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| 07:59.27 | golserma | Good morning. I have a question. I have a local asterisk server for my local phones (lan). And I have configured a trunk (receiving cals from landlines). The trunk is connecting to the server of my provider using NAT. The pinhole seems to work ok. I refresh every 30 seconds (port 5060) now I'm wondering how RTP works? is RTP used for normal SIP |
| 07:59.27 | golserma | cals or just if there is video? |
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| 08:48.14 | post-factum | rtp is used for media stream, be it audio or video |
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| 09:17.08 | golserma | thats strange for me i don't see any ports in the RTP range beeing used if I receive a call |
| 09:17.36 | golserma | are only the phones using the portrange or the servers too? |
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| 09:35.28 | golserma | OK solved the mistery using iptraf: So If I receive a call my pbx opens a udp connection in the middle of the RTP range to the trunk server. |
| 09:36.43 | golserma | I guess the trunk is using the udp connection to send audio back aswell. |
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| 12:01.57 | nbjoerg | yes, it is normally one "connection" |
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| 15:55.37 | dacod | hi all |
| 15:55.52 | dacod | I have some problems with activate realtime |
| 15:56.02 | dacod | i use 13.29.2 |
| 15:57.01 | dacod | and the same sip.conf work well with conf files, but with realtime db, REGISTER process works well |
| 15:57.34 | dacod | but, calls to peers/friends not work |
| 15:59.37 | dacod | and receive the "retrasmission" messages |
| 15:59.44 | dacod | like "Retransmission timeout reached on transmission " |
| 15:59.57 | igcewieling | that is a nat issue |
| 16:01.02 | dacod | igcewieling: yes, but this issue apeers just when I activate the realtime |
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| 16:01.39 | igcewieling | so the problem only happens when you replace your static config files with a database config? naw, nothing different at all. |
| 16:01.48 | dacod | yes |
| 16:02.28 | dacod | if rollback to conf files just work fine |
| 16:03.43 | igcewieling | any chance you are using a chan_sip realtime, with pjsip? I don't know how that might work. |
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| 16:04.33 | igcewieling | My experience with Realtime was so unhelpful, I switched back to using config files many years ago. |
| 16:04.59 | igcewieling | I actually generate my .conf files from the realtime tables, even though Asterisk doesn't use them anymore. |
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| 16:07.09 | dacod | chanip with pjsip, i will try some documentation, not work with pjsip yet |
| 16:08.33 | igcewieling | People mostly assume pjsip these days since chan_sip is considered "old" |
| 16:08.34 | dacod | I do the same, create "static" conf files from database |
| 16:09.27 | dacod | yes, I'm old to :-) like chan_sip |
| 16:09.31 | Kobaz | igcewieling: yup, that's what i'm doing |
| 16:09.44 | Kobaz | dacod: you won't like chan_sip anymore after you learn pjsip and realize what limitations chan_sip had |
| 16:10.29 | igcewieling | Kobaz: do chan_sip and pjsip use different tables? the config option names changed between the two. |
| 16:10.41 | Kobaz | igcewieling: nope |
| 16:10.53 | dacod | I read about this, but I'm trying not change the application logic in this moment |
| 16:11.10 | igcewieling | *nod* they must do some sort of translations. |
| 16:11.11 | Kobaz | igcewieling: my backend tables haven't changed, but i have a db function that reads the tables and outputs asterisk config. i only had to change that |
| 16:11.53 | Kobaz | so instead of calling asterisk.asterisk_exten_config('SIP'), it now handles asterisk.asterisk_exten_config('PJSIP_WIZARD') and 'PJSIP' for transports/etc |
| 16:17.41 | Samot | dacod: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Deprecations |
| 16:17.48 | Samot | dacod: You have three releases. |
| 16:18.00 | Samot | dacod: After that, Chan_SIP is gone. |
| 16:19.21 | dacod | yes, its time to get new chan and learn |
| 16:20.00 | dacod | thanks all |
| 16:20.31 | Samot | You should also review that page I linked. |
| 16:20.38 | Samot | Asterisk 21 is going to be a major house cleaning release. |
| 16:21.05 | igcewieling | eagerly awaits Asterisk 22. 8-| |
| 16:21.05 | Samot | As will 19 this year. |
| 16:21.47 | Samot | The next two Standard versions will clean up house in a big way. |
| 16:22.09 | dacod | lol |
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| 16:22.56 | Samot | Well I'm pretty sure res_monitor being removed in 21 will impact just as many as chan_sip being removed. |
| 16:23.12 | igcewieling | I sort of surprised iax hasn't been removed. |
| 16:23.33 | file | still core supported, for the moment |
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| 16:23.40 | igcewieling | It lost the Protocol Wars, accept it and move on. 8-) |
| 16:24.43 | Samot | It's not even that really. |
| 16:25.03 | Samot | A lot of the issues that IAX solved back in the day, are no longer issues that need to be solved really. |
| 16:25.38 | Samot | IAX, today, solves one major issue. The person being to cheap to have a real router/firewall in front of their PBX. |
| 16:26.18 | igcewieling | At least we are stuck with SIP and not H323 *shudder*. |
| 16:26.35 | igcewieling | Personally, I liked the design of MGCP. |
| 16:27.37 | igcewieling | MGCP is dumb-ish phones, smart PBX -- the way phones are supposed to work. Instead we get phones which are mini-PBXs on each desk. |
| 16:29.18 | Samot | Im not sure I follow that. |
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| 16:31.56 | Samot | I dont see IP phones as mini-pbx systems |
| 16:31.57 | igcewieling | SIP phones are basically a mini-PBX in the form of a phone. |
| 16:32.05 | Samot | Not really |
| 16:32.58 | igcewieling | They provide dialtone, they do digit mapping, digit translations, call routing. |
| 16:33.13 | Samot | Thats not a PB |
| 16:33.15 | Samot | PBX |
| 16:33.21 | igcewieling | they even do forwarding independent of a PBX. |
| 16:33.32 | Samot | Still not a PBX |
| 16:33.43 | Samot | A phone cannot replace a PBX |
| 16:34.18 | igcewieling | In MGCP the PBX handles all that. The PBX provides dialtone, the PBX does forwarding and call routing. |
| 16:34.26 | Samot | And what you described is SIP router/proxy functions |
| 16:34.42 | igcewieling | What I described can be done by any polycom phone. |
| 16:35.04 | Samot | And ATAs |
| 16:35.27 | Samot | I can do all that in ATAs |
| 16:36.19 | igcewieling | "Mini-PBX" might be a slight overstatement, but my point is that SIP devices are overly intelligent and provide many of the same functions as a PBX. |
| 16:36.49 | Samot | Well you are describing basic SIP functions. |
| 16:37.02 | Samot | None of which require a PBX |
| 16:37.07 | igcewieling | It is the whole "peer to peer voip" thing which never seemed to take off. |
| 16:37.33 | Samot | Peer to peer is perspective. |
| 16:37.34 | igcewieling | Samot: My point is that MGCP, in my opinion, was better designed and should have won. |
| 16:37.55 | Samot | None wanted all that came with it. |
| 16:37.56 | igcewieling | Samot: peer to peer requires intellegent endpoints. |
| 16:38.16 | Samot | My Poly to Asterisk is peer to peer |
| 16:38.27 | igcewieling | and both endpoints are intellegent |
| 16:38.37 | Samot | My poly to kamailio is peer to peer |
| 16:38.41 | Samot | Correct |
| 16:38.52 | Samot | Kamailio is not a PBX |
| 16:39.08 | igcewieling | As opposed to MGCP to Asteirsk where only Asterisk (server) is intellegent. The phone is as close to "dumb as a rock" as you can get still do RTP./ |
| 16:40.05 | Samot | So then an ATA |
| 16:40.27 | igcewieling | MGCP seemed to me to be a logical evolution from analog PSTN to IP. SIP seemed to be an attempt to make a usable variant of H323. |
| 16:40.39 | igcewieling | An ATA is still intellegent. |
| 16:43.09 | Samot | Well did you ever setup MGCP? |
| 16:43.45 | Samot | I did |
| 16:43.47 | igcewieling | Samot: back in 2002 and 2003 |
| 16:44.08 | Samot | Out of all three, SIP was both the most simplistic and feature rich |
| 16:45.46 | Samot | A part of me believes that if MGCP had one, POTS would still be around more. |
| 16:45.56 | Samot | Won* |
| 16:46.14 | file | MGCP sorta lives on in the end |
| 16:46.33 | Samot | Well its kinda like IPv6 |
| 16:47.19 | Samot | I have seen a few ISPs bitch about how much a PITA it is to deploy at their level |
| 16:47.46 | Samot | End user side, easy peasy |
| 16:47.57 | Samot | Their side, nightmare. |
| 16:48.06 | Samot | MGCP was the same way |
| 16:48.28 | file | PacketCable! that's what I was trying to remember |
| 16:58.01 | seanbright | we need an IAX3 |
| 16:58.11 | seanbright | or we can just skip right to 10 and call it IAXX |
| 16:58.23 | seanbright | or skip to 20... |
| 16:58.28 | seanbright | you see where this is going |
| 16:58.44 | file | seanbright: IAX tunneling protocol which encapsulates SIP! |
| 16:59.00 | seanbright | RTP media trunking |
| 16:59.04 | seanbright | that is all i need |
| 16:59.26 | seanbright | i don't even need it, but it would be cool. |
| 17:00.23 | seanbright | we can create an RTP extension. file you can write the RFC. |
| 17:00.31 | drmessano | Need IAX that integrates directly with neurons in the brain. Human Asterisk eXchange, call it HAX |
| 17:00.32 | seanbright | it's a perfect plan |
| 17:00.42 | file | request for cookies? |
| 17:01.08 | file | seanbright: 4170 btw |
| 17:01.26 | seanbright | file: my blood sugar? i know |
| 17:01.45 | seanbright | of course cisco would do it |
| 17:02.04 | seanbright | died on the vine |
| 17:13.53 | drmessano | It's just sad how Asterisk has fallen behind. No Snapchat integration, no "Stories", no "reacts" or comments for calls, no retweeting them. |
| 17:14.11 | drmessano | Can't even put a call in my Top 8 |
| 17:15.37 | drmessano | Maybe I need to implement "Likes" in dialplan. After a call hangs up, "If you wish to Like this call, Press 1" |
| 17:15.49 | drmessano | "To angry react to this call, press 2" |
| 17:15.51 | drmessano | Etc |
| 17:18.37 | Samot | "To ratio this call, press 7" |
| 17:34.29 | igcewieling | Talk about zero useful content: "Our flagship platform, Tesira is the world's only integrated, networked audio and video processing and distribution platform. Sophisticated, adaptive DSP is the core of what Biamp does and what Tesira delivers. ... With a single networked platform, Tesira truly is enterprise-wide media made simple." |
| 17:34.57 | igcewieling | I guess they have never heard of YoutTube. |
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| 17:42.05 | Samot | That's more something that drmessano would use. |
| 17:42.13 | Samot | Not really what YouTube does. |
| 17:44.54 | Samot | I mean YouTube could be involved if they were streaming live to it but that software is what they would use for the AV of stream to get it to YouTube. |
| 17:48.57 | igcewieling | They want to know how to connect it to their (FreePBX) PBX. Not quite so simple as that. |
| 17:50.46 | igcewieling | We could put them on the voice VLAN, but if the device needs to talk to an outside service over the internet, they will be disappointed. The voice vlan isn't allowed to access the Internet. The alternative would be to put the device on the customer LAN, then configure the device to use the external IP of the PBX. But then we have to rely on the customer firewall to not to screw it up. |
| 17:51.05 | Samot | What are they using it for? |
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| 17:53.17 | igcewieling | no idea. They are just asking for ip and login info to "integrate with the PBX" I told them the options and asked for more info. |
| 17:54.57 | igcewieling | they sent a config sheet for this device: https://www.biamp.com/products/tesira-configurable-audio-dsp#tesira-svc-2 |
| 18:04.17 | Samot | drmessano might have more insight but that device looks like something a radio station would use to mix in callers to the feed. |
| 18:05.56 | Samot | Based on what I'm seeing it supports SIP/FXS so yeah. |
| 18:06.09 | igcewieling | The customer is a "foodservice company distributing fresh produce, meats, & seafood along with a full line of food products", |
| 18:06.31 | drmessano | That's interesting |
| 18:06.55 | drmessano | Dante and CobraNet are a couple low-end AoIP protocols |
| 18:07.37 | Samot | So I'm close to my assessment on what it would be used for? |
| 18:07.50 | drmessano | Pretty much, yes |
| 18:08.12 | drmessano | Well |
| 18:08.43 | drmessano | Let me put it this way, i've never seen this products sold for Broadcast, so they may be used in audio production houses |
| 18:08.46 | igcewieling | "Thanks for calling into the Bobs Foods Show. Today we are going to talk about our meat." |
| 18:09.05 | Samot | Basically. |
| 18:09.18 | Samot | The SIP/FXS interfaces are the PSTN connection. |
| 18:09.31 | Samot | So this will probably need a trunk/peer off the PBX |
| 18:09.53 | Samot | drmessano: Could they be using this as their paging system? |
| 18:10.13 | Samot | If this is a warehouse setting. |
| 18:11.52 | Samot | Because that does seem like an option to use this for. |
| 18:13.59 | drmessano | That is beyond overkill, but I suppose |
| 18:16.07 | igcewieling | Any people using Adtran getting bogus LLDP speed/duples mismatch errors? I just found "no lldp receive 802.3-info mac-phy-config" to fix it. |
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| 20:15.47 | igcewieling | ha! from an article about AT&T: https://cdn.arstechnica.net/wp-content/uploads/2021/03/getty-snail-cable-800x600.jpg |
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