| 00:00.45 | simon | thanks. |
| 00:03.17 | simon | voxter, it seems that the sipura 3102 is a network unit in itself, correct? |
| 00:03.42 | voxter | it is, but you dont have to use it as such |
| 00:03.53 | voxter | its just a good all in one interface for connecting both a phone and a phone line to asterisk |
| 00:04.04 | voxter | but yes it is also a router if you choose to use it as one |
| 00:04.29 | *** join/#asterisk |dennis| (n=Dennis@200.32.217.34) |
| 00:05.10 | simon | how do you alternatively hook it up to a linux box; through the RJ45 plug into a second netcard? |
| 00:05.41 | simon | it doesn't seem to have a serial line or a usb interface. |
| 00:11.39 | TJNII | simon, Modems just arn't good for voice |
| 00:11.56 | TJNII | the * box will talk to the sipura over the network via VoIP |
| 00:13.32 | *** join/#asterisk seanmh (n=johndoe@216.31.9.161) |
| 00:15.52 | simon | TJNII, is there a standard protocol for that? |
| 00:16.01 | TJNII | Yes |
| 00:16.22 | TJNII | ~book |
| 00:16.23 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 00:16.28 | TJNII | simon, ^^^^^^ |
| 00:16.48 | TJNII | Take a look at it, it will give you a foundation on VoIP and Asterisk |
| 00:17.23 | simon | cool. I already had it loaded, but I skipped through sections. |
| 00:18.04 | NickKoji | anyone ever had a problem with tdm400p cards and asterisk now not being able to detect it? please send me references you used |
| 00:18.24 | xacatecas | does anybody know whether the 64-bit and 32-bit versions of the g729 codec is perhaps not quite compatible? |
| 00:18.50 | xacatecas | in other words, is there perhaps a bug in it that prevents my 64-bit switch from communicating with a 32-bit upstream? |
| 00:21.13 | Qwell | xacatecas: nope |
| 00:22.09 | *** join/#asterisk swiftkick (i=eeeeeeee@mail.beanproducts.com) |
| 00:22.17 | *** join/#asterisk pcrane (n=pcrane@202.49.106.158) |
| 00:22.25 | xacatecas | *sigh* ... i'm running out of ideas. http://voip.uls.co.za/g729/ ... mixmonitor.wav - the mixmonitor recorded file, the pcap is the captured rtp stream, then .g729 is extracted from that and then converted to wav - however I could not hear anything on my cell. |
| 00:23.18 | swiftkick | can someone point me to the specs of how to specify outbound caller ID? e.g. not only specs on calling the CALLERID function but also sip.conf and users.conf callerid = and cid_number = |
| 00:24.08 | swiftkick | i have a VOIP provider that claims to honor the CALLERID settings in the SIP headers but cant seem to find the right place wherer to specify the values per sip user etc. to asterisk |
| 00:25.15 | LemensTS | swift: exten => xxx,1,Set(CALLERID(num)=numberhere) |
| 00:25.43 | swiftkick | sure thats getting overwritten before it makes it to the SIP headers |
| 00:26.29 | LemensTS | swift: overwritten? it sets it when the extensions for that context is hit. |
| 00:26.52 | LemensTS | i have it working like that on 3 different voip providers on 2-3 different asterisks |
| 00:27.04 | swiftkick | if i place that right next to the outbound dial macro for that trunk, the SIP header does not show it being set |
| 00:27.27 | LemensTS | Do you see it being set in the CLI when the call is placed |
| 00:27.33 | swiftkick | zapata.conf shows caller id settings of the form callerid="Green Phone"<(256) 428-6121> |
| 00:27.46 | LemensTS | what asterisk version are you on |
| 00:27.54 | swiftkick | 1.4.18 i think |
| 00:28.22 | swiftkick | yep 1.4.18 |
| 00:28.28 | *** join/#asterisk PepOSX (n=angeldav@190.72.130.106) |
| 00:29.12 | LemensTS | it works in 1.2, im checking on 1.4 |
| 00:29.32 | swiftkick | thank you |
| 00:30.49 | LemensTS | I cant get to it right now, but Im sure it works i just set a new one up today that was on 1.4 Your provider offers it, but did you tell them to activate it on your acocunt? |
| 00:31.14 | swiftkick | yep |
| 00:31.33 | swiftkick | the issue is that it seems like its not making it out in the sip header... double checking on this right now |
| 00:31.58 | LemensTS | let me check my sip header too... |
| 00:35.22 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 00:36.02 | LemensTS | yea it shows up in my INVITE header of the sip |
| 00:39.40 | xacatecas | This paragraph is copyright (c) 2006 by Digium, Inc. In order for your module to load, it must return this key via a function called "key". Any code which includes this paragraph must be licensed under the GNU General Public License version 2 or later (at your option). In addition to Digium's general reservations of rights, Digium expressly reserves the right to allow other parties to license this paragraph under different terms. A |
| 00:39.41 | xacatecas | ny use of Digium, Inc. trademarks or logos (including "Asterisk" or "Digium") without express written permission of Digium, Inc. is prohibited. <-- found inside the binary-only g729 module. |
| 00:56.42 | *** join/#asterisk MrNaz (n=naz@ppp121-44-207-242.lns3.mel4.internode.on.net) |
| 00:56.55 | *** part/#asterisk carrar (i=tim@osburn.com) |
| 00:57.06 | *** join/#asterisk carrar (i=tim@osburn.com) |
| 01:00.06 | swiftkick | LemenTS: so yes it sets the name, but not the number, in the INVITE |
| 01:00.33 | swiftkick | is there a good document on the structure and contents of SIP headers somewhere? |
| 01:01.49 | Strom_M | swiftkick: you mean the RFCs aren't good enough? |
| 01:01.56 | *** join/#asterisk LemensTS (n=matthew@70.230.182.12) |
| 01:03.08 | swiftkick | LemenTS: so yes it sets the name, but not the number, in the INVITE |
| 01:03.25 | swiftkick | Strom_M: which RFC is it? ill gladly look at the RFC over at rfc-editor.org |
| 01:03.51 | swiftkick | INVITE sip:13122081208@sipconnect.chi0.cbeyond.net SIP/2.0 |
| 01:03.51 | swiftkick | Via: SIP/2.0/UDP 27.45.4.62:5060;branch=z9hG4bK4e49d5b0;rport |
| 01:03.51 | swiftkick | From: "Joe at NF" <sip:3126666666@sipconnect.chi0.cbeyond.net>;tag=as4011adf8 |
| 01:03.51 | swiftkick | To: <sip:13122081208@sipconnect.chi0.cbeyond.net> |
| 01:04.17 | swiftkick | this is outbound, to 13122081208, the callerid should be Joe at NF 312376xxxx |
| 01:04.36 | swiftkick | i dont know where the callerID data is supposed to go in this header. |
| 01:04.56 | swiftkick | 3126666666 is the sip username. |
| 01:06.00 | swiftkick | so is the FROM header also used to authenticate? |
| 01:06.09 | LemensTS | hey swift |
| 01:06.31 | file | ideally in an RPID header which is enabled using sendrpid=yes, but it can also be set in the From header but something providers require your authenticating username there... which means you can't put CID there |
| 01:06.35 | LemensTS | ~pb |
| 01:06.36 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines 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 |
| 01:06.36 | file | something? some |
| 01:07.47 | swiftkick | LemenTS: I figured that paste was less than 5 lines, not trying to annoy anyone |
| 01:08.55 | MrNaz | why was it thought a good idea to make Playback() take a filename with no extension and infer the "best" extension given the files in the path? |
| 01:09.40 | file | because then you can have files of multiple formats exist in a directory and not have to do anything funky to pick the right one? |
| 01:10.12 | swiftkick | file: thank you for the RPID info. that seems to have fixed it. |
| 01:11.34 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:11.51 | file | swiftkick: nifty that your provider supports it |
| 01:13.33 | MrNaz | file wow... i didnt know specifying the correct file extension was considered "funky" |
| 01:14.10 | file | MrNaz: take for example a call in the g729 audio format... and you have g729 audio files, you would want the g729 ones to be used so no transcoding is required |
| 01:14.14 | MrNaz | so Playback(intro.gsm) is funky compared to Playback(intro) ? |
| 01:14.17 | file | but if a ulaw call comes in you would want the ulaw ones to be used |
| 01:15.00 | MrNaz | surely there's a better method of selecting that rather than leaving it to the Invisible Backend ? |
| 01:15.09 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 01:15.12 | file | what would you propose? |
| 01:15.28 | MrNaz | has no idea, thus, shuts up |
| 01:15.45 | *** join/#asterisk alexhopper (n=a27386@mctnnbsa24w-142167043089.pppoe-dynamic.nb.aliant.net) |
| 01:16.00 | MrNaz | anyway.... at least now i understand why |
| 01:16.35 | swiftkick | are there docs anywhere on users.conf ? cant find any in the ancient asterisk manual on digium, not much on voip-info.org either |
| 01:16.51 | file | yup, and now you can effectively use that if you are doing audio playback and want to maximize CPU usage |
| 01:17.33 | file | swiftkick: there is a sample users.conf but I do not know of anything else... most people just use sip.conf/iax.conf/extensions.conf/etc instead of users.conf |
| 01:17.53 | swiftkick | file: sure, its just that asterisk-gui uses *both* sip.conf and users.conf |
| 01:18.15 | swiftkick | and nobody either there nor here has ever been able to answer what the actual purpose was originally supposed to be |
| 01:18.41 | Qwell | swiftkick: Asterisk-GUI only uses users.conf, and the general sections of the individual configs. |
| 01:18.44 | file | the original purpose was to have the idea of a "user" and simple options that automatically makes the required parts appear in the dialplan and channel drivers |
| 01:19.00 | *** join/#asterisk seanmh (n=johndoe@75.150.19.89) |
| 01:19.49 | swiftkick | Qwell: really??? its been a while but I recall noticing issues using only users.conf. (i inherited this asterisk install, now i get to sort it all out. fun fun!) |
| 01:20.06 | swiftkick | users.conf doesnt support the "register" directive does it? |
| 01:20.28 | swiftkick | it has registersip = |
| 01:20.36 | Qwell | That's all it should need. |
| 01:20.47 | swiftkick | but that doesnt seem to do the same thing. inbound trunks show up as '' using only registersip = in users.conf |
| 01:20.53 | swiftkick | adding a register line to sip.conf fixes that |
| 01:20.55 | swiftkick | thats just one issue |
| 01:20.58 | swiftkick | there are more |
| 01:21.02 | *** join/#asterisk hunmonk (n=hunmonk@drupal.org/user/22079/view) |
| 01:21.19 | swiftkick | inbound calls show up as being from '', rather. |
| 01:22.22 | Qwell | swiftkick: are you setting the contact? |
| 01:24.39 | swiftkick | Qwell: no, but i am setting the context |
| 01:25.02 | swiftkick | Qwell: is there a more up to date source of docs on sip.conf directives than the ancient and ugly page at http://www.voip-info.org/wiki-Asterisk+config+sip.conf ? |
| 01:25.22 | [TK]D-Fender | exothermc>[TK]D-Fender: When asterisk waits for a response. if the response is an exact match and there could be no other possible option within that context then shouldn't it go to that exten? <- Yes |
| 01:25.25 | Qwell | sip.conf.sample |
| 01:25.49 | swiftkick | lame |
| 01:25.53 | exothermc | [TK]D-Fender: Ya figured it out, I had some wide pattern matching going on which was stopping it. |
| 01:26.25 | [TK]D-Fender | swiftkick: Not applicable to GUI built configs. |
| 01:26.43 | swiftkick | Qwell: its painfully obvious that documentation is the least of Digium's concerns, dunno about the open source community... :) |
| 01:26.48 | [TK]D-Fender | And for global reference : users.conf is a fresh steamy pile of horse manure. |
| 01:27.46 | swiftkick | [TK]D-Fender: er... which are you saying is not relevant to gui built configs. sip.conf ? users.cconf ? registersip = ? |
| 01:27.56 | [TK]D-Fender | riddlebox: No application 'SetVar' for extension <- Since 1.2 you should have been using Set instead of SetVar. |
| 01:28.04 | voxter | users.conf for me is a confusing mess. |
| 01:28.13 | [TK]D-Fender | swiftkick: sip.conf |
| 01:28.32 | [TK]D-Fender | swiftkick: the GUI uses users.conf to set up registration and sip peers, etc. |
| 01:28.46 | [TK]D-Fender | swiftkick: Its trying to do too much all at one and can only do a shitty job of it. |
| 01:29.22 | [TK]D-Fender | voxter: No, its not just you, its the file :) |
| 01:29.22 | swiftkick | [TK]D-Fender: funny thing is that without the register line in sip.conf it doesnt work. Qwell was suggesting to set the contact = in users.conf, but I cant find any documentation on users.conf, let alone the directives it contains; im forced to assume they must be the same as sip.conf |
| 01:29.28 | JayTee52 | riddlebox? didn't he ask that question last nite? |
| 01:29.47 | [TK]D-Fender | swiftkick: Try to do it the 100% GUI way and learn by reverse engineering :) |
| 01:30.04 | JayTee52 | pain is fun |
| 01:30.08 | [TK]D-Fender | JayTee52: about 3 hours ago :) |
| 01:30.20 | JayTee52 | hahaha |
| 01:30.36 | JayTee52 | [TK]D-Fender, did you see my comment about my 800 number problem? |
| 01:30.51 | swiftkick | [TK]D-Fender : yah, it would help to have some info on the "contact =" directive. As much as I'd like to re-do this asterisk system from scratch using the GUI, not tonight.... definitely not tonight. :) :) :) |
| 01:30.58 | riddlebox | [TK]D-Fender, ok thanks |
| 01:31.02 | [TK]D-Fender | JayTee52: Yup. "r" <- wtf were you thinking? :) |
| 01:31.02 | swiftkick | its 90+% managed by the GUI as is |
| 01:31.22 | JayTee52 | [TK]D-Fender, I don't know! I've stripped it out of every Dial statement now |
| 01:31.30 | *** part/#asterisk Hydrant (n=aj@CPE0011950c737b-CM0012c90d1420.cpe.net.cable.rogers.com) |
| 01:31.32 | [TK]D-Fender | swiftkick: Better yet, do it from scratch yourself and screw the stupid GUI :) |
| 01:31.51 | riddlebox | JayTee52, I didnt really ask anything last night, I asked that this afternoon |
| 01:32.00 | JayTee52 | is r only for a zap analok channel? |
| 01:32.06 | JayTee52 | analog |
| 01:32.35 | swiftkick | [TK]D-Fender : i agree the GUI sucks. heres the funny part about users.conf vs sip.conf : it worked fine GUI only using users.conf until I started adding the 2nd line to these Cisco 7940s. The phones could not dial out until the account info for the 2nd line was added to sip.conf even though it was already in users.conf |
| 01:33.05 | JayTee52 | cuz I think I had to have it back when I was using TDM04b cards with FXO mods to analog lines. |
| 01:33.12 | [TK]D-Fender | swiftkick: You shouldn't normally be setting up multiple peers per single phone... |
| 01:33.24 | [TK]D-Fender | JayTee52: Still should never need it... |
| 01:33.35 | swiftkick | [TK]D-Fender : what is the suggested implementation for using a 2 line phone then? |
| 01:33.56 | JayTee52 | [TK]D-Fender, I think it was because I was connecting the FXO ports to analog lines from a Nortel pbx. |
| 01:34.02 | [TK]D-Fender | swiftkick: Its in your phone setup. it LOOKS like you're setting up the same peer MULTIPLE times to span the keys, and thats what it does |
| 01:34.11 | JayTee52 | and it wouldn't work right otherwise |
| 01:34.30 | JayTee52 | Nortel is such "teh suck" |
| 01:34.53 | [TK]D-Fender | swiftkick: Remember that a peer is a set of auth creds. It doesn't have an inherent limit to the number of calls that can be placed to & from it. |
| 01:34.53 | *** join/#asterisk BeeBuu (n=beebuu@218.13.89.223) |
| 01:34.56 | [TK]D-Fender | JayTee52: Last one nailed it. |
| 01:35.01 | swiftkick | [TK]D-Fender : well its not like I can give these phones a second IP addy! the "line 2" does have different creds |
| 01:35.26 | swiftkick | [TK]D-Fender: but, this was before i was getting used to viewing SIP headers from the CLI |
| 01:36.38 | JayTee52 | I've also got my Polycoms setup so they're in one VLAN while the computer that piggybacks off them is in another and each gets DHCP from a different server. |
| 01:37.00 | [TK]D-Fender | swiftkick: No, use the SAME ones. trust me, it'll "span". |
| 01:37.07 | [TK]D-Fender | swiftkick: its what everyone does. |
| 01:37.10 | JayTee52 | feeling less and less trepidation about this whole deal as time goes on |
| 01:37.26 | swiftkick | [TK]D-Fender you are saying use the same SIP creds? |
| 01:37.28 | [TK]D-Fender | swiftkick: Unless you really WANT it to be treated like a completely different identity |
| 01:37.39 | [TK]D-Fender | swiftkick: Normally, YES |
| 01:37.59 | swiftkick | [TK]D-Fender : well it essentially is a completely different entity. different DID, different outbound trunk |
| 01:38.15 | [TK]D-Fender | jayteeI usually one-up you and jsut shove them on entirely different LANs :) |
| 01:38.43 | [TK]D-Fender | swiftkick: YSo you want to treat the line-keys as separate "lines" with separate rules? |
| 01:39.10 | swiftkick | [TK]D-Fender : yes |
| 01:39.27 | swiftkick | they go to/come from separate contexts in extensions.conf |
| 01:39.28 | [TK]D-Fender | swiftkick: Oh... well never mind then :p |
| 01:39.28 | JayTee52 | [TK]D-Fender, you gonna pay for the cabling? we're a zoo with a large campus environment. shit, man you can't even afford to go to Astricon so I know you don't have the money and my boss's last name is Tightwadassclown |
| 01:39.48 | swiftkick | [TK]D-Fender : well, sure, but the original question remains: why did adding an entry to sip.conf fix the issue? |
| 01:39.57 | swiftkick | oh and btw |
| 01:40.05 | [TK]D-Fender | JayTee52: My company won't pay for it, and I CHOOSE not to spend MY money on it :) |
| 01:40.24 | swiftkick | godd*mn cisco for only supporting 3 stupid bellcore alternate ringtones via ALERT_INFO in their SIP firmware |
| 01:40.33 | unpaidbill | priority labels are the best thing since sliced bread |
| 01:40.38 | JayTee52 | [TK]D-Fender, looks like we both work for companies that undervalue us |
| 01:40.40 | Qwell | swiftkick: why not just add another users.conf entry? |
| 01:40.43 | unpaidbill | this is unpaidbill, signing off. |
| 01:40.45 | swiftkick | Qwell: I did. |
| 01:40.50 | JayTee52 | nite unpaidbill |
| 01:40.57 | swiftkick | Qwell: It didnt work until I added the corresponding extra entry to sip.conf also |
| 01:40.59 | [TK]D-Fender | JayTee52: And as for "cost" factor this : IP330 = $105, IP320 = $80. Thats 25$ per phone I can invest in wiring by keeping them separate. |
| 01:41.23 | swiftkick | is asterisk SCCP support ready for prime time? |
| 01:41.25 | [TK]D-Fender | JayTee52: and my infrastructure allows me to grow or run POTS over it as well. |
| 01:41.33 | [TK]D-Fender | swiftkick: LOL... |
| 01:41.45 | Wayhigh | anyone here seen the 'asteriskhunter' commercials in FL? |
| 01:41.47 | unpaidbill | i use SCCP without problems |
| 01:41.49 | Qwell | works fine in 1.6 |
| 01:41.52 | JayTee52 | [TK]D-Fender, I went down that road and I still have the teeth marks on my ass to show for it. You ever hear the old addage, "Never argue with a sick mind"? |
| 01:41.59 | unpaidbill | it's the SCCP-b thingy though, the one the 'other' guys made |
| 01:42.07 | Wayhigh | I totally thought they were about asterisk until it finally mentioned it being about contracts with hidden fees |
| 01:42.15 | *** part/#asterisk simon (n=simon@78.129.201.121) |
| 01:42.19 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 01:42.20 | unpaidbill | hopefully sccp in 1.6 works with all my crappy old phones |
| 01:42.34 | swiftkick | unpaidbill: we got these 7940's and flashed them right away to SIP 8.6, i was just hoping to get a little more functionality out of these damn things |
| 01:42.47 | [TK]D-Fender | JayTee52: Ever hear that the fastest way to a man's heart is with a sharp blade inserted horizontally through the ribcage? :) |
| 01:42.48 | unpaidbill | because i still cant get the damn 7935 (the polycom bastard child conferencing phone) to register without freezing |
| 01:42.56 | swiftkick | unpaidbill: i mean, after checking out the grandstream GXP 2000, we could have saved 2/3rds the cost by buying those instead of these ciscos |
| 01:43.01 | [TK]D-Fender | swiftkick: A sorry investment those are... |
| 01:43.15 | unpaidbill | swift SIP firmware on cisco phones was intentionally made shitty |
| 01:43.21 | jblack | I think my callerid is horked up. |
| 01:43.24 | [TK]D-Fender | swiftkick: Where are you located? |
| 01:43.25 | swiftkick | unpaidbill: GRANTED, the ciscos are physically constructed more like i expect a phone than the grandstream |
| 01:43.29 | JayTee52 | [TK]D-Fender, thought's crossed my mind but then the 300lb drooling guy named Bubba making me his prison bitch image kinda gave me second thoughts. |
| 01:43.33 | swiftkick | [TK]D-Fender: Chicago. you don't like Grandstream? |
| 01:43.37 | [TK]D-Fender | ~gs |
| 01:43.38 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 01:43.40 | swiftkick | unpaidbill: yeah i was kinda guessing that |
| 01:43.40 | [TK]D-Fender | ~grandstream |
| 01:43.41 | jbot | somebody said grandstream was the Yugo of VoIP hardware. Run. Run away now. |
| 01:43.42 | swiftkick | hahahahahah |
| 01:43.43 | [TK]D-Fender | ^^^^^^^^^^ |
| 01:43.45 | swiftkick | lol |
| 01:43.52 | [TK]D-Fender | swiftkick: their crapTASTIC! |
| 01:43.52 | jblack | exten => _NXXXXXX,1,Set(CALLERID(num)=${LOCALNUM}) |
| 01:43.57 | unpaidbill | the interface is different from SCCP in very annoying ways.. for example, in the 7912g the transfer buttons are on the first screen while you're on a call, in SIP firmware you have to press the 'more' button before you see the transfer button |
| 01:44.06 | swiftkick | [TK]D-Fender : i have a GXP-2000 and it works great |
| 01:44.10 | unpaidbill | i hate cisco. |
| 01:44.13 | gramulhaozin | ~cisco |
| 01:44.14 | jbot | cisco makes the routers that move the pr0n across the internet at a very high rate of exchange. a company full of patent protected punks!, or <reply>Cisco phones are expensive crap which should be avoided with extreme prejudice |
| 01:44.14 | [TK]D-Fender | swiftkick: Being in the USA you should never have considered anything except POLYCOM <- |
| 01:44.14 | jblack | Presuming LOCALNUM is NXXNXXXXXX, that looks about right, no? |
| 01:44.22 | edibrac | maybe when GrandStream phones first came out they were not too bad, compared to whatever else was available? |
| 01:44.25 | swiftkick | lol @ a company full of patent protected punks |
| 01:44.29 | unpaidbill | i have been buying polycoms lately |
| 01:44.38 | unpaidbill | i like them |
| 01:44.39 | gramulhaozin | so jbot says most of the stuff is crap |
| 01:44.42 | *** join/#asterisk moy (n=moy@189.169.95.179) |
| 01:44.44 | swiftkick | [TK]D-Fender : it wasnt me doin the considerin'. I inherited this asterisk system from a vendor who shall remain nameless |
| 01:44.46 | [TK]D-Fender | jblack: Yes, thats legal looking. |
| 01:44.49 | gramulhaozin | as jbot says Cisco or Grandstream all crap. |
| 01:44.51 | edibrac | i didn't get into asterisk until recently.. but we have some GS phones left lying around |
| 01:44.55 | JayTee52 | ~polycom |
| 01:44.55 | jbot | [polycom] the manufacturer of one of the best IP phones in the market. http://polycom.com - Note: Here is where you can get some downloads: http://www.polycom.com/resource_center/0,,pw-6812-12612,00.html |
| 01:45.05 | swiftkick | unpaidbill: (or anyone) you know any phone provisioning utils for these ciscos ? |
| 01:45.07 | JayTee52 | bias? naaaaaaaah |
| 01:45.07 | [TK]D-Fender | swiftkick: My God have mercy upon thee :p |
| 01:45.12 | unpaidbill | it's fucking beautiful how cheap polycom phones |
| 01:45.15 | LemensTS | ~linksys |
| 01:45.16 | jbot | linksys is probably a tool of satan |
| 01:45.22 | LemensTS | lmao |
| 01:45.27 | [TK]D-Fender | swiftkick: Cisco provisioning is fairly well covered on the WIKI actually. Go check it out |
| 01:45.30 | gramulhaozin | looks like someone ows polycom stocks |
| 01:45.33 | unpaidbill | swiftkick i can help you for 7912, 7902, 7960 |
| 01:45.39 | unpaidbill | 7940 i have never had the misfortune of dealing with |
| 01:45.42 | LemensTS | digium and polycom got a partnership |
| 01:45.49 | gramulhaozin | [TK]D-Fender: did you wrote jbot? |
| 01:45.51 | unpaidbill | err you for = you with |
| 01:45.52 | swiftkick | unpaidbill: yah I have 7940 and 7960, under sip 8.6 they are virtually identical |
| 01:45.58 | [TK]D-Fender | gramulhaozin: No, Tim Riker did |
| 01:46.09 | [TK]D-Fender | ~jbot |
| 01:46.10 | jbot | i heard jbot is a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or [TK]D-Fender's b*tch, or suck, or a pain in the ass |
| 01:46.10 | unpaidbill | yeah that's what i have on mine swift |
| 01:46.12 | gramulhaozin | Tim Riker owns Polycom Stocks. |
| 01:46.19 | [TK]D-Fender | gramulhaozin: jbots BY BITCH |
| 01:46.23 | [TK]D-Fender | MY* |
| 01:46.28 | unpaidbill | 8.9 actually |
| 01:46.32 | unpaidbill | but what can you do |
| 01:46.37 | gramulhaozin | Cisco phones are great and they are not expansive |
| 01:46.39 | swiftkick | unpaidbill: you use any opensource provisioning utilities? |
| 01:46.44 | [TK]D-Fender | unpaidbill: Kill'em all, let God sort'em out. |
| 01:46.45 | unpaidbill | for SIP? |
| 01:46.50 | Qwell | not expansive |
| 01:46.52 | Qwell | INDEED! |
| 01:46.54 | gramulhaozin | you get a BRAND NEW GREAT PHONE for $165, with a descent 320x240 PIXEL GRAPHICAL Screen |
| 01:47.00 | unpaidbill | i use dhcpd + tftpd |
| 01:47.02 | gramulhaozin | not a bullshit 3 lines TEXT Screen |
| 01:47.04 | Qwell | Your typo was amazing. |
| 01:47.13 | unpaidbill | there is extensive documentation on the wiki |
| 01:47.16 | swiftkick | unpaidbill: me too. i meant, more for serving up data to the various XML urls |
| 01:47.21 | unpaidbill | oh |
| 01:47.23 | swiftkick | unpaidbill: which wiki are you and [TK] talking about ? |
| 01:47.28 | [TK]D-Fender | ~wikis |
| 01:47.28 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 01:47.29 | unpaidbill | just use apache or lighttpd |
| 01:47.30 | [TK]D-Fender | ^^^ |
| 01:47.42 | swiftkick | oh yah im on there day and night. lately. :) |
| 01:47.44 | JayTee52 | Microsoft is now supporting Apache |
| 01:47.50 | unpaidbill | haha what |
| 01:47.58 | LemensTS | you mean the sip proxy formally known as OpenSER |
| 01:47.58 | JayTee52 | serious! |
| 01:48.01 | unpaidbill | what does supporting mean exactly |
| 01:48.14 | unpaidbill | did ms just give up on IIS? |
| 01:48.16 | JayTee52 | financially and also opening up code to them |
| 01:48.26 | unpaidbill | oh really |
| 01:48.30 | unpaidbill | interesting |
| 01:48.34 | swiftkick | IIS-- |
| 01:48.37 | swiftkick | apache++ |
| 01:48.41 | gramulhaozin | swiftkick: the cisco's have a built in browser and a big screen you can take advantage of it. |
| 01:48.49 | unpaidbill | unfortunately there's nothing microsoft does on the web that i'm even remotely interested in :P |
| 01:48.52 | unpaidbill | iis is crap |
| 01:48.57 | unpaidbill | asp is crap |
| 01:49.00 | JayTee52 | the two things I read that shocked me the most in tech news this week was that and that Nortel is providing code to SIPfoundry for sipXecs |
| 01:49.00 | gramulhaozin | swiftkick: and for $165 per phone, they are great. |
| 01:49.01 | unpaidbill | :/ |
| 01:49.10 | swiftkick | gramulhaozin: yah im running SIP firmware, know any utilities to help set that up?? |
| 01:49.19 | gramulhaozin | swiftkick: set up what ? |
| 01:49.24 | JayTee52 | and that they've abandoned BCM in favor of a branded version of sipXecs |
| 01:49.25 | swiftkick | e.g. all the now worthless buttons under the SIP firmware |
| 01:49.29 | swiftkick | gramulhaozin: e.g. all the now worthless buttons under the SIP firmware |
| 01:49.39 | unpaidbill | haha you cant do SHIT to the buttons swiftkick |
| 01:49.52 | gramulhaozin | swiftkick: just put your TFTP Server up and the phones are going to get the config, on the config you specify the buttons HTML URL |
| 01:49.52 | unpaidbill | the best you can do is set up line1, line2, ... line6 |
| 01:50.07 | gramulhaozin | swiftkick: when they click on SERVICES the phone grabs a URL |
| 01:50.14 | gramulhaozin | swiftkick: you can make the phone go to GOOGLE.COM |
| 01:50.19 | swiftkick | unpaidbill: yah i was hoping i could at least put a NAME next to the line buttons. :-) |
| 01:50.55 | unpaidbill | there isnt enough room, names would overlap the gigantic pointless logo! |
| 01:50.58 | gramulhaozin | swiftkick: yes you can, that's the extension name |
| 01:51.01 | unpaidbill | :P |
| 01:51.01 | swiftkick | gramulhaozin: sure, id like to set up personal phonebooks on the web server here, get the "mail" button to actually go to the persons voicemail, etc |
| 01:51.27 | JayTee52 | I could never get the BLF keys on the Grandstream GXP-2000's to work as pickup line even following a howto off voip-info.org |
| 01:51.33 | gramulhaozin | swiftkick: it's simple man |
| 01:51.44 | gramulhaozin | unpaidbill: you definately don't know much about cisco phones. |
| 01:51.53 | gramulhaozin | definitely 3 |
| 01:51.57 | unpaidbill | messages_uri: "500" |
| 01:52.03 | unpaidbill | that's for your VM button |
| 01:52.04 | swiftkick | gramulhaozin: url ? |
| 01:52.09 | gramulhaozin | swiftkick: URL yes URL |
| 01:52.13 | gramulhaozin | swiftkick: google URL |
| 01:52.16 | swiftkick | no i am asking :) |
| 01:52.21 | swiftkick | for a URL with some docs :) |
| 01:52.25 | unpaidbill | gramulhaozin ok |
| 01:52.27 | gramulhaozin | swiftkick: google |
| 01:52.34 | JayTee52 | or goosh.org |
| 01:52.45 | gramulhaozin | swiftkick: and you can set the url that you want for the buttons |
| 01:53.03 | gramulhaozin | the EXTRA SPACE where you usually have your logo |
| 01:53.09 | swiftkick | gramulhaozin: do you have a site that documents all the directives in the cisco config files? ? ? |
| 01:53.13 | gramulhaozin | Is where the HTML/XML is going to load |
| 01:53.22 | gramulhaozin | cisco.com |
| 01:53.32 | gramulhaozin | swiftkick: just go to voip-info.org |
| 01:53.35 | gramulhaozin | and read |
| 01:53.40 | swiftkick | ick cisco.com |
| 01:53.51 | gramulhaozin | I can make your stuff work if you want |
| 01:54.19 | gramulhaozin | you can set me up access and I can straight up your provisioning, write code to grab your ADdress book from a database and display on your phone |
| 01:54.52 | gramulhaozin | basically over services you should have |
| 01:55.07 | gramulhaozin | 1 - EXTENSIONS LIST for all the employees so they know eachothers extensions |
| 01:55.16 | gramulhaozin | 2 - Personal Phone list (each employee list) |
| 01:55.24 | *** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 01:55.26 | gramulhaozin | 3 - Company Phone Book |
| 01:55.47 | LemensTS | ~bored |
| 01:55.47 | jbot | La ... lalalala ... beer! |
| 01:55.51 | gramulhaozin | 3 - Phone Features and Codes so you can do a Call Forwarding on Busy without having to read 300 asterisk docs |
| 01:55.56 | gramulhaozin | ops that was 4 |
| 01:56.59 | gramulhaozin | swiftkick: got it ? |
| 01:57.16 | justdave | I'll agree on the Linksys = crap |
| 01:57.36 | justdave | got a WIP330 here (wifi-enabled phone) and the think can't even hold a wifi signal for more than 5 minutes |
| 01:57.47 | justdave | with it right under the access point the whole time |
| 01:57.54 | gramulhaozin | justdave: return it as deffective |
| 01:58.01 | swiftkick | gramulhaozin: yeah that sounds ambitious :) |
| 01:58.17 | gramulhaozin | swiftkick: ambitious ? that's what I do on all my setups |
| 01:58.27 | gramulhaozin | swiftkick: at least that |
| 01:58.48 | LemensTS | justdave: you need a phd to configure those linksys phones |
| 01:59.00 | *** join/#asterisk Olobola (i=Olobola@132.sub-75-210-199.myvzw.com) |
| 01:59.03 | justdave | yeah, took me about 20 minutes to get the first one working. |
| 01:59.12 | justdave | all their entry spaces need IP addresses |
| 01:59.17 | gramulhaozin | LemensTS: IQ higher than 80 is enough. |
| 01:59.18 | justdave | and they don't make an easy way to type a dot |
| 01:59.31 | justdave | it's three keystrokes every time you need a dot in the IP |
| 02:00.36 | unpaidbill | justdave: you should see the voice configuration menus in the 7902 |
| 02:00.39 | justdave | finally got all the connection info entered and had to reboot the thing to get it to actually connect |
| 02:00.44 | unpaidbill | it's amazing. |
| 02:01.06 | LemensTS | gramul: the developers must have had an iq <= to that |
| 02:01.50 | swiftkick | so anyone do anything with features.conf ? |
| 02:02.08 | justdave | heads out for dinner |
| 02:02.33 | swiftkick | it seems like a nifty to trigger dialplan functions from in-call DTMF. like, say, to activate call recording. |
| 02:02.37 | [TK]D-Fender | swiftkick: like? |
| 02:02.56 | [TK]D-Fender | LemensTS: Linksys isn't hard to configure, just hard to find the DOCS for. |
| 02:03.01 | unpaidbill | it is indeed nifty |
| 02:03.26 | [TK]D-Fender | swiftkick: Has some nominal value. |
| 02:04.00 | JayTee52 | i want to be able to press *3 or something during an active call and have Asterisk play a farting sound file, but only from my phone |
| 02:04.19 | unpaidbill | you can do that easily |
| 02:04.59 | gramulhaozin | you guys want to turn asterisk into a PrankMachine mmm |
| 02:05.24 | unpaidbill | no i want it to be a FM DJ station |
| 02:05.33 | unpaidbill | like when i make a joke i press *7 and it plays the clown horn sound |
| 02:05.34 | JayTee52 | ampache |
| 02:05.38 | unpaidbill | or a rimshot at *6 |
| 02:05.45 | unpaidbill | haha actually i kind of like this idea |
| 02:05.50 | unpaidbill | im doing it. |
| 02:05.59 | swiftkick | unpaidbill: features.conf is the way yes? |
| 02:06.30 | *** join/#asterisk seanmh (n=johndoe@75.150.19.89) |
| 02:06.49 | unpaidbill | yeah |
| 02:07.07 | JayTee52 | but will the other party hear the DTMF? |
| 02:07.47 | swiftkick | unpaidbill: so is there an asterisk function to dump the ongoing audio of a call, thru a codec (hopefully!) to a text file? |
| 02:07.57 | swiftkick | e.g. both caller and callee |
| 02:08.00 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 02:08.00 | unpaidbill | to a text file? |
| 02:08.02 | swiftkick | er |
| 02:08.05 | swiftkick | sorry not text file. |
| 02:08.10 | swiftkick | to a media file |
| 02:08.15 | swiftkick | (sorry - long day) |
| 02:08.20 | JayTee52 | I figure if people piss me off enough and I get another job the day before I leave I'll replace all the music on hold files with Slim Whitman tunes |
| 02:08.21 | unpaidbill | yes, Dial option W/w |
| 02:08.45 | unpaidbill | http://www.voip-info.org/wiki-Asterisk+cmd+Dial |
| 02:09.08 | swiftkick | W: Allow the calling user to start recording after pressing *1 or what defined in features.conf (Asterisk v1.2.x); requires Set(DYNAMIC_FEATURES=automon) |
| 02:09.11 | swiftkick | yah i see |
| 02:11.06 | JayTee52 | everytime I click on a link like that and see all these section titles in bold saying New In Asterisk v1.2.0 I want to puke |
| 02:14.45 | *** join/#asterisk bdolljr_ (n=bdolljr@64.154.93.78) |
| 02:17.06 | bdolljr_ | I know I've read somewhere that the "Missed Call" messages can be suppressed on a 79XX Cisco phone. I can't seem to recall where I saw that. Can someone tell me how to suppress these messages on the phone. Per dial would be preferred. |
| 02:22.29 | *** join/#asterisk LemensTS (n=matthew@70.230.182.12) |
| 02:23.08 | *** part/#asterisk bdolljr (n=bdolljr@64.154.93.78) |
| 02:24.03 | *** join/#asterisk bdolljr (n=bdolljr@64.154.93.78) |
| 02:26.34 | swiftkick | any ERP or CRM systems known to be integrated with asterisk? e.g. to dial directly out of a contact book, or better yet, to let a customer or vendor call up and check balances and have invoices read to them via tts |
| 02:27.02 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1279551595.dsl.bell.ca) |
| 02:28.07 | [TK]D-Fender | swiftkick: Anything like that is purely custom and not necessarily particularly hard. |
| 02:29.24 | bdolljr | any ideas on how to suppress the "missed call" messages on Cisco 79XX phones? I read there is a way... somewhere... and now i can't find it. |
| 02:29.47 | swiftkick | [TK]D-Fender: sure, obviously not too difficult in theory, i was just wondering if anyone knew of any one actually doing that with commercial or production open source |
| 02:30.01 | [TK]D-Fender | swiftkick: TONS of places are. |
| 02:31.26 | swiftkick | [TK]D-Fender: ok but you refer to "purely custom" implementations yes? |
| 02:31.40 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 02:31.45 | [TK]D-Fender | swiftkick: Everything with * is custom. |
| 02:32.05 | [TK]D-Fender | swiftkick: There is no such thing as "drop in" that does run the show from beginning to end. |
| 02:32.18 | [TK]D-Fender | not* |
| 02:33.09 | swiftkick | [TK]D-Fender well some of the asterisk GUI efforts seem to be farther along at that kind of integration than others, thats for sure |
| 02:34.07 | swiftkick | [TK]D-Fender : custom is fine and i personally dont mind editing .conf files in vi but its nice to be able to give the end user some nice web or app GUI tools |
| 02:34.27 | [TK]D-Fender | swiftkick: You can't really personalise this stuff off a cut&dry interface unless they provide all facets and * is just there foryou to point a call to. |
| 02:35.06 | [TK]D-Fender | swiftkick: You would be well served to throw the idea of "bolt-on" right out the window. go read up on AGI |
| 02:35.35 | swiftkick | [TK]D-Fender: i was thinking more of inbound calls than outbound. you know IVR's that call external apps etc. |
| 02:35.52 | [TK]D-Fender | swiftkick: Same answer, and yes I was thinking inbound |
| 02:37.43 | *** join/#asterisk xpot (n=jim@75.149.224.186) |
| 02:39.23 | swiftkick | [TK]D-Fender: i realize where you're coming from with this, and that each "bolt on" solution is likely going to need to interact with different parts of the asterisk subsystem. but that doesnt mean there arent still many milestones of usability that remain to be reached. |
| 02:40.03 | [TK]D-Fender | swiftkick: ......um... what exacty is that supposed to imply? |
| 02:40.15 | [TK]D-Fender | "milestons of usability"? huh? |
| 02:40.58 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 02:46.37 | swiftkick | [TK]D-Fender: many of the people who could benefit most from an asterisk system aren't ever going to edit extensions.conf by hand, much less think about contexts, dialplan functions, SIP, IAX, codecs, etc. as a result there are a few decent GUI's both open source and commercial to address the need , say, as an IT professional specializing in asterisk, not just to provide them with a working PBX, but to provide them with the abil |
| 02:46.37 | swiftkick | ity to manage at least some of it themselves |
| 02:48.38 | [TK]D-Fender | swiftkick: Umm... what does any of that have to do with CMS interaction? Does every CMS store the same information in the same way with the same processing concepts? No to all of the above. Hence if every one is different there isn't much to be said for "bolt-on: |
| 02:48.58 | [TK]D-Fender | swiftkick: Anything that exists is provided BY the CMS otherwise rests in YOUR hands to create. |
| 02:49.25 | riddlebox | [TK]D-Fender, that pastebin about the faxing, is rxfax incorrect too? |
| 02:49.58 | [TK]D-Fender | riddlebox: lost it. RxFax is still the app name though. |
| 02:50.13 | swiftkick | [TK]D-Fender : yes, I agree. I think we are using the concept of bolt-on in different ways. i am treating asterisk as the thing that is bolted-on to a "larger" system |
| 02:50.56 | JayTee52 | hmmm, if I bolt-on a flux capacitor to my * box I can call myself in the future to find out the winning Powerball number. |
| 02:50.58 | [TK]D-Fender | swiftkick: you don't by by a phone system as an addition to your CMS. You buy a phone system so you can TALK to your customers :) |
| 02:52.08 | [TK]D-Fender | swiftkick: thats like saying you bought a a pair of cup-holders... with a CHAIR attached to them. |
| 02:52.36 | riddlebox | [TK]D-Fender, I have rxfax in it, does it need to be RxFax? |
| 02:52.37 | [TK]D-Fender | Ohh look, a make-up mirror with a CAR attached! |
| 02:52.47 | [TK]D-Fender | riddlebox: case-insensitive |
| 02:53.04 | riddlebox | k |
| 02:57.00 | riddlebox | [TK]D-Fender, http://pastebin.ca/1086739 |
| 02:57.38 | [TK]D-Fender | riddlebox: and did you download the apps to compile in and compile SpanDSP as well? |
| 02:58.39 | swiftkick | [TK]D-Fender : I was really just asking if anyone knew of anybody that had ALREADY integrated any random open source or commercial ERP or CMS system with asterisk, preferrably open source, just to know what solutions are out there. i then contrasted this with asterisk GUI efforts. i wasnt trying to imply that asterisk was in any way not feature rich nor complex, potentially ridiculously so. |
| 02:58.40 | swiftkick | [TK]D-Fender : Rather I was suggesting that while the asterisk subsystem is a fully functional entity, even digiums own effort at a open source gui seems to have bogged down considerably, and so far none of the other GUI's or alternate PBX engines such as freeswitch have anything like multi-tenant capability capable of granting tiered access to end users. |
| 02:58.59 | riddlebox | [TK]D-Fender, nope, didnt see that part in the book, go ahead and slap me |
| 02:59.06 | swiftkick | [TK]D-Fender: so I am still particularly interested to hear of any efforts that are making their way into standard ERP/CRM distributions |
| 02:59.44 | [TK]D-Fender | swiftkick: Digium's GUI isn't so much a full GUI as it is a framework SAMPLE. |
| 03:00.15 | swiftkick | [TK]D-Fender: i was looking at phonecall recently. looks interesting |
| 03:00.23 | [TK]D-Fender | swiftkick: Trixbox includes SugarCRM or whatnot with some level of integration, but then again... thats packaged by someone else for some overly specific end. |
| 03:00.59 | *** join/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net) |
| 03:01.23 | swiftkick | [TK]D-Fender : do the people in #asterisk-gui know they are only working on a sample? that might explain the activity level there. :) :) :) |
| 03:01.37 | [TK]D-Fender | swiftkick: There is nothing I can think of that is any real use to a real * user. Companies that want integration PAY other companies to set things like this up, or have staff that are themselves capable. The variety is too wide. tehere is no generic "fill in 3 blanks account balance checker" |
| 03:02.20 | [TK]D-Fender | swiftkick: people using GUI's are typically the people who are too lazy to learn to configure * themselves, or unworthy to even admin a toaster-grade PBX |
| 03:02.31 | [TK]D-Fender | swiftkick: Morons should not run PBX's |
| 03:02.39 | [T]ank | hey all... have about 100 linksys spa 942 phone for sale. All are in nearly new condition. Paid about $150 for them. Asking $50. any takers? |
| 03:02.55 | Strom_M | [T]ank: go to ebay |
| 03:03.04 | [T]ank | got a bunch of them there too |
| 03:05.13 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 03:06.27 | *** join/#asterisk LemensTS (n=matthew@adsl-70-238-134-114.dsl.stlsmo.sbcglobal.net) |
| 03:06.34 | LemensTS | ~hello |
| 03:06.35 | jbot | Howdy Bub |
| 03:07.02 | riddlebox | [T]ank, all 100 for $50 |
| 03:07.51 | *** join/#asterisk roe_ (n=roe___@216-164-160-45.c3-0.eas-ubr10.atw-eas.pa.static.cable.rcn.com) |
| 03:08.25 | *** join/#asterisk d-k-t (n=dt@125.118.38.181) |
| 03:08.41 | *** join/#asterisk xpot (n=jim@75.149.224.186) |
| 03:09.50 | JayTee52 | "Welcome to the Asterisk IRC support & Tank's Used Surplus VOIP Equipment Channel!!! Our Mid-Summer sale is almost over so act now!" |
| 03:10.17 | outtolunc | almost over.. damn |
| 03:12.48 | JayTee52 | speaking of shameless self-promotion I have a 1999 Ovation Celebrity CC057 acoustic/electric with custom hard shell case going for only $120 plus shipping. |
| 03:14.55 | [TK]D-Fender | What'd you replace it with? |
| 03:15.55 | JayTee52 | an Ibanez acoustic electric |
| 03:17.27 | [TK]D-Fender | what prompted the change? |
| 03:18.28 | JayTee52 | my "best" friend borrowed my favorite Epiphone and left it at another guy's house who suddenly split with his wife and took off. The guitar is nowhere to be found like him. My best friend asked later to borrow a DVD and my answer was No F'n Way! |
| 03:19.02 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 03:19.05 | [TK]D-Fender | I bought an Ibanez RG 420 this Jan to replace my old Dean electric. Both were 24-fret double-locking Floyd designs. The Ibanez has a LOT more clearance on the cut-away however and it was impairing me on the Dean |
| 03:19.05 | JayTee52 | I'd bought the Ovation just as a travel guitar so I didn't have to take the Epi on planes |
| 03:19.38 | JayTee52 | but I don't like the rounded molded resin back |
| 03:19.59 | JayTee52 | some do but I'm used to a stock guitar shape |
| 03:20.25 | [TK]D-Fender | Oh yeah... meh |
| 03:20.28 | JayTee52 | and the frets on the fretboard sucked |
| 03:21.16 | JayTee52 | too high and most of them had sharp edges at each side. |
| 03:21.24 | [TK]D-Fender | I bought an acoustic in Jan as well (just before the electric actually), and wanted a folk cutaway, but ended up with a dreadnought cut-away instead because ti was the ONLY one that felt good to play and sound-wise |
| 03:21.50 | JayTee52 | I love the Ibanez. it's got great tone and keeps it's tune for longer than any other inexpensive acoustic I've owned. |
| 03:22.32 | JayTee52 | my a/e is cutaway but the dreadnought isn't but the dread has awesome bottom |
| 03:22.39 | [TK]D-Fender | unfortunately it has DOLPHIN inlays for fret-markings. So my great sounding and playing guitars makes me look plausibly of "alternative" lifestyle. |
| 03:23.23 | *** join/#asterisk devhen (n=devhen@72.8.124.55) |
| 03:23.26 | JayTee52 | well, after you code a GUI that actually works and make loads of money you can buy a Martin or a Breedlove |
| 03:23.33 | *** join/#asterisk EricL (n=eric@jarbeeg.chal.net) |
| 03:24.02 | EricL | Was there a change from 1.4.18 to 1.4.21.1 that would prevent phones from being able to register (without changing the configs)? |
| 03:24.14 | [TK]D-Fender | no need.... I have an acoustic I like, and the only electric I would consider would be a custom neck-through job that would go for about $1500 |
| 03:24.16 | EricL | ie Is there some config options that I need to change as a result of upgrading? |
| 03:24.25 | [TK]D-Fender | EricL: nope. |
| 03:24.36 | [TK]D-Fender | EricL: PASTEBIn the failed attempt |
| 03:24.58 | EricL | There is no failed attempt. They just aren't talking to the Asterisk server since I did the upgrade |
| 03:24.58 | JayTee52 | what's the acoustic you like? |
| 03:25.10 | EricL | Even when I do core set verbose/debug 10 |
| 03:25.14 | [TK]D-Fender | EricL: no packets = networking problem, and not *'s |
| 03:25.21 | [TK]D-Fender | EricL: SIP DEBUG <-------- |
| 03:25.40 | EricL | I didn't change any network setup. |
| 03:25.52 | JayTee52 | same IP for the server? |
| 03:26.09 | [TK]D-Fender | JayTee52: hold onn : EricL back everything up. if we can confirm it then its something completely different |
| 03:28.00 | EricL | [TK]D-Fender: I don't have the old install, just the old configs. |
| 03:28.04 | EricL | is an idiot |
| 03:28.35 | EricL | It only makes sense that its a networking issue, but I can ping from the * box to the phone. |
| 03:30.02 | [TK]D-Fender | EricL: ... FUCK PING. go look at some IMPORTANT shit on it. |
| 03:30.16 | *** join/#asterisk seanmh (n=johndoe@75.150.19.89) |
| 03:30.29 | [TK]D-Fender | EricL: iptables, routing. WAN IP, local IP. check your DAEMONS. |
| 03:31.03 | JayTee52 | [TK]D-Fender, when you're not too busy and want to look. http://www.ibanez.com/acoustic/guitar.aspx?m=AES10EAM I think I showed you this about a year ago. |
| 03:31.05 | [TK]D-Fender | EricL : and NEVER rely on useless statements like "I don't have the old install, just the old configs". |
| 03:31.29 | [TK]D-Fender | jayteeLast year, yup |
| 03:32.05 | jblack | ericl: I'd check lsof -i to make sure * is listening, I'd do a tcpdump on the phones and * too, to see who is talking and who isn't. |
| 03:33.48 | *** join/#asterisk keith4_ (n=kbe2@207-172-236-173.c3-0.eas-ubr9.atw-eas.pa.cable.rcn.com) |
| 03:33.55 | keith4_ | is there any way to auth sip users to ldap? |
| 03:34.30 | [TK]D-Fender | keith4 : realtime. |
| 03:35.26 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 03:36.03 | keith4_ | hmmm |
| 03:36.17 | keith4_ | can you explain how that's supposed to work, in once sentence? |
| 03:36.59 | keith4_ | sort of a JIT config? |
| 03:38.00 | jblack | Hmmm. |
| 03:38.08 | jblack | I don't quite get what realtime has to do with ldap. |
| 03:38.29 | keith4_ | it looks like it's just one more available "database" driver |
| 03:38.36 | keith4_ | http://free.oxymium.net/Asterisk/README.realtime-ldap |
| 03:39.20 | *** join/#asterisk TOrrIeri (n=torrieri@nelug/crew/torrieri) |
| 03:40.06 | *** join/#asterisk baliktad (i=baliktad@c-24-16-27-4.hsd1.mn.comcast.net) |
| 03:40.18 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 03:40.39 | jblack | I didn't know about that. That looks useful. |
| 03:40.47 | keith4_ | yes, it does |
| 03:42.43 | jblack | I need to find a new contract |
| 03:44.05 | keith4_ | ooh, the config ldap stuff is going to be in 1.6 |
| 03:46.46 | jblack | that's almost worth waiting for pre to become post if you need it. Perhaps even consider running pre. |
| 03:47.23 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 03:47.25 | EricL | jblack: Good call on the tcpdump. Don't know why I didn't do it earlier. |
| 03:47.54 | jblack | EricL: because the toughest bugs to find are the hidden basic assumptions. No worries. |
| 03:48.09 | EricL | The phone is sending SIP packets to the * server, but the * server isn't even saying its getting them. |
| 03:48.38 | [TK]D-Fender | EricL: Because you fel the "but I didn't change" anything schtick would actually mean something and that it would fix itself and you wouldn't have to look at it :) |
| 03:48.41 | jblack | server in which context? Asterisk itself, or the server the server is on? |
| 03:49.09 | [TK]D-Fender | EricL: PASTEBIN is your friend |
| 03:50.19 | *** join/#asterisk d-k-t-2 (n=dt@vpn21.witopia.net) |
| 03:50.58 | EricL | http://www.pastebin.ca/1086764 |
| 03:51.40 | EricL | That's the tcpdump from the * machine of 'tcpdump -i eth0 host 192.168.250.201' |
| 03:51.50 | jblack | So it is getting sip. |
| 03:52.00 | jblack | And it's trying to sip back, and getting blocked. |
| 03:52.02 | [TK]D-Fender | EricL: enable SIP debug in * CLI |
| 03:52.15 | jblack | [TK]D-Fender: Look at line 9 of his paste. |
| 03:52.17 | [TK]D-Fender | eircand please provider your sip.conf masking only passwords |
| 03:52.19 | EricL | [TK]D-Fender: I did and nothing shows up. |
| 03:53.26 | jblack | EricL: I'd check dns first, make sure things are resolving correctly. Second, I'd start checking iptables to see if things are being rejected. |
| 03:53.30 | [TK]D-Fender | EricL: Alongs with "netstat -an|grep 5060", "route -an", "iptables --list" , etc |
| 03:53.58 | [TK]D-Fender | EricL: You shuold be opening EVERYTHING that can be involved in this. |
| 03:54.51 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 03:55.10 | EricL | jblack: There are no iptables rules (you will see that in the pb in a moment). |
| 03:55.37 | *** join/#asterisk korihor (n=korihor@190.39.163.45) |
| 03:55.45 | jblack | I can't resolve that hostname, btw, so if you can add your /etc/hosts to that list, that'll be nice. |
| 03:56.17 | EricL | jblack: It's internal and it works inside. |
| 03:56.44 | jblack | I'm external, and it helps me if I know what has what addresses. |
| 03:57.43 | EricL | Yep, sorry. |
| 03:58.35 | jblack | tries to decide what he should try on hew new monitor next. |
| 03:58.41 | [TK]D-Fender | EricL: the rest please... |
| 03:58.53 | EricL | [TK]D-Fender: I am copying and pasting the sip.conf now. |
| 04:01.11 | EricL | http://www.pastebin.ca/1086768 |
| 04:01.26 | EricL | 192.168.0.205 = ast1nyc1.inside.eseoh.com |
| 04:02.15 | [TK]D-Fender | EricL: bindaddr=192.168.1.2 ; IP address to bind to (0.0.0.0 binds to all) <- WAKE UP TIME |
| 04:02.34 | [TK]D-Fender | EricL: udp 0 0 192.168.1.2:5060 0.0.0.0:* <-- should have tipped you off too |
| 04:02.45 | jblack | sigh |
| 04:02.54 | [TK]D-Fender | nobody EVER looks at the BIG PRINT |
| 04:03.09 | EricL | That's the way it has always been though. |
| 04:03.33 | [TK]D-Fender | EricL: 23:46:25.651205 IP 192.168.250.201.sip > ast1nyc1.inside.eseoh.com.sip: SIP, length: 588 <-- you WONDER why its ignoring THIS INTERFACE?! |
| 04:05.25 | JayTee52 | damn, it's late. I gotta get some zzzz's. nite all |
| 04:05.42 | EricL | [TK]D-Fender: So I should have it bind to 0.0.0.0 ? |
| 04:05.45 | JayTee52 | be well, citizens! |
| 04:06.06 | jblack | Dude, you're gonna kill [TK]D-Fender |
| 04:06.07 | *** part/#asterisk JayTee52 (n=jforde05@unaffiliated/jaytee) |
| 04:06.21 | [TK]D-Fender | EricL: ..... the "default" comments tell you to your face how it works. I'm going to pretend I didn't jsut read you asking that... |
| 04:07.31 | EricL | [TK]D-Fender: I understand that part of it. I guess my question is more specific to my setup. I'll just play around with it. |
| 04:08.08 | [TK]D-Fender | EricL: * listens on ONE IP or ALL. Your choice. You should already know which one will do the job for you |
| 04:11.20 | EricL | [TK]D-Fender: Alright. That fixed that problem. Now loading the ztdummy module isn't allow asterisk to start. |
| 04:12.04 | [TK]D-Fender | EricL: get reading... |
| 04:12.18 | [TK]D-Fender | EricL: go check your zaptel install, make sure you did * afterwards, etc... |
| 04:13.20 | baliktad | I am giving up trying asterisk in a virtual machine |
| 04:13.22 | EricL | [TK]D-Fender: Yep, * just dies if the zaptel module is loading. |
| 04:13.25 | baliktad | I'm willing to buy hardware |
| 04:13.34 | baliktad | someone just tell me what I need and I will buy it |
| 04:13.45 | jblack | asterisk works fine in a virtual machine, if you stay away from pri cards and such. |
| 04:13.56 | baliktad | it wasn't asterisk that was the problem |
| 04:14.16 | jblack | cpu starvation under xen? |
| 04:14.23 | [TK]D-Fender | baliktad: And what is the "problem"? |
| 04:14.55 | jblack | If so, try xm sched-credit -d pbxname -w 2048 |
| 04:15.14 | baliktad | the problem was vmware don't like virtual PC |
| 04:15.38 | [TK]D-Fender | baliktad: Um... virtualizing a virtual environment? |
| 04:15.43 | baliktad | no |
| 04:15.46 | baliktad | coexisting |
| 04:16.18 | legis | What should I edit/add to use a SIP DID for *? |
| 04:16.41 | [TK]D-Fender | legis: sip.conf & extensions.conf |
| 04:17.09 | legis | [TK]D-Fender: you know where there is a example for this? |
| 04:17.24 | baliktad | I don't wanna fight virtualization anymore |
| 04:17.32 | [TK]D-Fender | legis: check your ITSP for samples, or get Google-ing |
| 04:18.20 | legis | kk |
| 04:18.42 | legis | Can I make DID only ring on certain phones? |
| 04:19.17 | [TK]D-Fender | legis: You can make it dow just about whatever you want. |
| 04:20.15 | [TK]D-Fender | do* |
| 04:21.35 | legis | nice |
| 04:24.15 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 04:31.18 | *** join/#asterisk BBHoss (n=hoss@c-68-62-170-33.hsd1.al.comcast.net) |
| 04:31.46 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.138) |
| 04:33.08 | *** part/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net) |
| 04:40.02 | *** join/#asterisk crazygir (n=jason@c-24-63-114-26.hsd1.ma.comcast.net) |
| 04:42.17 | crazygir | I understand asterisk being used internally, like in a business environment, but is it relatively sane to setup asterisk running a phone network for a residential area.. like a small neighborhood? |
| 04:43.03 | [TK]D-Fender | crazygir: If you want to run a minor telco off it, sure, whatever |
| 04:43.21 | crazygir | I had the feeling, given asterisk can do anything :P |
| 04:43.32 | [TK]D-Fender | crazygir: everything depends on scale an approach. In many larger ones, * is only a piece with things like SER in front, etc. |
| 04:43.57 | crazygir | in telco-speak, what would that sort of system be? |
| 04:44.09 | crazygir | (i.e. I've got reading to do, what should I be looking for) |
| 04:44.23 | crazygir | particularly given how much info is available :) |
| 04:44.44 | [TK]D-Fender | crazygir: General rule is if you don't know the answer, you shouldn't be doing it in the first place. |
| 04:44.47 | jblack | crazygir: * works great in a home. =) |
| 04:45.12 | *** join/#asterisk djin (n=djin@84-104-110-116.cable.quicknet.nl) |
| 04:45.17 | crazygir | [TK]D-Fender: well of course I can figure it out myself |
| 04:45.35 | crazygir | it's a question of how much info to sort through *before* getting to the pertinent |
| 04:45.39 | crazygir | :) |
| 04:45.42 | [TK]D-Fender | crazygir: How many channels are you looking to support? |
| 04:45.43 | crazygir | jblack: so I've heard |
| 04:46.28 | crazygir | [TK]D-Fender: I'm only conceptualizing at this point. The end result would be rather large. But I'll be starting with my home first ;) |
| 04:47.00 | [TK]D-Fender | crazygir: Ah, another "I don't want to say anything" answer. Good luck with that. |
| 04:47.02 | crazygir | I was approached by someone with a possible market to serve |
| 04:47.23 | crazygir | [TK]D-Fender: well this gentlemen would like to serve a small country |
| 04:47.45 | jblack | I could use some money. Want help? |
| 04:47.54 | crazygir | eventually I will need it |
| 04:48.06 | crazygir | research is first :) |
| 04:48.23 | jblack | Ok. A good place to start is the asterisk book |
| 04:48.25 | jblack | ~book |
| 04:48.26 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 04:48.26 | crazygir | as well as figuring out what this guy really knows about the current technical aspects |
| 04:48.33 | crazygir | yes |
| 04:48.37 | crazygir | already got it :) |
| 04:48.38 | jblack | I suggest purchasing it, but the pdf is nice. |
| 04:48.55 | crazygir | thanks jblack :) |
| 04:49.24 | crazygir | and I'll definitely come here looking for asterisk hackers when/if the time comes ;) |
| 04:50.06 | jblack | sure. It's hit or miss here on behavior, but generally hit. |
| 04:50.48 | crazygir | hah |
| 04:50.54 | crazygir | ...like anywhere else :P |
| 04:51.06 | jblack | generally better here than elsewhere. |
| 04:51.35 | jblack | [TK]D-Fender: Took the time, months ago, to make sure I was spot on. Other times, particularly if someone isn't showing effort, he can burst a vein |
| 04:51.50 | jblack | That was meant to be a comma, not a colon. sorry tk. |
| 04:51.56 | crazygir | :) I hang out in #openbsd |
| 04:52.09 | jblack | oh, then you've nothing to worry about. |
| 04:52.13 | crazygir | I appreciate being set in line |
| 04:52.18 | crazygir | hahah |
| 04:52.31 | [TK]D-Fender | jblack: Yeah, blatant stupidity and failure to follow step-by-step spoon-fed instructions tends to piss me off :) |
| 04:52.44 | crazygir | understandably.. |
| 04:52.48 | *** join/#asterisk Itiliti (n=Itiliti@c-67-167-82-57.hsd1.il.comcast.net) |
| 04:52.58 | jblack | That guy earlier.. I honestly thought that was your last moment on earth. |
| 04:53.05 | Itiliti | I have local and 800 number did's form my SIp trunk provider. |
| 04:53.11 | [TK]D-Fender | jblack: jblack No, but it could have been HIS :D |
| 04:53.21 | jblack | you? BAH! |
| 04:53.23 | [TK]D-Fender | goes back to honing his sword. |
| 04:53.25 | Itiliti | I cant get the 800 numebr to generate ringtone to the calling partuy |
| 04:53.29 | *** join/#asterisk BBHoss (n=hoss@c-68-62-170-33.hsd1.al.comcast.net) |
| 04:53.54 | jblack | Itiliti: You generally should avoid using it, but check out the R option to dial |
| 04:54.07 | Itiliti | if I call the local did's, I get the ringing tone generated, and they are both coming in on the same context. |
| 04:54.29 | Itiliti | I thought was for outbound only calls? |
| 04:54.44 | Itiliti | nm |
| 04:54.55 | jblack | Dial is dial, no... But I see your problem. |
| 04:55.06 | jblack | You have different ITSPs for the local and the 800 number, I presume? |
| 04:55.28 | Itiliti | it makes no sense. why would it not generat ion an 800 number did, but work on a local did? they are both form the same provider? |
| 04:55.39 | Itiliti | same provider |
| 04:55.52 | jblack | They may not route the same internally. |
| 04:55.58 | Itiliti | true |
| 04:56.06 | jblack | Perhaps you're doing an answer() before you're dialing the internal number. |
| 04:56.32 | Itiliti | no answer until someone picks up the call |
| 04:56.49 | Itiliti | infact, if no one picks up, it never answers at all. |
| 05:00.34 | jblack | oh, you mean the _phone_ doesn't ring. |
| 05:00.47 | jblack | I'd check the ringer volume on the phone. ;) |
| 05:02.28 | *** join/#asterisk CunningPike_ (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 05:04.27 | Itiliti | I have like 10 phones that ring, but I am not performing the answer() in the dialplan until someone picks it up.. |
| 05:06.28 | pkunkra | hmmm |
| 05:06.38 | pkunkra | i would think you would want to answer it right away |
| 05:07.24 | pkunkra | Wait(1); Answer(); Wait(1); is the first thing I do. |
| 05:10.52 | [TK]D-Fender | ok, check-out time. Later all. |
| 05:11.31 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.138) |
| 05:14.46 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 05:16.19 | JerJer | I answer right away then do a Wait(2) |
| 05:16.50 | JerJer | if I am gonna do an IVR - that way the first word doesn't get clipped |
| 05:18.17 | *** join/#asterisk ragix (n=rage@125-238-133-94.broadband-telecom.global-gateway.net.nz) |
| 05:18.28 | ragix | Heya folks. :) |
| 05:26.25 | *** join/#asterisk gones (n=gones@203.193.37.251) |
| 05:30.24 | ragix | Tough crowd. |
| 05:44.04 | *** join/#asterisk [netman] (n=netman@12.Red-88-25-139.staticIP.rima-tde.net) |
| 05:44.06 | *** join/#asterisk pcrane (n=pcrane@202.49.106.158) |
| 05:44.42 | EricL | Anyone still around? |
| 05:45.27 | EricL | I keep getting "Asterisk ended with exit status 127" when I hang up a call. |
| 05:45.30 | EricL | Thoughts? |
| 05:51.43 | Levonk | you might want to see what it says when you do asterisk -vvvvvvvvvvvvvvvvvvvvvg |
| 05:52.23 | EricL | I am starting it with the init scripts. |
| 05:53.11 | Levonk | I don't know what distribution your using, but hopefully there is a variable up top to modify startup args |
| 05:53.29 | EricL | CentOS 5. |
| 05:53.44 | Levonk | I think it has something to do with not being able to load dependent modules |
| 05:53.50 | jblack | you? weren't you the guy that said "nothing has changed" only to indicate that you had changed networks? |
| 05:53.52 | Levonk | e.g. missing file |
| 05:54.33 | jblack | Get the debugging symbols, run it through gdb, and see were it's dying. |
| 05:54.37 | jblack | Or run it through systrace. |
| 05:55.06 | EricL | jblack: Yes. I am that asshole. I didn't change that network, one of my admins did and didn't tell me (found out a few min ago). |
| 05:56.10 | jblack | maybe you should wait for your admins show up, and tell them to get asterisk running. |
| 05:56.20 | EricL | Levonk: Yep, that told me the problem. |
| 05:56.39 | Levonk | Glad that worked |
| 05:56.46 | jblack | are you serious? You hadn't even turned on verbose and debug? |
| 05:57.10 | EricL | Apparently the odbc driver has unresolved symbols in libmysqlclient and mpg123 was then causing * to die. |
| 05:57.13 | jblack | steps away from irc. |
| 05:57.26 | jblack | Yeah. "Nothing changed" |
| 05:57.28 | EricL | jblack: I did but only through the * CLI and it wasn't giving me that informatoin. |
| 05:57.44 | EricL | jblack: This stuff is new. |
| 05:57.50 | EricL | (I am now making changes). |
| 06:00.09 | EricL | My old install had the MySQL addon for the CDR. This install I Am trying (unsuccessfully) to use ODBC. |
| 06:01.16 | jblack | see pages 265 - 270 of the book. |
| 06:02.19 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 06:04.06 | EricL | I must have the old version of the book, my 265-270 is an application reference. |
| 06:04.27 | EricL | checks the index |
| 06:05.18 | jblack | The current edition, 2nd, covers asterisk 1.4 |
| 06:05.46 | jblack | You can get the pdf of the 2nd edition by typing ~book |
| 06:05.57 | *** join/#asterisk sucituanbo (n=blah@c-24-21-121-148.hsd1.mn.comcast.net) |
| 06:06.01 | EricL | ~book |
| 06:06.01 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 06:08.21 | *** join/#asterisk nix8n8 (n=nate@63.162.28.92) |
| 06:10.30 | *** join/#asterisk foexle (n=heiko@router.moltomedia.de) |
| 06:11.58 | *** join/#asterisk gnorbert (n=gnorbert@228-177.static.ew.hu) |
| 06:12.33 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 06:12.39 | nix8n8 | Anyone have an opinion and the best settings for Answering Machine Detection for Asterisk 1.4 or know of a better way for AMD than the buitin application? |
| 06:13.50 | gnorbert | Hi, I have a running Asterisk server, that calls itself to play a sound file in a meetme conference. It works, until I try to call it with 2 linphones from the same computer (But each one is connected). Does somebody have an idea, what can be the problem? |
| 06:14.06 | *** join/#asterisk d-k-t (n=dt@vpn21.witopia.net) |
| 06:15.32 | jblack | make sure the two linphones have different accounts. asterisk and linphone both get screwy with double authentication |
| 06:16.01 | EricL | jblack: I followed everything exactly and still get the errors with setting up ODBC. |
| 06:16.10 | gnorbert | Ok, thanks. I try it. |
| 06:16.35 | EricL | http://www.pastebin.ca/1086856 |
| 06:16.44 | jblack | Maybe someone else will help you. |
| 06:17.29 | *** join/#asterisk rootlogin (n=root@saturn2.franken.de) |
| 06:21.13 | *** join/#asterisk BBHoss (n=hoss@c-68-62-170-33.hsd1.al.comcast.net) |
| 06:27.33 | *** join/#asterisk TOrrIeri (n=torrieri@nelug/crew/torrieri) |
| 06:27.40 | gnorbert | jblack: I call on two ports. The contact and the username is not the same at the two linphones (Actually one linphone with two config files) Shall I change something other too? |
| 06:28.17 | jblack | oh, I dunno. |
| 06:29.18 | gnorbert | jblack: It writes http://pastebin.com/d780ecac3 |
| 06:29.30 | gnorbert | I mean Asterisk writes this. |
| 06:30.19 | *** join/#asterisk CunningPike_ (n=arodgers@vpn.dnv.org) |
| 06:31.00 | nix8n8 | try using two different soft clients? |
| 06:32.15 | gnorbert | I should use this two. Anyway, it works with linphone and Ekiga from the same computer. And also the microphone can be heard with two linphones. |
| 06:32.37 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 06:35.45 | nix8n8 | just curious I don't get the purpose of having soft clients running on an asterisk server calling it self to play a sound file.. |
| 06:37.30 | gnorbert | nix8n8: If you have a better idea for this, I can try that too, just I thought it's the best. Anyway I'm not like an expert so there might be other solution too. |
| 06:38.40 | nix8n8 | ok so what are you trying to accomplish? |
| 06:39.37 | gnorbert | nix8n8: I have to play a sound file in a meetme conference to be heard by everyone. |
| 06:40.24 | gnorbert | The more linphones is because one computer has to be able to connect more conferences too. |
| 06:40.41 | gnorbert | In the same time of course. |
| 06:43.09 | gnorbert | nix8n8: Any other idea? Or anybody with an idea about more linphones? |
| 06:43.37 | nix8n8 | that sounds fun..how do you decide when the sound file gets played? |
| 06:43.55 | nix8n8 | and where is it coming from? |
| 06:45.40 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 06:47.07 | gnorbert | nix8n8: It depends on an other program, that's independent from asterisk and everything else like that. The program chooses a sound file from it's list and plays that. For this I use Asterisk to make other computers able to hear it. |
| 06:49.15 | gnorbert | Any (working) idea? :D |
| 06:49.41 | gnorbert | Either about this or with other solution. |
| 06:53.01 | nix8n8 | I would have to see the code that make it works to really understand what you are doing, but I would try using another soft phone from a different computer and see if it works that way |
| 06:55.17 | nix8n8 | you sure you have your timing set up right? |
| 06:56.54 | gnorbert | nix8n8: It works with other softphone, or with two different softphones. The timing must be good, I believe and when I used Zoiper, it could dial 6 different conferences without any trouble. |
| 06:58.32 | gnorbert | I should use a free softphone, that can be handled via console, that's why I use Linphone (Ekiga didn't work for some reasons). |
| 07:01.05 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 07:01.07 | *** join/#asterisk bazilek (n=bazil@mail.generation-p.com) |
| 07:04.02 | *** part/#asterisk baliktad (i=baliktad@c-24-16-27-4.hsd1.mn.comcast.net) |
| 07:06.41 | *** join/#asterisk h-idrisi (n=h-idrisi@212.103.170.132) |
| 07:06.41 | *** join/#asterisk d-k-t-2 (n=dt@vpn21.witopia.net) |
| 07:08.05 | nix8n8 | are you using a soft phone to play the sound file? |
| 07:09.12 | gnorbert | No, a .call file. |
| 07:09.26 | gnorbert | I use softphones to hear the sounds. :) |
| 07:09.38 | gnorbert | And of course to speak |
| 07:09.56 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-e9483ecfea8859c4) |
| 07:10.26 | nix8n8 | so what sets off the call file after everyone is in the conferance? |
| 07:13.14 | gnorbert | nix8n8: I guess I didn't fully understand this last question. The call file doesn't depend on any softphones. It depends only on the other program, that makes those call files in the directory of Asterisk. |
| 07:13.41 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:14.28 | nix8n8 | so this program calls a couple people then plays a sound file? |
| 07:15.18 | gnorbert | No, a couple people ar in the conference, where they want to be and this program plays a sound file in a conference. If somebody is in that conference, then he/she hears it. |
| 07:17.52 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:19.20 | foexle | hi guys, i've a question, it's possible a call forwarding with asterisk and ISDN or VoIP without to make a new connection ? i hope u understand :> my engl isn't good :( |
| 07:21.29 | foexle | caller <> Server <> Call center (false) --->> caller <> Server <> Server says no the caller must to call a call center, so caller <> call center |
| 07:23.25 | *** join/#asterisk h-idrisi (n=h-idrisi@212.103.170.132) |
| 07:24.34 | nix8n8 | well good luck gnorbert I'm going to take a nap |
| 07:24.59 | gnorbert | nix8n8: Thanks, Have a nice nap. :) |
| 07:28.12 | *** join/#asterisk baliktad (i=baliktad@c-24-16-27-4.hsd1.wa.comcast.net) |
| 07:30.21 | gnorbert | Does somebody have an idea, what can be the problem, if I can play sounds with a call file in a meetme conference, but if two linphones from the same computer is connected to the same server (but different conferences), it writes this: http://pastebin.com/d780ecac3 |
| 07:30.30 | gnorbert | I mean Asterisk writes this. |
| 07:37.26 | *** join/#asterisk BBHoss (n=hoss@c-68-62-170-33.hsd1.al.comcast.net) |
| 07:38.08 | gnorbert | I use the same linphone, just two different config files. Username, contact and the used ports are changed. |
| 07:39.21 | *** join/#asterisk ToTo (n=ToTo@209.8.41.141) |
| 08:00.18 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 08:07.55 | gnorbert | Does anybody have any idea? |
| 08:11.08 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 08:13.05 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
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| 08:21.41 | *** part/#asterisk thomas (i=tm@tm.muc.de) |
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| 08:58.49 | gones | I have a project about asterisk |
| 08:58.57 | creativx | nice. |
| 09:01.09 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 09:02.13 | gones | can somebody help me ? |
| 09:02.20 | gones | with me ! |
| 09:02.35 | creativx | help you with yourself? |
| 09:03.22 | gones | together with me . |
| 09:07.07 | gr0mit | gones, protocol is - you ask the question, and those who feel able to assist will do so. |
| 09:07.25 | Strom_M | gr0mit: shush, you |
| 09:07.34 | Strom_M | you're ruining the entertainment value :D |
| 09:07.39 | gr0mit | hehehe!!!! |
| 09:07.43 | *** part/#asterisk EricL (n=eric@jarbeeg.chal.net) |
| 09:07.55 | *** join/#asterisk AMUG (n=junky@modemcable156.137-20-96.mc.videotron.ca) |
| 09:08.22 | gones | gr0mit:yeah |
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| 09:10.18 | *** join/#asterisk StooJ (n=stooj@johnston37.plus.com) |
| 09:12.59 | gones | Strom_M, how about ? |
| 09:15.42 | *** join/#asterisk wonderworld (n=ww@ip-62-143-163-199.hsi.ish.de) |
| 09:21.06 | *** join/#asterisk arooni (n=arooni__@c-67-168-26-198.hsd1.wa.comcast.net) |
| 09:21.10 | arooni | how do i shut down asterisk? |
| 09:21.17 | arooni | and prevent it from running on startup? |
| 09:23.38 | yang | arooni: "stop gracefully" |
| 09:23.43 | yang | in CLI |
| 09:24.09 | arooni | ok how do i prevent it from starting up on default on a debian box |
| 09:24.53 | fors1 | arooni: remove files in /etc/rcX.d/ |
| 09:25.19 | yang | arooni: you need to edit sysinitrc get sysvconfig package |
| 09:25.33 | yang | and then (un)select services |
| 09:26.42 | fors1 | arooni: default runlevel on debian is 2, so if you remove any files named "asterisk" in /etc/rc2.d/ asterisk will not start on boot |
| 09:34.39 | *** join/#asterisk davix (i=bbbbbbb@89-138-156-39.bb.netvision.net.il) |
| 09:36.19 | davix | will asterisk be able to give me an easy free solution (other then needed hardware) to create a home pbx? (internal house calls, incoming calls on a regular line and directing it to the proper digital line, etc') |
| 09:36.25 | *** join/#asterisk torrikft (n=afj@m85-94-184-212.andorpac.ad) |
| 09:36.45 | Strom_M | davix: yes |
| 09:36.52 | torrikft | hey all, what dmtfmode is the correct one to use with g729? |
| 09:37.05 | *** join/#asterisk trafim (n=reallyma@212.200.84.70) |
| 09:37.14 | Strom_M | torrikft: depends on what your provider and/or phones want |
| 09:37.45 | Strom_M | the dtmfmode is independent of the voice codec |
| 09:37.50 | torrikft | with ulaw im using inbound and it works correctly |
| 09:37.59 | Strom_M | you mean inband? |
| 09:38.01 | *** join/#asterisk defswork (n=andy@mx1.3gcomms.co.uk) |
| 09:38.01 | torrikft | with g729 i get double dialing |
| 09:38.08 | torrikft | yes sorry inband |
| 09:38.12 | davix | thanks Strom_M, will I have to program it myself or does it come configured for basic work and all i need to get is a computer and some sort of extenstions card? |
| 09:38.15 | Strom_M | well, yeah, inband is the exception -- it only works with g711 |
| 09:38.32 | Strom_M | davix: you'll have to configure it yourself |
| 09:38.34 | Strom_M | see this |
| 09:38.36 | Strom_M | ~book |
| 09:38.36 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 09:38.39 | Strom_M | ^^^^ |
| 09:38.42 | torrikft | with g729 tried auto, info and rfc2833 and get double dialing |
| 09:39.00 | torrikft | every digit twice |
| 09:39.01 | Strom_M | torrikft: well, what does your provider and/or phone want? |
| 09:39.03 | davix | looking, thanks Strom_M |
| 09:39.32 | torrikft | its only an issue with callback |
| 09:39.46 | torrikft | dialout rfc2833 works fine |
| 09:40.09 | torrikft | lets see what nokia uses |
| 09:40.22 | Strom_M | what do you mean "callback"? |
| 09:40.28 | *** join/#asterisk admin0 (n=admin@bb121-7-229-228.singnet.com.sg) |
| 09:40.32 | Strom_M | what kind of terminal equipment are you using? |
| 09:41.15 | torrikft | ok i´ll explain the issue with more detail |
| 09:41.28 | torrikft | i can dial out and in and my web callback works fine |
| 09:41.51 | torrikft | but when doing cid callback, the system asks me to dial the number i want to call plus # sign |
| 09:42.01 | torrikft | and unless i use g711 |
| 09:42.11 | Strom_M | what is this "cid callback" thing? |
| 09:42.28 | Strom_M | i can't just divine what your setup looks like, you know |
| 09:42.37 | torrikft | yeah |
| 09:42.43 | torrikft | caller id callback |
| 09:42.52 | torrikft | i identify the caller, hangup and call him back |
| 09:43.02 | torrikft | ivr asks him to enter number to dial |
| 09:43.22 | torrikft | caller can be on any type of mobile phone |
| 09:43.45 | Strom_M | does the problem occur when using the service with wireline phones? |
| 09:44.01 | torrikft | let me test |
| 09:48.28 | torrikft | yeah same issue |
| 09:48.35 | torrikft | with wireline |
| 09:48.47 | Strom_M | i assume you're using an ITSP? |
| 09:49.17 | torrikft | yeah |
| 09:49.22 | torrikft | no pstn on that box |
| 09:49.32 | Strom_M | which ITSP? |
| 09:49.38 | torrikft | altecom.net |
| 09:49.50 | *** part/#asterisk davix (i=bbbbbbb@89-138-156-39.bb.netvision.net.il) |
| 09:50.12 | torrikft | but im able to pass the dtmf coding to them from my extensions.conf |
| 09:50.25 | Strom_M | what if you use rfc2833 with g711? |
| 09:50.34 | torrikft | i'll give that a try |
| 09:52.22 | torrikft | double dialing as well |
| 09:53.06 | torrikft | 00376112233 turns into 0000334466111122223333 |
| 09:53.26 | Strom_M | and, just to clarify, it works properly when you use inband? |
| 09:54.32 | torrikft | yes |
| 09:54.34 | torrikft | it does |
| 09:55.28 | Strom_M | does rfc2833 work properly on direct-dialed inbound and outbound calls? |
| 09:55.34 | torrikft | yes |
| 09:56.10 | Strom_M | are your completely sure? |
| 09:56.28 | Strom_M | s/your/you/ |
| 09:56.30 | *** join/#asterisk _Krieger_ (n=krieger@193.39.118.158) |
| 09:56.38 | torrikft | yes 100% |
| 09:56.45 | Strom_M | ok |
| 09:56.59 | Strom_M | pastebin your extensions.conf and sip.conf and your callfile |
| 09:57.01 | Strom_M | ~pb |
| 09:57.02 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines 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 |
| 09:57.57 | _Krieger_ | is it possible to suppress (at asterisk side!) tone sounds from a hardphone, produced at input to Read()? |
| 09:58.47 | Strom_M | _Krieger_: ? |
| 09:58.51 | Strom_M | why? |
| 09:59.42 | _Krieger_ | Strom_M: i have such request :) |
| 10:00.41 | _Krieger_ | i mean is it possible to change in a C code |
| 10:01.06 | Strom_M | _Krieger_: I don't understand what you're trying to do. |
| 10:01.16 | Strom_M | could you please explain it in more detail? |
| 10:03.08 | _Krieger_ | when Read() is processed in dialplan, and user inputs sth, every keystroke produces a tone sound heard by user himself. the request is to avoid these sounds heard by user |
| 10:03.24 | Strom_M | no, you can't do that |
| 10:03.28 | Strom_M | those tones are generated by the phone |
| 10:03.35 | Strom_M | asterisk has nothing to do with it |
| 10:04.56 | _Krieger_ | i have the same opinion. |
| 10:05.28 | Strom_M | that's not an opinion. |
| 10:07.27 | yang | For enabling the logging in asterisk is it only needed to modify logger.conf or also any other file? |
| 10:08.02 | Strom_M | yang: just logger.conf |
| 10:09.09 | yang | strange its not logging for me the CLI with full => notice,warning,error,debug,verbose |
| 10:09.59 | Strom_M | dogballs |
| 10:11.05 | *** part/#asterisk gones (n=gones@203.193.37.251) |
| 10:12.23 | yang | strange |
| 10:12.41 | _Krieger_ | yang, do logger reload |
| 10:12.46 | _Krieger_ | aybe |
| 10:13.34 | yang | yes i did, lets see now |
| 10:20.21 | gnorbert | Hi, I have an * server, that works well until I don't want to connect to it with 2 linphones (technically one linphone, from two config files). Then it doesn't play the sound from a .gsm file, that would be solved by call files. (call file dials in the conference and plays the sound there) Anyway it works with two different softphones (ekiga&linphone) and if I connect with two linphones, it writes that both are connected. When I try to move th |
| 10:20.21 | gnorbert | e call file in the * directory, it writes this: http://pastebin.com/d780ecac3 |
| 10:35.16 | gnorbert | Any idea from anybody? :) |
| 10:49.44 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:52.19 | *** join/#asterisk webman (n=adamg@200.179.233.220.exetel.com.au) |
| 10:52.37 | gr0mit | Am seeing much oddness. I am running Asterisk 1.2.29-BRIstuffed-0.3.0-PRE-1y-s in Venezuela attached to a PBX using Euro-ISDN with a Digium pri card. I see the IAX HANGUP, asterisk thinks call ended, but the HANGUP never gets sent out on the E1 to the PBX. Anyone else seen this? |
| 10:53.07 | gr0mit | this is only for outbound calls, inbound works fine. |
| 10:53.33 | webman | is it possible from the dialplan to see either the SIP username or the SIP IP address of the remote end ? (This is for incoming calls only) |
| 10:55.57 | tzafrir_laptop | gr0mit, can you please provide a pri debug? |
| 11:00.42 | gr0mit | sure. |
| 11:00.59 | webman | or is there a document somewhere that shows all SIP details which are available as channel variables ? |
| 11:01.27 | *** join/#asterisk anebi (n=anebi@87-126-80-161.btc-net.bg) |
| 11:01.37 | anebi | hi |
| 11:01.49 | anebi | i get warning messages in asterisk log like this: app_voicemail.c: Failed to obtain database object for 'asterisk'! |
| 11:01.56 | anebi | what can i do to solve this problem? |
| 11:02.09 | anebi | i don't use voicemail and recording at all |
| 11:02.39 | *** join/#asterisk The-Bat (n=karthik@203.199.114.33) |
| 11:03.39 | gr0mit | tzafrir_laptop, here it is http://www.pastebin.ca/1087028 |
| 11:04.50 | tzafrir_laptop | anebi, have you built app_voicemail with odbc support? |
| 11:05.14 | anebi | i think yes, i don't remember really |
| 11:05.22 | gr0mit | tzafrir_laptop, any clues here? |
| 11:05.52 | tzafrir_laptop | gr0mit, reading |
| 11:05.53 | anebi | tzafrir_laptop: probably i can disable this from some of the cofiguration files or not? |
| 11:06.50 | gr0mit | have tried Euroisdn, 4ess and 5ess |
| 11:06.51 | tzafrir_laptop | gr0mit, so nothing gets sent |
| 11:06.54 | gr0mit | exactly, tzafrir_laptop |
| 11:07.01 | tzafrir_laptop | don't they work there with euroisdn? |
| 11:07.15 | gr0mit | switch is supposedly euroisdn |
| 11:07.33 | *** part/#asterisk trafim (n=reallyma@212.200.84.70) |
| 11:07.59 | tzafrir_laptop | so nothing in the pri trace. Next thing to get some clues: debug logging |
| 11:08.39 | tzafrir_laptop | log "debug" and "verbose" (e.g. the sample "full" from logger.conf) and set debug to e.g. 10 |
| 11:09.06 | *** join/#asterisk h-idrisi (n=h-idrisi@212.103.170.132) |
| 11:09.33 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:10.55 | *** join/#asterisk zydoon (n=zydoon@213.150.170.26) |
| 11:11.01 | gr0mit | tzafrir_laptop, doing that now |
| 11:11.47 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 11:11.56 | *** part/#asterisk zydoon (n=zydoon@213.150.170.26) |
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| 11:17.04 | *** join/#asterisk acron17 (n=joe@p5089D53F.dip.t-dialin.net) |
| 11:17.29 | acron17 | hi there |
| 11:18.02 | acron17 | on our company we have an asterisk server for testing |
| 11:18.26 | acron17 | it was working very well, but someone seems to have messed up the conf |
| 11:19.13 | Strom_M | well, it's a good thing you have backups then, isn't it |
| 11:19.20 | acron17 | and now it's still working, but the "echo-service" and "time-service" the dialplans for having an echo or asterisk saying the time aren't working anymore |
| 11:19.37 | acron17 | well, i would be, wouldn't it? |
| 11:20.09 | acron17 | Strom_M: but life often isn't that way :-/ |
| 11:21.12 | Strom_M | this is the point where I point out what a stupid decision it was not to make backups |
| 11:21.17 | acron17 | i wonder if anyone can give me some hints, why astrisk is saying 404 when a dialplan for the number exists... |
| 11:21.20 | Strom_M | and then go out for a walk |
| 11:22.02 | *** join/#asterisk mattwj2002 (n=Matt@c-76-17-132-205.hsd1.mn.comcast.net) |
| 11:22.10 | mattwj2002 | hi guys |
| 11:22.29 | acron17 | Strom_M: i've done that before as well. still the admin has left the dev-server to the programers, no backup is done... |
| 11:24.19 | *** join/#asterisk PepOSX (n=angeldav@190.72.130.106) |
| 11:25.05 | gr0mit | tzafrir_laptop, here is the hangup part of the trace http://www.pastebin.ca/1087052 |
| 11:25.44 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 11:26.58 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 11:27.32 | gr0mit | tzafrir_laptop, bbl - gotta have food before the cafeteria closes. |
| 11:29.07 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 11:30.17 | pputman | I'm downloading and compiling asterisk 1.4.21.1 and for some reason after a make install it starts fine, but /usr/lib/asterisk/modules is empty? /etc/asterisk/asterisk.conf has it listed as the modules directory, so it looks like somehow asterisk is loading some modules, but I can't find them anywhere, did anything change relating to this in 1.4.21? |
| 11:31.34 | mattwj2002 | hey guys I have a question |
| 11:32.03 | mattwj2002 | what causes voice quality calls on a local asterisk box |
| 11:32.10 | *** join/#asterisk cplx (n=cplx@ettamo.lnk.telstra.net) |
| 11:32.24 | mattwj2002 | *qualiy issues |
| 11:32.50 | mattwj2002 | even a simple echo test sounds bad |
| 11:32.58 | mattwj2002 | I am getting something that sounds like static |
| 11:33.30 | cplx | hi guys.. i have a CallManager router talking to a Asterisk box (SIP Trunk configuration) - i make calls sometimes they get through and others dont (all to the same number) it just goes ...beep.beep.beep unknown number on the cisco ip phone - is this someting to do with the sip trunk configuration? retry invite? retry response? retry bye? retry cancel? timers trying ? values??? |
| 11:35.27 | gr0mit | tzafrir_laptop, back. any ideas?? |
| 11:35.54 | webman | mattwj2002: are you using voip phones or zap ? Please provide just a little information for us ? |
| 11:36.14 | mattwj2002 | sure |
| 11:36.37 | mattwj2002 | I am using a softphone X-lite to an Asterisk box on my local lan |
| 11:36.46 | mattwj2002 | my Asterisk box is slow |
| 11:36.53 | mattwj2002 | 600 MHz maybe |
| 11:37.01 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 11:37.26 | mattwj2002 | and I am using wireless on the computer that has the softphone (lack of QoS probably) |
| 11:37.39 | webman | mattwj2002: so all the same problems that can be had by using a slow asterisk box, network faults, and or workstation faults, and/or mic/speaker faults .... try to isolate each of those issues as much as possible |
| 11:38.16 | *** part/#asterisk anebi (n=anebi@87-126-80-161.btc-net.bg) |
| 11:38.46 | mattwj2002 | okay |
| 11:38.48 | mattwj2002 | :) |
| 11:39.49 | gr0mit | mattwj2002, start by using a network cable instead of wireless, i.e. eliminate things one by one |
| 11:40.05 | mattwj2002 | okay cool |
| 11:40.14 | gr0mit | what codec are you using? |
| 11:40.19 | mattwj2002 | G711 |
| 11:40.26 | gr0mit | ok good |
| 11:41.07 | mattwj2002 | yeah ulaw to be specific |
| 11:41.13 | mattwj2002 | well I have to go to work |
| 11:41.16 | mattwj2002 | thanks you guys :) |
| 11:41.19 | mattwj2002 | have a good day |
| 11:41.35 | *** part/#asterisk mattwj2002 (n=Matt@c-76-17-132-205.hsd1.mn.comcast.net) |
| 11:47.05 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 11:47.26 | XnOSX | anybody know any softphone in flash for web? |
| 11:47.32 | *** join/#asterisk scampbell (n=scampbel@199.105.195.156) |
| 11:48.28 | *** join/#asterisk dickyjoe (n=rlane@183.69.233.220.exetel.com.au) |
| 11:48.32 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:49.08 | dickyjoe | ~centos52bug |
| 11:49.09 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889 |
| 11:49.24 | tzafrir_laptop | XnOSX, flash and UDP are not the best of friends, from what I know |
| 11:49.38 | *** join/#asterisk C4colo (n=DJpyro@66.185.107.193) |
| 11:50.29 | dickyjoe | hi all i am bitten by the zaptel centos bug, how do i uncheck "xpp" in menuselect |
| 11:50.42 | XnOSX | tzafrir_laptop, ok! do you know other? in java or other language something is good for me too! |
| 11:51.17 | C4colo | I see we are up to beta9 on asterisk, is there an estimated release date for it yet? |
| 11:51.24 | C4colo | on asterisk 1.6 I mean |
| 11:52.05 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 11:52.24 | dickyjoe | ~centos52bug |
| 11:52.25 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889 |
| 11:53.11 | LemensTS | ~mornin |
| 11:53.24 | cplx | hi guys.. i have a CallManager router talking to a Asterisk box (SIP Trunk configuration) - i make calls sometimes they get through and others dont (all to the same number) it just goes ...beep.beep.beep unknown number on the cisco ip phone - is this someting to do with the sip trunk configuration? retry invite? retry response? retry bye? retry cancel? timers trying ? values??? |
| 11:54.35 | dickyjoe | hello? |
| 11:54.36 | C4colo | cplx: run tcpdump on the asterisk box and save it to a .pcap file, then look at it in wireshark |
| 11:54.53 | C4colo | dickyjoe: are you using menuselect when you build zaptel? |
| 11:55.31 | dickyjoe | i just issue make clean , make, make, make install |
| 11:55.52 | C4colo | cplx: I would use something like: tcpdump -s 2048 -w /tmp/sipdebug1234.pcap port 5060 -i eth0 |
| 11:56.28 | C4colo | the sftp that file off to a system with wireshark installed, open the .pcap file and look through the messages |
| 11:56.41 | C4colo | dickyjoe: make menuselect, make, make install, make configure |
| 11:56.48 | tzafrir_laptop | dickyjoe, there's also a simple patch to that issue in that link |
| 11:57.13 | *** join/#asterisk shtoom (n=shtoom@121.246.167.147) |
| 11:57.21 | dickyjoe | i tried to run the patch without luck |
| 11:57.22 | C4colo | during menuselect you can go through and use the spacebar/enter/etc to traverse the menu, [*] means it will be built/installed |
| 11:57.31 | dickyjoe | ok |
| 11:57.52 | dickyjoe | i will have a look see |
| 11:58.11 | tzafrir_laptop | patch -p0 <'http://bugs.digium.com/file_download.php?file_id=19260&type=bug' |
| 11:58.23 | tzafrir_laptop | and build as usual |
| 11:59.19 | C4colo | is that bug on the centos 5.2 side or in the TDM driver? |
| 11:59.22 | dickyjoe | thanks all |
| 11:59.31 | dickyjoe | i wasn't aware of make menuselect |
| 11:59.43 | C4colo | seems if it works on 5.1 and not 5.2 it would be on the centos side |
| 11:59.56 | tzafrir_laptop | C4colo, the bug is because CentOS 5.2 backported a new feature from kernel 2.6.19 to what it calls "2.6.18" |
| 12:00.19 | C4colo | interesting |
| 12:00.31 | C4colo | are there other drivers this could effect? |
| 12:00.42 | C4colo | I was in the process of upgrading all of my centos5 servers to 5.2 |
| 12:00.53 | cplx | C4colo - i don't have direct access to the asterisk box at the moment, anything you can think of that would be doing it? the first 2 calls i make go through fine, the next is 'declined, the fourth is 'unauthorized' |
| 12:01.01 | cplx | C4colo - according to the CM debug logs |
| 12:01.21 | C4colo | what version of asterisk? |
| 12:01.28 | tzafrir_laptop | It's a build issue. Those things will only potentially byte you when building the software, not when running |
| 12:01.58 | C4colo | cplx: in these cases it is nearly impossible to tell what the two boxes are saying to each other without actually capturing it and looking at it |
| 12:02.01 | tzafrir_laptop | This specific issue is only with the xpp driver |
| 12:02.13 | dickyjoe | if i only run a tdm400 digium card can i safely unselect everything else? |
| 12:02.27 | C4colo | everything else? |
| 12:02.29 | cplx | Asterisk 1.4.21.2 |
| 12:02.32 | C4colo | you mean the other devices? |
| 12:02.39 | tzafrir_laptop | C4colo, what modules do you have? FXS? FXO? |
| 12:02.40 | dickyjoe | in modulke selection |
| 12:02.47 | cplx | C4colo - hmm... yeah, but anything to do with register values, time out values on the callmanager etc? |
| 12:02.47 | C4colo | if you don't plan on using them yes |
| 12:02.55 | tzafrir_laptop | the modules you need: wctdm and zaptel |
| 12:03.13 | C4colo | it could be sip timer related |
| 12:03.20 | C4colo | doubtful though |
| 12:03.24 | cplx | C4colo - sip timer, on my end or the asterisk? |
| 12:03.30 | C4colo | yes |
| 12:03.31 | C4colo | lol |
| 12:03.40 | dickyjoe | i mean i will use iax2 and sip and i would like to look into bluetooth - its a home setup |
| 12:03.46 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
| 12:04.03 | C4colo | on the zaptel menuselect if you are not going to use the other cards you don't need to build the drivers |
| 12:04.14 | C4colo | not that it hurts to build them, they just sit there if you aren't using them |
| 12:04.24 | dickyjoe | ok |
| 12:04.36 | C4colo | cplx: do you have ssh access to the asterisk box? |
| 12:04.41 | dickyjoe | so maybe just unselect xpp |
| 12:04.44 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.141) |
| 12:04.54 | C4colo | yea |
| 12:05.36 | dickyjoe | what is pci radio all about? |
| 12:06.06 | C4colo | good question |
| 12:06.08 | C4colo | I don't know |
| 12:06.39 | dickyjoe | ham radio maybe? |
| 12:06.50 | C4colo | that is app_rpt.so |
| 12:07.27 | tzafrir_laptop | right. There are a few words about it in zaptel.conf.sample |
| 12:08.04 | dickyjoe | ok |
| 12:08.16 | tzafrir_laptop | Jim Dixon is the guy who actually started the Zapata Telephony projecy |
| 12:08.43 | C4colo | might be Wi-Fi or a DECT product .. or maybe for wireless/mobile phone devices? |
| 12:08.46 | *** join/#asterisk DSpair (n=D-Spare@163.muaa.syrc.chcgil24.dsl.att.net) |
| 12:08.52 | DSpair | Good morning all... |
| 12:09.18 | dickyjoe | is video hard to channel through asterisk? |
| 12:10.01 | DSpair | I'm having a problem with an automated phone service. When I dial this AT&T Account IVR, it answers before there is ever a single ring and so asterisk never recognizes the channel as completely up and after the timeout disconnects. |
| 12:10.13 | C4colo | I tried once to do video, gave up after a few hours |
| 12:10.26 | *** join/#asterisk joobie (n=joobie@joobie.org) |
| 12:10.32 | joobie | hey guys |
| 12:10.44 | joobie | anyone know *the* cheapest place to buy polycom 320's ? |
| 12:10.45 | DSpair | hello joobie |
| 12:11.01 | dickyjoe | joobie: back of the polycom truck |
| 12:11.01 | DSpair | joobie, I would probably search on Froogle. |
| 12:11.05 | *** join/#asterisk ManxPower (n=manxpowe@164.sub-70-222-104.myvzw.com) |
| 12:11.11 | DSpair | giggles. |
| 12:11.47 | DSpair | Mornig Manx... |
| 12:12.01 | ManxPower | 'morning |
| 12:12.12 | joobie | dickyjoe, :) |
| 12:12.19 | joobie | DS good idea :) ta |
| 12:12.31 | DSpair | Is there a way to make all calls out on a PRI experience a ring, no matter what the distant end does? |
| 12:13.12 | dickyjoe | thanks everyone for helping me out |
| 12:13.23 | dickyjoe | all compiled sweet and sound |
| 12:14.07 | ManxPower | DSpair: That is what the "r" option is for. But you won't like it. |
| 12:14.34 | ManxPower | well, the "r" option to Dial AND priindication=inband (make sure that's the correct spelling) in zapata.conf |
| 12:15.05 | joobie | im from AU hey |
| 12:15.08 | Strom_M | DSpair: why would you want to do that? |
| 12:15.13 | joobie | have you guys heard of onsale.com ? |
| 12:15.18 | joobie | http://www.onsale.com/shop/detail.aspx?dpno=7238939&store=onsale&source=bwbfroogle |
| 12:15.22 | joobie | looks like a US based site.. never heard of them |
| 12:15.29 | ManxPower | Strom_M: Because he doesn't know how bad doing that is. |
| 12:15.31 | joobie | thinking if i should buy 20 phones from them.. |
| 12:15.35 | joobie | anyone know of them? |
| 12:15.41 | ManxPower | ~phones |
| 12:15.41 | jbot | hmm... phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. places like such as |
| 12:15.56 | dickyjoe | joobie try techtopia |
| 12:16.22 | joobie | they are reputiable dicky? |
| 12:16.39 | dickyjoe | onlinr store |
| 12:16.40 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:16.47 | dickyjoe | i bought a few things through them |
| 12:16.55 | joobie | ahh |
| 12:16.59 | joobie | i used voipwarehouse.com.au |
| 12:17.01 | joobie | they aint bad |
| 12:17.01 | dickyjoe | austech were ok too |
| 12:17.11 | *** join/#asterisk shtoom (n=shtoom@123.176.41.46) |
| 12:17.22 | joobie | thanks dicky |
| 12:17.27 | dickyjoe | np |
| 12:17.31 | shtoom | Hi, I've a 6months old asterisk installation, which was working fine. for 15 days audio goes blank in the call for 6 secs or so, what might be the problem ? |
| 12:18.13 | ManxPower | shtoom: reboot |
| 12:18.14 | dickyjoe | shtoom: what sort of call connection? |
| 12:18.34 | ManxPower | shtoom: also canreinvite=no in sip.conf sections |
| 12:18.59 | acron17 | i've a dialplan for a "time-service" in extension.conf (see: http://rafb.net/p/3nPnsn58.html) |
| 12:19.06 | [TK]D-Fender | "May car doesn't always start when I turn the key, what could be the problem?" |
| 12:19.25 | dickyjoe | broken fret? |
| 12:19.43 | *** join/#asterisk angryuser (n=sldf@88.140.123.21) |
| 12:19.51 | acron17 | but calling 12@the-asterisk-server gives me 404. can someone point me some possible issues iout? |
| 12:20.18 | joobie | http://www.techtopia.com.au/product_info.php/products_id/1273 <-- that is interesting.. are there asterisk boxes that are solid state? i was thinking about investing around 2k for an asterisk box.. but looks like these all in ones can do it?? is that the case? If so, what's a good model for a sip uplink and sip to the lan?? |
| 12:20.32 | [TK]D-Fender | acron17: pastebin the sip debug of your failed attempt |
| 12:20.34 | [TK]D-Fender | ~pb |
| 12:20.34 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines 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 |
| 12:20.36 | [TK]D-Fender | ^^^^^^^^^^ |
| 12:20.54 | acron17 | [TK]D-Fender: thx. i will... |
| 12:21.47 | dickyjoe | joobie, i run a CFcard in my homebrew asterisk box, works ok most times |
| 12:22.27 | joobie | dickyjoe, i dont need cards atm.. just LAN cards really.. well 2.. one for uplinka nd one for lan going to a switch |
| 12:22.44 | joobie | but im wondering, rather than build a homebrew box for 2k.. if there's a better, solid state solution that can be bought |
| 12:22.47 | joobie | with asterisk |
| 12:23.03 | joobie | i saw that one above, it's 1.4k.. looks aite |
| 12:23.45 | dickyjoe | no idea |
| 12:23.54 | dickyjoe | looks pretty |
| 12:23.57 | DSpair | Strom_M, I am calling this automated telephone server from AT&T to check on our bill and the system answers before there is ever a ring. Because of that, asterisk never sees the connection as completely "UP" and so it hangs up after the timeout. |
| 12:24.15 | DSpair | s/server/service/ |
| 12:24.19 | [TK]D-Fender | joobie: What kind of interfaces will it need? |
| 12:24.30 | Strom_M | DSpair: um...what kind of connection are you calling it with? |
| 12:24.40 | joobie | Fender, RJ45 is fine |
| 12:24.43 | DSpair | Strom_M, PRI with national signaling |
| 12:25.01 | joobie | * connects as a sip client to my sip tisp.. |
| 12:25.04 | [TK]D-Fender | joobie: Just build an embedded box like the Alix, Soekris, etc... |
| 12:25.08 | joobie | phones in the LAN connect via sip to the * box |
| 12:25.16 | Strom_M | DSpair: I don't know what the hell you're smoking; asterisk should get answer supervision from the PRI |
| 12:25.33 | acron17 | <PROTECTED> |
| 12:25.35 | DSpair | Strom_M, I would think so as well, but it's not. |
| 12:25.42 | Strom_M | DSpair: what's teh number? |
| 12:25.53 | DSpair | I can pastebin a PRI debug if you like. |
| 12:26.04 | Strom_M | just PM me the telephone number. |
| 12:26.06 | joobie | Fender, do those boxes have enough processor speed? |
| 12:26.06 | DSpair | 18002357524 |
| 12:26.14 | [TK]D-Fender | acron17: the ENTIRE CALL. Stop trimming things if you expect to have this problem solved. |
| 12:26.18 | joobie | they look like very limited boards |
| 12:26.24 | [TK]D-Fender | joobie: For the basics yeah. |
| 12:26.29 | joobie | ahh |
| 12:26.35 | joobie | i want to do call recording for 20 calls at once |
| 12:26.36 | Strom_M | DSpair: it never supervises in the first place |
| 12:26.40 | joobie | + chanspy() |
| 12:26.47 | Strom_M | DSpair: you need to call AT&T and complain. |
| 12:27.00 | [TK]D-Fender | joobie: can probably handle a small MeetMe as long as you use an easy codec. AQvoid transcoding though |
| 12:27.34 | DSpair | Strom_M, Yeah, calling AT&T will really not help.... They don't give a crap. I've tried to complain to them about this a few times. |
| 12:28.01 | [TK]D-Fender | acron17: Mind you this should be a tip-off : "Jul 30 14:21:51 WARNING[1666]: chan_sip.c:2017 create_addr: No such host: 12" |
| 12:28.07 | DSpair | Strom_M, In the meantime, my boss is pissed because it works from his cell phone, but not from his desk. |
| 12:28.20 | acron17 | <PROTECTED> |
| 12:28.28 | Strom_M | DSpair: well, if you force ringing on the circuit, you will just hear ringing until the call is torn down. |
| 12:28.40 | DSpair | Strom_M, Can you imagine some way to "fudge" this? |
| 12:28.53 | Strom_M | DSpair: who is your PRI provider? |
| 12:28.55 | DSpair | Strom_M, Gotcha. |
| 12:29.01 | ManxPower | DSpair: It is not caused by the far end answering right away. |
| 12:29.03 | DSpair | Strom_M, Nuvox |
| 12:29.08 | joobie | thanks fender |
| 12:29.11 | Strom_M | DSpair: get a real telephone company. |
| 12:29.12 | acron17 | <PROTECTED> |
| 12:29.15 | [TK]D-Fender | acron17: Looking for 12 in default (domain devel.ac.axxteq.de) SIP/2.0 404 Not Found <- Dialplan error. Go fix it |
| 12:29.24 | DSpair | Strom_M, Sorry, stuck with the contract. |
| 12:29.54 | Strom_M | DSpair: well, then it sucks to be you, doesn't it? call nuvox and complain that they're tearing your unsupervised calls down too quickly. |
| 12:30.06 | ManxPower | DSpair: I recommend actually trying to diagnose the issue. What priindication= setting do you have? |
| 12:30.47 | ManxPower | DSpair: I suspect Strom_M's comments are correct and there is nothing you can do on the Asterisk side. HOWEVER, I won't give up on the Asterisk side, there are still options and things we can try. |
| 12:30.50 | DSpair | ManxPower, Which config file would that be in? |
| 12:31.09 | ManxPower | DSpair: /etc/asterisk/zapata.conf same place as all PRI settings in Asterisk |
| 12:31.23 | DSpair | ManxPower, That's where I looked, no setting for that option. |
| 12:31.36 | acron17 | <PROTECTED> |
| 12:31.38 | ManxPower | DSpair: when the far end hangs up, what is the HANGUPCAUSE? |
| 12:31.55 | DSpair | The far end doesn't hang up. Asterisk does. |
| 12:32.06 | Strom_M | DSpair: um, i doubt that. |
| 12:32.07 | [TK]D-Fender | acron17: Its looking for "12" in [default] and there is no match. Go fix it. |
| 12:32.38 | DSpair | Strom_M, Because asterisk never sees this as a connected call, it hangs up after 60 seconds. |
| 12:33.04 | Strom_M | DSpair: no, I suspect that's your provider tearing the unsupervised call down after 60 seconds |
| 12:33.07 | ManxPower | [root@bourbon configs]# grep priindication zapata.conf.sample |
| 12:33.07 | ManxPower | ; priindication = outofband |
| 12:33.11 | ManxPower | looks like it's in MY system. |
| 12:33.14 | Strom_M | what does the CLI say? |
| 12:33.30 | ManxPower | DSpair: OH! That's different! What's the HANGUPCAUSE?? |
| 12:33.34 | DSpair | ManxPower, Strom_M I |
| 12:33.43 | DSpair | ManxPower, Strom_M I'll pastebin a PRI debug session |
| 12:33.56 | ManxPower | DSpair: I don't want PRI debug. |
| 12:34.03 | ManxPower | I want the value of HANGUPCAUSE. |
| 12:34.03 | acron17 | <PROTECTED> |
| 12:34.19 | Strom_M | DSpair: just paste verbose CLI output and the hangupcause for a call |
| 12:34.20 | DSpair | ManxPower, Wait one.. |
| 12:34.55 | ManxPower | "(7:32:38 AM) DSpair: Strom_M, Because asterisk never sees this as a connected call, it hangs up after 60 seconds." You do not know enough to make this statement. |
| 12:34.59 | DSpair | If I turn up verbose, my fax server will drown out the output so that I cannot filter down to a single call. |
| 12:35.04 | [TK]D-Fender | acron17: ... you have no EXTENSION in there to match. |
| 12:35.36 | ManxPower | DSpair: OK. Well, I can start to help you when you can get me the value of hangupcause. Just what verbosity are you trying to run at? verbose 3 is plenty |
| 12:35.36 | DSpair | ManxPower, That's probably true, but thats what tzafrir determined last night. |
| 12:35.50 | ManxPower | DSpair: then mention him next time. |
| 12:36.11 | DSpair | ManxPower, 3 is where I tried, but my fax server generated LOTS of messages that make it VERY difficult to see a single call in that mess. |
| 12:36.18 | ManxPower | DSpair: Put a Noop(HANGUPCAUSE is ${HANGUPCAUSE}) right after the Dial. This is not rocket science. |
| 12:36.39 | DSpair | ManxPower, It isn't to you, but I'm basically a noob. |
| 12:36.44 | ManxPower | DSpair: It sucks to be you, but we need some info. If you can't get it to us then we can't help you. |
| 12:36.47 | Strom_M | surely this PRI will get me to the moon |
| 12:36.52 | DSpair | I working on it. |
| 12:37.07 | ManxPower | Strom_M: Imagine of the poor sod was using BRI. |
| 12:37.21 | Strom_M | DSpair: the less you argue with us and the more you do what we ask you to, the sooner we can fix your stupid problem |
| 12:37.37 | ManxPower | Yup! |
| 12:37.56 | acron17 | <PROTECTED> |
| 12:38.18 | ManxPower | You've already ignored my indication question/suggestion. Hell we can't even get the most basic info -- HANGUPCAUSE. This tells us the NETWORK HANGUPCAUSE AND REASON. |
| 12:38.25 | ManxPower | It's something we sort of need to know. |
| 12:38.33 | DSpair | ManxPower, I'm working on it. |
| 12:38.41 | *** join/#asterisk patrick-- (n=patrick@dev.basecom.eu) |
| 12:38.47 | [TK]D-Fender | acron17: Perhaps you forgot to APPLY your changes... |
| 12:39.13 | patrick-- | Hey all. im Having trouble with dialtones when calling hotlines and having to press buttons for menu choices. my sip phones wont react to those... anyone got an idea ? |
| 12:39.15 | *** join/#asterisk stimpie (n=stimpie@84-104-5-14.cable.quicknet.nl) |
| 12:39.21 | acron17 | <PROTECTED> |
| 12:39.35 | Strom_M | patrick--: dialtones? |
| 12:39.37 | Strom_M | that makes no sense |
| 12:39.44 | patrick-- | well |
| 12:39.47 | patrick-- | not dialtones |
| 12:39.49 | acron17 | <PROTECTED> |
| 12:39.49 | phpboy | What variable can I use to get the caller ID? |
| 12:39.50 | ManxPower | patrick--: figure out what your actual problem is. Your test setup is too complicated to diagnose. |
| 12:39.51 | patrick-- | DTMF thingy |
| 12:40.08 | ManxPower | phpboy: you need to read channelvariables.txt |
| 12:40.11 | patrick-- | ManxPower: you dont even know myy setup.. |
| 12:40.17 | Strom_M | ARE YOU TOO STUPID TO USE A TELEPHONE? TAKE THIS HANDY DANDY QUIZ |
| 12:40.19 | ManxPower | patrick--: then we can't help you |
| 12:40.28 | patrick-- | huh? |
| 12:40.36 | [TK]D-Fender | acron17: pastebin "dialplan show" |
| 12:40.42 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 12:40.46 | patrick-- | i know my setup but you dont. how can you judge the problem without any knowledge on my setup? |
| 12:40.46 | ManxPower | patrick--: if you don't know your setup, how do you expect to diagnose a problem? |
| 12:40.51 | patrick-- | YOU! |
| 12:40.56 | patrick-- | please read accurately |
| 12:41.32 | ManxPower | "(7:39:13 AM) patrick--: Hey all. im Having trouble with dialtones when calling hotlines and having to press buttons for menu choices. my sip phones wont react to those... anyone got an idea ?" Could be a port issue, an indications.conf issue, a hotline issue, immediate=yes, and DTMF issues. |
| 12:41.52 | patrick-- | well we're coming closer |
| 12:41.55 | ManxPower | Now you need to diagnose which one of those items is what you want to work on FIRST. We can't work on all these issues at once. |
| 12:42.00 | Strom_M | or it could be that patrick-- doesn't know any of the terminology and is describing an entirely different problem |
| 12:42.04 | *** join/#asterisk rootlogin (n=root@saturn2.franken.de) |
| 12:42.08 | patrick-- | no one said you should |
| 12:42.11 | acron17 | <PROTECTED> |
| 12:42.15 | Strom_M | i.e. "dialtones" for DTMF |
| 12:42.30 | Strom_M | "hotline" could mean anything, given that |
| 12:42.31 | patrick-- | indeed, thats what i meant, DTMF on hotline IV's |
| 12:42.45 | DSpair | ManxPower, OK, I ran the call at verbosity level 3 and level 5 and never saw anything about HANGUPCAUSE. |
| 12:42.48 | [TK]D-Fender | acron17: now pastebin "ls -la /etc/asterisk" |
| 12:42.59 | ManxPower | I think he means "When I call an IVR from my SIP phone it does not detect the DTMF: |
| 12:43.10 | patrick-- | thats it. |
| 12:43.22 | ManxPower | dialtone, hotline, are just words he threw in there. |
| 12:43.30 | Strom_M | ManxPower: I would be equally unsurprised if he was trying to dial a hamburger using a bottle of heinz ketchup |
| 12:43.33 | patrick-- | you can talk to me directly ;D |
| 12:43.53 | acron17 | <PROTECTED> |
| 12:43.56 | ManxPower | patrick--: a dialtone is the sound you hear when you pick up the phone. Why are you expecting to hear it again? "hotlines" are lines that automatically dial a predefined number when you go off hook. |
| 12:44.21 | patrick-- | right. maybe i got the terminology a bit wrong... hope to be excused |
| 12:44.43 | patrick-- | the IVR not detecting the DTMF's is the actual problem |
| 12:44.49 | patrick-- | (via my sip lines) |
| 12:45.17 | ManxPower | patrick--: It indicates it will be very, very difficult to help you as you don't know enough. There is one specific portion of the problem that I can help you with. Step 1 for YOU: What brand of SIP phone, what version of Asterisk, how are you connecting to the PSTN? |
| 12:45.21 | DSpair | ManxPower, In my CDR, it shows as "NO ANSWER" |
| 12:45.46 | ManxPower | DSpair: I cannot help you fuirther until I know what your hangupcause is. |
| 12:45.53 | ManxPower | Hangupcauses are NUMBERS |
| 12:45.56 | DSpair | ManxPower, How do I find that hangupcause? |
| 12:46.09 | Strom_M | DSpair: I'm privmsging you |
| 12:46.21 | ManxPower | (7:36:18 AM) ManxPower: DSpair: Put a Noop(HANGUPCAUSE is ${HANGUPCAUSE}) right after the Dial. This is not rocket science. |
| 12:46.25 | ManxPower | Shall I type it again? |
| 12:46.55 | patrick-- | Im Using a Snom 190 SIP Phone, asterisk version is 1.4.18 and the PSTN connection is PTP |
| 12:46.58 | patrick-- | @ ManxPower |
| 12:47.04 | ManxPower | PTP? |
| 12:47.33 | patrick-- | Point to Point |
| 12:47.33 | ManxPower | The only PTP I am aware of is a specific type of BRI ISDN setup. |
| 12:47.34 | C4colo | PPTP? or PPP? |
| 12:47.44 | C4colo | PPP then |
| 12:47.52 | C4colo | Point to Point Protocol |
| 12:47.56 | patrick-- | thats it. |
| 12:47.58 | ManxPower | I'll assume you mean "BRI ISDN". If so,. we need to know what CARD you have. |
| 12:48.14 | patrick-- | beroNet BN4S0 |
| 12:48.24 | ManxPower | C4colo: how do you connect to the PSTN via a PPP connection? Where would you plug the line into? |
| 12:48.25 | tzafrir_laptop | IIRC both PPP and PPTP stand for Point to Point Protocol , and PTP is PtP |
| 12:48.41 | ManxPower | patrick--: in the sip.conf entry for your SNOM what is the dtmfmode= set to? |
| 12:49.16 | patrick-- | inband |
| 12:49.36 | ManxPower | and what mode is the phone configured for? What codec is being used between the phone and Asterisk? |
| 12:49.48 | patrick-- | what mode? explain that.. |
| 12:50.00 | ManxPower | my phone I mean "what mode is configured on the phone using the phone configuration or provisioning interface. |
| 12:50.01 | patrick-- | codec is ulaw |
| 12:50.34 | patrick-- | you got me there.. |
| 12:50.35 | ManxPower | If your phone is using RFC2833 DTMF and Asterisk us expecting inband DTMF, well, that's like someone speaking french to someone that only speaks german. |
| 12:50.38 | patrick-- | i dont know what you mean |
| 12:50.38 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 12:50.48 | patrick-- | mhh okay |
| 12:50.53 | patrick-- | so what needs changing |
| 12:50.55 | ManxPower | patrick--: then get out the admin manual for the SNOM phone and find out. |
| 12:51.20 | patrick-- | so i guess the dtmfmode needs to be changed.. |
| 12:51.26 | ManxPower | You need to change the DTMF mode on the phone to use the same method as Asterisk is configured for. Usually you must use RFC2833, but if you are using ulaw or alaw you can use inband instead. |
| 12:51.50 | ManxPower | patrick--: There are a zillion things that could cause this issue. I'm just trying the most common solution first. |
| 12:51.52 | patrick-- | well, im using ulaw as the phones codec with inband dtmf.. |
| 12:52.15 | ManxPower | patrick--: How did you configure the phone? |
| 12:52.47 | ManxPower | patrick--: the PHONE is not configured by Asterisk. The phone is configured by whatever method the maker of the phone decided on. Asterisk cannot change the settings on your phone. |
| 12:53.02 | ManxPower | and you configured the phone. |
| 12:53.39 | [TK]D-Fender | acron17: Please try another call. |
| 12:54.27 | ManxPower | patrick--: if the issue is not the DTMF mode, then the most likely cause is a problem with your ISDN setup -- that is also not done by asterisk, you do it using the drivers for the card. |
| 12:55.19 | acron17 | <PROTECTED> |
| 12:55.41 | [TK]D-Fender | acron17: ANOTHER CALL. |
| 12:55.53 | Strom_M | I'm sing / ing in the stalls / just sing / ing in the stalls / what a glo / rious feeling / just me / and my balls |
| 12:58.27 | ManxPower | patrick--: shall I wait around for you to respond, or should to go to the stuff I've been putting off for the part half hour because I was trying to help you? |
| 12:58.34 | acron17 | <PROTECTED> |
| 12:58.57 | acron17 | <PROTECTED> |
| 12:58.59 | [TK]D-Fender | acoPlace another call so I can se IT fail. |
| 12:59.03 | [TK]D-Fender | see* |
| 12:59.03 | ManxPower | acron17: he wants you to try to make the problem happen again! |
| 12:59.17 | acron17 | ManxPower: ok, thx. |
| 12:59.43 | Strom_M | ah, #asterisk |
| 12:59.56 | Strom_M | it's like one giant boneramathon |
| 12:59.57 | patrick-- | ManxPower: sry... been busy |
| 13:00.13 | ManxPower | patrick--: So am I but I'm taking time out of my day to help you. |
| 13:00.26 | ManxPower | For free. I don't get paid for this. |
| 13:00.44 | ManxPower | patrick--: OK. now answer my questions. |
| 13:00.56 | acron17 | <PROTECTED> |
| 13:01.52 | acron17 | i wonder about: Jul 30 14:59:47 WARNING[5092]: chan_sip.c:2017 create_addr: No such host: 12 |
| 13:02.05 | [TK]D-Fender | acron17: Yeah, I'm wondering why I'm seeing *2* subnets in there |
| 13:02.11 | [TK]D-Fender | acron17: Care to explain? |
| 13:02.34 | acron17 | <PROTECTED> |
| 13:02.54 | [TK]D-Fender | acron17: Contact: <sip:12@192.168.101.14> <------Sending to 10.62.65.6 : 5600 (NAT) <---- WTF? |
| 13:03.27 | patrick-- | ManxPower: what exactly is it you need to know about the setup to be able to hint me to a solution? |
| 13:03.34 | ManxPower | Well, I must go back to work. Best of luck to everyone I was unable to complete solving the problem. |
| 13:03.56 | acron17 | <PROTECTED> |
| 13:04.24 | ManxPower | [TK]D-Fender: could you take over? patrick-- has a SIP->PSTN DTMF issue. He doesn't understand you set the DTMF mode on the phone as well as Asterisk. He is also using BRI ISDN. |
| 13:04.44 | [TK]D-Fender | patrick--: Go prove DTMF works in "voicemailmain" |
| 13:05.07 | ManxAway | [TK]D-Fender: he's using ulaw/inband on the Asterisk side. |
| 13:05.21 | [TK]D-Fender | ManxAway: prove the phone -> * first |
| 13:05.38 | acron17 | [TK]D-Fender: but it isn't working when directly connected to the LAN eigther... |
| 13:05.43 | ManxAway | [TK]D-Fender: I spent the last 30 mins trying to do that. He's too noob to do that easily. |
| 13:06.26 | *** join/#asterisk MrNaz (n=naz@ppp121-44-207-242.lns3.mel4.internode.on.net) |
| 13:06.58 | acron17 | [TK]D-Fender: as I said already: I wonder about: Jul 30 14:59:47 WARNING[5092]: chan_sip.c:2017 create_addr: No such host: 12 |
| 13:07.12 | [TK]D-Fender | acron17: I'm wondering too |
| 13:07.52 | acron17 | [TK]D-Fender: I would not expect the asterisk to create an address for something it should do itself... |
| 13:08.54 | [TK]D-Fender | acron17: Your READ's are coming from 10.XXX , Sending to 10.62.65.6 : 5600 (NAT), You're SENDING there... but then your AUDIO is miraculously ending up going to 192.168.xxxx |
| 13:09.07 | [TK]D-Fender | acron17: You have some %$#ed up networking going on there. |
| 13:09.09 | *** join/#asterisk ToTo (n=ToTo@209.8.41.181) |
| 13:09.20 | [TK]D-Fender | acron17: Looks like you are interacting with a really stupid proxy of some sort |
| 13:09.33 | [TK]D-Fender | acron17: And definitely nothing I want to touch |
| 13:10.31 | acron17 | [TK]D-Fender: Sorry I do not get the point. The setup is as this: |
| 13:11.10 | [TK]D-Fender | acron17: Reads shouldn't come sourced from 1 subnet and then audio, etc fall to another like that |
| 13:11.28 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:13.08 | acron17 | [TK]D-Fender: <Me with VPN-Address: 10.62.65.6> through vpn-gateway <Company astrisk with local address> |
| 13:13.47 | [TK]D-Fender | acron17: I'm sorry, but this is more than I'm prepared to continue with currently. |
| 13:14.33 | [TK]D-Fender | acron17: Wait a sec... |
| 13:15.07 | [TK]D-Fender | acron17: #@#$ |
| 13:15.19 | [TK]D-Fender | acron17: next time do this at VERBOSE 10 <--- |
| 13:15.28 | [TK]D-Fender | acron17: that has just wasted our time completely. |
| 13:16.06 | phpboy | hi |
| 13:16.17 | [TK]D-Fender | acron17: you dialed 12. SURE... [default] includes [time]... but thats not your problem |
| 13:16.38 | [TK]D-Fender | acron17: is ALSO includes [local] |
| 13:16.51 | acron17 | [TK]D-Fender: Sorry how can I increase the verbosity? |
| 13:16.52 | [TK]D-Fender | acron17: and just GUESS what we see in there? |
| 13:16.57 | [TK]D-Fender | acron17: '_X.' => 1. Dial(SIP/${EXTEN}) [pbx_config] |
| 13:17.04 | [TK]D-Fender | acron17: another ^#%$ing match for 12 |
| 13:17.19 | [TK]D-Fender | acron17: "set verbose 10" |
| 13:17.27 | [TK]D-Fender | acron17: Yuo have OVERLAPPING extensions |
| 13:17.39 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.141) |
| 13:17.51 | acron17 | [TK]D-Fender: Oh :-( sorry |
| 13:17.55 | seanbright | core set verbose 10 |
| 13:18.19 | [TK]D-Fender | acron17: helps when you can SEE the problem and you're running half blind |
| 13:19.02 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:19.49 | [TK]D-Fender | acron17: exten => _X.,1,Dial(SIP/${EXTEN}) <- dangerous pattern. |
| 13:20.08 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 13:20.10 | [TK]D-Fender | patrick--: So.... whats the story? |
| 13:20.42 | acron17 | [TK]D-Fender: ok. how can it be made less dangerous? |
| 13:21.16 | [TK]D-Fender | acron17: anythign 2 digits or longer is miraculously a SIP peer on your box? I don't think so... |
| 13:21.35 | [TK]D-Fender | acron17: Come up with a plan where your patterns don't everlap |
| 13:21.41 | *** join/#asterisk bdolljr (n=bdolljr@64.154.93.78) |
| 13:21.55 | [TK]D-Fender | acron17: a fixed "12" and "_X." don't mix |
| 13:22.14 | acron17 | [TK]D-Fender: ok, will do |
| 13:26.35 | patrick-- | [TK]D-Fender: works. |
| 13:26.46 | *** part/#asterisk bdolljr (n=bdolljr@64.154.93.78) |
| 13:26.56 | *** join/#asterisk bdolljr (n=bdolljr@64.154.93.78) |
| 13:27.15 | [TK]D-Fender | patrick--: Then its your other interface that has issues |
| 13:27.24 | patrick-- | "other interface"? |
| 13:28.41 | tzanger | has anyone here heard HPEC "hiss" instead of canceling echo? |
| 13:29.22 | gr0mit | tzafrir_laptop, resolved the issue |
| 13:29.36 | gr0mit | bristuff was upset by the STATUS message |
| 13:29.50 | [TK]D-Fender | patrick--: Your link to the PSTN |
| 13:30.26 | bdolljr | does anyone here know how to suppress the "Missed Call" messages that are displayed on a Cisco 79XX phone when a call is not answered? |
| 13:30.45 | tzafrir_laptop | gr0mit, how? |
| 13:31.21 | gr0mit | used native asterisk, not bristuff |
| 13:31.29 | gr0mit | bristuff breaks something |
| 13:31.52 | gr0mit | does not handle the state transitions correctly when it gets the STATUS message |
| 13:31.59 | gr0mit | I sent the traces to klaus |
| 13:32.21 | gr0mit | he looked at them and reckoned that was the issue |
| 13:32.42 | gr0mit | so tried same version, not bristuffed, and it works |
| 13:32.45 | *** join/#asterisk tapic (n=tap@88.255.77.200) |
| 13:33.32 | *** join/#asterisk jadams__ (n=jadams@rrcs-24-73-180-234.se.biz.rr.com) |
| 13:33.42 | ManxAway | bdolljr: IF that can be done it would be done via the phone config, not Asterisk |
| 13:34.36 | bdolljr | ok... thanks.. i thought i read somewhere about a way to do it on a dial by dial basis... i'll check the phone config. |
| 13:34.59 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:35.51 | *** join/#asterisk riddlebox (n=james@IP-16-115.wustl.edu) |
| 13:36.22 | ManxAway | bdolljr: you can with Call Manager, not with SIP. |
| 13:36.32 | *** join/#asterisk riddlebox (n=james@IP-149-215.wustl.edu) |
| 13:37.14 | bdolljr | Ahh.. maybe that is what I saw... I know i read it somewhere. Would be REAL nice for Operator and HelpDesk lines... |
| 13:38.03 | ManxAway | one would HOPE that operators and helpdesks answer every call |
| 13:38.47 | bdolljr | but if Operator and Helpdesk ring 6 different phones... 5 get Missed Call when 1 phone answers... :{ |
| 13:38.48 | ManxAway | At least at my clients, operators double as receptionists and they are not hired for their smarts, they are hired by who they are sleeping with or who wants to sleep with them. |
| 13:38.56 | Strom_M | ManxAway: "Thank you for calling nine one one. We're not here right now, so if you'll leave us a message, we'll get back to you when we can. Thanks! *BEEEEP* This mailbox is full." |
| 13:39.02 | ManxAway | bdolljr: well don't do that. |
| 13:39.38 | ManxAway | We just turn off that feature in our Polycom phones |
| 13:42.02 | bdolljr | i'll look for someway to turn it off in the phone config... thanks! |
| 13:43.12 | tapic | hi, I am searching for a way to originate call in an AGI script and interact with the called side and bridge parties if required |
| 13:43.40 | *** part/#asterisk bdolljr (n=bdolljr@64.154.93.78) |
| 13:44.07 | *** join/#asterisk rbd (n=rbd@adsl-074-229-183-112.sip.rmo.bellsouth.net) |
| 13:44.42 | keith4 | what's the meaning of this? "WARNING[23986] app_queue.c: The device state of this queue member, Sip/sollog, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings." |
| 13:45.02 | patrick-- | [TK]D-Fender: what kind of setting does that need for the pstn to work correctly |
| 13:45.10 | tapic | I am using FastAGI and can use AMI |
| 13:45.11 | rbd | hi guys. I'd like to get the number of active sip channels on a specific asterisk server via the manager API (or some equally easy external method). Would the most direct way be to run Execute for the 'sip show channels' command and parse out that? |
| 13:45.14 | [TK]D-Fender | keith4 : check UPGRADE.txt for correct configuration settings |
| 13:45.25 | [TK]D-Fender | patrick--: For yours, dunno |
| 13:45.36 | tapic | rbd : there is a Status command for AMI |
| 13:45.37 | rbd | I meant I'd like to programmatically do that, so my app can know how many sip channels are active |
| 13:46.28 | *** part/#asterisk keith4 (n=keith@lust.CC.Lehigh.EDU) |
| 13:46.33 | *** join/#asterisk keith4 (n=keith@lust.CC.Lehigh.EDU) |
| 13:46.35 | *** join/#asterisk riddlebox (n=james@IP-149-215.wustl.edu) |
| 13:46.54 | patrick-- | [TK]D-Fender: what does it generally need? |
| 13:47.11 | [TK]D-Fender | patrick--: Normally you'd think notihng, but I don't do BRI |
| 13:47.19 | rbd | tapic: I see....so just execute that and parse through the channel output returned? even with sip show channels the last line gives you the number of active channels though (and I only care about sip here) |
| 13:47.35 | patrick-- | okay |
| 13:47.44 | keith4 | [TK]D-Fender: this wasn't an upgrade |
| 13:47.49 | [TK]D-Fender | patrick--: pastebin a functional test with core debug 10, verbsoe 10, and then try a failed outbound call. |
| 13:48.05 | [TK]D-Fender | keith4 : it tells you to read a doc. GO READ IT |
| 13:48.11 | riddlebox | does anyone know where to get app_rxfax.c and app_txfax.c ? for faxing? when I go to soft-switch.org to download them, they are not there? |
| 13:48.40 | tapic | rdb : I don't know any other way. |
| 13:50.09 | patrick-- | [TK]D-Fender: right |
| 13:50.38 | rbd | ok thanks tapic |
| 13:50.57 | tapic | is it able to originate a call inside another call and play a music to the new channel and then connect this two channels on asterisk? |
| 13:51.11 | keith4 | [TK]D-Fender: could this be because the sip peers don't have call limits? |
| 13:51.33 | *** join/#asterisk riddlebox (n=james@IP-149-215.wustl.edu) |
| 13:51.38 | [TK]D-Fender | keith4 : get off your ass and read. |
| 13:51.45 | keith4 | i am |
| 13:51.56 | *** join/#asterisk hubguruJR (n=hubguruJ@mail.ntegratedsolutions.com) |
| 13:51.59 | keith4 | but it's 500 lines of stuff that seems not applicable |
| 13:52.06 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 13:52.06 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:52.22 | MaliutaLap | keith4: STFU +RTFM :) |
| 13:52.27 | MaliutaLap | to put it nicely |
| 13:52.41 | [TK]D-Fender | riddlebox: look HARDER |
| 13:54.08 | MaliutaLap | keith4: people can't always hold your hand. If you can't handle research and tech reading don't run a *nix OS or use OSS software, go pay for everything that comes with vendor hand holding and pointy clicky interfaces. Forget all about real tech and only take what you're spoon fed |
| 13:54.34 | keith4 | wow |
| 13:54.38 | keith4 | thanks, douche |
| 13:55.27 | keith4 | MaliutaLap!*@* added to ignore list. |
| 13:55.33 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.95.130) |
| 13:55.37 | tapic | how can I do this? I have searched originate,redirect and park AMI actions and still can not find a solution for a bright way of combining this actions to do this task |
| 13:55.46 | acron17 | [TK]D-Fender: thanks for your help i got i fixed. |
| 13:56.01 | acron17 | i'm off. bye |
| 13:56.32 | *** join/#asterisk riddlebox (n=james@IP-149-215.wustl.edu) |
| 13:57.04 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 13:57.16 | patrick-- | [TK]D-Fender: http://pastebin.com/m45b71826 excuse the mess, but i cannot keep people from placing calls ... |
| 13:57.42 | patrick-- | oad:14 dad:09113690 |
| 13:57.45 | patrick-- | that one it is |
| 13:57.46 | *** join/#asterisk CVirus (n=Satan@82.201.178.105) |
| 13:58.06 | CVirus | Qwell: did you try to run asterisk on OM ? |
| 13:58.35 | [TK]D-Fender | patrick--: Wheres the successful test in there? |
| 13:58.48 | patrick-- | ah successful... ? with voicemail? |
| 13:59.09 | patrick-- | thats the failed... ill do the second one now.. with VM |
| 13:59.09 | phpboy | How do I force asterisk do go through all the steps in a specific dial plan. see, I have a System() cmd right at the end of the dial plana nd I want that cmd to execute even if the caller hangs up from the other side. |
| 14:00.18 | patrick-- | [TK]D-Fender: http://pastebin.com/m29168422 |
| 14:00.45 | [TK]D-Fender | phpboy: go read "core show application dial" and go read up on "Asterisk Standard Extensions" on the WIKI |
| 14:00.47 | [TK]D-Fender | ~wikis |
| 14:00.47 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 14:01.16 | *** join/#asterisk djin (n=djin@84-104-110-116.cable.quicknet.nl) |
| 14:01.23 | *** join/#asterisk coppice (n=chatzill@27.202.17.210.dyn.pacific.net.hk) |
| 14:01.29 | [TK]D-Fender | patrick--: Ok, there's something with your BRI interface. I really don't have a clue what... |
| 14:01.32 | *** join/#asterisk riddlebox (n=james@IP-149-215.wustl.edu) |
| 14:02.46 | [TK]D-Fender | keith4 : pastebin your sip.conf entry for that user. |
| 14:03.02 | patrick-- | [TK]D-Fender: okay.. you see a way to find out what? |
| 14:03.15 | [TK]D-Fender | patrick--: Sorry, I don't. |
| 14:03.18 | patrick-- | right |
| 14:03.20 | patrick-- | thanks anyway |
| 14:03.30 | [TK]D-Fender | keith4 : Actually give me the WHOLE file. |
| 14:05.13 | keith4 | [TK]D-Fender: the only thing that seems relevant is this note about needing a call limit to allow sip channels to work with queues correctly |
| 14:05.20 | DSpair | ManxAway, Thanks for your help, and we did get the problem resolved. It was multiple configuration gaffs on the part of our vendor. |
| 14:05.28 | [TK]D-Fender | keith4 : PASTEBIN |
| 14:05.43 | keith4 | yah yah yah, i'm workin on it. gotta xxx out lots of passwords |
| 14:06.14 | [TK]D-Fender | keith4 : grep out the secret lines. |
| 14:06.27 | [TK]D-Fender | keith4 : there goes 5 valuable seconds |
| 14:06.32 | *** join/#asterisk riddlebox (n=james@IP-149-215.wustl.edu) |
| 14:08.47 | keith4 | [TK]D-Fender: http://pastebin.com/m4f7d433e |
| 14:09.38 | [TK]D-Fender | keith4 : Congratulations. Now go read the doc. |
| 14:09.57 | keith4 | the part I think is applicable is this: http://pastebin.com/m5abccf2e |
| 14:10.16 | keith4 | i've been trying, but I keep getting interrupted with requests to pastebin crap |
| 14:10.32 | ManxAway | DSpair: *nod* We should have been able to determine that in the first 10 mins, but since you don't know much about Asterisk..... |
| 14:10.33 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 14:10.33 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:11.34 | *** join/#asterisk riddlebox (n=james@IP-149-215.wustl.edu) |
| 14:11.37 | [TK]D-Fender | keith4 : I asked you for ONE pastebin which took entirely too long to get and it shows that you have not followed the more than explicit instructions the doc told you to read. The doc which was of course handed directly to you by the CLI notice |
| 14:11.51 | DSpair | ManxAway, Gotta start somewhere, but I do apologize for testing your patience. |
| 14:12.20 | ManxAway | I think [TK]D-Fender was just confirming you followed the docs, keith4. You didn't. Now he's annoyed. |
| 14:12.35 | keith4 | shrugs |
| 14:12.39 | ManxAway | DSpair: you had no business putting a server into production with your knowledge level. |
| 14:12.44 | keith4 | i haven't made any changes yet, don't know why he wanted to see sip.conf |
| 14:13.12 | ManxAway | keith4:L He told you to read the doc. Maybe he assumed you did so and made the changes? That was the point of having your read the doc. |
| 14:13.25 | DSpair | ManxAway, Well, we hired a DCAP provider to get this running, but they have proven their incompetence on many occasions. |
| 14:13.26 | *** join/#asterisk seanmh (n=johndoe@75.150.19.89) |
| 14:13.28 | [TK]D-Fender | keith4 : and I never looked at that doc myself until your error message made notice of it. I then instantly found the applicable passage. |
| 14:13.37 | ManxAway | DSpair: That is different then. 8-) |
| 14:13.48 | [TK]D-Fender | keith4 : Its not even vague about it |
| 14:13.56 | anonymouz666 | PRI got event: HDLC Abort (6) on Primary D-channel of span 1 -> this could happen due IRQ problems? Even if I don't see any IRQ misses in Zaptel? |
| 14:14.03 | keith4 | [TK]D-Fender: the suspense is killing me. is it call limit? |
| 14:14.26 | ManxAway | anonymouz666: yes, it could. It could be caused by a million things, all almost impossible to diagnose and fix on a production server. |
| 14:14.38 | DSpair | ManxAway, Prime example, the custom kernel he compiled was crashing every day during the logrotate process. |
| 14:14.55 | DSpair | ManxAway, I'm still trying to figure out how he accomplished that. |
| 14:15.03 | [TK]D-Fender | keith4 : Yes that passage says "If you want * to know that your %$#&#@^% phones are on calls an suppress these warnings enter this blatantly clear stuff into your config" |
| 14:15.03 | ManxAway | DSpair: I've not had to compile a custom kernel in years and years |
| 14:15.09 | DSpair | ManxAway, Me either. |
| 14:15.31 | *** join/#asterisk Superbartt (n=bart@84-245-35-223.dsl.cambrium.nl) |
| 14:15.33 | [TK]D-Fender | keith4 5 lines... 2 of which you already HAVE.... thats 3 little things... |
| 14:15.52 | DSpair | In every case I've had over the past few years, the distro I was using had a stock kernel that worked fine for all of my needs. |
| 14:15.52 | ManxAway | keith4: [TK]D-Fender usually requires dinner, drinks, and sweet nothings whispered in his ear before he will HOLD YOU HAND. |
| 14:16.07 | keith4 | well, he's a bit far from me to take him out to dinner |
| 14:16.23 | DSpair | snorts. |
| 14:16.39 | ManxAway | keith4: I guess you'll have to think and read for yourself then. He's not usually going to GIVE you the answer. He will GIVE YOU THE LOCATION of where you can read the answer. |
| 14:16.56 | keith4 | maybe we could share dinner via webcam or something |
| 14:17.10 | ManxAway | keith4: I suspect you're not his type. 8-) |
| 14:17.18 | keith4 | you don't know that |
| 14:17.18 | [TK]D-Fender | keith4 : I would be happy to simply see you read your own error messages and follow the explicit instructions * tells you on things like that and that you actually read the docs while the hamster is still on the wheel. |
| 14:17.49 | keith4 | the hamster is a lazy little bastard |
| 14:17.59 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 14:18.03 | [TK]D-Fender | keith4 : There are a lot of poorly documented things in *. this WASN'T one of them. |
| 14:18.53 | keith4 | can I set a global call-limit? or do I have to set it for each peer? |
| 14:19.13 | [TK]D-Fender | keith4 : jsut do what it SAYS. |
| 14:19.16 | ManxAway | I guess I could look in sip.conf.sample and tell you.....but I don't get paid for this. |
| 14:19.42 | ManxAway | [TK]D-Fender: he sure spends a lot of time talking not not much time doing. |
| 14:20.03 | [TK]D-Fender | ManxAway: But I don't want to do that, can I do it this way instead? |
| 14:20.08 | keith4 | oh, it's done. i'm just waiting for one of the queue members to actually get on the phone |
| 14:20.28 | *** join/#asterisk Great_Anta_Baka (i=c4219f6b@gateway/web/ajax/mibbit.com/x-3d8fd6306564e27f) |
| 14:21.35 | Great_Anta_Baka | i get the following error when trying to access the misdn config in the ast now gui --> For some reason, I could not grab scan.html, is the misdn-init tool installed? |
| 14:21.44 | Great_Anta_Baka | but misdn-init is installed |
| 14:21.49 | Great_Anta_Baka | what does this mean?\ |
| 14:22.24 | seanbright | #asterisknow |
| 14:22.50 | Great_Anta_Baka | :( channel is basically empty bleh... will just have too look elsewhere |
| 14:23.08 | seanbright | there are 8 people there |
| 14:23.24 | seanbright | including some that actually work on asteriskNOW |
| 14:23.50 | Great_Anta_Baka | i see well lets hope they can help |
| 14:24.04 | Great_Anta_Baka | otherwise more draconian bosses will be down my throat |
| 14:24.15 | Great_Anta_Baka | thanks sean |
| 14:24.25 | [TK]D-Fender | Great_Anta_Baka: Doesn't matter to us if their channel is empty. This is not 2nd-level support for any GUI |
| 14:24.52 | Great_Anta_Baka | well i thought you might know the reason why misdn-init is not being found |
| 14:25.34 | seanbright | Great_Anta_Baka: it's best sometimes just not to respond, a few people in here will take every opportunity to make you feel like shit |
| 14:25.44 | seanbright | (just an FYI) |
| 14:26.19 | fogo- | I just had a TE410P go bad this morning - that makes 2 so far... anyone have any reccomendations on quad-pri cards? Should I look at buying Sangoma? Reccomendations? |
| 14:26.49 | Great_Anta_Baka | thats the case in most irc channels these days |
| 14:27.04 | Great_Anta_Baka | i think people get a kick out of making n00bs feel bad |
| 14:27.11 | seanbright | Great_Anta_Baka: indeed. |
| 14:27.15 | gr0mit | Great_Anta_Baka, that is NOT true |
| 14:27.36 | [TK]D-Fender | seanbright: And Burger King managers LOVE it when people come in asking for a Big Mac. |
| 14:27.47 | ManxAway | Great_Anta_Baka: This channel is NOT 2nd level support for Asterisk GUI |
| 14:27.51 | seanbright | heh |
| 14:27.53 | seanbright | Great_Anta_Baka: see? |
| 14:27.57 | ManxAway | or AsteriskNOE |
| 14:28.06 | Great_Anta_Baka | ok i got that the first time man |
| 14:28.07 | [TK]D-Fender | fogo-: Sangoma A104d |
| 14:28.07 | seanbright | Great_Anta_Baka: the KEY words are the ones that are in ALL CAPS |
| 14:28.16 | gr0mit | what people here get sooo frustrated about is noobs starting out with a very complex gui-based system - when you can make your first call with about 5 lines in the right config files |
| 14:28.20 | seanbright | Great_Anta_Baka: they ARE very IMPORTANT |
| 14:28.25 | [TK]D-Fender | fogo-: I also went through 2 TE405P's back to back before switching |
| 14:28.34 | Great_Anta_Baka | hehe |
| 14:28.42 | gr0mit | fogo, i always buy sangomas |
| 14:29.22 | seanbright | Great_Anta_Baka: let me see if i can help you |
| 14:29.27 | seanbright | Great_Anta_Baka: gimme a sec |
| 14:29.30 | fogo- | I was beginning to wonder if I should be looking at sangomas.. thanks for the info :) |
| 14:29.33 | *** join/#asterisk moy (n=moy@nat/ibm/x-28d76d78f89975ed) |
| 14:29.40 | *** join/#asterisk watchy (n=watchy@adsl-69-152-41-251.dsl.ltrkar.swbell.net) |
| 14:29.40 | ManxAway | fogo-: I doubt you'll find Sangoma any more reliable than Digium. At least from the "card going bad" standpioint. You need to look at your line and surges on the lines |
| 14:29.45 | watchy | top of the morning to u |
| 14:29.51 | [TK]D-Fender | seanbright: Yes, usually key, in a few cases it can double for the "load" aspect as well (rare) |
| 14:30.11 | [TK]D-Fender | loud* |
| 14:30.12 | Great_Anta_Baka | thanks seanbright |
| 14:30.14 | *** join/#asterisk spokra (n=spokra@host093-179-146.sea0.speakeasy.net) |
| 14:30.18 | watchy | im having issues with * detecting fax machins, some seem to be %100 some seem to never work |
| 14:30.22 | watchy | is it the fax machine? |
| 14:30.38 | Great_Anta_Baka | maybe you should answer me in #asterisknow as to not upset anyone here |
| 14:30.43 | seanbright | Great_Anta_Baka: nah |
| 14:30.46 | seanbright | Great_Anta_Baka: here is fine |
| 14:30.52 | Great_Anta_Baka | kk |
| 14:31.10 | ManxAway | watchy: usually operator error. Fax sender dials the number on the handset on the phone, then waits to hear a fax tone before pressing SEND on the fax. Since the sending machine doesn't sent tones until you press START, Asterisk has no fax tone on the line to detect. |
| 14:31.13 | watchy | my parents can send a fax and i get %100 reception and detectiong |
| 14:31.28 | watchy | oh |
| 14:31.51 | watchy | so they arent using an automatic fax like my parents that just sucks in the paper and dials |
| 14:32.03 | *** join/#asterisk Rico29 (n=Rico@sto93-4-88-162-214-225.fbx.proxad.net) |
| 14:32.04 | watchy | that would make sense |
| 14:32.05 | fogo- | [TK]D-Fender: sangomas will work as a drop-in replacement for digium cards I assume? |
| 14:32.16 | [TK]D-Fender | Great_Anta_Baka: You already knew that GUI's aren't supported here yet you ask anyways. Not a first time offense either. Anta baka indeed. |
| 14:32.19 | ManxAway | watchy: They might be using an automatic machine, but they are using it manually |
| 14:32.28 | seanbright | fogo-: you shouldn't drop either brand of card, that could damage it |
| 14:32.31 | ManxAway | tell them stick it in, press the right buttons. |
| 14:32.34 | fogo- | lol |
| 14:32.48 | fogo- | seanbright: so that's why my cards have been going bad... ;) |
| 14:32.49 | [TK]D-Fender | fogo-: from * config poitn yes, They do need their own driver compiled and set up but that's rather minor |
| 14:32.55 | watchy | manx: ah thanks |
| 14:32.56 | tapic | hi, is there a way to transfer a channel to another channel? |
| 14:33.04 | watchy | i was thinking it might be that but was wondering |
| 14:33.12 | ManxAway | tapic: yes. press the transfer button on your phone. |
| 14:33.30 | keith4 | ManxAway: you're doing a lot of chatting for someone who's away, you know |
| 14:33.44 | tapic | sorry for insufficient information, I am trying to do that in AGI |
| 14:33.45 | ManxAway | watchy: zapscan or chanspy a call where the caller consistantly fails to send the fax. |
| 14:33.56 | ManxAway | tapic: I can't help you with that. |
| 14:33.59 | tapic | I use FastAGI and can also use AMI |
| 14:34.07 | [TK]D-Fender | tapic: AGI has nothing to do with redirecting a call |
| 14:34.08 | seanbright | Great_Anta_Baka: looks to be a web server configuration problem, based on a quick glance at the relevant file |
| 14:34.14 | ManxAway | watchy: I'll bet you don't hear fax tones on the bad calls. |
| 14:34.21 | [TK]D-Fender | tapic: Go read the AGI referece chapter in THE BOOK |
| 14:34.28 | watchy | hrm you know i dont think we are |
| 14:34.40 | seanbright | Great_Anta_Baka: actually, that might be a lie, hold. |
| 14:34.53 | watchy | man why are people idiots at using fax machines |
| 14:35.16 | [TK]D-Fender | watchy: because someone was dumb enough to give them one. |
| 14:35.19 | tapic | I use FastAGI and in that seperate machine I can issue AMI commands |
| 14:35.26 | Great_Anta_Baka | lol sean |
| 14:35.31 | ManxAway | watchy: We have fax detection on all our DIDs, but we also tell people that if they have problems sending a fax to the DID, to send it to the main fax number, which is on a POTS line not connected to Asterisk |
| 14:35.39 | seanbright | Great_Anta_Baka: run this command 'ls -al /var/lib/asterisk/static-http/config' |
| 14:35.39 | Great_Anta_Baka | well looking through the code myself to see where the problem is |
| 14:35.45 | seanbright | Great_Anta_Baka: and pastebin the output |
| 14:35.48 | seanbright | ~pb |
| 14:35.49 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines 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 |
| 14:35.49 | watchy | manx: ah |
| 14:35.50 | Great_Anta_Baka | k |
| 14:35.55 | ManxAway | seanbright: this would all be better on the correct channel. |
| 14:35.59 | tapic | so I am looking for a way of transferring a channel to another channel |
| 14:36.02 | seanbright | ManxAway: i couldn't agree with you more |
| 14:36.25 | *** join/#asterisk LND (n=Lee@89.192.17.57) |
| 14:36.44 | [TK]D-Fender | tapic: Perhaps you could explain to COMPLETE scenario and what you would like to have happen |
| 14:36.57 | Great_Anta_Baka | seanbright: http://pastebin.com/m10b4c01c |
| 14:37.00 | tapic | I will originate a call in an already existing call and than play a music to the new channel and than connect them together. is that possible on *? |
| 14:37.15 | keith4 | almost anything is possible |
| 14:37.19 | ManxAway | tapic: that sounds like a standard Dial to me. |
| 14:37.30 | seanbright | Great_Anta_Baka: gotcha... might be a permissions problem |
| 14:37.39 | Great_Anta_Baka | ah |
| 14:37.45 | fogo- | [TK]D-Fender: so it looks like once the wanpipe drivers are compiled/installed, all the configuration and so forth are the same? Is it going to cause problems if I have both installed in the case I have to swap the card? |
| 14:37.51 | seanbright | Great_Anta_Baka: there is javascript that is calling back to the box to generate 'scan.html' |
| 14:38.06 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 14:38.07 | seanbright | Great_Anta_Baka: which isn't there, so i'm assuming it's a permissions problem |
| 14:38.17 | [TK]D-Fender | fogo-: They work great. Nvere had a problem with them (analog or digital) either myself or any of my clients |
| 14:38.20 | ManxAway | so much for my suggestion to handle AsteriskNOW questions on #AsteriskNOW |
| 14:38.30 | Great_Anta_Baka | i see |
| 14:38.34 | seanbright | ManxAway: i couldn't agree with you more |
| 14:38.41 | tapic | I will originate a call in an already existing call and interact with the new channel (play music,get DTMF) and than connect them together if required. is that possible on *? |
| 14:38.42 | seanbright | shit |
| 14:38.48 | seanbright | sorry, that is an auto-responder i have set up |
| 14:38.57 | Great_Anta_Baka | hahaha |
| 14:39.09 | ManxAway | Well, I'll be watching #asterisk-cli for non-GUI questions. |
| 14:39.14 | [TK]D-Fender | tapic: "I will originate a call in an already existing call" <- please don't don't describe HOW you think you are going to do something. Describe what is happening NORMALLY and at what point you want something ELSE to happen. |
| 14:39.24 | seanbright | ManxAway: super! |
| 14:39.29 | fogo- | [TK]D-Fender: so I can have both compiled in; swap the card, and go? (sorry to ask again, just want to be sure here :) ) |
| 14:39.45 | [TK]D-Fender | fogo-: Largely yes. |
| 14:39.47 | ManxAway | seanbright: that way you GUI users can destroy this channel all you want. |
| 14:39.53 | seanbright | ManxAway: super! |
| 14:40.04 | Great_Anta_Baka | so seanbright do i just to a chown on that folder and its subfolders? |
| 14:40.08 | fogo- | [TK]D-Fender: Excellent. Thank you for the info :) |
| 14:40.22 | seanbright | Great_Anta_Baka: depends on what user the webserver is running as |
| 14:40.37 | Great_Anta_Baka | its running the asterisk one |
| 14:40.44 | seanbright | Great_Anta_Baka: ohhhh |
| 14:40.47 | Great_Anta_Baka | http.conf |
| 14:40.57 | Great_Anta_Baka | i tried apache once but to no avail |
| 14:40.59 | *** join/#asterisk ToTo (n=ToTo@207.176.6.40) |
| 14:41.00 | seanbright | Great_Anta_Baka: hrmm. well, i don't know the correct fix. but i'm 99% sure it's a permissions problem. |
| 14:41.11 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 14:41.11 | seanbright | Great_Anta_Baka: so hopefully that could get you going in the right direction |
| 14:41.33 | *** join/#asterisk Assimilate (n=Assimila@216.83.78.108) |
| 14:41.33 | Great_Anta_Baka | thanks for all the help man |
| 14:42.07 | seanbright | Great_Anta_Baka: no problem. good luck and god speed. and in the future, these types of questions should go to #asterisknow |
| 14:42.48 | tapic | ok. Mr.Caller will call my asterisk from ZAP. Then I will call Mr.Agent from ZAP and play a sound file to him and get his DTMF input. If Mr.Agent presses 5, then I want Mr.Caller and Mr.Agent start a conversation. |
| 14:43.07 | Great_Anta_Baka | will do.. think i will stick to the forums though |
| 14:43.14 | seanbright | Great_Anta_Baka: cool. |
| 14:43.20 | Great_Anta_Baka | never really had help in that channel but needed a quick response |
| 14:43.23 | *** join/#asterisk jgoddess (n=monkey@phrank.aus.us.siteprotect.com) |
| 14:43.44 | seanbright | Great_Anta_Baka: the digium guys tend to sleep in :) |
| 14:43.53 | Great_Anta_Baka | haha nice. |
| 14:43.55 | seanbright | not sure what every else's excuse is |
| 14:44.03 | seanbright | everyone* |
| 14:44.18 | Great_Anta_Baka | bah if i was good i think i would be reluctant to answer n00b questions myself |
| 14:44.40 | seanbright | Great_Anta_Baka: well that is why this channel and those channels exist... i think |
| 14:44.46 | Great_Anta_Baka | dont really blame people.. not as if they getting something out of it |
| 14:44.49 | seanbright | Great_Anta_Baka: in theory at least. |
| 14:44.51 | Great_Anta_Baka | mmm |
| 14:45.02 | Great_Anta_Baka | yeah... the theory part is the important bit there |
| 14:45.03 | Great_Anta_Baka | lol |
| 14:45.06 | seanbright | indeed. |
| 14:45.25 | [TK]D-Fender | tapic: so far there is no "transfer" in there at all. Just sounds like you want the callee to be able to choose wether or not to acccept the call. |
| 14:45.36 | dominic1 | hello guys, just a short question. If I execute a macro in the dialplan with a dialcommand. Where does the macro jump to, if the user hangs up? |
| 14:45.48 | [TK]D-Fender | tapic: For which there was already a more that easy solution : "core show application dial" |
| 14:45.50 | dominic1 | in ael there is no option to execute a h in the macro |
| 14:46.28 | [TK]D-Fender | dominic1: when a macro executes, it gets functionally merged with the calling context. it inherits the "h" from there. |
| 14:46.51 | [TK]D-Fender | dominic1: and you don't "execute" an extension. It goes there if its there. |
| 14:47.18 | *** join/#asterisk XnOSX (n=XnOSX@212.145.173.80) |
| 14:47.22 | [TK]D-Fender | dominic1: And if AEL can't let you fine-tune placement well enough, there's another reason why I would never recommend it. |
| 14:48.28 | codefreeze-lap | dominic1: You can. |
| 14:50.03 | codefreeze-lap | dominic1: the "catch" statement. See http://voip-info.org/wiki/view/Asterisk+AEL2#Macros |
| 14:50.58 | codefreeze-lap | [TK]D-Fender: my, my! feeling a bit anti-AEL today, aren't we? |
| 14:51.34 | dominic1 | hi codefreeze, thank you very much! |
| 14:51.51 | dominic1 | AEL is great! |
| 14:51.54 | seanbright | codefreeze-lap: we fear what we do not understand. |
| 14:53.55 | [TK]D-Fender | codefreeze-lap: No, its kinda a constant state :) |
| 14:54.55 | [TK]D-Fender | codefreeze-lap: And its not so much the idea of AEL its the fact that its limited by extensions.conf and can only offer less. Now mind you I'm not too happy about extensions.conf either, its just lets me do more :) |
| 14:55.37 | riddlebox | [TK]D-Fender, if I have the files for app_rxfax.c and app_txfax.c installed correctly I should see them in core show application rxfax right? |
| 14:55.57 | codefreeze-lap | [TK]D-Fender: Someday, when I get a chunk of time, I'd like to at try to make you a satisfied customer. I haven't forgotten your suggestions (er, complaints)... |
| 14:56.41 | dominic1 | I think if you are programming in other languages it's more easy to write a dialplan in ael than in extension.conf - format.... |
| 14:57.15 | ManxAway | I like the idea of AEL, but since I use 1.2, it's not really an option at this point. |
| 14:57.18 | *** join/#asterisk seanmh (i=seanmh@216.31.101.25) |
| 14:57.44 | [TK]D-Fender | codefreeze-lap: Other peoples problems offer me plenty of reason not to. Difficulty in debugging in CLI to backtrace problems is another. |
| 14:58.32 | [TK]D-Fender | dominic1: I've been programming for over 25 years now. Extensions.conf is jsut fine with me as far as our options go. |
| 14:58.49 | codefreeze-lap | [TK]D-Fender: Hmmm! that's a good point! |
| 14:58.53 | [TK]D-Fender | riddlebox: if you've done it all right, compiled and installed, yes |
| 14:59.01 | coppice | 25 years? youngsters |
| 14:59.31 | codefreeze-lap | coppice: how long have you been programming? |
| 14:59.34 | [TK]D-Fender | codefreeze-lap: if your code WORKS in AEL and you don't have to deal with it, AEL is just fine. Its the moment something goes wrong I don't want to think about. |
| 14:59.56 | Great_Anta_Baka | is the HFC-4S card the same as a b410p? |
| 14:59.56 | *** join/#asterisk wonderworld (n=ww@ip-62-143-163-199.hsi.ish.de) |
| 15:00.05 | coppice | about 40 |
| 15:00.06 | riddlebox | [TK]D-Fender, the only file I cannot find is apps_makefile.patch, I have found the other two, but on the wiki where it says the files should be, they arent |
| 15:00.26 | [TK]D-Fender | riddlebox: And where did you find those apps? |
| 15:00.48 | codefreeze-lap | coppice: I started as a teen in maybe '74 |
| 15:00.50 | seanbright | hasn't even been alive for 40 |
| 15:00.53 | riddlebox | [TK]D-Fender, http://www.voip-info.org/wiki/view/app_rxfax+and+app_txfax |
| 15:01.30 | riddlebox | the AGX Extra Addons for Asterisk |
| 15:01.35 | [TK]D-Fender | riddlebox: bad place. |
| 15:01.41 | riddlebox | crap |
| 15:01.50 | [TK]D-Fender | riddlebox: Sorry, technically a GOOD place. |
| 15:01.56 | LemensTS | heh im only 23 years young |
| 15:02.10 | ManxAway | The wiki is never a "good place". At most it is "the only place" |
| 15:02.16 | ManxAway | For some things |
| 15:02.47 | codefreeze-lap | LemensTS: enjoy it while it lasts. A snap of the fingers, and you're looking for reading glasses! |
| 15:02.47 | *** join/#asterisk kannan (n=kannan@121.247.254.20) |
| 15:02.48 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 15:03.06 | [TK]D-Fender | riddlebox: but you don't need any kind of patch. the tarball you SHOULD have picked up shouldn't need anything else |
| 15:03.21 | ManxAway | usually a divorce attorney before the reading glasses |
| 15:03.29 | riddlebox | [TK]D-Fender, the tarball for spandsp? |
| 15:03.40 | ManxAway | riddlebox: why don't you get it from coppice, he is the one that wrote it and hosts the source code. |
| 15:03.50 | ManxAway | He's the one with the official site for the software. |
| 15:04.27 | riddlebox | [TK]D-Fender, is that soft-switch.org? |
| 15:04.38 | [TK]D-Fender | riddlebox: for AGX |
| 15:04.56 | [TK]D-Fender | riddlebox: and its linked from there |
| 15:06.22 | Great_Anta_Baka | is using two bri cards on the same machine advisable? |
| 15:07.10 | tapic | I examined dial application, but how do I get dtmf from the callee? |
| 15:07.16 | florz | Great_Anta_Baka: at least there is no general reason for not doing so |
| 15:07.43 | [TK]D-Fender | tapic: M() <---- |
| 15:13.29 | dominic1 | I am 23 too and as I hear we are the youngster, I am feeling much better if I have to ask something ;-) |
| 15:13.59 | Great_Anta_Baka | thanks florz |
| 15:15.52 | rwaite | hmm. to anyone's knowledge is there a way to have the extension continue after calling voicemailmain? |
| 15:16.15 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 15:18.02 | rwaite | okay nevermind it does |
| 15:18.15 | riddlebox | [TK]D-Fender, I run into issues building that, and I tried copying the files to the asterisk source and building, but for some reason it wont work, I think I will just give up and continue to use the zap channel I have for the fax machine |
| 15:19.50 | [TK]D-Fender | riddlebox: If yo got it fromt he proper source you don'[t copy anything... |
| 15:20.44 | *** join/#asterisk fogo (n=fogo@rs-69-169-132-200-0003.broadweave.net) |
| 15:22.05 | *** join/#asterisk jpcansa (n=jpbenavi@201.198.58.30) |
| 15:22.23 | [TK]D-Fender | riddlebox: You seem to be mixing up the OLD instructions for when the apps was downloaded separately VS this new BUNDLE that has its own build process |
| 15:22.41 | *** join/#asterisk kensuke_ (i=c829e4f4@gateway/web/ajax/mibbit.com/x-d8151503ba6700b6) |
| 15:23.30 | kensuke_ | Hi, is bad idea user asterisk 1.2.29 and librpri 1.4 and zaptel 1.4? i need use all on 1.2 or 1.4 or i can "mix"? |
| 15:23.42 | jpcansa | Hi, any idea why my channels 8-1 and 3-1 got like this: http://pastebin.com/m6139d48e |
| 15:24.44 | [TK]D-Fender | kensuke_: No, you cannot mix versions like that |
| 15:25.14 | kensuke_ | [TK]D-Fender: ok, thanks |
| 15:25.44 | [TK]D-Fender | jpcansa: because you had one channel dial out another and my guess is that * didn't detect a hangup so its jsut sitting there. |
| 15:26.45 | jpcansa | [TK]D-Fender: didnt detect a hung up on the extension side or on the telco side? |
| 15:26.55 | jpcansa | *hang |
| 15:26.58 | tzafrir_laptop | riddlebox, I if you want RxFax that builds with latest spandsp: http://svn.debian.org/wsvn/pkg-voip/asterisk-spandsp-plugins/trunk/ |
| 15:27.16 | *** join/#asterisk StooJ (n=stooj@johnston37.plus.com) |
| 15:27.34 | [TK]D-Fender | jpcansa: Either side. |
| 15:28.29 | x86 | wtf is astcanary? |
| 15:28.30 | jpcansa | [TK]D-Fender: that was a dial out from the extension, so that means is not detecting a hang up from the telco? |
| 15:28.45 | [TK]D-Fender | jpcansa: Tahts what I jsut said |
| 15:28.54 | x86 | I started asterisk 1.6, and I see this is ps |
| 15:28.55 | x86 | root 9123 0.0 0.3 2604 808 tty8 S+ Jul27 0:00 astcanary /var/run/alt.asterisk.canary.tweet.tweet.tweet |
| 15:29.09 | [TK]D-Fender | x86: Sounds like another bird-brained idea ;) |
| 15:29.25 | x86 | hah |
| 15:29.41 | x86 | no results on google except some source code |
| 15:29.47 | [TK]D-Fender | x86: http://svn.digium.com/view/asterisk/team/russell/data_retrieval/utils/astcanary.c?view=log&pathrev=120108 |
| 15:30.10 | [TK]D-Fender | x86: was the first bloody result in http://www.google.ca/search?hl=en&q=astcanary&btnG=Google+Search&meta= |
| 15:30.20 | *** join/#asterisk wonderworld (n=ww@ip-62-143-163-199.hsi.ish.de) |
| 15:30.23 | x86 | actually http://svn.digium.com/view/asterisk/tags/1.6.0-beta1/utils/astcanary.c?view=log |
| 15:30.29 | x86 | that was the first result I got |
| 15:30.39 | x86 | 10:29 < x86> no results on google except some source code |
| 15:31.00 | x86 | yeah but that doesn't give me much info ;) |
| 15:31.02 | jpcansa | [TK]D-Fender: i dont understand why it uses 2 channelsto dial out: this is my zap sialout in extensions.conf [ext-zap] |
| 15:31.03 | jpcansa | exten => _XXXXXXXX,1,Answer() |
| 15:31.03 | jpcansa | exten => _XXXXXXXX,n,Set(CALLERID(name)=GrupoOrosi) |
| 15:31.03 | jpcansa | exten => _XXXXXXXX,n,Set(CALLERID(num)=22595895) |
| 15:31.03 | jpcansa | exten => _XXXXXXXX,n,ChanIsAvail(Zap/g1&Zap/g2&Zap/g3) |
| 15:31.04 | jpcansa | exten => _XXXXXXXX,n,NoOP(${AVAILCHAN}) |
| 15:31.06 | jpcansa | exten => _XXXXXXXX,n,Dial(${AVAILCHAN}/${EXTEN}) |
| 15:31.08 | jpcansa | exten => _XXXXXXXX,n,Hangup() |
| 15:31.10 | jpcansa | oopp sorry |
| 15:31.31 | [TK]D-Fender | jpcansa: Luck it finished before your ass skikked out of here :) |
| 15:31.35 | [TK]D-Fender | skidded* |
| 15:31.59 | [TK]D-Fender | jpcansa: and you don't jsue chanisavail with groups. you don't NEED it |
| 15:32.15 | jameswf-home | ~rtfc |
| 15:32.16 | jbot | i heard rtfc is read the fine code |
| 15:32.18 | [TK]D-Fender | jpcansa: and groups aren't channels anyways |
| 15:32.30 | x86 | groups are groups of channels |
| 15:32.37 | Qwell | jameswf-home: I like rtfsc better |
| 15:32.39 | [TK]D-Fender | x86: Read the ^%$# obvious paragraph at the enf of that link I gave you |
| 15:32.44 | *** join/#asterisk n3hxs (n=HAMming@151.196.87.132) |
| 15:33.14 | jpcansa | [TK]D-Fender: chanisavail is the reason it is using 2 channels? |
| 15:33.16 | x86 | jpcansa: a better idea ould be putting all the channels from g1, g2, and g3 into a single group, then dial on the single group... asterisk will hunt down a channel for you automagically |
| 15:33.28 | x86 | [TK]D-Fender: yeah, I did... I want more info than that ;) |
| 15:33.30 | *** join/#asterisk neurosys (n=neurosys@adsl-067-032-132-060.sip.bct.bellsouth.net) |
| 15:33.31 | [TK]D-Fender | x86: Add a canary process, for high priority mode (asterisk -p) to ensure that if Asterisk goes into a busy loop, the machine will be recoverable. We'd still need to do a restart to put Asterisk back into high priority mode, but at least a reboot won't be required. (Closes issue #11559) |
| 15:33.37 | x86 | yes, I can read :) |
| 15:33.46 | x86 | obviously, you can too... yay! |
| 15:33.49 | [TK]D-Fender | x86: DUBIOUS :p |
| 15:33.57 | x86 | gives [TK]D-Fender a cookie |
| 15:34.05 | [TK]D-Fender | jpcansa: just DON'T ok? |
| 15:34.35 | [TK]D-Fender | jpcansa: make your channels members of MULTIPLE groups in zapata.conf |
| 15:34.36 | x86 | jpcansa: the answer to your question, is that [TK]D-Fender is not sure why it's using two channels, but you're still doing it wrong ;) |
| 15:34.43 | [TK]D-Fender | jpcansa: and use that superset group. |
| 15:35.03 | x86 | [TK]D-Fender: didn't know you could have a channel in multiple groups... that's kinda rad |
| 15:35.20 | [TK]D-Fender | x86: True I have no proof as to why they got that way, but show me another probelm and I'll happily roast you separately for it :) |
| 15:35.34 | jpcansa | ok thanks, i´ll try that |
| 15:36.06 | [TK]D-Fender | x86: Its buried deep in sample configs, the WIKI page for zapata.conf and a bunch of other "z0mg I really should read these" sources. |
| 15:36.59 | x86 | [TK]D-Fender: you're awful snippy today... got your panties in a wad? :) |
| 15:37.33 | tapic | [TK]D-Fender : I use macro now, thanks, in macro I just play a file, *CLI reports that the file has played but I do not hear it |
| 15:38.00 | *** join/#asterisk bminish (n=bminish@brenbox.westnet.ie) |
| 15:38.09 | jameswf-home | I greped the asterisk source for dirty words and found none... I think asterisk lacks some of the spirit of open source such as disgrunteled sailors who code |
| 15:40.10 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:40.38 | coppice | that's corporate open source for you |
| 15:41.01 | x86 | coppice: i've seen some nasty comments from coders at IBM in the kernel ;) |
| 15:41.15 | x86 | among other corporate linux contributors |
| 15:42.34 | Qwell | coppice: mind a PM? |
| 15:43.47 | swiftkick | [TK]D-Fender: so I've definitely duplicated the issue that I mentioned last night with using users.conf over sip.conf (as asterisk-gui seems to prefer). |
| 15:43.59 | coppice | if the PM is Gordon Brown, then yes I do mind |
| 15:44.03 | swiftkick | and im wondering if you or anyone here could explain this to me: |
| 15:44.15 | Qwell | I don't know what that means, so I guess I'm clear :p |
| 15:44.22 | swiftkick | [TK]D-Fender: its a two line phone, two extensions, 39 and 201 |
| 15:44.26 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 15:44.26 | swiftkick | [Jul 30 10:36:29] NOTICE[29731]: chan_sip.c:13815 handle_request_invite: Failed to authenticate user "SB" <sip:41@192.168.1.254>;tag=001da266d7ed13b66d0da142-4f495b7e [Jul 30 10:36:37] WARNING[29731]: chan_sip.c:8377 check_auth: username mismatch, have <39>, digest has <201> |
| 15:45.16 | swiftkick | this is a result of starting with a blank sip.conf and letting asterisk-gui populate users.conf as you suggested |
| 15:45.59 | [TK]D-Fender | swiftkick: Feel free to pastebin your sip.conf and I hope you have WIPED users.conf for this... |
| 15:46.42 | *** join/#asterisk foexle (n=heiko@p5DC5E8FA.dip.t-dialin.net) |
| 15:46.56 | swiftkick | [TK]D-Fender no, its the other way around. you said "why not do the entire config with asterisk-gui" |
| 15:47.07 | swiftkick | [TK]D-Fender admittedly, that was before you suggested it was better i blow off the gui entirely |
| 15:47.18 | swiftkick | so i have a fully populated users.conf setup by the gui |
| 15:47.48 | swiftkick | there is really no good reason why there is no documentation on the wiki, and nobody in the defacto #asterisk channel can explain this error, there are even open trouble tickets about this issue on the bug tracker |
| 15:47.52 | LemensTS | swiftkick: is users.conf how asteriskNOW populates its sip users? |
| 15:48.04 | swiftkick | LemensTS: yes |
| 15:48.08 | [TK]D-Fender | swiftkick: Ok, well do a dump of "sip show peer [peerwithoutbraces]" for each |
| 15:48.35 | [TK]D-Fender | swiftkick: And of course include the full debug of your failed attempts |
| 15:49.03 | [TK]D-Fender | Qwell: Tony Blair's replacement |
| 15:50.07 | riddlebox | tzafrir_laptop, when I download it where do I put it ,in the asterisk source or spandsp source? |
| 15:50.10 | swiftkick | ~pb |
| 15:50.11 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines 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 |
| 15:51.15 | jameswf-home | keeps his source in /usr/src (seems logical) |
| 15:51.29 | tzafrir_laptop | basically in the asterisk source |
| 15:51.50 | tzafrir_laptop | but note that you need to apply the patch from debian/patches for it to actually build |
| 15:52.03 | tzafrir_laptop | app_fax.c it just the file from 1.6.0 |
| 15:52.08 | *** join/#asterisk Blackvel (n=blackvel@dslb-084-057-082-239.pools.arcor-ip.net) |
| 15:52.15 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 15:53.48 | Blackvel | hey all. trying to get bristuff 0.3.0 with asterisk 1.2 and zaptel 1.2.26 running on old linux system with 2.4 kernel. is there any EASY way just build zaptel module without all the card modules? found a note on bugs.digium.org that those card modules only work on zaptel 1.2. on kernel 2.6 |
| 15:54.25 | Blackvel | do I really have to do a full system upgrade to centos 5 with kernel 2.6? I don't want to :) |
| 15:56.25 | kannan | hello, to configure FXO card (TDM2400p) with FXO modules , the siganalling in zapata.conf shold be fxs_ks, is this correct? |
| 15:56.31 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 15:56.50 | [TK]D-Fender | kannan: Typically yes. |
| 15:57.50 | kannan | [TK]D-fender , thank you. I am getting the calls apparently as per the Asterisk CLI output, but the called party phone doesnt ring at all |
| 15:58.01 | *** part/#asterisk korihor (n=korihor@190.39.163.45) |
| 15:58.03 | *** join/#asterisk torrikft (n=afj@m85-94-184-212.andorpac.ad) |
| 15:59.45 | [TK]D-Fender | kannan: PASTEBIN is your friend. |
| 15:59.57 | kannan | http://pastebin.ca/1087277 |
| 16:02.03 | jameswf-home | kannan: your context is incorrect |
| 16:02.15 | [TK]D-Fender | kannan: Would be nice to see the FAILURE in there... like as in the WHOLE CALL. |
| 16:02.41 | kannan | [TK]D-fender , oh o, I will do it now |
| 16:02.45 | tzafrir_laptop | Blackvel, why do you need it? |
| 16:02.54 | jameswf-home | kanan, inbound calls would start with s on analog |
| 16:03.50 | Blackvel | tzafrir_laptop: bristuff 0.3? have 0.2 running but wanted to upgrade. afaik zaphfc needs zaptel (basic) module, e.g ztcfg command |
| 16:04.57 | tzafrir_laptop | Blackvel, anyway, just run ./download.sh to setup everything |
| 16:05.16 | tzafrir_laptop | If you want you can later build specific subdirectories |
| 16:05.24 | Blackvel | but with kernel 2.4 all the modules like wct4xxp, wctc4xxp fail with compile errors |
| 16:05.56 | kannan | [TK]D-fender , http://pastebin.ca/1087302 |
| 16:06.12 | kannan | jameswf-home , i am trying only outgoing for now |
| 16:06.17 | Blackvel | i am not a Makefile/Linux specialist, but it takes more just to fix MODULES:= zaptel variable in Makefile :) |
| 16:06.57 | kannan | jameswf-home , i ill re-check the context , i had it as test1 |
| 16:07.02 | kannan | will |
| 16:07.20 | Blackvel | one of the reason why i was a bit scared about upgrade as my linux system is too old :) |
| 16:07.50 | kannan | jameswf-home , could you tell me he specific error, i am not able to see it |
| 16:08.00 | Blackvel | could manually fix two errors, one in zaptel-base.c and gnu make 3.8 bug |
| 16:08.14 | jameswf-home | hmmm Unable to re-open DSP device /dev/dsp: No such file or directory |
| 16:08.40 | Blackvel | is asterisk 1.2.x and zaptel 1.2 even supporting officially linux kernel 2.4 or only 2.6 anymore? |
| 16:08.43 | jameswf-home | other than that looks as if it worked |
| 16:09.53 | jameswf-home | Blackvel: I think all the compat stuff is there but let go man why do you still un 2.4 |
| 16:09.55 | [TK]D-Fender | kannan: disable callprogress & busydetect |
| 16:10.36 | kannan | jameswf-home , that always shows up when the sound card is nt built and i do the console dial, but if i cald a sip phone or a number thru a PRI cad , it works, here it says ringing , but the called party phone does not ring |
| 16:11.08 | kannan | [TK]D-fender , i will try it |
| 16:11.19 | jameswf-home | kannan: if you watch zttool what does it do |
| 16:11.38 | tzafrir_laptop | Blackvel, zaptel 1.4 also supports kernel 2.4 |
| 16:11.44 | kannan | i am not able to make zttool |
| 16:11.49 | *** join/#asterisk pbrown985 (n=na@wh-gtw-0001.woolfharris.com) |
| 16:11.51 | kannan | it says some newt error |
| 16:12.01 | jameswf-home | kannan install curses |
| 16:12.05 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
| 16:12.06 | tzafrir_laptop | some of the drivers don't support it, but the specific one you need is not one of them |
| 16:12.10 | Qwell | or...install newt :D |
| 16:12.13 | Qwell | newt-devel |
| 16:12.31 | kannan | jameswf-home , @Qwell, thanks , i had done yum install newt |
| 16:12.33 | kannan | only |
| 16:12.43 | kannan | brb in a few :) |
| 16:13.00 | tzafrir_laptop | Blackvel, BTW: what distribution do you use? |
| 16:14.07 | Blackvel | don't ask :) you would be laughing hehe |
| 16:14.15 | Blackvel | this system is from 2004/2005 |
| 16:14.39 | Blackvel | and now i want to integrate IVR for my telco pbx for my little contracting company |
| 16:15.02 | tzafrir_laptop | 2005 and it was kernel 2.4? |
| 16:15.08 | Blackvel | RH 9 :) |
| 16:15.21 | Blackvel | tooooo old |
| 16:15.23 | Qwell | Blackvel: upgrade |
| 16:15.39 | tzafrir_laptop | It was too old in 2004 |
| 16:16.01 | Blackvel | I know....I read about recommendations for RH + asterisk + epia Via |
| 16:16.20 | Blackvel | but re-install costs me time ;) |
| 16:16.37 | Qwell | Blackvel: is this server on the internet anywhere? |
| 16:16.49 | *** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:16.49 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:17.09 | Blackvel | nat. behind linux firewall at my home |
| 16:17.51 | *** join/#asterisk shtoom (n=shtoom@121.246.167.147) |
| 16:18.06 | *** join/#asterisk angeldavid (n=angeldav@nelug/coreteam/pepo) |
| 16:19.36 | Blackvel | maybe I should just do a fresh centos 5 install at weekend and stop wasting time with workarounds/bugfixes and kernel 2.4 probs ;) but I had hope that there is a |
| 16:19.50 | Blackvel | an easy way to leave out some modules of an Makefile ;) |
| 16:20.09 | tzafrir_laptop | Blackvel, sure, you can edit the makefile |
| 16:20.13 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 16:20.18 | [TK]D-Fender | [12:08]<Blackvel>is asterisk 1.2.x and zaptel 1.2 even supporting officially linux kernel 2.4 or only 2.6 anymore? <- I was running 1.4 on Kernel 2.4 for a LONG time just fine. |
| 16:20.37 | *** join/#asterisk martha (n=martha@bzq-179-135-226.static.bezeqint.net) |
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| 16:24.59 | *** join/#asterisk macros73 (n=cs@dsl093-063-232.pit1.dsl.speakeasy.net) |
| 16:25.26 | dominic1 | I have a little problem with asterisk 1.4 and states. I want to monitor them via ami. Now I saw that if a queue is called the newstate event shows me the callerid of the person who called the queue. If the dialcommand is called it shows me the callerid of the extension which initiates the dialcommand. Is there any possibility to change the behaviour from one of these two? |
| 16:27.07 | *** join/#asterisk drumkilla (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:27.07 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 16:27.32 | *** part/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:27.34 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:30.47 | *** join/#asterisk cabbiepete (n=cabbiepe@83-244-133-169.cust-83.exponential-e.net) |
| 16:37.36 | *** join/#asterisk LND (n=Lee@89.192.150.56) |
| 16:37.50 | kannan | Regarding the TDM2400P with FXO mdules , I have a doubt now. Should I have got FXS mdules? I am connecting to Fixed Cellular Terminals that provide a RJ11 cable. The FCT has a GSM SIM card each. If i connect a Analog hne instrument to the FCT it works fne |
| 16:37.58 | kannan | analog pone |
| 16:38.00 | kannan | phone |
| 16:38.01 | *** join/#asterisk shtoom (n=shtoom@123.176.41.46) |
| 16:39.07 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:41.34 | [TK]D-Fender | kannan: FXO (red) modules are what you should be using. |
| 16:41.47 | tzafrir_laptop | kannan, if you can connect a normal analog phone instead of it, you need an FXO module |
| 16:42.25 | *** join/#asterisk InfiL00p (n=bowserj@S0106000c418bd9f7.vc.shawcable.net) |
| 16:43.57 | LND | Hi all, anyone had any experience of running asterisk on a eeepc (701)? |
| 16:44.30 | *** join/#asterisk korihor (n=korihor@201.211.168.130) |
| 16:44.30 | LND | have installed xubuntu and could get v1.4.10 from the ubuntu repositories |
| 16:45.24 | Qwell | LND: 1.4.10 is very old - you should install from source |
| 16:45.49 | keith4 | there must be a newer package than that for ubuntu, no? |
| 16:46.13 | tzafrir_laptop | http://packages.ubuntu.com/asterisk |
| 16:46.23 | keith4 | yah, even hardy has 1.4.17 |
| 16:46.34 | tzafrir_laptop | Or rebuild your own from the pkg-voip repo |
| 16:47.03 | tzafrir_laptop | run debian/backports/gutsy |
| 16:48.09 | shtoom | ManxPower:I've rebooted many times but this problem comes back again |
| 16:48.20 | LND | ok. Is the 600MHz chip upto the job.. Say 4 cncurrent calls SIP? |
| 16:48.45 | Blackvel | interesting |
| 16:49.29 | *** join/#asterisk bazilek (n=bazil@mail.generation-p.com) |
| 16:49.30 | shtoom | ManxPower:Also canreinvite is set to no in users.conf |
| 16:49.34 | jks | how would I move calls from one queue (app_queue) to another from the manager interface? |
| 16:50.16 | Qwell | jks: I don't think you can do that very easily. maybe some type of transfer...hmm |
| 16:50.45 | Qwell | I guess you could, you would just have to understand it fairly well |
| 16:50.55 | kannan | [TK]D-fender , tzafrir_laptop , anks |
| 16:50.58 | kannan | thanks |
| 16:51.09 | jks | Qwell, hmm, okay :-( sounds complicated |
| 16:51.15 | Qwell | jks: slightly |
| 16:51.33 | jks | Qwell, my goal is to be able to move them one queue to another while keeping everyone in the same position as before |
| 16:51.45 | Qwell | I can't think of the manager action name...redirect? something like that |
| 16:51.53 | Qwell | jks: now THAT is quite a bit trickier |
| 16:52.03 | Qwell | You'd need to setup your queues to have priorities and such.. |
| 16:52.13 | jks | yep, hmm |
| 16:52.21 | jks | perhaps I should start out just being able to move them |
| 16:52.25 | jks | and then do positions afterwards |
| 16:54.01 | legis | If my ISTP offers G.729, * can communicate with G.729 and my ATAs have G.729, I don't need a license for that right? |
| 16:54.45 | Qwell | legis: as long as you don't use any applications that might need to transcode (such as meetme) |
| 16:55.53 | legis | Qwell: ah I see, thx. |
| 16:56.12 | Blackvel | tzafrir_laptop: do you know how to re-write the Makefile just to compile the zaptel module and leave out all others? :) must be a RH9 2.4 problem then. http://bugs.digium.com/view.php?id=12220 |
| 16:57.18 | tapic | <[TK]D-Fender> : M() execute after the parties get bridged...what can I be doing wrong... |
| 16:57.24 | tzafrir_laptop | Blackvel, look for MODULES in Makefile |
| 16:57.35 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 17:02.38 | [TK]D-Fender | tapic: No, it gets called BEFORE they are bridged |
| 17:06.25 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 17:11.36 | Blackvel | right. uncommented the whole MODULES line and wrote my own. but still some modules are built which shouldn't be (wct4xxp) |
| 17:12.04 | Blackvel | hmmm... GOT IT. found a further modules prereq line which has for kernel 2.4 hardcoded wct4xxp in it ;) |
| 17:14.47 | *** join/#asterisk c4t3l (n=rcallico@mail.questia.com) |
| 17:16.44 | Blackvel | hmm... again an error :) now it wants to install fix wct4xxp. fixed... |
| 17:19.01 | Blackvel | crashes again.../fpga_load.c:236: undefined reference to `usb_get_string_simple in zaptel/xpp/utils |
| 17:19.02 | tapic | I use M() in Dial application and it gets executed after the parties have been bridged not before as the documentation says. Any ideas? |
| 17:19.04 | Blackvel | i give up :) |
| 17:25.53 | *** join/#asterisk Nugget (i=nugget@carrera.macnugget.org) |
| 17:26.02 | [TK]D-Fender | tapic: Documentation is not wrong. Its used for privacy functions expressly for that reason |
| 17:26.17 | [TK]D-Fender | tapic: PASTEBIN your diaplan and your CLI output for the failed attempt |
| 17:27.50 | *** join/#asterisk |||Mad||| (n=mad@mail.rubbusa.com) |
| 17:28.37 | |||Mad||| | Hi, all... quick question that I have been unable to find the answer to: How can I reduce the number of rings coming in to my analog card before Asterisk picks up? |
| 17:29.30 | [TK]D-Fender | |||Mad|||: Remove any delays you put in, and then relaize * waits for callerid typically <- |
| 17:30.02 | *** join/#asterisk torrikft (n=afj@m194-158-76-176.andorpac.ad) |
| 17:30.31 | |||Mad||| | Ahh! So if I disable callerid in Zapata.conf, that will help |
| 17:30.47 | |||Mad||| | The only delay I have is after Answer(), so that shouldn't affect it, right? |
| 17:32.10 | |||Mad||| | Wow, the callerid was it... picks up immediately now!!! Thanks a lot!! |
| 17:33.02 | |||Mad||| | Perfect, that's just what I needed... have a great day!! |
| 17:36.28 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 17:36.45 | [TK]D-Fender | pities the moron who pays for callerid and then IGNORES IT |
| 17:39.25 | Nugget | Last night, I finally gave up on imap voicemail storage. |
| 17:39.28 | Nugget | it's just too unstable. |
| 17:45.45 | Kobaz | is there a way to check if a context,extension,priority exists before jumping to it |
| 17:46.23 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 17:48.18 | *** part/#asterisk DSpair (n=D-Spare@163.muaa.syrc.chcgil24.dsl.att.net) |
| 17:52.56 | jblack | kobaz; There's the i extension if you go to something that doesn't exist. Perhaps you could use that with gosub. |
| 17:55.55 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 17:56.04 | *** join/#asterisk sucituanbo (n=blah@c-24-21-121-148.hsd1.mn.comcast.net) |
| 17:56.07 | [TK]D-Fender | Kobaz: "core show application chanisavail" <- |
| 18:00.10 | *** join/#asterisk i3inary (n=i3inary@cpe-66-91-231-125.san.res.rr.com) |
| 18:00.11 | *** join/#asterisk PakiPenguin (n=junaid@linuxpakistan/admin/pakipenguin) |
| 18:00.13 | PakiPenguin | hello |
| 18:01.59 | *** join/#asterisk angeldavid (n=angeldav@nelug/coreteam/pepo) |
| 18:11.58 | Blackvel | ~centos52bug |
| 18:11.59 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889 |
| 18:14.18 | *** join/#asterisk Man0l0 (n=Man0l0@246.Red-80-24-126.staticIP.rima-tde.net) |
| 18:16.11 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-198-76-rrdg-esr-2.dynamic.isadsl.co.za) |
| 18:20.48 | *** join/#asterisk Th3_KID (n=757@adsl-074-183-129-024.sip.gsp.bellsouth.net) |
| 18:21.40 | *** part/#asterisk Th3_KID (n=757@adsl-074-183-129-024.sip.gsp.bellsouth.net) |
| 18:22.42 | *** join/#asterisk angryuser (n=angryuse@88.140.123.21) |
| 18:23.02 | angryuser | fender are you here ? |
| 18:23.15 | *** join/#asterisk scampbell (n=scampbel@199.105.195.156) |
| 18:24.41 | angryuser | i have set simple script which uses 'asterisk-rx database put value key 1' when it is executed manully, no problems, but when set with crontab, the database gets not update |
| 18:25.10 | angryuser | i suppose some right's need to be set, but where ? |
| 18:26.23 | [TK]D-Fender | angryuser: Suppose you need to actually WACTH whats happening. |
| 18:26.58 | [TK]D-Fender | angryuser: If its crontab, go look at those users. |
| 18:27.27 | angryuser | i did, i am sure that when i execute my script manually, it works, and under crontab it dont, i recorded all in log |
| 18:27.55 | angryuser | i used crontab -e , i suppose the user used in that case is 'root' ? |
| 18:27.58 | [TK]D-Fender | angryuser: perhaps you should be paying more attention to users, etc. |
| 18:28.05 | [TK]D-Fender | ~assume |
| 18:28.06 | jbot | About assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight"e=assume.txt&file=assume.wav |
| 18:28.07 | angryuser | i was looged as 'root' |
| 18:28.07 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 18:28.55 | angryuser | so it is not ;) |
| 18:29.04 | angryuser | i will ask in #debian |
| 18:29.13 | angryuser | join #debian |
| 18:29.31 | Kobaz | [TK]D-Fender: ChanIsAvail isn't it |
| 18:29.51 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:30.00 | Kobaz | jblack: if you use goto, it changes the context before it runs any invalid handler |
| 18:30.14 | Kobaz | jblack: so i do have an 'i' exten, but that never gets hit |
| 18:30.23 | jblack | I'm thinking you can use Gosub, then set the i extension to return. |
| 18:30.42 | Kobaz | mmm |
| 18:30.46 | Kobaz | yeah that looks interesting |
| 18:30.53 | [TK]D-Fender | Kobaz: Chanisavail IS it and I've done it myself <- |
| 18:31.06 | Kobaz | <PROTECTED> |
| 18:31.11 | Kobaz | i'm not checking a channel |
| 18:31.19 | Kobaz | i'm checking a context,exten |
| 18:31.27 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 18:31.33 | [TK]D-Fender | Kobaz: It certainly CAN be, you just don't know *'s channel types well enough |
| 18:31.37 | Kobaz | or is that behavior not documented? |
| 18:32.01 | Kobaz | [TK]D-Fender: so you could use local or something? |
| 18:32.13 | [TK]D-Fender | Kobaz: Maybe there IS hope for you... |
| 18:32.20 | Kobaz | heh |
| 18:34.25 | swiftkick | anyone know what might cause the Name portion of Caller ID with Name to start coming in from a provider as "New User" but with the correct telephone number? |
| 18:35.08 | *** join/#asterisk bdolljr (n=bdolljr@64.154.93.78) |
| 18:35.20 | Kobaz | the provider borked something |
| 18:36.23 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 18:36.43 | Kobaz | oooooo |
| 18:36.50 | Kobaz | licks [TK]D-Fender |
| 18:36.59 | Kobaz | ChanIsAvail is lovely |
| 18:37.04 | swiftkick | well actually now that I look at the SIP headers, its coming in correctly from the provider, but * is somehow rewriting the name portion of the SIP header on its way to individual extensions |
| 18:37.12 | swiftkick | wtf |
| 18:37.39 | Kobaz | swiftkick: did you check your sip peers config, you dont have callerid forced in there somewhere |
| 18:37.39 | *** join/#asterisk Tili (n=tili@58.64.90.179) |
| 18:38.10 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 18:39.03 | swiftkick | kopbaz: you mean the peer defined in sip.conf/users.conf representing the inbound trunk??? |
| 18:39.11 | swiftkick | kobaz, rather |
| 18:40.16 | Kobaz | yeap |
| 18:40.29 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 18:41.12 | angryuser | nobody has any idea about crontab not working.... so astdb need some special right's or something ? |
| 18:42.17 | CanWood | "soft hangup SIP/1234-b582345" isn't ending a call. Any tips on something between that and a restart that may solve it? Reload soesn't do it either |
| 18:42.26 | Tili | is it possible to detect call diversion of outbound call on ISDN/Asterisk |
| 18:42.29 | Kobaz | CanWood: what asterisk version |
| 18:42.35 | CanWood | core 1.4.20 |
| 18:42.39 | Kobaz | ah |
| 18:42.52 | Kobaz | i've had problems with earlier 1.4.x where channels would get "stuck" |
| 18:42.59 | Kobaz | not sure about the latest versions |
| 18:44.15 | CanWood | I've had it happen once before, and couldn't solve it without a full restart. I'm trying to avoid that again. The extension can cal lout, but any calls to it go to voicemail and "core show channels" shows two active calls from the extension checking voicemail |
| 18:44.46 | [TK]D-Fender | angryuser: pipe the attempt into a file and see what it says |
| 18:45.19 | [TK]D-Fender | CanWood: Use AMI o redirect the channel to a hangup exten |
| 18:46.06 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 18:46.38 | CanWood | [TK]D-Fender: I haven't used AMI to do that before. could you direct me to some good docs I can read up on regarding doing so? |
| 18:46.47 | [TK]D-Fender | ~book |
| 18:46.48 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 18:46.49 | [TK]D-Fender | ^^^^^^^^^^^ |
| 18:48.39 | Man0l0 | anyone know about this mISDN error: register duplicate 50020100 e5e21200 e5e21200 |
| 18:48.47 | CanWood | looks like I have an old version. The index has no references to AMI. I'll check the pdf for the new version. tks |
| 18:49.11 | swiftkick | kobaz: nope there is nothing defined there that should affect caller ID that i know of |
| 18:52.15 | Kobaz | swiftkick: grep your entire asterisk config directory for "New User" |
| 18:52.26 | Kobaz | swiftkick: that's just very odd, it's gotta be a local setting somewhere |
| 18:53.37 | swiftkick | Kobaz: ummm. in the top of users.conf under [general] we have fullname = New User (????!) |
| 18:54.09 | swiftkick | which has been there for months without causing this effect |
| 18:54.17 | c4t3l | holla |
| 18:54.41 | swiftkick | but i have been forced to change a good twenty settings back and forth in sip.conf users.conf and extensions.conf since this time last week... |
| 18:55.01 | angryuser | by default where is asterisk execute file is stored ? |
| 18:55.18 | c4t3l | /usr/sbin/asterisk |
| 18:55.25 | angryuser | thx |
| 18:55.29 | c4t3l | :) |
| 18:55.36 | Kobaz | swiftkick: that might be it |
| 18:55.44 | Kobaz | swiftkick: have you upgraded asterisk lately? |
| 18:56.09 | [TK]D-Fender | angryuser: "which asterisk" |
| 18:56.09 | wasabi | any experience with asterisk in xen or kvm/ |
| 18:56.13 | swiftkick | kobaz: kobaz: not lately. 1.4.18 due to having a few things setup by asterisk-gui |
| 18:56.53 | CanWood | yeah, I've telnetted in to the AMI console, authenticated fine, and am trying Action: redirect Extension: 2017 to redirect the call to my own extension to get it off the stuck voicemail |
| 18:56.54 | Nugget | telnet is eeeeeeevil! |
| 18:56.55 | CanWood | no joy |
| 18:57.24 | [TK]D-Fender | CanWood: And as usually nothing to show us for the experience |
| 18:57.38 | swiftkick | kobaz: commenting out that fullname directive seems to have resolved the issue. |
| 18:57.42 | CanWood | was avoiding spamming the channel |
| 18:57.44 | Kobaz | swiftkick: weird |
| 18:58.00 | Kobaz | swiftkick: and that's been there the whole time? |
| 18:58.09 | swiftkick | kobaz: yea especially considering it was set by asterisk-gui in users.conf, one of the least documented files of everyting in /etc/asterisk. yep been there since day 1 |
| 18:58.26 | *** join/#asterisk DavidR2008 (n=chatzill@fw1.safedataisp.net) |
| 18:58.29 | Kobaz | i dont use users.conf at all, just sip/iax |
| 18:58.47 | swiftkick | kobaz: apparently asterisk-gui uses users.conf instead of sip.conf |
| 18:58.55 | DavidR2008 | I'm with more off the wall questions :-S |
| 18:59.02 | swiftkick | kobaz: for some reason nobody has ever been able to explain. :) |
| 18:59.07 | DavidR2008 | *back with |
| 18:59.11 | c4t3l | users.conf is more like a template |
| 19:01.31 | swiftkick | c4t3l: you know any online documentation on this??? (users.conf) |
| 19:01.31 | angryuser | finally got it, it was path problem, i have provided a relative, and absolute was needed |
| 19:01.47 | [TK]D-Fender | CanWood: PASTEBIN <- Sure ain't spam |
| 19:02.10 | DavidR2008 | has anyone ran a dial-up server (like a TC1000) off of an Asterisk box? I've got an Asterisk box that has several PRI ports on it. I'm simply bridging an incoming PRI call to another PRI channel on a port connected to my dial-up server. If I call the number I hear modem tones, yet can't connect. I'm able to connect if I connect a telco PRI directly to the dial-up server. |
| 19:02.19 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1279551595.dsl.bell.ca) |
| 19:02.40 | CanWood | I'm already in there TK |
| 19:02.53 | CanWood | just collecting ALL info bfore posting the url. t tks |
| 19:04.35 | CanWood | one attempt is here: http://pastebin.ca/1087480 but I am still reading up on the AMI use as I am not confident in my approach |
| 19:07.53 | c4t3l | swiftkick: i remember seeing it in the source docs |
| 19:07.53 | *** join/#asterisk pabelanger (n=pabelang@OTWAON23-1242541604.sdsl.bell.ca) |
| 19:07.54 | pabelanger | anybody have a url or tutorial using hints and polycom301? besides voip-info.org... thanks |
| 19:08.04 | *** join/#asterisk Blackvel (n=blackvel@dslb-084-057-082-239.pools.arcor-ip.net) |
| 19:08.05 | swiftkick | c4t3l: i would love to see a list of which directives work in sip and which in users. for instance users.conf doesnt seem to support the "register" directive of sip.conf, users.conf does have "hassip" and "registersip" but they dont seem to work the same |
| 19:08.45 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 19:09.07 | swiftkick | c4t3l: also, i cant seem to find documentation on voip-info for the myriad options possible for the insecure= directive. i see some old entries relating to asterisk 1.09 (!!! :-( ) |
| 19:09.52 | Qwell | swiftkick: sip.conf.sample |
| 19:09.58 | swiftkick | c4t3l: here is what the wiki on sip.conf has to say: insecure = very|yes|no|invite|port : Specifies how to handle connections with peers. Default no (authenticate all connections). (invite and port added in v1.2.x). <- 1.2.x ????????? |
| 19:10.05 | *** join/#asterisk bird_of_Luck (i=melifaro@secured.by.ipfw.ru) |
| 19:11.20 | Qwell | swiftkick: as far as registersip, it's because you don't know how to use it :) |
| 19:11.26 | jaytee | <PROTECTED> |
| 19:11.35 | Qwell | What is the exact register line you use in sip.conf (masking usernames/passwords)? |
| 19:11.48 | jaytee | anyone know where I can look up this warning message? |
| 19:12.29 | swiftkick | Qwell: register => 987654321:ZIPPITYDOODAH@some.sip.proxy.net |
| 19:13.00 | c4t3l | swiftkick: my understanding of users.conf is limited (since I don't really use it), but it is a shorthand way of provisioning your server more quickly |
| 19:13.13 | Qwell | username=987654321 \n secret=ZIPPITYDOODAH \n host = some.sip.proxy.net |
| 19:13.46 | CanWood | K. I'mm happy with my AMI syntax now. I run the "Hangup" it reports that it did, yet "core show channels" still indicates it has not hung up. http://pastebin.ca/1087491 |
| 19:14.12 | swiftkick | Qwell: thats exactly what i have in users.conf and it doesnt seem to ever actually register without the explicit register line in sip.conf . |
| 19:14.15 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 19:14.20 | Qwell | do you have registersip=yes? |
| 19:14.23 | swiftkick | Qwell: hassip = yes registersip = yes |
| 19:15.19 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 19:15.43 | c4t3l | swiftkick: i beleive that you must have the registration statement in sip.conf. what version are you running? |
| 19:16.10 | Qwell | swiftkick: double-check your config |
| 19:16.14 | swiftkick | c4t3l: yes no doubt re: users.conf as shorthand method. i just wish someone working on asterisk or asterisk-gui or digium.com or voip-info had a single shred of tangible info on users.conf :) i am running 1.4.18 |
| 19:16.19 | swiftkick | Qwell: I just did, in order to give you that info. |
| 19:16.46 | Qwell | I don't buy it :) |
| 19:16.50 | swiftkick | Qwell: that trunk's SIP configuration and credentials are in users.conf, duplicated nowhere except the single "register" line in sip.conf i quoted above, and without which it doesnt register |
| 19:17.15 | Qwell | rather than hack around it, why not figure out why it isn't working? |
| 19:18.03 | swiftkick | Qwell: i am totally down with figuring out why it isn't working :-) the original suggestion was, if I was going to use the gui at all, then to start with a fresh blank config and let the GUI fill it in. the GUI chose to fill in users.conf |
| 19:18.50 | *** join/#asterisk mandd (n=moo@bas1-toronto61-1279395000.dsl.bell.ca) |
| 19:18.52 | *** join/#asterisk CVirus (n=Satan@82.201.178.105) |
| 19:20.03 | swiftkick | Qwell: one thing I notice is that phones that have multiple lines with separate SIP accounts won't authenticate properly using only entries from users.conf .. these are Cisco phones so I cant change the port on a per account/line basis. adding the corresponding entries for both lines to sip.conf allows it to work. no good explanation why has been forthcoming |
| 19:20.41 | swiftkick | chan_sip.c:8377 check_auth: username mismatch, have <39>, digest has <201> |
| 19:20.41 | swiftkick | [Jul 30 11:08:09] NOTICE[29731]: chan_sip.c:13815 handle_request_invite: Failed to authenticate user "Joe" <sip:201@192.168.1.254>;tag=0019e7d0443dd0930b8f6bdd-2a257f17 |
| 19:21.24 | Qwell | so, you're using an old version of Asterisk. Upgrade. |
| 19:21.31 | CanWood | Anyone have any tips regarding http://pastebin.ca/1087491 before I do a full restart? |
| 19:21.33 | swiftkick | now I'm sure theres probably a setting for insecure = for that specific peer/phone, or perhaps a setting somewhere regarding authentication types, but as yet nobody has been able to suggest any |
| 19:21.57 | swiftkick | Qwell: the latest I can run is 1.4.x the gui doesnt support 1.6.x and besides, show me a changelog that says upgrading addresses this issue. |
| 19:22.05 | Qwell | 1.4.what? |
| 19:22.09 | swiftkick | 1.4.18 currently |
| 19:22.24 | swiftkick | happy to upgrade to the latest if you think it will address this issue |
| 19:22.44 | Qwell | you're using a 6 month old version... |
| 19:22.49 | LemensTS | canwood: you try to hang them up from the cli yet |
| 19:23.36 | CanWood | from a "telnet locathost 5038" I issues the Action: Hangup |
| 19:23.40 | swiftkick | Qwell: give me like 2 minutes and i can be running the latest. wanna bet its still there? there were open issues on the asterisk-gui (or maybe asterisk) bugtracker about this very issue, I replied to one of them suggesting that add the creds to sip.conf fixed the issue |
| 19:24.01 | Qwell | swiftkick: Sure. :) $20 says it's fixed. |
| 19:24.22 | swiftkick | Qwell: but, you know, if asterisk-gui is gonna bother using a file users.conf that is documented exactly nowhere, then thats pretty lame |
| 19:24.33 | russellb | Qwell: 611 fixes to 1.4 since 1.4.18 :) |
| 19:24.34 | c4t3l | swiftkick: if you really wanted the 'low-down' on the users.conf issue, maybe you could try running gdb on the asterisk proc (on startup) |
| 19:24.40 | Qwell | russellb: 104095 |
| 19:25.44 | c4t3l | swiftkick: perhaps strace the output to a file or something. |
| 19:25.47 | swiftkick | c4t3l: thing is, while i could probably find my way around gdb, or do an strace on /usr/sbin/asterisk, these are the kinds of issues i should just be able to find some docs on, if theyre in use on every "production" box running the ostensibly digium-endorsed asterisknow/asterisk-gui |
| 19:27.07 | swiftkick | c4t3l: im not opposed to helping out documenting this stuff, it just strikes me as rather late in th3e game for an end user to have to go digging around a source tree to "infer" documentation for a file in such common use by so closely related a project (in theory) as asterisk-gui |
| 19:27.56 | Qwell | "Parameter names here are the same as they appear in the other configuration files." |
| 19:28.02 | Qwell | I'd say that's extremely well documented |
| 19:28.04 | Strom_M | swiftkick: oh, come on, don't you know the open-source programmer mantra? ALL CODE AND NO DOCUMENTATION FOREVER !!!!!!!!!!" |
| 19:28.26 | Alan_Hicks | I asked this yesterday but didn't get much of a response. Hopefully some one will be able to provide further assistance today. I'm having a bit of difficulty with my asterisk box which I think may be hardware related. I have two PSTN lines hooked up to a TDM400 card with 4 FXO cards. |
| 19:28.35 | swiftkick | Strom: indeed :) :) :) |
| 19:28.44 | swiftkick | Qwell: except thats basically incorrect. |
| 19:28.53 | Qwell | it's...exactly correct. |
| 19:28.54 | swiftkick | Qwell yoiu getting that from what file? |
| 19:28.57 | CanWood | Well, I tested the same syntax on a call that was in progress and it hung up successfully. SO I give up |
| 19:29.02 | CanWood | restarting |
| 19:29.08 | swiftkick | Qewll I just grabbed 1.4.21.2 untarring |
| 19:29.19 | swiftkick | Qwell what file did you find that? users.conf.sample or a source file? |
| 19:29.24 | Qwell | users.conf.sample |
| 19:29.30 | Alan_Hicks | I am using Polycom IP 320 SIP phones. Most of the time everything works as expected, but every few days it is as if the TDM card locks up hard. When I try to call the asterisk box from another PSTN number, I get a busy signal. Any calls out of the asterisk box receive nothing but loud static. |
| 19:30.11 | swiftkick | Qwell: well, i can start listing the directives that users.conf doesnt seem to want to honor. register, callerid, cid_number to start with |
| 19:30.23 | Qwell | register is a general section option |
| 19:30.36 | swiftkick | its documented in sip.conf dox on voip-info |
| 19:30.42 | c4t3l | Alan: have you looked at the pots lines with a line-tester (bud set?) |
| 19:30.44 | Blackvel | is there a way to configure asterisk 1.2.29 and zaptel 1.2.26 to execlude xpp with menuconfig? how can I run this on a very old RH9 system and kernel 2.5? :) |
| 19:30.48 | Alan_Hicks | I have tried placing the PSTN lines on different FXO cards but the problem persists. The only way I have found to the machine back to a working state is to remove the wctdm module from the kernel and reload it, then restart asterisk. Any assistance would be appreciated. |
| 19:30.49 | Kobaz | it's a new major release after like 9 months |
| 19:30.50 | Kobaz | er |
| 19:31.15 | Qwell | kernel 2.5? |
| 19:31.15 | Alan_Hicks | c4t3l: yes. They are fine. Morever I have a few analogue phones attached to the same numbers. |
| 19:31.20 | Alan_Hicks | They all work fine. |
| 19:31.25 | Qwell | no, seriously, stop right now. Upgrade. 2.5 is absolutely not supported. |
| 19:31.47 | Kobaz | haha |
| 19:31.48 | Kobaz | 2.5 |
| 19:31.48 | Kobaz | wtf |
| 19:31.50 | Blackvel | qwell: o.k i meant 2.4 |
| 19:31.53 | Alan_Hicks | Except when this trouble is occuring, they have no sound or dialtone when I pick them up, not even static. |
| 19:32.02 | c4t3l | Alan: are you in houston,tx? This sounds like a problem I ran into before :) |
| 19:32.10 | Kobaz | 2.5 is oooooold (and very experimental) |
| 19:32.24 | c4t3l | what ver of zaptel driver r u using? |
| 19:32.26 | Alan_Hicks | c4t3l: Box is in Macon, Georgia. |
| 19:32.44 | DavidR2008 | anyone running dial-up through asterisk? |
| 19:32.47 | *** join/#asterisk iratik (n=itariki@12-226-116-3.client.mchsi.com) |
| 19:32.47 | Alan_Hicks | zaptel-1.4.10 |
| 19:32.59 | Alan_Hicks | I had the same problem with an older version of the drivers. |
| 19:33.03 | *** join/#asterisk n3hxs (n=HAMming@151.196.87.132) |
| 19:33.11 | iratik | Any recomendations for call recording? Open-source preferably ? |
| 19:33.28 | CanWood | MixMonitor |
| 19:33.29 | Qwell | iratik: Asterisk can record calls. |
| 19:33.30 | Alan_Hicks | c4t3l: Did you find a solution to your problem when you ran into it? |
| 19:33.36 | c4t3l | theres a little know test that you can enable with the zap drivers but it requires a recompile of zap |
| 19:33.38 | *** join/#asterisk LND (n=Lee@89.192.147.128) |
| 19:33.48 | iratik | Qwell: I know that, I need more re-assurance |
| 19:33.51 | *** part/#asterisk pabelanger (n=pabelang@OTWAON23-1242541604.sdsl.bell.ca) |
| 19:34.05 | Alan_Hicks | Recompile is no problem. I actually wrote the build script for Slackware and packaged it all myself, so I'm familiar with it. |
| 19:34.05 | Qwell | re-assurance? |
| 19:34.09 | [TK]D-Fender | iratik: MixMonitor is a good OSS recording option. |
| 19:34.17 | [TK]D-Fender | iratik: There, reassuring enough for you? |
| 19:34.28 | c4t3l | Alan: in my situation, we determined that the building wasn't grounded proberly. the demarc was messed up at the location |
| 19:34.32 | iratik | thats what I already use for recording |
| 19:34.56 | Qwell | iratik: but? |
| 19:35.10 | Alan_Hicks | c4t3l: I wouldn't rule that out in my case either. I'm at a residence, not a place of business, so the demarc ain't exactly the greatest in the world. |
| 19:35.26 | iratik | could never figure out how to get mixmonitor to send the calls somewhere else |
| 19:35.30 | *** join/#asterisk exothermc (n=miles@74.85.89.146) |
| 19:35.37 | iratik | have to use a process to move the calls out of /monitor manually |
| 19:35.38 | c4t3l | lemme dig through my notes and find that test for ya ;) |
| 19:35.46 | Alan_Hicks | c4t3l: Thanks so much! |
| 19:35.48 | exothermc | Any way to get asterisk to handle multiple registrations? |
| 19:35.50 | Qwell | iratik: nfs, cron+scp... |
| 19:36.01 | iratik | then i have to move the associated CRD records using a cron job and a mini mysql dump |
| 19:36.27 | iratik | this cron job moving stuff over is constantly running, taking up resources.... not very efficient |
| 19:37.33 | Alan_Hicks | I was wondering if my analogue phones might not be causing problems for the FXO cards. |
| 19:37.59 | hubguruJR | hello all, any t-38 guru's out there, wanting to setup: fax<>ATA<>*1.4<>Cisco-t.38-gateway |
| 19:38.11 | Alan_Hicks | I basically have three analogue phones plus the FXO line hooked to one PSTN number, and a fax machine and the other FXO card hooked to the secondary PSTN number. |
| 19:38.22 | iratik | I know one exists, its a I remember now,,, ... found it! |
| 19:38.23 | iratik | orecx |
| 19:38.42 | Alan_Hicks | And I'm pretty sure that's not a common configuration. I don't know if it's known to make the FXO cards freak out or not though. |
| 19:39.00 | [TK]D-Fender | iratik: pastebin what you're doing. |
| 19:40.28 | ManxAway | Alan_Hicks: I've had that problem before. It's the reason I stopped using analog cards. Does a reboot fix it? |
| 19:40.38 | c4t3l | Alan: if you look in zaptel<version>/wctdm.c at around line 850 you should see an 'if branch ' with a line containing voltage settings |
| 19:40.52 | *** join/#asterisk Dovid (n=Dovid@tony09-121-90.inter.net.il) |
| 19:40.56 | c4t3l | I'll just pastebin the snippit |
| 19:41.08 | Alan_Hicks | ManxAway: Yes it does, as does rmmod'ing wctdm then loading it back and re-running ztcfg. |
| 19:41.26 | Alan_Hicks | Thanks. |
| 19:41.33 | ManxAway | Alan_Hicks: I don't recall if rmmod/modprobe fixed the issue or not. |
| 19:41.40 | keith4 | are you getting power alarms? |
| 19:41.55 | Alan_Hicks | power alarms? |
| 19:42.01 | Qwell | Alan_Hicks: If you're having issues, I would highly recommend contacting support so they can help you |
| 19:42.07 | ManxAway | It only happened after we ran a fair number of calls thru the system. We just told them to reboot every monday morning. |
| 19:42.39 | Alan_Hicks | Qwell: Thanks. It's my experience that IRC channels usually know much more than phone support, but I'll try that if I don't get satisfaction here. |
| 19:43.00 | ManxAway | Alan_Hicks: If you are prepared to spend a week or two on the issue, let Digium connect into the system and take the system down at random times -- then call Digium support. Expect to spend most of the first day doing stuff you already tried. |
| 19:43.03 | Alan_Hicks | ManxAway: In my case it doesn't seem to matter how many calls. We're very low volume. This is just a dev box. |
| 19:43.23 | ManxAway | Alan_Hicks: then you have a good chance of solving the problem. |
| 19:44.10 | c4t3l | Alan: http://pastebin.com/m4d13985d |
| 19:44.16 | Alan_Hicks | Still, it only occurs once every few days. |
| 19:44.37 | c4t3l | this will enable a voltage readout to syslog (messages) |
| 19:45.10 | c4t3l | this is the second most likely thing that diguim would try anywayz |
| 19:45.34 | c4t3l | the first being a svn checkout of the latest zap stuff |
| 19:45.41 | Alan_Hicks | I'm not seeing anything that matches that. |
| 19:45.51 | Alan_Hicks | printk("Post-leakage voltage: %d volts\n", 376 * vbat / 1000); |
| 19:46.00 | Alan_Hicks | ^^^ That's the closest and it's way off. |
| 19:46.12 | c4t3l | Alan: grep for 'Voltage', anything there? |
| 19:46.54 | Alan_Hicks | http://pastebin.com/d425d9718 |
| 19:48.58 | iratik | [TK]D-Fender: sorry, got distracted |
| 19:49.10 | iratik | http://pastie.org/244333 |
| 19:49.18 | *** join/#asterisk LND (n=Lee@89.192.147.128) |
| 19:49.40 | c4t3l | Alan: gimme a sec i |
| 19:49.45 | c4t3l | i'll brb |
| 19:49.46 | Alan_Hicks | ok |
| 19:49.52 | [TK]D-Fender | iratik: I want to see your DIALPLAN that will do the monitoring. |
| 19:52.43 | Qwell | wonders where swiftkick has gone off to |
| 19:52.54 | swiftkick | Qwell: still here :) |
| 19:53.13 | Qwell | let me know when you want my paypal address |
| 19:53.14 | swiftkick | had to take a phone call. back to installing 1.4.21.2 |
| 19:53.18 | swiftkick | hahaha |
| 19:53.25 | *** join/#asterisk rukus (n=daniel@41.247.180.19) |
| 19:53.25 | ManxAway | I'm answering Non-GUI, non-package questions on #asterisk-cli Others also answer questions there. |
| 19:53.36 | *** join/#asterisk wonderworld (n=ww@ip-62-143-163-199.hsi.ish.de) |
| 19:53.48 | rukus | day three in asterisk world. |
| 19:53.50 | rukus | fun fun |
| 19:53.52 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
| 19:54.00 | Alan_Hicks | You mean people actually use GUIs for their server stuff? |
| 19:54.07 | Alan_Hicks | How Windows NT! |
| 19:54.43 | ManxAway | Alan_Hicks: The poor misguided souls. |
| 19:54.50 | Alan_Hicks | hehehe |
| 19:55.32 | ManxAway | Alan_Hicks: They here here often enough whining about not getting support on the channel for their GUI. I got tired of it and created a channel just for CLI stuff. The GUI people can have this channel for all I care. |
| 19:56.27 | *** part/#asterisk iratik (n=itariki@12-226-116-3.client.mchsi.com) |
| 19:56.33 | jaytee | I think the whole GUI vs NON-GUI topic is just plain sad. |
| 19:56.34 | rukus | oh ! |
| 19:56.36 | Alan_Hicks | Isn't that sort of the equivilant of asking the shadow devs why kdeadmin isn't working? |
| 19:56.43 | rukus | ManxAway: whats that channels name ? |
| 19:56.50 | jaytee | asterisk-cli |
| 19:56.52 | ManxAway | #asterisk-cli |
| 19:56.57 | rukus | the giu messed up my nice neat first test dialplan |
| 19:57.17 | jaytee | thats a feature |
| 19:57.18 | ManxAway | I don't have a real problem with people using GUIs, but they should do it on the correct channel. |
| 19:57.28 | jaytee | Manx, agreed |
| 19:57.29 | [TK]D-Fender | <--|iratik has left #asterisk ("Leaving") <- \o/ |
| 19:57.29 | tzafrir_laptop | ManxAway, huh? #asterisk-cli seems redundant |
| 19:57.41 | ManxAway | Since many of them refuse to do so. |
| 19:57.51 | tzafrir_laptop | unless it excludes the usage of a full-screen text editor |
| 19:57.56 | ManxAway | tzafrir_laptop: I thought that too until the GUI people started diluting this channel |
| 19:58.22 | tzafrir_laptop | ManxAway, and if someone asks here about Realtime? |
| 19:58.44 | ManxAway | tzafrir_laptop: What about it? |
| 19:58.44 | tzafrir_laptop | Or about Capi? or Unistim? |
| 19:58.54 | tzafrir_laptop | are those on-topic here? |
| 19:59.03 | tzafrir_laptop | can *you* answer them? |
| 19:59.04 | [TK]D-Fender | tzafrir_laptop: Unistim? Now you've gone TOO FAR!!!!!!!!!!! |
| 19:59.11 | ManxAway | I'm not going to /kick them from #asterisk-cli for asking. |
| 19:59.29 | jaytee | last night [TK]D-Fender and I were discussing guitars..... I'm still ashamed and feeling guilty. |
| 19:59.43 | ManxAway | tzafrir_laptop: I cannot answer them, but as the GUI users make this channel more and more useless I think more and more people that are TECHNICAL will be on the technical #asterisk-cli channel. |
| 20:00.06 | jaytee | I've never found this channel useless |
| 20:00.14 | [TK]D-Fender | just feels used |
| 20:00.15 | ManxAway | jaytee: not yet.... |
| 20:00.27 | ManxAway | [TK]D-Fender: we are nicer on #asterisk-cli 8-) |
| 20:00.58 | jameswf-home | so your saying #asterisk people are idiots? |
| 20:01.03 | tzafrir_laptop | ManxAway, the point is, that if you're running away from them, it won't help |
| 20:01.08 | [TK]D-Fender | ManxAway: I'm not quite at the point of kicking people... sometimes wish they WOULD cross the line... |
| 20:01.17 | ManxAway | tzafrir_laptop: fighting them does not help and REALLY does not help this channel. |
| 20:01.30 | Alan_Hicks | ManxAway: That's right. Running from them won't fix anything. You need to beat them with a stick! ;-) |
| 20:01.45 | ManxAway | tzafrir_laptop: I used to beat them with a stick here. |
| 20:02.38 | [TK]D-Fender | ManxAway: More like a wet noodle these day... |
| 20:03.12 | ManxAway | [TK]D-Fender: GUI Fatigue. |
| 20:03.27 | [TK]D-Fender | ManxAway: There are meds for that I think... |
| 20:03.28 | ManxAway | [TK]D-Fender: I'm here to help people, not berate them until they go to the correct channel. |
| 20:03.35 | *** join/#asterisk LND (n=Lee@89.192.147.128) |
| 20:04.20 | jameswf-home | heh newb |
| 20:04.38 | ManxAway | [TK]D-Fender: once I stopped getting universal support for getting GUI people out of here then I created the new channel |
| 20:04.58 | jameswf-home | ManxAway: they are like stray cats you feed them they breed and keep coming back |
| 20:05.15 | [TK]D-Fender | "Don't feed the troll" |
| 20:05.40 | seanbright | is a troll |
| 20:05.44 | hardwire | is there a sip spec for not allowing a phone call forward from the phone side? |
| 20:05.53 | [TK]D-Fender | hardwire: No |
| 20:05.56 | jameswf-home | trolls are also like stray cats you feed them they breed and keep coming back |
| 20:06.07 | [TK]D-Fender | hardwire: in SIP the endpoint is "king" |
| 20:06.13 | hardwire | [TK]D-Fender: sweet.. looks like I'm adding another line to that phone. |
| 20:06.24 | hardwire | sigh |
| 20:06.30 | hardwire | that needs to change IMHO |
| 20:07.08 | [TK]D-Fender | hardwire: protocol doesn't tell a phone what to do, its CONFIGURATION does. Read its manual and pray it offers you the choice. |
| 20:07.22 | [TK]D-Fender | hardwire: Another "line" to what "phone"? |
| 20:07.44 | hardwire | well, another line to a SPA-941 that has call-forward turned off |
| 20:07.55 | hardwire | specifically so I don't hit call-forward |
| 20:08.08 | hardwire | go go magic suck eggs. |
| 20:08.14 | *** join/#asterisk tekryan (n=AnonXXX@209.133.228.254) |
| 20:08.26 | hardwire | I may disable call forward all together, then require users to dial a feature code. |
| 20:08.29 | hardwire | hmmm |
| 20:08.32 | hardwire | that would be tasty. |
| 20:08.39 | hardwire | then I could chose what to do, whenever the hell I want. |
| 20:08.42 | tekryan | Hey all |
| 20:08.51 | *** join/#asterisk R_001 (n=kudare@a157-180.adsl.paltel.net) |
| 20:08.59 | *** part/#asterisk bdolljr (n=bdolljr@64.154.93.78) |
| 20:09.01 | tekryan | Need slight help, using asterisk. Whenever someone calls in it hangs up after 'x' seconds |
| 20:09.05 | tekryan | Here are the logs |
| 20:09.23 | tekryan | [Jul 30 15:07:49] NOTICE[2461] chan_sip.c: Registration from '<sip:710@IPHERE>' failed for 'IPHERE' - Device does not match ACL |
| 20:09.34 | jameswf-home | tekryan: NO |
| 20:09.35 | jameswf-home | ~pb |
| 20:09.36 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines 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 |
| 20:09.53 | ManxAway | tekryan: I can try to help you on #asterisk-cli, but you'll have to stop masking the error messages. |
| 20:10.09 | seanbright | ? |
| 20:10.18 | seanbright | bizarre |
| 20:10.20 | tekryan | ManxAway: ok |
| 20:10.43 | jameswf-home | How can Manx be away and here IDK wtf |
| 20:10.58 | ManxAway | I have long arms and good eyesight. |
| 20:11.02 | hardwire | bah.. spa-941's cant disable forwarding |
| 20:11.04 | Alan_Hicks | He's omniprescent. |
| 20:11.09 | c4t3l | back |
| 20:11.16 | Alan_Hicks | c4t3l: Welcome back. |
| 20:11.22 | [TK]D-Fender | tekryan: Registration has nothing to do with calls in progress |
| 20:11.23 | ManxAway | tekryan: the error message indicates you do not have a [701] section of sip.conf |
| 20:11.24 | c4t3l | hello ! |
| 20:11.38 | hardwire | it has a CFWD Notify Serv command. |
| 20:11.45 | hardwire | which seems mostly useless, except kinda useful. |
| 20:11.50 | [TK]D-Fender | ManxAway: ..... caffeinate.... |
| 20:11.55 | c4t3l | well, it turns out that I lied to you Alan |
| 20:11.59 | tekryan | ManxAway: why would it end calls ? |
| 20:12.00 | *** join/#asterisk anthm (n=anthm@mb70736d0.tmodns.net) |
| 20:12.01 | R_001 | hello every body, I want to use telephony system with a java application, found an API called JTAPI, for this operation, but actually, I can't find any usefull references, does anyone have references to help me? ...................., and thanks you in advance |
| 20:12.08 | Alan_Hicks | c4t3l: Liar! :-) |
| 20:12.22 | c4t3l | The voltage trick is in 1.2Zaptel so far as I can see |
| 20:12.35 | Alan_Hicks | Removed for 1.4 huh? |
| 20:12.38 | [TK]D-Fender | R_001: Youre question is entirely too vague. Please thy to rephrase it. |
| 20:13.03 | [TK]D-Fender | tekryan: it wouldn't. That message has nothign to do witha dropped call |
| 20:13.46 | [TK]D-Fender | tekryan: pastebin the ENTIRE CLI output for a failed call at verbose 10, SIP DEBUG enabled. |
| 20:14.29 | R_001 | [TK]D-Fender: I want to use JTAPI, but I can't find references that help me |
| 20:14.52 | Qwell | looks at swiftkick |
| 20:15.14 | hardwire | wazzat |
| 20:15.22 | c4t3l | Alan: i think there is another, but in 1.4 it may be enabled via DEBUG setting in manager.conf but dont qoute me on that just yet |
| 20:15.31 | c4t3l | quote** |
| 20:15.34 | Alan_Hicks | ok |
| 20:16.36 | [TK]D-Fender | R_001: "V0.2 release, 15th May 2006: The current code base is compliant to JTAPI 1.3.1 and tested with Asterisk Version 1.2.4. " <- this version is unsupported and it looks like this is a fairly dead project. |
| 20:16.57 | tekryan | [TK]D-Fender: how exactally do I go about doing what you just asked? |
| 20:17.17 | [TK]D-Fender | tekryan: Go to * CLI. Do "set verbose 10", "sip debug" |
| 20:18.12 | tekryan | [TK]D-Fender: and then try to call in? |
| 20:19.03 | [TK]D-Fender | tekryan: Yes |
| 20:19.19 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 20:19.37 | *** join/#asterisk friendly_fire (n=fraxff@129.21.56.132) |
| 20:19.58 | Alan_Hicks | Well thanks for the help guys, but I gotta get moving. Later. |
| 20:20.05 | c4t3l | Alan: later |
| 20:20.24 | tekryan | [TK]D-Fender: how many of these logs do you want me to put on pb |
| 20:20.36 | tekryan | [TK]D-Fender: I have like a whole page of errors |
| 20:20.39 | [TK]D-Fender | tekryan: I want the entire call from beginning to end |
| 20:21.48 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:21.48 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 20:23.12 | tekryan | [TK]D-Fender: http://pastebin.ca/1087582 |
| 20:24.23 | [TK]D-Fender | [TK]D-Fender>tekryan: Go to * CLI. Do "set verbose 10", "sip debug" |
| 20:24.28 | [TK]D-Fender | tekryan: I said SIP DEBUG <--- |
| 20:24.29 | swiftkick | blinks once |
| 20:24.33 | [TK]D-Fender | tekryan: Clearly not there |
| 20:24.56 | [TK]D-Fender | tekryan: And there is nothing there close to a "call" |
| 20:25.44 | Qwell | swiftkick: so |
| 20:25.56 | tekryan | [TK]D-Fender: ok let me try again |
| 20:27.38 | tekryan | [TK]D-Fender: did same thing, same error |
| 20:28.21 | [TK]D-Fender | tekryan: there is NOT SIP DEBUG in there. |
| 20:28.45 | [TK]D-Fender | tekryan: You have failed to folow simple instructions twice now. Go to * CLI. do "sip debug". PLACE A CALL |
| 20:29.11 | edwin_quijada | Hi! |
| 20:29.15 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 20:29.28 | tekryan | [TK]D-Fender: Command> sip debug / SIP Debugging re-enabled |
| 20:30.35 | *** part/#asterisk CVirus (n=Satan@82.201.178.105) |
| 20:31.03 | *** join/#asterisk albertoandrade (n=alberto@200.195.161.164) |
| 20:31.07 | [TK]D-Fender | ok, time's up. Heading home. BBIAB |
| 20:37.52 | *** part/#asterisk c4t3l (n=rcallico@mail.questia.com) |
| 20:38.11 | *** join/#asterisk VaNNi (n=VaNNi___@38.98.61.142) |
| 20:42.05 | friendly_fire | is it possible for asterisk to be on a vps, and relay incoming voip calls to cell phones? |
| 20:42.16 | friendly_fire | i've just started reading about this, and i'm curious |
| 20:45.17 | ManxAway | I'll be AFK quite a bit, but when I am ATK, I'll be watching #asterisk-cli |
| 20:47.20 | LemensTS | whats vps stand for |
| 20:48.03 | friendly_fire | sorry, a virtual private host. like a co-lo for vm's |
| 20:48.12 | friendly_fire | virtual private server |
| 20:48.21 | LemensTS | vmware? |
| 20:48.28 | friendly_fire | or xen |
| 20:48.38 | friendly_fire | what i mean, is a server off by itself in a datacenter |
| 20:48.41 | LemensTS | how are you going to send the call to the cell phones, pstn or voip |
| 20:48.46 | *** join/#asterisk Defraz (n=T0tal@fw.fuzecore.com) |
| 20:48.58 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 20:48.59 | LemensTS | cuz i dont think you can use pci cards in virtual machine right |
| 20:49.19 | LemensTS | you can do it in voip/out voip |
| 20:49.19 | [TK]D-Fender | LemensTS: pretty much a "no" |
| 20:49.54 | LemensTS | in voip - out voip - cell |
| 20:50.32 | friendly_fire | ok, that helps |
| 20:50.41 | friendly_fire | i will read more, and figure out exactly what i need |
| 20:50.42 | friendly_fire | thanks |
| 20:50.49 | LemensTS | np |
| 20:54.17 | *** part/#asterisk profxavier (n=me@unaffiliated/neverblue) |
| 20:57.26 | *** join/#asterisk spokra (n=spokra@host093-179-146.sea0.speakeasy.net) |
| 21:01.33 | hardwire | so.. hardware echo cancellation on TDM cards reduces interrupts/sec ? |
| 21:01.52 | ManxAway | hardwire: no |
| 21:02.07 | *** join/#asterisk InfiL00p (n=bowserj@S0106000c418bd9f7.vc.shawcable.net) |
| 21:02.46 | legis | How can I refresh SIP peers? |
| 21:03.00 | edwin_quijada | [TK]D-Fender: Thks! for your help yesterday! |
| 21:03.12 | edwin_quijada | hubguruJR: Hi! |
| 21:03.19 | [TK]D-Fender | legis: in what way? |
| 21:03.24 | [TK]D-Fender | edwin_quijada: You're welcome |
| 21:03.35 | edwin_quijada | hubguruJR: Thks! for assist me ! |
| 21:03.51 | legis | [TK]D-Fender: when I do 'sip show peers' is still showing old peers that are offline. |
| 21:04.16 | [TK]D-Fender | legis: Wait for qualify timeout. |
| 21:04.34 | [TK]D-Fender | legis: Or place a call to them |
| 21:04.43 | edwin_quijada | I owe you one! |
| 21:05.13 | edwin_quijada | Can can connect Asterisk using verifone? |
| 21:05.16 | legis | [TK]D-Fender: k, let me try |
| 21:05.43 | edwin_quijada | I have a system like credit card to authorize |
| 21:05.54 | edwin_quijada | i wanna do this using asterisk |
| 21:06.06 | edwin_quijada | ? |
| 21:06.46 | [TK]D-Fender | edwin_quijada: Verifone has nothing to do with telephony. |
| 21:07.12 | [TK]D-Fender | edwin_quijada: You can do whatever you feel like as far as interacting with them goes |
| 21:07.15 | ManxAway | [TK]D-Fender: the credit card machines make calls to the CC network |
| 21:07.35 | ManxAway | verifone is a brand of CC machine |
| 21:08.19 | [TK]D-Fender | edwin_quijada: Keep those machines as far away from * as possible |
| 21:08.38 | bkruse | [TK]D-Fender: heh :] |
| 21:10.29 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 21:11.35 | hardwire | I am learning so much about how people can really screw up everything. |
| 21:12.05 | hardwire | I'd like to thank our sales guy who hired a dude, without my knowledge, to be a dialer for us. |
| 21:12.34 | hardwire | I'd especially like to thank the dialer guy for maxing out or available channels. |
| 21:12.41 | hardwire | for the sake of 2 sales. |
| 21:12.57 | Qwell | swiftkick: so, how about that upgrade? |
| 21:16.49 | edwin_quijada | [TK]D-Fender: I understand but I want create my own network |
| 21:17.06 | [TK]D-Fender | edwin_quijada: And I fail to see what this has to do with * |
| 21:17.21 | edwin_quijada | so Asterisk will be receiving |
| 21:18.02 | edwin_quijada | I need a pull modem?> |
| 21:18.46 | [TK]D-Fender | edwin_quijada: * is NOT a modem and should not be allowed anywhere near these machines |
| 21:18.47 | *** join/#asterisk [Akemi] (n=akemi@206-248-164-136.dsl.teksavvy.com) |
| 21:19.50 | *** join/#asterisk moy (n=moy@nat/ibm/x-9f3db9982769755c) |
| 21:22.12 | swiftkick | Qwell: you still friends with Qwazaar and Denizen Kane? |
| 21:22.22 | *** join/#asterisk torrikft (n=afj@m194-158-76-176.andorpac.ad) |
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| 21:23.20 | tzafrir_laptop | hardwire, dialer? isn't that's what's asterisk is for? |
| 21:23.52 | hardwire | tzafrir_laptop: I guess it wasn't up to me. |
| 21:23.55 | tzafrir_laptop | (both for that and for screwing up badly for everyone, that is) |
| 21:24.11 | hardwire | It's not like we can't do it ourselves. |
| 21:24.16 | hardwire | we already have the list of numbers. |
| 21:24.40 | hardwire | I'm laughing a lot at this campaign Sr. McButtHat set up. |
| 21:25.09 | hardwire | It dials people, gives em some playback, then rings through to our phones. |
| 21:25.28 | hardwire | You answer, and immediately here "TAKE ME OFF YOUR LIST YOU EVIL **@#$! *click*" |
| 21:25.38 | hardwire | I don't think anybody prepared our sales drones for that sort of abuse. |
| 21:25.57 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 21:26.04 | hardwire | tzafrir_laptop: they probably didn't consult me because I would have told them "there are less obnoxious ways to get sales, like naked women." |
| 21:27.02 | *** join/#asterisk op3r (n=Op3r@222.127.152.250) |
| 21:27.47 | *** join/#asterisk angryuser (n=sldf@88.140.123.21) |
| 21:28.13 | swiftkick | Qwell: so when yr not emceeing with typical cats you work for digium? :-) |
| 21:28.48 | swiftkick | takes crosses that thin red line every now and then. |
| 21:28.52 | Qwell | Is that your way of saying "You were right, and I'm not paying up"? ;) |
| 21:29.08 | swiftkick | no actually i upgraded and the sip.conf/users.conf issue is still there, :-) |
| 21:29.13 | *** join/#asterisk tapic (n=b@88.255.77.200) |
| 21:29.40 | swiftkick | or were you referring to the registersip functionality? to test that will have to wait till after 5 PM central, phones up and down here too much during biz hrs already today |
| 21:30.42 | hardwire | pinging sangoma technical people.. gimme a privmsg when you get a chance. |
| 21:30.55 | swiftkick | Qwell: is this you? -> http://www.youtube.com/watch?v=mguuVmCXRo0&watch_response |
| 21:31.00 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 21:31.01 | hardwire | looking for a bit more info on the taps |
| 21:32.57 | op3r | Hello, im having problem with agents and call transfer . Here's the call scenario. agent login to the queue using agentcallbacklogin then answers an inbound call then it will conference to an external number once it is conference, the agent will now transfer the inbound call to that external number and its all good. But the agent will somehow logged off to the queue :( any ideas on why it happens? |
| 21:35.25 | Blackvel | n8 |
| 21:40.56 | *** join/#asterisk dogmeat (n=Bob@unaffiliated/dogmeat) |
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| 21:47.03 | scooby2 | Can anyone recommend a wireless headset that will work with the Linksys SPA941-NA? CEO is still using this ghetto plantronics hl10 that lifts the handset. |
| 21:59.48 | exothermc | When working with realtime registrations how do I clear one? |
| 22:06.45 | tapic | I can not play sound files (CLI reports that the file plays but it does not get heard) from macro when I call macro from dial function inside AGI using M(). Calling macro from an extension is fine. any ideas? |
| 22:08.56 | *** join/#asterisk StooJ (n=stooj@johnston37.plus.com) |
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| 22:11.57 | angryuser | nice kickin starting again ;) |
| 22:12.03 | Nkoji | hello, i cant get asterisknow to stop loading the ztdummy i am unsure of what to do |
| 22:13.48 | *** join/#asterisk Corydon76-dig (i=silver@pdpc/supporter/bronze/Corydon76-home) |
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| 22:13.49 | *** mode/#asterisk [+o Corydon76-dig] by irc.freenode.net |
| 22:15.46 | [TK]D-Fender | tapic: PASTEBIN your dialplan like I told you to hours ago |
| 22:16.45 | tapic | ok |
| 22:18.33 | *** join/#asterisk Tako-san (n=jmkiffia@24.108.192.144) |
| 22:21.58 | Nkoji | Can someone tell me how to stop ztdummy from booting up so i can load the wctdm and detect my hardware (tdm400p) |
| 22:23.38 | Nkoji | how do i stop zaptel? it gives me an error persmission denied |
| 22:23.40 | *** part/#asterisk hubguruJR (n=hubguruJ@mail.ntegratedsolutions.com) |
| 22:24.24 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 22:31.49 | [TK]D-Fender | Nkoji: "rmmod ztdummy" |
| 22:32.16 | *** join/#asterisk BBHoss (n=hoss@c-68-62-170-33.hsd1.al.comcast.net) |
| 22:32.32 | Nkoji | module does bot exist |
| 22:33.00 | [TK]D-Fender | Nkoji: then what makes you think its loading? |
| 22:33.02 | Nkoji | i try /etc/init.d/zaptel stop and i receive Module zaptel in use |
| 22:33.09 | Nkoji | how do i stop zaptel |
| 22:33.25 | [TK]D-Fender | Nkoji: Have you stopped Asterisk first? |
| 22:33.52 | Nkoji | command? |
| 22:34.22 | Nkoji | service asterisk stop? |
| 22:35.55 | seanbright | Nkoji: correct |
| 22:36.03 | seanbright | service asterisk stop; service zaptel stop |
| 22:36.39 | seanbright | of course that will kill any active calls |
| 22:36.47 | seanbright | but i assume you know that. |
| 22:36.54 | outtolunc | hehe |
| 22:36.56 | [TK]D-Fender | ~assume |
| 22:36.56 | jbot | About assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight"e=assume.txt&file=assume.wav |
| 22:37.12 | seanbright | ass-u-me |
| 22:38.02 | tapic | [TK]D-Fender : http://pastebin.com/m7ff69da5 |
| 22:38.37 | outtolunc | what no sally? |
| 22:42.03 | [TK]D-Fender | tapic: Don't use A() to play that file, do it in your MACRO |
| 22:42.24 | unpaidbill | sooo when is 1.6 coming out, i need t.38 :P |
| 22:42.49 | tapic | I already tried that the result is same |
| 22:43.08 | tapic | the file does not get played although CLI reports so |
| 22:43.33 | tapic | I tried Playback and Read to play file |
| 22:44.13 | tapic | after I restart asterisk the file gets played for 1 or 2 times |
| 22:44.59 | [TK]D-Fender | tapic: do Answer, Wait(1), then play them back in the macro. |
| 22:46.34 | tapic | ok |
| 22:46.40 | *** join/#asterisk n9urk (n=IceChat7@rrcs-70-62-74-122.midsouth.biz.rr.com) |
| 22:47.20 | n9urk | [root@vps8 asterisk-1.4.21.2]# asterisk |
| 22:47.20 | n9urk | bash: asterisk: command not found |
| 22:47.36 | n9urk | " when I run asterisk. Can anyone tell me what went wrong? |
| 22:47.37 | n9urk | thanks |
| 22:47.55 | [TK]D-Fender | n9urk: guess it isn't installed properly into your path |
| 22:48.01 | Nkoji | i get an error when starting asterisknow stating that no zaptel can be found so it loads ztdummy |
| 22:48.09 | Nkoji | why does it do that? |
| 22:48.19 | unpaidbill | do you have a zaptel card? |
| 22:48.23 | [TK]D-Fender | Nkoji: What should it find? |
| 22:48.38 | Nkoji | i have a tdm400p |
| 22:48.52 | n9urk | [TK]D-Fender: what can be done to cure this problem? |
| 22:48.53 | Nkoji | on sec im entering interactive start up so i can give you exact detail |
| 22:49.21 | [TK]D-Fender | Nkoji: pastebin "dmesg" |
| 22:49.23 | [TK]D-Fender | ~pb |
| 22:49.24 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines 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 |
| 22:49.25 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 22:49.33 | [TK]D-Fender | n9urk: Go recompile & install it |
| 22:49.41 | n9urk | I already did that 3 times |
| 22:50.07 | Nkoji | No functioning zap hardware found in /proc/zaptel, loading ztdummy |
| 22:50.16 | n9urk | (the last time I redownloaded the package) |
| 22:51.34 | tapic | <PROTECTED> |
| 22:51.39 | *** part/#asterisk keith4_ (n=kbe2@207-172-236-173.c3-0.eas-ubr9.atw-eas.pa.cable.rcn.com) |
| 22:52.30 | [TK]D-Fender | n9urk: download? What does that have to do with compile & install? |
| 22:52.33 | tapic | bridging happens immediately the callee picks up and he does not get the sound file played |
| 22:52.36 | [TK]D-Fender | tapic: PASTEBIN <- |
| 22:53.01 | *** join/#asterisk Bonix (n=Bonix@189.52.0.234) |
| 22:53.01 | [TK]D-Fender | Nkoji: I asked you for "dmesg" in a pastebin... please provide it. |
| 22:53.10 | n9urk | [TK]D-Fender -> to make sure I had the right source. but regardless I have recompilied it three times |
| 22:53.23 | [TK]D-Fender | n9urk: What did you do exactly? |
| 22:53.30 | [TK]D-Fender | n9urk: And on what distro? |
| 22:53.38 | n9urk | ./configure make make install |
| 22:53.44 | n9urk | CentOS |
| 22:54.11 | [TK]D-Fender | n9urk: for 1.4+ you should be doing "make menuconfig" |
| 22:55.29 | *** join/#asterisk l2trace99 (n=asd@75.112.133.235) |
| 22:55.30 | n9urk | [TK]D-Fender: cool, thanks, will give that a try. For future reference, how would have I known to have used menuconfig as the README only listed that as an option, implying that all would be installed with make install? |
| 22:55.50 | martha | anyone up for a weird quality problem? I'm running asterisk 1.4.20 on debian. all was fine until 2 days ago, when we started to get periodic massive quality drops in the middle of calls. Suddenly, you one side can't hear the other, just loud bursts of static. That happens for a few seconds, then recovers. This is happening even on calls that just go from channelbank (zhone) to channelbank, so there it doesn't go over the network |
| 22:56.10 | *** join/#asterisk cplx (n=cplx@59.167.199.141) |
| 22:56.24 | [TK]D-Fender | n9urk: its in all the docs and in the book. |
| 22:56.41 | Nkoji | is there a way i can get dmesg from the asterisknow GUI? |
| 22:57.13 | n9urk | [TK]D-Fender Where would I find it in the docs? I didn't see anything about it other then it listed as an option in the README? |
| 22:58.13 | *** join/#asterisk nix8n82 (n=nanderso@mo-65-41-196-62.sta.embarqhsd.net) |
| 22:58.26 | [TK]D-Fender | n9urk: pastebin "make install" |
| 22:58.49 | [TK]D-Fender | Nkoji: You DON'T. This is from Linux CLI. This is not a GUI support channel |
| 22:59.09 | Nkoji | yeah i know i just dont have any way of copying and pasting the text |
| 22:59.50 | [TK]D-Fender | Nkoji: connect via SSH & copy&paste like the rest of us |
| 23:00.12 | tapic | [TK]D-Fender : http://pastebin.com/m3969049 |
| 23:00.21 | n9urk | [TK]D-Fender: One second |
| 23:00.32 | nix8n82 | I keep having trouble with playback function in asterisk 1.4 and it goes to the h extension after that instead of completing the rest of the dialplan..how do I catch the error and make it retry the playback function? |
| 23:00.54 | *** join/#asterisk Linker3000L (n=chatzill@78.32.25.201) |
| 23:01.23 | [TK]D-Fender | tapic: Do you hear the file it says its playing? |
| 23:01.35 | tapic | no |
| 23:01.44 | [TK]D-Fender | nix8n82: PASTEBIN your failed attempt at verbose 10 |
| 23:02.02 | tapic | the file does not get played to callee |
| 23:02.05 | [TK]D-Fender | tapic: How long do you wait before hitting "5"? |
| 23:02.11 | n9urk | [TK]D-Fender: http://pastebin.ca/1087742 |
| 23:02.52 | tapic | about a second |
| 23:03.21 | *** join/#asterisk [tasty]freeze (n=yamahabr@204-181-48-126.skybest.com) |
| 23:03.44 | l2trace99 | has anyone here setup an asterisk cluster ? |
| 23:03.49 | [tasty]freeze | Is there a command to have asterisk drop calls, or all calls... for the need to connect an emergency call? |
| 23:04.06 | [TK]D-Fender | tapic: wait a little longer. Also try testing with another sound file |
| 23:04.29 | [TK]D-Fender | n9urk: verify your path |
| 23:05.00 | [TK]D-Fender | [tasty]freeze: "soft hangup [channel]" from * CLI |
| 23:05.08 | n9urk | [TK]D-Fender: How do I do that? |
| 23:05.28 | [tasty]freeze | [TK]D-Fender: Can I do that from the extensions.conf file? |
| 23:05.51 | *** join/#asterisk BBHoss (n=hoss@c-68-62-170-33.hsd1.al.comcast.net) |
| 23:05.54 | [TK]D-Fender | n9urk: "set|grep PATH" |
| 23:06.07 | tapic | the same file gets played to the caller. I tried with other gsm file and other file format(wav) too. Waiting longer is same... |
| 23:06.08 | [TK]D-Fender | [tasty]freeze: "core show application system" |
| 23:06.25 | *** join/#asterisk edibrac (n=edibrac3@75.149.50.41) |
| 23:06.43 | n9urk | [TK]D-Fender: PATH=/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/bin:/bin:/usr/bin:/home/lburton/bin |
| 23:06.51 | [TK]D-Fender | tapic: -- Executing [s@macro-agentmacro:3] Read("SIP/beri-081e8870", "ACCEPT|welcome|1") in new stack <- says its playing to the callee |
| 23:07.12 | tapic | but it does not get heard |
| 23:07.26 | n9urk | [TK]D-Fender should it only be "export PATH=$PATH:/usr/sbin/:/usr/local/bin"? |
| 23:07.50 | [TK]D-Fender | n9urk: And as you can clearly see you are MISSING "/usr/sbin" which is where it installed the binaries |
| 23:08.37 | *** part/#asterisk Deeewayne (n=Deeewayn@216.207.245.1) |
| 23:08.37 | tapic | sometime is gets heard, after restarting asterisk but not always. |
| 23:08.37 | [TK]D-Fender | tapic: Sometimes yes, sometimes no? |
| 23:08.37 | tapic | 99% no |
| 23:08.39 | [TK]D-Fender | tapic: Ok, may be a straight-up bug. |
| 23:08.58 | tapic | I think so. |
| 23:09.00 | [TK]D-Fender | tapic: I recommend posting your sample and explanation on Mantis (the bug tracker) |
| 23:09.17 | edibrac | in my dialplan, how do i jump to a menu, if I'm currently on the phone with someone? i.e. I'm talking to someone, and I'd like to press 4 to run a script -- what's the mechanism that says "if ext 100 is on a call" |
| 23:09.26 | n9urk | [TK]D-Fender: thanks, I don't know a lot about path. your help is much appreciated |
| 23:09.40 | n9urk | started now |
| 23:09.50 | [TK]D-Fender | n9urk: basic Linux stuff.... |
| 23:10.39 | n9urk | [TK]D-Fender: I know some linux things, but a lot is still beyond my grasp |
| 23:10.47 | *** join/#asterisk dmz (n=dmz@64.203.203.232.dyn-cm-pool-64.hargray.net) |
| 23:12.00 | martha | anyone? I thought that this might be a load problem, because there was another program that was causing a high load on the server, but I resolved the load problem and this is still happening :-( |
| 23:12.35 | [TK]D-Fender | n9urk: Your path is not normal for CentOS. |
| 23:12.52 | tapic | sorry but what is the address of mantis |
| 23:13.08 | [TK]D-Fender | tapic: www.asterisk.org . its linked there |
| 23:13.54 | [TK]D-Fender | edibrac: if you're on a call what are you jumping out to do? |
| 23:14.19 | [TK]D-Fender | martha: What card? |
| 23:15.33 | martha | [TK]D-Fender: we have 4 wct4xxp in this server |
| 23:16.27 | [TK]D-Fender | martha: You know every doc including Digiums says its recommended to use no more than 2 right? |
| 23:16.33 | martha | <PROTECTED> |
| 23:16.35 | [TK]D-Fender | martha: Does rebooting clear it off? |
| 23:16.42 | *** join/#asterisk AlexTO (n=alex@75.149.245.109) |
| 23:16.43 | edibrac | [TK]D-Fender: say i'm on a call, to kick off a script like to play a sound |
| 23:16.47 | martha | [TK]D-Fender: no, where do they say that? |
| 23:16.54 | [TK]D-Fender | edibrac: For who? |
| 23:17.08 | [TK]D-Fender | martha: asterisk.org. Its all over the place. |
| 23:17.10 | *** join/#asterisk mattwj2002 (n=Matt@c-76-17-132-205.hsd1.mn.comcast.net) |
| 23:17.11 | martha | [TK]D-Fender: we've been running with 4 cards for about a year |
| 23:17.45 | mattwj2002 | someone called my 866 number :S |
| 23:17.46 | edibrac | [TK]D-Fender: like to play back voicemail to a customer |
| 23:17.49 | nix8n82 | http://pastebin.ca/1087749 log for my playback failure |
| 23:18.03 | mattwj2002 | no one should know about it besides me |
| 23:18.32 | nix8n82 | there are random dialers out there |
| 23:18.54 | [TK]D-Fender | edibrac: If you want to go through the dialplan for other stuff your only choice is to start a 3-way call via your phone |
| 23:19.03 | nix8n82 | and it could be just a wrong number |
| 23:20.07 | mattwj2002 | okay |
| 23:20.14 | [TK]D-Fender | nix8n82: Looks like a normal hangup there. |
| 23:20.30 | mattwj2002 | it just sucks because I have to pay for these calls :( |
| 23:20.33 | [TK]D-Fender | nix8n82: Please pastebin your DIALPLAN and on the next call enable SIP DEBUG. |
| 23:20.57 | nix8n82 | where does asterisk go after Playback app has exited non zero? does it go directly to the h extension or can I intercept it and make it try to play the file again and or check the line status? |
| 23:21.00 | edibrac | [TK]D-Fender: ok the real reason is i want to play weird sounds while I talk .. say like.. that guy Jim Cramer on Mad Money |
| 23:21.42 | martha | [TK]D-Fender: I've have just rebooted, but I had to wait until all everyone else left. I haven't heard the problem since then, but there is no load on the system |
| 23:21.48 | [TK]D-Fender | edibrac: individual hard-coded choices can be made using "dynamic features" via features.conf. Look these up on the WIKI |
| 23:21.48 | edibrac | [TK]D-Fender: i think someone asked about this a few days ago here, but I didn't stick around to see the discussion (if any) |
| 23:22.03 | edibrac | ahh thanks |
| 23:22.16 | [TK]D-Fender | martha: I've heard of things like this just developing over time and reboot clearing it |
| 23:23.11 | martha | [TK]D-Fender: actually, this machine was just rebooted on sunday. so it wasn't up very long |
| 23:27.03 | *** join/#asterisk LinuxMafia (n=awatt@CPE000e081cbf7a-CM00159a642d7e.cpe.net.cable.rogers.com) |
| 23:27.07 | LinuxMafia | hi all |
| 23:27.26 | mattwj2002 | would a random dialer have an caller id of ""anonymous"" <anonymous>" |
| 23:27.29 | LinuxMafia | i have a question , dont know if it is related or no |
| 23:27.55 | *** join/#asterisk nix8n82 (n=nanderso@mo-65-41-196-62.sta.embarqhsd.net) |
| 23:28.07 | martha | [TK]D-Fender: if the reboot didn't fix it, any ideas on what else to check? I don't get any errors in the asterisk console/logs |
| 23:28.49 | [TK]D-Fender | martha: no idea. |
| 23:29.10 | martha | :-( |
| 23:29.18 | LinuxMafia | i have an linksys adoptor , and i have one cordless phone connected to it working properly , now i want to add another phone , it is nortel network phone what should i do? |
| 23:30.48 | nix8n82 | With priority jumping disabled in 1.4 is there a way to catch error from application such as playback,check to see if the line is still up and try the application again? |
| 23:31.55 | *** join/#asterisk macros73_ (n=cs@c-67-163-224-69.hsd1.pa.comcast.net) |
| 23:33.04 | martha | [TK]D-Fender: I can't find anything about only using two cards on asterisk.org |
| 23:34.25 | edibrac | hmm i never noticed this ..since I don't call that much -- a sales guy told me that when he sets call forwarding on his Cisco 7940 to his cell phone -- if he doesn't pickup, then our asterisk voicemail takes over, instead of his cell phone. |
| 23:35.00 | edibrac | he'd prefer it to be answered by his cell phone.. but then again, doesn't this depend on his cell phone's voicemail setting, compared to asterisk? |
| 23:36.16 | [TK]D-Fender | edibrac: * dial timeout VS Cell VM timeout |
| 23:36.37 | [TK]D-Fender | martha: I'll see if I can find it. But this is common knowledge going WAY back. |
| 23:37.22 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 23:37.29 | tapic | [TK]D-Fender : I have made a bug report, thank you for your helps |
| 23:37.36 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 23:37.44 | [TK]D-Fender | LinuxMafia: Nortel Digital (Meridian / Norstar) are not normally usable. Nortel SIP /UNISTIM are. |
| 23:38.48 | LinuxMafia | [TK]D-Fender, oh so that is why the lite does not turn on? |
| 23:38.57 | LinuxMafia | light* |
| 23:39.36 | LinuxMafia | M3904 is the one i have |
| 23:39.56 | LinuxMafia | but i have adaptor , so why it does not work? |
| 23:40.30 | [TK]D-Fender | LinuxMafia: What "adapter"? |
| 23:40.33 | LinuxMafia | [TK]D-Fender, i just want to make a call let's say without using * |
| 23:40.50 | LinuxMafia | [TK]D-Fender, linksys adaptor |
| 23:41.30 | LinuxMafia | [TK]D-Fender, like it turns the normal phone into SIP phone |
| 23:41.40 | [TK]D-Fender | LinuxMafia: that is a DIGITAL handset. it CANNOT be used period. |
| 23:41.58 | [TK]D-Fender | LinuxMafia: http://products.nortel.com/go/product_content.jsp?segId=0&parId=0&prod_id=52685 <--- DEAD END |
| 23:42.24 | *** join/#asterisk Nkoji (n=nicholas@168-215-215-195.static.twtelecom.net) |
| 23:42.52 | LinuxMafia | [TK]D-Fender, so it only works with normal phone lines? |
| 23:43.08 | Nkoji | [TK] D-Fender: pastebin of dmesg http://pastebin.com/m17369d61 |
| 23:43.31 | LinuxMafia | [TK]D-Fender, thanks alot |
| 23:43.34 | [TK]D-Fender | LinuxMafia: No it wokrs only with Nortel phone systems |
| 23:44.10 | LinuxMafia | [TK]D-Fender, oh so no one can use it at home? |
| 23:44.37 | *** part/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 23:45.05 | [TK]D-Fender | Nkoji: Freshmaker failed register test <- your card looks screwed up. |
| 23:45.17 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 23:45.22 | [TK]D-Fender | linuxmaWhat part of "useless" is not clear to you? |
| 23:46.21 | [TK]D-Fender | Nkoji: Call Digium support |
| 23:46.23 | Nkoji | screwed up in what manner like fried or something is set up wrong on it |
| 23:46.28 | Nkoji | ok thank you |
| 23:46.37 | [TK]D-Fender | Nkoji: KFC : Extra Tasty Crispy |
| 23:55.45 | *** join/#asterisk nebajoth (n=weechat@76-10-174-10.dsl.teksavvy.com) |
| 23:55.52 | nebajoth | whoa |
| 23:55.56 | nebajoth | this is a huge channel |
| 23:55.59 | nebajoth | sweet |
| 23:56.08 | nebajoth | I'm new to * |
| 23:56.36 | nebajoth | was thinking about putting in an embedded IP04 system for my company |
| 23:56.54 | nebajoth | don't mind hacking around on it, but was wondering, before I started |
| 23:57.10 | nebajoth | whether its possible to 'failover' from one busy incoming POTS line to another? |
| 23:57.18 | nebajoth | I'll have 3-4 |
| 23:57.34 | nebajoth | thanks for your consideration! |
| 23:59.55 | [TK]D-Fender | nebajoth: On outbound yes, assuming * is the only device using those lines. For INBOUND that it for you to ask of your telco |