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| 00:56.28 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.14.0 (2017/02/13), 11.25.1 (2016/12/08), Standard: 14.3.0 (2017/02/13); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 01:28.14 | eekrano | Hey all, trying to set up the latest FreePBX distro with SRTP and TLS, however calls donât go through and the asterisk logs show lines similar to: âsdp_srtp.c: Unsupported crypto suite: AES_256_CM_HMAC_SHA1_80â - Google canât seem to help. Any thoughts? Iâve tried with a LetâsEncrypt certificate as well as a self-generated one. Thank you |
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| 13:29.12 | forgotmynick | Hello |
| 13:30.17 | forgotmynick | Is anyone going to work on a replacement for google voice since google is shutting down google talk? |
| 13:37.42 | Samot | What do you mean a replacement? |
| 13:37.54 | freebs | yes flowroute |
| 13:37.56 | freebs | lol |
| 13:43.14 | Samot | I think the answer is no. There is no one working on a free phone service that requires XMPP to be used. |
| 13:43.20 | Samot | As far as I know. |
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| 20:21.13 | compu_85 | hiu |
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| 21:59.47 | dan_j | Whats the easiest way to extract the CID from an incoming P-Asserted-Identity header? |
| 22:03.39 | [TK]D-Fender | Depends |
| 22:03.58 | [TK]D-Fender | Easy for whom? |
| 22:04.08 | [TK]D-Fender | How mangled is its presentation? |
| 22:04.20 | [TK]D-Fender | There is a clear dialplan function and those to hack it up |
| 22:04.29 | [TK]D-Fender | Or you could do it on the outside |
| 22:04.35 | [TK]D-Fender | Depends on you |
| 22:04.55 | dan_j | The presentation in this case is controlled by me and being sent to asterisk from a freepbx server. |
| 22:05.49 | [TK]D-Fender | So pick how you want to mangle that variable |
| 22:06.23 | dan_j | In asterisk, I need to extract the CID from the PAI and check that it's approved for us. Eg. P-Asserted-Identity: <sip:01614321234@192.168.1.4> so i need the 01614321234 to compare with an approved list. |
| 22:06.36 | [TK]D-Fender | So got cut it up |
| 22:06.40 | [TK]D-Fender | go* |
| 22:06.55 | dan_j | Ok. Just checking there isn't anything simple like CALLERID(num) |
| 22:07.13 | dan_j | Thanks |
| 22:07.28 | [TK]D-Fender | unless it's counted at CID.... which you should already know the answer to. |
| 22:07.52 | [TK]D-Fender | * TRUSTING it and using it for CID is up to you. |
| 22:08.01 | [TK]D-Fender | so go look at your calls |
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| 22:12.01 | dan_j | I've done a dumpchan and the PAI is not listed so looks like im going to have to manipulate it myself and then set it in the channel if its accepted. |
| 22:15.40 | Samot | Is CALLERID() not getting the details? |
| 22:21.33 | dan_j | No that I can tell. |
| 22:21.36 | dan_j | not* |
| 22:22.11 | dan_j | RECEIVEDCALLERID=<sip:01614321234 |
| 22:22.21 | dan_j | How do I cut everything before and including the : ? |
| 22:22.42 | Samot | Well.. |
| 22:22.48 | Samot | How are you setting that? |
| 22:22.49 | dan_j | Already cut everything after and including the @ but cant seem to get cut to remove prior |
| 22:23.08 | dan_j | exten => _X.,5,ExecIf($[ "${RECEIVEDCALLERID}"!="" ]?Set(RECEIVEDCALLERID=${CUT(RECEIVEDCALLERID,@,1)})) |
| 22:23.52 | dan_j | This is how I set RECEIVEDCALLERID to start with |
| 22:23.52 | dan_j | exten => _X.,4,Set(RECEIVEDCALLERID=${PJSIP_HEADER(read,P-Asserted-Identity)}) |
| 22:24.42 | dan_j | Cant seem to get CUT to remove everything before a delimiter |
| 22:25.19 | [TK]D-Fender | Show us your attempt |
| 22:25.20 | dan_j | From asteriskguru.com, We have set this parameter to 1, which means that the application will Cut all characters before the first "/" character. |
| 22:25.43 | dan_j | But in my case, setting 1 removes everything after the first character |
| 22:25.55 | file | have you set trust_id_inbound to yes? |
| 22:26.40 | dan_j | file: no. it's on default |
| 22:26.48 | file | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_res_pjsip#Asterisk13Configuration_res_pjsip-endpoint_trust_id_inbound |
| 22:26.57 | dan_j | Will give it a go. Thanks. |
| 22:27.08 | file | if you set it to yes then CALLERID(num) will contain the value |
| 22:27.36 | Samot | But where are you setting RECEIVEDCALLERID? |
| 22:27.46 | file | otherwise you could use ${RECEIVEDCALLERID:5} to strip the first 5 characters |
| 22:27.52 | Samot | I can't find that setting/function anywhere in the docs... |
| 22:28.07 | Samot | Oh you pasted them out of order. |
| 22:28.35 | Samot | So you aren't using CALLERID() |
| 22:28.53 | Samot | You're grabbing a header and trying to CUT it.. |
| 22:29.35 | Samot | So you have to CUT it again |
| 22:29.41 | Samot | To get rid of <sip: |
| 22:32.24 | [TK]D-Fender | waits for an actual call |
| 22:33.41 | dan_j | [TK]D-Fender: Used file's solution. Thanks |
| 22:35.09 | Samot | Yes |
| 22:35.28 | Samot | Because now you can just use CALLERID(num) or CALLERID(name) |
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| 22:43.38 | dan_j | Is it normal for 'pjsip reload' to take a long time? I also have a problem with 'pjsip show endpoints' taking a long time. It actually causes calls to drop during peak usage. |
| 22:43.50 | dan_j | It = 'pjsip reload' |
| 22:45.12 | nickaugust | what linux softphone do you guys like? we've been using linphone which works well but doesnt seem to respond to presence messages from asterisk for some reason |
| 22:46.20 | dan_j | nickaugust: I use zoiper for testing but have had a few clients complain about it. they've switched to xlite or bria |
| 22:46.53 | nickaugust | yeah I dont think the counterpath software runs on linux any more |
| 22:47.12 | Samot | Are you setting up the hints? |
| 22:47.19 | nickaugust | yeah ive got the hints |
| 22:47.23 | nickaugust | and it works on other clients |
| 22:47.29 | nickaugust | just not linphone for some reason |
| 22:47.35 | dan_j | file: I've set debug to 3, but when I issue 'pjsip reload' the cli freezes until the command is complete. Is this a known issue? |
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| 22:48.05 | Samot | How long does it freeze? |
| 22:48.17 | file | any known issues would be present in JIRA, as to your previous question - it depends on your configuration |
| 22:48.26 | file | if you are using realtime for example and have a lot of endpoints |
| 22:48.33 | dan_j | I'm using static |
| 22:48.57 | Samot | How many endpoints? |
| 22:49.03 | dan_j | About 10 seconds till the command completes. The same with 'pjsip show endpoints'. Just checking the debug now to see if I can spot anything obvious. Not noticed anything in JIRA about it. |
| 22:50.14 | dan_j | Samot: 355 |
| 22:51.55 | file | do they have registered contacts and do you have qualify turned on? |
| 22:54.31 | dan_j | Qualify is turned on and most are registered. |
| 22:54.44 | dan_j | Some are unavailable because they aren't connected. |
| 22:54.48 | dan_j | powered |
| 22:54.48 | file | https://issues.asterisk.org/jira/browse/ASTERISK-26806 there's a few issues associated with it too, may be in the same area |
| 22:55.45 | dan_j | Thanks. might be related. |
| 22:56.01 | dan_j | Should I open an issue anyway or comment on that one? |
| 22:56.46 | file | you can comment on it |
| 22:56.55 | dan_j | np |
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| 23:56.53 | Lime123 | Hi, Hope you all having a great day. Is there anyone know what's the status of "freenum.org" ? |