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| 00:02.44 | apb1963 | rneese: where did you see that it supports 11.0 ? |
| 00:03.40 | rneese | I closed the page I will dig it up in a min .. |
| 00:03.50 | rneese | on phone at the min |
| 00:08.04 | apb1963 | rneese: ok, thank you |
| 00:09.42 | rneese | http://www.vicidial.org/VICIDIALforum/viewtopic.php?f=4&t=12501 |
| 00:09.54 | rneese | 1 page that says they moved to 1.8 |
| 00:10.00 | rneese | looking for the other page |
| 00:12.24 | apb1963 | rneese: :) |
| 00:12.55 | rneese | I read wron the guy installed 11 and found out that 2.7 vd would not work |
| 00:13.04 | rneese | http://forums.asterisk.org/viewtopic.php?p=188693 |
| 00:13.34 | apb1963 | yeah, I was kinda wondering... something like supporting 11.0 would of made the front page |
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| 00:14.50 | rneese | well with is supportint 1.8 should not be hard to get it to 11 it cant be that big of a hop |
| 00:15.03 | rneese | but I would not do it |
| 00:15.08 | rneese | I hate dialers |
| 00:15.12 | apb1963 | lol |
| 00:15.20 | rneese | I think they all should be destroyed |
| 00:15.34 | rneese | yes I use to do them to make money |
| 00:15.40 | rneese | but never liked them |
| 00:15.42 | apb1963 | they have their uses |
| 00:16.04 | WIMPy | Theyt certainly have their useRs. |
| 00:16.34 | apb1963 | When the doctor wants to remind you of your appointment... it frees the nurse from having to call you. |
| 00:16.34 | rneese | well the company I am working for now wants a sall router pbx in one we can put in home offices and link to the office pbx |
| 00:16.42 | rneese | thats true |
| 00:16.48 | apb1963 | sall? |
| 00:16.58 | rneese | small |
| 00:17.31 | rneese | the board we are workign on is a dualcore arm with a gig o fram |
| 00:17.35 | rneese | ram |
| 00:18.08 | apb1963 | cool |
| 00:18.11 | WIMPy | You call that small? |
| 00:18.56 | rneese | its only 4 port gig port and bgn wifi |
| 00:19.17 | rneese | mips routers are getting old and behind the time |
| 00:19.26 | rneese | we want to have the needed power |
| 00:22.22 | apb1963 | core restart |
| 00:22.23 | apb1963 | No such command 'core restart' |
| 00:22.26 | apb1963 | wth ?? |
| 00:22.44 | apb1963 | since when? |
| 00:22.46 | WIMPy | when? |
| 00:23.01 | WIMPy | No. When do you want it to restart? |
| 00:23.06 | apb1963 | ohhhh |
| 00:23.08 | apb1963 | lol |
| 00:23.15 | apb1963 | but it says no such command |
| 00:23.23 | apb1963 | not... command requires a time or some such |
| 00:24.55 | apb1963 | I'm not sure if it restarted... I reconnected immediately, so if it did restart... it only took 3 secs. |
| 00:25.25 | WIMPy | Why should it take longer? |
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| 00:26.26 | apb1963 | Dunno |
| 00:26.54 | apb1963 | I'm getting bizarre results, so I don't know what to think... that's why I restarted * |
| 00:27.02 | apb1963 | and, it didn't help |
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| 00:29.23 | apb1963 | is core restart the same as a complete reboot? |
| 00:30.18 | WIMPy | Off course not. |
| 00:30.23 | WIMPy | It only restarts Asterisk. |
| 00:30.45 | apb1963 | ok, let me rephrase |
| 00:31.32 | WIMPy | It restarts Asterisk, yes. |
| 00:31.47 | WIMPy | At least if it succeeds. |
| 00:32.18 | apb1963 | if I were to completely kill all asterisk processes, and cleanup any leftover droppings that might have been left... is that the same as restarting asterisk? |
| 00:32.33 | WIMPy | yes |
| 00:32.40 | apb1963 | then I'm a bit befuddled |
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| 00:49.43 | apb1963 | time for a reboot |
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| 02:07.23 | ruben23 | hi guys anyone cna help how to ad custom field on the the CDR of asterisk somehow.. |
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| 03:22.17 | ruben23 | :) guys how to setup a custom cdr on my asterisk 11 going to mysql...where do i define the custom field somewhow..?? |
| 03:23.19 | ruben23 | is it on cdr_custom.conf or |
| 03:23.23 | ruben23 | cdr_mysql.conf |
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| 04:46.06 | ChannelZ | Hrrm. Under the asterisk 12+ 'all calls are bridges' method, is there extra latency being added because of the bridge? |
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| 05:22.50 | ChannelZ | and how do you debug rtp with pjsip now? |
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| 09:52.45 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.13.0 (2014/09/24), 1.8.31.0 (2014/09/24); Standard: Asterisk 12.6.0 (2014/09/24); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
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| 09:55.37 | Zogot | what do people think is a good process to make a new rpm package of the build sources i have? |
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| 10:03.26 | SQUelcher | Hi |
| 10:04.30 | SQUelcher | I've just disabled SSLv3 on my XMPP server (Prosody), now res_xmpp can't connect anymore, can anyone confirm that it isn't supporting TLSv1? |
| 10:06.26 | TrevorH | SQUelcher: openssl s_client -connect your.hostname:port -ssl3 |
| 10:06.55 | TrevorH | or -tls1 |
| 10:15.29 | SQUelcher | TrevorH: I've to add -starttls xmpp there |
| 10:16.04 | SQUelcher | Do I need to add a legacy ssl port? |
| 10:16.39 | TrevorH | SQUelcher: no idea, sorry |
| 10:21.47 | SQUelcher | I've tried it, won't help either :/ |
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| 10:23.37 | vulpe | Hello everyone! I get CDR direct executed failed while my odbc write works |
| 10:23.40 | vulpe | any ideas? |
| 10:26.14 | tomekmatu | wdoekes: I dont expect for anyone holding my hand. I just wanted a bit more advice. for my small project is it better to use the latest ver of Asterisk or a special ver. of it. Also what did you mean when you said "you'll need the dialplan" |
| 10:33.23 | vulpe | @tomekatu is someone here? |
| 10:38.03 | wdoekes | tomekmatu: latest version of stable branch is best: i.e. 11.latest |
| 10:38.28 | Rac-on | tomekmatu: if you are going to connect a asterisk to the internet while having no clue where to start or how to configure |
| 10:38.38 | wdoekes | as for the dialplan, most of the work you'll need to do will be in extensions.conf |
| 10:38.40 | Rac-on | tomekmatu: be preparred to get hacked/abused within seconds |
| 10:42.20 | vulpe | Someone with a little bit of experience in recording CDR to a remote database? |
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| 10:54.45 | wdoekes | vulpe: increase logging/debugging in asterisk and/or unixodbc and/or sniff the db connection (if tcp) |
| 10:57.00 | vulpe | debuggig is set to level 5, from there I have the error text. I'm working in logging the odbc queries with not much luck yet |
| 10:57.33 | vulpe | However I thought that having [odbc write ....] working, from asterisk perspective there shouldn't be a connectivity issue |
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| 11:13.53 | wdoekes | could still be your cdr_odbc config |
| 11:14.10 | vulpe | @wdoekes |
| 11:14.24 | wdoekes | or it could be that your cdr table isn't correctly created |
| 11:14.24 | vulpe | thank you very much. You inspired me |
| 11:14.39 | vulpe | I actived logs for both odbc and freetds |
| 11:15.11 | vulpe | odbc more or less useless, but from freeetds I found out that a missing column in a differnet table prevented a stored procedure to run |
| 11:15.37 | wdoekes | hehe |
| 11:15.43 | vulpe | so not much from asterik's side, but from mine. Sorry for disturbing you |
| 11:15.48 | wdoekes | np |
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| 13:38.32 | gciotta | Hi, I'd like to playback a mp3 file during an already established call. Is this possible? I'm exploring using Confbridge but maybe there's an easier way. Thanks! |
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| 13:47.18 | newtonr | gciotta, do you care if one party in the call gets disconnected from the other? |
| 13:52.21 | gciotta | newtonr: I'd like the music to be played in the background |
| 13:52.28 | gciotta | while the two parties talk |
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| 13:55.19 | newtonr | gciotta, yeah I'm not sure how you would do that, unless you used a confbridge like you said |
| 13:55.58 | newtonr | gciotta, talking over music sounds like a bad idea for the users :D but I'm guessing there is a reason. |
| 13:56.26 | newtonr | Is it a phone conference that simulates being in a bar? |
| 13:56.40 | newtonr | Try to talk over this loud music! |
| 13:57.01 | WIMPy | alibi phone booth |
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| 14:55.33 | Zogot | anyone know if there is an issue with odbc and pjsip in asterisk 12.5? |
| 14:56.01 | Zogot | as i get an error when trying to register, it isn't adding the aor to the aor table |
| 14:56.10 | Zogot | or updating the aor table event |
| 14:56.11 | Zogot | even* |
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| 14:58.19 | file | what's the error? |
| 14:59.51 | Zogot | sql error with the put_statement, il redo the specifics, i log all queries on the db and dont see the put/update request, pjsip list aors works great |
| 14:59.59 | Zogot | if you give me 1 min, il get the specifics |
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| 15:04.50 | Zogot | file: https://gist.github.com/zogot/affc4f17576c0db6e748 |
| 15:04.55 | boch | hi all |
| 15:07.28 | boch | i am running an AGI script to bill calls on 'h' extension, and im getting duration differences between CDR(duration) variable and the one stored on database (database value is equal or bigger than variable) any one knows why and how to avoid this difference? |
| 15:09.47 | Zogot | file: from right now: https://gist.githubusercontent.com/zogot/184e90efe7ed29468e94/raw/fd30c2bc9eaebfc9ff5c76232afb3ba66a88b1c6/gistfile1.txt |
| 15:13.07 | Zogot | i saw there was a commit in 12.06 to attempt to stop sorcery from binding multiple pjsip loads |
| 15:14.04 | file | I have no suggestions on that. |
| 15:15.00 | Zogot | alright, no worries, thanks for looking anyway. perhaps you can answer another thing for me, the asterisk packages for centos havent been updated for 12.6 yet |
| 15:16.23 | Zogot | i built asterisk from the sources but the packages have been a testing thing for me at the moment, when trying to make an rpm based off the spec file from the SRPMS folder, it references other packages that I dont have, I would imagine I would have to look into making my own spec file |
| 15:16.50 | Zogot | or is there a public location of those other repositories, pjproject and the like? |
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| 15:24.07 | Katty | hello my asterisk does not work at all how to fix pls?? |
| 15:24.21 | Katty | is urgent thx. |
| 15:24.40 | WIMPy | What is it you want to fix? |
| 15:24.45 | Qwell | Katty: Please deposit $0.50. |
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| 15:29.22 | lvlinux | lol |
| 15:29.26 | Katty | Qwell: i appear to only have 27 cents. |
| 15:29.35 | Katty | Qwell: the asterisk gods will be displeased. |
| 15:29.54 | Katty | WIMPy: hmm. well. i guess my hair for starters. |
| 15:29.59 | Katty | WIMPy: it's a bit frizzy today. |
| 15:31.01 | WIMPy | That's what basecaps were invented for, I guess. |
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| 15:32.12 | Milenco | hey guys, i'm setting up queue cascading and have a question, perhaps you can help |
| 15:32.34 | Milenco | can i somehow see if all the members in the queue are busy before placing a new call in the queue? |
| 15:33.01 | Milenco | I would like to cascase callcentre calls to another location with has agents available if the first location is fully busy |
| 15:33.17 | WIMPy | You can check the device states if you know the agents. |
| 15:33.31 | Katty | WIMPy: yes indeed. |
| 15:33.37 | Milenco | I see that queues have an timeout but I rather have them send to an empty queue to prevent the waiting |
| 15:34.24 | Milenco | the agents vary WIMPy, we are using an external tool to manage the agents in our queue. |
| 15:35.13 | WIMPy | that makes it more complicated, but not impossible. |
| 15:35.47 | Milenco | but i can check a queue's agent status in my extensions? |
| 15:36.20 | WIMPy | If the channels have device status, i.e. sip. |
| 15:36.28 | [TK]D-Fender | Milenco: There is a queue option to kick them out if there are no free agents |
| 15:36.58 | Milenco | thanks [TK]D-Fender :) |
| 15:37.10 | WIMPy | only remembers the one for no agents. But I don't use them excessively :-) |
| 15:37.20 | Milenco | they do WIMPy, using IAX2 but I have no way of detecting the uids i think |
| 15:37.31 | Milenco | I'll take a look into that option |
| 15:38.06 | [TK]D-Fender | Milenco: Or you can script something up to check first. Three doesn't appear to be any app or function that could confirm this directly |
| 15:38.09 | [TK]D-Fender | there* |
| 15:38.32 | Milenco | the 'kick if no free agents' option would do it |
| 15:40.40 | Milenco | when is a queue considered 'full'? |
| 15:41.10 | WIMPy | There is a max calls limit. |
| 15:41.31 | Milenco | i dont see it anywhere in the documentation im looking at |
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| 15:41.57 | Milenco | ah, its QUEUEMAX |
| 15:42.20 | WIMPy | maxlen |
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| 16:06.26 | ChannelZ | Is there anyone who understands HFSC and/or can explain it in human english that can be understood? |
| 16:07.32 | WIMPy | h-fsc? |
| 16:08.11 | [TK]D-Fender | I understand HFCS.... that's pretty close... |
| 16:08.46 | WIMPy | I like HFC-S. |
| 16:13.24 | ChannelZ | Well, context is everything. I'm speaking of traffic shaping |
| 16:14.29 | WIMPy | That's what I thought. |
| 16:14.56 | WIMPy | At the time I set that up a combination of htb and sfq seemd like a good idea. |
| 16:15.05 | [TK]D-Fender | I'm speaking about diabetes.... |
| 16:16.02 | jeev | diabeetus |
| 16:17.40 | dwayne | I'm using the Blink softphone with Asterisk 11 configured for SIP TLS/SRTP. It's working great, except for one issue that I'm not sure if its a configuration issue or maybe a bug on either side. I have the Blink phone configured to register every 30 seconds and I'm seeing every other registration attempt being reported as failed on the phone even though most of the time it happens, sip show peers lists the peer |
| 16:18.42 | dwayne | when it fails, Asterisk is responding with a SIP 401 instead of the 200 and there is a notice about stale nonce |
| 16:18.56 | dwayne | I also have qualify on set to 30 seconds. Does this sound like a configuration issue? |
| 16:19.15 | ovoshlook | <PROTECTED> |
| 16:20.45 | orn | Does anyone know what asterisk does if it receives a malformed packet? Will it just ignore it and not log anything? |
| 16:22.37 | WIMPy | that would surely depend on how it was malformed. |
| 16:27.11 | orn | The tag is missing in the From header |
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| 17:28.44 | tafa2 | can anyone recommend a good SIP client for mac? |
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| 18:04.20 | boch | i am running an AGI script to bill calls on 'h' extension, and im getting duration differences between CDR(duration) variable and the one stored on database (database value is equal or bigger than variable) any one knows why and how to avoid this difference? |
| 18:06.57 | TazzNZ | boch: how big is the diff ? |
| 18:08.57 | boch | just a second or two, for calls that last 59 seconds is making a big difference :s |
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| 18:10.06 | eirirs | hehe, 0 or 1 minute are huge difference with whole 60 seconds |
| 18:10.07 | eirirs | :p |
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| 18:17.11 | WIMPy | Sure that it can be more than a second? |
| 18:17.38 | itsokrelax | Sorry if this has been asked 100 times. How are you guys sending MASSIVE amounts of faxes over long distance? I can only find ATT that offers true unlimited POTS lines |
| 18:17.49 | itsokrelax | 16 lines running 24/7 :( |
| 18:17.50 | WIMPy | Wasn't there a rounding option for CDR? |
| 18:29.29 | boch | WIMPy, not sure but ive seen calls with 1 or 2 seconds different |
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| 18:30.58 | boch | WIMPy, my script saves the call cost on ${CDR(userfield)} and asterisk does the rest (store on db, csv..) |
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| 18:52.49 | WIMPy | <wdoekes> (and if you use musiconhold, you may get chdir'd to somewhere else by |
| 18:52.53 | WIMPy | an accidental race condition, so the core_pattern works best) |
| 18:52.58 | WIMPy | Grrr |
| 18:53.00 | WIMPy | needs mre space |
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| 20:01.42 | justdave | Anyone know of any tips for troubleshooting DTMF issues? I seem to have exhausted what I can find on Google (most seem to talk about mismatched transmission types (sipinfo, rfc2833, inband)) and I've ensured we've got rfc2833 across the board. My IVR menus are dropping digits at random, even if I feature-code into it from a local phone. |
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| 20:14.54 | newtonr | justdave, you need to grab debug logs with DTMF logger channels enabled and perhaps a PCAP and throw it on the list where people will have time to look it over. |
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| 20:32.10 | justdave | newtonr: aha, logger is where that was |
| 20:32.32 | justdave | I knew I'd seen a way to get debug info on dtmf before, but couldn't find it. |
| 20:33.51 | justdave | hmm 'logger set debug level dtmf on' right? |
| 20:34.33 | newtonr | justdave, https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
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| 20:52.32 | justdave | ok, so the DTMF log shows every digit actually coming through correctly (I get a "passthrough" line for each digit), but the IVR still only sees 1 to 2 digits of it. what do I look for next? |
| 20:53.11 | justdave | I see all three digits of the extension number getting a passthrough line followed by "invalid extension" with 1 digit |
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| 21:01.25 | justdave | if I dial slowly (like a second or more between digits) it seems to work. |
| 21:01.42 | justdave | is there a timeout setting somewhere that needs to be shortened or increased? |
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| 21:02.04 | WIMPy | Do you use WaitExten? |
| 21:02.30 | justdave | Background() is what it's doing at the point where it's dropping stuff |
| 21:02.48 | WIMPy | Ok, so yiu use the dialplan/ |
| 21:03.01 | justdave | yeah |
| 21:03.20 | WIMPy | Do you have something included that does more than just match some digits? |
| 21:04.14 | WIMPy | I know that very well, as that's what makes it impossible for me to just dial. |
| 21:04.35 | WIMPy | But I'm using a combination of switch and DUNDi. |
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| 21:10.07 | justdave | yeah, there's a dundi switch in there |
| 21:10.16 | justdave | so dundi bustage might be causing it? |
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| 21:15.14 | WIMPy | It definitely does for me. |
| 21:15.33 | WIMPy | But for me it's for dialling, not for IVR. |
| 21:15.57 | WIMPy | But that goes the same way. |
| 21:16.38 | tafa2 | dont suppose anyone can recommend a good mac SIP clinet |
| 21:16.40 | tafa2 | *client |
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| 21:18.19 | justdave | tafa2: Telephone and YateClient both seem to work decently for me |
| 21:18.33 | justdave | tafa2: neither is very fancy but they get the job done and handle multiple accounts |
| 21:21.03 | tafa2 | nice thanks! |
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| 21:26.00 | WIMPy | justdave: If anyone finds out how to make asterisk not swallow digits, that would be a huge step towards usability. |
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| 21:30.32 | justdave | YateClient also does point-to-point sip without needing an account |
| 21:30.43 | justdave | (like for testing sip:... urls) |
| 21:31.12 | justdave | there's also X-Lite which is exceedingly more usable, but only does one account |
| 21:31.27 | justdave | (there's also Bria which is X-Lite with multiple accounts, but costs money) |
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| 23:48.21 | apb1963 | is there a "show settings" command that will show me nat settings? *1.8 |
| 23:49.20 | [TK]D-Fender | "core show settings" "sip show settings" |
| 23:49.44 | apb1963 | I've tried core show settings, sip show settings and core show settings misc |
| 23:50.17 | apb1963 | there's no indication of nat |
| 23:52.36 | [TK]D-Fender | I see indications... |
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| 23:54.59 | apb1963 | hmmm... none in mine :/ |
| 23:56.29 | apb1963 | there wouldn't be a difference in output if I use 'asterisk -rx "sip show settings" ' would there? Because that's what I'm checking... |
| 23:56.55 | [TK]D-Fender | I don't know what it'd take for you to see what you think you are expecting to see |
| 23:57.47 | apb1963 | which section are you seeing it in? |
| 23:58.37 | [TK]D-Fender | What is "it"? |
| 23:58.54 | [TK]D-Fender | You are talking about "it" as though this was a SINGLE settings |
| 23:59.04 | [TK]D-Fender | What SPECIFIC piece of information are you looking for? |
| 23:59.06 | file | the better question is - what are you expecting/wanting to see? |
| 23:59.37 | apb1963 | I'm looking for something that says "nat" |
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| 23:59.51 | file | and tells you... what |
| 23:59.54 | apb1963 | or, Network Address Translation perhaps |