| 00:14.51 | joepublic | two extensions are to be behind a nat router |
| 00:15.31 | joepublic | and thanks very much; a clear and concise explanation |
| 00:27.34 | Samot | Well here's another little nugger. |
| 00:27.41 | Samot | Well here's another little nugget. |
| 00:28.02 | Samot | Phone A calls Phone B. Phone A offers X codec, Phone B doesn't. No call can happen. |
| 00:28.21 | Samot | Both phones _must offer_ a matching codec. |
| 00:28.48 | Samot | A PBX in the middle will handle that for the phones. |
| 00:29.31 | Samot | Phone A offers g711 and Phone B offers g722. The PBX will transcode between G711<>G722. |
| 00:42.34 | joepublic | that is a direct_media=off sounding nugget? |
| 00:50.03 | Samot | That's a what happens if direct media is on. |
| 00:50.14 | Samot | Side A and Side B must have matching codecs |
| 00:50.34 | Samot | Because there's nothing in between them to deal with the transcoding. |
| 00:51.26 | Samot | And as it was pointing out, nothing to handle the NAT if the endpoints are behind NAT |
| 00:51.39 | Samot | In this day and age, that would be almost every endpoint. |
| 00:52.08 | Samot | Unless you have a public IPv4 slapped on it or an IPv6 address |
| 01:01.31 | joepublic | Your sage advise is helpful and I thank you |
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| 17:25.27 | sahmed | Hello, I am facing an issue, if 2 endpoint negotiated with ulaw codec, but during rtp communication, if one side sending something else like g719 packet, asterisk drop the call. |
| 17:26.00 | sahmed | I dont have this g719 codec in my allow list |
| 17:26.25 | sahmed | is there any configuration, that asterisk will ignore those packets? |
| 17:28.20 | sahmed | According to RFC-3551, I think it should ignore the packets, "A receiver must ignore packets with payload types that it does not understand" |
| 17:29.44 | igcewieling | no. |
| 17:30.12 | seanbright | https://issues.asterisk.org/jira/browse/ASTERISK-28139 |
| 17:30.22 | seanbright | sahmed: ^^^ |
| 17:33.42 | sahmed | Thanks a lot seanbright, Let me trying this, asymmetric_rtp_codec |
| 17:37.38 | joepublic | the definition of understand may be an issue in such broadly written things. |
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| 18:21.46 | seanbright | sahmed: i don't see that language in RFC 3551 |
| 18:23.34 | Samot | sahmed: Why don't you just disable that codec in said device? |
| 18:24.08 | seanbright | ah, it's in RFC 3550 |
| 18:26.10 | sahmed | oh sorry seanbright, its rfc-3550 sec. 5.1 |
| 18:27.21 | sahmed | Samot: yeah in endpoint we disabled that codec and the customer device its, beyond our scope |
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| 18:45.23 | seanbright | sahmed: what version of asterisk are you running? |
| 18:45.37 | seanbright | hoping you can test a patch for that issue i linked earlier |
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| 18:57.44 | sahmed | I am running 13.20 |
| 18:58.58 | sahmed | asymmetric_rtp_codec=yes worked good |
| 18:59.21 | sahmed | but now call disconnected after rtp-timeout |
| 19:02.54 | seanbright | are you able to test a patch? |
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| 19:09.14 | sahmed | yeah I can |
| 19:09.41 | seanbright | cd /path/to/asterisk/source |
| 19:09.46 | seanbright | wget -O - https://pastebin.com/raw/REceNu8V | patch -p1 |
| 19:10.13 | sahmed | okay let me check this and update you accordingly |
| 19:10.31 | seanbright | k. you can turn off asymmetric_rtp_codec |
| 19:11.43 | sahmed | okay |
| 19:19.39 | sahmed | seanbright: Its working fine, and even, the call dose not disconnected after rtp timeout. Thats cool |
| 19:19.44 | sahmed | Thanks a lot |
| 19:20.25 | seanbright | you get a g719 packet on each call? or does it happen sporadically? |
| 19:20.41 | seanbright | ... shouldn't the call disconnect after an rtp timeout? isn't that the point of the rtp timeout? |
| 19:21.38 | sahmed | I mean if the packet dosen't come from endpoint, rtptimeout working fine |
| 19:22.32 | sahmed | last time while using asymetric_rtp_codec then, the g719 packet didn't update last_rx_ts |
| 19:22.54 | sahmed | thats why call was disconnected after rtp-timeout |
| 19:24.05 | sahmed | I think its sporadically |
| 19:24.30 | seanbright | interesting |
| 19:24.45 | sahmed | because customer complained 5 call out of 10 call disconnected immediately |
| 19:25.24 | sahmed | and we captured one of failed call and found that issue |
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| 19:45.07 | sahmed | seanbright: So we going to use that patch for prod version, |
| 19:45.51 | seanbright | i will see about getting it included. res_rtp_asterisk has changed alot since asterisk 13, so patching 16 and 17 might be more difficult. |
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| 21:32.48 | Penguin | The commenting in sip.conf indicates that tlsclientmethod only controls the outgoing client connections. Is there a setting to control the protocol/version as a server? |
| 21:39.06 | Penguin | Asterisk keeps sending the following message back to a phone, seen on a tshark capture: TLSv1 73 Alert (Level: Fatal, Description: Protocol Version) |
| 21:40.46 | Penguin | And the asterisk CLI shows the following at the same time: |
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| 21:40.56 | Penguin | tcptls.c:731 handle_tcptls_connection: Problem setting up ssl connection with peer: error:00000001:lib(0):func(0):reason(1), Internal SSL error |
| 21:40.59 | Penguin | tcptls.c:823 handle_tcptls_connection: FILE * open failed from peer |
| 21:41.19 | Penguin | The phone is trying to register. |
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| 21:55.16 | Samot | Penguin: Show the config for this peer. |
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| 21:57.19 | Penguin | The problem comes from trying to set up a new asterisk on a different server host. Same config files, new host, new certificate (but also tried old certificate copied over). |
| 21:57.45 | Penguin | The sip peer is set to use tls only. transport=tls |
| 22:03.45 | Samot | OK so then show the one that isn't working. |
| 22:03.48 | Samot | From the other side. |
| 22:04.13 | Samot | the error is pretty clear, there's a protocol version issue. |
| 22:04.29 | Samot | So Asterisk or something doesn't like the cert. |
| 22:05.49 | Penguin | That code 73 doesn't really reveal very much useful information. |
| 22:08.58 | seanbright | that's openssl for ya |
| 22:09.00 | Penguin | What doesn't make sense is that I copied everything from one host which has asterisk working and the phones register and work. I copied asterisk configs from that good host to a new host. I copied my certificate and key, too. On the new asterisk, I have all these TLS/SSL problems. |
| 22:09.19 | Penguin | I created a new key and cert... same problem. |
| 22:12.59 | Samot | So you're not going to show the configs for each side? |
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| 22:37.13 | Samot | I'll take that as you're not showing things. |
| 22:39.31 | electronic_eel | Penguin: check that it is not a permissions problem on the certificate files. Also selinux could forbid accessing the files |
| 22:41.30 | electronic_eel | and btw, you write about "sip.conf" - that is the old chan_sip. If I'd migrate to a new server, I'd consider also moving to pjsip |
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| 23:14.01 | Casper | hi there, what would be a good voip ata nowadays? |
| 23:41.37 | cyclaw | I can't speak for nowadays, I'm using an old SPA3102, I think they have a V2 now. Also Grandstream seem pretty good. |
| 23:43.38 | Casper | I have the good old pap2t, but father want to move to voip, so... need a new one for him |
| 23:47.46 | cyclaw | Casper: if you only want FXS there's plenty to choose from, if you want a gateway/FXO Grandstream HT813 looks good. |
| 23:47.52 | Samot | They do not have a V2 of the 3102 |
| 23:48.05 | Casper | fxs only |
| 23:48.18 | Samot | It's been retired for a very long time and its replacement is also now retired. |
| 23:48.46 | Samot | The SPA1x2's are the only ones that are still supported right now. |
| 23:49.06 | Samot | The SPA8000, SPA8800, all the older models are no longer supported. |
| 23:49.27 | cyclaw | In that case I'd maybe look on amazon and pick up a grandstream or cisco. |
| 23:49.46 | Casper | grandstream ht802 maybe |
| 23:50.06 | cyclaw | Yeah I saw that, looks good. |