| 00:00.17 | fujin | -vvv should give you full protocol debug |
| 00:00.25 | fujin | I find -vv is usually enough |
| 00:04.13 | luisjose | hmm |
| 00:05.19 | luisjose | i think i can't get a SIP register password with tcpdump |
| 00:05.37 | saftsack | why? |
| 00:05.46 | saftsack | do you think it is encrypted? |
| 00:05.58 | luisjose | i don't think so |
| 00:06.06 | luisjose | but i don't know how i can get it |
| 00:06.12 | *** join/#asterisk pepo--- (n=pepOSX@190.72.158.147) |
| 00:06.14 | saftsack | why? |
| 00:06.24 | luisjose | its just a crappy ata |
| 00:06.41 | saftsack | if you know the password ... |
| 00:06.48 | luisjose | saftsack, negative |
| 00:07.03 | saftsack | then buy another ata which is similar |
| 00:07.09 | saftsack | turn tcpdump on and then the ata |
| 00:07.19 | saftsack | and then look at the logs where the password is shown |
| 00:07.30 | saftsack | and then you know when the password is sent |
| 00:08.56 | luisjose | saftsack, http://pastie.caboo.se/94432 |
| 00:09.39 | *** join/#asterisk ReD-MaN (i=daemon@CPE0002b38bce8b-CM0018c0b357cc.cpe.net.cable.rogers.com) |
| 00:10.00 | saftsack | did you try to change it from hexadecimal? :-P |
| 00:10.14 | Yourname` | sup fujin |
| 00:11.18 | saftsack | http://de.wikipedia.org/wiki/Session_Initiation_Protocol |
| 00:12.18 | saftsack | i sent the german page because of the good diagram |
| 00:12.26 | luisjose | saftsack, how :P |
| 00:12.44 | *** join/#asterisk craigk (n=ckowald@58.174.113.53) |
| 00:13.29 | saftsack | luisjose, did you see unscrambled words? |
| 00:13.48 | saftsack | or are there just hexadecimal letters? |
| 00:13.56 | saftsack | did you try wireshark? |
| 00:14.44 | luisjose | fucking mouse just died |
| 00:14.48 | fujin_ | pwned |
| 00:14.49 | luisjose | saftsack, no X |
| 00:15.03 | luisjose | saftsack, i was using options -vvv and -ttt |
| 00:15.57 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 00:17.38 | saftsack | http://ian.blenke.com/voip/tcpdump/ethereal/SIP/RTP/G.711/rtptools/quicktime/voipmp3.html |
| 00:17.47 | luisjose | wait |
| 00:17.53 | luisjose | can't do shit |
| 00:17.56 | luisjose | without the mouse |
| 00:17.57 | luisjose | damnit |
| 00:18.01 | luisjose | brb |
| 00:20.12 | Yourname` | fujin: Do you use QueueLog? |
| 00:23.25 | *** part/#asterisk lancey (i=lancey@support.net1.cc) |
| 00:23.33 | fujin_ | Yourname`, the application? |
| 00:23.35 | Yourname` | yessir |
| 00:23.49 | fujin_ | no, the standard queue logging facilities work to my requirements for Queuemetrics |
| 00:23.55 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 00:23.57 | Yourname` | ah |
| 00:24.03 | Yourname` | I'm trying to a build a custom log. |
| 00:24.09 | Yourname` | Like the log of our IVRs. |
| 00:24.14 | fujin_ | I see |
| 00:24.42 | Yourname` | For example, on a voicebroadcast application, how many people responded by DTMF to the first message played. How many people responded to the next message played to them after they responded, and so on, etc. |
| 00:24.56 | Yourname` | I think I can use Queuelog for that. Except I can't find much information on it.. |
| 00:25.12 | DrAk0 | MOUSE just died |
| 00:25.13 | fujin_ | show application QueueLog isn't verbose enough? |
| 00:25.17 | DrAk0 | had to find a new one... |
| 00:25.24 | saftsack | kk |
| 00:25.26 | fujin_ | QueueLog(101|${UNIQUEID}|${AGENT}|WENTONBREAK|600) |
| 00:25.26 | saftsack | http://ian.blenke.com/voip/tcpdump/ethereal/SIP/RTP/G.711/rtptools/quicktime/voipmp3.html |
| 00:25.30 | fujin_ | QueueLog(queuename|uniqueid|agent|event[|additionalinfo]): |
| 00:25.32 | saftsack | DrAk0, there you go |
| 00:26.19 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-92-106.no.no.cox.net) |
| 00:26.21 | Yourname` | I mean to be able to get the full thing, like all the way down to the CSVs that are created.. you know what I mean? |
| 00:27.01 | fujin_ | lol |
| 00:27.04 | fujin_ | can't say i do |
| 00:27.09 | *** join/#asterisk the_Goat1 (n=rsd095@firewall.turbolink.net) |
| 00:27.11 | Yourname` | lol |
| 00:27.15 | fujin_ | why not just setup some verbose logging to go to a mysql socket |
| 00:27.20 | fujin_ | which inserts rows into a db? |
| 00:27.35 | the_Goat1 | anyone here use cisco phones with asterisk |
| 00:27.36 | *** join/#asterisk snuff-work (n=bradl@61.29.30.137) |
| 00:27.37 | the_Goat1 | ?? |
| 00:27.53 | saftsack | the_Goat1 anyone here who has to much money? |
| 00:27.54 | DrAk0 | saftsack, but thats for record the call, i need the user and password |
| 00:27.54 | Yourname` | See, that sounds good too. For now, I'm generating verbose NoOps and saving CLI output and then searching for those NoOps, lol |
| 00:28.08 | fujin_ | lol |
| 00:28.15 | fujin_ | I have the problem of too much output |
| 00:28.19 | saftsack | DrAk0, yes the rtp part is for the audio |
| 00:28.28 | saftsack | the sip part is for initiating the call |
| 00:28.29 | fujin_ | for my queue delivery macro's, they flood the screen and make the logs unreadable |
| 00:28.29 | Yourname` | the_Goat: Had a few, didn't get to use them properly, and now they sit useless. Currently using Linksys SPA941s. |
| 00:28.50 | the_Goat1 | i am having an issue when i park or transfer a call. when i pick up the cisco phone, i can't hear anything but when italk on the phone, the person on the other phone can hear jsut fine |
| 00:28.56 | saftsack | DrAk0, o=icblenke 2890844526 2890842807 IN IP4 127.0.0.1 i think this is username + password in this example |
| 00:29.17 | Yourname` | I log it all down to to almost 3 gigs of log data per day. And then using UltraEdit look for the NoOps I created.. lol |
| 00:29.48 | slima | I have 2 accounts from the same sip provider, and inbound calls always match the first register, why? and how to fix it? my configs http://pastebin.com/d64bef9a0 (sorry for my english) |
| 00:30.04 | fujin_ | slima, use two seperate contexts |
| 00:30.09 | DrAk0 | saftsack, ok, im installing ethereal on my workstation and ill get the pcap from the server |
| 00:30.27 | fujin_ | Yourname`, are you using logrotate to rotate /var/log/asterisk/messages ? |
| 00:30.43 | saftsack | DrAk0, good thing ;) |
| 00:30.55 | slima | fujin_: for register => ? |
| 00:30.56 | saftsack | DrAk0, are you on a machine with X11 at the moment? |
| 00:31.04 | DrAk0 | saftsack, yes |
| 00:31.32 | saftsack | do you have ssh access to your dumping machine? |
| 00:31.40 | CCFL_Man2 | my cisco voice compression module will come tomorrow |
| 00:31.45 | DrAk0 | saftsack, yes |
| 00:32.02 | saftsack | if yes you can open your ethereal window from this pc on your X11 environment |
| 00:32.08 | Yourname` | fujin: Not yet. :S |
| 00:32.18 | fujin_ | slima, yes, if you register twice, register each one to generate messages onto a different context |
| 00:32.21 | Yourname` | Each box has 250gb HDD though, and just got asterisk |
| 00:32.26 | fujin_ | i see |
| 00:32.41 | fujin_ | I just put it a logrotate option which will stop asterisk, move the old messages log, compress, start asterisk |
| 00:32.47 | fujin_ | makes day-by-day diagnosis easier. |
| 00:33.01 | saftsack | DrAk0, http://www.cisl.ucar.edu/docs/ssh/guide/node29.html |
| 00:33.23 | Yourname` | Hmm, gotta read more about it then.. :D |
| 00:34.32 | DrAk0 | saftsack, im just gonna scp the pcap file to here |
| 00:35.23 | saftsack | or this way ;) |
| 00:36.01 | snuff-work | fujin_: i just use logger rotate on the * CLI |
| 00:36.56 | fujin_ | mm |
| 00:37.00 | fujin_ | can't really automate that though can ya |
| 00:37.10 | fujin_ | I guess I could automate `asterisk -rx "logger rotate"` |
| 00:37.21 | snuff-work | that's what i do in my crontab ;) |
| 00:37.27 | fujin_ | ta. |
| 00:37.42 | saftsack | DrAk0, please tell me then if you had success |
| 00:38.28 | [TK]D-Fender | fujin_, this is the part where you should relize the benefit of storing queue logs in a DB |
| 00:38.43 | DrAk0 | saftsack, im reading the pcap |
| 00:38.48 | DrAk0 | i think i got the user |
| 00:38.54 | saftsack | ;) |
| 00:39.00 | fujin_ | I shift my queue logs to db |
| 00:39.05 | fujin_ | but don't log directly to them |
| 00:39.19 | jwh | hm, is it possible to dump what extensions asterisk thinks it has defined (show dialplan dumps ael also) |
| 00:39.22 | fujin_ | [TK]D-Fender, is there some native functionality which will logger->db? |
| 00:39.24 | jwh | or should the extensions show there |
| 00:39.55 | fujin_ | jwh, Usage: core show dialplan [exten@][context] |
| 00:40.02 | fujin_ | you can drill down per-context. |
| 00:40.08 | jwh | oh, thanks |
| 00:40.09 | [TK]D-Fender | fujin_, not that I know. You should jsut go direct and save the transition. |
| 00:40.28 | fujin_ | [TK]D-Fender, The transition is required for queue metrics, it puts things into a database in a specific format |
| 00:40.36 | fujin_ | and that's something I unfortunately don't have the "power" to decide over. |
| 00:41.26 | [TK]D-Fender | fujin_, oh, ok, and outside program forces you toplay by its rules. |
| 00:41.33 | fujin_ | yeah |
| 00:41.37 | fujin_ | queuemetrics, closed source junk |
| 00:41.52 | fujin_ | the programming team didn't have time to write some in-house queue analysis software |
| 00:42.11 | jwh | ok then, i'm either getting the entire sql schema wrong, or something isn't working, i've tried both schemas I can find on the web, none work with 1.4 seemingly |
| 00:42.11 | fujin_ | it reads queue_log and fires stuff off to a database |
| 00:42.26 | fujin_ | jwh, the code should provide a schema |
| 00:42.33 | fujin_ | what are you using? |
| 00:42.34 | fujin_ | what code |
| 00:42.39 | fujin_ | what module rather |
| 00:43.07 | jwh | ah yes, its the same |
| 00:43.14 | jwh | i'm using realtime mysql |
| 00:45.18 | slima | fujin_: but, how I do it? http://pastebin.com/d7f9d2859 - doesn`t working |
| 00:45.48 | jwh | I have extensions => mysql,asterisk,extensions in extconfig.conf, sipusers/peers works fine, just extensions that don't work now |
| 00:46.47 | [TK]D-Fender | slima, http://pastebin.com/m7fa409a5 |
| 00:48.09 | *** part/#asterisk the_Goat1 (n=rsd095@firewall.turbolink.net) |
| 00:49.52 | jwh | fujin_: i'm using the schema provided by the retrieve_extensions_from_mysql.pl script, thats correct yes? |
| 00:51.36 | fujin_ | uh |
| 00:51.37 | fujin_ | le tme check |
| 00:52.10 | fujin_ | There is documentation of the SQL database in the file |
| 00:52.10 | fujin_ | doc/extconfig.txt in your Asterisk source code tree |
| 00:52.20 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 00:53.05 | jwh | that only appears to discuss voicemail and sip/iax |
| 00:53.07 | fujin_ | the schema for extensions should be like |
| 00:53.08 | slima | [TK]D-Fender: i know, but how I place register => to specified context? |
| 00:53.10 | fujin_ | dude |
| 00:53.11 | fujin_ | learn2read |
| 00:53.22 | fujin_ | "An extension table would look more like this:" |
| 00:53.26 | jwh | oh yes |
| 00:53.27 | fujin_ | context,exten,priority,app,appdata |
| 00:53.30 | jwh | yeah thats what i'm using |
| 00:53.40 | fujin_ | and you're using Switch => Realtime |
| 00:53.43 | fujin_ | in your dialplan? |
| 00:54.14 | jwh | in extensions.conf? yeah |
| 00:54.29 | jwh | im missing something really obvious somewhere |
| 00:54.30 | fujin_ | can you see asterisk querying the database? |
| 00:54.37 | fujin_ | well, correct schema, using switch |
| 00:55.00 | jwh | do I have to define Switch for every context in the database or just default? |
| 00:55.03 | fujin_ | I dunno, sorry, never bothered with realtime |
| 00:55.12 | jwh | it seems abit vague |
| 00:55.13 | jwh | hm |
| 00:55.14 | fujin_ | I'd go with *every* context. |
| 00:55.18 | jwh | yeah |
| 00:55.31 | fujin_ | generally, you can never overdefine. |
| 00:56.24 | jwh | yeah |
| 00:59.54 | *** join/#asterisk Rahail (n=rahail@c-68-43-176-199.hsd1.mi.comcast.net) |
| 01:00.01 | Rahail | ok i got prboelm people with dtmf |
| 01:00.04 | Rahail | can some one help me |
| 01:00.38 | Rahail | asterisk server a (old) its work very well with DID when people enter password from outside to chk voice mail it work |
| 01:00.59 | Rahail | today I changed to New server diffrent data center Now I got problem it send dubble digit |
| 01:02.53 | *** join/#asterisk Op3r (n=Op3r@121.97.145.174) |
| 01:02.57 | [TK]D-Fender | slima, you DON'T Registering has NOTHING to do with authing incoming and outgoing calls |
| 01:03.04 | *** join/#asterisk cmwt (n=bit_frog@dmz-emmb.redback.com) |
| 01:03.14 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 01:03.38 | GlobeTrotter | hola... what does this error mean?? translate.c:163 framein: no samples for g729tolin |
| 01:03.44 | GlobeTrotter | asterisk 1.4.11 |
| 01:03.56 | Rahail | any one |
| 01:04.29 | jwh | hm |
| 01:05.02 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 01:06.18 | tengulre | anybody like time away, I want to see, and see u!! |
| 01:06.38 | tengulre | OK, good morning everyone! |
| 01:07.38 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-3ced67dd0ecb6b38) |
| 01:09.11 | fujin_ | Afternoon :) |
| 01:09.27 | [TK]D-Fender | Evening :) |
| 01:09.37 | Op3r | anyone tried asterisell? |
| 01:10.10 | slima | eeh, i don`t understand |
| 01:10.41 | tengulre | how to buy digium card in asia? |
| 01:10.57 | Op3r | tengulre: get an asian reseller |
| 01:11.08 | fujin_ | someone in asian probably makes knockoffs |
| 01:11.08 | jwh | fujin_: the query seems to be exactly what the user dialled, so it will never match wildcard extensions |
| 01:11.08 | Op3r | tengulre: if I remember correctly there is one in singapore |
| 01:11.13 | jwh | that explains it |
| 01:11.24 | fujin_ | jwh, it *should* match wildcard extensions, that's what they're for |
| 01:11.48 | jwh | ah, I see |
| 01:11.58 | tengulre | my problem is that digital card could support chinese telecom signal? |
| 01:12.23 | fujin_ | in soviet china, telecom signals you |
| 01:12.48 | tengulre | E1 == Chinese E1? I don't know. |
| 01:13.02 | JT | probably ETSI. |
| 01:13.06 | JT | ETSI E1 |
| 01:13.21 | tengulre | I got a ISDN PRI lines from owener telecomm. |
| 01:13.30 | tengulre | so I want to buy a card, ... |
| 01:13.43 | tengulre | but I dont know which card is suit me? |
| 01:13.52 | jwh | ah |
| 01:13.54 | JT | most should work |
| 01:13.55 | jwh | excellent, working now |
| 01:13.56 | tengulre | 2E1 (resource) |
| 01:14.08 | jwh | thanks fujin_, very grateful |
| 01:14.14 | tengulre | ETSI =? |
| 01:14.31 | Rahail | can some one guide me how can i solve my dtmf issue |
| 01:14.33 | Rahail | :( |
| 01:14.39 | [TK]D-Fender | slima, registering is a SEPERATE action. All it does is tell your provider where to SEND calls to. It doesn't AUTH them when they come in or tell * what context to send the calls to.\ |
| 01:14.49 | JT | tengulre: easily googleable. |
| 01:14.59 | Op3r | <fujin_> in soviet china, telecom signals you <-- reminds me of slashdot |
| 01:15.30 | Op3r | like that I for one welcomes our China telecoms overlords |
| 01:16.57 | tengulre | JT: thanks I saw. |
| 01:17.06 | JT | tengulre: awesome. |
| 01:17.17 | tengulre | how to transfer some dialing when use agent? |
| 01:17.31 | tengulre | transfer to other agent? |
| 01:18.57 | tengulre | I think agent mode is bad, I must very be careful beacause I Habits click the hangup button when end . |
| 01:19.34 | tengulre | so like that ,I must relogin agent , input username, password. and so on. |
| 01:20.49 | Op3r | tengulre: are u talking about agents in queues? |
| 01:21.30 | fujin_ | tengulre, use agentcallbacklogin then |
| 01:21.32 | fujin_ | or aqm/rqm |
| 01:21.45 | fujin_ | aqm/rqm model is much more configurable, but agentcallbacklogin is better than agentlogin etc |
| 01:22.23 | pkunkra | out of curiosity, does anyone have a good link for a list of "good" area codes allow calls to? |
| 01:22.25 | tengulre | Op3r: yes. |
| 01:22.35 | tengulre | 2E1--->Queue--->Agents |
| 01:22.41 | fujin_ | pkunkra, international? |
| 01:22.50 | pkunkra | fujin_: just u.s. |
| 01:22.54 | fujin_ | ah |
| 01:22.56 | *** join/#asterisk frogzoo (n=Frogzoo@202.155.165.25) |
| 01:23.02 | tengulre | fujin_, OK.. |
| 01:23.12 | pkunkra | i don't think i'll allow international calls on my pbx |
| 01:23.28 | Op3r | tengulre: use what fujin suggested agentcallbacklogin allows you to hang up the phone and just wait for the phone to ring |
| 01:23.33 | pkunkra | i know i'm supposed to ignore the 809 area. |
| 01:23.38 | fujin_ | aqm/rqm is better, though :) |
| 01:24.18 | pkunkra | but i'd rather deny all and then allow calls to known good area codes , rather than just disallowing specific ones. |
| 01:24.27 | Strom_M | pkunkra: www.nanpa.com |
| 01:24.37 | tengulre | fujin_: aqm/rpm?? |
| 01:25.15 | fujin_ | addqueuemember/removequeuemember |
| 01:25.20 | fujin_ | dynamically adds <interface> to queue |
| 01:25.25 | fujin_ | via the dialplan |
| 01:25.35 | fujin_ | show application addqueuemember |
| 01:25.44 | fujin_ | there is a good txt file in doc/ aswell |
| 01:26.35 | pkunkra | Strom_M: well, the NANP might work, but i'm not sure if they'd just list all area codes or only good ones. the 809 area code *is* in the north american number plan. i don't see it on the nanp site but then, i'm not 100% sure. |
| 01:26.50 | Strom_M | pkunkra: NANP lists everything. |
| 01:26.59 | Strom_M | go search in detail before I have to do it for you :) |
| 01:27.32 | pkunkra | Strom_M: well, i've been poking around on google for 10 minutes but not a strong page yet. :-) alright. I'll keep looking. |
| 01:28.54 | pkunkra | there is this one.... http://www.consumer.att.com/global/english/usa_codes.html |
| 01:29.06 | Strom_M | pkunkra: ok, just hang on |
| 01:29.33 | pkunkra | i haven't found a page address the specific issue of which ones are good ones too call. plenty of pages that list all area codes. |
| 01:29.50 | Strom_M | http://nanpa.com/nas/public/npasInServiceByNumberReport.do?method=displayNpasInServiceByNumberReport |
| 01:30.11 | Strom_M | that's currently assigned geographic area codes and their locations |
| 01:30.18 | pkunkra | well, that one does have 809 in it. |
| 01:30.31 | Strom_M | http://nanpa.com/nas/public/npasInServiceByLocationReport.do?method=displayNpasInServiceByLocationReport |
| 01:30.37 | Strom_M | geographic area codes sorted by location |
| 01:30.51 | Strom_M | nonGeoNpasInServiceReport |
| 01:30.52 | Strom_M | er |
| 01:30.59 | Strom_M | http://nanpa.com/nas/public/nonGeoNpasInServiceReport.do?method=displayNonGeoNpasInServiceReport |
| 01:31.05 | Strom_M | non-geographic codes |
| 01:31.19 | Strom_M | etc etc etc etc etc |
| 01:31.32 | pkunkra | hmm ok. |
| 01:31.54 | Strom_M | NANPA is the end-all and be-all of code assignment information in North America. Period. :) |
| 01:32.01 | pkunkra | well, i guess i can just download the csv links at the bottom and take out all non-us state codes |
| 01:32.16 | pkunkra | i guess that would be pretty safe then. |
| 01:32.22 | pkunkra | thanks Strom_M |
| 01:32.29 | Strom_M | hence "North American Numbering Plan Administration" |
| 01:32.32 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 01:32.43 | pkunkra | hehe. yeah. if they don't know, it doesn't exist. :-) |
| 01:33.02 | Strom_M | no, that's the Pacific Bell Yellow Pages circa 1990 or so |
| 01:33.11 | Strom_M | "If it's not in here, it probably doesn't exist." |
| 01:33.17 | Qwell | http://www.voicecon.com/videos/playvideo/index.php?vid=vcsf07-summit-software-based-architectures-uc |
| 01:33.34 | Qwell | Mark "sharing the love" with MS, Cisco, Avaya, etc :D |
| 01:34.31 | pkunkra | Strom_M: i don't recall the yellow pages ever being that good. |
| 01:35.11 | Strom_M | pkunkra: it was an advertising slogan they used at the time |
| 01:35.21 | *** join/#asterisk rlama (n=rlama@cmodem-232-183.tricom.net) |
| 01:37.15 | pkunkra | Strom_M: yeah. I vaguely remember it. they had those annoying bell tones if i remember right. |
| 01:37.25 | *** join/#asterisk chendy (n=chendy@218.242.110.26) |
| 01:37.49 | Strom_M | pkunkra: you lived in california at the time? |
| 01:38.53 | slima | [TK]D-Fender okey, but I have two acounts from the same sip ISP and, the inbound calls are always directed through 1st peer, no matter if I call on the first account or the second account my config: http://pastebin.com/d24f1f81c whats wrong? |
| 01:39.16 | Qwell | Strom_M: s/it probably/maybe it doesn't/ |
| 01:39.28 | Qwell | erm, maybe |
| 01:39.33 | Strom_M | Qwell: oh, was that the slogan? |
| 01:39.40 | Qwell | sounds familiar |
| 01:39.42 | Strom_M | yeah |
| 01:39.47 | Strom_M | that's the gist of it anyway |
| 01:40.24 | Qwell | nothing |
| 01:40.30 | tengulre | hi,all ! which linux info website are you like visit? |
| 01:40.31 | [TK]D-Fender | slima, because both use "insecure" that way I don't think its using the user & pass to authenticate as the calls are coming from the SAME host. |
| 01:40.38 | Qwell | tengulre: google! |
| 01:40.59 | Qwell | [TK]D-Fender: Strom_M: See URL above |
| 01:41.12 | Qwell | hilarity |
| 01:41.43 | [TK]D-Fender | slima, do THIS instead for your register ans see if it helps : register=> user2:pass@host/12345 |
| 01:42.11 | [TK]D-Fender | slima, the /12345 tells your itsp what EXTENSION to dial instead of jsut sending calls to "s". That should help you differentiate between cals. |
| 01:42.57 | [TK]D-Fender | slima, so in that case your user2's calls would arrive to "exten => 12345,1,DoSomthing". If that work, do the same for user1. |
| 01:45.17 | slima | hm, thx. |
| 01:46.46 | *** join/#asterisk guillote_GNU (n=guillote@190.7.27.17) |
| 01:47.37 | [TK]D-Fender | slima, reload SIP afterwards and you should re-register. then test an incoming call. |
| 01:50.35 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 01:53.20 | Op3r | where do you find the asterisk logs? so that you can find the error that causes the shutdown of the asterisk? |
| 01:53.29 | Op3r | is it in /var/log/asterisk ? |
| 01:53.38 | slima | hm, when I add /12345 to register and add exten => 12345 i get 'busy' |
| 01:53.48 | jwh | if your compile/package installs in sensible places, then yes Op3r |
| 01:54.13 | Op3r | jwh: its in messages right? |
| 01:54.25 | jwh | should be |
| 01:56.15 | Yourname` | Hmm, I'd really like to log all disconnected phone numbers that I call, how can I do so? |
| 01:56.38 | Yourname` | Let's say I call 10 phone numbers in rapid succession, and I'd 3 are disconnected. I'd like to know which ones are disconnected. |
| 01:58.15 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128666903.dsl.bell.ca) |
| 02:00.23 | [TK]D-Fender | slima, do "sip debug" in * CLI and pastebin the incoming call. |
| 02:01.12 | slima | [TK]D-Fender: they looking for '12345' in context PEER 1 |
| 02:01.42 | [TK]D-Fender | slima, pastebin the output I requested and your dialplan. |
| 02:02.23 | slima | w8 |
| 02:08.15 | slima | [TK]D-Fender: http://pastebin.com/d1f92632f |
| 02:09.10 | [TK]D-Fender | slima, Looking for 100 in urahara-incomming (domain 85.89.167.139) <- this is the exten / context it needs to find |
| 02:09.12 | slima | incomming exten are in extensions.conf ofcourse |
| 02:09.30 | [TK]D-Fender | slima, SIP/2.0 404 Not Found <--- but DOESN'T |
| 02:09.54 | slima | but i call to slima-incomming! |
| 02:10.09 | [TK]D-Fender | slima, look at which peer entry it landed on! |
| 02:10.14 | Yourname` | russellb: You aroundish? |
| 02:10.33 | [TK]D-Fender | slima, its UN-AUTHED. it doesn't know which one to pick so its either random, first, or last. |
| 02:10.47 | [TK]D-Fender | slima, set them BOHT to the same context. |
| 02:11.09 | [TK]D-Fender | slima, the EXTEN you use will let you seperate your calls. you can have them both in the same context. |
| 02:12.24 | slima | yes, but i don`t want that. ;-) |
| 02:12.49 | slima | how AUTHED a incomming call? |
| 02:12.56 | slima | AUTH* |
| 02:13.41 | [TK]D-Fender | slima, comment out the "insecure" lines and restart. See if that helps. It doesn't really matter as long as you can seperate the calls. |
| 02:14.39 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 02:14.54 | slima | ok, thx. |
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| 02:37.42 | Blackthorn | how can you tell ther version of zaptel I am using? |
| 02:40.30 | [TK]D-Fender | Blackthorn, "ztcfg -vvvv |
| 02:40.44 | Blackthorn | hey thanks fender |
| 02:41.10 | WilliamK | hey fender, are you made outta steel like bender is? |
| 02:41.14 | WilliamK | stainless that is |
| 02:41.15 | WilliamK | :) |
| 02:41.29 | WilliamK | sorry - just had to ask :) |
| 02:42.36 | Blackthorn | just pulled a newer zaptel driver did the normal make clean, make install etc etc. do i need to rebuild asterisk as well for it to take effect? |
| 02:43.16 | [TK]D-Fender | WilliamK, No... cold carbon steel |
| 02:43.45 | WilliamK | hmmmmmmm.... bundled with blue label to show coldness? |
| 02:43.46 | WilliamK | :) |
| 02:43.52 | [TK]D-Fender | Blackthorn, the effect is what it is. if it is COMPATIBLE is another matter |
| 02:44.22 | WilliamK | (new beer commercial is advertising the blue label here) |
| 02:44.24 | [TK]D-Fender | WilliamK, http://gallery.aocomputing.net/index.php?album=2007-03-02+Oni+Forge+Bushi |
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| 02:45.00 | WilliamK | ah - jedi! |
| 02:45.58 | [TK]D-Fender | WilliamK, thats my favourite. I have yet to redo my gallery for my new Oriole |
| 02:46.44 | WilliamK | coool :) |
| 02:55.25 | fujin_ | is there any way to increase the volume of audio being played to a device? |
| 02:55.47 | fujin_ | the headset adapter I've got for these phones is still a bit too quiet (output) even when maxed out. |
| 02:55.52 | fujin_ | bloody things |
| 02:56.28 | Blackthorn | when i run the ztcfg -vvvv commands it shows that the new version is loaded but it says 0 channels configured. What do I do to configure the channels? |
| 02:57.14 | Blackthorn | i moved from zaptel 1.4.2.1 to 1.4.5.1 |
| 02:59.15 | [TK]D-Fender | Blackthorn, perhaps you should reload your MODULES and check yout configs |
| 03:00.10 | Blackthorn | ok it told me that chan_zap.so was unsuccessfull |
| 03:00.27 | [TK]D-Fender | Blackthorn, I'm referring to MODPROBE |
| 03:00.35 | [TK]D-Fender | Blackthorn, you may well have to reboot as well. |
| 03:01.55 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583591.dsl.bell.ca) |
| 03:03.15 | Blackthorn | ok rebooting now. but while thats going on let me ask. I pulled the new version of zaptel. then did a make clean then a make configure, then make install. is that correct order? |
| 03:03.49 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:04.37 | [TK]D-Fender | Blackthorn, should be "make clean", "make", "make install" , "make config" |
| 03:05.21 | Blackthorn | would that be the same order for asterisk as well? and thanks for the help |
| 03:05.33 | fujin_ | no, asterisk is different |
| 03:05.46 | fujin_ | make distclean && ./configure --bla && make && make install |
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| 03:25.45 | Blackthorn | i have done finished going through each of the steps to isntall zaptel and asterisk again. and reboot but I still can't get chan_zap.so loaded |
| 03:25.53 | Blackthorn | when i run ztcfg -vvv it tells me 0 channels configured. |
| 03:25.54 | Blackthorn | thoughts? |
| 03:26.30 | Blackthorn | ahh think i found the problem |
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| 03:27.45 | [TK]D-Fender | ... modprobe |
| 03:28.56 | Blackthorn | the adapter shows up in the modprobe fine |
| 03:29.05 | Blackthorn | but the zaptel file has been replaced |
| 03:29.10 | Blackthorn | gota setup the channels etc etc again |
| 03:33.11 | Blackthorn | yup that was the problem. didn't expect my config file to be over-writen. whew was worred. and i'm really ready to get some sleep. cya |
| 03:33.13 | Blackthorn | thanks again |
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| 03:50.34 | jgomo3 | What happen with http://www.asteriskdocs.org/ ? |
| 03:50.54 | jgomo3 | can you acces it? |
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| 04:10.37 | jiboneus | hi anybody here uses user.conf? where can I find more infomation on the configuration file? |
| 04:10.51 | jgomo3 | What happen with http://www.asteriskdocs.org/ ? |
| 04:10.56 | jgomo3 | can you acces it? |
| 04:11.26 | jiboneus | nope,... |
| 04:16.24 | [TK]D-Fender | jgomo3, Its gone..... find a therapist :) |
| 04:20.16 | jgomo3 | [TK]D-Fender: :( |
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| 04:54.58 | Yourname` | How come it's gone when there's a new version coming out! |
| 04:54.59 | Yourname` | :( |
| 04:55.20 | Yourname` | (of the book) |
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| 06:51.46 | dan__t | what's up, doods. |
| 06:52.06 | dan__t | I don't want to be polling or anything, just wondered what you guys think about Cisco 7940's and Asterisk, if they make a good combination or not? |
| 06:52.15 | dan__t | I found a good deal on a set of two, and I might go in for the steal. |
| 06:53.45 | dan__t | I've found some * documentation saying that they might be good candidates - but I wnat to make sure. |
| 06:54.07 | Nugget | Ciscos are a total pain in the ass to deal with, both from a firmware standpoint and a licensing standpoint. |
| 06:54.43 | Nugget | Cisco spends all their development energy on the SCCP firmware (used with their flagship call manager product) which Asterisk only half-assed supports. |
| 06:55.24 | Nugget | there is a SIP firmware for them, but it's definitely a second-class citizen in cisco world and is typically plagued with annoying little bugs and incomplete features (although it's gotten a lot better over the past year) |
| 06:55.56 | Nugget | Unless you're buying a bunch of phones, you'll have difficulty getting a vendor to set you up with the smartnet contracts you need to buy from cisco in order to even legally have access to the firmware files. |
| 06:56.23 | Nugget | it takes weeks to set up and nobody really seems to know exactly how the process is supposed to work, especially in the absence of a call manager license |
| 06:56.32 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 06:56.59 | Nugget | Unless you're dead-set on the cosmetics of having those prestigious cisco phones on everyones desks you're better off buying Polycoms |
| 06:57.25 | Nugget | also bear in mind that the 7940 is two versions old. |
| 06:57.41 | Nugget | there's a 7941 and now a 7942 (and companion 7960/7961/7962) |
| 07:01.28 | dan__t | Ok, that works for me |
| 07:02.49 | dan__t | So PolyCom phones are pretty much the authoritative answer for VoIP desktop phones? |
| 07:03.19 | Nugget | I've never owned one, so I'm just going on the strength of the channel consensus which seems to pretty clearly favor polycom and despise grandstream. |
| 07:03.28 | Nugget | so that much is just me parroting what other people say. |
| 07:03.35 | Nugget | my cisco advice is hard-won, though. :) |
| 07:04.52 | Strom_C | dan__t: for what it's worth, I have cisco and polycom phones here at my apartment, and I'm considering dumping the ciscos |
| 07:05.07 | tzafrir | Nugget, I have a feeling that this advice is somewhat amplified |
| 07:05.16 | tzafrir | over-amplified, that is |
| 07:05.19 | awk | how do I set verbosity in the manager |
| 07:05.19 | Nugget | I'm Mister Hyperbole. |
| 07:05.48 | Nugget | but I stand by the sentiment. It's a lot of work to keep cisco phones in line and the smartnet issues are very real. |
| 07:05.57 | tzafrir | awk, you can use an CLI command from the manager, using Command |
| 07:06.06 | *** join/#asterisk Rahail (n=rahail@c-68-43-176-199.hsd1.mi.comcast.net) |
| 07:06.09 | Nugget | especially for someone who "has a line on a really good deal on two 7940s" which are almost certainly global spare units. |
| 07:06.11 | Rahail | any one use veitelity |
| 07:06.24 | Rahail | i cant solve this DTMF issue |
| 07:06.27 | tzafrir | try the command "help" ? |
| 07:06.31 | dan__t | Ok, PolyCom it is then heh. |
| 07:06.36 | awk | k, let me try |
| 07:06.42 | dan__t | I'm needing a few phones for a few sales guys all over the country |
| 07:06.45 | *** join/#asterisk saftsack (n=oliver@p54A7D27C.dip.t-dialin.net) |
| 07:07.16 | Nugget | oh, ciscos are particularly bad at that. Running a remote cisco phone, off-site, with no tftp server on the same network can be a disaster. |
| 07:07.45 | Nugget | especially with a consumer appliance doing the dhcp |
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| 07:09.14 | awk | tzafrir: I dont want to set the asterisk debugging up |
| 07:09.17 | awk | but the manager debugging |
| 07:09.22 | awk | so I can see unique id's, etc |
| 07:09.44 | awk | Uniqueid: 1189062574.697 |
| 07:09.46 | awk | ahh, its there |
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| 07:14.17 | dan__t | Very good point, Nugget. |
| 07:14.20 | FlatFoot | good morning all |
| 07:14.31 | dan__t | Back in the day I got an Inter-Tel VoIP phone to work with Asterisk. |
| 07:14.45 | dan__t | It was a complete bitch, but it spoke pure SIP, and that ended up working out pretty well. |
| 07:14.55 | dan__t | Ever since, I've been looking for better (namely, CHEAPER) alternatives. |
| 07:15.11 | dan__t | They were solid phones, but they seemed to overheat - a lot. |
| 07:19.18 | FlatFoot | anyone got into using Cat6 yet ? |
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| 07:22.31 | henkoegema | <PROTECTED> |
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| 07:55.49 | hellop | hi |
| 07:55.53 | Beave | hello |
| 07:56.51 | awk | tzafrir: hmm, you dont have documentation on working with recordings with the uniqueid? |
| 07:56.58 | Strom_C | hello hellop |
| 07:57.07 | Strom_C | hellop hello |
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| 08:26.46 | tzafrir | anybody knows where I can find the "zeroconf for asterisk" page? |
| 08:27.08 | tzafrir | All links seem to lead to: http://www.astmasters.net/projects.html#zeroconf |
| 08:27.26 | tzafrir | But that one is no longer relevant |
| 08:29.50 | kaldemar | gah. astmasters.net has blocked crawlers in their robots.txt. :P no help from webarchive. |
| 08:34.36 | tzafrir | The developer of that was benjk? |
| 08:35.07 | tzafrir | [sp?] |
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| 08:59.04 | michaeljoser | Hi i am planning on installing a ISDN PRI line (24 channels) to be used with the DIGIUm TE120p and with asterisk.. i will be running VICIDIAL which is not compatible with chan_capi... now with my setup will I be running chan_capi? |
| 08:59.34 | michaeljoser | I am a bit confused with the reason why we use chan_capi |
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| 09:01.48 | Renacor | what value is supposed to be in ${CALLERIDNAME} if its from a sip phone |
| 09:01.54 | Renacor | the sip username? |
| 09:02.00 | kaldemar | michaeljoser: chan_capi is for BRI, you don't need it with PRI. |
| 09:03.12 | michaeljoser | kaldemar: ah ok so it should be fine then right, i will be using asterisk 1.2 series |
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| 09:04.19 | kaldemar | don't forget libpri then. |
| 09:04.44 | michaeljoser | ok thanks |
| 09:05.09 | kaldemar | and zaptel of course. |
| 09:06.00 | michaeljoser | yeah ^^ ouf i feel relieved |
| 09:06.11 | pkunkra | Argh! i've been trying to figure out UnpauseQueueMember() for the past hour. |
| 09:06.27 | pkunkra | anyone tried using that call before? |
| 09:06.43 | pkunkra | i'm calling UnpauseQueueMember(,SIP/chris) |
| 09:07.22 | pkunkra | its not recognizing that interface though. |
| 09:07.31 | pkunkra | i know that's what it *has* to be. |
| 09:07.37 | pkunkra | DEBUG[22050]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/chris |
| 09:08.02 | pkunkra | but i get this |
| 09:08.05 | pkunkra | WARNING[22203]: app_queue.c:3157 upqm_exec: Attempt to unpause interface SIP/chris}, not found |
| 09:08.12 | pkunkra | when i unpause it. |
| 09:08.23 | Strom_C | pkunkra: paste the actual line in your config file |
| 09:09.16 | pkunkra | exten => s,1,UnpauseQueueMember(,SIP/chris}) |
| 09:09.44 | pkunkra | just trying a straight unpause on it directly. |
| 09:09.55 | kaldemar | what's that } doing there? |
| 09:10.03 | Strom_C | exactly |
| 09:10.05 | Strom_C | TYPO CITY :) |
| 09:10.07 | pkunkra | ah.... |
| 09:10.23 | pkunkra | *smacks head on keyboard* |
| 09:10.49 | Strom_C | no, now you have exten => s,1,UnpauseQueueMember(,SIP/chris}eb4536ag*) |
| 09:10.59 | Strom_C | smacking your head on the keyboard is a bad idea :) |
| 09:11.32 | pkunkra | yeah, i'll get more typo's.... |
| 09:11.37 | pkunkra | lots of them. :-) |
| 09:11.43 | pkunkra | works perfectly now. |
| 09:11.52 | pkunkra | yeah, i'm an idiot. |
| 09:11.58 | pkunkra | someone shoot me now. |
| 09:13.59 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:23.22 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 09:26.57 | pkunkra | perfect! |
| 09:26.59 | pkunkra | exten => s,1,Set(UNPAUSE=${CUT(CHANNEL,-,1)}) |
| 09:26.59 | pkunkra | exten => s,n,UnpauseQueueMember(,${UNPAUSE}) |
| 09:27.05 | pkunkra | works like a charm. |
| 09:27.30 | pkunkra | turns a ""SIP/chris-08d41b38" into a "SIP/chris" |
| 09:30.52 | Strom_M | now the big question is why you're naming your peers "chris" and so on |
| 09:31.06 | Strom_M | it'd be better and more expandable to name them by their extension numbers |
| 09:31.26 | pkunkra | really? |
| 09:31.36 | pkunkra | hmmm |
| 09:32.17 | Strom_M | what if you have two people named "chris" using your system? |
| 09:32.23 | pkunkra | my thought was to assign that stanza to a specific person. |
| 09:32.25 | pkunkra | oh |
| 09:32.38 | pkunkra | i see your point. |
| 09:33.38 | pkunkra | yeah. that would be a problem. |
| 09:35.12 | *** part/#asterisk grndslm (n=grndslm@24-116-87-97.cpe.cableone.net) |
| 09:36.02 | Strom_M | better to just have SIP/2368 and so on |
| 09:36.56 | Wonka | what if you have someone named "h"? ;) |
| 09:37.36 | kaldemar | someone called "h" deserves to have problems. |
| 09:37.43 | Wonka | *g* |
| 09:37.43 | pkunkra | haha |
| 09:37.56 | Wonka | everything's possible in 'merika |
| 09:38.04 | pkunkra | so would someone called "2368" |
| 09:38.21 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-4caf7e16ad2b0c71) |
| 09:38.35 | Wonka | there's someone called "50 cent", i heard... |
| 09:38.37 | many | you dont name your boxes 172.21.1.2? |
| 09:38.51 | many | so give the phones hostnames and adress them by them |
| 09:39.52 | many | since boxes and phones are usually bound together, you can even use two different subdomains and use the boxen name for the phone too. so chris might have jupiter.office.yourcompany.com and the phone might be jupiter.phone.yourcompany.com |
| 09:40.19 | many | then just Dial(SIP/jupiter) |
| 09:40.28 | many | apply your naming scheme as appropriate |
| 09:40.44 | *** join/#asterisk chendy (n=chendy@218.242.110.26) |
| 09:41.24 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 09:42.10 | pkunkra | i normally pick names out of a common set. for example, i name all my servers after greek literary heros |
| 09:42.45 | Strom_M | yeah, but any naming scheme is going to fail miserably if you expand beyond a few dozen telephone sets |
| 09:42.55 | pkunkra | Dial(SIP/athena) for example. |
| 09:43.18 | pkunkra | hmm. i have a book with a few hundred greek names in it. |
| 09:43.36 | Strom_M | sigh |
| 09:43.52 | Strom_M | and you're going to cross-reference all those three hundred names against their extension numbers? |
| 09:43.58 | pkunkra | no |
| 09:44.03 | Strom_M | you're just creating extra work for yourself |
| 09:44.42 | pkunkra | yeah. extension numbers is probably the best bet. |
| 09:49.31 | *** join/#asterisk bintut (n=bintut@cm211.gamma177.maxonline.com.sg) |
| 09:51.05 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:55.17 | *** join/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
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| 10:01.14 | Strom_C | wheeee |
| 10:06.01 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 10:06.25 | Dr-Linux | anybody knows about Cepstral TTS price? |
| 10:06.58 | Strom_C | Dr-Linux: check their website |
| 10:07.08 | Strom_C | I recall a license being roughly US$20-$40 |
| 10:07.30 | Strom_C | also...when you ask that question in English, the proper phrasing is "Does anybody know about..." |
| 10:07.30 | Dr-Linux | Strom_C: i read everywhere per voice $30 |
| 10:07.56 | Strom_C | Dr-Linux: well, that would be between $20 and $40 then, wouldn't it. |
| 10:08.07 | Dr-Linux | but i'm confused if all i have to pay is for voice or i need some other license as well |
| 10:08.49 | Strom_C | the $30 license for the voice gives you unlmited TTS using that voice for one concurrent TTS session |
| 10:08.57 | Dr-Linux | Strom_C: well, i don't blame myself if i don't know english that's not my goal |
| 10:09.33 | Dr-Linux | english is not my native language |
| 10:09.46 | Dr-Linux | i'm sorry for wrong sentence |
| 10:10.06 | Strom_C | Dr-Linux: I'm not criticizing you; i'm giving you a suggestion to make yourself more intelligible in the future |
| 10:11.14 | Dr-Linux | Strom_C: Ohh i see, i'm sorry dude, i thought you said like i'm dumb or ... |
| 10:11.24 | Dr-Linux | Thanks for your suggestion |
| 10:11.51 | Dr-Linux | Strom_C: i like cepstral alot, now i wanna buy 2 voices |
| 10:13.30 | *** join/#asterisk zeeesh (i=zeeesh@202.166.161.45) |
| 10:33.31 | juuva | what's status of perl-agi with asterisk 1.4? |
| 10:39.16 | awk | hmm, anyone have a snom 300 here |
| 10:40.45 | awk | wondering if you redial button works when off hook ? |
| 10:40.58 | awk | as i dont want to be waisting my time here when it doesn't have that sort of functionality |
| 10:43.25 | *** join/#asterisk SA007 (n=sa007@ip565e0006.direct-adsl.nl) |
| 10:45.04 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 10:53.28 | *** join/#asterisk famicon (i=pastry@c51447ddc.cable.wanadoo.nl) |
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| 11:03.38 | WildPikachu | hrmmm, if I make zaptel channel 5 and 7 part of group 1 .... which one will it try first to dial out on? |
| 11:03.44 | WildPikachu | the on thats defined first? |
| 11:07.17 | Strom_C | depends on how you dial the group |
| 11:07.35 | Strom_C | if you Dial(ZAP/G1) it'll start with the high-numbered channel and work down |
| 11:07.50 | Strom_C | if you Dial(ZAP/g1) it'll start with the low-numbered channel and work up |
| 11:08.38 | WildPikachu | thanks a million man |
| 11:09.09 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128666903.dsl.bell.ca) |
| 11:13.08 | DrAk0 | anyone have any idea how to retrive/crack a SIP password ? |
| 11:13.27 | DrAk0 | my provider gave me a VoIP account but with an ATA ... |
| 11:14.53 | Strom_M | DrAk0: which provider? |
| 11:15.06 | DrAk0 | Strom_C, my local |
| 11:15.41 | DrAk0 | but is pointless plus i don't have any fxo available |
| 11:15.53 | Strom_M | "my local" is the name of the company? |
| 11:17.22 | Strom_M | DrAk0: you do realize you just messaged the account on the computer on the other side of my apartment, right? |
| 11:17.30 | Strom_M | strom_c != strom_m |
| 11:18.40 | DrAk0 | lol |
| 11:18.49 | DrAk0 | tab |
| 11:18.51 | DrAk0 | completation |
| 11:18.58 | Strom_M | paying attention, plzkthx |
| 11:19.05 | Strom_M | and why do I care who your ISP is? |
| 11:19.10 | Strom_M | I asked who your ITSP was |
| 11:19.14 | DrAk0 | same |
| 11:19.23 | DrAk0 | is the same -.- |
| 11:19.45 | Strom_M | I have no experience with that provider |
| 11:21.33 | *** join/#asterisk techy2U (n=techy@pool-72-84-19-208.pghk.east.verizon.net) |
| 11:32.53 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:44.57 | zapp-branigan | hi i have a problem loading g729 : -----> [Sep 6 13:24:29] WARNING[9562] loader.c: Error loading module 'codec_g729a.so': /usr/lib/asterisk/modules/codec_g729a.so: cannot restore segment prot after reloc: Permission denied |
| 11:45.56 | *** join/#asterisk blinky42 (n=me@67.106.175.130.ptr.us.xo.net) |
| 11:47.46 | *** join/#asterisk lincoln6e (n=lincoln6@ip68-227-216-225.dc.dc.cox.net) |
| 11:48.39 | lincoln6e | hello, anyone ever installed the Digium TE220 card in a PCI Express slot? |
| 11:51.55 | lincoln6e | PCI Express slot in a Dell rack server, that is |
| 11:54.03 | *** join/#asterisk cheGGo (n=cheGGo__@dslb-084-059-060-082.pools.arcor-ip.net) |
| 12:05.02 | *** join/#asterisk grEvenX (n=even@1ult2p8.ip.hipercom.no) |
| 12:05.23 | grEvenX | anyone using 1.4.X with peering of two or more asterisk servers? |
| 12:05.36 | grEvenX | got some problems at occations with re-invites |
| 12:06.03 | Daviey | asterisk 1.4 seems buggy with all types of re-invites imo |
| 12:08.54 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 12:09.54 | Wonka | hrmm |
| 12:10.07 | Wonka | i got an asterisk listening for SIP on udp/5061. |
| 12:10.14 | Daviey | cool |
| 12:10.24 | Wonka | "INVITE ext@asterisk" works |
| 12:10.34 | Wonka | "INVITE ext@asterisk:5061" is answered with 404 |
| 12:10.51 | Wonka | erm. "INVITE sip:..." in both cases |
| 12:11.15 | Wonka | ("asterisk" substituted for the fqdn, btw) |
| 12:11.38 | Wonka | i don't really like that |
| 12:12.10 | Daviey | Wonka: sip debug? |
| 12:12.42 | Wonka | haven't tried yet... only tcpdump |
| 12:14.53 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:15.08 | Wonka | »Found no matching peer or user for '213.178.67.244:5061'« |
| 12:15.12 | Wonka | might be a reason |
| 12:21.15 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:22.04 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:25.13 | Wonka | thx for the idea - problem solved by adding a friends entry to sip.conf |
| 12:27.50 | *** join/#asterisk Aeudian (n=Aeudian@74.92.134.190) |
| 12:27.55 | *** part/#asterisk Aeudian (n=Aeudian@74.92.134.190) |
| 12:28.44 | grEvenX | hope I don't have to go back to 1.2 just because of this :( |
| 12:29.24 | Uatec | ? |
| 12:37.16 | *** join/#asterisk Rawplayer (n=kevin@cp108757-a.landg1.lb.home.nl) |
| 12:38.32 | grEvenX | the re-invite issues we're having with 1.4 |
| 12:38.51 | *** join/#asterisk GaryH (n=chatzill@2001:618:42d:0:213:72ff:fecf:8262) |
| 12:46.38 | *** part/#asterisk heh_v_water (n=heh_v_wa@209-181-28-101.hlna.qwest.net) |
| 12:47.35 | *** join/#asterisk cavediver (i=jonas@trimix.eklof.eu) |
| 12:48.30 | Dr-Linux | does anybody using Cepstral? |
| 12:48.37 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-72-75-249-194.bflony.east.verizon.net) |
| 12:48.42 | cavediver | Just installed 1.4.11 and the beta-gui. Is there a manual/guide for the gui somewhere ? |
| 12:48.45 | cavediver | for beginners. |
| 12:48.46 | SuPrSluG | hello all |
| 12:49.11 | SuPrSluG | cave diver:yes check the digium forums |
| 12:49.25 | *** join/#asterisk guillote_GNU (n=bancaria@host185.190-136-203.telecom.net.ar) |
| 12:49.31 | SuPrSluG | cave diver:someone there wrote one |
| 12:49.46 | cavediver | Mkay. |
| 12:50.19 | SuPrSluG | sorry, i don't have the exact link. but it is there |
| 12:51.43 | SuPrSluG | almost have my nokia n80 working. anyone have any experience w/ nokia setup? |
| 12:51.53 | SA007 | k, maybe anyone here can give me a pointer in the right direction, incoming call's work fine, outgoing is ringing, but audio is fucked up |
| 12:52.21 | SA007 | (only one way and that is a bit jagged) |
| 12:52.50 | SA007 | nothing's behind nat |
| 12:53.10 | SuPrSluG | SA007:try using qualify=yes in sip.conf. if it is more than 200ms this could cause the problem |
| 12:53.33 | SA007 | what morge than 200ms? |
| 12:54.12 | SA007 | in what part btw? the outgoing line or the sip client(s) or both? |
| 12:55.46 | SA007 | ok, now the only audio i hear is the cell-buzz though my pots phone :P, no normal audio at all |
| 12:55.50 | SuPrSluG | i can register and call from an internal network. i can call my provider number and get audio both ways. when i make a call from the n80 from and outside wifi nothing happens. as if no tones are generated to tell asterisk to call. any ideas? |
| 12:55.53 | [TK]D-Fender | SuPrSluG: that has nothing to do with qualify.... |
| 12:56.31 | [TK]D-Fender | SuPrSluG: thats only a NAT keep-alive |
| 12:56.43 | *** join/#asterisk Shido6 (n=shido6@204.126.120.132) |
| 12:56.43 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 12:56.54 | SuPrSluG | won't it give an idea as to link to the provider? |
| 12:59.03 | *** join/#asterisk af_ (n=getsmart@81-174-47-128.dynamic.ngi.it) |
| 12:59.04 | [TK]D-Fender | SuPrSluG: sorry but you are simply not making any sense |
| 13:01.09 | *** join/#asterisk vader-- (n=me@c-71-226-197-0.hsd1.nj.comcast.net) |
| 13:01.59 | SuPrSluG | what i am asking is -> won't using qualify give you an ideas as to the quality of the link between your asterisk server and your service provider? |
| 13:02.02 | SA007 | ok? wtf, i turned sip debug on to see what's going on, and suddenly it works? |
| 13:02.23 | mocker | SA007: Asterisk has gnomes. |
| 13:02.37 | SA007 | gnomes? you meen gremlings? |
| 13:02.53 | mocker | They do the same thing. :) |
| 13:02.59 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:03.06 | mocker | %Except one uses gtk. |
| 13:03.18 | SA007 | asterisk doesn't use gtl |
| 13:03.20 | *** join/#asterisk umanghc (n=umanghc@ool-182fface.dyn.optonline.net) |
| 13:03.21 | SA007 | gtk |
| 13:03.27 | mocker | SA007: kidding. |
| 13:04.46 | SA007 | and now it stopped working again... |
| 13:08.13 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:14.14 | SA007 | anyone any idea to get the gremlins to reconnect everything and get them out? |
| 13:14.34 | threat | http://www.youtube.com/watch?v=HCJcvVWRABs |
| 13:14.37 | SA007 | i don't like a 'maybe you get sound, but probably not' phone :P |
| 13:15.07 | *** join/#asterisk the_Goat_ (n=chatzill@h-67-103-23-130.phlapafg.covad.net) |
| 13:15.32 | the_Goat_ | has anyone had experience using cisco 7960 phones with saterisk? |
| 13:16.34 | SA007 | threat: briljant video |
| 13:16.50 | threat | SA007: you like it? |
| 13:17.10 | SA007 | very funny :) |
| 13:17.34 | threat | good old Austrian comedy :) |
| 13:17.38 | threat | you are rober! |
| 13:17.52 | threat | Australian even |
| 13:18.24 | *** join/#asterisk michael-i (n=michael-@141.41.40.55) |
| 13:21.37 | michael-i | I'm having timing issues with IAX channels when I use ztdummy and also when I use a zaptel card for my timing source. Any "gotcha"s I need to know about? (using 1.4.11 on FreeBSD 6.2) |
| 13:22.20 | *** part/#asterisk Daviey (n=dave@ubuntu/member/daviey) |
| 13:23.37 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
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| 13:27.16 | *** part/#asterisk popey (n=alan@ubuntu/member/popey) |
| 13:27.45 | penguinFunk | anyone here had much luck with * and conference calling? |
| 13:29.13 | lirakis | yes |
| 13:30.31 | wwalker | anyone know the ballpark unit 1 pricing for the asterisk appliance 50? |
| 13:32.25 | tzafrir | penguinFunk, what sort of conference calling? |
| 13:33.16 | *** join/#asterisk crycos (n=crycos@72.54.46.18) |
| 13:33.51 | wwalker | nm, found it |
| 13:34.38 | penguinFunk | tzafrir: I want any persons who dial the conference call extension to all be on the same phone calls |
| 13:35.05 | penguinFunk | call* |
| 13:35.16 | penguinFunk | n-way calling effectively |
| 13:35.22 | penguinFunk | is there an easy way to do this? |
| 13:36.32 | tzafrir | penguinFunk, meetme? |
| 13:38.04 | Wonka | there was something with meetme that made me try app_conference, months ago... |
| 13:38.35 | *** join/#asterisk Daviey (n=dave@ubuntu/member/daviey) |
| 13:38.48 | penguinFunk | thanx, looks like this is what i need |
| 13:40.36 | *** join/#asterisk blitzrage[E61i] (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:40.36 | *** mode/#asterisk [+o blitzrage[E61i]] by ChanServ |
| 13:50.29 | bintut | tzafrir: regarding http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=438702 |
| 13:50.33 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.25.53) |
| 13:51.01 | bintut | tzafrir: i just noticed that it was tagged as unreproducible |
| 13:51.22 | tzafrir | What is the scenario again? |
| 13:52.17 | bintut | tzafrir: the asterisk crash when a wengophone inside the lan calls a pstn telephone number through a zap channel |
| 13:53.05 | bintut | tzafrir: i don't know if the main asterisk project knows about this crash problem or the problem only exist in debian |
| 13:54.42 | tzafrir | can you reproduce the same wiht your current installation? |
| 13:54.46 | tzafrir | what version is that? |
| 13:56.48 | crycos | SOS :) |
| 13:56.50 | crycos | How can I redirect incoming fax to a desired Shared Folder, instead of the standard /asteris/fax |
| 13:57.10 | bintut | tzafrir: i already posted all the information the package maintenainers need.. i remember that faidon was telling me that the crash problem only occurs on my setup because i built the asterisk 1.4.9 and 1.4.11 by myself.. but i used the binary packages you have in your build server and still, it crashed. |
| 13:57.35 | bintut | tzafrir: it is consistent.. it crashes all the time on my side.. |
| 13:58.00 | tzafrir | ok. I'll try reproducing it tonight on my home server |
| 13:58.08 | SuPrSluG | i got my n80 working!! |
| 13:58.37 | SuPrSluG | secret is not to enter any proxy registration server |
| 13:58.43 | SuPrSluG | info |
| 13:58.44 | bintut | tzafrir: please follow the scenario on how i was able to reproduce the crash issue |
| 13:58.59 | *** part/#asterisk blitzrage[E61i] (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:59.20 | *** join/#asterisk mog (i=mog@nat/digium/x-e407a9916f52d85c) |
| 13:59.20 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:59.20 | bintut | tzafrir: please refer the discussions at http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=438702 |
| 14:00.21 | bintut | tzafrir: i was thinking that maybe that problem is not being discussed here or in the mailing list maybe because no one's using the 1.4.11 in their production environment.. all are using the 1.2 series.. |
| 14:00.56 | bintut | ..and no one able to find it out that there's a consistent crash problem in the 1.4.11.. |
| 14:01.17 | bintut | ..or maybe because it's only specific to the debian unstable problem.. |
| 14:05.26 | bintut | tzafrir: thank you in advance. i know you're busy but hopefully you and your team will give time to fix that crash problem.. |
| 14:06.01 | bintut | tzafrir: i have to go now.. thank you once again.. :) |
| 14:07.00 | *** join/#asterisk jfitzgibbon (n=chatzill@64.72.237.130) |
| 14:12.24 | *** join/#asterisk coolbeans (n=null@adsl-070-148-053-190.sip.bhm.bellsouth.net) |
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| 14:13.16 | coolbeans | Does anyone have a SIP Cisco 7940 TFTP directory then can send me (Any recent firmware will do), someone just droped about 40 SCCP phones on my desk to convert to SIP and I'd rather not go the painful route of doing it the other way... |
| 14:16.28 | *** part/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
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| 14:27.12 | rody | quick q about ARI: should the default view have a "play" link like the Voicemail view is? |
| 14:27.20 | *** join/#asterisk umanghc (n=umanghc@ool-182fface.dyn.optonline.net) |
| 14:27.24 | rody | i see it in the code.. i just don't see it on my page |
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| 14:37.55 | rpyne | can anyone point me to any good mysql + asterisk demo applications with dial plan logic for learning? thanks |
| 14:38.36 | Shido6 | what do you want to do with asterisk? |
| 14:38.57 | rpyne | im trying to build a dial plan based lcr app using mysql |
| 14:39.01 | *** join/#asterisk jmesquita (n=jmesquit@c915230f.virtua.com.br) |
| 14:39.13 | Shido6 | the ones that exist just dont do it for you , eh? |
| 14:40.38 | rpyne | no, i have some additional logic to place in due to multiple switches and such, ive tried lcdial is there something else that i can check out? |
| 14:40.41 | *** join/#asterisk ming_zym (n=ming_zym@124.254.53.129) |
| 14:40.53 | *** join/#asterisk davevg-btwtech (n=davevg@nj-67-76-177-147.sta.embarqhsd.net) |
| 14:41.25 | *** join/#asterisk coolbeans (n=null@adsl-070-148-053-190.sip.bhm.bellsouth.net) |
| 14:41.51 | coolbeans | Hey guys, how do you reset an older Cisco 7940 SCCP phone to factory defaults? The "Hold down # on power up" method doesn't seem to work. |
| 14:42.07 | coolbeans | .. And Google isn't being very helpful this morning |
| 14:43.03 | watchy | where would i put something in gentoo to start automaticly? |
| 14:43.12 | *** join/#asterisk rody (i=netstati@neptune.negativeblue.com) |
| 14:46.06 | ozus | watchy : i think it should be rc-update add servicename default if if i remember correctly |
| 14:47.32 | watchy | hrm if theres a script for ity |
| 14:47.34 | *** join/#asterisk vlt (n=dm@suez.musketa.de) |
| 14:47.45 | SA007 | watchy: rc-update add <name> default |
| 14:48.15 | watchy | can i just put /usr/sbin/wanrouter start |
| 14:48.26 | watchy | in a file and add it to startu |
| 14:48.26 | watchy | p |
| 14:48.32 | SA007 | nope, that's just for /etc/init.d/ scripts |
| 14:48.56 | SA007 | so you'll have to make a script, or add it to local.start in /etc/conf.d |
| 14:49.05 | vlt | Hello. I have a "callgroup" and "pickupgroup" line in sip.conf. Where to put these lines for ZAP phones? |
| 14:49.27 | penguinFunk | vlt: zapata.conf? |
| 14:49.43 | watchy | hmm |
| 14:49.58 | watchy | does local.start start before /etc/init.d? |
| 14:50.09 | SA007 | after |
| 14:50.20 | watchy | well i need wanpipe to start before * |
| 14:50.23 | watchy | or else shit breaks |
| 14:50.26 | *** join/#asterisk Woifi1988 (n=wolfgang@M1027P029.adsl.highway.telekom.at) |
| 14:50.29 | Woifi1988 | hi |
| 14:50.38 | *** join/#asterisk UdontKnow (i=Evaldo@freenode/staff/udontknow) |
| 14:50.45 | SA007 | during boot jou get 'Starting Local' as last line, then it runs /etc/conf.d/local.start |
| 14:50.49 | Woifi1988 | anyone here who made expirensis with ekiga? |
| 14:50.59 | watchy | well i HAVE to have wanpipe start first |
| 14:51.03 | watchy | or else * wont even start |
| 14:51.29 | SA007 | it runs local.start as very last thing in you boot, so you should be fine |
| 14:51.53 | watchy | uh |
| 14:51.58 | watchy | so its gonna start asterisk first |
| 14:52.08 | watchy | then run wanpipe which i put in local.start |
| 14:52.15 | SA007 | ah |
| 14:52.25 | watchy | then i'm gonna have makor issues |
| 14:52.28 | watchy | major |
| 14:52.42 | SA007 | 2 choices: add wanpipe to the aserisk script (not a good idea) |
| 14:52.50 | watchy | i agree |
| 14:53.02 | SA007 | or make a wanpipe scipt that runs before asterisk |
| 14:53.14 | watchy | shit if i was on site i'd fuck with that |
| 14:53.18 | ManxPower | Uh, the wanpipe install process ASKS you if you want to start wanpipe on system boot. |
| 14:53.25 | watchy | but if they reboot this fucker today and something break |
| 14:53.26 | watchy | s |
| 14:53.31 | *** join/#asterisk Tili (n=tili@cm48.gamma244.maxonline.com.sg) |
| 14:53.35 | watchy | i'm going to be outta town |
| 14:53.39 | SA007 | just don't reboot it ;) |
| 14:53.43 | Woifi1988 | anyone here who made expirensis with ekiga? |
| 14:53.52 | watchy | they are having line issues |
| 14:54.01 | watchy | its the fucking telco though but they don't think it is |
| 14:54.38 | watchy | fuck them cock suckers i have to go outta town |
| 14:54.41 | watchy | i don't give a fuck |
| 14:54.53 | SA007 | watchy: lol |
| 14:55.03 | watchy | if they reboot it they will do without phones today |
| 14:55.59 | SA007 | just place a large red buttun in the hallway with a 'don't press this' sign next to it which reboots the server :P |
| 14:56.03 | Woifi1988 | why can't ekiga play a gsm file from asterisk? |
| 14:56.20 | [TK]D-Fender | watchy: Don't hold back, tell us how you REALLY feel.... |
| 14:56.24 | SA007 | Woifi1988: recode it? |
| 14:56.43 | Woifi1988 | SA007: the gsm file? |
| 14:56.45 | watchy | haha thanks tk i need someone to let my agression out on |
| 14:56.51 | watchy | i want to punch u in the head :) |
| 14:56.56 | SA007 | Woifi1988: why not |
| 14:57.11 | Woifi1988 | SA007: with X-Lite it works! |
| 15:00.27 | coolbeans | .. And Google isn't being very helpful this morning |
| 15:00.28 | coolbeans | Hey guys, how do you reset an older Cisco 7940 SCCP phone to factory defaults? The "Hold down # on power up" method doesn't seem to work. |
| 15:00.30 | *** join/#asterisk sevard (n=sev@192.235.0.85) |
| 15:01.10 | Qwell | hold down #, when the lights change, 123456789*0# I think |
| 15:01.38 | coolbeans | That works great on the newer firmware, but for some reason with this firmware, which is 3.1(MF.G2), the method doesn't do anything. |
| 15:01.50 | coolbeans | Qwell, thanks, though. |
| 15:02.24 | Woifi1988 | nobody who uses ekiga? |
| 15:04.15 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 15:04.19 | doughecka | Is it possible to have a analog phone plugged into asterisk, and when the handset is picked up, asterisk will automatically connect that line to a ring group? |
| 15:04.33 | doughecka | like a hot phone setting... |
| 15:04.35 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 15:04.57 | coolbeans | doughecka: No. |
| 15:05.36 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 15:05.59 | coolbeans | Someone needs to sell a "batphone" that works like that. We could use them. |
| 15:06.08 | coolbeans | a sip "batphone" that is... |
| 15:06.16 | doughecka | well, sip phones can do it |
| 15:06.19 | *** join/#asterisk cryc0s (n=crycos@72.54.46.18) |
| 15:06.23 | doughecka | my cheapy grandstream does it |
| 15:06.31 | doughecka | but I want analog, for a door phone |
| 15:06.33 | cryc0s | How can I redirect incoming fax to a desired Shared Folder, instead of the standard /asteris/fax |
| 15:06.40 | coolbeans | No, I mean with no buttons. Just pick up the handset and get connected ... |
| 15:07.05 | *** join/#asterisk exvito (n=exvito@195.245.132.93) |
| 15:07.18 | doughecka | I know viking sells a auto dialer for analog phones, but surely asterisk can handle something as simple as this... |
| 15:07.23 | doughecka | is ringdown the correct term? |
| 15:07.34 | coolbeans | Hey guys, how do you reset an older Cisco 7940 SCCP phone to factory defaults? The "Hold down # on power up" method doesn't seem to work. |
| 15:09.43 | exvito | hi, will asterisk be able to divert a call when it's already ringing in the destination extension ? example: A dials B, B rings, user in B interacts with system (maybe with an application via AMI/AGI) so as to divert the call to C, B stops ringing, C starts ringing... Is this possible ? |
| 15:11.54 | cheGGo | exvito, u can handle this via the Asterisk Manager API |
| 15:12.19 | exvito | cheGGo: hmmm... great, let me check... |
| 15:12.54 | cheGGo | not easy - but possible ;) |
| 15:13.03 | *** join/#asterisk saftsack (n=saftsack@pD9E06623.dip.t-dialin.net) |
| 15:13.03 | cheGGo | afk |
| 15:13.19 | exvito | cheGGo: thanks ! I got it... it's the "Redirect" manager command, I believe ! |
| 15:13.26 | *** join/#asterisk Op3r (n=Op3r@121.97.145.174) |
| 15:13.30 | cheGGo | exactly |
| 15:13.35 | cheGGo | now afk =) |
| 15:15.20 | *** part/#asterisk exvito (n=exvito@195.245.132.93) |
| 15:21.27 | doughecka | Its been awhile since I last messed with this, but someone tell me if I am correct... |
| 15:21.27 | doughecka | an zap FXS port on asterisk, if I set its context to, say, hotline, and then the hotline context had exten => s,1,Dial(SIP/100), as soon as the line was taken off hook, it would dial that number. |
| 15:21.27 | doughecka | Does that sound correct? |
| 15:21.57 | putnopvut | doughecka: the option you're thinking of is called "immediate" |
| 15:22.31 | putnopvut | It doesn't matter what you name the context. |
| 15:22.34 | ManxPower | doughecka: that is correct. |
| 15:23.24 | doughecka | ManxPower: and I set immediate in zapata.conf for that channel as well, correct? |
| 15:27.03 | *** join/#asterisk dr0ck (i=dr0ck@nat/digium/x-acdaf18b1b10b4bc) |
| 15:27.11 | ManxPower | yes. |
| 15:27.25 | neverblue | where are * error submitted to, I want to look at an error someone reported ? |
| 15:27.31 | neverblue | errors* |
| 15:29.13 | neverblue | nm, found it |
| 15:43.16 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 15:43.31 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
| 15:45.31 | Ryushin | Is there a linux softphones that uses alsa instead of oss? I know zoiper beta was just released, but it segfaults every time I try to run it. |
| 15:45.49 | Ryushin | Plus it's not open source which bothers me. |
| 15:49.31 | [TK]D-Fender | Ryushin: X-Lite & Wengo |
| 15:49.52 | [TK]D-Fender | Ryushin: Or Ekiga..... |
| 15:50.01 | [TK]D-Fender | Ryushin: Every heard of Google? :) |
| 15:50.28 | [TK]D-Fender | Ryushin: You'd be amazed at the instant gratification http://www.google.ca/search?hl=en&sa=X&oi=spell&resnum=0&ct=result&cd=1&q=alsa+softphone&spell=1 provided :p |
| 15:52.03 | *** join/#asterisk ware (i=w@shiz.nigz.mineyaown.biz) |
| 15:52.44 | andresdb | hole |
| 15:52.58 | UdontKnow | crater |
| 15:54.24 | andresdb | i have a linksys spa3102 and i need the calls from the pstn fxo forward to fxs, and the fxs forward to ivr in my asterisk |
| 15:54.39 | andresdb | its posible |
| 15:54.50 | *** part/#asterisk ware (i=w@shiz.nigz.mineyaown.biz) |
| 15:55.26 | andresdb | in spa i have configure 2 sip account |
| 15:55.37 | andresdb | 1 for fxo and 1 for fxs |
| 15:58.35 | Ryushin | [TK]D-Fender: I haven't been googling. The best resource was this that I found so far: http://www.voip-info.org/wiki-Asterisk+IAX+clients |
| 15:59.32 | *** join/#asterisk tsurko (n=tsurko@78.90.100.214) |
| 16:00.03 | Ryushin | The only IAX ALSA phone I've been able to find is zoiper, and it doesn't run yet. I don't want to use sip. |
| 16:05.07 | Ryushin | [TK]D-Fender: Sometimes I should read what I write before pressing enter. I mean I have been googling. A lot. Just not much luck. |
| 16:07.15 | penguinFunk | if you have an IAX2 trunk between 2 * boxes will sip users on one side be able to talk to sip users on the other side? |
| 16:07.25 | penguinFunk | or do you have to define a bunch of IAX2 users for this? |
| 16:07.49 | penguinFunk | meaning that every person will have to have a sip user and an IAX2 user |
| 16:07.51 | penguinFunk | :( |
| 16:08.10 | penguinFunk | defined* |
| 16:08.46 | *** join/#asterisk tsurko (n=tsurko@78.90.100.214) |
| 16:12.58 | penguinFunk | or can i somehow have sip phone/user <-> asterisk1 <-IAX2-> asterisk2 <-> sip phone/user ? |
| 16:13.48 | penguinFunk | is sip an IAX2 mutually exclusive? |
| 16:15.59 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 16:16.09 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:20.43 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 16:21.37 | Corydon76-vcch | penguinFunk: No, why would they be? |
| 16:21.50 | *** join/#asterisk jmls (n=jmls@62.49.235.130) |
| 16:21.58 | jmls | hey guys |
| 16:22.07 | penguinFunk | so you can have sip users communicating with each other over n IAX2 trunk between two * boxes? |
| 16:22.16 | dasuberdavid | yes |
| 16:22.22 | penguinFunk | excellent |
| 16:22.24 | penguinFunk | thanx |
| 16:23.30 | jmls | is there any circumstances where an isdn line would keep a call up after the call is "hung up" for 8, 16 or 32 hours, and then redial the desitnation number and start the call all over again ? |
| 16:24.55 | outtolunc | sure, most isdn modem/internet setups are like that |
| 16:25.18 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 16:25.23 | outtolunc | (especially idsl setups) |
| 16:25.39 | jmls | we are using a EuroISDN purely for voice calls |
| 16:26.18 | outtolunc | make sure you aren't using any DOV settings |
| 16:26.25 | jmls | ie no modem or internet access. All calls are originated from the asterisk AMI and sent to a SIP phone first, then dials the destination |
| 16:26.26 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 16:26.34 | jmls | outtolunc: what's a DOV ? |
| 16:26.40 | outtolunc | data over voice |
| 16:26.46 | jmls | nope. definately not. |
| 16:27.20 | outtolunc | then i'd have to say a config issue, either your config, the card config, or provider config |
| 16:27.34 | Corydon76-vcch | modem over voip is hot |
| 16:27.45 | *** join/#asterisk t3rror (n=t3rror@gateway.sscgp.com) |
| 16:28.00 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
| 16:28.22 | jmls | we've been running * since November 2004 : this is the first time we've had anything like this. During the period of these calls, we did not update or upgrade asterisk or any config |
| 16:28.45 | *** join/#asterisk davevg-btwtech (n=davevg@nj-67-76-177-147.sta.embarqhsd.net) |
| 16:28.52 | jmls | It seems to have happened since we moved our calls back to BT from a LCR company |
| 16:28.56 | outtolunc | well there are 2 issues i noted, the long disconnect, and the redial |
| 16:29.58 | jmls | outtolunc: I sent a mail to -users showing the dates / times and length of the calls |
| 16:30.00 | outtolunc | the long disconnect, can be a config/setup issue where even tho it was 'sent' a hangup, the other/both ends never agreed it was, and the redial as if it thought it should 'stay up' |
| 16:30.35 | jmls | outtolunc: what config options can alter these ? We make 5000+ calls per day, and it only happen |
| 16:30.43 | jmls | in a small number of cases |
| 16:31.26 | outtolunc | obviously there is a certain situation where the line cause is in a state that one end or both do not understand |
| 16:31.26 | *** join/#asterisk klictel (n=klictel@atelka.info) |
| 16:31.43 | jmls | yeah, that's what I'm trying to find :) |
| 16:32.12 | jmls | the cdr record states that the call lasted 4 minutes ... |
| 16:32.34 | outtolunc | remember that old channel bug where asterisk would bridge the call when it was still in proceeding, yet determine 20 secs later because of a timeout that oops, need to hangup... we think like that but in reverse |
| 16:32.46 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 16:33.25 | jmls | you think it's a chan_zap bug ? |
| 16:33.56 | *** join/#asterisk HCEvan (i=edude@pool-72-77-226-91.tampfl.fios.verizon.net) |
| 16:34.07 | outtolunc | it could be anywere in that setup, chanzap/libpri/channel |
| 16:34.18 | outtolunc | or not even in asterisk |
| 16:34.24 | jmls | ... |
| 16:34.37 | jmls | _could_ it be a BT bug ? |
| 16:34.47 | outtolunc | the only way you will find it is a full debug and do the call progress of that call |
| 16:35.01 | jmls | that's pretty hard 4 months down the line .. |
| 16:35.08 | outtolunc | i hear you |
| 16:35.42 | outtolunc | your easiest test, if you have access is do a test over an other provider |
| 16:36.39 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 16:39.13 | *** join/#asterisk ToTo (n=ToTo@host45-201-dynamic.2-87-r.retail.telecomitalia.it) |
| 16:40.26 | jmls | that's the issue - how can I make it happen if it's happened 10 times out of 500,000 calls ? |
| 16:41.06 | jmls | it never happened before - I've asked BT to tell me if it's happened since ... because in August we upgraded * to the latest 1.4 trunk |
| 16:41.46 | outtolunc | the obvious answer is you can't, so you would need to setup a tripwire of sorts to help you collect enough data to find the needle in that haystack |
| 16:42.03 | jmls | eep. |
| 16:42.16 | outtolunc | that or regress versions as test |
| 16:43.26 | *** join/#asterisk sekretarz (n=sekretar@gentoo/developer/sekretarz) |
| 16:43.31 | sekretarz | hi |
| 16:44.11 | sekretarz | i'm newbie in asterisk and i can't start it correctly, when loading asterisk in logs writes errors like that: |
| 16:44.15 | sekretarz | loader.c: Error loading module 'chan_sip.so': /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_park_call |
| 16:44.20 | sekretarz | i don't know what can it be |
| 16:44.24 | file | load res_features.so before chan_sip.so |
| 16:44.42 | sekretarz | in modules.conf i've added res_adsi.so |
| 16:44.51 | sekretarz | file: yeah, ;) |
| 16:44.59 | sekretarz | 'res_features.so': /usr/lib/asterisk/modules/res_features.so: undefined symbol: ast_monitor_st |
| 16:45.02 | sekretarz | op |
| 16:45.04 | file | load res_monitor.so |
| 16:45.17 | file | module dependencies are oh so fun |
| 16:45.50 | sekretarz | hehe |
| 16:45.52 | sekretarz | ok, it works |
| 16:45.54 | sekretarz | now |
| 16:46.05 | sekretarz | file: thanks |
| 16:46.25 | file | yw |
| 16:55.44 | *** join/#asterisk Arno[Slack] (n=hellSOUN@gre92-1-81-57-177-108.fbx.proxad.net) |
| 16:58.41 | *** join/#asterisk Tili (n=tili@cm48.gamma244.maxonline.com.sg) |
| 17:00.26 | *** join/#asterisk jmesquita (n=jmesquit@c915230f.virtua.com.br) |
| 17:02.11 | cryc0s | How can I redirect incoming fax to a desired Shared Folder, instead of the standard /asteris/fax |
| 17:04.30 | Corydon76-vcch | cryc0s: sounds like you're using some other product on top of Asterisk |
| 17:04.41 | cryc0s | well |
| 17:04.44 | cryc0s | here is the thing |
| 17:05.29 | cryc0s | faxes received by asterisk go into the asterisk/fax folder than are converted to pdf and sent to an email addr |
| 17:05.34 | *** join/#asterisk ming_zy1 (n=ming_zym@124.254.53.129) |
| 17:05.47 | Qwell | by what? Asterisk doesn't do anything with faxes |
| 17:05.50 | cryc0s | how can I change the asterisk/fax folder to something else ? |
| 17:06.11 | cryc0s | hmm |
| 17:06.21 | cryc0s | is it the faxrx then ? |
| 17:08.47 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 17:08.54 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 17:09.26 | *** join/#asterisk VJFROMGT (n=vjfromgt@static-68-161-227-229.ny325.east.verizon.net) |
| 17:09.51 | VJFROMGT | I want to say all cid that start with 876 must go to x |
| 17:09.53 | VJFROMGT | can that be done? |
| 17:09.55 | Qapf | what debug do i enable to see asterisk's logic in assigning a call a context? |
| 17:11.46 | iCEBrkr | set core verbose 9 |
| 17:11.48 | iCEBrkr | err |
| 17:11.51 | iCEBrkr | core set verbose 9 |
| 17:11.53 | iCEBrkr | or something |
| 17:12.01 | iCEBrkr | Qapf: it'll spam your console with all you need. |
| 17:12.07 | [TK]D-Fender | andresdb: Yes you can use the SPA-3102 ' FXO port both ways jsut fine. Go check out the forums at http://www.voxilla.com for instructions. |
| 17:12.18 | iCEBrkr | [TK]D-Fender: whassup homie |
| 17:12.35 | [TK]D-Fender | Qapf: Depends on what kind of channel the call is coming in on. |
| 17:12.38 | [TK]D-Fender | iCEBrkr: y0 |
| 17:12.58 | iCEBrkr | So what have I missed? |
| 17:12.59 | iCEBrkr | anything good? |
| 17:13.02 | iCEBrkr | :P |
| 17:13.09 | Qapf | [TK]D-Fender, i have voicepulse coming in on sip, and for some reason its getting from-sip-external meaning * thinks its an anonymous external call |
| 17:13.29 | Qapf | when it should be from-trunk indicating it is from one of my trunks, as it is |
| 17:13.33 | Qapf | or from-pstn |
| 17:14.09 | Qapf | im trying to see why asterisk is picking this considering the lines for vp say context=from-pstn |
| 17:14.21 | [TK]D-Fender | Qapf: enable sip debug and watch the call come in. |
| 17:14.35 | [TK]D-Fender | Qapf: and pastebin it along with your sip.conf masking only passwords |
| 17:14.37 | [TK]D-Fender | ~pb |
| 17:14.37 | jbot | A Pastebin is a place to paste your stuff without flooding 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://www.rafb.net/paste |
| 17:14.39 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^ |
| 17:14.45 | [TK]D-Fender | iCEBrkr: Not a whole hell of a lot. |
| 17:14.51 | Qapf | [TK]D-Fender, ok |
| 17:15.00 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 17:15.02 | [TK]D-Fender | iCEBrkr: more bugs, more emergency fixes, more CRITICAL bugs, etc... |
| 17:15.08 | iCEBrkr | Awesome |
| 17:15.13 | iCEBrkr | My Asterisk box just runs. |
| 17:15.19 | iCEBrkr | I forget about it |
| 17:16.25 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 17:16.28 | sekretarz | ok, next problem, i'm tring to use asterisk gui |
| 17:16.42 | sekretarz | when i execute install.html it shows only templete |
| 17:16.58 | VJFROMGT | can i use wildcard in caller id? |
| 17:17.00 | sekretarz | Next and Back buttons are disabled |
| 17:17.30 | sekretarz | when i'm logged off it shows login box but don't authenticate |
| 17:18.42 | *** join/#asterisk HeMan (n=jimmy@1-1-7-40a.far.sth.bostream.se) |
| 17:18.59 | sekretarz | whan can it be? |
| 17:19.04 | sekretarz | what* |
| 17:19.43 | *** join/#asterisk ygguh2 (n=concilio@ool-44c57e6c.dyn.optonline.net) |
| 17:21.00 | Qapf | [TK]D-Fender, im using trixbox, so im assuming you want the sip_additional file that actually contains the interesting things |
| 17:22.04 | [TK]D-Fender | ~trixbox |
| 17:22.05 | jbot | extra, extra, read all about it, trixbox is a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
| 17:22.05 | ygguh2 | Im having problems building zaptel-1.4.5.1, zaptel-1.4.4 and zaptel-1.2.20.1 on my redhat FC4 linux 2.6 server. The make works okay but when I run insmod ztdummy.ko I receive Unkown symbol in module. |
| 17:22.22 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 17:22.26 | Qwell | ygguh2: what symbol? |
| 17:22.32 | HeMan | Hi! Is there any way to tell asterisk to not send the entire caller id information to a SIP-phone? |
| 17:22.49 | ygguh2 | zt_receive, zt_transmit, zt_unregister, zt_register |
| 17:23.02 | Qwell | Use modprobe - it'll load zaptel also |
| 17:23.09 | HeMan | I don't want the "<sip:number@ip>" part |
| 17:23.17 | [TK]D-Fender | HeMan: Yes. CHANGE IT before you DIAL. |
| 17:23.27 | Qapf | [TK]D-Fender, so in short, go away? |
| 17:23.35 | ygguh2 | tried that to, insmod zaptel.ko, unknow symbol in module, crc_ccitt_table |
| 17:23.35 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 17:23.43 | Qwell | ygguh2: use modprobe |
| 17:24.01 | Qwell | it will resolve all of the symbols and load the appropriate modules |
| 17:24.04 | [TK]D-Fender | Qapf: It means forget about raw file manipulation, and there are tons of guides on how to do it properly through FreePBX |
| 17:24.13 | Qwell | s/properly// |
| 17:24.22 | Qwell | You can't do anything properly through freepbx |
| 17:24.26 | [TK]D-Fender | Qapf: http://aussievoip.com/wiki/LCR+With+FreePBX+and+VoicePulse |
| 17:25.10 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 17:25.11 | [TK]D-Fender | oops, bad link |
| 17:25.28 | [TK]D-Fender | Qapf: Easiest way : http://www.google.ca/search?hl=en&q=freepbx+voicepulse&btnG=Google+Search&meta= |
| 17:25.48 | [TK]D-Fender | Qapf: 2 words in goole gets you right to the way your should be doing it |
| 17:25.50 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 17:25.53 | HeMan | I've tried Set(CALLERID(all)=""), Set(CALLERID(name)="") and Set(CALLERID(number)="") but I still get "asterisk" <sip:asterisk@192.168.128.1> |
| 17:27.01 | Qapf | [TK]D-Fender, i am using the autoconfigure trinket that trixbox provides and my only issue seems to be my incoming calls are coming in via the anonymous context and not the proper internal one. i guess ill email them about it. thanks |
| 17:29.15 | [TK]D-Fender | HeMan: check your peer and dial statements |
| 17:29.24 | [TK]D-Fender | HeMan: it may be overriding things |
| 17:29.40 | [TK]D-Fender | Qapf: Not even going vanilla is going to make this even more painful. |
| 17:29.45 | ygguh2 | okay, I've done the make install which copied the ko files to /lib/modules/misc and not to /lib/modules/2.6.17-1.2142_FC4. this in not nice. |
| 17:31.49 | HeMan | [TK]D-Fender: I only have Dial(SIP/snom) as dial statement |
| 17:32.07 | ygguh2 | yup, did that already and ran modprobe zaptel or ztdummy, same results. KI dont get this. this is weird. |
| 17:32.38 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 17:32.47 | [TK]D-Fender | HeMan: then look at your peer setup |
| 17:33.25 | HeMan | [TK]D-Fender: and my peer is really simple, call-limit=1, busy-level=1, type=friend, secret=MySecret and host=dynamic, thats it |
| 17:34.00 | [TK]D-Fender | HeMan: pastebin a call at verbose 10 & sip debug enabled |
| 17:34.11 | *** join/#asterisk shinao1 (n=shinao1@77.220.1.216) |
| 17:35.52 | ygguh2 | its like modprobe doesnt see the modules. |
| 17:36.30 | HeMan | [TK]D-Fender: http://pastebin.com/m23c0f6be |
| 17:37.13 | HeMan | [TK]D-Fender: sorry, i only had sip debug for phone, not all sip |
| 17:37.44 | [TK]D-Fender | HeMan: its good |
| 17:38.07 | [TK]D-Fender | HeMan: works both ways. You are doing ALL, not the seperate name & number |
| 17:38.23 | [TK]D-Fender | HeMan: care to PB your sip.conf entry as well. |
| 17:38.34 | chemikk | [TK]D-Fender: hi :) |
| 17:38.52 | *** join/#asterisk lokadin (n=loki@209-161-212-129.dsl.look.ca) |
| 17:40.02 | HeMan | [TK]D-Fender: hang on, just removing my secrets |
| 17:40.21 | *** join/#asterisk mtaht4 (n=m@200.62.111.173) |
| 17:40.37 | lokadin | hey i need some software that can make outbound calls |
| 17:40.46 | Qwell | lokadin: asterisk can make outbound calls |
| 17:41.29 | lokadin | like many outbound calls in sequence from a list? as in an outbound calling centre |
| 17:42.07 | lokadin | or would i need some extension for that? |
| 17:42.09 | Qwell | well, you can use call files, or the manager interface to do that |
| 17:42.37 | lokadin | kk well i'll look into it thanks :-) |
| 17:44.14 | HeMan | [TK]D-Fender: http://pastebin.com/m67064adb |
| 17:44.48 | HeMan | [TK]D-Fender: I've tried with Set(CALLERID(name)="") and Set(CALLERID(number)="") as well |
| 17:45.23 | [TK]D-Fender | HeMan: do yourself a real favor and permanently remove everything commented out in there |
| 17:45.35 | [TK]D-Fender | HeMan: And ok, I'm stumped on this right now... |
| 17:45.57 | HeMan | [TK]D-Fender: will do |
| 17:47.15 | `Sean | Qwell do you use a card or how do you connect your IP phones? |
| 17:47.29 | chemikk | a have problem with phone: http://pastebin.ca/684318 |
| 17:48.33 | ygguh2 | okay, I finaly found a link which told me that for some unkown reason, depmod doesnt run and you need to manually run depmod. I check the modules.dep and it's date was over two weeks old. I ran depmod and then ran modprobe zaptel and ztdummy and they both loaded. |
| 17:49.52 | [TK]D-Fender | chemikk: Registration from '17 <sip:17@192.168.1.85>' <---- see that 17? change that to the USER |
| 17:50.09 | ygguh2 | dasuberdavid and Qwell thanks for you help. this was very very wierd. I run 6 differentr asterisk servers, all ES4, with out issues. but, FC4 had this issue. |
| 17:50.14 | HeMan | [TK]D-Fender: should i upload the cleaned one? |
| 17:52.50 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 17:52.51 | HeMan | [TK]D-Fender: http://pastebin.com/d6619f000, cleaned one |
| 17:53.26 | [TK]D-Fender | HeMan: Sorry, but I hit a wall on this one... you've done everything I could suggest... not sure what to advise from here |
| 17:54.12 | HeMan | ok, I'll try some more |
| 17:54.20 | HeMan | thanks for your time! |
| 17:58.19 | HeMan | I solved it! By cheating... |
| 17:58.44 | outtolunc | tattles |
| 17:58.52 | [TK]D-Fender | HeMan: Let me guess... 1 blank space for CID name? |
| 17:59.51 | HeMan | On the snom phones you could change "Number Display Style:" and if I set it to "Number" it doesn't show the "<sip...>"-part |
| 18:02.08 | ygguh2 | bye everyone |
| 18:04.44 | [TK]D-Fender | HeMan: even better |
| 18:07.47 | *** join/#asterisk barros (n=barros@189-19-24-162.dsl.telesp.net.br) |
| 18:10.05 | barros | Hi guys.. I'm having some of my calls droped by this error: Maximum retries exceeded on transmission. After a quick search, I found out that asterisk probes the other side of the call time to time to check if it is still up. If not, * drops the call. Two questions: Anyone here experienced this kind of error? How can I change the maximum retries count, or the delay between probes? |
| 18:13.39 | *** join/#asterisk elixer (i=elixer@65.207.74.18) |
| 18:16.40 | *** join/#asterisk Ebola (n=Ebola@host86-139-52-35.range86-139.btcentralplus.com) |
| 18:16.57 | Tili | i just installed asterisk-1.4 on a clean system and i am trying my agi script which worked fine with asterisk-1.2. the dtmf from iax client is not getting to script |
| 18:17.14 | Tili | in iax2 set debug i see dtmf being sent |
| 18:17.16 | Tili | received |
| 18:19.46 | *** join/#asterisk astguy (n=astguy@c-24-8-95-194.hsd1.co.comcast.net) |
| 18:21.19 | astguy | What will * do if I drop more .call files into the outgoing directory than I have channels? I will process them as channels become available (I hope) or will it error out the rest of the files once the channels are used up? |
| 18:23.31 | elixer | astguy: it will queue them up |
| 18:23.35 | viKing78 | I'm having problems hooking up a GS 4108 up to FreePBX. Can anyone lend a hand? |
| 18:23.44 | [TK]D-Fender | ~gs |
| 18:23.45 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 18:23.45 | astguy | Cool -- thanks. |
| 18:23.47 | [TK]D-Fender | ~freepbx |
| 18:23.48 | jbot | freepbx is, like, unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 18:24.04 | viKing78 | Thx anyway |
| 18:26.01 | *** join/#asterisk waltj (n=walt@216.179.31.170) |
| 18:27.05 | iCEBrkr | Hey! Grandstream phones are fine for the hobbiest. |
| 18:27.28 | *** join/#asterisk kkn088 (n=kikoun@88-136-53-187.adslgp.cegetel.net) |
| 18:27.40 | waltj | What happened to www.asteriskdocs.org? Is the server down or is these some orther reason it cannot be reached? |
| 18:27.55 | [TK]D-Fender | iCEBrkr: For a few bucks more you could get a REAL phone. |
| 18:27.59 | Strom_C | iCEBrkr: hobbiest? don't you mean hobbyist? :) |
| 18:28.19 | iCEBrkr | Strom_C: I knew something didn't look right. But someone started talking to me so I just jammed enter |
| 18:28.23 | iCEBrkr | :P |
| 18:28.30 | iCEBrkr | [TK]D-Fender: I'm jewish.. i can't stand spending a few bucks. |
| 18:28.45 | Strom_C | or is "hobbbiest" at the top of a continuum above "hobby" and "hobbier"? |
| 18:28.53 | iCEBrkr | Strom_C: I'm the hobbiest of them all! |
| 18:28.59 | Strom_C | iCEBrkr: don't be a meshugah schmuck |
| 18:29.17 | iCEBrkr | I'm not really jewish.. but ya know. I can't buy anything without talking myself into it. |
| 18:29.25 | iCEBrkr | Or I'd have a few polycoms around the house already |
| 18:29.35 | iCEBrkr | I've gone this long without it.. I don't need it |
| 18:30.05 | iCEBrkr | Heck, I only upgrade my Astrisk install because I was bored one night. Even upgraded the dialplan to use the newer methods. |
| 18:30.24 | iCEBrkr | my Sipura's + Asterisk + Voicepulse just works.. No issues. |
| 18:30.48 | Tili | wat is so wrong with asterisk 1.4 |
| 18:30.59 | Tili | it was supposed to be better |
| 18:31.10 | [TK]D-Fender | iCEBrkr: http://www.youtube.com/watch?v=fFyRohDBxfw |
| 18:31.33 | iCEBrkr | [TK]D-Fender: If only YouTube wasn't blocked here. |
| 18:31.45 | [TK]D-Fender | iCEBrkr: save it then. |
| 18:31.51 | iCEBrkr | k |
| 18:32.31 | chemikk | [TK]D-Fender: i fix my problem, change username to 17 and inside brackets [17] too |
| 18:33.33 | [TK]D-Fender | chemikk: should have just changed the 17 to the user name you were using up top, but same thing in the end, as long as it MATCHES |
| 18:35.03 | MooingLemur | are announcing holdtime and announcing queue position both tied to the same setting? |
| 18:35.55 | [TK]D-Fender | MooingLemur: the same FREQUENCY yes, but they are seperate |
| 18:36.49 | MooingLemur | which one toggles announce position? |
| 18:39.32 | [TK]D-Fender | MooingLemur: Go read the sample queues.conf * comes with. Its all laid out in there. |
| 18:40.19 | codec | can someone tell me what this means and how i can fix it? (appears as soon as i use Festival()) |
| 18:40.23 | codec | Sep 6 20:38:48 WARNING[17077]: utils.c:609 tvfix: warning too large timestamp 1836086386.859467873 |
| 18:42.31 | MooingLemur | [TK]D-Fender: I only see announce-holdtime as a toggle. |
| 18:42.57 | MooingLemur | seems I can't turn off position, just holdtime. |
| 18:43.24 | Corydon76-vcch | codec: it means the timestamp is invalid |
| 18:45.06 | MooingLemur | just wondering if you agree with my evaluation :) |
| 18:45.58 | putnopvut | MooingLemur: the position announcement is set with the announce-frequency setting |
| 18:46.10 | codec | Corydon76-vcch: which one? :p |
| 18:46.19 | MooingLemur | so I can't disable position announcements but keep estimated holdtime announcements |
| 18:46.26 | putnopvut | Let me check real quick... |
| 18:47.06 | putnopvut | MooingLemur: that would be correct, since the holdtime is announced along with the position. |
| 18:47.51 | MooingLemur | I suppose I could point the wavs to empty sounds, but I'll get the numeral announcement with the position |
| 18:48.02 | MooingLemur | thanks for checking |
| 18:48.10 | MooingLemur | I'll just leave it in |
| 18:48.45 | [TK]D-Fender | MooingLemur: tried not setting the recording variables? |
| 18:49.08 | MooingLemur | I haven't set recording at all |
| 18:49.23 | [TK]D-Fender | MooingLemur: hrm. |
| 18:52.02 | jwh | hm, has anyone developed a dynamic routing extension for asterisk yet? perhaps in a bgp/rip style format? |
| 18:52.13 | jwh | ie; to share prefixes with IAX peers or such |
| 18:52.58 | [TK]D-Fender | jwh: ....? |
| 18:53.08 | jwh | hm? |
| 18:53.40 | neverblue | Dr-Linux, you around ? |
| 18:55.24 | jwh | [TK]D-Fender: what was wrong with my question? |
| 18:55.39 | lirakis | jwh: .. you mean.. a routing engine.. where you can "dip" for routes or are you talking about balancing between internet connections.... because bgp and rip have nothing to do with routing phone calls |
| 18:56.05 | jwh | a routing engine yes |
| 18:56.14 | jwh | lirakis: I know, I was trying to think of an analogy |
| 18:56.15 | lirakis | jwh: i assume [TK]D-Fender said "...?" as in.. wtf are you talking about |
| 18:56.57 | jwh | as dialling prefixes and routes are the same in principal, both for phone and ip |
| 18:57.02 | putnopvut | jwh: the closest thing to that I know of is DUNDI. It's a decentralized way of discovering extensions. |
| 18:57.06 | jwh | ooh |
| 18:57.09 | lirakis | jwh: asterisk is class 5 pretty much... look into openSER if you want routing |
| 18:57.18 | jwh | thanks both |
| 18:57.34 | *** join/#asterisk shinao1 (n=shinao1@77.220.1.216) |
| 18:58.46 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 18:59.05 | russellb | DUNDi is awesome :) |
| 18:59.44 | jwh | it looks perfect |
| 18:59.46 | jwh | thanks guys |
| 19:01.01 | jwh | realtime integration would be even better, if it can dump current routing information to a database :D |
| 19:02.03 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 19:02.03 | *** mode/#asterisk [+o anthm] by ChanServ |
| 19:03.48 | *** join/#asterisk Lucky7 (n=Adam@207.200.28.175) |
| 19:04.14 | Lucky7 | hmm. |
| 19:04.28 | Lucky7 | Anyway to reset a Polycom to defaults before the boot? |
| 19:04.28 | *** join/#asterisk saftsack (n=saftsack@pD9E06623.dip.t-dialin.net) |
| 19:04.43 | Lucky7 | *468 doesn't seem to do the trick |
| 19:05.11 | mcab | Lucky7: what model? |
| 19:05.31 | Lucky7 | 330 |
| 19:05.33 | Lucky7 | IP-330 |
| 19:05.45 | Lucky7 | took D-Fender's advise and changed our order from 301's to 330's |
| 19:06.48 | mcab | Lucky7: try 1,3,5,7 |
| 19:07.07 | *** join/#asterisk kkn088 (n=kikoun@88-136-53-187.adslgp.cegetel.net) |
| 19:07.11 | Lucky7 | on boot? |
| 19:07.19 | mcab | yup |
| 19:07.30 | mcab | should work in the application too |
| 19:07.33 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 19:07.43 | Lucky7 | with or without * |
| 19:08.13 | mcab | just the 1,3,5,7 keys |
| 19:08.14 | mcab | no * |
| 19:08.33 | mcab | hold all 4 down at the same time for ~5 seconds |
| 19:09.16 | Lucky7 | hm |
| 19:09.52 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 19:09.57 | Lucky7 | hm |
| 19:10.01 | Lucky7 | doesn't seem to do anything |
| 19:10.11 | Lucky7 | now actually the phone doesn't do anytihng on boot |
| 19:10.13 | mcab | I just tested it on my 330 :-) |
| 19:10.19 | Lucky7 | just red light on the top blinks |
| 19:10.25 | mcab | what's on the display? |
| 19:10.36 | Lucky7 | lemme change my wiring around so it doesn't bump me off every few seconds |
| 19:10.48 | *** part/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 19:10.55 | Lucky7 | test |
| 19:10.56 | Lucky7 | k |
| 19:11.36 | Lucky7 | yea |
| 19:11.39 | Lucky7 | i plug it in |
| 19:11.48 | Lucky7 | i get a line 1 blink, and a line 2 blink |
| 19:11.51 | *** join/#asterisk shinao1 (n=shinao1@77.220.1.216) |
| 19:11.55 | Lucky7 | and now just a constant red blink at the top |
| 19:12.08 | mcab | nothing on the LCD screen? |
| 19:12.19 | Lucky7 | polycom appears for about 1/2 a second |
| 19:12.47 | mcab | then you should see a "x seconds until boot" screen |
| 19:12.53 | Lucky7 | normally yea |
| 19:12.56 | Lucky7 | not on this unit |
| 19:13.23 | *** join/#asterisk inv_arp[work] (n=junya@c-76-26-38-113.hsd1.fl.comcast.net) |
| 19:13.58 | mcab | Lucky7: try holding down volume up, volume down and the '0' key for a few seconds, then try pressing the volume up button repeatedly |
| 19:14.09 | mcab | that should let you try setting the contrast |
| 19:14.25 | *** join/#asterisk shinao1 (n=shinao1@77.220.1.216) |
| 19:14.34 | mcab | otherwise, it sounds like a hardware issue - I'd talk to your reseller... |
| 19:15.09 | Lucky7 | c'ya |
| 19:15.16 | Lucky7 | yea, looks like a dead phone-o |
| 19:16.34 | *** join/#asterisk michael-i (n=michael-@Wa5fa.w.pppool.de) |
| 19:18.01 | *** join/#asterisk Ebola (n=Ebola@host86-143-7-120.range86-143.btcentralplus.com) |
| 19:20.17 | *** join/#asterisk cryc0s (n=crycos@72.54.46.18) |
| 19:20.55 | [TK]D-Fender | Lucky7: IP 320 and 330: Volume-, Volume+, Hold, and Hands-free <----------------- |
| 19:21.10 | [TK]D-Fender | for reboot |
| 19:21.22 | [TK]D-Fender | Lucky7: IP 320, 330, and 430: 1, 3, 5, and 7 dial pad keys |
| 19:21.27 | [TK]D-Fender | for factory defaults |
| 19:22.23 | Yourname` | russellb: You free for little while? |
| 19:22.35 | elixer | Lucky7: 8, 6, 7, 5, 3, 0, 9 for tommy two-tone |
| 19:22.51 | [TK]D-Fender | for a good time call.... |
| 19:22.55 | Qwell | timmy |
| 19:23.00 | russellb | Yourname`: not really |
| 19:23.11 | [TK]D-Fender | russellb: affordable? ;) |
| 19:23.18 | Yourname` | russellb: When can I bug you for a bit? :D |
| 19:23.36 | russellb | heh, i don't commit to any time that i will be here helping. |
| 19:23.46 | russellb | i try when i can, but i stay pretty busy on the development side. |
| 19:23.50 | Yourname` | Let me spew |
| 19:23.57 | Yourname` | Thought so.. hence I asked. :D |
| 19:24.10 | Yourname` | Anyway, let me type out all my questions. |
| 19:24.16 | russellb | thanks for asking and not spamming me with private messages |
| 19:24.26 | Yourname` | (You're welcome.) |
| 19:24.39 | russellb | ::) |
| 19:24.44 | russellb | s/::/:/ |
| 19:25.36 | Yourname` | 1) If a 100 numbers were called rapidly, and 50 were disconnected. Asterisk rcvs the SIT back from provider, and once its done DIALING, spews out all the "call failed to go through, reason 0" messages. How can I catch those and know which number is that for? |
| 19:26.49 | Yourname` | 2) If I want to log particular events in an IVR, how can I do so? For example, a call goes out to a person. A message is played to press1 or 2. I'd like to log the phone number of the person, and what he pressed. And then, another msg is played, and again DTMF is needed.. I'd like to log that, etc. |
| 19:27.19 | Yourname` | ...and the rest are based on answers to these questions, lol |
| 19:29.04 | [TK]D-Fender | Yourname`: You have HANGUPCAUSE and DIALSTATUS you can process and in your IVR jsut log it using dialplan apps however you want |
| 19:30.15 | Yourname` | [TK]D-Fender: Going to read HANGUPCAUSE and DIALSTATUS. But what application would you use to log these events? |
| 19:31.08 | michael-i | Does anyone know any "gotcha"s with IAX channels and timing? I am getting very robotic / garbled audio on IAX channels which seems to be related to timing issues. (using 1.4.11 on FreeBSD 6.2) |
| 19:35.05 | *** join/#asterisk remi____ (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 19:35.43 | elixer | timmy... tommy... same diff |
| 19:41.05 | *** join/#asterisk hmmhesays (n=Neg@24-119-176-74.cpe.cableone.net) |
| 19:41.07 | hmmhesays | hey folks |
| 19:42.16 | *** join/#asterisk Lucky7 (n=Adam@207.200.28.175) |
| 19:42.26 | [TK]D-Fender | Yourname`: these are dialplan vars set after a DIAL. its all just DIALPLAN. |
| 19:42.48 | Yourname` | ok |
| 19:45.19 | hmmhesays | i'm having a hell of a time with this sangoma a200, it is randomly going static then disconnecting |
| 19:45.25 | hmmhesays | I don't know where to start troubleshooting |
| 19:48.00 | *** join/#asterisk Ebola (n=Ebola@host86-143-7-120.range86-143.btcentralplus.com) |
| 19:49.30 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 19:50.32 | *** join/#asterisk kkn088 (n=kikoun@88-136-53-187.adslgp.cegetel.net) |
| 19:51.32 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:53.43 | *** join/#asterisk davevg-btwtech (n=davevg@nj-67-76-177-147.sta.embarqhsd.net) |
| 19:55.53 | Lucky7 | Hm. |
| 19:56.05 | Lucky7 | this is retarded, i cant get this polycom 330 to register to asterisk |
| 19:56.29 | Lucky7 | http://rafb.net/p/ZM56Md61.html |
| 19:56.50 | Lucky7 | thats the Mac address config |
| 19:56.59 | Lucky7 | the mac address then pulls x141.cnf http://rafb.net/p/hqqRwt82.html |
| 19:57.27 | Lucky7 | then server.cfg http://rafb.net/p/vDRc7d93.html |
| 19:57.28 | [TK]D-Fender | Lucky7: careful as to which settings get overriden from one file to another. I personally only use 2 files. 1 master (sip.cfg) and a single phone entry (phoneXXX.cfg) |
| 19:57.42 | hmmhesays | can anyone give me a starting point for said problem? |
| 19:57.46 | _ShrikE | according to the paste it is pulling x140.cfg no x141.cnf |
| 19:57.56 | Lucky7 | 140, sorry |
| 19:58.02 | Lucky7 | fatfinger syndrome |
| 19:58.06 | [TK]D-Fender | cnf != cfg |
| 19:58.17 | _ShrikE | get rid of phone1.cfg also. |
| 19:59.07 | Lucky7 | done |
| 19:59.14 | Lucky7 | D-Fender |
| 19:59.15 | *** join/#asterisk bkruse_home (n=root@69.73.127.92) |
| 19:59.17 | wunderkin | no, you shouldn't remove phone1.cfg if you do it the proper way and only override the defaults |
| 19:59.19 | Lucky7 | Mind nopasting your two files |
| 19:59.35 | Lucky7 | or a default file that you work off of? |
| 20:01.21 | [TK]D-Fender | Lucky7: make a template out of the stock phone1.cfg and then make phone specific copies afterwards. then in sip.cfg add your server parms and everything else global. |
| 20:01.49 | [TK]D-Fender | Lucky7: I advise against doing them completely from scratch or in 10-layered override-style |
| 20:02.02 | *** join/#asterisk kiscokid (n=ron@208.106.35.66) |
| 20:03.18 | [TK]D-Fender | hmmhesays: pastebin your zapata.conf |
| 20:03.27 | kiscokid | Anyone understand how followme.conf works? There don't seem to be any per user entries in the sample file |
| 20:03.29 | *** join/#asterisk fatgoose (n=fg@206-248-175-211.dsl.teksavvy.com) |
| 20:05.38 | Lucky7 | reg.1.address="10.0.32.4" |
| 20:05.53 | Lucky7 | whats the difference between address and server.1.address? |
| 20:06.27 | kiscokid | Lucky on a polycom reg.1.address is the extension |
| 20:06.44 | kiscokid | go figure |
| 20:06.49 | Lucky7 | ah |
| 20:07.17 | [TK]D-Fender | correct |
| 20:08.51 | Lucky7 | http://rafb.net/p/Yxm1gQ37.html |
| 20:09.29 | Lucky7 | and the only place in sip.cfg is voIpProt.server.1.address to 10.0.32.4 |
| 20:10.06 | Lucky7 | hm |
| 20:10.11 | Lucky7 | Nope, didn't seem to do it |
| 20:11.13 | Corydon76-vcch | Could someone please explain overlapdial to me? |
| 20:12.14 | fatgoose | anyone known a SMS aggregator that can provide origination (shortcode)/termination in canada? |
| 20:13.42 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 20:14.56 | pkunkra | http://www.ubergizmo.com/15/archives/2006/07/skype_phone_converter.html |
| 20:15.03 | pkunkra | hmmm.... i have idea..... |
| 20:15.17 | pkunkra | rig a askterisk interface hack? |
| 20:15.36 | Strom_M | pkunkra: it'll sound like shit |
| 20:15.44 | pkunkra | plug it into an FXO... |
| 20:15.55 | pkunkra | really? |
| 20:16.08 | pkunkra | even if i kept the whole thing ulaw? |
| 20:16.13 | Strom_M | yes |
| 20:16.32 | [TK]D-Fender | "but don't want to plonk down a heft investment in a VoIP phone" <--- for $120!!! lol. RETARDS |
| 20:16.33 | *** join/#asterisk kkn088 (n=kikoun@88-136-53-187.adslgp.cegetel.net) |
| 20:16.44 | Strom_M | the increased latency and the unnecessary A/D conversion combined with skype's already poor quality == recipe for disaster |
| 20:16.54 | pkunkra | haha |
| 20:16.57 | pkunkra | alright. |
| 20:17.03 | pkunkra | interesting idea while it lasted. |
| 20:17.07 | [TK]D-Fender | 30 pounds... you can get a normal ata for that... |
| 20:17.37 | pkunkra | yeah, but the ata doesn't speak skype. |
| 20:17.44 | *** join/#asterisk guillote_GNU (n=bancaria@host185.190-136-203.telecom.net.ar) |
| 20:17.47 | cheGGo | anybody used proxy support for normal sip uri calls? |
| 20:17.54 | pkunkra | was thinking of rigging an incoming skype line |
| 20:17.56 | Lucky7 | D-Fender: I'm really confused, I've got 0004f216bea1.cfg, which loads x140.cnf (my copy of the phone1.cnf template, with extension edits) |
| 20:18.08 | Strom_M | pkunkra: skype, and anything to do with skype, is utter crap./ |
| 20:18.10 | Lucky7 | sory, .cfg |
| 20:18.21 | cheGGo | so that a call to SIP/bla@domain.tld is been routed via my proxy |
| 20:18.22 | Lucky7 | and sip.cfg, with address / proxy edits |
| 20:18.25 | cheGGo | nope |
| 20:18.35 | Lucky7 | but its not actaully registering to my system |
| 20:18.56 | cheGGo | i had set up the proxy in my sip.conf |
| 20:18.59 | Lucky7 | i can dial 7777, and i get a no service error, and i see "SIP/10.0.32.4-08af72c8" as a call |
| 20:19.10 | cheGGo | indeed |
| 20:19.22 | cheGGo | but not through a proxy |
| 20:19.28 | pkunkra | strom_m: true. but there's a large audience already using it. |
| 20:19.33 | cheGGo | asterisk - proxy - sip peer |
| 20:19.50 | cheGGo | i had set up the options in sip.conf |
| 20:19.53 | cheGGo | but didnt work |
| 20:20.07 | cheGGo | in the general section |
| 20:20.32 | *** join/#asterisk yannj_fr (n=yannj_fr@vpn.intelunix.fr) |
| 20:20.37 | cheGGo | anybody opened a branch with "ob_proxy" support |
| 20:20.51 | cheGGo | that works, but set a via header with 127.0.0.1 ;) |
| 20:21.19 | cheGGo | but i think the normal trunk of asterisk should support this too |
| 20:21.53 | Strom_M | pkunkra: so? |
| 20:22.03 | cheGGo | Lucky7? |
| 20:22.19 | Kwakwa | A lot of people use windows too :) |
| 20:22.29 | pkunkra | strom_m: never mind. :-P |
| 20:22.58 | pkunkra | i think i'll be butting heads with an ox if i try to argue it. ;-) |
| 20:23.20 | cheGGo | so nobody used asterisk with a proxy as NON peer? |
| 20:23.23 | Kwakwa | U don't know until u try pkunkra |
| 20:24.26 | pkunkra | kwakwa: perhaps. but i don't feel like trying right now. too damn tired. :-) |
| 20:24.57 | *** join/#asterisk the_Goat_ (n=chatzill@h-67-103-23-130.phlapafg.covad.net) |
| 20:25.06 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 20:25.35 | the_Goat_ | i have noticed that when i park a call, i go to the other phone and dial the specified part extensions, but when the person on the other end of the line is talking i can't hear them, but i can talk to them |
| 20:26.09 | the_Goat_ | any ideas? |
| 20:26.39 | deeperror | i'm getting an __zt_exception that seems to be in an endless loop printed to the CLI which fills my frame buffer quickly. Is there anyway to log CLI output to a file to capture what was going on prior to the exception occuring? |
| 20:33.56 | Lucky7 | I think Teliax can pass more then default CID info but I'm not sure on the details. |
| 20:33.56 | Lucky7 | * Bananaskin has quit IRC |
| 20:33.56 | Lucky7 | * cryc0s has joined #freepbx |
| 20:33.56 | Lucky7 | * jzakhar has quit IRC |
| 20:33.56 | Lucky7 | <Lucky7> hmm |
| 20:34.00 | Lucky7 | ACK |
| 20:34.03 | Lucky7 | Sorry... |
| 20:34.33 | Lucky7 | I'm trying to get a phone to properly register to my Asterisk System, and i seem to be missing something |
| 20:35.12 | Lucky7 | http://rafb.net/p/lYYOvk78.html // phones macAddress.cfg |
| 20:35.14 | Kwakwa | deeperror, have you enabled log files? |
| 20:35.30 | deeperror | yea, i'm going over messages now it's huge |
| 20:35.38 | Lucky7 | http://rafb.net/p/1yZRh735.html // x140.cfg |
| 20:35.55 | deeperror | but its the same error in there that is in CLI...but i don't know what was going on prior to this error occuring |
| 20:35.59 | Kwakwa | have you set the debug level to max as well? |
| 20:36.04 | Lucky7 | http://rafb.net/p/k1QUXF11.html // sip.cfg |
| 20:36.13 | deeperror | debug level where on the cli? |
| 20:36.13 | Kwakwa | `core set debug 10` or summat I think |
| 20:36.31 | deeperror | yea but this exception throws 100 errors a second |
| 20:36.33 | Lucky7 | http://rafb.net/p/xkt5bE56.html // log of what the system does when the phone with those configs tries to make a call |
| 20:36.43 | deeperror | quickly filling up the screen with the same error until i shut down |
| 20:36.52 | Kwakwa | yeah, I know what you mean |
| 20:36.54 | *** join/#asterisk w3pog (n=pgrace@66.92.234.76) |
| 20:37.06 | Kwakwa | have you searched bugs.digium.com for the error? to see if anyone else has the issue? |
| 20:37.15 | w3pog | does anyone happen to know in what structure the expected/received passwords are in chan_sip.c? |
| 20:37.21 | deeperror | chan_zap.c: We're Zap/1-2, not Zap/1-1<ZOMBIE> |
| 20:37.23 | w3pog | I'm trying to debug why my registers are failing with "wrong password" |
| 20:37.50 | deeperror | i think it has something to do with a 3-way call occuring and the first callee hanging up before the second callee can come into the conversation then things go crazy nuts |
| 20:38.04 | Kwakwa | What version are you running? |
| 20:38.09 | deeperror | 1.2.24 |
| 20:38.10 | pkunkra | i just thought of some funny april fools pranks with tt-monkeys. |
| 20:38.24 | Qwell | pkunkra: like playing it randomly during calls? |
| 20:39.08 | Kwakwa | Unless you're running the most recent version u won't get much help deeperror, most of problems in older versions of * may be fixed in 1.4* |
| 20:39.30 | pkunkra | qwell: nah, i was thinking of have the pbx call random friends of mine and set the callerid to another appropriate mutual friend. |
| 20:39.45 | pkunkra | then play tt-monkeys after they stop talking. |
| 20:39.56 | Kwakwa | pkunkra, u've got a long time to wait before u can use it as a legitimate april fool |
| 20:40.02 | pkunkra | right. |
| 20:40.11 | deeperror | i move to 1.4 and get different issues with sip packets and authentication |
| 20:40.12 | Kwakwa | and you can only do it before 12pm, so u gotta make sure these ppl will answer :) |
| 20:40.18 | pkunkra | i kinda think its silly to do it any other day. |
| 20:40.24 | Kwakwa | haha |
| 20:40.47 | Kwakwa | I think russellb has beaten you to it by the looks of things tho |
| 20:40.48 | Qwell | everybody expects things like that on April 1st |
| 20:40.55 | Qwell | but who's gonna expect it on Sept 6th? |
| 20:40.59 | Kwakwa | :) |
| 20:41.01 | pkunkra | before 12pm? people are more likely to answer in the morning? |
| 20:41.14 | Kwakwa | no, because u have to do it before the afternoon |
| 20:41.27 | Kwakwa | its some april 1st law |
| 20:41.46 | pkunkra | what did russellb do? :-) |
| 20:41.47 | pkunkra | oh. |
| 20:41.48 | Kwakwa | apparently if you do it after that time, you're the fool |
| 20:41.54 | pkunkra | i didn't know about that. |
| 20:42.51 | Kwakwa | I dunno if it applies to the US, it is in the UK, Canada n stuff |
| 20:43.17 | pkunkra | nothing on google. are you sure that's a law? |
| 20:43.24 | pkunkra | or are you pulling one over my head too? |
| 20:43.31 | Qwell | doing it after noon makes it no less funny |
| 20:43.35 | Kwakwa | Its not a real law, u won't have to do community service or anything |
| 20:43.38 | Kwakwa | http://www.bigdates.com/holidays/aprilfoolsday.asp |
| 20:45.13 | pkunkra | a friend of mine actually did research with monkeys. |
| 20:45.41 | Kwakwa | On April fools day a year ago my boss's wife thought it would be funny to tell me I was about to be fired coz of something I did so I went into the bosses room n told him to fuck off. |
| 20:45.44 | pkunkra | she'd be a perfect candidate to set the callerid to. |
| 20:45.57 | pkunkra | kwakwa: ouch. |
| 20:46.01 | pkunkra | were you fired? |
| 20:46.10 | Kwakwa | Turns out it was an april fool, fortunately he saw the funny side of it |
| 20:46.21 | Kwakwa | Felt good telling him to fuck off tho :) |
| 20:46.27 | pkunkra | haha |
| 20:46.30 | Kwakwa | He was lucky I didn't need a crap |
| 20:48.33 | the_Goat_ | i have noticed that when i park a call, i go to the other phone and dial the specified part extensions, but when the person on the other end of the line is talking i can't hear them, but i can talk to them |
| 20:48.34 | the_Goat_ | any ideas? |
| 20:50.57 | pkunkra | try speaking into your phone louder? |
| 20:51.20 | pkunkra | :-) |
| 20:51.42 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 20:52.00 | *** part/#asterisk w3pog (n=pgrace@66.92.234.76) |
| 20:52.06 | pkunkra | check the your phones for silence suppression |
| 20:52.44 | pkunkra | if you can't hear them, then is probably your handset. |
| 20:52.52 | pkunkra | google "comfort noise generation" |
| 20:57.13 | *** join/#asterisk w3pog (n=pgrace@66.92.234.76) |
| 20:59.00 | Corydon76-vcch | w3pog: you might want to start with a "sip set debug" |
| 20:59.17 | Corydon76-vcch | That will give you more information on the SIP dialog |
| 21:00.25 | w3pog | yeah |
| 21:00.34 | w3pog | I've been watching the debug on it, and have a few sip packet traces |
| 21:00.50 | w3pog | calls fail with a fast busy, and registers get bad auth |
| 21:00.51 | *** join/#asterisk saftsack (n=saftsack@pD9E06623.dip.t-dialin.net) |
| 21:00.58 | w3pog | the weirdest thing is, yesterday the phone was working fine |
| 21:01.02 | w3pog | it just stopped, out of nowhere. |
| 21:01.11 | w3pog | we had actually got it working right when we swapped the ip address |
| 21:01.25 | w3pog | but then "something" happened. I can't really identify that "something" |
| 21:01.32 | w3pog | oh |
| 21:01.52 | km- | heh |
| 21:02.28 | km- | Corydon76-dig: does bkw still hang out anymore or is he totally devoted to freeswitch or whatever now |
| 21:02.47 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:02.50 | *** join/#asterisk riddlebox (n=victoria@75-132-225-75.dhcp.stls.mo.charter.com) |
| 21:03.05 | riddlebox | how do I find out how my line is being seen by asterisk? |
| 21:03.22 | syzygyBSD | riddlebox: what do you mean by line? |
| 21:03.23 | riddlebox | like, when it rings in, what it is named? |
| 21:03.23 | [TK]D-Fender | riddlebox, .... huh? |
| 21:03.33 | riddlebox | [TK]D-Fender, I know I worded that horribly |
| 21:03.35 | fujin_ | riddlebox, depends on how you configure it |
| 21:03.37 | [TK]D-Fender | riddlebox, .... HUH?! |
| 21:03.38 | Corydon76-vcch | km-: he's in here from time to time, but we catch him trolling quite often |
| 21:03.43 | km- | Corydon76-dig: do you think if I pastebin'd my sip trace you might be able to help stare at it? |
| 21:03.43 | fujin_ | you can configure your inbound calls to present on a specific context |
| 21:03.50 | km- | Corydon76-dig: I'm sure stuck on what's going on |
| 21:03.50 | Kwakwa | When it guys busy on our production server we sometimes get one way audio for the callee, where the callee can hear the agent but the agent can't hear the callee. When its not busy its fine. I'm thinking iaxthreadcount / iaxmaxthreadcount might have something to do with it but they're at the default atm 10/0. I upped it to 30/100 but still hav the issues. I read that it can be as high as 200/1000. Is that reccommended? |
| 21:04.09 | Corydon76-vcch | km-: I can look, but I'm not the best at diagnosing sip problems |
| 21:04.32 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 21:04.34 | riddlebox | [TK]D-Fender, I have a pots line, which I configured a Quintum box to convert it to sip, but I need to know how asterisk is seeing it come in so when it rings I can have it ring my phones |
| 21:04.49 | km- | Corydon76-dig: that sure makes two of us |
| 21:05.01 | [TK]D-Fender | riddlebox, enable SIP debug and see what * is getting |
| 21:05.12 | km- | http://pastebin.com/m23f6fb20 |
| 21:05.15 | syzygyBSD | riddlebox: look at how to configure a dialplan |
| 21:05.21 | fujin_ | riddlebox, depends on how you set up the peer |
| 21:05.36 | km- | Corydon76-dig: that url has the registers |
| 21:05.38 | *** part/#asterisk javar (n=javar@69.79.134.24) |
| 21:05.48 | km- | Corydon76-dig: at first I was suspecting it was some weird nat issue |
| 21:05.54 | km- | but I'm starting to think it must be something else. |
| 21:05.55 | mvanbaak | bye all |
| 21:05.58 | SA007 | why can't i get audio when i dial out? very frustrating |
| 21:06.07 | fujin_ | SA007, you're doing it wrong |
| 21:06.09 | riddlebox | [TK]D-Fender, would it be under User-Agent? |
| 21:06.14 | km- | Corydon76-dig: starting to wonder if I wanna upgrade to the latest 1.2.x |
| 21:06.18 | mvanbaak | SA007: still fighting with budgetphone.nl ? |
| 21:06.26 | SA007 | fujin_: probably, but i can't find what i'm doing wrong |
| 21:06.39 | km- | Corydon76-dig: I'm also going to get the remote user to issue a factory reset on his phone |
| 21:06.42 | SA007 | mvanbaak: fixed the loop/busy error's, bu i don't get audio |
| 21:06.46 | [TK]D-Fender | riddlebox, if you want to see what comes in, SIP debug will tell you. |
| 21:06.48 | km- | see if for some reason the password got corrupted. |
| 21:06.57 | mvanbaak | SA007: cant say I didn't warn you |
| 21:07.19 | Corydon76-vcch | km-: I'd seriously consider upgrading to 1.4, if I were you |
| 21:07.31 | SA007 | mvanbaak: i haven't been able to find a good alternative, most are far more expensive or don't give a local numer (or any number) |
| 21:07.45 | mvanbaak | SA007: http://www.speakup.nl http://www.12connect.com |
| 21:07.52 | km- | Corydon76-dig: how production-ready is 1.4? |
| 21:08.01 | km- | Corydon76-dig: and is there a changelog on configs for 1.2 to 1.4? |
| 21:08.02 | fujin_ | lol |
| 21:08.03 | mvanbaak | km-: it's production ready |
| 21:08.14 | Corydon76-vcch | km-: UPGRADE.txt |
| 21:08.14 | mvanbaak | km-: check the UPGRADE.txt |
| 21:08.17 | km- | ok cool. |
| 21:08.43 | fujin_ | I probably wouldn't run 1.2 unless there was a specific business decision to do so. |
| 21:08.49 | fujin_ | 1.4 has been running fine in production here |
| 21:08.50 | SA007 | mvanbaak: looked at both, speakup is way to expensive and 12connect only does prepaid |
| 21:09.14 | km- | we have a ton of telephony here running through asterisk and I didn't want to upset anything with a major upgrade |
| 21:09.16 | mvanbaak | SA007: 12connect does postpaid as well |
| 21:09.16 | syzygyBSD | ya, all my new installs will be 1.4, but I won't upgrade from 1.2 until I reinstall |
| 21:09.18 | mvanbaak | SA007: call them |
| 21:09.32 | mvanbaak | SA007: tell them 'Michiel van Baak van Covide' sent you |
| 21:09.43 | km- | we also have all custom prompts |
| 21:09.51 | km- | the people at our office think alison sounds like a porn star :) |
| 21:09.53 | SA007 | mvanbaak: but still, i don't think this is a budgetphone issue, but more of my asterisk setup |
| 21:10.55 | mvanbaak | SA007: well, I had massive trouble to connect with them and even when I got a call going thru it was very unstable. most calls would fail or 1-way audio (if not reporting busy/loop) |
| 21:11.01 | SA007 | (like, switching provider doesn't solve the problem) |
| 21:11.06 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583591.dsl.bell.ca) |
| 21:11.20 | mvanbaak | SA007: both speakup and 12connect do IAX2 |
| 21:11.25 | fujin_ | get audio running locally |
| 21:11.29 | mvanbaak | works way better for an ITSP if you ask me |
| 21:11.31 | fujin_ | before you bring someone elses issues into it |
| 21:12.09 | SA007 | mvanbaak: jeah, but i don't have a phone right now, but i have a budgetphone account... |
| 21:12.28 | mvanbaak | SA007: did you call their support ? |
| 21:12.43 | mvanbaak | or mailed support ? |
| 21:13.04 | SA007 | no, looked at the website |
| 21:13.14 | mvanbaak | call/mail their support |
| 21:13.57 | mvanbaak | ok, I'm really off now |
| 21:14.31 | SA007 | speakup is just to expensive, but 12connect can be a good option if they really do postpaid, because i really don't want to continuesly add mony to the account |
| 21:15.26 | mvanbaak | SA007: it depends on the quality you want. speakup is totally redundant, 12connect is not |
| 21:16.09 | mvanbaak | SA007: speakup will give you 2 IAX2 switches and 2 SIP switches so you can do failover for calls. 12connect only has 1 registry/proxy box |
| 21:16.26 | fujin_ | ~cheap |
| 21:16.27 | jbot | methinks cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 21:16.35 | fujin_ | my two cents |
| 21:16.35 | SA007 | jeah, but my business isn't very profitable at the moment and the colocated server takes up about the entire income at the moment :P |
| 21:16.38 | fujin_ | ^5 jbot |
| 21:17.05 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 21:17.05 | mvanbaak | holy fuck, you only income is 70 euro/month ? |
| 21:17.13 | fujin_ | that's pretty crap |
| 21:17.25 | SA007 | for my company at the moment, yes, really |
| 21:17.32 | Kwakwa | U gotta start somewhere |
| 21:17.46 | [TK]D-Fender | Nowhere is somewhere too! ;) |
| 21:17.49 | SA007 | indeed, started the business like 4 monts ago now |
| 21:18.00 | mvanbaak | SA007: quit! even social security is like 15 times that income |
| 21:18.10 | Qwell | [TK]D-Fender: how was the openmoko demo thing? or was it today? |
| 21:18.17 | Kwakwa | [TK]D-Fender, nowhere is the lack of something :) |
| 21:18.22 | mvanbaak | openmoko > * |
| 21:18.24 | SA007 | mvanbaak: i'm not even looking for clients yet :P |
| 21:18.24 | Qwell | nowhere is the lack of somewhere |
| 21:18.31 | Kwakwa | that too |
| 21:18.39 | fujin_ | openmoko? |
| 21:18.42 | Qwell | nowhere is a place, thus it is somewhere |
| 21:18.45 | Qwell | thus, it cannot be nowhere |
| 21:18.54 | Qwell | fujin_: openmoko.org |
| 21:18.58 | fujin_ | o0o |
| 21:19.03 | fujin_ | mobile OS? |
| 21:19.04 | [TK]D-Fender | Qwell, Sunday evening. |
| 21:19.22 | Qwell | mvanbaak: if [ true ]; then echo false; fi ? |
| 21:19.26 | [TK]D-Fender | Qwell, and I added myself to the "want to come" list and haven't gotten confirmation. |
| 21:19.27 | mvanbaak | lol |
| 21:19.28 | fujin_ | what does it run on? |
| 21:19.37 | SA007 | i started the business, but i want to get stuff like the phones/website/email/etc working before i start looking for clients, which was planned for 2 monts ago but got delayed... |
| 21:19.38 | Qwell | fujin_: right now, only the Neo 1973 |
| 21:19.40 | mvanbaak | runkit_redefine |
| 21:19.43 | mvanbaak | meh |
| 21:19.46 | [TK]D-Fender | fujin_, Os *and* open hardware phone |
| 21:19.59 | [TK]D-Fender | Qwell, no, they got it working on a Palm or two ;) |
| 21:20.05 | [TK]D-Fender | (somewhat) |
| 21:20.07 | Qwell | well, yeah |
| 21:20.19 | Qwell | and really, people run it in emulators too |
| 21:20.29 | Kwakwa | Qwell: I was working from "darkness is the lack of light", light is something, darkness is the lack of it. Nowhere is the lack of something, if something was in nowhere it would be somewhere? |
| 21:21.13 | Qwell | if there was no light, then light wouldn't exist, therefore, darkness wouldn't exist |
| 21:21.15 | mvanbaak | SA007: well, if this is going to be for a business I have 1 advice: DONT use budgetphone.nl |
| 21:21.31 | Qwell | Kwakwa: quit while you're ahead :P |
| 21:21.37 | [TK]D-Fender | I doubt, therefor I may be. |
| 21:21.39 | SA007 | mvanbaak: it's better than having no phone at all |
| 21:21.44 | Kwakwa | :p |
| 21:21.54 | mvanbaak | SA007: you have it working now ? |
| 21:21.57 | fujin_ | hrmp |
| 21:22.00 | SA007 | which is the case at the moment |
| 21:22.03 | fujin_ | I wonder if they'll make it work on my htc tytn. |
| 21:22.13 | SA007 | i can receive call's but get no audio when calling |
| 21:22.24 | mvanbaak | SA007: non-woring budgetphone.nl is the same as no phone |
| 21:23.40 | SA007 | mvanbaak: true, but its the first time i'm doing anything with * and i'm sure it's a problem on my end which isn't solved by switching providers |
| 21:24.25 | mvanbaak | SA007: ok |
| 21:24.58 | mvanbaak | SA007: I had multiple working trunks, both SIP and IAX2 but still couldn't get budgetphone.nl to work |
| 21:25.05 | riddlebox | [TK]D-Fender, can you tell me what to look for in here? http://pastebin.ca/684601 |
| 21:25.15 | mvanbaak | SA007: I dont want to scare you, but that's how it is |
| 21:25.29 | SA007 | it worked once this afternoon, but i tried immediatly after that and it didn't work anymore |
| 21:25.56 | [TK]D-Fender | riddlebox, how about you include the part where the call comes IN. |
| 21:26.00 | mvanbaak | SA007: that's how they work. allow the testcall, fuckup everything after that |
| 21:26.10 | riddlebox | see thats the thing, I cant ever find that part |
| 21:27.00 | SA007 | mvanbaak: hu? it was like call 15 that suddenly worked and call 16 failed again |
| 21:27.49 | mvanbaak | SA007: lemme guess, you did not change anything between call 15 and 16 right |
| 21:28.07 | SA007 | and 12/13/14/15/16/17/17 etc |
| 21:28.38 | mvanbaak | and you still want them to handle your business phonecalls ? |
| 21:28.59 | SA007 | i'm sure it's a configuration error on my side |
| 21:29.24 | mvanbaak | ok |
| 21:29.45 | mvanbaak | good luck finding the error. |
| 21:29.51 | mvanbaak | can I ask you something ? |
| 21:29.53 | SA007 | tnx |
| 21:29.55 | SA007 | sure |
| 21:30.09 | mvanbaak | if you have it working reliable, can you document the setup on voip-info.org ? |
| 21:30.24 | SA007 | sure |
| 21:30.33 | mvanbaak | cool. thanks |
| 21:30.46 | mvanbaak | I'm off to zzzzzzzzzzz land now |
| 21:30.49 | [TK]D-Fender | riddlebox, that sure doesn't LOOK like "everything" |
| 21:31.04 | SA007 | bye |
| 21:31.12 | riddlebox | [TK]D-Fender, hows this, http://pastebin.ca/684610 |
| 21:31.28 | [TK]D-Fender | riddlebox, the 1st line of your PB shows the destruction of a call being initiated, followed by a READ related to that same call ID |
| 21:31.58 | [TK]D-Fender | riddlebox, Better, and OBVIOUS |
| 21:32.01 | [TK]D-Fender | riddlebox, Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x101 (g723|g729)/video=0x0 (nothing), combined - 0x0 (nothing) |
| 21:32.09 | [TK]D-Fender | riddlebox, CODEC mismatch |
| 21:32.25 | [TK]D-Fender | riddlebox, SIP/2.0 488 Not acceptable here <------------ |
| 21:32.39 | [TK]D-Fender | NEXT!!@!@@!@! (c) BKW |
| 21:32.40 | riddlebox | hrmm I will look into it |
| 21:33.14 | riddlebox | its either that or I will give up on this quintum box, and get a linksys one |
| 21:34.44 | km- | hahaha |
| 21:34.47 | *** join/#asterisk edwin_quijada (n=m@200.88.116.25) |
| 21:34.50 | [TK]D-Fender | riddlebox, Its very blatantly telling you that its allowing G.723 & G.729, and your * setup is saying GSM or ULAW only. This is not Raw-Cat Science |
| 21:34.54 | km- | bkw rocked with the NEXT. |
| 21:35.14 | edwin_quijada | Which card I must use to connect my * to PABX digital? |
| 21:35.27 | [TK]D-Fender | edwin_quijada, Depends what KIND of digital |
| 21:35.54 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 21:36.19 | edwin_quijada | [TK]D-Fender: I have a PABK Meridian and I have developed a IVR and I need to use 4 lines from PABX |
| 21:36.34 | jwh | keeps segfaulting :( |
| 21:36.42 | edwin_quijada | I dont know how connect my * with the extensions of PABX |
| 21:37.39 | [TK]D-Fender | edwin_quijada, that is going to be complex at BEST. you could PERHAPS do this VIA a T1/E1 trunk card, but forget about the "PBX Digital" port concept. |
| 21:37.58 | [TK]D-Fender | edwin_quijada, How big a Meridian setup do you have? |
| 21:38.59 | edwin_quijada | this has 48 lines 2 T1 |
| 21:39.24 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 21:39.25 | edwin_quijada | and I need take 4 extensions for her to coneect to my * |
| 21:39.30 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 21:39.35 | edwin_quijada | is it possible? |
| 21:40.05 | [TK]D-Fender | edwin_quijada, you'll nned to use a T1 link to * if you expect to get a call back INTO your PBX after taking one in. |
| 21:40.07 | km- | the "Right" way to do it would be to hook an extra T1 up to it and drop it to the asterisk box |
| 21:40.30 | km- | you could also kinda rig something with digital to analog converters and a TDM400p, but T1 is far, far superior to that |
| 21:43.03 | km- | what we actually did at my old job was have an asterisk box in front of the proprietary pbx |
| 21:43.10 | km- | and just dropped a T1 to the proprietary pbx |
| 21:43.23 | km- | the users on the NEC system never knew there was another pbx in front of the outside world |
| 21:43.33 | km- | meanwhile we were able to expand with voip without "upsetting the apple cart" |
| 21:44.11 | edwin_quijada | km-: so we can use ata and TDM cards? |
| 21:44.11 | km- | the executives got their favoritely annoying NEC Electra Elite system, and meanwhile those of us who needed to do actual work got 7960's :) |
| 21:44.36 | km- | edwin_quijada: can the meridian push calls out via voip? |
| 21:44.58 | km- | edwin_quijada: because short of that, what you really need is to run a T1 trunk from the meridian system to the asterisk box via a T1 card |
| 21:45.10 | [TK]D-Fender | edwin_quijada, if you use ATA's then you call is never coming back FROM *. is that acceptable for you? |
| 21:45.25 | km- | yeah, TBT would be pretty difficult in that situation :) |
| 21:46.21 | edwin_quijada | so we need to do trunk with asterisk and Meridian using a T1 card |
| 21:46.34 | edwin_quijada | [TK]D-Fender: No, we need the call |
| 21:46.44 | [TK]D-Fender | edwin_quijada, how big is your existing PBX? |
| 21:47.02 | edwin_quijada | [TK]D-Fender: I have 48 lines |
| 21:47.06 | edwin_quijada | 2 T1 |
| 21:47.11 | [TK]D-Fender | edwin_quijada, how many PHONES? |
| 21:47.22 | edwin_quijada | 40 +- |
| 21:47.40 | edwin_quijada | remember this is for IVR |
| 21:47.50 | [TK]D-Fender | edwin_quijada, ok, and how many T1 ports do you have FREE on your PBX currently? |
| 21:48.25 | edwin_quijada | really, I dont know |
| 21:49.03 | [TK]D-Fender | edwin_quijada, that is not a good answer. You'll need one... |
| 21:49.35 | fujin_ | just build a replacement PBX with asterisk |
| 21:49.38 | fujin_ | on a seperate T1 |
| 21:49.43 | fujin_ | which does the exact same thing |
| 21:49.53 | km- | fujin: while ideologically the right answer, may not be business feasible |
| 21:50.00 | fujin_ | get more money |
| 21:50.15 | km- | its not just money. It's testing time, downtime of migration, the need to get more phones |
| 21:50.20 | km- | meridian phones are proprietary digital |
| 21:50.23 | fujin_ | ah |
| 21:50.24 | km- | they wont work with asterisk |
| 21:50.25 | fujin_ | that's rather homosexual |
| 21:50.37 | km- | yeah, it's why we put an asterisk mothership pbx before our NEC |
| 21:50.42 | fujin_ | sell them all on ebay |
| 21:50.47 | km- | haha |
| 21:50.47 | fujin_ | lol :) |
| 21:51.02 | Lucky7 | in phone1.cfg |
| 21:51.04 | Lucky7 | it has a tag |
| 21:51.06 | Lucky7 | <phone1> |
| 21:51.09 | fujin_ | the engineer before me was sold a batch of mitel 5224,s they were complete rubbish. I sold them on the equivalent of ebay. |
| 21:51.09 | km- | ok... I've been waiting for this user to call me back for an hour |
| 21:51.11 | fujin_ | wasy fun :] |
| 21:51.12 | edwin_quijada | I thought that I can connect asterisk with it without more problem |
| 21:51.13 | Lucky7 | do i change that to w/e the extension is |
| 21:51.18 | Lucky7 | IE <phone140> |
| 21:51.42 | km- | edwin_quijada: anyone who gives you a quick answer is either stupid or trying to sell you something |
| 21:51.53 | km- | edwin_quijada: it's completely based on what you have in place already and how it would hook together |
| 21:52.06 | edwin_quijada | [TK]D-Fender: and i cant change the Pbx because this is for one client that just wants the IVR nothing about their PBX or phone systems |
| 21:52.38 | km- | edwin_quijada: now, this might not be the right answer for you, but it sounds like you need to do exactly what I did for the old company |
| 21:52.58 | edwin_quijada | <PROTECTED> |
| 21:53.16 | km- | edwin_quijada: you need to get an asterisk box with a 4port T1 card. You need the t1 settings from your telco as well. What you do is, create two T1 legs, one incoming to the asterisk box from the telco, one outgoing to the meridian |
| 21:53.24 | km- | find out how many digits they send for DNIS/DID |
| 21:53.38 | km- | and then just transparently forward all calls from T1-1 to T1-2 to begin with |
| 21:53.57 | km- | i.e. if there's a call on 1234567890, call 1234567890 on T1-2 |
| 21:54.14 | Lucky7 | what the crap, this is the wierdest thing i've ever seen. |
| 21:54.22 | edwin_quijada | km-: :( the big problem is that is not an option |
| 21:54.32 | Lucky7 | http://rafb.net/p/n83v2x48.html // macaddress.cnf |
| 21:54.34 | fujin_ | then you're doing it wrong |
| 21:54.35 | fujin_ | find another solution |
| 21:54.38 | km- | edwin_quijada: unfortunately it's the least invasive option |
| 21:54.45 | edwin_quijada | they dont do that |
| 21:54.48 | Lucky7 | http://rafb.net/p/nKxkc765.html // phone140.cfg |
| 21:54.58 | Lucky7 | http://rafb.net/p/bR3PXq12.html // sip.cfg |
| 21:55.00 | km- | hmm |
| 21:55.06 | km- | how would I solve your issue if I couldn't do that. |
| 21:55.25 | km- | edwin, is the IVR for their business to route calls to extensions? |
| 21:55.31 | *** join/#asterisk VJFROMGT (n=vjfromgt@static-68-161-227-229.ny325.east.verizon.net) |
| 21:55.35 | edwin_quijada | km-: just one way! |
| 21:55.48 | VJFROMGT | I am trying to do a soft hagup but keep getting is not a known channel |
| 21:55.49 | Lucky7 | Why wont the phone 1) get the proper time, 2) actually register with the Asterisk Server |
| 21:56.00 | edwin_quijada | km-: is for bussines to conect to database |
| 21:56.05 | VJFROMGT | what is time server on phone 1? |
| 21:56.18 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 21:56.42 | edwin_quijada | so they have inused extensions into pabx |
| 21:56.48 | tzanger | damn is coppice never around anymore? |
| 21:56.58 | edwin_quijada | and they want use it for this |
| 21:57.38 | Lucky7 | i went through the walkthrough on voipinfo, and i've got all those settings |
| 21:57.38 | edwin_quijada | so bougth anlaog lines and use TDM card is the only way , I think!! |
| 21:58.11 | [TK]D-Fender | edwin_quijada, like I said you can use analog + ATA's, but once calls enter your IVR, they're never coming BACK. |
| 21:59.12 | km- | it doesnt sound like thats an issue for him |
| 21:59.15 | edwin_quijada | [TK]D-Fender: but if the call never back i cant run my AGI script for? |
| 21:59.28 | *** part/#asterisk fatgoose (n=fg@206-248-175-211.dsl.teksavvy.com) |
| 21:59.44 | km- | what happens is |
| 21:59.49 | km- | user calls an extension on meridian |
| 21:59.59 | km- | the meridian system goes to an analog extension converter |
| 22:00.07 | km- | the analog extension dials the asterisk box |
| 22:00.17 | km- | the asterisk box then has that call, completely, until the user hangs up |
| 22:00.28 | km- | the call can never go back out to the meridian system |
| 22:00.38 | km- | i.e., there can never be a transfer-out of the IVR back to a hardline extension |
| 22:00.57 | km- | if that's not a big deal, then doing the above is fine |
| 22:01.10 | km- | but if it is a big deal, you'll need to use a T1 for it. |
| 22:01.17 | edwin_quijada | i should use ata |
| 22:01.49 | tzanger | eek |
| 22:01.50 | tzanger | meridian |
| 22:01.59 | edwin_quijada | if I use a T1 card can I do trunk with this |
| 22:02.09 | tzanger | km-: that's not true |
| 22:02.11 | edwin_quijada | and use the 4 extension |
| 22:02.18 | edwin_quijada | ? |
| 22:02.22 | tzanger | km-: you can hookflash the ATA to do things like park and transfer and so on |
| 22:02.25 | tzanger | it's just a PITA |
| 22:02.55 | km- | tzanger: hmm |
| 22:02.58 | edwin_quijada | tzanger: i can transfer the call from * to Meridian |
| 22:02.59 | edwin_quijada | ?\ |
| 22:03.09 | km- | tzanger: you can send a flash event back over a zap channel? |
| 22:03.10 | tzanger | edwin_quijada: if you tell * to hookflash the zap channel, yes |
| 22:03.19 | tsurko | hello |
| 22:03.22 | km- | tzanger: I wasn't even aware you could do that. |
| 22:03.31 | km- | tzanger: is it Hookflash(<channel>) or something else? |
| 22:03.33 | tzanger | km-: check the ATA and ATA2 user guides |
| 22:03.39 | tsurko | is it possible to create web based softphone with asterisk and ragi? |
| 22:03.41 | tzanger | km-: ZapFlash() I think |
| 22:03.44 | edwin_quijada | tzanger: I have never used hookflash |
| 22:03.45 | km- | wait, this ATA you're referring to |
| 22:03.50 | km- | you're not talking about SIP ATA's in this case? |
| 22:04.10 | tzanger | no, I'm talking about the Meridian ATA or ATA2 |
| 22:04.22 | km- | something makes me think I'm behind the times, I haven't really been on the cusp of new asterisk-related technologies since last time I stopped coming around here :) |
| 22:04.24 | km- | ahhhh. |
| 22:04.51 | km- | <PROTECTED> |
| 22:04.51 | km- | [Synopsis] |
| 22:04.52 | km- | Flashes a Zap Trunk |
| 22:04.56 | edwin_quijada | so km- your solution is posible? |
| 22:05.04 | km- | <PROTECTED> |
| 22:05.04 | km- | people who want to perform transfers and such via AGI and is generally |
| 22:05.05 | km- | quite useless oths application will only work on Zap trunks. |
| 22:05.34 | km- | edwin_quijada: tzanger's theory is credible however he is right, it could be a major PITA to implement |
| 22:05.51 | edwin_quijada | PITA? |
| 22:05.51 | km- | but as long as you don't need to transfer back out to a user |
| 22:05.55 | km- | Pain In The Ass |
| 22:06.06 | edwin_quijada | km-:jajajaja :0 |
| 22:06.44 | edwin_quijada | maybe just I need transfer to an agent if he need to talk with an agent |
| 22:06.55 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 22:07.07 | km- | I really think you'd be better served pushing an asterisk box in front of the meridian rather than behind it |
| 22:07.15 | km- | but failing that, it probably would be better the T1 route |
| 22:07.21 | km- | with the ATA thing and what tzanger said a distant third |
| 22:07.35 | Lucky7 | Can anyone here send me a working polycom config, so i can cross-compare? Preferablly the files for SIP2.1.0+ |
| 22:07.53 | VJFROMGT | i have setup an ivr, when ivr plays, i enter the extension of an ivr and it says not valid |
| 22:08.28 | km- | Is there a way to set the domain in cisco conifigs? |
| 22:08.46 | km- | sip domain I mean |
| 22:08.52 | km- | not dns |
| 22:09.59 | edwin_quijada | km- if the pabx has analog lines with can I use TDM cards without problems? |
| 22:12.02 | km- | you can, but you need to be sure you have enough analog lines to transfer back out |
| 22:12.05 | km- | if you are planning to transfer back out |
| 22:12.42 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583591.dsl.bell.ca) |
| 22:12.47 | edwin_quijada | km- they have 2 analog lines |
| 22:12.49 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 22:13.06 | edwin_quijada | the client call to company |
| 22:13.24 | edwin_quijada | respond IVR for attendance and info extensions |
| 22:13.41 | edwin_quijada | 221,222,223 ivrs service |
| 22:13.42 | SA007 | w00t, my phone works :) |
| 22:13.57 | edwin_quijada | the client press 221 |
| 22:14.07 | edwin_quijada | meridian forward to asterisk |
| 22:14.18 | edwin_quijada | astersk do the job |
| 22:14.37 | edwin_quijada | ask "u want a human operator? |
| 22:14.42 | edwin_quijada | client yes |
| 22:15.12 | edwin_quijada | asterisk trasnfer to 322 extension that is phisical extension in pbx and digital |
| 22:15.22 | edwin_quijada | km- it could be? |
| 22:21.38 | km- | if you only have two analog lines |
| 22:21.43 | km- | you're pretty screwed |
| 22:21.54 | km- | unless you're totally, totally sure that you will only ever get one client calling the IVR |
| 22:22.03 | km- | because you want that second analog line to transfer back out |
| 22:22.13 | edwin_quijada | jejje |
| 22:22.42 | km- | you really need to get asterisk in front of that meridian. I don't see any other way how you can make this work easily |
| 22:22.48 | km- | tzanger and the others may have other input |
| 22:22.56 | km- | I'm now an hour late however so I have to get going |
| 22:22.58 | edwin_quijada | ok if i use the solution that u said I must put the ATA into meridian and asterisk |
| 22:22.58 | km- | good luck! |
| 22:23.23 | Lucky7 | I'm using a Polycom IP330 Sip phone, and i keep getting this message on my asterisk box |
| 22:23.23 | Lucky7 | "Received incoming SIP connection from unknown peer to 7777") in new stack |
| 22:23.24 | edwin_quijada | thsks |
| 22:23.55 | Lucky7 | and then it says ss-noservice, instead of giving my expected "thank you for calling..." message |
| 22:24.03 | fujin_ | so configure the phone as a 'friend' |
| 22:24.12 | Lucky7 | its like my sip phone isn't actually registering in the system |
| 22:24.22 | edwin_quijada | tzanger: we can use the ata and dont came back |
| 22:24.34 | *** join/#asterisk nephfl (n=none@wsip-70-168-100-182.ga.at.cox.net) |
| 22:24.38 | Lucky7 | fujin > The phone, or the extension? |
| 22:24.39 | *** part/#asterisk kiscokid (n=ron@208.106.35.66) |
| 22:25.00 | nephfl | i cant get vtwhite to work on this system for anything |
| 22:26.13 | nephfl | sip show peer shows it as ok...but i get busy when i try to dial |
| 22:26.41 | Lucky7 | nephfl > Softphone? |
| 22:27.03 | nephfl | yes |
| 22:27.08 | nephfl | x-lite |
| 22:27.34 | Lucky7 | wierd, Xlite is normally one of the better ones that i've seen about that |
| 22:28.08 | nephfl | someone trying to connect remotely said they got someone from another country answer at their home number |
| 22:28.14 | nephfl | using an ata |
| 22:32.26 | Nichtwirklich | does chan_capi know/send that the outgoing call from a sip client is a fax, I mean, does it use the correct isdn service type? |
| 22:33.11 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 22:37.47 | *** part/#asterisk jmls (n=jmls@62.49.235.130) |
| 22:41.27 | *** join/#asterisk anthm (n=anthm@adsl-69-216-26-86.dsl.milwwi.ameritech.net) |
| 22:41.27 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:44.15 | Lucky7 | What the hell |
| 22:45.44 | *** join/#asterisk ReDNeQ (i=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 22:46.27 | *** join/#asterisk wundaboy (n=pat@pool-71-111-176-117.ptldor.dsl-w.verizon.net) |
| 22:47.37 | wundaboy | I am getting extremely long lag inbetween when i say something and the other person on the other side hears it |
| 22:47.43 | wundaboy | like 10 seconds or so |
| 22:49.57 | *** join/#asterisk ozus (n=ozus82@cpe-72-134-104-66.socal.res.rr.com) |
| 22:51.55 | Lucky7 | wundaboy > VOIP? T1? Analog? Softphones? Hardphones? |
| 22:52.08 | wundaboy | DSL / Polycom IP500 |
| 22:52.34 | wundaboy | I am in oregon and using an east coast provider ... but when I was on Cable it had about a 1 to 1.5 second lag |
| 22:52.56 | Lucky7 | So i assume then yes, it is a voip line |
| 22:52.57 | Qwell | Do you have like a 10 second jitter buffer or something? |
| 22:53.28 | wundaboy | Yes, it is a VOIP termination / origination |
| 22:53.46 | wundaboy | Qwell, what file would I set that in |
| 22:54.02 | wundaboy | I have about 120ms ping time in between me and my provider |
| 22:54.42 | Lucky7 | thats kinda gross |
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| 22:54.48 | Lucky7 | goto speedtest.net |
| 22:54.53 | Lucky7 | and see what your speeds are |
| 22:55.19 | Qwell | I don't trust speedtest.net - *AT ALL* |
| 22:55.27 | Qwell | They claim my speeds are 9mbit/2mbit...on consumer cable |
| 22:55.29 | Qwell | BS |
| 22:55.57 | Lucky7 | Qwell > then test it, see what you get uploading to a server capable of more then that |
| 22:55.59 | _x86_ | they got mine right |
| 22:56.02 | wundaboy | on speakeasy.net/speedtest I get 750 / 125 (SLOW! but should support voice) |
| 22:56.13 | Lucky7 | 125 is disgustingly slow |
| 22:56.14 | _x86_ | Qwell: my cable is 10mbit/1.5mbit |
| 22:56.17 | Qwell | Lucky7: I have, and I get *nowhere* near those speeds |
| 22:56.22 | Qwell | _x86_: I'm on comcast. enough said |
| 22:56.26 | _x86_ | haha |
| 22:56.30 | _x86_ | sorry to hear that :P |
| 22:56.32 | wundaboy | I am on verizon online dsl ... (not my choice) |
| 22:56.33 | Qwell | exactly |
| 22:56.35 | Lucky7 | same here |
| 22:56.36 | Qwell | so it's clearly wrong |
| 22:57.03 | wundaboy | jitterbuffer=no |
| 22:57.03 | Lucky7 | wundaboy > with 125kbps upload, if that PBX is not on a dedicated connection, thats probably where your suffering right there |
| 22:57.03 | wundaboy | forcejitterbuffer=no |
| 22:57.20 | wundaboy | Lucky7 ... my pbx is the computer I am on |
| 22:57.50 | Lucky7 | wunderkin > this isn't the verizon wireless internet is it? |
| 22:58.10 | wundaboy | Lucky7, no plain verizon online dsl over phone lines |
| 22:58.15 | Lucky7 | ok |
| 22:58.42 | wundaboy | although I am on wireless inbetween me and my modem ... but its like 1.2ms and a strong signal |
| 22:59.28 | Lucky7 | eh |
| 23:00.17 | Lucky7 | I'm not sure wundaboy. Out of personal experience, voip over a WIFI signal hasn't been very good |
| 23:00.24 | wundaboy | i know but its my only option |
| 23:00.30 | wundaboy | i think i just fixed it |
| 23:00.34 | wundaboy | bandwidth=low |
| 23:01.42 | wundaboy | yeah it calls out with GSM |
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| 23:05.02 | Lucky7 | what transport should i use with a Polycom phone? |
| 23:05.25 | *** join/#asterisk Yourname` (n=chatzill@76-10-128-178.dsl.teksavvy.com) |
| 23:06.07 | Yourname` | Hey [TK]D-Fender. Using call files and trying to use HANGUPCAUSE and DIALSTATUS isn't working out too well. Asked here too: http://tinyurl.com/2r5o6e |
| 23:08.47 | Yourname` | So how else can I get status of failed callfiles, especially when call returns call failed to go through, etc? |
| 23:10.08 | Lucky7 | many this is the most confusing thing ever... I stopped using server distrobution for my config method |
| 23:10.12 | Lucky7 | on my polycom's |
| 23:10.24 | Lucky7 | and now i'm JUST trying to hand program |
| 23:10.35 | Lucky7 | and they're still not registerring! lol, |
| 23:12.22 | fujin_ | fail |
| 23:12.23 | fujin_ | :\ |
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| 23:12.47 | dijungal | hi |
| 23:12.55 | Rahail | OK Ppl I am having problem with DTMF |
| 23:13.01 | Yourname` | very fail fujin_ very fail |
| 23:13.17 | Rahail | my Server A work fine with same DID but when i put that DID to server B it send extra digit |
| 23:13.22 | Rahail | this what i did for server B |
| 23:13.23 | Rahail | 1. Changed Asterisk Version 5 times |
| 23:13.23 | Rahail | 2.Changed codec go g729 ulaw alaw gsm |
| 23:13.23 | Rahail | 3. changed DTMF to infband auto rfc |
| 23:13.37 | dijungal | the host= parameter in a friend context of the iax.conf is the ip of the connecting host or the host to connect to? |
| 23:13.39 | Rahail | still same result |
| 23:14.14 | _ShrikE | err |
| 23:15.24 | dijungal | in other words, if i have server1 and server2, on server1 the "host=" should be the ip of server1 or 2 ? |
| 23:15.39 | dijungal | for type=friend |
| 23:17.04 | watchy | hey tk you there? |
| 23:17.24 | watchy | do you guys recommend trixbox for big installs? |
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| 23:17.39 | Sweeper | all signs point to no |
| 23:17.43 | Sweeper | ~trixbox |
| 23:17.44 | jbot | somebody said trixbox was a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
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| 23:18.23 | watchy | well this company wants to hire me and they use the shit outta trixbox |
| 23:18.43 | fujin_ | so turn the job down? |
| 23:18.43 | watchy | i like to install gentoo and install * myself |
| 23:18.54 | watchy | they offered me $100k |
| 23:19.00 | wundaboy | hey |
| 23:19.04 | wundaboy | they offer 100k you take the job. |
| 23:19.12 | watchy | yea thats what im saying |
| 23:19.28 | watchy | but they love some trixbox i dunno why |
| 23:19.39 | watchy | they say its easier to support then manually editing .confs |
| 23:20.04 | wundaboy | thats between them/you... |
| 23:20.17 | wundaboy | i dont think you are going to want to replace their pbx.... (that i am sure would be a headache) |
| 23:20.37 | Lucky7 | watchy > then I hate to say it, but who ever said that was a moron. |
| 23:20.42 | watchy | well they do phone systems for a living |
| 23:20.50 | watchy | and i'm coming in to start doing them for them |
| 23:20.58 | Lucky7 | I've built a few phone systems, and the one time i did a trixbox install, was a nightmare, for everything. |
| 23:21.03 | watchy | i went to checkout one of their installs today |
| 23:21.21 | watchy | they got 4 rhino channel banks |
| 23:21.26 | Sweeper | watchy: well, if it's working for them, fine. but you can always try to wean them off it slowly |
| 23:21.29 | Rahail | can some one give me hint |
| 23:21.31 | Rahail | about DTMF |
| 23:21.31 | watchy | and use rhino cards on a rhino built pc |
| 23:22.07 | watchy | i prefer sangnoma cards myself |
| 23:22.17 | watchy | gotta love HW echo cancellation |
| 23:22.31 | Sweeper | and 8x t1 cards ;) |
| 23:22.56 | watchy | sweeper: you ever setup fax 2 email on * 1.4? |
| 23:23.09 | Sweeper | watchy: nope. |
| 23:23.11 | watchy | the trixbox dude is having problems and has no idea to fix it |
| 23:23.14 | watchy | so i'm suppose to fix it |
| 23:23.41 | Sweeper | what's he using? rxfax or hylafax? |
| 23:23.41 | wundaboy | offtopic: anyone watching mid tenn and louisville? |
| 23:23.51 | watchy | fuck if i know |
| 23:23.52 | wundaboy | less than 3 minutes in and 3 touchdowns have been scored |
| 23:23.58 | watchy | he just said asterisk 1.4 on trixbox |
| 23:24.06 | watchy | hes pretty smart phone system wise |
| 23:24.14 | Sweeper | watchy: well, if it comes with trixbox, go ask on their forums, they're pretty good about stuff |
| 23:24.16 | watchy | but i think trixbox has fucked him up * wise |
| 23:24.32 | Sweeper | if it's something he added on later...still go ask in #trixbox XD |
| 23:24.37 | RypPn | what would cause the remote party to hear themselves again due to echo during a sip call? |
| 23:24.40 | watchy | man f some trixbox |
| 23:25.28 | Lucky7 | watchy |
| 23:25.35 | Lucky7 | i built a trixbox system a few weeks back |
| 23:25.42 | Lucky7 | and it was still 1.2.x |
| 23:25.56 | Lucky7 | so unless they've upgraded JUST recently, i'm pretty sure they're not 1.4.x yet |
| 23:26.03 | Lucky7 | I personally use Elastix now |
| 23:26.04 | watchy | i think he upgraded man |
| 23:26.08 | watchy | wtf is elastix |
| 23:26.09 | Lucky7 | ah |
| 23:27.29 | watchy | ~elastix |
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| 23:34.28 | fujin_ | any way with AEL |
| 23:34.30 | fujin_ | can I pause in a macro? |
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| 23:34.30 | fujin_ | my call delivery macro is causing my system to stutter a bit |
| 23:34.30 | fujin_ | cause it tries to deliver to all agents at once, lags it out a bit |
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| 23:36.23 | dug | I am getting the error file.c: File /var/lib/asterisk/sounds/custom/mainmenu.gsm does not exist in any format even though the file exists on the drive? |
| 23:37.14 | Strom_M | don't specify the extension when you call Playback() |
| 23:43.30 | dug | Strom_M now it doesnt give an error but I cannot hear anything |
| 23:44.08 | dug | now works |
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| 23:49.32 | johnadsfsdfdf | what does asterisk use to send voicemails as attachment |
| 23:50.43 | Corydon76-vcch | /usr/sbin/sendmail |
| 23:51.09 | Corydon76-vcch | The formatting of the message is done internally |
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| 23:58.03 | hmmhesays | what do you do to apply the settings once you change gains in zapata.conf |
| 23:58.28 | codefreeze | jbot: the new kid is fine, I hope |
| 23:58.29 | jbot | okay, codefreeze |
| 23:58.42 | hmmhesays | just ztconfig |
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