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| 03:24.26 | DannyA | hey all. im trying to use SendDTMF but the tones i hear are the shortest blips, barely registering to me. |
| 03:24.34 | DannyA | how do i get nice, long dtmf tones to play? |
| 03:35.36 | igcewieling | DannyA: call from a cell phone. |
| 03:35.53 | DannyA | i am |
| 03:36.48 | igcewieling | most sip phones will mute inband DTMF and Asterisk will mute inband DTMF and use rfc2833 DTMF. Enable DTMF debugging to see what is really happening with DTMF |
| 03:37.32 | igcewieling | you can't really "listen to dtmf audio" in VoIP. |
| 03:38.53 | DannyA | it's apparently working |
| 03:39.09 | DannyA | i made an automated system to ensure that audio is working in both directions |
| 03:39.22 | DannyA | the system initiates a test call from twilio, dialing our verizon did->asterisk every minutes |
| 03:39.31 | DannyA | there's a read, then a senddtmf |
| 03:39.37 | DannyA | it's meant to ensure audio works both ways |
| 03:39.53 | DannyA | even though i can't hear the dtmf tones when i call from my cell, apparently twilio is "hearing" them just fine |
| 03:40.22 | DannyA | is it still being sent via "audio"? meaning, as long as this "test" passes, i can assume human voice audio will also work in both directions? |
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| 04:02.55 | igcewieling | yes, you can assume that. |
| 04:03.10 | igcewieling | enable dtmf logging in logger.conf too. |
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| 13:45.50 | flok | how can I make all data from my iax phone go through asterisk instead of direct? |
| 13:51.15 | flok | I tried setting canreinvite=yes and directmedia=no |
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| 17:37.35 | igcewieling | "kernel:[Hardware Error]: MC4 Error (node 1): L3 data cache ECC error." |
| 17:37.47 | igcewieling | *sigh* I sometimes hate hardware. |
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| 18:57.13 | muks | grandstream support staff leave a lot to be desired.. they keep going on and on and on about changing config settings even after i provided enough info to demonstrate it's a firmware bug and asked the staff to forward the report to their developers |
| 18:57.56 | muks | asterisk logs errors like this when we use grandstream phones with 256-bit SRTP key (AES_256_CM) |
| 18:58.03 | muks | [2020-06-15 22:37:01] WARNING[38824] res_srtp.c: SRTP descriptions key length is '48', not '46' |
| 18:58.06 | 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 |
| 18:58.57 | muks | that inline: parameter of SDES is a base64 encoding of a concatenation of master key + salt |
| 18:59.15 | muks | >>> len(base64.b64decode('difEv6MhiHENEEqRBd1j5dWV1jtmGY8EL4t8RM4ix0RJiwTsrYxdupynS6GFroda')) |
| 18:59.18 | muks | 48 |
| 19:00.01 | muks | the AES-256 master key is 256 bits (32 octets) and its salt per RFC 6188 section 2 is 112 bits (14 octets) |
| 19:00.13 | muks | 32 + 14 = 46 whereas the phone is sending 46 bytes |
| 19:00.18 | muks | 32 + 14 = 46 whereas the phone is sending 48 bytes |
| 19:00.28 | muks | what more can i tell them? |
| 19:01.59 | drmessano | You seem to be answering your own question |
| 19:02.42 | muks | i want the support staff to fwd it on to their developers.. instead i get responses like: |
| 19:02.45 | muks | Could you: |
| 19:02.45 | muks | 2. provide the password of your phone config |
| 19:02.48 | muks | 1. please disable the "Crypto Life Time" in Asterisk and try again |
| 19:03.03 | drmessano | They're not competent enough to recognize the issue, so either wait until someone who buys their phones 20,000 at a time complains, or stick to 128 |
| 19:04.02 | drmessano | It's grandstream FFS |
| 19:04.15 | drmessano | Expect nothing, be happy with what you do get |
| 19:05.14 | drmessano | If I was buying for a buildout, I would have sent back these 50 or 100 phones and got a brand that did support 256 without a bug |
| 19:06.10 | sibiria | just an aside: 128 bit key is safe enough for rijndael |
| 19:06.20 | sibiria | there's no practically feasible attack against that key size |
| 19:06.43 | drmessano | My SB8200 cable modem shits the bed when I try to run LACP on it |
| 19:06.50 | drmessano | Known issue for 3 months |
| 19:06.54 | drmessano | Reported by many |
| 19:06.57 | drmessano | No fix |
| 19:07.04 | muks | this model of phone is kind of unique.. (the UI).. the phone works well mostly. that's the reason why i haven't given up.. it's an "almost there" |
| 19:07.23 | drmessano | and thats a cable modem, not a low-end phone with an obscurely high level of encryption |
| 19:07.34 | drmessano | It's not |
| 19:07.42 | drmessano | It's a "doesn't work" |
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| 23:32.14 | Casper | hi there, question about pap2t... I currently use a voip provider, which said that no issues has been reported... so I'm starting to doubt my pap2t... the symptoms is: intermittant no incomming audio, or dropping incomming audio. Like you pick up the line and you hear nothing but the caller hear you, or during a conversation you lose the incomming audio... |
| 23:32.24 | Casper | does this sound like a failing pap2t? |
| 23:33.40 | Samot | Well..I would say so. |
| 23:36.41 | Casper | and the grandstream ht-802 is still the recommended ata? |
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