Replies: 23 comments 31 replies
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If you are able to hear the tone, you should get inband dtmf to work. You can try with different dtmfmode settings for the chan_quectel trunk. |
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I already try relax and inband, result is the same. Is it possible to enable dtmf decode by the modem via At command? |
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If you are hearing the dtmf tone, you should get asterisk to recognize it working around with settings. AT+ QTONEDET=1 does enable detection and reporting by modem but without any additional code to read and take action on what is reported, it won't be of any use. |
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Where have you put relax and inband, in quectel.conf or in trunk settings? You should try with inband dtmf in trunk settings. |
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I put relax in quectel.conf. In freebpx I use a custom Trunk where i put "Quectel/quectel0/$OUTNUM$" as Custom dial string, but there aren't any settings about inband dmtf in custom trunks. Do you know if are there any conf. file where enable inband dtmf for custom trunks? |
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Do you thinks is possible to implement dtmt decode by modem? I'm quite familiar with C code and modem, may be with your help I can try to do it. |
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No need for it, lot of work will be required, just set dtmf mode correctly in trunk. If custom trunk is sip use dtmfmode=inband as part of trunk setting in conf file, if it's pjsip use dtmf_mode=inband and do a core restart |
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I've looked at the code and the copied code from chan_alsa does not handle dtmf data. Code will have to be modified for it to work with uac to handle non audio dtmf signalling. That is if it is working alright with serial audio port setting in the first place. Even without specific handling of dtmf, asterisk should be able to recognize audible dtmf with correct settings. Failing that, please disable uac and try with serial audio and default settings and let me know. If that works and you need it modified for uac, I'll try to do that in a couple of weeks. I think EG25-G should still have serial audio though it may no longer be documented. |
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I have 3 Trunks: one is the quectel, others two are PJSIP, but tath two are not involved in the IVR because asterisk route quectel inbound call to internal asterisk IVR. Anyway I triied to change PJSIP Trunks dtmf from RFC4733 to inband and put inband in general sip conf but the result is the same. Any others idea? |
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When you run tests with serial, make sure dtmf logging is enabled so that you can check whether any other setting (duration, gap etc) in quectel.conf needs to be changed if dtmf is detected but ignored |
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I can try with serial this afternoon for testing porpouse, but a the end I need UAC quecause I need serial port for GPS. |
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Code for dtmf frame is already present, it just needs to be enabled when using uac also, it's not inband alsa audio detection. Much easier than writing code for new response, parse, read and action. Anyway, need to know if you get it working with serial audio first (switch back to default settings and enable dtmf logging). |
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I tried now, seem something pass to asterisk but it ignore tone, do you know why? -- Executing [+XXXXXXXXXXX@from-mobile:1] NoOp("Quectel/quectel0-0100000005", "Entering user defined context from-mobile") in new stack |
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I change mindtmfduration=80 to mindtmfduration=50 , as you suggest and now it works: [2022-05-20 08:45:19] DTMF[14459][C-00000019]: channel.c:3896 __ast_read: DTMF end '2' received on Quectel/quectel0-0100000007, duration 51 ms So now can you enable same dtmf support for UAC? |
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Good. I'll get around to it in a few days. You could also make changes and compile. Presently this code block asterisk-chan-quectel/channel.c Lines 759 to 803 in b59d31e Another workaround could be sending call to another asterisk instance where ivr is located using a sip trunk whose dtmf is set to inband on both instances |
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Thanks be to God! |
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I can confirm that the problem is now solved. Thanks you very much IchthysMaranatha. |
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Very welcome. |
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Sorry for necroposting, but this discussion seems to be the most relevant to me. I have Asterisk 20.8.1 on Raspberry Pi 5, with EG25-G. I can make and receive calls and sms, have configured NetworkManager to connect through ModemManager to APN, everything seems to work, except for DTMF, it seems like Asterisk's deaf to the signals, as I can't see anything appear in the console after enabling the logging of dtmf. When I enable the |
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Actually the dtmf will be detected only on 2g/3g with this build as it processes the audible tones , with 4g/5g dtmf over ims is needed which requires a lot of changes. I'd made a working build for that around the same time as the uac dtmf fix, but don't seem to have uploaded it. I'd made it locally a long time ago, I'll check if I can find it. |
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That could be great, probably I'm not the only one who needs it. |
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Did not find the whole build, but had a folder with changed files. Have created a branch with those changed files, did not check deeply if there is a conflict with possible changes made later. Please test and see |
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Thanks, I'll let you know by tomorrow. |
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Well, couldn't wait... It looks like it recognizes the DTMF and extracts it, though I do have a problem here - the digit is not being handed to Asterisk. I defined a simple dial plan: But as soon as I dial 1(49) or 2 I'm getting disconnected with the following records in the console: |
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Oh, I'd coded and checked for outgoing dtmf using external ivr and it was working for that scenario, I'd put in the logs to check it was working correctly. My plan was to make a quectel.conf that could offer options for dtmf config, do some more testing, clean up the error logs and then publish it, but got busy with something else and never got around to it. I'd used microsip to test and it was working with one of the dtmf settings. Please try this fork, he seems to have integrated qtonedet https://github.com/RoEdAl/asterisk-chan-quectel |
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Well, I can confirm from the logs that their fork can decode DTMF successfully, the problem is with it 'as is' is that I have problems with sound ("[ALSA][PLAYBACK] Error - try again later"), and it requires exclusive lock on the ttyUSB2 which means that I can't use ModemManager to manage the APN connection. |
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I'll have a look at it and see if I can help, it might take a few days. |
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It can save, if not a life, then a few weeks for sure! |
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I went through the code again and actually it's already been put in the block here asterisk-chan-quectel/channel.c Lines 739 to 754 in 35abbf8 but not enabled for uac. Please check with serial audio. If it works as it should, we can enable it for uac also. |
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Ok, so I changed back the last bit of |
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Please test an incoming call that is answered using a softphone at an extension and then try the dtmf from the other end, I remember having tested it and working. Please check for proper working audio also. Please confirm with quectel status that serial is being used. |
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Well, anyway already disabled it, so will let it stay disabled until we'll get through this part. Didn't find quectel status, but the settings are as follows: Will route the call to SIP and let you know. |
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Ok, so I answered the call from MicroSIP, these are the logs when DTMF sent from mobile: And from softphone: |
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Forgot to add that the sound flows as expected. |
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So the error log in the block is not coming across? Is there a direct disconnection after dtmf event or manually hung up a little later? Seems like I'll have to setup the eqpt here and have a look. Haven't used it for a while, will get back to you after testing |
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Well, as we can see, the error logs do not appear, which is why I said it looks like the block is not reached. It's an immediate disconnection after the event in both cases. Thanks, much appreciate it |
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Took me a long time to get everything set up for testing as I was using this on a we826 router which I'd upgraded to openwrt 22 from 19. So the old binary could not be used. Built for openwrt 22 and results were what you experienced, channel crashes when dtmf is received. My recollection earlier of only testing outgoing dtmf was mistaken as qtonedet is used only for detection of incoming dtmf and this was tested and working. My problem with outgoing dtmf was sorted merely with microsip settings which is what came to mind when thinking of dtmf issues. I can only surmise that either I've not saved all the changed files or that the code does not work with Asterisk 18. I'll be able to carry out further testing on Saturday. Have you been testing with Asterisk > 16? If yes, if you could test with 16 it could save some time. I've the working binary for ramips router (mt7620) with openwrt 19 if you already have such a device. If it's an Asterisk version issue, problem is most likely with this line Line 208 in 35abbf8 |
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Looks to be working now, rolling forward to my original setup... |
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If you need it working with uac, the block marked earlier will have to be copied over just as it was done for non ims dtmf. |
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Ok, now it works with my original setup with serial audio port, later today I'll try to make it work with uac. Should I put the block(#31 (reply in thread)) right after line 910? Also it looks like a simple copy-paste would not be a good idea, as |
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Please try updated code, not tested myself Yes, depending on where block is copied, needs to be adjusted accordingly as done for non ims. You can look at how non qtonedet dtmf code was copied and adjusted. I think a single code block placed before branching for uac will also suffice. |
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Very good, moving the block outside the uac branching made it work. It seems that ModemManager sometimes eats DTFM, as when the service is running, not all Perhaps we could release them after they were consumed, but I don't think that would cause any problems. |
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Kudos for making the changes and getting it working! Yes, during the call, the dtmf digit remains in the structure while it's indicator is turned off, so that it's not sent again. After the call the entire structure is flushed out. You can now change all error logs to debug logs or delete them entirely, now that we know it's working. |
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You are the one to thank. Thank you for your time, effort and availability, much appreciated. |
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You're very welcome. |
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Hi, I have compiled chan-quectel module on freepbx with Asterisk 16.21.1 installed in a raspberry pi4. I'm using a quectel EG25-G modem conected via USB interface with UAC audio interface. Everything wark as expected: SMS, inboud audio call and outbond audio call, but when I route inboud call from quectel0 to my IVR, DTMF are not decoded. Using asterisk debug CLI noting appear if I press key from the caller phone. If I route inbound call to an extension I can hear dtmf tone. IVR work correctly with an SIP TRUNK. Can you confirm that DTMF decode should work with UAC audio interface in asterisk? I also try to enable TONE detection in the modem via AT+ QTONEDET command, in ASTERISK CLI (debug enabled) I can see the modem that recognize the DTMF tone because AT+ QTONEDET=X response appears, but IVR still do noting.
Please let me know.
Marco.
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