| 00:00.00 | Trioxin | lvlinux, well I remember the freepbx interface being very helpful |
| 00:00.07 | [TK]D-Fender | Do you SEE the option? |
| 00:00.15 | [TK]D-Fender | If you SEE it... then it is there |
| 00:00.25 | [TK]D-Fender | And if you don't see it ... it's probably still there |
| 00:00.34 | [TK]D-Fender | ...and you just need to have your eyes checked |
| 00:00.39 | lvlinux | Trioxin: ehh, depends on what you are trying to do. Just for an office style PBX I can see that. For me FreePBX has always been a hindrance. |
| 00:00.44 | [TK]D-Fender | </optometry> |
| 00:01.12 | lvlinux | Trioxin: but if you want to get something setup very quickly it may help you if you aren't doing anything non-officepbxlike :-) |
| 00:01.29 | wyoung | [TK]D-Fender: I am wearing my glasses so I should be right |
| 00:01.47 | [TK]D-Fender | ods++ |
| 00:01.51 | lvlinux | But you'd probably want to go with more RAM if you wanted to use FreePBX. Not sure how much it would take though. Might want to ask them in #freepbx |
| 00:01.51 | [TK]D-Fender | odds++ |
| 00:02.11 | wyoung | [TK]D-Fender: c: If the Dial() application cancels this call, always set HANGUPCAUSE to 'answered elsewhere' |
| 00:02.19 | wyoung | Is that it? |
| 00:02.26 | wyoung | I need to give Dial the c flag? |
| 00:02.41 | [TK]D-Fender | "ANSWERED ELSEWHERE" |
| 00:02.45 | Trioxin | lvlinux, well, extensions, ivr, departments, voicemails, and I was thinking about maybe using goautodial (Don't remember if that's asterisk) since I need an autodialer. Either that or I'll use ytel subscription for our autodialer. |
| 00:03.43 | wyoung | [TK]D-Fender: and obviously I use that in my incoming call context :) |
| 00:03.49 | lvlinux | Trioxin: extensions, ivr, departments, voicemails all sounds like regular PBX stuff. Autodialer is different and I dont' know if FreePBX will play nicely with that or not. |
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| 00:04.25 | Trioxin | lvlinux, I'm assuming they could run on the same machine and different ports. if not then a second vps for goautodial |
| 00:04.39 | Trioxin | or share an asterisk install idk |
| 00:05.43 | lvlinux | I'm not familiar w goautodial so idk. But I know with FreePBX, doing anything with Asterisk outside of the FreePBX interface is a total nightmare. |
| 00:05.44 | Trioxin | actually it sounds kind of nice and easy to setup http://goautodial.org/ |
| 00:06.46 | [TK]D-Fender | wyoung, You obviously put it in any dial that goes to multiple you want that effect to apply to |
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| 00:08.06 | lvlinux | Trioxin: looks like goautodial is a whole shebang in a box like Trixbox, Elastic. So might look into just installing it first and see if it gives you everything you need. If not you'll probably have to set things up manually. |
| 00:08.19 | Trioxin | yea |
| 00:08.32 | lvlinux | gtg... good luck :-) |
| 00:09.36 | mzbotr | I'm trying to connect mysql to ODBC to asterisk 11.17 in gentoo (all cmdline). I have a connection up to the database, but asterisk -vvvcr will repeatedly report remote unix connection open/disconnected. |
| 00:10.01 | mzbotr | every few seconds. The last conection attempt is also shown to have happened in 1969... |
| 00:10.06 | mzbotr | and that's not right. |
| 00:10.51 | [TK]D-Fender | Trixbox has been dead for years, Elastix should be dragged out and shot. |
| 00:10.53 | mzbotr | Decrementing the verbosity level makes those messages go away. |
| 00:10.56 | [TK]D-Fender | Survivors should be shot AGAIN |
| 00:11.52 | [TK]D-Fender | unix != ODBC |
| 00:12.06 | [TK]D-Fender | that would be AMI <- |
| 00:14.21 | mzbotr | so it's not ODBC acting up, only asterisk? |
| 00:15.17 | wyoung | [TK]D-Fender: obviously |
| 00:17.04 | mzbotr | yeah.. I've had so many things go horribly wrong w/ asterisk's internals. I'd rather run from read-only sources on asterisk with a VM/iso |
| 00:17.30 | mzbotr | with tmpfs or qcow2 mounted on /var or something. |
| 00:24.57 | wyoung | [TK]D-Fender: in Dial(), instead of doing Dial(SIP/1000&SIP/1001&SIP/1002&SIP/1003&SIP/1004&SIP/1005&SIP/1006&SIP/1007&SIP/1008) can I use Dial(meGroup) or get the list of extensions to dial from another means, database or whatnot? |
| 00:25.07 | wyoung | (for incoming calls that is) |
| 00:25.36 | [TK]D-Fender | Dial requires you pass it TEXT |
| 00:25.52 | WIMPy | You can use a variable or a function. |
| 00:25.58 | [TK]D-Fender | How you get that text is up to you but it had better evaluate out like you had in your first example |
| 00:26.10 | [TK]D-Fender | eg: SIP/1000&SIP/1001&SIP/1002&SIP/1003&SIP/1004&SIP/1005&SIP/1006&SIP/1007&SIP/1008 |
| 00:26.22 | [TK]D-Fender | Whether you read that value from a DB, or whatever, dial doesn't care. |
| 00:26.38 | [TK]D-Fender | just make sure that's what you're feeding dial |
| 00:26.56 | wyoung | [TK]D-Fender: ah ok, I Might set it has a global variable |
| 00:26.59 | wyoung | for now |
| 00:27.18 | [TK]D-Fender | If you use it in several places then it might be an idea |
| 00:32.00 | wyoung | Do I need to ' or " enclose a string when setting a global variable? |
| 00:32.13 | [TK]D-Fender | no |
| 00:36.00 | mzbotr | has anyone here experience with confbridge? |
| 00:37.11 | WIMPy | Ask what you want to know, not meta questions. |
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| 00:57.49 | wyoung | WIMPy: I like that word, meta questions. I might use it :) |
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| 01:58.02 | Jesterboxboy | i have a problem with understanding how asterisk handles calls that are buys when i use the Dial() Application |
| 01:58.20 | Jesterboxboy | when i use Dial() and the line is state busy, does the caller get the busy tone? |
| 01:58.54 | Jesterboxboy | or do i have to do that myself with checking the DIALSTATUS and then invoking Busy() ? |
| 01:59.06 | WIMPy | It will continue with whatever follows in your dialplan. |
| 01:59.45 | WIMPy | If you just Hangup() (i.e. without parameter) it will keep the cause from the last Dial() attempt. |
| 01:59.52 | Jesterboxboy | so i invoke Dial() and afterwords execif(Dialstuatu |
| 02:00.29 | Jesterboxboy | i have the problem that people dont hear a busy tone when the line is busy |
| 02:00.37 | WIMPy | No need unless you want to modify something or try something else. |
| 02:01.26 | Jesterboxboy | i just want to here the caller to get the busy tone when the line is busy |
| 02:01.28 | WIMPy | If it's on a SIP device, that would generate the busy tone. |
| 02:01.52 | WIMPy | SIP doesn't have a concept of sending tones after a call failed. |
| 02:02.13 | Jesterboxboy | no its a sip server that just hangs up and gives back exit status 16 i think( or whatever busy is) |
| 02:02.40 | WIMPy | 17 |
| 02:03.06 | WIMPy | And that's exactely what's expected. |
| 02:04.42 | Jesterboxboy | okay, so if i want the user to hear a busy tone, i have to check for the response on the asterisk server and then give him a busy with busy(). |
| 02:05.04 | WIMPy | That won't make a difference. |
| 02:05.28 | WIMPy | Check the phones config. There might be an option to immediately cancel failed calls or not. |
| 02:06.23 | WIMPy | Or you could Answer the call from our dialplan and use PalyTones or something. But then the call would appear to the user as answered. |
| 02:08.04 | Jesterboxboy | okay thanks |
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| 02:42.05 | epinky | where do I set ttl for multicast paging? what's the default ttl value? |
| 03:05.06 | wyoung | My SIP provider allows 4 concurrent "lines" through it. Can I use SLA for this? Or do I need to do this another way? |
| 03:05.18 | ChannelZ | A channel is a channel |
| 03:05.31 | ChannelZ | How you want to manifest it on your side is up to you |
| 03:06.06 | wyoung | so do I need to create 4 entries in sip.conf to represent each "line" / channel? |
| 03:06.19 | wyoung | or does SLA use the call counter? |
| 03:06.28 | wyoung | or max-lines config? |
| 03:07.31 | wyoung | the only SLA examples I can find are for DAHDI |
| 03:07.49 | ChannelZ | Well SLA is kind of complicated |
| 03:08.17 | wyoung | ok, where do I start? :) |
| 03:08.21 | ChannelZ | As far as limiting channels, you can count yourself (which I believe is the current way? I think call counter is kind of old/broken but I could be wrong) |
| 03:08.51 | ChannelZ | Your ITSP may simply reject calls with a congestion signal if you're over the limit, so you can just let it work its self out like that if you wanted. |
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| 03:08.55 | wyoung | count myself, as in I increment and decrement a global variable? |
| 03:09.09 | ChannelZ | It just depends on what you want. And even what your devices are. Are you using phones with multiple line keys? |
| 03:09.37 | wyoung | ChannelZ: I am using snom phones which support SLA. In particular they turn a red light on the "line" if it is in use |
| 03:10.41 | ChannelZ | That's more BLF than SLA. SLA is treating a channel as a 'party line' where you could actually pick up an in-progress call from any phone. |
| 03:10.51 | ChannelZ | like an analog line |
| 03:10.54 | wyoung | ah ok |
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| 03:11.18 | rhineheart_m | hello! all of a sudden my asterisk process won't start up |
| 03:11.24 | wyoung | I guess I would like that behaviour too |
| 03:11.41 | wyoung | rhineheart_m: what does /var/log/syslog and / or /var/log/asterisk/messages say? |
| 03:12.17 | ChannelZ | You can use your line keys with BLF on 4 different SIP peers on each phone and sort of pretend they're analog lines for that purpose, with them actually being shared in the traditional analog sense |
| 03:12.47 | ChannelZ | but personally I got rid of analog lines to escape their limitations :) |
| 03:13.26 | wyoung | ok so the 4 sip peers will contain the same information to my ITSP. Do I need to set maxlines or anything like that or will it figure that out? |
| 03:13.27 | ChannelZ | I don't want/shouldn't really care about what "line" I'm using. |
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| 03:14.22 | ChannelZ | Again, that depends on your ITSP. Chances are they'll just reject calls if you're over your channel limit |
| 03:14.36 | wyoung | hmmmm, ok |
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| 03:15.30 | Tagor | Is there a way to disable the 'please say your name' prompt when using confbridge? |
| 03:15.32 | ChannelZ | In your out-dialing extension(s), you can put some logic in there to check the result of the Dial() and if it's CHANUNAVAIL or something, maybe play a more helpful prompt to the user ("sorry all channels are in use" or something) or just let them get a busy signal |
| 03:16.19 | wyoung | Tagor: what is conf bridge? |
| 03:16.41 | Tagor | wyoung: a module that replaced meetme |
| 03:17.48 | ChannelZ | hmm there's 'quiet' but I didn't think it prompted for names or anything (besides PIN if you have one) on conferences.. |
| 03:17.59 | wyoung | ChannelZ: hmmm, I think I tried that once before and CHANUNAVAIL was being returned for BUSY as well by my ISTP |
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| 03:18.16 | wyoung | Tagor: ah ok, I still use meetme |
| 03:18.18 | ChannelZ | yeah that might be true |
| 03:18.23 | wyoung | is confbridge better? |
| 03:18.47 | ChannelZ | yes |
| 03:18.57 | Tagor | wyoung: I never used meetme, so can't tell. but from what I read, yes |
| 03:19.28 | ChannelZ | oh... Tagor, announce_join_leave under your user section |
| 03:22.00 | Tagor | ChannelZ: ah I thought that would also disable the sound when someone is entering, but it doesn't |
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| 03:22.27 | rhineheart_m | wyoung, I have this : WARNING[16701] db.c: Couldn't create astdb table: disk I/O error |
| 03:23.25 | WIMPy | Yes, ConfBridge is a lot better in all but one very important point: MeetMe is faster. |
| 03:23.43 | ChannelZ | probably /var/lib/asterisk or somesuch is not writeable |
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| 03:27.54 | WIMPy | Hmm. Looks like there have been some serious changes to chan_dahdi since I last tried it. |
| 03:28.43 | Tagor | is there a way to get the current number of participants in a confbridge conference room without actually joining the conference? |
| 03:29.07 | ChannelZ | through AMI I think.. |
| 03:29.18 | WIMPy | The CONFBRIDGE function. |
| 03:29.25 | ChannelZ | oh.. there's also CONDBRIDGE_INFO function |
| 03:29.33 | ChannelZ | *CONF even |
| 03:29.49 | WIMPy | Or rather CONFBRIDGE_INFO, yes |
| 03:30.02 | Tagor | thanks that's what I needed! |
| 03:34.25 | WIMPy | If that famous "glare" "feature" has also been fixed, I think I might find DAHDI usable. |
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| 03:35.48 | linocisco | hi |
| 03:36.04 | WIMPy | lo |
| 03:36.17 | linocisco | WIMPy, contra |
| 03:36.35 | WIMPy | Bock |
| 03:36.52 | linocisco | !Bock |
| 03:38.04 | drmessano | Contra? |
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| 03:38.20 | drmessano | UUDDLRLRBA-Start |
| 03:39.32 | linocisco | hi , is there any CRM + Asteriks integration that offer notification of how many times user had call whenever a new call received and for why and recording all cases regarding issues.? I want to generate report everymonth like what are the most common issues/problems, who is the top 1-10 callers. etc? |
| 03:41.21 | linocisco | and also the CRM+Asterisk portal should be accessible anywhere from internet while being providing service with GSM or WCDMA or PSTN trunks in our server room |
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| 03:53.27 | wyoung | rhineheart_m: type in dmesg. Do you have any of ATA RESTART or BAD SECTOR or stuff like that in it?? |
| 03:54.26 | wyoung | WIMPy: I have created a peer entry in sip.conf to my ITSP. In their example asterisk conf they have insecure=invite,port. Is that bad? |
| 03:55.35 | WIMPy | The invite part is pretty normal for ITSPs. They hardly ever authenticate towards their customers. |
| 03:55.56 | rhineheart_m | wyoung, I didn't notice 1. |
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| 03:56.23 | wyoung | WIMPy: ok, can that be abused? |
| 03:56.54 | WIMPy | Everything can be abused. |
| 03:57.01 | wyoung | WIMPy: I would like to allow users to take their SIP phone home and connect into office which means opening the SIP port |
| 03:57.03 | rhineheart_m | wyoung, there's none. |
| 03:57.12 | wyoung | rhineheart_m: you have permission? |
| 03:57.28 | rhineheart_m | wyoung, I don't know. How to check? |
| 03:57.30 | WIMPy | You can use ost or permit/demy to restritct the peer to their servers. |
| 03:58.55 | wyoung | hmmmmm |
| 03:59.09 | WIMPy | s/ost/host/ |
| 04:00.38 | rhineheart_m | wyoung, I am using elastix |
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| 07:04.12 | wyoung | hey, in a queue, is it possible to play announcements over the top of the music on hold music? |
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| 09:45.54 | Harm133 | Hey, anybody has knowledge about auto provisioning a yealing SIP-T38G? I know this isn't the channel for this exact issue but if, by any chance, someone knows about this I'd like to PM with you :) |
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| 10:10.53 | wyoung | any ideas gang? |
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| 10:26.15 | wyoung | WIMPy!!!!!!!!111 |
| 10:53.26 | phpboy | Harm133: I run yealinks with auto provisioning |
| 10:53.34 | phpboy | what is your question? |
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| 11:05.01 | Harm133 | I have the normal cfg files all set up, everything works, except for when I push a autoconfig setting the account which handles my linekeys jumps from 1 to 2. I haven't found a line that corresponds with this setting within the cfg itself but in the webinterface it is adjustable. |
| 11:07.44 | phpboy | it's possible that your devices firmware doesn't support it or it's a bug |
| 11:14.55 | wyoung | phpboy: Do you know anything about queues? |
| 11:19.02 | *** join/#asterisk scinawa (~scinawa@2-230-246-32.ip204.fastwebnet.it) |
| 11:23.27 | phpboy | I've done a little bit here and there |
| 11:23.32 | phpboy | what's the question |
| 11:24.25 | phpboy | Harm133: with the Yealinks specifically, I've found that it either works or doesn't work. No in between |
| 11:24.38 | phpboy | I've also picked up on a firmware bug or two in the past |
| 11:29.47 | wyoung | phpboy: When the periodic message plays it pauses the music on hold music then resumes when the message is finished. Can I just lower the volume of the music on hold music and play the message over top then raise the moh music after teh message? |
| 11:30.55 | Harm133 | phpboy: Thanks, and sorry for my late response |
| 11:30.56 | phpboy | I can't say I've ever had to do that before... I'm sure google can answer that question though? |
| 11:31.10 | wyoung | phpboy: yeah google was my first stop |
| 11:31.12 | Harm133 | phpboy: I'll try and look for another firmware and see if it works there |
| 11:31.21 | wyoung | I have about 32 tabs open of research into this and I can't find anything on it |
| 11:32.15 | phpboy | Harm133: look, you need to confirm that it's a known bug... don't assume that upgrading your firmware will fix it |
| 11:33.02 | phpboy | wyoung: We had a similar challenge at a customer and ended up making the ad part of the on hold music |
| 11:33.19 | phpboy | this wasn't an astrisk system though |
| 11:34.01 | wyoung | hmmmmm |
| 11:34.28 | wyoung | I could stream the music on hold and inject the ads over the top of the music |
| 11:34.43 | phpboy | that's another option |
| 11:35.08 | phpboy | Theorhetically I think you'll have to do that |
| 11:35.39 | wyoung | now to find some streaming software that does that or I open up a text editor and write some python :) |
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| 12:21.11 | Puck` | hi everyone |
| 12:21.37 | Puck` | is anyone using an asterisk AMI PHP library? Or do you guys know of any functioning ones? |
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| 13:14.10 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 13:14.45 | [TK]D-Fender | Puck`: PHP-AGI has a class for it |
| 13:26.48 | wyoung | hi [TK]D-Fender! |
| 13:27.23 | wyoung | I would like to inject ads into music on hold. Can asterisk do this natively or do I need to roll my own? |
| 13:28.56 | wyoung | [TK]D-Fender: Actually I lie, what started this was I wanted periodic announcements to play over the music on hold music. Although I may want to inject ads at a later time |
| 13:28.57 | [TK]D-Fender | Your streaming idea is all you've got. |
| 13:29.06 | [TK]D-Fender | It does nt play over |
| 13:29.12 | [TK]D-Fender | you're on your own to hack this up |
| 13:29.28 | wyoung | You would think asterisk would have an option for this |
| 13:29.50 | wyoung | lower volume of MOH, overlay announcement, return MOH to original volume |
| 13:31.31 | *** join/#asterisk Draecos (~Draecos@203-121-194-11.e-wire.net.au) |
| 13:31.58 | wyoung | Instead I have to deal with MOH, announcement with no background sound, MOH |
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| 13:34.54 | wyoung | very clunky |
| 13:37.08 | *** join/#asterisk bluemerlin (~bluemerli@office-nat.westpoint.gradwell.net) |
| 13:37.32 | [TK]D-Fender | Or you mix your own with the announcements already integrated. |
| 13:37.42 | *** join/#asterisk bluemerlin (~bluemerli@109.224.221.82) |
| 13:39.37 | wyoung | like a chump |
| 13:41.54 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 13:42.22 | [TK]D-Fender | I'm a musician so that'd be "like a CHAMP". |
| 13:43.24 | *** join/#asterisk scinawa (~scinawa@93-33-34-187.ip42.fastwebnet.it) |
| 13:43.28 | scinawa | hi |
| 13:43.33 | wyoung | hmmmmmmmm |
| 13:44.09 | scinawa | is it possible to write a call file that just spawn an exension in a context, without spawning a call? I have to set P-Preferred-Identity header and I cant add it after the call has been dialed |
| 13:44.38 | [TK]D-Fender | use a Local Channel as your Channel: |
| 13:44.39 | *** join/#asterisk leedm777 (leedm777@nat/digium/x-oxpgudqbsatnabmf) |
| 13:47.15 | scinawa | [TK]D-Fender: like a dialing to local/extension@context and pass all the parameters I need there? |
| 13:47.25 | scinawa | sounds a good idea, i'll try it now. tnx |
| 13:47.38 | [TK]D-Fender | yes including doing the actual idal, etc yourself |
| 13:47.40 | [TK]D-Fender | dial* |
| 13:47.42 | *** join/#asterisk acidfu (~netvirt@unaffiliated/acidmen) |
| 13:47.44 | acidfu | moo |
| 13:48.21 | wyoung | [TK]D-Fender: so being a musical type of person you can understand my delema |
| 13:48.29 | acidfu | on asterisk 1.8, configure.ac use the macro AST_EXT_LIB_SETUP ... where is that defined ? I'm getting an error when I do ./configure |
| 13:48.35 | wyoung | nn |
| 13:48.52 | acidfu | (im running debian wheezy) |
| 13:49.57 | [TK]D-Fender | wyoung: Nope, I can mix. Not a dilemma for me :) |
| 13:50.34 | [TK]D-Fender | You should not be touching 1.8 at all |
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| 13:53.40 | acidfu | [TK]D-Fender, ... ? |
| 13:53.58 | acidfu | so anyway idea for the macro problem ? :) |
| 13:54.43 | [TK]D-Fender | Stop using 1.8 and try something supported |
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| 13:56.38 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:02.07 | acidfu | ok I found the problem, we should not do "autoreconf -fvi" before ./configure |
| 14:02.41 | [TK]D-Fender | And should still not be using 1.8 at all.... seriously... don't |
| 14:03.06 | acidfu | take it easy, if Im using 1.8 it's because I have my reasons |
| 14:04.47 | acidfu | but yeah I agree with you, when possible it's better to use the newest version |
| 14:04.57 | acidfu | and in the future that might be possible for me |
| 14:05.28 | [TK]D-Fender | Not a matter of "newest". 1.8 is completely EOL and not supported. |
| 14:06.01 | [TK]D-Fender | 1.8 is 4 entire branches ago |
| 14:08.53 | acidfu | thank you for the history course |
| 14:14.14 | [TK]D-Fender | Correct, 1.8 is HISTORY :) |
| 14:15.42 | lvlinux | 1.8 is still supported according to the topic. ?? |
| 14:16.17 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 14:16.51 | [TK]D-Fender | 1.8.X LTS released 2010-10-21 sec fix only 2014-10-21 EOL 2015-10-21 |
| 14:16.56 | [TK]D-Fender | 1.8 = BAI BAI |
| 14:18.19 | lvlinux | yup looks like. Well then malcolmd or file probably should change the topic I guess. |
| 14:19.10 | [TK]D-Fender | Topic for #asterisk was set by file!~file@asterisk/developer-and-muffin-lover/file on Friday, October 9, 2015 7:47:47 PM |
| 14:19.23 | *** topic/#asterisk by file -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.6.0 (2015/10/09), 11.20.0 (2015/10/09); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 14:19.28 | [TK]D-Fender | :) |
| 14:19.32 | lvlinux | :-) |
| 14:19.36 | [TK]D-Fender | tosses file a muffin |
| 14:23.47 | [TK]D-Fender | file: perhaps you could arrange to have that WIKI page updated as we've passed the tentative dates on some already? |
| 14:26.39 | [TK]D-Fender | Wow, a whole year for 14 now :) |
| 14:27.00 | file | remains to be seen |
| 14:34.05 | *** part/#asterisk Harm133 (2e2caf3e@gateway/web/cgi-irc/kiwiirc.com/ip.46.44.175.62) |
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| 14:39.20 | *** join/#asterisk MarkS- (~mark@unaffiliated/mark21) |
| 14:40.53 | MarkS- | Hello, are there known issues with connecting 2 asterisk servers using multiple sip accounts? all sip accounts should have calls going both ways and for internal reasons it is not possible to combine it into 1 sip trunk |
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| 14:45.01 | [TK]D-Fender | Why would you need multiple |
| 14:47.05 | acidfu | [TK]D-Fender, let me help you for internal reasons it is not possible to combine it into 1 sip trunk |
| 14:47.35 | MarkS- | [TK]D-Fender: 1 server is used by multiple companies that each need their own account/trunk (different persons managing a certain part at that end, different invoices, legal reasons, etc.). if it is not possible I have to setup an extra asterisk server to go between the 2 servers for 1 account |
| 14:48.23 | [TK]D-Fender | just tag the calls |
| 14:49.21 | MarkS- | not something they seem to be willing to do :( they require seperate sip trunks |
| 14:49.35 | MarkS- | If it is possible to change the port for only one of the 2 it would probably work |
| 14:52.27 | [TK]D-Fender | Set them as type=friend on both and it should work. |
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| 14:54.31 | MarkS- | even with 2 times the same ip:port combination? that would be something that is easy to test/do |
| 14:56.10 | [TK]D-Fender | match by USER |
| 14:57.53 | MarkS- | [TK]D-Fender: Thank you for the pointer! Now looking for some documentation so I can get everything configured |
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| 16:15.18 | eth01 | hi, please can somebody tell me what the command 'asterisk -rx "core show calls"' actually shows? |
| 16:15.35 | eth01 | is it internal calls, or internal calls plus external inbound/outbound on our SIP trunk? |
| 16:15.48 | eth01 | many thanks |
| 16:16.54 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 16:17.07 | [TK]D-Fender | All calls |
| 16:17.07 | WIMPy | Asterisk has no concept of internal or external calls. A call is a call. |
| 16:17.10 | [TK]D-Fender | a call is a call is a call |
| 16:17.23 | *** join/#asterisk vader- (~Adium@50.232.174.194) |
| 16:17.47 | eth01 | could a call be a call into a telephone IVR , and if it rang multiple extensions in some kind of ring group, would it still represent one call? |
| 16:18.21 | [TK]D-Fender | A call is either 1 channel tht is unbridged, or 2 that are |
| 16:18.27 | [TK]D-Fender | It could be ANTYHING |
| 16:18.32 | [TK]D-Fender | Sitting in a prompt |
| 16:18.33 | [TK]D-Fender | Voicmail |
| 16:18.37 | [TK]D-Fender | bridged call |
| 16:18.45 | [TK]D-Fender | there is no such thing as in vs out |
| 16:18.54 | [TK]D-Fender | Could be sitting in a queue |
| 16:18.56 | [TK]D-Fender | or anywhere |
| 16:19.39 | [TK]D-Fender | 3 channels : #1 = SIP phone is in Voicemail(). #2 is another SIP phone that dialed a # that the dialplan bridged out to a carrier.. 3 channels. 2 calls |
| 16:20.12 | [TK]D-Fender | carrier leg there is also a channel. because they are bridged, those 2 = 1 call |
| 16:20.43 | WIMPy | channels - bridged channels = calls |
| 16:20.50 | eth01 | ok ok ok ok i get it |
| 16:20.59 | eth01 | very helpful |
| 16:21.04 | eth01 | ty |
| 16:22.27 | *** join/#asterisk Voyage (~user1@unaffiliated/voyage) |
| 16:22.51 | Voyage | What are the recommended ways to send and receive sms via asterisk? |
| 16:23.26 | WIMPy | Use SMS software. |
| 16:23.27 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-lfneabrkqcxkafqu) |
| 16:23.32 | [TK]D-Fender | Nothing is "recommended". |
| 16:23.43 | Voyage | WIMPy, Do you mean some SMS software other than asterisk? |
| 16:23.48 | [TK]D-Fender | the only ways that are supported are some European R1's, or SIP MESSAGE |
| 16:23.58 | eth01 | hopes that voyage doesn't work for voyage |
| 16:24.02 | WIMPy | Yes |
| 16:24.10 | [TK]D-Fender | E1* |
| 16:24.21 | Voyage | WIMPy, so there is no built-in way in asterisk? |
| 16:24.42 | Voyage | [TK]D-Fender, is E1 and Sip message not an SMS? |
| 16:24.42 | [TK]D-Fender | [11:23][TK]D-Fenderthe only ways that are supported are some European E1's, or SIP MESSAGE |
| 16:24.42 | WIMPy | Well that ETSI SMS stuf is actually analog modem shit. |
| 16:24.42 | eth01 | can't you use a SMS gateway box? |
| 16:24.50 | eth01 | filled with sim cards ? |
| 16:24.53 | Voyage | [TK]D-Fender, I read it the first time. |
| 16:25.05 | [TK]D-Fender | Then why did you ask again?> |
| 16:25.14 | Voyage | [TK]D-Fender, for clarity |
| 16:25.17 | [TK]D-Fender | [11:24]VoyageWIMPy, so there is no built-in way in asterisk? <- I jsut TOLD you there were 2 ways |
| 16:26.10 | WIMPy | The funny thing is that the GSM operators uesed to have V.110 or X.75 access to their SMSCs before SMS in the PSTN came up. |
| 16:26.15 | Voyage | [TK]D-Fender, no, you didn't "there is no facility in * to send normal SMS" |
| 16:26.43 | WIMPy | Voyage: What kind of SMS do you want to use? |
| 16:26.49 | Voyage | and I dont know if Sim or E1 can be used to send sms |
| 16:27.02 | Voyage | WIMPy, normal text message that we type from cell phone |
| 16:27.12 | Voyage | s/sim/sip |
| 16:27.13 | [TK]D-Fender | Voyage[TK]D-Fender, no, you didn't "there is no facility in * to send normal SMS" <- I did not say this |
| 16:27.19 | [TK]D-Fender | You asked if there was a way |
| 16:27.21 | [TK]D-Fender | I told you 2 |
| 16:27.27 | [TK]D-Fender | this is crystal clear |
| 16:27.50 | WIMPy | Voyage: And how do you want to send them? With some kind of GSM MT? Over a land line? Over some ITSP? |
| 16:27.56 | [TK]D-Fender | This is option #1 |
| 16:27.57 | Voyage | [TK]D-Fender, I dont know about technical terms for now. If something can help me send SMS, I will read about it. Thats my clear question "is there any way to send sms via *" |
| 16:27.58 | [TK]D-Fender | This is option #2 |
| 16:28.07 | [TK]D-Fender | and I said YES |
| 16:28.16 | Voyage | WIMPy, I have an account with flowroute as an ITSP. |
| 16:28.45 | Voyage | WIMPy, is there a possibility that it might help me? |
| 16:29.31 | WIMPy | I don't really know. The SIP stuff might be SMS or something else. It's some rather shady term there. |
| 16:29.53 | Voyage | hm. so what should I know |
| 16:29.57 | Voyage | or start with |
| 16:30.34 | WIMPy | 1. Does their service do what you want? |
| 16:30.50 | WIMPy | 2. Does their Service work with what Asterisk supports? |
| 16:30.52 | Voyage | WIMPy, I came here to know what technical terms should I use to ask them |
| 16:31.07 | Voyage | WIMPy, flowroute is providing good voice calls service to me. |
| 16:31.13 | Voyage | but I am not sure about SMS |
| 16:32.45 | Voyage | it would be good if I would be able to send SMS during a call with some certain conditions in dialplan |
| 16:33.13 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 16:33.39 | WIMPy | I've only tried direct access to SMSCs or via http (the easiest and best for higher volumes). Not sure it's available via SIP here anywhere at all. |
| 16:34.19 | Voyage | yes likely my worry |
| 16:34.49 | Voyage | I think a) I can't do sms via normal ITSP and sip b) asterisk don't have a SMS facility built-in |
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| 16:39.02 | Voyage | will be back |
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| 17:33.30 | Voyage | so it seems there is no proper way in * to send sms? |
| 17:34.09 | paraxor | oh! * = asterisk |
| 17:34.12 | paraxor | I just got that |
| 17:34.34 | Voyage | :) |
| 17:34.42 | [TK]D-Fender | Yes, it can |
| 17:34.43 | Voyage | yes |
| 17:34.45 | [TK]D-Fender | I told you 2 ways |
| 17:34.55 | [TK]D-Fender | many times. Over several days. |
| 17:35.00 | [TK]D-Fender | And we discussed carriers. |
| 17:35.13 | Voyage | [TK]D-Fender, those 2 ways can't send sms. If I am correct |
| 17:35.14 | [TK]D-Fender | This was answered literally weeeks ago |
| 17:35.21 | [TK]D-Fender | I told you they can. |
| 17:35.23 | [TK]D-Fender | repeatedly |
| 17:36.01 | Voyage | <WIMPy> Well that ETSI SMS stuf is actually analog modem shit. |
| 17:36.55 | Voyage | [TK]D-Fender, do I need to have special support from ITSP sip provider to use ETSI or SIP messaging to send sms (the 2 ways you told me 'repeatedly' that can send message)? |
| 17:37.14 | Voyage | or just normal sip ITSP service for calling will be suffice? |
| 17:37.33 | [TK]D-Fender | Yes, your provider obviously has to support SIP messaging... in order for you to send them SIP messages |
| 17:37.35 | [TK]D-Fender | DUH |
| 17:37.38 | [TK]D-Fender | We'd already been over that |
| 17:37.46 | [TK]D-Fender | aAND looked at the carrier offers for this |
| 17:37.49 | Voyage | and about ETSI? |
| 17:38.03 | [TK]D-Fender | You said you want to send SMS message |
| 17:38.08 | Voyage | yes |
| 17:38.10 | [TK]D-Fender | sSIP messaging allows this with providers who support it |
| 17:38.13 | [TK]D-Fender | The end |
| 17:38.17 | Voyage | and i said I am not,yet, familier with tech terms |
| 17:38.18 | [TK]D-Fender | https://botbot.me/freenode/asterisk/search/?q=SMS |
| 17:38.23 | Voyage | ok |
| 17:41.15 | Voyage | Does anyone knows, as a non-google, reputed recommendation that allows sip messaging? |
| 17:41.32 | [TK]D-Fender | huh? |
| 17:42.07 | Voyage | "any recommendations on providers that provide sip messaging support" |
| 17:43.38 | [TK]D-Fender | lots of providers. |
| 17:43.58 | Voyage | yes, only need the reputed ones with good rates |
| 17:44.10 | [TK]D-Fender | VApparently voip.ms does |
| 17:44.19 | [TK]D-Fender | which means upstream vitelity should as well |
| 17:46.02 | Voyage | hm. not flowroute it seems for now |
| 17:46.09 | Voyage | thanks. |
| 17:46.26 | Voyage | will check voip.ms and upstream and vitelity |
| 17:46.56 | [TK]D-Fender | "upstream" was not a name |
| 17:47.08 | [TK]D-Fender | voip.ms is a vitelity reseller |
| 17:47.18 | Voyage | oh |
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| 20:16.11 | K0HAX | Is there a limit to the number of simultaneous faxes with res_fax_spandsp in the current Git tree? |
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| 20:18.05 | [TK]D-Fender | Not inherently |
| 20:18.22 | K0HAX | It's not licensed like the res_fax_digium module was though, right? |
| 20:18.32 | K0HAX | where you had to pay for each channel |
| 20:19.07 | K0HAX | (I don't think it is, as I never had to install a license and I can use it) |
| 20:19.17 | [TK]D-Fender | no license |
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| 20:19.40 | K0HAX | :) That's what I thought. |
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| 20:21.22 | pjensen00 | When a blind transfer happens to a channel in the Stasis app, the two local channels spawn. One of them goes to my default dialplan which puts it goes back into the Stasis app. |
| 20:21.39 | pjensen00 | When I get the channel's information I don't see anything indicating it was a result of a SIP level refer. |
| 20:21.57 | pjensen00 | Is there a way to query a channel to see if it was created as a result of a SIP level xfer? |
| 20:25.19 | robmal | I don't know if it's the best way but i use CEL for this. |
| 20:28.19 | pjensen00 | Hrm. |
| 20:29.22 | pjensen00 | with chan_sip it'd just be a new sip channel which had the sip header SIPREFER in it. Obviously on LOCAL channels I can't be querying that kind of information. |
| 20:30.04 | robmal | Please check CEL specification in the wiki. |
| 20:30.29 | pjensen00 | aight |
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| 20:36.18 | pjensen00 | "THIS IS NO LONGER TRUE REWRITE" is not a good sign when reading the documentation for CEL |
| 20:37.28 | robmal | https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CEL+Specification |
| 20:37.30 | robmal | Try this. |
| 20:39.15 | igcewieling1 | gawd, please don't mess with CEL. |
| 20:39.42 | pjensen00 | I'm also shopping in the dialplan functions to see if there is something I can use out of there too |
| 20:41.10 | robmal | igcewieling1: Why? |
| 20:42.19 | rrittgarn | Does the new res_parking support realtime? Tried and failed to configure it in extconf |
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| 20:48.14 | caseyd | I have a copy of asterisk running, and I usually have asterisk -r up. I tried asterisk -rd for debug, and now it won't quit debugging. Even if I exit and go back to asterisk -r.. i figure its something simple.. can anyone help? |
| 20:48.59 | [TK]D-Fender | turn it off at CLI |
| 20:49.07 | [TK]D-Fender | core set debug 0 |
| 20:49.15 | [TK]D-Fender | if it's core (which I suspect that means) |
| 20:50.02 | caseyd | that was it, thanks! |
| 20:51.18 | [TK]D-Fender | You're welcome |
| 21:01.06 | igcewieling1 | robmal: because it works fine the way it is. |
| 21:02.35 | robmal | igcewieling1: I know, i'm just suggesting it has lots of useful info about transfers. |
| 21:04.04 | igcewieling1 | I was replying to "pjensen00: "THIS IS NO LONGER TRUE REWRITE" is not a good sign when reading the documentation for CEL" 8-) |
| 21:04.35 | igcewieling1 | I love CEL. It allowed me to build the sorts of logging I wanted. |
| 21:05.34 | robmal | Which is? ;-> |
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| 21:17.39 | ericNL | I need to create a conference room that calls (invites) 4 external numbers everyday at a certain time to invite them into the conference room. They need to press 1 to confirm, otherwise they won't be added to the conference. Something like that possible? |
| 21:18.54 | robmal | Yes. |
| 21:19.13 | ericNL | Any example scripts around or? |
| 21:19.16 | robmal | No. |
| 21:19.18 | ericNL | :) |
| 21:19.48 | robmal | Easiest way to do it is via cron and callfiles |
| 21:20.10 | ericNL | Place a copy of 4 callfiles everyday at time X in place to setup the calls. |
| 21:20.22 | [TK]D-Fender | yup |
| 21:20.31 | ericNL | follow me for call confirmation i think |
| 21:21.04 | robmal | Originate to an extension, which does playback and waits for dtmf, then jumps to confbridge/meetme if dtmf==1 |
| 21:22.32 | ericNL | I'll get to it, awesome. Thanks! |
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| 21:29.34 | qakhan | i am trying to config fsx port, when i connect the analog phone there is no dial tone. here is my config http://pastebin.com/mLuUKxfp |
| 21:31.09 | wyoung | qakhan: have you connected the molex power cable up to the card? |
| 21:31.22 | wyoung | Or are you using a different type of FXS / FXO card? |
| 21:31.40 | wyoung | Or an ATA? |
| 21:31.57 | qakhan | its Digium Wildcard A8B |
| 21:32.17 | wyoung | [TK]D-Fender: Can you teach me to mix? |
| 21:32.38 | [TK]D-Fender | http://sourceforge.net/projects/audacity/ |
| 21:32.48 | wyoung | In realtime? |
| 21:32.49 | [TK]D-Fender | get to it.. |
| 21:32.54 | [TK]D-Fender | heads home |
| 21:32.58 | wyoung | :O |
| 21:33.25 | wyoung | qakhan: ummm I am not familar with that card but I am assuming you have installed the DAHDI modules for it? |
| 21:34.03 | qakhan | yes this card has 4 FSX and 4 FXO module. i am getting calls on FXO lines |
| 21:34.26 | qakhan | but there is no dial tone on FSX lines |
| 21:34.55 | wyoung | qakhan: have you connected the molex power cable up to the card? |
| 21:35.48 | wyoung | qakhan: you have alot of channels there |
| 21:37.14 | qakhan | i have to check |
| 21:37.43 | qakhan | if i dont have power connected to it then that is the reason? |
| 21:39.01 | malcolmd | if you don't have the power supplied to the molex connector, then, if fas modules are to be used, they won't have the necessary power to provide battery or ringing voltage to the attached telephones. tl;dr, you need the molex for fxs |
| 21:40.07 | qakhan | ok and this also a reason that when i connect analog phone i dont see any alarm on cli that port is up |
| 21:40.08 | qakhan | ? |
| 21:48.09 | drmessano | Yes |
| 21:48.14 | drmessano | Power that molex |
| 21:51.55 | qakhan | ok i will check let you guys know |
| 21:55.56 | MaliutaLap | drmessano: I never liked that Mole X guy. I voted for Mole Y ;) |
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| 21:59.11 | [TK]D-Fender | MaliutaLap, https://www.youtube.com/watch?v=rc5G04nJecI |
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| 22:05.44 | lvlinux | Hmm, I have one server that is constantly (every few minutes) saying that "Peer 'voipms' is now Lagged. (2089ms / 2000ms)" and then "Peer 'voipms' is now Reachable. (68ms / 2000ms)" |
| 22:05.57 | lvlinux | Using chan_sip |
| 22:06.18 | lvlinux | How should I go about fixing that? |
| 22:08.04 | lvlinux | I have other Asterisk servers registering to the same voip.ms server so I'm pretty certain that there isn't a problem with them. |
| 22:08.30 | lvlinux | And the other servers don't report any issues with being lagged or not |
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| 22:42.06 | wyoung | ok next problem. When my queue retries my handsets I am getting multiple missed calls registered on my handset. Can I configure the queue to send a 'handled elsewhere' or equiv header instead? |
| 22:54.35 | [TK]D-Fender | "core show application queue" <- |
| 23:00.07 | wyoung | [TK]D-Fender: <3 |
| 23:02.23 | wyoung | The correct answer is C |
| 23:02.54 | [TK]D-Fender | Sounds like you're on a solid path to passing every multiple-choice standardized exam. |
| 23:03.17 | wyoung | wooooo!! |
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| 23:29.00 | igcewieling1 | A good exam woud have questions like "3:30pm, random calls get audio quality problems, there are two people calling to find out what the hell is going on. What do you do?" |
| 23:30.47 | file | drink? |
| 23:31.07 | WIMPy | Wait until rush hour is over and try again. |
| 23:31.12 | file | blame a saturated Level3 interconnect |
| 23:32.34 | pjensen00 | is there a way to persist channel variables when doing a SIP transfer? I know you can "__variable" to have that variable persist over to the next channel it creates. I was wondering if there was something similar for when a channel blind transfers |
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| 23:33.13 | MaliutaLap | tell each user it was the fault of the other, and then go to the pub |
| 23:35.31 | igcewieling1 | file: both are very good answers. |
| 23:35.33 | pjensen00 | Real life example, we flew one of our people out to the customer's physical site to troubleshoot audio quality issues. Turns out, people were chewing on the cords, stuffing the mic almost inside their mouths etc. Also the computer struggled to have an IE window open and the softphone at the same time. Needless to say, our man out there found a closet full of unused computers way more powerful than the call center's agent's computers. |
| 23:35.33 | pjensen00 | <PROTECTED> |
| 23:37.29 | robmal | What about the cord chewing part? Did you give them soothers? |
| 23:38.36 | WIMPy | Poisen the cords and wait for Evolution. |
| 23:38.59 | pjensen00 | They didn't last much longer. For... reasons. |
| 23:39.00 | WIMPy | Obviousely won't work for creationists, however. |
| 23:39.35 | pjensen00 | Well, it will. But more of a Jonestown.... ascension |
| 23:41.45 | pjensen00 | One fun one was complaints that their speaker had a rattling sound in the audio. This was a drive through speaker in Texas. Flew the same guy out to look at it. First thing he sees when he opens it is the cooling fan bouncing around inside. It fell off whatever mounting and was dangling by the power cable. |
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| 23:59.19 | wyoung | Context 'default' tries to include nonexistent context 'featuremap' |