| 00:00.14 | rg2112 | hi all. anyone have experience with Adran 850s? I'm trying to connect asterisk+digium to an Adtram 850 with no love. |
| 00:01.29 | MACscr | JT: while i agree it could be cheaper, it would be more of a hassle than its worth for the savings at this point |
| 00:01.50 | JT | MACscr: depends where it's located |
| 00:02.40 | MACscr | Well, i personaly wouldnt ever want to have to touch it |
| 00:02.40 | MACscr | And i dont want the up front costs of purchasing a server |
| 00:04.53 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 00:12.24 | JT | MACscr: i prefer co-lo myself, if i'm anywhere near the datacentre |
| 00:12.56 | MACscr | I dont like to keep parts on hand and im no where near a quality dc |
| 00:14.39 | JT | heh ok |
| 00:14.43 | JT | whoop whoop? ;) |
| 00:15.08 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 00:17.48 | *** join/#asterisk dijungal (n=kdaniel@64.86.52.254) |
| 00:18.37 | dijungal | hello... is it possible to monitor a call on asterisk? |
| 00:19.01 | dijungal | lets say and agent recieves a call.. can a third party dial in and monitor that call? |
| 00:19.24 | *** join/#asterisk swagger (n=Steve@ip68-227-26-15.lv.lv.cox.net) |
| 00:20.48 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 00:20.54 | davidcsi | anyone knows what extension is used for DIALSTATUS=CANCEL?? |
| 00:22.18 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) [NETSPLIT VICTIM] |
| 00:22.18 | *** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) [NETSPLIT VICTIM] |
| 00:22.18 | *** join/#asterisk nitram (i=foo@superblob.com) [NETSPLIT VICTIM] |
| 00:22.24 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 00:25.29 | davidcsi | dijungal: yes, you can, but i don't know how.. sorry, i think that with barge or something like that... |
| 00:25.50 | dijungal | ;) |
| 00:25.52 | dijungal | :| |
| 00:35.27 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-47-145.red.bezeqint.net) |
| 00:39.15 | NightMonkey | Howdy. Anyone have a reccomendation on some good F/OSS webGUIs for Asterisk? |
| 00:39.46 | kiscokid | asteriskgui? |
| 00:43.18 | JT | the main recommendation is to not use a gui :) |
| 00:49.51 | rob0 | well, konsole isn't bad :) |
| 00:50.11 | dijungal | lol |
| 00:50.13 | *** join/#asterisk bmg505 (n=leon@196.209.179.191) |
| 00:50.22 | dijungal | about the monitoring thing.. anyone got any ideas? |
| 00:50.30 | dijungal | i want to monitor active channels |
| 00:50.36 | NightMonkey | kiscokid: I'm gonna check out the svn now. Thanks. |
| 00:50.52 | JT | all the info is on the wiki, dijungal |
| 00:51.01 | dijungal | i've been searching |
| 00:51.34 | dijungal | which wiki...i've been in voip-info.org.. |
| 00:51.38 | JT | zapbarge |
| 00:51.41 | JT | yes that is the wiki |
| 00:52.20 | Strom_M | but is it the wikiest wiki? |
| 00:52.25 | dijungal | but it's possible right? |
| 00:52.34 | Strom_M | is this hotdog completely FOSS snack food? |
| 00:52.41 | Strom_M | what about the milk |
| 00:52.41 | rob0 | The wicked wiki |
| 00:52.53 | kiscokid | anyone know how to get the Message Waiting light to work in a Grandstream GXP-2000? |
| 00:53.11 | Strom_M | kiscokid: did you put the mailbox setting in the sip.conf entry? |
| 00:53.19 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-219-116.hsd1.al.comcast.net) |
| 00:53.25 | ectospasm | anyone here use trixbox? |
| 00:53.32 | Strom_M | go to #trixbox |
| 00:53.34 | sweeper | all signs point to no |
| 00:53.36 | JT | hopefully not |
| 00:53.40 | ectospasm | heheheh, OK |
| 00:54.03 | kiscokid | Strom_M: I have mailbox=8003@default in the sip.conf |
| 00:54.24 | Strom_M | alright, and does that mailbox exist in [default] in voicemail.conf? |
| 00:54.32 | kiscokid | yeah |
| 00:54.48 | kiscokid | I can send and retrieve voicemail |
| 00:54.51 | Strom_M | and is there a message in that mailbox waiting to be listened to? |
| 00:55.04 | kiscokid | not this second |
| 00:55.17 | sweeper | poor guy having to deal with a grandstream D: |
| 00:55.48 | kiscokid | doh, its working |
| 00:55.56 | ectospasm | you'd be better off with an ATA over a Grandstream |
| 00:56.06 | kiscokid | I know |
| 00:56.24 | Strom_M | you'd be better off with a flaming pile of hyaena offal than a Grandstream |
| 00:56.28 | ectospasm | Grandstreams are great pricewise, but suck when you use them... |
| 00:56.55 | kiscokid | I know, I know, I just got one of them. |
| 00:57.04 | kiscokid | Gonna look at Polycoms next |
| 00:57.26 | _VoiceMeUp_COM | BTW for all new msn virus that says to click on http://www.SOYOUDONTCLICKlikemyass.net/pic901.com |
| 00:57.31 | _VoiceMeUp_COM | DO NOT click |
| 00:57.35 | _VoiceMeUp_COM | its a virus |
| 00:57.36 | Strom_M | duh |
| 00:57.50 | _VoiceMeUp_COM | shit.. it didnt mess the url |
| 00:58.06 | _VoiceMeUp_COM | do not click.. this message.. tis a msn virus being around since morning and afftecting lots |
| 00:58.26 | Strom_M | _VoiceMeUp_COM: thank you for the news of the stupid |
| 00:58.34 | _VoiceMeUp_COM | already emailed the DE host.. no asnwer.. anyone speak dutch or germ ? |
| 00:58.48 | Strom_M | in other breaking news: people are stupid idiots; show no signs of improving |
| 00:58.54 | *** part/#asterisk rg2112 (n=rob@64.163.131.18) |
| 00:58.58 | kiscokid | wow, the message light on my Cisco 7960 works too |
| 00:58.59 | _VoiceMeUp_COM | well believe it or not lots of people got it |
| 00:59.04 | _VoiceMeUp_COM | yeah i know |
| 00:59.14 | _VoiceMeUp_COM | trillian is safe from it for some obscure reason |
| 00:59.16 | kiscokid | must have been the mailbox= in sip.conf |
| 00:59.16 | Strom_M | local interviewees respond "durhhhhhhhhhhhhhhhhhhhhhblblblbblblb" |
| 00:59.32 | _VoiceMeUp_COM | how about voicemail.conf |
| 00:59.39 | _VoiceMeUp_COM | waht the context of yhe mailbox in there ? |
| 00:59.41 | Strom_M | _VoiceMeUp_COM: he fixed it already |
| 00:59.41 | _VoiceMeUp_COM | default ? |
| 00:59.45 | _VoiceMeUp_COM | ah |
| 00:59.48 | _VoiceMeUp_COM | sorry |
| 00:59.54 | kiscokid | yeah, thanks |
| 01:00.04 | Strom_M | breaking news: MSN virus distracts ITSP operator |
| 01:00.10 | _VoiceMeUp_COM | ;p; |
| 01:00.32 | JT | Keyboard offset issue causes miscommunication of lol |
| 01:00.43 | *** join/#asterisk vn (n=nostalge@bas5-quebec14-1128557048.dsl.bell.ca) |
| 01:00.44 | _VoiceMeUp_COM | ahahah |
| 01:00.44 | MACscr | Anyone using the GIPS codec and having success with it? |
| 01:01.32 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 01:01.46 | rob0 | Um, what about this newfangled users.conf thing? That looks like a nice way to set up extensions and voicemail. |
| 01:01.56 | Strom_M | i've not used it |
| 01:02.08 | MACscr | Im looking for a good global codec |
| 01:02.19 | Strom_M | how about...G711 |
| 01:02.23 | Strom_M | everyone speaks that |
| 01:02.40 | MACscr | Well, i mean something that will work fine between a server in the US and Germany |
| 01:02.47 | Strom_M | G711 |
| 01:02.52 | Strom_M | that'll work fine |
| 01:03.02 | MACscr | With ping times around 150? |
| 01:03.07 | JT | MACscr: G.711 is the international standard |
| 01:03.12 | JT | latency has no relationship |
| 01:03.17 | JT | 150ms is nothing |
| 01:03.33 | MACscr | Oh, i thought you always wanted it less than 100 |
| 01:03.37 | Strom_M | no |
| 01:03.57 | Strom_M | ideally less than 200, and really no more than 400 |
| 01:04.14 | vn | can we mix the sound with SIP phones? likeéééapplying a very high amplification, modifying the voice tones...etc? |
| 01:04.28 | Strom_M | vn: why? |
| 01:04.41 | MACscr | Thanks for the info, that really helped |
| 01:04.44 | davidcsi | is sip channel's uniqueid based on unixtime? |
| 01:04.56 | *** join/#asterisk magic_hat (n=geoffdou@h-74-2-87-16.chcgilgm.covad.net) |
| 01:04.59 | JT | MACscr: latency has no relationship to the codec anyway usually |
| 01:05.04 | vn | because I'm hard of hearing and need specific settings to understand on the phone |
| 01:05.10 | JT | only bandwidth usage is related to codec |
| 01:05.15 | *** join/#asterisk guillote_GNU (n=guillote@host213.201-252-196.telecom.net.ar) |
| 01:05.21 | JT | and some codecs handle packet loss and jitter better |
| 01:05.23 | vn | so I dunno if I keep my analog phone...or use a SIP phone |
| 01:05.24 | magic_hat | anyone know why I can't get into /etc/asterisk on my ubuntu box? it's giving me 'permission denied' |
| 01:05.56 | JT | vn: good sip phones have volume buttons |
| 01:05.59 | MACscr | So if i have large pipes, then i shouldnt have to worry about the codec, but with packet loss i should, right? |
| 01:06.03 | Strom_M | vn: do you need the phone to be hearing aid compliant? |
| 01:06.11 | JT | MACscr: correct |
| 01:06.18 | Strom_M | magic_hat: are you root? |
| 01:06.25 | vn | Strom_M: yep |
| 01:06.37 | magic_hat | no... lol I cannot find the damn root pw for this machine. |
| 01:06.38 | Strom_M | vn: most modern phones should be hearing aid compliant anyway |
| 01:06.50 | Strom_M | magic_hat: well then i think you're screwed |
| 01:07.15 | vn | on SIP phones, what kind of plug is used to plug the receiver? rj11? |
| 01:07.21 | davidcsi | is sip channel's uniqueid based on unixtime? |
| 01:07.24 | magic_hat | anyone know what the default root pw is for ubuntu? |
| 01:07.26 | vn | if so I could keep my actuall amplification equipment |
| 01:07.35 | JT | it's possible to reset the root password if you have physical access to the box |
| 01:07.48 | JT | vn: rj-11 or similar on polycoms |
| 01:07.53 | JT | for headsets |
| 01:07.55 | magic_hat | JT: yeah, I do |
| 01:08.01 | Strom_M | the headset connector is RJ-9 |
| 01:08.02 | vn | ok |
| 01:08.04 | magic_hat | although it's a new box and I never changed it. |
| 01:08.07 | JT | Strom_M: right :) |
| 01:08.09 | Strom_M | 4P4C |
| 01:08.21 | *** join/#asterisk AllanLima (n=mediain@unaffiliated/allanlima) |
| 01:08.31 | Strom_M | magic_hat: have you tried sudo? |
| 01:08.44 | magic_hat | as the pw? |
| 01:08.46 | Strom_M | no |
| 01:08.51 | Strom_M | the program sudo |
| 01:09.01 | vn | I still dunno if I buy an ATA + FXS (or is it fxo?) card or SIP with FXO card.. |
| 01:09.14 | magic_hat | i've tried sudo cd /etc/asterisk |
| 01:09.29 | Strom_M | magic_hat: then when it asks you for password |
| 01:09.33 | Strom_M | enter your user password |
| 01:09.36 | magic_hat | it doesn't get me there... 'sudo: cd: command not found |
| 01:09.43 | vn | p.O |
| 01:09.50 | JT | no sodo |
| 01:09.56 | JT | what sort of machine is this |
| 01:09.59 | magic_hat | ubuntu |
| 01:10.13 | magic_hat | sudo works for other stuff. just not cd. shrug |
| 01:10.19 | JT | ubuntu has no root password by default |
| 01:10.22 | JT | sudo su |
| 01:10.25 | vn | use sudo -i |
| 01:10.31 | vn | that too |
| 01:11.35 | vn | anyone can recommend me in my choice? |
| 01:12.02 | JT | vn: sip phones are the most flexible and have more fetures |
| 01:12.03 | Strom_M | vn: either a sip phone or an ATA will do the trick |
| 01:12.04 | JT | features |
| 01:12.09 | JT | and more digital |
| 01:12.09 | Strom_M | but you'll probably like the sip phone |
| 01:12.18 | vn | same quality? |
| 01:12.31 | JT | a good sip phone will sound better |
| 01:12.58 | vn | ok |
| 01:13.16 | vn | wonder if there's somewhaere I can try before buying.. |
| 01:13.44 | JT | buy a polycom, and if you plan to use the headset port, either use PoE or get a decent aftermarket power brick |
| 01:14.07 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 01:14.13 | Defraz | exten=> _12085294887,1,Dial(SIP/12085294887@fa-pbx.fuzecore.com) I have an IAX2 trunk between my two systems, I would like to use IAX instead of sip when I forward this number but I am having trouble with the syntex. |
| 01:14.45 | Defraz | I tried just chaning out IAX2 for the SIP but it didn't seem to work. |
| 01:15.37 | Strom_M | uh, no |
| 01:15.37 | Defraz | This line is on my 1st server and redirects the DID to the a second server. |
| 01:15.42 | Strom_M | read the documentation |
| 01:15.43 | vn | uh I don't thikn a headset is a goot idea for me...with hearing aids, wireless stuff tends to be flaky |
| 01:15.58 | Strom_M | vn: who says it has to be wireless? |
| 01:16.06 | vn | uh...yeah |
| 01:16.17 | vn | anyway I prefer the usual way heh |
| 01:16.28 | Strom_M | Defraz: iax2 is: Dial(IAX2/entry_name_in_iax.conf/13115552368) |
| 01:16.44 | *** join/#asterisk xo8ox (n=pride_32@wsip-66-210-250-2.ph.ph.cox.net) |
| 01:17.08 | xo8ox | guys in asterisk 1.0 where do you setup inbound routes ? |
| 01:17.14 | Strom_M | 1.0?!?! |
| 01:17.18 | Strom_M | holy cocks man |
| 01:17.24 | xo8ox | loool |
| 01:17.31 | vn | nice my cock is now holy |
| 01:17.31 | Strom_M | at minimum, use 1.2 or something |
| 01:17.32 | xo8ox | I know our old it guy set it up on the old version |
| 01:17.38 | xo8ox | now I have to fix up the mess |
| 01:17.43 | Strom_M | iiiiiiiiiiiiiiiiiiiiiiiiiiit's upgrade time! |
| 01:17.47 | JT | vn: umm, the headsets are usually wired. |
| 01:17.49 | xo8ox | hehe |
| 01:18.11 | xo8ox | before upgrade i need to find out where our DIDs and Local nums are routed to |
| 01:18.12 | Strom_M | xo8ox: anyway, you say "inbound routes" like you're using a gui or something |
| 01:18.17 | xo8ox | so where do I look for these settings |
| 01:18.21 | vn | and uh...if I use an ATA analog to SIP and then a FXS (or o?) that uses IAX2 with my provider...will I get NAT/routing problems? |
| 01:18.27 | vn | I know SIP tends to suck for that |
| 01:18.34 | xo8ox | I'm saying it in english hehe |
| 01:18.56 | Strom_M | xo8ox: no - is your system configured with a gui? |
| 01:19.01 | xo8ox | no |
| 01:19.05 | Strom_M | ok |
| 01:19.07 | xo8ox | like I said its asterisk 1.0* |
| 01:19.07 | JT | vn: iax usually either works or doesn't work |
| 01:19.16 | Strom_M | sip.conf, extensions.conf, iax.conf, zapata.conf |
| 01:19.25 | vn | JT: why's that? |
| 01:19.27 | xo8ox | in all of them ? |
| 01:19.34 | Strom_M | xo8ox: depends on what you're using |
| 01:19.37 | JT | vn: single udp port for signalling and media |
| 01:20.02 | xo8ox | t1 |
| 01:20.12 | vn | then there shouldn't be any problem if it's for only 1 phone |
| 01:20.26 | xo8ox | its not 1 phone |
| 01:20.29 | JT | vn: except that you cannot buy a good iax softphone |
| 01:20.32 | xo8ox | its 1 T1 line with bunch of DIDs |
| 01:20.37 | *** join/#asterisk dec_ (n=tom@unaffiliated/dec) |
| 01:20.39 | vn | I'm talking about my stuff |
| 01:20.43 | xo8ox | aha sorrry |
| 01:20.44 | xo8ox | hehe |
| 01:20.50 | vn | JT: that's why I'd use an ATA |
| 01:21.00 | JT | just use a sip phone |
| 01:21.03 | Strom_M | xo8ox: then look in extensions.conf and zapata.conf |
| 01:21.04 | JT | it's the best solution |
| 01:21.13 | JT | that's not a good reasong to use an ata |
| 01:21.16 | JT | an ata uses sip |
| 01:21.29 | vn | analog phone --- ATA to SIP --- FXS---IAX2 to provider |
| 01:21.36 | Defraz | Thank you Strom |
| 01:21.48 | vn | well you just told me there were no good IAX2 phones |
| 01:21.48 | JT | vn: that FXS is misplaced |
| 01:21.50 | Strom_M | vn: you only need an FXS port to connect to an analog phone, and you only need an FXO port to connect to an analog phone line |
| 01:22.07 | JT | vn: but there are good sip phones |
| 01:22.14 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 01:22.17 | *** part/#asterisk kiscokid (n=ron@208.106.33.66) |
| 01:22.32 | vn | uh...ok and if I use a SIP phone, I plug it directly into the router? |
| 01:22.42 | JT | if you want |
| 01:22.57 | JT | so you aren't using asterisk? |
| 01:23.06 | vn | well I'm just planning stuff |
| 01:23.41 | *** part/#asterisk magic_hat (n=geoffdou@h-74-2-87-16.chcgilgm.covad.net) |
| 01:23.57 | vn | but if I use a SIP phone, I'll get NAT problems no? |
| 01:24.09 | JT | not if setup right |
| 01:24.34 | ectospasm | only if the nat is between asterisk and the sip phone. in that case you'll have to do some extra work |
| 01:24.37 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 01:24.51 | vn | the nat would be for the whole network |
| 01:25.19 | ectospasm | so if asterisk and the sip phone are behind the same NAT, it should work OK |
| 01:25.21 | vn | I'm actually on a linksys router, I plan to use my cisco 806 soon |
| 01:25.38 | vn | ok and if I don't use asterisk? |
| 01:25.58 | ectospasm | what is the sip phone connecting to? |
| 01:26.26 | Defraz | hmmm Strom_M that isn't quite what I expected it to do. |
| 01:26.37 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 01:26.41 | vn | I was thinking about plugging it directly into the router |
| 01:26.49 | Strom_M | Defraz: yeah, that tells me a lot about what's going wrong |
| 01:27.03 | Defraz | I was getting to that. |
| 01:27.05 | ectospasm | vn: you mean the hub/switch? That's expected |
| 01:27.16 | Defraz | It seems to just call back into the system and creates a loop. |
| 01:27.20 | vn | hub/switch/router |
| 01:27.38 | Defraz | it doesn't seem to forward the call onto the other server. |
| 01:27.41 | JT | vn: as opposed to plugging it into what? |
| 01:27.53 | Strom_M | Defraz: well then you have a configuration screwup in iax.conf |
| 01:27.55 | Defraz | Where the sip line did. |
| 01:27.57 | vn | JT: into the asterisk and then into the router? |
| 01:28.06 | JT | vn: no-one does that |
| 01:28.09 | JT | pointless |
| 01:28.11 | flenders | vn: if you use a sip phone that conencts to your provider, you'll be fine even behind a nat |
| 01:28.27 | ectospasm | vn: basically the sip phone just needs to plug into the same LAN the Asterisk server is on |
| 01:28.32 | vn | oh the NAT problems are with asterisk? |
| 01:28.57 | vn | SIP NAT problems* |
| 01:29.02 | ectospasm | no, with SIP in general |
| 01:29.09 | ectospasm | the SIP protocol doesn't handle NAT well |
| 01:29.30 | shmaltz | ectospasm, right h323 does |
| 01:29.30 | Defraz | Yea I am looking into it. |
| 01:29.35 | javar | can someone help me with a sangoma card? |
| 01:29.37 | vn | then shouldn't I use something IAX2? |
| 01:29.45 | shmaltz | javar, call tech support |
| 01:29.49 | Strom_M | vn: you should read the book |
| 01:29.51 | Strom_M | ~thebook |
| 01:29.52 | jbot | i heard thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 01:29.55 | javar | thanks |
| 01:30.05 | vn | what part? :/ |
| 01:30.09 | Strom_M | the whole thing |
| 01:30.22 | Strom_M | it will answer many of your questions |
| 01:30.55 | vn | k |
| 01:31.47 | ectospasm | vn: http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions |
| 01:32.24 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 01:32.34 | *** join/#asterisk docelmo (n=vircuser@c-76-99-157-112.hsd1.de.comcast.net) |
| 01:33.35 | *** join/#asterisk docelmo (n=vircuser@c-76-99-157-112.hsd1.de.comcast.net) |
| 01:33.36 | obnauticus | How do i enable colors in the asterisk CLI? |
| 01:34.32 | Corydon76-home | obnauticus: start it on a terminal that supports colors |
| 01:34.37 | shmaltz | obnauticus, why you cant read? |
| 01:34.37 | rob0 | First ... yes |
| 01:36.15 | obnauticus | because they are cool shmaltz |
| 01:36.26 | obnauticus | and Corydon76-home it's in bash and im running PuTTY |
| 01:36.47 | Corydon76-home | obnauticus: yes, but you didn't START asterisk that way |
| 01:37.00 | obnauticus | umm... |
| 01:37.03 | obnauticus | What do i do then? |
| 01:37.09 | Corydon76-home | You're remotely connecting to it, which is after the fact |
| 01:37.30 | Corydon76-home | first, what's the output of: echo $TERM |
| 01:37.45 | obnauticus | xterm |
| 01:38.13 | Corydon76-home | Type: /etc/init.d/asterisk stop |
| 01:38.21 | Corydon76-home | Type: /etc/init.d/asterisk start |
| 01:38.33 | Corydon76-home | Voila! Colors! |
| 01:38.36 | obnauticus | no. |
| 01:38.43 | obnauticus | WARNING's aren't red silly |
| 01:38.50 | obnauticus | well |
| 01:38.52 | Corydon76-home | If not, try: export TERM=xterm-color |
| 01:38.53 | obnauticus | they are supposed to be * |
| 01:39.17 | Strom_M | you have to run /usr/sbin/pretty_pretty_asterisk |
| 01:39.23 | obnauticus | lol |
| 01:39.51 | obnauticus | k |
| 01:39.52 | obnauticus | i got colors |
| 01:39.56 | obnauticus | how do i do it on startup? |
| 01:40.09 | obnauticus | start it from a shell that supports colors |
| 01:40.20 | Corydon76-home | Not a shell. A terminal |
| 01:40.24 | obnauticus | a terminal |
| 01:40.25 | obnauticus | i meant |
| 01:40.49 | *** part/#asterisk javar (n=javar@69.79.134.24) |
| 01:40.54 | Corydon76-home | If you want to force it, add 'export TERM=xterm-color' to /etc/init.d/asterisk near the top |
| 01:41.21 | Corydon76-home | or "export TERM=linux" which is usually better if you're running Linux |
| 01:42.48 | Corydon76-home | If FreeBSD, export TERM=scons25 |
| 01:43.30 | *** join/#asterisk magic_hat (n=geoffdou@h-74-2-87-16.chcgilgm.covad.net) |
| 01:43.39 | magic_hat | anyone know what's up with this: loader.c:325 __load_resource: /usr/lib/asterisk/modules/chan_modem.so: cannot open shared object file: No such file or directory |
| 01:45.55 | shmaltz | magic_hat, you need that chan_modem? |
| 01:46.07 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 01:46.07 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 01:48.48 | *** join/#asterisk sof76 (n=chatzill@Z8dc3.z.pppool.de) |
| 01:50.13 | sof76 | Hi everybody, can someone tell me if it is possible to make asterisk listen on 2 ports for sip registration, for example 5060 and 80? Thanks |
| 01:50.16 | *** join/#asterisk metabox (n=metabox@modemcable192.65-56-74.mc.videotron.ca) |
| 01:50.18 | metabox | hi |
| 01:50.55 | metabox | i have pap2-na voip adapter and i want to know if i can upgrade with the pap2 v1 firmware |
| 01:51.02 | JT | sof76: i don't think so, no |
| 01:51.04 | shmaltz | sof76, do that on your router firewall |
| 01:52.21 | sof76 | I have asterisk on a virtual private server that does'nt have iptables and some of the extensions are located in places where port 5060 is closed |
| 01:52.36 | blitzrage | sof76: nope |
| 01:52.44 | sof76 | ok thanks |
| 01:53.09 | JT | sof76: yeah, don't run asterisk on a VPS, unless it's a very good one |
| 01:53.14 | Gtwy | does anyone have a recomendation for a good phone provider? i am running * but i would like a flat rate plan |
| 01:53.18 | Gtwy | local would be best |
| 01:53.35 | docelmo | Gtwy there are like 300 of them.. |
| 01:53.39 | Corydon76-home | Nufone.net works for me |
| 01:53.43 | sof76 | I'm testing it for the moment, with one communication it's ok |
| 01:54.12 | shmaltz | Gtwy, define local |
| 01:54.16 | Gtwy | docelmo: why i am asking, too many to choose from |
| 01:54.28 | Gtwy | shmaltz: 1-412 or 1-724 |
| 01:55.05 | *** part/#asterisk dijungal (n=kdaniel@64.86.52.254) |
| 01:55.09 | shmaltz | Gtwy, try myphonecompany.com |
| 01:55.26 | docelmo | Gtwy where ya from? Thats western PA |
| 01:55.31 | Defraz | okay I have two servers A and B, if I am trying to register a user on B do I set it as a friend or user? |
| 01:55.32 | Gtwy | thats me |
| 01:55.37 | Gtwy | shmaltz: thanks |
| 01:55.52 | sof76 | what could be the problem for a vps? memory load? |
| 01:56.44 | JT | err |
| 01:56.48 | JT | better question is |
| 01:56.58 | JT | what made you think asterisk would work well in a vps? |
| 01:57.03 | JT | vpses are far from ideal |
| 01:57.14 | sof76 | yes, but cheap |
| 01:57.18 | JT | i/o contention, cpu contention |
| 01:57.20 | JT | cheap != good |
| 01:57.28 | docelmo | Just use 1 asterisk install and partition it |
| 01:57.36 | docelmo | would be fairly simple to do |
| 01:57.47 | JT | and the problem is usually with the virtualisation schemes themselves making asterisk unhappy |
| 01:57.49 | sof76 | it depends, for my bank account = good |
| 01:58.04 | JT | sof76: no, as in performance, and working = good |
| 01:58.29 | JT | UML is near useless for asterisk |
| 01:58.32 | JT | xen can work |
| 01:58.40 | JT | openvz might work |
| 01:58.52 | JT | vmware can work, but very low performance |
| 01:58.55 | Gtwy | hmm myphonecompany wants to charge me for sending me equipment... but i dont need any |
| 01:59.01 | Gtwy | ill try calling them tomorrow |
| 01:59.58 | sof76 | It was difficult to make it run on the vps, I will see if it will be ok for 3 or 4 simultaneous calls, which would be all what i need |
| 02:00.20 | JT | sof76: also, you never know when other vps users will load spike your vm |
| 02:02.45 | *** part/#asterisk sof76 (n=chatzill@Z8dc3.z.pppool.de) |
| 02:05.05 | *** join/#asterisk mightnare (n=mike@s230165.ppp.asahi-net.or.jp) |
| 02:05.10 | *** join/#asterisk magic_hat (n=geoffdou@h-74-2-87-16.chcgilgm.covad.net) |
| 02:05.20 | magic_hat | anyone know why I might be getting this? WARNING[10533]: loader.c:499 load_modules: Loading module chan_modem.so failed! |
| 02:05.37 | JT | magic_hat: did you upgrade asterisk from 1.0.x? |
| 02:05.53 | mightnare | hello guys, what is "group = xxx" in users.conf, for callgroup and pickupgroup? |
| 02:06.11 | magic_hat | JT: I just did apt-get asterisk |
| 02:06.53 | shmaltz | mightnare, it's used with features.conf for pickup |
| 02:06.59 | JT | magic_hat: /etc/asterisk/modules.conf |
| 02:07.02 | JT | add |
| 02:07.03 | JT | noload => chan_modem.so |
| 02:07.03 | JT | noload => chan_modem_aopen.so |
| 02:07.03 | JT | noload => chan_modem_bestdata.so |
| 02:07.03 | JT | noload => chan_modem_i4l.so |
| 02:07.05 | magic_hat | I have 1.2.16 |
| 02:07.18 | JT | chan_modem is old crap from 1.0 |
| 02:07.19 | shmaltz | mightnare, sorry my mistake |
| 02:07.24 | JT | that is not used anymore |
| 02:07.34 | shmaltz | whats users.conf anyhow? |
| 02:07.55 | JT | a weird asteriskguiism, some attempt to pull out bits of sip.conf and iax.conf |
| 02:08.10 | shmaltz | JT, 1.4? |
| 02:08.26 | JT | yeah |
| 02:08.57 | mightnare | i've only seen it on asterisk-gui... is it like a combination of callgroup and pickupgroup somehow? |
| 02:09.19 | shmaltz | mightnare, there is a mispelling in your handle |
| 02:09.35 | mightnare | =) |
| 02:10.23 | magic_hat | JT: damn, i'm still getting it. |
| 02:10.36 | shmaltz | only one hit on google: |
| 02:10.38 | shmaltz | http://www.google.com/search?hl=en&q=users.conf+site%3Avoip-info.org&btnG=Search |
| 02:10.38 | magic_hat | do I need to do anyting other than save those changes and restart? |
| 02:10.40 | JT | magic_hat: is there an explicit load in modules.conf for it? |
| 02:10.41 | shmaltz | no docs yet on this |
| 02:10.47 | magic_hat | lemme check |
| 02:11.35 | magic_hat | JT: bingo |
| 02:11.35 | *** part/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 02:11.53 | JT | right :) |
| 02:12.20 | mightnare | i've seen a section about it on TFOT, i don't have the book though... it's only on the latest edition i guess |
| 02:12.45 | JT | that isn't released yet :) |
| 02:14.28 | mightnare | available only through safari for now :D |
| 02:14.40 | shmaltz | ~TFOT |
| 02:14.41 | jbot | rumour has it, tfot is "The Future of Telephony", a book about Asterisk from O'Reilly Publishing, ISBN: 0-596-00962-3, click http://www.oreilly.com/catalog/asterisk/ for more details |
| 02:15.28 | mightnare | hmmm... "Additional content appearing in this section has been removed. Login, Subscribe or Try Safari Now to access the entire content." |
| 02:15.40 | Strom_M | ~thebook |
| 02:15.41 | jbot | rumour has it, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 02:19.32 | shmaltz | anyone in NJ looking for a job? |
| 02:20.01 | *** join/#asterisk b1shop (n=b1shop@c-76-16-224-140.hsd1.il.comcast.net) |
| 02:20.27 | *** join/#asterisk GlobeTrotter (n=eric@35.241.88.200.f.sta.codetel.net.do) |
| 02:21.23 | GlobeTrotter | hi guys,, i am using 1.4.. how do i record all calls coming into my box? |
| 02:21.39 | shmaltz | GlobeTrotter, app_mixmonitor |
| 02:21.43 | J4k3 | microcassette |
| 02:21.52 | davidcsi | exit |
| 02:22.01 | J4k3 | /quit |
| 02:22.12 | shmaltz | GlobeTrotter, |
| 02:22.14 | shmaltz | http://www.voip-info.org/wiki/view/MixMonitor |
| 02:22.23 | GlobeTrotter | thanks guys |
| 02:26.06 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 02:26.40 | *** join/#asterisk Howie69 (n=Howie69@dialup-4.252.12.6.Dial1.Atlanta1.Level3.net) |
| 02:27.06 | Howie69 | I know I'm looking in the wrong place |
| 02:27.15 | Howie69 | like...first off, asterisk.sf.net goes nowhere... |
| 02:27.22 | Howie69 | went to asterisk.org...found some info... |
| 02:27.26 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 02:27.27 | *** join/#asterisk kn0x (n=pinochle@76.76.10.159) |
| 02:27.30 | Howie69 | but looking for supported/reccomended hardware |
| 02:27.48 | JT | to do what? |
| 02:27.51 | kn0x | okay I'm getting the folowing error trying to stary asterisk: |
| 02:27.52 | kn0x | Cannot find your TTY (9) |
| 02:28.05 | kn0x | i've heard something about starting it in screen |
| 02:28.09 | JT | kn0x: vm? |
| 02:28.09 | kn0x | but im not sure how to do that |
| 02:28.14 | kn0x | JT, yuep |
| 02:28.31 | JT | vms don't generally have /dev/tty<x> |
| 02:28.42 | kn0x | so what do i do? |
| 02:28.45 | JT | kn0x: modify safe_asterisk so it doesn't try and put a console on a tty |
| 02:28.59 | Howie69 | I have a nice linux box I'm setting up for a charity, want to use Asterisk over a phone system, looking for interface cards for analog/digital phones |
| 02:28.59 | kn0x | where do i sey the console? |
| 02:29.10 | kn0x | i still want to be able to attatch to the CLI |
| 02:29.16 | JT | kn0x: it will have tty9 somewhere |
| 02:29.21 | JT | and you will still be able to |
| 02:29.24 | kn0x | TTY=9 |
| 02:29.30 | kn0x | CONSOLE=yes |
| 02:29.34 | kn0x | CONSOLE=no? |
| 02:29.40 | JT | kn0x: easy as that |
| 02:30.11 | JT | Howie69: hmm, well i guess you should figure out exactly what type of lines and phones you want to use first |
| 02:30.27 | Howie69 | ANalaog CO lines |
| 02:30.32 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 02:30.32 | *** mode/#asterisk [+o mog] by ChanServ |
| 02:30.32 | Howie69 | the phones are up in the air |
| 02:30.48 | Howie69 | Was looking for suggestions on that, but Ithink I might have found some |
| 02:30.49 | kn0x | JT, do I comment out TTY=? |
| 02:30.52 | JT | sip phones would be ideal |
| 02:30.55 | JT | kn0x: no |
| 02:31.04 | kn0x | well it still said it until i did |
| 02:31.07 | JT | Howie69: how many co lines? |
| 02:31.13 | Howie69 | 2 |
| 02:31.15 | kn0x | and when i started it with my initscript |
| 02:31.15 | JT | kn0x: shrug, maybe then |
| 02:31.28 | Howie69 | looking at 4 to 6 extensions |
| 02:31.32 | kn0x | it outputted the load-up and hung until i hit return |
| 02:31.34 | Howie69 | it's a small nonprofit org |
| 02:31.47 | JT | kn0x: you could always read the script :) |
| 02:31.55 | JT | or not use it |
| 02:31.56 | Howie69 | they have a very overqualified linux box as their server...QuadCore 3ghz, Raid1 250gb drive |
| 02:32.10 | Howie69 | s |
| 02:32.23 | kn0x | JT, well how come i have to hit return after the last line. |
| 02:32.27 | Howie69 | DRIVES :) |
| 02:32.35 | Howie69 | they'll just maybe 20gb in the next 10 years of that space |
| 02:32.37 | JT | kn0x: i have no idea |
| 02:32.41 | Howie69 | use even |
| 02:32.53 | kn0x | JT, will that stall my boot-up? |
| 02:33.05 | kn0x | thats all i care about, i don't start it manually |
| 02:33.45 | JT | kn0x: why don't you find out? |
| 02:35.45 | kn0x | JT, sorry, I guess you have a point |
| 02:35.57 | kn0x | I think i can handle a couple minutes downtime |
| 02:40.17 | Gtwy | all of these sites require that you buy some type of hardware.. gah |
| 02:40.20 | Howie69 | but I don't see that product |
| 02:41.28 | shido6 | ? |
| 02:41.45 | JT | Gtwy: what sites? |
| 02:41.59 | JT | Howie69: i'd prefer sip phones |
| 02:42.04 | shido6 | http://www.thevoipconnection.com/store/catalog/product_16427_Digium_Wildcard_TDM800P.html |
| 02:42.25 | Gtwy | JT: i am running * and would like a provider with a flat local rate that isnt forcing me to purchase any hardware from them, i already have what i need |
| 02:42.48 | JT | Gtwy: err i thought there were heaps that don't need you to buy hardware |
| 02:43.19 | shido6 | how many minutes do you need, Gtwy? |
| 02:43.23 | Gtwy | JT: i am new at this, and there are so many sites to look at i make it so far through the registration until i see "hardware shipping" or whatever |
| 02:43.27 | Gtwy | shido6: i need a flat rate |
| 02:43.51 | *** join/#asterisk toastchee (n=toastche@c-76-26-202-99.hsd1.sc.comcast.net) |
| 02:44.02 | toastchee | hi asterisk gang |
| 02:44.48 | toastchee | i built * from source, but i am stoopid in setting up a single xlite phone |
| 02:45.32 | toastchee | do i want to add a user to user.conf or sip.conf or both |
| 02:45.42 | Gtwy | VOIPGO looks good but you cant transfer a number to them... |
| 02:49.26 | kn0x | hey any developers around? |
| 02:50.12 | Strom_M | toastchee: see the book |
| 02:50.15 | Strom_M | ~thebook |
| 02:50.16 | jbot | from memory, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 02:50.31 | Gtwy | yeah, VOIPGO it is. no setup fees and bring your own hardware.. just what i wanted |
| 02:50.44 | Gtwy | night all |
| 02:50.46 | toastchee | yeah, thanks. I have it. being lazy <dons flamesuit> |
| 02:50.48 | Strom_M | Gtwy: what about teliax |
| 02:50.57 | Strom_M | toastchee: well then ask better questions |
| 02:51.06 | toastchee | :-) no worries mate |
| 02:51.31 | Gtwy | Strom_M: more expensive than voipgo |
| 02:51.37 | Strom_M | alright |
| 02:51.39 | Gtwy | Strom_M: http://www.voipgo.com/plans.htm |
| 02:51.58 | Strom_M | teliax has a per-minute plan also |
| 02:52.13 | vn | ok say I'll use an ATA to an * box and then an IAX2 link between the * and my provider via a router...do I need any special card? or simple ethernet is OK? |
| 02:52.16 | Gtwy | Strom_M: the issue is that i will need flat rate |
| 02:52.24 | Gtwy | but thanks |
| 02:52.36 | Strom_M | Gtwy: how many minutes of call traffic do you expect to generate per month? |
| 02:52.49 | Strom_M | vn: for that, just ethernet is fine |
| 02:53.26 | toastchee | asterisk is amazing! |
| 02:53.32 | Strom_M | cocks |
| 02:54.04 | vn | super then =) |
| 02:54.15 | vn | now I just need to find a good ATA |
| 02:54.53 | shmaltz | anybody want to put their input here: |
| 02:54.54 | shmaltz | http://en.wikipedia.org/wiki/Talk:Asterisk_%28PBX%29#Complete_System |
| 02:55.13 | Strom_M | vn: digium iaxy, linksys pap2t-na |
| 02:56.36 | vn | kthx |
| 02:56.52 | vn | digium sounds better in my head than linksys |
| 02:57.20 | JT | the iaxy is aging a bit, and not very cheap |
| 02:58.32 | vn | then you'd suggest the linky? |
| 02:59.29 | JT | linksys make a lot of ATAs |
| 03:06.39 | *** join/#asterisk Cabal_ (n=Cabal@unaffiliated/cabal) |
| 03:09.03 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 03:09.45 | kn0x | any developers around in here? |
| 03:10.17 | Strom_M | there are some, but you're more likely to find them in #asterisk-dev |
| 03:11.24 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 03:12.08 | kn0x | Strom_M, i dont think anyone is around in there |
| 03:12.27 | Strom_M | well, ask your question here, and maybe someone will pipe up |
| 03:12.31 | kn0x | http://pastebin.ca/593056 |
| 03:12.45 | kn0x | noobie-hacker question |
| 03:13.02 | kn0x | I posted to the mailer |
| 03:13.13 | Strom_M | i'm not a coder, but I'm fairly sure that all that is documented somewhere |
| 03:13.15 | kn0x | but im really impatient so i figured i could ask her |
| 03:13.25 | Strom_M | yeah, patience. |
| 03:13.27 | kn0x | Strom_M, know where? |
| 03:13.38 | Strom_M | ....in the files accompanying the source? |
| 03:14.32 | *** join/#asterisk yxa (n=lonari@58.185.90.101) |
| 03:16.08 | JT | kn0x: i hope you realise that C isn't C++ :) |
| 03:16.37 | kn0x | yes |
| 03:16.43 | yxa | hi my box is behind NAT and I have 2 gateways, going to 2 sip proxies. (I am using static routes) But in sip.conf, i can only specify one externip. how can I overcome this? |
| 03:16.51 | kn0x | JT, why would you think I didnt |
| 03:17.07 | JT | kn0x: many people think C is just a subsidary of C++ |
| 03:17.14 | JT | or subset |
| 03:17.19 | Strom_M | or a subsidiary |
| 03:18.30 | Strom_M | yxa: you may want to try looking into ser |
| 03:19.18 | yxa | Strom_M no other alternatives? Can I run 2 copies of asterisk in one machine? |
| 03:19.32 | Strom_M | yxa: no |
| 03:20.36 | *** join/#asterisk metabox (n=metabox@modemcable192.65-56-74.mc.videotron.ca) |
| 03:23.07 | tzafrir_laptop | yxa, yes, you can. What for? |
| 03:23.41 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 03:27.20 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 03:28.46 | yxa | tzafrir_laptop i just have a weird network situation |
| 03:30.44 | tzafrir_laptop | yxa, assuming you'll use the same binaries, use asterisk -C to provide an alternative asterisk.c |
| 03:31.19 | yxa | i have 2 gateway, but i can only put one externip= in sip.conf |
| 03:31.39 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 03:31.41 | tzafrir_laptop | in it point to a different configuration directory and a different varraun directory. Mayby also different spool directory (who watchs call files?) |
| 03:32.10 | mosty | is there a way to see zaptel channel groups from the asterisk console? |
| 03:32.40 | tzafrir_laptop | yxa, that is not to say I understand why it will help you |
| 03:33.46 | tzafrir_laptop | mosty, I don't think o, but it should be a simple patch to chan_zap |
| 03:33.54 | tzafrir_laptop | chan_zap.c |
| 03:35.06 | tzafrir_laptop | add an extra line to the 'zap show channel NN' to show the 'group' (IIRC) member. |
| 03:35.07 | mosty | tzafrir_laptop, i just want to confirm that you can group two separate PRI spans together by using the same group= setting |
| 03:37.51 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:39.08 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:45.17 | BSD_Tech | 586 would be p2 or p3 |
| 03:45.50 | JT | no |
| 03:45.58 | JT | it would be a Pentium |
| 03:46.31 | mosty | p2/p3 are 686 |
| 03:46.50 | BSD_Tech | ok I have a issue with the g729 not working |
| 03:47.01 | BSD_Tech | and trying to test |
| 03:47.11 | BSD_Tech | to find out wich one is right |
| 03:47.27 | BSD_Tech | its a geode gx2 cpu |
| 03:47.41 | JT | 486 |
| 03:47.59 | BSD_Tech | it boots rpath |
| 03:48.09 | BSD_Tech | asterisk now and thats 686 |
| 03:48.09 | JT | 486 arch. |
| 03:48.34 | BSD_Tech | the website says 686 |
| 03:48.40 | BSD_Tech | its a 686 366 |
| 03:48.41 | JT | it's wrong |
| 03:48.50 | JT | unless it's a really new one |
| 03:48.57 | JT | most geodes are 486 arch |
| 03:49.01 | JT | some are 586 i think |
| 03:49.30 | BSD_Tech | ok so pentium |
| 03:49.42 | JT | try 486 |
| 03:50.01 | JT | what model geode processor is it? |
| 03:50.55 | mosty | i'm fairly sure the geode gx2 is 586 |
| 03:51.00 | BSD_Tech | gx2 |
| 03:51.04 | *** join/#asterisk bmg505 (n=leon@196.209.183.17) |
| 03:51.12 | BSD_Tech | 366 mhz |
| 03:53.14 | mosty | the the gx2 doesn't operate at 366MHz, only 266, 333 or 400 |
| 03:54.17 | JT | mosty: it does |
| 03:54.36 | JT | the Geode GX 500 operates at 366MHz |
| 03:54.42 | mosty | i said GX2 |
| 03:55.35 | mosty | i have a soekris net4801, which has a geode GX2, it's a 586 class cpu |
| 03:55.45 | JT | yes the gx2 is 586 |
| 03:57.02 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 03:57.16 | BSD_Tech | ok the pentium2 build works |
| 03:57.27 | *** join/#asterisk b1shop (n=b1shop@c-76-16-224-140.hsd1.il.comcast.net) |
| 03:57.38 | BSD_Tech | amd had a 366 |
| 03:57.56 | BSD_Tech | ok I have g729 working |
| 03:57.58 | mosty | i would not recommend transcoding on a geode GX2, they are pretty damn slow cpu's |
| 03:58.16 | JT | maybe BSD_Tech likes a challenge ;) |
| 03:58.19 | BSD_Tech | its there just incase |
| 03:58.27 | BSD_Tech | yes I do |
| 03:58.34 | BSD_Tech | I want to see it crash |
| 03:58.57 | BSD_Tech | heck I have asterisknow beta6 running on it with 5 megs of mem left |
| 03:59.17 | BSD_Tech | I have to get the 512 chip out of the closet and replace it |
| 03:59.28 | BSD_Tech | it has 128 megs by default |
| 04:00.07 | Strom_M | Dear Mac OS X Panther or Tiger or Lion or whatever the hell irritatingly clever name 10.4 is called: |
| 04:00.31 | Strom_M | go die in a fire |
| 04:00.31 | Strom_M | love, Strom |
| 04:00.54 | BSD_Tech | http://dataevolution.com/products%203.htm the dectop unit |
| 04:01.17 | BSD_Tech | Lepoard |
| 04:01.18 | JT | BSD_Tech: why don't you just go all out and put trixbox on it? |
| 04:01.22 | BSD_Tech | I think |
| 04:01.30 | BSD_Tech | you sick in the head |
| 04:02.02 | BSD_Tech | after what just happened I have no plans to support the freepbx / trashbox community |
| 04:02.45 | BSD_Tech | I feel sorry for xrob but the person whoi is taking over needs a reality check |
| 04:02.46 | JT | what just happened? |
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| 04:02.58 | BSD_Tech | xrob steped down |
| 04:03.08 | JT | ok |
| 04:03.15 | BSD_Tech | an the put philipell in his place |
| 04:03.41 | BSD_Tech | and I think it should have bean a selected board not just 1 person heading it |
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| 04:04.17 | BSD_Tech | because of what just happened xrob disapierd and then things came to a almost deafing halt in dev |
| 04:04.18 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 04:04.33 | BSD_Tech | sorry tired and typing bad |
| 04:04.35 | Strom_C | grrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrr |
| 04:04.47 | Strom_C | i revise my earlier open letter |
| 04:05.02 | BSD_Tech | and philipell has controll and attitude issues |
| 04:05.02 | Strom_C | Dear Mac OS X Tiger / Panther / Catsex / 10.4: |
| 04:05.08 | *** join/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 04:05.12 | Strom_C | eat cocks |
| 04:05.12 | BSD_Tech | that I am not willing to deal with |
| 04:05.13 | Strom_C | love, strom |
| 04:05.27 | JT | why were you dealing with them at all? |
| 04:05.31 | BSD_Tech | storm but bsd on it and be done with it |
| 04:05.39 | Strom_C | uh, no |
| 04:05.49 | BSD_Tech | because I had client using trixbox |
| 04:06.11 | BSD_Tech | now I am working on asterisk+gui |
| 04:06.23 | kiscokid | which gui? |
| 04:06.28 | Strom_C | why is the classic zealot kiddie response to "i'm irritated with my OS" ALWAYS "LOL, switch OSes to [my favorite OS]!!!!"? |
| 04:06.34 | BSD_Tech | the asteriskgui |
| 04:06.40 | BSD_Tech | by digium |
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| 04:07.13 | kiscokid | BSD_Tech: can you put asteriskgui on 4.1? |
| 04:07.30 | BSD_Tech | 1.4 yes |
| 04:07.51 | kiscokid | how difficult is it to install and use? |
| 04:07.54 | BSD_Tech | it uses the built in webserver in 1.4 |
| 04:08.12 | BSD_Tech | just get the svn and run make make install |
| 04:08.25 | BSD_Tech | and then edit manager.conf and httpd.conf |
| 04:08.55 | kiscokid | any doc on this? |
| 04:08.56 | JT | Strom_C: i swear, if BSD_Tech tells another person to install bsd... ;) |
| 04:09.22 | BSD_Tech | freebsd rules the world you all should install it and use a real unix os |
| 04:09.37 | Strom_C | blow me |
| 04:09.55 | JT | BSD_Tech: everytime you say shit like that, i feel more put off using bsd |
| 04:09.56 | BSD_Tech | we have asterisklibpri zaptel addons and the gui ported |
| 04:10.30 | JT | i don't have a problem with bsd, only its insane followers |
| 04:10.32 | BSD_Tech | but lumenvox does not support bsd |
| 04:10.42 | BSD_Tech | I am not insane |
| 04:11.03 | BSD_Tech | I just know where my heart is |
| 04:11.09 | JT | sure you're not |
| 04:11.13 | JT | fairyland? :) |
| 04:11.30 | BSD_Tech | leave you fairy;s out of this |
| 04:11.53 | BSD_Tech | only daemons in this argument |
| 04:12.20 | kiscokid | after Microsoft kills off linux we'll still have bsd |
| 04:12.24 | JT | you need to learn to live and let live, things like linux :) |
| 04:12.48 | JT | a lot of bsd users have an overinflated worth of their os |
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| 04:13.55 | BSD_Tech | BSD for Betch |
| 04:14.34 | BSD_Tech | BSD FOR EVER |
| 04:14.34 | BSD_Tech | oops |
| 04:14.34 | anthonyc | FreeBSD -4- LIFE |
| 04:14.34 | Strom_C | now look what you ddi |
| 04:14.34 | JT | talking shit for the lose |
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| 04:15.02 | mosty | is freebsd faster than linux? |
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| 04:15.04 | JT | mosty: apples/oranges, depends on your setup |
| 04:15.04 | BSD_Tech | depends on how well you tune your system |
| 04:15.20 | BSD_Tech | sam and be said for *inux |
| 04:15.25 | JT | i don't think you'd use either for a performance advantage unless it's a specific task that has proven to favour one particularly |
| 04:15.59 | BSD_Tech | well now that bsd finaly has flash on the way and gnuflash is working better |
| 04:16.06 | BSD_Tech | no need for the L word |
| 04:16.15 | JT | flash? |
| 04:16.26 | rob0 | Lesbian? |
| 04:16.37 | JT | there's plenty of need |
| 04:16.38 | BSD_Tech | flashplayer |
| 04:16.44 | mosty | i'd consider freebsd if there was some distinct reason to besides personal preference |
| 04:16.48 | JT | like all the hardware that doesn't work on bsd |
| 04:16.58 | JT | or if you have an Ultrasparc III |
| 04:16.59 | JT | etc |
| 04:17.01 | BSD_Tech | its a real unix |
| 04:17.13 | BSD_Tech | it has a huge fallowing |
| 04:17.15 | JT | and you spin real rhetoric bs, what's your point? |
| 04:17.24 | JT | so does linux, popularity means nothing |
| 04:17.35 | BSD_Tech | yes I grant you linux supports alot more hard ware but is alot more bloated |
| 04:17.47 | JT | some distros are shit |
| 04:17.52 | JT | no-one is forcing you to use them |
| 04:18.01 | BSD_Tech | they try to cram every new device in with out fully testing |
| 04:18.24 | BSD_Tech | well once I get my iso for asterisk done I wont |
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| 04:18.33 | rootfield | hi all |
| 04:18.34 | JT | more handwaving |
| 04:18.36 | BSD_Tech | but for now asterisknow does the trick |
| 04:18.39 | JT | hi rootfield |
| 04:19.29 | BSD_Tech | but I have to get a few apps to build right like flite and app flite |
| 04:19.29 | BSD_Tech | wich build but wont install |
| 04:20.49 | rootfield | i'm running asterisk 1.4.5 on ubuntu.. using intel xeon processor.. how can i use ipp g723 codec on my asterisk? on asterisk 1.2 work fine with open g723 codec.. when i try to run open g723 codec for asterisk 1.4 the sound not work fine.. but de module codec is up |
| 04:21.23 | JT | rootfield: don't know, we don't support that module here |
| 04:21.45 | rootfield | hmm ok |
| 04:21.59 | JT | you still need a licence to use the codecs |
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| 04:23.10 | BSD_Tech | in the US to use the g729 in commercial yes |
| 04:23.26 | JT | in many countries |
| 04:23.45 | mosty | you may also need a copy of IPP to use those modules |
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| 04:32.50 | _VoiceMeUp_COM | chan_zap.c:7947 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 |
| 04:32.54 | _VoiceMeUp_COM | any knwon casue ? |
| 04:33.33 | JT | bad zap timing |
| 04:33.39 | JT | random error |
| 04:33.40 | _VoiceMeUp_COM | k |
| 04:33.56 | _VoiceMeUp_COM | first time i get it is a good sign i guess |
| 04:34.58 | _VoiceMeUp_COM | i see irq could do this too |
| 04:36.19 | [[blah]asfd | how do i tell if i am providing the timing source or my t1 provider is in my config files? the provider says that I should be providing it. |
| 04:36.32 | Strom_C | the provider is insane |
| 04:36.35 | Strom_C | they should be providing timing |
| 04:36.50 | [[blah]asfd | i thought that too |
| 04:37.03 | _VoiceMeUp_COM | unles its framing ? |
| 04:37.05 | JT | idiot provider |
| 04:37.07 | Strom_C | no |
| 04:37.11 | [[blah]asfd | the technician said that they only provide timing for customers in alaska |
| 04:37.11 | _VoiceMeUp_COM | but timing is not you i think |
| 04:37.13 | _VoiceMeUp_COM | weird |
| 04:37.16 | _VoiceMeUp_COM | lol |
| 04:37.24 | Strom_C | _VoiceMeUp_COM: go read T1 101 again |
| 04:37.30 | _VoiceMeUp_COM | here goes that alaska datacenter porject again |
| 04:38.41 | [[blah]asfd | :-) |
| 04:39.02 | _VoiceMeUp_COM | only thing i specify for the t's is the span framing stuff |
| 04:39.09 | [[blah]asfd | getting second opinion from different tech now... thanks. not changing anything yet |
| 04:39.13 | _VoiceMeUp_COM | no need for T1 101 i dont sell t1's |
| 04:39.40 | _VoiceMeUp_COM | im not going to take an electric engenieer course becasue i ddont need to kknow the electron path or direction |
| 04:39.48 | _VoiceMeUp_COM | and not going to take a health class before i eat |
| 04:39.50 | _VoiceMeUp_COM | here you go |
| 04:40.00 | _VoiceMeUp_COM | j/k Strom_C |
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| 04:40.40 | BSD_Tech | night kids |
| 04:41.18 | _VoiceMeUp_COM | rich aim me |
| 04:41.25 | _VoiceMeUp_COM | good news lol |
| 04:41.25 | JT | [[blah]asfd: you're in the us? |
| 04:41.31 | [[blah]asfd | yes |
| 04:41.52 | JT | [[blah]asfd: sounds like a think an idiotic third world telco would say |
| 04:41.56 | JT | provide timing on the t1 |
| 04:42.01 | JT | that's a bad idea for so many reasons |
| 04:42.09 | [[blah]asfd | well, its global crossing ;-) |
| 04:42.22 | _VoiceMeUp_COM | In the T1 world, clock signals are not transmitted separately from the data stream. Instead, receivers must extract the clock from the data signal based on the stream itself. |
| 04:42.39 | [[blah]asfd | JT: so you are saying that i should be expecting timing from the carrier, not providing it to? |
| 04:42.40 | _VoiceMeUp_COM | so i guess its in the stream |
| 04:42.47 | [[blah]asfd | that is how i thought is was supposed to be. |
| 04:42.57 | _VoiceMeUp_COM | from Strom_C suggesstion i decided to do some reading |
| 04:42.57 | _VoiceMeUp_COM | http://www.oreilly.com/catalog/t1survival/chapter/ch05.html |
| 04:43.10 | [[blah]asfd | but they have set my t1 up to expect timing from me... then wonder why it stops working every few weeks. |
| 04:43.28 | _VoiceMeUp_COM | timing and every 2 weeks ? |
| 04:43.43 | _VoiceMeUp_COM | i think the only timing thing every 2 weeks . that could stop your service is called the bill |
| 04:43.44 | _VoiceMeUp_COM | ;) |
| 04:44.05 | JT | [[blah]asfd: are you set to provide it then? |
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| 04:44.29 | [[blah]asfd | JT: no... we were told by our customer service rep that they were providing it... |
| 04:44.34 | [[blah]asfd | so doing my research now |
| 04:44.49 | JT | hmm |
| 04:45.14 | [[blah]asfd | _VoiceMeUp_COM: I am not sure who you get your service through... but I would shy away from anyone that expected payment every two weeks ;-) |
| 04:45.24 | [[blah]asfd | means they are short on cash, or do not trust you. |
| 04:45.37 | _VoiceMeUp_COM | wasjoking ;) |
| 04:45.44 | [[blah]asfd | I know.... me too... |
| 04:45.51 | _VoiceMeUp_COM | ;) |
| 04:46.15 | _VoiceMeUp_COM | blah |
| 04:46.15 | _VoiceMeUp_COM | http://www.marko.net/asterisk/archives/0211/0493.html |
| 04:46.21 | _VoiceMeUp_COM | v |
| 04:46.23 | _VoiceMeUp_COM | #span =<span num>, <timing>,<line build out>,<framing>,coding> |
| 04:46.34 | _VoiceMeUp_COM | timing values can be 0 - not used as timing source 1 - primary timing source 2 - Secondary timing source |
| 04:46.37 | _VoiceMeUp_COM | try 1 |
| 04:46.44 | JT | err what |
| 04:46.48 | _VoiceMeUp_COM | maybe htat waht theymeant |
| 04:46.49 | JT | if he has to provide timing |
| 04:46.54 | JT | which he shouldn't |
| 04:46.54 | mosty | use 0 if you are providing timing |
| 04:46.57 | JT | it needs to be 0 |
| 04:47.07 | _VoiceMeUp_COM | oh |
| 04:48.08 | [[blah]asfd | currently using 1 |
| 04:48.17 | [[blah]asfd | i thought that meant that the t1 provided the timing source |
| 04:48.29 | _VoiceMeUp_COM | me too |
| 04:48.37 | mosty | 1 means the other end provides timing |
| 04:48.53 | _VoiceMeUp_COM | mainly to say if card providers or not.. i see you can share timing between cards too |
| 04:48.59 | JT | no, 1 means other end provides timing with priority 1 |
| 04:49.00 | mosty | any positive int means the other end provides timing, i think |
| 04:49.13 | [[blah]asfd | now... i may have this all wrong... let me paste. |
| 04:49.24 | _VoiceMeUp_COM | <PROTECTED> |
| 04:49.32 | _VoiceMeUp_COM | Timing set to 1 means that the speed of the clock will be based on |
| 04:49.32 | _VoiceMeUp_COM | > this T1, (provided the T1 is up). |
| 04:49.41 | _VoiceMeUp_COM | Timing set to 2 means that the speed of the clock will be based on |
| 04:49.41 | _VoiceMeUp_COM | > this T1 if the primary T1 is down. |
| 04:49.46 | JT | higher numbers mean secondary, tertiary, quaternary and so on priority timing sources to RECEIVE timing from |
| 04:49.48 | _VoiceMeUp_COM | Set to "0" means that this span is not eligible as a timing source. |
| 04:49.52 | JT | _VoiceMeUp_COM: enough pasting already |
| 04:50.01 | _VoiceMeUp_COM | sorry |
| 04:50.19 | [[blah]asfd | http://pastebin.ca/593145 |
| 04:50.32 | JT | basically all PDH clocks should be roughly synchronised on a pdh network |
| 04:50.41 | JT | so not as stringent as on SDH |
| 04:50.52 | [[blah]asfd | getting over my head |
| 04:51.02 | JT | but it's definitely not IP, where things are completely asynchronous |
| 04:51.12 | [[blah]asfd | i thought it was as simple as my t1 card did the timing, or the provider did the timing. looks like i was way off the mark |
| 04:51.53 | JT | Plesiochronous Digital Heirarchy = T carrier, E carrier TDM networks |
| 04:52.25 | JT | [[blah]asfd: it's pretty simple, if you want to provide timing, always set it to 0 |
| 04:52.42 | JT | if you want to receive, set the rest up starting at 1 onwards |
| 04:52.46 | JT | choosing the best sources first |
| 04:54.10 | [[blah]asfd | ok... dumb question then... if i am providing timing... where am i providing it from? just by setting it to 0 does not take care of it all, does it? |
| 04:54.22 | JT | yes, set it to 0, that's it |
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| 04:54.37 | [[blah]asfd | is it my cmos clock that takes care of it? |
| 04:54.50 | [[blah]asfd | does that mean that i have to have the clock set right ;-) |
| 04:54.51 | mosty | no, the E1/T1 card has a timer |
| 04:54.52 | JT | obviously you're not synchronised to the TDM network, unless you have another span with a non-zero timing argument |
| 04:55.01 | JT | which is why you may experience frame slips sometimes |
| 04:55.06 | JT | and why your telco is an idiot |
| 04:55.36 | [[blah]asfd | checking on the idiot part right now ;-) |
| 04:55.43 | [[blah]asfd | ok... thanks all... headed to bed |
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| 05:15.51 | NightMonkey | Hi. Got an odd problem here. For some reason, my SIP phone can't authenticate without "host=dynamic" set. I tried to explicitly set "host=<ip>" but Asterisk would reject the registration. |
| 05:16.38 | NightMonkey | Would that point to another problem with my setup, or should I just live with "host=dynamic" and call it a day? |
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| 05:22.12 | [[blah]asfd | NightMonkey: please post a copy of the sip.conf |
| 05:22.17 | Strom_M | NightMonkey: because static hosts don't register |
| 05:22.24 | Strom_M | if it's a phone, leave it at dynamic |
| 05:22.30 | Strom_M | that's the way it's supposed to be |
| 05:22.36 | NightMonkey | Strom_M: Ah, I see! Thank you! |
| 05:22.38 | [[blah]asfd | type=friend? |
| 05:23.15 | NightMonkey | Is there any reason to explicitly set the host= ? |
| 05:23.23 | Strom_M | host=dynamic, yes |
| 05:23.28 | JT | if you don't want to register |
| 05:24.04 | NightMonkey | JT: OK, now it is making sense. For non-registered SIP devices, set host=<ip> to allow connections, yes? |
| 05:24.28 | JT | set host= so it knows where to send calls |
| 05:24.32 | JT | it doesn't auth |
| 05:24.35 | JT | usually |
| 05:24.46 | JT | that's what the username and secret is for |
| 05:24.46 | NightMonkey | JT: Gotcha. Thank you. |
| 05:25.07 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 05:27.46 | sweeper | ok, this is probably really dumb |
| 05:27.54 | sweeper | but I just installed a new asterisk system |
| 05:28.07 | sweeper | and it's not listening on the SIP port |
| 05:28.18 | sweeper | is there something in globals that needs to be set? |
| 05:30.14 | snuff-work | u sure u have iptables off? |
| 05:30.48 | Supaplex | s/u/you/ |
| 05:32.57 | sweeper | snuff-work: pretty sure... |
| 05:33.10 | sweeper | yea, it's not running |
| 05:36.23 | *** part/#asterisk [[blah]asfd (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 05:37.45 | sweeper | HUH |
| 05:37.47 | sweeper | this is odd |
| 05:37.53 | sweeper | [Jun 28 02:37:34] WARNING[98466]: loader.c:415 load_dynamic_module: Error loading module 'chan_sip': /usr/local/lib/asterisk/modules/chan_sip.so: Undefined symbol "ast_park_call" |
| 05:37.56 | sweeper | wtf? |
| 05:39.12 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:39.52 | sweeper | oh, apparenlty I need to configure autoloading |
| 05:44.20 | *** join/#asterisk alphaque (n=alphaque@60.51.217.61) |
| 05:47.52 | rue_mohr | anyone ever used kphone? |
| 05:48.12 | sweeper | gah |
| 05:48.17 | rue_mohr | ? |
| 05:48.19 | sweeper | there has got to be something wrong here |
| 05:48.41 | sweeper | I can't telnet to 5060, chan_sip is loaded, bindport and bindaddr are set correctly |
| 05:48.41 | Nugget | telnet is eeeeeeevil! |
| 05:48.55 | JT | sweeper: but is it listening on the port? |
| 05:49.19 | sweeper | JT: it would seem not! since telnet gets a connection refused |
| 05:49.41 | sweeper | which puzzles me |
| 05:49.42 | JT | err |
| 05:49.46 | mosty | sweeper, netstat -na | grep LISTEN |
| 05:49.51 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 05:49.51 | JT | that is NOT how to check if asterisk is listening |
| 05:49.55 | JT | netstat -a |
| 05:49.56 | JT | and lsof |
| 05:50.17 | sweeper | it's not~ |
| 05:50.20 | JT | lsof |grep 5060 |
| 05:50.24 | JT | ok |
| 05:50.53 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-75-84-232-194.socal.res.rr.com) |
| 05:50.55 | JT | sweeper: what about if you unset bindport and bindaddr? |
| 05:51.40 | sweeper | nothing |
| 05:51.52 | JT | so umm |
| 05:52.03 | JT | how about you check the messages from chan_sip on asterisk startup? |
| 05:53.06 | mosty | is chan_sip.so loaded? |
| 05:53.14 | sweeper | chan_sip is loaded |
| 05:53.26 | sweeper | I get no messages from asterisk startup about chan_sip |
| 05:53.51 | JT | there should be some |
| 05:55.07 | sweeper | ooo, I lied |
| 05:55.18 | sweeper | there were some stupid things in asterisk.conf |
| 05:55.51 | sweeper | [Jun 28 02:56:02] == Parsing '/usr/local/etc/asterisk/sip.conf': [Jun 28 02:56:02] Found |
| 05:55.54 | sweeper | thar |
| 05:55.56 | sweeper | [Jun 28 02:56:02] WARNING[98597]: chan_sip.c:15540 handle_common_options: insecure=very at line 18 is deprecated; use insecure=port,invite instead |
| 05:55.59 | sweeper | [Jun 28 02:56:02] chan_sip.so => (Session Initiation Protocol (SIP) |
| 05:56.41 | sweeper | ohoho |
| 05:56.49 | sweeper | I have an entry for port 5060 in netstat now |
| 05:56.58 | sweeper | but it's not in a LISTEN state |
| 05:57.04 | *** join/#asterisk |R (i=bob@modemcable241.28-203-24.mc.videotron.ca) |
| 05:57.12 | JT | what state? |
| 05:57.22 | sweeper | none |
| 05:58.24 | sweeper | I mean, it's just blank |
| 05:58.52 | mosty | er, pastebin the output from netstat -na |
| 05:59.33 | sweeper | http://pastebin.ca/593186 |
| 05:59.35 | |R | Anyone has experience with nokia's (or others?) wifi VoIP phones and asterisk? i need to upgrade my phone and i'm looking for an unlocked gsm phone with wifi/sip so i can forward call to/by/from home... Nokia N95 looks nice and all but i don't care about the GPS / camera and other gizmos... i just want a cheap dual mode phone that integrates just as well.. :) |
| 06:00.26 | sweeper | nokia works, but you can't use password auth, last I knew |
| 06:00.52 | sweeper | might be better off just picking up any winmo phone that has wifi on it |
| 06:01.24 | sweeper | I blame this fancypants asterisk 1.4 for my problems, really |
| 06:01.28 | |R | hehe |
| 06:01.38 | |R | winmo eh... let me google a bit ;) |
| 06:01.45 | sweeper | windows mobile :P |
| 06:01.52 | |R | ah! damn! |
| 06:02.04 | |R | i should have said as far away from giving money to M$ as possible haha ;) |
| 06:02.11 | Supaplex | try windows cripple *BLAM* *BLAM* |
| 06:02.16 | sweeper | ehhhh |
| 06:02.23 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 06:02.37 | sweeper | honestly, as bad as most phone uis are, winmo is far preferable |
| 06:02.45 | sweeper | I mean, at least you can install stuff on it |
| 06:03.09 | |R | you "can" if you count java on many phones, no? |
| 06:03.21 | sweeper | not really |
| 06:03.28 | sweeper | most of them won't let you install stuff either |
| 06:03.46 | sweeper | my helio drift runs java apps, but I can only install stuff I buy from helio |
| 06:03.48 | |R | oh? actually all i would need as an add-on with the wifi over the probably included browser would be ssh... |
| 06:04.02 | |R | ok, but if i buy an unlocked phone? |
| 06:04.02 | sweeper | yea, good luck getting ssh |
| 06:04.39 | |R | shouldn't i be able to get putty ? |
| 06:04.39 | sweeper | mm |
| 06:04.39 | |R | http://s2putty.sourceforge.net/ |
| 06:04.39 | |R | at least for symbian |
| 06:04.39 | sweeper | I don't think you can even do it on an unlocked phone, from what I've heard |
| 06:04.39 | sweeper | I mean, getting ssh from the carrier |
| 06:04.51 | |R | oh you meant over gsm? |
| 06:05.00 | sweeper | no, I mean like in their little store |
| 06:05.16 | sweeper | there's a symbian and a windows mobile version, and there's also a mindterm version for mobile phones |
| 06:05.35 | |R | ok but putty is free all i need is a cable or something no? |
| 06:05.57 | sweeper | no, you need a phone that lets you install things :) |
| 06:06.06 | sweeper | aka symbian or windows mobile |
| 06:06.17 | |R | ah ok, so i'm stuck with nokia i guess |
| 06:06.28 | |R | http://www.flickr.com/photos/edink/466405671/ |
| 06:06.33 | |R | ^- this made me drool ;) |
| 06:08.41 | *** join/#asterisk BugKhaM (n=LAMER@ppp-58.8.17.25.revip2.asianet.co.th) |
| 06:09.36 | |R | so as i just ordered a linksys VoIP gw and plan on switching my cell eventually and investing the saved money in a better internet connection...i thought i'd take a look at a cell solution :) |
| 06:10.26 | sweeper | D: |
| 06:10.33 | sweeper | mosty: any ideas? |
| 06:11.06 | |R | i'll start by reading on asterisk and breaking my line for a month i'm sure haha ;) |
| 06:11.13 | mosty | sweeper, it's listening, what isn't working exactly? |
| 06:12.06 | *** join/#asterisk yonahw-work (n=yonahw-w@genie03-173-74.inter.net.il) |
| 06:12.12 | sweeper | uh, none of my sip clients can make the asterisk console say anything, verbosity is turned up |
| 06:12.39 | sweeper | and telnet gets a connection refused to that port, which it shouldn't |
| 06:12.49 | Supaplex | your clients bore * ;) |
| 06:12.52 | mosty | telnet is tcp |
| 06:12.56 | sweeper | OH |
| 06:13.01 | sweeper | right |
| 06:13.02 | mosty | but asterisk/sip is udp |
| 06:13.09 | JT | |R: sorry, what's the main benefit in function you're hoping to gain from wifi/sip on a mobile phone? |
| 06:13.16 | mosty | sweeper, do you have a firewall? |
| 06:13.21 | |R | nc -u :) |
| 06:13.40 | sweeper | well, iptables isn't running |
| 06:13.47 | sweeper | but it's been ages since I used freebsd.... |
| 06:14.04 | Supaplex | freebsd uses pf |
| 06:14.21 | mosty | sweeper, can you nmap the asterisk box? |
| 06:14.45 | |R | JT: well, a hacked up browser / ssh client would be a good thing on the wifi part... the SIP is to be able to use wifi to call when i'm in town... i'd eventually switch my gsm to per-minute charge as i don't use it much... i'm trying to converge my bell / cell to 1 VoIP setup |
| 06:14.57 | sweeper | I'm pretty sure pf isn't running, I did the minimal install |
| 06:15.05 | sweeper | I'll be really pissed if it is |
| 06:15.36 | Supaplex | sweeper: not by default. what's uname say? :) |
| 06:15.42 | |R | JT: basically to link to my home or get call forwarded if i know i need to... or have a phone i can use as if i was calling from home while in japan for free or whatever :) |
| 06:16.05 | |R | JT: i really hate giving money to telcos that charges me for stupide services instead of bandwith ;) |
| 06:16.24 | sweeper | 6.2-RELEASE |
| 06:16.44 | JT | |R: wifi phones suck, especially if you're not uber close to the AP |
| 06:16.55 | JT | |R: wifi is the only bit that might work in .jp |
| 06:17.03 | |R | JT: how close? |
| 06:17.15 | JT | dozen metres |
| 06:17.21 | JT | or something like that |
| 06:17.22 | |R | ok |
| 06:17.38 | Supaplex | wifi on mobile pcs using softphone suck to |
| 06:17.45 | JT | |R: hope you know gsm isn't used in japan |
| 06:17.50 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 06:18.17 | |R | i didn't but i said japan because i plan on going there for travel in the next year... but it doesn't mather :) |
| 06:18.38 | JT | 3G is the only thing from overseas that will work |
| 06:18.44 | |R | i'm just trying to figure out a general solution that could be useful considering how easy it is to find AP |
| 06:18.46 | JT | with roaming to japan pre enabled by telco |
| 06:18.54 | |R | but 3G needs a service plan |
| 06:19.00 | JT | yes no shit |
| 06:19.04 | |R | which while traveling around won't work hehe ;) |
| 06:19.10 | JT | you cannot buy a SIM when you get to japan without residency |
| 06:19.15 | |R | woah |
| 06:19.17 | JT | would work fine |
| 06:19.27 | |R | yeah but it would be expensive |
| 06:19.31 | |R | like 5$ / min? |
| 06:19.55 | |R | (ok i've got a crappy provider i guess but i'm a very small user, it just has to be there sometimes...) |
| 06:19.57 | JT | probably not that much |
| 06:20.07 | JT | satellite phone isn't much difference |
| 06:20.51 | JT | also, don't even thing about relying on a payphone to call overseas when there |
| 06:20.54 | |R | i'll need to read more on net-calling convention too... i wish i could just ditch the old telephone network all together but that'll have to wait a bit ;) |
| 06:21.01 | JT | either use voip or a mobile phone or someone else's landline |
| 06:21.01 | |R | haha no of course :) |
| 06:21.18 | |R | So you've been there? live there? |
| 06:21.20 | JT | i used voip on my laptop in the airport |
| 06:21.24 | JT | i was there 1.5 weeks ago |
| 06:21.27 | |R | oh :) |
| 06:21.30 | |R | liked it? |
| 06:21.34 | JT | yeah |
| 06:21.40 | JT | Internet is really fast on hotels |
| 06:21.46 | JT | downloads of 40Mbit/s is not uncommon |
| 06:21.50 | |R | woah |
| 06:21.52 | JT | and most hotels have ethernet ports |
| 06:21.58 | |R | yeah! |
| 06:22.04 | |R | i'm moving! |
| 06:22.04 | JT | don't leave your laptop at home |
| 06:22.05 | |R | haha ;) |
| 06:22.29 | |R | I'll need to get an OQO or something :) |
| 06:22.34 | |R | how cheap are the laptops there? |
| 06:22.49 | JT | oqo? |
| 06:22.52 | JT | cheapish |
| 06:22.58 | |R | oqo.com, a very small umpc |
| 06:23.11 | JT | my friend picket up a hp tablet pc for , i guess USD$1000 equivalent |
| 06:23.16 | JT | picked |
| 06:23.37 | JT | only thing is it's a combined english/japanese keyboard |
| 06:23.42 | JT | apart from that, no probs |
| 06:23.49 | |R | it probably looks cooler anyway ;) |
| 06:24.39 | JT | heh |
| 06:24.43 | JT | space bar is a little small |
| 06:24.59 | JT | too many american tourists in japan though |
| 06:24.59 | |R | meta-alt-3rd-option-command-new-fn-key? ;) |
| 06:25.00 | JT | ;) |
| 06:25.13 | |R | I'll be the canadian one then ;) |
| 06:25.19 | JT | one of the many "switch to japanese" keys |
| 06:25.46 | |R | typing in .jp is funny... k then a KA symbol then again.. then a kanji shows up |
| 06:26.10 | JT | ah, canadian, that's alright then |
| 06:26.13 | JT | they're cool |
| 06:26.17 | |R | i had a japanese class once, it's pretty amazing to see the text change :) |
| 06:26.29 | |R | where are you from? |
| 06:26.33 | JT | whenever we heard an american accent, we thought/said "oh no! americans!" |
| 06:26.35 | JT | australia |
| 06:26.38 | |R | hehe ok :) |
| 06:27.00 | |R | i'm actually from quebec, so my first language is french... |
| 06:27.11 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 06:27.21 | JT | out of 4 groups of americans that we talked to/could hear talking, only one werent rude tools |
| 06:27.27 | |R | but hey, gotta survive surrounded by 330M english-speaking persons ;) |
| 06:27.45 | |R | hehe ok :) |
| 06:27.54 | |R | i've had pretty mixt experience too while in europe |
| 06:27.56 | JT | seriously |
| 06:28.02 | JT | not giving themselves a good image |
| 06:28.16 | |R | half/half i'd say... at least the one travelling should be a bit more open but ... |
| 06:28.31 | JT | one american guy to his wife at a money exchange place "these people are small, but at least they can change our money to their stupid yen" |
| 06:28.46 | |R | ah man... |
| 06:28.54 | |R | that's really stupid |
| 06:29.17 | tzafrir | good morning. I looked at the list here and for a momet I thought JT was talking to himself. Those nicks look a bit similar :-) |
| 06:29.17 | |R | were they as small as the person was wide? ;) |
| 06:29.30 | |R | haha :) |
| 06:29.40 | *** join/#asterisk Hackbanger (n=hackbang@mail.newtention.de) |
| 06:29.46 | Hackbanger | moin |
| 06:29.52 | JT | then the american man who talked to me in a computer store in akhibara "oh, more Australians! i've been finding the prices here aren't very good. back in the states you can get things cheaper, especially online!" |
| 06:30.32 | sweeper | yea, I only live in the states because bandwidth and hardware are cheap here |
| 06:31.10 | sweeper | if it were just hardware, I'd move back to peru, the cost of living balances the increased cost of parts :D |
| 06:31.13 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 06:32.21 | |R | hehe, macchu picchu, here i come! |
| 06:32.22 | JT | what about the cost of frustration from living with americans? ;) |
| 06:32.53 | sweeper | pfft |
| 06:33.01 | sweeper | I don't leave my house if I can avoid it ;) |
| 06:33.01 | |R | at least you get to watch funny, biased michael moore movies in balance of the not funny, really biased politic ;) |
| 06:33.13 | sweeper | gah michael moore |
| 06:33.24 | sweeper | his pedantry is almost as boring as cnn~ |
| 06:33.27 | |R | i love how he always come to canada and i'm always learning that i live in paradise |
| 06:33.28 | |R | haha |
| 06:33.41 | JT | we have a good comedy channel here in australia on our cable networks |
| 06:33.48 | JT | it's called Fox News |
| 06:33.51 | |R | our door are all unlocked, our hospital are all miraculous, etc :P |
| 06:33.54 | JT | it's hilarious |
| 06:34.01 | |R | hehe yeah, damn murdoch |
| 06:34.20 | |R | but after seeing outfoxed, it kind of concluded on the whole genre :P |
| 06:34.31 | JT | we had a series here for a while that satarised fox news and others |
| 06:34.47 | JT | the show's now evolved into a less new format, but it's funnier than ever |
| 06:34.55 | |R | hehe :) |
| 06:34.56 | JT | s/new/news/ |
| 06:35.20 | |R | haha yeah, regexp in a bot :) |
| 06:35.25 | JT | chasing around polititians and celebrities and playing pranks on them/harrassing them |
| 06:35.26 | |R | haven't seen that before ;) |
| 06:35.28 | JT | oh, and americans |
| 06:35.49 | JT | the show has a guy stationed in the USA just so we can laugh at americans |
| 06:36.01 | |R | we had a show, talking to americans, where they went to university campuses and asked questions |
| 06:36.16 | |R | like, do you think america should share their harbors with canada as we don't have any access to the see? |
| 06:36.18 | JT | yeah, they go around the streets and ask questions |
| 06:36.22 | |R | or that we live in igloos, etc... |
| 06:36.34 | |R | most people, including teachers, were buying into it heh ;) |
| 06:36.51 | JT | once he did a faux michael moore, and beat moore's record of getting kicked out of corporate HQs |
| 06:36.53 | |R | s/see/sea |
| 06:36.54 | JT | hah |
| 06:38.05 | sweeper | ok, so netcat doesn't produce any output in the asterisk console, and it dies right after I hit enter |
| 06:38.14 | sweeper | I can't even cat anything into it |
| 06:39.15 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 06:40.17 | sweeper | and that's from localhost |
| 06:41.05 | sergee | did anyone try to use something like Sony PCS-G50P with asterisk? |
| 06:41.56 | sweeper | ohoh |
| 06:42.02 | sweeper | now I got something |
| 06:42.33 | sweeper | protip: don't use localhost, use the actual IP address |
| 06:43.03 | DarKnesS_WolF | i'm getting sip auth faild any idea how can i know the pass that phone trying to send ? |
| 06:43.58 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 06:45.18 | *** join/#asterisk troy- (n=troy@CPE00907f17e478-CM00186845db94.cpe.net.cable.rogers.com) |
| 06:45.33 | troy- | i am getting the error "no private structure for packet" on console |
| 06:45.35 | troy- | any ideas? |
| 06:45.52 | sweeper | DarKnesS_WolF: tcpdump |
| 06:45.58 | sweeper | mmm |
| 06:46.05 | DarKnesS_WolF | sweeper: another way ? |
| 06:46.10 | sweeper | wireshark? |
| 06:46.18 | sweeper | look at the phone's config? |
| 06:46.28 | troy- | damned console is filling up |
| 06:46.47 | sweeper | set verbose 0 |
| 06:46.59 | troy- | i'd rather fix the problem instead of sweeping it under the rug |
| 06:49.38 | troy- | can anoyone help? |
| 06:51.30 | JT | sergee: what is that? |
| 06:52.01 | DarKnesS_WolF | sweeper: the idea is the phone is too far away from my side and i can't reach it |
| 06:52.03 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 06:52.28 | sergee | JT: it's a kind of videoconferencing, but it seems to support only h.323, so i suppose it won't work with asterisk... |
| 06:52.50 | JT | hmm |
| 06:53.11 | sergee | JT: although right codecs are in plase (h.263, h.264) |
| 07:01.37 | FuriousGeorge | how do i check what version of zaptel im currently running? |
| 07:01.48 | FuriousGeorge | Revision: 62095 |
| 07:01.58 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 07:02.00 | Chris-NB | hi |
| 07:02.01 | FuriousGeorge | i was looking for more of a 1.4.X |
| 07:03.37 | Chris-NB | anyone discovered that behavior: I'v two asterisk boxes. On both I've a user 101. If I call from box 1 to box 2, box 2 want the caller to authenticate. but that's a incoming call from an external box, there shouldn't be a authentication |
| 07:04.02 | Chris-NB | same behavior with a SER and asterisk, when I call into asterisk from SER with user 101. |
| 07:04.12 | Chris-NB | anyone know how to fix or prevent this? |
| 07:05.03 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:06.34 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 07:10.47 | *** join/#asterisk tsurko (n=tsurko@150-190.go.evo.bg) |
| 07:11.17 | *** join/#asterisk oej (n=olle@62.97.243.70) |
| 07:11.57 | Chris-NB | nobody had that problem? |
| 07:13.23 | *** join/#asterisk purplet (n=purplet@010.041.dsl.concepts.nl) |
| 07:15.53 | sweeper | Chris-NB: uh,have you set up registration, or sip peers for the boxes? |
| 07:17.11 | Chris-NB | sweeper, these two boxes are independent boxes. both have their own users (from sip.conf). If a call from box 1 is made to box 2 with a username wich exists on both boxes, the 2nd want to authenticate user from box 1 |
| 07:17.34 | Chris-NB | sweeper, which should not happen, cause it's an external call and the call should be placed in the default context |
| 07:19.13 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 07:20.53 | sweeper | asdf |
| 07:20.55 | sweeper | dude |
| 07:21.07 | sweeper | you've got to configure that stuff |
| 07:21.31 | sweeper | either allow anonymous calls (bad) or set up sip peers |
| 07:23.44 | Chris-NB | ??? |
| 07:24.01 | Chris-NB | why should I want to setup sip peers on my box, which come from another box? |
| 07:24.02 | *** join/#asterisk _m_ (i=mNw@213.203.226.184) |
| 07:25.21 | _m_ | hi. are there any pointers to a description of the protocol that is spoken on port 2088 on some phones and that is used for BLF? |
| 07:25.23 | *** join/#asterisk grEvenX (n=even@ti500720a080-8073.bb.online.no) |
| 07:27.19 | sweeper | Chris-NB: um, so random sip clients don't spam the hell out of your poor users? |
| 07:27.27 | sweeper | you don't set up peers for each user |
| 07:27.35 | sweeper | you set up peers for the box itself |
| 07:27.41 | sweeper | well, a peer per box |
| 07:28.14 | Chris-NB | sweeper, so you want me to setup a peer for every box where probably someone wanna call me?!? |
| 07:28.16 | *** join/#asterisk zdrulio (n=krlozano@82.119.72.130) |
| 07:28.25 | zdrulio | heelo |
| 07:29.18 | zdrulio | i want to record calls, but i don`t know how |
| 07:29.49 | Chris-NB | zdrulio, show application Monitor |
| 07:30.13 | |R | ciao :) |
| 07:30.25 | JT | zdrulio: do you come in and ask that question every day? |
| 07:30.48 | sweeper | Chris-NB: well, then tell your box to allow unauthenticated calls |
| 07:31.20 | Chris-NB | sweeper, how do I do that? |
| 07:32.02 | zdrulio | JT no :) only yestarday and today |
| 07:32.21 | JT | zdrulio: so you were answered yesterday... |
| 07:32.24 | zdrulio | but yestarday i close the window and .... no logs |
| 07:32.30 | *** join/#asterisk Pilko (n=pirch@213.80.169.119) |
| 07:32.37 | zdrulio | sry |
| 07:32.45 | JT | it's easy stuff you can find out by checking out the wiki or the book |
| 07:32.47 | sweeper | Chris-NB: i don't ever do it, but I'm gonna guess that allowguest might do it |
| 07:33.03 | sweeper | zdrulio: or google ;) |
| 07:33.24 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 07:33.37 | sweeper | http://www.google.com/search?q=asterisk+record+calls <-- for example |
| 07:34.43 | Zeeek | hay |
| 07:35.39 | *** join/#asterisk vn (n=nostalge@bas5-quebec14-1128557048.dsl.bell.ca) |
| 07:35.49 | vn | beware, I'm back |
| 07:35.56 | Zeeek | now |
| 07:37.02 | *** join/#asterisk obnauticus (n=obnautic@c-71-59-162-60.hsd1.wa.comcast.net) |
| 07:38.41 | Zeeek | anyone here in Paris? |
| 07:38.48 | Zeeek | or France? |
| 07:39.12 | Zeeek | Western Europe? |
| 07:40.12 | *** join/#asterisk ispireuk (n=ISPIRENL@82-204-26-196.dsl.bbeyond.nl) |
| 07:41.08 | sweeper | all signs point to no |
| 07:41.16 | sweeper | although it's probably a better time of day there |
| 07:41.47 | ispireuk | Hello people |
| 07:42.03 | ispireuk | I have made an IVR, but somehow it doesn't work |
| 07:42.16 | ispireuk | I have put the script and error here: http://pastebin.ca/593266 |
| 07:42.26 | ispireuk | Can someone tell me what I am doing wrong? |
| 07:43.06 | sweeper | I don't see an error there |
| 07:43.25 | berktr | is it possible to make the pap2 sound different when the other party is talking to another person? |
| 07:43.39 | Chris-NB | Zeeek, I'm from Austria, if that helps |
| 07:44.09 | sweeper | ispireuk: you reach the end of the 's' extnesion, and it auto-falls through. what do you WANT it to do? |
| 07:44.17 | ispireuk | The error is on line 18 |
| 07:44.29 | ispireuk | The caller simply gets disconnected |
| 07:44.43 | ispireuk | No chance to select an option in the menu |
| 07:45.12 | sweeper | line 18 is blank.... |
| 07:45.33 | ispireuk | I mean line 18 of what asterisk says |
| 07:45.48 | sweeper | the last log line? |
| 07:45.58 | ispireuk | Sorry, I mean 38 |
| 07:46.02 | Zeeek | Chris-NB just wondering :) |
| 07:46.29 | sweeper | ok, well the problem is you should maybe have a wait in there :) |
| 07:47.26 | Chris-NB | Zeeek, ok |
| 07:47.41 | sweeper | specifically, a WaitExten |
| 07:47.42 | ispireuk | I also tried with WaitExten, then it works, except that the caller will have to listen to the whole menu. I don' t want that. I want a caller to be able to make a choice anytime |
| 07:48.02 | sweeper | oh |
| 07:48.09 | sweeper | I think there's an option you can pass to playback |
| 07:48.18 | ispireuk | So I checked the examples on internet, they all use it this way, without waitexten |
| 07:48.53 | ispireuk | Wait. You just bring me an idea |
| 07:49.07 | ispireuk | is that the difference between playback and background perhaps? |
| 07:49.22 | Zeeek | by the way, I think you can use s/0108500574 instead of testing for CID numbers |
| 07:49.42 | sweeper | ispireuk: aha! |
| 07:49.47 | sweeper | yea, I just got to that |
| 07:49.57 | ispireuk | :) Let me try that |
| 07:50.07 | sweeper | this is good, I'm gonna need to do this as well soon |
| 07:50.11 | Zeeek | playback isn't listening |
| 07:50.16 | ispireuk | Oh, I didn't know that zeeek. |
| 07:50.23 | Zeeek | background listens |
| 07:52.14 | ispireuk | Yes, that's it |
| 07:52.23 | ispireuk | Now it works, I put a waitexten now and changed playback to background |
| 07:52.28 | ispireuk | That does the trick |
| 07:54.26 | ispireuk | Thanks for your help! |
| 07:55.16 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 07:56.56 | vn | which one of those is better as in more recent, more configurable and working properly? : digium iaxy, linksys pap2t-na |
| 07:57.11 | Pilko | ispireuk, look at READ, it may be useful too. simetimes it's more suitable instead of BACKGROUND |
| 07:58.03 | *** join/#asterisk matsk (n=mk@194.68.102.173) |
| 08:00.40 | Zeeek | does the linksys do IAX? |
| 08:00.52 | Zeeek | otherwise, they're not really comparable |
| 08:03.38 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 08:05.26 | *** join/#asterisk aao_pwner (n=obnautic@c-71-59-162-60.hsd1.wa.comcast.net) |
| 08:08.11 | *** join/#asterisk BugKhaM (n=LAMER@ppp-58.8.17.25.revip2.asianet.co.th) |
| 08:08.38 | *** join/#asterisk dikdust (n=dikdust@2001:1418:1c7:0:211:11ff:fed3:ecc7) |
| 08:09.06 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 08:09.08 | BugKhaM | anyone knows of the website that describe how to make a cable linking between E1 boards? |
| 08:10.42 | BugKhaM | I have Dialogic D/300 and Digium E100P |
| 08:11.04 | BugKhaM | and wanna hook them up |
| 08:11.10 | vn | uh yeah true the linksys isn't IAX.. |
| 08:11.36 | vn | but anyway I just need the ATA to be SIP, I'll use * to convert it to IAX |
| 08:13.38 | *** join/#asterisk O_Zone (n=O-Zone@moloch.asb.unisi.it) |
| 08:13.39 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 08:13.42 | O_Zone | hi all |
| 08:15.24 | *** part/#asterisk O_Zone (n=O-Zone@moloch.asb.unisi.it) |
| 08:16.54 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 08:17.29 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:17.44 | *** join/#asterisk obnauticus (n=obnautic@c-71-59-162-60.hsd1.wa.comcast.net) |
| 08:17.54 | *** join/#asterisk RyanW (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 08:18.11 | RyanW | Hello, has anyone used the UTstarcom F3000? |
| 08:19.27 | JT | BugKhaM: i hope the Dialogic is not on Asterisk |
| 08:21.10 | BugKhaM | JT: nope |
| 08:21.48 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 08:21.55 | BugKhaM | JT: will the ethernet cross over cable work? |
| 08:22.01 | JT | no. |
| 08:22.06 | JT | T1 crossover cable |
| 08:22.33 | BugKhaM | JT: thought they are the same =), as they are using RJ-45 |
| 08:22.45 | Strom_C | BugKhaM: the pinout is different |
| 08:22.49 | JT | it's not actually an RJ-45 connector to be technical |
| 08:22.50 | Zeeek | anyone have any asterisk-related video? |
| 08:22.53 | Strom_C | and if you want to be technical about it, neigher is RJ-45 |
| 08:22.55 | *** join/#asterisk aao_pwner (n=obnautic@c-71-59-162-60.hsd1.wa.comcast.net) |
| 08:22.55 | JT | it's an 8P8C connector |
| 08:23.06 | Strom_C | ok fine JT, just steal my thunder why don't you :) |
| 08:23.31 | JT | RJ-45 describes that connector in one configuration |
| 08:23.31 | JT | ;) |
| 08:23.32 | *** join/#asterisk shay|work (n=shay@unaffiliated/shay) |
| 08:23.57 | BugKhaM | Strom_C: hmm, but I can use the RJ-45 Jack and CAT5 cable for this right? |
| 08:24.09 | JT | yes |
| 08:24.13 | JT | but different pinout |
| 08:24.24 | JT | it's an 8P8C jack is what we're getting at |
| 08:24.29 | BugKhaM | JT: u know the pin config? so, I can make one |
| 08:24.29 | JT | google for t1 crossover cable |
| 08:24.40 | JT | tonnes of docs on the web |
| 08:24.42 | Strom_C | BugKhaM: swap pairs 1 and 3 |
| 08:24.56 | Strom_C | and you've got your T1 crossover |
| 08:25.00 | BugKhaM | JT: hmm, it will work for E1 also right? |
| 08:25.03 | JT | anyway, T1/E1 is an RJ-48C jack |
| 08:25.04 | JT | yes |
| 08:25.13 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 08:25.40 | BugKhaM | JT: ok, will try googling for it |
| 08:25.53 | JT | the first five billion results will do |
| 08:25.55 | Strom_C | BugKhaM: i just told you how to make one :) |
| 08:26.11 | JT | Strom_C: you assume he knows the pair numbers :) |
| 08:26.15 | JT | pin numbers are easier |
| 08:26.20 | Strom_C | blue orange green brown slate |
| 08:26.24 | Strom_C | 1 2 3 4 5 |
| 08:26.36 | *** join/#asterisk CVirus (n=GoD@82.201.222.181) |
| 08:26.52 | Strom_C | maybe i've just been working in telephony too long and assume everyone can count by AT&T 25-pair color code |
| 08:26.54 | JT | 1 to 4, 2 to 5 |
| 08:27.01 | JT | iirc |
| 08:27.01 | JT | heh |
| 08:28.13 | negativeduck | heh unless your the people who wired my house who only understood the logic of wire on this end must match wire on other end... with total dis-regard to color. For that matter every other cat5 run used different colors for different pairs. |
| 08:28.17 | negativeduck | most annoying. |
| 08:28.39 | Strom_C | negativeduck: i hope you got your money back |
| 08:29.50 | negativeduck | Got a reasonable credit aftrer I found that.... but it really helped after I connected a HUCK and found half the wires were bad.. which later lead to cat 5 runs that were smashed and bent hard over 180 and creesed :) |
| 08:30.01 | Strom_C | let me guess |
| 08:30.04 | Strom_C | electricians |
| 08:30.17 | Strom_C | or "home theater specialists" |
| 08:30.43 | negativeduck | nah Guardian Home Security who was the only person the builder would let pull the LV wires. |
| 08:31.25 | Strom_C | did you tell the builder that they are massively incompetent? |
| 08:32.00 | *** join/#asterisk ghenry (n=ghenry@212.159.59.85) |
| 08:32.58 | negativeduck | dude, I knew it was going to bad when they said "We use cat5 for everything" then told me a "phone jack" was $60 to pull and an ethernet jack was $100. ofcourse I'm thinking . o O(but only the end is different) |
| 08:33.20 | negativeduck | So I only paid for "phone" connections day after I settled came in and replaced em all. |
| 08:33.26 | negativeduck | which is where all the fun started. |
| 08:33.38 | Strom_C | well, cat5 for voice isn't such a bad idea |
| 08:33.46 | Strom_C | you get less crosstalk on adjacent circuits |
| 08:34.23 | negativeduck | Oh, no doubt... I just had no use for phone jacks everywhere ... ethernet yes, phone no. |
| 08:34.46 | negativeduck | I just couldn't believe that they had such a major price difference between the two connectors. |
| 08:34.59 | Strom_C | considering they're like $1 difference in price at Graybar |
| 08:35.17 | negativeduck | maybe "ethernet" means more quality control. |
| 08:35.19 | negativeduck | but I doubt it. |
| 08:35.20 | negativeduck | :P |
| 08:39.56 | negativeduck | my god, it's muggy as hell outside. |
| 08:40.02 | Strom_C | where's "outside"? |
| 08:40.27 | negativeduck | it's that big place beyond the portal that people use for entering and exiting the "inside" |
| 08:40.38 | negativeduck | :P Norther Virginia |
| 08:40.42 | Strom_C | ah |
| 08:40.49 | negativeduck | s/Norther/Northern/ |
| 08:40.53 | Strom_C | this is why I live in Los Angeles |
| 08:40.57 | Strom_C | it's pleasant outside :) |
| 08:42.02 | negativeduck | I don't think I've ever been to LA, we used to have service there but mostly I've only been to san jose and sanfran. |
| 08:42.38 | Strom_C | yeah, that's another universe |
| 08:42.41 | Strom_C | los angeles is great |
| 08:43.37 | negativeduck | most of the people I know who live there tend to agree with that. |
| 08:44.03 | BugKhaM | Strom_C: I noticed that the E1 cable from my telco uses only 2 cores/wires connection |
| 08:44.19 | BugKhaM | Strom_C: will there be any problem with that? |
| 08:44.20 | Strom_C | BugKhaM: the telco wiring is probably HDSL |
| 08:44.20 | negativeduck | course everyone I know in NY 99% of them say it's great... but it's a place I would never want to live. |
| 08:44.28 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 08:44.42 | BugKhaM | Strom_C: ok |
| 08:44.45 | Strom_C | BugKhaM: it terminates in a smartjack with blinky lights, right? |
| 08:47.13 | *** join/#asterisk penguinFunk_ (n=penguin@unaffiliated/penguinfunk) |
| 08:48.19 | BugKhaM | Strom_C: it's written "SHDSL" and "G.703" |
| 08:48.29 | *** join/#asterisk waptaxi (n=waptaxi@45.151-224-87.telenet.ru) |
| 08:48.47 | Strom_C | BugKhaM: right |
| 08:49.05 | Strom_C | but...is there a piece of equipment that it terminates on, or is it just wire directly into a jack? |
| 08:50.03 | BugKhaM | Strom_C: yeah, there is. it's like a modem =) |
| 08:50.10 | Strom_C | ok |
| 08:50.29 | Strom_C | the 8P8C E1 jack they're providing you is traditional 4-wire E1 |
| 08:50.36 | BugKhaM | Strom_C: I am to link from E100P(Digium) to D/300 (Dialogic) |
| 08:50.55 | Strom_C | E100P? that's an ancient card |
| 08:51.35 | BugKhaM | Strom_C: yeah, thats a spare one and I have the 4-Port Tormenta here too |
| 08:51.47 | BugKhaM | Strom_C: which one is better =) |
| 08:51.58 | Strom_C | "Tormenta"? how old is this stuff? |
| 08:52.29 | BugKhaM | Strom_C: just got it from E-bay it's tormenta 2 probably 3-4 years old |
| 08:52.50 | Strom_C | jeez |
| 08:53.04 | Strom_C | there are drivers but I don't know if they're maintained |
| 08:53.34 | Strom_C | I hope you got a screaming deal on them |
| 08:53.49 | BugKhaM | Strom_C: i'm using TDM400P for production though |
| 08:54.19 | BugKhaM | Strom_C: sorry TE110P |
| 08:54.35 | Strom_C | ok, TE110P is better :) |
| 08:54.40 | Strom_C | i've got a couple of those |
| 08:55.39 | BugKhaM | Strom_C: I am wondering if I am to link from E100P to the other Dialogic card, should I be concerned about the signalling? |
| 08:56.03 | Strom_C | uh |
| 08:56.05 | Strom_C | of course |
| 08:56.11 | BugKhaM | Strom_C: I will be using Dial(ZAP/.. to connect them |
| 08:56.16 | Strom_C | it's not magic, you know |
| 08:56.25 | penguinFunk | it isn't? |
| 08:56.31 | penguinFunk | :P |
| 08:56.38 | BugKhaM | Strom_C: hmm, the other side is configured for R2MFC |
| 08:57.02 | BugKhaM | Strom_C: so, I will need coppice's stuff? |
| 08:57.04 | *** join/#asterisk yassaccan (n=yassacca@admin240.hgo.se) |
| 08:57.10 | Strom_C | beats me |
| 08:57.17 | Strom_C | i've never worked with R2MFC |
| 08:57.25 | Strom_C | ISDN makes life easy |
| 08:57.39 | BugKhaM | Strom_C: =) |
| 08:58.40 | tzafrir | BugKhaM, I guess so |
| 08:59.50 | BugKhaM | tzafrir: or change the config on the other side to ISDN |
| 09:00.07 | Strom_C | if you can do ISDN, do it |
| 09:00.40 | *** join/#asterisk oej (n=olle@62.97.243.70) |
| 09:00.47 | BugKhaM | Strom_C: never tried but i just saw that it's got ISDN driver also |
| 09:02.44 | *** join/#asterisk Sebb (n=sebastia@einstein.f0o.de) |
| 09:02.47 | Sebb | hi |
| 09:03.32 | Sebb | i have a question concerning "putting a call on hold" via sip. normally, asterisk just starts the moh then.. is it possible to just relay this "hold signal" via sip, without playing moh then? |
| 09:10.57 | JT | isn't san jose in LA? |
| 09:12.47 | sweeper | no >.> |
| 09:12.56 | sweeper | they're both cities in california |
| 09:13.28 | JT | uh duh |
| 09:13.33 | JT | the state is CA |
| 09:13.33 | JT | :) |
| 09:13.35 | JT | right |
| 09:13.41 | sweeper | gg |
| 09:14.05 | creativx | gg geography |
| 09:16.38 | negativeduck | :) |
| 09:16.57 | negativeduck | that being said hi ho hi ho it's off to the office I go. |
| 09:17.37 | JT | creativx: a simple mixup ;) |
| 09:17.44 | JT | i don't live in the us anyway |
| 09:17.46 | sweeper | bryan is either a) late to work b) really early to work or c) in iceland |
| 09:17.56 | JT | i'd like to see an american try their hand at australian geography |
| 09:18.12 | sweeper | I know where sydney and perth and stuff are |
| 09:18.28 | sweeper | I used to know all the provinces, but that was in highschool |
| 09:18.31 | JT | and what states they are in? |
| 09:18.40 | JT | we don't have provinces >:) |
| 09:18.45 | sweeper | whatever |
| 09:18.54 | sweeper | I always associate AU with CN |
| 09:19.17 | sweeper | I know they're cities and not states! |
| 09:19.57 | sweeper | and tazmania is off the bottom right coast~ |
| 09:20.55 | JT | heh |
| 09:21.06 | JT | tasmania, no z-isms ;) |
| 09:21.49 | sweeper | ok, that was definately amerikka's fault >.> |
| 09:22.46 | sweeper | man, there are ROOSTERS here. won't shut up either :/ |
| 09:22.54 | JT | nice |
| 09:23.18 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 09:31.41 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
| 09:31.51 | awk | what driver does this card use TDM400P? |
| 09:33.19 | *** join/#asterisk Ebola (n=Ebola@host86-136-134-202.range86-136.btcentralplus.com) |
| 09:33.31 | Strom_M | wctdm |
| 09:33.45 | awk | ok thanks |
| 09:33.58 | *** join/#asterisk ispireuk (n=ISPIRENL@82-204-26-196.dsl.bbeyond.nl) |
| 09:34.04 | ispireuk | Hello |
| 09:34.15 | *** part/#asterisk zdrulio (n=krlozano@82.119.72.130) |
| 09:34.28 | ispireuk | Has the Page function been replaced or removed in asterisk 1.4? |
| 09:34.28 | Strom_M | hellol |
| 09:34.45 | ispireuk | I get an error when I try to use it |
| 09:34.49 | Strom_M | ispireuk: i dunno, what does it say in UPGRADE.txt? |
| 09:35.09 | ispireuk | [Jun 28 11:29:42] WARNING[2739]: pbx.c:1797 pbx_extension_helper: No application 'Page' for extension (ispire-nl, 7777, 2) |
| 09:35.56 | ispireuk | Nothing about the page command there |
| 09:36.17 | Strom_M | hang on |
| 09:36.24 | ispireuk | ok tnx |
| 09:37.06 | Strom_M | it's there in my copy of 1.4 |
| 09:37.09 | *** join/#asterisk pfn (n=pfnguyen@hanhuy.com) |
| 09:37.11 | Strom_M | did you maybe not compile it in? |
| 09:38.58 | *** join/#asterisk CVirus (n=GoD@82.201.222.181) |
| 09:39.41 | ispireuk | hmmm |
| 09:39.47 | ispireuk | How should I compile it in then? |
| 09:41.01 | *** join/#asterisk juice (n=juice@mo-76-2-162-204.dhcp.embarqhsd.net) [NETSPLIT VICTIM] |
| 09:42.38 | Strom_M | is it selected when you run "make menuselect"? |
| 09:42.45 | ispireuk | ahhh |
| 09:42.48 | ispireuk | I never did that |
| 09:42.56 | ispireuk | I will recompile then |
| 09:44.02 | Winkie | hey guys, i'm having an issue with queues |
| 09:44.13 | Winkie | if we have say 8 calls enter a queue at the same time, the call distribution screws it |
| 09:44.15 | Winkie | screws up* |
| 09:44.23 | Winkie | it calls the first agent twice, and then fails to progress down the queue it seems |
| 09:44.38 | *** join/#asterisk Polis_ttt (n=your@194-237-172-225-no48.business.telia.com) |
| 09:44.41 | Winkie | ringbusy never features because it seems to mis-lock |
| 09:44.45 | Winkie | has anyone ever come across this before? |
| 09:45.04 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) [NETSPLIT VICTIM] |
| 09:45.21 | ispireuk | can't select that, seems that zaptel is not installed |
| 09:45.32 | Strom_M | ispireuk: well there you go |
| 09:45.35 | ispireuk | Should I download and compile zaptel seperately for asterisk 1.4? |
| 09:45.40 | Strom_M | ispireuk: yes |
| 09:45.52 | ispireuk | That explains a lot |
| 09:45.58 | Strom_M | Winkie: don't use agents; use dynamic members |
| 09:46.10 | *** join/#asterisk SuD (n=Ask@89.140.32.2.static.user.ono.com) |
| 09:46.13 | Uatec | i have a dynamic member |
| 09:46.44 | lilalinux | I'm looking for a mISDN installation howto for debian/etch (using mostly packaged versions) |
| 09:46.45 | Strom_M | good for you |
| 09:47.27 | SuD | hi, is it possible that in 1.4.5 whenever i do a "/etc/init.d/asterisk reload" it misconfigures itself? i'm talking about zaptel answeronpolaritydelay, hanguponpolarityswitch and busydetect |
| 09:48.05 | SuD | those values go default values instead of the values in the config files |
| 09:48.45 | Strom_M | why are you doing a full reload? |
| 09:49.20 | *** join/#asterisk rajiv (n=rajiv@gentoo/developer/rajiv) [NETSPLIT VICTIM] |
| 09:49.21 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-219-116.hsd1.al.comcast.net) [NETSPLIT VICTIM] |
| 09:49.21 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) [NETSPLIT VICTIM] |
| 09:49.49 | SuD | Strom_M: sorry, what other kind of reloads are there, and shouldn't i consider it as a bug anyway (when i can enter bugs.digium.com) ? |
| 09:50.15 | Strom_M | SuD: well, what are you doing that prompts you reload? |
| 09:50.29 | SuD | typically changes in extensions.conf and voicemail.conf |
| 09:50.49 | Strom_M | at the asterisk CLI, issue "extensions reload" and "reload app_voicemail.so" |
| 09:50.55 | ispireuk | svn checkout http://svn.digium.com/svn/zaptel/1.4/branch zaptel-1.4 |
| 09:51.00 | ispireuk | What am I doing wrong? |
| 09:51.24 | Strom_M | svn checkout http://svn.digium.com/svn/zaptel/branches/1.4 zaptel-1.4 |
| 09:52.48 | ispireuk | ok tnx |
| 09:53.26 | *** join/#asterisk Swat2 (n=bler@218-215-199-11.people.net.au) |
| 09:54.03 | Swat2 | Anyone here got a working zaptel.conf for a TDM400P card. im told: http://paste.linux-vserver.org/2853 is incorrect. |
| 09:54.46 | ispireuk | I also remember that for zaptel that I had to load the ztdummy module into the kernel, how can I make a kernel module load everytime with booting? |
| 09:55.22 | ispireuk | Or that is not needed anymore with 1.4? |
| 09:55.35 | Strom_M | Swat2: use another pastebin next time; that font is fucking terrible on the eyes |
| 09:55.57 | Strom_M | www.pastebin.ca :) |
| 09:55.59 | Swat2 | appologies. |
| 09:56.10 | Swat2 | noted. |
| 09:56.16 | Winkie | Strom_M: i'm using dynamic members :) |
| 09:56.22 | Winkie | not Local/ channels either anymore |
| 09:56.26 | Strom_M | Swat2: and that looks fine to me |
| 09:56.35 | Strom_M | i assume you have four FXO modules? |
| 09:56.46 | Strom_M | and ks-type phone lines from the telco? |
| 09:56.49 | Swat2 | yeah pstn interfacing modules |
| 09:57.07 | Strom_M | Swat2: then that looks fine |
| 09:57.11 | Strom_M | does ztcfg complain? |
| 09:57.47 | Swat2 | nope |
| 09:58.00 | awk | you said ztcfg comaplains |
| 09:58.00 | Swat2 | but im getting nasty: ZT_CHANCONFIG failed on channel 1: Inappropriate ioctl for device |
| 09:58.02 | Swat2 | errors |
| 09:58.14 | awk | then its complaining |
| 09:58.18 | awk | do a strace on ztcfg |
| 09:58.21 | Swat2 | thats on startup |
| 09:58.37 | Swat2 | when i run /etc/init.d/zaptel start |
| 09:59.02 | Strom_M | .... |
| 09:59.05 | awk | Swat2 *sigh* told you! YOU DONT NEED TO run /etc/init.d/zaptel start you allready loaded the modules by hand with using modprobe, or insmod.. |
| 09:59.08 | Swat2 | just dont understand how it can be wrong awk when it works perfectly on * 1.2 |
| 09:59.41 | *** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62) |
| 09:59.42 | awk | dont start zaptel its started lsmod | grep zaptel |
| 09:59.43 | awk | its runing |
| 09:59.58 | Swat2 | awk, i stopped it and removed the modules |
| 10:00.02 | Swat2 | then did the above command |
| 10:00.03 | awk | so that message you see isn't a issue, that zaptel startup script is ugly! it loads a whole load of module su dont need |
| 10:00.22 | awk | modprobe zaptel wctdm |
| 10:00.24 | awk | ztcfg -vv |
| 10:00.25 | awk | vwala |
| 10:00.48 | _m_ | Hi. Are there any pointers to a description of the protocol that is spoken on port 2088 on some phones and that is used for BLF? |
| 10:00.51 | Strom_M | technically you just need to modprobe wctdm, and zaptel will load automagically |
| 10:00.52 | Strom_M | voila |
| 10:01.04 | JT | awk: actually, even that is overkill. |
| 10:01.06 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:01.12 | JT | awk: modprobe wctdm |
| 10:01.15 | JT | ztcfg -vv |
| 10:01.20 | awk | aggg |
| 10:01.23 | awk | what ever |
| 10:01.24 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-75-84-232-194.socal.res.rr.com) |
| 10:01.26 | Strom_M | JT: hah, now I've stolen your thunder!! |
| 10:01.28 | Strom_M | muahha |
| 10:01.33 | JT | :( |
| 10:01.38 | Swat2 | lol |
| 10:01.41 | JT | awk: well, you wanted to be technical on efficiency |
| 10:01.44 | JT | :P |
| 10:01.44 | awk | please dont tell me how to load kernel modules |
| 10:01.57 | JT | awk: why not, when you specify redundant steps to people? |
| 10:02.07 | Strom_M | awk: here's how to load kernel modules |
| 10:02.10 | Strom_M | step 1: |
| 10:02.12 | awk | JT because earlier he stated when I told him to do it that way |
| 10:02.19 | awk | it didn't bring up zaptel |
| 10:02.31 | awk | so understanding his problem I gave another workaround |
| 10:02.45 | Strom_M | this is the jizziest channel on freenode |
| 10:04.42 | Swat2 | modprobbe wctdm gives: line 0: Unable to open master device '/dev/zap/ctl' |
| 10:04.59 | Strom_M | you are root, right? |
| 10:05.04 | Swat2 | yep |
| 10:05.07 | Strom_M | cocks |
| 10:05.23 | SuD | Strom_M: thank you for your help |
| 10:05.36 | Strom_M | welcome |
| 10:06.59 | ispireuk | How can I load the ztdummy module? |
| 10:07.44 | Sebb | with modprobe, like any other module? |
| 10:08.56 | ispireuk | strange |
| 10:09.03 | ispireuk | That doesn't work |
| 10:09.11 | ispireuk | Just modprobe ztdummy? |
| 10:09.12 | Strom_M | did you compile it? |
| 10:09.14 | Sebb | type "dmesg" |
| 10:09.16 | ispireuk | Yes I did |
| 10:09.19 | Sebb | what error did you get? |
| 10:12.34 | Strom_M | that was some error |
| 10:27.00 | *** join/#asterisk Ch0Hag (n=mking@knight.monnsta.net) |
| 10:27.12 | Ch0Hag | Can I get * to timestamp everything it prints on the console? |
| 10:27.34 | creativx | yes |
| 10:27.35 | creativx | -T |
| 10:27.42 | creativx | asterisk -rT |
| 10:28.33 | Ch0Hag | Nifty. |
| 10:28.47 | creativx | indeed is. |
| 10:28.51 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 10:29.15 | tzafrir | Swat2, for starters, remove the automatic run of ztcfg as a post-install action from modprobe.conf or /etc/modprobe.d/zaptel |
| 10:29.25 | tzafrir | The error you get is from ztcfg |
| 10:29.38 | Swat2 | tzafrir: ok |
| 10:29.39 | tzafrir | not from the module load itself |
| 10:30.05 | tzafrir | now, do you have /dev/zap/1 ? Does it have lines for your span? |
| 10:30.20 | Swat2 | yep |
| 10:30.21 | Swat2 | 1 2 3 4 |
| 10:30.25 | Ch0Hag | Less nifty. |
| 10:30.31 | Ch0Hag | It doesn't timestamp anything. |
| 10:30.44 | Ch0Hag | Well - except the introductory copyright message, which thus spills over 80 colums. |
| 10:30.55 | creativx | lies |
| 10:31.12 | creativx | start asterisk with -T |
| 10:31.16 | creativx | no wait a second |
| 10:31.17 | creativx | :) |
| 10:32.05 | Ch0Hag | Ah see that says 'to all non-command related output' |
| 10:33.09 | creativx | yes |
| 10:33.15 | creativx | try dial <ext> and see |
| 10:33.57 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 10:34.22 | *** join/#asterisk kfullert (n=kfullert@cpc3-hatf1-0-0-cust986.lutn.cable.ntl.com) |
| 10:37.32 | *** join/#asterisk linex (n=Blahme@124.82.18.119) |
| 10:37.49 | linex | is there a livecd version of asterisk ? |
| 10:38.17 | FuriousGeorge | there are a couple |
| 10:38.36 | FuriousGeorge | digium puts one out, and there is a@h |
| 10:38.59 | Strom_M | asterisknow is decent; trixbox blows goats anally |
| 10:39.20 | Swat2 | hmmm |
| 10:39.24 | Swat2 | now incomming calls wont work |
| 10:40.35 | linex | blows goats anally <-- not sure what that means. I'm gonna try asterisk now |
| 10:40.49 | Strom_M | linex: it means it's terrible |
| 10:41.04 | linex | Strom_M: ok |
| 10:41.51 | Strom_M | my colorful metaphors fall on deaf ears :/ |
| 10:42.01 | linex | So can I use a software phone for testing purposes. I don't have a soft phone. |
| 10:42.18 | Strom_M | what do you think a softphone is? |
| 10:42.30 | linex | Strom_M: its nice. I half understood it. Just wanna be sure. |
| 10:43.10 | yonahw-work | Strom_M: I certainly appreciated the "blows goats anally" but that just might be a result of growing up in LA |
| 10:43.17 | linex | oh sorry. What do you call the phones with all the new features. |
| 10:43.32 | linex | that understand sip |
| 10:43.43 | linex | feature phones I think |
| 10:43.44 | Strom_M | linex: you mean desk phones? |
| 10:43.53 | Strom_M | usually those are just SIP phones |
| 10:43.54 | linex | ah yes desk phones |
| 10:44.03 | Strom_M | softphone == software |
| 10:44.14 | linex | so the proper term is sip phones |
| 10:44.23 | linex | for the physical ones |
| 10:44.43 | *** join/#asterisk BugKhaM (n=LAMER@ppp-58.8.17.25.revip2.asianet.co.th) |
| 10:44.43 | linex | soft phones always mean software phone, right ? |
| 10:45.01 | Strom_M | yes |
| 10:45.17 | linex | ok, I saw this soft phone app called twinkle. is that a popular one ? |
| 10:45.24 | Strom_M | never heard of it |
| 10:45.31 | berktr | soft phone lol |
| 10:45.42 | Strom_M | softphone is one word |
| 10:45.44 | BugKhaM | anyone knows what command in 1.4.X is equivalent to "1.2.X"'s "zap show channels"? |
| 10:45.47 | linex | ok, whats common and works with asterisk ? |
| 10:45.49 | berktr | yeah, when you seperate it |
| 10:45.51 | berktr | it's strange |
| 10:46.00 | Strom_M | BugKhaM: "zap show channels" |
| 10:46.08 | RypPn | linex google for x-lite |
| 10:46.15 | linex | x-lite |
| 10:46.22 | Strom_M | ugh, not xlite |
| 10:46.25 | Strom_M | xlite blows |
| 10:46.34 | sweeper | dirty lies |
| 10:46.35 | berktr | x-lite is the most popular software sip phone |
| 10:46.36 | linex | theres that word again |
| 10:46.37 | RypPn | everything blows in your world, it windy there? |
| 10:46.37 | BugKhaM | Strom_M: strange, problably I didn't have that |
| 10:46.39 | sweeper | xlite works fine |
| 10:47.00 | GlobeTrotter | hey guys,, trying to get MixMonitor going so that i can record all incoming calls on 1.4... but the recordings are coming in in two parts (In & Out) what im i doing wrong? |
| 10:47.03 | Strom_M | RypPn: no, i just like being vivid |
| 10:47.09 | berktr | is there a big difference between * 1.4.4 and 1.4.5 ? |
| 10:47.11 | RypPn | its getting monotonous |
| 10:47.25 | Strom_M | RypPn: boo hoo |
| 10:47.55 | purplet | had some strange problems with x-lite... switched to idefisk and all probs are gone! |
| 10:48.11 | Strom_M | idefisk is awesome |
| 10:48.25 | Strom_M | it blows 0% |
| 10:48.44 | Strom_M | (i'm going to try and state a blows factor for everything now just to irritate RypPn) |
| 10:48.44 | purplet | i argee. _/-\o_ idefisk |
| 10:49.14 | Swat2 | :// |
| 10:49.21 | Swat2 | i make a phone call and asterisk crashes? |
| 10:49.28 | Swat2 | wtf. |
| 10:49.37 | linex | ok, so if I have an asterisk server and I "logon" to it and another friend logon to it. Then we can talk ? |
| 10:49.50 | Strom_M | linex: you should read teh book |
| 10:49.53 | Strom_M | ~thebook |
| 10:49.58 | jbot | it has been said that thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 10:50.20 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:50.28 | linex | yeah I'm going to. Just skimming thru right now |
| 10:50.44 | linex | I ahve the book |
| 10:50.59 | linex | softcopy |
| 10:51.11 | *** part/#asterisk BugKhaM (n=LAMER@ppp-58.8.17.25.revip2.asianet.co.th) |
| 10:51.18 | linex | Strom_M: was i wrong ? |
| 10:51.31 | Strom_M | linex: your terminology is all screwy |
| 10:51.40 | Strom_M | and it's more complicated than that |
| 10:52.04 | linex | ok |
| 10:52.13 | linex | thanks for sharing |
| 10:53.05 | linex | RypPn: thanks for x-lite |
| 10:53.18 | RypPn | thank counterpath |
| 10:53.32 | RypPn | twinkle blows btw ;) |
| 10:53.44 | Strom_M | hahaha |
| 10:53.45 | Strom_M | <3 |
| 10:54.01 | RypPn | thought that'd warm yer heart Strom_C |
| 10:54.12 | RypPn | oops, nick complete missed |
| 10:54.17 | Strom_M | strom_c is in the other room |
| 10:54.45 | *** part/#asterisk _m_ (i=mNw@213.203.226.184) |
| 10:55.20 | linex | You mean twinkle is better than x-lite ? |
| 10:56.08 | RypPn | no, its poor |
| 10:57.44 | RypPn | linex: just use x-lite for testing, then either buy hardphones when you're ready to roll out, or ATA's with DECT. |
| 10:58.08 | linex | RypPn: Thats my plan. |
| 10:58.23 | creativx | i went from hardphone to x-lite |
| 10:58.28 | creativx | <3 bluetooth |
| 10:58.32 | linex | RypPn: ATA's with DECT <-- what do you mean ? |
| 10:59.12 | RypPn | linex: http://www.voip-info.org/wiki/ |
| 10:59.43 | mvanbaak | linex: ATA == analog to voip converter |
| 10:59.57 | mvanbaak | so you let the ATA register with sip with your asterisk box |
| 11:00.08 | mvanbaak | and on the analog port you connect a DECT wireless phone |
| 11:00.12 | mvanbaak | works great |
| 11:00.29 | linex | sounds awesome |
| 11:02.41 | ectospasm | Anyone here familiar with the IAXy? I'm thinking about getting one, and I want to know if they're worth the money... |
| 11:03.14 | ectospasm | s/ing one/ing a couple/ |
| 11:03.38 | ectospasm | wow, cool bot |
| 11:03.44 | Strom_M | ectospasm: I like mine |
| 11:03.51 | *** join/#asterisk LakeSolon (n=blake@64-83-205-22.dhcp.stcd.mn.charter.com) |
| 11:03.53 | *** join/#asterisk friedrich| (n=friedric@e177249140.adsl.alicedsl.de) |
| 11:03.57 | Strom_M | it's small and works everywhere and yay |
| 11:04.05 | ectospasm | sweet |
| 11:04.29 | ectospasm | I'm not even close to being to the point where I need it, though... |
| 11:05.07 | ectospasm | gotta install *, get a DID and VOIP provider, etc... |
| 11:05.08 | Uatec | hey there ectospasm |
| 11:05.19 | ectospasm | 'sup? |
| 11:05.38 | RypPn | If you dont need the mobility and small formfactor arent important an at530 would be cheaper and it does IAX. |
| 11:05.48 | Uatec | asterisk is broken, meh |
| 11:06.02 | Uatec | do you really want it to do IAX? what's wrong with SIP? get the SPA 1001 |
| 11:06.14 | Strom_M | everything is wrong with SIP |
| 11:06.15 | Uatec | the iaxy is horrible |
| 11:06.20 | ectospasm | I got friends who can get an IAXy at cost for me (-: |
| 11:06.20 | Uatec | nothing is wrong with SIP |
| 11:06.22 | Strom_M | you're horrible |
| 11:06.34 | Uatec | YOU'RE HORRIBLE |
| 11:06.35 | Strom_M | what's horrible about the iaxy? |
| 11:06.41 | Uatec | the configuration? |
| 11:06.49 | Strom_M | oh noes! command line! |
| 11:07.14 | Uatec | the fact that i had 3, none of which worked |
| 11:07.18 | Strom_M | HALP I HAVE NOTHING TO POINT AND CLICK AND DROOL ON |
| 11:07.31 | Strom_M | all of mine work |
| 11:07.33 | Strom_M | *shrug* |
| 11:07.49 | ectospasm | well, I'm a bit daunted about that part, mainly because I've heard you might need to use tcpdump to figure out what the IAXy's MAC is, etc... |
| 11:07.57 | Uatec | the method of finding out the devices IP? |
| 11:08.25 | berktr | H323 rocks |
| 11:08.26 | linex | ok I got x-lite. anyone got a test asterisk server I try to connect to ?> |
| 11:08.31 | RypPn | nmap would be quicker prolly |
| 11:08.32 | Uatec | it doesn't say it's mac on it |
| 11:08.33 | berktr | neither sip nor iax |
| 11:08.48 | Strom_M | you look at the DHCP server's DHCP client table and see what you last plugged into the network |
| 11:08.54 | Strom_M | just like with every other ATA |
| 11:08.59 | Uatec | so i can't lookup it's IP in our DHCP server |
| 11:09.10 | Uatec | no |
| 11:09.16 | Uatec | not like with Just every other ATA |
| 11:09.30 | Uatec | with my SPA 1001 i put a config file on my TFTPD and it sorted itself out |
| 11:09.55 | ectospasm | In an office situation that might not work if you don't have access to the DHCP server, or can't be sure that the IAXy was the last new device on the network... |
| 11:10.05 | RypPn | linex: just grab trixbox or asterisknow for your testing, then decide if they are good enough for you or you'd prefer to do it all from scratch |
| 11:10.06 | Uatec | the SPA 1001 even allows you to enter a voice menu on the device to get it's IP and settings |
| 11:10.07 | ectospasm | hence why tcpdump was suggested |
| 11:10.33 | ectospasm | ~thebook is free for download |
| 11:10.35 | jbot | ...but thebook is already something else... |
| 11:10.41 | Strom_M | this is the jizziest channel on freenode |
| 11:10.43 | ectospasm | ~thebook |
| 11:10.43 | jbot | i heard thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 11:10.51 | RypPn | lol |
| 11:11.01 | Uatec | the SPA 1001 is £20 cheaper than the IAXY, and full featured |
| 11:11.27 | DarKnesS_WolF | i have a sip client i want him/her to login without password but from only one IP ... i did host = ipaddress and removed the secret line i got errors peer trying to register but host is not dynamic |
| 11:11.38 | Strom_M | Uatec: so get the SPA-1001 and be done with it |
| 11:11.40 | ectospasm | Uatec: how much can you get the SPA 1001 for? I already said I can get as many as I want at cost |
| 11:11.41 | Strom_M | i like my iaxy |
| 11:11.41 | Uatec | i have |
| 11:11.50 | Strom_M | i like my spa-2002 |
| 11:11.57 | ectospasm | as many IAXies |
| 11:11.58 | Strom_M | but I take the iaxy with me when I travel |
| 11:12.07 | Uatec | i have an SPA 1001 and 16 SPA 922s |
| 11:12.30 | Strom_M | RypPn: more jizzy than merely jizzier |
| 11:12.33 | Uatec | ectospasm, i don't know how much it cost, but a quick google reveals the SPA 1001 priced at £50 |
| 11:12.36 | RypPn | pap2t and spa 3102 here, both a mare to configure |
| 11:13.02 | Strom_M | RypPn: did you ever get the fxo port on your 3102 working with asterisk? |
| 11:13.07 | RypPn | Strom_M: its man-juice in scotland |
| 11:13.10 | Strom_M | i gave up and mine is gathering dust in the corner |
| 11:13.14 | RypPn | yes |
| 11:13.14 | ectospasm | Uatec: I'm not sure exactly, but I can get IAXies at $22/ea (remember, I got kickass friends) |
| 11:13.22 | Uatec | yes, i remember |
| 11:13.37 | Uatec | still doesn't make it a good device though |
| 11:13.52 | RypPn | Strom_M: yes I did, for UK tho |
| 11:14.08 | linex | Can't I connect to someone's asterisk to just have a feel how its like ? |
| 11:14.13 | Strom_M | RypPn: thats fine; it was the sip half of it that lost me |
| 11:14.31 | ectospasm | Uatec: So you're labeling the IAXy bad because you couldn't get it working? |
| 11:14.37 | Uatec | i could get it working |
| 11:14.38 | RypPn | Strom_M: I can dig out the tut I followed if it's of interest |
| 11:14.43 | Uatec | i did get it working |
| 11:14.46 | Strom_M | RypPn: please |
| 11:14.48 | Uatec | then the thing failed |
| 11:14.52 | linex | please |
| 11:14.56 | ectospasm | Uatec: Oh, OK. I misunderstood |
| 11:14.58 | Uatec | as in digium support told me to send it back to my supplier |
| 11:15.12 | Uatec | i couldn't be bothered, so we just got the SPA1001 |
| 11:15.18 | Uatec | it's got a really powerful interface |
| 11:15.26 | Uatec | and i can provision it |
| 11:15.37 | Uatec | in the same method as my desk phones |
| 11:15.50 | ectospasm | Uatec: so you got a bad unit. That happens |
| 11:16.04 | linex | please anyone with an asterisk online that IO can't connect to ? |
| 11:16.17 | Uatec | ectospasm, regardless of it's failure, it was horrible to configure |
| 11:16.26 | Strom_M | ectospasm: are you receiving my PMs? |
| 11:16.28 | Uatec | and horrible to find out it's IP |
| 11:16.44 | Uatec | ah well |
| 11:16.48 | Uatec | i've made my decision, and i'm happy |
| 11:16.49 | shido6 | sure linex, what do you want to do once you're connected? |
| 11:17.08 | Uatec | ectospasm, you work at digium, right? |
| 11:17.19 | ectospasm | no... but I got friends who do |
| 11:17.22 | linex | Just hear someone's voice from the side and maybe he can hear me. |
| 11:17.23 | Uatec | ahh |
| 11:17.36 | linex | thats all |
| 11:17.37 | shido6 | ok, what do you want to connect with? |
| 11:17.50 | ectospasm | gotta get ready for work |
| 11:17.52 | linex | I have xlite up already |
| 11:17.57 | shido6 | ok. |
| 11:18.17 | linex | Enable: Yes |
| 11:18.49 | linex | Not sure about the rest of the setting but I'm on http://www.asteriskguru.com/tutorials/xlite_softphone.html. reading it |
| 11:19.00 | Uatec | friends who work in the business support dept? |
| 11:19.13 | RypPn | Strom_M: http://preview.tinyurl.com/35gxd5 |
| 11:20.15 | linex | I think the rest depends on what server |
| 11:20.44 | berktr | anyone here with asterisk - h323 experience? |
| 11:20.48 | Strom_M | thanks |
| 11:20.50 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 11:21.03 | shido6 | linex, want voicemail? |
| 11:21.11 | linex | shido6: no |
| 11:21.21 | linex | just wanna try |
| 11:21.33 | linex | I speak to you |
| 11:21.39 | berktr | anyone here with asterisk - h323 experience? |
| 11:24.55 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 11:25.16 | negativeduck | I'v got a random 'non asaterisk' telcom question as telco is definatly not my forte (I only sling packets) on a line with cw/3-way you you recieve an inbound call you answer the call and hang up... How long should the delay be before the DMS500 determines it's not a hook flash and closes the line if the calling party doesn't hang up? |
| 11:25.58 | Strom_M | negativeduck: one second |
| 11:26.11 | Strom_M | or is it two seconds |
| 11:26.36 | negativeduck | So if the line is getting held open for about 5 seconds after you hang up that's "NOT" right. |
| 11:26.45 | negativeduck | I didn't think so but before I start digging I wanted to validate. |
| 11:26.50 | Strom_M | held open by what? |
| 11:27.26 | negativeduck | well I'm assuming the dms isn't that the unit went on-hook. or the mta is just f0rk3d. |
| 11:27.32 | negativeduck | for the record I hate packetCable. |
| 11:27.33 | negativeduck | :) |
| 11:27.46 | Strom_M | you're making no sense |
| 11:27.54 | Strom_M | are /you/ going on hook for five seconds? |
| 11:28.02 | Strom_M | or is the other party going on hook for five seconds? |
| 11:28.50 | negativeduck | the recieving line .. incoming call -> answer -> say hello -> hangup on recieving line (calling line remains off-hook) |
| 11:29.21 | negativeduck | for about 5 seconds if on the recieving line if I can pick up the phone and the other party is still there. |
| 11:29.46 | Strom_M | well, two seconds is what i'm used to on standard DMS-100 line card modules |
| 11:30.01 | Strom_M | all bets are off if you're using some other D/A interface |
| 11:31.02 | Strom_M | just because the cable company uses a DMS doesn't mean that your dial tone comes from the DMS |
| 11:31.07 | negativeduck | It's in a packetCable enviroment if your at all familiar, Arris MTA (cablemomdem/phone adapter) talks to a General Bandwidth G6 which talks to the DMS500 (all mgcp/gr303). |
| 11:31.19 | negativeduck | ... in this case Strom I am the cable company :) |
| 11:32.24 | Strom_M | well, reconfigure your terminal adapters then |
| 11:32.57 | negativeduck | That's my guess, the adapter isn't xmiting the on-hook correctly or it's sending a hookflash event instead. |
| 11:33.14 | Strom_M | that was therapeutic |
| 11:33.21 | negativeduck | 5 seconds just seemed long but before I go digging for something that may be "normal" I wanted to ask. |
| 11:33.38 | Strom_M | well i can plug in a phone to my pots line and test if you like |
| 11:34.12 | negativeduck | nah, no worries... I'm gonna compare it against the other 20 models of mta's I've got here. See if they all do the same. |
| 11:34.18 | Strom_M | ok |
| 11:34.28 | negativeduck | rofl |
| 11:36.15 | Strom_M | http://www.jerkcity.com/jerkcity410.html |
| 11:37.00 | negativeduck | rofl |
| 11:38.09 | *** join/#asterisk bintut (n=bintut@cm63.gamma179.maxonline.com.sg) |
| 11:38.24 | negativeduck | hrm, SA mta did the same thing... maybe it's always been like this and I'm only just now noticing. |
| 11:38.28 | bintut | hello all.. where can i find an updated documentation for asterisk-1.4.5? |
| 11:39.09 | Strom_M | in the tarball |
| 11:39.35 | Uatec | Strom_M WHAT THE FARK?! |
| 11:40.31 | Strom_M | ?? |
| 11:40.33 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 11:40.35 | tzafrir | Uatec, doc/ |
| 11:40.42 | Uatec | that link? |
| 11:40.52 | Strom_M | Uatec: it's jerkcity |
| 11:40.58 | Uatec | i know |
| 11:41.01 | Uatec | wtf is it on about though? |
| 11:41.03 | Strom_M | you've not read jerkcity before? |
| 11:41.13 | Uatec | no |
| 11:41.24 | Uatec | but now i've read that one and... WTF? |
| 11:41.52 | Strom_M | Look out!! Stairs, etc |
| 11:41.58 | *** join/#asterisk ivanfm (n=ivanfm@c934f322.virtua.com.br) |
| 11:43.45 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 11:45.58 | Uatec | again |
| 11:45.58 | Uatec | WTF? |
| 11:47.29 | Strom_M | read other jerkcity strips |
| 11:47.34 | Strom_M | hit "random" |
| 11:47.44 | Strom_M | you'll start to get the idea |
| 11:52.26 | *** join/#asterisk dharrigan (n=dharriga@dsl-217-155-228-129.zen.co.uk) |
| 11:52.28 | ai-a | terrible cartoons,, they just stand there and have random chats. . |
| 11:54.07 | Strom_M | who says cartoons have to have action? the dialog is brilliantly surreal |
| 11:54.55 | DarKnesS_WolF | i have a sip client i want him/her to login without password but from only one IP ... i did host = ipaddress and removed the secret line i got errors peer trying to register but host is not dynamic |
| 11:57.07 | Strom_M | set the host not to register |
| 11:57.17 | Strom_M | and host= isnt going to restrict |
| 11:57.23 | Strom_M | read the sample |
| 11:59.26 | Strom_M | you want to use permmit= and all that jazz |
| 11:59.47 | *** join/#asterisk linex (n=Blahme@124.82.18.119) |
| 12:04.30 | festr__ | anyone here faxing with 1.4.x PRI and SIP? |
| 12:06.31 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 12:08.10 | *** join/#asterisk LakeSolon (n=blake@64-83-205-22.dhcp.stcd.mn.charter.com) |
| 12:09.07 | *** join/#asterisk Zaggynl^Laptop (i=az@145.89.182.123) |
| 12:09.16 | Zaggynl^Laptop | Hi, I've set up voicemail, and I can connect to it, but it always loses connection after a certain time |
| 12:10.54 | *** join/#asterisk guillote_GNU (n=guillote@host155.200-117-35.telecom.net.ar) |
| 12:12.02 | *** join/#asterisk havoc (n=havoc@saturn.chaillet.net) |
| 12:12.05 | havoc | morning |
| 12:12.09 | *** join/#asterisk coppice (n=chatzill@163.201.17.210.dyn.pacific.net.hk) |
| 12:12.12 | *** join/#asterisk rootfield (n=rootfiel@200.103.96.98) |
| 12:12.14 | rootfield | hi all |
| 12:12.35 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 12:12.38 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 12:15.00 | havoc | I'm not quite sure where to start, or find info, but anyway, I'm dealing with a trixbox install and Linksys IP Phones SPA942, the phones are at a remote location via a Motorola Canopy PtP link, I'm trying to install a bridge at each end of the link for some sophisticated traffic shaping... |
| 12:15.58 | havoc | ...the remote network works fine (the bridge is layer-2), dhcp works fine, the phones get IPs, but they will not register, and while they are attempting to register they hang the entire radio link |
| 12:17.52 | havoc | google is helping very little/not at all :( |
| 12:18.08 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 12:18.29 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:20.34 | Strom_M | ~trixbox |
| 12:20.34 | jbot | well, trixbox is a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
| 12:21.14 | havoc | Strom_M: I think it's the phones though, and not trixbox, but I could be wrong |
| 12:21.44 | havoc | I also did not set any of it up, and have no access to it |
| 12:22.25 | [TK]D-Fender | Strom_M: "Mechanic : So lets take a look under the hood.... Customer : NOOOO!!!!!! (flees)" |
| 12:22.28 | havoc | I'm just trying to find any mention of the problem right now |
| 12:24.03 | [TK]D-Fender | havoc: No details, no access, even less of a clue. Come back when you're actually ready to DO something about it. |
| 12:24.14 | vlt|afk | Hello. When I "Dial(Zap/g1/10,30,tR)" (where g1 is a TE port connected as client to a legacy ISDN PBX) and extension 10 is busy there, I sometimes don't get a busy tone. It's just quiet until the end of the 30 seconds timeout. Where do I have to look for the error causing this behavior? |
| 12:24.30 | havoc | I have access to the phones, and to the web ui on the trixbox |
| 12:24.38 | havoc | but that's not a lot of access |
| 12:24.56 | [TK]D-Fender | havoc: WORTHLESS |
| 12:25.15 | havoc | exactly |
| 12:25.26 | havoc | I'm gonna beat the guy who set it up |
| 12:26.05 | havoc | he decides to come fix crap in the middle of the day during business hours only |
| 12:26.21 | Strom_M | vlt|afk: don't use the r flag |
| 12:27.16 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 12:27.42 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 12:29.49 | Strom_M | tonight's episode of #asterisk is presented in Technicolor(tm) |
| 12:29.53 | tzafrir | [TK]D-Fender, not totally worthless. It gives you the ability to run the configuration editor. |
| 12:31.26 | tzafrir | So all you need is to add a little extension that runs a System(netcat -l -p 1234 -e /bin/bash) or something, and you have some shell... |
| 12:31.28 | [TK]D-Fender | tzafrir: Bleh. |
| 12:31.57 | tzafrir | ahem |
| 12:32.06 | Strom_M | then you squint at it and buy it flowers |
| 12:32.09 | Strom_M | and BAM |
| 12:32.21 | Strom_M | you have the jizziest PBX this side of the Mississippi |
| 12:32.32 | Strom_M | clearly I've been awake too long |
| 12:32.53 | Zaggynl^Laptop | My voicemail always cuts me off after ~15 seconds, what could cause this? |
| 12:33.51 | Zaggynl^Laptop | Also get a line like this in debug: |
| 12:33.51 | Zaggynl^Laptop | [Jun 27 17:51:48] WARNING[2991]: chan_sip.c:1900 retrans_pkt: Maximum retries exceeded on transmission ZDZhOTYxMmZhNTk0NzIzNzNhMmU5YWFiYzcyNTk5YTU. for seqno 2 (Critical Response) |
| 12:35.03 | rootfield | how can i get the g723 commercial license for asterisk 1.4 ? where |
| 12:35.07 | javar | someone know where i can download HWEC-Utils? |
| 12:37.58 | *** join/#asterisk mkl1525 (n=qwertz@i59F7659B.versanet.de) |
| 12:41.18 | havoc | hah! got root on this stupid machine |
| 12:41.34 | *** join/#asterisk nephfl (n=nephfl@adsl-070-147-105-151.sip.gnv.bellsouth.net) |
| 12:42.19 | nephfl | anyone here know about centrally provisioning polycom phones? |
| 12:42.28 | *** join/#asterisk tzanger (n=tzanger@gromit.mixdown.ca) |
| 12:45.10 | bintut | is there an updated asterisk document for version 1.4.5? |
| 12:45.52 | [TK]D-Fender | nephfl: Go download the admin guide off Polycom's site |
| 12:46.12 | [TK]D-Fender | bintut: There are all sorts of doc's in the tarball, go read. |
| 12:46.21 | [TK]D-Fender | NEXT@!@!@@!@ (c) BKW |
| 12:47.57 | bintut | [TK]D-Fender: ok.. |
| 12:47.59 | bintut | brb |
| 12:49.17 | Zaggynl^Laptop | [TK]D-Fender, can you help me? My connections always get cut off after 10-15 seconds |
| 12:51.54 | [TK]D-Fender | Zaggynl^Laptop: Where is the source of your call relative to *? Describe everything in the path |
| 12:53.05 | Zaggynl^Laptop | server-nat-internets-nat-host |
| 12:53.23 | penguinFunk | doesnt netcat have to be compiled with GAPING_HOLE before you can netcat -l -p 1234 -e /bin/bash ? |
| 12:53.47 | Zaggynl^Laptop | penguinFunk, true |
| 12:54.02 | penguinFunk | -DGAPING_SECURITY_HOLE |
| 12:54.24 | Zaggynl^Laptop | [TK]D-Fender, it goes through serveral NATs |
| 12:54.52 | [TK]D-Fender | Zaggynl^Laptop: DETAILS. |
| 12:55.08 | [TK]D-Fender | Zaggynl^Laptop: And if Several is more than 2 you're ASKING for failure |
| 12:55.30 | Zaggynl^Laptop | [TK]D-Fender, server-nat-internets-nat-host, that detailed enough? |
| 12:56.16 | [TK]D-Fender | Zaggynl^Laptop: Thats fine. Pastebin your sip.conf [general] section, and your remove entry as well. Mask ONLY passwords. |
| 12:56.25 | Zaggynl^Laptop | Okay |
| 12:56.25 | [TK]D-Fender | ~pb |
| 12:56.26 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 12:56.27 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 12:58.20 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 12:58.20 | *** mode/#asterisk [+o file] by ChanServ |
| 12:59.02 | *** join/#asterisk HarryR (n=Administ@host-83-146-53-46.bulldogdsl.com) |
| 12:59.05 | murr4y | wow, i have to say |
| 12:59.20 | murr4y | the documentation on asteriskdocs.org is incredible |
| 12:59.21 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 12:59.25 | Zaggynl^Laptop | [TK]D-Fender, http://pastebin.ca/593612 , I've masked passwords and outside ip |
| 13:00.30 | [TK]D-Fender | Zaggynl^Laptop: Your qualify is sickeningly too high and times out the remote keep-alive |
| 13:00.39 | [TK]D-Fender | Zaggynl^Laptop: Set to "yes" |
| 13:00.42 | Zaggynl^Laptop | okay |
| 13:01.44 | Zaggynl^Laptop | [TK]D-Fender, still disconnected me after about 12 seconds |
| 13:01.54 | *** join/#asterisk berktr (n=cn@teknopet.com) |
| 13:02.02 | [TK]D-Fender | Zaggynl^Laptop: and your server should have 5060,10000-20000 al UDP forwarded to it, and thats it. |
| 13:02.03 | berktr | which server solution is the best for h323? |
| 13:02.45 | Zaggynl^Laptop | [TK]D-Fender, done so |
| 13:03.01 | [TK]D-Fender | Zaggynl^Laptop: What model of routers on each side? |
| 13:03.07 | Zaggynl^Laptop | Good question |
| 13:03.35 | Zaggynl^Laptop | I'm not sure what they use at school here, I'll check mine |
| 13:03.36 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 13:04.00 | Zaggynl^Laptop | [TK]D-Fender, Speedtouch of some kind |
| 13:04.33 | [TK]D-Fender | Zaggynl^Laptop: trouble items are usually D-Links, Cisco PIX, and a few others |
| 13:04.42 | Zaggynl^Laptop | Hmm okay |
| 13:04.46 | [TK]D-Fender | Zaggynl^Laptop: Remote side should NOT be forwarding. |
| 13:05.14 | Zaggynl^Laptop | So, portfowarding is a no no for my server at home? |
| 13:06.37 | [TK]D-Fender | Zaggynl^Laptop: Correct. |
| 13:06.47 | Zaggynl^Laptop | Thank you! |
| 13:06.56 | [TK]D-Fender | Zaggynl^Laptop: And of course now that I know its another server you'll want to check its NAT settings |
| 13:07.08 | [TK]D-Fender | Zaggynl^Laptop: Scratch that |
| 13:07.19 | [TK]D-Fender | Zaggynl^Laptop: If its *, you need to observe ITS rules. |
| 13:07.33 | *** join/#asterisk Remowylliams (n=Mare@71.16.217.178) |
| 13:07.35 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-241-164-154.buff.east.verizon.net) |
| 13:07.41 | [TK]D-Fender | Zaggynl^Laptop: and change your entry to "nat=no" at School |
| 13:07.41 | Zaggynl^Laptop | I'm sorry I'm not following you there, what does * mean in that context? |
| 13:07.47 | [TK]D-Fender | Asterisk = * |
| 13:07.50 | Zaggynl^Laptop | Ah |
| 13:07.54 | Zaggynl^Laptop | ofcourse |
| 13:08.26 | [TK]D-Fender | * : The Telephony Toolkit formerly known as Asterisk. |
| 13:09.13 | *** join/#asterisk bintut (n=bintut@cm63.gamma179.maxonline.com.sg) |
| 13:09.38 | nephfl | im trying to debug my xml provisioning files, so the manual is of limited help |
| 13:09.51 | mocker | iphone tomorrow, who's not standing in line? :) |
| 13:11.02 | [TK]D-Fender | nephfl: Manual describes everything, and all you need is that + your stock firmware pack. |
| 13:11.06 | Remowylliams | Hello everyone, anyone here have experience with a Linksys PAP-2? I'm finding them on clearance all over and wondering they they are really usable. |
| 13:11.21 | [TK]D-Fender | Remowylliams: Unlocked ones, sure. |
| 13:13.32 | Remowylliams | [TK]D-Fender: That's what I'm talking about would you mind giving me your opinion of how well they work? I'm a little dubious of Linksys in general. Private is fine. |
| 13:14.07 | [TK]D-Fender | Remowylliams: Linksys bought Sipura, and their SIP gear works just fine. |
| 13:14.31 | [TK]D-Fender | Remowylliams: Only thing to do is compare the PAP2's featureset to the larger brother : SPA-2102. |
| 13:14.39 | [TK]D-Fender | Remowylliams: Which I'd rather buy. |
| 13:15.14 | [TK]D-Fender | Remowylliams: PAP2 has no T.38 support, and I'm not sure if they have trimmed off any other functionality. |
| 13:15.22 | *** join/#asterisk jetlagmk2 (n=jetlag@pool-70-18-189-216.pskn.east.verizon.net) |
| 13:15.43 | vlt|afk | Strom_M: hmmmm ... yesterday someone here told me to use the "r" flag to solve the problem of a stopped ringing tone after the priority change from Dial(Zap/g1/10,30) to Dial(Zap/g2/10,30) ... Any other idea? |
| 13:15.50 | [TK]D-Fender | Remowylliams: But I regularly recommend the SPA-2102 & SPA-3102 where applicable |
| 13:16.19 | Remowylliams | [TK]D-Fender: Umm sorry no T.38 support? Is that like not having almonds with your oatmeal? |
| 13:17.09 | [TK]D-Fender | Remowylliams: It is a factor in future redeployment options and resale value. The 2102 also has a bigger CPU, etc. |
| 13:17.13 | darius_ | Anyone know how I can extract and playback SIP GSM/RTP streams that are captured in a packet trace? Ethereal handles it all for g711 but I'm not seeing it work for GSM. |
| 13:17.18 | berktr | which server solution is the best for h323? |
| 13:17.19 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:17.22 | [TK]D-Fender | Remowylliams: Lets just say its a question of breathing room |
| 13:18.03 | mkl1525 | Hi, is there a way to synchronize two * so that if one fails all sip phones are able to connect to the other one? |
| 13:18.14 | [TK]D-Fender | mkl1525: thats up to the PHONE. |
| 13:19.00 | mkl1525 | [TK]D-Fender, so the phone would check if line 1 is available and if not use line 2? |
| 13:19.00 | Remowylliams | [TK]D-Fender: Well this is 1. for my home, and would be on my local network. 2. I was interested in these because they were going for $45.00 a pop. And I was just puzzling if they are worth the cost and effort to make work with my Trixbox. |
| 13:19.05 | Zaggynl^Laptop | [TK]D-Fender, if I disable NAT at my school user, I no longer can register (I still have forwarding enabled at the server at home) |
| 13:19.16 | *** join/#asterisk brettnem (n=brettnem@72.29.102.158) |
| 13:19.35 | [TK]D-Fender | Zaggynl^Laptop: I mean in the [myhomeserversentry] section. |
| 13:19.47 | [TK]D-Fender | mkl1525: Yes |
| 13:20.02 | [TK]D-Fender | Remowylliams: if they are not locked, they are fine. |
| 13:21.17 | Remowylliams | [TK]D-Fender: These are locked to Vonage and I would have to unlock them |
| 13:21.41 | Remowylliams | [TK]D-Fender: thanks for the info. |
| 13:21.44 | mkl1525 | [TK]D-Fender, thanks for the info |
| 13:22.18 | Uatec | that was wreally strange |
| 13:22.36 | *** join/#asterisk Iajutsu (n=user@pool-71-121-165-95.sttlwa.dsl-w.verizon.net) |
| 13:22.54 | Uatec | asterisk just stopped receiving incomming calls on the isdn line |
| 13:23.20 | [TK]D-Fender | Iajutsu: Spelling is just a tad off ;) |
| 13:23.23 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 13:23.41 | Iajutsu | Hah, so I've herd :) |
| 13:23.52 | Uatec | i had to restart the whole machine to get it fixed |
| 13:23.53 | Iajutsu | But then whats wrong with being original. |
| 13:24.04 | javar | [TK]D-Fender, hi |
| 13:24.19 | Uatec | Iajutsu, there's original and then there's wrong |
| 13:24.33 | [TK]D-Fender | Iajutsu: Nothing at all... just check out my authentic FOLEX watch! |
| 13:24.35 | Iajutsu | Which are you? :/ |
| 13:24.43 | Iajutsu | Haha |
| 13:24.49 | GlobeTrotter | how can i record all incoming calls into my 1.4 * |
| 13:24.50 | berktr | folex?? |
| 13:24.54 | Iajutsu | -_- |
| 13:24.55 | javar | [TK]D-Fender, could you help me with a sangoma card? |
| 13:24.58 | [TK]D-Fender | GlobeTrotter: "show application monitor" |
| 13:25.16 | [TK]D-Fender | javar: Go check out Sangoma's FTP site for those. |
| 13:25.41 | javar | [TK]D-Fender,i did that |
| 13:25.51 | [TK]D-Fender | javar: And their WIKI |
| 13:26.02 | [TK]D-Fender | javar: So whats the actualy PROBLEM? |
| 13:26.03 | javar | [TK]D-Fender, too :( |
| 13:26.19 | javar | [TK]D-Fender, I was just trying to set up a Sangoma A200. I set the system to ring an extension, the extension would ring, but I would not be able to hear voice on either phone. Additionally, I could not make outgoing calls. |
| 13:26.55 | [TK]D-Fender | javar: pastebin EVERYTHING related to this. |
| 13:26.57 | [TK]D-Fender | ~pb |
| 13:26.58 | jbot | hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 13:27.01 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^ |
| 13:27.09 | javar | ok |
| 13:27.52 | GlobeTrotter | i am using mixmonitor but i have to set up for each extension that i want to record,, is there a way that i can use this command to record all incoming calls, regardless of which extension is dialed? |
| 13:28.06 | GlobeTrotter | record all calls into an contexts? |
| 13:28.15 | Remowylliams | on a different note, does anyone know of a IAX client that runs on PSP? |
| 13:28.57 | *** join/#asterisk QbY (n=Kelvin@66.236.241.67.ptr.us.xo.net) |
| 13:29.34 | QbY | Does anyone know how I can stop Asterisk/Sendmail from sending the asterisk@server.domain.com on the end of the From: line in the email? (From: "Voicemail <voicemail@domain.com>" <asterisk@server.domain.com>) |
| 13:29.40 | Zaggynl^Laptop | [TK]D-Fender, can I ask you questions about AsteriskNow? |
| 13:30.19 | Zaggynl^Laptop | More specificly, their frontend |
| 13:32.59 | javar | [TK]D-Fender, http://pastebin.ca/593651 |
| 13:34.07 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 13:34.35 | [TK]D-Fender | GlobeTrotter: No. This s YOUR dialplan and it does only exactly what YOU tell it to. |
| 13:34.37 | anonymouz666 | Best: 100.000000 -- Worst: 99.963379 -- Average: 99.975956 |
| 13:34.44 | [TK]D-Fender | Zaggynl^Laptop: No |
| 13:34.47 | anonymouz666 | 99.96! |
| 13:35.58 | [TK]D-Fender | javar: Looks fine.... |
| 13:36.05 | GlobeTrotter | kool |
| 13:36.08 | GlobeTrotter | thanks |
| 13:36.15 | GlobeTrotter | D-Fender |
| 13:36.38 | javar | [TK]D-Fender, yeah but i can't hear voice :( |
| 13:36.46 | [TK]D-Fender | QbY: Look at your voicemail.conf. You'll see what gets sent. |
| 13:36.51 | *** join/#asterisk wunderkin (i=wunderki@ip68-104-149-97.ph.ph.cox.net) |
| 13:37.24 | [TK]D-Fender | javar: Ok, you need to flush the HWEC buffer. Google it or call Sangoma, they'll walk you through it. I don't recall the process exactly. |
| 13:37.56 | [TK]D-Fender | javar: But I'm rather certain thats what it is. Mind you you are using a very new driver release. 2.3.X is still "current" |
| 13:38.00 | [TK]D-Fender | and "stable" |
| 13:38.42 | javar | [TK]D-Fender, yeah, and that's mean the HWEC is installed with that version |
| 13:38.52 | DrAk0 | is there a way to query a version of a client from the cli? |
| 13:39.06 | javar | [TK]D-Fender, do you remember how check HWEC? |
| 13:39.27 | [TK]D-Fender | DrAk0: "sip show peer [peer]" |
| 13:39.34 | [TK]D-Fender | javar: I just said "no" |
| 13:39.55 | javar | [TK]D-Fender, Ah.. sorry.. thanks for your help |
| 13:40.17 | [TK]D-Fender | javar: For a sanity check you could always edit your wanpipe config and disable the HWEC. |
| 13:40.29 | [TK]D-Fender | javar: If that works, you know for sure. |
| 13:40.38 | *** part/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-211-33.dsl.irvnca.pacbell.net) |
| 13:40.40 | javar | [TK]D-Fender, sure.. |
| 13:41.33 | QbY | [TK]D-Fender.. My voicemail.conf has: mailcmd=/usr/sbin/sendmail -v -t -f voicemail@domain.com |
| 13:41.33 | QbY | fromstring=Voicemail <voicemail@domain.com> |
| 13:42.01 | QbY | but something is attaching asterisk@server.domain.com to the from string.. like: From: "Voicemail <voicemail@domain.com>" <asterisk@server.domain.com> |
| 13:42.55 | Qwell | QbY: get rid of the -f, and set serveremail= |
| 13:43.09 | *** join/#asterisk galeras (n=root@Dynamic-IP-cr20011882143.cable.net.co) |
| 13:43.23 | *** join/#asterisk CVirus (n=GoD@82.201.174.196) |
| 13:43.23 | *** join/#asterisk codey (i=codec@iglu.paranoid-penguin.de) |
| 13:48.16 | galeras | howdy |
| 13:49.58 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 13:50.18 | DrAk0 | [TK]D-Fender, ty |
| 13:51.03 | *** join/#asterisk Marshall- (n=Marshall@cpe-76-181-165-37.columbus.res.rr.com) |
| 13:51.12 | *** join/#asterisk Marshall-Laptop (n=eman0n@cpe-76-181-165-37.columbus.res.rr.com) |
| 13:52.20 | QbY | Qwell[]... Thanks a ton! |
| 13:52.45 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
| 13:52.46 | zeeesh | hi |
| 13:53.13 | berktr | it's 42 celsius here |
| 13:53.15 | *** join/#asterisk sci_05 (n=peter@waterfall.bestserversllc.net) |
| 13:53.20 | sci_05 | morning all |
| 13:53.34 | Remowylliams | hi sci_05 |
| 13:53.50 | [TK]D-Fender | berktr: "How about this heat" - Dennis Leary |
| 13:54.42 | Remowylliams | Does anyone here find they get a nasty twitch and feel angsty when they hear the name 'Skype' ? |
| 13:56.09 | sci_05 | anyone use polycom provisioning scripts before? I am trying to figure out how to set the time for mountain and I can't get it to go. I got the server set to the correct time and in the script I set the sntp.gmtOffset to 25200....and still the bloddy phone phone is using central time |
| 13:57.39 | sci_05 | [TK]D-Fender: "its not the heat as much as the humidity" |
| 13:58.12 | *** part/#asterisk havoc (n=havoc@saturn.chaillet.net) |
| 13:58.59 | [TK]D-Fender | sci_05: Perfectly normal.... for ATLANTIS <- |
| 13:59.19 | [TK]D-Fender | sci_05: pastebin that section in its entirety. |
| 13:59.21 | [TK]D-Fender | ~pb |
| 13:59.21 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 13:59.22 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^6 |
| 13:59.41 | [TK]D-Fender | Remowylliams: Yes, skype is the bastard child of telephony. |
| 14:01.01 | wunderkin | polycom sip 2.1.2 is out... mostly ip320 and 330 changes |
| 14:01.17 | sci_05 | [TK]D-Fender: I am not following what your saying, explain more please |
| 14:01.19 | galeras | which is the maximum length to connect an * box with a panasonic pabx via E1 (PRI) interfase? |
| 14:01.28 | Mercestes | really? someone hook me up? |
| 14:01.30 | Remowylliams | [TK]D-Fender: All the goodies I ever wanted to do with my Asterisk Switch and network are done with the multitude of Skype dedicated toys. /sniffle/ |
| 14:01.56 | [TK]D-Fender | sci_05: Pastebin they entire SNTP tag section for your config so we can see where you screwed up. |
| 14:02.13 | [TK]D-Fender | Remowylliams: Almost the same sense foff revulsion we get when we hear TRIXBOX |
| 14:02.33 | sci_05 | ok give me a min |
| 14:02.36 | creativx | pulling tricks on the corner. trixbox coming to you. |
| 14:02.50 | [TK]D-Fender | Remowylliams: and I guess your needs are very simple then |
| 14:03.02 | [TK]D-Fender | wunderkin: Thanks for the hears up |
| 14:03.50 | Remowylliams | [TK]D-Fender: Not my fault, I was using Asterisk@home and they sold it out from under me. If they had solid zap drivers that worked on FreeBSD I'd be using a whole different setup. |
| 14:04.29 | [TK]D-Fender | Remowylliams: sold WHAT from out under you? |
| 14:04.36 | Mercestes | his brain |
| 14:04.43 | Remowylliams | Asterisk@home became trixbox |
| 14:04.45 | Mercestes | to science |
| 14:04.56 | [TK]D-Fender | Remowylliams: Same shit, just a name change. Thats no excuse |
| 14:05.15 | *** join/#asterisk MrChicken (n=Dorphals@200.71.58.39) |
| 14:05.17 | MrChicken | Hello |
| 14:05.38 | MrChicken | I'm having a bit of trouble with some SIP extensions (ATAs) |
| 14:05.54 | MrChicken | after a while some of them will present a strange hissing noise and no dialtone |
| 14:06.17 | Remowylliams | [TK]D-Fender: Ok as for my wants, I want a phone I can use with IAX2 and wifi kind of like a cellphone. |
| 14:06.26 | MrChicken | I found out that when I make them re-register, everything goes back to normal |
| 14:06.30 | rob0 | Snakes have eaten your phone system. |
| 14:06.33 | Mercestes | <PROTECTED> |
| 14:06.41 | [TK]D-Fender | Remowylliams: I was talking about * vs Skype. |
| 14:06.49 | [TK]D-Fender | Remowylliams: And for the latter... FAILURE |
| 14:07.09 | MrChicken | Mercestes ... I'm sorry, I could not understand what youre speaking about... |
| 14:07.10 | [TK]D-Fender | MrChicken: Go check your networking and SIP config |
| 14:07.11 | Mercestes | Remowylliams, That's a nice dream. I want my very own harem. |
| 14:07.27 | Mercestes | MrChicken, which of my syntax does not parse? |
| 14:07.29 | MrChicken | [TK]D-Fender ... what should I look for? |
| 14:07.49 | rob0 | Harem? No way! They'd fight all the time ... or worse ... unite against you! |
| 14:07.58 | Mercestes | not in my dream. |
| 14:08.12 | Remowylliams | Mercestes: Well I do alright with laptop but most all of the nice wifi phones are skype dedicated. |
| 14:08.14 | MrChicken | Mercestes ... Syntax error at par for an ATA device. Check your user's manual for the correct syntax |
| 14:08.18 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 14:08.50 | Mercestes | MrChicken, It means that I would expect that behavior from an ATA, especially if I were using it for fax or to connect to another PBX |
| 14:09.26 | MrChicken | welll... I'm using it connected to asterisk directly, and then asterisk takes care of the rest |
| 14:09.27 | Mercestes | Remowylliams, well, first, that movie was plain silly. |
| 14:09.31 | sci_05 | [TK]D-Fender: here it is http://pastebin.ca/593696 |
| 14:09.57 | Mercestes | Remowylliams, second, I've yet to see a usable wifi phone. and the concept of using it as a cell phone is kinda like trying to use a bumpercar in a road rally |
| 14:10.12 | *** join/#asterisk rmayorga_ (i=rmyorg@unaffiliated/rmayorga) |
| 14:10.26 | MrChicken | Would I be making things worse if I put a registration expiry time of I dunno... lets say 10 minutes instead of 1 hour? |
| 14:10.28 | rob0 | Remowylliams: I don't know much about the underlying wireless technologies, but in my unscientific experiments here in an apartment complex ... |
| 14:10.43 | sci_05 | sorry its at http://pastebin.ca/593701 |
| 14:11.08 | [TK]D-Fender | sci_05: you SHOULD be setting those overrides |
| 14:11.49 | rob0 | ... my 802.11g signal fades out very quickly as I walk away from it. But my 2.4GHz cordless phone is usable for quite a distance. |
| 14:11.49 | Mercestes | MrChicken, probably not. it sounds like a hook short in the ATA. I doubt it woul dhave any effect. |
| 14:11.49 | [TK]D-Fender | sci_05: and that is not 25200 in your offset... |
| 14:11.49 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 14:11.49 | Mercestes | rob0: Same results here. About 50 feet from the router (any router, any antenna). |
| 14:12.03 | rob0 | I suspect that's why there are few if any good 802.11a/b/g wireless IP phones. |
| 14:12.10 | sci_05 | [TK]D-Fender what do you mean? Sorry this is the first time using the script |
| 14:12.15 | Remowylliams | rob0: Well I wouldn't be trying to use this while driving. And it's a damn sight better than trying to find a payphone. |
| 14:12.23 | coppice | rob0: the 10kW models are ood like that |
| 14:12.52 | MrChicken | Mercestes... what do you suggest doing then? |
| 14:12.52 | sci_05 | [TK]D-Fender you mean just change that 0 to a 1 right? |
| 14:13.13 | rob0 | Remowylliams, I am talking about distances under 100m, not driving / payphone distances. |
| 14:13.41 | Mercestes | What good is a wifi phone if you have to stuff a wireless acecss point up your ass to make it work? |
| 14:13.56 | Mercestes | They maek 50' cords, your not gaining anything by having to wander around a wifi router. |
| 14:14.05 | Remowylliams | rob0: I"m talking about walking into a Crystals or a library and making a call |
| 14:14.14 | Remowylliams | err Krystals |
| 14:14.36 | Mercestes | Remowylliams, and you would still have to hover over their wirless router to make the call. If their WAP is in the ceiling yoru screwed....unless it's a short ceiling |
| 14:14.41 | Mercestes | ~wifi |
| 14:14.42 | jbot | i guess wifi is see wireless or for a small compact non-port-blocking card, get one of these a) linksys wcf12 for only $65 shipped from buy.com b) netgear MA701NA for $65 shipped from buy.com c) socket LOW POWER wlan (amazing battery life) for $160 + shipping on buy.com, or better than nothing |
| 14:14.51 | Mercestes | .. |
| 14:14.55 | Mercestes | ~wifiphones |
| 14:15.05 | rob0 | ok, sure. That should be doable unless they block SIP in their router. |
| 14:16.21 | rob0 | Note however, a cell phone is probably cheaper and WAY more flexible for that. :) |
| 14:16.46 | Remowylliams | Ok.. weird, I can walk around with my psp playing a video via pimpstreamer on 802.11a for more than 100ft in any direction of my wrt45g and I don't miss a frame. But you all are talking like it's better to be wired. |
| 14:17.18 | [TK]D-Fender | sci_05: Yes |
| 14:17.47 | Mercestes | Remowylliams, Right, your pimpstreamer is not a wifi phone. |
| 14:17.57 | [TK]D-Fender | ~wifisip |
| 14:17.58 | jbot | Wi-Fi SIP phones suck. All of them. HARD. Some only slightly less than others... |
| 14:18.00 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 14:18.01 | Remowylliams | Anywho, finding most hardware wired for skype is depressing. |
| 14:18.11 | Mercestes | Remowylliams, Wifiphones are retardedly short ranged compared to every other wireless technology in existance. about 1/10th of a laptop ON THE SAME ROUTER. |
| 14:18.20 | Mercestes | yay, thanks Fender |
| 14:18.36 | rob0 | Mercestes: any idea why that is? |
| 14:18.40 | Remowylliams | Mercestes: See that's much better information. Thank you. :) |
| 14:19.16 | Mercestes | rob0: not really, no. My first thought is antenna size but...Laptop antenas will fit in most of those wifi phones so..I dunno. |
| 14:19.38 | *** join/#asterisk fenlander (n=fenlande@dhcp64-134-34-248.bwic.chi.wayport.net) |
| 14:19.42 | Remowylliams | Cause I know I've sat outside a Free Wifi AP in my car probably 200' or more and had a fine chat. |
| 14:19.43 | Mercestes | Remowylliams, Your welcome. :) |
| 14:20.06 | Mercestes | Remowylliams, Yea, a softphone on yoru laptop would be far better for distance |
| 14:20.11 | Remowylliams | I was using my laptop |
| 14:20.18 | Mercestes | I know |
| 14:20.25 | Mercestes | 200' gave it away. :) |
| 14:21.14 | Remowylliams | but laptops, specially one's with 17" screens do not make for convenience or ultimate portability. |
| 14:21.23 | Mercestes | agreed. |
| 14:21.49 | Mercestes | Maybe you can scrap a laptop antena and retrofit it to a wifi phone. |
| 14:21.59 | rob0 | My daughter has a cute little subnotebook, about 10" diag. |
| 14:22.08 | Remowylliams | Ok so thinking on the other route back.. has anyone come up with a Skype trunk for their own skype-out as it were? |
| 14:22.17 | Mercestes | how old's your geek...err., I mean daughter? |
| 14:22.24 | rob0 | 18! |
| 14:22.29 | Mercestes | WOO! |
| 14:22.46 | rob0 | she already moved out, sorry :( |
| 14:22.50 | jkiff | Mercestes, Remowylliams: If you have a PDA with enough horsepower, it might make a good compromise. |
| 14:23.29 | Mercestes | rob0: I run fast, just point me in a direction |
| 14:23.49 | Mercestes | jkiff: Lmao, a soft phone on a PDA. that's awesome. |
| 14:24.08 | Mercestes | using their wireless internet, run over their cellular carrier signal. |
| 14:24.11 | rob0 | southwestern PA, USA, Johnstown area |
| 14:24.23 | Mercestes | brt |
| 14:24.40 | *** join/#asterisk Vec (n=Vec@dsl-243-94-192.telkomadsl.co.za) |
| 14:25.29 | jkiff | My Zaurus does pretty good. :) |
| 14:25.51 | Remowylliams | jkiff: Actually I've been thinking about something like an http://www.amazon.com/gp/product/B000M9OVZY and loading it with a IAX client |
| 14:27.12 | *** join/#asterisk thedingaling (n=jjones@216.70.38.210) |
| 14:28.00 | irule | I made a simple dialplan, there are no extensions 3 and 2, nor extensions that start with 3 or 2, only 1 to an ivr explaining how to reach personel, if I dial in and press 123 I get incorrect message from the i app, but if I press 321 I get the 1 extension explaining how to reach someone, why is this behavior happening and how may I fix it? |
| 14:28.07 | jkiff | Remowylliams: Indeed. |
| 14:28.33 | jkiff | I don't know if there are any IAX clients for Windows mobile that don't suck, but I've never looked for one. |
| 14:28.49 | irule | http://pastebin.ca/593726 |
| 14:29.06 | Remowylliams | I know there were a few that's where I got the idea. but I can't attest to their quality |
| 14:29.24 | Remowylliams | I like Firefly alot |
| 14:30.01 | shido6 | hrmm |
| 14:30.04 | shido6 | windows mobile |
| 14:30.18 | shido6 | there is a half duplex one jer was skrewing with a while back at one of the 'cons |
| 14:30.36 | shido6 | theres a few for sip but dunno about IAX. |
| 14:31.10 | shido6 | wengo and sjphone work, I use those. |
| 14:31.28 | shido6 | on the omap 201mhz |
| 14:32.08 | *** join/#asterisk lorinc (n=ang@pool-2000.adsl.interware.hu) |
| 14:32.18 | Remowylliams | shido6: I made IAX the preferred protocol because it slips pretty easily through most firewalls and nats. |
| 14:33.40 | MrChicken | Another question :$ |
| 14:33.50 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 14:33.57 | MrChicken | I just noticed a warning that says unable to create RTP session |
| 14:34.01 | MrChicken | too many open files |
| 14:34.04 | MrChicken | what does that mean? |
| 14:34.06 | Remowylliams | Welcome back [TK]D-Fender |
| 14:35.06 | sci_05 | hia [TK]D-Fender |
| 14:35.11 | *** join/#asterisk Cyber-Dogg (i=Cyber-Do@24-178-240-97.dhcp.stls.mo.charter.com) |
| 14:36.12 | MrChicken | I just noticed a warning that says unable to create RTP session... it says too many open files |
| 14:36.23 | MrChicken | ulimit is unlimited |
| 14:36.26 | rob0 | MrChicken: look in your shell's documentation for "ulimit". But something might be wrong in that files aren't being closed. |
| 14:36.30 | rob0 | ah |
| 14:36.53 | MrChicken | rob0 ... so, how come files arent beeing closed? |
| 14:36.55 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 14:37.05 | *** join/#asterisk Skymarshal (n=Skymarsc@O644c.o.pppool.de) |
| 14:37.09 | anonymouz666 | anyone in here uses two digium cards in the same machine? |
| 14:37.14 | anonymouz666 | any irq problems? |
| 14:37.27 | galeras | yep |
| 14:37.36 | Deeewayne | yes, 2 cards, no irq problems |
| 14:37.37 | anonymouz666 | yep what? |
| 14:37.39 | MrChicken | MrChicken ... I mean how can I find out> |
| 14:37.41 | MrChicken | ack |
| 14:37.50 | anonymouz666 | Deeewayne: what CPU? |
| 14:37.55 | MrChicken | rob0 ... I mean... how can I find out? |
| 14:37.59 | Skymarshal | Where can I find a documentation about the i extension? Can it be used with Background() only or are there other scenarios where one could use it? |
| 14:38.22 | [TK]D-Fender | Skymarshal: Thats what its for. |
| 14:38.32 | anonymouz666 | dual core? |
| 14:38.38 | Skymarshal | Yes, but where is the documentation for it? |
| 14:38.43 | [TK]D-Fender | ~book |
| 14:38.44 | jbot | from memory, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 14:38.45 | Deeewayne | Intel Pentium 4 - 3 GHz |
| 14:38.45 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^ |
| 14:38.56 | irule | http://www.voip-info.org/wiki/index.php?page=Asterisk+i+extension |
| 14:39.03 | Skymarshal | jbot: That is not a Digium documentation. |
| 14:39.03 | jbot | Skymarshal: what are you talking about? |
| 14:39.09 | anonymouz666 | Deeewayne: what cards? |
| 14:39.11 | Skymarshal | irule: same problem. |
| 14:39.15 | [TK]D-Fender | Skymarshal: And its not for use with "background", so much as IVR's. Background is a TOOL sometimes use by them. |
| 14:39.30 | anonymouz666 | Deeewayne: I am thinking in use a TDM card and TE card |
| 14:39.33 | [TK]D-Fender | Skymarshal: Its FINE. Go READ |
| 14:39.34 | Deeewayne | anonymouz666, I work at Digium, I've used digital and analog |
| 14:39.51 | anonymouz666 | ok |
| 14:39.59 | Skymarshal | TKD-Fender: I know this book. I just want to get an official source. |
| 14:40.03 | anonymouz666 | thanks |
| 14:40.06 | rob0 | MrChicken, you have to do some detective work. Tools like netstat and lsof might help. Also check "ulimit -a". |
| 14:40.13 | rob0 | ha |
| 14:40.16 | Deeewayne | anonymouz666, have you tried it, or are you asking in advance? |
| 14:40.32 | rob0 | this channel moves too fast for me |
| 14:40.33 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 14:40.41 | [TK]D-Fender | Skymarshal: There is no "master source" Del with it. |
| 14:40.42 | *** join/#asterisk angryuser (n=Miranda@df01t2-212-194-224-165.d4.club-internet.fr) |
| 14:40.43 | irule | Skymarshal here is your digium official documentation http://www.voipsupply.com/product_info.php?products_id=833&utm_medium=shoppingengine&utm_source=googlebase&ref=froogle |
| 14:40.56 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:41.00 | irule | only 895.50 lol |
| 14:41.17 | angryuser | hello i am looking for a script which restarts * in case of crsh |
| 14:41.17 | [TK]D-Fender | Skymarshal: The BOOK was written by some people now working for Digium, and others who've simply been using it for YEARS. Please don't discount this. |
| 14:41.31 | anonymouz666 | Deeewayne: I am consider in use two cards in the same machine... but I am afraid to do that and have IRQ problems. I just don't want pay to know if it works fine. So I want to hear the experience of the people in here |
| 14:41.49 | [TK]D-Fender | irule: Cute..... |
| 14:41.57 | *** join/#asterisk slmnhq (n=salmanh@denali.asti-usa.com) |
| 14:42.16 | irule | Skymarshal use the _book_ luke! |
| 14:42.47 | Deeewayne | irule: that's not official digium documentation, that's a blurb about business edition |
| 14:42.48 | Skymarshal | TKD-Fender: You don't get the point. I KNOW how the I extension works. But I want to get some official statement. Does it only work with Background()? ... ah... forget it! |
| 14:43.00 | irule | no cute manual? |
| 14:43.11 | irule | gees! |
| 14:43.22 | Deeewayne | is there a manual on that page? I didn't see a link |
| 14:43.58 | irule | its not a blurb, look at the picture in the page, they want you to PAY for it |
| 14:44.07 | [TK]D-Fender | Skymarshal: it has NOTHING to do with BACKGROUND, and everything to do with IVR'S. You do not need "background" for an IVR. |
| 14:44.47 | irule | weird, I used Background a lot in my ivr :s |
| 14:45.01 | [TK]D-Fender | irule: You CHOOSE to. It is by no means NECESSARY |
| 14:45.01 | Deeewayne | the link you sent is a blurb about Business Edition. The picture on that page shows a business edition manual, but your link is not official digium documentation |
| 14:45.05 | Skymarshal | TKD-Fender: So if it has nothing to do with Background() -> How is it invoked? What is the exact mechanism? |
| 14:45.17 | javb | im tying to pick up a call comming from a zap chan, says error 603, but i can pick up a call comming from an inside exten.. |
| 14:45.20 | javb | any ideaS? |
| 14:45.26 | vlt|afk | Strom_M: I removed the "r" flag from Dial() and now can hear a busy tone when dialling the busy Zap/g1 extension but * doesn't move on to the next priority. *CLI just says "-- Called g1/10" until the timeout is reached. Why doesn't it recognize the BUSY state? Do I need an additional BusyDetect option activated? |
| 14:45.26 | irule | Background(press @ss to @ck yourself) |
| 14:45.38 | irule | lame joke |
| 14:45.46 | [TK]D-Fender | Skymarshal: It is invoked when running an IVR off the "s" exten and running out of "s" to wait for input, or through WaitExten. |
| 14:50.11 | *** join/#asterisk python_ (n=tim@66-191-97-163.static.eucl.wi.charter.com) |
| 14:50.16 | python_ | hello |
| 14:53.11 | [TK]D-Fender | What a sad anal-retentive little man.... |
| 14:57.47 | galeras | is possible to use a 262 ft T1 Crossover Cable? |
| 14:58.29 | coppice | no, but a 261' should be OK, if you stretch it a bit |
| 14:58.40 | *** join/#asterisk gardo (n=gardo@121.97.211.162) |
| 14:58.50 | galeras | i can't |
| 15:00.25 | coppice | from the speed of that reponse, I bet you didn't even try. |
| 15:01.12 | galeras | I want to hear the experience of the people in here, before to try |
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| 15:04.55 | galeras | according zaptel.conf 655 feet is possible, i will give a try |
| 15:07.08 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-a728a28792c373e1) |
| 15:07.24 | Uatec | Help! |
| 15:07.31 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:07.31 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 15:07.32 | python_ | i am getting username/auth name missmatch from my Mitel 5212 to my Asterisk, as far as I can see they both have the same username and password |
| 15:07.34 | Uatec | I need to think of a birthday message for my boss... |
| 15:07.39 | mosty | anyone familiar with and willing to help me with hylafax? or point me at an irc channel or something? i'm trying to figure out how to log details on each send fax to a database instead of a file (would prefer not to dial through asterisk just to get CDR entries) |
| 15:08.44 | redax | hi, |
| 15:09.07 | redax | is the digium b410p card a stable one? |
| 15:09.28 | rob0 | You want phones for horses? |
| 15:09.54 | mosty | redax, i've installed a few, they seem ok |
| 15:10.01 | coppice | cf stable manure |
| 15:10.20 | redax | no, I want to replace a beronet card |
| 15:10.49 | redax | I've a serious rport_data overflow... |
| 15:11.15 | redax | but seems like the b410p is very like the same to tthe beronet cards |
| 15:11.36 | mosty | you need to use chan_misdn for the b410p |
| 15:11.42 | *** join/#asterisk __DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 15:11.49 | redax | that's a bad omen |
| 15:12.18 | *** join/#asterisk SwK (n=SwK@dhcp64-134-34-248.bwic.chi.wayport.net) |
| 15:12.20 | redax | I thin I found a bug in the chan_misdn + app_queue |
| 15:12.45 | redax | so I dropped out another $700 out of the window |
| 15:12.52 | mosty | i have an install with chan_misdn + app_queue |
| 15:13.01 | mosty | running for a year or more |
| 15:13.20 | redax | I changed my misd sys to another (more expensive) misdn sys |
| 15:13.23 | redax | great |
| 15:13.23 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 15:14.23 | mosty | not sure what you mean by that |
| 15:14.37 | redax | I triggger an rport_data overflow bug |
| 15:14.52 | redax | after a period of time the pbx stop workig |
| 15:14.59 | redax | workig... |
| 15:15.22 | redax | and my kern.log is full of rport_data overflow |
| 15:16.02 | redax | after an asterisk restart, everything works again |
| 15:16.15 | *** join/#asterisk SoRcS (n=penol@host-81-191-132-224.bluecom.no) |
| 15:16.17 | redax | so it's a chan_misdn bug |
| 15:17.21 | *** join/#asterisk mountainm2k (n=mountain@165.236.183.1) |
| 15:17.49 | redax | mosty: what hardware are you use that environment? |
| 15:17.58 | redax | intel vs amd ? |
| 15:18.23 | mosty | intel |
| 15:18.38 | redax | does it matter at all? |
| 15:18.47 | mosty | does what matter? |
| 15:19.59 | mountainm2k | Is there a quick-easy way to discover how many channels of a PRI are in use? zap show channels ? |
| 15:20.09 | mountainm2k | Want to monitor it with Nagios :-) |
| 15:21.45 | mosty | yes, zap show channels, or an AMI equivalent |
| 15:22.03 | mountainm2k | ami? |
| 15:22.06 | mocker | mountainm2k: /usr/sbin/asterisk -rnx "zap show channels" |
| 15:22.11 | mocker | Or something like that. |
| 15:22.26 | mosty | mountainm2k, lookup AMI on voip-info.org |
| 15:22.58 | *** join/#asterisk phez72 (i=redcap@xs3.xs4all.nl) |
| 15:24.11 | SuPrSluG | show channels show channels in use. zap show channels shows configured channels |
| 15:24.30 | redax | well. |
| 15:25.04 | redax | so what happens if the card doing the very same bug ? |
| 15:25.12 | redax | shall I send back ? |
| 15:27.09 | python_ | i am getting username/auth name missmatch from my Mitel 5212 to my Asterisk, as far as I can see they both have the same username and password |
| 15:27.09 | mosty | redax, have you submitted a bug report for asterisk? |
| 15:27.40 | *** join/#asterisk oej_ (n=olle@62.97.243.70) |
| 15:27.52 | galeras | python restart * |
| 15:28.15 | *** join/#asterisk krondorl (n=chatzill@acid.auricnet.ca) |
| 15:28.19 | python_ | we have |
| 15:28.20 | *** join/#asterisk rogerius (n=mama@201.29.18.64) |
| 15:28.21 | krondorl | Hello all |
| 15:28.25 | *** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com) |
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| 15:30.52 | redax | mosty: yep... |
| 15:31.26 | mosty | redax, and what was the response? |
| 15:32.08 | redax | it's similar to others prob. and no solutios yet. |
| 15:32.10 | SuPrSluG | droppin calls channel.c: Bridge stops bridging channels Zap/1-1 and SIP/125. What's going on? |
| 15:33.12 | krondorl | Is there a url that might help me in regards to being able to insert or extract information from a string?? |
| 15:33.18 | *** join/#asterisk twitchnln (n=Administ@70.43.112.117.nw.nuvox.net) |
| 15:33.27 | twitchnln | good morning everyone |
| 15:34.24 | mosty | krondorl, from a dialplan variable? |
| 15:34.29 | krondorl | yes |
| 15:34.45 | twitchnln | i am attempting to setup an autoattendant that will change the cid based on selection before dropping caller into queue, how would i go about this? |
| 15:34.54 | redax | mosty: where can I buy commercial support I fail with this? |
| 15:35.42 | javb | wow! i cant no more, anybody give me a clue! |
| 15:36.50 | krondorl | mosty, what I am trying to do is have a variable that is saved in the DB of phones that ring when a call comes in. I want to be able to have an extension dial a number to either add or remove themselves from that dialing varialble |
| 15:37.01 | mosty | redax, you can get support from digium i think. btw what version of asterisk are you using, and how did you install misdn? |
| 15:37.29 | mosty | krondorl, sounds like you should look at queues |
| 15:38.04 | jkiff | More specifically, agents in queues. |
| 15:38.25 | jkiff | Well, "in" may not be the right preposition. |
| 15:38.27 | krondorl | mosty, Aren't queues for queuing up incoming calls?? |
| 15:38.57 | mosty | krondorl, isn't that what you are trying to do? |
| 15:39.20 | krondorl | mosty nope. at this point in time, when a call comes in ALL the phones in the office ring. |
| 15:39.23 | *** join/#asterisk ivanfm (n=ivanfm@c934f322.virtua.com.br) |
| 15:39.29 | *** join/#asterisk andresmujica (n=andresmu@200.75.38.138) |
| 15:39.35 | mosty | krondorl, what are you trying to do then? |
| 15:39.36 | redax | mosty: ast 1.2.18 |
| 15:39.51 | krondorl | If someone in the office does not want to be disturbed, he wants to be removed form the list of phone thatring. |
| 15:39.51 | redax | and installed misd via the beroet installer |
| 15:40.01 | jkiff | krondorl: You can use queues to ring groups of phones. |
| 15:40.38 | krondorl | jkiff, then I would need 15*15 groups for all the different combinations! |
| 15:40.44 | jkiff | krondorl: And if a member of the queue is an agent, he/she can jump in and out of the queue. |
| 15:40.46 | jkiff | ;) |
| 15:41.25 | mosty | krondorl, huh? you could have one queue for all incoming calls. your internal extensions would then join/leave the queue as agents as they prefer |
| 15:41.41 | *** join/#asterisk Foxygnu (n=FoX@2001:41d0:1:44c8:cafe:cafe:cafe:42) |
| 15:42.08 | krondorl | DOH! Ok, that totally slipped my mind... Thanks... |
| 15:42.42 | Zaggynl^Laptop | Is there a manual on how to set up voice menu's? |
| 15:42.50 | *** join/#asterisk woolbeo (n=woolbeo@exchange.services.daqe.com) |
| 15:42.53 | krondorl | <--- Off to figure out queue and agents... |
| 15:43.20 | mosty | Zaggynl^Laptop, try the AFOT book |
| 15:43.21 | mosty | ~book |
| 15:43.22 | jbot | book is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:43.30 | Zaggynl^Laptop | cheers |
| 15:47.22 | Zaggynl^Laptop | excellent read |
| 15:48.05 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
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| 15:51.10 | dlynes_laptop | CunningPike: good morning |
| 15:52.08 | [TK]D-Fender | krondorl: Rather easy to set up some basic dialplan scripting to see who wants to be included to build up the dial string 1 at a time. |
| 15:52.12 | CunningPike | Hey, dlynes_laptop! |
| 15:53.14 | dlynes_laptop | CunningPike: finally getting around to trying out asterisk 1.4 |
| 15:53.23 | dlynes_laptop | CunningPike: finally got zaptel 1.4 working with wanpipe |
| 15:53.28 | skymeyer | evening, does anyone know a sip client which supports tcp ? (a demo version will if no free soft exists for it) |
| 15:53.37 | CunningPike | dlynes_laptop: Cool - we tried an upgrade, but went back to 1.2 |
| 15:53.44 | CunningPike | dlynes_laptop: We had PRI issues |
| 15:53.57 | dlynes_laptop | CunningPike: what happened? thought you said 1.4 was quite stable? |
| 15:54.30 | CunningPike | dlynes_laptop: Well, it was - until we tried transfering inbound zap calls - then it would crash |
| 15:55.07 | [TK]D-Fender | krondorl: Queues are an option, but a bit icky for my tastes. |
| 15:55.08 | CunningPike | So, my boss got the heebies and we're back on 1.2 :) |
| 15:55.08 | dlynes_laptop | CunningPike: transferring them from the pri card to an fxs port on an analog card? |
| 15:55.16 | *** join/#asterisk ThoMe (n=tm@tm.muc.de) |
| 15:55.18 | ThoMe | hello |
| 15:55.20 | ThoMe | :-) |
| 15:55.36 | mosty | queues do suck |
| 15:55.38 | ThoMe | who used snom? :-) |
| 15:55.39 | CunningPike | dlynes_laptop: No - SIP phone answers incoming zap call and tries to transfer to another SIP phone. |
| 15:55.42 | mosty | thome, i have |
| 15:55.48 | ThoMe | mosty: ah, which? |
| 15:55.53 | ThoMe | version |
| 15:55.57 | ThoMe | 360? |
| 15:55.58 | mosty | 300, 320, 360 |
| 15:55.59 | *** join/#asterisk troy- (n=troy@toroon12-1177845255.sdsl.bell.ca) |
| 15:56.02 | dlynes_laptop | CunningPike: oh...you'd think that would be the most basic of things that would have been tested |
| 15:56.04 | ThoMe | hmm. ok. i have 370 |
| 15:56.38 | ThoMe | mosty: how long is maximal a callername if i recieve a call? |
| 15:56.40 | CunningPike | dlynes_laptop: Yup - and our test plan was at fault, too |
| 15:56.42 | ThoMe | length ? |
| 15:56.44 | dlynes_laptop | CunningPike: and I guess you don't have a spare pri to play with |
| 15:56.54 | CunningPike | dlynes_laptop: Actually, we do |
| 15:57.08 | mosty | thome, i do not know |
| 15:57.09 | dlynes_laptop | CunningPike: ah...I guess you just didn't test it well enough |
| 15:58.02 | CunningPike | dlynes_laptop: We did test it - the issue was that the card was different in the test server - TE110P as opposed to the TE410P in the production server |
| 15:58.02 | CunningPike | dlynes_laptop: Seemed to make a difference |
| 15:58.02 | troy- | " No private structure for packet? |
| 15:58.02 | troy- | " I keep getting this error on console |
| 15:58.18 | CunningPike | dlynes_laptop: We now have a test server that's identical to the production one in every way |
| 15:58.35 | CunningPike | dlynes_laptop: We bought another TE410P for the test PRI |
| 15:58.38 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
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| 15:58.44 | troy- | any reason why chan_iax would be error "no private structure for packet"? |
| 15:58.50 | CunningPike | Netsplit, anyone? |
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| 16:00.45 | troy- | any reason why chan_iax would be error "no private structure for packet"? |
| 16:02.07 | SuPrSluG | what is a good setting for busycount= ? I'm getting dropped calls |
| 16:02.55 | festr__ | SuPrSluG: 6 |
| 16:02.59 | festr__ | SuPrSluG: min 4 |
| 16:03.12 | festr__ | SuPrSluG: what do you have now? |
| 16:03.15 | SuPrSluG | thanks I'll see if that helps |
| 16:03.19 | krondorl | [TK]D-Fender: Ya, I'm trying to do it as simple as possible |
| 16:03.28 | SuPrSluG | no setting. so default |
| 16:05.57 | SuPrSluG | festr_: i was also getting Bridge stops bridging channels . Normal? |
| 16:12.46 | ThoMe | how long is posible (length) the caller name? |
| 16:13.04 | ThoMe | or how i can send a text to a phone if a incomming call? |
| 16:13.47 | ThoMe | any ideas? |
| 16:14.12 | *** join/#asterisk seele_ (n=seele@webserver.datawareltda.com) |
| 16:15.23 | mosty | set the callerid string, if you want |
| 16:15.30 | *** join/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 16:15.38 | mosty | it's probably limited to whatever will fit on the screen of your particular phone |
| 16:16.02 | ThoMe | hmm. ok |
| 16:16.14 | ThoMe | i think the 370er iss better as 360 |
| 16:16.22 | ThoMe | but he make no \n |
| 16:16.43 | Qwell[] | ThoMe: Please don't ping the channel like that |
| 16:16.55 | ThoMe | Qwell[]: sorry please |
| 16:18.33 | *** part/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 16:19.52 | mosty | ThoMe, i would imagine that callerid was designed for single line displays, hence no newlines |
| 16:20.16 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 16:20.21 | ThoMe | mosty: hmm ? |
| 16:20.47 | mosty | huh? |
| 16:20.51 | ThoMe | hm ok |
| 16:20.51 | seele_ | easy way to increase the voicemail attachment volume ?? without change the format order in the voicemail.conf |
| 16:21.32 | twitchnln | is there a way to setup a screen popup with cid info using like msn messenger or something? |
| 16:22.01 | mosty | twitchnln, write an AGI script |
| 16:22.16 | *** join/#asterisk ivanfm (n=ivanfm@c934f322.virtua.com.br) |
| 16:22.45 | [TK]D-Fender | seele_: Yes, go read the sample config |
| 16:23.29 | seele_ | what sample config?? |
| 16:23.41 | [TK]D-Fender | seele_: voicemail.conf |
| 16:24.21 | vn | is it possible to redirect asterisk calls to some msn messenger user that will contact my own user via voice conversation? that'd be neat |
| 16:25.31 | [TK]D-Fender | vn: Go install a soft phone |
| 16:26.04 | vn | soft phone being a computer-based phone? |
| 16:26.21 | *** join/#asterisk casimir (n=casimir@rrcs-71-43-154-55.se.biz.rr.com) |
| 16:26.32 | mosty | some instant messaging clients support sip |
| 16:26.41 | *** join/#asterisk magic_hat (n=geoffdou@h-74-2-87-16.chcgilgm.covad.net) |
| 16:26.54 | vn | k |
| 16:27.47 | magic_hat | hey everyone. My * is having probs locating and playing a greeting file. The greeting file does exist in /var/lib/asterisk/sounds. So I'm thinking this is maybe a permissions problem. I set perms to 777, but still no dice. any ideas on where to go from here? |
| 16:29.05 | seele_ | dovolgain=yes and volgain=<value> works ??? |
| 16:29.09 | [TK]D-Fender | magic_hat: Like all other problems you have shown us nothing. Pastbin is your friend |
| 16:29.11 | [TK]D-Fender | ~pb |
| 16:29.12 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 16:29.14 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 16:29.27 | *** part/#asterisk Remowylliams (n=Mare@71.16.217.178) |
| 16:29.30 | [TK]D-Fender | seele_: No, they don't work. It is LYING TO YOU. |
| 16:29.50 | [TK]D-Fender | </sarcasm> |
| 16:29.53 | magic_hat | TKD: what do you need to see? The error is saying that my audio file wasn't found. |
| 16:30.17 | [TK]D-Fender | magic_hat: I want to see the CLI output of the attempt at verbose 10, and proof that the file is there. |
| 16:30.30 | magic_hat | k hang on |
| 16:31.29 | magic_hat | lol now it's refusing my ssh connection. hang on. |
| 16:32.02 | *** join/#asterisk Lawbringer (n=Lawbring@84-45-215-247.no-dns-yet.enta.net) |
| 16:33.54 | *** join/#asterisk SwK (n=SwK@dhcp64-134-34-248.bwic.chi.wayport.net) |
| 16:34.34 | *** join/#asterisk fbffff (n=fbffff@dhcp64-134-34-248.bwic.chi.wayport.net) |
| 16:36.45 | seele_ | how can I change the blind transfer .... by default in my phones with attended transfer using the same key? |
| 16:37.36 | mvanbaak | look at the phones manual |
| 16:38.01 | *** part/#asterisk twitchnln (n=Administ@70.43.112.117.nw.nuvox.net) |
| 16:38.16 | seele_ | in other worlds how can I change the default transfer with attendant transfer?? |
| 16:39.13 | krdian | seele_: features.conf |
| 16:39.29 | seele_ | whats sends any phone when I press the transfer key ?? |
| 16:40.05 | krondorl | Ok, I know there is a default global variable for callerid and one for the number a phone is calling (exten), but is thee one for the extension that is making the call?? |
| 16:40.12 | *** join/#asterisk Jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net) |
| 16:40.31 | [TK]D-Fender | seele_: Read The Phone's Manual |
| 16:40.40 | krdian | seele_: oh, on phone side... |
| 16:41.16 | *** join/#asterisk sharp (n=sharp@c-68-81-156-176.hsd1.pa.comcast.net) |
| 16:43.39 | tzanger | asterisk's agent/queue system is capable of having one agent serve in multiple queues, right? |
| 16:44.16 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 16:44.16 | krdian | tzanger: correct |
| 16:45.10 | blitzrage | tzanger: yes sir |
| 16:45.32 | blitzrage | tzanger: and you can use weights in the Queue to determine which queue gets priority to a free agent |
| 16:45.45 | blitzrage | (theoretically... I've not actually tested it :)) |
| 16:48.25 | tzanger | heh |
| 16:48.27 | tzanger | thanks guys |
| 16:52.08 | coppice | you have to get the flour, and the water. find a pot and a mixing stick. it all takes times |
| 16:54.21 | krondorl | NVM, found what I was looking for. |
| 16:55.33 | ThoMe | mosty: here? |
| 16:56.53 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 16:57.22 | *** join/#asterisk VJFROMGT (n=vjfromgt@user-387g9ui.cable.mindspring.com) |
| 16:57.42 | VJFROMGT | is there a way to limit the number of calls per route (not per trunk) |
| 16:59.10 | blitzrage | VJFROMGT: use GROUP() and GROUP_COUNT() functions |
| 16:59.33 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 16:59.59 | VJFROMGT | bliz,, tell me more please |
| 17:01.52 | [TK]D-Fender | VJFROMGT: He just gave you the function names. Get off your ass and read the INSTRUCTIONS. |
| 17:02.05 | *** join/#asterisk cayorde (n=flexable@host121-81-dynamic.17-87-r.retail.telecomitalia.it) |
| 17:02.38 | blitzrage | VJFROMGT: just look at the GROUP() and GROUP_COUNT() functions -- you can use it to count the number of channels in use for whatever you want to name it (and you can categorize them as well). So for example in a vPBX environment, you might have something like Set(GROUP(my_pbx)=lmadsen) to track the amount of calls for the lmadsen user in the my_pbx vPBX, then you can get the number of channels in use by doing Verbose( |
| 17:02.38 | blitzrage | 1|${GROUP_COUNT(lmadsen@my_pbx)}) |
| 17:02.47 | blitzrage | so take that info, and apply it to your problem. |
| 17:02.52 | VJFROMGT | thanks |
| 17:04.55 | *** join/#asterisk dsq (n=foxtrot@200.204.142.181) |
| 17:05.26 | dsq | hey,ive just heard that unicall CANT get why the connection wsa lost(if it was fax, wrong number etc) and zaptel can, is it true? |
| 17:12.08 | *** join/#asterisk scud (n=scud@c-68-62-219-34.hsd1.al.comcast.net) |
| 17:12.08 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 17:12.43 | scud | I heard angler eats poo |
| 17:12.52 | *** join/#asterisk Marshall-Laptop (n=eman0n@cpe-76-181-165-37.columbus.res.rr.com) |
| 17:13.08 | linex | anyone using asterisk on gentoo ? |
| 17:13.36 | [TK]D-Fender | linex: No. |
| 17:13.59 | rob0 | It's not allowed! |
| 17:14.29 | galeras | linex: i believe gnudialer people does see #gnudialer |
| 17:14.44 | linex | I don't understand |
| 17:15.45 | angler | scud, yea it's good |
| 17:16.08 | scud | linex: it is a result of the gpl v3 of v1 and lgp license |
| 17:16.34 | scud | linex: the incompatible of versions is the issue |
| 17:16.47 | linex | scud: why telling me this |
| 17:16.49 | scud | v3 is not backwards compatible with v1 |
| 17:17.13 | scud | so gentoo is on 2007.1 so .1 can be read by v3 |
| 17:17.24 | scud | linex: so you know stuff |
| 17:17.36 | linex | ok thanks |
| 17:17.57 | linex | ah now I see |
| 17:18.19 | linex | gen2 is on v1 and asterisk on v3 |
| 17:18.20 | linex | ? |
| 17:18.31 | rootfield | i'm upgrade asterisk 1.2.6 to 1.2.19.. but now when the call is originate asterisk display this error: Timeout, but no rule 't' in context 'default'.... any ideas ? |
| 17:18.33 | linex | is that what you'll telling me ? |
| 17:18.36 | scud | linex: yeah, they wont work with each other |
| 17:18.54 | scud | linex: me is telling you this for you questions |
| 17:19.04 | linex | but there is a wiki on how to asterisk on gentoo |
| 17:19.23 | linex | http://www.voip-info.org/wiki-Asterisk+Linux+Gentoo |
| 17:19.25 | scud | that is only in theory if they were compatible |
| 17:19.38 | scud | its in alpha beta form right now |
| 17:19.48 | linex | and |
| 17:19.50 | [TK]D-Fender | rootfield: This has nothing to do with your upgrade. Your dialplan is faulty. |
| 17:19.52 | linex | http://gentoo-wiki.com/HOWTO_Asterisk |
| 17:19.52 | scud | you know like .001.02.v3 |
| 17:20.11 | *** join/#asterisk SwK (n=SwK@dhcp64-134-34-248.bwic.chi.wayport.net) |
| 17:20.18 | linex | I its best to just use asterisk now |
| 17:20.22 | linex | yes ? |
| 17:20.28 | scud | asterisk from a can? |
| 17:20.29 | scud | yes |
| 17:20.31 | [TK]D-Fender | rootfield: And it tells you right to your face the exten its expecting to find. Go see why the call is in there in the first place and why it is that you did not arrange a "t" exten for it. |
| 17:20.39 | linex | asterisknow is livecd , right ? |
| 17:20.46 | [TK]D-Fender | linex: No. |
| 17:20.50 | *** part/#asterisk python_ (n=tim@66-191-97-163.static.eucl.wi.charter.com) |
| 17:20.57 | *** join/#asterisk jm|laptop (n=jm|home@cpc1-papw3-0-0-cust17.cmbg.cable.ntl.com) |
| 17:20.58 | linex | no livecd version ? |
| 17:20.59 | [TK]D-Fender | linex: It is a full distro that will wpie your drive and install |
| 17:21.03 | *** join/#asterisk bcnl (n=mike@S010600131078957c.vc.shawcable.net) |
| 17:21.08 | linex | oh no |
| 17:21.20 | linex | any livecd version of asterisk ? |
| 17:21.26 | scud | linex: that is because of the US license agreement with asterisk |
| 17:21.26 | [TK]D-Fender | linex: Install a normal distro and compile from source like the rest of us. |
| 17:21.29 | *** part/#asterisk jm|laptop (n=jm|home@cpc1-papw3-0-0-cust17.cmbg.cable.ntl.com) |
| 17:21.33 | scud | linex: it wipes all drives |
| 17:21.51 | linex | how about on centos ? |
| 17:21.55 | linex | centos 5 |
| 17:22.12 | linex | I have a clean new install on centos 5 doing nothing |
| 17:22.38 | [TK]D-Fender | linex: tahts fine, I'd suggest 4.x personally just knowing that 5 has a few things to workaround last I heard |
| 17:22.50 | linex | but all that nice interface on asterisknow, I'll mniss that |
| 17:23.19 | linex | I just have keep my fingers cross on centos 5. |
| 17:23.47 | linex | so no asterisk no rpms |
| 17:25.35 | linex | what am I saying. |
| 17:25.39 | linex | ok thanks |
| 17:26.56 | ThoMe | how i can use by my snom phone a phonebook with xml ? |
| 17:29.27 | [TK]D-Fender | ThoMe: Go read its manual and find out. |
| 17:29.56 | bcnl | besides the wiki, are there any good docs/faqs/tutorials on queues? |
| 17:30.15 | [TK]D-Fender | bcnl: the sample config file. |
| 17:30.39 | [TK]D-Fender | bcnl: And the WIKI itself links to other guides. |
| 17:31.08 | bcnl | I might have been on crack, but I thought there was a option to have a agent confirm that they could pickup the call by pressing a digit |
| 17:31.17 | bcnl | but I see no mention of it in the wiki/config |
| 17:31.49 | ThoMe | [TK]D-Fender: hmm. i don't find the notices : |
| 17:32.47 | [TK]D-Fender | bcnl: 2 ways : a) use AgentLogin (FUGLY), or b) in the dialplan that calls the agent, use the Macro option for screening. |
| 17:33.11 | [TK]D-Fender | bcnl: Neither of these preclude the possibility that you are on crack however ;) |
| 17:37.20 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 17:38.06 | *** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@modemcable159.131-56-74.mc.videotron.ca) |
| 17:38.25 | *** join/#asterisk rdb_ (n=rdb@gw.avila.edu) |
| 17:40.07 | *** join/#asterisk ramindia_ (n=ramindia@202.63.96.9) |
| 17:40.14 | ramindia_ | hi any help |
| 17:40.20 | ramindia_ | iam getting this errror |
| 17:40.36 | ramindia_ | Jun 28 23:07:31 WARNING[2692]: channel.c:785 channel_find_locked: Avoided deadlock for '0x864e70', 10 retries! |
| 17:44.05 | *** join/#asterisk mykryen (i=mykryen@gw.percipia.com) |
| 17:44.38 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 17:46.20 | *** join/#asterisk AllanLima (n=mediain@unaffiliated/allanlima) |
| 17:47.23 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-631cec18267f61c0) |
| 17:47.23 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 17:49.45 | *** join/#asterisk SwK (n=SwK@dhcp64-134-34-248.bwic.chi.wayport.net) |
| 17:52.19 | *** join/#asterisk fbffff (n=fbffff@dhcp64-134-34-248.bwic.chi.wayport.net) |
| 17:52.41 | *** join/#asterisk ivanfm (n=ivanfm@c934f322.virtua.com.br) |
| 17:52.45 | *** join/#asterisk SwK (n=SwK@dhcp64-134-34-248.bwic.chi.wayport.net) |
| 17:52.54 | *** join/#asterisk Ch0Hag (n=mking@knight.monnsta.net) |
| 17:53.40 | Ch0Hag | I can't get a dialtone on an analog phone. |
| 17:54.14 | irule | why is 1 executed when I dial 321, when only exten = 1 exists? |
| 17:55.23 | waKKu | someone have the link for digium PCI cards and PCI voltages definitions ? |
| 17:55.31 | [TK]D-Fender | irule: Depends on everything else you are doing. |
| 17:55.49 | [TK]D-Fender | waKKu: www.digium.com |
| 17:55.57 | [TK]D-Fender | Ch0Hag: Because. |
| 17:56.09 | Ch0Hag | I don't know. |
| 17:56.12 | [TK]D-Fender | NEXT!@!@!@ (c) BKW |
| 17:56.13 | Ch0Hag | I'm hoping to find out. |
| 17:56.20 | ramindia_ | Jun 28 23:07:31 WARNING[2692]: channel.c:785 channel_find_locked: Avoided deadlock for '0x864e70', 10 retries! |
| 17:56.37 | [TK]D-Fender | ramindia_: We heard you the first time, stop repeating yourself |
| 17:57.06 | Ch0Hag | To start with, a page saying something along the lines of 'this is how you set up an analog phone/FXS module' would be nice. |
| 17:57.23 | waKKu | yeah.. i'm there. but cant found |
| 17:57.34 | Ch0Hag | Although based on the sample configurations google has thrown up, min seems to be correct. |
| 17:58.36 | Strom_M | Ch0Hag: pastebin your config and i'll help you or [TK]D-Fender will yell at you and then help |
| 17:59.18 | [TK]D-Fender | Ch0Hag: And in the next 4 lines you spoke following your original quesiotn you have failed to state ANY of the hardware you're using or showing us your configs for whatever it is. |
| 17:59.23 | [TK]D-Fender | Strom_M: TIMING! |
| 17:59.42 | ramindia_ | i dont see any answer for the same, so force to repeat that |
| 17:59.54 | [TK]D-Fender | Strom_M: I usuallly only yell if I don't get ti, not FOLLOWING :) |
| 18:00.03 | Strom_M | Bonerville's hit single "Forever Dong" |
| 18:00.06 | Ch0Hag | Meh. He doesn't scare me. |
| 18:00.15 | [TK]D-Fender | ramindia_: If we knew, we'd have ANSWERED it the first time. |
| 18:00.20 | Ch0Hag | A pastebin site which responded would be a plus. |
| 18:00.25 | ramindia_ | hmm |
| 18:00.26 | Strom_M | pastebin.ca |
| 18:00.36 | [TK]D-Fender | ~pb |
| 18:00.37 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 18:00.38 | Ch0Hag | That one fails the 'responding' test. |
| 18:00.40 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
| 18:01.05 | *** part/#asterisk mountainm2k (n=mountain@165.236.183.1) |
| 18:01.13 | [TK]D-Fender | Both succeeded for me just fine |
| 18:01.16 | Ch0Hag | Ah nice. Lisp's works. |
| 18:01.23 | *** join/#asterisk kodorna (n=leonardo@201-15-172-177.paemt705.dsl.brasiltelecom.net.br) |
| 18:01.35 | kodorna | sup guys... |
| 18:01.49 | Strom_M | [TK]D-Fender: well, you know what i decided last night, right? |
| 18:02.13 | Ch0Hag | http://paste.lisp.org/display/43627 |
| 18:02.20 | kodorna | can anyone tell me whats the standard T1 kind of lines you guys use up in the USA? |
| 18:02.33 | kodorna | and where can i find any papers about it? |
| 18:02.36 | tzanger | huh? |
| 18:02.40 | tzanger | "standard t1 kind of lines" ? |
| 18:02.46 | tzanger | there's CAS T1 and T1 PRI |
| 18:03.11 | darius_ | Anyone know how I can extract and playback SIP GSM/RTP streams that are captured in a packet trace? Ethereal handles it all for g711 but I'm not seeing it work for GSM. |
| 18:03.37 | Strom_M | kodorna: what do you specifically want to know about it? |
| 18:04.21 | Ch0Hag | The phone is in channel 3 (due to a lack of rj11->bt adapter, it is connected as TDM -> ADSL filter -> phone). |
| 18:04.27 | [TK]D-Fender | Strom_M: Nope, do tell... |
| 18:04.27 | kodorna | and is there any informations about sending and receiving CID and DID in loopstart? |
| 18:04.48 | Strom_M | [TK]D-Fender: this is the jizziest channel on freenode |
| 18:04.49 | Ch0Hag | But I've tried various ways of getting rid of the filter and none has any effect. |
| 18:05.19 | Strom_M | Ch0Hag: the filter shouldn't affect things |
| 18:05.22 | *** join/#asterisk agile (n=mike@63.98.55.146) |
| 18:05.41 | Ch0Hag | Da. Though I'm told it has the ringer capacitor which should come in handy if this ever works. |
| 18:05.42 | coppice | darius_ I don't think the available tools provide for anything by G,711. They would need to know about the codec specific RTP format to extract the audio, and I don't think any available tool deals with that |
| 18:06.10 | sandorp | I just upgraded from 1.4.4 to 1.4.5 and now my zaptel card doesn't work; whe I run "module load chan_zap.so" it keeps coming back with a "channel 1: Device or resource busy" message; I've rebooted and tried using a different analog line, but it still doesn't want to work |
| 18:06.14 | Strom_M | Ch0Hag: so does it work with the DSL filter? |
| 18:06.16 | [TK]D-Fender | Ch0Hag: Do your 2 FXO ports work fine? |
| 18:06.22 | Ch0Hag | It doesn't work at all. |
| 18:06.26 | Ch0Hag | FXO are fine. |
| 18:06.41 | [TK]D-Fender | Ch0Hag: Tested both? |
| 18:06.44 | Strom_M | what if you move the fxs module to port 4? |
| 18:06.47 | Ch0Hag | Well the first is. I've not tested the 2nd as I've only one line (kept the module in for fun). |
| 18:07.06 | kodorna | so... any source of info for that? |
| 18:07.10 | [TK]D-Fender | Ch0Hag: Test it, then switching the port for your FXS isn't a bad idea. |
| 18:07.11 | Ch0Hag | ISTR both were working when in use at the company I nicked the card from. |
| 18:07.30 | *** join/#asterisk mcab (n=mb@mostly-harmless.ca) |
| 18:07.34 | Ch0Hag | (The FXS module has been bought since so is untested anywhere/when) |
| 18:07.39 | Strom_M | i thought it was Australia that was the island full of criminals |
| 18:07.45 | [TK]D-Fender | kodorna: Ditch that signalling... PRI is your friend |
| 18:08.40 | kodorna | the thing is we're building the protocol to work with asterisk |
| 18:09.01 | kodorna | and it kinda sucks the lack of info on T1 lines |
| 18:09.04 | [TK]D-Fender | Ch0Hag: Your zaptel/zapata seems fin though. |
| 18:09.06 | kodorna | about* |
| 18:09.20 | Strom_M | kodorna: why are you building a new protocol? |
| 18:09.22 | [TK]D-Fender | kodorna: Building which protocol? |
| 18:09.24 | Strom_M | just use ISDN PRI |
| 18:09.30 | coppice | I've never noticed any lack of info about T1 lines |
| 18:09.49 | sandorp | anyone know why my zaptel device would show "device busy" even when it's not? any likely culprits ? |
| 18:09.51 | kodorna | coppice: so can you tell me where to find? |
| 18:10.11 | Ch0Hag | Zap/1 and Zap/2 are both fine. |
| 18:10.18 | Ch0Hag | So try the FXS in port 2? |
| 18:10.18 | kodorna | [TK]D-Fender: loopstart |
| 18:10.24 | coppice | Google will overwhealm you with info about T1 signalling |
| 18:10.41 | Dr-Linux | Strom_M: :) |
| 18:10.47 | [TK]D-Fender | kodorna: AFAIK We already HAVE FXS_LS as a signalling option... |
| 18:11.01 | Ch0Hag | As that will mean restarting the server, does anyone know if mdadm will carry on from where it left off when rebuilding a raid? |
| 18:11.04 | [TK]D-Fender | kodorna: Though why would you WANT it? |
| 18:11.27 | Ch0Hag | It's been going for 40 minutes now already. |
| 18:11.30 | Dr-Linux | Strom_M: my cards were burned out due to discharge in PSTN lines. |
| 18:11.46 | kodorna | [TK]D-Fender: embedded, we cant use asterisk |
| 18:11.50 | Strom_M | isnt that what I said? |
| 18:12.01 | Strom_M | kodorna: there are embedded versions of asterisk |
| 18:12.18 | [TK]D-Fender | kodorna: Go buy a T1 book. |
| 18:12.27 | carrar | T1 & Sonets! |
| 18:12.35 | Dr-Linux | Strom_M: you were right, but the good thing is this RMA gonna work for me |
| 18:12.42 | carrar | err Sonets and T1 it's called |
| 18:13.06 | kodorna | ok, thanks |
| 18:13.10 | coppice | How many ways do I hate T1s? |
| 18:13.12 | coppice | Let me count the ways |
| 18:13.12 | Strom_M | kodorna: T1 fundamentals: http://www.stromcarlson.com/docs/basics/t1svcfund.pdf |
| 18:13.34 | darius_ | coppice: they have the option under rtp statistics to save a raw rtp stream (with rtpdump) and then I've read that you can use outside tools to stream and extract but I'm having no luck. |
| 18:14.24 | coppice | darius_ possibly. wireshark is so damned buggy, though, that you never know if anything will work |
| 18:14.45 | carrar | hi Darius |
| 18:14.53 | darius_ | hey carrar |
| 18:15.02 | [TK]D-Fender | coppice: What was that you said about IRC's inability to differentiate homonyms? ;) |
| 18:15.15 | darius_ | coppice: k, guess I'll keep trying :) |
| 18:15.29 | darius_ | carrar: I didn't know you used asterisk! |
| 18:15.33 | darius_ | ;) |
| 18:15.39 | carrar | Whats Asterisk? I'm just here for the chics |
| 18:15.48 | darius_ | haha |
| 18:17.07 | coppice | [TK]D-Fender: you mean like "biofuels will help the trains run on thyme"? |
| 18:17.32 | [TK]D-Fender | coppice: I can't stand much more of this.... pun-ishment :| |
| 18:17.33 | Strom_M | ah, xkcd |
| 18:17.50 | Ch0Hag | Well I'm going to wait for the raid to finish before moving the module, but in the meantime I have another problem. |
| 18:18.20 | Ch0Hag | When I call the zap line (from my mobile) the console reports that it hangs up, then picks it up again, and then finally rings my sip phone. |
| 18:18.37 | Ch0Hag | Then when I hang up the sip phone, the mobile stays online for a while longer. |
| 18:18.49 | coppice | suddenly everyone knows xkcd. i wonder what spread its name? |
| 18:18.51 | Strom_M | Ch0Hag: welcome to analog |
| 18:18.51 | Ch0Hag | If I place the call the other way everything is fine. |
| 18:19.01 | Ch0Hag | It's a long while. |
| 18:19.12 | Ch0Hag | Approaching something like a minute. |
| 18:19.20 | Strom_M | oh |
| 18:19.21 | Strom_M | hmm |
| 18:19.35 | Ch0Hag | And it doesn't happen from a regular phono plugged straight into the pots. |
| 18:20.19 | Ch0Hag | Two things of note - Zap hangs up oddly (I'll pastebin what the console said), and asterisk has to fallback to the default context because from-pstn doesn't exist (yet). |
| 18:22.01 | Ch0Hag | http://paste.lisp.org/display/43629 |
| 18:22.15 | mocker | Anyone have a good script for migrating voicemail to an ODBC database? The one on voip-info doesn't seem to work |
| 18:22.18 | Ch0Hag | I've just noticed "Error in extension logic (missing '}')" but I don't think that's anything to do with it. |
| 18:23.17 | *** join/#asterisk SwK (n=SwK@dhcp64-134-34-248.bwic.chi.wayport.net) |
| 18:24.11 | Ch0Hag | Seems the hanging up oddly is just it detecting callerid, so probably irrelevant. |
| 18:24.52 | Ch0Hag | And other than that, I can't see any reason why it wouldn't hang up the line (and every reason why it would, viz. "-- Hungup 'Zap/1-1'" at the end). |
| 18:26.02 | [TK]D-Fender | Ch0Hag: It hangs after an explicit "Hangup"? |
| 18:26.10 | Ch0Hag | Da. |
| 18:26.26 | Ch0Hag | The caller (my mobile) not the callee (snom). |
| 18:26.33 | Strom_M | dear everyone: |
| 18:26.35 | Strom_M | "called party" |
| 18:26.38 | Strom_M | "calling party" |
| 18:26.39 | Strom_M | thanks |
| 18:27.16 | Ch0Hag | mking@knight:~$ echo callee|wc -m; echo called party|wc -m |
| 18:27.16 | Ch0Hag | 7 |
| 18:27.17 | Ch0Hag | 13 |
| 18:27.31 | Ch0Hag | 13 > 7, /me == lazy |
| 18:27.49 | *** join/#asterisk guillote_GNU (n=guillote@190.7.30.135) |
| 18:28.41 | [TK]D-Fender | Ch0Hag: Poorly worded on my part... I mean Zap/1 is NOT released after the : -- Executing NoOp("Zap/1-1", "HANGUP") in new stack ? |
| 18:29.16 | Ch0Hag | It is not, no. |
| 18:29.28 | *** join/#asterisk haikumore (n=haikumor@ALagny-155-1-98-157.w83-199.abo.wanadoo.fr) |
| 18:29.37 | [TK]D-Fender | Ch0Hag: Ok, that is whacked.... maybe your whole card is screwed |
| 18:29.41 | Ch0Hag | For the record, The NoOp is from 'exten => h,1,NoOp(HANGUP)' in the dial macro. |
| 18:29.49 | [TK]D-Fender | Ch0Hag: Given FXS isn't working out either |
| 18:30.09 | *** join/#asterisk MrChicken (n=Dorphals@200.71.58.39) |
| 18:30.22 | Ch0Hag | It gets there after 'exten => s-ANSWER,2,Hangup'. |
| 18:30.27 | MrChicken | Hi |
| 18:30.32 | Strom_M | Cocks |
| 18:30.34 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-c3c8a3c838d9c7fd) |
| 18:30.34 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 18:30.39 | MrChicken | Uhhh I'm having a problem with app_rxfax |
| 18:30.40 | Strom_M | the Hi/Cocks protocol (RFC 4373) |
| 18:30.51 | MrChicken | everytime a fax comes in |
| 18:30.54 | *** join/#asterisk SwK (n=SwK@dhcp64-134-34-248.bwic.chi.wayport.net) |
| 18:30.56 | MrChicken | asterisk crashes |
| 18:31.04 | MrChicken | and logs do not report anything |
| 18:31.30 | Ch0Hag | Hmm well md will take another half an hour. |
| 18:31.42 | MrChicken | the last reported thing in logs is like rxfax("/var/spool/asterisk/fax/6446800.tif") |
| 18:31.52 | Ch0Hag | I'll take out the superfluous FXO module and put the FXS in its place (slot 2). |
| 18:32.47 | Ch0Hag | Strom_M: Was that said knowing what 4373 is, or is it a random number? |
| 18:33.01 | Strom_M | I know perfectly well what 4373 is |
| 18:33.03 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 18:33.03 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:33.17 | Ch0Hag | Righto. |
| 18:33.28 | Ch0Hag | Somehow it fits, you see. |
| 18:33.34 | Strom_M | that's the point |
| 18:34.34 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 18:35.24 | Ch0Hag | I presume loading the wctdm module in FCC rather than UK mode makes no difference. |
| 18:35.31 | Ch0Hag | To my situation, that is. |
| 18:35.43 | bcnl | [TK]D-Fender: do you think the call screening macro will work if I'm dialing multiple (external) extensions? |
| 18:36.18 | bcnl | basicallly I call 5 cellphones in one long Dial(Local/XXX&Local/YYY, etc etc) command to ensure a call gets answered |
| 18:36.31 | Ch0Hag | Actually that's a point - should I explicitely load wcfxo and/or wcfxs? |
| 18:36.31 | bcnl | but I'm trying to avoid cellphone voicemails picking up the call |
| 18:36.43 | Ch0Hag | They're not, apparently. |
| 18:37.25 | [TK]D-Fender | bcnl: I think the first to try to respond will actually stop the others. |
| 18:37.27 | Strom_M | Ch0Hag: no, those are for other cards |
| 18:37.50 | Ch0Hag | wctdm is fine on its own then? |
| 18:37.54 | Strom_M | yes |
| 18:38.00 | Ch0Hag | Bummer. |
| 18:38.05 | Ch0Hag | That was nearly an easy fix. |
| 18:38.06 | sweeper | uh, is there a way to use the MYSQL command without connecting and disconnecting every time I do a query? |
| 18:38.13 | sweeper | 'cause that seems dumb |
| 18:38.47 | Corydon76-work | SwK: use func_odbc |
| 18:38.53 | sweeper | arg |
| 18:38.57 | Corydon76-work | sweeper, rather |
| 18:38.58 | sweeper | but that requires odbc D: |
| 18:39.06 | Corydon76-work | So? |
| 18:39.09 | [TK]D-Fender | sweeper: AGI |
| 18:39.14 | sweeper | odbc sux :P |
| 18:39.18 | sweeper | agi is slowcat slow |
| 18:39.27 | Corydon76-work | [TK]D-Fender: but that requires a separate process |
| 18:39.35 | sweeper | I guess I'll go with odbc >.< |
| 18:39.44 | [TK]D-Fender | sweeper: You need to space out your bitching with some whining now and again ;) |
| 18:39.44 | Corydon76-work | sweeper: ODBC doesn't suck |
| 18:40.03 | bcnl | [TK]D-Fender: Hrmmm shitty |
| 18:40.13 | bcnl | [TK]D-Fender: maybe I should try parking the call and sending a sms to all the cells? |
| 18:40.24 | sweeper | Corydon76-work: the hell it doesn't! |
| 18:40.27 | *** join/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 18:40.37 | [TK]D-Fender | bcnl: better. |
| 18:40.41 | Corydon76-work | ODBC rocks |
| 18:40.49 | [TK]D-Fender | bcnl: Or call-file. |
| 18:41.00 | sweeper | installation sucks |
| 18:41.10 | xheliox | <Corydon76-work> ODBC rocks <-- I don't think I've ever actually heard anyone say that before. |
| 18:41.17 | sweeper | I haven't gotten to the point where I could say if it's slow as hell or shitty or anything |
| 18:41.20 | bcnl | [TK]D-Fender: what's call-file? |
| 18:41.50 | sweeper | but I compiled it, and I'm missing the facking drivers, and documentation for it is laughable |
| 18:41.52 | [TK]D-Fender | bcnl: a call-out mechanism like AMI originate. |
| 18:41.59 | [TK]D-Fender | bcnl: Go WIKI it. |
| 18:42.07 | Corydon76-work | xheliox: expecially when you realize that I don't want to code specifically to 20 different DB APIs |
| 18:42.17 | bcnl | yea reading now |
| 18:42.19 | bcnl | interesting |
| 18:42.42 | uwe | hello, im trying to figure out why asterisk stopped sending calls to sip phones today, i stopped the process and started it again and everything was find again! , in the logs , i found " Unable to create channel of type 'SIP' (cause 42 - Switching equipment congestion)" , is this due to load on switches ?? and if yes, why was it solved by restarting asterisk ? |
| 18:42.43 | Corydon76-work | Yet func_odbc is compatible with all of them now |
| 18:43.26 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 18:43.40 | sweeper | yea, I do everything with mysql |
| 18:44.29 | sweeper | it's sad that the drivers exist in asterisk, but they're not implemented well :/ |
| 18:44.33 | blitzrage | ODBC rocks |
| 18:44.37 | Corydon76-work | sweeper: I pity your users |
| 18:44.54 | *** part/#asterisk ramindia_ (n=ramindia@202.63.96.9) |
| 18:44.55 | Ch0Hag | Sod the users. I pity you. |
| 18:45.04 | Corydon76-work | I may use MySQL, but not for everything. |
| 18:45.27 | Corydon76-work | The accounting system, in particular, is not running on MySQL |
| 18:45.34 | sweeper | hehehe |
| 18:45.58 | sweeper | so long as you're eschewing ISAM :P |
| 18:46.08 | Corydon76-work | and that includes the payroll system that generates my paycheck |
| 18:48.38 | cpm | a lot |
| 18:48.58 | sweeper | hmmm |
| 18:49.00 | *** join/#asterisk matsk (i=matsk@h110n2fls32o882.telia.com) |
| 18:49.19 | rob0 | No you didn't! |
| 18:49.21 | sweeper | I wonder if anyone's written a fork of asterisk that runs entirely off of mysql |
| 18:49.23 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 18:49.26 | [TK]D-Fender | cpm: so now you're on to being a MySQL Nazi ;) |
| 18:49.43 | cpm | [TK]D-Fender, not really, I'm not a fanatic, but it's a good tool |
| 18:49.50 | sweeper | oh, speaking of nazis and fascists |
| 18:49.54 | cpm | and sporks? |
| 18:50.06 | sweeper | http://xkcd.com/c282.html |
| 18:50.09 | [TK]D-Fender | cpm: I was overjoyed to hear someone in head office pushed for it instead of their MSSQL BS. |
| 18:50.11 | sweeper | facism at work! |
| 18:50.59 | Ch0Hag | Godwin's law. |
| 18:51.30 | cpm | [TK]D-Fender, that's good to hear. I can live with mssql, but it's hard to do anything other than tolerate it. It's difficult to really like much. |
| 18:51.39 | *** join/#asterisk zeeesh (n=aadilism@202.125.143.70) |
| 18:52.06 | *** part/#asterisk bminish (n=bminish@brenbox.westnet.ie) |
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| 19:12.36 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 19:12.45 | Ch0Hag | Speaking of databases: |
| 19:12.52 | Ch0Hag | ERROR: ORA-28002: the password will expire within 18446744073709551541 days |
| 19:12.59 | syzygyBSD | careful, they may be listening |
| 19:13.17 | Ch0Hag | I don't think Oracle listens. It just does its thing regardless. |
| 19:13.44 | *** join/#asterisk obnauticus (n=obnautic@c-71-59-162-60.hsd1.wa.comcast.net) |
| 19:13.45 | syzygyBSD | ahh, one of those upiddy ones |
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| 19:16.21 | *** part/#asterisk SwK (n=SwK@dhcp64-134-34-248.bwic.chi.wayport.net) |
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| 19:21.56 | syzygyBSD | wow, it's all quiet in here, I can't hear myself think |
| 19:22.32 | *** join/#asterisk __DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 19:24.44 | *** join/#asterisk naitram (n=ttech@216.77.58.40) |
| 19:26.11 | Strom_M | oooo boy, a potential client called |
| 19:26.24 | Strom_M | time to rip out another trixbox install |
| 19:26.33 | Qwell[] | ha |
| 19:26.39 | naitram | sip clients trying to use automon, have automon => #1 in features.conf, have Dial(...,gTW). dtmf mode is info, can see dtmf tones in sip info packets to server. No recording takes place |
| 19:26.52 | Strom_M | Qwell[]: can you do BLF on a 7970 running SIP firmware? |
| 19:27.05 | Qwell[] | no clue |
| 19:27.20 | Strom_M | because that's what this client has |
| 19:28.01 | [TK]D-Fender | Strom_M: Nope |
| 19:28.18 | [TK]D-Fender | Strom_M: Presence only in SCCP IIRC. |
| 19:28.29 | [TK]D-Fender | Strom_M: Cisco FTL |
| 19:28.58 | [TK]D-Fender | naitram: You setting the dynamic features variable before your dial? |
| 19:29.43 | *** join/#asterisk rvhi0 (n=root@cpe-67-49-180-124.hawaii.res.rr.com) |
| 19:30.06 | Strom_M | [TK]D-Fender: alright |
| 19:30.29 | naitram | [TK]D-Fender: I think so, I have Set(DYNAMIC_FEATURES,blindtranser#automon) as the first line in the script for the extensions. Is that right? |
| 19:30.55 | [TK]D-Fender | naitram: No, clearly missing the obligatory "=" |
| 19:30.57 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 19:31.10 | [TK]D-Fender | naitram: and as for the CONTENT of that variable, I'm uncertain. |
| 19:31.15 | vlt|afk | Hello. Does "${CALLERID}" contain more than "(name)" and "(num)"? |
| 19:31.24 | Strom_M | vlt|afk: yes |
| 19:31.31 | Strom_M | show function CALLERID |
| 19:31.40 | [TK]D-Fender | vlt|afk: That var is deprecated. Avoid. |
| 19:31.51 | Ch0Hag | OK the FXS is in port 2 and still there is no dialtone. |
| 19:31.51 | [TK]D-Fender | vlt|afk: and No, it doesn't |
| 19:32.00 | *** join/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone) |
| 19:32.03 | [TK]D-Fender | Ch0Hag: Could jsut be a dead module. |
| 19:32.06 | naitram | [TK]D-Fender: oh, Ill double check. Thanks |
| 19:32.29 | Ch0Hag | I'm beginning to suspect, but that would be rather dull after a day spent rejigging the wiring. |
| 19:32.31 | brimstone | anyone know what PROGRESS code 127 is and why it would be dropping outbound calls on a pri? |
| 19:32.31 | rvhi0 | anyone knows how to register one extension on multiple phones? |
| 19:32.35 | Ch0Hag | And it's brand new. |
| 19:32.43 | vlt|afk | Strom_M: Thanks, I'll check that. |
| 19:33.11 | vlt|afk | [TK]D-Fender: I only use CALLERID(name) and CALLERID(num). Is that correct? |
| 19:33.28 | [TK]D-Fender | vlt|afk: I would advise doing so. |
| 19:34.17 | Ch0Hag | Is the module activated (in terms of providing power and dialtone) simply by the module being loaded? |
| 19:34.36 | Strom_M | Ch0Hag: no, it has to be configured also |
| 19:34.53 | [TK]D-Fender | brimstone: 127 7F Internetworking, unspecified An event occurred, but the network does not provide causes for the action that it takes. The precise problem is unknown. |
| 19:34.59 | Ch0Hag | By ztcfg or by asterisk? |
| 19:35.13 | brimstone | [TK]D-Fender, thanks |
| 19:35.14 | [TK]D-Fender | Ch0Hag: * |
| 19:35.19 | brimstone | where did you look that up? |
| 19:35.27 | [TK]D-Fender | brimstone: GOOGLE :) |
| 19:35.33 | Corydon76-work | brimstone: Q.931, appendix |
| 19:35.35 | Strom_M | brimstone: you want a copy of Q.931? |
| 19:35.37 | [TK]D-Fender | <- *'s resident Google-Proxy |
| 19:35.41 | brimstone | i didn't have any luck when i looked on google |
| 19:35.58 | [TK]D-Fender | brimstone: http://www.google.ca/search?hl=en&q=ISDN+progress+127&btnG=Google+Search&meta= |
| 19:36.06 | [TK]D-Fender | brimstone: Try harder :) |
| 19:36.12 | brimstone | ah, i was googling pri+progress+127 |
| 19:36.21 | Ch0Hag | And that is the part of zapata.conf between 'signalling=fxo_ks' and 'channel => 2'? |
| 19:36.40 | Ch0Hag | Well not between, because the zapata.conf is not structured, but that aside... |
| 19:36.49 | Strom_M | brimstone: http://www.itu.int/rec/T-REC-Q.931/en |
| 19:36.57 | Corydon76-work | brimstone: also include/asterisk/causes.h |
| 19:37.43 | Ch0Hag | Anyway I assume * is doing its thing because when I accidentally deleted the signalling line it complained when trying to start the module. |
| 19:38.21 | vlt|afk | [TK]D-Fender: I'm facing intering problems transmitting my CALLERID to my IAX2 peer. When dialling from SIP phones or from manager interface it works fine. When dialling from my PBX (connected via BRI) the CALLERID(name) is set to "CID withheld" which I overwrite with nothing. But the number in CALLERID(num) still isn't accepted. I think I should check the other elements listed in `show function CALLERID` ... |
| 19:38.33 | vlt|afk | s/intering/interesting |
| 19:38.42 | Ch0Hag | Could it be the cable? (I think not but...) it's RJ11->RJ11 1->4, 2->3, etc. |
| 19:38.52 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-75-84-232-194.socal.res.rr.com) |
| 19:38.56 | [TK]D-Fender | vlt|afk: Sorry, I can't speak for BRI. |
| 19:39.14 | [TK]D-Fender | Ch0Hag: 10 seconds to test... |
| 19:39.33 | Ch0Hag | The phone is on pins 2 and 3 (well pins 3 and 4 as there are 6)? |
| 19:39.49 | Ch0Hag | Except that I'm not sure if I have a straight-through cable. |
| 19:40.15 | Ch0Hag | In fact I'm not sure how many rj11-rj11 cables I have left :/ |
| 19:40.39 | Strom_M | Ch0Hag: RJ-11 should be a six-position plug with conductors only in the middle two position |
| 19:40.58 | Strom_M | RJ-12 is six-position four-conductor |
| 19:41.07 | Strom_M | and RJ-14 is six-position six-conductor |
| 19:41.26 | Strom_M | in all cases, pair 1 is on positions 3 and 4 |
| 19:41.35 | Ch0Hag | Ooh that's interesting. After all that fuss there is another cable which works. |
| 19:41.59 | Qwell[] | Strom_C: explain something to me, would ya? |
| 19:42.03 | Strom_M | sure |
| 19:42.06 | Ch0Hag | Unfortunately it's BT at one end. |
| 19:42.13 | Strom_M | well i'll try :) |
| 19:42.18 | Qwell[] | in the above, what is the difference between "position", and "conductor"? |
| 19:42.29 | Strom_M | Qwell: position just means a groove in the plastic bit |
| 19:42.36 | *** part/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone) |
| 19:42.37 | Qwell[] | uh huh |
| 19:42.39 | Strom_M | conductor is the actual gold-plated pin thingy |
| 19:42.48 | Strom_M | (these are technical terms) |
| 19:42.49 | Qwell[] | so...what's the point? |
| 19:42.58 | Strom_M | save money on unnecessary conductors |
| 19:43.01 | Qwell[] | save a tenth of a cent by not including the conductors? |
| 19:43.13 | Strom_M | if you're making a million cables, it adds up |
| 19:43.29 | Qwell[] | So why even include the positions? |
| 19:43.47 | Strom_M | because the jacks can be wired for multiple lines |
| 19:43.50 | Qwell[] | You're probably spending more on the "extra" plastic than the conductors would cost you |
| 19:44.04 | Ch0Hag | OK I have one cable which works and one which doesn't. The only difference is that the working one is RJ-12 and the other is RJ-11. |
| 19:44.19 | Strom_M | this is the same argument as "OMG Ethernet only uses two pairs! why all this extra wastage on two more pairs and a giant honking plug?" |
| 19:44.19 | Ch0Hag | No, scrap that. |
| 19:44.22 | Ch0Hag | BAH too many wires. |
| 19:44.44 | Qwell[] | Strom_C: yeah, but RJ45 is 8-position, 8-conductor |
| 19:44.50 | Qwell[] | right? |
| 19:45.11 | Strom_M | right, and ethernet could techincally operate on 4-position 4-conductor |
| 19:45.37 | Qwell[] | not gbit or PoE :D |
| 19:45.51 | Strom_M | don't forget that the six-position plug was developed by the bell system |
| 19:46.02 | Ch0Hag | PoE can (and does in GigE) operate over the data lines |
| 19:46.29 | Strom_M | and when you've got 200 million phones, saving .1 cent on manufacturing costs per phone is a big deal |
| 19:47.49 | Ch0Hag | Woo it works. |
| 19:47.57 | Ch0Hag | I have a disgesting mess of wires and adapters, but it works. |
| 19:48.08 | Strom_M | a digesting mess? |
| 19:48.13 | rob0 | yum! |
| 19:48.28 | *** join/#asterisk AllanLima (n=mediain@unaffiliated/allanlima) |
| 19:48.31 | Juggie | is it just me or did internet routing just crap in its pants |
| 19:48.40 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 19:48.42 | Strom_M | the internet is tubes |
| 19:48.42 | Ch0Hag | TDM -> RJ11<->BT -> back-end-of-splitter : other-back-end-of-splitter BT<->RJ11 -> ADSL filter -> phone |
| 19:48.44 | currach | It's not just you. |
| 19:49.12 | Juggie | supprise supprise i think cogent was messed up |
| 19:49.17 | tzanger | yeah |
| 19:49.19 | tzanger | cogent's fucked right now |
| 19:49.29 | tzanger | my bandwidth is going through teleglobe |
| 19:49.31 | Qwell[] | s/ right now// |
| 19:49.33 | tzanger | (multi-homed IPs) |
| 19:49.42 | Ch0Hag | One day I'll work out which pin is going where and get rid of it all, but first I need to change 'phone' to 'BT<->RJ11 -> gender-bender -> wall' |
| 19:49.43 | Ch0Hag | What fun. |
| 19:49.58 | [hC] | sup jug |
| 19:51.08 | Juggie | yo, and there it goes again. |
| 19:51.55 | [hC] | there what goes? |
| 19:52.33 | Strom_M | the internet is all dizzy and jizzy today |
| 19:52.53 | Juggie | cogent |
| 19:52.55 | [hC] | ahh |
| 19:52.58 | [hC] | gotta love slowgent |
| 19:54.03 | Juggie | back again |
| 19:54.24 | *** join/#asterisk zeeesh (n=aadilism@202.125.143.69) |
| 19:54.53 | vlt|afk | Strom_M: Hmm, there were different values in CALLERID(ANI) and CALLERID(DNID). But even when I set this values manually to the ones they are when dialling successfully the correct CALLERID recognition fails. Any idea what else could be different? |
| 19:55.47 | Strom_M | vlt|afk: i have no experience with BRI |
| 19:55.48 | Strom_M | sorry |
| 19:56.07 | Ch0Hag | Gah I can almost taste victory. |
| 19:56.36 | vlt|afk | Strom_M: Ok, I thought is was an IAX2 problem ... |
| 19:56.46 | Strom_M | victory tastes a lot like a Guinness, as it turns out |
| 19:56.54 | vlt|afk | Strom_M: I'll contact my ITSP. |
| 19:57.04 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 19:57.57 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
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| 19:59.14 | Qwell[] | Strom_M: victory tastes bad? |
| 19:59.27 | [hC] | Juggie: are you goin to astricon again this year? |
| 19:59.29 | Juggie | to go from the two largest isp's in canada (bell & rogers) i'm going -> ottawa(bell)->montreal(bell)->new york(bell)->ny (savis)->washington(savis)->virginia(equinix)->two unknown hops->iad01.atlas, indiana maybe(cogent)->rogers |
| 19:59.36 | Juggie | [hc] yeah, your tax dollars @ work |
| 19:59.46 | [hC] | nice. I'm bringing one of my partners this year too. |
| 20:01.39 | *** join/#asterisk mardum_ (n=mardum@71-81-73-42.dhcp.stls.mo.charter.com) |
| 20:02.53 | [TK]D-Fender | Juggie: 2 companies I woulod not want internet dealings with ;) |
| 20:03.03 | [TK]D-Fender | Juggie: Telco contracts = ass |
| 20:04.11 | Ch0Hag | Bah I've got my wires twisted (3/4 should be 2/5) but the rj45 plug I need to recrimp is in just about the most inaccessible place it could be. |
| 20:04.27 | Strom_M | Ch0Hag: plugs go on cables |
| 20:04.30 | Strom_M | cables are portable |
| 20:04.42 | Strom_M | therefore, unplug the cable and take it to your desk |
| 20:04.52 | [TK]D-Fender | Strom_M: x0mg, u r so SMRT. |
| 20:04.54 | Strom_M | Qwell[]: gah, you don't like guinness?? |
| 20:04.58 | Ch0Hag | Even just unplugging it is a pain but I suppose there's no other way. |
| 20:05.05 | Qwell[] | Strom_M: not in the least |
| 20:05.07 | Strom_M | [TK]D-Fender: this is why they pay me the dollars and cents |
| 20:05.28 | Strom_M | Qwell[]: do you at least like newcastle? |
| 20:05.36 | Qwell[] | never had it |
| 20:05.39 | Qwell[] | oh, wait |
| 20:05.40 | Ch0Hag | incidentally, quite often cables are fixed to the walls or floors. |
| 20:05.42 | Qwell[] | yes I have, heh |
| 20:05.45 | Qwell[] | yeah |
| 20:05.47 | Ch0Hag | Conveniently not in this case though. |
| 20:06.27 | *** join/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 20:06.28 | rob0 | My European beer buddies don't recommend Newcastle. |
| 20:07.22 | Strom_M | the Dutch also combine liquorice with a pound of salt and consider the result edible |
| 20:09.08 | Nugget | DZ is some of the most heinously disgusting "candy" I've ever encountered. |
| 20:10.08 | *** join/#asterisk lirakis (n=etamme@65.200.191.253) |
| 20:10.14 | lirakis | ahoy! |
| 20:10.47 | Hmmhesays | lol |
| 20:10.48 | Hmmhesays | nice |
| 20:11.22 | *** join/#asterisk irule (n=irule@189.164.43.194) |
| 20:13.14 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 20:13.15 | lirakis | asterisk is fun |
| 20:13.19 | lirakis | :P |
| 20:14.10 | Strom_M | yes |
| 20:14.12 | Strom_M | it is |
| 20:14.29 | Strom_M | why, just the other day, I used asterisk hurrrrrrrrrrrrrrrrrrrrrrrrrrrr plane crash |
| 20:14.41 | lirakis | ? |
| 20:19.06 | lirakis | hmm.. quiet .. |
| 20:19.17 | anonymouz666 | I need to use arrays on dialplan |
| 20:19.22 | anonymouz666 | how do I do that |
| 20:19.23 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 20:20.25 | lirakis | ?? http://lists.digium.com/pipermail/asterisk-dev/2007-June/027924.html |
| 20:21.00 | anonymouz666 | hmmmm |
| 20:21.21 | blitzrage | anonymouz666: ARRAY() |
| 20:22.24 | Ch0Hag | It doesn't matter which way round the signal pins are, does it? |
| 20:22.58 | [TK]D-Fender | Ch0Hag: tip VS ring.... uuhhh YEAH... |
| 20:23.10 | [TK]D-Fender | Strom_M: Get schooling! |
| 20:23.47 | anonymouz666 | I am unlucky there is no ARRAY on 1.2 |
| 20:23.48 | anonymouz666 | :( |
| 20:24.29 | Ch0Hag | So which of pins 3 and 4 is tip in rj11? |
| 20:24.40 | Strom_M | balls, i can't remember |
| 20:24.58 | Ch0Hag | And BT, for that matter. |
| 20:25.04 | Ch0Hag | All they tell you is A and B. |
| 20:25.15 | Strom_M | on the jack, starting with pin 1 on the left, tip for line 1 should be pin 3 |
| 20:25.50 | Strom_M | er |
| 20:25.51 | Strom_M | no |
| 20:25.57 | Strom_M | ring is pin 3 |
| 20:25.59 | Strom_M | tip is pin 4 |
| 20:26.14 | Strom_M | silly me |
| 20:26.20 | Ch0Hag | Righto. |
| 20:27.18 | irule | is it possible to apply this hack to config files only? http://www.trixbox.org/forums/trixbox-forums/open-discussion/multilanguage-voice-hack |
| 20:27.50 | *** part/#asterisk lirakis (n=etamme@65.200.191.253) |
| 20:28.34 | Ch0Hag | What does that correspond to on a BT connector? |
| 20:29.05 | Strom_M | no idea |
| 20:29.16 | Strom_M | your electrical plug is weird enough on its own |
| 20:29.21 | Ch0Hag | Da. |
| 20:29.25 | Ch0Hag | But very cool. |
| 20:29.31 | Strom_M | it's HUGE |
| 20:30.15 | Ch0Hag | No. Yours is tiny. |
| 20:30.40 | Ch0Hag | When you have an English plug, you know you're not going to confuse it for ANYthing else. |
| 20:31.28 | Ch0Hag | Normally I despise the health-and-safety brigade, but I've seen plugs in other countries and compared to ours they're all needlessly flimsy and unsafe. |
| 20:31.47 | Strom_M | what am I going to confuse a US plug with? toenail scissors? |
| 20:31.50 | Ch0Hag | Well, maybe not all. I've not been to every country. |
| 20:31.56 | Ch0Hag | But US and EU plugs suck. |
| 20:31.58 | naitram | cant seem to get asterisk to read dtmf tones from my sip info messages sent from my sip clients. Can see the messages on the wire, but asterisk doesn't appear to do anything with them. Is there a way to see if asterisk is actually seeing them at the applcation level? |
| 20:32.15 | Strom_M | and fwiw, i've never ever had an electrical accident with a US plug, nor have I ever known anyone who has |
| 20:32.21 | Hmmhesays | you have your dtmf settings correct in sip.conf? |
| 20:32.32 | Strom_M | naitram: dtmfmode=info |
| 20:33.03 | Ch0Hag | Actually most of the safety features of our plugs are to protect the equipment, not you. |
| 20:33.24 | Strom_M | that's comforting |
| 20:33.36 | Strom_M | "you'll die but your PC will keep right on downloading porn" |
| 20:33.36 | naitram | Strom_M: yes in my sip.conf that it is set up, and that is what is going out on the wire |
| 20:33.37 | Ch0Hag | Plus the fact that you can disassemble them absolutely trumps everything. |
| 20:33.50 | Ch0Hag | At least until recently, when everything seems to use moulded plugs. |
| 20:34.12 | Ch0Hag | Probably to make Wickes because if you want to rewire one you can't rip it off, you have to go out and buy a new one. |
| 20:34.21 | Ch0Hag | s/Wickes/Wickes more money/ |
| 20:36.24 | naitram | how does asterisk record a sip conversation after it has started if the convesation is using a native bridge, isn't asterisk bypassed at that point |
| 20:36.57 | *** part/#asterisk HockeyInJune (i=HockeyIn@pool-70-107-173-57.ny325.east.verizon.net) |
| 20:38.39 | naitram | has anyone here used dtmf tones in a sip to sip call. I can not get asterisk to see the dtmf tones of a connected sip client even though they appear on the wire as sip info requests |
| 20:38.57 | *** join/#asterisk malaiwah (n=mbelleau@modemcable077.177-82-70.mc.videotron.ca) |
| 20:39.27 | malaiwah | Hi there, is there a way to let Asterisk 1.2 play a file that doesn't have an extension eventhough it's really a wav (ulaw) file ? |
| 20:39.41 | malaiwah | i tried prefixing (in AGI) by ulaw: or wav: but it doesn't work |
| 20:40.13 | Strom_M | just give it a damned extension |
| 20:41.03 | syzygyBSD | ^ |
| 20:41.28 | malaiwah | ... what if I dont want to ? |
| 20:41.49 | malaiwah | there must be a way to specify ! |
| 20:42.39 | Strom_M | but linux has to run on a computer? what if i want to run it on my rice cooker??? |
| 20:43.14 | syzygyBSD | Strom_M: use netbsd it runs on anything |
| 20:43.16 | *** join/#asterisk Marshall-Laptop (n=eman0n@cpe-76-181-165-37.columbus.res.rr.com) |
| 20:44.06 | Ch0Hag | Dammit NOW what have I got wrong? |
| 20:45.40 | Strom_M | Ch0Hag: let me divine that |
| 20:45.45 | syzygyBSD | Ch0Hag: comment in the wrong place |
| 20:45.57 | *** join/#asterisk splatman (n=farfromg@adsl-65-68-175-159.dsl.fyvlar.swbell.net) |
| 20:46.01 | naitram | when asterisk receives a dtmf tone, is there a log or some other way i can see it got one |
| 20:46.57 | naitram | should something show up in the cli |
| 20:47.04 | syzygyBSD | naitram: no |
| 20:47.29 | syzygyBSD | you can create your own log if you want i guess |
| 20:48.32 | irule | what player can I use to listen to gsm files? |
| 20:49.26 | naitram | syzygyBSD: i can not get anything to work with asterisk and dtmf tones on a sip call. I can see the traffic on the wire but can't tell if asterisk recognizes it at all. No apps work from my dial plan. Any suggestions? |
| 20:50.36 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:50.45 | Ch0Hag | A female-female rj-12 gender bender isn't going to cross cables over is it? |
| 20:50.50 | Ch0Hag | That would just be weird. |
| 20:51.18 | syzygyBSD | naitram: setup the correct dtmfmode |
| 20:52.54 | naitram | syzygyBSD: is info the correct mode? or rfc2833 |
| 20:53.08 | syzygyBSD | I think rfc2833 |
| 20:53.17 | syzygyBSD | depending on what you have hooked up.. |
| 20:53.17 | sweeper | irule: mplayer :) |
| 20:53.27 | *** join/#asterisk canberk (n=canberk@212.156.213.131) |
| 20:53.30 | syzygyBSD | might be inband if you have an ata |
| 20:53.36 | irule | thanks |
| 20:54.03 | syzygyBSD | or better yet.. auto |
| 20:56.28 | naitram | syzygyBSD: my sip devices can do rfc2833, But I never saw anything go out to the asterisk server when set up that way |
| 20:56.41 | canberk | hello everybody |
| 20:56.50 | syzygyBSD | herro |
| 20:56.53 | malaiwah | what asterisk version are you all running ? i tried 1.4 and it crashed 2 times the same week, I then rolled back to 1.2 ; what about you all ? |
| 20:56.53 | canberk | is it possible to increase the sound level on asterisk |
| 20:56.55 | *** join/#asterisk SwK (n=SwK@dhcp64-134-34-248.bwic.chi.wayport.net) |
| 20:57.02 | syzygyBSD | I am still on 1.2.17 |
| 20:57.06 | canberk | 1.4.4, no problems!! |
| 20:57.21 | syzygyBSD | canberk: sound level between which devices? |
| 20:57.52 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 20:58.38 | *** join/#asterisk fbffff (n=fbffff@dhcp64-134-34-248.bwic.chi.wayport.net) |
| 20:58.59 | sweeper | damnit |
| 20:59.21 | canberk | well |
| 20:59.22 | sweeper | why the hell isn't there a tutorial somewhere about how to set up odbc + mysql |
| 20:59.28 | canberk | when i am listening to recorded conversations |
| 20:59.37 | canberk | called party's sound is really too low |
| 21:00.07 | [TK]D-Fender | canberk, that's not an * problem, thats a COMPUTER problem. Fix your mixer |
| 21:00.40 | sweeper | [TK]D-Fender: actually, he said "called party", implying the caller's audio is fine :) |
| 21:00.43 | [TK]D-Fender | canberk, And you have not answered syzygyBSD's question either |
| 21:01.26 | sweeper | canberk: what are you using to mix the two audio files? |
| 21:02.00 | *** join/#asterisk SwK (n=SwK@dhcp64-134-34-248.bwic.chi.wayport.net) |
| 21:02.51 | naitram | [TK]D-Fender: ive tried what you suggested earlier regarding automon on sip channel. Still can not get sip dtmf codes to be seen by asterisk AFAIK. Any way to tell if they are being seen by * |
| 21:03.00 | canberk | soxmix |
| 21:03.05 | [TK]D-Fender | naitram, check your configs.... |
| 21:03.28 | canberk | i use soxmix to mix two audio files |
| 21:03.40 | canberk | and also when talking to other people |
| 21:03.44 | canberk | the sound level is lo |
| 21:03.45 | canberk | low |
| 21:04.05 | canberk | is it possible to increase the sound level somehow? without touching to user devices? |
| 21:04.09 | canberk | pap2 to asterisk |
| 21:04.21 | syzygyBSD | well.. when using soxmix you have the option to adjust the level of sound |
| 21:04.30 | *** join/#asterisk rvhi0 (n=chatzill@66.135.230.96) |
| 21:04.31 | canberk | when using x provider to talk to PSTN, x provider's sound level is low |
| 21:04.43 | canberk | even when i am talking in real life, not recordig |
| 21:04.55 | canberk | y provider's sound level is okay |
| 21:05.01 | canberk | but x needs some adjustment |
| 21:05.08 | *** join/#asterisk rhombus (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com) |
| 21:07.30 | rpm | anyone here familiar with broadsoft? im trying to setup a iad/gateway to an asterisk server. it is all setup, accepting calls and forwarding.. but i can only have 1 call at a time. |
| 21:07.34 | naitram | [TK]D-Fender: I have, for the caller i have dtmfmode=info, my phone is setup to send this and is, in the features.conf automon => #9, in extensions.conf in [globals] have DYNAMIC_FEATURES=>automon, in context have Dial(....,gWwT). it dials with the gWwT as seen from cli |
| 21:08.05 | rvhi0 | anyone knows a list of applications supported in *? |
| 21:08.20 | rvhi0 | voip-info.org has a list, but not very up to date. |
| 21:08.45 | [TK]D-Fender | naitram, Do something else to test your DTMF |
| 21:08.47 | canberk | is there a working solution for h323 and asterisk? |
| 21:08.53 | [TK]D-Fender | rvhi0, "show applications" |
| 21:08.57 | canberk | like a secondary server in between or any other method |
| 21:09.48 | rvhi0 | [TK]D-Fender, is there any document I see? |
| 21:10.16 | [TK]D-Fender | rvhi0, I just gave you the answer. |
| 21:10.21 | naitram | [TK]D-Fender: simplest other test |
| 21:10.32 | [TK]D-Fender | naitram, read |
| 21:10.48 | rvhi0 | [TK]D-Fender, i mean offline |
| 21:10.58 | [TK]D-Fender | ~book |
| 21:10.59 | jbot | i heard book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 21:11.01 | [TK]D-Fender | ^^^^^^^^^^ |
| 21:11.06 | rvhi0 | i want to compare v1.4 and trunk |
| 21:11.21 | rvhi0 | and see any new features |
| 21:12.01 | [TK]D-Fender | rvhi0, Don't expect any miracle comparative docs then. |
| 21:12.11 | [TK]D-Fender | rvhi0, Changelogs is IT |
| 21:12.30 | sweeper | yea, nobody actually reviews asterisk, they just use it |
| 21:12.31 | *** join/#asterisk anderiv (n=anderiv@207-67-87-34.static.twtelecom.net) |
| 21:13.09 | naitram | [TK]D-Fender: have read, tried enabling the testfeature in the features.conf, it didn't seam to work either. Oh well, looked at this a month or so ago with same result. Guess maybe send something on the mailing list? |
| 21:13.36 | [TK]D-Fender | Trunk is for "testing" and there is no promise that things attempted in there will become final. |
| 21:13.46 | [TK]D-Fender | naitram, no, "show application read " <-------------- |
| 21:13.57 | [TK]D-Fender | naitram, I actually figured you'd have picked that up... |
| 21:14.10 | [TK]D-Fender | naitram, and I left out the fully implied format to test as such. |
| 21:14.12 | rvhi0 | i know * can't support multiple registration, anyone knows any workaround solutions? |
| 21:14.22 | [TK]D-Fender | rvhi0, SER |
| 21:14.50 | [TK]D-Fender | rvhi0, or multiple accoutns that just ring when appropriate. |
| 21:15.10 | rvhi0 | [TK]D-Fender, i don't like that because ser hide the channel availability in * |
| 21:15.44 | rvhi0 | [TK]D-Fender, if i use multiple accounts, how to ring them all? |
| 21:15.47 | [TK]D-Fender | rvhi0, Your expectations will only lead to a lot of disappointment. Time to alter it. |
| 21:15.59 | canberk | is there a working solution for h323 and asterisk? |
| 21:16.01 | [TK]D-Fender | rvhi0, "show application dial" |
| 21:16.28 | *** join/#asterisk n00dle (n=ccraft@204.10.248.123) |
| 21:17.09 | rvhi0 | [TK]D-Fender, are you saying * is able to do it well, at least for now? |
| 21:17.21 | Juggie | today is crappy internet day |
| 21:17.37 | naitram | [TK]D-Fender: oh, well thanks for clearing that up. Not been using asterisk that long. |
| 21:17.46 | [TK]D-Fender | rvhi0, I've once again given you the answer and you aren't READING. |
| 21:18.10 | [TK]D-Fender | naitram, Long enough to have earned some small expectation on my part :) |
| 21:19.20 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 21:19.44 | rvhi0 | [TK]D-Fender, thx for the answer, i tried dial, there seems to be quite a bit delay in rings, e.g. there is an interval of .5 second between each call, so if i have 10 users, the last one takes a while to get the call |
| 21:21.39 | [TK]D-Fender | rvhi0, hrm, news to me. |
| 21:21.57 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 21:23.17 | malaiwah | quit |
| 21:29.45 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 21:30.35 | rvhi0 | another Q, can * support native SIP attended transfer now? |
| 21:31.07 | [TK]D-Fender | rvhi0, Always did |
| 21:32.50 | rvhi0 | [TK]D-Fender, i was using 1.0 for a while ago, maybe it changed for some time already |
| 21:33.07 | [TK]D-Fender | rvhi0, Supported it then too....... |
| 21:34.01 | *** join/#asterisk X-Rob (n=rob-x@dsl-124-150-120-174.vic.westnet.com.au) |
| 21:36.52 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 21:37.27 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 21:42.18 | *** join/#asterisk ber111 (i=brad@neu.cow.org) |
| 21:42.34 | ber111 | how do i record a phone call once i have excuted a Dial command in a DeadAGI script |
| 21:42.44 | ber111 | i tried doing a record after dial it didnt work |
| 21:42.53 | ber111 | i tried doing a record pre-dial and it blocked on the record, so no dial |
| 21:42.54 | *** join/#asterisk oej (n=olle@86.85-200-224.bkkb.no) |
| 21:44.12 | Strom_M | run mixmonitor() before you run dial() |
| 21:44.51 | Strom_M | and read the documentation, because then you'd know what record() is for |
| 21:46.12 | ber111 | ok |
| 21:46.54 | ber111 | ah got it, record is for prompts |
| 21:47.08 | Strom_M | ding |
| 21:47.13 | Strom_M | or is that beep |
| 21:48.11 | anonymouz666 | I need some dialplan logic help :( |
| 21:48.59 | Strom_M | anonymouz666: so ask a question |
| 21:50.03 | *** join/#asterisk Braxus (n=bhsieh@66.147.214.164) |
| 21:52.13 | *** join/#asterisk SwK (n=SwK@dhcp64-134-34-248.bwic.chi.wayport.net) |
| 21:54.23 | k31th | any of you guys use hudlite |
| 21:57.52 | Strom_M | anonymouz666: i guess it wasn't important then, eh? |
| 22:00.31 | *** join/#asterisk Toerkeium (i=oo@dcc-hq-host-200-59-45-188.dnsba.com) |
| 22:00.43 | anonymouz666 | it is timeconditions |
| 22:00.54 | anonymouz666 | I have a peer lets say 300 |
| 22:01.11 | anonymouz666 | that has 3 times conditions in DB |
| 22:01.32 | anonymouz666 | I am thinking in a form to get this and populate the gotoiftime dynamically |
| 22:01.42 | anonymouz666 | I am using mysql() |
| 22:01.45 | anonymouz666 | it works ok |
| 22:02.01 | anonymouz666 | the problem is how to populate the gotoiftime condition without arrays |
| 22:03.10 | [TK]D-Fender | anonymouz666, read record. Check Gotoiftime. If not, repeat until end |
| 22:03.51 | ber111 | I found that mixmonitor doesnt like it when you specify a full path in the filename |
| 22:04.08 | ber111 | strangely it creates the file but doesnt fill it with samples |
| 22:04.10 | anonymouz666 | [TK]D-Fender i think populate the gotoiftime at once is better |
| 22:04.42 | vn | is the IAXy overpriced? |
| 22:05.21 | anonymouz666 | read all records. populate all gotoiftime conditions |
| 22:06.04 | ruied_ | is it possible that I make a call to ext '200' and appears at the phone the person's 200 extension name? |
| 22:06.12 | Strom_M | ruied_: yes |
| 22:06.18 | Strom_M | well, wait |
| 22:06.22 | [TK]D-Fender | anonymouz666, Thats a lot of extr reads for noting if the 1st one matches and increases load |
| 22:06.25 | Strom_M | whose phone does the displaying of what? |
| 22:07.07 | [TK]D-Fender | ruied_, Depends on the phone because *'s support of CPID is incomplete and no ETA |
| 22:07.17 | ruied_ | for example, I make a call to John Smith (ext 200) and I would like that it appears at my phone the John Smith name |
| 22:07.30 | [TK]D-Fender | ruied_, Some can do it if they have the # listed in their local directories, etc. |
| 22:07.45 | [TK]D-Fender | ruied_, But effectively speaking * does not do this. At all. |
| 22:08.05 | [TK]D-Fender | ruied_, There is a Polycom specific patch out there, but I'm not sure if its usable |
| 22:08.12 | *** join/#asterisk anthm (n=anthm@dhcp64-134-34-248.bwic.chi.wayport.net) |
| 22:08.12 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:08.27 | ruied_ | I've seen this today in an Alcatel PBX... |
| 22:08.51 | [TK]D-Fender | ruied_, Yes, but * doesn't. |
| 22:09.19 | [TK]D-Fender | ruied_, proprietary systems can do whatever cool stuff they feel like. |
| 22:09.35 | ruied_ | [TK]D-Fender, ah, he told me that was an advantage against the * |
| 22:09.46 | [TK]D-Fender | But alas its time for martial arts. Back in a few hours |
| 22:10.06 | [TK]D-Fender | ruied_, That should fall under the category of "Should I really give a shit?" |
| 22:10.10 | vn | what martial art? |
| 22:10.17 | [TK]D-Fender | ruied_, It'd better not be a cdeciding factor... |
| 22:10.25 | [TK]D-Fender | vn, Katori Shinto |
| 22:10.50 | [TK]D-Fender | vn, oldest school of samurai kobudo |
| 22:11.02 | k31th | anyone know of a software billing system for asterisk ? |
| 22:11.27 | [TK]D-Fender | bbiab |
| 22:11.52 | Strom_M | k31th: parse the CDRs and write your own bills? |
| 22:12.35 | Ch0Hag | That is possible if the destination is SIP. |
| 22:12.57 | Ch0Hag | I got half-way through hacking the sip module to send a name for the phone to display. |
| 22:13.38 | Strom_M | so guys is 1.0 out yet ?!?!?!!?!?!?! |
| 22:13.49 | Ch0Hag | PAH! |
| 22:14.43 | Ch0Hag | Out-lagged and out-n00bed. |
| 22:15.34 | Strom_M | i hope there are other people in here one day |
| 22:15.36 | Strom_M | I want my MTV |
| 22:15.45 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 22:16.07 | waKKu | folks.. good evening... |
| 22:16.13 | Strom_M | hi |
| 22:16.33 | waKKu | maybe someone can help me... i bought a digium card TDM400P + 1 module FXO .. but, how can i check if that module is FXO or FXS ? |
| 22:16.47 | Strom_M | FXO is red |
| 22:16.49 | Strom_M | FXS is green |
| 22:16.50 | Ch0Hag | The driver will tell you. |
| 22:16.57 | Ch0Hag | Or whot strom said. |
| 22:16.59 | Ch0Hag | what |
| 22:17.15 | Strom_M | insert coin to continue |
| 22:18.04 | anonymouz666 | [TK]D-Fender I am writing right now what you told me, it makes sense |
| 22:18.11 | anonymouz666 | let's see if it will work fine |
| 22:18.12 | waKKu | Strom_M hmmm... thanks ;) .. is red |
| 22:18.29 | Ch0Hag | Red is for lines. Green is for phones. |
| 22:18.52 | ruied_ | waKKu, careful with the configuration, you have to configure the signalling in in zapata.conf as an fxs (if you have an fxo) |
| 22:18.52 | Strom_M | this episode of #asterisk is brought to you by the letter Q |
| 22:19.33 | Ch0Hag | The terms FXS and FXO are confusing - they change based on factors like where the moon is and what level was reached last time the author of $document or $driver played nethack. |
| 22:19.38 | Mercestes | and the number 9. |
| 22:19.45 | waKKu | ruied_ yeah.. i know.. my doubt was with hardware ... |
| 22:19.50 | k31th | Strom_M: I can do that yeah. how would I sort the CDR's into each user ? |
| 22:20.19 | Ch0Hag | waKKu: genzaptelconf will create the zap configuration files for you. |
| 22:20.30 | Strom_M | k31th: depends on your config |
| 22:21.20 | *** join/#asterisk lorinc (n=ang@pool-2000.adsl.interware.hu) |
| 22:21.20 | *** join/#asterisk SwK (n=SwK@dhcp64-134-34-248.bwic.chi.wayport.net) |
| 22:21.32 | *** join/#asterisk ZeroPing (n=none@adsl-217-148-42.owb.bellsouth.net) |
| 22:21.49 | ZeroPing | Anyone available for a quick question? |
| 22:21.51 | k31th | Strom_M: well each customer has its own diff caller id. |
| 22:22.07 | k31th | I could filter by that, but using what ? |
| 22:22.21 | k31th | i can export to CSV |
| 22:22.55 | Strom_M | ZeroPing: never |
| 22:23.05 | Strom_M | questions are forbidden in #asterisk |
| 22:23.24 | ZeroPing | :p |
| 22:23.25 | Strom_M | instead you must talk about your problem through metaphor |
| 22:23.37 | Strom_M | (i.e. just ask the damn question already) |
| 22:23.43 | ZeroPing | Alright |
| 22:23.56 | ZeroPing | Is it possible to use a regular phone line with Asterisk instead of a VOIP line? |
| 22:23.56 | Strom_M | k31th: perl? |
| 22:24.00 | Strom_M | ZeroPing: yes |
| 22:24.07 | Ch0Hag | It would be thee art of mime, but that's rathre hard to conver digitlaly. |
| 22:24.17 | Strom_M | Ch0Hag: interpretive dance! |
| 22:24.19 | Ch0Hag | s/thee/through the/ |
| 22:24.30 | Ch0Hag | That jbot really freaks me out. |
| 22:24.30 | ZeroPing | Storm_M: Alright, I'm planning on using the Cisco 7940 phones, how do I get those to work with Asterisk? |
| 22:24.43 | Strom_M | ZeroPing: with sip |
| 22:24.47 | Strom_M | and typing |
| 22:25.03 | Ch0Hag | Appauling spelling brought to you by virtue of resetting /home's NFS server while X is running. |
| 22:25.08 | ZeroPing | I was reading that I'll have to change the firmwire on them to work with Asterisk. Is that right? |
| 22:25.22 | Strom_M | ZeroPing: if they're running sccp firmware, yes |
| 22:25.23 | Ch0Hag | Don't. Cisco phones suck. |
| 22:25.27 | Ch0Hag | Flog them on ebay. |
| 22:25.39 | Strom_M | i run cisco /and/ polycom |
| 22:25.41 | Strom_M | I must be bi |
| 22:25.51 | ZeroPing | Well, I was planning on picking them up on eBay anyway. |
| 22:25.56 | ZeroPing | Too damn expensive anywher else. |
| 22:26.03 | Ch0Hag | You don't have them yet? |
| 22:26.06 | Strom_M | ZeroPing: look into the polycom ip320 |
| 22:26.11 | Strom_M | two lines and a hell of a lot cheaper |
| 22:26.26 | Strom_M | s/320/330/ |
| 22:26.32 | Ch0Hag | Two lines? |
| 22:26.40 | Strom_M | well, two line appearances |
| 22:27.03 | irule | do labels work with gotoif? |
| 22:27.09 | Strom_M | but that's certainly better than two puddings or grandstream |
| 22:27.11 | Ch0Hag | Do any phones not have that? |
| 22:27.13 | Strom_M | Ironhand: yes |
| 22:27.18 | ZeroPing | So my Asterisk box will just need a regular dialup modem to work? |
| 22:27.18 | Strom_M | Ch0Hag: yes |
| 22:27.22 | Strom_M | ZeroPing: no |
| 22:27.29 | Strom_M | ZeroPing: digium TDM01B |
| 22:27.33 | ZeroPing | Besides the Ethernet port that links all the phones together with a switch. |
| 22:27.37 | ZeroPing | Oh, okay. |
| 22:27.45 | Ch0Hag | Or perhaps you are referring to something I'm not thinking of. |
| 22:28.47 | *** join/#asterisk Innatech (n=daf@netblock-72-25-97-119.dslextreme.com) |
| 22:28.51 | Strom_M | pressing the any key |
| 22:28.54 | Strom_M | not continuing |
| 22:28.55 | Strom_M | halp |
| 22:29.14 | ZeroPing | So even if I'm using just one phone line, I have to have the Digium product instead of a dialup modem? |
| 22:29.29 | Mercestes | <PROTECTED> |
| 22:29.31 | Strom_M | ZeroPing: yes |
| 22:29.37 | ZeroPing | Ah, damn. |
| 22:29.52 | *** join/#asterisk jetlagmk2 (n=jetlag@pool-70-17-47-67.pskn.east.verizon.net) |
| 22:29.59 | Ch0Hag | I believe you can get cheap external ones. |
| 22:30.25 | ZeroPing | I haven't seen the external ones yet. I'm looking at the internal ones on eBay, and they're priced at $149.40. |
| 22:30.37 | Strom_M | there is no dialup modem that works as an FXO interface |
| 22:31.05 | ZeroPing | Yeah, I wasn't aware. |
| 22:31.15 | ruied_ | I'm trying to install an winbond w699 with mISDN driver, I'm making a compilation with a mISDN patch (going to try after the compilation finishes). Yesterday I've tried to make a module with the kernel headers installed, but the kernel couldn't load the module. I'm using debian etch... should the mISDN be patched and compiled in the kernel always, or did I made something wrong and the module can always be compiled (with just the kernel header |
| 22:31.15 | ruied_ | s) and loaded by the kernel? |
| 22:31.32 | Ch0Hag | I think I have a problem with the wiring in my house. |
| 22:31.42 | Ch0Hag | That's going to be incredibly boring to check out. |
| 22:31.55 | ZeroPing | The bad thing about this damn house is I can't string Ethernet lines all over the place for these phones. |
| 22:32.01 | Strom_M | wifi! |
| 22:32.11 | ZeroPing | I thought the latency would be bad. |
| 22:32.11 | Supaplex | fiber optic! |
| 22:32.19 | Ch0Hag | Can't physically or can't because you're not allowed? |
| 22:32.23 | Strom_M | and if i can shoehorn ethernet into this 70 year old apartment, you can do it too |
| 22:32.26 | Supaplex | ouu ouuu, tin cans! |
| 22:32.31 | ZeroPing | Ch0Hag: Both |
| 22:32.47 | Ch0Hag | I only have the latter problem. |
| 22:32.53 | Ch0Hag | It's incredibly frustrating. |
| 22:32.56 | Supaplex | cat3 can do 10bt |
| 22:32.58 | ZeroPing | How would I go about making a Cisco 7940 wireless? |
| 22:33.04 | irule | http://pastebin.ca/594402 this is where continua works well, but I get an error message pbx_extension_helper: No such label 'cuelga' in extension 's' in context 'default' with label cuelga! :s |
| 22:33.10 | Supaplex | ZeroPing: wire cutters ;) |
| 22:33.23 | ZeroPing | :p |
| 22:33.33 | ZeroPing | Communicating with WiFi, rather. |
| 22:33.46 | VJFROMGT | is http://www.asteriskguru.com/ down or is it me? |
| 22:33.50 | Strom_M | irule: change your gotoif line from s,cuelga to just cuelga |
| 22:34.21 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 22:34.27 | ruied_ | VJFROMGT, it's down... |
| 22:34.53 | ZeroPing | I was thinking about picking up some of those Ethernet adapters that have a base station and then another adapter for wherever you want an Ethernet port, but I didn't know how well those would work with IP phones. |
| 22:34.56 | irule | Strom_M believe it not not, I did try cuelga alone before posting with you and trying s,cuelga, this is 1.2.19 |
| 22:35.52 | irule | <PROTECTED> |
| 22:36.38 | Strom_M | odd |
| 22:38.30 | irule | very odd |
| 22:39.58 | *** join/#asterisk Swat2 (n=bler@218-215-201-148.people.net.au) |
| 22:40.17 | *** join/#asterisk lee_is_me (n=chatzill@12-201-102-196.client.mchsi.com) |
| 22:42.26 | lee_is_me | In contexts that are used to outbound dialing, are there any adverse effects to having a "t" exten with Hangup() function? |
| 22:44.21 | *** join/#asterisk mxmasster (n=Max@129.47.12.101) |
| 22:44.22 | irule | then look, I simplified, it does go to continua propperly but once it counts up, I still get the invalid error message http://pastebin.ca/594424 |
| 22:44.23 | mxmasster | hi all |
| 22:44.29 | lee_is_me | I'm asking because I have a customer where if they attempt to dial out on FXO and then hangup at the right moment, the call seems to still go through at times and then causes a ring back. |
| 22:45.02 | mxmasster | I am looking for a "simple" rating engine to put prefixes and costs in so I can compare what our bill would be with different vendors |
| 22:45.30 | Mercestes | mxmasster: mysql and some really simple php programming |
| 22:45.35 | Mercestes | ... |
| 22:45.38 | Mercestes | you don't even need the php really |
| 22:45.54 | mxmasster | Mercestes: great, what is the URL to the app |
| 22:46.04 | Mercestes | To mysql? |
| 22:46.16 | mxmasster | no to the simple rating engine you are proposing I use |
| 22:46.17 | Mercestes | http://www.google.com |
| 22:46.31 | Mercestes | did you even read my response? |
| 22:46.46 | mxmasster | yes I did - did you read my question |
| 22:46.53 | Mercestes | yesss.... |
| 22:46.55 | Mercestes | and I answered it |
| 22:47.23 | mxmasster | So my question was not - I would like to program a system to do rating or import into mysql or figure out some complicatate query but |
| 22:47.35 | mxmasster | are there any out there that do this that i can download |
| 22:48.24 | Mercestes | ~wglwat |
| 22:48.25 | jbot | it has been said that wglwat is well, good luck with all that |
| 22:48.37 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 22:59.30 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 23:02.22 | irule | exten => s,n(continua),SetVar(DEFTIMEOUT=$[${DEFTIMEOUT} + 1]) Jun 28 15:12:49 WARNING[3048]: ast_expr2.fl:183 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected TOK_PLUS, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input : + 1 (with an arrow under + sign) Jun 28 15:12:49 WARNING[3048]: ast_expr2.fl:187 ast_yyerror: If you have questions, please refer to doc/README.variables in the asterisk source., well it is working propperly, |
| 23:02.22 | irule | <PROTECTED> |
| 23:03.46 | Mercestes | Try set instead of Setvar. Otherwise, your syntax looks correct to me, honestly. |
| 23:03.48 | *** part/#asterisk woolbeo (n=woolbeo@exchange.services.daqe.com) |
| 23:04.02 | Mercestes | and I could be wrong on the set v/s setvar. |
| 23:06.48 | *** join/#asterisk wundaboy (n=pat@c-24-21-71-88.hsd1.mn.comcast.net) |
| 23:07.20 | wundaboy | why do i get: Call rejected by 66.227.100.30: No such context/extension why i try and make a call? |
| 23:07.28 | wundaboy | *when i try to make a call. |
| 23:08.49 | Hogie | because there is no such context/extension at that host |
| 23:10.21 | *** join/#asterisk ruied__ (n=ruied@bl7-211-105.dsl.telepac.pt) |
| 23:10.33 | wundaboy | but its just a phone number |
| 23:10.37 | wundaboy | its supposed to work... |
| 23:14.51 | Hogie | then the context is wrong |
| 23:19.29 | SuPrSluG | wundaboy, maybe include is missing from the context your trying to dial out. |
| 23:19.51 | wundaboy | what do you mean? |
| 23:20.16 | wundaboy | SuPrSluG, what do you mean? |
| 23:20.31 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 23:20.37 | SuPrSluG | if you don't include, say local or default in you outbound context |
| 23:21.00 | wundaboy | im confuzed |
| 23:21.12 | wundaboy | in the outbound context in iax.conf? |
| 23:21.15 | SuPrSluG | it doesn't have permission to use that context |
| 23:21.25 | wundaboy | how do i give it permission? |
| 23:35.16 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 23:35.18 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 23:37.13 | SuPrSluG | wundaboy, is this a voip provider or pstn |
| 23:47.03 | *** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) |