| 00:00.29 | bkruse_home | svn, always |
| 00:00.35 | bkruse_home | but........try both |
| 00:00.36 | *** join/#asterisk Zorix (n=Brandon@c-76-101-72-47.hsd1.fl.comcast.net) |
| 00:00.38 | bkruse_home | is it an agent deadlock? |
| 00:01.01 | Zorix | anyone familiar with unlocking an spa2002-er for asterisk use? |
| 00:01.23 | bkruse_home | Zorix: you have to unlock it? |
| 00:01.25 | FuriousGeorge | bkruse_home: i know the symptoms but i dont know what causes it. |
| 00:01.26 | bkruse_home | dang, didnt know that |
| 00:01.32 | FuriousGeorge | i have suspicions |
| 00:01.35 | Zorix | yea its set to be provisioned by earthlink |
| 00:01.36 | bkruse_home | FuriousGeorge: do you have agents/queues? |
| 00:01.39 | FuriousGeorge | no |
| 00:01.48 | bkruse_home | oh, that would be the problem :P |
| 00:01.50 | bkruse_home | IAX?! |
| 00:01.50 | bkruse_home | FuriousGeorge: is it dumping cores? |
| 00:02.09 | FuriousGeorge | no, the cli just gets non-responsive. i cant take or make calls |
| 00:02.23 | Zorix | it doesnt seem like it can be unlocked but i figured i would ask |
| 00:02.24 | FuriousGeorge | sometimes its prefaced by an incoming phantom call no one can answer |
| 00:02.43 | FuriousGeorge | ~s/preceeded/prefaced |
| 00:03.47 | FuriousGeorge | i also notice that by default asterisk built with -march=k8, whereas gentoo recommends using -march=opteron if anything |
| 00:04.27 | FuriousGeorge | now that i read that i remember once i accidentally set my -march to athlon64 and a lot of linux instability happened |
| 00:05.42 | bkruse_home | can anyone give me temporary access to a box with at t1 card in it? |
| 00:07.01 | bkruse_home | i guess thats a ...............no ;[ |
| 00:07.34 | anonymouz666 | I am not crazy enough :) |
| 00:07.38 | anonymouz666 | lol=very |
| 00:11.06 | FuriousGeorge | i should ask easier questions :) |
| 00:12.02 | FuriousGeorge | i want to go away from tdm400p, which i have had nothing but problems with, especially when it comes to fxs |
| 00:12.07 | *** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@145-27.mc.cite.net) |
| 00:12.38 | FuriousGeorge | im looking at these linksys pap2s, which are super cheap. you get two fxs channels for ~60USD |
| 00:12.59 | FuriousGeorge | can someone tell me why this is inadvisable, b/c it must be too good to be true |
| 00:13.04 | Zorix | i got screwqed with a locked version |
| 00:13.20 | _VoiceMeUp_COM | send it back |
| 00:13.25 | FuriousGeorge | Zorix: fair enough, but lets assume i dont buy the locked version |
| 00:13.26 | _VoiceMeUp_COM | chargeback your card if they dont want to |
| 00:13.35 | Zorix | i already opened it i didnt know for sure |
| 00:13.48 | Zorix | it was $28 so i cant complain |
| 00:14.00 | _VoiceMeUp_COM | so what.., if the sellign ocmpany knew and didd that to you then send it back |
| 00:14.01 | FuriousGeorge | omg, you have to open them to unlock them? |
| 00:14.03 | _VoiceMeUp_COM | ? |
| 00:14.13 | _VoiceMeUp_COM | Zotrix.. send me 28$ if you dont mind it |
| 00:14.20 | Zorix | no idea i played around with the internal jumper |
| 00:14.20 | _VoiceMeUp_COM | my paypal is .... |
| 00:14.35 | Zorix | i usually dont back away from a hardware hacking challenge |
| 00:14.42 | _VoiceMeUp_COM | people go around and screw everyone and the reason is no one acts on it |
| 00:15.03 | _VoiceMeUp_COM | so then we have nigeria frauds still scamming us.. until one guy told chris hansens and dateline made a show.. |
| 00:15.51 | J4k3 | how exactly can a pap2 be 'locked' |
| 00:15.57 | J4k3 | ng/AndyCap] has quit [Nick |
| 00:15.57 | J4k3 | <PROTECTED> |
| 00:15.57 | J4k3 | ng/AndyCap] has quit [Nick |
| 00:15.57 | J4k3 | <PROTECTED> |
| 00:16.06 | FuriousGeorge | if i wanted 4 fxs channels using a tdm400p, i would have to pay at least 300USD, but i can do the same thing with pap2s for ~120 dolloars. other than a few more wires, is there any negative here for a 4 fxs or less setup? |
| 00:16.07 | _VoiceMeUp_COM | ? |
| 00:16.23 | _VoiceMeUp_COM | they set it to a provider and trow away the key |
| 00:16.25 | FuriousGeorge | J4k3: the firmware is hardlocked to use only, for instance, vonage |
| 00:16.48 | _VoiceMeUp_COM | then your screwed witha crappy vonage like service that needs to say that the quality is ok on commercials they are so bad |
| 00:18.04 | FuriousGeorge | see, yesterday i had two fxs modules just stop working. they are no longer providing dialtone, much less enough voltage to ring the phone, but asterisk doesnt know it. the channel just rings, but the phone is dead |
| 00:18.23 | _VoiceMeUp_COM | yeah |
| 00:18.45 | FuriousGeorge | this is the second time it has happened to on this particular card, so that makes it my 3rd and fourth fxs module that just fried |
| 00:18.48 | _VoiceMeUp_COM | the manufacturers put a end of service in the eeproms.. usually is the 22rd of the month |
| 00:19.16 | _VoiceMeUp_COM | j/k |
| 00:19.34 | _VoiceMeUp_COM | again |
| 00:19.51 | _VoiceMeUp_COM | FuriousGeorge , i would go a200's or something for 4 analog lines |
| 00:19.52 | FuriousGeorge | _VoiceMeUp_COM: heh. im gonna call digium tomorrow, see how their customer service is |
| 00:20.00 | _VoiceMeUp_COM | pap's are.. hmm not as good |
| 00:20.01 | _VoiceMeUp_COM | imho |
| 00:20.06 | FuriousGeorge | _VoiceMeUp_COM: ive thought of that |
| 00:20.07 | _VoiceMeUp_COM | oh and good luck |
| 00:20.27 | bkruse_home | please..........anyone, you can screen in, just need to look at /proc/zap, gah |
| 00:20.29 | FuriousGeorge | but why would a sangoma hardware work be any better with asterisk than hardware from digium |
| 00:20.38 | _VoiceMeUp_COM | ;) |
| 00:20.44 | _VoiceMeUp_COM | #1 echo can on board |
| 00:20.51 | _VoiceMeUp_COM | no need for buying a software one ;) |
| 00:21.00 | _VoiceMeUp_COM | #2 , no reason it just works |
| 00:21.09 | _VoiceMeUp_COM | #3 i dotn know waht im talking ab out i never buy dig hard |
| 00:21.28 | Qwell | _VoiceMeUp_COM: then go ahead and bite your tongue |
| 00:21.32 | _VoiceMeUp_COM | i did that |
| 00:21.33 | _VoiceMeUp_COM | ;) |
| 00:21.40 | _VoiceMeUp_COM | hurts like qwell |
| 00:21.46 | Qwell | I mean, you don't want me to say how horrible voicemeup.com is compared to nufone |
| 00:21.54 | _VoiceMeUp_COM | ;) lol |
| 00:21.57 | Mercestes | <PROTECTED> |
| 00:22.00 | _VoiceMeUp_COM | i never said digium was horrible |
| 00:22.08 | _VoiceMeUp_COM | he asked me difference.. i gave him one |
| 00:22.24 | bkruse_home | Mercestes: just to read and look at what it looks like when a digital card is loaded/synced up, vs not |
| 00:22.28 | _VoiceMeUp_COM | but i dont really want to start a piss contest ;) so ill /quiet now |
| 00:22.34 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 00:22.48 | dlynes_laptop | _VoiceMeUp_COM: too late....moggie already left |
| 00:23.07 | bkruse_home | dlynes_laptop: no, his box just froze, haha |
| 00:23.10 | dlynes_laptop | oh |
| 00:23.11 | dlynes_laptop | heh |
| 00:23.18 | _VoiceMeUp_COM | lol |
| 00:23.35 | _VoiceMeUp_COM | google for your comaprison youll see |
| 00:23.40 | dlynes_laptop | I guess he's running Linux in a vm under Windows? |
| 00:23.59 | bkruse_home | dlynes_laptop: ha, no, i think he just had to restart, or just exited out of gajim |
| 00:24.23 | bkruse_home | Mercestes: can you help me out? |
| 00:24.46 | Mercestes | Hm, a company, who advertises via an IRC nick, promoting Sangoma over Digium....and has never used a Digium card.... |
| 00:25.05 | _VoiceMeUp_COM | http://www.google.com/history |
| 00:25.07 | Mercestes | Nice. how does a VoIP provider never manage to get their hands on a digium card? :D |
| 00:25.13 | bkruse_home | Mercestes: that is kinda lame, i agree |
| 00:25.14 | FuriousGeorge | sorry i walked away for a sec |
| 00:25.20 | _VoiceMeUp_COM | is oole now logging your web history ? |
| 00:25.28 | FuriousGeorge | Qwell, that was a bit harsh of you |
| 00:25.32 | _VoiceMeUp_COM | You know that great web site you saw online and now can't find? From now on, you can. With Web History, you can view and search across the full text of the pages you've visited, including Google searches, web pages, images, videos and news stories. Y |
| 00:25.33 | _VoiceMeUp_COM | wow |
| 00:25.39 | Mercestes | bkruse_home, Sec. |
| 00:25.42 | bkruse_home | Mercestes: woot |
| 00:25.48 | bkruse_home | you can screen, just look at an idea i had |
| 00:25.57 | Mercestes | bkruse_home, I can't give you access bu tI can pastebin the contents fo ryou |
| 00:26.08 | bkruse_home | Mercestes: that would be great |
| 00:26.11 | _VoiceMeUp_COM | talk about invasion..im sure htey had 20% of surfers habits now they needed a way to ask perm for the last 80% to put spy ads on your sites |
| 00:26.30 | _VoiceMeUp_COM | Mercestes i configure them for clients, |
| 00:26.36 | _VoiceMeUp_COM | but we buy from one source only |
| 00:26.49 | _VoiceMeUp_COM | its easier to handle rma's but never needed yet.. |
| 00:27.01 | _VoiceMeUp_COM | ony get 102d's and 104'd yet, all rest is cisco's 54xx |
| 00:27.06 | Mercestes | You seem to talk a great deal about things you have no experience in. |
| 00:27.19 | _VoiceMeUp_COM | so we use those as testing boxes at new pri location until we provision thebig boys |
| 00:27.27 | _VoiceMeUp_COM | ? |
| 00:27.32 | _VoiceMeUp_COM | omg lol |
| 00:28.31 | Mercestes | bkruse_home, http://pastebin.ca/455107 |
| 00:28.38 | bkruse_home | Mercestes: did you get my pm? |
| 00:28.38 | Mercestes | Two different PRis, EoF'd by ============ |
| 00:29.14 | Mercestes | _VoiceMeUp_COM, seriously. You prefer Sangoma over Digium, but you've never owned a Digum, and you say Sangoma RMAs are easier (but you've never used it.) |
| 00:29.27 | FuriousGeorge | dlynes_laptop: if you were referring to me, im not using asterisk in a vm, ive just had nothing but problems with systems with tdm400p. im not making a statement about all digium hardware, or even saying its all digiums fault. perhaps an engineer of some sort would have better luck than myself. |
| 00:29.34 | _VoiceMeUp_COM | no i said its easier for us to deal with one company |
| 00:29.35 | _VoiceMeUp_COM | then 2-3 |
| 00:29.41 | _VoiceMeUp_COM | so the rma process will be easier |
| 00:29.43 | FuriousGeorge | but i cant be all my fault either. maybe asterisk/zaptel doesnt like amd/nforce platform, which is what i always use. all i know is that more use of tdm400p->more deadlocks. |
| 00:29.50 | dlynes_laptop | FuriousGeorge: no...it was a joke in reference to mog's computer going down :) |
| 00:29.51 | Mercestes | _VoiceMeUp_COM, I'm not trying to troll or start a fight here, but, you are promoting a business and kinda.....casting yourself (and your business) in an unfavorable light. |
| 00:30.11 | FuriousGeorge | dlynes_laptop: oh, :) since i was asking about pap2s i thought it might could have been for me |
| 00:30.14 | bkruse_home | Mercestes: this is awesome......could you also show one thats not synced up? |
| 00:30.17 | _VoiceMeUp_COM | [20:27] _VoiceMeUp_COM: but we buy from one source only |
| 00:30.17 | _VoiceMeUp_COM | [20:28] _VoiceMeUp_COM: its easier to handle rma's but never needed yet.. |
| 00:30.19 | _VoiceMeUp_COM | that what i said |
| 00:30.27 | dlynes_laptop | FuriousGeorge: nah...how would a pap2 crash your system? |
| 00:30.29 | Mercestes | FuriousGeorge, What problems are you having? I'm coming in late. |
| 00:30.32 | bkruse_home | Mercestes: or is that impossible |
| 00:30.54 | Mercestes | bkruse_home, Not synced up as in clock master? |
| 00:31.05 | Mercestes | bkruse_home, Or not synced up as in unplugged? |
| 00:31.11 | dlynes_laptop | FuriousGeorge: but, personally, i'd be surprised if you could get sangoma or digium hardware working in a vm |
| 00:31.22 | FuriousGeorge | Mercestes: the problem is i cant tell you what the problem is. i have two almost identical mbs (one has onbard sound) one doesnt. one has been in production two weeks, the other 4 months. the former has deadlocked twice already |
| 00:31.24 | dlynes_laptop | FuriousGeorge: I wouldnm't think either one would work very well in a vm |
| 00:31.48 | Mercestes | FuriousGeorge, I think you've already decided what's wrong. |
| 00:31.52 | FuriousGeorge | they only other difference is that the former also has two maxed out tdm400p cards that get used basically with every incomming call |
| 00:31.53 | bkruse_home | Mercestes: just unplugged, or put them in the wrong signalling type would be even better |
| 00:32.00 | _VoiceMeUp_COM | dlynes_laptop i heard people got it stable under xen.. |
| 00:32.08 | _VoiceMeUp_COM | where vmware had irq sharing problems |
| 00:32.08 | FuriousGeorge | dlynes_laptop: im assuming you're joking again :) |
| 00:32.11 | FuriousGeorge | so its funny |
| 00:32.18 | Mercestes | bkruse_home, Like from CPE to NET maybe? |
| 00:32.25 | _VoiceMeUp_COM | notice i said heard,, and i never ued any of the 2 for hardware so i coudnt say |
| 00:32.37 | bkruse_home | Mercestes: sure, or anything besides b8zs/esf |
| 00:32.48 | bkruse_home | so i can see what it look like when it errors, and when it synces would be awesome, much appreciated |
| 00:32.49 | dlynes_laptop | FuriousGeorge: nope...I just can't see how anything that uses a lot of interrupts would function well in a vm |
| 00:32.59 | _VoiceMeUp_COM | yeah i know.. |
| 00:33.02 | FuriousGeorge | dlynes_laptop: again, im not using a vm |
| 00:33.15 | _VoiceMeUp_COM | i think its a bad idea all together ..but on the lists some had it ok with xen .. |
| 00:33.20 | dlynes_laptop | FuriousGeorge: ah...ok |
| 00:33.22 | FuriousGeorge | gentoo linux, nforce/amd platform |
| 00:33.31 | dlynes_laptop | FuriousGeorge: sorry for the confusion |
| 00:33.35 | FuriousGeorge | np ;) |
| 00:33.40 | Mercestes | bkruse_home, I can't unsync it because I have a caller on the phone. I *can* unplug it real quick however if you wish. |
| 00:33.41 | _VoiceMeUp_COM | also heard people saying asteirsk wont run on a embedeed VIA board |
| 00:33.48 | _VoiceMeUp_COM | ill have to test this tonight |
| 00:33.50 | dlynes_laptop | _VoiceMeUp_COM: yes it will |
| 00:34.02 | bkruse_home | Mercestes: that will work |
| 00:34.11 | _VoiceMeUp_COM | cool then i wont waste time |
| 00:34.16 | dlynes_laptop | _VoiceMeUp_COM: and it has problems with both digium and sangoma cards |
| 00:34.24 | bkruse_home | unplug one, itll be the same thing, not getting async, then show me cat /proc/zaptel/span# of the one you unplugged, thanks man |
| 00:34.36 | FuriousGeorge | i just had what alcoholics refer to as a moment of clarity. im oging to buy the cheapest snom i can find, and sell the tdm400p with the fxs in it |
| 00:34.39 | dlynes_laptop | _VoiceMeUp_COM: it has less problems with sangoma cards, but it still has problems with both |
| 00:34.42 | bkruse_home | s/thanks man/thanks :]/g |
| 00:35.02 | _VoiceMeUp_COM | yeah i assumed that , could it be a timing thing ? |
| 00:35.09 | *** join/#asterisk Sonjoshuaz (i=PJirc@pool-71-110-74-237.lsanca.dsl-w.verizon.net) |
| 00:35.19 | Sonjoshuaz | Hello |
| 00:35.32 | dlynes_laptop | _VoiceMeUp_COM: It's because those embedded via systems usually have 3 or 4 pieces of hardware all sharing the same interrupt |
| 00:35.49 | _VoiceMeUp_COM | so i take it the trixbox applicance and digium applicance are both full scaled motherboards ? |
| 00:35.49 | dlynes_laptop | _VoiceMeUp_COM: and so the digium and sangoma cards don't get the interrupts when they expect to receive them |
| 00:36.05 | Sonjoshuaz | Any one Nows Sip For Free |
| 00:36.15 | _VoiceMeUp_COM | or did digium make a version that works on them for that hardware |
| 00:36.18 | bkruse_home | Mercestes: and, if possible, a dump of /dev/zap/ctl, if not/doesnt work, thats fine, i appreciate it |
| 00:36.26 | dlynes_laptop | _VoiceMeUp_COM: sangoma cards are a little less timing sensitive for interrupts, but even sangoma cards still expect to receive the interrupts within a reasonable amount of time |
| 00:37.11 | dlynes_laptop | Sonjoshuaz: advertising is not accepted in this channel, I don't think |
| 00:37.40 | Sonjoshuaz | I do not want to Advertis |
| 00:37.45 | dlynes_laptop | _VoiceMeUp_COM: both cards have their strengths |
| 00:37.47 | Qwell | dlynes_laptop: I don't think he is, heh |
| 00:37.51 | _VoiceMeUp_COM | i think he wants a FEE sip provider |
| 00:37.52 | _VoiceMeUp_COM | free |
| 00:37.55 | Sonjoshuaz | Iam looking for a Sip Provider |
| 00:38.16 | dlynes_laptop | Qwell: I thought he was advertising for a service with the motto 'Now Sips For Free' |
| 00:38.17 | Sonjoshuaz | Free if Possible |
| 00:38.32 | _VoiceMeUp_COM | well you get waht you pay for.. |
| 00:38.44 | Sonjoshuaz | I am new to Trixbox and Asterisk |
| 00:38.54 | Sonjoshuaz | I know |
| 00:38.57 | Mercestes | bkruse_home, It's my president that's on the phone and I've had some sync issueson my PRI so lte me wait until he's off and I'll break it for you |
| 00:38.58 | Sonjoshuaz | ha ha |
| 00:39.03 | Mercestes | ~trixbox |
| 00:39.06 | jbot | Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org |
| 00:39.06 | bkruse_home | Mercestes: ha, no problem :] |
| 00:39.24 | _VoiceMeUp_COM | free = zero support , gamble on line quality and avail , cheap = some or no suport and downtime that would make a crack addict a happy camper.. and expensive = tdm , support and fun |
| 00:39.25 | Mercestes | bkruse_home, /zap/ctl before or after I braek it? |
| 00:40.13 | bkruse_home | Mercestes: either or, shouldnt matter |
| 00:40.16 | Sonjoshuaz | I want to see that my Trixbox Works i set it up allready |
| 00:40.33 | Sonjoshuaz | on a spare machine |
| 00:40.44 | bkruse_home | Sonjoshuaz: http://asteriskNOW.org |
| 00:41.10 | Sonjoshuaz | Thanks bkruse home :) |
| 00:41.24 | bkruse_home | ;] np |
| 00:42.12 | Sonjoshuaz | i allready set up Trixbox i need a Sip Line |
| 00:43.15 | *** join/#asterisk hfb (n=hfb@pool-72-67-156-130.lsanca.dsl-w.verizon.net) |
| 00:46.49 | *** join/#asterisk kavit (n=kavit@ppp167-236-231.static.internode.on.net) |
| 00:47.06 | Mavvie | on a Quad PRI card... if none of the spans are receiving a clock, will the card provide its own clock then to all spans? |
| 00:47.58 | JerJer[mobile] | load average: 61.03, 129.95, 94.48 |
| 00:48.28 | JerJer[mobile] | <PROTECTED> |
| 00:48.31 | _VoiceMeUp_COM | jerjer lol |
| 00:48.34 | _VoiceMeUp_COM | nice |
| 00:48.56 | JerJer[mobile] | v1.4 |
| 00:49.12 | bkruse_home | JerJer[mobile]: nice nice |
| 00:49.25 | JerJer[mobile] | dual xeon 3gig |
| 00:49.29 | JerJer[mobile] | sip |
| 00:49.34 | JerJer[mobile] | ulaw |
| 00:49.45 | _VoiceMeUp_COM | real or generated? |
| 00:49.52 | JerJer[mobile] | playing white noise in both directions |
| 00:49.56 | JerJer[mobile] | out of a ram disk |
| 00:51.04 | JerJer[mobile] | _VoiceMeUp_COM: simulated traffic |
| 00:51.53 | _VoiceMeUp_COM | ah;) k |
| 00:52.13 | JerJer[mobile] | MOS testing will come in a few hours |
| 00:52.18 | JerJer[mobile] | maybe tomorrow |
| 00:53.18 | Mavvie | JerJer[mobile]: is somebody of RAD there too? |
| 00:54.47 | JerJer[mobile] | doesn't sound familiar |
| 00:55.16 | Mavvie | JerJer[mobile]: they have a product called IPMux which is supposed to be PRI over IP networks. |
| 00:55.35 | Mavvie | it euhm... doesn't play nice with Digium products yet. |
| 00:55.48 | JerJer[mobile] | ahh |
| 00:56.07 | Mavvie | or to be more clear: I haven't been able to get it working without hundreds of HDLC errors. |
| 00:56.38 | JerJer[mobile] | talk to creslin |
| 00:56.50 | JerJer[mobile] | better yet send him a trace |
| 00:57.53 | Mavvie | the Matthew Fredrickson creslin one? |
| 00:58.08 | *** join/#asterisk cspot (i=cspot@ip68-109-8-207.pn.at.cox.net) |
| 00:59.44 | *** join/#asterisk bmd (n=bmd@72.54.252.34) |
| 00:59.54 | shmaltz | anybody here have a good link for guidelines what a local governments bids should look like? |
| 00:59.55 | JerJer[mobile] | yup |
| 01:01.33 | _VoiceMeUp_COM | her egov bids are a strip club tour and a handshake followed by an enveloppe filled with green |
| 01:01.40 | _VoiceMeUp_COM | probably same in usa and everywhere else |
| 01:02.24 | _VoiceMeUp_COM | meaning in my experience, with them it always went to shady companies that did hosting @ 1500 per page.. 400$ counters.. and a 2.500$ update fee |
| 01:02.35 | _VoiceMeUp_COM | now that cant be bidded.. |
| 01:02.56 | shmaltz | _VoiceMeUp_COM, this is not the case here |
| 01:03.07 | drfreeze | Can someone tell me what it means when teh Polycom phone says "Waiting for network to intialize"? |
| 01:03.09 | _VoiceMeUp_COM | k |
| 01:03.12 | _VoiceMeUp_COM | good to hear |
| 01:03.48 | shmaltz | it's just that I need to compete against someone that has written a proposal |
| 01:04.12 | shmaltz | and the gov office that needs the system insist it has to be those items that are on the first proposal |
| 01:04.20 | *** join/#asterisk thoughtpolice (n=austin@c75-111-146-82.plaicmtc01.tx.dh.suddenlink.net) |
| 01:04.51 | shmaltz | while I argue that with a system like that all that has to be provided in the RFP is the system requirements and it's up to the bidders to decide on the system |
| 01:05.03 | shmaltz | any help with this? |
| 01:07.09 | JT | drfreeze: it means you should plug the phone into an ethernet port |
| 01:12.41 | mcab | drfreeze: it's waiting for DHCP |
| 01:13.29 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:16.22 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.169.195) |
| 01:18.06 | *** part/#asterisk danicholson (n=danichol@203.89.191.222) |
| 01:18.32 | Mercestes | shmaltz, What do they *want*? |
| 01:18.57 | shmaltz | Access control integrated with the fire/burglar alarm and video system |
| 01:19.53 | *** join/#asterisk khronos (n=khronos@c-76-110-134-230.hsd1.fl.comcast.net) |
| 01:20.02 | Mercestes | Is this all supposed to be integrated or seperated systems? |
| 01:20.22 | Mercestes | Is * even involved in this bid? LOL |
| 01:21.03 | khronos | Hi guys. |
| 01:21.36 | khronos | Am placing some calls between a couple of servers and if I call from my machine the audio breaks up on my sending after a couple mins. |
| 01:21.49 | khronos | The log file on the other server says: |
| 01:21.51 | khronos | [Apr 23 18:10:16] WARNING[18724] chan_iax2.c: Resyncing the jb. last_delay 118943, this delay -11600, threshold 101292, new offset 21303 |
| 01:22.08 | khronos | This only happens when I place an iax call to the server. |
| 01:22.21 | khronos | If the server calls mine the audio is fine for as long as we wish to talk. |
| 01:25.38 | _VoiceMeUp_COM | i thin kthat oculd mean the jitter bucket was overwelmed by the lag and decided it coudnt keep up |
| 01:28.18 | khronos | What system do I need to change the setting on? Mine or the remote? |
| 01:28.20 | khronos | or both? |
| 01:30.01 | *** join/#asterisk kd6cae (n=fileshar@71-83-150-196.dhcp.rvsd.ca.charter.com) |
| 01:30.12 | kd6cae | hi all |
| 01:30.44 | drfreeze | Helo |
| 01:31.03 | kd6cae | howdy, new to asterisk but absolutely love what it can do! |
| 01:31.21 | drfreeze | A person plugged a polycom 501 phone into an analog port today, now the phone won't get an ip address via dhcp |
| 01:31.37 | drfreeze | did they toast the phone? |
| 01:32.18 | JT | probably |
| 01:32.20 | JT | what an idiot |
| 01:32.45 | JT | an analogue port has a DC battery voltage of -48VDC and a ring voltage of 90VAC@20Hz |
| 01:33.56 | hal2k | they shouldn't use the same plugs |
| 01:34.51 | khronos | If we're talking about an ip phone here the ip phone should have an rj45 and analog rj11. |
| 01:35.10 | JT | khronos: err analogue is rj-45 in a lot of places now |
| 01:35.15 | JT | like businesses |
| 01:35.17 | hal2k | yes, but rj11 fits perfectly into rj45 |
| 01:35.25 | JT | that too |
| 01:37.01 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 01:37.59 | ManxPower | Qwell Qwell[] Looks like I found some software to do what I want with the media conversion |
| 01:38.17 | Qwell | cool |
| 01:38.27 | Qwell | I flash on the former, fyi |
| 01:40.38 | *** join/#asterisk plattypus1 (n=venom@71-83-150-196.dhcp.rvsd.ca.charter.com) |
| 01:41.40 | kd6cae | hi there |
| 01:41.57 | kd6cae | Tom was here think he had to go though darn |
| 01:42.26 | Qwell | huh? |
| 01:42.49 | kd6cae | I was talking to my friend Plattypus1 who just joined |
| 01:43.29 | Qwell | reppin the 909 I see |
| 01:43.35 | *** join/#asterisk iwes (n=iwes@rrcs-24-106-129-123.central.biz.rr.com) |
| 01:44.20 | plattypus1 | 951, yo. :) |
| 01:44.20 | Qwell | pfft |
| 01:44.20 | kd6cae | anyone know why when I call a sip friend via my asterisk box he can hear me but I can't hear him? |
| 01:44.21 | plattypus1 | Oooh! Oooh! Pick me! Pick me! |
| 01:44.21 | plattypus1 | Port forwarding maybe?! |
| 01:44.26 | kd6cae | ok go ahead Plattypus1 what's the answer? lol |
| 01:44.40 | plattypus1 | Sounds like the inbound SIP ports arent |
| 01:44.45 | plattypus1 | getting forwarded to Asterisk. |
| 01:44.59 | plattypus1 | But that's coming from just a general geek-of-all-trades, so it could be wrong. |
| 01:45.04 | kd6cae | should be, my box anyway should be in dmz |
| 01:45.56 | plattypus1 | Oh. I'm out. :) |
| 01:45.56 | kd6cae | We must all be working on the next crazy thing to do with asterisk, it's like way quiet in here, hehe |
| 01:48.20 | Mercestes | kd6cae, What about yoru friend? He DMZ? |
| 01:48.55 | kd6cae | I think so? I mean he says he's able to receive calls from others via SIP so I have to assume at least his ports are forwarded correctly |
| 01:49.49 | Mercestes | kd6cae, what isyour OS? |
| 01:50.00 | kd6cae | fedora core 5 |
| 01:50.03 | Mercestes | Hrm. |
| 01:50.13 | Mercestes | I would've guessed Windows XP for some reason. |
| 01:50.46 | kd6cae | well that's the OS I'm running IRC on, but my asterisk is on fedora, maybe I misunderstood what OS you wanted to know about lol |
| 01:50.47 | plattypus1 | His desktop is, evil Windozer... |
| 01:50.51 | Mercestes | lesse. One way audio is usually caused by NAT or a firewall (or a pissy router) |
| 01:51.05 | plattypus1 | (Got that right!) |
| 01:51.06 | kd6cae | we definetly have a pissy router here hehe |
| 01:51.23 | Mercestes | kd6cae, It's rarely a pissy router but it *can* be that so I wouldnt' rule it out. |
| 01:51.38 | kd6cae | That's why I have to force my linux box in to dmz because of our pissy router |
| 01:51.40 | ManxPower | Qwell: sadly I'm trying Windows Media Encoder |
| 01:51.40 | plattypus1 | What ports does SIP usually run? |
| 01:51.53 | Mercestes | oh, somewhere around 10k to 30k |
| 01:51.54 | kd6cae | I think it's like 5060 udp |
| 01:51.57 | ManxPower | plattypus1: sip signaling or audio? |
| 01:52.05 | Mercestes | you can google sip ports and get it from voip info.odrg |
| 01:52.14 | plattypus1 | Alrighty, thankies. |
| 01:52.21 | Mercestes | voip-info.org should have it too. |
| 01:52.39 | plattypus1 | I figure an existing port forward is conflicting with whatever inbound port it's trying to use. |
| 01:53.10 | kd6cae | yes I've had all kinds of fun trying to forward ports especially UDP ones with our fine router, yet services don't show any conflicts that I can see, so go figure |
| 01:53.29 | kd6cae | I should just solve this by getting the asterisk box on it's own public IP that'll fix that |
| 01:53.48 | ManxPower | kd6cae: or a real router |
| 01:54.09 | kd6cae | that too, never use the netgear WGR614V6 if you can avoid it |
| 01:54.12 | plattypus1 | Routers don't grown on trees, unfortunately. |
| 01:54.27 | Mercestes | heh |
| 01:54.33 | plattypus1 | And Monty, I'd be happy to install a brand new shiny router for you. You buyin'? |
| 01:54.47 | kd6cae | but we've got tons of money Plattypus1, we can get a real router like a dlink Di-624 |
| 01:54.52 | kd6cae | lol |
| 01:54.53 | Mercestes | I wish watermelons grew on trees |
| 01:55.14 | plattypus1 | That's still not a real router. The Cisco... now THAT'S a router. |
| 01:55.26 | Mercestes | Cisco isn't even a real router |
| 01:55.39 | kd6cae | I'll agree there, and while we're at it, let's get a full T1 line installed eh? |
| 01:55.41 | ManxPower | I personally own a Cisco 1720 and a Cisco Catalyst 2205 |
| 01:55.43 | plattypus1 | What kind of a router do you want Mercestes? |
| 01:56.02 | ManxPower | ..er... not 2205, 5505 |
| 01:56.03 | Mercestes | The one after Cisco. hasn't been invented yet. |
| 01:56.25 | plattypus1 | Lol, the NeverCrashes 2000? :) |
| 01:56.36 | kd6cae | Plattypus1, what was that router Jim has connected to his T1 line? That's what I want, complete with T1 line too! lol |
| 01:56.38 | ManxPower | Mercestes: The chronoton based one? It's so fast packets arrive before they are sent |
| 01:56.52 | Mercestes | Cisco is a severe discombobulated PITA with firmware written by a deranged 4-year old with such an insanely non-intutiive CLI that it defines *why* tab completion is useful (required). |
| 01:56.58 | plattypus1 | It was a Cisco somethin'or'other, don't remember the model number. |
| 01:57.04 | Mercestes | Cisco is Cisco because it comes the *closest* to adequacy |
| 01:57.04 | ManxPower | kd6cae: Uh, several people on this channel have personal T-1 lines |
| 01:57.14 | Mercestes | ManxPower, Yes, that one! :D |
| 01:57.31 | kd6cae | cool I am totally blind but my friend's T1 rack was fun to feel |
| 01:58.02 | kd6cae | I'm a networking geek, I love high speed symmetrical networking the way it should be! |
| 01:58.17 | kd6cae | I think some year I'm gonna join the ranks and have my own personal T1 |
| 01:58.41 | plattypus1 | kd6cae, get you the fastest residential 'net available for 30 miles on either side of us... and you're still complainin'. I ought to downgrade you to Charter Lite. |
| 01:58.42 | _VoiceMeUp_COM | yeah got a full rack a@home |
| 01:58.47 | Mercestes | Cisco *is* the best but they are way off the mark in stabilty/customer service. And someone will pwn them one day by making something that works the *first* time, in multiple intuitive configurations instead of quirky nuances of each model/implementation, with a friendly and responsive customer support team who will answer questions for people just for buying their product instead of requiring expensive certifications. |
| 01:58.47 | _VoiceMeUp_COM | neat with mac servers |
| 01:59.04 | _VoiceMeUp_COM | just need a video module to run the whole homw theater |
| 01:59.24 | _VoiceMeUp_COM | also need a san.. but that gonna be $$ |
| 01:59.48 | kd6cae | Plattypus1, you downgrade me to Charter lite and I'll have to shut down the whole asterisk box as punishment lol |
| 02:00.05 | plattypus1 | Darn, I guess that repeater of yours... :) |
| 02:00.23 | kd6cae | Oh crap forgot about that gurrr next idea hmm |
| 02:00.50 | plattypus1 | Have you tried watching the console output during a SIP call? Any interesting info? |
| 02:01.01 | plattypus1 | Do you maybe want to make a SIP call so I can watch it for interesting info? :) |
| 02:01.25 | kd6cae | Haven't tried that yet, though that's my next plan of attack? also need to figure out why when I call my buddy Tom via IAX2, |
| 02:01.42 | kd6cae | it sounds fine but if he calls me, it breaks up after a few seconds |
| 02:01.48 | kd6cae | says something about timing |
| 02:01.52 | Mercestes | kd6cae, Something unfortunate involving System(), his /dev/urandom, and his /dev/hda3 I hope? :D |
| 02:02.03 | ManxPower | kd6cae: 99% of asterisk issues are solved, in part, by watching the Asterisk console |
| 02:02.23 | Mercestes | kd6cae, The other 50% is solved by doing an MTR from your box to your endpoint. Do you have DSL? |
| 02:02.31 | kd6cae | I'll have to get better details later |
| 02:02.36 | plattypus1 | Mercestes, I have root on his asterisk box. Such action would be very unwise. Especially since /dev/hda3 is swap. :D |
| 02:02.47 | kd6cae | I have cable here and sure I can attempt the sip call Plattypus1 if you want to watch |
| 02:02.48 | Mercestes | bwahaha. |
| 02:03.19 | kd6cae | Plattypus1, I heard that, you be nice to that there box and leave swap lol |
| 02:03.22 | plattypus1 | Go for it Monty. |
| 02:03.24 | Mercestes | kd6cae, I've seen trash routers on the Level3 network before. Do an MTR from your * box to your endpiont, I bet you see some lost packets. |
| 02:03.38 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:03.42 | kd6cae | Plattypus1 stand by for sip attempt |
| 02:03.51 | Mercestes | anyways, I'm outies. Goodnight. |
| 02:03.53 | plattypus1 | bbr2.losangeles1.level3.net I can tell you that one. :) |
| 02:03.54 | Mercestes | good luck |
| 02:03.59 | J4k3 | Mercestes: the problem is the "IT" dorks in the corporate world insist on buying who kicks them back the most |
| 02:04.01 | Mercestes | Fight well, sip warriors |
| 02:04.03 | J4k3 | which is losers like cisco, IBM, etc. |
| 02:04.07 | J4k3 | Microsoft. |
| 02:04.09 | Mercestes | plattypus1, bwahaha. pwned. |
| 02:05.19 | ManxPower | One of the things I like about Ciscos is that they don't discontinue their most popular boxes after 6 months. |
| 02:05.41 | J4k3 | but Cisco won't ever discount their most popular anything |
| 02:05.49 | J4k3 | ie - the Cisco 2501 @ $4200 MSRP in 1998 |
| 02:06.05 | ManxPower | Just try buying two exact same servers 6 months apart. |
| 02:06.15 | J4k3 | but Cisco doesn't see their products as what they are, they see them as what the customer thinks they will get in value. |
| 02:06.18 | J4k3 | why would you want to? |
| 02:06.35 | J4k3 | in 6 months, especially PC-based servers, you can get double the horsepower for the same price :P |
| 02:06.48 | ManxPower | J4k3: uh, so you don't have to do custom installs for every single different server you have |
| 02:06.51 | _VoiceMeUp_COM | hmm wahts best san solution for the price |
| 02:06.56 | JT | the more switched on people in IT love and hate cisco at the same time |
| 02:07.03 | JT | only silly fanboys just love them |
| 02:07.22 | J4k3 | JT: Yep. Theres no doubt that when you choose cisco you get the job done. |
| 02:07.34 | ManxPower | One of my clients has 28 servers. No two of them are the same. incompatable memory, CPUs, disk interfaces, chipsets. |
| 02:07.43 | J4k3 | but it might not work *the best*, and it'll definetly cost the most. |
| 02:08.05 | JT | it really shits me when fanboys tell me how stable they are |
| 02:08.12 | ManxPower | J4k3: We let the suckers pay MSRP and buy 1 or 2 generations behind current on the used market. |
| 02:08.12 | JT | cisco aren't even close to carrier grade |
| 02:08.13 | J4k3 | ManxPower: this sounds like a good place for some virtualization |
| 02:08.13 | flenders | I' |
| 02:08.15 | JT | they are high end IT |
| 02:08.17 | flenders | I'm a fanboy |
| 02:08.22 | JT | which is not carrier grade |
| 02:08.24 | flenders | I love cisco |
| 02:08.25 | ManxPower | heck you can buy two of each and still save money |
| 02:08.26 | flenders | :D |
| 02:08.28 | JT | heh |
| 02:08.42 | ManxPower | J4k3: Yeah, like the large batch of servers our linux auto install did not work on. |
| 02:09.02 | JT | IT is a collection of hacks that hopefully doesn't disintegrate too often |
| 02:09.09 | JT | carrier grade MUST work |
| 02:09.23 | J4k3 | ManxPower: I recently installed a server that runs WinXP Pro + VMWare + FreeBSD better than it runs FreeBSD native... Damned depressing :P |
| 02:09.49 | J4k3 | well, ok 'faster'... 'better' and running under WinXP is a damned awful thing to say. |
| 02:11.10 | *** join/#asterisk arcanine (n=arcanine@203.82.44.179) |
| 02:11.23 | arcanine | does anyone know vici dial? |
| 02:11.45 | ManxPower | J4k3: how long did it take to do the basic install/ |
| 02:12.56 | J4k3 | the XP+vmware install? eh... 1.5 hours or so with a whole pile of windowsupdates |
| 02:12.59 | J4k3 | on a 1.5mbit line. |
| 02:13.16 | ManxPower | Until none of the new servers worked with our auto install we basically booted from an autoinstall floppy and CD-ROM, install is done in 15 mins |
| 02:13.53 | ManxPower | then they call me, tell me what the IP address of the new server is, what it will be used for and I install whatever packages are required. |
| 02:14.05 | J4k3 | vmware emulates very 'fixed' devices... like an Intel BX chipset. |
| 02:14.07 | ManxPower | the base install is exactly the same between server. |
| 02:14.14 | J4k3 | its an odd thing to see "Opteron 1210" on a BX chipset :) |
| 02:14.29 | ManxPower | I don't really see the need for VMware in our enviropment |
| 02:14.33 | Zorix | hey guys, any idea why when i power on my asterisknow machine and go to the gui it only shows the users and the rest of the menus just sit there.. i even updated from beta 4 to beta 5 and no change |
| 02:15.04 | J4k3 | yeah, I don't like the overhead of vmware |
| 02:15.04 | shmaltz | anybody using the 5xi phones from aastra? |
| 02:15.27 | ManxPower | Management had one of two options: 1) spend lots of money on bandwidth between all the offices and same money on servers or 2) spend lots of money on servers and very little on bandwidth between offices. |
| 02:15.34 | ManxPower | Management picked option 2 |
| 02:15.52 | ManxPower | Most of the WAN is 384K links |
| 02:18.04 | _VoiceMeUp_COM | yeah i use vmware daily |
| 02:18.10 | _VoiceMeUp_COM | and it doesn tend to mem leak and all |
| 02:18.16 | Iamnacho | vmware r0x! |
| 02:18.17 | _VoiceMeUp_COM | windows in unix.. |
| 02:18.27 | _VoiceMeUp_COM | seems after a while i loose gui |
| 02:18.39 | ManxPower | I don't manage any Windows boxes. |
| 02:18.44 | _VoiceMeUp_COM | also around half the time after 3-4 horus it starts hoging host cpu |
| 02:18.52 | J4k3 | Windows would not be the optimal OS to run on either side of the VM. |
| 02:19.14 | J4k3 | in my case it 'worked better' because the drivers for my hardware sucked for both linux and bsd, but good drivers in windows. Bad hardware choices. |
| 02:19.22 | J4k3 | my fault. |
| 02:20.59 | ManxPower | My policy is "If you did not use my standardized install disks to create the server then I am not supporting that server" |
| 02:21.18 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 02:21.18 | *** mode/#asterisk [+o mog] by ChanServ |
| 02:23.34 | J4k3 | ManxPower: well, with that scenario, unless they're willing to replace whatever hardware your intaller doesn't like, vmware is the answer IMHO. |
| 02:23.50 | J4k3 | unless, of course, vmware causes too much overhead for the task. |
| 02:23.59 | J4k3 | (or your installer doesn't work properly on old P3 boxes) |
| 02:27.31 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 02:29.11 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 02:40.48 | *** join/#asterisk lwh (n=lwh192@rdsl-0593.tor.pathcom.com) |
| 02:46.59 | JT | oh how we love autoaway! |
| 02:47.17 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 02:47.28 | Nivex | http://sackheads.org/~bnaylor/spew/away_msgs.html |
| 02:50.35 | *** join/#asterisk Avochelm (n=damo@gw-morphett.koalatelecom.com.au) |
| 02:50.58 | JT | Nivex: so true |
| 02:57.03 | khronos | Anyone know what I can do to make an Asterisk server connect to a sip endpoint? |
| 02:57.24 | khronos | I have an asterisk server behind a router and the asterisk server is on the dmz of this router. |
| 02:57.45 | *** join/#asterisk rmayorga (n=rmayorga@168.243.73.11) |
| 02:57.51 | khronos | When I connect to the sip phone on another network the call connects but there isn't any audio. |
| 02:58.28 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 02:58.33 | khronos | I tried setting the externip variable to the public interface of the connection but this really didn't seem to work. |
| 02:58.37 | JT | clearly the RTP traffic is not making it through |
| 02:59.34 | drfreeze | What to do when Polycom 501 is stuck in "Updating initial configuration..." mode..? |
| 02:59.50 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 03:00.08 | JT | drfreeze: i'd thought the conclusion was the phone had been blown up |
| 03:00.31 | drfreeze | JT this is another phone |
| 03:00.44 | JT | ah |
| 03:00.45 | drfreeze | and the jury is still out on 'the other' phone |
| 03:01.00 | JT | why did it get plugged into the wrong socket? |
| 03:01.35 | drfreeze | the analog CC line was out and they were trying to test for a dial tone with the polycom phone |
| 03:01.50 | JT | cc? |
| 03:01.56 | JT | that wasn't awfully bright of them |
| 03:01.56 | drfreeze | credit card |
| 03:02.09 | drfreeze | most people don't know if a phone is digital or analog |
| 03:03.16 | JT | the line musn't have been completely out |
| 03:03.18 | drfreeze | need to implement a policy though, that they don't do anything electrically until that ok it with me first |
| 03:03.37 | drfreeze | the analog line was out - jumper was off |
| 03:03.48 | JT | then how did it cause damage? |
| 03:04.19 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:04.22 | drfreeze | they might have plugged it into the fax line, which was not out |
| 03:04.32 | JT | hmm |
| 03:05.19 | drfreeze | JT: Know why I would get "Waiting for network to initialize" after the phone has got it's IP address? |
| 03:05.40 | JT | no idea |
| 03:06.15 | drfreeze | JT: know the diff between TrivialFTP and FTP in the phone config? |
| 03:06.50 | JT | totally different protocols |
| 03:06.53 | JT | tftp is shit |
| 03:07.12 | *** join/#asterisk darkgamer20 (i=darkgame@adsl-71-146-157-38.dsl.pltn13.sbcglobal.net) |
| 03:07.45 | darkgamer20 | is there any way to get the sunrocket voip service to work with asterisk? |
| 03:07.49 | drfreeze | JT: I am using vsftp |
| 03:07.56 | drfreeze | should that work with either protocol? |
| 03:07.59 | JT | yes that's ftp |
| 03:08.00 | JT | no |
| 03:08.01 | JT | ftp. |
| 03:08.12 | drfreeze | ok, thx |
| 03:10.53 | *** join/#asterisk darkgamer20 (i=darkgame@adsl-71-146-157-38.dsl.pltn13.sbcglobal.net) |
| 03:11.15 | shmaltz | funny: |
| 03:11.17 | shmaltz | http://www.liveleak.com/view?i=cb5_1177378417&p=1 |
| 03:12.41 | JerJer[mobile] | darkgamer20: i highly doubt it |
| 03:13.19 | *** join/#asterisk rmayorga (n=rmayorga@168.243.73.11) |
| 03:13.38 | JerJer[mobile] | i get dozens (if not more) of hits per day on "asterisk sunrocket" on my blog, so you aren't the only on lookin |
| 03:14.11 | JT | what is sunrocket? |
| 03:14.26 | JerJer[mobile] | so naturally i inquired with Sunrocket and whoever they have answering the phone had no clue what Asterisk is, then nobody has called me back |
| 03:14.33 | darkgamer20 | JerJer[mobile]: why so? |
| 03:14.33 | JerJer[mobile] | JT: a voip provider |
| 03:14.48 | JT | what's so good about them? |
| 03:14.54 | darkgamer20 | JerJer[mobile]: i see |
| 03:14.55 | JerJer[mobile] | no clue |
| 03:14.57 | JT | do they use a proprietary protocol? |
| 03:15.01 | darkgamer20 | JT: they have good plans |
| 03:15.18 | JerJer[mobile] | they lock you into their devices |
| 03:15.30 | JT | i see |
| 03:15.43 | JerJer[mobile] | darkgamer20: they are the next largest loss leader in the VoIP Game |
| 03:15.43 | JT | sounds non-optimal |
| 03:15.57 | darkgamer20 | JerJer[mobile]: after vonage? |
| 03:16.08 | JerJer[mobile] | yep |
| 03:16.55 | darkgamer20 | JerJer[mobile]: well the reason i choose sunrocket is because they have good rates for calling india, only 8 cents a minute, which is less than most voip providers |
| 03:17.30 | LeddyHM | Can anyone explain why my "GoTo"'s aren't working? http://www.pastebin.ca/455285 |
| 03:18.21 | drfreeze | Anyone know what the problem is for Polycom when you get the message: "waiting for network to initialize" |
| 03:18.29 | JT | darkgamer20: probably won't last for long |
| 03:18.42 | blitzrage | LeddyHM: first thing you should use is not a Goto(*41,2), but a Goto(*42,some_label_you_make_up) |
| 03:18.43 | darkgamer20 | JT: probably, but until then |
| 03:18.56 | blitzrage | drfreeze: your phone can't get an IP |
| 03:19.07 | JT | pretty sure it would be an illegal connection into india |
| 03:19.10 | JerJer[mobile] | LeddyHM: and i see no 's' exten |
| 03:19.28 | LeddyHM | hmm |
| 03:19.35 | LeddyHM | where do I define that? |
| 03:19.45 | drfreeze | blitzrage: it's got an IP |
| 03:19.57 | blitzrage | drfreeze: if its sitting at that screen, no it doesn't |
| 03:20.12 | blitzrage | or the cable is unplugged, or you have some other network issue |
| 03:20.23 | darkgamer20 | well thanks everyone for your help |
| 03:20.31 | drfreeze | blitzrage: then when I press the about button, why does it give me an IP address? |
| 03:21.04 | drfreeze | blitzrage: I watch /var/log/messages give it an address, |
| 03:21.07 | blitzrage | you have some other network problem |
| 03:24.06 | drfreeze | should cfg files for polycom phones be writable by the phone? |
| 03:28.01 | JerJer[mobile] | smells like a firewall blocking a response |
| 03:34.51 | LeddyHM | damn it was the goto(s) |
| 03:35.00 | LeddyHM | forgot to change it to *41 |
| 03:36.21 | *** join/#asterisk PakiPenguin (i=uppal@linuxpakistan/admin/pakipenguin) |
| 03:36.53 | PakiPenguin | hi there, I am having an issue where every SIP call of mine disconnects after 20 secs , what could be wrong , i am using 1.2.17 |
| 03:38.35 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
| 03:45.43 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 03:49.15 | illsci | anyone still awake? |
| 03:49.55 | illsci | chamber*CLI> iax2 show peers |
| 03:49.55 | illsci | Name/Username Host Mask Port Status |
| 03:49.55 | illsci | iaxuser 68.227.204.129 (D) 255.255.255.255 4569 UNREACHABLE |
| 03:49.56 | aptura | yea everyone here is. |
| 03:49.57 | aptura | :) |
| 03:50.00 | illsci | :) |
| 03:50.10 | illsci | I don't understand why that is unreachable... |
| 03:50.27 | aptura | illsci open up that port on the firewall |
| 03:50.27 | illsci | on the console it keeps saying its registered |
| 03:50.27 | illsci | and and then its not.. |
| 03:50.52 | illsci | i think it is opened.... do you mean on the one here on the network im using kiax or on the box running asterisk in the colo... |
| 03:51.35 | illsci | I have it opened up on the box running asterisk... |
| 03:52.07 | illsci | I thought iax2 worked with nat... |
| 03:52.30 | *** join/#asterisk bmg505 (n=leon@196.209.183.243) |
| 03:55.44 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 03:55.53 | illsci | so I can't "peer" with this asterisk box unless the asterisk box can initiate sessions with the other side |
| 03:56.26 | aptura | illsci everyone has mostly gone home or retired for the evening. |
| 03:56.36 | aptura | Is this for a business |
| 03:56.43 | illsci | no |
| 03:57.17 | illsci | well, not an important one.. |
| 03:57.29 | PakiPenguin | umm can anyone help me with the 20sec call issue? |
| 03:59.57 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:00.40 | *** join/#asterisk Mattwj2005 (n=Matt@user-38q4155.cable.mindspring.com) |
| 04:00.54 | Mattwj2005 | Good evening everyone :) |
| 04:02.04 | *** part/#asterisk kopeah (n=kopeah@cpe-70-115-242-122.satx.res.rr.com) |
| 04:04.22 | JT | aptura: some people are wroking right now |
| 04:05.37 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 04:06.12 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.206.181) |
| 04:08.17 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.149) |
| 04:10.06 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 04:11.15 | CunningPike | Vancouver Canucks ftw |
| 04:12.33 | russellb | what's hockey? |
| 04:12.35 | russellb | :-p |
| 04:12.35 | *** join/#asterisk HockeyInJune (i=HockeyIn@pool-68-161-140-134.ny325.east.verizon.net) |
| 04:16.28 | blitzrage | CunningPike: it was so sweet |
| 04:17.26 | Scrye | im in Vancouver |
| 04:18.57 | Juggie | CunningPike, now if they could only find selanne, neidermyer (x2) and pronger, and possibly giguere, and hold them hostage in a warehouse for the next 2 weeks, they MIGHT have a chance in round two :) |
| 04:20.56 | CunningPike | Juggie: lol |
| 04:21.33 | Juggie | seriously though, they are going out in 5-6 games |
| 04:21.45 | Juggie | anaheim is just much much better |
| 04:21.58 | Juggie | i am hopeing for a anaheim/sj western conference final. |
| 04:22.07 | blitzrage | makes sense |
| 04:22.18 | blitzrage | anaheim is my pick for the finals |
| 04:22.21 | CunningPike | Well, I'll respectfully hope for something else |
| 04:22.23 | CunningPike | :) |
| 04:22.29 | Juggie | that series would be good for hockey |
| 04:22.44 | Juggie | gets some usa fans into the game, sets up a california rivlirary etc. |
| 04:22.44 | blitzrage | but no one would watch it |
| 04:23.01 | Juggie | plus the cities are close enough to do it all in HD with one truck. |
| 04:23.23 | Mattwj2005 | russellx |
| 04:23.44 | Mattwj2005 | hockey is too guys fighting over a rubber disc with sticks |
| 04:23.48 | Mattwj2005 | *two |
| 04:25.33 | CunningPike | Mattwj2005: Actually, 10 guys |
| 04:25.49 | Mattwj2005 | indeed |
| 04:26.03 | flenders | is it 10 on each side? |
| 04:26.20 | CunningPike | flenders: 5 skaters plus the goalie per side |
| 04:26.26 | Mattwj2005 | basically guys with sticks and a rubber disc |
| 04:26.42 | CunningPike | Mattwj2005: Basically |
| 04:26.47 | JT | a steel disc would be more fun |
| 04:26.52 | CunningPike | Pretty good entertainment, all the same |
| 04:26.55 | flenders | ice hockey is a funny sport |
| 04:27.25 | CunningPike | I want to see basketball on ice - now _that_ would be funny |
| 04:28.02 | flenders | :D |
| 04:28.21 | JT | lawn bowls |
| 04:28.30 | CunningPike | No it doesn't |
| 04:28.41 | JT | ? |
| 04:28.43 | CunningPike | Lawn darets |
| 04:28.46 | CunningPike | darts |
| 04:28.48 | Mattwj2005 | actually the most fun hockey would be in outspace |
| 04:29.04 | Mattwj2005 | no gravity style |
| 04:29.08 | JT | heh |
| 04:29.24 | JT | or ice up a velodrome, goals at the top of each side |
| 04:29.45 | ManxPower | sports? What is this "sports" thing you speak of? |
| 04:31.14 | drfreeze | Any polycom experts out there? |
| 04:31.39 | drfreeze | Wondering if "Waiting for network to initialize" means the phone is corrupted? |
| 04:31.57 | CunningPike | drfreeze: No - it's waiting for a DHCP lease |
| 04:32.53 | drfreeze | CunningPike: it's getting the lease |
| 04:33.11 | drfreeze | I have 1 phone that works, and a couple that just keep cycling |
| 04:33.23 | drfreeze | setups are the same, best I can tell |
| 04:33.46 | drfreeze | have plugged the phones into the others network, and get the same respone from the phone |
| 04:34.22 | drfreeze | CunningPike: after the waiting msg, it says "Updating initial configuration..." |
| 04:34.32 | CunningPike | OK, and then? |
| 04:35.05 | drfreeze | just a minute, and I'll let you know. Is in the update phase now |
| 04:35.51 | CunningPike | OK |
| 04:37.08 | drfreeze | CunningPike: I think the phones got the power pulled during an update |
| 04:37.11 | dlynes_laptop | good evening, cp |
| 04:37.29 | CunningPike | dlynes_laptop! |
| 04:37.37 | dlynes_laptop | CunningPike: btw |
| 04:37.44 | dlynes_laptop | CunningPike: I'm getting married this summer :0 |
| 04:37.52 | CunningPike | Wow! |
| 04:37.59 | CunningPike | Congratulations! |
| 04:38.04 | aptura | CunningPike people honking outside your place? |
| 04:38.15 | drfreeze | dlynes_laptop: congrats! :) |
| 04:39.34 | CunningPike | Not right outside - we live in a respectible neighborhood, you know |
| 04:39.35 | aptura | hope she is the right one for you. |
| 04:39.35 | CunningPike | Hope you're the right one for her - it goes easier that way :) |
| 04:39.35 | CunningPike | drfreeze: That's sometimes not good....... |
| 04:39.41 | CunningPike | drfreeze: Can you get to the setup menu on the phone? |
| 04:39.41 | aptura | my wife never told me she was in debt untill after then engagment. |
| 04:40.04 | dlynes_laptop | aptura: well, she already knows about my issues before marriage |
| 04:40.05 | drfreeze | CunningPike: yes |
| 04:40.12 | drfreeze | CunningPike: still in update mode |
| 04:40.20 | dlynes_laptop | CunningPike: Yeah...we're both pretty compatible, I think |
| 04:40.29 | dlynes_laptop | CunningPike: I'm asianized, she's westernized :) |
| 04:40.38 | CunningPike | drfreeze: Hmmm - double check your settings - maybe try a factory reset |
| 04:40.44 | dlynes_laptop | CunningPike: we were both born and raised on farms, ... |
| 04:40.45 | CunningPike | drfreeze: What model? |
| 04:41.01 | CunningPike | dlynes_laptop: That's great - I'm very happy for you |
| 04:41.07 | CunningPike | Set a date yet? |
| 04:41.10 | dlynes_laptop | CunningPike: Yeah...a super sweet cantonese girl :) |
| 04:41.15 | aptura | what is the typical range of wifi line of sight |
| 04:41.18 | drfreeze | CunningPike: and finally, it says - couldn't contact boot server .... loading app |
| 04:41.32 | dlynes_laptop | CunningPike: Well, we were planning on today or tomorrow, but the pastor wouldn't marry us without going through marriage counselling first |
| 04:41.37 | drfreeze | running App=sip.ld |
| 04:41.54 | dlynes_laptop | CunningPike: so, off to marriage counselling first I guess :) |
| 04:41.57 | aptura | dlynes_laptop that helps if you want to live in the country |
| 04:42.04 | dlynes_laptop | aptura: huh? |
| 04:42.18 | dlynes_laptop | aptura: who? what country? |
| 04:42.18 | aptura | both lived on fams |
| 04:42.23 | CunningPike | dlynes_laptop: Hmm - well, their rules, I guess |
| 04:42.25 | aptura | farms |
| 04:42.28 | aptura | both of you |
| 04:42.30 | dlynes_laptop | aptura: oh...no...compatible way of thinking |
| 04:42.36 | aptura | my wife would hate that. |
| 04:42.39 | dlynes_laptop | aptura: not necessarily about where we would live |
| 04:43.02 | aptura | my wife said if we live on a farm I will divorse you |
| 04:43.03 | aptura | :) |
| 04:43.06 | dlynes_laptop | aptura: hahahaha |
| 04:43.27 | dlynes_laptop | aptura: anyways...maybe 15 to 20 years down the road we might live on a farm |
| 04:43.35 | drfreeze | aptura: ick. Had enuf farm life. Too many stinky animals and flys |
| 04:43.45 | dlynes_laptop | aptura: but not any time soon...I work on computers all day...kinda hard to find work in the bush for that |
| 04:44.16 | aptura | drfreeze I would live on a farm only to grow a large garden. |
| 04:44.30 | aptura | dlynes_laptop I know |
| 04:44.48 | drfreeze | aptura: take out the grass at your house and put in a garden. :) |
| 04:44.53 | aptura | Better yet, hydroponic garden. Alot less maintence. |
| 04:44.53 | JT | aptura: i read that as "I would live on a farm only to grow a large chicken" |
| 04:45.40 | aptura | I lived in the country growing up it was a bit lonely. |
| 04:46.31 | aptura | dlynes_laptop what type of work |
| 04:46.35 | CunningPike | drfreeze: Does it complete the boot up? |
| 04:47.19 | aptura | CunningPike whats your take on shaw taking over telus and small voip companies providing phone service? |
| 04:47.22 | *** join/#asterisk shadou (n=aj@unaffiliated/dj-fu) |
| 04:47.49 | drfreeze | CunningPike: after Running App, it's back to the "Waiting for network to initialze.." |
| 04:48.09 | CunningPike | aptura: Telus have wanted out of the last mile business for years - they want to be a carrier and wholesale their capacity |
| 04:48.25 | CunningPike | drfreeze: OK - what model phone? |
| 04:48.26 | drfreeze | with 3 buttons, start, setup and about |
| 04:48.32 | drfreeze | Polycom 501 |
| 04:48.57 | CunningPike | drfreeze: OK - during the countdown to reboot, press and hold the 4, 6, 8 and * keys |
| 04:49.08 | CunningPike | drfreeze: That will reset the phone to it's factory defaults |
| 04:49.16 | CunningPike | s/it's/its/ |
| 04:49.33 | CunningPike | Can't believe I did that |
| 04:50.10 | dlynes_laptop | aptura: system administration, network administration, voip administration, programming, cabling, ... |
| 04:51.16 | drfreeze | CunningPike: ok, it said something about reseting and now is "Uploading log file..." |
| 04:51.25 | CunningPike | drfreeze: Excellent |
| 04:51.47 | CunningPike | What is your provisioning setup? Are you using Option 66 to point to an FTP server? |
| 04:52.07 | drfreeze | CunningPike: Yes. What is that BTW? |
| 04:52.50 | illsci | hey if you have an mp3 how can you convert it to some format you can play with Playback() |
| 04:53.14 | illsci | or is there some codec I can allow that will just work with mp3's |
| 04:53.52 | CunningPike | drfreeze: Option 66 is a piece of information that gets sent in the DHCP settings that tells the phone where to go to get its configuration |
| 04:54.23 | drfreeze | CunningPike: I'm providing the IP of the host server, not the name. Do I still use 66? |
| 04:54.35 | aptura | CunningPike updated to latest kernel and zap no more echo or tx issues. |
| 04:54.51 | CunningPike | illsci: Better off to use sox to convert to the same format that you use for your calls |
| 04:55.19 | CunningPike | drfreeze: Yes - IP or hostname - IP is better as it eliminates a dependancy on DNS |
| 04:55.28 | CunningPike | aptura: Excelent |
| 04:55.48 | aptura | but its to late to sway the wife. she went shaw digital. |
| 04:55.50 | CunningPike | aptura: We are in the throes of upgrading to 1.4.2 - we had a few glitches unrelated to Asterisk |
| 04:55.59 | CunningPike | aptura: Bummer |
| 04:56.05 | aptura | shaw digitals terminal or ata has a built in ups. |
| 04:56.08 | drfreeze | CunningPike: ok, now it's back to "Waiting for network to init..." |
| 04:56.32 | CunningPike | OK - and you're sure it's getting a valid lease from the DHCP server - it sure doesn't sound like it |
| 04:56.46 | illsci | is there a comparrison of the different formats.. |
| 04:56.55 | CunningPike | illsci: What format are your calls |
| 04:57.04 | illsci | i allow a bunch of formats |
| 04:57.08 | illsci | ulaw is the first i think |
| 04:57.31 | CunningPike | aptura: Which is as much use as a chocolate teapot unless your Shaw modem, and the repeater on the pole outside your house are on UPS also |
| 04:57.46 | CunningPike | illsci: Use ulaw then |
| 04:57.52 | illsci | is that the best format... |
| 04:58.04 | illsci | least cpu intesive... best quality sound? |
| 04:58.06 | CunningPike | illsci: Define best :) |
| 04:58.13 | illsci | i dont know what other characteristics to care aobut |
| 04:58.14 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-40-188.socal.res.rr.com) |
| 04:58.20 | *** join/#asterisk snoopster (n=santa@cpe-76-187-204-88.tx.res.rr.com) |
| 04:58.30 | CunningPike | Oh, you just did - the best is arguably G.729, but you have to pay for that |
| 04:58.39 | CunningPike | illsci: We use ulaw |
| 04:59.07 | CunningPike | illsci: It's probably the best solution for most purposes |
| 04:59.11 | illsci | in the asterix book it says something about G.729 and it crippling a box if you try to converence 10 calls with that format |
| 04:59.12 | J4k3 | I got bandwidth to burn. |
| 04:59.18 | CunningPike | J4k3: Bummer |
| 04:59.19 | CunningPike | :) |
| 04:59.33 | J4k3 | illsci: if you transcode 10+ calls, sure. |
| 04:59.39 | J4k3 | transcoding any codec 'hurts' performance |
| 04:59.49 | illsci | what would be the best way to do that |
| 04:59.50 | J4k3 | if your extensions and your 'trunks' are the same codec, theres no transcoding issue. |
| 04:59.58 | illsci | then if you were going to have like 20 people on a conference call |
| 05:00.01 | J4k3 | use whatever codec both ends support |
| 05:00.02 | J4k3 | hrm |
| 05:00.10 | illsci | oh hmmm |
| 05:00.11 | J4k3 | ulaw would give you the best performance |
| 05:00.22 | J4k3 | but figure 80kbit per call, in each direction, simultaniously. |
| 05:00.26 | snoopster | Is it possible to compile asterisk 1.4.2 and have it not know about sip? |
| 05:00.38 | illsci | just dont load the sip module |
| 05:00.44 | illsci | in modules.conf... |
| 05:04.05 | drfreeze | CunningPike: it rebooted, but is back to waiting for network |
| 05:04.22 | CunningPike | OK - and you're sure it's getting a valid lease from the DHCP server - it sure doesn't sound like it |
| 05:04.26 | drfreeze | I can hold down the about button and see the IP address that it gets and ping the phone |
| 05:04.33 | CunningPike | drfreeze: Hmm |
| 05:04.43 | drfreeze | CunningPike: is a valid lease not the same thing as an address? |
| 05:05.19 | drfreeze | the /var/lib/dhcp/dhcpd.leases shows 3 occurances for the lease |
| 05:05.33 | CunningPike | drfreeze: Does xferlog on the FTP server show anything? |
| 05:05.59 | CunningPike | drfreeze: OK - if you can ping the phone, it's got an IP address |
| 05:06.21 | CunningPike | drfreeze: Next, take a look at the FTP server's log and see if the phone is making contact |
| 05:07.28 | illsci | wow |
| 05:07.35 | illsci | that sounded demonic |
| 05:07.43 | illsci | i converted a mp3 to gsm with sox |
| 05:07.56 | illsci | that did not sound anywhere near the what the mp3 sounds like |
| 05:08.19 | drfreeze | CunningPike: getting nothing from the vsftpd log |
| 05:08.25 | drfreeze | last access was Apr 3 |
| 05:08.36 | drfreeze | However, I can ftp from teh command line |
| 05:08.43 | CunningPike | drfreeze: OK - that tells me that the phones are connecting to it |
| 05:08.52 | CunningPike | s/are/aren't/ |
| 05:09.36 | drfreeze | option tftp-server-name "ftp://polycom:password@192.168.50.1"; |
| 05:09.52 | FuriousGeorge | any digium guys wanna comment on which is the lesser of two evils: open asterisk on (insert your) distro, or asterisknow |
| 05:09.54 | drfreeze | CunningPike: that's the line in /etc/dhcpd.conf |
| 05:10.19 | CunningPike | drfreeze: Try it with just the IP address |
| 05:10.34 | drfreeze | CunningPike: how does it get the username and password? |
| 05:10.39 | drfreeze | from the phone? |
| 05:10.46 | *** join/#asterisk kiwoneka (n=kiwoneka@KTNRON06-1168103823.sdsl.bell.ca) |
| 05:10.52 | kiwoneka | good morning to all |
| 05:11.21 | kiwoneka | what does gtalk.conf do? |
| 05:12.04 | CunningPike | drfreeze: Yes - the default is PlcmSpIp/PlcmSpIp |
| 05:12.27 | CunningPike | drfreeze: So, set up a user on your FTP server called PlcmSpIp, and set its password to the same |
| 05:12.45 | CunningPike | drfreeze: Make its home directory the folder where your config files live |
| 05:13.15 | CunningPike | kiwoneka: Configures gtalk |
| 05:13.17 | CunningPike | :) |
| 05:13.24 | kiwoneka | nice |
| 05:13.36 | *** join/#asterisk p0g0 (n=pogo@madwifi/support/p0g0) |
| 05:14.00 | kiwoneka | that means google talk on my polycom601 |
| 05:14.04 | kiwoneka | hmm |
| 05:15.55 | kiwoneka | CunningPike: you have an tutorials on that? |
| 05:16.16 | kiwoneka | also, paging howto with the polycom601 |
| 05:16.22 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.149) |
| 05:16.43 | CunningPike | So _that's_ what 'Destroy' does |
| 05:16.55 | mcab | the polycom should use the username and password in the URLs it gets |
| 05:17.04 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-136017.home.otenet.gr) |
| 05:17.25 | mcab | drfreeze: is this still the phone that got plugged into the analog line? |
| 05:17.45 | drfreeze | mcab: nope, it's still sitting next to me. |
| 05:17.55 | drfreeze | But, this may fix it. |
| 05:18.17 | drfreeze | I could have just been unplugged while updating and I need to reset the firmware |
| 05:19.04 | mcab | bootrom update? |
| 05:19.25 | kiwoneka | i need some directions on how to setup paging |
| 05:19.40 | mcab | polycoms are pretty tolerent of being unplugged during updates |
| 05:20.01 | mcab | sometimes the filesystem can get a little messed, but generally a reformat will fix that |
| 05:20.36 | CunningPike | kiwoneka: What sort of paging? |
| 05:20.43 | drfreeze | mcab: I've got 3 phones updating right now. Just taking awhile |
| 05:21.17 | Mattwj2005 | hi room :) |
| 05:21.27 | Mattwj2005 | how is everything going in Asterisk land? |
| 05:21.32 | kiwoneka | well, i have 6 polycom 601s and i need to setup paging,extensions to extension(intercom) and anouncement |
| 05:22.02 | Mattwj2005 | how is that going kiwoneka? |
| 05:22.04 | mcab | drfreeze: slow connection? |
| 05:22.04 | kiwoneka | i have set my phone to auto answer |
| 05:22.13 | [TK]D-Fender | kiwoneka, go look up "polycom auto-answer" on the wiki |
| 05:22.27 | [TK]D-Fender | kiwoneka, plenty of instructions there |
| 05:22.57 | kiwoneka | i might have missed it all, may you paste me a link please |
| 05:23.27 | kiwoneka | its the dial plan that is beating me up |
| 05:23.59 | Mattwj2005 | what are you trying to do that isn't work? |
| 05:24.11 | Mattwj2005 | *working |
| 05:24.45 | kiwoneka | i believe i just went about entirly the wrong way |
| 05:24.46 | Mattwj2005 | I am not an expert...but if I can help I would like to :) |
| 05:24.55 | Mattwj2005 | ok? |
| 05:25.18 | kiwoneka | now iam hoping to gain some insight |
| 05:25.41 | kiwoneka | which wiki? |
| 05:25.53 | Mattwj2005 | well a polycom is generally used for business conferences |
| 05:26.28 | CunningPike | ~wiki |
| 05:26.29 | Mattwj2005 | push some buttons and it should dial |
| 05:26.30 | *** join/#asterisk asteriskguy (n=learnast@cpe-75-80-111-113.socal.res.rr.com) |
| 05:26.30 | Mattwj2005 | :) |
| 05:26.39 | Mattwj2005 | www.voip-wiki.org ? |
| 05:26.50 | asteriskguy | voip-info.org |
| 05:27.09 | Mattwj2005 | oops |
| 05:27.25 | Mattwj2005 | thanks asteriskguy |
| 05:27.32 | asteriskguy | np |
| 05:28.08 | asteriskguy | anyone know how to use callgroup? |
| 05:28.35 | asteriskguy | I have two frontdesk phones and need to have them able to pickup each other's phone call |
| 05:29.13 | Mattwj2005 | I have an idea! |
| 05:29.32 | asteriskguy | I'm open for ideas |
| 05:29.39 | Mattwj2005 | exten => 5000,1,Dial(SIP/5001&SIP/5002) |
| 05:29.53 | Mattwj2005 | exten => 5000,2,Hangup |
| 05:30.11 | asteriskguy | that only get the calls to ring on both phones |
| 05:30.12 | Mattwj2005 | that style of code is what I use |
| 05:30.22 | Mattwj2005 | this is true |
| 05:30.24 | kiwoneka | me three |
| 05:30.28 | kiwoneka | call group |
| 05:30.40 | J4k3 | call gruppe. |
| 05:30.56 | asteriskguy | But I need Person A to pickup a call, put that call on hold, and person B should be able to pickup that call |
| 05:31.25 | asteriskguy | I put them both in the same callgroup and pickup ground using callgroup=1 and pickupgroup=1 in sip.conf file |
| 05:31.32 | [TK]D-Fender | asteriskguy, Not doable with * yet. |
| 05:31.38 | asteriskguy | but it doesn't work. I think I'm missing something |
| 05:31.40 | asteriskguy | oh |
| 05:31.54 | asteriskguy | hey [TK]D-Fender |
| 05:32.09 | [TK]D-Fender | asteriskguy, Anything you think you've seen to the contrary is not related to your request. |
| 05:32.13 | asteriskguy | how you've been? |
| 05:32.19 | [TK]D-Fender | still breathing :) |
| 05:32.22 | [TK]D-Fender | late nights |
| 05:32.42 | asteriskguy | not quite understand what you said ^ |
| 05:33.31 | asteriskguy | how about faxing TK? |
| 05:33.44 | Mattwj2005 | asteriskguy |
| 05:33.57 | asteriskguy | yeah Mattwj2005? |
| 05:34.05 | Mattwj2005 | in that idea I wrote about |
| 05:34.15 | [TK]D-Fender | asteriskguy, Being able to pick up somebody elses calls = Not doable. Everything that SOUNDS like its related to this taks, ISN'T. |
| 05:34.25 | Mattwj2005 | can it be changed or some how add the ability to do transfers? |
| 05:35.01 | asteriskguy | TK, I saw on the SIP Admin guide for the Polycom something about shared line vs private line? |
| 05:35.05 | Mattwj2005 | I don't know the syntax but I know you can transfer between numbers |
| 05:35.18 | [TK]D-Fender | asteriskguy, As for faxing, the 2 key options are 1. SpanDSP + RxFax/TxFax. 2. IAXMODEM (uses SpanDSP seperately) + Hylafax (a fax server app) |
| 05:35.38 | [TK]D-Fender | asteriskguy, Polycom supports this funcitonality, ASTERISK does not. |
| 05:35.51 | asteriskguy | Yeah, I tried SpanDSP, compiled it but unable to get the patch working |
| 05:36.05 | Mattwj2005 | anyone ever get modem over voip to work? |
| 05:36.10 | asteriskguy | oh... |
| 05:36.21 | drfreeze | Mattwj2005: ie, fax? |
| 05:36.40 | [TK]D-Fender | Mattwj2005, You might be able to get it to work over ULAW in the BEST of cases, but only at LOW speeds. |
| 05:36.48 | Mattwj2005 | okay |
| 05:36.56 | Mattwj2005 | that would be great for networking guys like me |
| 05:37.01 | [TK]D-Fender | maybe a fax might work, but I SERIOUSLY wouln't bet on it. Expect a bad failure rate |
| 05:37.14 | asteriskguy | hehe I tried ULAW with the AIXy device from Digium. Works fine for fax but bad for Data connex |
| 05:37.22 | Mattwj2005 | dialup networking devices on the cheap :) |
| 05:38.16 | asteriskguy | TK do you know of a good site that help with SpanDSP + RxFax/TxFax or the IAXMODEM + Hylafax? |
| 05:38.21 | Mattwj2005 | a practical use would be if you had a dialup backdoor on a router...you could use a voip phone line |
| 05:38.33 | asteriskguy | I tried asteriskguru.com but failed following their method |
| 05:38.37 | *** part/#asterisk snoopster (n=santa@cpe-76-187-204-88.tx.res.rr.com) |
| 05:38.59 | [TK]D-Fender | asterisk checkout iaxmodem on sourceforge |
| 05:39.00 | CunningPike | asteriskguy: We kind of kludged the call pickup you mean using some dialplan foo |
| 05:39.02 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 05:39.25 | asteriskguy | will do TK |
| 05:39.29 | [TK]D-Fender | asteriskguy, as for spandsp + rx/txfax... thats challenging.. not sure where to go for that. its been a serious PITA since 1.2.9.1 |
| 05:39.30 | CunningPike | asteriskguy: PM me on Wednesday with your email address and I'll send it to you |
| 05:40.55 | drfreeze | What sets the GMT in a polycom phone? I tried: http://pastie.textmate.org/56090, but didn't work |
| 05:40.55 | kiwoneka | hmm |
| 05:40.55 | drfreeze | [TK]D-Fender: Hi |
| 05:40.56 | asteriskguy | PITA TK? |
| 05:40.56 | kiwoneka | i guess the element i am missing is allcall.agi |
| 05:40.56 | asteriskguy | Ok CunningPike |
| 05:40.56 | asteriskguy | thanks |
| 05:40.56 | kiwoneka | is there an updated version for 1.4 |
| 05:40.56 | [TK]D-Fender | drfreeze, wrong unit of measure, and your tag isn't closed right. Thats just MESSY... go read the admin guide again. |
| 05:40.56 | CunningPike | ~pita |
| 05:41.05 | jbot | [pita] pain in the ass, or a bread-like food |
| 05:41.08 | [TK]D-Fender | asteriskguy, Pain In The Ass |
| 05:41.10 | drfreeze | never mind, I see a bug |
| 05:41.11 | asteriskguy | oh....haha |
| 05:41.32 | asteriskguy | I saw a weird problem on Asterisk ABE 1.3 the other day |
| 05:42.23 | asteriskguy | we opened port 5038 on manager.conf for one of our developer to connect to the server |
| 05:42.50 | asteriskguy | as soon as he attempted to connect, * died with a core dump, segmentation fault |
| 05:42.54 | asteriskguy | every single time |
| 05:44.51 | asteriskguy | but gotta admit, I must give props to Digium's tech support |
| 05:45.14 | asteriskguy | patient and knowledgeable |
| 05:47.09 | drfreeze | [TK]D-Fender: too many standards to choose from. :) |
| 05:48.02 | kiwoneka | are there any good resources to aquire agi scripts for asterisk? |
| 05:48.12 | [TK]D-Fender | drfreeze, its all documented in black & white. Try actually READING the admin guide :) |
| 05:49.04 | drfreeze | [TK]D-Fender: what admin guide are you referring to? The voip-info pages or some polycom guide? |
| 05:49.21 | [TK]D-Fender | drfreeze, The polycom SIP admin guide. |
| 05:49.50 | asteriskguy | There's a new SIP firmware just came out |
| 05:49.52 | asteriskguy | 2.1.1 |
| 05:50.25 | asteriskguy | suppose to fix the polycom locking up and restarting when use along with G729 |
| 05:50.42 | asteriskguy | for Polycoms |
| 05:52.18 | [TK]D-Fender | ben out for at least 2 weeks now |
| 05:53.01 | asteriskguy | yeah, I pushed it out to three phones to test it |
| 05:54.13 | kiwoneka | if iam running 1.6.7 do i need upgrade |
| 05:54.27 | asteriskguy | ok TK, thanks for the reference on IAXMODEM, I'll do some more reading on that |
| 05:54.50 | [TK]D-Fender | kiwoneka, Do you THINK you need to upgrade? |
| 05:54.52 | drfreeze | <PROTECTED> |
| 05:55.17 | drfreeze | mcab: the phone that was plugged into the analog jack won't get an IP address |
| 05:55.24 | kiwoneka | all is well in my little word |
| 05:55.31 | kiwoneka | just seeking an opinion |
| 05:55.40 | kiwoneka | world |
| 05:57.02 | asteriskguy | you should upgrade |
| 05:57.45 | kiwoneka | explain |
| 05:57.52 | asteriskguy | at least polycom seem to be recommending to upgrade |
| 05:58.26 | asteriskguy | try this kiwoneka, download the admin guide for SIP 2.0 and read it, see what's the difference between the two and decide for yourself |
| 05:58.56 | asteriskguy | alright [TK]D-Fender & CunningPike, thanks for everything. Good night |
| 06:11.02 | kiwoneka | ok, 1 down, intercom now works |
| 06:11.12 | kiwoneka | now to paging |
| 06:17.32 | JT | kiwoneka: how did you do intercom? |
| 06:18.13 | kiwoneka | [intercom] |
| 06:18.13 | kiwoneka | exten => _*7XX,1,SIPAddHeader(Alert-Info: Ring Answer) ;Polycom |
| 06:18.13 | kiwoneka | exten => _*7XX,n,Dial(sip/${EXTEN:1}) |
| 06:18.13 | kiwoneka | exten => _*7XX,n,Hangup |
| 06:18.13 | kiwoneka | exten => _*7XX,102,Hangup |
| 06:18.25 | kiwoneka | i will use pastebin nextime |
| 06:18.38 | *** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net) |
| 06:18.43 | JT | kiwoneka: does that auto answer? |
| 06:18.49 | kiwoneka | and i included that in my trusted contxt |
| 06:18.53 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:19.00 | kiwoneka | yes |
| 06:19.04 | JT | cool |
| 06:19.06 | kiwoneka | works very well |
| 06:19.18 | JT | does the phony ring at all? |
| 06:19.23 | JT | phone |
| 06:19.24 | kiwoneka | no |
| 06:19.28 | JT | hmm |
| 06:19.34 | kiwoneka | you can make it |
| 06:19.42 | kiwoneka | ring if you chose |
| 06:19.48 | JT | different header? |
| 06:20.21 | kiwoneka | in sip.cfg |
| 06:20.30 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 06:20.36 | JT | ah ok |
| 06:20.52 | JT | kiwoneka: what happens if you try to intercom a phone that is on a call? |
| 06:21.02 | kiwoneka | this is the resource i just used http://www.voip-info.org/wiki/view/Polycom+auto-answer+config |
| 06:21.29 | kiwoneka | you can set that too, in the dial plan |
| 06:21.45 | kiwoneka | i have hinting working |
| 06:21.51 | kiwoneka | with my buddies |
| 06:22.02 | JT | in your experience though... if the phone is already on a call, and you try to intercom to it? |
| 06:22.12 | kiwoneka | so, naturally you would not intercom someone that is on the phone |
| 06:22.21 | kiwoneka | vm |
| 06:22.30 | kiwoneka | that is what i have it set to |
| 06:22.33 | *** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net) |
| 06:22.49 | kiwoneka | i am working on paging right now |
| 06:23.17 | JT | kiwoneka: you might not know if someone is on the phone |
| 06:23.22 | JT | especially if the company is big |
| 06:23.34 | *** join/#asterisk Bazy (i=bazy@80.96.184.61) |
| 06:23.43 | kiwoneka | but i do |
| 06:23.49 | kiwoneka | the phones tell me |
| 06:24.09 | kiwoneka | all the phones subscribe to the state of each extension |
| 06:24.52 | kiwoneka | if your using the polycom5xx, 6xx they tell you |
| 06:25.04 | *** join/#asterisk lenne_dk (n=leif@cpe.atm2-0-74391.0x535cc77e.hknxx4.customer.tele.dk) |
| 06:25.14 | JT | it's limited though isn't it? |
| 06:25.20 | kiwoneka | no |
| 06:25.37 | JT | you can subscribe to unlimited extension states? |
| 06:25.39 | kiwoneka | i have not run into any complaints yet |
| 06:25.53 | mcab | JT: there used to be a limit of 7, but that was fixed ages ago |
| 06:26.05 | JT | it's infinity now? |
| 06:26.07 | *** join/#asterisk zoranoth (n=gla@139.sub-75-202-208.myvzw.com) |
| 06:26.11 | kiwoneka | well |
| 06:26.24 | mcab | JT: 48, I think |
| 06:26.39 | kiwoneka | only limiting factor, is how many sidecars you have |
| 06:26.44 | mcab | so, unless you can hack a 601 to take more than 3 EMs... :-) |
| 06:26.45 | JT | mcab: oh that needs lots of attendant consoles too doesn't it? |
| 06:26.49 | JT | heh |
| 06:27.11 | kiwoneka | i also understand what you mean |
| 06:27.25 | JT | let's assume you're not the receptionist or someone with an attendant console |
| 06:27.37 | JT | most SIP phones don't have that much info |
| 06:27.39 | kiwoneka | its entirly up to you, how you handle it |
| 06:27.43 | kiwoneka | the dial plan |
| 06:27.53 | JT | build it yourself, yes i know |
| 06:27.55 | kiwoneka | yes |
| 06:28.10 | JT | i just want to know what happens with the polycom, if it barges in or what |
| 06:28.27 | kiwoneka | then, if you intercom, and i am on the phone, i will get notifyied as if it were any other call coming in |
| 06:28.40 | mcab | JT: if it's just coming in as another INVITE, I suspect it will behave as if a 2nd call is coming in |
| 06:28.57 | kiwoneka | that is what happens |
| 06:29.27 | kiwoneka | but, that is dial plan dependant |
| 06:29.27 | mcab | but, it won't disrupt the current call |
| 06:29.35 | kiwoneka | no |
| 06:29.43 | kiwoneka | i imagine it can |
| 06:29.56 | mcab | kiwoneka: how so? |
| 06:30.26 | kiwoneka | you can probably force a hold command |
| 06:30.33 | kiwoneka | i dont know |
| 06:32.04 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 06:32.14 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 06:33.49 | kiwoneka | i am stuck |
| 06:34.07 | kiwoneka | i need to create a pagegroup |
| 06:34.18 | kiwoneka | exten => _*7243,n,Page(${PAGE_GROUP}) |
| 06:34.42 | *** join/#asterisk grEvenX (n=even@ti500720a080-4710.bb.online.no) |
| 06:35.43 | lenne_dk | Hi channel. I want to automatically call phone A and playback a message if phone B is not registered. I have made an extension in the dialplan, which checks chanisavail, and plays the apprpriate message. I then make a call-file, which uses this extension. But it calls ext A first, and then runs my extension. |
| 06:36.21 | lenne_dk | I'd like to check first, and then dial out if needed. |
| 06:36.49 | lenne_dk | Can I run an extension from a callfile, without calling a number first? |
| 06:36.58 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.169.195) |
| 06:38.10 | JT | run an extension? |
| 06:38.13 | JT | you mean call it? |
| 06:38.51 | lenne_dk | No, do the stuff in the dialplan, which checks if chan B is available. |
| 06:39.20 | lenne_dk | Then the exten => in the dialplan can decide if to call chan A or not. |
| 06:39.57 | *** join/#asterisk sgrover (n=sgrover@70.73.128.163) |
| 06:40.07 | JT | so put the checking stuff first in the dialplan? |
| 06:41.02 | lenne_dk | Yes, but how do I trigger running the ext from a call-file without calling a phone first? |
| 06:41.37 | JT | exactly how long does it take to check if an extension is available? |
| 06:41.38 | lenne_dk | I'd periodically copy a callfile to the spooldir to trigger the "call"/test |
| 06:41.44 | JT | i see |
| 06:41.53 | JT | maybe a callfile is the wrong way then |
| 06:41.56 | lenne_dk | milliseconds. I guess |
| 06:42.02 | JT | not sure if you can make a call file not call |
| 06:42.34 | lenne_dk | So an idea to get the chanisavail info to a script? |
| 06:42.51 | JT | perhaps use AMI |
| 06:43.22 | flenders | AMI => nasal delivery technology?? |
| 06:43.38 | flenders | jt: heard that on the radio? |
| 06:43.39 | flenders | :D |
| 06:43.40 | JT | Asterisk Manager Interface |
| 06:43.47 | JT | someone's been listening to too much radipo |
| 06:43.49 | JT | yes :P |
| 06:44.02 | lenne_dk | Or just run "sip show peers" on the commandline and grep for the channel. |
| 06:44.16 | flenders | listening to radio instead of doing the 343 things on my list before I go away |
| 06:44.35 | JT | heh |
| 06:45.59 | lenne_dk | Actually, I'd prefer chanisavail, because I the remote office has both a sip and an iax2 phone. And a POTS, which I want to make the announcement "Reboot the f&!§$ ip-phone again" on |
| 06:46.15 | kiwoneka | 2/2 |
| 06:46.19 | kiwoneka | i win |
| 06:46.29 | kiwoneka | i also have paging working |
| 06:46.36 | kiwoneka | i am happy guy |
| 06:46.40 | kiwoneka | this moring |
| 06:46.50 | kiwoneka | omg it 2.46am |
| 06:46.56 | kiwoneka | bed time |
| 06:47.16 | lenne_dk | Happy dreams |
| 06:47.27 | kiwoneka | will be |
| 06:47.33 | kiwoneka | i was successful |
| 06:47.36 | kiwoneka | tonight |
| 06:47.48 | kiwoneka | could not sleep anyway |
| 06:47.57 | JT | kiwoneka: how hard was paging/ |
| 06:48.02 | kiwoneka | might as well be productive |
| 06:48.10 | kiwoneka | not really |
| 06:48.12 | JT | lenne_dk: ami might be able to do it |
| 06:48.16 | mosty | lenne_dk: what kind of ip phone is it? |
| 06:48.16 | lenne_dk | Yes yes, that's enough bragging, or Nemesis will strike :-) |
| 06:48.31 | kiwoneka | did not mean to |
| 06:48.43 | kiwoneka | it took me a few days |
| 06:48.47 | lenne_dk | That's ok, no worry :-) |
| 06:49.12 | lenne_dk | Congrats anyway. |
| 06:49.39 | kiwoneka | you must have page.agi in /var/lib/asterisk/agi-bin |
| 06:49.49 | kiwoneka | that is the biggie |
| 06:49.57 | kiwoneka | hte rest iall dial plan |
| 06:50.02 | *** part/#asterisk zoranoth (n=gla@139.sub-75-202-208.myvzw.com) |
| 06:50.19 | kiwoneka | ;paging |
| 06:50.19 | kiwoneka | exten => _*7243,1,Set(TIMEOUT(absolute) = 15) |
| 06:50.19 | kiwoneka | exten => _*7243,n,AGI(page.agi|XXX|XXX) ; where XXX is extension(s) to always exclude, optional |
| 06:50.19 | kiwoneka | exten => _*7243,n,SetCallerID("Page:${CALLERIDNAME}"<${CALLERIDNUM}>) |
| 06:50.19 | kiwoneka | exten => _*7243,n,SIPAddHeader(Alert-Info: Ring Answer) |
| 06:50.20 | kiwoneka | exten => _*7243,n,Page(${PAGE_GROUP}) |
| 06:50.21 | kiwoneka | exten => _*7243,99,Hangup |
| 06:50.36 | JT | i assume you get the agi from somehrre |
| 06:50.39 | JT | somewhere |
| 06:50.41 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 06:50.46 | kiwoneka | yes |
| 06:50.57 | kiwoneka | i can pasebin |
| 06:51.02 | kiwoneka | if you like |
| 06:51.49 | JT | is it not already online? |
| 06:52.06 | kiwoneka | i lost the link |
| 06:52.08 | kiwoneka | :( |
| 06:52.26 | JT | is it on voip-info? |
| 06:52.38 | kiwoneka | no |
| 06:53.34 | kiwoneka | http://www.voip-info.org/wiki/view/Script+to+page+mixed+SIP+%252F+SCCP+system |
| 06:53.45 | kiwoneka | you were right JT |
| 06:53.50 | FuriousGeorge | kiwoneka: are you trying to devise a way of paging a phone vs just calling it? |
| 06:54.02 | kiwoneka | no |
| 06:54.26 | FuriousGeorge | n, then :) |
| 06:54.29 | kiwoneka | think of a warehouse senerio |
| 06:54.31 | FuriousGeorge | *nm |
| 06:54.36 | FuriousGeorge | thinkinb |
| 06:54.38 | FuriousGeorge | thinking |
| 06:55.01 | FuriousGeorge | a horn, so to speak? |
| 06:55.07 | kiwoneka | reception paging bob to pick up a parked call |
| 06:55.33 | FuriousGeorge | kiwoneka: ive done some looking into this |
| 06:55.39 | kiwoneka | next challeng |
| 06:55.45 | kiwoneka | streaiming moh |
| 06:55.57 | kiwoneka | but not tonight |
| 06:56.04 | FuriousGeorge | companies like valecom (or something) make these horns, and they also make adapters for them to be on fxs/fxo channels |
| 06:56.15 | kiwoneka | i am going to try get my slimserver |
| 06:56.15 | FuriousGeorge | obviously i meant fxs xhannels |
| 06:56.20 | kiwoneka | in the mix of things |
| 06:56.44 | kiwoneka | works there too if you have a phone that can auto answer |
| 06:57.00 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 06:57.49 | kiwoneka | good night to all |
| 06:57.58 | kiwoneka | i thank you for the inspiration |
| 06:58.00 | kiwoneka | :) |
| 06:59.11 | sumasuma | when asterisk appliance will be there for sale ? |
| 06:59.34 | kiwoneka | just build one, then you dont have to wait |
| 07:00.08 | sumasuma | kiwoneka: i have one running with EPIA mother board, but i have seen one of the appliance it is robust enough |
| 07:00.59 | sumasuma | kiwoneka: you know how to build one similar to asterisk appliance with the same capabilities ? |
| 07:01.05 | sumasuma | don't suggest me big PC ! |
| 07:01.10 | kiwoneka | lots of exciting things happening on the appliance dvelopment end |
| 07:01.27 | sumasuma | kiwoneka: what you mean ? |
| 07:01.41 | kiwoneka | my good sir, i am a novice |
| 07:01.45 | sumasuma | you mean it is still under development ? |
| 07:02.16 | kiwoneka | but i happen to know of a gentalman in this here very channel that does that sort of thing |
| 07:02.24 | kiwoneka | tzanger: |
| 07:02.41 | sumasuma | ha ha |
| 07:02.41 | kiwoneka | talk to him, he is a mad scientist |
| 07:03.41 | uwe | hello, i have asked this question yesterday, but got disconnected hard, i have bad voice quality, with inturruptions and apparently a very annoying experience, when i do sip show peers, i see in the status high number in milliseconds , like >500 ms, and ping is ~60 ms, when i posted yesterday, the status was ~150 ms and ping <2 ms ... so is there a tool to measure that time instead of using asterisk to see it, in order to do testing on the network? and bt |
| 07:03.41 | uwe | w, what exactly are these numbers, any keywords would be helpful ! |
| 07:03.56 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 07:04.27 | JT | well >500ms is bad definitely |
| 07:05.11 | uwe | JT, now its back to 140-190 ms |
| 07:05.13 | *** join/#asterisk af_ (n=getsmart@81-174-46-10.f5.ngi.it) |
| 07:12.01 | lenne_dk | A quick, partially OT: in bash, how do I output a here-document to a file? echo > myfile.call <<STOP Channel: SIP/123 etc STOP doesn't work |
| 07:12.47 | mosty | lenne use cat, not echo |
| 07:13.25 | JT | here-document? |
| 07:15.17 | lenne_dk | mosty: elaborate. cat myfile <<STOP Channel: etc STOP doesn't work either |
| 07:15.35 | JT | lenne_dk: have no idea what you're talking about |
| 07:15.52 | lenne_dk | Then I'l just talk to mosty :-) |
| 07:16.05 | JT | or perhaps you could explain better |
| 07:16.10 | *** join/#asterisk mvanbaak (n=mafkees@vanbaak.xs4all.nl) |
| 07:16.11 | JT | what is a "here-document"? |
| 07:16.47 | tris | JT: cat <<HEREDOC |
| 07:16.51 | tris | maybe |
| 07:17.00 | lenne_dk | exactly |
| 07:17.29 | JT | still doesn't explain what the hell a here document is |
| 07:17.33 | tris | lenne_dk: you do have carriage returns in there, right? |
| 07:18.22 | tris | a way of specifying a multi-line string literal which is terminated by the given string (such as HEREDOC in my above example, although it's usually something like "EOF") |
| 07:18.34 | tris | in a shell |
| 07:18.53 | JT | i see |
| 07:19.32 | lenne_dk | I do have cr. I'm looking for pastebin... |
| 07:20.00 | tris | I think what you want is cat >myfile <<EOF |
| 07:20.29 | mosty | lenne_dk: what tris said |
| 07:20.51 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 07:21.03 | lenne_dk | Right. |
| 07:21.41 | lenne_dk | Darn, why do I have a poppwd in /tmp created right now? |
| 07:22.02 | lenne_dk | Containing lines of \nAT ixuCheteki |
| 07:22.08 | lenne_dk | Rooted? |
| 07:23.07 | lenne_dk | Oh, silly me... <blush> |
| 07:23.49 | drfreeze | Anyone know if the 301's take a different config file than the 501's? |
| 07:27.25 | *** join/#asterisk dhakatel (n=root@58.65.224.5) |
| 07:28.11 | JT | root@ |
| 07:31.59 | *** join/#asterisk mkl1525 (n=mkl1525@pD9532807.dip0.t-ipconnect.de) |
| 07:33.53 | lenne_dk | A minor annoyance, why do callfiles need to be owned by asterisk? why isn't mode rwrwrw enough? |
| 07:34.39 | JT | what about mode x? |
| 07:36.14 | lenne_dk | x is not relevant, as the os i not executing it, asterisk is merely reading it. But rwxrwxrwx is no better. Just strange |
| 07:36.14 | mosty | lenne_dk: does asterisk need to be able to remove call files? |
| 07:36.33 | *** join/#asterisk Keltus (n=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 07:36.43 | mosty | in which case, you will probably want to look at the permissions on the directory |
| 07:36.45 | lenne_dk | It do remove the files, it doesn't execute it |
| 07:37.11 | mosty | what does the verbose log from asterisk show? |
| 07:37.51 | lenne_dk | <PROTECTED> |
| 07:38.26 | lenne_dk | a file perms 666, owned by nagios. |
| 07:39.12 | lenne_dk | nagios (network monitor script) calls me with alerts :-) |
| 07:40.18 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 07:40.38 | drfreeze | d |
| 07:40.49 | lenne_dk | nagios calls me and reads the errormessages using festival. |
| 07:41.13 | lenne_dk | No big deal, just have to remember doing a chown. |
| 07:45.39 | mosty | maybe you can get away with a chgrp, or even simpler setgid on the dir |
| 07:48.26 | lenne_dk | mode 4777 ? drwxrwsrwx ? |
| 07:49.06 | JT | no can't you simply chown the directory? |
| 07:49.21 | JT | or is file creation overriding the defaults? |
| 07:49.29 | lenne_dk | the dir is owned by asterisk |
| 07:49.39 | JT | look at your script then |
| 07:49.42 | JT | run it as asterisk |
| 07:50.32 | mosty | if a dir is setgid, when you create files inside, they will be owned by the same group as the dir (assuming you are in that group) |
| 07:50.43 | lenne_dk | Can't, it runs as nagios. But as long as I can chown it, that's the solution. Just wondered why asterisk can't read it when it can delete it. |
| 07:51.01 | JT | security reasons |
| 07:53.44 | lenne_dk | OK, next problem: Can this be done: If A and B is talking with each other, if either is called, break the conversation and connect the incoming call? |
| 07:54.42 | JT | pretty hard i think |
| 07:54.48 | JT | to do nicely anyway |
| 07:58.09 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-208647.home.otenet.gr) |
| 08:00.20 | mkl1525 | Hi,(* 1.2) trying to enable attended transfer in a queue (atxfer => *2 in features.conf) but when I press * call is gone, although the h and H options are not set in Queue() command but the t options. Any hints what could go wrong? |
| 08:03.35 | mosty | mkl1525: how quickly are you diallin the 2? * by itself is "disconnect call" |
| 08:03.36 | lenne_dk | Try using another sequence than *2. eg 99. And check the spacing between the digits, some phones doesn't send tones quickly when in a call. |
| 08:10.13 | *** join/#asterisk xermesx (n=ermsewrk@217.220.121.62) |
| 08:14.08 | mkl1525 | thanks seems that this is the problem using "11" seems to works |
| 08:15.15 | mosty | i wonder, is it possible to disable builtin features (disconnect call) for example, or can you only remap them? |
| 08:16.03 | lenne_dk | Well, if you remap it to 8980283128302131283091 it is disabled effectively :-) |
| 08:16.43 | mosty | i don't know why you would ever need to use * to disconnect a call, my phones have distinct buttons for that |
| 08:17.45 | lenne_dk | It's for call-center use, where the drones wear headsets, and dial using a softphone, I believe. |
| 08:18.57 | *** join/#asterisk ltd (n=z@ppp167-251-11.static.internode.on.net) |
| 08:23.38 | *** join/#asterisk allankardec (n=root@189-19-59-138.dsl.telesp.net.br) |
| 08:23.44 | allankardec | hello all |
| 08:24.42 | creativx | drones |
| 08:24.43 | creativx | teheh |
| 08:24.50 | *** join/#asterisk pvanstam (n=Pim@dsl-083-247-065-012.solcon.nl) |
| 08:25.09 | allankardec | is there anyone that know to play with playback functions the format g729? |
| 08:25.19 | allankardec | sorry, I am brazillian |
| 08:25.45 | mosty | you need to buy a license to use digium's g729 codec |
| 08:26.45 | allankardec | I bought of digium |
| 08:26.55 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:27.19 | pvanstam | you can also transcode the messages to g729 and then play it natively |
| 08:27.23 | mosty | download the codec_g729a.so into the correct location, then run the registration util? |
| 08:27.46 | mosty | pvanstam: you need to have a g729 licence to do that in the first place though, heh |
| 08:28.02 | allankardec | |
| 08:28.02 | allankardec | already I bought |
| 08:28.49 | pvanstam | mosty: ok, I don't do it myself, I prefer pcm |
| 08:29.46 | pvanstam | BTW, decoding can be done via the web. There are sites where you can decode it to other formats. In that case you don't need to do it yourself |
| 08:30.15 | pvanstam | I remember doing that once. I don't have a license, since I'm using FreeBSD 6 |
| 08:31.11 | mosty | yeah you can do that |
| 08:32.23 | allankardec | I receive a call sip using g729 codec, at this moment i try to touch an archive with gsm format, |
| 08:32.51 | mosty | allankardec, did you do this: <mosty> download the codec_g729a.so into the correct location, then run the registration util? |
| 08:33.21 | allankardec | the asterisk information "(format 0x100 (g729)): No such file or directory" |
| 08:34.14 | allankardec | mosty, this already i made |
| 08:34.36 | *** join/#asterisk X-Rob_ (n=rob-x@CPE-58-167-128-40.qld.bigpond.net.au) |
| 08:35.34 | pvanstam | allankardec: it's clear that g729 transcoding is not done in the *, so is the code active? |
| 08:36.03 | allankardec | the codec is active |
| 08:37.48 | allankardec | the call between two sips with g729 codec is functioning |
| 08:38.16 | allankardec | my problem is to play archieve |
| 08:39.16 | pvanstam | well there is no transcoding in *, so this will work |
| 08:39.16 | pvanstam | You should set one SIP-phone to PCM or GSM and then check if a call between the 2 phones can be made |
| 08:39.44 | allankardec | I am connected to the vonage provider |
| 08:40.42 | mkl1525 | Is there another way to "finish" a number in attended transfer than waiting for digit timeout? # doesn't work |
| 08:40.56 | pvanstam | Try "Send" on Grandstream |
| 08:41.10 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 08:41.13 | allankardec | vonage provider to send the call for my asterisk server, my * then call one sip client, is function |
| 08:42.16 | allankardec | the vonage provider to send call with g729 |
| 08:42.21 | tsurko | Hello, everybody |
| 08:43.15 | tsurko | I've an asterisk set up with sip softphones on windows and linux. On linux (i use twinkle) everything is fine, but on windiws (with x-lite) i got this on CLI when try to dial through zap channel: |
| 08:43.20 | tsurko | Apr 24 11:37:50 WARNING[11056]: channel.c:2570 ast_request: No translator path exists for channel type Zap (native 68) to 1024 |
| 08:43.20 | tsurko | Apr 24 11:37:50 NOTICE[11056]: app_dial.c:1057 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) |
| 08:43.32 | tsurko | could you tell me what's wrong? |
| 08:43.45 | jql | codec mismatch |
| 08:44.14 | tsurko | jql, that was for me? |
| 08:44.49 | jql | yeah |
| 08:45.20 | tsurko | thank you, i'll check it out |
| 08:47.22 | mkl1525 | pvanstam, thanks for the suggestion having snoms here |
| 08:47.56 | pvanstam | On Snom you press the Ok button (v) |
| 08:48.01 | *** join/#asterisk Ahrimanes (n=ma@81.7.159.2) |
| 08:48.22 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 08:48.31 | jql | one of my cow orkers stole my test snom... I'm still ticked |
| 08:48.39 | allankardec | thanks all |
| 08:49.11 | pvanstam | mkl1525: which Snom? |
| 08:54.51 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 08:56.39 | mkl1525 | pvanstam, 300 and 360 |
| 08:59.59 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 09:00.24 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 09:00.46 | e-ddie | 0 |
| 09:01.44 | pvanstam | ok, both have the tick key (v) and the abort key (x). Dial a number and then press the tick key. On both 300 and 360 they are next to the arrow keys |
| 09:02.38 | *** join/#asterisk SoMeOnEnUlL (i=morris@p1920-adslbkksp7.C.csloxinfo.net) |
| 09:03.05 | *** part/#asterisk SoMeOnEnUlL (i=morris@p1920-adslbkksp7.C.csloxinfo.net) |
| 09:03.47 | pvanstam | allankardec: audio conversion of archive can be done at: http://www.asteriskguru.com/audio_conversion.php |
| 09:04.41 | tsurko | jql, everything is fine now, thank you a lot |
| 09:05.35 | tsurko | in x-lite 3, codecs selection is in advanced tab in the bottom of the options window. It never crossed my mind that this is a button and it's supposed to be clicked |
| 09:16.18 | *** join/#asterisk mquin (n=mike@pdpc/supporter/active/mquin) |
| 09:16.33 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 09:18.38 | *** join/#asterisk darviria (n=darviria@194-105-181-29.ifb.co.uk) |
| 09:20.13 | mkl1525 | pvanstam, tried it but seems not to work |
| 09:22.14 | pvanstam | mkl1525L that is very strange. We have several Snoms too and use the tick key all the time. Does it work with normal phone call's? |
| 09:24.15 | pvanstam | With attended transfer you should perhaps just hangup. Depending on the config the call is transfered. You could also use the "Transfer" button. Third possibility (depending on config) is that you can use the "Cancel" button (x). That's a signal to your phone that you want to transfer and not stay in the middle of conversation. |
| 09:25.25 | *** join/#asterisk Zdrulio (n=sux_@82.119.72.130) |
| 09:25.27 | Zdrulio | hello all |
| 09:25.33 | mkl1525 | pvanstam, thanks for the hints will try them, tick key works normally as expected just in attended transfers from a queue it fails |
| 09:32.24 | *** part/#asterisk Geert (i=geert@irssi/staff/geert) |
| 09:54.41 | *** join/#asterisk mquin (n=mike@pdpc/supporter/active/mquin) |
| 09:55.04 | *** join/#asterisk Strom_M (n=strom@212.188.130.178) |
| 09:56.32 | Zdrulio | i have one question |
| 09:58.04 | Zdrulio | i have private numbers for office |
| 09:58.13 | Zdrulio | 102,103,104,105 |
| 09:58.29 | Zdrulio | i`m 101 |
| 09:58.41 | Zdrulio | i wish call to 102 |
| 09:59.20 | Zdrulio | i dial 102 and i wait 10sec |
| 09:59.30 | Zdrulio | can i decrease this 10 sec for respond |
| 10:00.46 | pvanstam | press #, Send or tick key on the phone |
| 10:00.55 | Zdrulio | i know that |
| 10:00.56 | Zdrulio | but |
| 10:01.07 | Zdrulio | i want to decrease wait time |
| 10:02.23 | pvanstam | make your dialplan in such a way that the internal numbers are not part of any outside number. I guess you have '.' as part of a trunk. |
| 10:02.36 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.voop.net) |
| 10:03.32 | JT | pvanstam: basically it's the difference between timeout dialling and explicit match dialling |
| 10:03.42 | JT | pvanstam: are they analogue FXS ports? |
| 10:04.25 | *** join/#asterisk mquin (n=mike@pdpc/supporter/active/mquin) |
| 10:04.28 | pvanstam | JT: right |
| 10:04.49 | nasls_lsa | does anyone have problems with: Grandstream GXW-4008 ? stacks and needs reboot to work again ? |
| 10:04.53 | JT | pvanstam: use more explicit dialplan patterns |
| 10:05.39 | JT | Zdrulio: i mean |
| 10:06.11 | Zdrulio | i don`t understand |
| 10:06.16 | pvanstam | JT: thought so |
| 10:06.38 | JT | Zdrulio: ok, paste your extensions.conf into pastebin.ca and we'll have a look |
| 10:06.54 | *** join/#asterisk misc-- (n=misc@203.87.183.218) |
| 10:07.23 | JT | Zdrulio: are they analogue FXS ports? |
| 10:07.49 | misc-- | hi, I want to install the g729 codec but its giving me unresolved errors indicating its the wrong arch. The g729 codec works with the Pentium D doesn't it? If so, then which codec should I use? |
| 10:08.11 | Zdrulio | mm |
| 10:08.16 | Zdrulio | asterisk |
| 10:08.21 | JT | Zdrulio: analogue phones? |
| 10:08.25 | Zdrulio | yes |
| 10:08.39 | Zdrulio | linksys ATA |
| 10:08.40 | JT | well obviously it uses asterisk, otherwise it wouldn't be relevant to this channel |
| 10:08.43 | JT | ok |
| 10:08.44 | Zdrulio | and phones behind |
| 10:08.54 | JT | so you need to modify the dialplan in the linksys ATA |
| 10:09.01 | JT | for more explicit pattern matching |
| 10:09.25 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:09.28 | Zdrulio | hm |
| 10:10.44 | *** join/#asterisk mendol (i=mendol@bobas.nowytarg.pl) |
| 10:10.51 | mendol | Afternoon |
| 10:11.04 | mendol | how can I enable iax2 trunk monitoring? :-/ |
| 10:11.42 | mkl1525 | from reading the comments in queue.conf I'd suppose that after "retry"-time all available agents are called again. But when a caller comes into queue and an additional agent logs in later it takes about 75 seconds till he gets a call signal - didn't find any variable with this value, any suggestions? |
| 10:12.08 | Zdrulio | JT are you know from where in ATA admin panel change taiming ? |
| 10:22.27 | *** join/#asterisk zogulus (n=zogulus@58.98.adsl.brightview.com) |
| 10:22.44 | *** part/#asterisk misc-- (n=misc@203.87.183.218) |
| 10:23.07 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 10:25.02 | *** join/#asterisk dj-fu (i=aj@unaffiliated/dj-fu) |
| 10:26.27 | nasls_lsa | does anyone have problems with: Grandstream GXW-4008 ? stacks and needs reboot to work again ? |
| 10:32.28 | *** join/#asterisk darkmug (n=dennis@143.106.7.170) |
| 10:42.55 | *** join/#asterisk claude81 (n=claude_t@claudetoma.plus.com) |
| 10:44.36 | *** join/#asterisk Ebola (n=Ebola@host86-136-130-202.range86-136.btcentralplus.com) |
| 10:46.03 | mosty | mkl1525: queues suck. reduce the retry time |
| 10:46.32 | mkl1525 | mosty, retry time is at 1 - that's why im wondering |
| 10:46.46 | mosty | 1 is probably too short |
| 10:46.49 | mosty | try 10 |
| 10:47.33 | *** join/#asterisk darviria (n=darviria@194-105-181-29.ifb.co.uk) |
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| 10:50.07 | mosty | there are a few other timeout settings from memory, have you tried playing with those? |
| 10:53.40 | mkl1525 | had a look at them but nothing in the near of 75 seconds |
| 10:56.31 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 10:56.37 | Zeeek | hi |
| 10:56.49 | Zeeek | anyone using or heard of Adhearsion.com ? |
| 10:56.53 | *** join/#asterisk JoanaDoe (n=yvonne@193.174.25.23) |
| 10:57.06 | Zeeek | or Ruby for that matter? |
| 10:57.27 | mosty | ruby, yes |
| 10:57.34 | Zeeek | anyone unsing asterisk? |
| 10:57.45 | pvanstam | duh |
| 10:57.48 | Zeeek | mosty how do you like ruby? |
| 10:58.21 | Zeeek | if you really like it the thing I mention above works with asterisk |
| 10:58.24 | mosty | i'm not a huge fan. i prefer strongly typed languages, much easier to debug |
| 10:58.47 | Zeeek | I haven't ever tried it (mostly because it requires way too much installation for someone as lazy as me) |
| 10:59.10 | Zeeek | I haven't updated linux in like three years! |
| 10:59.49 | Zeeek | ruby looks like "fun" |
| 10:59.59 | Zeeek | I'm reading this: http://www.linuxjournal.com/article/9519 |
| 11:13.13 | *** join/#asterisk rodwish (n=dfas@212.17.56.97) |
| 11:13.22 | rodwish | hi |
| 11:13.30 | rodwish | need some help |
| 11:13.38 | Zeeek | ask and you shall |
| 11:13.39 | rodwish | i got a quintum connected to an asterisk box |
| 11:13.44 | rodwish | thanks |
| 11:14.04 | rodwish | p[roblem is the quintum is getting answer code when the phone is just ringing |
| 11:14.19 | rodwish | i have a2billing install on the asterisk |
| 11:14.39 | rodwish | the calls are coming from a client who has quintum ax series fxs 8 port |
| 11:15.17 | rodwish | sip codec g729 |
| 11:15.50 | pvanstam | g729 codec on asterisk installed? |
| 11:16.04 | pvanstam | * has to trancode g729 and pcm i guess? |
| 11:16.40 | pvanstam | NAT in between perhaps? |
| 11:17.44 | rodwish | yes codec is installed |
| 11:17.59 | rodwish | the quintum is behind nat |
| 11:18.12 | rodwish | i tried on fix ip as well but same problem |
| 11:19.21 | pvanstam | did you take a look in the logs? |
| 11:19.33 | rodwish | if you check the cdr on the asterisk box u dod not see the call unless it is answered but if you check the quintum u see the call was answered |
| 11:19.38 | rodwish | yes i did |
| 11:19.46 | rodwish | never saw that problem b4 |
| 11:20.02 | pvanstam | could be the codec not finding it's way or NAT |
| 11:20.15 | rodwish | the astrerisk seem fine but the quintum getting answer code when the phone just ringing |
| 11:20.26 | pvanstam | can you do tcpdump, you can see if SIP and/or RTP is going well or not |
| 11:20.38 | rodwish | i tried without nat, nat not the issue |
| 11:20.49 | pvanstam | ack |
| 11:20.58 | rodwish | i tried g711 codec, same problem |
| 11:21.22 | rodwish | i contacted a2b they could not fix |
| 11:21.51 | pvanstam | how does it come into *, is there answer at the context where it comes in? |
| 11:24.27 | rodwish | no |
| 11:24.56 | rodwish | maybe quintum firmware |
| 11:25.03 | rodwish | i dont know |
| 11:25.47 | rodwish | the billing system on the quintum side sees the call as connected even just ringing whereas u can see the logs in the * fine |
| 11:26.47 | pvanstam | strange thing |
| 11:27.01 | pvanstam | asterisk is not dropping into voicemail or something? |
| 11:27.50 | rodwish | no nothing |
| 11:28.13 | rodwish | everthing works fine except the billing side on teh quintum detects the call as connected |
| 11:29.07 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:30.03 | rodwish | i tried to send call to another gateway but not asterisk they seem to work fine, so i guess it not the quimtum and the billing on the quintum side |
| 11:30.16 | rodwish | i dont know where to troubleshoot |
| 11:31.05 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 11:33.11 | rodwish | any1 |
| 11:33.12 | rodwish | ???? |
| 11:33.40 | DrukenHME | lets see a pastebin of the cli output for an incoming call |
| 11:36.21 | rodwish | 1 sec |
| 11:38.30 | *** join/#asterisk sorend (i=sorend@195.140.132.34) |
| 11:43.26 | rodwish | just a quick question how do we copy from mc in ssh to windows notepad so i can paste it here |
| 11:44.24 | *** join/#asterisk Dibbler_ (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com) |
| 11:58.01 | tzanger | wow.. 1700 messages in -users |
| 11:59.39 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 12:03.41 | pvanstam | rodwish: try Ctrl-Ins & Ctrl-V |
| 12:06.31 | rodwish | i just got a call from the people that made makes billing for the quintum |
| 12:06.47 | rodwish | they saying they did come across that problem b4 |
| 12:07.05 | rodwish | i need to remove asnwer in the a2billing as i am not using the ivr |
| 12:07.14 | rodwish | thats why it is doing that |
| 12:07.27 | rodwish | but they dont know where exactly to remove the asnwer |
| 12:07.49 | *** join/#asterisk Jynger (n=rip@tti.tt.ee) |
| 12:08.03 | Jynger | hi |
| 12:08.08 | rodwish | because a2b is made for calling card |
| 12:08.26 | rodwish | so no matter what the ivr plays, for the quintum box is sees it as connected |
| 12:08.36 | rodwish | thats why its doing that |
| 12:08.53 | Jynger | does asterisk realtime database users have NAT keepalive ? |
| 12:08.56 | rodwish | so i need to remove that option...now the problem is where to remove that option |
| 12:09.07 | rodwish | hi jynger |
| 12:09.22 | rodwish | its not nat like i tried with and without, im having same problem |
| 12:09.35 | Jynger | hello, i read from wiki: 'The database peers/users are not kept in memory. These are only loaded when we have a call and then deleted, so there's no support for NAT keep-alives (qualify=) or voicemail indications for these peers' |
| 12:09.46 | rodwish | is a config in the asterisk a2billing that i need to delete |
| 12:10.06 | rodwish | or to change from yes to no |
| 12:10.54 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:11.17 | LeddyHM | OMG |
| 12:11.26 | [TK]D-Fender | LeddyHM: y0 |
| 12:11.27 | LeddyHM | I think pigs can fly |
| 12:11.34 | [TK]D-Fender | LeddyHM: Get my msg yesterday? |
| 12:11.38 | LeddyHM | yeah |
| 12:11.41 | LeddyHM | already implemented |
| 12:11.49 | [TK]D-Fender | LeddyHM: Everyone happy? |
| 12:12.04 | LeddyHM | No one gets here for another 2-3 hours |
| 12:12.07 | [TK]D-Fender | :O |
| 12:12.16 | LeddyHM | damn slackers |
| 12:12.30 | [TK]D-Fender | Slackware = c00l |
| 12:12.42 | tzanger | [TK]D-Fender: amen |
| 12:12.47 | tzanger | Damn I didn't know you were a slacker |
| 12:13.08 | LeddyHM | he's a centos then slacker |
| 12:13.09 | Kigh | ive been using slackware for a long time and it sucks .. really :) |
| 12:13.09 | [TK]D-Fender | tzanger: "it just works". |
| 12:13.15 | tzanger | [TK]D-Fender: yep |
| 12:13.29 | tzanger | I'm looking at the SIP conntrack support in 2.6.19/20 right now |
| 12:13.33 | LeddyHM | I can't even remember the first version of slack I used |
| 12:13.34 | Kigh | today if i want it raw, i use BSD |
| 12:13.48 | tzanger | gonna see if I can hook that in to my rc.tc script |
| 12:14.00 | tzanger | LeddyHM: I started with the first version of slackware, right after SLS |
| 12:14.08 | [TK]D-Fender | tzanger: Sadly yes, I am going more down the CentOS route now. I do like some small level of package management, plus better order control on startup. I don't feel less in control, and slack is smaller and a little fast in some ways.... |
| 12:14.24 | tzanger | oh? |
| 12:14.28 | [TK]D-Fender | tzanger: but I'm also afraid to compile my own kernel and I have ZTDUMMY now :) |
| 12:14.37 | tzanger | I just use slackpkg, it is teh awesome |
| 12:15.26 | tzanger | [TK]D-Fender: heh; I lost all fear of compiling and breaking things when I made the great a.out -> elf transition |
| 12:15.27 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 12:15.38 | tzanger | and then again when I did libc5 -> libc6 |
| 12:15.41 | tzanger | that one was particularly painful |
| 12:15.45 | LeddyHM | tz: I was using it pre 1.0 kernel |
| 12:15.51 | LeddyHM | on my 386 baby! |
| 12:15.55 | [TK]D-Fender | tzanger: Swaret killed my server about 3 weeks ago and I was forced to rebuild. Was waiting to go CentOS for a while and got the unavoidable push... |
| 12:15.56 | tzanger | I learned about statically linked "ln" "mv" "ls" and "cp" :-) |
| 12:16.10 | tzanger | LeddyHM: yep, 0.99.6 I think is where I started |
| 12:16.21 | LeddyHM | damn, we old |
| 12:16.30 | tzanger | I still have my 80386DX33 (complete with double-sigma mark to show it could handle 32bit mul) |
| 12:16.36 | tzanger | actually no I don't have that anymore |
| 12:16.40 | tzanger | I just got rid of it in the last move |
| 12:16.42 | tzanger | it was a beast |
| 12:16.45 | LeddyHM | mine was a dx40 |
| 12:16.51 | LeddyHM | took 6 hours to compile |
| 12:16.56 | LeddyHM | and it HATED my sb card |
| 12:17.11 | tzanger | Datatech motherboard, SIPP memory (8M max), 64k cache with the logic implemented in discrete logic... wowza |
| 12:17.13 | LeddyHM | so inactuality it really took 3-4 days to get it just right |
| 12:18.14 | tzanger | yep |
| 12:18.39 | tzanger | about 5 years ago I actually found the DTK memory expansion card for it and a Weitek 387 coprocessor at a junkyard. I put it up to 16M of memory, which was the maximum it could take. :-) |
| 12:18.44 | tzanger | didn't make it much faster though |
| 12:18.47 | tzanger | morning file |
| 12:18.58 | LeddyHM | yeah mine had a coproc too |
| 12:19.01 | LeddyHM | ohh the days |
| 12:19.33 | tzanger | now I've got a MCF5282 (ColdFire) board I've designed... back to using the kernel software coprocessor, heh |
| 12:20.52 | tzanger | hmm |
| 12:21.00 | tzanger | 2.6.20.4 has a "working" SIP nat-t |
| 12:21.07 | tzanger | does that mean that 2.6.19 and 2.6.20's implementation don't work? |
| 12:21.55 | *** part/#asterisk sorend (i=sorend@195.140.132.34) |
| 12:23.31 | tzanger | ok 2.6.20 base will accept that sip alg patch... 2.6.19 won't, at least not cleanly |
| 12:47.12 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:54.21 | *** join/#asterisk saravia (n=jovannot@190.84.99.36) |
| 12:54.55 | *** join/#asterisk VJFROMGT (n=vjfromgt@user-387g9ui.cable.mindspring.com) |
| 12:55.00 | VJFROMGT | Apr 24 08:54:06 WARNING[28864] chan_visdn.c: Unable to load config visdn.conf, VISDN disabled |
| 12:55.06 | VJFROMGT | any ideas? |
| 12:56.44 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 12:57.39 | [TK]D-Fender | VJFROMGT: First guess.... make a config file for that channel driver if you expect it to work. |
| 12:58.06 | VJFROMGT | file is present |
| 12:58.39 | JT | wow, the only person on the planet running visdn :P |
| 12:58.59 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 12:59.13 | VJFROMGT | its actually vdsn which works with visdn driver |
| 12:59.18 | VJFROMGT | i mean vgsm |
| 12:59.27 | VJFROMGT | calls via sim card |
| 12:59.50 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 13:01.10 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 13:08.44 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 13:09.31 | VJFROMGT | when i do amportal start asterisk tries to load certian modules, one of them crash every time, how can i prevent it from trying to load? |
| 13:11.10 | JT | amportal? |
| 13:11.23 | VJFROMGT | service asterisk start |
| 13:11.45 | JT | you said amportal |
| 13:12.01 | VJFROMGT | same issue when i run service asterisk start |
| 13:12.09 | JT | what is amportal? |
| 13:12.24 | VJFROMGT | freepbx |
| 13:12.33 | JT | i see, that's not supported here |
| 13:12.38 | VJFROMGT | i know |
| 13:12.38 | saravia | what is VISDN ? virtual ISDN lines ? |
| 13:12.47 | VJFROMGT | which is why i mention service asterisk start |
| 13:12.56 | JT | VJFROMGT: trying to trick us, nice |
| 13:13.07 | JT | saravia: no, it's an ISDN channel driver |
| 13:13.08 | VJFROMGT | i am running visdn driver for a vgsm card |
| 13:13.21 | VJFROMGT | (use sim card to dial out) |
| 13:13.26 | saravia | if you want to avoid to load some module, just remove it from /usr/lib/asterisk/modules VJFFROMGT |
| 13:13.34 | VJFROMGT | thanks |
| 13:14.11 | VJFROMGT | you mean like delete the file? |
| 13:14.34 | JT | which is bad advice |
| 13:14.40 | JT | better to make it not load |
| 13:14.52 | VJFROMGT | i prefer to make it not load,, |
| 13:15.03 | VJFROMGT | which file script calls the file? |
| 13:15.11 | saravia | well if you prefer just move to another place, and test it |
| 13:15.32 | JT | modules.conf |
| 13:15.41 | JT | saravia: no, you can also do it properly. |
| 13:15.42 | saravia | I dont know the way to not allow to load an module, of course it will be the better choice |
| 13:17.17 | tzanger | wtf... |
| 13:17.22 | tzanger | exten => 123,1,... |
| 13:17.27 | tzanger | exten => 456,1,... |
| 13:17.30 | tzanger | include = foo |
| 13:17.42 | JT | tzanger: hmm? |
| 13:17.43 | tzanger | exten => _XXX,1,NoOp(wildcard match) |
| 13:17.47 | tzanger | [foo] |
| 13:17.57 | tzanger | exten => 789,1,NoOp(foo context match) |
| 13:17.57 | saravia | ok JT, then you write noload in modules, and thats it, right ? |
| 13:18.05 | JT | yes |
| 13:18.17 | tzanger | dialplan show bar (the first context) |
| 13:18.23 | tzanger | shows include foo AFTER the _XXX |
| 13:18.25 | saravia | yes,is a better way ! |
| 13:18.29 | VJFROMGT | modules.conf does not have an entry for visdn |
| 13:18.31 | tzanger | that is not right |
| 13:18.38 | *** join/#asterisk NormanAthol (n=filenotf@203.208.76.227) |
| 13:18.41 | tzanger | I included before the wildcard, it should search the included context BEFORE the wildcard matches |
| 13:18.43 | JT | VJFROMGT: then you add one, jeebus |
| 13:18.49 | NormanAthol | hi |
| 13:19.15 | saravia | noload => chan_visd.so or something like that ! |
| 13:19.31 | VJFROMGT | but where is it been called from at this time? |
| 13:19.40 | JT | he uses freepbx, no wonder he has some much trouble with config files |
| 13:19.48 | JT | s/some/so/ |
| 13:19.48 | saravia | autoload=yes |
| 13:20.10 | VJFROMGT | i am a linux noob |
| 13:20.32 | Corydon76-home | The list is generated automatically from the directory listing |
| 13:20.52 | NormanAthol | i was wondering if someone could help me out i am having trouble getting asterisk going on debian 4 i have done this before but the only difference this time is that the machine is running samba which is AD intergrated my problem is that when asterisk strats is dosent apear to be listening on port 5060 |
| 13:21.36 | *** join/#asterisk mquin (n=mike@pdpc/supporter/active/mquin) |
| 13:21.50 | JT | NormanAthol: udp 5060? |
| 13:22.09 | NormanAthol | http://pastebin.ca/455582 < that is a copy of runnying asterisk -vvvv it says in there that sip is listening on 5060 but netstat seems to prove this wrong |
| 13:22.23 | NormanAthol | its for sip whihc is tcp isnt it |
| 13:22.29 | JT | no, udp |
| 13:22.57 | NormanAthol | ok there is something new ok then it is listening |
| 13:23.37 | JT | it's not new, sip has always been udp as a standard |
| 13:24.01 | NormanAthol | but l cant connct to the box it always times out there is no firewall set up on the debian box at this point and i have tried connecting to it from a local computer on there network and also remotely on my network |
| 13:24.06 | Corydon76-home | Actually, the standard is both, but most implement only UDP |
| 13:24.36 | NormanAthol | i thought sip was TCP but then the actual data was send through RTP which was udp |
| 13:24.45 | JT | i realise that, it's irrelevant though |
| 13:24.49 | JT | as asterisk doesn't support it |
| 13:24.58 | JT | and lots of SIP clients dont support it |
| 13:25.13 | JT | NormanAthol: better to check for sure than to imagine ;) |
| 13:25.26 | NormanAthol | yeah ok so i know that bit now |
| 13:26.01 | JT | the only reason i've ever heard of anyone playing with SIP over TCP is to connect to some stupid Microsoft groupware thing |
| 13:26.03 | Corydon76-home | NormanAthol: no, that's H.323 |
| 13:26.27 | NormanAthol | but it still dosent explain why its not working i have used the config files as reference from my server which works fine |
| 13:26.53 | Corydon76-home | Are you perhaps binding to a particular IP address instead of 0.0.0.0 ? |
| 13:27.03 | NormanAthol | the only differency between the 2 machines is that one is AD intergrated but i fail to see how that could really change anything |
| 13:27.09 | Corydon76-home | Check bindaddr in sip.conf |
| 13:27.25 | NormanAthol | Corydon-w, i have tried 0.0.0.0 192.168.0.9 |
| 13:27.27 | tzanger | this is *not* right |
| 13:27.31 | NormanAthol | both do not work |
| 13:27.57 | saravia | please send the output of: "nestat -an | grep 5060" |
| 13:28.05 | saravia | netstat I mean |
| 13:28.26 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 13:28.37 | tzanger | http://www.pastebin.ca/455772 |
| 13:28.47 | tzanger | how the hell are you supposed to nest contexts with any kind of priority? |
| 13:34.14 | *** join/#asterisk etfonhomey (n=etfonhom@74-140-213-69.dhcp.insightbb.com) |
| 13:34.32 | *** join/#asterisk coppice (n=chatzill@94.143.17.210.dyn.pacific.net.hk) |
| 13:34.46 | saravia | why you have 2 times exten 789 with priority 1. that must be an error |
| 13:35.04 | *** part/#asterisk nettie (n=nettie@ns.coolgadgets.it) |
| 13:39.03 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
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| 13:43.11 | *** mode/#asterisk [+o anthm] by ChanServ |
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| 13:45.19 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:47.35 | *** join/#asterisk joe-f (n=joef@c-71-201-188-239.hsd1.il.comcast.net) |
| 13:53.28 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
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| 13:57.25 | AndrewGearhart | any opinions on Linksys VoIP phones? |
| 13:57.45 | [TK]D-Fender | AndrewGearhart: Where are you located again? |
| 13:58.07 | AndrewGearhart | Saint Marys, Pennsylvania |
| 13:58.10 | joe-f | anyone know whats the best format for my mp3's to be in for my MusicOnHold? |
| 13:58.30 | AndrewGearhart | you're not asking so you can drive to my house and kill me for asking stupid questions are you? (/me is sorry if he's annoying you) |
| 13:58.32 | [TK]D-Fender | AndrewGearhart: Seriously... Linksys is not worth it in North America. Polycom or bust.... |
| 13:58.51 | AndrewGearhart | hm. |
| 13:58.52 | joe-f | i converted them with sox (they're in .raw format), and they play on my softphone, but not over my cellphone.. |
| 13:59.22 | [TK]D-Fender | joe-f: just leave them as MP3 or you'll end up double transcoding for SOMEONE. |
| 13:59.34 | [TK]D-Fender | joe-f: and the quality will turn to crap |
| 13:59.48 | joe-f | is the processing for that bad? |
| 14:00.00 | [TK]D-Fender | joe-f: How many channels of MOH at a time? |
| 14:00.20 | [TK]D-Fender | joe-f: I've never heard of any significance to the load. |
| 14:01.31 | errr | is there an eaiser way to get centralized voicemail than just making a dedicated VM server and making either IZX2 or SIP trunk between the phone systems you have and the VM server? |
| 14:01.43 | errr | IAX2* |
| 14:02.37 | joe-f | [TK]D-Fender: ok, well right now the load is nothing, but i need to plan ahead.. ill leave them alone and see what happens |
| 14:04.25 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 14:04.30 | NormanAthol | one more question while i am here i have seen in trixbox manuals that if you dial 777 it simulates an incomming call does anyone know how they do this i do have trixboxrunning on vmware but i cant seem to find where they set it up |
| 14:04.39 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 14:05.29 | joe-f | [TK]D-Fender: that doesnt work, if it's not encoded and i leave it as an .mp3, this is what i get: WARNING[8187]: res_musiconhold.c:232 ast_moh_files_next: Unable to open file '/var/lib/asterisk/moh-native/my-audio-file': No such file or directory |
| 14:05.33 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 14:06.13 | joe-f | what should my musiconhold.conf look like? it's: [default] \n mode=files \n directory=/var/lib/asterisk/moh-native \n random=yes |
| 14:06.20 | [TK]D-Fender | joe-f: Then you might want to have read the big print and installed asteris-addons , as * on its own does not support MP3... |
| 14:06.46 | joe-f | i have format_mp3 installed |
| 14:06.58 | [TK]D-Fender | joe-f: Then something else is wrong. |
| 14:07.14 | [TK]D-Fender | joeKeep in mind your MP# must not be VBR, or have ID3 tags |
| 14:07.22 | joe-f | plus, in that warning message - the file's name is actually 'my-audio-file.mp3', it's obviously not recognizing mp3s.. or something |
| 14:07.40 | joe-f | hmm, ok, i'll look at that |
| 14:07.42 | joe-f | it might |
| 14:07.45 | *** join/#asterisk Mad|Cow (n=madcow@63.96.151.145) |
| 14:08.16 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 14:08.16 | *** mode/#asterisk [+o denon] by ChanServ |
| 14:08.27 | *** join/#asterisk SnrWup (i=Neil@i-83-67-202-134.freedom2surf.net) |
| 14:08.28 | SnrWup | hi |
| 14:08.35 | flujan | hi guys... |
| 14:08.37 | SnrWup | does anyuone know how i can track SIP call transfers? |
| 14:09.00 | flujan | I am using asterisk to have a queue and using x-lite as a softphone. |
| 14:09.21 | flujan | x-lite is freezing a lot of times... The channel remains busy... |
| 14:09.26 | flujan | any ideas? |
| 14:10.43 | *** join/#asterisk Dovid (n=Dovid@barak.cellcom.co.il) |
| 14:10.47 | Mad|Cow | Is it possiable to have my agents automatically login? I am using agent callback login but I want all my agents to allways be apart of the queue (I dont want them to have to remember to login/logout each day) |
| 14:11.03 | Dovid | how would I so an RTP trace from my box to some one else ? |
| 14:11.22 | Dovid | can i use wireshark via ssh ? |
| 14:13.02 | *** join/#asterisk Goodjoke (n=Goodjoke@mail.theenergynetwork.com) |
| 14:17.34 | illsci | you can tcpdump -w to a file and then load the pcap with wireshark |
| 14:18.14 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:18.32 | [TK]D-Fender | flujan: stop using X-Lite :) |
| 14:18.57 | [TK]D-Fender | Mad|Cow: Don't use callback login, make them static members |
| 14:19.05 | illsci | last night I scared myself trying to use Playback to play mp3's |
| 14:19.07 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 14:19.19 | *** join/#asterisk iulius (n=iulius@adsl-33-146-148.asm.bellsouth.net) |
| 14:19.24 | Dovid | illsci: and in wireshark remove everything but the RTP streams (this is for the provider to look at) |
| 14:19.27 | Dovid | nm |
| 14:19.27 | [TK]D-Fender | illsci: Works fine. |
| 14:19.33 | illsci | after I used sox to turn it into a gsm format it sounded like demonic spirts |
| 14:19.35 | Dovid | i can filter in in wireshark by IP |
| 14:19.57 | illsci | well you can do that or do it with tcpdump with you do the packet capture |
| 14:20.09 | *** join/#asterisk rmayorga (i=rmyorg@168.243.89.17) |
| 14:20.13 | *** join/#asterisk joshaidan (n=brianj@thunderbay-voip-4.vianet.ca) |
| 14:20.47 | illsci | [TK]D-Fender: it sounded horrible |
| 14:21.28 | [TK]D-Fender | illsci: Horrble as MP3, and demonic transcoded to GSM? |
| 14:21.36 | illsci | no the mp3 sounds fine |
| 14:21.53 | illsci | http://www.hack3rs.org/~illsci/media/music/ilong2seduceu.mp3 |
| 14:22.01 | flujan | [TK]D-Fender, which softphone do you recommend? |
| 14:22.04 | illsci | i just copied that to /usr/share/asterisk/sounds/ |
| 14:22.19 | illsci | and then sox blah.mp3 blah.gsm |
| 14:22.34 | illsci | and then i had Playback(ilong2seduceu) and it sounded retarded |
| 14:22.47 | [TK]D-Fender | flujan: Buy real phones... soft-phones suck. Idefisk is better than X-Lite though. Has natiove transfer, etc, and has IAX2 for mobile users |
| 14:22.51 | illsci | I tried MP3Player too and that didnt work.... Playback atleast played it |
| 14:23.26 | [TK]D-Fender | illsci: Check for the usual problems, VBR / ID3. |
| 14:23.31 | illsci | [TK]D-Fender: do you have an opinion on real phones? like which ones to get.. |
| 14:23.50 | [TK]D-Fender | illsci: Polycom > all |
| 14:24.01 | illsci | yeah it's probably id3 tags... this was the first time asterisk worked for me... |
| 14:24.13 | illsci | i think my kiax client at home doesn't work because its behind a nat... |
| 14:24.24 | illsci | it registers but iax2 show peers shows it as unreachable |
| 14:24.50 | illsci | so testing stuff from my laptop at home doesn't work out... im not sure how to fix that.. |
| 14:25.45 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 14:29.02 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:29.19 | illsci | what's a way to remove VBR/ID3 tags on linux? |
| 14:29.23 | [TK]D-Fender | blitzrage: I don't want to be at work! |
| 14:29.44 | blitzrage | I just want... |
| 14:30.01 | flujan | [TK]D-Fender, do you recommend some IP hardphone? A cheap one with headset support? |
| 14:30.08 | blitzrage | ! ! ! |
| 14:30.09 | [TK]D-Fender | blitzrage: ! ! ! |
| 14:30.11 | blitzrage | [TK]D-Fender: yer a tease |
| 14:30.13 | blitzrage | yay! |
| 14:30.28 | blitzrage | SPA942 is a good phone for cheap |
| 14:30.28 | [TK]D-Fender | blitzrage: You still work with Sokol, right? |
| 14:30.32 | blitzrage | aye |
| 14:30.35 | mosty | illsci: you have to re-encode to change from vbr to cbr |
| 14:30.59 | [TK]D-Fender | blitzrage: Do you know who's coming down for the May Montreal 3-day training seminar? |
| 14:31.11 | illsci | so re-encode from vbr to cbr and remove id3 tags.. |
| 14:31.14 | blitzrage | [TK]D-Fender: no idea to be honest |
| 14:31.22 | blitzrage | I haven't been asked, so it won't be me |
| 14:31.43 | [TK]D-Fender | blitzrage: Can you poke your head around and get me a roster? :) |
| 14:31.49 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 14:31.54 | blitzrage | [TK]D-Fender: if I remember :) |
| 14:31.55 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-208647.home.otenet.gr) |
| 14:32.02 | blitzrage | [TK]D-Fender: your best bet is to just email Lisa directly |
| 14:32.50 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 14:32.51 | [TK]D-Fender | blitzrage: Is she single, hot, and local? ;) |
| 14:32.52 | mosty | illsci: you can remove id3 tags easily, but to turn vbr into cbr re-encode |
| 14:33.33 | illsci | mosty: do you know a linux util that can do both? or one of each? |
| 14:34.37 | *** join/#asterisk plasmid (n=noway@c-68-46-97-136.hsd1.pa.comcast.net) |
| 14:35.25 | mosty | illsci: use sox to convert to alaw or ulaw, and use that instead (unless you need to save as much disk space as you possibly can) |
| 14:36.02 | illsci | mosty: is that doing somethign different than converting from vbr to cbr and then removing the id3 tags? |
| 14:36.25 | mosty | yes |
| 14:36.44 | illsci | well which one is it? |
| 14:37.02 | plasmid | when running an asterisk system for home use.... is the following ok? an olde 500 mhz Intel Board compu with 384 PC100 RAM and 20 gig HD. |
| 14:37.34 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 14:37.41 | mosty | illsci: it's converting it to an uncompressed, low-fi version that's the same quality as the phone call you are sending it down |
| 14:39.50 | illsci | is there a place i can read about playing mp3's over the phone specifically where this stuff is documented? |
| 14:40.36 | SnrWup | does anyuone know how i can track SIP call transfers? |
| 14:40.47 | [TK]D-Fender | illsci: Only catches are the ones I've listed beyond any standard copyright / broadcast laws in your area |
| 14:41.09 | [TK]D-Fender | plasmid: Fo minimal usage without transcoding, sure |
| 14:41.13 | illsci | ok... i'm still looking for a way to go from vbr to cbr and then remove the tags |
| 14:41.47 | aydiosmio | pretty sure asterisk uses mpg123 no? |
| 14:42.09 | aydiosmio | oh |
| 14:42.16 | aydiosmio | that kind of thing |
| 14:42.22 | plasmid | [TK]D-Fender, minimal usage without transcoding? what if I want to make 3 simultaneous outgoing calls? |
| 14:42.37 | mosty | illsci: there's not much point in playing mp3's, because you have to downsample them anyway |
| 14:42.42 | SnrWup | PSTN Caller calls SIP1, SIP1 transfers PSTN Caller to SIP2, SIP2 Transfers PSTN Caller to SIP3. I need some way of a) recording the entire process from beginning to the time SIP3 finally hangs up and b) a way to log it in a database as a series of records all detailing call times etc |
| 14:43.02 | illsci | mosty: i just want it to work once and not sound horrible and then I'll start doing other things.. |
| 14:43.11 | illsci | im just going through the asterisk book |
| 14:43.21 | illsci | which reminds me... |
| 14:43.30 | uwe | i have cisco 7940 phones, i have two firmwares, P0S3-06-3-00 and P0S3-08-2-00, the version 8 ones, when doing sip show peers, show larger number in ms in the status section, the 6 show less , ie, 200 and 40 ... anyone knows what these numbers represent? its irrelevent to the ping value , and is it better when the number is less, and any idea which firmware works better with asterisk |
| 14:43.36 | SnrWup | so that later on i can see that x number called our office, they spoke to SIP1 then they were transfered to SIP2 then to SIP3 etc, and then i can play a recording of the whole thing to hear what happened |
| 14:43.37 | mosty | illsci: you may as well convert your mp3's to alaw, sox foo.mp3 foo.al ; mv foo.al foo.alaw |
| 14:43.39 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 14:43.47 | illsci | ahhh hehehe |
| 14:43.52 | illsci | tahts why it didn't work |
| 14:43.53 | aydiosmio | SnrWup: I think you can use ForkCDR in your transfer context to do what you want to do |
| 14:43.55 | [TK]D-Fender | plasmid: transcoding is what would kill that ssystem, just pumping packets though you'll be fine |
| 14:43.59 | illsci | i was trying to use alaw and not al |
| 14:44.05 | aydiosmio | ForkCDr should create a new CDR record for each transfer leg |
| 14:44.23 | mosty | illsci: for some reason sox uses a different extension |
| 14:44.25 | illsci | I have a question... In the asterisk book is shows to use like exten => s,1,Answer() |
| 14:44.25 | [TK]D-Fender | uwu : I find no small irony that Cisco's SIP firmware starts with POS ;) |
| 14:44.27 | SnrWup | aydiosmio the SIP phones dont use an asterisk transfer context. they just hit the transfer button, dial SIP2 and then hang up |
| 14:44.31 | illsci | but that doesnt work |
| 14:44.46 | illsci | I had to use exten => _XX.,1,Answer() |
| 14:44.56 | illsci | in the book that wasn't ever mentioned |
| 14:45.07 | [TK]D-Fender | illsci: Dangerously vague pattern matches, yay! |
| 14:45.08 | illsci | it says that the first extension used is s |
| 14:45.15 | *** join/#asterisk corrupt (i=user@128.227.22.108) |
| 14:45.25 | illsci | Well in the book the dangerously vague pattern matches it tells you to use is s |
| 14:45.28 | illsci | and that doesn't work |
| 14:45.31 | mosty | illsci: that's because some calls come in to a particular extension (eg voip) and others dont (eg analogue telephone) |
| 14:45.37 | [TK]D-Fender | illsci: thats because "s" is used almost exclusivly by ANALOG ZAPTEL only. |
| 14:45.46 | aydiosmio | SnrWup: do they even use asterisk for the transfer at all? I'm sure there's a way to stick that fork in there using the extension's context |
| 14:46.03 | aydiosmio | but that's beyond my scope of knowledge |
| 14:46.06 | illsci | So like a softphone |
| 14:46.15 | [TK]D-Fender | illsci: And thats the kind of setup they are describing. I do not necessarily endorse the way that book is layed out, but its the best we've got. |
| 14:46.47 | illsci | so what kind of incomming calls end up in the s extension |
| 14:46.50 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:46.51 | *** join/#asterisk darylvoip (n=darylvoi@c-71-224-42-97.hsd1.pa.comcast.net) |
| 14:47.16 | SnrWup | aydiosmio not that i know of. they just hit the hardkey which says transfer, they then dial the extension, then hang up, and the call is transfered. |
| 14:47.24 | mosty | illsci: calls coming in from an analogue telephone line |
| 14:47.35 | illsci | weird... |
| 14:47.35 | SnrWup | asterisk then seems to initiate some sort of call between SIP1 to SIP3 after SIP2 has hung up |
| 14:47.53 | illsci | I was calling from my cell phone to a number I have with voicepulse.com |
| 14:48.22 | illsci | so I guess I am never going to get an analogue telephone call am i |
| 14:48.24 | corrupt | can Asterisk Gateway Interface scripts be written in python? |
| 14:48.30 | SnrWup | seems like a very simple concept to me, to somehow know when a calls is transfered and get data about the transfer |
| 14:48.35 | SnrWup | but asterisk doesnt seem to be able to do that? |
| 14:48.37 | mosty | illsci: sip servers dont send the extension to sip phones |
| 14:48.46 | illsci | im using iax2 |
| 14:48.52 | illsci | same deal? |
| 14:48.59 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 14:49.03 | mosty | depends how the server is setup |
| 14:49.37 | illsci | well... wtf |
| 14:49.41 | [TK]D-Fender | illsci: sip, PRI, IAX all target extensions, so the dialed number is known. you should read up on your standard extensions.... |
| 14:49.44 | illsci | the whole example in the book is using sip |
| 14:50.03 | *** join/#asterisk csaba (i=HydraIRC@adsl6-203.ptt.yu) |
| 14:50.04 | [TK]D-Fender | corrupt: Yes, virtually any language. |
| 14:50.20 | mosty | [TK]D-Fender: sip doesnt necessarily send target extensions |
| 14:50.20 | uwe | hmmm |
| 14:50.29 | illsci | Well after reading that all calls come to the s extension in the book |
| 14:50.35 | corrupt | cool. |
| 14:50.45 | illsci | i figured that it was going to work... |
| 14:50.53 | csaba | hello, I've just heard about Asterisk... is there a way to call multiple phone numbers at the same time? |
| 14:50.56 | [TK]D-Fender | illsci: They don't. Its a poor generalization. |
| 14:51.02 | mosty | corrupt: you can write AGI programs in any language that can read stdin and write to stdout |
| 14:51.07 | uwe | [TK]D-Fender, what do you mean, sorry, i dont get it ? i got that by telnetting into the phones and doing show config |
| 14:51.25 | [TK]D-Fender | csaba: yes, depends on what you are expecting to have happen exactly. |
| 14:51.57 | [TK]D-Fender | uwe: I was making fun of Cisco. Forget it :) |
| 14:52.08 | csaba | [TK]D-Fender: I expect the phone to ring max 30 seconds, and if someone picks it up the program should read out a message from a wav file... I can do this with Skype, but it can only call 1 phone number at a time |
| 14:52.23 | uwe | oh ... ok |
| 14:53.00 | Uatec_ | hi |
| 14:53.06 | csaba | and I mean a phone number, not another computer |
| 14:53.16 | Uatec_ | i'm trying to make my extension dial a sip device |
| 14:53.25 | Uatec_ | but whenever i try it says WARNING: No channel type registered for ' SIP' |
| 14:53.33 | darylvoip | csaba: Without knowing too much more......yeah. That's not too hard. |
| 14:53.34 | Uatec_ | why is it coming up as ' SIP' not 'SIP'? |
| 14:53.38 | Uatec_ | i reckon that's the reason but i don't know why |
| 14:53.52 | [TK]D-Fender | csaba: Oh you want to annoy people like a telemarketer.... sure you can do that. Go read up on .call files. |
| 14:54.00 | illsci | what other reasons would you ever want to buy channel banks or fxo fxs cards... other than to use existing phones |
| 14:54.14 | illsci | when you can just do everything digitally with regular switches |
| 14:54.20 | darylvoip | You can generate calls on Asterisk with call files. If you write something to generate a call file (one per number you want to call) and make an extension that plays your wav file, you can have Asterisk make the call and connect them to that extension. |
| 14:54.28 | mosty | Uatec_: your extensions.conf probably has a typo |
| 14:54.35 | csaba | well not like a telemarketer, but yeah the idea is the same... is there an extension for Java? |
| 14:56.41 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 14:57.04 | Uatec_ | never mind |
| 14:57.04 | [TK]D-Fender | Uatec_: pastebin your dialplan, and the CLI output of the failed call. |
| 14:57.04 | Uatec_ | i fixed it :) |
| 14:57.04 | [TK]D-Fender | ~pb |
| 15:03.53 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 15:03.53 | Uatec_ | i had a space in it becuase i was tyring to make it look tidy |
| 15:03.53 | csaba | oh, and does Asterisk work under windows? |
| 15:03.53 | darylvoip | csaba: Extnsion for Java to do what? |
| 15:03.54 | Uatec_ | but the space was being included |
| 15:03.54 | Uatec_ | csaba, not |
| 15:03.54 | Uatec_ | -t |
| 15:03.54 | *** join/#asterisk darkmug (n=dennis@143.106.7.170) |
| 15:03.54 | [TK]D-Fender | csaba: this has nothing to do with Java or any other language necessarily. |
| 15:03.54 | csaba | ok, but I guess it's written in c++ |
| 15:03.54 | [TK]D-Fender | csaba: You can also use the AMI Originate command. |
| 15:03.55 | csaba | ok i've downloaded Asterisk so I'll play around a bit thanks for the info :) |
| 15:03.55 | [TK]D-Fender | csaba: NO. a .call file is just a flat text file describing the channel for * to creat & dialout from. |
| 15:03.55 | illsci | even after changing it with sox from mp3 to ulaw you cant even make out the sounds.... |
| 15:03.55 | illsci | it sounds demonic |
| 15:03.55 | illsci | heh |
| 15:03.55 | darylvoip | csaba: basic answer - Yes, it will do that. No, not out of the box. No, you don't want to try to run it on windows, but you could do it with a VMWare image of Asterisk and the VMWare Player app. |
| 15:04.21 | [TK]D-Fender | illsci: get a better audio manipulation tool. |
| 15:04.22 | illsci | what do you use? |
| 15:04.22 | [TK]D-Fender | illsci: Cool Edit 2000 and Audacity |
| 15:04.22 | mosty | illsci: test the ulaw in an audio player first |
| 15:04.24 | aydiosmio | I'm running * in a windows VMware vm, great for testing |
| 15:04.25 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:04.25 | darylvoip | Yeah...that used to be pretty rough for audio quality, but it seems to be pretty acceptable now. |
| 15:04.25 | SnrWup | =[ |
| 15:04.26 | Goodjoke | anyone here got experience with polycom 601s? I need to find out if it is possible to 'intercom' between two 601s |
| 15:04.27 | SwK | goodjoke: define 'intercom'... |
| 15:04.27 | SwK | you mean like call one of them and have it automagically go offhook? |
| 15:04.28 | Goodjoke | so when you dial an extension, it the ext auto picks up on speakerphone |
| 15:04.28 | SwK | yes.. look for polycom paging and Alert_info on the wiki |
| 15:04.28 | pvanstam | goodjoke: no experience with polycom, but I do with both grandstream and snom. They do it |
| 15:04.29 | pvanstam | I guess polycom should be able to it too |
| 15:04.30 | *** join/#asterisk `pariah (n=josh@unaffiliated/pariah) |
| 15:04.31 | [TK]D-Fender | Goodjoke: Yes. Go lookup "polycom auto-answer" on the WIKI |
| 15:04.31 | [TK]D-Fender | ~wikis |
| 15:04.36 | jbot | rumour has it, wikis is http://www.voip-info.org |
| 15:04.36 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-208647.home.otenet.gr) |
| 15:04.36 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 15:04.53 | aydiosmio | we're outsourcing our pastes to canada now |
| 15:04.53 | aydiosmio | I think this is a risk to US National Security |
| 15:05.13 | Mercestes | I always thought .ca meant California |
| 15:05.38 | darylvoip | That would be .ca.us ;) |
| 15:07.08 | aydiosmio | don't get me started on Ottawa, CA |
| 15:07.24 | illsci | dont kid yourself |
| 15:07.29 | illsci | .ca is just northern .us |
| 15:07.36 | aydiosmio | there's one in Cali and on in Canada |
| 15:08.13 | Mad|Cow | [TK]D-Fender: Could you point me in the direction of some documentation about how to create a "static" agent? |
| 15:09.23 | tzanger | ouch |
| 15:09.27 | tzanger | unlimitel lost a DS3 |
| 15:09.32 | [TK]D-Fender | Mad|Cow: this is in the SAMPLE file already... : member => SIP/someguy |
| 15:10.46 | [TK]D-Fender | aydiosmio: You might want to add a P{RIVINCE to that last one :) |
| 15:11.09 | [TK]D-Fender | asd;jklas;ldjdf |
| 15:11.12 | [TK]D-Fender | can't type today... |
| 15:11.18 | [TK]D-Fender | tzanger: OUCH |
| 15:11.32 | tzanger | [TK]D-Fender: come on, practise with me... a a a ; ; ; s s s l l l d d d k k k f f f j j j |
| 15:11.33 | [TK]D-Fender | tzanger: Causing contention issues I take it? |
| 15:11.47 | tzanger | dunno my PRI failover works seamlessly and automatically |
| 15:11.48 | Mad|Cow | [tk]d-fender: ahh... very cool... thanks... didnt realize I could do that |
| 15:12.24 | *** join/#asterisk mquin (n=mike@pdpc/supporter/active/mquin) |
| 15:14.54 | [TK]D-Fender | tzanger: Good, now multiply that by 28, and you'll be able to relate to Unlimitel's plight ;) |
| 15:15.45 | tzanger | [TK]D-Fender: :-) |
| 15:15.49 | tzanger | oh I've lost DS3s before |
| 15:15.55 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 15:16.06 | tzanger | in fact the first DS3 I ever had went into a M13 to 14 AS5248s |
| 15:16.17 | tzanger | the M13 has a hot-fail controller and a hot-fail DS2 card |
| 15:16.32 | tzanger | we lost controller#1, it failed over to controller#2... which we didn't have. |
| 15:17.21 | tzanger | 28 DS1s down |
| 15:17.21 | tzanger | got the second controller shipped up from CAC... UPS lost the fucking package... |
| 15:17.21 | tzanger | Day #3 saw us up and running again |
| 15:17.21 | aydiosmio | lol |
| 15:17.22 | aydiosmio | cac |
| 15:17.22 | tzanger | there was a lot of liquid poop those days |
| 15:17.22 | plasmid | [TK]D-Fender, after upgrading the memory and e-mailing my voip provider's (they assure me that they allow unlimited outgoing/incoming channels on my DID#) I still get that infamous: "Everyone is busy/congested at this time (1:0/0/1)" on my CLI. What should be my first/second step in tracking down the problem? I checked my trunks and I allow upto 20 channels. |
| 15:17.51 | tzanger | plasmid: (1:0/0/1) means 1 call, 0 busy, 0 congested, 1 unavailable. |
| 15:18.45 | [TK]D-Fender | plasmid: that error says nothing, check your SIP debug, look at your dialplan and peer setup. |
| 15:19.01 | plasmid | [TK]D-Fender, checking. |
| 15:20.32 | mosty | and see what tethereal/tshark thinks is happening |
| 15:20.55 | *** join/#asterisk hfb (n=hfb@pool-72-67-156-130.lsanca.dsl-w.verizon.net) |
| 15:21.51 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 15:22.33 | *** join/#asterisk karlhaines (n=karl@209.12.254.71) |
| 15:25.06 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 15:27.14 | *** join/#asterisk allankardec (n=root@189-19-59-138.dsl.telesp.net.br) |
| 15:32.12 | DoDaT69 | who has unlimited incoming/outgoing for DID's? |
| 15:32.34 | DoDaT69 | anyone know of a provider? All I can seem to find is pay for outgoing.. |
| 15:33.05 | nasls_lsa | how do I de-activate some codecs ? for example PCMU ... people can hear me but I can't .. |
| 15:34.20 | illsci | you know what... this ulaw file sucks.. |
| 15:34.45 | mosty | nasls_lsa: sip.conf / iax.conf disallow/allow settings |
| 15:34.57 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:35.48 | plasmid | DoDaT69, vitelity.net and les.net |
| 15:36.10 | nasls_lsa | iax.conf or misdn.conf ? |
| 15:36.48 | mosty | nasls_lsa: you only get to choose codecs on voip channels, i believe |
| 15:37.10 | nasls_lsa | I have problems with one grandstream device, |
| 15:37.20 | nasls_lsa | disallow=pmcu ? |
| 15:37.33 | *** join/#asterisk littleball (n=littleba@bb220-255-154-109.singnet.com.sg) |
| 15:37.40 | *** join/#asterisk zuez (i=steve@66.103.132.86) |
| 15:37.43 | mosty | that's a sip phone right? see the docs on sip.conf in the wiki |
| 15:37.56 | littleball | hello, how to register one asterisk IAX on anther asterisk server? |
| 15:37.58 | nasls_lsa | yes , SIP phone , |
| 15:38.00 | nasls_lsa | ok , thanks |
| 15:38.09 | littleball | because the first box is behind firewall |
| 15:38.31 | zuez | Hi all. technically if I do a 'show sip peers' on asterisk's CLI and see my 7960G registered, shouldn't I be able to dial '2' on my handset to hear a recording on the asterisk server? |
| 15:40.42 | mosty | littleball: you want to register to a firewalled iax server? then setup port forwarding on the firewall |
| 15:41.10 | nasls_lsa | mosty: if I put disallow in iax.conf will be apply to all my phones / lines / connections ? |
| 15:41.21 | *** join/#asterisk gustavoz (n=gustavoz@gentoo/developer/pdpc.active.gustavoz) |
| 15:41.58 | [TK]D-Fender | zuez: Just because a phone is registered doesn't mean it can DIAL anything productive. Go check your DIALPLAN 9extensions.conf) |
| 15:41.58 | mosty | nasls_lsa: that would apply to whichever iax channel section you do it in |
| 15:42.11 | nasls_lsa | aha .. |
| 15:42.40 | zuez | [TK]D-Fender, thanks, I thought that might have some implications. :-) |
| 15:43.00 | illsci | hey if ulaw is a better format for audio over the phone why are all the asterisk sound files in .gsm? |
| 15:43.13 | littleball | mosty, no, my local asterisk is behind firewall, |
| 15:43.27 | littleball | and i want to registered this to remote asterisk server with public ip |
| 15:43.45 | littleball | but just got registration rejected info message |
| 15:43.47 | DoDaT69 | We are currently running some great prices on Linksys VoIP Phones-- http://www.digitalson.com/catalog |
| 15:44.06 | darylvoip | illsci: First, we have to deal with the definition of "better". If bandwidth is no concern, sure....ULAW is "better". |
| 15:44.20 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-152dc2febf517563) |
| 15:44.26 | ChkDigit | Has anyone played with HD Voice yet? |
| 15:44.31 | illsci | darylvoip: I have yet to create a decent ulaw file.... |
| 15:44.53 | illsci | they all sound like garbage |
| 15:46.04 | darylvoip | Well, that sounds more like a procedural/recording issue than anything else. Do your GSM files sound better? |
| 15:46.38 | mosty | littleball: what does the error message say? |
| 15:47.28 | uwe | im sorry, i repeat this question for the 3rd time here, but really i cant fine documentation about what the status field in milliseconds(i suppose) means in the output of the sip show peers cli command is asterisk, does any one know ? |
| 15:48.20 | littleball | mosty, 'iax127topstn148' rejected: 'Registration Refused' from: '20 |
| 15:48.22 | littleball | ... |
| 15:48.31 | *** join/#asterisk docelm0 (n=vircuser@c-76-99-157-112.hsd1.de.comcast.net) |
| 15:48.31 | SwK | uwe: thats how long it took to get an answer to the qualification ping |
| 15:48.50 | [TK]D-Fender | ChkDigit: waste of time & $ |
| 15:48.50 | uwe | thank you very much SwK ! |
| 15:49.05 | SwK | uwe: you'll notice qualify=no makes that go away |
| 15:49.10 | mosty | littleball: username/password wring? |
| 15:49.33 | littleball | yes |
| 15:49.49 | littleball | this is only username, password is omitted |
| 15:49.54 | littleball | of course it is there |
| 15:49.57 | littleball | in the configuration file |
| 15:50.04 | nasls_lsa | well , I have a probem: when I call my PBX ( <- an ISDN number ) from my mobile phone , in my mobile I can listen and speak , but from the VoIP phone that is on the asterisk I can't hear anything :( |
| 15:50.23 | *** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@145-27.mc.cite.net) |
| 15:50.44 | littleball | register => iax127topstn148:pasword@202.XX.XXX.148 |
| 15:51.06 | littleball | mosty, this is in the iax.conf |
| 15:51.07 | ChkDigit | [TK]D-Fender: Yeah? I'm assuming it quadruples the bandwidth, so unless people are saying WOW!!!! It would not be worthwhile. |
| 15:51.07 | AndrewGearhart | any opinions on wireless VoIP phones? I'm specifically thinking about VoIP Wi-Fi not Skype. |
| 15:51.14 | *** join/#asterisk chefrs (n=joe@c-24-8-226-145.hsd1.co.comcast.net) |
| 15:51.23 | mosty | littleball: confirm the username and password with the admin |
| 15:51.33 | [TK]D-Fender | ChkDigit: its only useful within your PBX at best. Waste of time. |
| 15:51.58 | plasmid | AndrewGearhart, Hitachi IP5000A |
| 15:52.05 | ChkDigit | AndrewGearhart: Tried a couple and they mostly suck. |
| 15:52.18 | chefrs | Got a weird T1 issue if anyones happened to work with them in the past. |
| 15:52.19 | littleball | mosty, i am admin :-) |
| 15:52.23 | littleball | i configure myself |
| 15:52.28 | ChkDigit | Presently I'm fighting with the Aastra 480iCT |
| 15:52.37 | mosty | littleball: then what do the verbose logs say on the server? |
| 15:52.52 | AndrewGearhart | plasmid: have you tried it yourself? |
| 15:53.18 | plasmid | AndrewGearhart, I own one. |
| 15:53.46 | plasmid | AndrewGearhart, a tad expensive but you get what you pay for. |
| 15:53.53 | AndrewGearhart | plasmid: what is your biggest complaint and the thing you like the most about it? |
| 15:54.10 | [TK]D-Fender | AndrewGearhart: ... |
| 15:54.13 | [TK]D-Fender | ~wifisip |
| 15:54.24 | jbot | Wi-Fi SIP phones suck. All of them. HARD. Some only slightly less than others... |
| 15:54.56 | AndrewGearhart | lol |
| 15:55.13 | littleball | mosty, the same msg |
| 15:55.29 | codefreeze | it's frightening, how fast jbot learns |
| 15:55.29 | [TK]D-Fender | ChkDigit: I've got a 57i CT on my desk right now. I took it in place of the IP 600 I had. I can't get the wireless to ring on the lines I dedicated to it (#9 in my case), without rining the BASE as well. Total piss-off |
| 15:55.38 | littleball | mosty, is this due to firewall? |
| 15:55.43 | [TK]D-Fender | codefreeze: I teach well ;) |
| 15:55.47 | plasmid | AndrewGearhart, my biggest complain is that the phone only allows WEP/WPA PSK encryption only. And it only allows ONE registration (provision). Other than that, it's smooth in detecting AP's and registering. |
| 15:55.53 | mosty | littleball: the server must say why, set verbose higher |
| 15:55.57 | mosty | turn debug on |
| 15:56.03 | chefrs | [TK]D-Fender: Desolder the speaker in the base? |
| 15:56.19 | littleball | mosty, i know.... -vvvvvv |
| 15:56.19 | littleball | c |
| 15:56.21 | ChkDigit | [TK]D-Fender: Yeah, the 480iCT too. They are really just extensions of the same phone. |
| 15:56.55 | AndrewGearhart | the reason I ask is that we want to be able to give the secretary/receptionist the ability to go and clean/cook/file/make promo materials away from her desk... but still stop the phone from ringing off the hook. |
| 15:57.01 | nasls_lsa | people call me, they hear me but I can't hear them .. |
| 15:57.21 | mosty | nasls_lsa: sounds like a firewall issue |
| 15:57.33 | aptura | morning. Anyone seen a case of the dial logic refering to the 1NXX logic rather then the 18XX logic when dialing out with a 1800 number? My system is doing this now. |
| 15:57.34 | [TK]D-Fender | ChkDigit: It should be the smallest of amtters to disable that... I hope Aastra Wakes up or I may have to ditch this phone... |
| 15:57.42 | [TK]D-Fender | (from MY desk that is...) |
| 15:57.52 | nasls_lsa | it can't be, all devices are in lan , and the line is iSDN , directly on the asterisk .. |
| 15:57.52 | *** join/#asterisk unik-rados (n=rados@c-68-62-71-239.hsd1.mi.comcast.net) |
| 15:58.01 | plasmid | AndrewGearhart, from an office standpoint I reckon the HitachiIP5000A will do the job even with a residential wireless router like D-link. |
| 15:58.26 | unik-rados | how come I can't get music on hold working with Meet Me on asterisk BE? |
| 15:58.31 | [TK]D-Fender | AndrewGearhart: Thats what VOICEMAIL is for :) |
| 15:58.33 | plasmid | AndrewGearhart, originally the Hitachi was designed for office use only. |
| 15:58.36 | mosty | nasls_lsa: internal lan calls dont work? what codecs are your phones using? |
| 15:59.09 | plasmid | [TK]D-Fender, dunno.. some people like their phones to ring so that they know they matter. |
| 15:59.23 | nasls_lsa | at my grandstream screen appears PCMA .. |
| 15:59.40 | AndrewGearhart | re voicemail vs wireless phone... it's more an issue of making our customers feel like they matter |
| 15:59.53 | nasls_lsa | strange , in the same test , after 5 times I heared , without any restart or changing anything .. |
| 16:00.27 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 16:00.30 | [TK]D-Fender | AndrewGearhart: Have it fallback to calling someone else then. Next thing you know your receptionist will be answering calls from the can. You INCONSIDERATE BASTARD! |
| 16:00.39 | [TK]D-Fender | :D |
| 16:00.49 | AndrewGearhart | LOL :-D |
| 16:01.20 | [TK]D-Fender | AndrewGearhart: Why don't you just break out the S&M gear and strap her to the desk (she might like that kind of thing... who knows...) |
| 16:01.34 | nasls_lsa | mosty: it works for internal lines , but doesn't for mISDN calls .. |
| 16:06.47 | mosty | heh, anyone know how to open the pci card latches on a dell 2950? the picture in the manual is microscopic |
| 16:06.49 | nasls_lsa | mosty: and the problem is when I call my asterisk -> grandstream , not when I call from asterisk |
| 16:06.56 | AndrewGearhart | S&M = Service & Marketing? |
| 16:06.56 | AndrewGearhart | ;-) |
| 16:06.58 | nasls_lsa | <PROTECTED> |
| 16:06.59 | AndrewGearhart | I'm going to be hard pressed to get them to accept "auto attendant" |
| 16:06.59 | [TK]D-Fender | AndrewGearhart: Yeah... umm... thats it.... |
| 16:06.59 | AndrewGearhart | can asterisk do the "thank you for holding, there are currently X people ahead of you, you should have called sooner." kind of messages for people on hold? |
| 16:07.00 | nasls_lsa | AndrewGearhart: yes , but I am not sure how .. |
| 16:07.43 | [TK]D-Fender | AndrewGearhart: When I call my head office, I get their receptionist. I often have to call several times in a given day and I feel guilty as hell at having to ask for the same people over and over. Then the actually RING delay thta I get instead of a simple IVR where I can just enter their extension that I know by heart anyways. In these cases, humans slow me don, and drag me down. |
| 16:07.43 | [TK]D-Fender | AndrewGearhart: Yes. Thats called QUEUES |
| 16:08.02 | [TK]D-Fender | AndrewGearhart: And your idea one-ups my scenario by making me wait FOREVER before getting where I want to go! |
| 16:08.23 | ChkDigit | [TK]D-Fender: I've got a customer that dedicates one of their hunt trunk lines (the last) to IVR. |
| 16:08.35 | LeddyHM | when asterisk crashes an email is sent to X@Y.com where is that email address stored? I didn't see anything in the config files |
| 16:08.50 | ChkDigit | They just train their customers/employees family/staff that want immediate IVR to call the number assinged. |
| 16:08.59 | GreyFoxx | LeddyHM: voicemail.conf |
| 16:09.17 | [TK]D-Fender | ChkDigit: a single analog line for that? total waste, and the last caller who might WANT the receptionist won't get him/her. Inconsistant experience. Poor choice... |
| 16:09.37 | ChkDigit | No, when all other lines are busy, the receptionist is too.\ |
| 16:09.46 | ChkDigit | They always have the option of pressing 0. |
| 16:09.48 | [TK]D-Fender | ChkDigit: thats just horrid... |
| 16:09.59 | LeddyHM | this isn't when a voicemail is received, this is when asterisk "crashed" |
| 16:10.03 | ChkDigit | Well, it is what they wanted. |
| 16:10.09 | [TK]D-Fender | ChkDigit: Simple IVR with the ability to hit 0 at any time. the ONLY way to fly. |
| 16:10.19 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 16:10.22 | *** join/#asterisk tuan_modulis (n=chatzill@3-82-252-216-static.enter-net.com) |
| 16:10.44 | [TK]D-Fender | ChkDigit: Some poeple come up with the most hair-brained crap when left to their own devices. thats why idiots should be kept out of system design. |
| 16:10.55 | ChkDigit | Speaking of which, what would cause VoiceMail() to not respond to 0 or *? |
| 16:10.59 | GreyFoxx | LeddyHM: Sorry, I should read the whole message next time. Asterisk doesn't email you, whatever you installed is doing it. Using some prebuilt script or distro ? |
| 16:11.24 | [TK]D-Fender | ChkDigit: lack of "a", "o" and for "o", an improper box setup. |
| 16:11.59 | ChkDigit | So an a extension and o, from the context in which VoiceMail was called is all that is necessary, right? |
| 16:12.24 | zuez | [TK]D-Fender: I was just reviewing the VoIP Hacks text on asterisk, and apparently only config changes to sip.conf were made in order to test asterisk's automated message. |
| 16:12.37 | LeddyHM | grey: that's what I was afraid of, I'll keep digging :) |
| 16:12.43 | [TK]D-Fender | ChkDigit: Go read all this up on the WIKI. "asterisk standard extensions" , "cmd voicemail" |
| 16:13.10 | ChkDigit | Yes done. I'm just still not sure why it is not working. |
| 16:13.12 | [TK]D-Fender | zuez: ...huh? What are you talking about? |
| 16:13.41 | [TK]D-Fender | ChkDigit: Perhaps you could try showing us all the related configs in a pastebin since we're not PSYCHIC. |
| 16:13.42 | [TK]D-Fender | ~pb |
| 16:13.45 | jbot | methinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 16:13.49 | tuan_modulis | hi, I'm looking for a way to allow a calling user to dial a digit during conversation (bridged by Dial) in order to end channel and dial another number (customer service). Is this possible? |
| 16:13.52 | zuez | [TK]D-Fender: Just reading VoIP hacks, it shows an example on how to configure sip.conf and have your phone register with asterisk. That's all that's apparently required to dial '2' and hear asterisk's automated message. |
| 16:14.00 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 16:14.36 | [TK]D-Fender | zuez: I have no idea what documents you are referring to, and there is no "super magic" anything that you don't code yourself |
| 16:14.49 | tuan_modulis | but hmm, ending the channel will simply end the call... |
| 16:14.56 | tzanger | [TK]D-Fender: question for you |
| 16:15.06 | [TK]D-Fender | tzanger: 42 |
| 16:15.14 | [TK]D-Fender | FTW! |
| 16:15.35 | tzanger | asterisk[public ip] -- internet -- NATting firewall -- Asterisk[192.168.2.x] -- ip601 |
| 16:15.42 | tzanger | works fine |
| 16:15.57 | tzanger | the natting firewall has 5060 forwarded to asterisk, and asterisk has nat=yes/canreinvite=no |
| 16:16.01 | tzanger | now |
| 16:16.01 | AndrewGearhart | [TK]D-Fender: trying to say that queues aren't a good idea? ;-) |
| 16:16.08 | Goodjoke | not really an * question... but know of any services that will transcribe recorded calls? |
| 16:16.21 | AndrewGearhart | honestly... we'd probably use it for just one of our people. |
| 16:16.25 | tzanger | ip501[192.168.1.x] -- NATting firewall -- internet -- NATting firewall -- Asterisk[192.168.2.x] - ip601 |
| 16:16.34 | tzanger | no audio at all |
| 16:16.50 | tzanger | the ip501 is registering to the far-end asteirsk box fine |
| 16:16.59 | [TK]D-Fender | AndrewGearhart: I'm saying that if you force all incoming calls to queue into a single receptioninst, I'll hunt you down and gounge your eyes out with a rusty spork! ;) |
| 16:17.02 | tzanger | call progress works (I even hear ringback) |
| 16:17.26 | AndrewGearhart | Goodjoke: there are several websites in a google search that will do transcription services. |
| 16:18.04 | zuez | [TK]D-Fender: just a small excerpt from this text: In its default configuration, Asterisk has an auto-attendant that can route calls. To try it out, take the IP phone off the hook and dial 2. Then dial the BudgeTone's Send button. You will hear a friendly voice saying, "Asterisk is an open source, fully featured PBX and IVR platform…." |
| 16:18.06 | mog | that is not a very good joke AndyCap |
| 16:18.11 | mog | er AndrewGearhart |
| 16:18.16 | [TK]D-Fender | tzanger: canreinvite=no, nat=yes, make sure your localnet & externip are right. thats all.... |
| 16:18.31 | AndrewGearhart | mog: lol |
| 16:18.33 | Goodjoke | AndrewGearhart: ok..the question should have said... are there any services that anyone would recommend |
| 16:18.35 | tzanger | [TK]D-Fender: canreinvite=no, nat=yes... ahah I bet localnet is wrong, I saw it was odd |
| 16:19.03 | *** join/#asterisk tbic (n=tbic@protious.fciautomation.com) |
| 16:19.04 | [TK]D-Fender | zuez: You sound like you have clearly NOT configured * at all... go make a dialplan. |
| 16:19.29 | zuez | Uh, I'm just going based off of what I read in a published book. |
| 16:19.43 | AndrewGearhart | [TK]D-Fender: we have one person that has been doing technical support for about 5 years... and most of our clients know her by name... and would select her from a menu of employees if given the chance... |
| 16:19.48 | tzanger | [TK]D-Fender: fixed the localnet with no work |
| 16:19.50 | tzanger | er no success |
| 16:19.52 | tzanger | damn |
| 16:19.55 | AndrewGearhart | [TK]D-Fender: so I'd put a queue up just for her |
| 16:20.11 | *** join/#asterisk lude (i=dolsen@smoke.isprime.com) |
| 16:20.21 | [TK]D-Fender | zuez: time to actually CONFIGURE *. Go download and read THE BOOK. Whatever guide you're looking at seems largely useless, and you/it are making large and poor assumptions about *. |
| 16:20.23 | [TK]D-Fender | ~!book |
| 16:20.32 | GreyFoxx | AndrewGearhart: That sounds like a crappy deal for her :) |
| 16:21.10 | GreyFoxx | Just because users know my name, doesn't mean I have to be the one to help them and end up with a lineup of users waiting for me while my coworkers twiddle their thumbs :) |
| 16:21.12 | AndrewGearhart | [TK]D-Fender: You are 18th in line to speak to "Jane". If you would like to speak with a different person, press 1 and you will be directed to an employee with a shorter wait time, or press 2 to go to "Jane's" voicemail |
| 16:21.41 | AndrewGearhart | GreyFoxx: that's why I would setup the queue... because largely... the clients just don't know that there are other people that can help |
| 16:21.50 | tzanger | [TK]D-Fender: every time I see FTW I read it as "fuck the web" |
| 16:21.57 | [TK]D-Fender | AndrewGearhart: Thats WORSE than an IVR, and is one in its own perverse way. |
| 16:22.38 | [TK]D-Fender | tzanger: does the remote phone Echo properly? |
| 16:22.42 | GreyFoxx | We use a few queues here, thankfully users can't call me directory otherwise I'd be the only one they call for and be quite cranky :) |
| 16:22.52 | tzanger | AndrewGearhart: You are 18th in line to speak to Jane. SHe's a chatty lass, and as such we've determined that you would be waiting at least another 3 hours... |
| 16:23.01 | tzanger | [TK]D-Fender: polycom phones have an internal echotest? |
| 16:23.10 | [TK]D-Fender | tzanger: "show application echo" :) |
| 16:23.13 | AndrewGearhart | [TK]D-Fender: it goes right along with the problem of the dialog box that never gets read by users. "are you sure you want to delete that data?" and the user automatically clicks ok withotu reading |
| 16:23.22 | tzanger | [TK]D-Fender: ohhhhhhh, I thought you meant the polycom |
| 16:23.23 | [TK]D-Fender | tzanger: make sure the 501 is clear to *, forget the 601 from your test |
| 16:23.26 | tzanger | I am trying that now |
| 16:23.31 | tzanger | yeah I hear ya |
| 16:23.53 | AndrewGearhart | [TK]D-Fender: our recep. will tell them that she has a backlog of 10 calls, but they don't listen. |
| 16:24.38 | mosty | poor jane |
| 16:24.50 | mosty | if i were her i'd ask for a big raise |
| 16:24.50 | AndrewGearhart | so, right now they take a message, print out a "telephone call follow-up" ticket and it gets shuffled around the office to somebody that has time to answer the calls |
| 16:24.57 | [TK]D-Fender | AndrewGearhart: Something is terribly wrong with the way you represent yourself to your callers.... you are engendering, fostering, heck even BREWING ignorace. You should open a bottling company to reduce production costs... |
| 16:25.26 | tzanger | heh |
| 16:25.30 | AndrewGearhart | [TK]D-Fender: lol |
| 16:25.33 | tzanger | echo() needs a parameter to introduce delay |
| 16:25.48 | AndrewGearhart | [TK]D-Fender: the idea is to pander to their ignorance while making an end-run around it |
| 16:26.01 | [TK]D-Fender | AndrewGearhart: And you turn he into a post-it-note delivery machine, soon to be taking calls from the can? My respect for you is falling faster than the anvil follow Wile-E-Coyote off a cliff... |
| 16:26.18 | [TK]D-Fender | *poof* |
| 16:26.23 | tzanger | [TK]D-Fender: nope, echo and milliwatt both give no audio so it's that stupid firewall |
| 16:26.29 | [TK]D-Fender | (inster mushroom dust could here) |
| 16:26.46 | [TK]D-Fender | tzanger: Not a PIX / D-Link involved I hope |
| 16:27.01 | mosty | AndrewGearhart: you should designate this person as top-level support, and charge a massive premium to talk to them |
| 16:27.12 | tzanger | [TK]D-Fender: linksys, which is bad enough |
| 16:27.13 | AndrewGearhart | hehe... yeah... the secretary does the paper delivery... to make it worse... "Jane" works upstairs.... so the secretary runs up the stairs for each message. |
| 16:27.19 | tzanger | [TK]D-Fender: I think I've convinced him to get rid of it |
| 16:27.40 | [TK]D-Fender | tzanger: Linksys is usually painless.... |
| 16:27.50 | [TK]D-Fender | tzanger: I suspect something else is amiss.... |
| 16:27.50 | tzanger | [TK]D-Fender: yes, but not when asterisk is behind it |
| 16:27.58 | [TK]D-Fender | tzanger: pastebin what you can. |
| 16:28.14 | tzanger | [TK]D-Fender: I have zero issues with [slew of polycoms] - linksys - adsl modem - internet - asterisk_with_public_ip |
| 16:28.28 | [TK]D-Fender | tzanger: and do you have any other function remote phones on that box? |
| 16:28.35 | tzanger | but this is polycom - linuxNat - internet - linksys - asterisk_with_rfc1918_ip |
| 16:29.21 | tzanger | [TK]D-Fender: nope can't get any of the others to work, and that's through a mixture of linuxnat, linksysnat, maybe even a direct connection, not sure about the last one |
| 16:29.59 | thevoke | anyone here an idea why io::socket doesnt work from an perl agi ? |
| 16:31.34 | chefrs | Got a weird PRI T1 issue and caller ID if anyones happened to work with them in the past. |
| 16:31.53 | [TK]D-Fender | tzanger: I'd now suspect your* server side at fault. Silly thought... you DID forward DIP+RTP, right? |
| 16:32.00 | Mercestes | chefrs: Did you just ask for help from anyone whose worked with a PRI with CallerID in the past? |
| 16:32.19 | [TK]D-Fender | Mercestes: You mean like ALL OF US?! ;) |
| 16:32.38 | Mercestes | [TK]D-Fender, Yea, really. |
| 16:32.42 | tzanger | udp5060 is forwarded for sure, but I can't verify if a udp port range for rtp was forwarded at the moment |
| 16:32.46 | chefrs | Well, if I send just the #, it works. If I try to send the name + the #, it shows "Hidden" or "Private" on other lines. |
| 16:33.06 | [TK]D-Fender | tzanger: You know just answering that... I should smack you :) |
| 16:33.08 | Mercestes | chefrs, are you sure your provider supports CIDName transmission? |
| 16:33.09 | tzanger | chefrs: I have PRI and set callerid without issue. whos' the telco? |
| 16:33.39 | tzanger | [TK]D-Fender: no... as I said I have absolutely no problem the other way 'round (slew of phones going through one consumer router) -- no forwarding, no nothing |
| 16:33.48 | tzanger | however asterisk-behind-nat is always a pain in the ass |
| 16:33.48 | chefrs | Mercestes: Had them watch our line the other day, he said he got everything fine. He said it looked like I was "Sending along a flag that said 'I'm providing CID info but not the name' and when it provided the name the Telco apparently goes "Wha?" and drops the whole thing. |
| 16:33.52 | tzanger | phones-behind-nat is a non-issue by and large |
| 16:33.55 | mosty | thevoke: no, but why dont you print the error messages to a file you can read afterwards to find out? |
| 16:34.10 | [TK]D-Fender | tzanger: I'm talking * side, not 501 side. 501 side should NOT have forwarding, and just qualify=yes to keep it open. |
| 16:34.22 | tzanger | [TK]D-Fender: right... |
| 16:34.32 | tzanger | *-side nat, reinvite, externip and localnet are all set right |
| 16:34.42 | [TK]D-Fender | tzanger: so remote should not forward anything at all. * side = 5060 + rtp |
| 16:35.11 | tzanger | *nods* I think I've convinced him to get rid of the linksys and use a linux router, which I can forward ports or use conntrack_sip |
| 16:36.31 | *** join/#asterisk agile (n=mike@63.98.55.146) |
| 16:36.40 | [TK]D-Fender | tzanger: You have a linksys in front of your *? |
| 16:36.53 | tzanger | [TK]D-Fender: this particular guy does, yes |
| 16:36.56 | Mercestes | chefrs: Could you maybe give us the original terminology? |
| 16:36.59 | tzanger | I stick my * right on the public IP |
| 16:37.04 | tzanger | it's also the router/firewall :-) |
| 16:37.13 | Mercestes | chefrs: We don't understand "layman." |
| 16:37.19 | [TK]D-Fender | tzanger: Dittot for me at home. On M S518 to boot ;) |
| 16:37.25 | chefrs | Mercestes: Yeah well I can only give you what the tech gave me. |
| 16:37.29 | [TK]D-Fender | mt* |
| 16:37.29 | chefrs | Mercestes: Unfortunately. |
| 16:37.34 | tzanger | [TK]D-Fender: wow we have identical systems :-) |
| 16:37.52 | Mercestes | chefrs: Yea, but we don't have a "flag" that says "we are sending CID info, but not the name" with an error message that goes "wha?" |
| 16:38.17 | Mercestes | I'd have to google '"caller ID without the name" flags' just to try and get the tech-speak for that. |
| 16:38.21 | chefrs | The Telco apparently gets a flag in my setup that says "I'm providing CID, but not the name" |
| 16:40.19 | [TK]D-Fender | tzanger: We share the sme first name, and in the same country! (insert creepy Rod Serling music here) |
| 16:40.20 | Mercestes | I wasn't aware there *was* a flag that said "We are sending CID info without the name." |
| 16:40.20 | chefrs | Then my system sends along the name, Telco sees the name and the # and apparently gets "confused" |
| 16:40.20 | tzanger | hahahahha |
| 16:40.20 | aptura | tzanger, I was thinking of installing the firewall on my ast box but was a bit leary of that. Is your fw hand written or some third part app installed? |
| 16:40.20 | tzanger | [TK]D-Fender: next thing you know we'll discover that we're just talking to ourselves and that we're actually quite insane |
| 16:40.20 | tzanger | aptura: handwritten |
| 16:40.21 | tzanger | it's not much of one to be honest |
| 16:40.21 | aptura | k |
| 16:40.21 | tzanger | I just do SNAT, have a default ALLOW forward policy and drop anything coming in ppp0 that is RFC1918 |
| 16:40.21 | aptura | Its a good way to cut down on the power consumption. |
| 16:40.21 | tzanger | it won't prevent a real attack but honestly I am not concerned about that at this time |
| 16:40.21 | tzanger | I could use a default DROP policy and use port knocking and a dozen other layers but it's just too much effort for me at this time |
| 16:40.21 | aptura | yea :) |
| 16:40.31 | aptura | for new installs with no fw this is probebly a good way to go. |
| 16:42.28 | [TK]D-Fender | tzanger: yeah, knock off private IP spoofs on ext if's + SYN attack covers most of it. Disabling undeeded service most of the rest. |
| 16:42.46 | *** part/#asterisk infernix (i=nix@unaffiliated/infernix) |
| 16:42.54 | [TK]D-Fender | tzanger: I still need to make some port-firwarding lines to my primary PC.... |
| 16:43.03 | tzanger | [TK]D-Fender: precisely. I have 5060, 22... and that's it. |
| 16:43.07 | tzanger | oh yeah, 5900 for vnc |
| 16:43.15 | _VoiceMeUp_COM | vnxc has a hack |
| 16:43.17 | aptura | btw my ast box is acting a little odd this morning. I think its possibly a bug but dont know. Dial pattern matching is a little crossed. Dialing a local 1800 number is going out my sip provider in the states. It does show up as a 1800 number with 9 being stripped off. |
| 16:43.35 | aptura | it should be going out my zap. |
| 16:43.47 | [TK]D-Fender | tzanger: thats what I want to forward. VNC to my internal system so clients can call-out and give control to me at home. |
| 16:44.21 | tzanger | [TK]D-Fender: "call out and give control to me at home" ? |
| 16:44.32 | tzanger | ... mine just forwards to my windows PC so I can control emule from work :-) |
| 16:44.44 | Qwell[] | nx > vnc |
| 16:45.03 | tzanger | yeah it's not bad |
| 16:45.05 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-208647.home.otenet.gr) |
| 16:47.15 | *** join/#asterisk saravia (n=jovannot@lan8.att.net.co) |
| 16:47.17 | chefrs | Hmm. My dialparties.agi isn't saying returned with no extensions to call... |
| 16:47.33 | chefrs | err, "is saying" returned with no extensions to call |
| 16:49.21 | hmm-home | summer is upon us |
| 16:50.38 | Uatec_ | hey |
| 16:50.42 | Uatec_ | i've got multiple lines on my phone |
| 16:50.54 | Uatec_ | how can i make line 2 ring when i'm already busy? |
| 16:50.59 | Uatec_ | SIP this is |
| 16:51.25 | Hmmhesays | depends on how your phone handles incoming calls when line 1 is busy |
| 16:51.41 | Uatec_ | oh |
| 16:51.42 | [TK]D-Fender | tzanger: You can have a VNC Server call out to a listening view you know... |
| 16:51.50 | Uatec_ | would i have to set it up in my phone settings then? |
| 16:51.54 | [TK]D-Fender | chefrs: .... |
| 16:51.55 | [TK]D-Fender | ~freepbx |
| 16:52.07 | jbot | from memory, freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 16:52.07 | hmm-home | yeah that is a life saver sometimes |
| 16:52.20 | [TK]D-Fender | Uatec_: Depends on your phone and how you set it up (duh) |
| 16:52.20 | *** join/#asterisk hrmphh (i=patrick@notchill.com) |
| 16:52.23 | tzanger | [TK]D-Fender: ah yes |
| 16:52.26 | tzanger | I never use it |
| 16:52.27 | Uatec_ | O RLY?!?!? |
| 16:52.28 | hrmphh | anyone know where i can find "tone on hold" wavs/mp3s? |
| 16:52.48 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 16:52.53 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-208647.home.otenet.gr) |
| 16:53.02 | [TK]D-Fender | tzanger: I do all the time. Its a real boon so you don't have to walk every idiot through callit it up. Single incon to hand over control, no questions asked. |
| 16:53.14 | [TK]D-Fender | Uatec_: INDEED! ;) |
| 16:53.28 | [TK]D-Fender | Uatec_: So yeah.. some DEATIALS might be nice ;) |
| 16:56.08 | aptura | to bad there wanst a asterisk sound called trunkld down |
| 16:56.44 | tzanger | aptura: make one |
| 16:56.55 | tzanger | aptura: I just play we're having trouble, please call again later |
| 16:57.02 | *** join/#asterisk rrocha (n=ruyrocha@201.22.48.149.adsl.gvt.net.br) |
| 16:58.23 | aptura | hahah |
| 16:58.46 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:58.51 | aptura | I dont think somone would like that and loose confidence in the system. |
| 16:59.25 | aptura | I should get my voipjet account going again and just use that if vitelity is down. |
| 17:00.07 | aptura | tzanger, what is the reason to include 9 in the dial plan anyway if ast does not really need it? |
| 17:00.22 | tzanger | ... I dn't know why, I wouldn't |
| 17:00.31 | [TK]D-Fender | aptura: You don't need it. Never did. |
| 17:02.09 | hrmphh | is there a "tone on hold" avail for asterisk? |
| 17:02.09 | errr | When I call my DID from the outside and I am put itno voicemail and I press * it prompts me for my password. What is the setting for that called? |
| 17:02.12 | hrmphh | or do i have to create it myself? |
| 17:02.20 | aptura | Mabey its for those clients who need it becasuse thay keep forgetting :) |
| 17:02.43 | mosty | errr:setting for what? |
| 17:03.04 | errr | mosty: so I can enable/disable it |
| 17:04.06 | mosty | errr: enable what? |
| 17:04.27 | errr | mosty: what part of my question didnt you understand.. Ill try to eplain it better |
| 17:04.41 | [TK]D-Fender | errr: "show application voicemail" |
| 17:05.13 | errr | [TK]D-Fender: Ill read that, thanks |
| 17:06.47 | mosty | errr: are you trying to setup voicemail accounts, or direct callers to retrieve voicemail for a specific account, or something else? |
| 17:06.48 | aptura | Anyway goto go. |
| 17:07.10 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 17:08.05 | puzzled | hi |
| 17:10.55 | *** join/#asterisk zim (n=chatzill@zimonline.demon.co.uk) |
| 17:12.41 | LeddyHM | you guys have any links on accessing the astdb for scripting purposes? |
| 17:13.42 | mosty | LeddyHM: what kinds of things are you trying to do? |
| 17:13.42 | *** join/#asterisk falco_toadfoot (i=papir@160.80-202-240.nextgentel.com) |
| 17:13.55 | LeddyHM | pull out info |
| 17:14.23 | LeddyHM | want to grab cols where data has been set |
| 17:15.03 | LeddyHM | isn't it just a mysql db? |
| 17:15.07 | mosty | no |
| 17:15.09 | errr | mosty: Im trying to make it so when someone calls in from outside the office to check their voicemail and they call their extension instead of the global VoiceMailMain() extension I have setup they are able to press * and be prompted for their voicemail password |
| 17:15.18 | *** join/#asterisk Meaty (n=meaty3@office.abi.ca) |
| 17:15.36 | errr | it seems I dont have my a extension setup properly though |
| 17:15.40 | LeddyHM | hmm found res_odbc.conf |
| 17:15.42 | mosty | LeddyHM: use asterisk realtime if you want to use mysql or postgresql, but astdb is a bdb |
| 17:16.08 | LeddyHM | has info in it on mysql |
| 17:23.11 | *** join/#asterisk X-TAR (i=SiLENT@81-178-73-39.dsl.pipex.com) |
| 17:23.26 | errr | [TK]D-Fender: how do I make it so the user will be prompted for their password only instead of being asked for their mailbox number as well.. |
| 17:23.32 | X-TAR | is this about the telephone system ? |
| 17:24.07 | errr | basicly how do I know what mailbox they are hitting * from |
| 17:24.12 | mosty | errr: that's an option of the VoicemailMain dialplan command |
| 17:24.34 | errr | mosty: right but I dont know how to know what mailbox the person is hitting * from |
| 17:25.00 | [TK]D-Fender | errr: "show application voicemail" <------------ |
| 17:25.01 | errr | I figured it might be some globla var like ARG1 or something |
| 17:25.17 | [TK]D-Fender | errr: No. READ THE INSTRUCTIONS. |
| 17:25.29 | errr | [TK]D-Fender: yeah I read that, thats how I got it to the point of getting me in the mail voicemail |
| 17:25.39 | mosty | errr: use a channel variable, possibly set for each sip client, equal to their voicemail account, then use that |
| 17:27.00 | errr | ah ok thanks |
| 17:28.39 | *** part/#asterisk chefrs (n=joe@c-24-8-226-145.hsd1.co.comcast.net) |
| 17:31.06 | *** join/#asterisk Bazy (n=bazy@89.137.178.124) |
| 17:33.36 | Uatec_ | i hate aastras |
| 17:35.02 | _VoiceMeUp_COM | err. VoicemailMail(MAILBOXNUMBER) |
| 17:35.36 | *** part/#asterisk X-TAR (i=SiLENT@81-178-73-39.dsl.pipex.com) |
| 17:37.07 | *** join/#asterisk Arrick (n=Arrick@about/windows/regular/arrick) |
| 17:37.08 | Arrick | hi all |
| 17:37.18 | Arrick | is there a version of asterisk for windows XP yet? |
| 17:38.01 | _VoiceMeUp_COM | winast |
| 17:38.07 | gambolputty | don't know, but * runs best on linux |
| 17:38.18 | _VoiceMeUp_COM | ~google winast |
| 17:41.30 | Arrick | what is the best and easiest distro to setup asterisk on? |
| 17:41.49 | _VoiceMeUp_COM | some say debian , some ubuntu others centos |
| 17:42.01 | _VoiceMeUp_COM | i prefer centos since its Red hat enterprise trimmed |
| 17:42.11 | _VoiceMeUp_COM | dpeends on what you want to install afterwords |
| 17:42.19 | _VoiceMeUp_COM | centos is limited on ports |
| 17:42.31 | _VoiceMeUp_COM | but stable as it can be |
| 17:42.31 | Arrick | well, I am not going to run anything else on it |
| 17:42.35 | Arrick | just asterisk |
| 17:42.47 | _VoiceMeUp_COM | id suggest cent but hat my opinion |
| 17:43.08 | Arrick | ok |
| 17:43.12 | Arrick | I will look at it |
| 17:43.17 | _VoiceMeUp_COM | np |
| 17:43.27 | Arrick | whatever I do, I am installing in a VM, so it is easily undone lol |
| 17:44.54 | errr | where do I find docs on using strip in my dialplan? |
| 17:45.24 | LeddyHM | centos isn't a "trimmed" RHEL |
| 17:46.52 | *** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
| 17:49.49 | Arrick | lol |
| 17:50.30 | *** join/#asterisk hijacked (i=oC9o@cerebus.clandestineresearch.com) |
| 17:54.23 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:58.06 | falco_toadfoot | anyone have any idea why my dundi lookups return 0 results every time? |
| 17:58.29 | falco_toadfoot | I have 2 hosts running asterisk 1.2 |
| 17:58.42 | falco_toadfoot | both are set up as symmetric dundi peers |
| 17:58.49 | falco_toadfoot | keys are generated |
| 17:59.14 | falco_toadfoot | host A has 1111 in dialplan, host B has 2222 in dialplan |
| 17:59.27 | Arrick | ok, i found the windows version on http://www.asteriskwin32.com/ |
| 17:59.28 | falco_toadfoot | and a switch=>DUNDi/search |
| 17:59.34 | Arrick | anyone know how to set this up? |
| 17:59.34 | karlhaines | anyone in here use les.net services? |
| 18:01.07 | *** join/#asterisk ManOfMilk (n=CpnPlnet@70-56-26-12.eugn.qwest.net) |
| 18:01.15 | *** join/#asterisk chefrs (n=joe@c-24-8-226-145.hsd1.co.comcast.net) |
| 18:01.33 | chefrs | Any idea why my channels on my PRI reset periodically? |
| 18:07.36 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:07.43 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:08.14 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-62f7039a6d80159c) |
| 18:15.20 | *** join/#asterisk s0lid (n=jlq@202.124.153.100) |
| 18:15.32 | *** part/#asterisk chefrs (n=joe@c-24-8-226-145.hsd1.co.comcast.net) |
| 18:16.02 | *** join/#asterisk paavum (n=Dorphals@200.71.58.39) |
| 18:16.53 | paavum | hello |
| 18:17.08 | naitram | anyway to force a hangup of caller in dial plan |
| 18:18.41 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 18:18.54 | *** part/#asterisk naitram (n=ttech@216.77.58.40) |
| 18:19.02 | paavum | I have asterisk 1.4.1 and zaptel 1.4.0 connected to two E1s by a 1st Gen TE410P |
| 18:19.11 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 18:19.13 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 18:19.45 | paavum | however I get two strange behaviours |
| 18:19.53 | paavum | first, calls get randomly hung up |
| 18:20.20 | falco_toadfoot | does anyone know how to get dundi to work? |
| 18:20.38 | paavum | and second when asterisk starts, I get this message Cant call Zap/XXX |
| 18:23.04 | *** join/#asterisk dfgas (n=dfgas@adsl-75-44-39-127.dsl.milwwi.sbcglobal.net) |
| 18:23.06 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 18:23.06 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:23.24 | *** join/#asterisk joe (n=nnnnjsau@ip66-107-33-195.z33-107-66.customer.algx.net) |
| 18:23.37 | paavum | why does this happen? |
| 18:23.47 | paavum | and how can I prevent this from happening? |
| 18:29.44 | paavum | Looking at my /varlog/asterisk/full I can only see my sips trying to register |
| 18:29.49 | paavum | but no analog errors or anything |
| 18:31.39 | *** join/#asterisk c4t3l (n=c4t3l@cpe-72-181-205-77.houston.res.rr.com) |
| 18:31.40 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 18:32.48 | c4t3l | greetings. Has anyone noticed a polycom bug in the last couple of days |
| 18:33.00 | c4t3l | or a windoze worm for that matter. |
| 18:33.17 | c4t3l | related to registration |
| 18:34.02 | c4t3l | oh well, geuss not |
| 18:34.20 | mcab | c4t3l: I'd wait a little longer for a response than that :-) |
| 18:34.29 | darylvoip | Sorry....I'm masochistic enough to be a Cisco guy. |
| 18:34.42 | mcab | c4t3l: but, I can't say *I've* heard of an issue, but that's just me... |
| 18:36.59 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 18:38.43 | AndrewGearhart | mcab: what are you talking about? 1.5 minutes is more than enough time to expect, no, demand a response in IRC! |
| 18:38.52 | AndrewGearhart | mcab: right? |
| 18:38.59 | AndrewGearhart | mcab: aren't you going to say something? |
| 18:39.02 | AndrewGearhart | ;-) |
| 18:39.05 | mcab | AndrewGearhart: of course! how silly of me :-) |
| 18:39.33 | AndrewGearhart | yep... still fits |
| 18:41.38 | paavum | hello I have asterisk 1.4.1 and zaptel 1.4.0 connected to two E1s by a 1st Gen TE410P however I get two strange behaviours first, calls get randomly hung up and second when asterisk starts, I get this message Cant call Zap/XXX why does this happen? and how can I prevent this from happening? Looking at my /varlog/asterisk/full I can only see my sips trying to register |
| 18:41.54 | paavum | can anybody gimme a hand |
| 18:41.56 | paavum | ? |
| 18:43.36 | *** join/#asterisk Meaty` (n=meaty3@office.abi.ca) |
| 18:44.22 | *** join/#asterisk bmg505 (n=leon@196.209.176.127) |
| 18:45.30 | zuez | the init script shipped with 1.4.2 run safe_asterisk, however as a privileged user |
| 18:47.09 | tzafrir | edit it not to run safe_asterisk |
| 18:47.13 | tzafrir | problem solved |
| 18:47.35 | zuez | I suppose I can write my own init script |
| 18:49.00 | *** join/#asterisk khronos (n=khronos@c-76-110-134-230.hsd1.fl.comcast.net) |
| 18:53.51 | *** join/#asterisk froguz (n=alvaro@pc-69-217-46-190.cm.vtr.net) |
| 18:55.10 | drfreeze | Is there a way to hang up a line from 'asterisk -r'? |
| 18:55.24 | Qwell[] | drfreeze: soft hangup |
| 18:55.59 | *** join/#asterisk BSD_Tech (n=bsdtech@ppp-69-239-112-47.dsl.irvnca.pacbell.net) |
| 18:57.18 | paavum | Qwell ... dude can you give me a hand? I have a prob with asterisk. I give you the login pass info. you look at my system and then you charge me :P |
| 18:57.39 | paavum | I have asterisk 1.4.1 and zaptel 1.4.0 connected to two E1s by a 1st Gen TE410P however I get two strange behaviours first, calls get randomly hung up and second when asterisk starts, I get this message Cant call Zap/XXX why does this happen? and how can I prevent this from happening? Looking at my /varlog/asterisk/full I can only see my sips trying to register |
| 18:57.45 | paavum | I keeo getting these msgs |
| 18:57.57 | paavum | Warning: zthook failed device or resource busy |
| 19:00.22 | hrmphh | question |
| 19:00.26 | hrmphh | if i have two files in my moh dir |
| 19:00.33 | hrmphh | and im using the default setup for musiconhold.conf |
| 19:00.41 | hrmphh | i.e. mode=files |
| 19:00.52 | hrmphh | itll just play those two files over and over in the same order right? |
| 19:01.09 | froguz | i have configured my PAP2 to log into the office's Asterisk. i'm pretty sure i've set port 5060 on PAP, however "sip show peers" shows me port 63001. Why? |
| 19:05.19 | froguz | my PAP2 is at home, behind nat. |
| 19:05.28 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 19:08.06 | *** join/#asterisk pvanstam (n=Pim@dsl-083-247-093-018.solcon.nl) |
| 19:10.27 | *** join/#asterisk Bobocop (n=Bobocop@public-gprs33843.centertel.pl) |
| 19:13.16 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 19:14.23 | *** join/#asterisk bmg505 (n=leon@196.209.176.127) |
| 19:14.51 | *** join/#asterisk flewid (n=flewid@216.145.22.66) |
| 19:14.55 | flewid | hello |
| 19:15.04 | flewid | anyone here using a cisco 7970 with asterisk 1.2.x ? |
| 19:16.42 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 19:17.17 | iCEBrkr | I got a Ciso 7940, but it's not hooked to Asterisk :) |
| 19:18.14 | _VoiceMeUp_COM | froguz |
| 19:18.24 | _VoiceMeUp_COM | froguz its because router maps that port to the phone |
| 19:18.43 | flewid | ice: aha |
| 19:19.05 | froguz | _VoiceMeUp_COM, you mean, the router at home (PAP2)? |
| 19:19.13 | flewid | my issue is that i can get it to register to 1.4.x, but if i try and register to a 1.2.x server (that's not behind nat, but the phone is) it won't work |
| 19:19.17 | flewid | i get sip 401 unauth errors |
| 19:21.20 | _VoiceMeUp_COM | yes |
| 19:21.46 | _VoiceMeUp_COM | if a pa2 try setting ext pot to 5060 |
| 19:21.47 | _VoiceMeUp_COM | port |
| 19:22.35 | _VoiceMeUp_COM | to try to force it but maybe not |
| 19:22.35 | _VoiceMeUp_COM | dlink? |
| 19:23.33 | froguz | linksys PAP2 ATA and dlink router |
| 19:23.48 | _VoiceMeUp_COM | hehe dlink does that |
| 19:25.08 | *** join/#asterisk s0lid (n=jlq@210.213.241.246) |
| 19:25.26 | froguz | i can't access my router now, but i'll try to do something to avoid this when i get home |
| 19:25.32 | froguz | thanks _VoiceMeUp_COM |
| 19:27.56 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 19:32.19 | hmm-home | does anyone use perl to connect to the asterisk manager interface? |
| 19:32.43 | *** join/#asterisk naitram (n=ttech@216.77.58.40) |
| 19:33.10 | naitram | is there a way for me to send a sip bye message from asterisk to a specific device |
| 19:33.15 | hmm-home | I can't figure out why I can't get this to log in |
| 19:34.50 | *** join/#asterisk Ravi1974 (n=I@static-70-19-119-112.ny325.east.verizon.net) |
| 19:35.06 | Ravi1974 | Hi, I was looking to buy the LINKSYS SRW224P. A 24 port POE switch. I understand it is very loud. Apart from this does anyone have any comments or other brand/model recommendations |
| 19:38.14 | Mercestes | c4t3l, windows *is* a worm, and what polycom registration bug? |
| 19:43.18 | *** join/#asterisk Bobocop_ (n=Bobocop@public-gprs24483.centertel.pl) |
| 19:44.54 | c4t3l | Mercrestes: EVERY single customer is having issues today with pcoms registering and un-registering |
| 19:45.02 | Mercestes | All of them? |
| 19:45.05 | c4t3l | yes |
| 19:45.14 | c4t3l | did you do something?? :) |
| 19:45.15 | Mercestes | ? weird. Any updates? |
| 19:45.16 | c4t3l | jk |
| 19:46.47 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:47.18 | Mercestes | Hey, what's the name o fthe carrier with a DS3 out? |
| 19:48.49 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:48.53 | *** join/#asterisk perimus (n=happy@mail.serverific.com) |
| 19:50.24 | flewid | so nobody's gotten a 7970 on 1.2 workin eh? |
| 19:54.02 | *** join/#asterisk corrupt (n=user@n128-227-69-72.xlate.ufl.edu) |
| 19:54.09 | *** part/#asterisk corrupt (n=user@n128-227-69-72.xlate.ufl.edu) |
| 19:54.11 | *** join/#asterisk corrupt (n=user@n128-227-69-72.xlate.ufl.edu) |
| 19:54.27 | corrupt | what is asterisk mainly use for? |
| 19:54.36 | flewid | telephony |
| 19:56.35 | LeddyHM | gettin chicks |
| 19:56.52 | corrupt | can asterisk be used to send sms and mms to cellular telephones? |
| 19:57.20 | *** join/#asterisk Goodjoke (n=Goodjoke@74.202.86.23) |
| 19:58.36 | Nugget | there's some talk of the (currently pre-beta) chan_cellphone being able to do that via bluetooth to a cooperative mobile handset, but I doubt it's the sort of solution anyone would want to rely on for production |
| 19:58.53 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 19:58.57 | Nugget | I'm still anxious to see chan_cellphone make it into svn |
| 19:59.35 | tzanger | Nugget: amen |
| 19:59.36 | tzanger | it is awesome |
| 19:59.58 | Nugget | yeah |
| 20:00.07 | corrupt | why would you doubt it's the sort of solution anyone would want to rely on for production, Nugget? |
| 20:00.18 | Nugget | just the inherent flakiness of the solution. |
| 20:00.21 | tzanger | corrupt: for production you don't want to fuck around iwth a cellphone much less bluetooth |
| 20:00.25 | Arrick | asterisk for windows sucks |
| 20:00.35 | Arrick | I have learned that in the last 1.5 hours |
| 20:00.44 | BSD_Tech | why would you run anything on windows |
| 20:01.07 | Nugget | I can see it now... "Bug # 19291 -- sms gateway fails whenever someone heats a burrito in the lunchroom microwave" :) |
| 20:01.22 | corrupt | tzanger, what do you mean when you say, "for production"? |
| 20:01.23 | Arrick | BSD_Tech, because all the networks I run are windows? |
| 20:01.45 | BSD_Tech | time to face the truth and change them |
| 20:01.46 | tzanger | corrupt: as in to rely on, for say a business or serious use |
| 20:02.25 | BSD_Tech | MicroSoft thw swiss cheese of os's |
| 20:02.32 | Arrick | BSD_Tech, there isnt any truth to linux being better, uhmm, all OS's have their uses, troll elsewhere please |
| 20:02.42 | corrupt | a lot of people rely on cellular communications for business use. |
| 20:02.56 | *** join/#asterisk pigpen2 (n=pigpen@207.71.33.114) |
| 20:03.20 | pigpen2 | anyone have a workable idefisk for the mac? I was emailed one, but the dmg is corrupt...and they don't know why... |
| 20:03.56 | pigpen2 | sorry: 2.0 beta |
| 20:04.14 | BSD_Tech | BSD is not LINUX |
| 20:04.41 | Arrick | BSD is crap |
| 20:04.53 | corrupt | what's so bad about BSD? |
| 20:04.54 | Arrick | especially the support for it |
| 20:05.00 | BSD_Tech | not even much more stable then linux and is a true unix |
| 20:05.37 | corrupt | i thought everyone and their mom was in love with FreeBSD. |
| 20:05.59 | BSD_Tech | I my BSD Asterisk Box seems to have less issuse then my linux asterisk boxes |
| 20:06.12 | Arrick | kinda funny when the OpenBSD and FreeBSD support cant even get it running enough to install apache2 with php.... yet it's supposed to be the "best" for web servers. |
| 20:06.24 | pigpen2 | bsd is a great solution...just many like the tools and community around linux. |
| 20:06.27 | BSD_Tech | what crack you on |
| 20:06.40 | BSD_Tech | apache and php are easy to install on bsd |
| 20:06.41 | Arrick | I consider anything that uses bash to be linux though |
| 20:06.42 | aydiosmio | Arrick: methinks that's apache's fault |
| 20:06.48 | aydiosmio | 2 has always been a mess |
| 20:06.58 | BSD_Tech | we dont use bash unless you install it |
| 20:07.02 | russellb | all software sucks |
| 20:07.04 | russellb | ok, we're even |
| 20:07.07 | BSD_Tech | it is not installed by default |
| 20:07.08 | pigpen2 | russellb, I agree. |
| 20:07.08 | aydiosmio | russellb sucks! |
| 20:07.11 | russellb | moving on |
| 20:07.12 | Nugget | none of my machines have bash installed. |
| 20:07.15 | russellb | aydiosmio: thank you, sir! |
| 20:07.18 | Goodjoke | @russellb... :) |
| 20:07.18 | corrupt | macs rule, pcs drool. |
| 20:07.20 | Nugget | except my two linux boxes. |
| 20:07.29 | BSD_Tech | macs are based on BSD |
| 20:07.30 | pigpen2 | corrupt, I agree... |
| 20:07.32 | BSD_Tech | so get a grip |
| 20:07.36 | aydiosmio | pork > chicken |
| 20:07.42 | corrupt | i'm joking. |
| 20:08.02 | Goodjoke | you guys have been watching too many commercials |
| 20:08.07 | pigpen2 | BSD_Tech, we know...it is just fun poke at the people that don't know...many think mac runs linux...poor bastards. |
| 20:08.12 | *** part/#asterisk Arrick (n=Arrick@about/windows/regular/arrick) |
| 20:08.56 | corrupt | i hate it when mac fanboys talk about how they're so much better than pcs then proceed to flame an operating system microsoft produces assuming that's the only thing that runs on a pc. |
| 20:09.04 | Nugget | We got an email to flightaware support, groping for a job I guess, from some mouth-breathing linux user. He said "Aha! I see you guys are using Linux, MySQL, PHP, and mrtg." He was wrong on all four. |
| 20:09.30 | pigpen2 | Windows has it's place...just not at my office. |
| 20:10.18 | pvanstam | hey, i thought this was about asterisk |
| 20:10.34 | pigpen2 | asterisk what? :) |
| 20:10.43 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 20:10.54 | Nugget | there's always room for an os war. |
| 20:10.56 | pvanstam | the key used for forwarding a phone call |
| 20:11.17 | pigpen2 | oo..not to mention a db war... (postgresql rules bty...) |
| 20:11.17 | *** join/#asterisk JerJer[mobile] (n=jj@199.45.11.90) |
| 20:11.20 | Nugget | Someday mankind will find a way to harness the energy from irc dicksize wars and bring about an end to the global petroleum crisis. |
| 20:11.46 | pvanstam | Nugget: ok, whp has asterisk on BSD running (my finger is up now) |
| 20:11.52 | *** join/#asterisk _DAW (n=chatzill@adsl-222-56-130.msy.bellsouth.net) |
| 20:11.53 | JerJer[mobile] | ohhh a flame war? dammit i'm late |
| 20:12.15 | Nugget | my production asterisk box is Linux, though. |
| 20:12.37 | pvanstam | what's the bsd system used for then? |
| 20:12.38 | pigpen2 | Oh..hey..how about a dist war? Gentoo anyone? |
| 20:12.49 | pvanstam | fedora and centos |
| 20:12.50 | Nugget | personal use. |
| 20:13.02 | darylvoip | There's always got to be atl least one Gentoo guy in the growd. |
| 20:13.09 | flewid | daryl: what's up |
| 20:13.10 | pvanstam | nugget: ok, i'm on production though with several servers |
| 20:13.28 | flewid | oh |
| 20:13.32 | Nugget | I have two Linux asterisk servers, a FreeBSD, and I run it on my OS X laptop. |
| 20:13.34 | flewid | ahha, i thought you were asking for gentoo help, sorry |
| 20:13.34 | flewid | :) |
| 20:13.37 | pigpen2 | darylvoip, yeah..I don't have a choice...my business partner is a gentoo dev (kernel stuff) |
| 20:13.47 | darylvoip | lol...then you're stuck. |
| 20:13.54 | BSD_Tech | I bet if half of you tried building a bsd and a linux box at the same time base up bsd is 1000% and I am talking by source not pkgs |
| 20:14.18 | jm|laptop | hmm |
| 20:14.20 | pigpen2 | how about this: I run Gentoo, how long does it take to compile Open Office? haha! |
| 20:14.20 | Qwell[] | 100% what? |
| 20:14.25 | BSD_Tech | and like I said my bsd asterisk box is way more stable then my linux asterisk box |
| 20:14.27 | jm|laptop | what is port 2000? It's listening on all interfaces and I'd rather it didn't ... |
| 20:14.35 | Qwell[] | jm|laptop: skinny |
| 20:14.45 | jm|laptop | Qwell[]: many thanks |
| 20:15.52 | flewid | SIP/2.0 401 Unauthorized - anyone know why this happens when trying to register a cisco 7970 that's behind nat, to a * server that isn't behind nat? |
| 20:19.17 | _DAW | With asterisk 1.4.2 and polycoms, is presence broken or considerably changed? I've tried everything but states never change from idle. |
| 20:19.26 | *** join/#asterisk menfin (n=cray@LNeuilly-152-21-127-49.w193-253.abo.wanadoo.fr) |
| 20:20.42 | BSD_Tech | ? |
| 20:20.52 | BSD_Tech | I have 1.4.2 and 6 polycoms |
| 20:20.59 | BSD_Tech | I love polycoms |
| 20:21.11 | _DAW | So do I. This is the first time I have had a presence issue. |
| 20:21.26 | pvanstam | did u restart the phones after upgrade of * |
| 20:21.46 | _DAW | Yeah. I read that call-limit is now required in 1.4. I am goint to try that. |
| 20:21.48 | BSD_Tech | what firmware on the poly ? |
| 20:21.50 | _DAW | er.. going |
| 20:22.03 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 20:22.08 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 20:22.19 | _DAW | 2.0.2.0076 |
| 20:22.32 | BSD_Tech | 2.1.1 is out and its alot better |
| 20:22.43 | pvanstam | u sure need the call-limit. I have snoms. The LED's didn't work until call-limit was set. You can find the hints in the log btw |
| 20:23.34 | _DAW | I am trying it now... |
| 20:23.38 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 20:26.36 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 20:29.32 | _DAW | That was it... |
| 20:31.00 | pvanstam | great |
| 20:31.01 | darylvoip | flewid: Did you do a SIP debug of the registration? |
| 20:31.15 | flewid | daryl: yah, i'm just running through it again to make sure i didn' tmake a lame mistake |
| 20:31.37 | flewid | i've tried nat=never, nat=yes, nat=no, defaultip=<ip>, host=<ip>,dynamic |
| 20:31.37 | darylvoip | I don't really thing NAT should have much to do with that, but I guess anything is possible. |
| 20:31.44 | flewid | nothing seems to work |
| 20:31.49 | flewid | yeah, i'm just getting sip/401 reg errors |
| 20:31.53 | flewid | if i register with a softphone, works fine |
| 20:32.00 | darylvoip | Ugh. Of course. |
| 20:32.32 | flewid | unfortunately I don't have a local 1.2 box or i'd test against it as well |
| 20:32.39 | flewid | i know it works on 1.4.1 and 1.4.2 tho |
| 20:32.55 | darylvoip | Are you aware that on some of the Cisco firmware loads it seems to try to use linex_name at the linex_authname? |
| 20:32.58 | *** join/#asterisk cspot (i=cspot@ip68-109-8-207.pn.at.cox.net) |
| 20:33.13 | flewid | hmm, no i wasn't |
| 20:33.15 | darylvoip | If you just want to throw somthing at it, use yout authname n both fields for the phone and see if it works |
| 20:33.27 | darylvoip | I've run into that one a few times now. |
| 20:33.32 | flewid | i'm generating my password with echo -n "800:asterisk:800" | md5sum |
| 20:33.35 | pvanstam | try downgrading sip on the cisco. I had an issue with cisco's and * 1.2.13 |
| 20:33.53 | flewid | sure, let me try that |
| 20:33.54 | flewid | sec |
| 20:33.56 | pvanstam | shouldn't you better run skinny on * |
| 20:34.03 | darylvoip | Yeah...I had 8.2 and it was doing that. Upgraded to 8.6 and it does the same thing. |
| 20:34.10 | darylvoip | I'm thinking it broke right around 8.x |
| 20:34.12 | flewid | 8.6? |
| 20:34.21 | flewid | i tihnk i'm running 8.0.2s1 |
| 20:34.39 | darylvoip | Yeah....most likely it was an 8.0 bug. |
| 20:34.48 | *** join/#asterisk ToTo (n=toto@host232-161-dynamic.0-87-r.retail.telecomitalia.it) |
| 20:34.49 | flewid | authnae, name are both the same thing |
| 20:34.51 | flewid | that's currect eh |
| 20:34.53 | ToTo | hi all |
| 20:34.56 | darylvoip | Because I know that doesn't happen on my customer's phones and didn't use to happen on mine (at 7.x) |
| 20:34.57 | flewid | in the SEPxxx.xml |
| 20:35.22 | joe-f | anyone know what would prevent my asterisk server from accepting voice/digit entry from a hard phone? I'm testing voxbone's VOIP origination.. |
| 20:35.31 | ToTo | does someone use asterisk snmp support with cacti? |
| 20:35.38 | flewid | all the details in SEPxxx.xml are the same, but if i change the reg server it won't work :/ |
| 20:35.41 | flewid | toto: yup |
| 20:36.29 | joe-f | does anyone use asterisk and VOIP originiation with someone like voxbone? |
| 20:37.17 | *** join/#asterisk mquin (n=mike@pdpc/supporter/active/mquin) |
| 20:38.03 | ToTo | flewid: i have some question, i wold to see current channles , i use generic oid templeate and in OID space i put asterisk.5.1 is ti correct? |
| 20:38.06 | darylvoip | flewid: You're using it? I haven't done that yet...been meaning to. Messy setup? I've got 8 * boxes I've been meaning to roll it out to. |
| 20:39.36 | flewid | it was kinda a bitch, and I find that i must start asterisk first, then the snmp daemon |
| 20:39.36 | flewid | otherwise no data gets pulled |
| 20:39.36 | flewid | but there's a howto on the voip-info site |
| 20:39.37 | flewid | i think i may have written about it on my blog too |
| 20:39.37 | flewid | sec |
| 20:39.37 | darylvoip | Ugh...yeah...the howto was what made me think I didn't want to bother on a production network. |
| 20:39.41 | flewid | http://www.voipphreak.ca/archives/382 |
| 20:39.41 | flewid | that's what i did to make it work |
| 20:39.41 | darylvoip | I've got 100k minutes a day that need to pass. |
| 20:39.41 | darylvoip | Nice..thanks. |
| 20:39.41 | flewid | np |
| 20:39.41 | ToTo | flewid: with snmpwalk i can get data |
| 20:39.46 | flewid | toto: if you're following mark's howto, the OID's have all changed |
| 20:39.46 | flewid | check ine out for the new descirptions |
| 20:39.58 | darylvoip | So I'm downgrading my phone to 7.4 to see if that linex_name/authname thing isnt broken anymore, now that you've reminded me of it. |
| 20:40.16 | flewid | :) i've never tried a 7.x image |
| 20:40.24 | flewid | i wonder if that would fix my registration problems |
| 20:40.27 | darylvoip | Well, it's been on my nagging todo list. |
| 20:40.30 | ToTo | flewid: is OID string for cacti asterisk.x.x correctt/ |
| 20:40.38 | ToTo | ? |
| 20:40.39 | darylvoip | I'll let you know if it EVER GETS PAST CONFIGURING VLAN. |
| 20:40.49 | flewid | from the forum entries i've found, it seems that people have the 7970 working with trix 1.2 |
| 20:40.51 | pvanstam | flewid: I had cisco's with SIP on 8.0.x and needed to downgrade. 7.4 did work for me |
| 20:40.53 | flewid | which was 1.2.x series asterisk |
| 20:40.58 | flewid | yah eh |
| 20:41.02 | flewid | anyone got the 7.4 img? :) |
| 20:41.06 | flewid | toto: sec |
| 20:41.17 | ToTo | flewid: ok |
| 20:41.38 | flewid | ASTERISK-MIB::astNumChannels.0 |
| 20:41.40 | pvanstam | flewid: i have 7.5 |
| 20:41.40 | flewid | or |
| 20:41.44 | flewid | .1.3.6.1.4.1.22736.1.5.1.0 |
| 20:41.47 | pvanstam | sorry 7.4 |
| 20:41.47 | flewid | is your channels |
| 20:42.28 | ToTo | flewid: where i can get this one .1.3.6.1.4.1.22736.1.5.1.0? |
| 20:42.44 | flewid | read my howto :) it's all explained there |
| 20:42.51 | ToTo | ok |
| 20:43.07 | *** join/#asterisk Fieldy (i=w2GXwAvr@gentoo/contributor/Fieldy) |
| 20:44.15 | darylvoip | OK....flewid...that didn't fix it |
| 20:45.16 | flewid | darn |
| 20:45.23 | flewid | brb |
| 20:45.43 | joe-f | if anyone could point me in ANY direction, on where to find out how i can get my voip origination -> sip -> asterisk server to pick up my voice? I can hear the MOH, etc.. but cant send anything to it.. is this something with codecs?? |
| 20:46.41 | darylvoip | So you have one-way audio? |
| 20:46.48 | darylvoip | That's usually a NAT issue. |
| 20:47.09 | shido6 | ZZzZz |
| 20:50.08 | _VoiceMeUp_COM | darylvoip;) one way audio is 99% nat 1% code 18 |
| 20:50.33 | darylvoip | Well, true enough. |
| 20:50.35 | _VoiceMeUp_COM | the 1% is .. people have the mute button enabled |
| 20:50.38 | _VoiceMeUp_COM | ;) |
| 20:51.06 | _VoiceMeUp_COM | im aliasing that situation to synaptic storm in the users left brain |
| 20:51.15 | darylvoip | Doesn't it still count as code 18 when you're the guy configuring the box? |
| 20:51.42 | _VoiceMeUp_COM | and it happens around 1 time every day on our support line.. also had a client say voip was down and they had a power loss at theyr home ;) |
| 20:52.36 | _VoiceMeUp_COM | with no ups of course.. |
| 20:52.36 | darylvoip | Well.....it WAS down at their house, now wasn't it? |
| 20:52.36 | joe-f | darylvoip: yes, i have one-way audio.. however when i use x-lite, i have two-way |
| 20:52.36 | _VoiceMeUp_COM | heeheh, yes but where not the power corp.. yet |
| 20:52.36 | joe-f | darylvoip: does that make sense? |
| 20:52.39 | _VoiceMeUp_COM | yes |
| 20:52.44 | darylvoip | Sure |
| 20:52.53 | _VoiceMeUp_COM | also note , sometime xlite will take control of some audio on lan |
| 20:53.17 | joe-f | yeah, so try to use it outside of my lan? |
| 20:53.25 | _VoiceMeUp_COM | EX.. if you have 2 phones reg'ed to asterisk.. and xlite reg'es then you might have fails on other phone |
| 20:53.26 | darylvoip | definitely |
| 20:53.34 | joe-f | ok |
| 20:53.39 | joe-f | sweet, this helps me already |
| 20:53.41 | darylvoip | Yeah...depending on your router....what voicemeup said. |
| 20:53.44 | drfreeze | Anyone seens this problem when building Asterisk? http://pastie.textmate.org/56276 |
| 20:53.47 | darylvoip | Gotta love uPnP |
| 20:53.56 | paavum | I keep seeing this in my cli |
| 20:54.05 | paavum | ZtHook Failed: Device or resource busy |
| 20:54.08 | paavum | what can it be? |
| 20:54.11 | joe-f | so to deal with nat issues, would this issue be related to sip_nat.conf ? |
| 20:54.57 | joe-f | what ports besides 5060 are usually the case? |
| 20:55.39 | darylvoip | 5060 is just SIP. |
| 20:55.46 | darylvoip | One-way audio is an RTP issue. |
| 20:56.00 | darylvoip | If you completed the call, the SIP is fine. That's just call signaling. |
| 20:56.21 | joe-f | ok, sweet |
| 20:56.26 | joe-f | ahhh thanks for this input |
| 20:56.52 | darylvoip | If you have some crappy home router, just put the asterisk box in the "dmz" or whatever they call it. |
| 20:57.08 | darylvoip | It will likely go away. Just make sure you * box is properly protected. |
| 20:57.40 | darylvoip | Or you could forward all of the rtpo ports in rtp.conf to the asterisk box. |
| 20:57.48 | darylvoip | not the best way to do things, but brute force works sometimes. |
| 20:59.16 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:59.21 | joe-f | yeah, so i'd prefer to not open it up completely |
| 20:59.49 | darylvoip | Well, you can give it a try for a test, and at least figure out if that's the issue or not |
| 20:59.59 | joe-f | yeah i'll do that right now, thanks a ton |
| 21:00.07 | BSD_Tech | back |
| 21:00.13 | J4k3 | heh... if you can't trust your * box, why do you trust your router? :) |
| 21:00.37 | J4k3 | I mean, your router is probably running Linux or VXWorks, or some proprietary crap with limited testing ;) |
| 21:01.23 | [TK]D-Fender | "Trust Ivanova. Trust yourself. Anybody else: shoot 'em!" |
| 21:01.56 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-190-47.nycmny.east.verizon.net) |
| 21:03.46 | *** join/#asterisk RevRob (n=RevRob@212.183.134.129) |
| 21:05.17 | zuez | Argh, why does FWD insist on rejecting authorization :-) |
| 21:05.50 | *** join/#asterisk Bobocop (n=Bobocop@public-gprs48217.centertel.pl) |
| 21:07.32 | flewid | okay |
| 21:07.37 | flewid | so lets try this downgrade :) |
| 21:07.52 | darylvoip | lol...I think I just figured out what I was doing wrong. |
| 21:07.56 | darylvoip | linex_name is the name it registeres with |
| 21:08.17 | darylvoip | linex_authname and linex_password is what is uses if it is challenged by a proxy |
| 21:08.33 | darylvoip | if you don't specify for lines 2-6, it uses the authname and password from line 1 |
| 21:09.02 | flewid | oh? hmm let me check y sep |
| 21:09.15 | darylvoip | for whatever reason, it will use linex_authname as the phone label. If you want a meaningful label, you need to specify linex_shortname |
| 21:09.33 | darylvoip | I knew that at one point, and had to go back and look it up :) |
| 21:11.08 | *** join/#asterisk tuxinator_linux (n=tuxinato@128.187.169.195) |
| 21:13.50 | *** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@145-27.mc.cite.net) |
| 21:14.17 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 21:14.53 | *** join/#asterisk galeras (n=root@200.31.204.42) |
| 21:15.55 | *** join/#asterisk xlyz (n=xz@host-84-223-89-30.cust-adsl.tiscali.it) |
| 21:16.02 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 21:16.07 | *** part/#asterisk xlyz (n=xz@host-84-223-89-30.cust-adsl.tiscali.it) |
| 21:20.33 | galeras | In a 3-party conference: User, Agent and IVR,i need to avoid that the agent listen DTMF tones pressed by the user, but i need to process that tones by the ivr. Any suggestion? |
| 21:22.48 | galeras | init 3 |
| 21:26.05 | zuez | [Apr 24 17:24:03] NOTICE[9465]: chan_iax2.c:7368 socket_process: Registration of '841968' rejected: 'Registration Refused' from: '192.246.69.186' <- anyone had that issue with FWD before albeit using the correct user/pass? |
| 21:26.05 | zuez | I'm assuming it could be a firewall issue as well |
| 21:26.06 | _VoiceMeUp_COM | no |
| 21:26.06 | _VoiceMeUp_COM | not a frewall |
| 21:26.06 | _VoiceMeUp_COM | theyr reg is down |
| 21:26.06 | _VoiceMeUp_COM | or |
| 21:26.06 | _VoiceMeUp_COM | seomthign messed up |
| 21:26.12 | _VoiceMeUp_COM | Registration Refused' from means its aCtively refused |
| 21:26.12 | _VoiceMeUp_COM | not like cant connect or timeouts |
| 21:26.40 | zuez | yeah, I haven't configured extensions.conf yet, but I figured the iax2 peer can take place without it if iax.conf is properly configured |
| 21:26.56 | drfreeze | Can anyone help me get zaptel to compile? |
| 21:27.37 | drfreeze | http://pastie.textmate.org/56290 |
| 21:27.57 | Mercestes | drfreeze, Well, first you install gentoo. :D |
| 21:28.26 | drfreeze | Mercestes: :) seems like we've had this conversation before. |
| 21:28.32 | Mercestes | Aye. |
| 21:28.36 | Mercestes | do you need xbus core? |
| 21:28.59 | drfreeze | there was a problem with the build soft link inside /lib/modules dir |
| 21:29.31 | _DAW | Finally.. all my polycoms have working presence with 1.4.2 :) |
| 21:29.48 | drfreeze | if pointed to the 2.6.18-8.el5-i686 kernel |
| 21:29.48 | drfreeze | but, the 2.6.18-8.1.1.el5-i686 was what existed |
| 21:29.57 | *** part/#asterisk falco_toadfoot (i=papir@160.80-202-240.nextgentel.com) |
| 21:30.56 | *** join/#asterisk zogulus (n=zogulus@58.98.adsl.brightview.com) |
| 21:35.31 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-136017.home.otenet.gr) |
| 21:40.41 | |BLiX| | anyone know why I am getting this... |
| 21:40.42 | |BLiX| | <PROTECTED> |
| 21:40.42 | |BLiX| | <PROTECTED> |
| 21:40.42 | |BLiX| | <PROTECTED> |
| 21:40.42 | |BLiX| | [Apr 20 08:28:46] ERROR[16925]: chan_h323.c:1377 build_user: A dynamic host on a type=user does not make any sense |
| 21:40.55 | |BLiX| | the host is an ip address |
| 21:41.47 | |BLiX| | it happens as soon as I uncomment the line ;hash323 = yes in users.conf |
| 21:42.11 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 21:45.24 | nasls_lsa | I think type is only peer or friend |
| 21:46.51 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:46.53 | blitzrage | tzanger: ping |
| 21:47.25 | blitzrage | anyone in Toronto have the Rogers SMS number? |
| 21:47.40 | |BLiX| | the type is friend |
| 21:48.40 | |BLiX| | its like chan_h323 is ignoring the host field |
| 21:49.02 | ManxPower | I am starting to think I am experiencing bt rot on my XP box |
| 21:51.03 | illsci | i love toronto.. |
| 21:52.32 | Mercestes | illsci, when I first glanced at that I read "I love torture." Gotta love my mind. |
| 21:53.15 | *** join/#asterisk thoughtpolice (n=austin@c75-111-146-82.plaicmtc01.tx.dh.suddenlink.net) |
| 21:55.45 | *** join/#asterisk cyburdine (n=cyburdin@fonix.com) |
| 21:56.12 | *** join/#asterisk axisys (n=axisys@c-69-143-190-152.hsd1.va.comcast.net) |
| 21:56.17 | *** part/#asterisk slmnhq (n=salmanh@denali.asti-usa.com) |
| 21:57.51 | *** join/#asterisk dotSlashW (n=HTP@200.80.197.5) |
| 21:59.00 | dotSlashW | I need a device to connect my trixbox to the cellular network, something like the Nexo FCT, any tips on what to look for, prices, etc ? |
| 21:59.30 | jm|laptop | the http server on 8088 doesn't seem to do much :S |
| 21:59.54 | BSD_Tech | it servs the web pages |
| 22:00.02 | jm|laptop | well; yes. |
| 22:00.10 | cyburdine | didn't cingular have a product that connected your cell to your home line? |
| 22:00.13 | BSD_Tech | if you install the asterisk gui |
| 22:00.17 | jm|laptop | ah |
| 22:00.18 | jm|laptop | I didn't |
| 22:00.45 | jm|laptop | vi ftw .... or something |
| 22:00.49 | *** join/#asterisk amegyeri (i=amegyeri@catv-50634218.catv.broadband.hu) |
| 22:00.53 | BSD_Tech | ? |
| 22:00.59 | jm|laptop | quite. |
| 22:01.01 | *** join/#asterisk kusr (i=rehn@nat/google/x-f43306e65ef3dbaa) |
| 22:01.08 | tzafrir | jm|laptop, what do you want it to do? |
| 22:01.18 | BSD_Tech | give him a bj |
| 22:01.31 | jm|laptop | tzafrir: I'm on Debian and Asterisk in apt just went 1.2 --> 1.4 I'm fiddling with the new things |
| 22:01.34 | jm|laptop | BSD_Tech: ?! |
| 22:01.39 | ManxPower | I'm impressed. Got a response from Digium support in less than 1 hour |
| 22:02.55 | mog | woohoo |
| 22:03.01 | cyburdine | is it possible to perform a blind transfer in AGI? I'm having a hell of a time getting caller id to pass correctly. |
| 22:03.49 | ManxPower | mog: And the tech ACTUALLY READ MY MESSAGE and sent a fix. |
| 22:03.55 | ManxPower | First time that's happened in a while. |
| 22:04.02 | mog | ouch... |
| 22:04.08 | mog | well im glad you got help |
| 22:04.49 | ManxPower | mog: last time I had to get kpflemming to spank the tech and get my solution |
| 22:04.49 | cyburdine | I have two asterisk boxes (switchvox as our PBX and a custom asterisk box acting as our auto attendent) |
| 22:05.13 | cyburdine | and the caller id seems to be getting lost when i dial between them |
| 22:06.02 | flewid | guh |
| 22:06.15 | flewid | yeah 7.4 and 7.5 and 8.2.1S1 7970 no reg to 1.2.16 |
| 22:06.26 | flewid | maybe 8.2.2s2 will fix |
| 22:06.27 | flewid | :/ |
| 22:06.47 | ManxPower | flewid: IT is externip= not externalip= |
| 22:08.47 | *** join/#asterisk magikxx (n=tawandax@209.88.93.62) |
| 22:09.12 | cyburdine | I am able to do a blind transfer manually and all things work |
| 22:10.03 | cyburdine | hence my curiosity if anyone knows how to implement a blind transfer in AGI |
| 22:10.07 | puzzled | evening ManxPower. how's life? |
| 22:10.23 | blitzrage | flewid: you have to use a fairly old version to make the 7970 connect to Asterisk -- the voip-info wiki will tell you the correct version |
| 22:10.50 | puzzled | I thought that 7.4 generally works ok with 1.2 |
| 22:11.00 | *** join/#asterisk Frosh (n=Frosh@unaffiliated/frosh) |
| 22:11.28 | ManxPower | puzzled: I am hating microsoft more every moment. |
| 22:11.33 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:11.37 | flewid | manx: yah, i mistyped :( |
| 22:11.48 | flewid | will that actually make a diff for the registration? |
| 22:11.57 | flewid | i just posted a message to the mailing list with the sip debug from 1.4 vs 1.2 |
| 22:12.02 | flewid | it looks to me like it's something with digest auth |
| 22:12.15 | ManxPower | puzzled: been trying for DAYS to get DVDs converted in a format that can stream to Windows and Mac. No luck. |
| 22:12.21 | puzzled | ManxPower: I can imagine. did they trick you into becoming responsible for some MS stuff? |
| 22:12.24 | flewid | blitz: i'm connecting to 1.4.2 fine using the latest SIP |
| 22:12.26 | kusr | Hello. I think I have a firewall problem. My asterisk server is behind a NAT/firewall router. I have two service providers (A and B) that my server talks to (via SIP). I can receive and place calls from both of them fine. But when I try to forward an incoming call like this: caller->A->my server->B->me, the caller can hear me fine but I can hear him only for half a second, then the sound is cut off. I have opened ports 5060-5100 and |
| 22:12.26 | kusr | 10000-20000 (=my RTP range in rtp.conf) in my NAT/firewall. To connect the call I use Dial(). I have nat=no and use externhost=hostname.tld to let people outside of the NAT find me. What could be wrong/how do I debug? |
| 22:12.29 | flewid | it's just 1.2.16 that's being the bitch :) |
| 22:12.42 | ManxPower | flewid: I started to read the message then realize that your sip.conf info was not there so I deleted the message. |
| 22:12.50 | ManxPower | puzzled: this is a personal project |
| 22:13.12 | puzzled | ah right |
| 22:13.12 | flewid | manx: sec, i'll paste it to pastbin |
| 22:13.17 | puzzled | never did that, converting dvd's into streamable stuff |
| 22:13.44 | bmd | cyburdine: show application transfer |
| 22:14.02 | ManxPower | flewid: too late now |
| 22:14.19 | ManxPower | puzzled: I would be happy to be able to convert ANYTHING into streaming |
| 22:14.20 | _VoiceMeUp_COM | could be |
| 22:14.32 | _VoiceMeUp_COM | we are now dedicating a 1.4 box to trasnalte back to our systems.. |
| 22:14.41 | _VoiceMeUp_COM | else we hav elocks and all |
| 22:14.45 | flewid | manx: :/ |
| 22:15.17 | ManxPower | cyburdine: try the "o" option to dial |
| 22:15.34 | _VoiceMeUp_COM | <PROTECTED> |
| 22:15.34 | _VoiceMeUp_COM | <PROTECTED> |
| 22:15.34 | _VoiceMeUp_COM | <PROTECTED> |
| 22:15.46 | _VoiceMeUp_COM | sorry had to look too ;) |
| 22:16.29 | bmd | that'll work too, but Transfer() actually does a SIP REFER (ie: blind transfer) |
| 22:16.32 | puzzled | ManxPower: http://www.peerstream.net/ it's payware but not too expensive |
| 22:17.34 | ManxPower | puzzled: the last FOUR pieces of commercial software I paid for did not do what they promised |
| 22:18.04 | puzzled | ManxPower: heheh been there too. you can try them out for free iirc |
| 22:18.46 | *** join/#asterisk [Outcast] (n=bill@219-89-206-239.adsl.xtra.co.nz) |
| 22:19.00 | ManxPower | The most recent: Paid for Quicktime Pro so I can convert media files from 1 format to another format. Turns out it does not support converting from MPEG2. So I bought the MPEG2 module for quicktime. Turns out it can transcode MPEG2 video, but does nothing with mpeg2 audio (how useless is that?) |
| 22:20.17 | [Outcast] | anyone else here using broadvoice? |
| 22:20.23 | flewid | manx: now you have pastebin and -list :) |
| 22:20.50 | _VoiceMeUp_COM | nope |
| 22:21.01 | ManxPower | puzzled: I need 2Mbps streams at least, and it has to run on Linux |
| 22:21.02 | _VoiceMeUp_COM | lol |
| 22:21.03 | flewid | outcast: we used to use them, but have since switched |
| 22:21.08 | _VoiceMeUp_COM | ManxPower i use hold on |
| 22:21.59 | _VoiceMeUp_COM | squeeze |
| 22:22.00 | ManxPower | this stuff will be on a 100Mbps switched lan with a GigE backbone. It would be silly to use any lower of a bitrate |
| 22:22.16 | _VoiceMeUp_COM | sorenson squeeze |
| 22:22.22 | _VoiceMeUp_COM | it also can monitor a folder and ocnvert |
| 22:22.40 | ManxPower | _VoiceMeUp_COM: does it run under Linux? |
| 22:22.53 | _VoiceMeUp_COM | does mp4,swf/flv/mov//wmv/.rm/mpg/mp3 |
| 22:22.57 | _VoiceMeUp_COM | nah |
| 22:22.57 | ManxPower | and does it support ASF contrainers |
| 22:23.03 | _VoiceMeUp_COM | unles you use.. hold on again lol |
| 22:23.11 | _VoiceMeUp_COM | crossover |
| 22:23.15 | _VoiceMeUp_COM | that urns iwndows on unix |
| 22:23.21 | _VoiceMeUp_COM | windows apps i mean |
| 22:23.23 | ManxPower | since ASF is, as far as I can tell is the only format media player will use as a stream |
| 22:23.27 | _VoiceMeUp_COM | but yuo could put a win2k server |
| 22:23.32 | _VoiceMeUp_COM | and moutn the unix drive |
| 22:23.36 | _VoiceMeUp_COM | dir |
| 22:23.50 | ManxPower | _VoiceMeUp_COM: Hell will freeze over before I manage or install a microsoft server |
| 22:23.53 | _VoiceMeUp_COM | no idea i uguess theres modules for it |
| 22:23.59 | _VoiceMeUp_COM | lol i feel you |
| 22:24.04 | _VoiceMeUp_COM | that why i dont use |
| 22:24.32 | cyburdine | thanks guys, yeah I tried the o option, but that isn't doing it. transfer may be exactly what we need. I'll try that and let you know my results. |
| 22:24.47 | ManxPower | I have a few other things I can try |
| 22:24.52 | plasmid | any users here who can give some experiences on asterisknow? I am contemplating on installing it. |
| 22:25.02 | _VoiceMeUp_COM | cough |
| 22:26.18 | ManxPower | plasmid: the asterisk now channel was not helpful? |
| 22:26.22 | plasmid | wow. The channel went completely comatose. asterisknow = sacrilege here? |
| 22:26.38 | plasmid | ManxPower, they are all dead/ afk |
| 22:27.16 | ManxPower | plasmid: the last person that asked about Asterisk now was exiled to sibera |
| 22:27.40 | plasmid | ManxPower, it's quite alright. I got a compass and a US passport. |
| 22:27.46 | ManxPower | plasmid: I think everyone is trying to figure out where you are physically located. |
| 22:28.06 | plasmid | ManxPower, I can save them the trouble. Philadelphia, PA. |
| 22:28.18 | Brandon_W | oh, that explains a lot |
| 22:28.23 | Brandon_W | :) |
| 22:28.27 | plasmid | Brandon_W, sure does. :-) |
| 22:28.37 | Corydon-w | wudder world |
| 22:29.28 | plasmid | anyways, I reckon asterisknow is not something to be triffled with. A shame... not all of us are 100% console editing gnomes. |
| 22:29.44 | Brandon_W | is #asterisknow a dead channel? |
| 22:29.58 | ManxPower | plasmid: I suggest the Nortel BCM then . |
| 22:30.00 | plasmid | Brandon_W, try saying something. |
| 22:30.03 | Corydon-w | I'm rather proud of my color syntax highlighting file for vim, though |
| 22:30.19 | plasmid | Corydon-w, along with your TAB completion. |
| 22:30.19 | Brandon_W | ah, so I see |
| 22:30.30 | Corydon-w | plasmid: no, that's not mine |
| 22:30.37 | Brandon_W | I had asterisknow installed on one of my play boxes |
| 22:30.41 | Brandon_W | it was ok |
| 22:30.59 | Brandon_W | I liked freepbx over asterisknow |
| 22:30.59 | russellb | there is nothing wrong with asking about asterisknow here |
| 22:31.02 | Corydon-w | plasmid: I'm talking about the file that I wrote |
| 22:31.08 | Brandon_W | and just normal CLI editing over both. :P |
| 22:31.17 | plasmid | Corydon-w, i c |
| 22:31.35 | russellb | the asterisk gui with asterisknow has a built-in fancy way to manually edit the config files :) |
| 22:31.38 | russellb | it's hot |
| 22:31.51 | Brandon_W | AsteriskNow seems like A Good Start; but not quite there yet... |
| 22:32.19 | russellb | so you can use the GUI for some things, and still manually hack at the config or look at what it builds for learning purposes |
| 22:32.38 | plasmid | Brandon_W, isn't it a tad more efficient in managing the configuration files as opposed to trixbox configuration files "management"? |
| 22:33.01 | russellb | it communicates directly with the asterisk manager interface over HTTP |
| 22:33.10 | Brandon_W | eh, I can't really speak to it completely. I've only used either for a short time before just going totally CLI for any edits I Need |
| 22:33.10 | russellb | so it asks asterisk to directly modify the config files |
| 22:33.55 | plasmid | Brandon_W, so when you say "...not quite there yet" are u referring to some features, usability or? |
| 22:34.15 | Brandon_W | usability mainly, least from the time I Played with it |
| 22:34.29 | Brandon_W | It could have just been me, don't get me wrong |
| 22:34.53 | _VoiceMeUp_COM | hehe maybe some day digium could bpatent the manager api or cli ;0 |
| 22:35.09 | Corydon-w | You can't patent software |
| 22:35.21 | _VoiceMeUp_COM | You can patent drivers no ? |
| 22:35.21 | Brandon_W | patent the API would be silly |
| 22:35.31 | Corydon-w | No, you cannot. |
| 22:35.33 | ManxPower | Corydon: tell that to Microsoft |
| 22:35.37 | Brandon_W | the whole point of an API is so people can interface with your system |
| 22:35.41 | Corydon-w | You can copyright drivers, not patent them |
| 22:35.45 | _VoiceMeUp_COM | hmm oh, hmm g729 is not a driver then , |
| 22:35.55 | _VoiceMeUp_COM | or what is difference, sorry im dumb ;) |
| 22:36.02 | russellb | no, it is an algorithm |
| 22:36.09 | Brandon_W | All of you owe me $.05 a breath |
| 22:36.16 | _VoiceMeUp_COM | ok , algorithm is software |
| 22:36.20 | _VoiceMeUp_COM | or math |
| 22:36.22 | russellb | no it's not :) |
| 22:36.26 | _VoiceMeUp_COM | or math applied to software ;) |
| 22:36.28 | Brandon_W | and with that, I'm out |
| 22:36.32 | _VoiceMeUp_COM | i just dont get it |
| 22:36.36 | russellb | software is just the specific implementation of the algorithm |
| 22:36.40 | _VoiceMeUp_COM | ok |
| 22:36.44 | *** join/#asterisk saftsack (n=saftsack@pd9e06842.dip.t-dialin.net) |
| 22:36.53 | _VoiceMeUp_COM | so software is the application using the algo |
| 22:36.56 | JerJer[mobile] | I am considering filing a patent on a business method that involves the filing of patents that are trivial, and to some, obvious extensions of what has been done before. That way I can enjoin those stupid companies that do business this way. :) |
| 22:37.00 | _VoiceMeUp_COM | like a car uses a moto |
| 22:37.02 | _VoiceMeUp_COM | motor |
| 22:37.53 | _VoiceMeUp_COM | so alotugh you could patent the mechanism to create a deadlock , you couldnt patent asterisk |
| 22:37.55 | _VoiceMeUp_COM | right ? |
| 22:38.07 | Corydon-w | Correct |
| 22:38.08 | _VoiceMeUp_COM | j/k |
| 22:38.09 | _VoiceMeUp_COM | ;) |
| 22:38.20 | _VoiceMeUp_COM | i amuse myself more hten i should |
| 22:38.31 | Corydon-w | Specific implementations can be protected with copyright, not with patents |
| 22:38.57 | _VoiceMeUp_COM | ok |
| 22:39.12 | jm|laptop | ok 1.4 is officially weird |
| 22:39.19 | BSD_Tech | 1.4.2 rocks |
| 22:39.26 | _VoiceMeUp_COM | jm|laptop ;) |
| 22:39.28 | dotSlashW | <PROTECTED> |
| 22:39.35 | russellb | 1.4.3 coming soon to a store near you |
| 22:39.36 | Corydon-w | You can't patent me; I am prior art. |
| 22:39.41 | jm|laptop | I have 1.4.2 and it's being weird |
| 22:39.42 | BSD_Tech | google |
| 22:39.46 | _VoiceMeUp_COM | bah oyu a clone ;) |
| 22:39.52 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 22:40.00 | ManxPower | dotSlashW: I doubt anyone here even uses Trixbox |
| 22:40.07 | BSD_Tech | google gsm gateway |
| 22:40.07 | _VoiceMeUp_COM | dotSlashW , let me know what you need got a wharehouse full of stuff m,, from nortel lucient cisco etc |
| 22:40.20 | _VoiceMeUp_COM | also a sattelite modem bank |
| 22:40.20 | Corydon-w | You're like the third person recently who has proposed cloning me |
| 22:40.31 | russellb | ~trixbox |
| 22:40.33 | jbot | Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org |
| 22:40.40 | *** join/#asterisk krapper (n=krapper@wsip-64-58-154-130.oc.oc.cox.net) |
| 22:40.40 | BSD_Tech | freepbx is coming out with a non trixbox iso |
| 22:40.42 | _VoiceMeUp_COM | ~noob |
| 22:40.44 | jbot | ACTION has died. |
| 22:40.44 | BSD_Tech | called isopbx |
| 22:40.46 | flewid | druidgui is nice if you want a gui |
| 22:40.50 | _VoiceMeUp_COM | lol |
| 22:40.50 | JerJer[mobile] | speaking of 1.4 weirdness - about 90% of the commands i am so used to (in 1.2) have been munged or outright removed :( |
| 22:40.51 | flewid | cli > * tho :) |
| 22:41.06 | russellb | JerJer[mobile]: I know </3 |
| 22:41.11 | russellb | I'm still not used to it |
| 22:41.16 | krapper | anyone seen where asterisk 1.4.2 will all of the sudden forget it's global variables? |
| 22:41.20 | _VoiceMeUp_COM | ~trixbox |
| 22:41.24 | jbot | Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org |
| 22:41.27 | krapper | this just happened on a production box of mine |
| 22:41.27 | dotSlashW | russellb, yeah ok, lets say I'm running * then, anyway everything is working, I just need a new device |
| 22:41.28 | _VoiceMeUp_COM | oh jbot really did die |
| 22:41.36 | _VoiceMeUp_COM | ~noob |
| 22:41.37 | jbot | ACTION has died. |
| 22:41.40 | _VoiceMeUp_COM | hehe |
| 22:41.41 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 22:41.45 | JerJer[mobile] | russellb: it would be nice if there was some consistency |
| 22:41.53 | russellb | JerJer[mobile]: that was the goal ... |
| 22:41.56 | jm|laptop | lol |
| 22:41.58 | JerJer[mobile] | like sip set debug on (which btw, on is not listed) |
| 22:42.07 | JerJer[mobile] | but then its still rtp debug |
| 22:42.57 | Corydon-w | JerJer[mobile]: it's supposed to be consistent now |
| 22:42.57 | Corydon-w | JerJer[mobile]: everything is of the form: MODULE ACTION ARGS |
| 22:43.06 | JerJer[mobile] | sip set debug versus rtp debug |
| 22:43.06 | jm|laptop | awww man; something has broken *everything* :( |
| 22:43.11 | JerJer[mobile] | <PROTECTED> |
| 22:43.13 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 22:43.31 | Corydon-w | JerJer[mobile]: yeah, we'll have to fix that |
| 22:43.35 | jm|laptop | hmm it might be DNS |
| 22:43.43 | JerJer[mobile] | i must say though that 1.4 does scale quite a lot better |
| 22:43.47 | Corydon-w | JerJer[mobile]: but it'll wait until 1.6 now that 1.4 is already released |
| 22:44.15 | Corydon-w | it had been a lot worse |
| 22:44.37 | BSD_Tech | call it gasterisk and be done with it because its bloated |
| 22:44.40 | jm|laptop | (it was DNS) |
| 22:44.43 | jm|laptop | << |
| 22:45.11 | Corydon-w | "sip set debug off" is far nicer than "sip no debug", for example. |
| 22:45.33 | JerJer[mobile] | yeah |
| 22:45.38 | russellb | JerJer[mobile]: scales a lot better, huh? good to hear ... |
| 22:45.40 | JerJer[mobile] | tru dat |
| 22:45.46 | flewid | manxpower: did you get a chance to look at my pastebin for the 7970 ? |
| 22:45.50 | flewid | ah damn |
| 22:45.56 | Corydon-w | and I'm going home |
| 22:45.56 | JerJer[mobile] | 967 active calls |
| 22:46.02 | russellb | hot |
| 22:46.31 | BSD_Tech | I still think everyone should move to asterisk on bsd |
| 22:46.33 | JerJer[mobile] | dual xeon, 3gig, sip, ulaw, playing whitenoise |
| 22:46.38 | JerJer[mobile] | out of a ramdisk |
| 22:47.20 | *** join/#asterisk Strom_M (n=strom@135.196.213.180) |
| 22:47.46 | JerJer[mobile] | the box had a load average of like 85 but we could place a call between two attached phones and call quality was rock solid |
| 22:47.51 | JerJer[mobile] | sip phones |
| 22:48.43 | SwK | anyone have a Polycom 501 or 550 they want to sell cheap? |
| 22:49.04 | BSD_Tech | you can have my polycom when I am dead |
| 22:49.10 | SwK | heh |
| 22:49.47 | jm|laptop | does 1.4 not do mp3 moh? |
| 22:49.55 | BSD_Tech | http://ipphone-warehouse.com/ |
| 22:49.57 | BSD_Tech | yes |
| 22:50.04 | BSD_Tech | 1.2 does also |
| 22:50.32 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 22:50.32 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:50.43 | BSD_Tech | swk look at thier prices |
| 22:50.57 | BSD_Tech | theres a name I have not seen in a long time |
| 22:51.02 | BSD_Tech | hey Anthm |
| 22:51.26 | jm|laptop | [Apr 24 23:50:55] WARNING[9730]: file.c:553 ast_openstream_full: File /var/lib/asterisk/moh/se2 does not exist in any format |
| 22:51.51 | anthm | hi |
| 22:52.27 | BSD_Tech | <=== Formerly Darwin35 |
| 22:52.34 | BSD_Tech | longtime no speek |
| 22:53.24 | SwK | darwin you still working at teliax? |
| 22:53.32 | BSD_Tech | nope |
| 22:53.43 | BSD_Tech | not worked there for over a year |
| 22:53.54 | BSD_Tech | long story |
| 22:54.04 | SwK | so thats why their service got better :P |
| 22:54.16 | BSD_Tech | hell if I know I dont use them |
| 22:54.19 | SwK | heheheh |
| 22:55.00 | BSD_Tech | tight now I use RNK |
| 22:55.29 | BSD_Tech | and a sangoma tdm card |
| 22:56.01 | BSD_Tech | I kinda went anti voip for the last 7 months |
| 22:56.18 | BSD_Tech | and now I am making a swing back to it |
| 22:56.28 | BSD_Tech | but never gave up on asterisk |
| 22:57.43 | plasmid | what configuration file do I need to manually edit so that I can accept 5 incoming calls and make 5 outbound calls on the same DID#? |
| 22:57.47 | BSD_Tech | and I am looking at VP but they dont use paypal |
| 22:57.59 | jm|laptop | ok now it's SO lying |
| 22:58.03 | jm|laptop | [Apr 24 23:59:21] WARNING[9789]: res_musiconhold.c:247 ast_moh_files_next: Unable to open file '/var/lib/asterisk/moh/se2': No such file or directory |
| 22:58.04 | _VoiceMeUp_COM | darnwin = rich ? |
| 22:58.05 | anthm | SwK, that other software runs on the n800 now |
| 22:58.21 | jm|laptop | macintel:/var/lib/asterisk/moh# ls -lh /var/lib/asterisk/moh/se2 |
| 22:58.21 | jm|laptop | -rw-r--r-- 1 jamie jamie 11M 2007-04-24 23:58 /var/lib/asterisk/moh/se2 |
| 22:58.38 | BSD_Tech | hey Frank |
| 22:58.40 | flewid | maybe it needs the ext? |
| 22:59.23 | jm|laptop | -rwxrwxrwx 1 asterisk asterisk 11M 2007-04-24 23:58 se2 |
| 22:59.26 | jm|laptop | now I know it's lying |
| 23:02.02 | BSD_Tech | ? |
| 23:02.15 | BSD_Tech | what was it your thumb in the pipe |
| 23:03.36 | jm|laptop | mode=mp3 :/ |
| 23:03.42 | BSD_Tech | lol |
| 23:03.50 | jm|laptop | not sure what mode=files filetypes are supported |
| 23:03.53 | jm|laptop | I made it a .wav! |
| 23:03.55 | BSD_Tech | gee helps to read the conf files |
| 23:04.04 | BSD_Tech | aaaa |
| 23:07.10 | BSD_Tech | friends , Europiens, Americans, Fellow Canadians lend us your dial plans |
| 23:07.22 | Hmmhesays | heh |
| 23:07.27 | BSD_Tech | let us merge to make a fully functional dialplan |
| 23:07.43 | BSD_Tech | for all to use and enjoy |
| 23:08.05 | jm|laptop | ffs |
| 23:08.09 | jm|laptop | now my IAX isn't working :| |
| 23:08.14 | jm|laptop | it's reporting busy when it isn't |
| 23:08.19 | BSD_Tech | hmm leg of hmmmesays for dinner |
| 23:08.37 | jm|laptop | <PROTECTED> |
| 23:08.38 | jm|laptop | <PROTECTED> |
| 23:08.38 | jm|laptop | <PROTECTED> |
| 23:08.38 | jm|laptop | <PROTECTED> |
| 23:09.05 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 23:09.16 | jm|laptop | I'm beginning to wish I had never updated :S |
| 23:09.43 | flewid | so i'm assuming that getting a cisco 7970 with ast 1.2 is just not possible? |
| 23:09.48 | flewid | i'll have to upgrde the remote box to 1.4? |
| 23:09.51 | BSD_Tech | yes it is |
| 23:09.57 | russellb | 1.4 should work with it |
| 23:10.00 | BSD_Tech | is it sip or mgcp |
| 23:10.05 | flewid | 1.4 works i know, i have it working here |
| 23:10.08 | BSD_Tech | 1.4 should work fine |
| 23:10.12 | flewid | but it's j ust this remote 1.2 box that's being the pain |
| 23:10.16 | flewid | and yah, it's sip |
| 23:10.51 | flewid | i have the phone behind nat here, and it reg's fine to the local 1.4 server, but registering over the net to a remote box (with no nat) won't work |
| 23:10.57 | flewid | sip 401 unauth |
| 23:11.22 | *** join/#asterisk Gpl_Source (n=The_natu@unaffiliated/gplsource) |
| 23:12.17 | BSD_Tech | thats ugly |
| 23:12.27 | jm|laptop | [Apr 25 00:13:46] NOTICE[9673]: chan_iax2.c:1744 iax2_destroy: Avoiding IAX destroy deadlock |
| 23:12.29 | jm|laptop | Jebus |
| 23:12.39 | flewid | bsd: what is |
| 23:13.19 | flewid | brb i need to get a coffee and a cigarette this is driving me nuts |
| 23:14.19 | BSD_Tech | life |
| 23:15.05 | russellb | jm|laptop: you can ignore that ... it should be a debug message |
| 23:15.41 | jm|laptop | russellb: sure. But the IAX call keeps retrying and I get a tiny ring from my phone each time :S |
| 23:17.38 | jm|laptop | <PROTECTED> |
| 23:17.38 | jm|laptop | <PROTECTED> |
| 23:17.45 | jm|laptop | how does that even make sense?! |
| 23:18.05 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX -=- Asterisk 1.2.18, 1.4.3, and Zaptel 1.2.17, 1.4.2 (April 24, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/#trixbox for trixbox support. |
| 23:18.26 | BSD_Tech | 1.4.3 already |
| 23:18.28 | BSD_Tech | wow |
| 23:18.38 | russellb | it has been a month since 1.4.2 |
| 23:19.03 | puzzled | ugh, I just built 1.2.17 |
| 23:19.10 | russellb | tarballs haven't made it to the ftp site yert |
| 23:19.22 | russellb | getting there ... |
| 23:19.41 | jm|laptop | I don't underSTAND :( |
| 23:19.45 | Nivex | Wow, you get it here before..... |
| 23:19.57 | russellb | heh, yep |
| 23:20.13 | russellb | waiting for some other developers to sign the tarballs with their gpg keys ... |
| 23:20.59 | jm|laptop | I can't even call MYSELF via IAX now :S |
| 23:22.40 | Strom_M | Look on the bright side - talking to yourself is really boring |
| 23:22.48 | Nivex | jbot: chan_cellphone? |
| 23:23.13 | flewid | hm |
| 23:23.21 | flewid | maybe 1.2.18 will fix the cisco :) |
| 23:23.29 | russellb | heh |
| 23:23.37 | russellb | svn co http://svn.digium.com/svn/asterisk/tags/1.2.18 |
| 23:24.12 | Nivex | oh bother, what was that bug # ? |
| 23:25.13 | Nivex | praise the wiki! |
| 23:26.38 | flewid | meh i think i'll just upgrade to 1.4.x |
| 23:26.40 | flewid | it's a new box |
| 23:26.50 | flewid | i wanted to stay 1.2 cause that's what the gui "supports' |
| 23:26.54 | flewid | but 1.4 should be alright |
| 23:28.27 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 23:29.24 | tzafrir | ~chan_cellphone |
| 23:34.20 | *** join/#asterisk cspot (i=cspot@ip68-109-8-207.pn.at.cox.net) |
| 23:48.03 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:51.30 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 23:52.00 | JerJer[mobile] | ~chan_dishwasher |
| 23:52.04 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-70-232-29-69.dsl.hstntx.sbcglobal.net) |
| 23:53.47 | krapper | anyone seen where asterisk 1.4.2 will all of the sudden it's loses it's global variables? |
| 23:53.56 | krapper | and a dialplan reload is required |
| 23:54.23 | tzafrir | krapper, anything from the manager interface? any explicit set? |
| 23:55.10 | krapper | well, we a global variable for all phones ${EXTALL} and a global variable for each extension ${EXT100} etc... |
| 23:56.50 | krapper | exten => s,n,Dial(${EXTALL}) |
| 23:57.06 | krapper | EXTALL=${EXT100}&${EXT101}&${EXT102}&${EXT103}&${EXT104} <--- global variable |
| 23:57.09 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.169.195) |
| 23:57.22 | krapper | then all of the sudden today... -- Executing [s@incoming-cox:3] Dial("Zap/1-1", "&&&&") in new stack |
| 23:57.32 | krapper | followed by of course a dial application error |
| 23:58.25 | krapper | dialplan reload started working like a charm |
| 23:58.55 | krapper | box has been running with no problems for about a week and a half prior to today |