| 00:17.14 | *** join/#asterisk alexmontoanelli (n=alexm@alexmm.unetvale.com.br) |
| 00:28.57 | *** join/#asterisk njan (n=james@about/security/staff/njan) |
| 00:34.30 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 00:36.03 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 00:40.51 | *** join/#asterisk xanderp (n=xanderp@74.133.15.46) |
| 00:44.03 | *** join/#asterisk Mad_guy (n=austin@ip68-1-213-212.dl.dl.cox.net) |
| 00:45.09 | *** join/#asterisk ShipHead (i=ShipHead@gateway/tor/x-c930ad3e5e6a9599) |
| 00:49.32 | *** join/#asterisk MartianLobster (n=clarks@m815f36d0.tmodns.net) |
| 00:50.12 | MartianLobster | I have an asterisk server (actually trixbox) and I logged in via ssh. How can I check the log of incoming phone calls? |
| 00:50.34 | DrkShdw | MartianLobster: 'asterisk -rvvvvv |
| 00:50.40 | DrkShdw | err, remove the ' |
| 00:51.44 | MartianLobster | so I type "asterisk -rvvvvv" ? |
| 00:52.06 | MartianLobster | what does the vvvvv do after the -r ? |
| 00:53.24 | mog | verbose |
| 00:53.35 | mog | adds more verbosity |
| 00:53.37 | *** join/#asterisk MoutaPT (n=MoutaPT@a213-22-41-37.cpe.netcabo.pt) |
| 00:54.13 | MartianLobster | ok so I started the asterisk cli |
| 00:55.00 | MoutaPT | Hi i've been trying to compile and install cdr_addon_mysql.c from asterisk-addons on Asterisk 1.4 but i never get cdr_addon_mysql.so module ... Mysql is running fine |
| 00:55.05 | MoutaPT | any guess? |
| 00:55.15 | MartianLobster | so now I do "show history" ? |
| 00:55.58 | xanderp | anyone here used the DPH-549 wifi sip phone from D-link? wondering if it's good. |
| 00:57.02 | MartianLobster | nope, that isn't what I should do. How do I print the history of incoming phone calls once I have started the asterisk cli? |
| 00:57.08 | DrkShdw | MartianLobster: just call it. you'll see the incoming call |
| 00:57.17 | DrkShdw | oh, you want the history? |
| 00:57.26 | *** join/#asterisk Tili (n=tili@202.133.65.150) |
| 00:57.41 | teknoprep | is there a metermaid patch for 1.2.12.1 ? |
| 00:58.01 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 00:58.13 | MartianLobster | DrkShdw: yes I missed a call today around 2:00pm, I want to see who called |
| 00:58.15 | DrkShdw | MartianLobster: trixbox has a CDR somewhere, in the web interface. I forget where |
| 00:58.55 | teknoprep | its under the freepbx section |
| 00:59.26 | MartianLobster | I don't have access to the web interface from here. I am offite. I can ssh to the box and that is it |
| 00:59.51 | xanderp | i have a newbie question. I would like to call from a wifi sip phone from the internet into my asterisk box, but don't want anyone else to be able to call through my asterisk box from the Internet. What do I need to do to secure the asterisk box to only be used by MY devices to make calls out my trunks? |
| 01:00.37 | denon | set a very complex secret |
| 01:00.46 | denon | and dont allow any guest access |
| 01:01.06 | xanderp | ok, so it's a matter of hoping someone doesn't do a brute force against the secret. |
| 01:01.33 | denon | pretty much .. if you cant do a vpn, and you dont know what ip block they're coming from |
| 01:01.46 | denon | the phone probably wont support any kind of secure auth |
| 01:01.51 | MoutaPT | MartianLobster try this: /var/log/asterisk/cdr-csv/ |
| 01:01.52 | denon | so a complex secret it is |
| 01:01.53 | *** part/#asterisk njan (n=james@about/security/staff/njan) |
| 01:02.06 | MoutaPT | there you will find at least csv |
| 01:02.12 | MoutaPT | or log into mysql |
| 01:02.18 | xanderp | i'm thinking that wifi phones aren't going to be powerful enough to run vpn and voip at the same time. |
| 01:02.19 | MoutaPT | and query it |
| 01:02.24 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 01:02.39 | xanderp | especially if they are having to do WPA encryption to the AP at some hotspot... |
| 01:02.40 | denon | xachen: if the phone has a web browser, you could do something cheezy like make them hit a page in their favorites, that requires SSL auth .. then that page could update your firewall with the IP of the phone |
| 01:03.05 | denon | that'd be reasonably secure |
| 01:03.15 | xanderp | denon: good thought |
| 01:03.28 | denon | you would, of course, want to make sure that it happens over ssl |
| 01:03.38 | denon | so that whoever's running the hotspot cant do it again later |
| 01:03.48 | xanderp | I can't believe that they "gurus" haven't done more to secure this stuff. |
| 01:03.50 | denon | and you'd want to expire that firewall rule after a few hours or day |
| 01:04.11 | denon | xanderp: we've done stuff like this .. it's a bit cobbled though |
| 01:04.15 | denon | you're much better off with a real vpn |
| 01:04.25 | denon | and it takes a bit of user education |
| 01:04.43 | xanderp | i've got a good vpn, but i doubt any wifi phones will have a vpn client! |
| 01:04.45 | denon | some users are lucky to dial a phone .. making them browse to a web page AND dial would be unheard of |
| 01:05.00 | MartianLobster | MoutaPT: hmmmmmm somehow I don't know the mysql password :( dang |
| 01:05.48 | denon | xanderp: well, sip has some secure extensions .. |
| 01:05.48 | denon | that's where things will head eventually |
| 01:05.48 | denon | but you still need to leave a port open .. |
| 01:05.48 | xanderp | so i should just sit and wait. :( oh well... |
| 01:05.48 | denon | which leaves you at risk |
| 01:05.48 | Jason99 | Hello, I'm using the manager API and when doing an Action: Originate you can only place one call at a time. Does anyone know how to bypass this? |
| 01:05.48 | denon | xanderp: what's wrong with my idea? if you have a bright user .. |
| 01:05.57 | MoutaPT | MartianLobster, is this a kind of Newbie install with default pass? |
| 01:06.01 | xanderp | denon: the web cert? |
| 01:06.08 | denon | yeah |
| 01:06.09 | xanderp | denon: nothing, great idea. |
| 01:06.10 | MartianLobster | MoutaPT: it probably is |
| 01:06.16 | denon | so why wait? :) |
| 01:06.24 | MoutaPT | mysql -u roo -p |
| 01:06.29 | MoutaPT | mysql -u root -p |
| 01:06.37 | MoutaPT | try then passw0rd |
| 01:07.09 | xanderp | work work work... ;) i was hoping someone had made it brainless for me already. i have to think about security all day at work. I like to sit in the corner and drool after hours :) |
| 01:07.09 | MartianLobster | MoutaPT: you got it, ^5 |
| 01:07.29 | denon | xanderp: so which wifi phones are you using? |
| 01:07.29 | *** part/#asterisk alexmontoanelli (n=alexm@alexmm.unetvale.com.br) |
| 01:07.31 | MoutaPT | you probalby can get it from web portal |
| 01:07.40 | MoutaPT | if your http is running |
| 01:07.53 | xanderp | none yet... that's the rub. I don't want to buy anything until i figure out how to secure it, and if the phone will support it. |
| 01:08.16 | MartianLobster | MoutaPT: ok this is great, that is exactly what I need thank you so much |
| 01:08.19 | xanderp | I was thinking about the DPH-549 from D-link |
| 01:08.35 | MoutaPT | u r welcome |
| 01:08.38 | xanderp | it looks 'sturdy' |
| 01:08.55 | denon | xanderp: have you looked at the linksys WIP330s? |
| 01:09.06 | xanderp | will do so now... |
| 01:09.08 | MoutaPT | Is there any one here that used volgain_6237_u1.patch ?? |
| 01:09.14 | teknoprep | w0ot |
| 01:09.20 | MoutaPT | I'm not used with aplying patches |
| 01:09.36 | MoutaPT | i found a problem running the patch |
| 01:10.00 | MoutaPT | but i think is only the reference about the diff that's wrong, is only when 2 arrays are defined |
| 01:10.20 | MoutaPT | i think i can get into C file and define those two |
| 01:10.33 | MoutaPT | any one experienced on this? |
| 01:10.41 | *** join/#asterisk saftsack (n=saftsack@p54A7DDA5.dip.t-dialin.net) |
| 01:11.19 | saftsack | hi if i dial a number on my phone i dont hear a tone when pressing a button on the phone. what do i need to hear this? (its an analog phone) |
| 01:14.26 | *** join/#asterisk hyphen (n=hyphen@71.224.213.97) |
| 01:16.18 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 01:18.49 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 01:21.40 | JT | well i doubt it'd be more reliable in most cases :P |
| 01:22.43 | xanderp | hehe |
| 01:23.00 | denon | xanderp: actually, in many cases a sip desktop phone is MUCH cheaper than, say, a nortel desk phone |
| 01:23.08 | denon | or pretty much any proprietary digital phone |
| 01:23.22 | denon | you can get a nice sip phone under $150 |
| 01:23.28 | denon | PoE even |
| 01:23.36 | JT | yeah but then compare that to an analogue home phone cheapie |
| 01:23.39 | denon | and with a switch on the back, saving them another eth run for the phone |
| 01:23.40 | xanderp | denon: i'm talking to home use |
| 01:23.42 | denon | er computer |
| 01:23.43 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 01:23.48 | denon | oh, home users are a waste |
| 01:23.49 | justdave | yeah, for office phones, SIP is quite cost-effective. Doesn't seem to have effectively hit the home market yet though |
| 01:24.02 | denon | home users whine about there $25/mo phone service |
| 01:24.09 | denon | their |
| 01:24.11 | EyeCue | jt :~) |
| 01:24.13 | justdave | SIP phones for the home are darn expensive. Even ATA boxes are. |
| 01:24.21 | JT | hi EyeCue |
| 01:24.33 | xanderp | they need to get unlocked sip phones in the electronics stores so the home users can drive down the price. |
| 01:24.40 | JT | i don't have any sip phones at home |
| 01:24.49 | JT | just a T1 channel bank, good enough :P |
| 01:24.51 | denon | nobody knows what to do with a sip phone, if it isnt tied to vonage |
| 01:24.56 | denon | home users are too stupid |
| 01:25.13 | *** join/#asterisk tuck3r (n=tuck3r@unaffiliated/tuck3r) |
| 01:25.28 | xanderp | all the general public has access to easily are locked to vonage/packet8/etc... stuff... Fry's Electronics now has the dlink wifi phone that's not locked. |
| 01:25.36 | justdave | but when you get phone service over SIP for $14.95/mo with free long distance to the entire state (or all of north america in some cases) you make up for the $80 you paid for your phone in less than a year of phone bills. |
| 01:26.18 | denon | vonage/skype/etc has given voip a bad name, "yeah, I tried that voip thing .. it didnt sound very good" |
| 01:26.27 | justdave | an $80 SIP phone is a piece of crap though. |
| 01:26.40 | tuck3r | denon: no, their shitty internet did |
| 01:26.49 | denon | tuck3r: bit of each .. |
| 01:27.03 | xanderp | justdave: my problem is moderation. I don't understand it. I would want like 8-10 sip phones scattered all over the house, and a few wifi handsets sprinkled on top. |
| 01:27.05 | denon | even great Internet connections still aren't 100% for voip |
| 01:27.21 | tuck3r | whats wrongs with skype or vonage on a good connection? |
| 01:27.34 | justdave | I got a couple $80 Grandstream phones in my house. Sound quality is great. Usability not so much. |
| 01:27.35 | denon | tuck3r: the public Internet isnt designed for voip |
| 01:27.42 | denon | it's just not |
| 01:27.44 | xanderp | tuck3r: unreliable |
| 01:27.51 | tuck3r | denon: true, but it still works |
| 01:27.56 | denon | you can have an empty OC3 and you still will have issues |
| 01:28.02 | EyeCue | works and works well enough are different things |
| 01:28.18 | denon | it may work for the few hundred minutes you use it for .. |
| 01:28.22 | denon | but that's just luck |
| 01:28.34 | justdave | found it difficult to make them behave like a normal household phone. Trying to teach people who have never used a cell phone how to hit "Send" after dialing the number is fun. :) |
| 01:28.47 | xanderp | the wife has 'suffered' me to have vonage for several years... I just have to get the asterisk online and stable and I can move us to a better provider. |
| 01:28.52 | denon | justdave: just let dialplans and timeouts do the work |
| 01:28.56 | denon | dont bother making them learn new tricks |
| 01:29.05 | tuck3r | vonage/skype are home provider, so "well enough" is a enough |
| 01:29.14 | justdave | denon: yeah, but the phones time out too quick. |
| 01:29.23 | denon | justdave: then use a dialplan |
| 01:29.29 | denon | if they're dialing LD, they'll start with 1 .. |
| 01:29.34 | xanderp | justdave: change the timeout? |
| 01:29.35 | denon | if they're dialing local, expect XXXXXXX |
| 01:29.43 | justdave | If I turn on autodial and let asterisk do the work, then they're sitting there wondering why they got a busy signal after dialing 2 digits |
| 01:29.45 | denon | if it's 1, assume 1XXXXXXXXXX |
| 01:29.53 | denon | if it's 0, look for 011. |
| 01:30.05 | justdave | denon: right, but these phones don't let you play with the dialplan |
| 01:30.14 | denon | ug |
| 01:30.24 | xanderp | justdave: you do that on the server, not the phone |
| 01:30.25 | denon | good, fast, cheap - pick 2 |
| 01:30.30 | JT | perhaps skype is unreliable |
| 01:30.33 | denon | xanderp: no, im talking about doing it on the phone |
| 01:30.34 | justdave | all they do is attempt to dial what you've typed so far and keep waiting for more digits as long as asterisk says "that doesn't work yet" |
| 01:30.35 | JT | but voice quality is fairly good |
| 01:30.57 | xanderp | oh, gotcha |
| 01:31.06 | justdave | (or until it times out, which is about 2 seconds and doesn't appear to be adjustable) |
| 01:31.13 | denon | xanderp: you can, of course, do both .. but doing a good dialplan on the phone makes it easier before sip gets ahold of it |
| 01:31.49 | justdave | I also have a polycom phone, and that one lets me put a dialplan in the phone itself, so it won't even both sending it to asterisk until it knows it has a dialable number |
| 01:31.50 | tuck3r | any reason deny=0.0.0.0/0.0.0.0, permit=192.168.1.1 is letting 192.168.1.2 in? |
| 01:32.01 | xanderp | justdave: so the phone doesn't 'dial' the server until you hit send, or the phone's dialplan thinks it knows what to do right? |
| 01:32.13 | tuck3r | this is slightly making me mad |
| 01:32.28 | CrazyTux | Hey, does anyone in here know if you can control the number of rings, instead of number of 'seconds to ring'. |
| 01:32.47 | tuck3r | no, you cant |
| 01:32.47 | justdave | xanderp: right, for the polycom. For the Grandstream phones, it either attempts to dial after each digit, or waits until you hit send. |
| 01:32.56 | CrazyTux | tuck3r, why not, companies such as verizon, and sprint do this? |
| 01:33.09 | justdave | SIP phones don't have a concept of ring counts. that's a POTS thing. |
| 01:33.15 | tuck3r | asterisk != verzion? |
| 01:33.20 | justdave | for a SIP phone, it's either been invited to the call or it hasn't. |
| 01:33.42 | CrazyTux | justdave, Any algorithm to match it up as close as to x count of rings? |
| 01:33.45 | justdave | how long the ring lasts is a cosmetic thing on the phone itself |
| 01:33.54 | CrazyTux | justdave, i.e. 5 rings... 2 secs per ring = 10 secs? |
| 01:33.58 | xanderp | i'm wondering if cable providers partnering with voip will have their QOS hacked down to not give the 'best effort' to alternate providers that it does to their partners. |
| 01:34.08 | CrazyTux | justdave, is there any close 'call judgement' |
| 01:34.09 | justdave | CrazyTux: sure, then put in that number of seconds on your dialplan :) |
| 01:34.14 | tuck3r | xanderp: some do, some don't |
| 01:34.45 | justdave | I'm currently using Charter cable, who just recently started offering their own phone service. |
| 01:35.04 | justdave | I've got VoIP to two different places from here (neither through them), and haven't had any degraded service yet. |
| 01:35.09 | justdave | but I'm probably just lucky. |
| 01:35.21 | justdave | one is my home phone service, the other is a remote extension from work |
| 01:36.07 | *** join/#asterisk Mad_guy (n=austin@ip68-1-213-212.dl.dl.cox.net) |
| 01:36.43 | xanderp | i'm wondering if they generally just do QOS against SIP packets, or if they only QOS based on a filter match of destinations to their partners. Could be a very dirty pool to swim in if they did that and anyone found out that they were intentionally degrading service.... |
| 01:37.31 | JT | it'd be QoS against RTP packets more likely |
| 01:38.01 | xanderp | JT: that'd make sense, but we're talking greedy businesses here... :) |
| 01:38.18 | xanderp | i meant RTP, not SIP... sorry |
| 01:38.27 | JT | well QoSing SIP packets wouldn't do that much |
| 01:38.29 | JT | heh |
| 01:38.40 | xanderp | really clean session initiations :) |
| 01:38.51 | xanderp | and then horrible audio ;) |
| 01:39.34 | JT | yeah but it'd hard to catch cases where people are manually controlling stuff to evade any such Qos |
| 01:39.35 | xanderp | i overgeneralize alot. i will say sip to mean VOIP/RTP/SIP/IP Telephony, etc... |
| 01:39.53 | JT | unless they had across the board packet application inspection |
| 01:40.08 | JT | you could use a different range of ports |
| 01:40.20 | JT | may be hard when connecting to a voip provider though |
| 01:41.07 | xanderp | i could imagine just putting QOS against traffic between them and their providers though, and leaving everything else as is... that's actually NOT dirty pool, so I wouldn't be pissed. |
| 01:41.25 | JT | sure |
| 01:42.09 | xanderp | unless they intentionally injected some packet jitter that wouldn't really effect browsers/downloads/etc... on their 'non-QOS' routes. |
| 01:42.20 | JT | yeah |
| 01:42.30 | JT | that's all basically intentional sabotage |
| 01:43.06 | xanderp | why is xanderphillips so apt to see the possible negative of companies? Hmmm maybe a lifetime of experience. |
| 01:43.40 | *** join/#asterisk ShadowHntr (i=sentinel@c-68-52-48-169.hsd1.tn.comcast.net) |
| 01:43.40 | xanderp | although i do want to tell the world that Hauppauge is AWESOME! they have awesome customer support. |
| 01:44.26 | xanderp | i had a tv tuner card that died and they gave me an RMA no questions asked, without even a receipt! they are shipping my replacement card now, free of charge. |
| 01:47.32 | charles___ | Anyone had problems with IVR's getting mute ? |
| 01:47.37 | charles___ | <PROTECTED> |
| 01:47.40 | charles___ | But I don't hear anything |
| 01:50.13 | charles___ | Even tried restarting |
| 01:50.16 | charles___ | but no sound at all |
| 01:52.37 | charles___ | just hear a gap when it starts and then mute |
| 01:55.40 | *** join/#asterisk blebleble (n=ble@d149-67-99-160.col.wideopenwest.com) |
| 01:56.04 | JT | charles___: what channel are you using to listen to it? |
| 01:56.14 | charles___ | JT Sip |
| 01:56.24 | orlock | ngaargh |
| 01:56.28 | JT | have you tried something else? |
| 01:57.17 | *** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com) |
| 01:57.21 | charles___ | JT I only have sIp |
| 01:57.27 | charles___ | no IAX devices |
| 01:57.35 | JT | softphone for testing? |
| 01:57.38 | charles___ | everything sip here |
| 01:58.04 | tuck3r | use a IAX softphone |
| 01:58.28 | JT | charles___: does application Echo() work? |
| 02:00.13 | charles___ | let me see |
| 02:00.21 | charles___ | I can talk on the phone |
| 02:00.28 | charles___ | I can hear the on hold messages also |
| 02:00.40 | charles___ | but Playing 'vm-intro' (language 'en') Does not work |
| 02:01.09 | JT | maybe there is a problem with the sound file |
| 02:01.22 | charles___ | I can play it with play |
| 02:01.26 | JT | try making a backup of it and replacing it with a copy of a sound file that works |
| 02:01.29 | JT | hrm ok |
| 02:01.29 | charles___ | and permissions are good too |
| 02:01.40 | charles___ | not only that no IVR's work |
| 02:01.44 | charles___ | asterisk say playing blah |
| 02:01.49 | charles___ | and reports no error |
| 02:03.32 | charles___ | when I try some files it gives me a GAP and then gets dead |
| 02:06.22 | *** join/#asterisk Winkie (n=urmom@cpc2-stre3-0-0-cust344.bagu.cable.ntl.com) |
| 02:07.43 | charles___ | really weird |
| 02:08.37 | charles___ | JT something with zaptel |
| 02:08.51 | charles___ | JT I reloaded asterisk and runned ztcfg |
| 02:09.07 | JT | works now? |
| 02:11.37 | charles___ | JT yes it does |
| 02:11.39 | charles___ | very weird |
| 02:16.45 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 02:16.45 | *** mode/#asterisk [+o mog] by ChanServ |
| 02:18.57 | teknoprep | i have the meter maid patch installed on 1.2.12.1 |
| 02:19.22 | teknoprep | when i do an restart on asterisk inside the CLI it no longer works for BLF of parked calls |
| 02:21.09 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 02:22.10 | findlay | why is is that some numbers I call the other person can't hear me at all yet others numbers I call the other person can hear me just fine? |
| 02:23.47 | quid246 | Anybody know of a billing module/app for Asterisk? |
| 02:27.06 | carrar | SELECT * FRMO cdr; |
| 02:27.11 | carrar | SELECT * FROM cdr; |
| 02:27.13 | carrar | heh |
| 02:27.37 | JT | charles___: what do you use zaptel for, just timing? |
| 02:28.08 | carrar | quid246, check out http://www.voip-info.org/wiki-Asterisk |
| 02:28.27 | quid246 | Dude, I know that site inside and out. |
| 02:28.45 | charles___ | JT I used to have ztdummy but today I installed a te110p |
| 02:28.47 | carrar | then you know the general ones then |
| 02:29.00 | carrar | oh |
| 02:29.02 | carrar | module |
| 02:29.07 | quid246 | Yeah, somethig has to be better than AGI. |
| 02:29.10 | carrar | was thinking of stand alone software |
| 02:29.15 | quid246 | Yeah module as in .so |
| 02:29.20 | carrar | heh |
| 02:29.33 | carrar | wouldn't that be nice |
| 02:29.49 | quid246 | Yeah, or just wait for FreeSwitch. |
| 02:29.52 | carrar | so you can tag it to keep the incoming did all the way through |
| 02:29.55 | JT | charles___: i'd say that'd have something to do with it |
| 02:42.22 | charles___ | JT now working fine |
| 02:42.24 | charles___ | JT thanks |
| 02:42.38 | *** join/#asterisk dieguito (n=diego@201.230.106.54) |
| 02:42.42 | *** join/#asterisk Libila (n=vye@ip68-101-240-176.sd.sd.cox.net) |
| 02:43.46 | dieguito | hello, I think I read that it's possible to turn a modem card into a card that asterisk can use, did I read wrong or is this possible? |
| 02:47.35 | carrar | http://www.voip-info.org/wiki/view/X100P+clone |
| 02:48.13 | carrar | yes, some cards you can |
| 02:48.55 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 02:48.59 | dieguito | carrar: thanks. I'll take a look :) |
| 02:57.09 | teknoprep | in 1.4 do parked hints work better then the patch for 1.2 ? |
| 02:57.19 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 02:57.19 | *** mode/#asterisk [+o mog] by ChanServ |
| 02:57.23 | teknoprep | the patch for 1.2 makes all my lights light up on first boot of the phone |
| 02:57.42 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 02:58.30 | *** join/#asterisk _Vile (i=_Vile@198.175.14.242) |
| 03:02.29 | *** join/#asterisk aptura (n=s_a_l_e_@S010600a0c93f6f7e.vs.shawcable.net) |
| 03:16.57 | *** part/#asterisk hyphen (n=hyphen@71.224.213.97) |
| 03:19.16 | *** join/#asterisk SECGOD (i=SECGOD@c-71-57-36-106.hsd1.il.comcast.net) |
| 03:29.36 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 03:36.40 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 03:37.49 | *** join/#asterisk HaMYaI (i=HaMYaI@202.8.86.164) |
| 03:38.06 | HaMYaI | anyhere using ooh323 of oh323? |
| 03:38.29 | HaMYaI | s/anyhere/anyone here |
| 03:40.40 | HaMYaI | I'd like to know what to use in order to restrict incoming connections like using "host" or "ip" configuration |
| 03:50.44 | *** join/#asterisk MercRey (n=rcflyr@66.17.33.138) |
| 03:51.48 | MercRey | Anyone know if SLA is working enough for testing in 1.4 beta ? |
| 03:53.35 | MercRey | no one here? |
| 03:55.11 | mog | dont use it yet |
| 03:55.18 | mog | sla that is |
| 03:55.45 | MercRey | sad :( |
| 03:55.50 | MercRey | why is that? |
| 03:56.52 | mog | the current implementation does not allow for several cases to act as intended |
| 03:58.32 | MercRey | is there any documentation on it yet? or will that come after a working implementation is released? |
| 03:58.54 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 03:58.59 | mog | documentation is ther |
| 03:59.00 | mog | e |
| 03:59.10 | mog | in configs/sla.conf |
| 04:00.24 | *** join/#asterisk ltd (n=z@202-161-26-249.dyn.iinet.net.au) |
| 04:00.25 | MercRey | I have seen that, but I don't seem to have much luck while playing with it. |
| 04:00.56 | MercRey | I have someone that's been wanting SLA for a while so I have been watching and waiting to play with it. |
| 04:01.34 | mog | yup |
| 04:01.44 | mog | its a frequently requested feature |
| 04:02.33 | sbingner | heh I need to add a "database show" command to the manager interface |
| 04:02.38 | sbingner | <-- been lazy |
| 04:03.32 | *** join/#asterisk HaMYaI_ (n=HaMYaI@ppp-58.8.13.67.revip2.asianet.co.th) |
| 04:03.57 | MercRey | well, thanks for the feedback. I really wish I could get even an unfinished example working to play with. |
| 04:03.59 | sbingner | does unixODBC leave a bad taste in anybody else's mouth? ... ODBC just sounds evil heh, but I haven't used it so that's not really fair... heh |
| 04:04.04 | JT | what does sla do? |
| 04:04.13 | sbingner | service level agreement? |
| 04:04.27 | HaMYaI_ | any one here has an experience with h323 in asterisk? |
| 04:04.35 | mog | <PROTECTED> |
| 04:04.52 | sbingner | a keysystem? |
| 04:05.02 | JT | a keysytem pabx |
| 04:05.14 | mog | cheapo oldschool pbx |
| 04:05.20 | JT | mog: what does it do that asterisk doesn't previously provide? |
| 04:05.42 | HaMYaI_ | I'm wondering which one is a more featureful btw ooh323 and oh323 |
| 04:06.30 | mog | http://en.wikipedia.org/wiki/Key_telephone_system |
| 04:06.45 | sbingner | http://forums.digium.com/viewtopic.php?t=891 |
| 04:06.47 | mog | its basically each phone has access to x number of lines |
| 04:06.53 | mog | and they all have access to same lines |
| 04:07.22 | file | mog! |
| 04:07.27 | sbingner | purpose: put on hold on any phone, pick up on any other phone |
| 04:07.27 | mog | file! |
| 04:07.46 | file | mog: zomg 11 days |
| 04:08.00 | mog | i know |
| 04:08.10 | *** join/#asterisk jusse (n=jusse@201.240.241.199) |
| 04:08.13 | jusse | hola??? |
| 04:08.15 | mog | you speaking this year file? |
| 04:08.18 | mog | hiya justdave |
| 04:08.19 | file | I am, with Russell |
| 04:08.21 | mog | er jusse |
| 04:08.26 | jusse | algien habla espaņol |
| 04:08.29 | mog | about? |
| 04:09.17 | file | we're doing development stuff, optimizations in 1.4, some other stuff |
| 04:09.29 | jusse | somebody speack spanish??? |
| 04:09.56 | file | mog: are you going to speak jingle? |
| 04:09.58 | file | :D |
| 04:10.01 | mog | yes |
| 04:10.04 | mog | and jabber |
| 04:10.14 | mog | some of that stuff we talked about asterdevcon europe |
| 04:10.18 | carrar | Nihongo ga sukoshi wakarimasu |
| 04:10.39 | file | I'm looking forward to the development summit too |
| 04:10.41 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 04:10.50 | mog | ditto |
| 04:11.06 | file | but there won't be any gelato :\ |
| 04:11.41 | justdave | I speak Spanish enough to know that the systran translator is probably screwed up in telling me to say this: No hablo probablemente espaÃąol bien bastante a la ayuda con un problema |
| 04:11.55 | justdave | but that's about it :) |
| 04:12.41 | MercRey | anyone know when SLA should be functional for testing? |
| 04:13.45 | mog | MercRey, kevin has planned how to deal with it but has been too busy to write the code |
| 04:14.22 | mog | it will be working in the 1.4 final release at the latest |
| 04:14.32 | MercRey | I wish I knew more about this area or I would offer to help with some coding... :( |
| 04:14.48 | MercRey | thanks again. |
| 04:14.58 | mog | no problem MercRey |
| 04:16.19 | [TK]D-Fender | mog : is the "october" expected release date still feeling viable so far? |
| 04:16.42 | sbingner | MercRey: ony way to learn :b |
| 04:16.54 | mog | i think we need more testing there have not been an abundant ammount of bug reports regarding it |
| 04:16.58 | mog | which is discouraging |
| 04:17.04 | mog | we should have another beta soon |
| 04:17.21 | sbingner | mog: I'd have to have a specific SIP phone though, yes? |
| 04:17.34 | sbingner | or you mean 1.4 in general |
| 04:17.35 | mog | for? |
| 04:17.50 | sbingner | heh |
| 04:18.03 | mog | im confused as to your question |
| 04:18.23 | sbingner | mog: you said you needed more feedback but I thought you meant in SLA code not just general 1.4 code |
| 04:18.47 | mog | 1.4 in general i wish we were getting more test reports |
| 04:19.01 | mog | i know at least in my section of code there is still more work to be done in bugs zone |
| 04:19.53 | sbingner | I'll load it up anyways, the constant depreciation of commands is annoying lol mostly why I didn't (yes, stupid reason--I just don't like it whining that I need to type the command THIS way instead) |
| 04:20.23 | mog | well its just a warning , in 6 months or so said commands wont exist |
| 04:20.43 | *** part/#asterisk dieguito (n=diego@201.230.106.54) |
| 04:20.58 | sbingner | sip show peers / sip list peers --- why bother removing the old one -- not like it's really going to hurt anything ;) |
| 04:21.19 | mog | because its not uniformed |
| 04:21.22 | sbingner | sorry, I like cisco heh |
| 04:21.27 | mog | better to have everything make since |
| 04:21.31 | sbingner | then why isn't iax2 show peers depreciated? |
| 04:21.34 | file | sense |
| 04:21.36 | mog | then have 8million ways to execute same command |
| 04:21.44 | mog | it should be then sbingner |
| 04:21.48 | mog | report a bug ^_^ |
| 04:21.51 | sbingner | heh |
| 04:22.00 | *** part/#asterisk DasTech (n=DasTech@ppp-71-128-114-49.dsl.irvnca.pacbell.net) |
| 04:22.26 | mog | seriously |
| 04:22.27 | sbingner | it'd make more sense for all the info to be show like on cisco IMO :b "show iax2 peers" |
| 04:22.33 | sbingner | I know |
| 04:22.51 | sbingner | but I had this conversation like 2 years ago, and it's not gonna happen |
| 04:23.18 | sbingner | I'll submit a bug when I upgrade and verify that -- I'm not sure it's not depreciated since that was from looking at somebody else's system |
| 04:23.44 | mog | im just saying if its not consistant verbage its a bug |
| 04:24.14 | sbingner | I agree with that, I just dont see the real advantage of changing one that made perfect sense :b |
| 04:24.35 | mog | well the problem was it was inconsistant across the board |
| 04:24.40 | mog | like show meetme |
| 04:24.44 | sbingner | just more ways of saying something to remember -- if you use show for list and anything that shows info, it's easier to remember |
| 04:24.47 | mog | instead of meetme show conferences |
| 04:24.54 | sbingner | yea, that's bad |
| 04:25.00 | mog | there was a good bit of that |
| 04:25.11 | mog | so while fixing that Qwell also fixed the verbage |
| 04:25.20 | sbingner | that's what |
| 04:25.34 | sbingner | I disagree with :b -- but hey |
| 04:26.00 | mog | not my choice either though ^_^ i dont like the core blah instead of blah for the core commands |
| 04:26.07 | mog | but overall id say its better now |
| 04:26.41 | sbingner | you're probably right there... I just don't want to see it get where there's a COMPLETELY different command for everything |
| 04:27.36 | sbingner | sip list peers / sip show peer / sip info peerspeed (making last up) |
| 04:27.41 | mog | it is the first time we have done this in 5 years so its pretty good run so far |
| 04:28.58 | sbingner | unless you count the messing with apps/variables(functions) :b |
| 04:29.00 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-42.ph.ph.cox.net) |
| 04:29.06 | kronic | anyone setup dynamic realtime for queueing and queue members? |
| 04:29.16 | kronic | on 1.2.12 that is |
| 04:30.14 | sbingner | kronic: not me, sorry |
| 04:30.39 | mog | whats your question kram |
| 04:30.42 | mog | er kronic |
| 04:31.20 | HaMYaI_ | any one here has an experience with h323 in asterisk? |
| 04:31.26 | HaMYaI_ | I'm wondering which one is a more featureful btw ooh323 and oh323 |
| 04:31.27 | kronic | I've setup dynamic realtime for queue_members via odbc into a mysql db, I can't get it working (in the sense I have rows in the table, but show queues has no members) |
| 04:31.39 | kronic | I was "told" it doesn't work in this version |
| 04:32.05 | sbingner | mog: if I present a good case, think anybody would consider using show everywhere instead of list? -- I mean which one with be show stats? It's plural, but are they really a list? |
| 04:32.11 | HaMYaI_ | what I need is to restrict in-coming connection by their ip addresses |
| 04:32.23 | mog | o dp |
| 04:32.25 | mog | er i do |
| 04:32.32 | mog | i think it would preferred at least to me |
| 04:32.50 | mog | and even if you dont succeed the intentions will be more clearly layed out |
| 04:33.04 | mog | i would reccomend asterisk-dev mailinglist for that question |
| 04:33.14 | sbingner | mog: aiight, thanks |
| 04:33.26 | kronic | I'm able to get an odbc connection, yet nothing is pulled from the db |
| 04:39.06 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:54.35 | *** join/#asterisk shellshark (n=x86@get.hooked.on.voip.with.shellshark.net) |
| 04:55.29 | *** part/#asterisk Tili (n=tili@202.133.65.150) |
| 05:01.26 | *** join/#asterisk markhubbard (n=markhubb@adsl-226-140-231.int.bellsouth.net) |
| 05:02.20 | stephane_ | jour |
| 05:02.35 | sbingner | bon |
| 05:04.37 | *** part/#asterisk markhubbard (n=markhubb@adsl-226-140-231.int.bellsouth.net) |
| 05:08.57 | kronic | mog: was that asterisk-dev mailing list suggestion for me? |
| 05:09.11 | mog | no |
| 05:09.33 | kronic | any suggestions? |
| 05:09.38 | kronic | as I'm really stuck |
| 05:13.16 | X-Rob_ | I wouldn't use realtime then, if you're really stuck |
| 05:13.17 | X-Rob_ | 8) |
| 05:14.28 | *** join/#asterisk jusse (n=jusse@200.121.165.178) |
| 05:14.32 | jusse | Hi |
| 05:15.03 | jusse | somebody work with asterisk on debian |
| 05:15.28 | mog | <PROTECTED> |
| 05:15.36 | *** join/#asterisk inv_Arp (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 05:15.45 | kronic | I do |
| 05:16.00 | kronic | I'm attempting to setup queue-based routing with multiple agents in queues |
| 05:16.51 | *** join/#asterisk Tond (n=tond@206.53.54.138) |
| 05:17.40 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 05:19.33 | Tond | Which flavour of Linux works best with asterisk? I have used RedHat before.. Just wanted to check.. also for RH there is AS, ES, WS and desktop. Which ons is mostly recommended for a box that will act as PBX and no development work? |
| 05:19.59 | carrar | centos! |
| 05:20.22 | h3x | LFS ! hahahah |
| 05:20.26 | carrar | hahah |
| 05:20.47 | carrar | ... and I mean from SCRATCH!!! |
| 05:21.08 | h3x | hell yeah |
| 05:22.05 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 05:22.57 | kronic | openbsd! |
| 05:23.22 | Tond | I thought * didn't work so well with BSD.. but then again my info is outdated.. |
| 05:23.23 | Tond | :) |
| 05:23.32 | kronic | works fine on obsd |
| 05:23.45 | kronic | I wouldn't trust anything else for my server(s) |
| 05:23.58 | kronic | well certainly not windows |
| 05:24.11 | Tond | I ahve very little experience with BSD, but have heard many good things about it... |
| 05:24.29 | Tond | I am thinking of using RH, what do you think? |
| 05:24.38 | kronic | I don't like red hate |
| 05:24.45 | kronic | that's my opinion obviously |
| 05:24.45 | brookshire | debian! |
| 05:24.56 | kronic | yeah, if your linux centric, use debian |
| 05:25.16 | kronic | the bsds are functionality approached |
| 05:25.37 | Tond | I see.. What makes debian better..? ( i know it may sounds doumb, but sort of at rokkie stage) |
| 05:25.37 | Tond | lol |
| 05:25.41 | Tond | rookie |
| 05:25.47 | kronic | while gnu/linux distros seems to be perhaps feature approached (if I can described it as that) |
| 05:26.01 | Tond | ya I know what u mean |
| 05:26.18 | kronic | well, they have a good philosophy towards free software, you'll probably never have to build anything from source, its binary package based |
| 05:26.29 | h3x | the freebsd port of asterisk is great |
| 05:26.33 | h3x | as long as you dont need tdm cards |
| 05:26.41 | kronic | I think this is getting off topic ey? |
| 05:26.43 | h3x | well, zaptel . besides x101p or whatever |
| 05:26.51 | Tond | well I already have one... |
| 05:26.54 | *** join/#asterisk DanTMG (n=danielga@124-168-31-109.dyn.iinet.net.au) |
| 05:27.01 | Tond | the 4 port one |
| 05:27.14 | h3x | i dont think that would work on bsd then |
| 05:27.25 | brookshire | i think it does |
| 05:27.29 | brookshire | the te110p works with bsd |
| 05:27.44 | h3x | he didnt say t1 |
| 05:27.55 | kronic | I haven't used any telephony h/w with bsd, so I have no knowledge on it |
| 05:27.56 | brookshire | though.. i hear it's kinda flakey :) |
| 05:28.00 | Tond | I see, so then if I am going to use Linux, debian is mostly used and recommended over RH? |
| 05:28.13 | h3x | redhat sucks in all forms |
| 05:28.17 | Tond | lol |
| 05:28.28 | kronic | well, redhat supports it, debian does, a linux kernel is a linux kernel |
| 05:28.35 | kronic | more or less |
| 05:28.38 | h3x | really what distro you pick should have more to do with what you are doing besides asterisk itself |
| 05:28.55 | h3x | not really, asterisk had problems with uClibc until recently |
| 05:29.01 | h3x | userland counts more than kernel |
| 05:29.14 | jusse | when i hung up the phone, i dont listen some marked tone |
| 05:29.22 | jusse | when i hung up the phone, i dont listen some marked tone // why???? |
| 05:29.34 | Tond | well I guess I won't be doing much more than Asterisk on that box. I just want the setup to be as easy as possible. I remember with RH i had to get so many different updates and rebuilding of source, etc.. |
| 05:30.10 | jusse | ztcfg -vvv show 3 channels configured : 2 FXO 1 FXS |
| 05:30.29 | jusse | i have a TDM400P |
| 05:30.32 | jusse | when i hung up the phone, i dont listen some marked tone // why???? |
| 05:30.37 | kronic | if you want something similar to redhat, use centos |
| 05:31.09 | Tond | Ok.. |
| 05:31.10 | kronic | see, too many choices, its free or open for me |
| 05:31.17 | mog | jusse, i have no idea what you are saying |
| 05:31.36 | jusse | hahahaha |
| 05:31.40 | *** part/#asterisk DanTMG (n=danielga@124-168-31-109.dyn.iinet.net.au) |
| 05:31.43 | jusse | ok ok |
| 05:31.48 | jusse | relax |
| 05:31.57 | *** join/#asterisk DanTMG (n=danielga@124-168-31-109.dyn.iinet.net.au) |
| 05:32.02 | Tond | hrm.. Is debian's GUI and ease of use the same as RH? How about finding packages and binary files? |
| 05:32.04 | jusse | i need coffe |
| 05:32.12 | brookshire | tond: apt-get! |
| 05:32.17 | mog | brookshire, ! |
| 05:32.21 | brookshire | mog! |
| 05:32.39 | Tond | apt-get?! |
| 05:32.49 | mog | apt-get is debian pkg manager |
| 05:32.53 | mog | its very good |
| 05:32.53 | brookshire | well |
| 05:32.57 | Tond | ah i see |
| 05:33.02 | Tond | lol.. tnx |
| 05:33.06 | brookshire | dpkg is debian's pkg manager |
| 05:33.12 | *** join/#asterisk doolph (n=doo@200.124.20.212) |
| 05:33.12 | kronic | Tond: that question is incorrectly phrased |
| 05:33.32 | brookshire | apt-get is just a utility for updates and new installs :) |
| 05:33.34 | kronic | a GUI is independent of the US |
| 05:33.39 | doolph | hello, my isp has some welcome message each time I dial a phone number, how can I disable it? |
| 05:33.46 | kronic | are you talking about the installer? |
| 05:34.30 | Tond | I am talking about, installation and finding independant binary files and not having to compile everything all the time, system stability, etc.. |
| 05:35.19 | Tond | I dont mind doing the work, just want to make sure that all will be compatible and will have lots of options later and not be limited... ex. how u say the cards may not work so well on BSD, etc.. |
| 05:35.21 | brookshire | all of them are pretty stable |
| 05:35.31 | Tond | Although BSD works great for many other stuff |
| 05:36.26 | brookshire | tond: you could try poundkey.. everything works out of the box |
| 05:36.28 | brookshire | http://www.rpath.org/rbuilder/project/asterisk/ |
| 05:36.38 | brookshire | at least for asterisk, lol |
| 05:36.43 | kronic | debian's installation is text-based, redhat has the anaconda installer |
| 05:37.02 | kronic | honestly if you can't install a distro via text prompts etc.. you shouldn't be using a *nix os |
| 05:37.09 | Tond | well even on RH i always used the text based anyways |
| 05:38.22 | Tond | see like some codes i remember were optimized for RH and for other packages had to patch / twick them to work with debian and other flavours, etc.. |
| 05:38.35 | Tond | I want to avoid taking the rocky way if u know what i mean..? :) |
| 05:40.10 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 05:42.02 | Tond | and is TrixBox a build of asterisk with its hand and legs cut in terms of debug, edit and development? |
| 05:50.35 | *** join/#asterisk Inverted (n=Inverted@66-90-148-38.dyn.grandenetworks.net) |
| 05:50.39 | brookshire | trixbox is like a distro of linux targeted towards asterisk |
| 05:50.59 | *** join/#asterisk VibroMax (n=phisto@83.229.70.154) |
| 05:51.31 | Inverted | anyone have any idea why asterisk wouldnt be playing the file at the offset specified? |
| 05:51.31 | Inverted | AGI Rx << CONTROL STREAM FILE /tmp/tmpstream1857 # 10000 2 1 |
| 05:53.59 | *** join/#asterisk mbit (n=nothing9@218-214-57-65.people.net.au) |
| 05:54.03 | *** join/#asterisk LoneShadow (n=duh@c-67-188-235-220.hsd1.ca.comcast.net) |
| 05:54.34 | mbit | hi guys is there any obvious answers when you try to execute an agi script and it returns 0? |
| 05:56.11 | HaMYaI_ | anyone, please advise me a good free h323 softphone |
| 05:56.20 | *** join/#asterisk lorinc (n=ang@caracas-4756.adsl.interware.hu) |
| 06:02.30 | carrar | stay away from h323 |
| 06:04.40 | Inverted | mbit: that it executed properly? |
| 06:04.59 | shellshark | HaMYaI_: H323 is nasty |
| 06:05.06 | shellshark | HaMYaI_: you want an IAX softphone ;) |
| 06:05.15 | Inverted | http://www.msnbc.msn.com/id/15220450/ |
| 06:06.15 | kronic | how do you define group memebership in agents.conf, maybe I'm missing something |
| 06:06.18 | kronic | :S |
| 06:10.13 | mog | well that sucks Inverted |
| 06:12.02 | Inverted | mog: agreed |
| 06:12.16 | ido | asterisk is so fucking cool. |
| 06:12.50 | mog | agreed ido |
| 06:12.58 | Inverted | its fucking insane |
| 06:13.02 | ido | mog, how unpatriotic. |
| 06:13.11 | Inverted | i cannot believe what is happening |
| 06:13.17 | mog | when my liberties are violated |
| 06:13.18 | ido | criticizing bush is letting the terrists win |
| 06:13.25 | mog | the offender should be charged |
| 06:13.27 | *** join/#asterisk bobloblian (n=bob@net-252-14.northwestel.net) |
| 06:13.29 | Inverted | mog: you will also probably like rbnlive.com and prisonplanet.com |
| 06:13.39 | mog | my bad |
| 06:14.06 | *** join/#asterisk copantl (n=galel@207.13.77.101) |
| 06:14.14 | mog | i love listening to all sides of politcal radio and news |
| 06:14.33 | Inverted | alex jones has a radio show that plays on the rbn network in the afternoon. hes out there sometimes, but has some good info |
| 06:14.35 | ido | :) |
| 06:14.52 | Inverted | that radio station doesnt really take political sides, they distrust republicans and democrats |
| 06:15.07 | mog | lol everyone takes political sides |
| 06:15.18 | mog | as no one can be completely impartial |
| 06:15.23 | Inverted | ok well they dont side with those two parties |
| 06:15.24 | mog | except of course fox news |
| 06:15.30 | mog | fair and balanced and all |
| 06:15.40 | copantl | any body know howto compile ss7 ? |
| 06:15.46 | mog | make |
| 06:15.47 | Inverted | mainstream news is a complete waste |
| 06:15.49 | mog | make install |
| 06:15.51 | Inverted | i quit watching it |
| 06:15.54 | mog | agreed |
| 06:16.00 | mog | but at same time |
| 06:16.11 | mog | its important to know what is being fed to general public as well |
| 06:16.22 | mog | it amazes how little news is reported in main stream news |
| 06:18.02 | bobloblian | can someone point me at an article that explains the differences between modems/modules/ata/etc...? |
| 06:18.15 | bobloblian | I seem to be bogging down in all the info... |
| 06:18.27 | mog | what do you want to know |
| 06:18.50 | bobloblian | well, I see some guys say its best to use ata's instead of 2 4port modems... |
| 06:19.05 | tzafrir_home | bobloblian, it depends on what you need |
| 06:19.06 | mog | use what ever makes you happy |
| 06:19.10 | bobloblian | and I am looking at setting up a system with maybe a dozen lines, so I want to have an idea what all the options are |
| 06:19.18 | mog | you have couple options |
| 06:19.23 | mog | t1 with channelbank |
| 06:19.26 | Inverted | anyone have any idea why asterisk wouldnt be playing the file at the offset specified? |
| 06:19.31 | mog | high density card |
| 06:19.31 | Inverted | AGI Rx << CONTROL STREAM FILE /tmp/tmpstream1857 # 10000 2 1 |
| 06:19.36 | mog | or ata gateway |
| 06:19.46 | mog | each have disadvantages and advantages |
| 06:20.01 | mog | disadvantage of cards is everything is in one box |
| 06:20.08 | mog | t1 and channel bank is expensive |
| 06:20.21 | bobloblian | what about ata? |
| 06:20.33 | mog | and problem with gateways are they can be expsensive, and its another lay between you in the pbx |
| 06:20.36 | tzafrir_home | bobloblian, how many phones do you need? |
| 06:20.37 | mog | but that doesnt matter much |
| 06:21.05 | tzafrir_home | if 4 or less, T1 is out of the question... |
| 06:21.15 | mog | i would reccomend getting a 2400p |
| 06:21.18 | mog | but i work for digium |
| 06:21.18 | bobloblian | how many has not been decided yet, but minimum 3 lines, maximum 15... |
| 06:21.22 | mog | so my view is slanted |
| 06:21.26 | mog | ^_^ |
| 06:21.55 | tzafrir_home | well, I would recommend a slightly different hardware, for similar slnted reasons... |
| 06:22.21 | bobloblian | ok, what would you recomend tzafrir? |
| 06:22.50 | shellshark | mog == mark spencer? :) |
| 06:22.56 | mog | lol |
| 06:22.58 | mog | no |
| 06:23.03 | mog | kram is mark spencer |
| 06:23.06 | bobloblian | and mog, how do the digital modem/modules work that I saw on the digium site? are they significantly different than analog modems? |
| 06:23.10 | mog | mog is Matt O'Gorman |
| 06:23.13 | mog | or mogorman |
| 06:23.24 | mog | ? |
| 06:23.24 | Inverted | were you at phreaknic last year? |
| 06:23.32 | mog | nope |
| 06:23.38 | mog | im not big fan of phreaknic |
| 06:23.41 | mog | i dont drink |
| 06:23.44 | mog | so not a lot to do there |
| 06:24.02 | kronic | like what does this line establish "group=1" in agents.conf, a group called 1 or what |
| 06:24.22 | Inverted | someone from digium was there, mark and some other guy were eating ice cream and prank called him. he was a fairly young guy 22-24 |
| 06:24.38 | Inverted | mog: yea its pretty boring |
| 06:24.41 | mog | probably mattf |
| 06:24.45 | Inverted | yep matt |
| 06:24.49 | file | <insert applicable Matt> |
| 06:25.04 | mog | thin guy black hair Inverted ? |
| 06:25.06 | Inverted | he was doing php at the time |
| 06:25.16 | shellshark | file: you work for digium also? |
| 06:25.21 | Inverted | he seemed embarassed of that :) |
| 06:25.22 | tzafrir_home | bobloblian, http://www.xorcom.com/astribank/features-16.html . But then again, there are also the Digium 2400 card and the Sangoma A200 card. If you need 3 extensions now but may need 15 later, those cards may be more extendible |
| 06:25.29 | file | shellshark: I do. |
| 06:25.34 | shellshark | file: cool :) |
| 06:25.40 | mog | oh tzafrir is right there is the usb channelbank too |
| 06:25.44 | mog | i forgot about that |
| 06:25.52 | shellshark | usb channelbank? |
| 06:25.56 | shellshark | no way! |
| 06:26.02 | mog | xorcom astribank |
| 06:26.12 | shellshark | and asterisk supports such a beast? |
| 06:26.20 | mog | yes |
| 06:26.24 | mog | its a zaptel device |
| 06:26.52 | Inverted | so neither of you digium guys know why control stream file wouldnt be playing the file at the offset but rather at the beginning no matter what value is sent for it? |
| 06:27.12 | *** join/#asterisk vexorg (n=vexorg@CPE0003478eef7c-CM0016b531e87c.cpe.net.cable.rogers.com) |
| 06:27.31 | mog | what version Inverted ? |
| 06:28.17 | Inverted | 1.2.11 |
| 06:28.57 | mog | im not aware of any such problem |
| 06:29.08 | mog | although i dont have any devices to lab it up at the moment |
| 06:29.12 | mog | will do so tommorrow at work |
| 06:29.41 | Inverted | am I doing something dumb here? print "CONTROL STREAM FILE /tmp/tmpstream$random # $skipms 2 1\n"; $result = <STDIN>; |
| 06:31.23 | *** join/#asterisk doolph (n=doo@200.124.20.212) |
| 06:32.21 | doolph | There's anyway to dial a number then asterisk gives me a ring tone instead of the provider ring tone |
| 06:32.40 | shellshark | Ringing() |
| 06:32.44 | shellshark | oh |
| 06:33.12 | shellshark | Dial(IAX2/provider/12125552424|tr) |
| 06:33.23 | shellshark | the 'r' in the dial options tell it to spoof ringtone |
| 06:33.24 | doolph | let me see |
| 06:33.29 | doolph | ah |
| 06:33.45 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 06:33.53 | sbingner | doolph: you may or may not need the t ;) "show application dial" |
| 06:34.16 | bobloblian | tzafrir_home: thanks, I have looked over the three things you mentioned. I am apparently still not getting something, though, from the pictures and descriptions, I am just not understanding how 24 ports can be gotten into a single pci slot... |
| 06:34.30 | sbingner | bobloblian: dongle cables |
| 06:34.31 | bobloblian | is there another piece of hardware that goes with those two cards? |
| 06:34.43 | shellshark | sbingner: yeah I always use the t as a force of habit ;) |
| 06:34.50 | brookshire | Inverted: i was at phreaknic last year |
| 06:34.53 | brookshire | mattf was not |
| 06:35.08 | bobloblian | dongles? they all pile out the back of the computer? |
| 06:35.08 | brookshire | but then again.. neither was mark |
| 06:35.16 | Inverted | brookshire: you handed out the stuff out of your knapsack? |
| 06:35.37 | brookshire | yup |
| 06:35.44 | brookshire | there were 3 of us actually |
| 06:36.12 | Inverted | did you go eat with howard? |
| 06:36.15 | brookshire | yes |
| 06:37.58 | sbingner | bobloblian: I'm assuming it's a cable that connects tot he card and comes out as cables, but I can't find anything that actually states that |
| 06:38.02 | doolph | nice |
| 06:38.04 | doolph | many thanks |
| 06:38.36 | sbingner | doolph: make sure you know if you want the t or not -- in case you missed earlier |
| 06:39.08 | doolph | I don't know if I need the t |
| 06:40.05 | sbingner | doolph: show application dial |
| 06:40.13 | sbingner | t = called party can xfer the call |
| 06:40.20 | doolph | ah |
| 06:40.22 | doolph | then no |
| 06:40.51 | doolph | it works perfectly now |
| 06:40.52 | doolph | heh |
| 06:41.05 | shellshark | doolph: happy? :) |
| 06:41.16 | doolph | but the dial tone is a bit different |
| 06:41.36 | doolph | its like CISCO gateways |
| 06:43.00 | sbingner | doolph: you from outside USA? |
| 06:43.07 | doolph | yes |
| 06:43.19 | sbingner | it's a US dialtone probably, you may be able to change that |
| 06:43.27 | doolph | where |
| 06:43.34 | bobloblian | sbingner and others, thank you, I think I am pointed in the right direction now... |
| 06:43.54 | *** join/#asterisk linagee (n=linagee@unaffiliated/linagee) |
| 06:44.24 | sbingner | doolph: I think indications.conf may be of use -- not quite sure but that should be a good place to start |
| 06:44.41 | sbingner | it may have something for your country, you just need to tell it to use it heh |
| 06:45.58 | doolph | i think its there |
| 06:46.46 | doolph | ho ho ho |
| 06:46.50 | doolph | oh man you are so geek |
| 06:47.01 | sbingner | you find it? |
| 06:47.04 | sbingner | http://www.voip-info.org/wiki/index.php?page=Asterisk+config+indications.conf |
| 06:47.14 | doolph | yes my country is using chinese dialtone lol |
| 06:47.25 | doolph | I thought it was uk |
| 06:47.27 | sbingner | lol |
| 06:47.37 | doolph | because we are Cable&Wireless |
| 06:48.49 | *** join/#asterisk skirmisha (n=viki@87.126.55.7) |
| 06:48.55 | skirmisha | hello guys |
| 06:49.14 | skirmisha | does anyone know about sip messaging |
| 06:49.37 | skirmisha | and how is implemented in asterisk |
| 06:49.51 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 06:50.05 | skirmisha | i tried some patch but i get a problem |
| 06:50.12 | skirmisha | system send msg 2 times |
| 06:51.35 | skirmisha | huh nobody home |
| 06:52.48 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 06:54.21 | sbingner | I'm here, but I can't help ya :\ |
| 06:54.46 | kristalino | hi. is there a free software for speech recognition ? |
| 06:56.26 | skirmisha | sbingner where are u from |
| 06:57.48 | *** join/#asterisk skywriter (n=test@mail.splendor.net) |
| 06:58.05 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 06:58.14 | sbingner | ... why? lol |
| 06:58.30 | sbingner | kristalino: yes, but the name escapes me... look on freshmeat |
| 06:58.33 | skirmisha | just asking |
| 06:58.40 | sbingner | hawaii |
| 06:58.57 | doolph | anybody know any good standlone Callshop Software |
| 06:58.59 | skirmisha | lol |
| 06:58.59 | kristalino | sbingner, sphinx maybe ? |
| 06:59.30 | skywriter | can someone explain what s the difference between [sip-proxy] and [sip-proxy-out] context in sip.conf |
| 07:00.26 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 07:01.32 | sbingner | skywriter: two different examples, _out is outbound only |
| 07:02.23 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 07:02.37 | skirmisha | damn |
| 07:03.02 | sbingner | skirmisha: ? lol |
| 07:03.12 | sbingner | kristalino: http://freshmeat.net/search/?q=speech+recognition§ion=projects ;) |
| 07:03.25 | sbingner | like 3 or 4 there |
| 07:04.27 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 07:04.33 | sbingner | skywriter: I mean it sends calls out but does not recieve them |
| 07:04.44 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 07:05.01 | sbingner | skywriter: http://www.voip-info.org/wiki-Asterisk+config+sip.conf |
| 07:06.00 | kristalino | does any of you use these speech recognition tools ? |
| 07:06.10 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 07:12.48 | skirmisha | i need to get this msg thing working |
| 07:12.56 | skirmisha | but seems there is bug |
| 07:14.31 | *** join/#asterisk af_ (n=af@ip-193-108.sn2.eutelia.it) |
| 07:16.04 | *** join/#asterisk nextime (n=nextime@unaffiliated/nextime) |
| 07:16.27 | nextime | hi, anyone have tryed chan_skype in svn trunk? |
| 07:18.34 | af_ | nextime, do you need a sound card for skype clients? |
| 07:19.11 | nextime | af_ _ apparently no, you can use a fake oss sound card and a fake X server like xvfb |
| 07:19.41 | af_ | oh that's good |
| 07:20.26 | nextime | but it seem to work only on * 1.2.9, not in 1.4 or svn trunk ( and i use svn trunk on my home * server, sigh ) |
| 07:21.06 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 07:24.08 | *** join/#asterisk cian (n=cian@cian.ws) |
| 07:24.47 | *** join/#asterisk cian (n=cian@cian.ws) |
| 07:30.52 | *** join/#asterisk M_at (n=matt@lrfadsl01.demon.co.uk) |
| 07:34.44 | skirmisha | anyone familar with sip messaging |
| 07:40.39 | *** join/#asterisk dpryo (i=hn@donatello.ondskap.net) |
| 07:41.14 | dpryo | Hi. Why is the cdr duplicated when using agent/queues? A call comes in to the queue (one record), and the queue dials the agent (another record). |
| 07:41.21 | dpryo | Makes billing confusing :( |
| 07:41.54 | benjk | CDRs are not meant to be convenient to people who like it easy to do billing |
| 07:42.12 | benjk | they are meant to give an accurate account of the true events |
| 07:42.57 | *** join/#asterisk jmls (n=asterisk@host81-159-167-80.range81-159.btcentralplus.com) |
| 07:44.27 | *** join/#asterisk M_at (n=matt@lrfadsl01.demon.co.uk) |
| 07:45.09 | dpryo | Well. It says we're having about double as much calls than we really are, so it isn't really accurate :P |
| 07:46.35 | M_at | Wouldn't you want to know how many calls enter the queue, how many operators take calls and how many are bounced to voicemail? |
| 07:46.44 | sbingner | dpryo: no, one call is to the system... another call is to the agent |
| 07:47.04 | benjk | it is accurate |
| 07:47.14 | sbingner | you just need to parse it properly |
| 07:47.18 | dpryo | Yeah |
| 07:47.20 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 07:47.32 | benjk | get somebody who actually understands mediation and billing processes |
| 07:48.12 | dpryo | right ;P |
| 07:48.22 | dpryo | You are not helpful. |
| 07:48.26 | dpryo | Thanks anyway. |
| 07:48.27 | *** part/#asterisk dpryo (i=hn@donatello.ondskap.net) |
| 07:48.42 | M_at | The world doesn't work the way I want! Fix it! |
| 07:49.33 | benjk | indeed |
| 07:49.40 | benjk | the Microsoft generation |
| 07:50.46 | Inverted | heh STREAM FILE will work with an offset, but CONTROL STREAM FILE will not |
| 07:50.49 | Inverted | bug! |
| 07:57.52 | skirmisha | anyone familar with sip messaging |
| 07:58.22 | M_at | skirmisha - If you describe your problem a little people may be able to help |
| 08:01.25 | skirmisha | the problem is that i used that patch for asterisk sip message |
| 08:01.37 | *** join/#asterisk bmg505 (n=leon@c1-43-4.rndf.isadsl.co.za) |
| 08:01.52 | skirmisha | but now system send message 2 times , as far as i understand it doea not clean queue properly |
| 08:02.22 | skirmisha | it's always send message to queue and if the phone is registered at the time of msg been sent it get it twice |
| 08:02.46 | *** part/#asterisk mbit (n=nothing9@218-214-57-65.people.net.au) |
| 08:02.50 | skirmisha | if phone is not registered and msg been sent when phone is registered it gets msg only once |
| 08:02.57 | skirmisha | any workaround on this |
| 08:05.29 | skirmisha | i think chan_sip.c does not check whether the phone is registered upon msg receive |
| 08:06.56 | skirmisha | any ideas |
| 08:06.59 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:07.36 | *** join/#asterisk jgoo (n=e4b80e21@athedsl-25955.otenet.gr) |
| 08:09.13 | jgoo | hey guys - I have one * setup using G.711 ulaw, and have issues with G.711 polycoms talking to G.729 polycoms (one issue) But, right now I am considering making one office use G.729 between them and us, to improve latency by compressing... but I cannot find specifics on the compression to make sure the networking and traffic shaping is right |
| 08:09.21 | *** join/#asterisk adorah (n=admin@87.68.168.214.cable.012.net.il) |
| 08:09.36 | jgoo | is G.711 is 100kbs, how much will I save compressing to G.729 ?? |
| 08:10.12 | skirmisha | g711 is 64 |
| 08:10.16 | sbingner | jgoo: http://lists.digium.com/pipermail/asterisk-users/2003-June/007594.html |
| 08:10.17 | skirmisha | g729 is 8 |
| 08:10.28 | adorah | g.711 is 64kbs+overhead about 87 |
| 08:10.38 | sbingner | it probably won't help you with latency tho |
| 08:10.39 | psk | jgoo: http://www.asteriskguru.com/tools/bandwidth_calculator.php |
| 08:11.06 | adorah | g.729 is about 30-40kbs inc. overhead |
| 08:11.46 | adorah | GSM is about 15-22kbs |
| 08:11.48 | *** join/#asterisk tobben (n=tobben@194.54.103.22) |
| 08:12.11 | sbingner | I've always got the best sound by using ilbc but most devices don't support that |
| 08:12.17 | *** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 08:12.53 | tobben | When doing a status action, I get message: Channel status will follow . Where can I get the channel status-info ? |
| 08:13.57 | tobben | Would it appear as an event? |
| 08:14.06 | sbingner | g.729 is about 20.5 Kbps |
| 08:14.51 | sbingner | vs. ulaw at about 76.5 Kbps |
| 08:15.10 | jgoo | aaaah |
| 08:15.24 | sbingner | anyways the bw calc has all that :b |
| 08:15.25 | jgoo | nice, thanks guys |
| 08:16.00 | sbingner | jgoo: if you have bandwidth problems it might help, but latency problems are something else -- unless they're caused by oversaturation of the pipe |
| 08:16.16 | jgoo | sbingner: oversaturation |
| 08:16.22 | jgoo | I shot 3 people who were using skype |
| 08:16.30 | jgoo | and I am putting a linux box up for traffic shaping |
| 08:16.39 | jgoo | so our up pipe won't queue any voip packets |
| 08:16.48 | *** join/#asterisk vgster (n=vgster@170.252.64.1) |
| 08:17.13 | jgoo | but office to office we had problems, not office out, so that is why we can take a hit to quality office to office |
| 08:17.13 | *** join/#asterisk darkskiez (n=mbryars@195.11.205.216) |
| 08:17.44 | sbingner | jgoo: you might want to use ilbc between the offices -- it's free and sounds good.... it's a bit cpu intensive tho |
| 08:17.57 | sbingner | I don't like the way GSM sounds heh |
| 08:18.15 | jgoo | I see, didn't realise there was such a choice :-) |
| 08:18.34 | sbingner | jgoo: I'm of coutse assuming you have an * box at each office, and a trunk between? |
| 08:18.35 | jgoo | I noticed I needed to buy a license for g.729 (which I did) |
| 08:18.46 | jgoo | sbingner: how do you mean 'a trunk between' |
| 08:18.53 | sbingner | a connection between the 2 * boxes |
| 08:19.02 | jgoo | ah you mean, I dial an extension and it knows to pass that to the * box |
| 08:19.06 | qdk | jgoo: IAX2 trunk |
| 08:19.16 | sbingner | iax2 trunk is most efficient way |
| 08:19.21 | jgoo | yeah, something like that, although right now we were using a VPN, which I heard can also cause some issues |
| 08:19.50 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 08:19.59 | sbingner | then I'd suggest using ULAW at both offices, and ilbc or GSM on the trunk between |
| 08:20.22 | sbingner | you can play with different codecs and see which gives you the best quality/bandwidth trade off |
| 08:20.33 | jgoo | great, very informative, thanks guys. was really impressed at the features and ease of this |
| 08:20.35 | ernie_ | FXO PCI Master abort on a x100p from x100p.com while loading wcfxo module (zaptel version 1.2.9.1). Any clues? |
| 08:20.55 | sbingner | ernie_: IRQ conflicts |
| 08:21.32 | jbsolutios | hi everyone - does anyone know how to set the type of ring-tone (UK) used in queues, please? |
| 08:21.36 | sbingner | ernie_: some MLB completely don't work... you have to try to get it on it's own IRQ and try APCI on and off |
| 08:21.38 | ernie_ | sbingner: ok, i checked in my lspci output, only that card is using irq 185? |
| 08:21.45 | tobben | Doing a Status action.. Where does the Channel info go ? |
| 08:21.50 | sbingner | ernie_: cat /proc/interrupts |
| 08:22.14 | sbingner | ACPI? heh typo |
| 08:22.31 | ernie_ | sbingner: wcfxo does not show up there? |
| 08:23.05 | sbingner | does the x100p still use wcfxo not wctdm? |
| 08:24.02 | ernie_ | wcfxo accordig to voip-info.org |
| 08:24.07 | ernie_ | according that is |
| 08:24.14 | sbingner | anyways I gotta sleep... x100p can be a little finiky, might even try moving it to another PCI slot to get it isolated |
| 08:24.26 | ernie_ | ok |
| 08:24.36 | ernie_ | i expected some issues when i bought it |
| 08:24.49 | ernie_ | so i'll try playing arround with it |
| 08:35.29 | *** part/#asterisk jmls (n=asterisk@host81-159-167-80.range81-159.btcentralplus.com) |
| 08:36.31 | M_at | Just a quickie - the 1.2 branch of asterisk from the svn repository is the same as downloading the current tar file? |
| 08:38.08 | benjk | ernie_, a dog shit on my carpet and now it stinks. Any clues? |
| 08:40.23 | FuriousGeorge | M_at: you want ...asterisk/tags/1.2.11 i think is the newest |
| 08:40.41 | FuriousGeorge | M_at: not branches/1.2 |
| 08:41.19 | *** join/#asterisk jmls (n=asterisk@host81-159-167-80.range81-159.btcentralplus.com) |
| 08:41.41 | M_at | Just out of curiosity - what is branches/1.2 - the point at which 1.2 became live? |
| 08:43.38 | FuriousGeorge | not sure if ur asking me |
| 08:43.47 | FuriousGeorge | ~g7.29 |
| 08:44.39 | FuriousGeorge | ~g.729 |
| 08:44.40 | jbot | extra, extra, read all about it, g.729 is It was in November 1995 that the G.729 standard, also referred to as CS-ACELP was adopted by the ITU, a United Nations organization. Similar, quality-wise, to 32 kbps ADPCM, G.729 offers toll quality speech. Furthermore, being only an 8 kbps codec, G.729 offers opportunities for significant increases in bandwidth ... |
| 08:52.05 | *** join/#asterisk ltd (n=z@202-161-26-249.dyn.iinet.net.au) |
| 09:01.02 | FuriousGeorge | someone recommended i use g.729 to help with upstream packets that dont seem to make it to my provider (outside parties say we are breaking up). now that i think about it, woudlnt that make it worse? in other words, those same packets that are getting lost now contain even more of the conversation? |
| 09:01.07 | FuriousGeorge | i use ulaw now |
| 09:01.45 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 09:03.16 | FuriousGeorge | - |
| 09:04.48 | Makenshi | maybe this is an old question, but i dont suppose anyone knows of a sip modem, preferably in the form of a windows driver, but posix would also do |
| 09:07.43 | james_ | FuriousGeorge: the codec selection doesnt have anything to do with dropped packets, you have an underlying problem... figure out what it is, then you can figure out the most suitable codec for the situation |
| 09:08.04 | *** join/#asterisk oliviert (n=olivier@194.2.122.6) |
| 09:08.58 | oliviert | hi guys |
| 09:09.07 | james_ | if it's lack of bandwidth, then yeah, g729 might help |
| 09:09.46 | oliviert | my asterisk is working well, i do use ring group, it works, and also use queue when nobody answer on ring group, |
| 09:10.28 | oliviert | my problem is, using queue, when an agent get a line from the queue, he can not hear the correspondant, and on the correspondant line, the sound is weird |
| 09:10.40 | oliviert | any idea where that could come ? |
| 09:12.45 | shellshark | perhaps a firewall / packet filter is dropping some RTP packets? |
| 09:12.52 | *** join/#asterisk ShadowHntr (i=sentinel@c-68-52-48-169.hsd1.tn.comcast.net) |
| 09:12.56 | shellshark | are you behind a NAT? |
| 09:12.58 | Makenshi | hmm i think the enum.at tapi driver is broken, every time i use it to place a call it causes an error in svchost |
| 09:13.20 | oliviert | no, not behind a nat |
| 09:13.35 | oliviert | i saw that jitterbuffer are created and destroyed |
| 09:14.34 | oliviert | when the agent hang up the phone, jitterbuffer is created for agent extension, and destroyed for my zap channel |
| 09:14.52 | oliviert | well the contrary |
| 09:15.22 | oliviert | created for zap , and destroyed for agent extension |
| 09:15.43 | shellshark | what version of asterisk? |
| 09:17.00 | oliviert | asterisk 1.2.11 |
| 09:17.21 | *** join/#asterisk sumasuma (n=sumkji@61.14.86.23) |
| 09:18.10 | sumasuma | Is stun supported with asterisk ? |
| 09:19.58 | *** join/#asterisk Avochelm (n=damien@CPE-60-226-238-40.sa.bigpond.net.au) |
| 09:20.12 | *** join/#asterisk Avochelm (n=damien@CPE-60-226-238-40.sa.bigpond.net.au) |
| 09:22.30 | FuriousGeorge | james_: i realize the codec has nothing to do with how many packets are lost |
| 09:23.13 | FuriousGeorge | what im saying is that since packets are occasionally being lost, wouldnt using a codec with more compression cause the packets that are lost to contain more of what im saying than if i use ulaw |
| 09:24.02 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 09:24.23 | M_at | But it would also decrease the chance of the packets being los being voice ones |
| 09:24.37 | *** join/#asterisk ltd (n=z@202-161-26-159.dyn.iinet.net.au) |
| 09:25.17 | FuriousGeorge | M_at: yeah but ninety something percent of the packets are media, right |
| 09:25.20 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) |
| 09:25.43 | FuriousGeorge | whats the overhead for sip? |
| 09:25.47 | FuriousGeorge | ~sip |
| 09:25.50 | jbot | [sip] http://www.cs.columbia.edu/sip/ X11 PPP dialer interface written in gtk+. URL: http://www.geocities.com/SiliconValley/Campus/3104/sip/ Session Initiation Protocol (see RFC 3261) |
| 09:26.14 | FuriousGeorge | whatever, im going to bed |
| 09:27.20 | M_at | Night! |
| 09:36.45 | *** join/#asterisk RaYmAn-Bx (i=rayman@kbhn-vbrg-sr0-vl212-213-185-15-16.perspektivbredband.net) |
| 09:40.45 | *** join/#asterisk jaike (n=jaike@125.5.144.90) |
| 09:49.23 | *** join/#asterisk Makenshi (n=chaz@2001:630:1c0:2001:20c:29ff:fe4d:1bd5) |
| 09:54.22 | *** join/#asterisk flackes (n=flack@host-80-194-73-131.static.telewest.net) |
| 09:54.29 | flackes | Hello there does anyone use SOX ? |
| 09:54.51 | flackes | i cant seem to find a way to converting many wav files to gsm |
| 09:54.54 | Makenshi | Sarbanes Oxley? |
| 09:55.11 | hegemoOn | hehe |
| 09:55.16 | hegemoOn | vast program |
| 09:55.26 | Makenshi | thankfully, no, we don't even use itil here |
| 09:55.32 | hegemoOn | s/program/topic/ |
| 09:55.45 | M_at | Can anyone advise on good places to look for zaptel compilation errors on CentOS 4.4? |
| 09:55.46 | hegemoOn | right jbot |
| 09:56.16 | flackes | Could any one advise me on a good method of changing wav to gsm ? |
| 09:56.29 | Makenshi | M_at, i don't know, but are pre-compiled packages no good for you? |
| 09:56.37 | M_at | No |
| 09:56.44 | M_at | I need to compile in Sangoma support |
| 09:57.06 | Makenshi | flackes, did you try `sox foo.wav -r 8000 -c1 foo.gsm -resample -ql` ? |
| 09:57.28 | flackes | will that change all the files in my directory? |
| 09:57.34 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 09:57.34 | flackes | as i have like 100 that need doing |
| 09:57.38 | flackes | do i have to do them 1 at a time? |
| 09:57.48 | hegemoOn | for i in *.wav |
| 09:58.05 | flackes | ? |
| 09:58.28 | hegemoOn | do sox $i -r 8000 -c1 $i.gsm -resample -ql |
| 09:58.30 | hegemoOn | done |
| 09:58.36 | hegemoOn | or something like that |
| 09:59.14 | flackes | -r must be given as a postive intiger? |
| 09:59.15 | hegemoOn | need to remove .wav to turn into .gsm |
| 09:59.26 | hegemoOn | let me dig a bit |
| 09:59.40 | flackes | the help file is evil for this software :P |
| 10:00.57 | hegemoOn | well im not talking about sox, just about the shell script |
| 10:02.43 | hegemoOn | someting like |
| 10:03.11 | hegemoOn | for i in *.wav |
| 10:03.40 | hegemoOn | do gsm="`basename $i .wav`.gsm" |
| 10:04.01 | hegemoOn | sox $i -r 8000 -c1 gsm -resample -ql |
| 10:04.05 | hegemoOn | done |
| 10:04.44 | hegemoOn | this would create toto.gsm from toto.wav |
| 10:05.53 | flackes | will this work with the windows sox ? |
| 10:06.54 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 10:07.18 | hegemoOn | you need bash |
| 10:07.19 | hegemoOn | so |
| 10:07.20 | hegemoOn | dunno |
| 10:07.35 | hegemoOn | may be with cygwin |
| 10:07.42 | flackes | which is? |
| 10:07.46 | hegemoOn | sorry i though you were under linux |
| 10:07.55 | flackes | well i have a linux box too |
| 10:07.58 | *** join/#asterisk xnon (n=xnon@200.82.222.85) |
| 10:08.01 | hegemoOn | cygwin is linux env for windows |
| 10:08.05 | hegemoOn | you can run shell |
| 10:08.08 | flackes | just have the sounds files on this pc |
| 10:08.11 | hegemoOn | and run ls on windows |
| 10:08.28 | hegemoOn | i dont know windows scripting |
| 10:08.36 | flackes | nerf |
| 10:14.25 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) |
| 10:15.55 | *** join/#asterisk xnon (n=xnon@200.82.222.85) |
| 10:16.32 | *** join/#asterisk flackes (n=flack@host-80-194-73-131.static.telewest.net) |
| 10:16.42 | flackes | Any one use SOX with xp |
| 10:19.00 | jmls | flackes: ffs - copy the files to your linux box, and follow the instructions given by hegemoOn and copy them back |
| 10:19.04 | M_at | flackes: It's probably easier to just use a script as suggested by hegemoOn on the Linux box itseld |
| 10:19.06 | M_at | Too late |
| 10:19.10 | jmls | :) |
| 10:20.05 | M_at | No I mean * flackes has quit IRC (Read error: 104 (Connection reset by peer)) |
| 10:20.24 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 10:20.55 | backblue | morning |
| 10:20.56 | jmls | oh damn. git asks a question and then hightails it out. |
| 10:21.03 | jmls | (yes, I know, reset by peer) |
| 10:21.18 | backblue | ppl. what it's that software for testing sip? sipp or something? |
| 10:21.19 | M_at | Connection reset by more intelligent user |
| 10:21.29 | jmls | lol |
| 10:21.45 | M_at | backblue: Google seems to think so |
| 10:21.48 | jmls | next he'll be wanting to install asterisk on his windows 95 pc |
| 10:22.35 | M_at | Nothing wrong with that. As long as he uses something like trixbox first so he has no choice but to wipe the HD ;) |
| 10:22.37 | backblue | yes, it's that. |
| 10:22.43 | backblue | thks |
| 10:22.51 | M_at | Don't thank me - thank google |
| 10:23.05 | jmls | ha! |
| 10:25.00 | *** join/#asterisk addisonzinser (i=addisonz@dynamic-acs-24-144-221-113.zoominternet.net) |
| 10:25.42 | *** join/#asterisk skywriter (n=test@mail.splendor.net) |
| 10:26.43 | skywriter | can we start asterisk with sip debug enabled??i need to see what happen directly at startup |
| 10:28.47 | *** part/#asterisk addisonzinser (i=addisonz@dynamic-acs-24-144-221-113.zoominternet.net) |
| 10:29.13 | *** join/#asterisk SimonK_ (n=SimonK@host86-144-75-221.range86-144.btcentralplus.com) |
| 10:30.13 | SimonK_ | Anyone point me to best info about setting up Granstream GXP-2000 with Asterisk? |
| 10:32.01 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) |
| 10:33.36 | benjk | jmls, I think Patrick Deruel's Windows port might actually run on W95 |
| 10:33.52 | M_at | God I hope not |
| 10:34.04 | benjk | at the very least it will work on W98 |
| 10:34.21 | benjk | he's even got ISDN and voice modem support |
| 10:34.25 | ksh | seriously, is this a joke ? |
| 10:34.29 | benjk | nope |
| 10:34.38 | ksh | who wants this ? |
| 10:34.44 | benjk | I dont know |
| 10:35.17 | M_at | Got a link to it? |
| 10:35.25 | benjk | asteriskwin32.com IIRC |
| 10:35.33 | ksh | absolutly nogo running network software on windows w* |
| 10:36.07 | benjk | I wouldn't consider it for any serious use |
| 10:36.08 | M_at | Nah - 2k, XP or 2k3 it says |
| 10:36.15 | benjk | oh |
| 10:36.27 | M_at | It runs in cygwin |
| 10:36.30 | *** join/#asterisk tmjb (n=tmjb@mail.grappoloin.com) |
| 10:36.35 | benjk | I thought it went back to W98, oh well |
| 10:36.41 | benjk | yes it uses Cygwin |
| 10:36.44 | ksh | ah, ok :) cygwin is ok |
| 10:36.48 | ksh | on 2k+ |
| 10:37.03 | benjk | well the nice thing about it is that it is a single installer |
| 10:37.09 | M_at | Gah! Bloody zaptel compilation |
| 10:37.20 | hegemoOn | damn, it isnt often im inclined to help people, and this guy runs windows |
| 10:37.23 | benjk | so if somebody wants to play around and check out the software, they can do that very easily |
| 10:37.42 | benjk | also for learning and testing dialplan stuff etc etc |
| 10:37.45 | hegemoOn | silly me |
| 10:37.51 | tmjb | who is the woman that recorded default voice messages on Asterisk ? |
| 10:37.59 | benjk | Allison Smith |
| 10:37.59 | hegemoOn | my gf |
| 10:38.31 | M_at | What have CentOS/RedHat changed to make zaptel fall over on CentOS 4.4? |
| 10:38.36 | hegemoOn | http://www.allisonsmith.org/ |
| 10:38.37 | hegemoOn | ? |
| 10:38.44 | tmjb | oh great :) |
| 10:39.13 | linagee | allisonsmith is da man! (er, woman) |
| 10:39.20 | linagee | allison smith in da house |
| 10:39.36 | M_at | If only we could get her to say hash instead of pund |
| 10:39.50 | linagee | M_at: it's pound though, not has. :p |
| 10:39.55 | linagee | h |
| 10:39.59 | M_at | Not in my country it isn't |
| 10:40.11 | linagee | M_at: which country is that? |
| 10:40.14 | M_at | UK |
| 10:40.22 | linagee | ah |
| 10:40.27 | M_at | You may see the confusion |
| 10:40.33 | linagee | M_at: why not call it by the real name. :) |
| 10:40.38 | linagee | what is it, octothorp or something |
| 10:40.41 | M_at | yup |
| 10:40.48 | M_at | But even less people over here know that |
| 10:40.50 | linagee | that will really throw people off. :-pD |
| 10:41.03 | M_at | But that doesn't solve my zaptel compilation error |
| 10:41.21 | linagee | "Which key is the octothorp? ack!" "you have ten seconds to press the octothorp before explosion..." |
| 10:42.48 | linagee | look at all the incorrect definitions on urbandictionary. http://www.urbandictionary.com/define.php?term=octothorp |
| 10:43.49 | benjk | octothorpe |
| 10:44.08 | benjk | sharp |
| 10:44.46 | benjk | but Allison Smith calling it "pound" wouldn't be right |
| 10:44.52 | benjk | it would be very bad taste |
| 10:45.04 | benjk | Better use Cepstral Lawrence |
| 10:45.16 | benjk | Queens English |
| 10:45.42 | benjk | or use a UK narrator |
| 10:45.57 | M_at | http://pastebin.ca/200742 Can anyone identify the probelm stopping my zaptel compilation? Warning it's long. |
| 10:46.07 | benjk | there should be British English recordings for Asterisk somewhere |
| 10:46.19 | linagee | hegemoOn: allison smith is your girlfriend? lol |
| 10:46.40 | M_at | There are - not great ones though, levels are all over the shop |
| 10:46.57 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) |
| 10:46.59 | benjk | ok |
| 10:47.10 | *** join/#asterisk drcode (n=chatzill@87.69.59.186.cable.012.net.il) |
| 10:47.12 | linagee | hegemoOn: does she say, "please press one to get my phone number. please press two to ask me out on a date." etc? |
| 10:47.14 | benjk | well, I use Cepstral Lawrence to record my British prompts |
| 10:47.15 | drcode | hi all |
| 10:47.28 | drcode | carrar: hi |
| 10:47.29 | M_at | I use an ex BBC Radio 4 announcer ;) |
| 10:47.29 | benjk | sounds splendid |
| 10:47.59 | benjk | nice |
| 10:48.13 | linagee | benjk: what do you think your friends would say if you called them with the soundex codec? LOL. "hello? is this a robot? hello?" |
| 10:48.22 | drcode | some one can do checking with my ast in x-lite client? |
| 10:48.24 | Makenshi | i recall a while ago a few of us put in money to have a british lady do some recordings |
| 10:48.31 | drcode | I want to check it it working from internet |
| 10:48.36 | benjk | they'll hangup immediately |
| 10:48.40 | Makenshi | my request was for "I.T. Services" |
| 10:48.51 | linagee | Makenshi: a british lady on my IVR would make me seem like I have more class! |
| 10:48.58 | benjk | heh |
| 10:49.10 | benjk | use Cepstral Lawrence |
| 10:49.10 | Makenshi | as i live in the midlands perhaps i should see about finding a brummie to use |
| 10:49.16 | linagee | benjk: "press one for tea and crumpets" |
| 10:49.23 | Makenshi | lol |
| 10:50.08 | M_at | Press 1 for beef tea, press 2 for yam yam yam |
| 10:50.09 | linagee | that would be kind of funny to make a bot like IVR. "press one to feed the bot. press two to check when linagee was last on IRC. press three..." |
| 10:50.36 | linagee | feed the bot would be like a cookie monster recording |
| 10:50.36 | kristalino | does anyone use speech recognition here ? |
| 10:50.54 | linagee | kristalino: we don't believe in speech recognition here |
| 10:51.04 | skywriter | can anyone point me a link on how to configure asterisk behind a NAT? |
| 10:51.04 | linagee | blasphemy! |
| 10:51.13 | kristalino | linagee, really ? Why ? |
| 10:51.15 | M_at | skywriter - SIP or IAX? |
| 10:51.19 | skywriter | sip |
| 10:51.21 | linagee | skywriter: use IAX2 |
| 10:51.24 | M_at | Don't |
| 10:51.30 | drcode | there is good speech recognize for ast. that realy recognize? |
| 10:51.40 | skywriter | my voip doesn t support IAX |
| 10:51.42 | M_at | Use a SIP proxy on the NAT edge |
| 10:51.50 | linagee | kristalino: because speech recognition is really hard and cpu intensive. :-D |
| 10:51.52 | skywriter | how ? |
| 10:52.18 | benjk | speech recognition sucks ballz |
| 10:52.20 | drcode | I didin';t saw yet software that can recognize perfect |
| 10:52.22 | linagee | skywriter: voip provider doesn't support SIP = get a new provider. ;) |
| 10:52.23 | drcode | yes |
| 10:52.28 | kristalino | heh |
| 10:52.29 | linagee | skywriter: er, IAX |
| 10:52.38 | benjk | everybody with a foreign accent will not be recognised |
| 10:52.38 | drcode | but if I use g.729 , it dosnt suppoort dtfm |
| 10:52.39 | skywriter | it worked fine ,but i put it behind a NAT,nothing i configure van make him work |
| 10:52.47 | drcode | so how pepole can press 1. or 2.? |
| 10:52.53 | benjk | can you afford to lock out that many customers? |
| 10:52.58 | benjk | piss them off even? |
| 10:53.06 | linagee | benjk: you are wrong! |
| 10:53.07 | kristalino | benjk, linagee drcode did you ever try speech recognition tools ? |
| 10:53.12 | benjk | people get very very annoyed if speech recognition doesn't work for them |
| 10:53.15 | drcode | yes |
| 10:53.16 | linagee | benjk: speech recognition does NOT suck balls! |
| 10:53.17 | benjk | yes |
| 10:53.21 | linagee | benjk: look, i'll prove it |
| 10:53.22 | drcode | kristalino: |
| 10:53.23 | benjk | it sucks ballz |
| 10:53.29 | linagee | benjk: here, i'll type a letter. (you're wrong!) |
| 10:53.32 | linagee | benjk: "Dear aunt, let's set so double the killer delete select all" |
| 10:53.34 | benjk | machine speech recognition sucks ballz |
| 10:53.35 | drcode | I did try 5 software |
| 10:53.36 | linagee | </humor> |
| 10:53.52 | drcode | include dragon,viavoice cmu speenx |
| 10:53.54 | drcode | and so |
| 10:54.00 | linagee | benjk: Dear aunt, let's set so double the killer delete select all! :-D |
| 10:54.08 | drcode | I didn't find any software that work perfect |
| 10:54.20 | linagee | benjk: i don't hear laughing |
| 10:54.21 | skywriter | is IAX better than Sip? |
| 10:54.32 | benjk | mostly yes |
| 10:54.34 | drcode | if there is somone that use ast with speech recognize that relay work , its intersting me |
| 10:54.37 | shellshark | drcode: nothing will work perfect, especially if you speak english as bad as you type it |
| 10:54.44 | benjk | but it depends what you need to do |
| 10:54.45 | linagee | shellshark: Dear aunt, let's set so double the killer delete select all |
| 10:54.52 | drcode | well |
| 10:54.57 | linagee | benjk: you don't get the reference, do you. :-/ |
| 10:55.01 | drcode | it's not only english |
| 10:55.20 | drcode | what program do U know that work good with ast? |
| 10:55.29 | shellshark | well that's where those applications excel, is with the English language |
| 10:55.41 | linagee | benjk: here, it's a google video. it will explain. :-D http://video.google.com/videoplay?docid=-1123221217782777472 |
| 10:55.42 | shellshark | I always just use Sphinx |
| 10:55.50 | shellshark | works fine for my needs |
| 10:56.03 | skywriter | i can call my phone through asterik , but one someone response after ringing the line is cutted |
| 10:56.06 | JT | drcode: sending dtmf out of band will with with g.729 |
| 10:56.06 | drcode | well with ast? |
| 10:56.10 | skywriter | what could be the problem? |
| 10:56.33 | drcode | is there info how to setup sphinx with ast? |
| 10:56.35 | kristalino | shellshark, does sphinx recognize french language ? |
| 10:56.36 | drcode | intersting |
| 10:56.37 | linagee | benjk: the joke is that microsoft fscked up with their speech recognition demo. ;) |
| 10:56.42 | drcode | well |
| 10:56.53 | linagee | benjk: that was what the computer put. "Dear aunt...etc" |
| 10:56.53 | drcode | I check about languges with sphinx |
| 10:56.57 | drcode | it is possible |
| 10:57.09 | drcode | but no docnument that expline it |
| 10:57.16 | shellshark | kristalino: I have no idea |
| 10:57.19 | benjk | Its so easy to wreck a beach |
| 10:57.20 | *** join/#asterisk saftsack (n=saftsack@p54A7DDA5.dip.t-dialin.net) |
| 10:57.22 | shellshark | kristalino: I use it for English |
| 10:57.30 | benjk | wreck a nice beach |
| 10:57.30 | kristalino | shellshark, ok |
| 10:57.38 | JT | drcode: is the only reason you want speech recognition because dtmf isn't working at the moment? |
| 10:57.41 | linagee | benjk: wreck a beach? more speech recognition blunders? heh |
| 10:57.45 | drcode | shellshark: do U have good document on setting ast with sphinx |
| 10:57.47 | drcode | ? |
| 10:57.48 | benjk | wreck a nice beach |
| 10:57.54 | drcode | yes |
| 10:57.57 | drcode | in g729 |
| 10:58.10 | linagee | "How to wreck a nice beach. How to recognize speech" |
| 10:58.13 | JT | drcode: just use out of band dtmf, that's what it's there for |
| 10:58.14 | linagee | :-D |
| 10:58.18 | drcode | if there where codec like g.729 in open source , it will be good |
| 10:58.32 | shellshark | drcode: voip-info.org has all the tutorials that I went off of |
| 10:58.32 | JT | drcode: dtmf gets sent in the packet headers with out of band |
| 10:58.38 | drcode | JT: is there doc about ? |
| 10:58.39 | shellshark | drcode: search around |
| 10:58.43 | JT | and gets reinserted at the other end |
| 10:58.44 | drcode | k |
| 10:58.45 | drcode | thanx shellshark |
| 10:58.48 | JT | drcode: lots on voip info |
| 10:58.50 | shellshark | no prob :) |
| 10:58.58 | linagee | benjk: i find it very strange that he says "Dear Mom" and it puts "Dear aunt". LOL |
| 10:59.17 | linagee | maybe the computer knows best |
| 10:59.59 | ksh | drcode: maybe you should check the payload length. |
| 11:00.13 | ksh | i was experiencing the same issues with g729 |
| 11:00.31 | benjk | drcode, g729 is patent encumbered |
| 11:00.36 | drcode | is there codec like g729 but in open source? |
| 11:00.38 | benjk | open source or not, doesn't matter |
| 11:00.51 | ksh | g729 is still patented. |
| 11:00.51 | benjk | you still have to pay the patent royalties |
| 11:00.54 | drcode | I understand the that I need 88k for regular codec |
| 11:00.57 | benjk | even if you write it yourself |
| 11:00.57 | drcode | yes |
| 11:01.08 | drcode | I know its patent |
| 11:01.09 | benjk | use Speex |
| 11:01.18 | benjk | or GSM 06.10 |
| 11:01.28 | linagee | benjk: use speex to sound like a robot. :-D |
| 11:01.33 | drcode | benjk: how much bandwith speex or GSM need? |
| 11:01.42 | *** join/#asterisk ShipHead (i=ShipHead@gateway/tor/x-f9d92aeea7e8d588) |
| 11:01.43 | benjk | then something is wrong with it, speex sounds as good as g720 |
| 11:01.47 | benjk | 729 |
| 11:01.51 | linagee | drcode: less than a dime's worth of bandwidth |
| 11:02.05 | drcode | k |
| 11:02.05 | linagee | drcode: speex is probably the worst codec for quality. lol |
| 11:02.14 | drcode | and gsm? |
| 11:02.15 | linagee | you WILL sound like a robot. ;) |
| 11:02.26 | linagee | it's almost fun |
| 11:02.26 | drcode | k |
| 11:02.36 | benjk | I was using IAX+ILBC between Egypt and Japan on an analog dialup connection that was so bad that it only got about 18-19Kbps |
| 11:02.38 | linagee | drcode: http://www.speex.org/samples/ |
| 11:02.52 | benjk | it was clear as an ISDN call |
| 11:03.19 | drcode | so GSM is better qulty? |
| 11:03.32 | benjk | GSM is pretty good |
| 11:04.11 | linagee | those demos of speex are not fair. those people already sound like robots in the original. lol |
| 11:04.28 | ksh | isnt GSM the mobile phone codec? |
| 11:04.44 | linagee | sure why not |
| 11:04.45 | benjk | it was developed for the GSM mobile phone system |
| 11:04.55 | drcode | how much bandwith GSM take? |
| 11:05.07 | linagee | drcode: lots of things are encoded in GSM |
| 11:05.15 | ksh | just use a voip bandwidth calculator |
| 11:05.21 | linagee | benjk: will GSM sound double-crappy over a cell phone? |
| 11:05.46 | drcode | ksh: where I can download voip band calculator? |
| 11:06.10 | linagee | do a google search for: speex robot. hehehe. lots of matches |
| 11:06.21 | drcode | thanx |
| 11:06.25 | drcode | I will check it out |
| 11:08.17 | *** join/#asterisk saftsack (n=saftsack@p54A7DDA5.dip.t-dialin.net) |
| 11:08.51 | linagee | that is wierd. the "speex sounds like a robot" has been taken down from the wiki. heh |
| 11:08.52 | ksh | drcode: google |
| 11:09.20 | ksh | cisco got one on their site. afaik its for registered customers only |
| 11:09.36 | *** join/#asterisk Champi (i=Champi@damn.e-leet.be) |
| 11:10.04 | linagee | interesting... counterstrike uses speex. US Army uses speex... hah |
| 11:10.18 | linagee | it's a great protocol if you have very little bandwidth and don' |
| 11:10.25 | linagee | don't care about quality. ;) |
| 11:10.40 | benjk | speex is not a protocol, its a codec |
| 11:10.56 | linagee | benjk: bah. protocol/codec. same difference. :p |
| 11:11.10 | linagee | (yes, i realize there is a difference) |
| 11:11.23 | linagee | microsoft xbox live uses speex |
| 11:13.20 | *** join/#asterisk creativx (n=creadure@196.82-134-19.bkkb.no) |
| 11:16.12 | *** join/#asterisk linagee (n=linagee@unaffiliated/linagee) |
| 11:16.17 | linagee | a blip in the matrix? |
| 11:18.57 | drcode | realy |
| 11:21.53 | skywriter | i m using allow=all for codecs |
| 11:22.14 | skywriter | and i m trying to make outbaound call through sip |
| 11:22.26 | skywriter | how can i know which codec i musing?????? |
| 11:30.07 | skywriter | <skywriter> i m using allow=all for codecs |
| 11:30.07 | skywriter | <skywriter> and i m trying to make outbaound call through sip |
| 11:30.07 | skywriter | <skywriter> how can i know which codec i musing?????? |
| 11:34.18 | *** join/#asterisk Ebola (n=Ebola@host86-138-123-50.range86-138.btcentralplus.com) |
| 11:35.28 | ksh | sÞ9Ãiow channels |
| 11:35.31 | ksh | sip show channels |
| 11:39.45 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:40.45 | hegemoOn | linagee: sorry i was out for lunch |
| 11:41.01 | hegemoOn | well this was a bad joke about allison smith |
| 11:41.08 | hegemoOn | my gf is really better ;) |
| 11:42.39 | skywriter | i m getting :called failed, request time out while the phone i m calling is still ringing??? |
| 11:46.54 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) |
| 11:49.10 | *** join/#asterisk ltd (n=z@202-161-26-159.dyn.iinet.net.au) |
| 11:54.41 | M_at | I thought one of the whole points of using a UNIX like system was not having to reboot so often ;) |
| 11:56.02 | xnon | with a TDM400P i can conect my Asterisk PC with a PSTN line |
| 11:56.05 | xnon | ??????? |
| 11:58.04 | M_at | With a TDM400p you can connect your Asterisk PC to a PSTN line |
| 11:58.17 | *** join/#asterisk tamp4x (n=syntheti@vonmail.vonworldwide.com) |
| 11:58.34 | tamp4x | http://bugs.digium.com/view.php?id=8081 try it buy it |
| 11:58.38 | benjk | not necessarily |
| 11:58.53 | M_at | I was just correcting the grammar. |
| 11:59.10 | benjk | if you only have fxs modules then you cannot connect your tdm400 to a pstn line |
| 12:00.02 | benjk | xnon, you need 1 fxo module for each pstn line |
| 12:01.40 | xnon | well TDM400P have 4 modules FXS or FXO |
| 12:02.05 | xnon | can be FXS or FXO |
| 12:02.12 | benjk | you have to choose |
| 12:02.38 | benjk | the modules are either fxs modules or they are fxo modules |
| 12:02.55 | benjk | they are not capable of doing both |
| 12:11.32 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 12:13.48 | *** join/#asterisk Tili (n=tili@202.133.65.114) |
| 12:17.17 | *** join/#asterisk rata (n=rcampos@princed/developer/rata) |
| 12:17.36 | rata | hi |
| 12:18.39 | rata | i just upgrade asterisk and zaptel (i use debian) and it does not start. Asterisk depends on zaptel to start ? could be perhaps a zaptel issue which is causing this ? or it may start with/without zaptel ? |
| 12:19.25 | *** join/#asterisk bmg505 (n=leon@c1-106-14.rndf.isadsl.co.za) |
| 12:22.22 | hank | rata: did you install the zaptel modules? |
| 12:24.28 | rata | hank, yes, sorry i just find out that is a problem of the script it starts asterisk |
| 12:27.05 | *** join/#asterisk s0lid (n=jlq@210.213.198.20) |
| 12:27.49 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:30.42 | *** join/#asterisk fanfoue (n=admin@lev92-1-81-57-180-205.fbx.proxad.net) |
| 12:31.00 | fanfoue | hi |
| 12:31.46 | fanfoue | iive got a small question |
| 12:31.55 | fanfoue | i'm quite new to sip and asterisk |
| 12:32.08 | fanfoue | i juste set up an Asterisk@home |
| 12:32.46 | M_at | You may find more Asterisk@home help in #freepbx |
| 12:32.47 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 12:33.13 | fanfoue | and i don't understand why all my rtp communications are going through the asterisk |
| 12:33.25 | fanfoue | and not directly from one pc to another |
| 12:33.29 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 12:33.33 | fanfoue | (all pcs are in the same subnet) |
| 12:33.42 | M_at | Are they the same model phones or same setup softphones? |
| 12:33.50 | fanfoue | yes |
| 12:33.59 | fanfoue | sotfphone |
| 12:34.06 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-152-190-223.red.bezeqint.net) |
| 12:34.09 | M_at | definately not got one softphone using a different codec to the other? |
| 12:34.13 | fanfoue | i've tried sjphone, xlite and ekiga |
| 12:34.23 | fanfoue | i'll check that |
| 12:35.32 | fanfoue | i have to check that on each softphone ? |
| 12:35.40 | fanfoue | or can this be done via asterisk |
| 12:36.29 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.178) |
| 12:36.50 | M_at | the phone should specify which codecs it supports and prefers so check there |
| 12:37.25 | fanfoue | ok |
| 12:37.38 | *** join/#asterisk kumbalae (n=suma@cm53.omega182.maxonline.com.sg) |
| 12:38.01 | fanfoue | so normally it should not go through asterisk if every softphone uses the same codec ? |
| 12:38.02 | M_at | also check to see if you have canreinvite in the appropriate entries in sip.conf |
| 12:38.04 | *** join/#asterisk kippi (n=none@untrust-gct.equinoxit.net) |
| 12:38.09 | kippi | hey |
| 12:38.41 | M_at | If the softphones support the same codecs and reinvite |
| 12:38.51 | kippi | is there a install guide for linux somewhere? |
| 12:39.02 | M_at | Which distribution kippi? |
| 12:39.07 | kippi | ubuntu |
| 12:39.24 | fanfoue | i've check on asterisk, every extension have canreinvite=yes |
| 12:39.26 | kumbalae | kippi: it must be the one the same as mentioned in the asterisk website |
| 12:39.45 | bXi | yo |
| 12:39.50 | M_at | kippi: guide to installing ubuntu or guide to installnig * on ubuntu? |
| 12:40.03 | bXi | i got an issue with visdn |
| 12:40.06 | *** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 12:40.08 | kippi | installing * ubuntu |
| 12:40.16 | bXi | running exactly the same configuration on a server |
| 12:40.21 | M_at | Just use the instructions on asterisk.org |
| 12:40.34 | bXi | oh wait nvm |
| 12:40.50 | kippi | looked at the cvs ones but its says it can't find cvs.digium.com |
| 12:41.11 | *** join/#asterisk nortex (n=nortex@64.136.65.144) |
| 12:42.55 | M_at | http://www.asterisk.org/download |
| 12:42.57 | M_at | This page? |
| 12:44.31 | *** join/#asterisk jm|home (n=jamiem@dilbert.jamiem.com) |
| 12:44.50 | jm|home | anyone useing a 7905 or 7912 ip phone? |
| 12:44.53 | jm|home | -e |
| 12:45.19 | Qwell | jm|home: afaik, 7905 is sccp only |
| 12:45.27 | kippi | cvs down then? |
| 12:45.36 | jm|home | Qwell: I have a 7912 |
| 12:45.40 | noky | hi |
| 12:45.41 | Qwell | kippi: has been for months. we use svn now |
| 12:46.18 | kippi | ah ha |
| 12:46.36 | jm|home | afacs it supports sip but not from the box |
| 12:46.49 | jm|home | I'm having problems getting the SIP firmware going :( |
| 12:48.30 | Qwell | jm|home: the 7912? Yeah, it should support sip |
| 12:48.37 | Qwell | BUT, sccp is way cooler :p |
| 12:48.42 | jm|home | er ... |
| 12:48.57 | jm|home | so I need to read up on Asterisk's sccp support? |
| 12:49.03 | Corydon76-home | jm|home: one of the things about Cisco phones is that you can only upgrade by a single major version at a time |
| 12:49.15 | Qwell | Corydon76-home: nah, not with the newer firmwares |
| 12:49.22 | jm|home | oh :S |
| 12:49.28 | Corydon76-home | Qwell: oh, they finally fixed that? |
| 12:49.31 | jm|home | hmm |
| 12:49.34 | Qwell | You can pretty much go straight to the latest version, *IF* you stay on the same protocol |
| 12:49.40 | jm|home | I had 'apploadid' and 'bootloadod' ... |
| 12:49.49 | Qwell | so, sccp 2.x > sip 8.x ain't gonna work |
| 12:49.49 | Corydon76-home | Qwell: right, but he's going from SCCP to SIP |
| 12:49.56 | jm|home | a version of "60.01.00(051208A) |
| 12:49.57 | Qwell | but sccp 2.x > sccp 8.x should |
| 12:50.07 | jm|home | 6.01.00, even |
| 12:50.23 | Qwell | then sccp 8.x > sip 8.x is simple |
| 12:50.47 | jm|home | can you guys help? |
| 12:50.51 | jm|home | I have tftpd going |
| 12:50.56 | festr__ | does 1.4 beta dunamic codec change ? |
| 12:51.02 | Corydon76-home | jm|home: what version of SCCP do you have? |
| 12:51.12 | jm|home | Corydon76-home: how can I check? |
| 12:51.15 | *** part/#asterisk ManxPower (n=manxpowe@69-2-85-41.wan.networktel.net) |
| 12:51.16 | jm|home | oh, it's on the back? |
| 12:51.25 | Qwell | jm|home: in the menus |
| 12:51.35 | jm|home | k |
| 12:51.42 | festr__ | i mean if iax -> iax is g711 -> sip is g729 and iax can fo g729 too so established iax will change to g729? |
| 12:51.42 | Qwell | grr |
| 12:51.45 | jm|home | Version 6.01.00 (051208A) |
| 12:52.00 | Qwell | I can't tell if my ears are ringing, or if something in my house is emiting a really high freq |
| 12:52.00 | jm|home | does that help? |
| 12:52.10 | jm|home | Qwell: tinitus :) |
| 12:52.59 | jm|home | so this PDF is for entirely the wrong model :S |
| 12:53.44 | Qwell | and of course, nobody else here would be able to hear it if it was real...wtf |
| 12:53.47 | *** join/#asterisk McLazarus (n=mcallist@64.94.197.130) |
| 12:56.43 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 12:57.56 | Corydon76-home | festr__: they are two different protocols, so what would be the point of one changing codecs? They still have to go through the gateway |
| 13:00.18 | kippi | i am getting this error when I try to build zaptel, any ideas? make: *** [kdist_build] Error 2 |
| 13:02.43 | *** join/#asterisk dasenjo (n=dasenjo@63.245.86.151) |
| 13:02.53 | *** join/#asterisk mutilator (n=ana@65.111.222.120) |
| 13:03.30 | oliviert | hi guys |
| 13:03.41 | mutilator | hola |
| 13:04.15 | oliviert | i do have crappy sounds using queue, well when an agent hang up a line from the queue, the agent can not hear the caller !! |
| 13:04.19 | oliviert | any dea ? |
| 13:04.25 | oliviert | any idea |
| 13:06.13 | *** join/#asterisk trelane (n=trelane@unaffiliated/trelane) |
| 13:06.34 | *** join/#asterisk serge-v (n=serge@64.61.117.138) |
| 13:08.37 | jm|home | it does even correlate :( |
| 13:08.45 | jm|home | I don't /have/ a sip menu |
| 13:09.42 | nortex | Can any one give me a ball park figure for * consulting in the US? |
| 13:11.32 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:11.34 | *** part/#asterisk serge-v (n=serge@64.61.117.138) |
| 13:13.59 | trelane | nortex, take a really large number, I mean big... really big, and devide it by another large (But not quite as large number) |
| 13:14.04 | trelane | something around that price |
| 13:14.17 | trelane | $75-150/hour depending on the consultant and the work |
| 13:14.51 | trelane | basic dialplan being cheaper, realtime asterisk, asterisk development, asterisk manager development or asterisk embedded development costing significantly more |
| 13:15.08 | trelane | those however are just a guideline, and not necessarily set in stone |
| 13:15.17 | trelane | I know people here who are certainly worth more |
| 13:15.22 | trelane | and some that are worth less |
| 13:15.28 | trelane | and I shaln't name names!!! |
| 13:19.03 | *** join/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
| 13:19.43 | *** join/#asterisk Blok42 (n=admin@c-b120e353.1257-1-64736c12.cust.bredbandsbolaget.se) |
| 13:20.31 | Juggie | trelane, where do i put my name to get 75$ an hour? :P |
| 13:20.53 | trelane | Juggie, sokol and associates? |
| 13:21.00 | rob0 | Qwell: http://zapatopi.net/afdb.html :) |
| 13:21.07 | trelane | actually unless you're an independant consultant you usually don't make that yourself |
| 13:21.18 | trelane | but then you might get 5 hours of work a week for a long time |
| 13:21.31 | trelane | 5 hours at $75 isn't much money |
| 13:21.41 | Juggie | it is when you already work 37.5 somewhere else. |
| 13:22.18 | Juggie | i'll take 375$ us to supliment my income for sure :) |
| 13:22.49 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 13:23.10 | Blok42 | I would like to have asterisk call a c-program and get some return data, preferrably in the extension file. Is it possible? |
| 13:23.24 | Juggie | explain more. |
| 13:23.45 | Juggie | what kind of c program |
| 13:23.49 | *** part/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 13:24.33 | Blok42 | Hmm, I'll give a case... |
| 13:25.34 | Blok42 | Say I have a program that takes the CID and runs it through a db and then returns the callers destination. How would you implement this? |
| 13:25.40 | Juggie | well, i ask because if its a program you intend to write then the answer is very easily yes. |
| 13:25.58 | Juggie | you could use ODBC/MYSQL from the dialplan |
| 13:26.21 | Blok42 | The main problem is that the dialplan gives me a headache ;) |
| 13:26.23 | Juggie | or, write a AGI program to interface w/ asterisk. |
| 13:26.42 | Juggie | then write an AGI, personally, i'm lazy, (sometimes) i use php+phpagi |
| 13:27.03 | Juggie | http://www.voip-info.org/wiki-Asterisk+AGI |
| 13:27.11 | Blok42 | php would be the first hand choice, I spent years using it.. |
| 13:27.40 | Juggie | check out that link, tons of resources, have fun. |
| 13:28.01 | Blok42 | bullseye, sweet, thanks! |
| 13:28.26 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:28.30 | Juggie | i'd recomend phpagi->http://sourceforge.net/projects/phpagi/ |
| 13:29.16 | Blok42 | I have another question though. I use the SIP-3000, are you familliar with it? |
| 13:30.27 | mutilator | anyone recommend any model adsl/adsl2+ modem? for cisco as5800 and/or a lucent stinger |
| 13:31.20 | Blok42 | I want it to forward DTMF-tones from the analog phone connected to it. When I use my analog phone to call and the asterisk sends it through VOIP I can't get any DTMF-tones passed through. |
| 13:31.49 | Juggie | Blok42, no, i'm not... but on that voip-info.org page i gave you |
| 13:32.07 | Juggie | search for SIP-3000 and you'll find something |
| 13:32.38 | Juggie | who makes it? |
| 13:33.10 | Blok42 | Linksys/Sipura |
| 13:33.43 | toerkeium | guys, what kind of card do I need to connect 4 E1 to * ? |
| 13:33.58 | Juggie | a 4span carD? :) |
| 13:34.12 | Juggie | that answer depends on your hardware |
| 13:35.04 | toerkeium | Juggie: what do you mean? I don't have any hawdware already |
| 13:35.36 | Juggie | on your host system |
| 13:35.36 | Juggie | eg, 3.3 or 5v slot etc... |
| 13:35.51 | Juggie | though 5v is usually a safe bet since most 3.3v slots are backwords compatible |
| 13:36.05 | Juggie | http://www.digium.com/en/products/hardware/digitalcards.php |
| 13:36.08 | toerkeium | Juggie: which are the 5v slots? |
| 13:36.14 | Juggie | TE4XXX |
| 13:36.39 | Juggie | read the product description |
| 13:36.52 | toerkeium | nice |
| 13:37.03 | toerkeium | and what about hardware? what should I use for that kind of channels? |
| 13:37.14 | toerkeium | at max useability |
| 13:37.31 | Juggie | once again, the answer is non conclusive |
| 13:37.47 | Juggie | it depends entirely on what your application is. |
| 13:38.09 | *** join/#asterisk ellisdee (n=ellisdee@69.15.174.114) |
| 13:38.27 | toerkeium | I have readed in Asterisk TFOT their recommendations, but they speak about small channels quantities |
| 13:39.38 | *** part/#asterisk rata (n=rcampos@princed/developer/rata) |
| 13:39.51 | toerkeium | well. thanks juggie |
| 13:42.40 | *** join/#asterisk StyleWarz (i=stylewar@euphoria.evil-packet.org) |
| 13:43.40 | StyleWarz | Hi, anyone using RxFax with Zaphfc? |
| 13:43.48 | StyleWarz | my fax hangs up after every page |
| 13:44.47 | jtexter3 | Anyone here ever had Asterisk crash in app_voicemail with a SIGABRT on 1.2.10? |
| 13:46.44 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 13:47.23 | jtexter3 | It happened on line 5516, free(vms.deleted); Right before this line is a check for if( vms.deleted ) |
| 13:48.18 | *** join/#asterisk lorinc (n=ang@caracas-0505.adsl.interware.hu) |
| 13:48.21 | Juggie | toerkeium, if you tell me your application i will tell you. |
| 13:48.59 | Juggie | eg, if you are doing 4 spans of T1 to SIP but the sip phones are all G711.alaw |
| 13:49.08 | Juggie | then, the server doesnt do much work. |
| 13:49.21 | Juggie | however, if you planned to transcode, thats another story. |
| 13:49.27 | Juggie | so theres lots of variables.. |
| 13:51.36 | *** join/#asterisk jeremy_g (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se) |
| 13:51.37 | jeremy_g | hi |
| 13:52.11 | jeremy_g | exten=> something,1,Set(VAR1=123 & VAR2=124 & VAR3=125) |
| 13:52.13 | jeremy_g | is this correct |
| 13:52.41 | tzanger | [TK]D-Fender: around? |
| 13:53.17 | eltech | whats is the real need of a digium/zap card? |
| 13:53.26 | tzanger | TDM connectivity |
| 13:53.29 | jeremy_g | come one guys, i am assigning variables in parallel |
| 13:53.36 | jeremy_g | am i doing it right or is it | i gotta use |
| 13:54.22 | eltech | tzanger so this card doesnt necessarily replace your network card and connection to broadband .. |
| 13:54.45 | tzanger | eltech: no not at all, not unless you're also getting internet from your T1 provider (at the cost of voice channels) |
| 13:55.23 | eltech | tzanger .. so what would be plugged into the zap/digium card? |
| 13:55.41 | *** join/#asterisk oliviert (n=olivier@194.2.122.6) |
| 13:55.42 | eltech | thanks for the help btw |
| 13:56.02 | tzanger | eltech: POTS FXS or FXO for the TDM series, or T1/E1 and maybe J1 to the TE series cards |
| 13:56.21 | *** join/#asterisk mattchis (n=mattchis@216.54.143.242) |
| 13:56.53 | mattchis | I have a quick question: If is issue a reload command will it restart *? |
| 13:57.31 | tzanger | mattchis: no, it'll reload * |
| 13:57.38 | tzanger | mattchis: if you want to restart it, then issue a restart command |
| 13:58.13 | eltech | tzanger .. so if i had an fxs card, id be able to plug in regular phones to this card correct? thats considered a zap card |
| 13:58.33 | tzanger | no |
| 13:58.40 | tzanger | eltech: if you have an FXS card, you can plug TELEPHONES into it |
| 13:58.45 | tzanger | FXS cards generate ringing voltage |
| 13:58.50 | tzanger | FXO cards accpet ringing voiltage |
| 13:59.20 | *** join/#asterisk lero (n=rootz@200.192.160.100) |
| 14:02.27 | eltech | so the fxo is for a fax basically |
| 14:02.35 | *** join/#asterisk mattchis (n=mattchis@216.54.143.242) |
| 14:03.08 | eltech | and so without these cards what does the accepting and generating.. when youre just using a regular nic in the box |
| 14:03.46 | tzanger | eltech: no |
| 14:03.50 | tzanger | the FXS would be for a fax machine |
| 14:03.57 | tzanger | FXO = connect to Central Office |
| 14:04.05 | tzanger | FXS = connect to Telephone Set or Station |
| 14:05.13 | eltech | ah, sorry .. so thats why an fxs for home/small office on broadband |
| 14:05.21 | eltech | fxo = t1 card or similar |
| 14:05.50 | eltech | and either way these are considered zap cards correct? |
| 14:06.35 | [TK]D-Fender | tzanger: Yup |
| 14:06.50 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:07.43 | *** join/#asterisk SeicherlBoB (n=seicherl@85-126-76-170.work.xdsl-line.inode.at) |
| 14:07.57 | [TK]D-Fender | eltech: No, T1 is T!. FXO is an ANLOLG port. |
| 14:08.17 | eltech | ok, im sorry |
| 14:08.20 | tzanger | eltech: Zaptel cards = cheap TDM interfaces |
| 14:09.31 | eltech | ok thanks guys |
| 14:09.39 | SeicherlBoB | hi there! i wanted to set up some kind of queue for incoming calls so the caller hears something like "your call is beeing proceeded to the next available ..." and music on hold and stuff. does anybody know a good starting-point? cause oreiley doesnt handle queues in his book. do i need agents for that? |
| 14:09.45 | eltech | appreciate the patience |
| 14:10.34 | tzanger | eltech: no worriesw |
| 14:10.36 | tzanger | er worries |
| 14:13.30 | jeremy_g | damn cudnt get an answer for such a simple question |
| 14:14.05 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:14.05 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:14.51 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:17.27 | [TK]D-Fender | jeremy_g: No you CAN'T do them in parallel. "show application set" |
| 14:17.38 | [TK]D-Fender | jeremy_g: What does it say, hmmm??? |
| 14:17.55 | jm|home | anyone got a Cisco CCO? |
| 14:18.10 | Qwell[] | jm|home: buy one |
| 14:18.11 | [TK]D-Fender | jeremy_g: " Set(name1=value1|name2=value2|..[|options])" |
| 14:18.17 | jm|home | I tried |
| 14:18.27 | jm|home | Cisco didn't know what I was talking about |
| 14:18.30 | [TK]D-Fender | not with an & sign :) |
| 14:18.32 | jm|home | Cisco UK & Ire |
| 14:18.33 | Qwell[] | jm|home: smartnet |
| 14:18.39 | jm|home | yeah; I said that, too |
| 14:18.51 | jm|home | they directed me to my Cisco reseller ... who I'm waiting to come back to me |
| 14:19.05 | jm|home | if it's like just a tenner, I will buy one |
| 14:19.10 | Qwell[] | it's not |
| 14:19.15 | Qwell[] | Not if you want to be legal |
| 14:19.18 | jm|home | well, for the firmware I need, I mean |
| 14:19.31 | Qwell[] | You need the one that's like $90 :) |
| 14:19.32 | Qwell[] | <3 cisco |
| 14:19.33 | jm|home | rather than a company-wide 4hr support contract |
| 14:19.38 | jm|home | ?45! |
| 14:19.42 | jm|home | cksake |
| 14:19.57 | jm|home | that's more than the phone was |
| 14:20.08 | jm|home | ok - so how well does SCCP driver work in FreeBSD? |
| 14:20.13 | jm|home | maybe I could try a different route |
| 14:20.21 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 14:20.26 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 14:20.56 | SeicherlBoB | can someone tell me a good tutorial for queues and agents? |
| 14:22.45 | *** join/#asterisk ms345 (n=mike_sim@64.74.198.10) |
| 14:26.13 | SeicherlBoB | if i want to build a "small callcenter" with only one agent: should i use queues for the waiting calls if he's busy or just hold incoming calls until the agent is free again (or is there another solution for this, i dont know)= |
| 14:26.14 | SeicherlBoB | ? |
| 14:26.26 | *** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net) |
| 14:28.34 | ms345 | anyone use a cisco 7940 or 7960 as a receptionist phone with asterisk? my receptionists are complaining about the behavior when many calls come in around the same time. "Secondary calls are still interrupting the transfer process of an initial call. A new call stops you dead in the middle of a transfer. If calls start backing up due to activity, you can't clear any of them for being interrupted mid-process by a new call." Can anyone point me to some |
| 14:29.44 | [TK]D-Fender | ms345: You want a real receptionist phone, get a Polycom IP 601 + Attendant Modules. |
| 14:29.51 | *** join/#asterisk flackes (n=flack@host-80-194-73-131.static.telewest.net) |
| 14:29.54 | flackes | hello there |
| 14:30.14 | flackes | When the dial command times out what priority does it go to? |
| 14:30.33 | *** join/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com) |
| 14:30.45 | flackes | <PROTECTED> |
| 14:30.45 | flackes | <PROTECTED> |
| 14:30.45 | flackes | Oct 13 15:26:44 NOTICE[25990]: chan_local.c:498 local_alloc: No such extension/context 1907740352412@inbound-from-stem creating local channel |
| 14:30.45 | flackes | Oct 13 15:26:44 NOTICE[25990]: app_dial.c:474 wait_for_answer: Unable to create local channel for call forward to 'Local/1907740352412@inbound-from-stem' (ca |
| 14:30.46 | flackes | use = 0) |
| 14:30.48 | *** join/#asterisk inv_arp[work] (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 14:30.54 | flackes | keep getting that error and dont know why |
| 14:31.08 | *** join/#asterisk Avochelm (n=damien@CPE-60-226-238-40.sa.bigpond.net.au) |
| 14:31.39 | SeicherlBoB | flackes: timeout will lead to the next one (n+1). busy or unavail will lead to n+100 (afaik) |
| 14:33.02 | flackes | ok |
| 14:33.09 | ms345 | I'm stuck with the cisco for now. This is the only comlaint: when new calls come in it interupts them if they are in the middle of the transfer. right now I have all 6 lines using the same SIP extension. should I try individual SIP extensions for each line? |
| 14:33.31 | flackes | thought busy was n + 101? |
| 14:33.57 | flackes | where can i past a macro |
| 14:35.07 | *** join/#asterisk droops (n=droops@adsl-074-245-001-031.sip.jan.bellsouth.net) |
| 14:35.14 | SeicherlBoB | flackes: yeah. 101. sorry |
| 14:35.26 | *** join/#asterisk yogurt2ungue (n=yogurt2u@201.231.48.24) |
| 14:35.26 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 14:35.28 | [TK]D-Fender | flackes: Clearly you are forwarding from SIP/7003 and your target doesn't exist in that context |
| 14:35.31 | PakiPenguin | hello :) |
| 14:36.29 | flackes | [TK]D-Fender that the thing... it should not be forwarding |
| 14:36.37 | flackes | it should just bypass that |
| 14:36.41 | flackes | Now forwarding Zap/3-1 to 'Local/1907740352412@inbound-from-stem' (thanks to SIP/7003-08ab5678) |
| 14:36.44 | [TK]D-Fender | flackes: Its your phone thats doing it, not * |
| 14:36.49 | ellisdee | got a problem where one user (ext 2000) attempts to call another user (2001), and 2001's phone does not ring however if 2001 picks up the phone they can talk to 2001. |
| 14:36.49 | flackes | oO |
| 14:37.01 | ellisdee | 2001 can call 2000, and 2000 gets a ring. |
| 14:37.13 | flackes | why would the phone do that? |
| 14:37.22 | SeicherlBoB | can i setup queues without agents? like writing "member=>SIP/123" instead of "member=>agent/123" ? |
| 14:37.35 | [TK]D-Fender | SeicherlBoB: Yes. |
| 14:37.35 | flackes | yes SeicherlBoB |
| 14:37.44 | SeicherlBoB | thx |
| 14:37.50 | flackes | [TK]D-Fender can i past you the macro |
| 14:37.53 | bXi | ellisdee: using a grandstream 286 by any chance? |
| 14:37.59 | *** join/#asterisk creativx (n=creadure@196.82-134-19.bkkb.no) |
| 14:38.03 | flackes | im using a gxp - 2000 |
| 14:38.29 | [TK]D-Fender | flackes: I want you to pastebin [inbound-from-stem] and everything included by it. www.pastebin.ca |
| 14:39.00 | bXi | anybody here familiar with visdn? |
| 14:39.05 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 14:39.47 | ellisdee | bxi, polycom 501 |
| 14:39.55 | bXi | ellisdee: then i dont know |
| 14:40.02 | ellisdee | i am about to try a format of the filesystem on the poly. |
| 14:40.19 | ellisdee | then reattempt the download of everything |
| 14:40.23 | ellisdee | see if this resolves the issue. |
| 14:40.38 | ellisdee | if i perform a touch on the *.wav files |
| 14:40.50 | ellisdee | will phones attempt to "update" these files? |
| 14:40.54 | flackes | [TK]D-Fender http://pastebin.ca/200988 |
| 14:41.05 | flackes | - Now forwarding Zap/3-1 to 'Local/1907740352412@inbound-from-stem' (thanks to SIP/7003-08ab5678) |
| 14:41.05 | flackes | Oct 13 15:33:04 NOTICE[26139]: chan_local.c:498 local_alloc: No such extension/context 1907740352412@inbound-from-stem creating local channel |
| 14:41.05 | flackes | Oct 13 15:33:04 NOTICE[26139]: app_dial.c:474 wait_for_answer: Unable to create local channel for call forward to 'Local/1907740352412@inbound-from-stem' (ca |
| 14:41.05 | flackes | use = 0) |
| 14:41.14 | flackes | i only get that error when i let the phone ring and ring and ring |
| 14:41.25 | flackes | and i have not put any divert numbers in |
| 14:41.40 | flackes | so it should not even know what 1907740352412 is |
| 14:42.43 | *** join/#asterisk alkiser (n=email@bdsl.66.14.163.224.gte.net) |
| 14:43.04 | [TK]D-Fender | flackes: Holy crap that is so nested I'm getting dizzy. what line in your pastebin is that # supposed to match against? |
| 14:43.17 | flackes | well that the thing its not |
| 14:43.19 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-6.rcmdva.fios.verizon.net) |
| 14:43.25 | SpaceBass | hey folks |
| 14:43.38 | flackes | i have no idea where its getting the idea im asking it to forward |
| 14:43.46 | flackes | i only get a problem if i dont answer the phone |
| 14:43.47 | [TK]D-Fender | flackes: * is telling you to your face that the PHONE is doing the forward. You are mistaken. |
| 14:44.05 | flackes | i understand that |
| 14:44.07 | [TK]D-Fender | flackes: Somewhere in there you have things redirected. |
| 14:44.13 | alkiser | is it possible to put a 'minimum wait time' on queues, lets say 30 secs, even if there are available members? |
| 14:44.32 | flackes | hmm not sure where. |
| 14:45.27 | *** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
| 14:46.11 | tzanger | what's the difference between reinvite and canreinvite? reinvite = other end can ask us to send RTP elsewhere, canreinvite = we can ask other end to send RTP elsewhere? |
| 14:47.46 | flackes | cant see anywhere else in my code that would divert |
| 14:48.11 | SpaceBass | anyone know a source for free DIDs in spain? |
| 14:48.51 | flackes | [TK]D-Fender lets say i comment out everything in the divert macro except |
| 14:48.54 | flackes | exten => s,1,DBdel(DEV/${ARG1}) |
| 14:48.54 | flackes | exten => s,2,DBdel(TIM/${ARG1}) |
| 14:48.54 | flackes | exten => s,3,DBget(fwdexten=CFU/${ARG1}) |
| 14:48.54 | flackes | exten => s,4,DBget(fwdcontext=CFUC/${ARG1}) |
| 14:48.54 | flackes | exten => s,5,Goto(${fwdcontext}|${fwdexten}|1) |
| 14:48.54 | flackes | exten => s,104,Goto(s|6) |
| 14:48.56 | flackes | exten => s,6,Dial(${ARG2},${ARG3},${ARG4}) |
| 14:49.10 | flackes | if i let the phone ring and ring and ring... it should not divert right? |
| 14:49.16 | *** join/#asterisk Whisk (n=a@195.60.27.193) |
| 14:51.50 | [TK]D-Fender | flackes: Divert isn't a damned macro. its a phone function! |
| 14:53.19 | *** join/#asterisk lsl23 (n=chatzill@141.214.234.28) |
| 14:53.39 | *** join/#asterisk luke-jr_work (n=luke-jr@2002:4335:4375:0:20d:60ff:fe60:756a) |
| 14:53.43 | luke-jr_work | why is lists.digium.com down? |
| 14:54.15 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-42.ph.ph.cox.net) |
| 14:54.21 | *** join/#asterisk cian (n=cian@cian.ws) |
| 14:54.55 | *** join/#asterisk flackes (n=flack@host-80-194-73-131.static.telewest.net) |
| 14:55.02 | flackes | Any ideas [TK]D-Fender |
| 14:55.13 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 14:55.25 | StyleWarz | Anyone here worked with RxFax? |
| 14:55.31 | SeicherlBoB | I dont get a MoH in my queue. can someone look at my conf and log?: http://pastebin.ca/201004 |
| 14:56.19 | flackes | did you set it in queue.conf? |
| 14:56.33 | SeicherlBoB | look at the pastebin |
| 14:56.41 | *** join/#asterisk oQPa (i=Ftv@1.Red-83-35-130.dynamicIP.rima-tde.net) |
| 14:58.37 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 14:59.11 | *** part/#asterisk Whisk (n=a@195.60.27.193) |
| 14:59.15 | *** join/#asterisk Whisk (n=a@195.60.27.193) |
| 15:01.02 | [TK]D-Fender | flackes: Divert isn't a damned macro. its a phone function! |
| 15:01.25 | file | dejavu |
| 15:01.52 | *** join/#asterisk ccesario (n=ccesario@matriz.isic.com.br) |
| 15:03.03 | SeicherlBoB | anybody got a clue, why MoH is not playing here? |
| 15:03.14 | [TK]D-Fender | flackes: Here's a hint : All of your pattern matches for outbound calls begin with a "9" prefix. that phone is forwarding to a normal looking number WITHOUT the prefix. hence no match. |
| 15:03.36 | [TK]D-Fender | SeicherlBoB: You may want to actually pastebin USEFUL stuff for us to see.... |
| 15:03.40 | flackes | [TK]D-Fender i understand... but i dont know where in the phone that would have been set |
| 15:04.01 | [TK]D-Fender | flackes: Thats fo you to find out, because its not an * problem. |
| 15:04.08 | SeicherlBoB | [TK]D-Fender: sure. what do you need? |
| 15:04.24 | flackes | mahhh :D |
| 15:04.46 | [TK]D-Fender | SeicherlBoB: The extensions.conf that calls it (everything slightly related) |
| 15:05.42 | SeicherlBoB | [TK]D-Fender: ok. here i have the queues and moh-conf: http://pastebin.ca/201014 |
| 15:06.34 | SeicherlBoB | [TK]D-Fender: here is the extension: http://pastebin.ca/201016 |
| 15:06.41 | [TK]D-Fender | SeicherlBoB: Does MoH work in general? |
| 15:06.45 | *** join/#asterisk lorinc (n=ang@caracas-0505.adsl.interware.hu) |
| 15:07.13 | SeicherlBoB | [TK]D-Fender: guess not. havent had a situation to test. now i got one and its not working |
| 15:07.22 | SeicherlBoB | i guess it never was working. |
| 15:08.16 | *** join/#asterisk olivier__ (n=olivier@obs92-4-82-239-116-113.fbx.proxad.net) |
| 15:09.30 | *** join/#asterisk tuck3r (n=tuck3r@unaffiliated/tuck3r) |
| 15:10.49 | [TK]D-Fender | NEXT !!!@!@!@! (c) BKW |
| 15:11.15 | SeicherlBoB | ? are you okay? |
| 15:12.18 | *** join/#asterisk SplasPood (n=jwb@nat-loopback.lga4.us.corp.voxel.net) |
| 15:14.42 | [TK]D-Fender | SeicherlBoB: Thats me asking the next person in line to ask their question since we've figured your problem out. |
| 15:15.19 | *** join/#asterisk jarg (n=jarg@200.56.225.61) |
| 15:15.23 | *** join/#asterisk Dossy (n=dossy@aolserver/dossy) |
| 15:15.54 | Dossy | Has anyone here gotten Asterisk 1.2.12.1 to work with X-Lite softphone w/ DTMF? |
| 15:15.55 | jarg | hello all, some one can recomend me a stress test tool for asterisk? |
| 15:16.04 | luke-jr_work | why is lists.digium.com down? |
| 15:16.13 | tuck3r | Dossy: uh, yeah |
| 15:16.23 | Dossy | tuck3r: what's the trick? |
| 15:16.36 | Dossy | It just will not work. |
| 15:16.50 | SeicherlBoB | [TK]D-Fender: erm.... ok. can you give me a hint on how to get MoH running? |
| 15:16.56 | tuck3r | the sample sip.conf has the proper settings i think |
| 15:17.00 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
| 15:17.59 | Dossy | tuck3r: nope :( |
| 15:18.03 | Dossy | been trying for two days now. |
| 15:18.34 | tuck3r | you don't see the commneted out settings? |
| 15:19.32 | file | luke-jr_work: it's working fine for me... and I can visit the archives fine too |
| 15:19.32 | luke-jr_work | ... |
| 15:19.37 | luke-jr_work | file, hrm |
| 15:20.11 | luke-jr_work | file, it's the only site I'm having problems with |
| 15:20.21 | luke-jr_work | I can telnet and do a HTTP request, but I never get the response |
| 15:20.30 | tuck3r | file: i can't get to it also |
| 15:21.19 | murf | luke-jr_work: Unrelated matter: did you file bug 8128 on ael? |
| 15:21.30 | luke-jr_work | murf, if it says I filed it... O.o |
| 15:21.57 | *** join/#asterisk mattchis (n=mattchis@216.54.143.242) |
| 15:22.02 | luke-jr_work | it that the one w/ switch fallthrough on patterns? |
| 15:22.25 | murf | luke-jr_work: hmm, when I updated the bug, it complained it couldn't email to you. Could be a network prob at digium. |
| 15:22.37 | luke-jr_work | murf, sounds like it |
| 15:22.39 | file | murf: we know about that already, you're behind |
| 15:22.53 | [TK]D-Fender | SeicherlBoB: Sorry, you are going to actually have to READ up on this. |
| 15:22.54 | [TK]D-Fender | ~book |
| 15:22.56 | jbot | extra, extra, read all about it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:22.57 | [TK]D-Fender | ~wikis |
| 15:22.58 | jbot | from memory, wikis is http://www.voip-info.org |
| 15:23.03 | *** join/#asterisk xezz (n=xez@serial.trust-it.ondsl.gr) |
| 15:23.07 | Juggie | ~RTFM |
| 15:23.08 | jbot | somebody said rtfm was Read The F*cking Manual (TM). It is a suggestion to do your homework before posting a question. Sometimes used as RTFM $SPECIFIC_MANUAL to refer to a specific source of information. See also http://en.wikipedia.org/wiki/RTFM |
| 15:24.02 | SeicherlBoB | [TK]D-Fender: ok. thanks. thought it was an easy hack getting it running |
| 15:24.02 | murf | I was just going to say: or a prob with Mantis! --- at any rate, luke-jr_work-- I've taken on the bug, and spotted a prob in the 1.4 version. |
| 15:24.23 | luke-jr_work | murf, does that mean AEL won't work until 1.6? :( |
| 15:24.35 | *** join/#asterisk jake1932 (n=Administ@pool-68-236-19-153.phil.east.verizon.net) |
| 15:24.46 | *** join/#asterisk afrosheen (n=cj@txprotoa2.august.net) |
| 15:24.48 | [TK]D-Fender | SeicherlBoB: Its not a "hack" its a TASk, and one you should know to go read about. It can be simple, or less than simple, all depending,a nd I'm not about to hand-hold you when you haven't been working on it for a while. |
| 15:24.52 | jake1932 | can you load SIP on the 7970? |
| 15:25.13 | luke-jr_work | jake1932, wtf is a 7970? |
| 15:25.21 | Juggie | jake1932, go buy a phone thats been designed in the last 5 years :) |
| 15:25.29 | SeicherlBoB | [TK]D-Fender: ok, i got it ;) thanks |
| 15:25.43 | jake1932 | Juggie: already did |
| 15:25.49 | luke-jr_work | Juggie, you have a suggestion re that? I was about to tell someone they want a 7960 ;p |
| 15:25.51 | jake1932 | Juggie: quite a few times |
| 15:26.08 | murf | luke-jr_work: Heavens, no! I'll fix the 1.4 in the next few hours at the most. I'll even take a look at the 1.2 version, and if it's not too hairy to fix, I might make a change there, too. Otherwise, I'll make an AEL2 update for 1.2 with the fix in it... whatcha think? |
| 15:26.10 | jake1932 | i have a 7960 and it's fine |
| 15:26.11 | jeremy_g | can * be configured to dial a certain number on encountering a 480 error message |
| 15:26.29 | Dossy | relaxdtmf doesn't do anything, dtmfmode=inband, dtmfmode=rfc2833 ... |
| 15:26.29 | [TK]D-Fender | Cisco = too expensive for nothing and a pain to get working with things it wasn't designed for (namely SIP) |
| 15:26.30 | luke-jr_work | murf, 1.2+AEL in Debian? ;p |
| 15:26.33 | Dossy | nothing makes a difference. |
| 15:26.44 | afrosheen | any recommendations for a pri card? |
| 15:26.44 | Dossy | Maybe X-Lite just isn't sending DTMF across. |
| 15:26.48 | jeremy_g | if(480-error encountered) Dial(SIP/freedom) |
| 15:26.51 | luke-jr_work | ok, so what's a good SIP/IAX phone? ;p |
| 15:27.04 | luke-jr_work | my experience is only PAP2-NA and 7960 =p |
| 15:27.05 | [TK]D-Fender | luke-jr_ : Polycom / Aastra |
| 15:27.22 | Dossy | oh well. |
| 15:27.25 | [TK]D-Fender | PAP2 is a tuned-down version of the SPA-2002. Should never be purchased. |
| 15:27.42 | Juggie | jake1932, cisco phones are a nightmare to manage |
| 15:27.42 | murf | luke-jr_work: I understand not...? Is there something "special" about debian? |
| 15:27.54 | brad_mssw | [TK]D-Fender: what's the problem with the PAP2? |
| 15:27.58 | Juggie | i've had all kinds of problems with their tftp client, etc.. regardless thats just my opinion. |
| 15:28.00 | luke-jr_work | murf, nothing more than that it's what I use for production servers |
| 15:28.02 | jake1932 | it's for a demo - i want something cool |
| 15:28.17 | Juggie | your willing to buy a phone for a demo? |
| 15:28.23 | luke-jr_work | [TK]D-Fender, it's cheaper, and unless the SPA-2002 is more reliable, makes better financial sense |
| 15:28.27 | jake1932 | uh - for this one - yes :) |
| 15:28.34 | afrosheen | jake1932, must be high profile :) |
| 15:28.38 | jake1932 | indeed |
| 15:28.51 | luke-jr_work | [TK]D-Fender, PAP2 hardware is buggy in my experience |
| 15:28.56 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:28.56 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:29.14 | jeremy_g | pap2 suxs big time |
| 15:29.20 | Juggie | jake1932, go get yourself a Mitel 5235 |
| 15:29.27 | jeremy_g | i wrote a provisioning system for that shit |
| 15:29.28 | brad_mssw | hmm, interesting, haven't had any problems with pap2's |
| 15:29.34 | murf | luke-jr_work: if AEL+1.2 debian gives you fits, let me know. |
| 15:29.37 | brad_mssw | though we only have a couple |
| 15:29.49 | luke-jr_work | brad_mssw, I've had to RMA my first, and might need to RMA my second at the rate its going |
| 15:29.54 | jake1932 | hmm- that mitel looks decent |
| 15:29.57 | luke-jr_work | murf, a package, I mean |
| 15:30.18 | luke-jr_work | luckily, the first was within the 1 yr warranty period |
| 15:30.23 | Juggie | i have it on my desk here and it rocks :) i'll admit i havnt tried the sip FW yet... but i've used the sip FW on the 5225 and its fantastic so i'd imagine its much along the same lines. |
| 15:30.53 | jake1932 | the speaker is a little weired - but otherwise - very nice |
| 15:31.18 | luke-jr_work | 7960s have crappy speakerphones |
| 15:31.25 | [TK]D-Fender | jake1932: Stop aiming for proprietary based phones. Doesn't matter if they say they support SIP. ODds are they will suck in some serious capacity and you'll regret it. |
| 15:31.30 | tzanger | hmm apparently "reinvite" has been deprecated, only "canreinvite" is around now |
| 15:31.34 | Juggie | jake, look @ the 5340 |
| 15:31.45 | luke-jr_work | [TK]D-Fender, some alternative is open? |
| 15:32.09 | Juggie | [TK]D-Fender, i cant speak for any other vendors, but the Mitel SIP Implementation is fantastic, it does not suck at all. |
| 15:32.12 | jake1932 | TK - it's just for the demo to get the guy intersted in the tech... too expensive to do an install with |
| 15:32.14 | afrosheen | the polycoms are a little more open :) |
| 15:32.14 | [TK]D-Fender | luke-jr_ : Both Polycom & Aasta have very strong SIP implementations and are very flexible. |
| 15:32.15 | murf | luke-jr_work: Ahhhhhhhhhhh! I am not a debian packaging sort of guy! Hopefully, there's someone around who can whip up something! |
| 15:32.27 | luke-jr_work | [TK]D-Fender, I'd prefer source ;p |
| 15:32.45 | luke-jr_work | hm |
| 15:32.51 | luke-jr_work | does Digium host voipsupply.com? |
| 15:33.04 | *** join/#asterisk Nixis (n=Nixis@81.10.34.82) |
| 15:33.12 | Qwell[] | luke-jr_work: no... |
| 15:33.17 | jake1932 | hopefully not their phone system - i got busies this morning |
| 15:33.22 | Juggie | jake1932, http://sipdnld.mitel.com/Downloads/Release%20Notes%20for%20Rel%206.0.0.19%20(06-5177-0008_2).doc |
| 15:33.29 | luke-jr_work | hm |
| 15:33.30 | Juggie | for all the standards/drafts that the phones support. |
| 15:33.34 | *** join/#asterisk pdtwork (n=ptinsley@209.12.249.243) |
| 15:33.36 | luke-jr_work | its having network problems too |
| 15:33.37 | Coriantum | <PROTECTED> |
| 15:33.47 | Qwell[] | Coriantum: should be, sure |
| 15:33.50 | Juggie | of course |
| 15:33.55 | pdtwork | are there any known issues with caller id showing up as being set in the console but not changing when it gets to the sip phone on the other end? |
| 15:33.56 | luke-jr_work | where's a good place to buy phones? |
| 15:34.00 | Juggie | considering E1 is >24 channels |
| 15:34.01 | Coriantum | where's the setting? |
| 15:34.22 | Coriantum | <PROTECTED> |
| 15:34.25 | Juggie | its in zaptel.conf in /etc/asterisk isnt it |
| 15:34.36 | Juggie | Coriantum, yes of course |
| 15:34.38 | Coriantum | yeah your right juggie thanks |
| 15:34.50 | *** join/#asterisk intralanman (n=lanman@pool-72-82-69-74.nrflva.east.verizon.net) |
| 15:35.09 | Juggie | you just do, group=1!channel=1-23!channel=25-47 |
| 15:35.12 | Juggie | where !=new line |
| 15:35.22 | pdtwork | it's zap to sip and the caller id is being set before the sip phone is rung, and it shows up as being changed in the console but the original caller id still shows up |
| 15:35.57 | Juggie | pdtwork, pastebin.ca your console log. |
| 15:36.08 | luke-jr_work | pdtwork, o - Specify that the CallerID that was present on the *calling* channel |
| 15:36.08 | luke-jr_work | <PROTECTED> |
| 15:36.08 | luke-jr_work | <PROTECTED> |
| 15:36.37 | Coriantum | thanks juggie! |
| 15:36.47 | Juggie | np. |
| 15:37.15 | *** part/#asterisk kevin_m (i=spyro2@3digit.de) |
| 15:38.09 | pdtwork | Juggie, http://pastebin.ca/201042 |
| 15:38.21 | Juggie | pdtwork, did you try what luke-jr_ just pasted? |
| 15:38.21 | *** join/#asterisk paryl (n=chatzill@www.admiralexpress.com) |
| 15:38.41 | pdtwork | i need some context around that, is that a dial option? |
| 15:38.55 | Juggie | yes |
| 15:39.19 | mutilator | "we havn;t been getting emailed our voicemail messages and its not letting me delete my old ones" |
| 15:39.23 | mutilator | i login to their mailbox |
| 15:39.29 | mutilator | "you have 100 new messages and 9 old messages" |
| 15:39.34 | mutilator | O_O |
| 15:39.44 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 15:39.50 | afrosheen | so..nobody can recommend a card to use with a PRI? |
| 15:40.05 | mutilator | www.voip-info.org |
| 15:40.10 | luke-jr_work | afrosheen, maybe you should ask before coming to that conclusion |
| 15:40.20 | afrosheen | I did earlier smartass |
| 15:40.20 | Juggie | afrosheen, i could recomend something but it may not be right for you |
| 15:40.31 | Juggie | because you havnt stated what your requirments are |
| 15:40.31 | luke-jr_work | not in my backscroll |
| 15:41.02 | afrosheen | luke-jr_: murf, 1.2+AEL in Debian? ;p |
| 15:41.04 | afrosheen | 2 lines below that |
| 15:41.18 | Juggie | afrosheen, what are your requirements. |
| 15:41.32 | afrosheen | Juggie, we're going to have a 24 channel t1 pri coming in, that's about it |
| 15:41.40 | Juggie | any planned expansion. |
| 15:41.54 | *** join/#asterisk FlatFoot (n=simon@80.88.192.113) |
| 15:41.56 | afrosheen | Juggie, I think 24 channels will be good for at least another year |
| 15:42.19 | pdtwork | adding o didn't help |
| 15:42.22 | Juggie | then you can go with a single span, or maybe a dualspan card so you at least have the ability to upgrade to two pri's |
| 15:42.22 | luke-jr_work | FlatFoot, |
| 15:43.00 | Juggie | pdtwork, what does the other end see? |
| 15:43.06 | Juggie | the dialed sip side |
| 15:43.09 | pdtwork | i get WIRELESS CALLER since i am calling from my cellphone |
| 15:43.22 | pdtwork | where as the console says i should be getting HFB |
| 15:43.37 | paryl | I'm getting digital choppiness on my TDM2400p, and I'm wondering if it could have something to do with the echo cancellation? how does zapata.conf need to be configured as far as echo goes? |
| 15:43.44 | afrosheen | Juggie, I'm considering the TE207 |
| 15:43.47 | Juggie | pdtwork, show me that part of your dialplan |
| 15:44.53 | Juggie | hmmmm i see your problem |
| 15:45.05 | Juggie | put HFB in quotes |
| 15:45.19 | Juggie | so, Set(CALLERID(name)="HFB") |
| 15:45.57 | pdtwork | Juggie, nope |
| 15:46.05 | pdtwork | http://pastebin.ca/201050 there it is pre quotes |
| 15:46.08 | jtexter3 | I need some help troubleshooting an issue if anyone has some advice. I have a customer where sometimes callers cannot hear the receptionist. She has tried a Polycom 601 and 501. It's very sporadic, so it's not like it happens once and then continues to happen until she does something. Unfortunately, I don't really know where to start. Anyone have any ideas? |
| 15:46.34 | Juggie | pdtwork? |
| 15:46.36 | r0d3nt | jtexter3, replace the user and try again ? |
| 15:46.44 | pdtwork | with quotes o or no o still get WIRELESS CALLER |
| 15:47.04 | jtexter3 | r0d3nt: tried that, but it continued :-( |
| 15:47.12 | r0d3nt | jtexter3, i bet it's her headset or some other issue... |
| 15:47.20 | r0d3nt | hmmm |
| 15:47.36 | pdtwork | Juggie, is it getting lost because of the goto? |
| 15:47.49 | pdtwork | i mean it says it sets it but still nothing on the other end |
| 15:47.54 | jtexter3 | I'm thinking it may be networking/wiring, but I don't know where to look to prove such a theory |
| 15:47.57 | Juggie | pdtwork, loose the spaces |
| 15:48.03 | Juggie | do exactally as i said |
| 15:48.08 | Juggie | Set(CALLERID(name)="HFB") |
| 15:48.50 | [TK]D-Fender | remove the quotes. |
| 15:49.07 | [TK]D-Fender | and NEVER ADD whitespace.... |
| 15:49.22 | Juggie | voip-info had an example w/ quotes. i wasnt sure which was right. |
| 15:49.37 | pdtwork | i want whitespace |
| 15:49.47 | pdtwork | li am trying to prepend something on the callerid |
| 15:49.56 | pdtwork | so the person that gets the call knows what "line" it came from |
| 15:50.09 | pdtwork | so the end result is really HFB REAL CALLERID |
| 15:50.44 | Juggie | there cant be any white space around the = |
| 15:51.13 | *** join/#asterisk Bentley (n=Bentley@S01060080c8135e6a.cg.shawcable.net) |
| 15:51.32 | pdtwork | <PROTECTED> |
| 15:51.40 | pdtwork | it works as long as it's past the = it seems |
| 15:52.42 | pdtwork | is it just me or does it seem like it should bitch instead of parroting your broken command back at you? |
| 15:52.48 | pdtwork | ok doing somethign that won't work |
| 15:53.15 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 15:53.28 | *** join/#asterisk postel (n=jp@wikimedia/Postel) |
| 15:54.08 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 15:54.16 | pdtwork | Juggie, thanks for you help |
| 15:55.37 | Inverted | mog: you around? |
| 15:55.48 | mog | never |
| 15:55.59 | murf | afrosheen: as to deb packages for 1.2 fixes, I am told that, for a price, a person in Digium support can generate one for you. |
| 15:56.22 | afrosheen | murf, that wasn't for me, I was reminding Luke-jr_ that I had asked a question earlier :) |
| 15:56.54 | *** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 15:57.14 | murf | OK, then, luke-jr_work: as to deb packages for 1.2 fixes, I am told that, for a price, a person in Digium support can generate one for you. |
| 15:57.43 | Inverted | mog: I upgraded to version 1.2.12.1 and the problem I described last night is still occuring, however the offset works fine with stream file |
| 15:58.01 | mog | interesting |
| 15:58.24 | Juggie | Inverted, whats the problem? |
| 15:58.39 | Inverted | i was going to submit a bug, but the mailer for bug registration was broken |
| 15:59.03 | Inverted | Juggie: I am supplying an offset for control stream file, but it just plays the sound file from the beginning |
| 15:59.17 | luke-jr_work | murf, heh, well, I've already got workarounds in place so I'll probably just keep using them until 1.4 is ready I guess |
| 15:59.17 | Juggie | via dialplan, agi? |
| 15:59.24 | Inverted | agi |
| 15:59.31 | *** join/#asterisk Qwell[] (i=qwell@unaffiliated/qwell) |
| 15:59.31 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 15:59.42 | Juggie | should be a simple bug |
| 16:00.45 | Inverted | calling them like: |
| 16:00.46 | Inverted | print "CONTROL STREAM FILE /tmp/tmpstream$random \"\" 40560\n"; $result = <STDIN>; |
| 16:00.46 | Inverted | <PROTECTED> |
| 16:01.07 | afrosheen | has anyone dealt with conferening delays since 1.2.11? |
| 16:01.08 | Inverted | stream file works, control stream file doenst |
| 16:01.08 | Juggie | unless you are providing a position thats beyond the end of the file |
| 16:02.02 | Inverted | Juggie: no, I checked that |
| 16:02.47 | Bentley | hello all, I have a box running that has a problem with multiple versions of asterisk starting after a certain period of time. Looking at the options that * is running as, it appears that safe_asterisk is responsible for starting the new instances. Anyone ever seen this? |
| 16:03.03 | *** join/#asterisk DrkShdw (n=scorpio@unaffiliated/drkshdw) |
| 16:03.11 | Juggie | Inverted, according to the code control stream file doesnt support an offset. |
| 16:04.14 | Inverted | hmmm I was going off of http://www.voip-info.org/wiki/view/control+stream+file |
| 16:04.30 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 16:04.46 | Juggie | right, but where there does it say it accepts an offset |
| 16:05.10 | Juggie | when you pass 40560 in there, all you are doing is setting the skipms to that, not setting an offset |
| 16:05.27 | *** join/#asterisk oQPa (i=Spain@1.Red-83-35-130.dynamicIP.rima-tde.net) |
| 16:05.59 | Inverted | Juggie: well what I am wanting it to do is start playing the file from that point |
| 16:06.29 | Juggie | and the users have to haev the ability to control file playback themselves? |
| 16:06.44 | *** join/#asterisk Jamez^7 (n=martini@modemcable131.214-131-66.mc.videotron.ca) |
| 16:07.08 | Inverted | speedup or slowdown yea |
| 16:07.26 | Juggie | then you are going to have to patch res_agi.c |
| 16:07.59 | Inverted | broken? |
| 16:08.27 | Juggie | its not broken |
| 16:08.32 | Juggie | its just not supported the way you want to do it |
| 16:09.05 | Inverted | Juggie: then why with stream file does <skipms> cause the file to skip that # of milliseconds, but not with control stream file? |
| 16:09.52 | *** join/#asterisk pph (n=pph@81.255.164.157) |
| 16:09.53 | Juggie | sigh, skipms = how far to skip in the file if the user presses fwd/back button |
| 16:10.00 | Juggie | not the same as offset which = start here. |
| 16:10.26 | Inverted | I see. well you can see where this definition is misleading: If <skipms> is provided then the audio will seek to sample offset before play starts |
| 16:10.50 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 16:11.23 | *** part/#asterisk Bentley (n=Bentley@S01060080c8135e6a.cg.shawcable.net) |
| 16:11.26 | Juggie | yes it would appear someone just coppied the stream file docs. |
| 16:12.28 | Juggie | if you have any programming knowledge at all, this shoudnt be a hard change, but since it would be disruptive it would never get accepted into 1.2/1.4 |
| 16:13.35 | Juggie | looks like you would really have to change two functions. |
| 16:13.47 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 16:13.54 | *** join/#asterisk mattchis (n=mattchis@216.54.143.242) |
| 16:14.30 | Juggie | mod, ast_control_streamfile to support an offset, and then handle_controlstreamfile to read the offset from "CONTROL STREAM FILE" command line and pass it in. |
| 16:16.37 | Inverted | one could set it to accept a skipms value as another optional argument at the end, that shouldnt be disruptive |
| 16:17.02 | Juggie | right, but then it would be two similar functions with different parameter orders |
| 16:17.04 | Juggie | would be dumb. |
| 16:17.12 | *** join/#asterisk oQPa (i=Spain@1.Red-83-35-130.dynamicIP.rima-tde.net) |
| 16:17.12 | *** join/#asterisk |Vulture| (n=_Vulture@101.222.121.70.cfl.res.rr.com) |
| 16:22.47 | Inverted | Juggie: passing a value of 40500 for skipms to stream file, one would think it should skip 40.5 seconds right? |
| 16:22.57 | *** join/#asterisk mattchis (n=mattchis@216.54.143.242) |
| 16:24.26 | Juggie | IF the file >that size and IF you press the skip button |
| 16:24.28 | Juggie | then yes. |
| 16:25.02 | xezz | some1 uses voipbuster ? |
| 16:25.17 | Inverted | Juggie: not control stream file, just stream file |
| 16:25.48 | Inverted | when I pass stream file the offset it starts playing from that point in the record automatically, without have to press a skip button |
| 16:25.55 | Juggie | right |
| 16:26.00 | Juggie | because its a totally different function |
| 16:26.06 | Juggie | and serves a different purpose |
| 16:26.32 | Juggie | if you want, implement your own functionality. |
| 16:26.41 | Inverted | exactly. now to my point. that value of 40500 causes it to skip about 4-5 seconds worth of audio |
| 16:26.46 | Juggie | right |
| 16:26.49 | Juggie | thats what it should do |
| 16:27.30 | Inverted | 4000ms = 4 secs |
| 16:27.59 | Juggie | right |
| 16:28.03 | Juggie | but offset is byte offset |
| 16:28.05 | Juggie | not ms. |
| 16:29.20 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 16:31.08 | xheliox | Can pridialplan= be set per call in extensions.conf? |
| 16:31.54 | Juggie | negative. |
| 16:31.57 | oliviert | guys, i'm becoming mad |
| 16:32.04 | *** join/#asterisk sloth (n=sloth@pool-162-83-147-88.ny5030.east.verizon.net) |
| 16:32.08 | Juggie | we have mad people in here all the time. |
| 16:32.09 | Juggie | no big deal. |
| 16:32.25 | oliviert | queues are working but, when the agent hang up the phone he can not hear the caller |
| 16:32.56 | oliviert | and if it transfer the call to another agent, or phone, then everything is working |
| 16:33.15 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.178) |
| 16:33.40 | Inverted | oliviert: if I was you, I'd ask for a refund! |
| 16:33.48 | oliviert | when the agent can not hear the caller, the caller can hear the agent but with crappy sounds |
| 16:34.10 | oliviert | Inverted: hehe, joking is good for health |
| 16:34.57 | oliviert | Inverted: any other idea, something specific to queues and config ? |
| 16:37.02 | StyleWarz | does asterisk_cdr write its stuff upon hangup? |
| 16:37.03 | Inverted | oliviert: i'd start with my config |
| 16:37.41 | *** join/#asterisk vader-- (n=johndoe@204.183.88.101) |
| 16:38.17 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.178) |
| 16:39.01 | [TK]D-Fender | oliviert: Sounds like a NAT issue. Describe what situation your phone is in, and then your server. Please do not hold back any info on this. |
| 16:39.38 | jake1932 | is anyone able to get the poly 650? |
| 16:40.13 | [TK]D-Fender | jake1932: I believe that it only becomes available at the end of the month. |
| 16:40.16 | murf | luke-jr_work: I think I just nailed the problem in trunk; I'll be committing it soon. On to the 1.2 & 1.4 side. |
| 16:40.35 | jake1932 | damn - i just read something that said it was released |
| 16:40.36 | [TK]D-Fender | jake1932: Then again I find it to be "overkill". Not worth the added expense over the 601 |
| 16:40.47 | oliviert | [TK]D-Fender: well NAT should not be the problem, because with IVR or Ring Group, i do not have this issue |
| 16:40.48 | jake1932 | it's got HD Voice!! |
| 16:41.06 | [TK]D-Fender | oliviert: IVR isn't you other phone. |
| 16:41.21 | [TK]D-Fender | jake1932: yuo = winnar! |
| 16:41.23 | jake1932 | but seriously, if anyone can get it, i'd take it |
| 16:41.29 | [TK]D-Fender | </sarcasm> |
| 16:41.53 | [TK]D-Fender | jake1932: Yeah I'm sure its better than all the other models, I just can't justy the cost. |
| 16:42.06 | jake1932 | neither can I uless it lands this deal |
| 16:42.14 | oliviert | [TK]D-Fender: yes, but IVR or Ring Group, are making the phone ring after they operate, and then communication is really good, crystal clear |
| 16:42.45 | jake1932 | heck - i'm fine with my linksys 921 |
| 16:42.50 | jake1932 | and 7960 |
| 16:42.54 | oliviert | But could it be a Context problem ? |
| 16:42.58 | *** join/#asterisk mattchis (n=mattchis@216.54.143.242) |
| 16:43.09 | [TK]D-Fender | oliviert: Listen up. If you have audio problems its your phones that are at fault. Not Queues or anything else. paste your sip.conf masking ONLY passwords |
| 16:43.22 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.178) |
| 16:46.07 | oliviert | [TK]D-Fender: http://pastebin.ca/201126 this is the sip_additional.conf where the sip devices are declared |
| 16:47.50 | oliviert | [Tk]D-Fender: http://pastebin.ca/201127 and the sip.conf |
| 16:50.51 | [TK]D-Fender | oliviert: where are your phones relative to *? |
| 16:51.06 | oliviert | [TK]D-Fender: and i also can add that when the agent hang up with X-Lite as SIP phone, it works, but when he hang up with Thomson ST2030, he can hear the caller and the caller got crappy sounds |
| 16:51.23 | [TK]D-Fender | oliviert: Also how are your calls coming in? |
| 16:51.36 | oliviert | [TK]D-Fender: from PSTN lines |
| 16:51.48 | oliviert | [TK]D-Fender: TDM04B |
| 16:52.04 | [TK]D-Fender | oliviert: Pastebin a call for me. |
| 16:52.07 | *** join/#asterisk luke-jr_work (n=luke-jr@adsl-70-128-250-252.dsl.ksc2mo.swbell.net) |
| 16:54.47 | oliviert | [TK]D-Fender: here is a pastebin of 2 calls Ii made few mins ago, the first one answer with X-lite, and the 2nd one with ST 2030 http://pastebin.ca/201137 |
| 16:54.52 | oliviert | let me set verbose to 10 |
| 16:57.51 | oliviert | [TK]D-Fender: here is a better pastebin with verbose 10 of a call http://pastebin.ca/201138 |
| 16:58.12 | [TK]D-Fender | oliviert: where are your phones relative to *? <- please answer this. |
| 16:58.27 | [TK]D-Fender | oliviert: and that last pastebin sucks. please remove the crap chars as well |
| 16:58.58 | oliviert | done, look at the last one |
| 16:58.59 | oliviert | SIP/101-08d57788 answered Local/101@from-internal-29ae,2 |
| 16:59.16 | oliviert | this is relative to the phone SIP/101 |
| 16:59.59 | oliviert | SIP/101 is a ST2030, and in that call, the agent can not hear the caller |
| 17:03.39 | afrosheen | any nat involved here |
| 17:04.12 | oliviert | [TK]D-Fender: the jitterbuffer for 101 is destroyed before the jitterbuffer for Zap/4-1 is created, and that seems to be the problem, but i don't know where to go to fix that, or what is the reason |
| 17:05.39 | oliviert | well i'm lost in that problem |
| 17:05.59 | *** part/#asterisk zparta (i=zparta@146.net90.skekraft.net) |
| 17:06.03 | [TK]D-Fender | oliviert: You laste pastebin was worthless. And answer the damned question already. |
| 17:06.10 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 17:06.44 | afrosheen | it's like troubleshooting with a deaf mute ;) |
| 17:07.06 | [TK]D-Fender | afrosheen: No.... mutes are quiet and easier to ignore ;) |
| 17:07.13 | oliviert | [TK]D-Fender: then it's a nat problem you said ? |
| 17:08.03 | afrosheen | lol |
| 17:08.03 | afrosheen | oliviert, one way audio is NAT related most of the time in my experience |
| 17:08.03 | [TK]D-Fender | oliviert: I strongly suspect that, but I'd have confirmation if you WOULD JUST ANSWER MY DAMNED QUESTION |
| 17:08.36 | [TK]D-Fender | oliviert: Ouvre tes oreilles tabarnac! |
| 17:08.50 | [TK]D-Fender | :O |
| 17:09.00 | oliviert | [TK]D-Fender: c'est ce que je fais, mais trouve pas le *? dont tu parles |
| 17:09.57 | [TK]D-Fender | oliviert: J'attends le description COMPLET de tous-sce-qui-est reseaux entre ton serveur pis ton maudite telephone! |
| 17:10.49 | oliviert | ils sont sur le meme segement: serveur: 10.160.2.199 tel: 10.160.2.194 |
| 17:11.48 | oliviert | tous les 2 sur un switch catalyst 2950 |
| 17:11.54 | oliviert | => cisco |
| 17:12.07 | oliviert | [TK]D-Fender: c'est la rÃĐponse a la question ? |
| 17:13.03 | *** join/#asterisk pifiu (n=someone@216.5.79.1) |
| 17:13.26 | pifiu | any idea when 1.4 stable comes out? |
| 17:14.12 | *** join/#asterisk drcode (n=chatzill@87.69.59.186.cable.012.net.il) |
| 17:14.14 | drcode | hi all |
| 17:14.17 | drcode | again |
| 17:20.07 | drcode | with http://www.e164.org/ I can find number on internet? |
| 17:20.16 | pdtwork | xten 10.x.x.x --- iptables --> (internet) --- linksys --> asterisk 192.168.x.x whats the magic trick to getting phone call across |
| 17:21.05 | drcode | hi pdtwork |
| 17:21.15 | pdtwork | and xten could be sjphone, expresstalk whatever.. doesn't matter |
| 17:21.36 | pdtwork | drcode, hello |
| 17:21.56 | *** join/#asterisk tuck3r (n=tuck3r@unaffiliated/tuck3r) |
| 17:21.58 | drcode | pdtwork: can U help me with testing my server? |
| 17:22.03 | *** join/#asterisk mattchis (n=mattchis@216.54.143.242) |
| 17:22.13 | pdtwork | what does that consist of? |
| 17:22.19 | pdtwork | do i have to sacrifice monkeys? |
| 17:22.41 | pdtwork | if so you'll have to pay me |
| 17:22.46 | drcode | k |
| 17:22.51 | pdtwork | i don't kill monkeys for free |
| 17:22.53 | drcode | I just want to check if my server work |
| 17:22.54 | jake1932 | you'd sacrific monkeys for money? |
| 17:22.56 | drcode | k |
| 17:23.09 | tuck3r | jake1932: you wouldn't? |
| 17:23.12 | pdtwork | jake1932, if the price was right |
| 17:23.19 | drcode | any way, what e164.org can help? |
| 17:23.28 | jake1932 | hmm.. |
| 17:23.32 | drcode | I thote I am in open source channel |
| 17:23.34 | jake1932 | i couldn't |
| 17:23.51 | pdtwork | drcode, what do you need |
| 17:23.57 | pdtwork | i'm just being an idiot |
| 17:24.04 | drcode | well |
| 17:24.10 | drcode | I have setup ast server |
| 17:24.23 | drcode | I have linux gw + ast server and lan |
| 17:24.27 | oliviert | [TK]D-Fender: another pastebin call with verbose set to 17, i don't know if it could help http://pastebin.ca/201165 |
| 17:24.28 | drcode | I connect to ast from my lan |
| 17:24.39 | drcode | I need some one to try connect from outside |
| 17:25.05 | drcode | and to check if I can speak with pepole out side |
| 17:25.28 | *** part/#asterisk jake1932 (n=Administ@pool-68-236-19-153.phil.east.verizon.net) |
| 17:26.13 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 17:26.18 | *** join/#asterisk drig (n=drig@c-67-161-144-187.hsd1.co.comcast.net) |
| 17:26.20 | luke-jr_work | anyone at Digium here want to know what their network problem is? |
| 17:26.27 | luke-jr_work | file? |
| 17:26.32 | file | sure |
| 17:26.37 | luke-jr_work | "Since the packet is too big for your connection, your upstream router (run by your ISP) will send a ICMP 3:4 (fragmentation needed) packet back to the remote WWW / FTP server. Within this packet is a recommended smaller MTU size to retry with. The problem is that either your local upstream router, some router between you and the remote server, or the remote HTTP / FTP server is either misconfigured or has a firewall in |
| 17:26.37 | luke-jr_work | <PROTECTED> |
| 17:26.47 | *** join/#asterisk sloth (n=sloth@64.3.170.100.ptr.us.xo.net) |
| 17:27.09 | luke-jr_work | it affects dialup or DSL users |
| 17:27.49 | pdtwork | drcode, i am trying to do the same thing at the moment |
| 17:27.55 | drcode | k |
| 17:28.13 | drig | Hi, I'm a software developer. I'm trying to add dial-out/IVR capabilities to my app. Can someone tell me if there is a way to use an asterisk server to do this without the complexities of SIP? I've tried with JAIN-SIP (my app is in Java) and it's just too complex. Just an FM to read would be great! |
| 17:28.18 | pdtwork | sip gets across but rtp gets blocked on the client side firewall because the firewall has no damn clue what to do with that port traffic |
| 17:28.19 | luke-jr_work | file, so find out who's blocking that ICMP and yell at em ;) |
| 17:28.28 | file | me? |
| 17:28.31 | file | IT can do that :P |
| 17:28.33 | luke-jr_work | sure, why not? =p |
| 17:28.37 | luke-jr_work | well, tell IT what I told you |
| 17:28.49 | luke-jr_work | blocking ICMP is very BAD |
| 17:29.06 | tuck3r | file: a software dev company has IT staff? shameful :P |
| 17:29.25 | file | Digium makes hardware... does support... of course we have an IT staff |
| 17:29.54 | wunderkin | computers should run themselves |
| 17:30.36 | [TK]D-Fender | oliviert: I don't see the dial command being called or any other dialplan application your verbose is bad |
| 17:31.49 | oliviert | [TK]D-Fender: it was set to 17 in the last one |
| 17:33.25 | tuck3r | oliviert: is that freepbx? |
| 17:33.56 | oliviert | tuck3r: yes, but the olg come from asterisk CLI |
| 17:34.42 | tuck3r | #freepbx? |
| 17:35.01 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 17:35.21 | [TK]D-Fender | oliviert: Well something was bad because it showed nothig ov any use |
| 17:35.27 | *** join/#asterisk afrosheen (n=cj@txprotoa2.august.net) |
| 17:35.48 | oliviert | [TK]D-Fender: here is a complete log from beginning to end : http://pastebin.ca/201177 |
| 17:37.44 | xheliox | Hrm. |
| 17:38.37 | oliviert | i got to go, i'll come back on this problem late, thanks for all guys, ciao |
| 17:39.34 | xheliox | http://bugs.digium.com/view.php?id=5331 -- A quick scan of chan_zap indicates this isn't included in the latest stable version of Asterisk, right? |
| 17:41.26 | *** join/#asterisk cekc (n=cekc@66-17-9-220.biz.bkfd.arrival.net) |
| 17:41.57 | *** join/#asterisk sp0n9e- (n=sp0n9e@phpurge.com) |
| 17:42.00 | cekc | how do I match a dialplan to any number that is dialed? also does putting a * in a dialplan act as an asterisk or a wildcard |
| 17:42.11 | drcode | carrar: hi |
| 17:42.14 | tuck3r | _X. |
| 17:42.25 | sp0n9e- | what's the command to show realtime sip users? they don't show up in sip show users |
| 17:45.23 | drig | doh. sorry to bother you. asterisk-java was what I needed (didn't think to search freshmeat.net until just now). thanks! |
| 17:45.34 | ellisdee | man, those asterisk's error messages can cry wolf. |
| 17:45.53 | ellisdee | or can be decieving. |
| 17:46.20 | ellisdee | Oct 13 12:46:07 NOTICE[30148]: app_dial.c:1040 dial_exec_full: Unable to create channel of type 'Sip' (cause 3 - No route to destination) |
| 17:50.58 | ellisdee | basically the phone was not powered on. |
| 17:51.16 | ellisdee | for that particular ext that was being called. |
| 17:52.32 | *** join/#asterisk vivis (n=vivis@81.22.63.104) |
| 17:52.51 | sbingner | waah they're hijacking my show vs list thread lol, not that I disagree with the cisco-like behavior suggested... it's just OT heh |
| 17:53.23 | drcode | any one can try register my ast server? |
| 17:53.35 | drcode | I want to check if my fw dosnt block ? |
| 17:56.11 | *** join/#asterisk _alex_mx_ (n=alex@dsl-200-67-125-45.prod-empresarial.com.mx) |
| 18:08.20 | *** join/#asterisk C6Vette (n=info@72-166-37-114.dia.static.qwest.net) |
| 18:08.55 | *** join/#asterisk justinu|laptop (n=Justin@12.44.122.130) |
| 18:09.17 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
| 18:09.55 | Defraz | I have an ATA that registers with my Asterisk Box |
| 18:10.44 | Defraz | I am using ULAW so I can fax |
| 18:11.00 | Defraz | on this ATA ULAW is needed |
| 18:11.14 | Defraz | on my other ATA's I use g729 and it works great |
| 18:11.20 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 18:11.46 | Defraz | No my question is If I allow ULAW and g729 to my VoIP provider it always transecodes |
| 18:11.59 | Defraz | to g729 can I maek it do what the ATA is. |
| 18:14.31 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 18:25.11 | *** join/#asterisk ToTo (n=ToTo@host203-49-dynamic.0-87-r.retail.telecomitalia.it) |
| 18:29.00 | *** join/#asterisk fiber0pti (n=John@207.114.199.107) |
| 18:31.41 | ellisdee | exten => s,1,Voicemail(u500) |
| 18:31.48 | ellisdee | how do i add a timer to this line |
| 18:31.56 | ellisdee | i want the line to ring 4-5 times |
| 18:32.03 | ellisdee | then the voicemail to play |
| 18:34.43 | ellisdee | err |
| 18:34.50 | ellisdee | that wasn't enough nfo =P |
| 18:35.26 | CunningPike | ellisdee: The place to do that is in your Dial() command........ |
| 18:36.30 | ellisdee | let me elaborate a little more. |
| 18:36.39 | ellisdee | i have a context with the hrs |
| 18:36.49 | ellisdee | i also have a context for closed. |
| 18:37.10 | ellisdee | in my context for closed. |
| 18:37.20 | ellisdee | i have didnumber,1,Goto(mainnumberclosed,s,1) |
| 18:38.27 | ellisdee | i want it to ring a couple times instead of just shoot straight to voice mail |
| 18:38.41 | ellisdee | when my store is closed |
| 18:39.11 | *** part/#asterisk jarg (n=jarg@200.56.225.61) |
| 18:39.15 | xheliox | I'm trying to create an account on the Digium bug tracker, but it's giving me an error about not being able to send an e-mail (to me). Who should I contact that about that? |
| 18:44.33 | file | xheliox: we already know |
| 18:44.41 | xheliox | file: oh. :) |
| 18:48.11 | *** join/#asterisk cekc (n=cekc@rrcs-24-199-36-210.west.biz.rr.com) |
| 18:48.30 | *** join/#asterisk xnon (i=xnon@200.82.222.85) |
| 18:49.04 | *** join/#asterisk Egonis (n=root@TOROON12-1177740455.sdsl.bell.ca) |
| 18:49.22 | Egonis | When I try to 'wanrouter start' I get tons of zt_* unknown symbols in dmesg, what am I missing? |
| 18:49.24 | cekc | is there a way to have an extension match for null? My system works fine with the telco dials the DID, but can I have it send calls to an extension if no DID is dialed? |
| 18:50.07 | *** part/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
| 18:52.31 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 18:53.22 | luke-jr_work | does anyone use Aastra 480i-CT? |
| 18:54.51 | Egonis | luke-jr_work: I have 3 of them |
| 18:54.51 | Inverted | is there a way with asterisk to play a file back to the user but have it speed up or slow down? |
| 18:55.10 | Egonis | luke-jr_work: They work very nicely |
| 18:55.54 | luke-jr_work | Egonis, can the handsets be seperated? |
| 18:56.00 | luke-jr_work | Egonis, eg, have them on seperate SIP accounts |
| 18:56.16 | luke-jr_work | or do all the handsets share the same accounts? |
| 18:58.23 | [TK]D-Fender | luke-jr_ : What I'm told is they incrementally steal call appearances away from their "host" making the base phone less and less useful. |
| 18:58.28 | Egonis | luke-jr_work: No, but each handset can select a line, of which you have up to four on the 480iCT Base |
| 18:59.05 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 18:59.07 | Egonis | [TK]D-Fender: Not true, they are simply accessory to the base, the base controls all calls and whatnot |
| 18:59.08 | [TK]D-Fender | Egonis: So basically it steals a line key from the bottom for its won use? |
| 18:59.16 | Egonis | [TK]D-Fender: Precisely |
| 18:59.27 | Egonis | [TK]D-Fender: There are lines 1-4, public, and private and intercom |
| 18:59.42 | [TK]D-Fender | Egonis: And if the base triess to use that line key what happens to the handset that would associate with it normally? |
| 18:59.49 | Egonis | [TK]D-Fender: Public is for conf, private is between sets, and intercom is obvious |
| 18:59.55 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 18:59.56 | *** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
| 19:00.01 | Egonis | [TK]D-Fender: It will say 'line in use' |
| 19:00.12 | Egonis | [TK]D-Fender: And that goes both ways |
| 19:00.21 | Egonis | [TK]D-Fender: Until you put the call on hold |
| 19:00.28 | [TK]D-Fender | Egonis: So basically you have to tell the user "don't use this button or the handsets will be useless"? |
| 19:01.04 | Egonis | [TK]D-Fender: no, the handset will pick the next available line if the desired one is busy |
| 19:01.23 | [TK]D-Fender | Egonis: And a handset can only handle 1 "line"? Can it handle multiple calls on it? |
| 19:01.47 | Egonis | [TK]D-Fender: It can handle multiple calls, yes, but the 480iCT base is actually doing the work, it's not that complicated |
| 19:02.14 | [TK]D-Fender | Egonis: This sounds like a real headache when trying to give the handset a different "identity" from the bse. Its a race condition. |
| 19:02.16 | Egonis | [TK]D-Fender: Think of the handset as a simple remote from the base, that's all it is, no real configurability in the handset itself |
| 19:02.48 | Egonis | [TK]D-Fender: You can't really give any handset separate identity, they are intended as an extension of the base only, if you want separate identity, I suggest a simple WiFi Phone |
| 19:03.13 | [TK]D-Fender | Egonis: Yeah I suspected as much, and unfortunately, WiFi phones SUCK. |
| 19:03.26 | [TK]D-Fender | Egonis: But thanks for the first-hand recounting of their usability |
| 19:03.34 | Egonis | :) |
| 19:03.37 | luke-jr_work | Egonis, but if line 1 rings, all handsets must ring? :| |
| 19:03.44 | Egonis | correct |
| 19:03.48 | Egonis | or line 3, etc |
| 19:04.09 | Egonis | they make great home ip phones, I have the base in my office, and the handset goes with me, works real nice. |
| 19:04.18 | drcode | [TK]D-Fender: wifi dosnt work good? |
| 19:04.50 | drcode | I mean wifi phones? |
| 19:06.28 | denon | [TK]D-Fender: have you played with the new linksys one? |
| 19:06.31 | denon | well, not soo new .. but .. |
| 19:06.37 | [TK]D-Fender | Egonis: Yeah, thats basically what it is........ a "home" solution. Or for a guy who works at a desk, but needs mobility too and save you buying a seperate ATA and having to dual-dial all the time |
| 19:06.49 | [TK]D-Fender | denon: I have a WIP-300. One word : BLEH! |
| 19:07.05 | denon | [TK]D-Fender: isnt that their old one? |
| 19:07.07 | [TK]D-Fender | denon: No transfer or conference, interface is slow. |
| 19:07.17 | [TK]D-Fender | denon: New. Same age as the 330 |
| 19:07.29 | [TK]D-Fender | denon: 330 is full WinCE, the 300 is its own deal. |
| 19:07.35 | denon | yeah, but the 330 has a more powerful cpu etc |
| 19:07.50 | [TK]D-Fender | denon: And as far as I can see, not call waiting or much of anything. DUMB! |
| 19:08.05 | [TK]D-Fender | denon: Yeah well noone should even make a phone as useless as the WIP-300 |
| 19:08.05 | denon | how 'bout the dlink? |
| 19:08.32 | [TK]D-Fender | denon: From what I hear its effectively the same phone in clam-shell, but no personal experience or reports to confim |
| 19:09.41 | *** join/#asterisk afrosheen (n=cj@txprotoa2.august.net) |
| 19:09.59 | Egonis | I am getting tons of zt_{name it} unknown symbols in dmesg, any ideas? |
| 19:10.12 | *** join/#asterisk JakBeatZ (n=JakBeatZ@beta.arionetworks.ca) |
| 19:10.26 | luke-jr_work | know any wireless (non-WiFi or interfering-with-Wifi) IP phones? |
| 19:10.36 | luke-jr_work | preferably *just* a handset standalone |
| 19:10.47 | denon | luke-jr_: engenius? |
| 19:11.03 | denon | or pretty much any 900mhz phone, or 5ghz |
| 19:11.20 | denon | personally, I kinda like the 900mhz band .. not much down there anymore |
| 19:11.27 | *** join/#asterisk bhrobinson (n=brobinso@northtx-cust-rtr.telwestonline.com) |
| 19:11.28 | denon | but obviously security is a bit of an issue |
| 19:12.11 | luke-jr_work | 5ghz interferes with WiFi, no? |
| 19:12.19 | denon | no . . |
| 19:12.20 | denon | not b/g anyway |
| 19:12.35 | luke-jr_work | supposedly it does in my dad's case |
| 19:12.51 | denon | does your dad have a wap that he got free after rebate? |
| 19:12.59 | bhrobinson | I have an asterisk server that will not start... it is unfortunately in another country and I am trying to hit it remotely. |
| 19:12.59 | denon | that probably interferes with air traffic control and nasa too |
| 19:12.59 | luke-jr_work | no, WRT54GL |
| 19:13.09 | luke-jr_work | from newegg |
| 19:13.22 | bhrobinson | can anyone tell me what to do for a "Unregistered channel x" on startup of asterisk? |
| 19:13.29 | tzanger | [TK]D-Fender: I think there's something fucked in the default polycom dialplan.. *98<send> and it sends just * to asterisk |
| 19:13.30 | denon | I wouldnt call those broadcom radios top notch .. |
| 19:13.30 | luke-jr_work | anyway, where would I find a 900mhz SIP phone? :) |
| 19:13.34 | denon | but they shouldnt interfere |
| 19:13.39 | denon | oh, you want sip? |
| 19:13.45 | denon | good luck with that |
| 19:14.03 | denon | just get a regular cordless phone and an ATA |
| 19:14.18 | luke-jr_work | that won't handle multiple lines |
| 19:14.19 | bhrobinson | luke-jr_work: voipsupply has them... but they are outrageous in expense |
| 19:14.24 | denon | luke-jr_: sure they do .. |
| 19:14.31 | denon | you can do like 4 line cordless phones |
| 19:14.37 | luke-jr_work | and 2 ATAs each? |
| 19:14.41 | luke-jr_work | that's insane |
| 19:14.49 | denon | well, a multi-port ata, or jack em into a channel bank |
| 19:15.04 | denon | how many calls do you plan on taking at a time? |
| 19:15.14 | denon | you can use call waiting to accomplish most of what people want |
| 19:15.18 | denon | everything else should happen in a meetme |
| 19:15.27 | denon | and you can obviously use your dialplan to handle as many inbound lines as you want |
| 19:16.16 | luke-jr_work | bhrobinson, where? |
| 19:16.34 | bhrobinson | can anyone tell me what to do for a "Unregistered channel x" on startup of asterisk? |
| 19:17.09 | bhrobinson | luke-jr_work: http://www.voipsupply.com/index.php?cPath=270_279 |
| 19:17.26 | luke-jr_work | bhrobinson, I only see wifi thre |
| 19:17.27 | luke-jr_work | there |
| 19:17.35 | *** join/#asterisk sloth (n=sloth@mail.fex.org) |
| 19:18.02 | denon | bhrobinson: he wants a proprietary wireless @900mhz, but doing sip to the handset |
| 19:18.41 | bhrobinson | oh my bad |
| 19:19.07 | denon | I'm sure someone used to do a proprietary thing like that, before wifi got so cheap |
| 19:19.28 | denon | but I doubt anyone does now -- APs are such an easy way to get good coverage |
| 19:19.51 | luke-jr_work | denon, everyone says WiFi phones suck |
| 19:20.06 | bhrobinson | you want 900MHz to the base, instead of wired? |
| 19:20.07 | denon | luke-jr_: a 900mhz proprietary phone would suck equally |
| 19:20.10 | bhrobinson | for distance? |
| 19:20.16 | luke-jr_work | bhrobinson, no idea |
| 19:20.26 | denon | bhrobinson: no, he wants to get away from 2.4ghz |
| 19:21.06 | denon | [TK]D-Fender: so what about utstarcom and hitcachi? |
| 19:21.18 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:21.24 | denon | those utstarcom ones are cheap .. almost too cheap |
| 19:23.36 | luke-jr_work | ? |
| 19:23.52 | ms345 | anyone know if callwaiting=no in a valid sip.cnf entry? |
| 19:23.59 | ms345 | is valid? |
| 19:25.12 | bhrobinson | so no one have an idea on the unregistered channel? |
| 19:25.18 | [TK]D-Fender | denon: UTstarcom seemed kinda flakey, and I have a Hitachi WIP-5000 that I am still unable to get running. |
| 19:25.43 | [TK]D-Fender | ms345: Lookup "config sip.conf" on the WIKI. All your answers are there |
| 19:25.48 | ms345 | thx |
| 19:25.53 | denon | [TK]D-Fender: hmm .. did you have the new hardware rev of the utstarcom? does 802.11g etc |
| 19:26.51 | denon | [TK]D-Fender: a $120 wifi phone with 4 hour talk sure is appealing |
| 19:26.52 | [TK]D-Fender | denon: No, it was one of my clients who had them. Constant time-outs, etc. Sound quality was OK from what he said. Never asked about functionality. |
| 19:27.05 | [TK]D-Fender | denon: Thats how the fuckers suck you in ;) |
| 19:27.09 | denon | hehh |
| 19:27.47 | [TK]D-Fender | denon: I recall encountering some bad reviews but encourage you to search for yourself. |
| 19:27.48 | denon | timeouts as in sip timeouts? |
| 19:27.56 | [TK]D-Fender | denon: Yeah. |
| 19:28.09 | denon | seems like you could just crank them up to be more aggressive .. |
| 19:28.20 | denon | I mean, you've got 54Mbps to play with |
| 19:28.41 | sp0n9e- | i'm having issues with asterisk trying to answer the line when no one is calling. |
| 19:29.09 | [TK]D-Fender | sp0n9e-: Pastebin your zapata.conf |
| 19:29.29 | denon | [TK]D-Fender: dunno, I really want to replace some old siemens cordless phones here .. they're good phones, but they've had a good life too |
| 19:29.31 | sp0n9e- | i'll grep out he lines that are comments |
| 19:29.45 | Egonis | when I try to start zaptel, I get 'unable to open master device '/dev/zap/ctl' although I have tried chmod'ding 777 /dev/zap/* -R |
| 19:30.22 | *** join/#asterisk DanTMG (n=danielga@124-168-31-109.dyn.iinet.net.au) |
| 19:30.51 | *** join/#asterisk _DAW (n=_DAW@adsl-222-12-239.msy.bellsouth.net) |
| 19:31.09 | _DAW | hello everyone |
| 19:32.07 | *** join/#asterisk DarKnesS_WolF (n=wolf@217.54.209.1) |
| 19:32.12 | denon | [TK]D-Fender: hmm, reviews I'm seeing are kind of indifferent .. I wonder if it says anything that vonage is using em |
| 19:32.58 | [TK]D-Fender | denon: Best to ignore that. |
| 19:33.32 | denon | hard to say .. odds are they dont want to support something that's total crap |
| 19:34.35 | sp0n9e- | [TK]D-Fender: http://paste.devpipe.com/1136 <<< zapata.conf |
| 19:34.38 | ms345 | [TK]D-Fender - thx, call-limit=1 worked for me.... and that answers why I couldn't search and find sip.conf callwaiting=no |
| 19:36.07 | tzanger | [TK]D-Fender: what do you use for a *polycom* dialplan? dialing *98 seems to only send the "*" to asterisk |
| 19:37.25 | [TK]D-Fender | sp0n9e-: Kill 6,10-16. But I'm not sure that will affent the "big picture". The rest looks fine. Could be shaky lines to someone is calling you and hanging up around the time CID is being passed. |
| 19:37.57 | denon | [TK]D-Fender: I find it hard to base much on people's opinions on the web .. especially when they're asking, "Does this phone come with free long distance?" |
| 19:38.01 | [TK]D-Fender | tzanger: The only dialplan worth having : "x.T|*.T|#.T" |
| 19:38.02 | sp0n9e- | well, CID isn't present (which i figured out after pluggin in a CID phone to the line) |
| 19:38.08 | sp0n9e- | so, i have to have a talk with the phone co. |
| 19:38.22 | denon | and saying, "you would probably still want a cell phone, as you cant get wifi everywhere" |
| 19:38.33 | [TK]D-Fender | sp0n9e-: IF you don't have CID, set it to NO then. that will help debugging. |
| 19:38.41 | [TK]D-Fender | sp0n9e-: And let you answer faster. |
| 19:38.50 | tzanger | [TK]D-Fender: never used this stuff on the phone :-) |
| 19:38.54 | sp0n9e- | ah, so that's why the delay on answering |
| 19:38.57 | sp0n9e- | i thought i had it |
| 19:39.10 | sp0n9e- | the box isn't local to me anymore, so i've been running around |
| 19:39.21 | [TK]D-Fender | tzanger: That dialplan is my "STFU and take whatever I feel like giving you" strategy |
| 19:39.26 | tzanger | hahaha |
| 19:39.58 | tzanger | <PROTECTED> |
| 19:40.01 | tzanger | ? |
| 19:40.22 | [TK]D-Fender | tzanger: Endpoint dialplans don't do much for me. means tweaking 2 sides. also there is no "safe" value in my mind. better just timeout everything or wait for "send" |
| 19:40.33 | tzanger | yeah I do agree with that 100% |
| 19:40.36 | [TK]D-Fender | tzanger: You need to fix your match handling as well |
| 19:40.39 | tzanger | asterisk matching is enough :-) |
| 19:41.01 | tzanger | impossibleMatchHandling needs to be set to 1 (default is 0) |
| 19:41.02 | [TK]D-Fender | tzanger: Exactly... why force yourself to do it in multiple places? INSANITY.... |
| 19:41.21 | [TK]D-Fender | tzanger: No... "2" and removeendofdial="0" |
| 19:43.12 | denon | [TK]D-Fender: hmm .. these zyxels let you do ad-hoc wireless .. wonder if you could hack em up to do WDS <G> |
| 19:43.15 | denon | that'd be a nice intercom system |
| 19:44.08 | [TK]D-Fender | tzanger: The last comment was so that you can have a TRAILING "#" as part of your dial string. its necessary to enforce the "STFU" method |
| 19:44.38 | tzanger | jesus... these phones are trying ot be TOO smart |
| 19:46.19 | [TK]D-Fender | tzanger: They're only too smart if you're not smart enough ;) |
| 19:46.44 | tzanger | as I said... too smart :-) |
| 19:47.15 | [TK]D-Fender | :O |
| 19:48.29 | ellisdee | how do i reload vmail config from cli? |
| 19:50.32 | *** join/#asterisk DrkShdw (n=scorpio@unaffiliated/drkshdw) |
| 19:51.52 | tzanger | sp0n9e-: I get that on the last two FXO channels on my channel bank |
| 19:52.55 | sp0n9e- | well...no more random calls |
| 19:53.00 | [TK]D-Fender | ellisdee: "reload app_voicemail.so" |
| 19:53.04 | sp0n9e- | just a lot of CID errors, trying to track down where it's enabled |
| 19:53.43 | [TK]D-Fender | sp0n9e-: "usecallerid=no", "callerid=INCOMING" |
| 19:53.57 | ellisdee | fender, thanks |
| 19:54.30 | *** join/#asterisk _alex_mx_ (n=alex@dsl-200-67-125-45.prod-empresarial.com.mx) |
| 19:58.19 | [hC] | anyone have any clue as to why after starting asterisk with a pri in, thats up, and it claims to see just fine, i dont get any messages regarding restarting the b/d channels like i usually do? |
| 19:58.19 | *** join/#asterisk BigBadHoss[1] (n=hoss@24-178-92-14.static.dctr.al.charter.com) |
| 19:58.37 | BigBadHoss[1] | hey anyboady avaliable to help me setup up a polycom mgcp? |
| 19:58.43 | BigBadHoss[1] | or convert it to sip? |
| 19:58.55 | *** join/#asterisk __freedom__lover (n=eduardo@clipper.provale.com.br) |
| 20:00.57 | *** join/#asterisk jm|home (n=jamiem@dilbert.jamiem.com) |
| 20:01.41 | *** join/#asterisk _alex_mx_ (n=alex@dsl-200-67-125-45.prod-empresarial.com.mx) |
| 20:02.02 | *** join/#asterisk __freedom__lover (n=eduardo@clipper.provale.com.br) |
| 20:02.10 | [TK]D-Fender | BigBadHoss[1]: What model #? |
| 20:02.55 | sp0n9e- | mmm, no more random calls |
| 20:03.02 | jm|home | so I managed to get firmware for my Cisco 7912 |
| 20:03.26 | jm|home | I can call FROM it but when I try to call TO it I get (cause 3 - No route to destination) |
| 20:04.12 | *** join/#asterisk __freedom__lover (n=eduardo@clipper.provale.com.br) |
| 20:04.20 | __freedom__lover | hi all |
| 20:04.46 | [TK]D-Fender | jmls: Its either not registered, or has a timeout qualify against it, or is simply NAT'd with a keep-alive so no path |
| 20:04.56 | jm|home | hmm |
| 20:04.58 | [hC] | hmm. trying to dial out a pri that im not even sure is fully lit up, i get Call failed to go through, reason 8 |
| 20:05.13 | [hC] | I dont have any alarms, is it possible that the loop is good, but the telco hasnt put any service on it yet? |
| 20:07.16 | BigBadHoss[1] | its a 500 |
| 20:08.13 | BigBadHoss[1] | u still there fender |
| 20:08.28 | [TK]D-Fender | BigBadHoss[1]: Ok, download the SIP firmware and bootrom you want to run on it, and go set up a provisioning server and it should replace the apps pretty painlessly |
| 20:08.40 | BigBadHoss[1] | i have it setup like that |
| 20:08.56 | BigBadHoss[1] | i already have a sip phone running, just bought 9 mgcp :( |
| 20:08.57 | *** join/#asterisk oliviert (n=olivier@82.245.68.209) |
| 20:09.03 | BigBadHoss[1] | at least they were cheap :) |
| 20:09.19 | sp0n9e- | man, still getting random phone calls that don't exist. |
| 20:09.32 | BigBadHoss[1] | will it work through manual configuration, or do i need to do dhcp |
| 20:09.40 | BigBadHoss[1] | it uploads a log file too |
| 20:10.05 | [TK]D-Fender | BigBadHoss[1]: You can do it manual just fine |
| 20:10.35 | BigBadHoss[1] | lemme look at my log file |
| 20:10.46 | jm|home | [TK]D-Fender: you're right: it's not registering :| |
| 20:11.01 | BigBadHoss[1] | Could not get bootrom.ver. |
| 20:11.16 | sp0n9e- | wtf, it's happening almost every minute now. |
| 20:12.16 | BigBadHoss[1] | do i need to rename sip.ver to bootrom.ver? |
| 20:12.48 | BigBadHoss[1] | or bootrom.ld to bootfrom.ver? |
| 20:13.17 | [TK]D-Fender | BigBadHoss[1]: no, you should have your own ver file. |
| 20:13.24 | BigBadHoss[1] | from where? |
| 20:13.40 | BigBadHoss[1] | the sip or bootrom file? |
| 20:13.40 | [TK]D-Fender | download BR 2.6.1 for this phone first. that will have both files. I believe after that point you no longer need the .ver file |
| 20:13.46 | [TK]D-Fender | bootrom. |
| 20:13.58 | BigBadHoss[1] | k lemme check brb |
| 20:14.46 | sp0n9e- | [TK]D-Fender: still getting random calls |
| 20:15.03 | sp0n9e- | random rings really |
| 20:15.27 | [TK]D-Fender | sp0n9e-: Ok, NOW bitch at your telco :) |
| 20:15.45 | [TK]D-Fender | sp0n9e-: Maybe plug an analog phone in parallel with that line to test. |
| 20:15.48 | sp0n9e- | doesn't happen on the a physical phone though |
| 20:15.51 | sp0n9e- | did that |
| 20:16.02 | sp0n9e- | unless i called, it wouldn't pick up |
| 20:16.10 | [TK]D-Fender | sp0n9e-: Then the card may be spastic or not well set for your region. |
| 20:16.17 | sp0n9e- | or wouldn't ring at all, despite the console saying "Phone is ringing" |
| 20:16.37 | sp0n9e- | it's the US and i'm using sangoma card |
| 20:16.43 | sp0n9e- | the funny part it was working fine before |
| 20:17.00 | *** join/#asterisk synthetiq (n=tampon@66.159.21.2) |
| 20:17.18 | synthetiq | why would i not be able to hear any .wav fiels playing? |
| 20:17.20 | synthetiq | files |
| 20:18.10 | [TK]D-Fender | synthetiq: Your description is less than useful. |
| 20:19.22 | synthetiq | i dial the voicemailmain |
| 20:19.24 | aydiosmio | Why would my fingers be cold? |
| 20:19.24 | synthetiq | no sound |
| 20:19.38 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 20:19.39 | synthetiq | i have multiple extension to play wav files |
| 20:19.41 | synthetiq | no dice |
| 20:19.46 | synthetiq | i can make outbound calls |
| 20:19.54 | synthetiq | but i jsut cant hear wav files on my server |
| 20:19.58 | synthetiq | could it be permissions |
| 20:20.25 | [TK]D-Fender | synthetiq: Possible. You know..... have you considered SHOWING us a failed call? |
| 20:21.39 | *** join/#asterisk lorinc (n=ang@caracas-2476.adsl.interware.hu) |
| 20:22.16 | synthetiq | calls do not fail |
| 20:22.23 | synthetiq | it says the wav file is playing |
| 20:22.30 | synthetiq | but i hear no sound |
| 20:23.20 | BigBadHoss[1] | alright back |
| 20:23.47 | aydiosmio | synthetiq: what codec is your call? |
| 20:23.57 | BigBadHoss[1] | it now says cant contact provisioning server |
| 20:24.03 | BigBadHoss[1] | after it downloaded the new bootrom |
| 20:24.47 | synthetiq | g729 |
| 20:25.05 | aydiosmio | hahaha |
| 20:25.14 | aydiosmio | how many g729 licenses do you have? |
| 20:25.53 | synthetiq | 15 |
| 20:25.55 | synthetiq | but |
| 20:26.05 | Dossy | so, is anyone here using X-Lite as their softphone with Asterisk? |
| 20:26.06 | synthetiq | i cannot har the wav files when i call in via zap |
| 20:26.10 | synthetiq | sp that is not the problem |
| 20:26.12 | Dossy | I'd really like to ask if you've gotten DTMF working. |
| 20:26.13 | synthetiq | sp=so |
| 20:26.18 | Dossy | please /msg me. thanks. |
| 20:26.44 | jm|home | :( |
| 20:27.10 | [TK]D-Fender | BigBadHoss[1]: You need to set it up again. New BR means your old BR's config got FLUSHED. |
| 20:27.26 | aydiosmio | down the POOPER |
| 20:27.28 | BigBadHoss[1] | but it says it cant contact the server period |
| 20:27.37 | aydiosmio | metaphorically SPEAKING |
| 20:27.41 | [TK]D-Fender | BigBadHoss[1]: Make sure you have an appropriate SIP ver in there waiting. |
| 20:27.41 | BigBadHoss[1] | all of the setting are right |
| 20:27.52 | BigBadHoss[1] | ok what ver goes with 2.6.1 |
| 20:27.55 | synthetiq | anyone else have any ideas? |
| 20:27.57 | [TK]D-Fender | 1.5 + |
| 20:28.56 | [TK]D-Fender | ok, heading home, back later |
| 20:29.02 | *** join/#asterisk aptura (n=s_a_l_e_@S010600a0c93f6f7e.vs.shawcable.net) |
| 20:29.07 | oliviert | [TK]D-Fender: hi i'm back. I was reading the queue module doc. And all my extension in the queue are defined as static agents, do you think the problem could come from that ? |
| 20:29.35 | *** join/#asterisk oQPa (i=Ftv@1.Red-83-35-130.dynamicIP.rima-tde.net) |
| 20:29.46 | BigBadHoss[1] | it formatted the fs like 6 times, is that normal? |
| 20:30.03 | BigBadHoss[1] | shit |
| 20:31.40 | bhrobinson | I am getting errors on startup... "Unregistered Channel X" |
| 20:31.45 | bhrobinson | anyone able to help? |
| 20:33.10 | sp0n9e- | i'm too new to even begin to offer help |
| 20:33.21 | jm|home | WORKIES! |
| 20:33.55 | *** join/#asterisk Fnc (n=tivo1@66-190-65-139.dhcp.ftwo.tx.charter.com) |
| 20:34.33 | Fnc | afternoon all.. |
| 20:35.15 | Fnc | i was wondering for a cheap pstn connection if you could remend some hardware? |
| 20:38.03 | jm|home | what do I put in the extensions.conf to make it pickup the FXO card? |
| 20:38.08 | jm|home | when it's not ringing? |
| 20:39.34 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 20:41.49 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.71.181) |
| 20:45.34 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 20:50.48 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
| 20:51.52 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 20:51.54 | *** join/#asterisk mattchis (n=mattchis@216.54.143.242) |
| 20:53.10 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 20:53.51 | *** join/#asterisk lunaphyte_ (n=lunaphye@70.90.148.1) |
| 20:54.18 | jm|home | o.O |
| 20:54.27 | jm|home | ZapBarge doesn't seem to help me |
| 20:55.45 | *** join/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 20:56.46 | hoobastooba | can i create a line in sip.conf and call it as a variable in extensions.conf? for example in sip.conf: mycell=5554444 and in extensions.conf: exten => 220,1,Macro(follow-me|${EXTEN}|${mycell}) |
| 20:57.36 | [TK]D-Fender | hoobastooba : yes. |
| 20:57.42 | Dossy | grr. |
| 20:57.51 | jm|home | [TK]D-Fender: thanks for your help - I have managed to make it register, now |
| 20:57.52 | Dossy | is there a better VMB app for Asterisk |
| 20:58.00 | hoobastooba | does it work just like that or is my syntax off? |
| 20:58.03 | Dossy | or does everyone run that crappy comedianmail? |
| 20:58.09 | *** part/#asterisk Fnc (n=tivo1@66-190-65-139.dhcp.ftwo.tx.charter.com) |
| 20:58.22 | [TK]D-Fender | hoobastooba : Its off. Go read up on sip.conf on the wiki. |
| 20:58.34 | [TK]D-Fender | Dossy : works jsut great. Whats your beef? |
| 20:58.46 | Dossy | TKD-Fender: you ever use Verizon home voicemail? |
| 20:59.05 | [TK]D-Fender | Dossy : nope |
| 20:59.07 | Dossy | ... |
| 20:59.19 | Dossy | that's what I have now, and I want to save money and switch to my Asterisk setup. |
| 20:59.33 | Dossy | I'll get an extra DID and have it ring into Asterisk, directly into voicemail |
| 20:59.34 | aptura | Hi [TK]D-Fender sent my email yesterday regarding my ip500 setup have yet to get a responce. |
| 20:59.37 | [TK]D-Fender | Dossy : .......... |
| 20:59.53 | [TK]D-Fender | aptura : I never really got home yesterday |
| 21:00.00 | Dossy | it'd be nice to be able to send DTMF "#" to skip outgoing message, "*" to enter admin/request PIN code, etc. |
| 21:00.01 | aptura | ohh really |
| 21:00.13 | aptura | slept at work" ;) |
| 21:00.24 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 21:00.39 | [TK]D-Fender | Dossy You mean you just don't like the keys they chose to move around? |
| 21:00.51 | jm|home | [TK]D-Fender: is there anyway I can just "pickup" an already picked up call on the PSTN? |
| 21:01.04 | [TK]D-Fender | Dossy : Because you CAN do all of those things. just with not THOSE keys |
| 21:01.19 | lunaphyte_ | is there a channel where folks familiar w/ traditional PBXs might be lurking? |
| 21:01.20 | [TK]D-Fender | jm|home : ... huh? |
| 21:01.24 | *** join/#asterisk nortex (n=nortex@72.16.95.84) |
| 21:01.39 | Dossy | TKD: Oh? |
| 21:01.58 | *** part/#asterisk jmls (n=asterisk@host81-159-167-80.range81-159.btcentralplus.com) |
| 21:02.01 | aptura | [TK]D-Fender so mabey we can discuss it here. I already have Line 2 configured to recieve incomming DID calls just not sure how to setup for outbound on that DID. I have two sip.conf context one for my DID incomming and one for the phones line 2. |
| 21:02.06 | Dossy | Also, when someone leaves VM, I've got it configured to ring my cell. |
| 21:02.07 | jm|home | [TK]D-Fender: yeah :S I have an Asterisk box with a FXO card in it. An IP phone attached and two other 'normal' phones on extensions of the line going into the FXO card |
| 21:02.24 | Dossy | So, it rings my cell and asks for my PIN code. If I enter my PIN, it puts me in my VMB so I can listen to the new messages. |
| 21:02.29 | jm|home | [TK]D-Fender: if GF has picked up downstairs, but it's for me, how can I 'pick up' the FXO card? |
| 21:02.30 | [TK]D-Fender | lunaphyte : Most traditional PBX's don't have *'s complexity so less need for "live" type support. Many have popular mailing list / message board communities |
| 21:03.01 | Dossy | I should look at the AGI stuff, write my own VMB setup. |
| 21:03.42 | [TK]D-Fender | jm|home : are the other phones on the * system, or seperate of it? |
| 21:03.51 | jm|home | [TK]D-Fender: separate |
| 21:03.57 | jm|home | they're just plain old phones |
| 21:04.04 | jm|home | on the analogue line coming into the house |
| 21:04.23 | lunaphyte_ | [TK]D-Fender: yeah, found a few of those. just have a bit of an urgent problem. :) |
| 21:04.34 | [TK]D-Fender | Dossy : So far I've confirmed you can do everything you have asked. Whats the problem? |
| 21:04.36 | jm|home | but you know how you can pick up the phone normally and if someone is already on it you can keep the line open while the other person hangs up |
| 21:04.47 | lunaphyte_ | on a partner-II that has been grudgingly inherited. |
| 21:04.51 | [TK]D-Fender | lunaphyte : Google time..... |
| 21:05.05 | lunaphyte_ | [TK]D-Fender: has been google time for the last hour ;) |
| 21:05.07 | *** part/#asterisk Egonis (n=root@TOROON12-1177740455.sdsl.bell.ca) |
| 21:05.25 | aptura | [TK]D-Fender whats luns issues |
| 21:05.28 | aptura | issue |
| 21:06.17 | aptura | mmmm meridian or what model? |
| 21:06.35 | lunaphyte_ | partner-II |
| 21:06.52 | *** join/#asterisk blaylock (n=blaylock@snap.helixsystems.com) |
| 21:06.53 | lunaphyte_ | lucent |
| 21:08.37 | jm|home | :S: |
| 21:08.39 | aptura | lunaphyte dont know what to say. Alot of these pbx vendors are propiatory and dont really release info about there gear unlike asterisk. I guess you have googled away and other engines. BTW I came across a propriatory PBX news group with a wealth of info. It may take me some time to find it. |
| 21:10.04 | [TK]D-Fender | aptura : you phone1.cfg is bad. "reg.1.address" is NOT a server IP address it is inface supose to be your SIP user name. |
| 21:10.14 | aptura | I see |
| 21:10.22 | [TK]D-Fender | aptura : hence "200" |
| 21:10.30 | aptura | right |
| 21:11.07 | aptura | That ip address is the phones ip ? |
| 21:11.09 | jm|home | BAH |
| 21:11.17 | jm|home | I thought Zapbarge might help me there |
| 21:11.18 | aptura | Thats what its setup as and works fine. |
| 21:11.19 | jm|home | but no :( |
| 21:12.44 | [TK]D-Fender | aptura : Um... where did you get that sip.cfg from?! I recognize the format but would like some confirmation..... |
| 21:13.20 | aptura | From somone else that comes here from time to time. |
| 21:13.36 | [TK]D-Fender | aptura : who? |
| 21:14.10 | aptura | Its been some time so dont recall. |
| 21:14.37 | *** join/#asterisk tRSS (n=tRSS@193.220.221.2) |
| 21:14.38 | aptura | Actually i have his real name and number :) |
| 21:14.52 | aptura | okay yea, Arial |
| 21:14.53 | [TK]D-Fender | aptura : Ok, well its horrifically bare. You are losing out on a lot of functionality in there. |
| 21:14.59 | tRSS | is there a way for me to send sms to my cell if I miss a call on my extension through asterisk? |
| 21:15.17 | aptura | Yea it probebly is. |
| 21:15.22 | tzanger | tRSS: of course |
| 21:15.33 | aptura | Lack of any real xml configs online is probebly why. |
| 21:16.01 | BigBadHoss[1] | what goes in the phone1.cfg file in the reg.1.server.address |
| 21:17.54 | tRSS | my macro forwards the call to my cell, but I want it to send me an sms before forwarding the call to my cell. how do I go about it? |
| 21:20.07 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 21:20.13 | aptura | tRss do you have the right version for sms capability? |
| 21:20.56 | aptura | [TK]D-Fender are you editing the file? |
| 21:21.09 | [TK]D-Fender | aptura : Nope. |
| 21:21.16 | aptura | I see |
| 21:21.28 | aptura | Well I made that minor change |
| 21:22.01 | [TK]D-Fender | aptura : thats what you should need. now mind you, you didnt' show me [200]'s config in that e-mail so I can't verify that it'll work |
| 21:22.18 | aptura | 200 works fine |
| 21:22.30 | [TK]D-Fender | aptura : Perhaps but *I* don't know that. |
| 21:22.47 | aptura | In fact everything work fine for inbound local and outbound. Only line two outbound on the DID is the issue. |
| 21:23.16 | aptura | outbound on line 1 works is what I was saying. That goes oout my zap card. |
| 21:23.48 | aptura | Its line two that is a issue concerning outbound. Want the calls to go out my DID provider for both local and long distance. |
| 21:23.58 | aptura | and that is a local DID for my area |
| 21:24.12 | [TK]D-Fender | aptura : well just setup your reg.2 accordingly. |
| 21:24.31 | [TK]D-Fender | or if you want it going through *, make another entry for it (where I think we left off) |
| 21:24.33 | jm|home | :S |
| 21:25.27 | aptura | <PROTECTED> |
| 21:25.30 | tRSS | aptura: how can I tell about the right version? what is the right version? |
| 21:26.16 | aptura | TRss, you may have to look at voip-info.org and find that info but I did see it once. Unless somone here can give you a answer off the top of there mind. |
| 21:26.25 | toerkeium | does anyone know what should I check to verify if I am sending the outgoing CID to my ITSP ? as I can't make it work |
| 21:27.02 | *** join/#asterisk Fnc (n=tivo1@66-190-65-139.dhcp.ftwo.tx.charter.com) |
| 21:27.18 | aptura | [TK]D-Fender so line two is registered. |
| 21:27.19 | tRSS | sure, sounds good. thanks! |
| 21:27.39 | Fnc | can anyone tell me if the linksys verizon pap2 can be made to work with asterisk |
| 21:27.41 | Fnc | ? |
| 21:27.46 | aptura | excepts inbound as per extension config but outbound is another issue. |
| 21:28.18 | toerkeium | Fnc: I have linksys pap2 working, not "verizone" |
| 21:28.39 | Fnc | i read that the vonage ones you can not flash |
| 21:29.13 | Fnc | but havent seen anything specifically about the verizon/voicewind |
| 21:29.19 | Fnc | er voicewing |
| 21:29.42 | aptura | [TK]D-Fender what last bit of info do you need? |
| 21:29.58 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 21:30.22 | *** join/#asterisk vader-- (n=johndoe@204.183.88.101) |
| 21:30.51 | *** join/#asterisk oQPa (i=Ftv@1.Red-83-35-130.dynamicIP.rima-tde.net) |
| 21:33.13 | *** part/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 21:35.14 | tRSS | aptura: may be I was presented my question. Infact, I want to be able to send en email if I miss a call (this email will eventually reach me as an sms) |
| 21:35.31 | tRSS | *i present the wrong question |
| 21:36.48 | tzanger | heh |
| 21:36.48 | tzanger | Q: What do you call a half-dozen Indians with Asian flu? |
| 21:36.49 | tzanger | A: Six sick Sikhs |
| 21:37.54 | aptura | Well found my issue seems I made some mistakes in the config. |
| 21:39.30 | CunningPike | tzanger: We had a tongue-twister when I was in school: The sixth sick sheik's sixth sheep's sick |
| 21:40.03 | aptura | Hi CunningPike |
| 21:40.08 | CunningPike | Hey aptura |
| 21:40.13 | aptura | How was IFAL2? |
| 21:40.23 | aptura | or was that L3? |
| 21:44.14 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.170) |
| 21:54.57 | TrixVox | Does anyone have an unmodified Polycom template xml file? For any phone? |
| 21:59.47 | *** join/#asterisk epablo (n=epablo@WLL-24-pppoe202.t-net.net.ve) |
| 22:02.22 | *** join/#asterisk mattchis (n=mattchis@216.54.143.242) |
| 22:04.44 | CunningPike | TrixVox: Freedomphones should have them |
| 22:09.29 | *** join/#asterisk S^P (n=Masood@203.148.73.236) |
| 22:09.58 | S^P | hi, do we have any limit for number of uniques extensions in *. |
| 22:10.10 | *** part/#asterisk Fnc (n=tivo1@66-190-65-139.dhcp.ftwo.tx.charter.com) |
| 22:11.12 | *** join/#asterisk jgoo (n=249fe70a@athedsl-114613.otenet.gr) |
| 22:11.28 | jgoo | bo peeps. Anyone using fastAGI, with festival / flite? |
| 22:12.37 | *** join/#asterisk techie (n=gus@pluto.voipops.net) |
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| 22:13.25 | *** part/#asterisk epablo (n=epablo@WLL-24-pppoe202.t-net.net.ve) |
| 22:16.49 | *** part/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com) |
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| 22:28.34 | *** part/#asterisk yogurt2ungue (n=yogurt2u@201.231.48.24) |
| 22:29.15 | CunningPike | We have an issue with blind transfer on Polycoms and voicemail - If personA blind transfers a call to personB, and personB does not answer, the caller gets personA's voicemail. We'd prefer that they get personB's voicemail - can that be done? |
| 22:29.57 | [hC] | that sounds like a typo in your dial plan |
| 22:30.10 | [hC] | why would the dialing party get their own voicemail? |
| 22:32.31 | [TK]D-Fender | CunningPike : pastebin the call & your dialplan |
| 22:34.14 | CunningPike | nm - it's not happening - I'm getting bogus info from our users :S |
| 22:35.41 | justinu|laptop | heh |
| 22:36.09 | *** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 22:36.33 | *** join/#asterisk JoseBravo (n=JoseBrav@ut-cable-126-44.uniweb.net.co) |
| 22:36.55 | JoseBravo | What hadware I need to connect 10 phone lines to asterisk? |
| 22:38.29 | [TK]D-Fender | JoseBravo : Sangoma A220, Digium TDM2400P, or an external gateway. T1 card + Channel bank is an option, but overkill |
| 22:39.29 | file | [TK]D-Fender: I see alive people |
| 22:40.43 | [TK]D-Fender | file : http://www.thinkgeek.com/tshirts/frustrations/6b6e/ |
| 22:40.56 | file | :D |
| 22:42.23 | JoseBravo | [TK]D-Fender what is the hadware where I connect the lines? |
| 22:43.45 | JoseBravo | Or I need 3 TDM2400P? |
| 22:45.05 | [TK]D-Fender | JoseBravo You need an amphenol patch panel or a break-out box for the TDM2400. And only ONE of those |
| 22:45.22 | [TK]D-Fender | JoseBravo : Or an A200 w/ 2 loaded Remora boards |
| 22:46.48 | JoseBravo | Ok, if I need connect RJ11 line? |
| 22:46.54 | JoseBravo | 10 RJ11 lines? |
| 22:47.11 | *** join/#asterisk lg` (i=lg@CPE00112f6dfb41-CM000f9f778bae.cpe.net.cable.rogers.com) |
| 22:47.12 | [TK]D-Fender | JoseBravo : this is common telco stuff. |
| 22:47.31 | [TK]D-Fender | JoseBravo : thats what the break-out box is for |
| 22:47.57 | lg` | Hi. I cannot get asterisk to start at boot on debian. I have done a update-rc.d on it and it created all symlinks. When i reboot the computer i can see "Starting Asterisk PBX... asterisk", so it looks like it should have started |
| 22:48.26 | JoseBravo | Then, I need break-out box and TDM2400P if I have 10 RJ11 lines? |
| 22:48.48 | [TK]D-Fender | JoseBravo : thats one way. the other is the Sangoma A200 solution |
| 22:49.12 | *** join/#asterisk Dude34 (n=Aces1UP@ip68-96-234-176.lv.lv.cox.net) |
| 22:50.08 | *** join/#asterisk Gabriel25 (n=gabriel@user-387g7ff.cable.mindspring.com) |
| 22:50.11 | Gabriel25 | hi guys |
| 22:50.16 | Gabriel25 | I try to copile zapata |
| 22:50.20 | Gabriel25 | and I get this error |
| 22:50.21 | Gabriel25 | http://pastebin.ca/201669 |
| 22:51.01 | [TK]D-Fender | ~centosbug |
| 22:51.06 | jbot | rumour has it, centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/*/include/linux/spinlock.h" |
| 22:51.06 | JoseBravo | With Sangoma A200 I need break-out box too, right? |
| 22:51.08 | [TK]D-Fender | JoseBravo : nope |
| 22:51.24 | JoseBravo | Then, where I connect my 10 lines? |
| 22:51.39 | [TK]D-Fender | JoseBravo : right into the card with the little lead cables they give you |
| 22:52.28 | JoseBravo | Its the card: http://www.voipsupply.com/product_info.php?products_id=1336 ? |
| 22:53.00 | [TK]D-Fender | JoseBravo : right series, wrong card. Look at the FXO count |
| 22:55.08 | *** join/#asterisk Ebola (n=Ebola@host86-138-123-50.range86-138.btcentralplus.com) |
| 22:57.02 | JoseBravo | I see, Im very thanks for your help. |
| 22:59.19 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 23:02.27 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 23:04.31 | *** join/#asterisk hoobastooba (n=ckwall@c-67-169-248-217.hsd1.ut.comcast.net) |
| 23:04.39 | hoobastooba | anyone here used gnudialer? |
| 23:05.02 | hoobastooba | or vicidial for that matter |
| 23:05.03 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 23:09.21 | hoobastooba | is anyone here? |
| 23:11.40 | hoobastooba | ahhhhh, all alone :-( |
| 23:12.06 | bhrobinson | I am here... but I always seem to give the wrong answers :) |
| 23:12.23 | hoobastooba | did you see my questions? |
| 23:13.20 | bhrobinson | yeah... but I use xlite, ekigo, and idefisk |
| 23:13.42 | bhrobinson | and I think yours are just dialers |
| 23:13.43 | bhrobinson | right? |
| 23:13.58 | bhrobinson | not softphones |
| 23:15.00 | hoobastooba | actually, i was just wondering about how people liked each program for powerdialer.. I am trying to see which one to install... since both are such a pain. i dont want to do both just to see which i like |
| 23:15.28 | bhrobinson | sorry boss... not my forte |
| 23:17.52 | S^P | hi, do we have any limit for number of uniques extensions in *. |
| 23:18.13 | hoobastooba | no |
| 23:18.29 | hoobastooba | do this.... exten => XXXXXXXXXXXXXXXXXXXXXXXX :-D |
| 23:18.52 | denon | of course there are limitations .. but they're really, really high |
| 23:19.14 | S^P | let say I have 1 million extensions, does it effect asterisk performance? |
| 23:19.27 | denon | it'll affect load/reload, certainly |
| 23:19.43 | denon | but if you had a need for 1M extensions, you'll have other scalability issues |
| 23:19.57 | S^P | like? |
| 23:19.59 | justinu|laptop | has to affect dialplan hunt times too |
| 23:20.06 | denon | uh,well, 1 million phones registering to a single * box |
| 23:20.24 | *** join/#asterisk BlepsoaF (n=pbaker@nnat-gw.adeptra.com) |
| 23:20.41 | BlepsoaF | anyone around that can help with a quick polycome provisioning question? |
| 23:20.48 | BlepsoaF | polycom |
| 23:20.49 | denon | there'd come a time when you'd probably want to do that logic via a lookup, rather than using * extensions |
| 23:20.56 | denon | if there were literally millions |
| 23:21.06 | BlepsoaF | well, I |
| 23:21.13 | justinu|laptop | i wonder if a million phones could register with openser |
| 23:21.36 | denon | not all at once :) |
| 23:21.36 | S^P | lol |
| 23:21.55 | justinu|laptop | would be fun to torture it |
| 23:21.58 | denon | perhaps 1-2 at a time.. |
| 23:22.02 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 23:22.05 | denon | or a second between each |
| 23:22.12 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 23:22.22 | denon | toward the millionth or so, maybe a few seconds between each :) |
| 23:22.23 | justinu|laptop | what happens when the entire eastern seaboard goes down/back up and all your millions of customers reregister? ;) |
| 23:22.36 | justinu|laptop | seaboard power grid that is |
| 23:22.44 | BlepsoaF | I'm following the setup instructions that are currently on the wiki about having a separate Polycom directory then under that having a directory for each phone and the related configuration file. my question is do I then need to add something to 000000000000.cfg to point to /var/ftp/Phones/0004f21139c7.cfg, which will then inturn point to /var/ftp/Phones/x2632/x2632.cfg |
| 23:22.51 | denon | justinu|laptop: they find their respective regionalized switch, and each register with one of those |
| 23:23.06 | *** join/#asterisk alerios (n=alerios@208.195.215.170) |
| 23:23.44 | justinu|laptop | distributed network, too complicated for me |
| 23:24.01 | denon | multiple points of failure .. allows me to sleep when on-call |
| 23:24.51 | [TK]D-Fender | BlepsoaF : just ask it |
| 23:25.03 | hoobastooba | well... your issue would really be the number of concurrent calls you could do. |
| 23:25.20 | denon | hoobastooba: I think he's talking about ser doing the reg, and something else handling the calls |
| 23:25.21 | denon | or reinvites |
| 23:25.26 | BlepsoaF | [TK]D-Fender: I dont understand.. |
| 23:25.27 | hoobastooba | ah |
| 23:25.43 | justinu|laptop | still, even AT&T lost 114 switches on one day in 1990 |
| 23:26.02 | denon | qwest lost half their gear one day too .. due to a stupid switch update |
| 23:26.13 | [TK]D-Fender | BlepsoaF : you shouldn't be using the defaults "000000000000.cfg" and "000000000000-directory.xml" for anything. |
| 23:26.17 | justinu|laptop | cause? a misplaces break; statement in the code :P |
| 23:26.38 | hoobastooba | so has anyone tried the gnudialer? I just wonder if anyone likes it. |
| 23:26.39 | denon | yup, it happens |
| 23:26.41 | BlepsoaF | [TK]D-Fender: Ok so I should basically toss those files out then |
| 23:26.55 | *** join/#asterisk Tili (n=tili@202.133.65.79) |
| 23:27.02 | hoobastooba | or if they have tried both gnudialer and vicidial and if they have a preference |
| 23:27.23 | [TK]D-Fender | BlepsoaF : basically they are samples. It is also used for an initial directory load. Once the phone saves its own back, then it should be referenced. |
| 23:28.17 | BlepsoaF | [TK]D-Fender: then I'm confused, should I delete those files? |
| 23:28.40 | BlepsoaF | [TK]D-Fender: and if not, should I leave them as they came? |
| 23:29.36 | [TK]D-Fender | BlepsoaF : leave them. |
| 23:30.13 | BlepsoaF | [TK]D-Fender: ok, then I guess my last question is..how do I tell my phone to look for its mac.cfg file in the ftproot/phones directory |
| 23:31.07 | [TK]D-Fender | BlepsoaF : you can't. the <mac>.cfg file must be in the base of the provisioning folder. the files it REFERS to may be in folders however |
| 23:31.56 | *** join/#asterisk juanjoc (n=juanjoc@201.216.212.113) |
| 23:32.42 | BlepsoaF | [TK]D-Fender: ok great now I get it, should I remove APPLICATION XML block from the 000000000 file? |
| 23:32.57 | BlepsoaF | cuz right now it has CONFIG_FILES="phone1.cfg, sip.cfg" |
| 23:33.02 | [TK]D-Fender | BlepsoaF : Don't believe thats a healthy idea.... |
| 23:33.40 | [TK]D-Fender | the 00000000 files should not be called if you have setup your phone's ones |
| 23:34.00 | BlepsoaF | ok great I understand now, thanks for all your help |
| 23:34.41 | [TK]D-Fender | the 000000000 are called if there is no MAC math to provide a MINIMUM default config. this can help diagnose that your configs aren' right |
| 23:35.11 | BlepsoaF | [TK]D-Fender: ok cool, so I will leave the sip.cfg and the phone1.cfg alone along with those 0000 files |
| 23:35.29 | *** part/#asterisk hoobastooba (n=ckwall@c-67-169-248-217.hsd1.ut.comcast.net) |
| 23:36.25 | *** join/#asterisk ManxPower (n=manxpowe@84.sub-70-216-8.myvzw.com) |
| 23:36.32 | ManxPower | join/join #asterisk-dev |
| 23:37.32 | *** join/#asterisk mountainm2k (n=mountain@165.236.183.1) |
| 23:51.57 | *** join/#asterisk kimosabe (n=kimosabe@189.156.235.21) |
| 23:52.37 | kimosabe | is there a guie for my asterisk box that i can addon ? and where 2 download it from please |
| 23:53.59 | BigBadHoss[1] | freepbx |
| 23:54.11 | BigBadHoss[1] | itttssss greeeaaat! |
| 23:55.24 | kimosabe | bigbadhouse do i have to add php and all that or can i just run the gui |
| 23:57.17 | *** part/#asterisk mountainm2k (n=mountain@165.236.183.1) |
| 23:57.32 | [TK]D-Fender | LOL |
| 23:57.38 | [TK]D-Fender | crap-tastic! |
| 23:58.01 | [TK]D-Fender | kimosabe : it has all osrts of requirements. www.freepbx.org |
| 23:58.39 | kimosabe | thanks |