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| 05:04.01 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.34.0 (2020/06/11) 16.11.0 (2020/06/11) Standard: 17.5.0 (2020/06/11); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
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| 18:56.39 | hardwire | Anybody else noticed ARI channels/create in Asterisk 16 (only one I have access to right now) sets up the call ready for channels/dial but it's missing all the channel vars and such. |
| 18:57.23 | hardwire | I've been using a POST to channels to just directly originate for now. I might have my eyes crossed or something trying to create channels using channel/create in a similar fashion. |
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| 19:25.52 | file | define channel vars |
| 19:26.03 | file | it's only as of the latest release that it supports setting channel variables |
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| 23:07.45 | muks | i get the following asterisk log messages when selecting "256 bit" SRTP in my SIP phone: |
| 23:07.53 | muks | [2020-06-15 22:37:01] WARNING[38824] res_srtp.c: SRTP descriptions key length is '48', not '46' |
| 23:07.56 | muks | [2020-06-15 22:37:01] WARNING[38824] res_pjsip_sdp_rtp.c: Ignoring crypto offer with unsupported parameters: 1 AES_CM_256_HMAC_SHA1_80 inline:difEv6MhiHENEEqRBd1j5dWV1jtmGY8EL4t8RM4ix0RJiwTsrYxdupynS6GFroda|2^32 |
| 23:08.04 | muks | [2020-06-15 22:37:01] WARNING[38824] res_srtp.c: SRTP descriptions key length is '48', not '46' |
| 23:08.05 | muks | [2020-06-15 22:37:01] WARNING[38824] res_pjsip_sdp_rtp.c: Ignoring crypto offer with unsupported parameters: 2 AES_CM_256_HMAC_SHA1_32 inline:RF2WqnYlrqawKup/TLHDlT3HgupT36Twh++RDJ4YZuJ2/IzsIjqS0mR8UbEuFUZr|2^32 |
| 23:08.22 | muks | and after placing a call, media is not audible |
| 23:09.30 | muks | it appears that the asterisk build supports the 256-bit ciphers, but doesn't like what the SIP client has sent it |
| 23:09.51 | muks | it is logged here: https://github.com/asterisk/asterisk/blob/16/res/res_srtp.c#L1039 |
| 23:10.39 | muks | the expected key length is assiged here: https://github.com/asterisk/asterisk/blob/16/res/res_srtp.c#L912 |
| 23:11.23 | muks | is asterisk at fault here, or the SIP phone client? (i'll create a ticket for the respective product for it) |
| 23:13.10 | muks | ("128 bit" selection for SRTP works fine) |
| 23:14.04 | drmessano | What SIP client is this? |
| 23:15.10 | muks | Grandstream GXP-2160 phone |
| 23:16.35 | muks | it is the same issue as another user's report here for freepbx: https://issues.freepbx.org/browse/FREEPBX-20284 |
| 23:17.10 | muks | but that user seems to think that the asterisk build does not support 256 bit key lengths, which is incorrect from the logged messages above. the logged messages can only be logged if support is present. |
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| 23:26.13 | muks | i like this device, but it seems to have some other bugs. it supports TLS v1.2 and ECDSA certificates, but in the current firmware, it doesn't appear to be able to validate the certificate chain. one has to keep validation disabled (default), or the validation fails (e.g., with a lets encrypt certificate or custom CA certificate) |
| 23:27.31 | muks | another nit is that in domain name validation from the certificate CN, it doesn't support subjectAltName (but that's easy to work around) |
| 23:29.08 | muks | have filed a bug with them for the lack of certificate chain validation which is a serious bug. if this SRTP issue is also their bug, i'll file another ticket with them. |