On 8/17/21 4:22 PM, Antony Stone wrote: Asterisk was just one possibility of course. I don't know your case in detail and probably best to direct that question to an Asterisk group of course... This said, Asterisk if as a b2bua you may want to look at moh_passthrough and ARI. -- Fred Posner -- www.palner.com Matrix: @fred:matrix.lod.com

On Tuesday 17 August 2021 at 21:22:37, Antony Stone wrote: Sorry for telling you this, but if you think that, you don't know Asterisk at all. You don't need to send a REINVITE to put a call on hold, just change how you handle the RTP of the call if you SIP endpoint doesn't support REINVITES/UPDATES. Asterisk knows about presence, it doesn't have full implementation, but you either need it. Also transfers doesn't force you to do them throught REFERS The thing you need it's a B2BUA, and Asterisk, FreeSwitch, YATE, SEMS, either a simple python script using pjsua2 or any of the dozens SIP frameworks out there, will allow you to acomplish your goals If you go the Asterisk road, any of the AMI/ARI documentation will explain to you how to handle ongoing calls. If you prefer FreeSwitch, you will have to surf the ESL seas. I you take the SEMS aproach, SBC + DSM/Python/xmlrpc You could take any road you like, but not Kamailio, as it's not a B2BUA. I suggest you, first undestand the concepts you need to manage, then you will be able to pick up a tool for solving the problem. Saludos -- Raúl Alexis Betancor Santana Serlink Telecom S.R.L.U.

On Wednesday 18 August 2021 at 09:36:02, Antory Stone wrote: Just tells Asterisk to put the call on hold. On your scenario of controlling the call from outside the SIP channel, Custom Dynamic Features (https://wiki.asterisk.org/wiki/display/AST/Custom+Dynamic+Features) and AMI PlayDTMF (https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerAction_PlayDT…) could do the trick That's not true, you could use res_pjsip and RLS, It will need some dialplan and script coding but could be done. Either way I don't undestand why you whant Presence handling if your are using a soo dumb SIP Client. I've told you how to doit: CDF + AMI PlayDTMF Partially true. NO, what you first need is to undestand how SIP protocol works, then which role could play any of the available tools, next design how your are going to get you problem solved and which tools with which roles you need to use to acomplish your goals, and finaly build and test your solution. There is no pre-build tool to solve your problem and no magic wand to move and get it solved. I have pointed you to how to solve it, it's up to you to read the docs and undestand how to get to the final solution. I'm not going to argue agains what people from the Asterisk lists says, there are too many things done wrong inside Asterisk since it was first launched, and lot of things that it's supposed it could not do, it does, if you know how to deal with it. So read the docs of that SIP Server and how it allows you to put a call on hold, Asterisk, for example, allows your to do so, playing DTMF codes on the on-going call, no matter that the SIP endpoint does not have a "hold button". If your SIP endpoint it's soo dumb that it could not play DTMF, your could simulate the "hold" button throught CDF+AMI PlayDMTF, I've told you so. Just rethink what you need and what are you doing. Seems to me, like your are trying to build the Piramids and neither you know how a hammer works. Saludos -- Raúl Alexis Betancor Santana Serlink Telecom S.R.L.U.

On Wednesday 18 August 2021 at 13:40:25, David Villasmil wrote: But, would that not perform the hold function on the FreeSwitch server? And, if after putting it on hold, I need to transfer a call to another number / channel, that transfer would also take place on the FreeSwitch server? Or am I misunderstanding, and FreeSwitch can pass the instruction to "place the call (channel if you prefer) on hold" on to the PBX system which my existing client application is placing its calls through? This all sounds as though I am getting FreeSwitch to "do the work" and manage the hold/transfer/resume/conference/whatever itself. I could do that with Asterisk (which I already know, whereas I haven't used FreeSwitch), but it isn't a solution to my needs. I need something which will *tell the existing PBX* to do these things (and it's probably not running FreeSwitch), in just the same way as pressing the buttons on a competent SIP phone such as Yealink or Polycom would tell the PBX to do it. Antony. -- The Magic Words are Squeamish Ossifrage. Please reply to the list; please *don't* CC me.

On Wednesday 18 August 2021 at 15:11:54, Raúl Alexis Betancor Santana wrote: Not an option, as already stated. Precisely what I am asking how to do, thank you. Agreed, I realise now that Kamailio is not the solution to my requirements, however people here seem to believe they know what *would* be a solution to my requirements, but so far nobody has pointed me at anything specific which I can use. Hints are all very well, but telling me "put a B2BUA in front of that SIP Endpoint, and use API, DTMF, RPC or witchever method that B2BUA gives you" doesn't exactly help when I've made it perfectly clear that I don't know how to solve the problem. If it really is that simple, please just point me at one example of how to actually do it. Antony. -- How many Prolog programmers does it take to change a lightbulb? No. Please reply to the list; please *don't* CC me.

On Wednesday 18 August 2021 at 16:16:29, Raúl Alexis Betancor Santana wrote: No, he told me how to get FS to put a call *which it is processing* on hold *inside FreeSwitch* - that is *not* what I need to do - I need to send a command to the *existing PBX* which is handling the call so that *it* puts it on hold, just the same as it would if I had standard SIP phone and could press the "hold" button. If you are referring to: "The thing you need it's a B2BUA, and Asterisk, FreeSwitch, YATE, SEMS, either a simple python script using pjsua2 or any of the dozens SIP frameworks out there, will allow you to acomplish your goals "If you go the Asterisk road, any of the AMI/ARI documentation will explain to you how to handle ongoing calls." then this neither tells me how to do it, nor is it even what I need to do (see above about FreeSwitch). Please show me just one of them. Which ones, please? You seem to know, why not just tell me? If I find someone who says they can do it, that's definitely an option. Try what? I'm just saying that if you know how it can be done, please show me an example. You really don't get what it is I need to do. I do NOT need to build myself a B2BUA using any of the above tools and get *that machine* to put calls on hold, transfer them, conference, etc. I can, and have, done that perfectly well using Asterisk. Despite your opinion, I do in fact know both SIP and Asterisk pretty well. What I need is something which can tell *the existing server* to put the call on hold, resume it, or transfer it, in just the same way that a competent SIP telephone can tell the server to do that. How does playing a DTMF tone down a channel tell some other server to put a call on hold? That makes no sense at all. I'm looking for the textbook. (Oh, and if your response is "buy the Kamailio book", I have.) I have tried to be polite in my responses to you; I would appreciate if you did the same. Regards, Antony. -- Ramdisk is not an installation procedure. Please reply to the list; please *don't* CC me.

On Wednesday 18 August 2021 at 17:18:20, Fred Posner wrote: Yes, I understand that. My responses to these suggestions have largely been based on people who are experts with these B2BUAs telling me specifically that they can NOT do what I want. A B2BUA can definitely do all the call processing I want, but internally. I need something which can *instruct the existing PBX* to do it. That's the bit which Raúl in particular did not seem to grasp. I have not yet asked FreeSwitch, but the responses from the Asterisk and PJSIP projects indicate that their software cannot achieve what I am looking for. I shall continue my search, and I definitely accept that Kamailio is not the right tool. Thank you - that is appreciated. Regards, Antony. -- Wanted: telepath. You know where to apply.

Hello Anthony, As discussed earlier, Kamailio is not the right tool for your specific requirements. Here on this list are people naturally most experienced with this software. We can also not provide a detailed solution to any problem. Regarding about telling the PBX to put the call on hold, one way how to do it is by sending a Re-INVITE with a changed SDP from one of the called parties. Maybe you can get more detailed and useful answers for you at the Asterisk or Freeswitch user lists. If hiring somebody is an option for you, we've is a business list available where you can just ask for people to contact you regarding your particular requirements. You will find similar lists also for other projects. Best regards, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com -----Original Message----- From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Antony Stone Sent: Wednesday, August 18, 2021 5:01 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] Is this idea even feasible? On Wednesday 18 August 2021 at 16:16:29, Raúl Alexis Betancor Santana wrote: No, he told me how to get FS to put a call *which it is processing* on hold *inside FreeSwitch* - that is *not* what I need to do - I need to send a command to the *existing PBX* which is handling the call so that *it* puts it on hold, just the same as it would if I had standard SIP phone and could press the "hold" button. If you are referring to: "The thing you need it's a B2BUA, and Asterisk, FreeSwitch, YATE, SEMS, either a simple python script using pjsua2 or any of the dozens SIP frameworks out there, will allow you to acomplish your goals "If you go the Asterisk road, any of the AMI/ARI documentation will explain to you how to handle ongoing calls." then this neither tells me how to do it, nor is it even what I need to do (see above about FreeSwitch). Please show me just one of them. Which ones, please? You seem to know, why not just tell me? If I find someone who says they can do it, that's definitely an option. Try what? I'm just saying that if you know how it can be done, please show me an example. You really don't get what it is I need to do. I do NOT need to build myself a B2BUA using any of the above tools and get *that machine* to put calls on hold, transfer them, conference, etc. I can, and have, done that perfectly well using Asterisk. Despite your opinion, I do in fact know both SIP and Asterisk pretty well. What I need is something which can tell *the existing server* to put the call on hold, resume it, or transfer it, in just the same way that a competent SIP telephone can tell the server to do that. How does playing a DTMF tone down a channel tell some other server to put a call on hold? That makes no sense at all. I'm looking for the textbook. (Oh, and if your response is "buy the Kamailio book", I have.) I have tried to be polite in my responses to you; I would appreciate if you did the same. Regards, Antony. -- Ramdisk is not an installation procedure. Please reply to the list; please *don't* CC me. __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions * sr-users(a)lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe: * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

On Wednesday 18 August 2021 2021 5:18:51, Antony Stone wrote: And again arguing ... your are asking for the FULL solution, and we refuse to give it to you. David have pointed you to a way of how to solve it, you just DON'T undestand how to do it, but that's your fault, not our. A B2BUA MAN! ... how many times you need to hear that? A sip proxy, could not inject things in the dialog whitout braking it, that why we told you, you need a B2BUA, but it's clear, that you don't get the point. And I have told you, that and Asterisk its perfectly capable as a client to do so, the problem is that you NEED TO KOWN how to control Asterisk for that, and I have told you how. CDF+AMI, there is no prebuild solution for that, so get your hand on if you are willing to have a solution. With FreeSwitch it's lot easier to do manipulations on A-Leg or B-Leg of the call from the ESL than with Asterisk, but both could do it. Best regards

On Wednesday 18 August 2021 at 19:11:02, Fred Posner wrote: Thank you Fred, sincerely. I also see no further purpose in continuing this conversation here. Antony. -- #define SIX 1+5 #define NINE 8+1 int main() { printf("%d\n", SIX * NINE); } - thanks to ECB for bringing this to my attention

The SIP protocol does not allow you to inject messages in the middle of a SIP dialog. The CSEQs will be out of order on one end and other messages will be ignored. That’s why everyone points you to the b2bua solutions where you have different dialogs on both ends. Start with accepting that, then look for a small b2bua that fits your needs, that isn’t configured to be a full featured PBX, which you clearly stated that you don’t want. /O

Guys, I just ACKed the stuck post from Raul in the list manager, should be end up on the list in some minutes. Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com -----Original Message----- From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Raúl Alexis Betancor Santana Sent: Wednesday, August 18, 2021 6:44 PM To: Kamailio <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] Is this idea even feasible? On Wednesday August 18, 2021 4:01:10, Antony Stone Man, really, you don't know how things works so STOP saying that something is not what you need, only because you don't have the knowleage. David, told you EXACTLY, what you need, send the API call to hold the B-Leg on the FreeSwitch. YDSC ---- > FS ---- > PBX If you send the HOLD api commando to FS, and told it to doit on the B-Leg of and outgoing call, FS will put the leg PBX on hold, EXACTLY as if you have a "hold" button on your YDSC (Your Dumb SIP Client). But as is more than clear, that you don't have a f**k idea how things works on the SIP world, you keep saying "that's not what I was asking for". No, I was refering to the reply that still is not on the list, where I pointed you to and way to solve your issue using Asterisk, pointing you to the DOCS where it's explaixed how that AMI commands works and what you need to take into account on the dialplan to get it working. That's point you to the docs WHERE it explained how it works. But you whant me to give you the full solution, that will not happen. Again ... RTFM, I've told you more than tree times, that the solution is on the docs. Because I do not work for free, and your attitude on the thread, not ever reading the docs we point you, don't give me any incentive to do it so. I could count at least 5 people on this list, that have told you, that could be done, but will be easier do it using a B2BUA. Try to make me doing your job, and that will not happen, I've told you why. I've point you to the docs where it's explained how to put a call on hold, in Asterisk throught the AMI and enabling some features on the dialplan manager. I'm not going to give you a full step-by-step guide on how to do it. I got it perfectly, It's just I don't want to give it to you, as simple as that. Your answers show to me, and the rest of the list, that what you call "knowing Asterisk" it's probably nothing deeper than runing an Issabel or FreePBX, if you would REALLY know Asterisk and the SIP protocol, you will not insist that what we have told you is not what you need. Again an again ... YOU COULD DO THAT WITH A CUSTOMIZED B2BUA. Get it or leave it So you say you know Asterisk and you don't know that on Asterisk and on any other SIP pbx, you could control the call flow using DTMF? ... Really? http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/Additio… Please read the [featuremap] section. You could do that control on the A-Leg or on the B-Leg, if your upstream PBX doesn't support DTMF call-control, THATS where your customized B2BUA take into place. If your still doesn't unsdestand that. Just give up on this. And if you have it, why are you asking if Kamailio could do a B2BUA Role feature, like putting a call on hold? ... just because you don't undestand a word of what you have read from that book. I'm been very polite on my replies, better you don't get see my BOFH side ;-) There is thousands ways of skinning a cat, and you insist on the only one that don't work ... ;-) Best Regards, __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions * sr-users(a)lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe: * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

On Wednesday 18 August 2021 at 13:40:25, David Villasmil wrote: Thank you for your assistance, and pointing in a helpful direction. I now have a working solution based on FreeSwitch. This is indeed what happens - FreeSwitch sends a re-invite containing sendonly to the upstream PBX, which then puts the other party on hold and plays its own hold music at them. Exactly what I needed. Thanks, Antony. -- I want to build a machine that will be proud of me. - Danny Hillis, creator of The Connection Machine Please reply to the list; please *don't* CC me.