| 00:55.03 | *** join/#asterisk fuselage (~metarzan@cpe-65-28-165-112.wi.res.rr.com) |
| 00:58.28 | fuselage | i currently use callcentric, a softphone app, and google voice for my sip/voip telephony |
| 00:58.48 | fuselage | trying to figure out if adding asterisk to the mix could somehow improve things |
| 00:58.54 | fuselage | any input on that? |
| 01:04.52 | jm|laptop | fuselage: it could add 'features'. Depends what you need, really. |
| 01:05.05 | jm|laptop | e.g. centralised voicemail |
| 01:05.59 | fuselage | thanks jm|laptop. my needs are really pretty basic. this is just a home phone line |
| 01:06.45 | jm|laptop | then you might have all you need already! What drew you to Asterisk in the first place? |
| 01:06.48 | fuselage | we've always had reliability issues to a greater or lesser extent like missed calls and audio lag |
| 01:07.32 | fuselage | like somone will call us when we're home but the phone doesn't ring and they wind up getting voicemail |
| 01:07.43 | jm|laptop | fuselage: assuming your internet is capable enough, that sounds more like a sip client issue to me? |
| 01:08.01 | fuselage | audio lag is worst when speaking with cell phone users |
| 01:08.44 | jm|laptop | and that sounds like the lag is in the cellular leg? |
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| 01:08.45 | fuselage | yeah. we switched to grandstream wave some months ago and things do seem better on the missed calls front |
| 01:09.39 | fuselage | also i have a vps i need to repurpose and wondered whether setting up asterisk on it might be worthwhile |
| 01:09.47 | jm|laptop | I live in the middle of a forest and cell signal is pretty poor [indoors particularly], if someone else even remotely rural rings me on my mobile it's almost unbearable! |
| 01:11.10 | jm|laptop | fuselage: NAT is often a killer with these things. SIP likes direct routes. Shoving a VOIP server endpoint on a VPS with proper and good connectivity is probably a good idea, but if you're still doing odd NAT/SNAT/STUN for the last bit it might not help |
| 01:11.50 | fuselage | yeah. i've got a nat router here in the home (openwrt) |
| 01:11.59 | jm|laptop | fuselage: Asterisk is IPv6 compliant which can sometimes help assuming your internet and all your local devices support it too, and assuming you don't have an routtable IPv4 subnet to carve up. |
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| 01:13.15 | jm|laptop | fuselage: if you slap Asterisk on that VPS it might at least be a decent learning exercise and you might realise features you didn't expect to need :) |
| 01:13.25 | fuselage | my lan is ipv4 but the router of course handles ipv6. as well as the "home phone" (samsung edge 7 mobile phone that has wifi but no connection to cellular network) |
| 01:14.32 | jm|laptop | does the SE7 VOIP client support IPv6? I take it you're using the one built-into Dialer? Or did you say you're using a third party app? |
| 01:14.35 | fuselage | i've wanted for years to do some experimentation with asterisk but stuck with ata's so far as they were easier to start with |
| 01:14.55 | jm|laptop | ATAs can connect to Asterisk :) |
| 01:14.59 | fuselage | the phone supports ipv6. will take a look at the sip app to see |
| 01:15.51 | jm|laptop | "some" missed calls smacks a little of NAT/STUN timeouts to me but I'm no expert on the matter because I made sure I didn't need to rely on them before I rolled things out |
| 01:17.14 | jm|laptop | if your router, ISP and SIP broker all support IPv6 that's probably a Good Thing. As more things started supporting IPv6 my job got a lot easier (I support a handful of VOIP servers for small businesses) |
| 01:18.29 | jm|laptop | I found even that if you have to use RFC1918 locally on your LAN, if everything from a local Asterisk server upwards if 'proper' IP[v6] I got better results, even if the local Asterisk was still proxying, effectively |
| 01:19.02 | fuselage | yeah. i first ran across this idea in a post someone made on the openwrt forum about sip telephony and how he was thinking of setting up ipv6 to answer to problems he was having with sip telephony |
| 01:19.07 | fuselage | made sense to me |
| 01:22.48 | jm|laptop | the irony is that most [all?] "voip" providers won't/don't do SIP anyway. The business model is a bit weird, if you think about it. If all they're doing is translating a 'telephone' number into a SIP endpoint and then RTP happens directly between everyone (in a perfect all-routed-IP[v6]-world) how do you cost that? |
| 01:23.57 | jm|laptop | (obviously, they /employ/ SIP as part of the general process - but they effectively proxy everything so they can call charge.) |
| 01:24.25 | jm|laptop | mumbles something about E.164 |
| 01:25.46 | jm|laptop | fuselage: do you currently leverage¹ free calls? |
| 01:25.51 | jm|laptop | ¹ not a verb |
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| 01:26.30 | jm|laptop | running your own local voip server can permit internet calling from various devices that allow SIP URI calling |
| 01:26.54 | jm|laptop | or if you have friends/relatives with SIP clients you can all mush up nicely in a PBX |
| 01:28.55 | jm|laptop | I used to use chan_mobile aeons ago, before it was properly mature, because I had 'hundreds of minutes' of calls to other mobile phones and landlines on my monthly tariff and this let us leverage¹ that from any of our home phones |
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| 04:51.09 | LiuYan | If a phone respond 302 to asterisk to redirect to another number, asterisk then call a Local/the-redirected-number@my-context, how can I know the original called number in the second call (@my-context)? |
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| 04:54.14 | LiuYan | I didn't see datatype name contains 'orig' in CALLERID function. https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Function_CALLERID |
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| 08:54.26 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 18.3.0, 16.17.0 (2021/03/25) Final Bugfix: 13.38.2, 17.9.3 (2021/03/04); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
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| 09:04.21 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 18.3.0, 16.17.0 (2021/03/25) Final Bugfix: 13.38.2, 17.9.3 (2021/03/04); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
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| 10:14.19 | sibiria | LiuYan: one way would be to store the number in a channel variable for later keeping |
| 10:14.37 | sibiria | though i think you can refer to it using the original channel as well |
| 10:14.46 | sibiria | for later use* |
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| 12:47.19 | LiuYan | sibiria: but how to refer it using the orginal channel? >> though i think you can refer to it using the original channel as well |
| 12:57.07 | sibiria | LiuYan: with the MASTER_CHANNEL() function |
| 12:58.08 | sibiria | it works the same way CHANNEL() does, but it points to the originating channel rather than current channel |
| 12:58.50 | sibiria | or specifically, it points to the oldest channel |
| 12:59.31 | sibiria | it might be a simpler and cleaner solution to just set an inheritable channel variable in the start instead |
| 13:15.30 | LiuYan | sibiria: Thanks, this function is new to me, it seems is the proper function to get the original callled number. >> with the MASTER_CHANNEL() function >> it works the same way CHANNEL() does |
| 13:21.36 | sibiria | LiuYan: it depends a bit on how the call was set-up and how the dial plan looks. usually you will be able to get it from the EXTEN variable, but you can also look at e.g. endpoint / contact |
| 13:21.55 | sibiria | check out the "Function_CHANNEL" documentation on wiki.asterisk.org |
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| 13:30.57 | sibiria | i find it easier to just set the variable myself. less hunting for it depending on the scenario of how the call was set up |
| 13:33.57 | LiuYan | sibiria: Okay, thanks a lot. I'll check if MASTER_CHANNEL() works on our dialplan. Our dialplan is logically simple: A call is in from PSTN, then transform the called number to SIP phone number of employee, the phone rings but it returns 302 redirect to the mobile phone number of the employee, then asterisk starts the local channel to call the mobile phone number |
| 13:36.59 | LiuYan | sibiria: and when call the mobile phone number, the CALLERID(num) is the CALLERID(num) of the original channel, which is not allowed by the SP, so I need to know the EXTEN of the original channel, and set it as CALLERID(num) of the local channel. |
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| 13:37.59 | sibiria | yeah, as said, i'd just save it to an inheritable channel variable right at the start when the extension is known |
| 13:38.42 | sibiria | saves the trouble of trying to fish for it later on |
| 13:40.56 | LiuYan | sibiria: yes, that's already done in our dialplan, the variables includes even the employee's department & seat information, lol |
| 13:42.49 | sibiria | shoe size... eye color... etc. |
| 13:44.37 | igcewieling | Don't forget current GPS coordinates! |
| 13:46.57 | Samot | This is why I disable CF at the phones |
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| 14:07.42 | LiuYan | sibiria: that's not necessary, I only use these information to append to CALLERID(name), so that the callee will know which department & building you're from >> shoe size... eye color... etc |
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| 15:19.04 | Kobaz | woah, i just had a sip stack freeze on 16 suddenly |
| 15:19.28 | Kobaz | unfortunately i don't have a core file |
| 15:21.22 | Samot | That's odd. |
| 15:21.45 | Kobaz | i was trying to troubleshoot why asterisk wasn't responding to this invite from a new endpoint i set up |
| 15:22.08 | Kobaz | i was tcpdumping... saw it coming in and asterisk ignoring it, and then i checked asterisk and it stopped responding to sip entirely |
| 15:22.30 | Kobaz | my health check restarted it... so i'll add a core dump prior to restart |
| 15:23.01 | Samot | Was it just not responding or was it completely dead? Not sending out requests? |
| 15:24.15 | Kobaz | the console was functional |
| 15:24.18 | Kobaz | probably a deadlock |
| 15:24.49 | Kobaz | everything was slowly going unreachable |
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| 15:29.40 | Kobaz | i saw this type of thing once before when pjsip reloads took too long |
| 15:33.19 | Kobaz | i'll see if i can break it again |
| 15:56.08 | Samot | I have faith. |
| 15:58.15 | Kobaz | me too |
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