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| 00:58.51 | rpifan | asterisk 16 is now avaiable as a package |
| 00:58.52 | rpifan | interesting |
| 01:11.04 | Samot | From where? |
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| 01:29.49 | rpifan | is chan sip gonna be deprecated? |
| 01:29.51 | rpifan | debian apt |
| 01:30.59 | Samot | Chan_SIP *is* deprecated. |
| 01:33.00 | rpifan | ah |
| 01:33.08 | rpifan | so i should be making my configs in pjsip? |
| 01:33.17 | rpifan | last time i used asterisk they were still just getting into pjsip |
| 01:35.22 | Samot | Well Chan_SIP has to be configured to be installed during compile time as of Asterisk 17. |
| 01:35.42 | Samot | It can officially be removed from Asterisk as of 2023 or so. |
| 01:36.45 | rpifan | ah |
| 01:36.46 | rpifan | i got time |
| 01:41.46 | rpifan | so i got two asterisk boxes i need ot connect |
| 01:41.56 | rpifan | one is on a dedicated vps the other is in my house with a dynamic ip |
| 01:42.00 | rpifan | is there a way to get this to work |
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| 02:56.41 | drmessano | You shouldn't be using chan_sip at this point, at all |
| 03:09.37 | rpifan | hm |
| 03:09.41 | rpifan | well i was playing with iax2 |
| 03:09.50 | rpifan | with one machine with dynamic ip and the other fixed |
| 03:09.54 | rpifan | i got something from a website |
| 03:10.15 | rpifan | and it shows as connected on the dynamic side but in the fixed ip it shows 0 online 0 offline 1 unmonitored |
| 03:19.29 | drmessano | iax2 is even worse |
| 03:19.36 | drmessano | Just use chan_pjsip |
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| 14:14.36 | igcewieling | Nah, use MGCP or SKINNY if you really want to use a dead protocol. |
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| 17:00.03 | spagheti | is there a way I can set a variable (via SetVar) as the output of a shell (bash) command via AMI? I've been looking around for it but nothing I've found is working when I test through AMI |
| 17:01.42 | igcewieling | spagheti: "core show functions like SHELL" |
| 17:02.30 | jsmith | spagheti: Also, pay attention to the documentation for that dialplan function. It will only work via AMI if you set a flag in asterisk.conf. |
| 17:02.52 | spagheti | thanks igcewieling and jsmith |
| 17:06.16 | igcewieling | "core show functions" and "core show applications" is your friend. |
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| 17:32.48 | jsmith | spagheti: You're welcome! |
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| 19:14.18 | rpifan | drmessano, but my machine is behind a firewall |
| 19:14.23 | rpifan | and i tried to open ports and such and it was a mess |
| 19:14.33 | rpifan | so the idea is to use iax2 to bind two boxes |
| 19:14.39 | rpifan | and then plug my soft phone into my local box |
| 19:16.34 | igcewieling | rpifan: SIP works just fine with NAT |
| 19:17.24 | *** join/#asterisk Janos (~textual@201.204.94.76) |
| 19:24.33 | seanbright | eh... "just fine" is subjective |
| 19:24.43 | seanbright | you have to know what you're doing |
| 19:24.49 | seanbright | at least to some degree |
| 19:26.02 | rpifan | igcewieling, it didnt that was the whole issue |
| 19:26.11 | rpifan | unless pjsip is better then old sip |
| 19:26.30 | seanbright | it is, but that probably isn't why it didn't work |
| 19:28.46 | rpifan | oh? |
| 19:40.42 | seanbright | 'and i tried to open ports and such and it was a mess' |
| 19:41.00 | seanbright | chan_sip nat setup is fairly straight forward (externaddr and localnet primarily) |
| 19:41.18 | seanbright | assuming those were correct, that leaves firewall config |
| 19:45.20 | drmessano | IAX2 hasn't been a good excuse for firewall transversal in maybe 15 years |
| 19:45.42 | seanbright | IAX3 is going to be dope |
| 19:46.08 | drmessano | It's going to auto-create IPv6 tunnels? |
| 19:46.59 | seanbright | that's page 1 of the RFC |
| 19:47.11 | drmessano | If someone wanted to really piss off the "Just use SIP" crowd, something really sexy for firewall transversal in IAX would be trololol for the next 15 years |
| 19:47.35 | drmessano | "I heard IAX3 creates magic IPv6 tunnels, so I used it" |
| 19:47.53 | drmessano | I can feel Samot's aorta tearing, slowly |
| 19:48.26 | drmessano | No no no |
| 19:48.41 | drmessano | IAX3 needs to incorporate WireGuard |
| 19:48.53 | drmessano | Done.. I just murdered Samot |
| 19:49.01 | drmessano | RIP ThunderBuddy |
| 19:49.06 | *** join/#asterisk eXistenZ (~pectic@bzq-109-67-186-167.red.bezeqint.net) |
| 19:49.42 | Samot | You are a mean person. |
| 19:49.59 | Samot | file: Help me. I need a safe place from the bad man. |
| 19:50.16 | seanbright | git checkout -b iax3 |
| 19:50.20 | seanbright | shit, wrong window, sorry |
| 19:50.49 | drmessano | app_iax3_wgproxy |
| 19:51.34 | drmessano | iax3.conf > wireguard=veryyes |
| 19:51.37 | Samot | Why don't you just embed it in pfsense? |
| 19:51.48 | drmessano | Now who is being mean |
| 19:52.10 | Samot | Oh it's all fun and games until pfsense is involved, huh? |
| 19:52.16 | drmessano | I heard you're a big fan of PFSense |
| 19:52.23 | drmessano | See what I did there? huh? huh? |
| 19:52.39 | Samot | No, you heard wrong. |
| 19:52.46 | Samot | It's you need a big fan for pfsense. |
| 19:53.53 | Samot | We all know pfsense users follow the open box deployment. |
| 19:54.14 | Samot | You know, where they don't cover the box because the fan needs to blow on the system for cooling. |
| 19:54.34 | drmessano | PFSense is very efficient when you overclock it |
| 19:55.12 | Samot | Or as Stefon would say "It's this thing where a midget sits on the server and blows into it to keep it cool" |
| 19:56.36 | drmessano | hahaha |
| 19:57.34 | drmessano | "New Yorks hottest new club is called PFSense" |
| 19:59.47 | drmessano | "All of their servers are sweaty, overheating midgets, and there's giant fans blowing down from above. The servers drag old UPSes around by jumper cables clamped to their heels, and they savagely attack anyone who uses the word SIP" |
| 19:59.51 | drmessano | Hows that? |
| 20:00.39 | Samot | "A an group of furries cosplaying as The IT Crowd" |
| 20:00.44 | Samot | And* |
| 20:01.40 | drmessano | "They only serve one drink, and it's milk mixes with the tears of admins" |
| 20:01.45 | drmessano | mixed* |
| 20:23.32 | igcewieling | I fsck'ing hate salespeople. |
| 20:27.23 | igcewieling | The "public static IP address", which we require for service, provided was 192.168.x.x |
| 20:38.57 | rpifan | is there really something wrong with iax2? |
| 20:39.03 | rpifan | that i should use pjsip for instead? |
| 20:40.15 | igcewieling | rpifan: Using IAX2 is sort of like using an 8-track tape. Sure, 8-track tapes are OK, but since nobody makes them anymore, nobody makes equipment to use them, and nobody can help you with them, you'll have trouble getting help. |
| 20:42.29 | rpifan | really? i thought it was the standard to connect two asteriks boxen |
| 20:43.41 | igcewieling | rpifan: only if you read old information or maybe browse voip-info |
| 20:46.08 | rpifan | hm |
| 20:48.16 | seanbright | rpifan: iax2 is no longer maintained |
| 20:48.48 | seanbright | the mose recent change to chan_iax2.c was 11 months ago... by me |
| 20:49.07 | seanbright | and it didn't make it suck less |
| 20:49.08 | seanbright | heh |
| 20:50.01 | rpifan | seanbright, oh really? is there gonna be an iax3 then |
| 20:50.09 | seanbright | nope |
| 20:50.17 | seanbright | i mean, it's a community project, so i guess there could be |
| 20:50.46 | seanbright | but sangoma resources are focused on chan_pjsip |
| 20:51.04 | seanbright | and i don't know if any community developers that are interested |
| 20:58.06 | rpifan | is ther anything that iax can do better then sip then? |
| 20:58.43 | seanbright | trunking is awesome |
| 20:58.58 | seanbright | single port makes network configuration a bit easier |
| 20:59.00 | seanbright | that's probably it |
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| 22:27.43 | *** join/#asterisk bbhoss (sid18216@gateway/web/irccloud.com/x-pnyrdxevuhizhxtb) |
| 22:27.56 | bbhoss | So, who's got the best ip phones these days? |
| 22:35.49 | rpifan | hm so why is it my sip connection to call centric just works but the one ot my asterisk box doesnt |
| 22:35.50 | *** join/#asterisk verzo (~verzo@gateway/tor-sasl/verzo) |
| 22:36.01 | rpifan | the softphone works just fine on wifi to my call centric account |
| 22:36.07 | rpifan | what am i doing wrong seanbright |
| 22:36.49 | *** join/#asterisk drathir_tor (~drathir@gateway/tor-sasl/drathir) |
| 22:42.10 | seanbright | unpacks his crystal ball... |
| 22:42.28 | seanbright | shrugs |
| 22:42.32 | seanbright | have a good weekend |
| 22:53.54 | rpifan | hm |
| 22:53.57 | rpifan | how do i debug then |
| 22:57.24 | igcewieling | Polycom has had the best SIP phones since at least 2003 |
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| 23:01.49 | rpifan | sip requires ports of course but how can call centric work without me punching them open |
| 23:05.18 | igcewieling | rpifan: the outbound RTP will automatically open up the NAT ports |
| 23:05.41 | igcewieling | So port 5060/udp is the only thing you would normally need to port forward. |
| 23:06.00 | igcewieling | the inbound RTP will flow over those opened ports. |
| 23:06.21 | rpifan | but with my softphone and my call centric sip account ive not opened any ports its just worked automatically, then again maybe smoething was opened by default |
| 23:06.27 | rpifan | but asterisk is sdoing something else? |
| 23:06.58 | igcewieling | rpifan: You only need to forward port 5060 if you are registering TO a server behind NAT. |
| 23:07.15 | igcewieling | and that port forwarding would have to be done on the device which handles NAT for the Asterisk server. |
| 23:07.35 | igcewieling | No special config is needed to run a SIP client behind NAT. |
| 23:08.17 | igcewieling | The outbound registration will open port 5060 for replies and incoming calls. |
| 23:08.20 | rpifan | hm |
| 23:08.25 | rpifan | well the asterisk box is on a vpn |
| 23:08.29 | rpifan | with open everything |
| 23:08.43 | igcewieling | now you've just changed from simple to very complicated. |
| 23:10.05 | *** join/#asterisk mducharme (uid303982@gateway/web/irccloud.com/x-vlmruhutguklbsac) |
| 23:10.08 | rpifan | how so |
| 23:21.19 | *** join/#asterisk verzo (~verzo@gateway/tor-sasl/verzo) |
| 23:26.18 | rpifan | oh the other thing is that it works fine on LTE but not on wifi |
| 23:40.18 | Janos | I have a funny behavior with some realtime queue_members, I have the following agent in a queue: 'Jane Doe <4353> (Local/4353@agents from PJSIP/7675) (ringinuse enabled) (realtime) (Unavailable) has taken 60 calls (last was 296 secs ago)'. This agent is marked as Unavailable yet the system is still sending it calls, any advice on how to prevent this from happening ? |
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