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09:49.27 | utrack | Howdy! Is there a way to make CDR count sanely when making a call with Originate? |
09:50.16 | utrack | At the moment it counts everything as BillableSeconds (i.e. when the 1st side picks up the call BillableSeconds counter starts) |
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13:16.00 | talntid | I don't know how to propose changes to the asterisk.org wiki, but.... |
13:16.02 | talntid | https://wiki.asterisk.org/wiki/display/AST/Extension+State+and+Hints |
13:16.15 | talntid | that page shows the syntax being exten = XXXX |
13:16.20 | talntid | instead of exten => XXXX |
13:18.19 | newtonr | talntid, simply comment on the page |
13:18.28 | newtonr | talntid, or for big changes file an issue on issues.asterisk.org/jira |
13:18.47 | newtonr | talntid, both syntax are valid |
13:19.20 | newtonr | talntid, newer examples on the wiki will use = instead of => as the > is unnecessary. |
13:22.58 | [TK]D-Fender | I like my arrows. Feels classier |
13:25.06 | newtonr | :D |
13:43.34 | talntid | ahh, i see. yes, i like the arrows too :P |
13:44.17 | talntid | got my * box all integrated tonight... it emits events over XMPP, line presence over XMPP, etc... |
13:44.33 | talntid | ain't i speshul... |
13:44.58 | talntid | AMI events, that is |
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14:03.51 | stephanj | newtonr, [TK]D-Fender: i am automatically generating asterisk sip.conf, voicemail.conf and extensions.conf for a project... i spend 2 hours looking on how to *cleanly* hack the ini writer to use '=>'. a day later i read the proper docs and they told me that the arrow was not mandatory.. i was a bit annoyed ;) |
14:04.21 | newtonr | :D |
14:05.08 | newtonr | I'm sure you'll use those skills you enhanced in the future! |
14:05.48 | stephanj | i still have neither read the complete docs or the book |
14:06.03 | newtonr | They are helpful! |
14:06.21 | stephanj | probably! |
14:07.30 | newtonr | stephanj, So why are you using chan_sip instead of chan_pjsip? |
14:08.25 | stephanj | easy because i still run debian jessie with asterisk 11 |
14:08.42 | stephanj | and chan_pjsip is a invention from asterisk 13? |
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14:10.36 | newtonr | Yes the pjsip resources and channel driver were added in 12. |
14:11.17 | stephanj | i tested 13 briefly when messing with TLS in asterisk |
14:11.26 | newtonr | When you think about moving to 13 you'll want to use chan_pjsip as that is where the development focus is for the future of SIP connectivity in Asterisk. |
14:11.32 | stephanj | cool |
14:11.51 | stephanj | until i found out that me not being able to provide a correct cert file does not varrant keeping a self compiled asterisk in production :) |
14:12.39 | stephanj | we use a config writer which pulls stuff out of ldap and into flatfiles.. so supporting a new backend is quite easy |
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14:14.59 | newtonr | stephanj, :) cool. |
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15:05.20 | Phil-Work | I'm trying to automatically originate 2 outbound calls and conference them together on an inbound call... to create a 3 way call |
15:05.36 | Phil-Work | presume I need to use ConfBridge but I'm not entirely sure how to implement it |
15:05.59 | Phil-Work | if I Answer(), ConfBridge() then Dial()... the Dial() isn't executed |
15:09.53 | [TK]D-Fender | You don't dial |
15:10.23 | [TK]D-Fender | What is the starting trigger for this process? |
15:11.40 | Phil-Work | [TK]D-Fender, call coming into exten 11111 on the "alarm" context |
15:12.15 | Phil-Work | that should simultaniously dial 2 numbers on a SIP trunk and conference them all together |
15:12.59 | Phil-Work | *simultaniously |
15:14.01 | [TK]D-Fender | that's a 2-way call |
15:14.41 | [TK]D-Fender | So in this case I'd just directly Dial() and redirect the 1st to the second after answer and confirmation |
15:14.56 | Phil-Work | there's 3 parties - the original caller and then the two that are dialled |
15:15.22 | [TK]D-Fender | ok, original + 2 |
15:15.27 | Phil-Work | yes |
15:15.51 | [TK]D-Fender | So in that case trigger ther 2 originates and continue on to dump the caller into the conf immediately awaiting their arrival |
15:17.00 | Phil-Work | how do I originate those calls if not using Dial() |
15:17.13 | Phil-Work | oh, Originate()? |
15:19.26 | [TK]D-Fender | use AMI or call files. The dialplan app version is blocking |
15:19.46 | Phil-Work | hm, ok |
15:19.59 | [TK]D-Fender | or CLI call |
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16:17.32 | craigify | that's a lot of simultaneous ping timeouts |
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16:37.41 | Phil-Work | is it possible to get the number of users in a given confbridge as a variable? |
16:38.57 | WIMPy | CONFBRIDGE_INFO |
16:41.06 | igcewieling | Phil-Work: you should read the output of "core show functions" |
16:42.03 | Phil-Work | ok, thanks |
17:04.29 | Phil-Work | I have something like this to create a 3 way conference call on an inbound call - https://gist.github.com/phil-lavin/b52ed32395238d3854b7 |
17:04.41 | Phil-Work | the NoOp doesn't happen, presumably because the ConfBridge() blocks |
17:04.45 | Phil-Work | what's the way around this? |
17:06.59 | Phil-Work | I guess I could originate a 4th call to another local channel which handles the NoOp |
17:07.35 | igcewieling | Phil-Work: I assume the problem is the truely horrible way you are using Originate |
17:07.53 | igcewieling | Phil-Work: "core show application Originate" |
17:08.32 | Phil-Work | igcewieling, I've read the manual but not doing it via System() also blocks until the call is answered |
17:09.03 | igcewieling | Phil-Work: correct. |
17:09.22 | Phil-Work | so what's the non-blocking alternative? |
17:09.36 | igcewieling | I normally use .call files or AMI to Originate. |
17:10.07 | Phil-Work | those were amoung [TK]D-Fender's suggestions - I'll look into those as less dirty alternatives |
17:10.12 | Phil-Work | thanks |
17:10.56 | igcewieling | Phil-Work: using System() like that will cause all sorts of issues. |
17:13.58 | lvlinux | Phil-Work: I have N-way calling setup---is that what you are needing? |
17:15.13 | Phil-Work | lvlinux, yes |
17:15.14 | lvlinux | I set it up for an office where the recptionist answers the phone, the caller needs to talk to both the receptionist and the owner at the same time. |
17:15.40 | Phil-Work | similar concept - this is for an intruder alarm |
17:15.57 | Phil-Work | the alarm dials asterisk over PSTN and Asterisk automatically calls n people and conferences them together |
17:16.15 | lvlinux | I'll put up the config a bit later---have something I have to do now but if you'll be here a while I'll get back with you. |
17:16.29 | lvlinux | That's not exactly how my thing works but you should be able to adapt it pretty easily. |
17:16.44 | Phil-Work | I'll be online but possibly not here - just highlight me and I'll pick it up when I get home |
17:16.52 | Phil-Work | thanks :) |
17:17.20 | lvlinux | k |
17:17.38 | WIMPy | wonders if the classic conference could somehow be emulated with Bridge(). |
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17:19.09 | lvlinux | Bridge() doesn't do mixing does it? |
17:20.14 | WIMPy | No, but it allows you to connect to other channels that aren't part of the current call (yet). |
17:21.06 | lvlinux | hmm, interesting |
17:46.18 | Phil-Work | am I right in assuming that Asterisk cannot distinguish a call that was legitimately answered and one that went to voicemail when calling, for example, a cell phone over a SIP trunk? |
17:47.54 | [TK]D-Fender | It is answered |
17:47.56 | [TK]D-Fender | one way or the other |
17:47.59 | lvlinux | ehh, I _think_ there are methods for determining if it's a real person on the other end, but I don't know how effective they are. |
17:48.05 | [TK]D-Fender | WHAT answered ... is a "point of view" |
17:48.11 | [TK]D-Fender | "core show application amd" <----- |
17:48.14 | igcewieling | Phil-Work: correct. |
17:48.56 | igcewieling | Phil-Work: you should read the output of "core show applications" 8-) |
17:49.39 | Phil-Work | hm, amd looks interesting |
17:49.42 | Phil-Work | I'll give it a try |
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17:51.04 | igcewieling | Phil-Work: Tuning AMD is not easy. I used to work for a company which considered their AMD settings to be a trade secret. 8-| |
17:52.19 | igcewieling | Phil-Work: you could use answer confirmation, do a search for: asterisk answer confirm |
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17:58.23 | Phil-Work | I'll have a play with AMD |
17:58.43 | Phil-Work | it'll be the same 2 voicemails that are hit so once it's tuned for those, it should be alright |
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19:16.48 | jmordica | Does anyone know if homer by sipcapture has an irc channel? |
19:16.59 | jmordica | or does anyone use homer here? |
19:36.28 | rrittgarn | i'm starting to implement homer in my environment - seems like it will be useful. I do not know about the IRC channel though |
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19:37.03 | rrittgarn | Question about app_followme in 13 - does the syntax in the file work like queues or sip.conf where you can have templates? Or has anyone played with it in realtime? |
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19:55.47 | dan_j | Hi. Is phpagi compatible with asterisk 13? |
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20:01.41 | copystring | my asterisk seems to shut down for no apparent reason but the process doesn't get killed. on top of that another instance of asterisk is startet after that and that instance of asterisk is then no longer able to connect to the trunks. the two instances then result in this: Â Failed to bind to 0.0.0.0:59361: Address already in use. any ideas why this happens? |
20:02.49 | WIMPy | Well, you can't hace two instances running. |
20:03.11 | WIMPy | You need to find out what starts the new instance. |
20:03.34 | copystring | yes. i've been trying that. the logs seem to show nothing :\ |
20:07.45 | mjordan | do you have safe_asterisk running? |
20:07.58 | mjordan | (granted, it should only fire a new Asterisk instance up when the old one dies...) |
20:08.09 | copystring | no. what's that? |
20:08.25 | mjordan | "man safe_asterisk" |
20:08.54 | copystring | not sure if that helps since the old process never actually dies ... it just sits there |
20:09.01 | copystring | doing nothing |
20:09.12 | copystring | (apart from blocking the port) |
20:09.35 | cmendes0101 | ps uxa | grep asterisk |
20:09.42 | cmendes0101 | who owns the other process? |
20:09.52 | copystring | both are owned by root |
20:10.30 | copystring | ah no. both are owned by asterisk |
20:11.11 | cmendes0101 | hmm, just kill both and try to run it again |
20:11.26 | cmendes0101 | or does that always happen? |
20:11.45 | copystring | it happens all the time |
20:11.55 | cmendes0101 | how are you running it? |
20:11.59 | copystring | it runs for like 20-30 minutes then boom. tits up |
20:12.33 | cmendes0101 | oh so you have one process and everything is happy for 20-30? |
20:12.46 | copystring | yes, roughly |
20:12.59 | copystring | it's running on pfsense via the official package from pfsense. also tried the freebsd package (version 1.8 and version 13) |
20:13.21 | cmendes0101 | what level are you logging? |
20:13.57 | copystring | i've connected to the console of asterisk with log level 80 and waited for it to happen. want a pastebin of it? |
20:14.00 | cmendes0101 | you can run asterisk with -g and that should dump core if it crashes |
20:15.16 | cmendes0101 | does the logs in /var/log/asterisk show anything else from what your seeing remote? |
20:15.42 | copystring | yes. the log level was only in the console |
20:18.32 | copystring | i'll make a new log with log level 80 in the asterisk.conf |
20:18.38 | cmendes0101 | try running with -c -g. When I have modules having issues the -c helped show that since it wasn't logging for me |
20:19.03 | copystring | ok. i'll do that |
20:20.04 | cmendes0101 | just curious what do you mean log level in asterisk.conf? |
20:20.14 | cmendes0101 | I haven't really seen that before |
20:20.29 | copystring | oh. i meant verbosity |
20:21.46 | cmendes0101 | I think the max on verbose is like 10 but that setting is just for CLI i believe. For changing logging its in logger.conf |
20:23.25 | copystring | odd. when i changed verbosity in the asterisk.conf it said in the console the log level was whatever i've set it to |
20:24.01 | mjordan | Just as an FYI: max debug is 5; max verbose is 4. |
20:24.16 | cmendes0101 | oh there we go lol |
20:24.20 | mjordan | Once upon a time some really spammy junk was tossed out at 10 for some reason, but that's been removed for ages. |
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20:24.42 | mjordan | You can always specify higher; it just won't do anything |
20:25.08 | copystring | yeah. i thought so. i've just set to 80 because i didn't bother finding out what the max was :) |
20:26.12 | copystring | got it running with -c -g now. just waiting for it to happen again |
20:34.33 | rrittgarn | when using extconf to load things into real time - how does that work programmatically? I know that some modules have native support for it like sip.conf. I'm trying to get app_followme to work. I am not getting SQL errors on it, so i know its connecting, but when i query for an ID it isn't found |
20:35.16 | rrittgarn | I'm not entirely sure that app_followme supports realtime, but i don't know where to start fixing if there is fixing required at the dev level... i'd love to contribute if necessary though |
20:36.15 | rrittgarn | disregard - googled better than i did before |
20:36.25 | rrittgarn | fail on my part for not finding https://wiki.asterisk.org/wiki/display/AST/Followme+-+Realtime |
20:49.07 | igcewieling | jmordica: I use pcapsipdump but it is not designed to handle the scale Homer is designed for. |
20:50.38 | igcewieling | copystring: there is only one verbosity level, it applies to all log files and the CLI. Make sure Asterisk is not blocking on a DNS request. |
20:51.29 | copystring | blocking on a dns request? |
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21:10.20 | igcewieling | copystring: Asterisk can stop processing calls if it is having trouble looking up hostnames for ip addresses configured on the system. |
21:10.45 | igcewieling | Usually this happens when the server's internet connection is down. Even extenstion to extension calls fail. |
21:11.44 | igcewieling | copystring: I recommend running a local caching only name server, point /etc/resolv.conf to it Update /etc/hosts to have a name for every ip on the system. Enable dnsmgr in Asterisk. |
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21:16.52 | copystring | sounds a lot like that might be it. the dnsmgr is enabled and the resolv.conf is configured to the caching dns already. just the hosts file isn't. i'll give that a try next time it goes weird |
21:17.28 | igcewieling | copystring: the key is to do it before it freaks out. Many things could cause the problem, DNS is only one. |
21:19.43 | copystring | yup. i've just tried something else just now and i'm waiting to see if that already did it. |
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21:30.23 | lvlinux | copystring: it's a major pain. I think using pjsip fixes it though since I don't believe pjsip DNS lookup is blocking. |
21:32.08 | copystring | i thought about using pjsip. does it work with a normal sip trunk and is it any good? |
21:33.50 | copystring | hm, asterisk 1.8 doesn't seem to support pjsip |
21:37.56 | file | chan_pjsip only exists in 13+ |
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21:46.29 | copystring | right. i have another log now with the issue and the -c -g parameters. it does a few dnsmgr_lookup and then:-- Remote UNIX connection |
21:46.30 | copystring | Beginning asterisk shutdown.... |
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23:03.42 | Encrypt | Hum, so boritos isn't registered :( |
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