23, 2018 1:17:04 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Server driven calls see two_step_dial here: https://github.com/voxserv/freeswitch-helper-scripts/tree/master/esl it's very likely what you want. On Tue, Mar 13, 2018 at 5:34 PM, Paul Mateer wrote: > I've been using FreeSWITCH for a while and I now have a query about whether > something is possible. > > > > I have clients that can place calls to one another, but it's a user driven > action – one user has to deliberately place a call to the other. > > > > I have a server component which monitors certain activities and when > specific events occur, I'd like the server to initiate a call between two > parties. Ideally, the server would initiate the call to each party and then > when the call is established (both parties have at least reached the > “ringing” stage) it would plug the two calls together and step back. > > > > I think it might be possible to do this by constructing an appropriate > conference call and then having the call initiator hang up, but I haven't > tried this to confirm if it's possible and I'm not sure if there's a more > appropriate way to achieve this. > > > > Does anyone have any thoughts on this? > > > > Sent from my Windows 10 phone > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.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 http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From vma at vallimamod.org Tue Mar 27 09:49:20 2018 From: vma at vallimamod.org (Vallimamod Abdullah) Date: Tue, 27 Mar 2018 11:49:20 +0200 Subject: [Freeswitch-users] sip_hangup_disposition In-Reply-To: References: Message-ID: Hi, You cannot send a CANCEL to an incoming call as it is a sip request. You reject it with a 4XX or 6XX sip response code like 480 Temporarily Unavailable or 486 Busy Here. In this case, you will see send_refuse in sip_hangup_disposition. You will see send_cancel if the caller ends an outgoing call before it is answered, so only on leg A. Best Regards, -- Vallimamod Abdullah SIP Solutions linkedin.com/in/vallimamod . > On 27 Mar 2018, at 09:52, Gregor Nanger wrote: > > Thank u Serge for answer. What if callee cancel incoming call on phone, would I see in CDR as send_cancel? > > > 2018-03-27 1:59 GMT+02:00 Serge Yuriev : > Hi, > > You SEND cancel so it’s initiated on your side. %) > btw leb B just can't send you CANCEL. > >> On 26 Mar 2018, at 16:11, Gregor Nanger wrote: >> >> Hi! >> >> I am bridging incoming call to provider. In CDR of b leg I see send_cancel. But how do I know which side initiated cancel (originator or provider)? >> >> Thank you, Gregor >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.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 >> http://www.freeswitch.org > > -- > Serge S. Yuriev > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > > > > -- > Gregor Nanger > > CTO > t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 > • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia > • www.infomedia.si > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org From vbvbrj at gmail.com Tue Mar 27 05:51:05 2018 From: vbvbrj at gmail.com (Mimiko) Date: Tue, 27 Mar 2018 08:51:05 +0300 Subject: [Freeswitch-users] Using lcr in lua session and fail_on_single_reject problem Message-ID: <3af44928-2411-d6ec-33cf-11e064b39b17@gmail.com> Hello. I want to originate a call using lcr from lua script: freeswitch.Session("{ignore_early_media=true,originate_timeout=15,fail_on_single_reject=^^:NO_ANSWER:CALL_REJECTED:NORMAL_CLEARING:USER_BUSY}lcr/default/"..number) Using this command, originate_timeout is not taken into account and all routes are tried regardless of response reason, for example NO_ANSWER. If I remove fail_on_single_reject, then originate_timeout works but same problem: all routes are tried, except when the call was answered. In diaplan this works fine. Maybe its due to using bridge, not lcr directly. But in lua ${lcr_auto_route} variable is not available. Any suggestion? FreeSWITCH Version 1.5.6b+git~20130928T022323Z~6b9382290d (git 6b93822 2013-09-28 02:23:23Z) Yes, it is old, but can't update at this time. From steveayre at gmail.com Tue Mar 27 22:29:46 2018 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 27 Mar 2018 23:29:46 +0100 Subject: [Freeswitch-users] using ssh connection in module python in dialplan In-Reply-To: References: Message-ID: Careful running system commands from dialplan (on a remote system in this case), or you can create a fork bomb if you allow enough calls into that rule quickly enough. On 25 March 2018 at 12:39, Morteza Bashsiz wrote: > Hi everybody > I have an asterisk(172.16.1.33) and freeswitch(172.16.1.30) > My scenario : > > 1. Calls come to freeswitch ivr menu > 2. In ivr calls transfer to asterisk after running an python script > freeswitch start recording voice > 3. In asterisk calls push to an queue > > my python script is : > > ############################################################ > import paramiko > > hostname = '172.16.1.33' > port = 3022 > username = 'root' > pkey_file = '/root/.ssh/id_rsa' > > def get_agent_id(num): > command="asterisk -rx \"pjsip show channels\" | grep -a1 \"CLCID: > \\\"%s\\\"\" | grep Channel | cut -d \/ -f2 | cut -d\- -f1 " % (num) > key = paramiko.RSAKey.from_private_key_file(pkey_file) > s = paramiko.SSHClient() > s.load_system_host_keys() > s.connect(hostname, port, pkey=key) > stdin, stdout, stderr = s.exec_command(command) > result=str(stdout.read()) > s.close() > return result > > #if __name__ == "__main__": > # agent_id=get_agent_id('1000') > # print (agent_id) > import freeswitch as freeswitch > def handler(session, args): > caller_num = session.getVariable("caller") > agent_num = get_agent_id(caller_num) > session.setVariable("AGENT_NUM", agent_num); > session.setVariable("CALLER_NUM", caller_num); > ############################################################ > > > my dial plan is : > > ############################################################ > > > > > > > > > > > > > > > ############################################################ > > > *MY PROBLEM IS:* > > when i want to use my variable AGENT_NUM it is null > BUT > when i use the python manually it returns true value > > please help me what is my mistake ?????? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bob.mccarthy at westtel.com Wed Mar 28 00:34:23 2018 From: bob.mccarthy at westtel.com (Bob McCarthy) Date: Wed, 28 Mar 2018 00:34:23 +0000 Subject: [Freeswitch-users] Adding a Parked Call into a conference Message-ID: Anyone have a best practice for taking a channel out of a parking lot and adding it to a conference? I have successfully transferred into the parking lot but the results are less than desired. The BLF remains active until the conference ends and pressing it give you hold music. Looking for an API / lua solution. Thanks in advance, Bob McCarthy -------------- next part -------------- An HTML attachment was scrubbed... URL: From michael at mailworks.org Thu Mar 29 07:13:53 2018 From: michael at mailworks.org (Michael Avers) Date: Thu, 29 Mar 2018 00:13:53 -0700 Subject: [Freeswitch-users] Multiple concurrent Verto calls in browser In-Reply-To: References: <1521446608.822385.1307854600.6BAD2C6E@webmail.messagingengine.com> <7B8D444E-7087-450C-96B8-AC7755BAB74D@gmail.com> Message-ID: <1522307633.2221439.1319939472.62B616C1@webmail.messagingengine.com> How do you handle the state of the multiple calls? Say user is on a call, and a new incoming call comes in and they want to answer it - do you place the currently active call on hold before you answer the new one? Thanks Mike On Wed, Mar 21, 2018, at 3:36 AM, Gregor Nanger wrote: > @Tihomir > Sorry, to advanced question for me :-) We just insert new video canvas > for each call and hide, since we only use audio and keep own array > list with call guid and states as we want.> > Gregor > > > 2018-03-21 7:19 GMT+01:00 Tihomir Culjaga : >> @Michael, >> We did the same thing as Gregor, implemented verto api in our own web >> interface... a single wss is enough :)>> >> @Gregor, sorry to hijack the mail ;) >> >> But did you happen to be able to set googDscp in RTCPeerConnection >> and chrome actually to mark rtp packets?>> My chrome (latest) on win10 keeps ignoring it :(. >> >> Tihomir. >> >> >> >> Sent from my iPhone >> >> On 19 Mar 2018, at 09:56, Gregor Nanger wrote:>>> Hi, >>> >>> we are using such scenario, but it is not based on Verto >>> Communicator. We took core libraries (javascript) and implement GUI >>> on our own. Everything what happens on verto triggers event and you >>> can implement own logic and callstate for each call in array. For >>> example, when user picks up a call, put active on hold via verto >>> command.>>> >>> And for info, you can issue as many calls you want through one verto >>> connection.>>> >>> >>> >>> >>> 2018-03-19 9:03 GMT+01:00 Michael Avers : >>>> Hello, >>>> >>>> I'm using Verto Communicator as a base for building a simple app >>>> to make/receive calls - and it works great.>>>> >>>> I'm trying to add the ability to manage multiple concurrent calls.>>>> >>>> My first thought is to create a new Verto service (in Angular) >>>> that maintains call state and reference to currently active call >>>> and otherwise keeps all calls in an array. Then to switch calls >>>> it'd place the currently active one on hold and then unhold any of >>>> the other calls in the array. Does this make sense?>>>> >>>> Any tips on handling multiple calls with Verto? Do I need a >>>> separate websocket connection for each? (they're all using the >>>> same proxy and credentials).>>>> >>>> Thanks! >>>> Mike >>>> >>>> _________________________________________________________________- >>>> ________>>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.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>>>> http://www.freeswitch.org >>> >>> >>> >>> -- >>> Gregor Nanger >>> >>> *CTO* >>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 • Infomedia >>> d.o.o. • Jerebova 3, Novo mesto, Slovenia • www.infomedia.si>>> ___________________________________________________________________- >>> ______>>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.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>>> http://www.freeswitch.org >> >> __________________________________________________________________- >> _______>> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.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>> http://www.freeswitch.org > > > > -- > Gregor Nanger > > *CTO* > t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 • Infomedia > d.o.o. • Jerebova 3, Novo mesto, Slovenia • www.infomedia.si> ___________________________________________________________________- > ________> Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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> http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From gregor at infomedia.si Thu Mar 29 07:32:08 2018 From: gregor at infomedia.si (Gregor Nanger) Date: Thu, 29 Mar 2018 09:32:08 +0200 Subject: [Freeswitch-users] Multiple concurrent Verto calls in browser In-Reply-To: <1522307633.2221439.1319939472.62B616C1@webmail.messagingengine.com> References: <1521446608.822385.1307854600.6BAD2C6E@webmail.messagingengine.com> <7B8D444E-7087-450C-96B8-AC7755BAB74D@gmail.com> <1522307633.2221439.1319939472.62B616C1@webmail.messagingengine.com> Message-ID: We are dealing same way as multiline phone. We are showing to user in GUI waiting call (can be many of them) and it is up to user what he wants to do. If he picks up call and one is active, we put active on hold. There can be only one active at a time and others are on hold (if they were pickedup) or on queuee. 2018-03-29 9:13 GMT+02:00 Michael Avers : > How do you handle the state of the multiple calls? Say user is on a call, > and a new incoming call comes in and they want to answer it - do you place > the currently active call on hold before you answer the new one? > > Thanks > Mike > > > On Wed, Mar 21, 2018, at 3:36 AM, Gregor Nanger wrote: > > @Tihomir > Sorry, to advanced question for me :-) We just insert new video canvas for > each call and hide, since we only use audio and keep own array list with > call guid and states as we want. > > Gregor > > > 2018-03-21 7:19 GMT+01:00 Tihomir Culjaga : > > @Michael, > We did the same thing as Gregor, implemented verto api in our own web > interface... a single wss is enough :) > > @Gregor, sorry to hijack the mail ;) > > But did you happen to be able to set googDscp in RTCPeerConnection and > chrome actually to mark rtp packets? > My chrome (latest) on win10 keeps ignoring it :(. > > Tihomir. > > > > Sent from my iPhone > > On 19 Mar 2018, at 09:56, Gregor Nanger wrote: > > Hi, > > we are using such scenario, but it is not based on Verto Communicator. We > took core libraries (javascript) and implement GUI on our own. Everything > what happens on verto triggers event and you can implement own logic and > callstate for each call in array. For example, when user picks up a call, > put active on hold via verto command. > > And for info, you can issue as many calls you want through one verto > connection. > > > > > 2018-03-19 9:03 GMT+01:00 Michael Avers : > > Hello, > > I'm using Verto Communicator as a base for building a simple app to > make/receive calls - and it works great. > > I'm trying to add the ability to manage multiple concurrent calls. > > My first thought is to create a new Verto service (in Angular) that > maintains call state and reference to currently active call and otherwise > keeps all calls in an array. Then to switch calls it'd place the currently > active one on hold and then unhold any of the other calls in the array. > Does this make sense? > > Any tips on handling multiple calls with Verto? Do I need a separate > websocket connection for each? (they're all using the same proxy and > credentials). > > Thanks! > Mike > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > > > > > -- > Gregor Nanger > > *CTO* > t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 > • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia > • www.infomedia.si > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > > > > > -- > Gregor Nanger > > *CTO* > t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 > • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia > • www.infomedia.si > *_________________________________________________________________________* > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > -- Gregor Nanger *CTO* t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia • www.infomedia.si -------------- next part -------------- An HTML attachment was scrubbed... URL: From achinthau at gmail.com Thu Mar 29 07:16:13 2018 From: achinthau at gmail.com (Achintha) Date: Thu, 29 Mar 2018 12:46:13 +0530 Subject: [Freeswitch-users] Freeswitch Crashed- Segmentation Fault Message-ID: Hi all, Our freeswitch server handles 3000 – 4000 calls per day. We have custom module to rout calls. Anyway freeswitch was crashed two times I have attached core dump on “ https://pastebin.freeswitch.org/view/4463025f ” Our freeswitch version is “1.6.20-37-987c9b9~64bit (-37-987c9b9 64bit)” Please advise to solve this matter asap. -- Best Regards.. Achintha Udukumbura -------------- next part -------------- An HTML attachment was scrubbed... URL: From matthew at brightfire.net Wed Mar 28 19:53:02 2018 From: matthew at brightfire.net (Matthew Grooms) Date: Wed, 28 Mar 2018 14:53:02 -0500 Subject: [Freeswitch-users] Chat between SIP and Verto In-Reply-To: <21265e38-649f-8655-8369-6a167e07742f@brightfire.net> References: <21265e38-649f-8655-8369-6a167e07742f@brightfire.net> Message-ID: <2d516247-372d-55cc-2f97-ab271fa1ab82@brightfire.net> Hello everyone, Does anyone have any insight into how 'SIMPLE MESSAGE' delivery can be achieved between sip and verto clients? The verto documentation is very light on detail especially when it comes to messaging. There are references to using chat in conferences but that's about it. I've tried appending the protocol in the "to" value ( ie. "verto+9905 at voip.gh.bf" ) but that doesn't appear to help much. It does change the to_proto to verto, but the message doesn't end up being delivered to the verto client. Any help would be greatly appreciated. -Matthew On 3/21/2018 3:03 PM, Matthew Grooms wrote: > > Hello All, > > We have a few hundred agents on a custom built call center platform > that leverages freeswitch and mod_callcenter. Up until now, agents > have used a custom built Windows Client based on pjsip. I'm trying to > replace that with a new client built with WebRTC and mod_verto. The > last piece of the puzzle I'm trying to solve is sending chat messages > between Verto and SIP clients. I can use the chat command from fs_cli > to send messages to freeswitch to a Verto client, but I can't seem to > send messages from a SIP client to a Verto client or a Verto client to > a SIP client. After spending a few days on this, I thought I'd appeal > to the list to see if someone can help provide some guidance. > > # show registrations > reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname,metadata > > 1000,voip.gh.bf,0d5831a1b9274063a1e06237f47a68af,sofia/external/sip:1000 at X.X.X.X:49524;transport=TCP;ob,1521650573,X.X.X.X,49524,tcp,voip2.gh.bf, > > > # verto status >                      Name Type                                       > Data      State > ================================================================================================= > >                      mine       profile ws:0.0.0.0:8081 RUNNING >                      mine       profile wss:0.0.0.0:8082 RUNNING > mine::9905 at voip.gh.bf client X.X.X.X:60992      CONN_REG (WSS) > ================================================================================================= > > 1 profile , 1 client > > > Messages are delivered when I try the following ... > > CHAT TO VERTO: > -------------------- CONSOLE -------------------- > # chat verto|1000 at voip.gh.bf|9905 at voip.gh.bf|test > 2018-03-21 11:20:48.170147 [ALERT] mod_verto.c:604 WRITE X.X.X.X:49548 [{ >         "jsonrpc":      "2.0", >         "id":   6, >         "method":       "verto.info", >         "params":       { >                 "msg":  { >                         "from": "1000 at voip.gh.bf", >                         "to": "9905 at voip.gh.bf", >                         "body": "test" >                 } >         } > }] > -------------------- CHROME -------------------- > onVertoMessage from:1000 at voip.gh.bf to:9905 at voip.gh.bf body:test > > > Messages are not delivered when trying to send from client to client ... > > SIP TO VERTO: > -------------------- EVENT -------------------- > KV -> Chat-Send-From: "A" > ;tag=759225c356d545d2b8c83b284063cac2 > KV -> Chat-Send-Profile: external > KV -> Chat-Send-To: 9905 at voip.gh.bf > KV -> Content-Length: 4 > KV -> Core-UUID: 74a1329b-6a85-4078-9c0d-f9af5972bf0a > KV -> Error-Reason: no recipient > KV -> Error-Type: chat > KV -> Event-Calling-File: sofia_presence.c > KV -> Event-Calling-Function: sofia_presence_chat_send > KV -> Event-Calling-Line-Number: 276 > KV -> Event-Date-GMT: Wed, 21 Mar 2018 15:44:13 GMT > KV -> Event-Date-Local: 2018-03-21 10:44:13 > KV -> Event-Date-Timestamp: 1521647053570639 > KV -> Event-Name: CUSTOM > KV -> Event-Sequence: 299577 > KV -> Event-Subclass: sofia::error > KV -> FreeSWITCH-Hostname: voip2.gh.bf > KV -> FreeSWITCH-IPv4: X.X.X.X > KV -> FreeSWITCH-IPv6: ::1 > KV -> FreeSWITCH-Switchname: voip2.gh.bf > -------------------- CONSOLE -------------------- > 2018-03-21 10:44:13.570639 [WARNING] sofia_presence.c:221 Not sending > to local box for 9905 at voip.gh.bf > 2018-03-21 10:44:13.570639 [ERR] sofia_presence.c:272 Chat proto [sip] > from ["A" ;tag=759225c356d545d2b8c83b284063cac2] > to [9905 at voip.gh.bf] > test > Nobody to send to: Profile external > > VERTO TO SIP: > -------------------- EVENT -------------------- > KV -> Content-Length: 22 > KV -> Core-UUID: 74a1329b-6a85-4078-9c0d-f9af5972bf0a > KV -> Event-Calling-File: mod_verto.c > KV -> Event-Calling-Function: verto__info_func > KV -> Event-Calling-Line-Number: 3374 > KV -> Event-Date-GMT: Wed, 21 Mar 2018 15:50:56 GMT > KV -> Event-Date-Local: 2018-03-21 10:50:56 > KV -> Event-Date-Timestamp: 1521647456550625 > KV -> Event-Name: MESSAGE > KV -> Event-Sequence: 299780 > KV -> FreeSWITCH-Hostname: voip2.gh.bf > KV -> FreeSWITCH-IPv4: X.X.X.X > KV -> FreeSWITCH-IPv6: ::1 > KV -> FreeSWITCH-Switchname: voip2.gh.bf > KV -> Nonblocking-Delivery: true > KV -> dest_proto: GLOBAL > KV -> from: 9905 at voip.gh.bf > KV -> from_full: 9905 > KV -> from_host: voip.gh.bf > KV -> from_user: 9905 > KV -> proto: verto > KV -> to: 1000 at voip.gh.bf > KV -> type: text/plain > KV -> verto_jsock_uuid: 85eef9c8-54c2-f4a7-eb95-ef9f5f1e6ff8 > KV -> verto_profile: mine > -------------------- CONSOLE -------------------- > freeswitch at voip2.gh.bf> 2018-03-21 10:50:56.550625 [ALERT] > mod_verto.c:1384 READ X.X.X.X:64275 [{ >         "jsonrpc":      "2.0", >         "method":       "verto.info", >         "params":       { >                 "msg":  { >                         "to": "1000 at voip.gh.bf", >                         "from": "9905 at voip.gh.bf", >                         "body": "This is a test message" >                 }, >                 "sessid": "85eef9c8-54c2-f4a7-eb95-ef9f5f1e6ff8" >         }, >         "id":   5 > }] > 2018-03-21 10:50:56.550625 [ALERT] mod_verto.c:604 WRITE X.X.X.X:64275 [{ >         "jsonrpc":      "2.0", >         "id":   5, >         "result":       { >                 "message":      "SENT", >                 "sessid": "85eef9c8-54c2-f4a7-eb95-ef9f5f1e6ff8" >         } > }] > > Thanks in advance, > > -Matthew > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Thu Mar 29 15:24:27 2018 From: mike at jerris.com (Michael Jerris) Date: Thu, 29 Mar 2018 11:24:27 -0400 Subject: [Freeswitch-users] Adding a Parked Call into a conference In-Reply-To: References: Message-ID: are you transferring out of the parking lot? sounds like maybe you are executing extension while its parked. > On Mar 27, 2018, at 8:34 PM, Bob McCarthy wrote: > > Anyone have a best practice for taking a channel out of a parking lot and adding it to a conference? > I have successfully transferred into the parking lot but the results are less than desired. The BLF remains active until the conference ends > and pressing it give you hold music. Looking for an API / lua solution. > > Thanks in advance, > Bob McCarthy -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Thu Mar 29 15:26:10 2018 From: mike at jerris.com (Michael Jerris) Date: Thu, 29 Mar 2018 11:26:10 -0400 Subject: [Freeswitch-users] Freeswitch Crashed- Segmentation Fault In-Reply-To: References: Message-ID: <29BC9A0C-00CE-43D7-9D0D-07610BD347E6@jerris.com> You have no debug symbols installed so it is impossible to tell what is going on. Please install those and reproduce the bug report with debug symbols on lira. > On Mar 29, 2018, at 3:16 AM, Achintha wrote: > > > Hi all, > > Our freeswitch server handles 3000 – 4000 calls per day. We have custom module to rout calls. > > Anyway freeswitch was crashed two times I have attached core dump on “ https://pastebin.freeswitch.org/view/4463025f ” > > Our freeswitch version is “1.6.20-37-987c9b9~64bit (-37-987c9b9 64bit)” > > Please advise to solve this matter asap. > > > > -- > Best Regards.. > Achintha Udukumbura > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From donguyenha at gmail.com Thu Mar 29 15:40:32 2018 From: donguyenha at gmail.com (Do Nguyen Ha) Date: Thu, 29 Mar 2018 22:40:32 +0700 Subject: [Freeswitch-users] need help to run python script with mod_python Message-ID: hi i user python_example.py script to test with freeswitch https://gitlab.k-net.fr/shimaore/freeswitch/blob/c6be531b4b6c38b69274057b19d73c7b15a07b14/src/mod/languages/mod_python/python_example.py i installed freeswitch-python and check the scripts folder is in : [root at localhost site-packages]# fs_cli -x 'global_getvar'|grep script script_dir=/usr/share/freeswitch/scripts and the freeswitch-python package: [root at localhost site-packages]# rpm -ql freeswitch-python /etc/freeswitch/autoload_configs /etc/freeswitch/autoload_configs/python.conf.xml /usr/lib/python2.7/site-packages/freeswitch.py /usr/lib/python2.7/site-packages/freeswitch.pyc /usr/lib/python2.7/site-packages/freeswitch.pyo /usr/lib64/freeswitch/mod/mod_python.so /usr/lib64/python2.7/site-packages/ESL.py /usr/lib64/python2.7/site-packages/ESL.pyc /usr/lib64/python2.7/site-packages/ESL.pyo /usr/lib64/python2.7/site-packages/_ESL.so where do i place the python-example.py script for freeswitch working properly Thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: From joel at gogii.net Thu Mar 29 16:54:12 2018 From: joel at gogii.net (Joel Serrano) Date: Thu, 29 Mar 2018 09:54:12 -0700 Subject: [Freeswitch-users] need help to run python script with mod_python In-Reply-To: References: Message-ID: Put the script in: /usr/share/freeswitch/scripts/python_example.py Then call it from the dialplan without the extension (.py) For example: You can also place it wherever you want and use the full path as long as the user you are running freeswitch with has the correct permissions to access it. On Thu, Mar 29, 2018 at 8:40 AM, Do Nguyen Ha wrote: > hi > > i user python_example.py script to test with freeswitch > https://gitlab.k-net.fr/shimaore/freeswitch/blob/c6be531b4b6c38b69274057b19d73c7b15a07b14/src/mod/languages/mod_python/python_example.py > > i installed freeswitch-python and check the scripts folder is in : > [root at localhost site-packages]# fs_cli -x 'global_getvar'|grep script > script_dir=/usr/share/freeswitch/scripts > > and the freeswitch-python package: > [root at localhost site-packages]# rpm -ql freeswitch-python > /etc/freeswitch/autoload_configs > /etc/freeswitch/autoload_configs/python.conf.xml > /usr/lib/python2.7/site-packages/freeswitch.py > /usr/lib/python2.7/site-packages/freeswitch.pyc > /usr/lib/python2.7/site-packages/freeswitch.pyo > /usr/lib64/freeswitch/mod/mod_python.so > /usr/lib64/python2.7/site-packages/ESL.py > /usr/lib64/python2.7/site-packages/ESL.pyc > /usr/lib64/python2.7/site-packages/ESL.pyo > /usr/lib64/python2.7/site-packages/_ESL.so > > > where do i place the python-example.py script for freeswitch working > properly > > Thank you > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org From brian at freeswitch.com Thu Mar 29 17:48:10 2018 From: brian at freeswitch.com (Brian West) Date: Thu, 29 Mar 2018 12:48:10 -0500 Subject: [Freeswitch-users] List update... gmail addresses with plus signs are banned. Message-ID: If you have subscribed via a gmail.com address to the mailing list in the format of user+tag at gmail.com, You're ok, but I had to block all gmail addresses that contain a plus sign, it was being abused, 1000's of emails were being added and used to target specific gmail users so their email would stop working because we were flooding them with subscribe confirmation emails. It was a total of three target addresses coming from 1000's of different IP addresses. This was also adding to our bounce issue with google. Thanks, -- 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: color-facebook-96.png] [image: color-twitter-96.png] -------------- next part -------------- An HTML attachment was scrubbed... URL: From enp at itx.ru Thu Mar 29 17:56:47 2018 From: enp at itx.ru (Eugene Prokopiev) Date: Thu, 29 Mar 2018 20:56:47 +0300 Subject: [Freeswitch-users] Raw SIP request text in Lua (or any other) script Message-ID: Hi, Is it possible to get raw SIP request (INVITE or so) text with headers and body as is in Lua (or any other) script? -- WBR, Eugene Prokopiev From roman at dissauer.net Thu Mar 29 18:14:18 2018 From: roman at dissauer.net (Roman Dissauer) Date: Thu, 29 Mar 2018 20:14:18 +0200 Subject: [Freeswitch-users] postgres db contains non existent channels Message-ID: <6C59B394-8BCE-4316-AE52-D2883ECB23D0@dissauer.net> Hi, we have about 800 connected users, 150 simultaneous calls and some user agents send sip messages with non UTF-8 characters. When FreeSWITCH tries to alter the channel information in the postgres db it throughs an error: 2018-03-29 15:47:46.794346 [ERR] switch_pgsql.c:656 Error executing query: ERROR: invalid byte sequence for encoding "UTF8": 0xf6 0x63 0x68 0x6c The Problem now is that we see some calls in the database which do not exist anymore (uuid_exists returns false). This are other calls, not the one which cause the error above! It seems that everytime the error happens, Freeswitch cannot delete other channels from db when the bye is arriving +/- 100ms near the error. We tried to set client_encoding in the postgres connection string to some options e.g. LATIN1 but that didn’t solve our problem. Whichever client_encoding we used, we always had some user agents with some unknown characters. How should we deal with that? Thanks, Roman RDI SOLUTIONS e.U. Hollenthon 105 2812 Hollenthon, Österreich T: +43 1 3530349 - 10 F: +43 1 3530349 - 99 roman.dissauer at rdi.at www.rdi.at -------------- next part -------------- An HTML attachment was scrubbed... URL: From francesco at delagarda.com Fri Mar 30 08:20:16 2018 From: francesco at delagarda.com (Francesco Facco de Lagarda) Date: Fri, 30 Mar 2018 10:20:16 +0200 Subject: [Freeswitch-users] javascript calculate duration of call In-Reply-To: <05ab01d3c7ff$5dbb4be0$1931e3a0$@shenker.it> References: <05ab01d3c7ff$5dbb4be0$1931e3a0$@shenker.it> Message-ID: <05ba01d3c7ff$f05f4640$d11dd2c0$@delagarda.com> Good morning all, I will be using both digital (which knows when the other part answers) and analogue (when, at least in Italy, doesn't!) I need to calculate the total duration of an outgoing call. Calls also must have a preset max duration. I need to schedule a warning message (1 min before timeout) and a hangup (on timeout) for BOTH digital and analogue lines. This is the core of the code, reduced to the bone: var allowedSecs = 300; // max length 5 min . session.answer(); var sessOut = new Session("sofia/gateway/realtoneFXO/" + dialedNum + "@192.168.0.216:5060"); if (trunkIsDigital) { sessOut.execute("set", "execute_on_answer_1=sched_hangup " + allowedSects + " alloted_timeout") sessOut.execute("set", "execute_on_answer_2=sched_broadcast + " + (allowedSects-10) + " playback::" + soundDir + "one_min_left.wav both"); } else { // will schedule message and hangup on allowedSecs + graceTime to be calculated according to destintation (time to ring/answer) } if (sessOut.ready()) { bridge(session, sessOut); } sessOut.hangup(); session.hangup(); /*** HOW LONG DID THE CALL (bridge) ACTUALLY LAST?? .. for billing purposes **/ Thanks and Happy Easter! -------------- next part -------------- An HTML attachment was scrubbed... URL: From lists at pbaines.com Fri Mar 30 12:18:48 2018 From: lists at pbaines.com (Peter Baines (lists)) Date: Fri, 30 Mar 2018 13:18:48 +0100 Subject: [Freeswitch-users] Setting hold_music variable Message-ID: Hello, I am trying to set hold music per channel but cannot override the default that is set in vars.xml. I am setting the default hold_music variable in vars.xml, then in a dialplan I am overriding like so: I can see it hitting the logline in fs_cli, I am them putting it on hold like so: The music from vars.xml is played, even though the log line above tells me that ${hold_music} is set to my custom wav. I have also tried the following which also plays the default music on hold: Should I be able to overwrite the hold_music variable ? Regards, Peter -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Fri Mar 30 15:57:20 2018 From: mike at jerris.com (Michael Jerris) Date: Fri, 30 Mar 2018 11:57:20 -0400 Subject: [Freeswitch-users] Raw SIP request text in Lua (or any other) script In-Reply-To: References: Message-ID: <3507398E-073F-46B8-B1B8-868D9EC85EF5@jerris.com> no, just the specific headers that we extract into variables. > On Mar 29, 2018, at 1:56 PM, Eugene Prokopiev wrote: > > Hi, > > Is it possible to get raw SIP request (INVITE or so) text with headers > and body as is in Lua (or any other) script? > > -- > WBR, > Eugene Prokopiev From brian at freeswitch.com Fri Mar 30 17:52:49 2018 From: brian at freeswitch.com (Brian West) Date: Fri, 30 Mar 2018 12:52:49 -0500 Subject: [Freeswitch-users] javascript calculate duration of call In-Reply-To: <05ba01d3c7ff$f05f4640$d11dd2c0$@delagarda.com> References: <05ab01d3c7ff$5dbb4be0$1931e3a0$@shenker.it> <05ba01d3c7ff$f05f4640$d11dd2c0$@delagarda.com> Message-ID: You're already doing it wrong if you're taking this approach. Review our existing CDR modules that would accomplish this for you. /b On Fri, Mar 30, 2018 at 3:20 AM, Francesco Facco de Lagarda < francesco at delagarda.com> wrote: > Good morning all, > > I will be using both digital (which knows when the other part answers) and > analogue (when, at least in Italy, doesn’t!) > > > > I need to calculate the total duration of an outgoing call. > > > > Calls also must have a preset max duration. I need to schedule a warning > message (1 min before timeout) and a hangup (on timeout) for BOTH digital > and analogue lines. > > > > This is the core of the code, reduced to the bone: > > > > var allowedSecs = 300; // max length 5 min … > > > > session.answer(); > > var sessOut = new Session("sofia/gateway/realtoneFXO/" + dialedNum + "@ > 192.168.0.216:5060"); > > > > if (trunkIsDigital) { > > sessOut.execute("set", "execute_on_answer_1=sched_hangup " + > allowedSects + " alloted_timeout") > > sessOut.execute("set", "execute_on_answer_2=sched_broadcast + " + > (allowedSects-10) + " playback::" + soundDir + "one_min_left.wav both"); > > } else { > > // will schedule message and hangup on allowedSecs + graceTime to be > calculated according to destintation (time to ring/answer) > > } > > > > if (sessOut.ready()) { > > bridge(session, sessOut); > > } > > sessOut.hangup(); > > session.hangup(); > > > > */**** > > *HOW LONG DID THE CALL (bridge) ACTUALLY LAST?? .. for billing purposes* > > ***/* > > > > > > > > Thanks and Happy Easter! > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > -- 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: color-facebook-96.png] [image: color-twitter-96.png] -------------- next part -------------- An HTML attachment was scrubbed... URL: From donguyenha at gmail.com Sat Mar 31 02:28:35 2018 From: donguyenha at gmail.com (Do Nguyen Ha) Date: Sat, 31 Mar 2018 09:28:35 +0700 Subject: [Freeswitch-users] need help to run python script with mod_python In-Reply-To: References: Message-ID: yeah it is working :) thank you On Thu, Mar 29, 2018 at 11:54 PM, Joel Serrano wrote: > Put the script in: > > /usr/share/freeswitch/scripts/python_example.py > > Then call it from the dialplan without the extension (.py) > > For example: > > > > You can also place it wherever you want and use the full path as long > as the user you are running freeswitch with has the correct > permissions to access it. > > > > > On Thu, Mar 29, 2018 at 8:40 AM, Do Nguyen Ha > wrote: > > hi > > > > i user python_example.py script to test with freeswitch > > https://gitlab.k-net.fr/shimaore/freeswitch/blob/ > c6be531b4b6c38b69274057b19d73c7b15a07b14/src/mod/languages/ > mod_python/python_example.py > > > > i installed freeswitch-python and check the scripts folder is in : > > [root at localhost site-packages]# fs_cli -x 'global_getvar'|grep script > > script_dir=/usr/share/freeswitch/scripts > > > > and the freeswitch-python package: > > [root at localhost site-packages]# rpm -ql freeswitch-python > > /etc/freeswitch/autoload_configs > > /etc/freeswitch/autoload_configs/python.conf.xml > > /usr/lib/python2.7/site-packages/freeswitch.py > > /usr/lib/python2.7/site-packages/freeswitch.pyc > > /usr/lib/python2.7/site-packages/freeswitch.pyo > > /usr/lib64/freeswitch/mod/mod_python.so > > /usr/lib64/python2.7/site-packages/ESL.py > > /usr/lib64/python2.7/site-packages/ESL.pyc > > /usr/lib64/python2.7/site-packages/ESL.pyo > > /usr/lib64/python2.7/site-packages/_ESL.so > > > > > > where do i place the python-example.py script for freeswitch working > > properly > > > > Thank you > > > > ____________________________________________________________ > _____________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.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 > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From donguyenha at gmail.com Sat Mar 31 03:16:09 2018 From: donguyenha at gmail.com (Do Nguyen Ha) Date: Sat, 31 Mar 2018 10:16:09 +0700 Subject: [Freeswitch-users] freeswitch segfault with mod_python Message-ID: hi list i am testing freeswitch with mod_python and freeswitch sometime dead :( Mar 30 23:05:08 localhost kernel: freeswitch[1961]: segfault at 0 ip 00007f97a936ec31 sp 00007f97a4b24ba8 error 4 in libc-2.17.so [7f97a92e8000+1b8000] [root at localhost ~]# rpm -qi glibc.x86_64 Name : glibc Version : 2.17 Release : 196.el7_4.2 Architecture: x86_64 Install Date: Sat 03 Feb 2018 09:48:58 PM EST Group : System Environment/Libraries Size : 14107624 License : LGPLv2+ and LGPLv2+ with exceptions and GPLv2+ Signature : RSA/SHA256, Sat 02 Dec 2017 09:35:03 AM EST, Key ID 24c6a8a7f4a80eb5 Source RPM : glibc-2.17-196.el7_4.2.src.rpm Build Date : Thu 30 Nov 2017 01:53:33 PM EST Build Host : c1bm.rdu2.centos.org Relocations : (not relocatable) Packager : CentOS BuildSystem Vendor : CentOS URL : http://www.gnu.org/software/glibc/ Summary : The GNU libc libraries Name : freeswitch Version : 1.6.20 Release : 1.el7.centos Architecture: x86_64 Install Date: Sun 11 Mar 2018 04:34:58 AM EDT Group : Productivity/Telephony/Servers Size : 12776118 License : MPL1.1 Signature : DSA/SHA1, Wed 24 Jan 2018 02:16:20 PM EST, Key ID d76edc7725e010cf Source RPM : freeswitch-1.6.20-1.el7.centos.src.rpm Build Date : Wed 24 Jan 2018 02:14:46 PM EST Build Host : b-centos7-2.freeswitch.org Relocations : /usr Packager : Ken Rice Vendor : http://www.freeswitch.org/ URL : http://www.freeswitch.org/ Summary : FreeSWITCH open source telephony platform -------------- next part -------------- An HTML attachment was scrubbed... URL: From krice at freeswitch.org Sat Mar 31 03:23:45 2018 From: krice at freeswitch.org (Ken Rice) Date: Fri, 30 Mar 2018 22:23:45 -0500 Subject: [Freeswitch-users] freeswitch segfault with mod_python In-Reply-To: References: Message-ID: <116801d3c89f$adc96a90$095c3fb0$@freeswitch.org> See the troubleshooting documentation at https://freeswitch.org/confluence/ Also bugs get reported to jira From: FreeSWITCH-users On Behalf Of Do Nguyen Ha Sent: Friday, March 30, 2018 10:16 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] freeswitch segfault with mod_python hi list i am testing freeswitch with mod_python and freeswitch sometime dead :( Mar 30 23:05:08 localhost kernel: freeswitch[1961]: segfault at 0 ip 00007f97a936ec31 sp 00007f97a4b24ba8 error 4 in libc-2.17.so [7f97a92e8000+1b8000] [root at localhost ~]# rpm -qi glibc.x86_64 Name : glibc Version : 2.17 Release : 196.el7_4.2 Architecture: x86_64 Install Date: Sat 03 Feb 2018 09:48:58 PM EST Group : System Environment/Libraries Size : 14107624 License : LGPLv2+ and LGPLv2+ with exceptions and GPLv2+ Signature : RSA/SHA256, Sat 02 Dec 2017 09:35:03 AM EST, Key ID 24c6a8a7f4a80eb5 Source RPM : glibc-2.17-196.el7_4.2.src.rpm Build Date : Thu 30 Nov 2017 01:53:33 PM EST Build Host : c1bm.rdu2.centos.org Relocations : (not relocatable) Packager : CentOS BuildSystem Vendor : CentOS URL : http://www.gnu.org/software/glibc/ Summary : The GNU libc libraries Name : freeswitch Version : 1.6.20 Release : 1.el7.centos Architecture: x86_64 Install Date: Sun 11 Mar 2018 04:34:58 AM EDT Group : Productivity/Telephony/Servers Size : 12776118 License : MPL1.1 Signature : DSA/SHA1, Wed 24 Jan 2018 02:16:20 PM EST, Key ID d76edc7725e010cf Source RPM : freeswitch-1.6.20-1.el7.centos.src.rpm Build Date : Wed 24 Jan 2018 02:14:46 PM EST Build Host : b-centos7-2.freeswitch.org Relocations : /usr Packager : Ken Rice Vendor : http://www.freeswitch.org/ URL : http://www.freeswitch.org/ Summary : FreeSWITCH open source telephony platform -------------- next part -------------- An HTML attachment was scrubbed... URL: From donguyenha at gmail.com Sat Mar 31 03:45:49 2018 From: donguyenha at gmail.com (Do Nguyen Ha) Date: Sat, 31 Mar 2018 10:45:49 +0700 Subject: [Freeswitch-users] how to access the XML_REQUEST in mod_python Message-ID: hi list i want to use mod_python to dynamic load user directory and read the wiki but no documents for access XML_REQUEST how to get the XML_REQUEST in python script thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: From francesco at delagarda.com Sat Mar 31 07:21:54 2018 From: francesco at delagarda.com (Francesco Facco de Lagarda) Date: Sat, 31 Mar 2018 09:21:54 +0200 Subject: [Freeswitch-users] javascript calculate duration of call In-Reply-To: References: <05ab01d3c7ff$5dbb4be0$1931e3a0$@shenker.it> <05ba01d3c7ff$f05f4640$d11dd2c0$@delagarda.com> Message-ID: <53B23D7D-0E29-4418-88AC-00E1A5841429@delagarda.com> Thanks Brian for l’Inter, but exactly what am I doing wrong? Francesco Facco de Lagarda > On 30 Mar 2018, at 19:52, Brian West wrote: > > You're already doing it wrong if you're taking this approach. Review our existing CDR modules that would accomplish this for you. > > /b > > >> On Fri, Mar 30, 2018 at 3:20 AM, Francesco Facco de Lagarda wrote: >> Good morning all, >> >> I will be using both digital (which knows when the other part answers) and analogue (when, at least in Italy, doesn’t!) >> >> >> >> I need to calculate the total duration of an outgoing call. >> >> >> >> Calls also must have a preset max duration. I need to schedule a warning message (1 min before timeout) and a hangup (on timeout) for BOTH digital and analogue lines. >> >> >> >> This is the core of the code, reduced to the bone: >> >> >> >> var allowedSecs = 300; // max length 5 min … >> >> >> >> session.answer(); >> >> var sessOut = new Session("sofia/gateway/realtoneFXO/" + dialedNum + "@192.168.0.216:5060"); >> >> >> >> if (trunkIsDigital) { >> >> sessOut.execute("set", "execute_on_answer_1=sched_hangup " + allowedSects + " alloted_timeout") >> >> sessOut.execute("set", "execute_on_answer_2=sched_broadcast + " + (allowedSects-10) + " playback::" + soundDir + "one_min_left.wav both"); >> >> } else { >> >> // will schedule message and hangup on allowedSecs + graceTime to be calculated according to destintation (time to ring/answer) >> >> } >> >> >> >> if (sessOut.ready()) { >> >> bridge(session, sessOut); >> >> } >> >> sessOut.hangup(); >> >> session.hangup(); >> >> >> >> /*** >> >> HOW LONG DID THE CALL (bridge) ACTUALLY LAST?? .. for billing purposes >> >> **/ >> >> >> >> >> >> >> >> Thanks and Happy Easter! >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.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 >> http://www.freeswitch.org > > > > -- > > 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... 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