| 00:02.28 | *** join/#asterisk threesome (~threesome@ip-94-113-13-143.net.upcbroadband.cz) |
| 00:07.18 | *** part/#asterisk navaismo (~navai_000@189.241.90.55) |
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| 00:53.56 | volga629 | Hello Everyone, I did small dial plan to send messages through sip, trying fix issue with CID. When message arrive remote end they don't see original sender to reply, only trunk ID |
| 00:54.16 | *** join/#asterisk Linkforsoad (~Linkforso@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 00:54.42 | volga629 | https://fpaste.networklab.ca/yO2r/ |
| 00:55.14 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 00:55.14 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 00:56.21 | volga629 | any help tnk |
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| 02:05.37 | raden | is it normal for sip show peers to show the router IP for externally registered phones ? |
| 02:05.42 | raden | everyhting is working |
| 02:06.00 | raden | changed from cisco to mikrotik router and now asterisk shows 172.17.1.1 instead of the external IP like it used to |
| 02:07.39 | Penguin | It is not normal for your own router address to be there. |
| 02:08.13 | Penguin | If you're talking about a peer on the other side of the public internet, showing your own local router for the peer's address doesn't make any sense. |
| 02:09.52 | volga629 | It depend if NAT enabled from outside |
| 02:10.06 | Penguin | No it wouldn't. |
| 02:10.39 | volga629 | on router |
| 02:11.20 | Penguin | If you registered your phone to my asterisk, there is nothing you could do with a nat setting to make my router's IP address show up for your phone's address. |
| 02:13.48 | volga629 | external client should be they public ip show as in registration, if on router enabled nat from Outside which is not correct then it will use LAN gw ip on router and that what you will see on in asterisk |
| 02:14.13 | Penguin | Never. |
| 02:14.18 | Penguin | It has never worked that way. |
| 02:14.45 | Penguin | It will either show the phone's private address or the router's public address. |
| 02:15.13 | Penguin | It will never show the router's gateway private address. |
| 02:16.04 | volga629 | then it router configuration wrong |
| 02:16.13 | Penguin | There is also a chance that he did a terrible job explaining the problem and he didn't mean what I think he meant. |
| 02:16.35 | volga629 | which cisco device it is ? |
| 02:17.18 | Penguin | "changed from cisco to mikrotik" |
| 02:17.28 | volga629 | if it asa then need disable all sip filtering because by default it enabled for cisco voice manager and acting as proxy |
| 02:17.43 | Penguin | It's not Cisco. |
| 02:19.23 | *** join/#asterisk jasonwert (~w3rt@71-89-135-86.static.aldl.mi.charter.com) |
| 02:20.59 | volga629 | I have some issue with dial plan to send messages through SIP https://fpaste.networklab.ca/yO2r/ can't get set CID that remote side can reply |
| 02:21.40 | volga629 | it always set trunk ID and delivery message |
| 02:32.03 | *** join/#asterisk cradek (~chris@emc/board-of-directors/cradek) |
| 02:37.32 | cradek | Hi all! I found the telephony card selector, but not a list of supported hardware. I found some cards (Dialogic D/4PCI) cheap and wonder how to find out whether they are supported for interfacing analog phones. |
| 02:49.21 | *** join/#asterisk RaNa (~WinNT@97-71-95-51.res.bhn.net) |
| 02:49.55 | RaNa | hey guys i have a bit of a problem when im trying to run a command |
| 02:50.02 | RaNa | tcpdump -s 3000 -C 10 -W 50 -w ~/sip.pcap port 5060 or portrange 10000-35000 |
| 02:50.15 | RaNa | tcpdump: /root/sip.pcap00: Permission denied |
| 03:06.56 | phix | cool |
| 03:17.13 | *** join/#asterisk linocisco (~linocisco@193.134.242.12) |
| 03:17.42 | linocisco | hi |
| 03:18.08 | linocisco | hi all |
| 03:19.40 | phix | werd |
| 03:20.43 | linocisco | asterisk HD videos on youtube is very few |
| 03:42.35 | ChannelZ | It's not really a visual product. |
| 03:48.00 | WIMPy | I didn't know dialogic did analog stuff. |
| 04:00.26 | volga629 | I am working setup SIMPLE. I can send message through trunks no problem, but I can't get user status |
| 04:01.02 | *** join/#asterisk brutal_lobster (~brutal_lo@sapmail.pbconsulting.ru) |
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| 04:02.31 | *** join/#asterisk brutal_lobster (~brutal_lo@sapmail.pbconsulting.ru) |
| 04:02.51 | volga629 | any suggestion what need to be done ? |
| 04:04.47 | *** join/#asterisk brutal_lobster (~brutal_lo@sapmail.pbconsulting.ru) |
| 04:09.03 | phix | linocisco: True, but there is alot of text and writing examples on the subject. |
| 04:09.45 | phix | linocisco: If you are having issues with anything specific then let us know :) |
| 04:11.38 | linocisco | just want to know how to configure HD video conferencing with asterisk |
| 04:15.33 | volga629 | I see I can do with corosync is any other way it possible ? |
| 04:16.43 | phix | linocisco: And what equipment do you have? just computers / tablets / smart phones? or specialised devices? VoIP phones with video support? |
| 04:17.08 | linocisco | phix, just PC with webcam. zoiper installed |
| 04:19.22 | *** join/#asterisk benklop (~quassel@2001:470:f400:47:ae81:12ff:fe31:668b) |
| 04:19.29 | benklop | hello! |
| 04:20.20 | phix | ok, what protocols does zopier support? |
| 04:20.22 | benklop | i've set up a very simple asterisk box with a couple extensions, and I'm sure I've got somthing wrong, because i'm getting a 401unauthorized whenever i try to call from one extension to the other |
| 04:20.52 | phix | wrong password? |
| 04:21.41 | benklop | phix: if i was using the rong password, would my sip client be able to register successfully? |
| 04:22.01 | benklop | *wrong |
| 04:22.56 | phix | nope |
| 04:23.42 | phix | hmmmmm |
| 04:25.23 | benklop | phix: i can't even call an extension that just plays a sound at me |
| 04:25.36 | ChannelZ | Look at the console. It's telling you why. |
| 04:25.40 | ChannelZ | core set verbose 3 |
| 04:25.46 | phix | yay ChannelZ! |
| 04:26.13 | benklop | hmm |
| 04:26.15 | benklop | just a sec |
| 04:29.45 | benklop | phix: here's hat it logged about that: http://pastebin.com/4z6biSVk |
| 04:29.55 | benklop | i don't doubt it's telling me, but I just dont see it |
| 04:31.48 | ChannelZ | Found peer '101' for '101' from 192.168.45.3:5060 |
| 04:32.18 | ChannelZ | Assuming that's the peer you're intending, it seems like a bad password. |
| 04:32.43 | ChannelZ | though if you had verbose on you'd probably see an auth message of some sort. |
| 04:33.02 | benklop | 101 is the peer i called from |
| 04:33.52 | benklop | and, I tried re-entering the password, and I've definitely got the right one because if i put in the wrong one the thing won't even let me try to make a call |
| 04:34.11 | benklop | ChannelZ: that was on core set verbose 3 |
| 04:34.40 | *** join/#asterisk vinhdizzo (~vinh@cpe-98-154-220-20.socal.res.rr.com) |
| 04:35.49 | ChannelZ | let's see your sip.conf |
| 04:37.30 | snadge | i have discovered that digium packages asterisk for both debian/ubuntu and centos |
| 04:37.50 | snadge | but not fedora.. fedora comes with the latest 11.5.. should i just use that? |
| 04:38.06 | Penguin | And before you go any further, I strongly suggest that you read the README-SERIOUSLY.bestpractices.txt |
| 04:38.13 | benklop | ChannelZ: just a sec |
| 04:38.14 | snadge | or should i download the .tar.gz and build it? |
| 04:38.14 | ChannelZ | Or use the distro you want and build the asterisk version you want. |
| 04:38.43 | Penguin | snadge: I thought we already covered this. |
| 04:39.25 | snadge | we probably have.. i was under the impression that there were official asterisk pacakges from digium for fedora |
| 04:39.48 | snadge | but it seems this is the case for rhel/centos 6 only |
| 04:39.57 | Penguin | benklop: README-SERIOUSLY.bestpractices.txt was for you. |
| 04:40.26 | Penguin | It's exactly the same. They provide RPMs in the repo. |
| 04:40.37 | benklop | Penguin: hmm. well then I'd better read it. |
| 04:40.54 | snadge | the fedora people have basically chastised me for wanting to install asterisk from source, and said just use the rpm |
| 04:41.00 | benklop | ChannelZ: http://pastebin.com/Wr5tcH0w |
| 04:41.06 | Penguin | snadge: I said that, too. |
| 04:41.19 | snadge | ~. |
| 04:41.20 | infobot | methinks ~. is not the escape sequence you're looking for. |
| 04:42.19 | benklop | Penguin: reading |
| 04:43.44 | ChannelZ | did you add those // lines just in your paste or are they really there? |
| 04:44.55 | benklop | ChannelZ: they are really there |
| 04:45.29 | benklop | ChannelZ: there are also some additional extensions, but they are all like that one and I have no logged phones into them currently |
| 04:46.11 | benklop | erm, extension is probably not the term, but you get what i mean |
| 04:46.22 | *** join/#asterisk NoobSaibot (~n00b_Saib@24-213-19-74.static.mrqt.mi.charter.com) |
| 04:46.29 | linocisco | hello |
| 04:46.38 | NoobSaibot | hi |
| 04:46.41 | ChannelZ | well hopefully the parser is just skipping them and not barfing. // are not legitimate comments in asterisk configs |
| 04:46.54 | ChannelZ | does 'sip show peers' actually show 101? |
| 04:46.57 | benklop | ChannelZ: i just realized that and am now replacing them with ; |
| 04:49.02 | benklop | ChannelZ: http://pastebin.com/7Px4sfR3 |
| 04:49.56 | ChannelZ | you have two peers with the same IP and port |
| 04:50.01 | ChannelZ | 101 and 102 |
| 04:50.09 | linocisco | hello |
| 04:50.24 | ChannelZ | OH HAI! |
| 04:51.57 | ChannelZ | benklop: Do you have 2 softphones running on the same machine or something? |
| 04:52.46 | benklop | ChannelZ: those are both on the same NA-PAP2T adapter |
| 04:53.24 | Penguin | North America! |
| 04:53.33 | Penguin | Revision PAP2T |
| 04:54.25 | ChannelZ | benklop: they should be different ports then |
| 04:55.01 | benklop | ChannelZ: okay, let me just disable one of the lines for the moment. I'm not certain how they got set that way, the thing is pretty much at its defaults |
| 04:56.22 | benklop | ChannelZ: i just changed the second to use a 5061 |
| 04:57.43 | benklop | ChannelZ: the one that says unmonitored is the line i'm trying to use |
| 04:58.06 | ChannelZ | the '101' one |
| 04:58.12 | benklop | yeah |
| 04:58.29 | ChannelZ | Unmonitored is not necessarily an error |
| 04:58.46 | Penguin | I don't think it's even possibly an error. |
| 04:58.52 | benklop | re-ran sip show peers and now 102 shows 5061 |
| 04:58.58 | Penguin | It just means you do not have qualify enabled on it. |
| 04:58.59 | ChannelZ | It would be if you told it to monitor I suppose :) |
| 04:59.31 | ChannelZ | When you changed the port, it rebooted right? Did you see 101 register on the console? |
| 05:00.53 | benklop | hmm i'll have to reboot and see |
| 05:03.09 | benklop | ChannelZ: i think so, but there's enough crap flying by its hard to tell |
| 05:03.22 | ChannelZ | turn off sip debug if it's still on |
| 05:03.28 | ChannelZ | sip set debug off |
| 05:06.06 | benklop | yes, it does show the registration |
| 05:06.58 | ChannelZ | ok.. so do you still get rejected? |
| 05:07.28 | benklop | yeah, it appears so. |
| 05:08.10 | benklop | double checking |
| 05:08.26 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-hemnfvthzxegskxx) |
| 05:08.27 | ChannelZ | and the console is saying NOTHING when you try? |
| 05:08.29 | benklop | the phone doesn't tell me, just beeps. i have to look in the log to find the 403 |
| 05:09.02 | *** join/#asterisk taylorbyte2013 (~cyberninj@139.218.237.26) |
| 05:09.24 | benklop | ChannelZ: logging is still pretty much exactly hat i pasted earlier |
| 05:09.30 | benklop | do you want me to re-paste? |
| 05:11.03 | ChannelZ | that was sip debug - which the first time I didn't see the initial invite and auth rejection.. but with sip debug off it should spit out _something_, a note about the rejected auth.. |
| 05:11.29 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 05:12.19 | ChannelZ | actually.. |
| 05:13.02 | ChannelZ | look at /etc/asterisk/logger.conf for a console= line.. if it's commented out, uncomment it and then on the console do 'logger reload' |
| 05:13.24 | benklop | just a moment |
| 05:14.47 | benklop | i don't have that file currently, I only hve barebones config fies so it uses defaults for everything else |
| 05:14.56 | benklop | i'll put it in from the sample tho |
| 05:17.27 | benklop | ok, that file is in place, console line is there uncommented |
| 05:17.37 | benklop | config reloded |
| 05:17.43 | benklop | i'll try again in just a moment |
| 05:19.56 | benklop | AHA, thank you, i feel dumb for not having logging enabled like i sohlud have. |
| 05:20.23 | benklop | [Oct 3 23:19:15] ERROR[27015][C-00000006]: rtp_engine.c:259 ast_rtp_instance_new: No RTP engine was found. Do you have one loaded? |
| 05:20.39 | benklop | and [Oct 3 23:19:15] NOTICE[27015][C-00000006]: chan_sip.c:25282 handle_request_invite: Failed to authenticate device Asterisk Line 1 <sip:101@192.168.45.15>;tag=3b891845f9101ef8o0 |
| 05:21.09 | benklop | there's the authenication failure you knew had to be there |
| 05:21.38 | ChannelZ | You need rtp.conf |
| 05:22.25 | ChannelZ | for the first one that is. |
| 05:23.24 | benklop | i have an rtp.conf file - http://pastebin.com/4Hseb9Sw |
| 05:23.48 | NoobSaibot | Anybody having luck with Asterisk on ESXi 5.5? Any caveats to watch out for? |
| 05:23.56 | ChannelZ | You probably don't have modules.conf either |
| 05:24.20 | ChannelZ | I'm guessing a bunch of modules are not loading |
| 05:24.48 | benklop | modules.conf exists, and basically contains autoload=yes |
| 05:25.06 | benklop | so modules load.. oterwise i dont even get sip |
| 05:25.09 | benklop | :) |
| 05:26.11 | benklop | i'll take a look and see if there's something else i'm missing in rtp.conf though.. just your realizing i was not getting all the logging i was supposed to have, and telling me how to fix that has been a huge help |
| 05:26.20 | ChannelZ | I'm not sure why you would get 'RTP engine not found' |
| 05:27.21 | ChannelZ | but the other bit, 'failed to authenticate' still looks like a bad password or something, or now it's not matching the right peer either (your previous paste implied that it was matching 101 before though) |
| 05:27.23 | benklop | I think i might actually be missing a module entirely |
| 05:28.08 | ChannelZ | module show like rtp |
| 05:28.35 | benklop | ChannelZ: the error here makes it look like the second can be caused by the first: http://forums.asterisk.org/viewtopic.php?f=1&t=86518 |
| 05:29.06 | ChannelZ | probably. I've not quite encountered this combination of fail before |
| 05:29.24 | benklop | chan_multicast_rtp.so and res_rtp_multicast.so |
| 05:29.43 | benklop | looking at a mailing list post, i think maybe i'm supposed to have another |
| 05:30.01 | benklop | res_rtp_asterisk.so ? |
| 05:30.25 | ChannelZ | that's the big one yes |
| 05:30.51 | benklop | okay then. I wonder why that didn't build.. well. sorry for the trouble! |
| 05:32.05 | ChannelZ | you need libuuid |
| 05:32.14 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 05:32.55 | ChannelZ | or uuid-dev rather |
| 05:33.18 | ChannelZ | install that, then re- ./configure asterisk and rebuild |
| 05:33.44 | benklop | am doing :) |
| 05:33.57 | benklop | i was looking for libuuid1-dev, turns out there is no such thing. |
| 05:34.13 | benklop | wish that library packages and such were named a bit more consistently |
| 05:34.22 | benklop | oh well |
| 05:35.01 | benklop | also kind of wish that asterisk would tell me 'hey you've got sip but not all the stuff sip needs - this probably won't work' |
| 05:35.06 | ChannelZ | uuid is a bit of an oddity |
| 05:35.48 | ChannelZ | The requirement is also relatively new, 11.5 if I remember. |
| 05:36.35 | ChannelZ | http://burner.com/asterisk-primer/building-asterisk/ |
| 05:36.47 | benklop | well that would make sense then since I'm instaling 11.5.1 :) |
| 05:37.10 | benklop | and i was probably referencing old docs |
| 05:38.11 | benklop | looks like that was the only one i didn't have\ |
| 05:38.19 | ChannelZ | probably |
| 05:38.36 | ChannelZ | lack of the others should fail a config |
| 05:39.54 | benklop | well that quite likely explains an awful lot. |
| 05:40.05 | benklop | i'm recompiling now, so we'll see |
| 05:40.17 | benklop | but i anticipate it will work a lot better with all its dependencies |
| 05:40.19 | benklop | :) |
| 05:41.13 | benklop | at least i kno it wasn't some super secret 'you have to be in the club to know about it' option |
| 05:42.11 | benklop | 'you're unauthorized ot use this until you'e cool enough to know the handshake' |
| 05:43.31 | ChannelZ | Well.. sort of. libuuid is only mentioned in ChangeLog which most people don't read (including me unless I'm upgrading) |
| 05:44.31 | benklop | hum. well, I guess i learned the handshake then. |
| 05:45.04 | benklop | my phone just started playing monkeys at me |
| 05:45.11 | benklop | pretty sure that's a good sign :) |
| 05:47.47 | ChannelZ | yay! |
| 05:51.54 | benklop | and phones can cal leach other too - I have to say this is working almost infinitely better than before :) |
| 05:55.15 | ChannelZ | Of course. |
| 05:57.18 | *** join/#asterisk hypknight (~hypknight@unaffiliated/hypknight) |
| 05:59.42 | benklop | well yes, but i just wanted to test it all :) |
| 06:00.13 | benklop | it seems my android device is a little finnicky - if the screen is off it doesn't respond t othe call right away, and the caller is notified the call failed |
| 06:00.21 | benklop | then after that the android device starts ringing |
| 06:01.12 | benklop | or at least it did that once. not reproducible now |
| 06:02.42 | *** join/#asterisk taylorbyte2013 (~cyberninj@139.218.237.26) |
| 06:05.47 | ChannelZ | what app |
| 06:07.47 | snadge | apparently digium have a voip providing business in australia |
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| 08:32.23 | D30 | hi all, good day.. just need to ask for some advice... what SIP / DID providers can you recommend to try out?? |
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| 09:00.23 | linocisco | hi |
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| 09:14.41 | As001 | Hello does asterisk manager interface have any limit how many events it can handle per second ? |
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| 13:05.43 | Katty | g'morning |
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| 13:15.31 | bananapie | is there an asterisk console command that allows me to list all presence data that asterisk has in memory? |
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| 13:20.16 | [TK]D-Fender | bananapie: core show hints |
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| 13:22.08 | bananapie | ok. If I have no hints, does asterisk still collect presence info ? |
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| 13:25.19 | [TK]D-Fender | :/ |
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| 13:37.44 | qakhan | call usp_SaveTrip @iTreatedID=${itc_1_tid},@vPhoneNo=${callerid},@dtDate=${date},@dtTime=${time} |
| 13:38.15 | qakhan | is it correct syntax for store procedure in func_odbc.conf |
| 13:38.52 | qakhan | Set(DBINST=${ODBC_sql(${itc_1_tid},${callerid},${date},${time})}) |
| 13:39.05 | qakhan | extensions.conf |
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| 13:57.34 | volga629 | how I can extend device state across multiply boxes ? I so it possible with corosync, is the anther approach ? |
| 13:57.38 | volga629 | another |
| 14:01.02 | file | XMPP can be used as a pubsub mechanism for that. |
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| 14:09.32 | volga629 | for SIMPLE or only XMPP ? |
| 14:10.16 | volga629 | I have SIMPLE setup and I can send messages across the trunk, but user can't get device state working |
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| 14:23.31 | file | it is used as a transport mechanism for the internal device state of the Asterisk core, it does not know or care what triggers said device state changes internally |
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| 14:42.50 | gbit | Hello, I need some help to set CallerID from the output of the script, I'm using exten => 2000, n, Set(CALLERID(name)=/usr/local/bin/contacts.py ${CALLERID(num)}) |
| 14:45.52 | WIMPy | Make that an AGI |
| 14:45.58 | [TK]D-Fender | gbit: You can't just execute shell command in free-form like that |
| 14:46.05 | [TK]D-Fender | And no need for AGI |
| 14:46.17 | gbit | How can I do it? |
| 14:46.28 | [TK]D-Fender | gbit: "core show function SHELL" <- |
| 14:47.21 | gbit | ok, but how I can set the output from the script to CALLERID? |
| 14:47.52 | [TK]D-Fender | gbit: that's why it is a FUNCTION |
| 14:52.57 | gbit | Thanks |
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| 15:04.07 | cusco | hi folks |
| 15:04.28 | cusco | can I list the order that queue is about to serve next calls to members? |
| 15:07.02 | qakhan | can anyone config store procedure in syntax in func_odbc.conf |
| 15:07.12 | qakhan | <PROTECTED> |
| 15:07.14 | qakhan | ? |
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| 15:15.59 | cusco | qakhan: exec sp param1, param2; |
| 15:16.00 | cusco | ? |
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| 15:26.10 | qakhan | cusco should i use exec instead of call? |
| 15:27.51 | cusco | not sure, in mssql we use exec bd.dbo.sp-name param |
| 15:29.03 | qakhan | can you confirm dialplan syntax |
| 15:29.10 | qakhan | <PROTECTED> |
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| 15:48.25 | diatonic | Where do I go to file a bug report? |
| 15:50.27 | mjordan | issues.asterisk.org |
| 15:50.35 | mjordan | diatonic: please read the instructions on filing an issue |
| 15:50.36 | newtonr | diatonic, go here first: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines :D |
| 15:50.43 | mjordan | exactly :-) |
| 15:50.49 | diatonic | thanks |
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| 15:51.30 | diatonic | I believe there is a legitimate bug in ChanSpy though that I can recreate easily |
| 15:51.36 | diatonic | on 1.8.23 |
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| 15:53.24 | [TK]D-Fender | Try on 1.8.23.1 |
| 15:53.46 | diatonic | That is the version I am testing on |
| 15:54.38 | file | what's the bug? |
| 15:54.43 | diatonic | there is an option: x(digit): digit - Specify a DTMF digit that can be used to exit the application. |
| 15:54.57 | diatonic | it only works if there is a call in progress on the cannel you are spying on |
| 15:55.06 | diatonic | if they are not in a call you can not exit |
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| 15:58.52 | cusco | sorry qakhan I'm not using that with asterisk |
| 15:59.00 | cusco | We use old mysql() stuff |
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| 16:00.11 | cusco | is there a way to list the current order of members in a queue? We're using 'leastrecent' strategy, but sometimes seems that the call doesn't go to the leastrecent member ... |
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| 16:14.36 | monsterco | Hi everyone - is it easy to integrate Asterisk with SalesForce? is there a module for it? or it just about pointing the CLID number/CLID name to SalesForce servers? |
| 16:18.12 | [TK]D-Fender | monsterco: No. The only integration that exists is that which you invent yourself |
| 16:18.43 | [TK]D-Fender | monsterco: There is no "pointing a CLID number" concept |
| 16:24.16 | monsterco | [TK]D-Fender - thanks for the info. I never used SalesForce but want to use it - however, if it doesn't pull the screen based on incoming CLID then it would be useless to me - hence the question |
| 16:25.32 | [TK]D-Fender | monsterco: You missed the point... its YOUR job to create the lookup. |
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| 16:26.07 | [TK]D-Fender | monsterco: Asterisk itself has no knowledge of Salesforce. Everything that happens to process your calls is YOUR job. |
| 16:26.47 | [TK]D-Fender | monsterco: Dialplan = everything |
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| 16:30.08 | file | I'm sure consultants/businesses have done that, but such integrations are often not shared |
| 16:32.50 | monsterco | SalesForce people are saying there should be a CTI plugin for Asterisk |
| 16:33.14 | leifmadsen | we integrated with SalesForce, but had to build our own REST service |
| 16:33.37 | leifmadsen | regardless, integration with SF isn't going to be that trivial |
| 16:33.38 | monsterco | @leifmadsen - I see - so it's not a quick one day job then |
| 16:33.46 | leifmadsen | well depends how good you are I guess :) |
| 16:35.07 | monsterco | @leifmadsen - I guess I have to read on salesforce connection methodes first. So, does your setup pull up the screen once a call comes in? |
| 16:35.24 | leifmadsen | it's not that kind of integration |
| 16:35.30 | leifmadsen | it uses information from SF to handle queue routing |
| 16:35.35 | leifmadsen | based on the CID |
| 16:36.36 | monsterco | oh, I see. I think I need just the reverse |
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| 17:17.06 | qakhan | [TK]D-Fender do you know how to call Store Procedure in func_odbc.conf |
| 17:17.07 | qakhan | ? |
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| 17:24.05 | [TK]D-Fender | qakhan: No. I'd just as soon do it in AGI |
| 17:29.43 | qakhan | ok |
| 17:31.19 | wltjr | does it make sense to have to keep turning up the volume for each call on a polycom vvx 300? seems it should stay where you left it, up/down, what ever, but seems to reset on each call/extension/line |
| 17:32.20 | wltjr | might be a bug, but polycom is a pita to report such to, have to go through reseller for any support, if not a reseller or certified polycom tech or some bs |
| 17:33.27 | WIMPy | Don;t know Polycom, but usually I expect the volume to reset after a call unless you save it. |
| 17:33.50 | Penguin | Save it after setting it if you want it to stay. |
| 17:34.01 | wltjr | save? never seen that before, most phones remember their volume, its not a setting but a button on the front I will look to see if it can be saved |
| 17:34.34 | WIMPy | Usually you just press the OK button after changing the volume to save it. |
| 17:35.21 | wltjr | looking for that, but the phones have a up/down button for volume, most things stay that way when you turn it up or down, withouth having to go save that, its not even an option in web interface, might be in UC for provisionining, last phones I could provision that and had setting in web interface, Cisco 501gs, but most SPA phones have that |
| 17:35.32 | Penguin | On my Cisco phones, as soon as you push the volume up/down, a Save softkey appears on the screen for a few seconds. If you do not press it before it goes away, when you end the call the volume goes back to where it was before. |
| 17:36.43 | wltjr | no softkey just checked, not sure how to get the volume to remain, the ringer volume seems to remain, but not in call volume |
| 17:37.19 | WIMPy | Does that phone not have an OK/Enter button? |
| 17:37.27 | wltjr | Penguin: ok so maybe reseting volume is normal, wish there was a way to make it permanent, the other phones were like that, guess nothing is perfect |
| 17:37.53 | wltjr | WIMPy: no, it has a navigation thing, 4 direction button with center button, but does nothing when in call or messing with volume and no soft keys show up |
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| 17:39.59 | wltjr | just went through, manual searched for volume, no way to save or anything, need to make feature request/bug report to polycom, but its likely this way by design, so feature request |
| 17:40.04 | wltjr | ty |
| 17:42.12 | [TK]D-Fender | wltjr: SPIP's have a persistent flag. I suspect VVX's have the same |
| 17:47.27 | Penguin | I was thinking the last time I installed some SPIPs that I turned up the in-call volume on the handset, then separately on the speakerphone, then they were set forever. But it's been a while, so thought maybe I pressed a save/ok button; couldn't really remember. |
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| 18:06.44 | pzn | I'm trying to use a Queue without success, when someone dials 799, the queue gets the call, the person hears the music, but the phones 811 813 and 817 do not ring. any hints about how to debug this? thanks in advance. http://pastebin.com/K9kUKe9D |
| 18:07.14 | [TK]D-Fender | pzn: you posted configs... but not actual status |
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| 18:07.42 | pzn | [TK]D-Fender, what should I post? asterisk -rvvvv then make a call? |
| 18:08.00 | [TK]D-Fender | pzn: as well as "queue show" |
| 18:08.44 | pzn | let me get them |
| 18:09.47 | fnsound | I have jackd running, asterisk CLI is giving me Client Open Status: Failure, Server Failed when I run JACK or JACKHOOK. jackd shows no connection on the console. Is there something I am missing? |
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| 18:16.17 | pzn | [TK]D-Fender, I don't know why... but when I tried to get all the debug information for you help to analyse, sudenly all started working ok!!! |
| 18:16.27 | pzn | [TK]D-Fender, thanks anyway. |
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| 18:19.46 | wltjr | [TK]D-Fender: thanks again, I will try it, http://wiki.freeswitch.org/wiki/Polycom_configuration#Example_local-sip.cfg assuming the vvx will respect that config |
| 18:20.50 | [TK]D-Fender | wltjr: Well it is the SPIP spec.. not sure what the commonalities/differences are |
| 18:21.27 | Penguin | pzn: Maybe you forgot to reload the queues or forgot to reload something else important. |
| 18:23.16 | wltjr | [TK]D-Fender: no clue, but seems there are differences, but maybe not, I haven't found a universal file name I can use for common settings, thought I could use 0000000.cfg but phones never ask for that |
| 18:23.55 | pzn | Penguin, yes, that may be what happened. I had the problem before lunch. just got back from lunch and posted it here. the computer was rebooted at lunch... |
| 18:23.58 | [TK]D-Fender | That'st 1st load only |
| 18:24.03 | [TK]D-Fender | Every other time was <mac>.cfg |
| 18:24.26 | [TK]D-Fender | Have you downloaded the firmware for it? It is generally complete |
| 18:25.46 | wltjr | [TK]D-Fender: so no common file, spas would look for and use like spa501g.cfg or something, I haven't downloaded firmware, I upgraded it via the web interface, hear about these .ld files but haven't seen them or something like that |
| 18:25.51 | Penguin | pzn: I'm sure that's what happened, then. Remember to run queue reload ... when changing queues.conf. |
| 18:26.15 | wltjr | think other ciscos look for common file as well, guess I will just have to copy/paste, and diff to make sure all are same, kinda pita for common stuff, ntp, syslog, etc |
| 18:26.33 | [TK]D-Fender | wltjr: well common can be LINKED by the mac file... |
| 18:26.51 | [TK]D-Fender | wltjr: typical deployments would have 2 files, 1 for common settings, and the other device-specific |
| 18:27.32 | pzn | Penguin, tks |
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| 18:31.11 | wltjr | [TK]D-Fender: that's what I am thinking i created a link to another file but its not using those settings... files requested http://pastebin.com/zRPZUUQU I tried to link the mac-phone.cfg to the common file, but those settings dont show, let me double check, rebooted phones earlier |
| 18:32.40 | wltjr | I had to link the -web one to the mac.cfg it seems to pull those but doesn't always use one or the other, I had just mac.cfg and sometimes it would not use those settings, and sometimes it would not use mac-web.cfg odd stuff |
| 18:32.58 | [TK]D-Fender | wltjr: Oct 2 03:32:16 asterisk in.tftpd[26149]: RRQ from 10.4.2.6 filename 0004f2829d16.cfg Oct 2 03:32:16 asterisk in.tftpd[26150]: RRQ from 10.4.2.6 filename 0004f2829d16-phone.cfg Oct 2 03:32:16 asterisk in.tftpd[26151]: RRQ from 10.4.2.6 filename 0004f2829d16-web.cfg |
| 18:33.13 | [TK]D-Fender | wltjr: those are the 3. Don't forget they could be layered overrides |
| 18:33.30 | [TK]D-Fender | wltjr: Which is why you never touch the configs via web/phone-direct |
| 18:33.38 | wltjr | [TK]D-Fender: yes I see, but the stuff in my common file does not overrite the other stuff, unless default settings in phone override what I am pushing out |
| 18:34.04 | [TK]D-Fender | That's where layer order comes in |
| 18:34.22 | wltjr | could be why I had to do mac-web.cfg though, ok let me check out the layered order and see whats up |
| 18:34.59 | wltjr | [TK]D-Fender: do the SPIP phones require you to hit send when you dial every time? not sure if there is some interdigit time out to auto send/dial or what |
| 18:35.22 | Penguin | digitmap should control that problem. |
| 18:35.53 | [TK]D-Fender | wltjr: these should be just as configurable as the SPIP's.. so it's all up to you |
| 18:36.20 | wltjr | [TK]D-Fender: ok will look for something, haven't seen phones do that before, usually you dial if it matches a pattern it does its thing |
| 18:36.45 | [TK]D-Fender | And what have you got in there, and what are you dialing? |
| 18:37.16 | wltjr | [TK]D-Fender: normal call 7 digit #, and I have the standard stuff in there, plus some patterns for matching 1-2 digits for internal calls |
| 18:37.32 | [TK]D-Fender | Show it... |
| 18:37.59 | wltjr | dialplan.digitmap="x|xx[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT|**x.T" |
| 18:38.18 | Penguin | I don't see a 7-digit dial string there. |
| 18:38.20 | wltjr | maybe there isn't one for 7 digits |
| 18:38.22 | [TK]D-Fender | no 7-digit there... |
| 18:38.32 | wltjr | yeah jsut seeing that... duh... wtf why did the leave that out... freaking polycom :) |
| 18:38.37 | wltjr | let me sed... |
| 18:38.48 | [TK]D-Fender | so I hurd |
| 18:38.56 | Penguin | 7-digit isn't common everywhere. |
| 18:39.02 | wltjr | really a lvoe hate relationship |
| 18:39.11 | wltjr | 7 digit is common in US all local calls are 7 digits |
| 18:39.20 | Penguin | Not everywhere in the US. |
| 18:39.34 | wltjr | unless long distance |
| 18:39.39 | Penguin | Some places require 10 digits to call next door. |
| 18:39.59 | wltjr | yeah I think s.f. is getting that way, I assume ny is that way now |
| 18:40.12 | Penguin | So 7-digit dial isn't really normal anymore. |
| 18:40.14 | wltjr | i always put 10 in my cell for when I travel |
| 18:40.19 | Penguin | same |
| 18:40.34 | wltjr | it still is for local call, the vast majority of numbers I call are local and same with calls recieved |
| 18:41.02 | Penguin | So it's typical for YOUR area. Polycom didn't know you were still using 7-digit dial! |
| 18:41.04 | wltjr | anyway nice homer simpson moment, still can't believe polycom left that out, kinda funny |
| 18:41.17 | wltjr | sure but can't really hurt to have it in there, the others are |
| 18:41.25 | wltjr | I get its made for global distribution |
| 18:44.05 | WIMPy | Which make a total fail. |
| 18:46.04 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 18:53.06 | wltjr | looks like feature.nonVolatileRingerVolume.enabled |
| 18:53.19 | *** join/#asterisk threesome (~threesome@ip-94-113-13-143.net.upcbroadband.cz) |
| 18:53.28 | wltjr | er might help with persisent volume, not seeing the other in the current UCS guide |
| 18:55.00 | Penguin | Seems reasonable to me. |
| 18:55.17 | wltjr | it says it retains between reboot, I want it retained between calls :) |
| 18:55.37 | Penguin | backward capable setting? |
| 18:55.38 | wltjr | noisy env staff talks loud some hard of hearing etc... |
| 19:02.35 | *** part/#asterisk monsterco (~monsterco@64.231.101.21) |
| 19:02.51 | *** join/#asterisk Ahrotahntee (~rhjones@unaffiliated/ahrotahntee) |
| 19:12.29 | wltjr | these vvx series phones are really frustrating, they do not want to be provisioned, I have tried both mac.cfg, and mac-web.cfg and both are being ignored, new settings not showing in phone, I see files being requested... grrr |
| 19:13.12 | wltjr | I had both before -web was symlink it ignored both, I tried just one of each and same thing... maybe the 0000000.cfg is causing problems |
| 19:16.24 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 19:17.34 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 19:18.33 | wltjr | hmmm think I found another bug... seems these things digitmap timeout is set to 3|3|3|3|3|3|3, and that is supposed to be seconds, hardly a valid parameter value |
| 19:18.36 | [TK]D-Fender | wltjr: dump the folder for us |
| 19:18.47 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 19:18.52 | wltjr | [TK]D-Fender: its a mess still have cisco stuff in there |
| 19:19.15 | [TK]D-Fender | wltjr: DNDC :) |
| 19:20.00 | wltjr | [TK]D-Fender: ? |
| 19:20.29 | [TK]D-Fender | wltjr: Didn't know, don't care :) |
| 19:21.02 | wltjr | ah, well might if/when you work with vvx unless its resolved by then, I would report to Polycom if there was an easy way |
| 19:22.17 | Penguin | N? Know starts with K. |
| 19:22.41 | [TK]D-Fender | Penguin: Good, you are paying attention ;) |
| 19:22.55 | Penguin | I sometimes do that. |
| 19:25.59 | *** join/#asterisk mokmeister (~mokmeiste@86-40-248-153-dynamic.b-ras2.lmk.limerick.eircom.net) |
| 19:31.35 | wltjr | this is my tftp root, adding stuff to files rebooting phones and settings are not there |
| 19:31.39 | wltjr | http://pastebin.com/M0u2f4Mg |
| 19:32.19 | Penguin | Why do you have both the regular and the -web configs? |
| 19:32.45 | wltjr | Penguin: idk, I have tried just going with one or the other and sometimes things are not applied, thus rocking both... |
| 19:33.10 | wltjr | also seems order is web stuff overrides provisining, would assume it be the opposite... |
| 19:33.35 | wltjr | modified digitmap initially via web interface, and because of that its ingoring my provisining changes |
| 19:33.44 | Penguin | I don't know about that... if you provision it and then change the web settings, the web settings override. |
| 19:34.13 | wltjr | web settings first long ago, now provisioning might reset, but have to use web interface to configure provisioning server, unless I can put that in 00000000.cfg and have it picked up that way |
| 19:34.31 | Penguin | If I were fighting it, I would be sure my files are as they need to be and then do a factory reset on the phone. |
| 19:34.59 | wltjr | Penguin: that would be ideal, but config is evolving... :) |
| 19:35.34 | *** join/#asterisk jetlag (~jetlag@pool-71-168-241-76.cmdnnj.east.verizon.net) |
| 19:36.04 | *** join/#asterisk Linkforsoad (~Linkforso@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 19:36.43 | wltjr | reseting one will do others next, lots of learning with polycom, definitely very picky on order, read that in admin ucs guide, but order seems a bit odd, I guess it makes sense |
| 19:38.08 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 19:43.20 | wltjr | phones don't seem to respect dhcp next server, have to always configure provisioning server after reset |
| 19:44.05 | *** join/#asterisk boom^time (~boom^time@75-151-20-174-Michigan.hfc.comcastbusiness.net) |
| 19:47.18 | *** join/#asterisk FinboySlick (~shark@74.117.40.10) |
| 19:50.41 | drmessano | You need option 160 for the Polycoms |
| 19:50.59 | drmessano | and it needs to be formatted correctly. |
| 19:51.52 | wltjr | drmessano: ty for the heads up will look into that, seems a bit non-standard, but guess its that way in case your using ftp or something other than tftp |
| 19:52.18 | FinboySlick | I'm trying to setup an extremely minimal asterisk 11.5. Basically, starting from an empty /etc/asterisk and getting it to register to a sip server. I assume that I need a sip.conf and load => chan_sip.so in modules.conf |
| 19:52.43 | FinboySlick | (previous experience is with 1.4 so I'm a bit unfamiliar with all that changed since) |
| 19:54.10 | Penguin | Ah, not 66 and not 150, but 160 for Polycom. Interesting. |
| 19:54.44 | *** join/#asterisk vinhdizzo (~vinh@cpe-98-154-220-20.socal.res.rr.com) |
| 19:54.46 | malcolmd | FinboySlick: it'll load by itself if autoload's turn on |
| 19:55.27 | FinboySlick | malcolmd: Okay. How do I check though? I'm used to having a 'sip ... ' command from 1.4 but that apparently moved. |
| 19:56.17 | malcolmd | you should still, if you don't, that likely means chan_sip wasn't loaded or wasn't built. if module load chan_sip.so fails, it wasn't built, or it's trying to load a version compiled against a different build than you're actually running |
| 19:57.17 | Penguin | This guide I am looking at says that option 66 is mandatory. Set your tftpd address in option 66. |
| 19:57.41 | Penguin | Then it indicates that you use option 160 to tell the phone to use tftpd. |
| 19:57.50 | Penguin | I have no idea how you define it, but that's what this guide says. |
| 19:58.08 | FinboySlick | malcolmd: Hmmm... module show like chan_sip.so says that it's laoded with use count 0 |
| 19:58.29 | Penguin | I guess option 160 should be "tftp://serveradress" |
| 19:58.43 | malcolmd | then you should be able to start typing "sip" from the asterisk cli and see stuff |
| 19:59.05 | FinboySlick | malcolmd: that'S what I thought too but apparently not. |
| 19:59.17 | Penguin | Option 66 would be "serveraddress" only. |
| 19:59.21 | malcolmd | something's amiss for you. good luck, sorry. |
| 19:59.25 | FinboySlick | I'll restart it from scratch and see what I might have done wrong. |
| 20:00.00 | Penguin | wltjr: Try that procedure. |
| 20:01.02 | FinboySlick | malcolmd: could something wrong in my sip.conf cause it not to load properly? |
| 20:01.34 | malcolmd | possibly, i'm sorry, i can't carry you forward on this one, it's the friday before astricon, things are busy :D |
| 20:02.09 | Penguin | If your sip.conf is broken, of course it will cause the channel driver not to load. |
| 20:02.21 | FinboySlick | malcolmd: No problem, this isn't an urgent project and I'm just tossing questions at the tree to see what falls. |
| 20:02.28 | Penguin | But if you said chan_sip.so shows up, then it is loaded. |
| 20:03.42 | FinboySlick | Penguin: module show says that it's loaded (along with chan_local, res_crypto and res_http_websocket) |
| 20:03.56 | FinboySlick | use count at 0 for all of them. |
| 20:04.39 | FinboySlick | Note that this is a completely empty /etc/asterisk I'm very likely missing a lot of 'default' stuff. |
| 20:05.34 | leifmadsen | FinboySlick: check out the Asterisk book which documents Asterisk 11 and shows exactly how to get a "bare minimum" /etc/asterisk directory setup |
| 20:05.46 | leifmadsen | fair notice: I co-authored |
| 20:05.51 | leifmadsen | I'm pimping my warez. |
| 20:06.13 | leifmadsen | Asterisk: The Definitive Guide 4th Edition is what you want |
| 20:06.17 | mjordan | leifmadsen: so you're the book's sugar daddy? |
| 20:06.23 | leifmadsen | mjordan: hells ya I am |
| 20:06.25 | FinboySlick | leifmadsen: *laughs* Okay. Is it linked in the /topic ? |
| 20:06.35 | leifmadsen | FinboySlick: it's linked on the O'Reilly site :) |
| 20:06.38 | mjordan | ~book |
| 20:06.39 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 20:06.40 | leifmadsen | or google will find it easily |
| 20:06.45 | FinboySlick | leifmadsen: If you wrote the one that was around when 1.4 came out, I quite enjoyed it. |
| 20:06.47 | mjordan | ~buybook |
| 20:06.47 | infobot | You can buy "Asterisk: The Definitive Guide" at http://oreilly.com/catalog/0636920025894 so go buy it SERIOUSLY |
| 20:07.01 | leifmadsen | FinboySlick: yep, same book, but twice as thick :) |
| 20:07.05 | leifmadsen | much like my gut now |
| 20:07.16 | leifmadsen | 800 something pages now |
| 20:07.28 | FinboySlick | leifmadsen: It made me go from complete noob to just good enough to make it work flawlessly for our setup. |
| 20:07.34 | FinboySlick | leifmadsen: So big thanks. |
| 20:08.12 | *** join/#asterisk deegen (~deegen@S01060023bee90320.gv.shawcable.net) |
| 20:09.40 | *** part/#asterisk navaismo (~navai_000@189.241.24.0) |
| 20:11.17 | wltjr | Penguin: craziness, I will check it out when I get a chance, pausing on that have a phoen at my office will mess with it later or over weekend, moving on to other stuff, doubling vlans... fun... |
| 20:12.01 | Penguin | The mention of 160 made me go look to see how it affected the phones. |
| 20:17.58 | *** join/#asterisk navaismo (~navaismo@189.241.24.0) |
| 20:18.50 | SuperNull | hey guys, when ever an ATA changes its ip address i have to manually prune the peer.. from realtime.. i think one of these options is just not right... http://pastebin.com/ATfBnTQ4 |
| 20:19.06 | SuperNull | if i dont prune the peer they fail registration. |
| 20:24.11 | *** join/#asterisk jpcansa (~jpcansa@201.199.100.178) |
| 20:36.07 | FinboySlick | leifmadsen: For the record, I think it's the fact that my startup script had a -C /etc/asterisk/asterisk.conf which didn't exist. |
| 20:36.15 | FinboySlick | sip works now. |
| 20:36.30 | leifmadsen | ya that would do it :) |
| 20:39.31 | *** join/#asterisk danjenkins (~danjenkin@cpc3-folk2-2-0-cust34.1-2.cable.virginmedia.com) |
| 20:40.19 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-igfbmnofbgtdwleg) |
| 20:50.03 | SuperNull | leifmadsen oh man.. i love your books sir. |
| 20:50.13 | leifmadsen | heh |
| 20:50.25 | SuperNull | i paid for mine even ;) |
| 20:50.38 | leifmadsen | w00t |
| 20:50.59 | leifmadsen | ok, well I'm out for the weekend |
| 20:51.06 | SuperNull | have a good one man. |
| 20:51.09 | leifmadsen | peas out |
| 20:53.13 | *** join/#asterisk mwo (~Mark@unaffiliated/mwo) |
| 20:58.12 | FinboySlick | Hmmm, is autoload=yes in modules.conf supposed to load *everything* or just what your config files would suggest is needed (eg look at what apps you are using in extensions.conf and only grab those). |
| 20:59.43 | WIMPy | everything |
| 20:59.46 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:00.00 | WIMPy | No such thing as parsing all configs to see what's needed. |
| 21:01.37 | FinboySlick | WIMPy: Alright. I'll try to cherry-pick myself then. At least it got my basic setup working but it sort of defeats my initial aim of 'minimal' ;) |
| 21:02.02 | *** join/#asterisk Dovid (~Dovid@ool-457f6ac5.dyn.optonline.net) |
| 21:03.52 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-kvercuewuocqmzpu) |
| 21:03.53 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 21:07.39 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 21:08.43 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 21:09.27 | *** join/#asterisk danjenkins (~danjenkin@cpc3-folk2-2-0-cust34.1-2.cable.virginmedia.com) |
| 21:11.49 | *** join/#asterisk kbarry (~chatzilla@12.164.169.142) |
| 21:12.56 | kbarry | We are using d40s, we have noticed that a small number of our phones do automatically dial out. We removed the "dial 9" requrement from the outgoing call rules, and for that matter, they are all getting their rules from the server, |
| 21:13.26 | kbarry | so the behavior of some of the phone is as expected, you pick up the reciever, dial a number to texas, for instance, |
| 21:13.27 | kbarry | and it calls |
| 21:13.48 | kbarry | on others, you pick up, dial a number, and must then press "dial" softkey to make it perform the call. |
| 21:14.16 | kbarry | Can't find anyone with this problem, and truth be told i'm no pbx expert. |
| 21:15.14 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 21:15.24 | ChannelZ-Wk | Well the phone usually has a dialplan of its own so that it can figure out when to send the string to Asterisk or if it should wait for other digits to be dialed |
| 21:15.34 | ChannelZ-Wk | I'm guessing the phone's dialplans are different. |
| 21:16.06 | ChannelZ-Wk | That said I have no experience with Digium phones |
| 21:16.56 | ChannelZ-Wk | (and thus whatever magical integration they have) |
| 21:17.08 | WIMPy | Well, tehy have a dialplan like any sip phone and like most of them it can't ask the server. |
| 21:17.31 | *** join/#asterisk taylorbyte2013 (~cyberninj@139.218.237.26) |
| 21:22.15 | newtonr | kbarry, are you setting the digit_maps for their lines with DPMA, or is each phone configured manually from the phone's GUI ? |
| 21:22.40 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:24.44 | newtonr | kbarry, if using DPMA and res_digium_phone.conf, you can configure the digit_map for the "line" section types to set the phones diaplan for that line. There is the potential for misconfiguration in there of course. You could use templates to reduce the chances of misconfiguration for individual lines or phones. |
| 21:25.09 | newtonr | kbarry, If you get confused, just contact Digium support and they'll look at it. It sounds like you are able to reproduce it easily. |
| 21:27.56 | newtonr | kbarry, https://wiki.asterisk.org/wiki/display/DIGIUM/DPMA+Configuration#DPMAConfiguration-LineConfigurationOptions for reference |
| 21:31.27 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 21:58.02 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 22:00.31 | *** join/#asterisk redotis (~redotis@nc-76-5-176-123.dhcp.embarqhsd.net) |
| 22:00.38 | redotis | Hello fellers. |
| 22:00.58 | redotis | How do you get asterisk to send voicemail to email in mp3 format instead of wav? |
| 22:01.48 | navaismo | converting the wav to mp3 |
| 22:01.49 | WIMPy | You have to script something yourself. |
| 22:02.18 | WIMPy | And don't forget to buy a licence. |
| 22:02.28 | _Corey_ | WIMPy: Don't be lame |
| 22:02.48 | RaNa | hello need some help to capture call logs my system keeps hanging up calls how can i get the log for that or see why its doing what its doing |
| 22:03.02 | _Corey_ | hahaha... ;) too late on a Friday. |
| 22:03.10 | newtonr | RaNa, https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 22:03.28 | redotis | license for what |
| 22:03.30 | RaNa | thanx newtonr |
| 22:03.32 | navaismo | mp3 |
| 22:03.39 | redotis | there is such a thing? |
| 22:03.56 | WIMPy | Yes |
| 22:04.03 | *** join/#asterisk xzarth (~krikkit@dh207-37-181.xnet.hr) |
| 22:04.23 | WIMPy | It's a commercial codec. |
| 22:04.25 | newtonr | RaNa, no problem, pay close attention to your verbose and debug levels as set in asterisk.conf and show on the CLI with "core show settings" |
| 22:04.35 | redotis | mp3 is a commercial codec? |
| 22:04.37 | redotis | seriously |
| 22:04.41 | redotis | i did not know that |
| 22:04.54 | WIMPy | Has always been. |
| 22:05.07 | redotis | you know what...i hate to say it but..fu#$ the police |
| 22:06.26 | _Corey_ | redotis: The patent holder is the one who would be suing you, not the police |
| 22:06.35 | WIMPy | That's why we have things like vorbis. Which I think is better anyway. |
| 22:06.36 | redotis | sounds good to me |
| 22:06.58 | redotis | What plays well straight from gmail? |
| 22:08.31 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-kvercuewuocqmzpu) |
| 22:08.42 | WIMPy | That obviousely depends on your client. |
| 22:09.30 | redotis | gmail client |
| 22:09.32 | redotis | chrome |
| 22:09.42 | redotis | I think it's built into the browser |
| 22:09.56 | redotis | wav files don't play for me for some reason |
| 22:10.35 | redotis | Apparently http://www.voip-info.org/wiki/view/Asterisk+Voicemail recommends a script from generationd.com |
| 22:10.58 | redotis | I downloaded it. What do you have to do in voicemail.conf to get it to work? |
| 22:11.10 | redotis | The site doesn't seem to say. |
| 22:12.54 | redotis | mailcmd=perl |
| 22:13.29 | redotis | mailcmd=/path? |
| 22:15.59 | *** join/#asterisk danjenkins (~danjenkin@cpc3-folk2-2-0-cust34.1-2.cable.virginmedia.com) |
| 22:17.13 | *** join/#asterisk Nemus (~Nemus@unaffiliated/nemus) |
| 22:18.02 | Nemus | hello I'm wondering how you can load your sip.conf or iax.conf file from a mysql database in real time. I am having a hard time understanding where I would start. |
| 22:19.59 | newtonr | Nemus, https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration |
| 22:20.08 | newtonr | And probably want to read relevant sections in |
| 22:20.09 | newtonr | ~book |
| 22:20.09 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 22:21.46 | *** join/#asterisk Carlos_PHX1 (~Carlos@ip68-104-246-231.ph.ph.cox.net) |
| 22:30.40 | Nemus | newtonr I've seen that link, but I don't really understand how sip realtime is linked to the mysql database. |
| 22:31.55 | navaismo | anyone has experience with snom and openvpn, i just entered in an endless loop with it |
| 22:35.13 | newtonr | Nemus, I'm not sure how to explain it better than that page.. you could take a look at the chapters on realtime in the book. |
| 22:52.09 | *** join/#asterisk monsterco (~monsterco@static-173-212-179-145.ptr.terago.net) |
| 22:52.36 | monsterco | Hi everyone - can someone please tell me how to match this DID range in inbound: 3156313041-3156313069 |
| 22:53.26 | navaismo | AGAIN--->http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html#asterisk-DP-Basics-SECT-3.6 |
| 22:53.33 | WIMPy | With two patterns. |
| 22:53.49 | navaismo | this is question coming from #freepbx |
| 22:54.01 | navaismo | seems like he want the exact response |
| 22:54.58 | monsterco | navaismo - I am asking an Asterisk question not a FreePBX question |
| 22:55.09 | navaismo | is the same answer anyway |
| 22:55.20 | monsterco | FreePBX is not same as ASterisk |
| 22:55.23 | navaismo | that i gave you before |
| 22:55.26 | monsterco | totally differnet projects |
| 22:55.35 | navaismo | dont you say |
| 22:55.39 | monsterco | answer maybe same but I am asking the Asterisk community |
| 22:56.01 | navaismo | walks away |
| 22:57.04 | WIMPy | You need one extension for ...41-49 and one for ...50-69 |
| 23:02.42 | monsterco | WIMPy - thanks. so like _315631[41-49] and another like _315631[50-69] ? I am all confused about this |
| 23:04.27 | WIMPy | That's not how patterns work. |
| 23:06.07 | monsterco | what do the numbers inside [] signify? |
| 23:07.07 | WIMPy | A range for a single digit. |
| 23:07.15 | WIMPy | Or a list. |
| 23:11.27 | monsterco | so that is why the two sets are needed? |
| 23:12.11 | WIMPy | Exactely |
| 23:12.32 | monsterco | so something _315631304[1-9] <<< meaning 315631304 |
| 23:12.48 | monsterco | 3156313041 - 315631049 |
| 23:13.02 | WIMPy | yes |
| 23:13.03 | monsterco | WIMPy ^^^ tnx |
| 23:13.57 | WIMPy | Where you can write [1-9] as Z. |
| 23:15.33 | monsterco | what is Z? |
| 23:15.43 | WIMPy | [1-9] |
| 23:16.09 | WIMPy | X=[0-9], Z=[1-9] and N=[2-9] |
| 23:16.12 | monsterco | right |
| 23:16.24 | monsterco | so my example above is right for 41-49? |
| 23:16.35 | WIMPy | Yes |
| 23:17.42 | monsterco | thank you WIMPy |
| 23:17.48 | monsterco | and so I need three set |
| 23:17.51 | monsterco | sets* |
| 23:17.54 | monsterco | one for 50 series |
| 23:17.58 | monsterco | and one for 60 series as well |
| 23:18.10 | WIMPy | No, only two. |
| 23:18.21 | monsterco | hmmmm |
| 23:18.35 | WIMPy | 304Z and 30[56]X |
| 23:18.58 | monsterco | I Don't understand the 30[56]X meaning |
| 23:19.31 | WIMPy | See above. Read it as 30[56][0-9]. |
| 23:20.24 | monsterco | but what is 56? |
| 23:20.35 | monsterco | that is the digit position? |
| 23:20.38 | WIMPy | A lost of digits. |
| 23:20.50 | WIMPy | At that position. |
| 23:21.25 | monsterco | k got it |
| 23:21.25 | monsterco | thanks |
| 23:21.41 | WIMPy | You could write [5-6] instead of [56] as well. |
| 23:25.55 | gusto | hi WIMPy |
| 23:26.16 | WIMPy | hi gusto |
| 23:30.18 | newtonr | hi WIMPy |
| 23:30.38 | WIMPy | Hi Rusty |
| 23:31.36 | WIMPy | is not that much on the keyboard lately. |
| 23:40.50 | gusto | me neither |
| 23:41.01 | gusto | at least not at asterisk |
| 23:41.25 | gusto | lately i even got the asterisk 4th edition book |
| 23:41.56 | gusto | i do not know why, but it came by |
| 23:42.25 | gusto | there should be some minor demages to the book, but i havent discovered any yet |
| 23:42.37 | gusto | thats why i got it for almost nothing |
| 23:43.36 | navaismo | ey guys do you have an android phone I made an app to chanspy sip channels if you can please test it and sen me some feed back---> http://asterisktools.blogspot.mx/2013/10/aplicacion-android-para-espiar.html |
| 23:43.58 | navaismo | see you all |
| 23:44.35 | gusto | well, i do not have a smartphone at all |
| 23:45.22 | gusto | and i am not planning to buy one, i found a new destination to spend money to, donating for projects, i donated to every project i stumbeled upon last days |
| 23:45.52 | gusto | that makes more sense than wasting money on a smartphone |
| 23:47.13 | WIMPy | doesn't own a smartphone, either. |
| 23:47.55 | gusto | yes |
| 23:48.16 | newtonr | gusto, what kind of projects do you donate to? |
| 23:48.23 | gusto | however, i discovered that the NOKIA mobile phone i privatized from my father has SIP capabilities |
| 23:48.43 | gusto | today i donated for that lawsuit against tempora |
| 23:49.02 | gusto | before for one computer security website in slovakia |
| 23:49.22 | gusto | and of course all BSD projects ... thats a must |
| 23:50.06 | WIMPy | I'm donating time. For a project opening up some buildings to the pubic for open workshops or performing arts. |
| 23:50.30 | WIMPy | Well, legally it's squatting, but I think there's a good chance for it to stay. |
| 23:50.39 | gusto | but i am going to expand these activities as soon as i earn some more money, because now i am unemployed so i do not have a lot to donate, however, when i get new wage i will plan some 10% for donating into it |
| 23:51.56 | newtonr | i think last thing I donated too was Wikimedia foundation |
| 23:51.59 | gusto | because i do not like it any more how i started and i feel something like is best described in german by "bringschuld" that i ll make it better next time by planning and making a clear budget plan |
| 23:52.17 | gusto | i do not plan to donate for mainstream projects |
| 23:52.28 | gusto | i thaught about donating to CentOS, but then i said NO |
| 23:52.46 | newtonr | I use wikipedia like 500 times a day, so I donate |
| 23:53.05 | gusto | i have to make a clear transparent plan, because this time, i really donated very chaotically |
| 23:53.24 | WIMPy | has donated to wikimedia as well. |
| 23:53.29 | gusto | lol |
| 23:53.39 | gusto | a lot of ppl are donating to wikimedia so i will not |
| 23:53.54 | gusto | of course i use wikipedia too, but i can live without it as well |
| 23:54.16 | WIMPy | Well, I find it veru usefull. Well, often :-) |
| 23:54.25 | gusto | apart from that, wikipedia would need work to invest in and not money |
| 23:54.38 | gusto | because the only english language wikipedia is usable |
| 23:54.47 | gusto | other languages pages are useless |
| 23:55.56 | WIMPy | does not think so |
| 23:56.21 | WIMPy | But most of the content is in english, off course. |
| 23:56.29 | gusto | yes |
| 23:56.48 | gusto | do not start defending german wikipedia |
| 23:56.54 | gusto | dont even try it |
| 23:59.16 | *** join/#asterisk serafie (~erin@24.96.64.240) |