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| 01:18.55 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.19.0 (2018/01/11), Standard: 15.2.0 (2018/01/11); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 01:49.27 | Samot | degenerate: That's not a full call log of the call. Can't tell what happened. |
| 02:04.29 | degenerate | Samot yeah, i'm tarded. i realized that afterwards. thanks for looking at that log though. |
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| 03:15.31 | Turaiel | Any OBi users that might be able to interpret a problem I'm having? |
| 03:32.20 | Turaiel | In case anyone sees this later, I've got the logs from my OBi110 here: https://pastebin.com/9g16xzpU. My issue is that once it answers a call from my apartment's intercom system (on FXO), the line goes silent but stays connected. No audio in either direction except for a short blip at the beginning of the call. |
| 04:04.06 | Samot | a=ice-ufrag:2562fa1479186ac3558025e9305558cb |
| 04:04.07 | Samot | a=ice-pwd:709c382a0f112c3e659ca4743de578c0 |
| 04:04.07 | Samot | a=candidate:Hc0a8011a 1 UDP 2130706431 192.168.1.26 14614 typ host |
| 04:04.07 | Samot | a=candidate:Hc0a8011a 2 UDP 2130706430 192.168.1.26 14615 typ host |
| 04:04.29 | Samot | Why do you have ICE enabled on the Asterisk peer? |
| 04:26.53 | LiuYan | One of my asterisk servers seems doesn't accept incoming socket anymore after several days. The output of `nmap -T4 -sU -p5060 <my-asterisk-ip-address>` show the socket is closed, but asterisk server is running. |
| 04:28.40 | LiuYan | Restart asterisk service (systemctl restart asterisk) or issue `sip reload` in asterisk console will fixed this issue, but what could the reason & how to avoid it? |
| 04:29.50 | LiuYan | Currently I'm running asterisk 13.18.5~dfsg-1 in debian unstable. |
| 04:34.05 | Samot | Is the Asterisk server behind NAT? |
| 04:38.25 | LiuYan | Samot: Yes, |
| 04:38.38 | Samot | And you're scanning your WAN IP? |
| 04:38.43 | Samot | The external IP? |
| 04:39.03 | LiuYan | Samot: Yes, nmapping the external IP |
| 04:39.10 | Samot | Fix your NAT. |
| 04:39.37 | Samot | The reason the sip reload/restart works is because you force Asterisk to resend the REGISTER / outbound request. |
| 04:39.48 | Samot | Which is opening a fresh NAT hole in your router. |
| 04:40.22 | Samot | You can't get inbound requests because your NAT table is not staying updated. |
| 04:42.59 | LiuYan | Samot: So you mean it could caused by the NAT, I mean the router devie? |
| 04:46.19 | Samot | It's "could" it "is" NAT |
| 04:46.26 | Samot | And yes, that would be your router/firewall |
| 04:47.12 | LiuYan | Samot: I was told it is a entry level (cheap) router :( |
| 04:47.33 | Samot | K. |
| 04:47.42 | Samot | It's not keeping the NAT open. |
| 04:53.18 | LiuYan | Samot: Thanks, I'll tell the other side to replace the router. By the way, the router model is TL-WVR1300G: http://www.tp-link.com.cn/product_509.htmlhttp://www.tp-link.com.cn/product_509.html <-- You're guity, router :D |
| 05:01.14 | drmessano | Yeah |
| 05:01.20 | drmessano | That'll do it |
| 05:01.24 | drmessano | Toilet Paper Link |
| 05:01.37 | LiuYan | lol |
| 05:03.10 | drmessano | Im at a point now where I wont even service a SIP connection unless there's a Mikrotik on it |
| 05:03.21 | drmessano | Anything else is questionable |
| 05:09.11 | LiuYan | The other side is a small office, they don't even have a firewall device. I can see lots of crack(not sure what word should I use) connection in asterisk console. |
| 05:10.59 | drmessano | They dont? All their devices are on public IPs? |
| 05:13.34 | LiuYan | Devices are not on public IPs: {internet} -- [Router] -- [Terminals (include asterisk server)] |
| 05:14.21 | drmessano | Ok, the other side of the router |
| 05:14.29 | LiuYan | Part of firewall function is inside of the router, but it's not a dedicated firewall device |
| 05:14.46 | drmessano | So where is the TP Link? |
| 05:15.17 | LiuYan | [Router] is the TP Link |
| 05:15.53 | drmessano | You speak as though you're referring to multiple devices and multiple locations |
| 05:15.57 | drmessano | Its confusing as hell |
| 05:16.08 | drmessano | How is the TP Link not a dedicated firewall device? |
| 05:16.18 | Samot | Well |
| 05:16.23 | Samot | It's a TP-Link? |
| 05:17.12 | drmessano | 00:14:29 <LiuYan> Part of firewall function is inside of the router, but it's not a dedicated firewall device |
| 05:17.20 | drmessano | ^ Clarify |
| 05:17.27 | LiuYan | drmessano: Sorry for my poor English. I mean, it's not a "professiinal" firewall device |
| 05:17.34 | drmessano | Sigh |
| 05:17.58 | drmessano | Well, in this context, it IS your firewall |
| 05:18.03 | drmessano | It's the only firewall |
| 05:18.06 | drmessano | and it's the problem |
| 05:19.13 | drmessano | 00:09:11 <LiuYan> The other side is a small office, they don't even have a firewall device. <- confusing |
| 05:21.31 | LiuYan | drmessano: Sorry about that. That's my lack of knowledge on hardwre, I thought firewall is another device (not the router itself). |
| 05:22.22 | drmessano | Something HAS to be providing a NAT/Firewall.. You kinda have to have it |
| 05:22.33 | drmessano | So yes, it's a firewall |
| 05:22.41 | drmessano | and a NAT device |
| 05:23.00 | drmessano | Just a really shitty one |
| 05:25.53 | LiuYan | Copy that. I'll tell them to replace the router, or bypass the router. |
| 05:26.59 | drmessano | They cant |
| 05:27.13 | drmessano | They literally need the damn router for NAT |
| 05:27.24 | LiuYan | drmessano: I mean setup another line from ISP |
| 05:31.03 | LiuYan | drmessano: So the asterisk server is on another public IP, no router upside. |
| 05:32.25 | drmessano | Wouldnt it be a lot cheaper to just replace that shitty $20 router? |
| 05:45.21 | LiuYan | \ |
| 05:48.36 | LiuYan | drmessano: We are two friendly companies, they will pay for replacing router, while we will pay for another line, so I think they may choose the second option. |
| 05:49.44 | drmessano | LiuYan: Are the clients connecting to the PBX still going to be behind the TP Link? |
| 05:50.05 | drmessano | Clients <---> TP Link <--> Internet <----> PBX |
| 05:56.34 | LiuYan | drmessano: Which PBX do you mean here? Currently, I setup two asterisks in two cities, using SIP trunk to link two asterisk server, so that we(in city A) can make call through PSTN line in city B. |
| 05:57.12 | drmessano | LiuYan: Will there be ANY users with IP Phones behind the TP Link? |
| 05:57.17 | drmessano | Yes or no |
| 05:58.30 | LiuYan | drmessano: No. Currently we don't use IP phones. |
| 05:59.24 | drmessano | All analog huh |
| 06:01.41 | LiuYan | drmessano: City A: [Agents] --> [Traditional PBX] ----E1 Trunk---- [Asterisk 1] ----------{internet}---------- City B: [Router, the guity one] -- [Asterisk 2] ----E1 Trunk---- {PSTN} |
| 06:02.08 | drmessano | ok |
| 06:15.17 | [TK]D-Fender | So in City A; you have an * server with no LAN, and a WAN IP directly on it? |
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| 12:23.00 | zicada | Hey guys. We're having some random segfaults where the core lists this: |
| 12:23.07 | zicada | #0 0x00000000005eff20 in ast_say_date_with_format_ja (chan=0x0, time=0, ints=0x0, lang=0x284e5a0 "", format=0x0, timezone=0x0) at say.c:8737 |
| 12:23.36 | zicada | this is in 15.2.0 |
| 12:24.16 | zicada | We don't use say_date and we certainly don't use it in japanese, but this method is consistently mentioned in every core dump |
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| 12:47.18 | stefan27 | I never used ast 15 and I'm not that involved with asterisk source code, but I'm still curious what the rest of the coredump says? you got a pastebin? |
| 13:04.04 | zicada | it says surprisingly little even with a binary combiled with the debug compiler flags |
| 13:04.39 | zicada | https://pastebin.com/jpkyVS8u |
| 13:05.04 | zicada | made some changes to the server. It was using 'apport' so didn't seem to generate "real" cores. Waiting for another segfault now. Usually a few each day |
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| 15:21.28 | chl_ | hi, I keep getting Congestion on dials, is there any debug options you can enable? |
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| 15:27.00 | [TK]D-Fender | depends on the channel type |
| 15:27.07 | [TK]D-Fender | Should always be at verbose 10 |
| 15:27.17 | [TK]D-Fender | and then channel-debug for whatever type you're using |
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| 16:31.07 | jeffspeff | is there a way to display on the phone the extension a call is parked at? |
| 16:32.23 | [TK]D-Fender | Thre are Parking hints for BLF, or things like "XML scripting possible with some special phones |
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| 16:51.32 | jeffspeff | thanks [TK]D-Fender |
| 16:54.17 | jeffspeff | i'm also playing with BLF states with call pickup. I'm using a polycom vvx 411 phone. i have the line key subscribed to the state and i can answer/pickup calls that come in to that exten. when a call comes in, the phone displays something like 'call to: EXTEN from: ' where the "from" is blank. but after i answer/pickup the call the CID updates on my phone. how can i get the CID to show in the "from" section so i know if i want to answer that call or not? |
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| 17:06.00 | [TK]D-Fender | read the sample config. |
| 17:06.04 | [TK]D-Fender | there is a setting regarding this |
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| 17:11.10 | jeffspeff | sample config for sip.conf? |
| 17:11.32 | [TK]D-Fender | yes |
| 17:11.50 | jeffspeff | btw, i added notifycid=yes to sip.conf and that fixed the problem, but now the pickup tries to dial *97EXTEN whereas before it was just *97. |
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| 17:54.10 | jeffspeff | can you use patterns with hints like exten=_XXXX,hint,SIP/${EXTEN} ? |
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| 18:21.40 | jeffspeff | can you specify multiple values for subscribecontext? |
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| 18:37.31 | [TK]D-Fender | of course not |
| 18:37.47 | [TK]D-Fender | like any context it's always just 1 |
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| 18:59.49 | eromo | EHLO. I have sip trunk with my PSTN, when my internet fail my inbound and outbound calls audio stop working in one way(cannot hear us) and DTMF stops working. I have realize if I remove my default gateway happens the same. What can I do?. I am using asterisk 11.25 |
| 19:09.19 | igcewieling | eromo: run a local dns server and make sure the IP of EVERY interface is listed in /etc/hosts |
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| 20:53.20 | eromo | EHLO. I have sip trunk with my PSTN, when my internet fail my inbound and outbound calls audio stop working in one way(cannot hear us) and DTMF stops working. I have realize if I remove my default gateway happens the same. What can I do?. I am using asterisk 11.25 |
| 20:53.32 | igcewieling | eromo: run a local dns server and make sure the IP of EVERY interface is listed in /etc/hosts |
| 20:54.30 | igcewieling | For reasons nobody knows, Asterisk (maybe the OS) likes to do reverse DNS lookups against the IP of each interface. This is such a common issue, the FreePBX distro includes a local dns server. |
| 21:01.56 | eromo | Strange Ok that worked |
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| 23:58.41 | n\ack | Asterisk features' in-band DTMF tones are subtracted from the media stream (of course) but there is a smidge of the tone that makes it through (naturally). Question. Is there a way by which I can add a few MS of latency in exchange for removing these tones completely? It would be akin to the LIVE BROADCAST dump key in Radio and TV. Is this a thing? |