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| 01:00.40 | trapa | I have a nat question i Think, I have some phones.. that can dial the echo-test and get perfect results every time, but if another phone calls them or they call another phone directly, no audio is passed |
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| 01:06.10 | ChannelZ | well that would depend on the network topology |
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| 01:12.53 | trapa | Server located in .. while essentially a colo |
| 01:13.17 | trapa | and each person with their phones at their house with likely a linksys type router between their phone and the internet |
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| 01:33.37 | ChannelZ | well I'm assuming each client is behind NAT then |
| 01:33.45 | ChannelZ | but the * server is not |
| 02:01.43 | citywok | trapa: if some phones work, and some don't, and all the configs are identical it's likely an issue with the router. |
| 02:10.49 | trapa | nods. |
| 02:11.05 | *** part/#asterisk Mhaddog_ (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 02:11.15 | trapa | Asterisk server is not behind nat .. and citywok, probably right. But can you successfully do echotest with incorrect nat settings? |
| 02:11.29 | trapa | i thought that nat was either a "Either audio is passed, or not" kind of situation |
| 02:12.26 | antiwire | nope |
| 02:12.37 | citywok | trapa: it can be one way, both ways, not work at all |
| 02:12.46 | citywok | depends on how crappy the router is |
| 02:13.29 | trapa | well crap |
| 02:13.59 | citywok | for home users i'd recommend providing them a router w/ the phone |
| 02:14.10 | citywok | that's how we handle our WFH users. give them a router we know works / we like. |
| 02:14.27 | citywok | $40 for a router is nothing in the scheme of support calls. |
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| 02:18.47 | trapa | nods. |
| 02:18.48 | trapa | Suppose. |
| 02:18.59 | trapa | If we were a business and not a non-profit |
| 02:19.22 | trapa | Was everything we could do to get the board to spend 50$ per phone .. |
| 02:19.28 | trapa | But maybe there's another thing going on here. |
| 02:19.44 | trapa | I just dailed the extension in question (2208) that was broken, and 2213 rang |
| 02:19.46 | citywok | dear god, what phones did you get for $50? |
| 02:19.56 | trapa | Polycom Soundpoint IP 330's |
| 02:20.11 | antiwire | for 330's that's not too bad |
| 02:20.31 | citywok | yea, those things are like $100 phones aren't they? |
| 02:20.32 | trapa | And they have the central provisioning what i thought would be awesome because i can re-config them remotely |
| 02:20.57 | trapa | we bought 10 for 500$ and I talked the guy into free shipping. |
| 02:20.59 | citywok | trapa: yea, the polycom provisioning is okay. not as good as aastra, but it's powerful. |
| 02:21.25 | antiwire | The Zultys phones are just aastra phones |
| 02:21.35 | antiwire | I like the polycom phones more than aastra |
| 02:21.55 | citywok | i like the polycom browser wayyyy more than the aastra. but i like most everything else about the aastra over the polycom. |
| 02:22.04 | trapa | These are neat phones. We were considering Snom phones since they're the same pricepoint |
| 02:22.23 | trapa | cisco we use at the office but is out because they require too much central hardware to set up remotely |
| 02:28.27 | raden | where is p3nguin ? |
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| 02:34.02 | citywok | trapa: i just got an ip650 to play with, i use aastra 57i's at my office. for provisioning the aastra wins, heh. |
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| 03:40.54 | delroy | any using TDM cards? |
| 03:41.19 | delroy | analog cards? like a digium or sangoma pci card> |
| 03:41.21 | delroy | ? |
| 03:49.16 | fenrus | yes, someone is probably using them |
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| 04:13.33 | citywok | delroy: you're better off just asking your question |
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| 04:35.29 | dive | Hi, I have an account at draytel but can't get asterisk to connect. Console shows it's registered ok and it dials the number but it never connects. |
| 04:35.34 | dive | [outgoing_calls] exten => _X.,1,Dial(SIP/${EXTEN}@draytel.org) exten => _X.,n,hangup() |
| 04:35.59 | dive | Any clues? |
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| 05:21.04 | citywok | dive it could be a nat issue |
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| 05:52.36 | dive | citywok, should the console show dialed sip:0123435@draytel.org or just 012344@... ? |
| 05:53.33 | dive | I keep getting busy/congested |
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| 06:37.18 | kaldemar | dive: you should dial using the peer defined in sip.conf. see the sample sip.conf for examples on dialstrings. |
| 06:37.48 | dive | ok will have a read through |
| 06:40.02 | kaldemar | if you can't get it to work, enable sip debug to see what's wrong. |
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| 06:49.37 | dive | I'm seeing a lot of this: |
| 06:49.38 | dive | Retransmitting #6 (NAT) to 192.168.1.3:5060: |
| 06:49.38 | dive | SIP/2.0 403 Forbidden |
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| 06:55.42 | kaldemar | dive: retransmissions to a private address sounds like a nat issue. |
| 06:57.59 | dive | I've tried both nat=yes/no |
| 06:58.09 | dive | doesn't seem to make any difference |
| 06:58.24 | kaldemar | where? and with what kind of a dialstring? |
| 06:58.46 | dive | in sip.conf section |
| 06:59.41 | kaldemar | where in sip.conf? |
| 06:59.51 | kaldemar | ~pb |
| 06:59.51 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
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| 07:00.51 | dive | this is commented out but shows you |
| 07:00.52 | dive | http://pastebin.slackadelic.com/p/VIkfoB52.html |
| 07:01.13 | dive | I also tried in [general] |
| 07:01.19 | dive | (nat) |
| 07:02.02 | kaldemar | in [general], it means that your asterisk box is behind a NAT. in a peer section it means that a peer is behind a NAT. |
| 07:03.26 | dive | ah |
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| 07:05.04 | dive | same |
| 07:05.30 | kaldemar | your dialstring is to blame. the peer definition is not used. |
| 07:06.14 | dive | I don't really see what I should use from that sample sip.conf |
| 07:06.33 | dive | numbers are usually sip:012345678:draytel.org |
| 07:07.33 | kaldemar | SIP/devicename/extension |
| 07:08.15 | kaldemar | do they register to you? if not, you need to define host= as something else than dynamic. meaning an ip address or a host name. |
| 07:10.55 | dive | well with draytel I register with them, my laptop/softphone registers ok with the pbx |
| 07:11.16 | dive | draytel is a voip provider |
| 07:11.28 | dive | so there is no /device/extension |
| 07:11.43 | dive | just a regular phone number @draytel.org |
| 07:11.46 | kaldemar | irrelevant. your asterisk box needs to know their address to know where to send your invites. |
| 07:12.21 | dive | ok I can put in hostname of laptop |
| 07:12.25 | kaldemar | the device is where you define where the messages go and how asterisk authenticates. |
| 07:12.41 | kaldemar | no, put the host of their service, be that draytel.org or whatever. |
| 07:12.59 | dive | ah right |
| 07:13.29 | kaldemar | and then dial SIP/draytel/<number> |
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| 07:18.02 | dive | my softphone can't register with host=draytel.org - forbidden |
| 07:19.39 | kaldemar | why should it? |
| 07:19.49 | kaldemar | you have asterisk to connect to them, right? |
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| 07:21.07 | dive | yeah but I need to laptop/softphone to register with asterisk? |
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| 07:22.01 | kaldemar | technically no, but that's most likely what you want. |
| 07:23.08 | kaldemar | registrations are separate from making actual calls. registrations are just a way to let the other end know where you are. |
| 07:23.32 | dive | right |
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| 08:03.48 | dive | kaldemar, go it. Thanks for your help. exten => _X.,n,Dial(SIP/${EXTEN}@draytel,wW) and an extra sip.conf section |
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| 08:04.27 | dive | IE Before I was trying to use the same one for my device as draytel |
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| 08:07.24 | schmidts | good morning |
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| 08:26.06 | dive | well outgoing is working but now incoming is showing always busy |
| 08:27.24 | dive | but when I call in nothing shows at in console |
| 08:27.30 | dive | at all* |
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| 09:40.34 | _omer | Hello |
| 09:41.09 | _omer | how to get the 10 digits from right from ${CALLERID(NUM)} ??? |
| 09:41.23 | kaldemar | which 10? |
| 09:41.28 | _omer | right |
| 09:41.35 | _omer | from right side |
| 09:41.38 | kaldemar | 10 last? |
| 09:41.50 | tuxx- | ${CALLERID(num):-10} |
| 09:41.53 | tuxx- | me thinks |
| 09:42.02 | _omer | for example: 1-212-5551212 ... I am not sure about the lenght of caller id num .... so I need 2125551212 |
| 09:42.32 | _omer | ${CALLERID(num):-10} ..... ok let me check .. |
| 09:43.27 | _omer | Works! ... thanks |
| 09:43.29 | _omer | take care.bye |
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| 10:05.42 | AdvoWork | does anyone know if its possible with trixbox, to listen to recordings(from the call monitor) on an actual phone? i.e Snom 300? or can you only do this via the web interface? |
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| 10:06.13 | WIMPy | 2nd door left, #trixbox |
| 10:19.53 | schmidts | wimpy the other left or the "right" left :D |
| 10:20.40 | WIMPy | Now that you ask, it might have been two floord down. |
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| 11:06.36 | AdvoWork | dont suppose anyone has ever used snom 300 phones? |
| 11:09.57 | schmidts | only if i need something below a table or something if it was wiggling :D |
| 11:10.11 | schmidts | but i heard they are also good to keep doors open ;) |
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| 11:24.20 | pluesch0r | hi everybody! |
| 11:25.34 | pluesch0r | unfortunately, we were a victim of sip hijacking today and lost some money .. now i'd like to close the hole, obviously. i got the tip to add allowguest=no to the general section of sip.conf - unfortunately, if i do that, we are no longer able to receive inbound calls. |
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| 11:27.34 | pluesch0r | http://www.pastie.org/1444927 <= our sip.conf file .. |
| 11:27.40 | pluesch0r | any idea what i'm doing wrong there? |
| 11:27.56 | kaldemar | pluesch0r: make unauthenticated calls land to a context that doesn't include extensions allowing outbound calls. |
| 11:28.41 | pluesch0r | kaldemar: sounds reasonable. how do i do that? can you point me to a tutorial or something? |
| 11:29.45 | kaldemar | how is your dialplan structured now? |
| 11:30.31 | kaldemar | to what context do unauthenticated calls go to? |
| 11:31.49 | pluesch0r | as far as i can see, there's some std-dialout function that gets called for _+X. |
| 11:32.12 | pluesch0r | but that's inside the from-sip context. |
| 11:35.28 | pluesch0r | actually .. i'm not even sure if unauthenticated calls get a special treatment, just by looking at this. |
| 11:36.14 | kaldemar | just a matter of context in your dialplan. |
| 11:37.49 | pluesch0r | since 'from-sip' is defined as context in the general config part of sip.conf, i take it that that's the general context. |
| 11:38.13 | pluesch0r | would it help to change the name of that context, in the general section? |
| 11:38.29 | pluesch0r | so that undefined sip logins get that context? |
| 11:39.23 | pluesch0r | or better yet: if i define allowguest=no, i get messages like that in my log: chan_sip.c:19171 in handle_request_invite: Failed to authenticate device - how do i allow those devices, then? |
| 11:41.19 | kaldemar | change the context under general to something that does not allow outbound calls. |
| 11:42.08 | kaldemar | there is no such thing as an undefined sip login. or a sip login for that matter. just authentication upon calls or registrations. |
| 11:43.04 | kaldemar | if you define allowguest=no, you need to define every peer that you accept calls from. |
| 11:43.12 | pluesch0r | kaldemar: so .. i've renamed the general context to 'alien' |
| 11:43.29 | pluesch0r | i've copied my dialplan settings for the from-sip context and named the new context alien, obviously. |
| 11:43.52 | kaldemar | also make sure that you don't have anything in a "default" context, in case of a fallback. |
| 11:43.57 | pluesch0r | after that, i've disabled those _+X. extensions. as, obviously, aliens shouldn't be able to call out. |
| 11:44.16 | pluesch0r | there's no default context entry in my extensions.ael file. |
| 11:45.31 | schmidts | pluesch0r maybe in your extensions.conf file? |
| 11:45.42 | pluesch0r | schmidts: that's empty. |
| 11:46.07 | pluesch0r | other than that .. i've got an extensions.lua file. seems to be unused. |
| 11:46.09 | schmidts | pluesch0r what does your dialplan show gives you? do you see a default there |
| 11:46.14 | pluesch0r | is that one interpreted as well? |
| 11:46.40 | schmidts | yes but normaly there shouldnt be a default extension in it AFAIK |
| 11:47.01 | pluesch0r | nope, no default context. |
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| 11:57.25 | WIMPy | AdvoWork: Only 320/360/370 |
| 12:00.21 | AdvoWork | WIMPy, just having the weird issue where the update settings url keeps getting overwritten so i cannot do a firmware upgrade |
| 12:02.16 | WIMPy | Settings or Firmware? |
| 12:04.30 | AdvoWork | firmware, but i tried to do it automatically, just testing manually to see if anything happens |
| 12:05.10 | WIMPy | Never specify a version in the settings. |
| 12:10.01 | AdvoWork | why? |
| 12:12.56 | WIMPy | Because that will override the real version. Only use the firmware update URL. |
| 12:13.43 | AdvoWork | ahh ok |
| 12:14.00 | AdvoWork | argh, its not updated after all that |
| 12:14.38 | AdvoWork | basically followed the full guide: http://wiki.snom.com/Firmware/V7/Update_Description#Manual_Update and have specified each link, in its correct order, from the software update/firmware page as it specifies.. rebooted, loaded the files, but still states old version :S |
| 12:15.14 | AdvoWork | hmm but: Firmware-URL:http://provisioning.snom.com/download/fw/snom300-7.3.30-SIP-bf.bin thats what it shows, somethings not right |
| 12:18.05 | WIMPy | You can eiter use the manual update and specify the firmware directly. Or you specify and update URL to do it automatically. You can use one from the snom server to always get the latest release offe |
| 12:18.08 | WIMPy | offered. |
| 12:22.10 | AdvoWork | ive done it manually, specifying the urls, ie 3 different urls in the order it states, it says loading etc and seems to work, but still keeps the same versioning |
| 12:23.26 | WIMPy | Why three URLs? |
| 12:24.52 | AdvoWork | its just what it says to do: http://wiki.snom.com/Firmware/V7/Update_Description#Manual_Update |
| 12:25.05 | AdvoWork | "The full process of updating from v6 to v7 requires the following steps" and ive done all 3.. |
| 12:26.26 | WIMPy | Ah, V6. Ok. It's easier to use the automatic update. You just have to give it the update URL and the rest will happen automatically. |
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| 12:32.22 | AdvoWork | i had tried that originally though, i keep giving it the update url, save, reboot, but same problem, it didnt appear to even start updating |
| 12:33.16 | WIMPy | After you give it the URL, you should see "SW" on your display. |
| 12:34.16 | WIMPy | Then you get an update option in the maintenance menu. |
| 12:35.13 | WIMPy | Does the phone have internet acces? |
| 12:35.18 | WIMPy | s |
| 12:36.38 | AdvoWork | i saw SW when i did it manually, and what i just did was: Advanced, its set to update automatically, i added the url: http://provisioning.snom.com/update6to7/update_once.php pressed Save, Reboot, but i didnt see SW this time |
| 12:38.47 | *** join/#asterisk Poincare (~jefffnode@213.219.184.23.wls.msr03bkc3.adsl.static.edpnet.net) |
| 12:40.03 | WIMPy | It's been quite some time since I saw such an ancient version. I used the atutomatic update URL. |
| 12:40.45 | WIMPy | It took quite some time incl. some reboots, but it all happened automatically. |
| 12:40.48 | AdvoWork | hmm actually, i see: Application-Version:snom300-SIP 6.5.13 Firmware-URL:http://provisioning.snom.com/download/fw/snom300-7.3.30-SIP-bf.bin so is that correct? or still wrong |
| 12:42.27 | AdvoWork | hmm nope, its still the wrong firmware, doh |
| 12:43.25 | *** join/#asterisk E-bola (~bola@188.120.76.228) |
| 12:44.38 | *** join/#asterisk SeTTleR (~bernd@p5DDEE030.dip.t-dialin.net) |
| 12:45.31 | WIMPy | I can't find that URL :-( |
| 12:46.04 | WIMPy | Did it try to flash at all? |
| 12:47.22 | AdvoWork | didnt appear to, when i do it manually, it does say SW, automatically it wont which is wierd |
| 12:48.08 | WIMPy | But did it stat to update after you saw SW? |
| 12:48.34 | AdvoWork | yeah, it says loading file(for say 5 minutes), and then boots back up properly, but then shows the old firmware version again :S |
| 12:49.06 | WIMPy | Then it didn't like the file. |
| 12:49.53 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 12:49.54 | WIMPy | After loading the file it should go on with erasing and flashing along with the warning not to switch off power during the process. |
| 12:50.36 | AdvoWork | ahh, so what can i do? can i find out why it doesnt like the file? |
| 12:51.11 | WIMPy | Not sure you find that in the log. |
| 12:52.29 | *** join/#asterisk m_tadeu (~quassel@89.180.156.159) |
| 12:53.56 | WIMPy | Updates from Versions <V7 are a little special. |
| 12:56.16 | *** join/#asterisk Visine (~Visine@unaffiliated/visine) |
| 12:56.53 | AdvoWork | i dont know what else to try |
| 12:56.58 | Visine | is there any standard tweaks to reduce echo for trixbox? |
| 12:57.42 | kaldemar | Visine: #trixbox |
| 12:57.56 | Visine | ty |
| 12:58.33 | WIMPy | If you try it manually, you should find something in the log, I guess. |
| 12:59.37 | *** join/#asterisk Tim_Toady (~moi@178.128.140.163.dsl.dyn.forthnet.gr) |
| 13:03.10 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.89) |
| 13:14.01 | *** join/#asterisk ironm (~ironm@fwj00.e-fon.ch) |
| 13:15.23 | *** join/#asterisk asterisk-learner (~chatzilla@77.42.241.114) |
| 13:24.32 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 13:26.48 | *** join/#asterisk clintc (~clintc@n128-227-125-254.xlate.ufl.edu) |
| 13:27.50 | *** join/#asterisk danielwa (~user@188-192-59-240-dynip.superkabel.de) |
| 13:30.34 | *** join/#asterisk jkroon (~jkroon@dsl-241-233-157.telkomadsl.co.za) |
| 13:33.53 | *** join/#asterisk russellb (~russell@asterisk/digium-open-source-team-lead/russellb) |
| 13:33.53 | *** mode/#asterisk [+o russellb] by ChanServ |
| 13:33.59 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:21b:63ff:feb5:e13b) |
| 13:44.54 | *** join/#asterisk fofware (~Fabian@200.117.29.167) |
| 13:47.07 | *** join/#asterisk BlackBishop (dexter@d3xt3r01.tk) |
| 13:47.09 | *** join/#asterisk Ox41464b (~chatzilla@unaffiliated/Ox41464b) |
| 13:47.29 | BlackBishop | anyone can recommend a cheap (usb preferably) gsm module for asterisk ? |
| 13:48.37 | fofware | hello guys, I'm looking for info about database definition for Asterisk 1.6, i did found it for sipusers suppeers, but I don't find for sipregs. Can any one give a link? |
| 13:49.01 | Ox41464b | Hi, im looking for asterisk implementations which works both ways phone, ip and easy to plug my own works (or build in for CRM), any suggestion where should i start comparing ? |
| 13:51.21 | schmidts | fofware sipregs are only stored in the astdb not in realtime AFAIK |
| 13:54.33 | fofware | schmidts: thaks, so I will keep sipregister in my sipconf_db |
| 13:56.07 | kaldemar | actually there is a sipregs map for realtime. |
| 14:00.36 | *** join/#asterisk recluze (~recluze@175.145.83.247) |
| 14:01.12 | BlackBishop | well .. any recomandations ? |
| 14:02.56 | fofware | kaldemar: in 1.6 exists sipregs to map them, but I don't have idea how define db_table |
| 14:07.36 | *** join/#asterisk Arsenick (~y@fedora/Arsenick) |
| 14:10.31 | Katty | gooooooooooooooooooooooooooooooooooood monday morning everyone! |
| 14:10.46 | *** join/#asterisk joachim_- (~joachim@post.comvie.no) |
| 14:10.53 | Katty | it's time to wake UP! |
| 14:10.56 | Katty | and go OUTSIDE |
| 14:10.59 | Katty | and build SNOWMEN! |
| 14:11.13 | joachim_- | hahah.. gotta finish work first |
| 14:11.47 | joachim_- | got some problems with the Asterisk Manager interface |
| 14:12.16 | joachim_- | how can i filter out unknown events so it doesnt fill up my logs with "!#¤#"¤%"%"¤#% ? |
| 14:12.22 | joachim_- | e.g. Ignoring unknown event 'RTCPSent' |
| 14:12.45 | joachim_- | same goes for 'RTCPreceieved" and DTMF |
| 14:15.22 | *** join/#asterisk Jasnejac (~kvirc@81.91.106.59) |
| 14:15.30 | schmidts | Katty its almost time to go to bed (atleast i fell like this) and we have 10? C now, so no way to build a snow man ;) |
| 14:16.52 | *** join/#asterisk skrusty (~skrusty@83.166.176.39) |
| 14:17.02 | drmessano | No work today (unless someone goes off the air) WOOOOOO |
| 14:22.41 | BlackBishop | goes google for some asterisk compatible usb gsm dongles |
| 14:30.20 | Katty | i read that as usb gsm goggles. |
| 14:30.21 | *** join/#asterisk oej (~olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
| 14:30.41 | BlackBishop | :)) |
| 14:30.45 | BlackBishop | might as well... |
| 14:30.47 | BlackBishop | can't find any .. |
| 14:31.24 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 14:31.45 | WIMPy | There are many but most don't support voice or have it disabled. |
| 14:31.54 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 14:33.33 | BlackBishop | well .. I'd like one that supports voice .. |
| 14:33.55 | WIMPy | thought so |
| 14:33.58 | BlackBishop | can ya' give some examples ? ( I wanna see some prices .. ) |
| 14:34.50 | WIMPy | I know that E1550 can support voice. |
| 14:35.11 | WIMPy | I guess you should look at the documentation of cahn_datacard. |
| 14:35.27 | BlackBishop | huawei ? |
| 14:36.01 | BlackBishop | what I'll try to do is have 3 of these .. and be able to call from my cell to one of them .. and then ( typing some code ) redirecting my call through one of the other 2 |
| 14:38.01 | WIMPy | I'm not sure there are any other vendors left. |
| 14:38.12 | WIMPy | So, yes, Huawei. |
| 14:38.40 | BlackBishop | and you say I could do what I want with 3 of those thingies and an asterisk box ? :D |
| 14:39.12 | WIMPy | Yes |
| 14:39.14 | *** join/#asterisk salimb (~chatzilla@81.5.139.17) |
| 14:39.18 | BlackBishop | interesting... |
| 14:39.53 | BlackBishop | googles some prices |
| 14:40.02 | WIMPy | But I couldn't try. My E1550 does not have voice enabled. |
| 14:41.08 | BlackBishop | hardware locked ? :/ isn't it something that the sim in it would lock ? |
| 14:41.10 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 14:41.53 | WIMPy | I don't think so. |
| 14:42.35 | BlackBishop | what if I buy some 1552 off ebay ? |
| 14:42.39 | BlackBishop | they say "unlocked" :/ |
| 14:43.16 | WIMPy | Unlocked usually only means without SIMlock or Netlock. |
| 14:43.33 | BlackBishop | ahm .. |
| 14:43.40 | WIMPy | And there are code generators available for that part. |
| 14:44.05 | BlackBishop | hmmm |
| 14:44.07 | BlackBishop | "Huawei E1550 HSPA 3G Modem USB Voice Broadband Unlocked" |
| 14:44.11 | BlackBishop | found one of these... |
| 14:44.26 | WIMPy | But I haven't managed to get at the voice features. Might even be another firmware. |
| 14:44.50 | WIMPy | The whole voice thing seems very grey. Not much to be found about that. |
| 14:45.02 | BlackBishop | http://cgi.ebay.com/Huawei-E1550-HSPA-3G-Modem-USB-Voice-Broadband-Unlocked-/140395225568?pt=LH_DefaultDomain_0&hash=item20b03521e0 |
| 14:45.53 | WIMPy | The only thing that seems to be clear is that Huawei won't give any support. |
| 14:46.23 | BlackBishop | I'll buy one if you want to try it out on my server and see how it goes .... |
| 14:46.24 | BlackBishop | :D |
| 14:46.48 | WIMPy | Looks like that should work. |
| 14:47.09 | WIMPy | But as I said: You should check the documentation of chan_datacard. |
| 14:47.28 | BlackBishop | googles |
| 14:48.59 | drmessano | Personally, I think something like a single GSM channel is a waste of time |
| 14:48.59 | BlackBishop | ok .. so I should buy some of those .. |
| 14:49.54 | drmessano | SIP over the data connection of a GSM dongle gives you many more options than using a single GSM channel |
| 14:49.59 | drmessano | voice* |
| 14:50.14 | BlackBishop | I wanna redirect my calls to one of the other 2 sims depending on my destination .. |
| 14:52.35 | *** join/#asterisk draicone (~draicone@wikimedia/Draicone) |
| 14:55.04 | *** join/#asterisk edwin_quijada (~edwin_qui@190.166.166.254) |
| 14:55.53 | edwin_quijada | I have a Openvox A800P card with 8 analog ports but I have problem installing DAHDI |
| 14:56.07 | edwin_quijada | I get an error when I load the module |
| 14:56.11 | edwin_quijada | I get this |
| 14:56.13 | edwin_quijada | ERROR[3798]: chan_dahdi.c:7979 mkintf: Signalling requested on channel 1 is FXO Kewlstart but line is in FXS Kewlstart signalling |
| 14:59.25 | kaldemar | edwin_quijada: you have the channel defined to use fxs signalling in DAHDI's system.conf but fxo in chan_dahdi.conf. |
| 15:01.31 | edwin_quijada | I fix it |
| 15:02.30 | edwin_quijada | but now I get thid http://pastebin.com/M1SCBAnp |
| 15:02.47 | salimb | can anyone point me to a good guide on asterisk 1.6 voicemail config with cyrus imap and ldap authentication? |
| 15:05.35 | salimb | i have cyrus authenticating with ldap and i was hoping to integrate my asterisk voicemail feature to use it too |
| 15:07.10 | salimb | not having much joy with my google searches |
| 15:08.10 | kaldemar | edwin_quijada: are the lines connected? |
| 15:08.13 | Chainsaw | salimb: You may want to speak to kerframil in #gentoo-voip when he's around. |
| 15:08.31 | edwin_quijada | yes |
| 15:08.31 | Chainsaw | salimb: He's big on that IMAP voicemail stuff. |
| 15:08.57 | edwin_quijada | the cli doesnt show anything when i call |
| 15:11.45 | salimb | Chainsaw: thanks |
| 15:12.12 | Chainsaw | salimb: Perhaps use memoserv to send a message? I don't have his e-mail on me right now. |
| 15:12.53 | salimb | Chainsaw: ok I will continue to investigate |
| 15:17.09 | *** join/#asterisk w0ls0n (~w0ls0n@94-120-181-66.dsl.sacoriver.net) |
| 15:17.15 | w0ls0n | good morning all |
| 15:17.17 | w0ls0n | are there any sip peers that offer free service just for one line? I am just looking at testing out asterisk |
| 15:17.29 | w0ls0n | US Phone # if possible |
| 15:18.00 | russellb | it's not SIP, but google voice. |
| 15:21.46 | w0ls0n | I could use asterisk with google voice? |
| 15:22.21 | russellb | yes |
| 15:22.23 | russellb | ~googlevoice |
| 15:22.23 | infobot | For information on setting up calls with google using Asterisk 1.8, see https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google, or see the External Services chapter: http://ofps.oreilly.com/titles/9780596517342/ch17.html |
| 15:29.02 | edwin_quijada | There is something way to use Magic Account with Asterisk ? |
| 15:30.06 | *** join/#asterisk Tim_Toady (~moi@178.128.140.163.dsl.dyn.forthnet.gr) |
| 15:31.35 | *** join/#asterisk davevg-btwtech (~davevg__@24.115.249.195.res-cmts.senj.ptd.net) |
| 15:32.13 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 15:35.39 | drmessano | Asterisk 1.8 is badass |
| 15:35.46 | drmessano | Google Voice calling ++++ |
| 15:36.17 | w0ls0n | can I use it with 1.4? I had some issues with 1.8 |
| 15:36.52 | drmessano | No |
| 15:37.31 | drmessano | What issues have you had with 1.8? |
| 15:41.32 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 15:45.20 | w0ls0n | oh that sounds like a deal |
| 15:45.50 | w0ls0n | I had issues just starting it on freebsd. I went to centos and installed 1.4 and it installed flawlessly and had it running in about 10-15 mins |
| 15:46.34 | *** join/#asterisk The_Boy_Wonder (~The_Boy_W@asterisk/batman-developer/dvossel) |
| 15:50.55 | lirakis | hmm... i just updated a dns record and its propagated to my name servers. None of my polycom phones have updated though. |
| 15:51.08 | lirakis | does any one know if/how long polycom phones cache dns for? |
| 15:51.17 | lirakis | they dont seem to be honoring the ttl |
| 15:51.44 | *** join/#asterisk corretico (~corretico@201.201.44.82) |
| 15:54.46 | russellb | reboot them |
| 15:54.55 | russellb | you can force them to reboot using the "sip notify" CLI command |
| 15:55.17 | beek | russellb: Their config file needs to change for that to have any effect. |
| 15:55.24 | lirakis | russellb, not really an option. |
| 15:55.29 | fenrus | sounds a bit harsh |
| 15:55.36 | russellb | beek: ah |
| 15:56.13 | fenrus | might be able to disable client side caching somehow |
| 15:56.23 | *** join/#asterisk WindBack (~quassel@kirk.capitalinasdc.com) |
| 15:56.32 | lirakis | im curious how long they cache for |
| 15:56.45 | lirakis | it doesnt matter to me really if it is a 24 hours. |
| 15:56.58 | lirakis | i just wanted to know if its indefinite till reboot or some period of time |
| 15:57.18 | WindBack | Is there any documentations to do a correct billing when the caller transfer? |
| 15:57.41 | lirakis | i thought some one in here might have some expertise, since I do not know polycoms very well |
| 15:58.57 | fenrus | lirakis, and you are certain that the dns responds according to the new configuration? |
| 15:59.14 | lirakis | fenrus, yes 100% positive |
| 15:59.29 | fenrus | the well known fortgot-to-update-serial bug ;) |
| 15:59.31 | fenrus | "bug" |
| 15:59.47 | lirakis | fenrus, emacs has a zone mode that auto updates serial |
| 15:59.49 | lirakis | ;) |
| 16:00.01 | fenrus | one plus for emacs i guess :) |
| 16:01.47 | *** join/#asterisk evharten (~evharten@vpn.evertje.net) |
| 16:02.13 | beek | You mean the emacs operating system? |
| 16:02.24 | fenrus | of course. |
| 16:04.11 | fenrus | lirakis, http://support.polycom.com/global/documents/support/technical/products/voice/DNS_Cache_TB36033.pdf seems to contain alot of information about this |
| 16:04.23 | lirakis | fenrus, yeah i was lookin at that |
| 16:05.00 | lirakis | meh. i use both vim and emacs. they both have their own + and - |
| 16:05.33 | lirakis | fenrus, that seems to tell you how to setup a static dns cache though ... which is basically the opposite of what i want |
| 16:05.44 | lirakis | and im mostly curious about what is "default" behavior |
| 16:06.15 | WindBack | In which way can i do a correct billing when the caller transfer the call? |
| 16:07.14 | fenrus | lirakis, yea |
| 16:07.34 | fenrus | lirakis, i guess it does not state any default settings of the internal dns cache.. |
| 16:10.44 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 16:13.55 | *** join/#asterisk tzanger (~tzanger@gromit.mixdown.ca) |
| 16:14.56 | *** join/#asterisk am1n0 (yiffstar66@unaffiliated/devemo) |
| 16:20.47 | Katty | hai |
| 16:20.53 | Katty | distributes granola bars |
| 16:21.11 | Katty | sweet and salty cashew ones, to be specific |
| 16:23.55 | w0ls0n | Oats and Honey pls kthnx |
| 16:23.56 | lirakis | hmmm ... this seems to be an issue others have had too |
| 16:24.06 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-172.dslextreme.com) |
| 16:24.38 | *** join/#asterisk sourcode (~code@ppp-58-8-160-95.revip2.asianet.co.th) |
| 16:25.32 | *** join/#asterisk evharten (~evharten@vpn.evertje.net) |
| 16:26.56 | Katty | wow. i just learned someone i went to high school died in the military |
| 16:27.04 | Katty | and recently |
| 16:27.25 | Katty | he had a crush on me in 1st grade. |
| 16:27.31 | Katty | i feel awful :< |
| 16:29.41 | BlackBishop | well .. I wonder if the girl I had a crush in the 1st grade will ever remember me .. |
| 16:29.47 | BlackBishop | I ate her chocolate ... |
| 16:30.00 | BlackBishop | and ran her down the stairs when going out from school... |
| 16:32.48 | beek | hugs katty |
| 16:33.22 | BlackBishop | can I hug Katty too ? |
| 16:39.37 | Katty | hugs beek |
| 16:39.42 | Katty | eyes BlackBishop suspiciously |
| 16:40.44 | *** join/#asterisk DaneoShiga (~DaneoShig@187.66.86.169) |
| 16:43.56 | *** join/#asterisk micols (~mio@rlogin.dk) |
| 16:44.01 | *** join/#asterisk VenomX (~venomx@189.110.39.203) |
| 16:44.19 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 16:47.18 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 16:47.27 | VenomX | Hi :) . Quick question with a possibly not so quick answer: Does dahdi wcfxo (X100P clone/MD3200) have the capability of decoding CallerID from DTMF? I'm in Brazil and I wonder if this feature will work withou using a DTMF to FSK converter. |
| 16:47.45 | russellb | Katty: sorry to hear that ... war is terrible. |
| 16:49.43 | Chainsaw | Can anyone tell me what this Cisco handset is doing wrong please: |
| 16:49.44 | Chainsaw | http://pastie.org/1445615 |
| 16:49.52 | Chainsaw | (It's an attempt at a blind transfer, which is failing miserably) |
| 16:52.27 | *** join/#asterisk WonTu (~WonTu@p57B55FAB.dip.t-dialin.net) |
| 16:52.41 | *** part/#asterisk WonTu (~WonTu@p57B55FAB.dip.t-dialin.net) |
| 16:53.08 | drmessano | Existing |
| 16:53.13 | drmessano | </sorry> |
| 16:53.32 | russellb | drmessano: that should be <sorry/> |
| 16:53.36 | beek | Katty -- mind a PM? |
| 16:53.48 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 16:53.58 | drmessano | 10 PRINT "SORRY" |
| 16:54.02 | *** part/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 16:54.02 | drmessano | 20 GOTO 10 |
| 16:54.04 | drmessano | RUN |
| 16:54.39 | drmessano | IS THAT GOOD ENOUGH FOR YOU? |
| 16:55.20 | drmessano | flicks the power switch and takes it back |
| 16:56.27 | *** join/#asterisk oej (~olle@2001:470:1f15:d79:225:ff:fe44:74ec) |
| 17:03.32 | ChannelZ | Chainsaw: well the 'retransmitting' is a bit of a worry |
| 17:05.36 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:05.52 | Chainsaw | ChannelZ: First time I'm finally getting an error message: |
| 17:05.53 | Chainsaw | [2011-01-10 17:05:13] WARNING[16471]: rtp_engine.c:1206 remote_bridge_loop: Channel 'SIP/45999-00000040<ZOMBIE>' failed to break RTP bridge |
| 17:06.50 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 17:14.22 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.113.14.dsl.dyn.forthnet.gr) |
| 17:15.02 | drmessano | Anyone using Asterisk + IPv6 here? |
| 17:16.02 | *** join/#asterisk McLazarus (~McLazarus@dogpile.mcallister.ws) |
| 17:16.36 | *** join/#asterisk moy (~moy@CPE003048b11058-CM0026f396812d.cpe.net.cable.rogers.com) |
| 17:16.57 | drmessano | I know there was a bug report opened/closed about using :: as bindaddr and externip/externhost no longer being honored on IPv4.. anyone run into a problems with this? |
| 17:29.06 | Visine | does anyone know of anyway to monitor if an asterisk phone server is up? |
| 17:29.16 | Visine | like a tool that monitors if calls can go our or in? |
| 17:32.25 | pabelanger | Visine: nagios |
| 17:32.44 | Visine | well I do use nagios |
| 17:33.04 | Visine | but my problem is somtimes calls arnt working outgoing or incoming |
| 17:33.09 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 17:33.16 | Visine | is there a bot then can make a test phone call in or out |
| 17:33.19 | Visine | and report back? |
| 17:33.30 | w0ls0n | ok people help me out. What is the issue when I have 2 softphones connected but we cannot hear each other? Port 5060 and 10000-11000 are forwarded to the right place. Please let me know |
| 17:33.31 | Visine | i use nagios to detect if our phone system is up or no |
| 17:35.23 | pabelanger | Visine: Shouldn't be to hard to write something. |
| 17:35.41 | pabelanger | we basically do the same thing with the Asterisk testsuite using python and starpy |
| 17:36.11 | pabelanger | w0ls0n: sounds like a NATing issue |
| 17:36.13 | pabelanger | ~sipnat |
| 17:36.13 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 17:36.13 | fenrus | if you find something pre-written, or if you start a project - gimme a shout |
| 17:37.53 | Visine | im digging around on google |
| 17:37.57 | w0ls0n | thanks |
| 17:37.59 | Visine | can't believe this isnt already out there |
| 17:38.15 | fenrus | it probably is |
| 17:38.37 | pabelanger | http://starpy.sourceforge.net/ |
| 17:42.40 | *** join/#asterisk Da-Geek (~Da-Geek@11.74.155.90.in-addr.arpa) |
| 17:45.25 | *** join/#asterisk clintc (~clintc@n128-227-5-241.xlate.ufl.edu) |
| 17:48.57 | Katty | beek: sure |
| 17:49.49 | jaytee | anyone know how to get around a Polycom 330 giving a Config file error. Error is 0x0? |
| 17:55.30 | carrar | fix the error? |
| 17:56.32 | *** part/#asterisk salimb (~chatzilla@81.5.139.17) |
| 18:01.41 | *** join/#asterisk coppice (~chatzilla@210.17.255.115) |
| 18:11.15 | beek | waves to jaytee |
| 18:13.34 | *** join/#asterisk visik7 (~Adium@unaffiliated/visik7) |
| 18:14.20 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
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| 18:31.47 | citywok | jaytee: sounds like a config issue |
| 18:33.07 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 18:33.08 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 18:33.14 | *** join/#asterisk dmz (~dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) |
| 18:43.17 | *** join/#asterisk nny (~Scott@cpe-174-107-201-103.sc.res.rr.com) |
| 18:44.01 | citywok | getting the polycom configs right took me a while the first time |
| 18:44.43 | *** join/#asterisk cusco (~tralala@49.192.54.77.rev.vodafone.pt) |
| 18:44.45 | cusco | hi |
| 18:44.49 | jaytee | citywok, yeah but I can't get into the menu to do a file system format |
| 18:44.52 | cusco | in a dialplan pattern |
| 18:45.05 | cusco | how do I match any number up to 9 chars long? |
| 18:45.12 | citywok | jaytee: do you have a tftp server set up that it's trying to hit? |
| 18:45.17 | cusco | 9 chars long OR less ? |
| 18:45.17 | jaytee | and the bootlog file keeps saying it can't find the "macaddress".cfg file |
| 18:45.24 | citywok | cusco _X. will get you 1 or more |
| 18:45.32 | jaytee | citywok, I'm using vsftpd |
| 18:45.34 | cusco | yes but then 10 is still valid |
| 18:45.42 | cusco | and I want to limit at 10 |
| 18:45.49 | citywok | then specify each valid length? |
| 18:45.55 | cusco | ok.. |
| 18:45.58 | citywok | you may have to just do 6, 7, 8, 9 seperately |
| 18:46.01 | cusco | is there function LENGHT() ? |
| 18:46.25 | citywok | jaytee: make sure there isn't a mac.cfg, if there is that will cause problems and override the 00000.cfg |
| 18:47.02 | cusco | there is LEN |
| 18:47.03 | cusco | ok thanks |
| 18:48.26 | citywok | cusco: nice. then you can use _X. and use LEN to make sure it's between 1 and 9. |
| 18:59.10 | leifmadsen | citywok: _X. actually will get you 2 or more |
| 18:59.15 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 18:59.32 | citywok | oh yea. hmm. any way to get digit only 1 or more? _. will get letters and other garbage. |
| 18:59.37 | leifmadsen | citywok: . means "match any 1 character" and ! means match "any 0 or more characters" |
| 18:59.40 | leifmadsen | _X! |
| 18:59.53 | leifmadsen | any length string starting iwth 0-9 |
| 19:00.03 | citywok | neat, thanks for the tip |
| 19:00.10 | leifmadsen | any 2 or more character string starting with 0-9 would be _X. |
| 19:00.23 | citywok | yea, makes sense. i've never had to worry about that so never dug in to it. |
| 19:00.43 | leifmadsen | Whenever using . or ! make sure you use FILTER() to get rid of characters you don't expect |
| 19:01.12 | leifmadsen | README-SERIOUSLY.bestpractices.txt |
| 19:01.15 | leifmadsen | in your source |
| 19:01.24 | citywok | i only use . for international 011 dialing, thankfully. |
| 19:01.30 | leifmadsen | E1 ISDN is 30B + 2D is it not? |
| 19:01.39 | leifmadsen | PRI |
| 19:01.47 | citywok | no idea |
| 19:02.02 | leifmadsen | I'm sure someone from Europe can confirm if I'm crazy or not.... |
| 19:02.09 | citywok | hmm. i do have a couple E1's plugged in to cisco's |
| 19:02.12 | citywok | want me to go look at the configs? |
| 19:02.23 | citywok | i didn't configure them, but i have access to them. |
| 19:02.29 | WIMPy | What? |
| 19:03.22 | WIMPy | Ah, no 30B, 1D and one unused or for sync. |
| 19:03.48 | leifmadsen | ok, so still 32 channels then... |
| 19:03.55 | leifmadsen | regardless of how they are config'd I guess |
| 19:03.58 | citywok | controller E1 0/0 channel-group 0 timeslots 1-31 |
| 19:04.22 | WIMPy | correct |
| 19:05.03 | leifmadsen | correct? |
| 19:05.08 | leifmadsen | me? |
| 19:05.15 | WIMPy | yes :-) |
| 19:10.39 | *** join/#asterisk mr_ian_ (~mr_ian@S0106001b63f49383.du.shawcable.net) |
| 19:13.56 | WIMPy | Shit. SIP channels piling up again :-( |
| 19:14.18 | ChannelZ | GIT DEH PLUNGER, MABEL! |
| 19:14.33 | citywok | WIMPy: ? |
| 19:14.39 | citywok | WIMPy: channels not closing out properly? |
| 19:15.08 | WIMPy | I've got a box with lots of stuck PUBLISH dialogs. |
| 19:15.57 | citywok | from the phone, or provider? |
| 19:16.12 | WIMPy | I think it could have something to do with the fact that they are associated with peer <guest> and I don't allow guests on that box. |
| 19:16.17 | WIMPy | From phones. |
| 19:18.50 | WIMPy | ... because it doesn't happen here and that's the most obvious difference that came to my mind. |
| 19:19.16 | citywok | yea not sure, i've never worked with guest. is that used for some of the automated provisioning or? |
| 19:19.52 | WIMPy | That must be about device states. |
| 19:20.16 | WIMPy | I told the phones to shut up, but that didn't help. |
| 19:21.28 | WIMPy | Oh, BTW: Is there a way to set a custom devstate at startup? |
| 19:22.33 | citywok | WIMPy: does this help any? http://www.asterisk.org/node/48325 |
| 19:24.04 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 19:24.58 | WIMPy | That's about what I have running, but I'd like that devstate to be INUSE when Asterisk is started. |
| 19:28.57 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v000-079.mobile.uci.edu) |
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| 19:50.31 | ScanEarth | hello all |
| 19:51.02 | ScanEarth | anyone here with knowledge aboute app_rpt module? |
| 19:51.09 | ScanEarth | -e |
| 19:52.05 | *** join/#asterisk WindBack (~quassel@kirk.capitalinasdc.com) |
| 19:52.26 | cusco | hi... |
| 19:53.17 | WindBack | Why when I do a tranfer, the tranfered channel is renamed to AsyncGoto/Channel N<ZOMBIE> |
| 19:53.18 | WindBack | ? |
| 19:53.19 | cusco | in a call file, I commented out "#Channel: Local/90210332311@incoming-pri" and added "Application: Playback" and "Data: /tmp/cusco" |
| 19:53.25 | WindBack | This is during the transfer |
| 19:53.29 | cusco | asterisk now states: pbx_spool.c:242 apply_outgoing: At least one of app or extension must be specified, along with tech and dest in file /var/spool/asterisk/outgoing/tecnomaster.call |
| 19:53.40 | cusco | what am I doing wrong? |
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| 20:20.12 | *** join/#asterisk cyberscan (~cyberscan@8.26.248.214) |
| 20:20.55 | cyberscan | Anyone here know anything about the Asteriskwin32 project? |
| 20:21.25 | bjhaid | not sure think asteriskwin32 was discontinued, google it |
| 20:21.54 | cyberscan | Is it a true port of the Asterisk software, a complete rewrite, or an entirely different project? |
| 20:29.37 | tzafrir_laptop | cyberscan, a port of asterisk |
| 20:30.05 | tzafrir_laptop | If anybody really cares about it, the best place to start is by reviving the cygwin port |
| 20:30.13 | tzafrir_laptop | making it at least build, for starters |
| 20:30.16 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 20:31.56 | leifmadsen | cyberscan: that project is probably very old, very out of date, and very unsupported :) |
| 20:32.36 | leifmadsen | wow, 1.2.14 |
| 20:32.42 | leifmadsen | ya, that is crazy old |
| 20:32.46 | cyberscan | tzafrir_laptop Thanks for the information. I have used the Linux version for years and was trying to find out if this was the real deal for my students who do not want to install Linux. |
| 20:33.15 | cyberscan | I didn't find a mention of it on the Asterisk.org website. |
| 20:33.23 | tzafrir_laptop | some sort of virtual machine is probably a simpler way to go |
| 20:33.33 | leifmadsen | agree |
| 20:33.35 | leifmadsen | agreed * |
| 20:33.46 | cyberscan | I also know how malware writers can load search engines with fake results too. |
| 20:33.49 | leifmadsen | they could probably even follow the documentation at http://ofps.oreilly.com |
| 20:34.07 | leifmadsen | basically asteriskwin32 has been defunct for a few years now, so I wouldn't bother (and it probably won't work) |
| 20:34.09 | cyberscan | One of my students has it on their vbox. |
| 20:34.31 | leifmadsen | using VMware Player or something is probably a better way to go |
| 20:34.52 | leifmadsen | tzafrir_laptop: thanks for your comments -- I think I got them all integrated today |
| 20:35.14 | tzafrir_laptop | ok, great |
| 20:35.17 | cyberscan | Ok. Thanks, I will discourage them from even trying it. |
| 20:35.22 | cusco | in a call file, I commented out "#Channel: Local/90210332311@incoming-pri" and added "Application: Playback" and "Data: /tmp/cusco" |
| 20:35.28 | cusco | asterisk now states: pbx_spool.c:242 apply_outgoing: At least one of app or extension must be specified, along with tech and dest in file /var/spool/asterisk/outgoing/tecnomaster.call |
| 20:35.36 | cusco | what am I doing wrong? |
| 20:35.48 | *** join/#asterisk i_heart_asterisk (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 20:35.52 | i_heart_asterisk | hi |
| 20:36.16 | Corydon76-home | cusco: you cannot comment out Channel. Channel is mandatory |
| 20:37.01 | cusco | err.. sorry I thought I read that on the wiki I could replace channel by app |
| 20:37.02 | tzafrir_laptop | leifmadsen, will those changes be available on-line? |
| 20:37.18 | Corydon76-home | No, you can replace Extension with Application |
| 20:37.42 | Corydon76-home | The wiki is often wrong. |
| 20:38.39 | i_heart_asterisk | anybody notice when in the voicemail menu, if you push '2' to save message during message it just repeats message from beginning? |
| 20:39.18 | Corydon76-home | 2 isn't for saving messages |
| 20:39.52 | Corydon76-home | 9 is for saving messages |
| 20:40.28 | i_heart_asterisk | ah sorry, im confused because i changed asterisk's app_voicemail.c |
| 20:40.36 | i_heart_asterisk | i made 2 save message |
| 20:40.42 | Corydon76-home | Then it's unsupported |
| 20:40.51 | i_heart_asterisk | buti i see now it must be something i did in code |
| 20:40.56 | i_heart_asterisk | thanks tho |
| 20:43.20 | cusco | thanks Corydon76-home |
| 20:43.24 | cusco | another question... |
| 20:43.48 | Katty | pants |
| 20:43.50 | Katty | i ares jello |
| 20:43.58 | cusco | Unable to load module cdr_csv.so |
| 20:43.59 | cusco | Command 'module load cdr_csv.so' failed. |
| 20:44.07 | cusco | I can't figure out why I can't load that module |
| 20:44.41 | cusco | I think I need it to be able to use CDR variables on hangup |
| 20:45.17 | cusco | on 'h' extension, I have: NoOp(dialed time: ${DIALED_TIME} or $[${CDR(duration)}-${CDR(billsec)}] and total: ${DIALEDTIME} or ${CDR(duration)} ); |
| 20:45.42 | *** join/#asterisk freakynl (~freaky@unaffiliated/freakynl) |
| 20:46.23 | freakynl | Looking for an ATA that can use 2 SIP accounts on 1 line with a dial-plan |
| 20:46.25 | cusco | if the outbound call isn't answerd these comes almost correct like: |
| 20:46.27 | cusco | NoOp("SIP/218-00000a98", "dialed time: 33 or 33 and total: or 33 ") in new stack |
| 20:46.45 | cusco | but if call is answered... Executing [h@agents:12] NoOp("SIP/224-00000a8b", "dialed time: or 0 and total: 58 or 0 ") in new stack |
| 20:46.51 | cusco | dialed time is always 0 |
| 20:46.57 | cusco | what can I do about it? |
| 20:47.15 | cusco | I think I need to load that module |
| 20:47.47 | cusco | I don't know why it fails to load cdr_csv.so |
| 20:48.43 | cusco | :( |
| 20:48.43 | cusco | brb |
| 20:49.14 | leifmadsen | tzafrir_laptop: yes, the next time the book is updated from subversion they will be available |
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| 21:18.45 | *** join/#asterisk w0ls0n (~w0ls0n@94-120-181-66.dsl.sacoriver.net) |
| 21:18.55 | w0ls0n | sorry about that |
| 21:21.23 | *** join/#asterisk fullstop (~fullstop@static-173-210-91-4.ngn.onecommunications.net) |
| 21:21.48 | fullstop | does anyone here use vestech speech recognition software? |
| 21:22.40 | *** join/#asterisk Dovid (Dovid@213.8.121.90) |
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| 21:30.28 | bspidel | have a call center client using queues for skill sets of members. lower queues are getting greater service. any support for round robin queues so all get equal service? |
| 21:34.05 | citywok | bspidel: in a single queue you can use rrmemory as the type |
| 21:35.31 | bspidel | yes, but how can queue associate caller to member with correct skills? |
| 21:36.13 | bspidel | currently we use ivr script to place caller in correct queue |
| 21:36.46 | citywok | all * can do is weighted members, you have to do custom skills (agent one knows iphone, agent two knows droid) on your own. |
| 21:38.36 | bspidel | what is meant by custom skills? |
| 21:40.35 | bspidel | using queues to model skills works fine with 35 queues and 200 callers but lower numbered queues get better response. |
| 21:43.22 | citywok | well you said skill sets, so i'm assuming you mean skills that are custom to your business |
| 21:44.55 | *** join/#asterisk Lann (ab457d6c@gateway/web/freenode/ip.171.69.125.108) |
| 21:44.58 | Lann | heyas |
| 21:45.17 | Lann | Is astereisk capable of playing an audio file to a conference call? |
| 21:46.17 | WIMPy | sure |
| 21:46.44 | Lann | I have this project idea i've been kicking around |
| 21:47.01 | Lann | to make a game that you play by calling in and basically a MUD meant to be played over audio |
| 21:47.08 | Lann | which i'd probably call TrafficMUD heh |
| 21:47.18 | Lann | i wonder if it's been done |
| 21:47.47 | bspidel | citywok: I really meant, what do you mean by "on your own" did you have a technique in mind? |
| 21:48.13 | Lann | but the asterisk side would just mean that I need to have the ability to create conference calls and move people between the conference calls at will |
| 21:48.21 | citywok | i'm not really sure what your requirements are, so no i have no technique in mind |
| 21:48.48 | Lann | I think each "room" of the game would be a conference call that I'd create on-the-fly |
| 21:48.54 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 21:49.50 | fenrus | sounds nice :D |
| 21:49.59 | fenrus | doing input with DTMF? :D |
| 21:50.16 | bspidel | I am about to write some queue code to support this feature if I cannot find a workaround. |
| 21:50.40 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 21:50.48 | Lann | well that was my next question |
| 21:50.59 | Lann | what is the best voice to text service that works with asterisk? |
| 21:51.18 | Lann | i've already thoguht o use google text to speach for the other way around as well as pre-recorded audio |
| 21:52.20 | Lann | err well voice recog |
| 21:52.50 | Lann | I wish i could find one capable of calibrating to each user |
| 21:53.00 | Lann | because in this case i could save the calibration with a user's account |
| 21:53.20 | Lann | make them read the terms of service lol |
| 21:54.13 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 22:06.30 | Dovid | anyone know what would of caused this core dump ? http://dovid.net/backtrace.txt |
| 22:07.57 | citywok | 1.4.26 is ancient |
| 22:08.40 | Dovid | citywok: That I know. I want to know if there is something that is known abou it. i dont want to upgrade but will if I must |
| 22:08.56 | Dovid | i leave things as is unless there is an issue |
| 22:09.04 | Dovid | hey. where is TK ? |
| 22:09.09 | citywok | is it reproduceable |
| 22:09.17 | citywok | he got muted, so he doesn't come around anymore |
| 22:09.29 | Corydon76-home | TK was banned for being continuously rude to people |
| 22:09.37 | Lann | I'm thinking google TTS + cmusphinx + asterisk |
| 22:10.03 | Corydon76-home | Lann: There are multiple, none of which are Sphinx |
| 22:10.23 | Corydon76-home | Lann: I believe Vestec and Lumenvox are the two best known |
| 22:10.36 | Lann | cmusphinx is open source it seems. and seems to be well reccomended |
| 22:10.39 | Dovid | wow. TK was real helpful. I wonder what happend. he was Mr. Niceguy and know it all. |
| 22:10.45 | Lann | if it outputs an audio file i can pipe it into asterisk right? |
| 22:10.56 | citywok | Dovid: he was everything but mr nice guy |
| 22:11.04 | citywok | he was mister know it all butthead |
| 22:11.06 | Dovid | citywok: Happening more and more these days. |
| 22:11.24 | Dovid | he was too the point. didnt see it as rude. he did help a lot but then again.... |
| 22:11.43 | Dovid | Corydon76-home: Can you have a look at my back trace? |
| 22:11.45 | citywok | he was very rude, lol. he was warned quite a few times and never got off his high horse. |
| 22:11.55 | Dovid | oh well. |
| 22:12.13 | Corydon76-home | Dovid: I remember something about that, but an upgrade is in order. If an upgrade still doesn't fix it, open an issue for it |
| 22:12.36 | Lann | i wonder how i could control the relative volume of the audio i play to a connected user vs other the speech of other conference callers |
| 22:12.50 | citywok | lann meetme has volume controls |
| 22:13.29 | Lann | cool |
| 22:13.30 | Lann | thanks |
| 22:13.31 | Corydon76-home | There's also a VOLUME() dialplan function that sets up an audiohook |
| 22:13.41 | Dovid | Corydon76-home: Any key words I can look for on the bug tracker. also any idea what would cause it so I know what code to avoid in the AGI ? |
| 22:14.19 | Corydon76-home | Dovid: I did not fix it myself. Look in the Changelog |
| 22:15.10 | *** join/#asterisk m_tadeu (~quassel@89.180.156.159) |
| 22:15.25 | Dovid | Corydon76-home: I can build a full blown PBX with all the bells and whistles but SUCK at searching (it's a handycap). any thing in aprticular u remember that I should search for besides pickup ? |
| 22:16.04 | citywok | Dovid: there have been a _LOT_ of bug fixes since .26 came out, your best bet would probably be to upgrade. |
| 22:16.09 | Dovid | and it seems that I can't spell either ;) |
| 22:16.31 | Dovid | citywok: Like I said If I have to I will but I need to test. i woudl rather know what the issue is so i know what version to go to |
| 22:16.46 | citywok | but as Corydon76-home suggested you can step through the changelogs and see what major changes were made. |
| 22:17.09 | Lann | i'm surprised nobody has tried this |
| 22:17.14 | Lann | at least that I'm aware of |
| 22:17.19 | Dovid | doing that now |
| 22:17.21 | citywok | you may find something related to your crash and then look up the corresponding bug |
| 22:18.00 | bspidel | if two queues have same weight, will * process calls alternatively from each queue? |
| 22:18.24 | citywok | bspidel: that doesn't make sense. |
| 22:18.30 | citywok | bspidel: a call can only be in one queue at a time |
| 22:18.32 | csnook | I hate the telecom industry. |
| 22:19.03 | citywok | csnook: yes |
| 22:19.33 | bspidel | 10 calls arrive and are split evenly between 2 queues. will they be processed evenly if queue weights are the same? |
| 22:19.42 | csnook | I have 3 different bills for different amounts, on the same account, while their web interface shows two different numbers for balance due (on the same page) and this is our more customer-friendly provider |
| 22:20.26 | csnook | Also, they will only give me usage logs for July of last year, so there's no way to verify their numbers |
| 22:20.33 | citywok | bspidel: the queues are independent of each other. so just ignore the extra queue. one queue will rrmemory one call at a time unless you enable the option that tells it to do more than one at a time (i don't remember what it is). |
| 22:20.46 | csnook | but, hey, the people we just dumped laughed at us when we asked for usage logs |
| 22:22.17 | bspidel | citywok: the queues are used to model different skill sets. Not having weights, the lower queue numbers service more calls, but will they be serviced fairly if I make the weights the same? |
| 22:23.24 | citywok | do you mean the queues get serviced fairly, or the members serviced fairly? |
| 22:28.11 | bspidel | I mean the queues because some members are in multiple queues and if a member is in 2 queues the lowest queue always gets him never the highest until no calls remain in lowest queue. |
| 22:28.34 | WIMPy | Does anyone have any idea why I get SIP PUBLISH dialoges piling up or how to get rid of them? |
| 22:29.54 | *** join/#asterisk russellb (~russell@asterisk/digium-open-source-team-lead/russellb) |
| 22:29.54 | *** mode/#asterisk [+o russellb] by ChanServ |
| 22:32.13 | citywok | bspidel: you have to use logic to load the queues |
| 22:33.34 | bspidel | the queues are loaded by ivr which determines what skills are required |
| 22:33.36 | citywok | bspidel: asterisk won't put the callers in them for you, you have to handle whether they get loaded evenly or not. |
| 22:33.45 | citywok | bspidel: then they will be loaded as evenly as your ivr |
| 22:34.21 | bspidel | yes but they are not processed evenly, one queue will be emptied before the other is serviced. |
| 22:35.17 | citywok | what do you mean by lowest/highest queue? |
| 22:36.21 | bspidel | this client numbers his queues so first is lowest number beginning with 200. Weights are not currently used. I am wondering if they will help |
| 22:37.46 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 22:37.48 | bspidel | a very capable member may be in as many as 10 of the 35 skill set based queues |
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| 22:42.40 | citywok | oh, so they are NAMED 1-200? |
| 22:42.52 | bspidel | correct. |
| 22:43.05 | bspidel | actually 200-300 |
| 22:44.51 | citywok | gotcha. leifmadsen any idea if you have 10 queues, with the same member in each queue, why all callers get serviced out of the first queue before the callers in the second queue? |
| 22:46.26 | *** join/#asterisk Defraz (~Defraz@63-226-95-152.dia.static.qwest.net) |
| 22:47.41 | leifmadsen | citywok: do you have shared_lastcall=yes and weight set the same on all queues? |
| 22:47.55 | citywok | leifmadsen: this isn't for me, it's a summary of bspidel's problem |
| 22:48.11 | citywok | leifmadsen: he has weights set the same, never heard of shared_lastcall before, i'm guessing not. |
| 22:48.42 | bspidel | citywok,leifmadsen: have not set weights, fillall is on and shard_lastcall is yes |
| 22:48.59 | citywok | fillall may override and cause you problems |
| 22:49.05 | leifmadsen | only other thing I can think of is the prompts are setup differently where the first queue plays less, or shorter length, prompts |
| 22:49.08 | citywok | i'd try it without fillall |
| 22:49.20 | leifmadsen | I think he means autofill |
| 22:49.31 | leifmadsen | fillall is not an option :) |
| 22:49.39 | citywok | yea, i figured. i just didn't remember what that option was. |
| 22:49.46 | *** join/#asterisk m_tadeu (~quassel@89.180.156.159) |
| 22:49.47 | leifmadsen | note that queues will not attempt to deliver calls to agents during prompts |
| 22:50.08 | leifmadsen | make sure queues are not trying to play prompts too often, etc... |
| 22:50.13 | citywok | leifmadsen: ahh, i never thought about that. i playback my announcements before i send my calls to the queues. |
| 22:50.18 | leifmadsen | (see the queues chapter in the new book about that ;)) |
| 22:50.28 | citywok | and we're small enough we don't need to use the periodic announcements |
| 22:50.32 | bspidel | yes autofill. we have 200 simultaineous calls with carefull attention to call delays. we are afraid turning autofill off will add excessive delay. |
| 22:50.33 | leifmadsen | citywok: right, but things like periodic announcements could screw you up if they play often |
| 22:50.42 | leifmadsen | it probably will |
| 22:50.55 | citywok | bspidel: yea, but i'd turn it off to test the theory. |
| 22:51.03 | citywok | bspidel: do it on a test queue not on your production :P |
| 22:51.06 | leifmadsen | read the queues chapter at http://ofps.oreilly.com and read the section on prompt control |
| 22:51.06 | bspidel | prompts happen before call is queued. announcements do seem to affect call processing. |
| 22:51.24 | leifmadsen | you never know, maybe a particular queue has callers changing more often, and is blocking calls from the agents? |
| 22:51.31 | leifmadsen | yes, announcements will |
| 22:51.56 | leifmadsen | I'd try and make them all as similar as possible, maybe even to the point of disabling all announcements in the queue for the time being |
| 22:52.23 | leifmadsen | beyond that, I've never had the particular issue you describe |
| 22:52.52 | bspidel | The only place I see this behavior is on high volume production server. I need to feel very warm and fuzzy before I make a change. |
| 22:53.09 | leifmadsen | then my work here is done :) |
| 22:53.23 | citywok | bspidel: then recreate it with a handful of fake calls on a test system :P |
| 22:53.42 | citywok | this should be easy to reproduce with a handful of calls, 2 queues, and 1 member in both of them. |
| 22:53.53 | leifmadsen | SIPp ftw |
| 22:54.15 | leifmadsen | capture some existing call flows and reproduce them via SIPp on a development server, recreate problem, fiddle with bits |
| 22:54.20 | leifmadsen | wash, rinse, repeat |
| 22:54.23 | bspidel | let me restate, I can try it on smaller test system. but I dont know if it will affect scaled system |
| 22:54.55 | leifmadsen | I don't think we can offer any additional suggestions at this point |
| 22:54.57 | citywok | bspidel: that's why he suggested sipp |
| 22:55.50 | bspidel | Thanks for the help. I will review your suggestions and try some tests. |
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| 23:02.42 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
| 23:03.07 | IsUp | hi |
| 23:03.34 | WIMPy | Looks like I can't get any subscriptions with the current CVN version. |
| 23:04.29 | WIMPy | SVN |
| 23:06.09 | citywok | cute |
| 23:07.03 | WIMPy | is going to sleep. |
| 23:08.10 | IsUp | gnite |
| 23:08.12 | jaytee | is it possible to use the * as the first character in a patter match in the dialplan? |
| 23:08.28 | jaytee | s/patter/pattern |
| 23:08.37 | WIMPy | sure |
| 23:09.56 | jaytee | I keep getting this: chan_sip.c:20223 handle_request_invite: Call from '206' to extension '*' rejected because extension not found in context 'unrestricted'. |
| 23:10.55 | IsUp | jaytee: use '.' instead of '*'? |
| 23:11.19 | citywok | WIMPy: i'll compile svn real quick and try |
| 23:11.24 | jaytee | I've added [*]2xxT to my dialplan digit map on the two Polycom phones I'm using to test with and I've got _*2xx as the pattern in extensions.conf in the unrestricted context |
| 23:11.32 | jaytee | but I still get that message |
| 23:11.42 | citywok | jaytee: it says '* |
| 23:11.50 | citywok | ' which makes it look like it didn't get the 2xx |
| 23:11.53 | pabelanger | jaytee: *CLI> dialplan show unrestricted |
| 23:11.56 | citywok | so you need dialplan for _* |
| 23:12.12 | pabelanger | actually |
| 23:12.21 | pabelanger | *CLI> dialplan show *@unrestricted |
| 23:13.31 | Chainsaw | ChannelZ: Thanks for having a look eariler. |
| 23:13.32 | jaytee | so you're saying try _*. instead of _*2xx? |
| 23:13.39 | *** part/#asterisk bspidel (~IceChat77@63.134.128.4) |
| 23:13.45 | Chainsaw | ChannelZ: Earlier, even. Looks like intermittent packet loss affecting >500 byte packets on an intermediate switch. |
| 23:13.50 | citywok | jaytee: well it says it only got '*' so... you need that. |
| 23:14.24 | jaytee | not sure why it's not getting the rest and tries to dial immediately |
| 23:15.02 | citywok | can't help ya on that part. |
| 23:15.36 | IsUp | why you need '*'? |
| 23:15.39 | IsUp | i didnt get it |
| 23:15.43 | jaytee | I don't |
| 23:15.49 | jaytee | just thought I'd use it |
| 23:16.00 | IsUp | ah, okay |
| 23:16.03 | jaytee | but I can use another number |
| 23:16.58 | IsUp | _[*].,1,Dial(... |
| 23:17.37 | jaytee | so I need brackets in the pattern match in extensions.conf? I've only used them in the Polycom digit map |
| 23:18.00 | IsUp | well take a look into http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns personally ive never used * or # in dialplan |
| 23:18.17 | jaytee | probably not a good idea I guess |
| 23:18.57 | IsUp | yeah probably |
| 23:19.41 | citywok | has anybody succesfully used calendaring w/ ews? |
| 23:27.07 | citywok | WIMPy: i just compiled trunk and my Aastra 57i registered just fine. |
| 23:40.18 | jaytee | be back later |
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