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| 08:17.54 | _ganapathi_ | am using AGI on my IVR. suddenly am getting error as 510 invalid command for every command to asterisk. |
| 08:18.04 | _ganapathi_ | any idea to traceback and solve this. |
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| 11:50.24 | velix | Is there a stable plugin to link the user table of sip.conf or pjsip.conf with a database? I think, postgreSQL and MySQL are compatible? |
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| 12:09.16 | Samot | What user table? |
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| 13:47.19 | Zerg | Hello all. How do I set up asterisk to register towards a trunk that has a non fqdn domain? Currently I have a registry string user:password@domain.com but asterisk throws me errors that it could not resolve it |
| 13:48.35 | Zerg | I wish to push registraation towards lets say 192.168.1.10 with the domain domain.com |
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| 13:53.46 | [TK]D-Fender | Go make a host entry on your server for it then. |
| 13:54.44 | velix | Samot: I mean, can I read the username & password for the devices (peers) from a database? |
| 13:55.15 | velix | [device](!,preset) <-- is the preset applied after settings in [devices]? |
| 13:58.09 | [TK]D-Fender | velix, yes your configs can be DB based. Go read the WIKI on this |
| 13:58.34 | velix | [TK]D-Fender: I did, I just didn't know if it's up to date. |
| 13:58.45 | velix | Maaany information in the wiki are for 1.2 |
| 13:58.54 | velix | canreinvite= etc. are outdated. |
| 13:58.58 | [TK]D-Fender | velix, those templates only apply to fixed file configs, and are overridden by anything in the [device] section. |
| 13:59.08 | [TK]D-Fender | <velix> Maaany information in the wiki are for 1.2 <- the OFFICIAL one |
| 13:59.24 | velix | [TK]D-Fender: Oh, I thought 1.4/1.6 ? |
| 13:59.27 | [TK]D-Fender | .... |
| 13:59.32 | Samot | There is no backend database. |
| 13:59.35 | [TK]D-Fender | no, not the voip-info WIKI |
| 13:59.39 | Samot | Not for static configs. |
| 13:59.45 | [TK]D-Fender | the OFFICIAL Asterisk wiki |
| 13:59.55 | velix | [TK]D-Fender: I see. |
| 14:00.03 | Zerg | [TK]D-Fender: Ok, I did that. But it is also a non standard port, even though I have port=5066 in my trunk configuration the registry is still sending it towards 5060 |
| 14:00.05 | velix | [TK]D-Fender: Ah okay, I thought the order is important (!,preset) vs. (preset,!) |
| 14:00.06 | Samot | You use use SQL backends for Real-Time. |
| 14:00.18 | velix | Samot: Ahh yeah, that's what I've read. |
| 14:00.28 | [TK]D-Fender | Zerg, Show your actual configs |
| 14:00.28 | velix | Right now, I'm adding stuff to sip.conf by script and do a reaload. |
| 14:00.36 | Samot | OK. |
| 14:00.37 | velix | It works, but... #ugly. |
| 14:00.57 | igcewieling | That is how FreePBX does it. |
| 14:01.22 | velix | We have processed 520 calls in 3 days. |
| 14:01.28 | velix | with 8 students |
| 14:01.30 | Samot | Zerg: Your trunk config doesn't mean anything in Chan_SIP for registers. |
| 14:01.36 | velix | On a $2.5 vps :) |
| 14:01.44 | velix | Asterisk is incredible. |
| 14:01.45 | Samot | Zerg: That is what the register string is for. |
| 14:04.51 | Zerg | [TK]D-Fender: https://pastebin.com/raw/a4LscBqN |
| 14:05.13 | [TK]D-Fender | Zerg, Samot has already said it. |
| 14:05.22 | [TK]D-Fender | Zerg, that peer config has no impact on your register |
| 14:05.35 | [TK]D-Fender | Zerg, You want that sent to another port? Sepcify it in the register |
| 14:05.49 | [TK]D-Fender | specify* |
| 14:06.39 | Zerg | [TK]D-Fender Samot thanks, that did it. Changing my hosts file to make it resolveable and setting port in the registry |
| 14:10.12 | [TK]D-Fender | Zerg, glad to hear |
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| 15:07.04 | velix | In chan_sip you cannot restart a single trunk, can you? |
| 15:11.44 | [TK]D-Fender | no |
| 15:11.59 | [TK]D-Fender | What do you actually even mean by "restart"? |
| 15:14.48 | igcewieling | Now that hanguphanders have existed for a long time, is the 'h' extension eventually going away? |
| 15:14.54 | velix | [TK]D-Fender: One of my peers sometimes get a hickup because it's configured wrong. |
| 15:15.14 | velix | [TK]D-Fender: sometimes, a "sip reload" fixes it, but sometimes, it's completly gone away. |
| 15:15.29 | velix | [TK]D-Fender: Since I've got multpiple peers, I cannot restart the whole core. |
| 15:15.42 | velix | Then the calls on the other peers will get cancaled. |
| 15:15.43 | igcewieling | velix: if you are constantly rewriting sip.conf in production then you have a problem. |
| 15:15.54 | [TK]D-Fender | velix, a sip reload doesnt' trash calls in progress |
| 15:16.00 | igcewieling | velix: no, reloading chan sip won't kill active calls. |
| 15:16.09 | velix | [TK]D-Fender: I'm not talking about reload, this works. |
| 15:16.18 | [TK]D-Fender | Where's the actual problem? |
| 15:16.18 | velix | But I need a restart in this case. |
| 15:16.22 | [TK]D-Fender | fix your peer. reload |
| 15:16.27 | velix | It's not my peer. |
| 15:16.27 | [TK]D-Fender | Why a full restart? |
| 15:16.31 | [TK]D-Fender | huh? |
| 15:16.35 | velix | [TK]D-Fender: That fixed the problem with the peer. |
| 15:16.42 | velix | Uh wait, it's my peer. I mean the other side :D |
| 15:16.53 | velix | Not my trunk, where I'm connecting to. It's broken. |
| 15:16.57 | velix | igcewieling: Kobaz and I tried intesively to go to pjsip without success. |
| 15:17.08 | velix | Innovaphone behind FortiNetfirewall. |
| 15:17.21 | file | igcewieling: I doubt "h" will go away |
| 15:17.25 | igcewieling | I've been running pjsip for at least a year, not on FreePBX though. |
| 15:18.14 | [TK]D-Fender | velix, I don't understand what you're asking for at this point |
| 15:19.12 | velix | Okay, let me explain it, 1 sec. |
| 15:20.06 | velix | Our university gave me a SIP account for their PBX. I've integrated it as a peer via chan_sip. It sometimes (really sometimes) goes away and shows error 403 forbidden without a reason. |
| 15:20.12 | velix | The admin also doesn't know. |
| 15:20.25 | velix | Sometimes, a "sip reload" fixes this. Sometimes it does not. |
| 15:20.45 | velix | The other peer on my system works perfectly stable. |
| 15:20.55 | velix | (I've got an external provider as a backup) |
| 15:20.56 | igcewieling | file: I think I see the issue, it is actually AEL which is generating the h extensions. Executing [h@sm_dial:9991] Set("PJSIP/level3-00000011", "~~parentcxt~~=switch_manager") in new stack |
| 15:21.15 | velix | [TK]D-Fender: I've done TCPdump, Kobaz consulted me. |
| 15:21.24 | velix | It's not on my side. No nat, no firewall, no filterings. |
| 15:21.27 | igcewieling | A 403 is normal at the start of a call. |
| 15:21.46 | Samot | No it is not. |
| 15:21.49 | Samot | A 401 is. |
| 15:21.56 | igcewieling | I sit corrected. |
| 15:22.06 | velix | Yeah, I'm getting 401 all the time, but suddenly 403. And then it's over. |
| 15:22.18 | velix | register_retry_403=yes doesn't help |
| 15:22.20 | Samot | So let me guess... |
| 15:22.35 | Samot | If you need more voice channels they give you a new account? |
| 15:22.50 | velix | No, there are 15 channels on this account. |
| 15:22.59 | Samot | So why do you need two? |
| 15:23.23 | velix | In the beginning he gave me four. But he configured one to have multiple channels on it now. |
| 15:23.38 | velix | But that's also not the problem. The problem also occures when there are no calls it all. |
| 15:23.48 | velix | In idle mode, it sometimes (!) jumps from 401 to 403. |
| 15:23.59 | Samot | Because they are rejecting the registers. |
| 15:24.01 | velix | And then on the next morning, no calls are possible until core restart. |
| 15:24.04 | velix | Samot: yep. |
| 15:24.09 | velix | A restart fixes this. |
| 15:24.12 | Samot | So that is on them. |
| 15:24.18 | velix | Yep. I think FortiNet. |
| 15:24.32 | Samot | How is it FortiNet? |
| 15:24.46 | velix | Samot: it's their outgoing content firewall |
| 15:24.54 | Samot | That doesn't mean anything. |
| 15:24.58 | velix | He's using a SIP preset because of all the attackers. |
| 15:25.03 | Samot | Their PBX is sending the 403. |
| 15:25.18 | velix | Samot: Fortinet is a package firewall. It goes into streams. It also can give 403 |
| 15:25.28 | velix | content firewall* |
| 15:25.36 | Samot | That's for HTTP |
| 15:25.40 | Samot | Not SIP |
| 15:25.44 | velix | Samot: naaah, it does anything. |
| 15:25.49 | Samot | OK man. |
| 15:25.59 | velix | Samot: but yeah, it's not on my side :) |
| 15:26.42 | velix | https://help.fortinet.com/fos50hlp/54/Content/FortiOS/fortigate-voip-guide-52/sessionhelper-disenable.htm |
| 15:26.43 | velix | See? |
| 15:26.51 | velix | It even has SIP addons ... very annoying. |
| 15:27.12 | velix | It can modify headers, rewrite stuff etc. |
| 15:28.03 | velix | okay, then I need to tell my students to not start new calls. I'll reboot it then. |
| 15:28.07 | velix | restart ;) |
| 15:31.21 | velix | I'm sorry if I offended you. But the problem is definitely not on my side. The connection with the professional drink works great and is stable. |
| 15:34.37 | velix | My config isn't anything special: https://bpaste.net/B5OQ |
| 15:35.10 | velix | whoops: https://bpaste.net/ON4Q |
| 15:35.31 | velix | damn: https://bpaste.net/EJTA |
| 15:35.36 | igcewieling | Nobody is going to help you try and fix a problem with a remote server not under your control. |
| 15:35.50 | velix | Yeah, I didn't ask for help fixing this :) |
| 15:36.02 | velix | I just asked if I can restart ONE special peer. |
| 15:36.27 | joepublic | many words, little content, high noise to signal ratio, glad you are enthusiastic however--genuinely--and wish you all success. |
| 15:36.31 | igcewieling | you can't. |
| 15:37.57 | velix | But what makes me wonder: why a restart of the core works. Does asterisk send an "unregister" or anything? |
| 15:38.39 | igcewieling | a restart stops Asterisk, then starts it. Generally Asterisk will unregister when it stops. |
| 15:41.15 | velix | Okay, all students have stopped talk, let's do a restart |
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| 16:18.52 | velix | Yep, the PBX has a hickup. |
| 16:18.58 | velix | Some calls go through, some don't. |
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| 16:27.04 | velix | Ah, when looking at the logs, "Locally RTP bridged" doesn't get established on failed calls. |
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| 17:04.20 | Kobaz | Samot: i've been migrating to SIP TLS for this very reason |
| 17:04.24 | Kobaz | fscking firewalls |
| 17:04.31 | Kobaz | stop modifying my traffic! |
| 17:05.01 | Samot | Uhm. |
| 17:05.03 | Samot | OK. |
| 17:05.49 | Kobaz | velix: correct, there is no RTP on a call that never completed |
| 17:08.32 | velix | Hehe, just talked with the admin "we didn't have problems today". blahhh |
| 17:08.42 | velix | It works perfectly over my backup line. |
| 17:08.54 | velix | The conference rooms in Asterisk are damn nice. |
| 17:08.55 | *** part/#asterisk MLC (~MLC@63.249.40.11) |
| 17:09.09 | velix | I can press "5" to record important conferences and send it to the other participants. |
| 17:09.12 | velix | \o/ |
| 17:09.50 | velix | I've also set an IPfilter to allow German IPs only. Intruders dropped to less than 1% |
| 17:11.43 | *** join/#asterisk ruied (~ruied@PARSIS.PT) |
| 17:11.48 | ruied | [2020-03-27 17:11:04] WARNING[27516]: res_pjsip_pubsub.c:777 subscription_get_handler_from_rdata: No registered subscribe handler for event call-info from 50 |
| 17:14.23 | ruied | ups... I have a polycom5000 when registered it reports the warning no registered "subscribe handler for event call-info from 50". is this something that the phone is sending to asterisk? |
| 17:14.52 | Kobaz | ruied: you can easily tell if you dump the sip traffic |
| 17:15.27 | Kobaz | but typically, if something is subscribing to something that doesn't exist, you will get some sort of warning |
| 17:15.38 | Kobaz | asterisk doesn't do much on its own without being told otherwise |
| 17:17.39 | ruied | the phone is a polycom5000, and not much info about the config files (loading from tftp)... it is working but trying to figure it out what might be this warning... :\ |
| 17:18.21 | Kobaz | [2020-03-27 13:14:52] <Kobaz> ruied: you can easily tell if you dump the sip traffic |
| 17:18.43 | file | it means that the endpoint "50" sent a SUBSCRIBE request using the call-info event, that is not supported by Asterisk so a warning was output |
| 17:19.49 | ruied | ok, thanks, going to take a look at sip traffic... |
| 17:20.11 | file | according to Polycom, that is involved with "Shared Call Appearances" |
| 17:20.45 | Kobaz | mmm SLA |
| 17:21.15 | Kobaz | i wrote this whole fancy dancy sla solution in dialplan/ami/agi etc, but it would be nice to have native support at some point |
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| 20:34.10 | TheFatherMind | Hello? Hello?? Hello???? can you hear me??? |
| 20:34.14 | TheFatherMind | stupid phone system does not work |
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| 20:35.09 | file | blinks |
| 20:37.09 | Samot | hahahahahaha |
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| 21:56.09 | Slade | whoa. that was kinda meta |
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| 23:51.53 | thansen | I've got a situation where externip isn't getting set on the sip traffic (trying to migrate to a new server from an old one) |
| 23:53.41 | thansen | The general setup with how things are NAT'd etc is exactly the same and I've copied the config file essentially verbatim minus tweaking IPs etc as appropriate...the version has been upgraded from 11.17.1 to 11.25.2...anyone got any tips to share to figure out why it's not setting the value in sip traffic as expected? |