| 00:07.14 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 00:08.45 | mchou | is there a firewall keep alive on the iax port if I have a valid "register =>" statement in iax.conf? |
| 00:09.43 | mchou | or must the default iax port be specifically port forwarded from the nat? |
| 00:10.33 | *** join/#asterisk ZenBSDi (n=zenbsdi@unaffiliated/ZenBSDi) |
| 00:21.21 | davidstrauss | mchou: you should try to forward |
| 00:21.29 | davidstrauss | mchou: though there is a keepalive |
| 00:23.47 | *** join/#asterisk delrocas (n=IceChat7@168-103-31-91.spkn.qwest.net) |
| 00:35.23 | *** join/#asterisk Paige_ (n=Paige@208.89.241.31) |
| 00:35.33 | *** join/#asterisk natmlt (n=natmlt@ip68-109-184-144.ph.ph.cox.net) |
| 00:44.54 | Katty | hmm. |
| 00:46.00 | *** part/#asterisk korihor (n=korihor@201.211.168.130) |
| 00:46.55 | *** join/#asterisk JenniferAkemi (n=akemi@72.60.168.132) |
| 00:51.34 | *** join/#asterisk nebajoth (n=weechat@12-180-163-230.alphacomm.net) |
| 00:54.59 | nnY_2 | ok so I have logging on full for the day tomorrow. I have been actively trying to hunt down why* asterisk keeps the sip channel open even after the client sends BYE |
| 00:55.30 | nnY_2 | should i tunr sip history on for the day tomorrow too? |
| 00:55.33 | nnY_2 | turn* |
| 00:56.40 | natmlt | I just started to try and switch over to using AEL. Does anyone know now to convert this context name [section](+) to AEL format? |
| 00:57.24 | natmlt | Or another way to extend a context? |
| 00:57.58 | *** join/#asterisk orionr (n=orionr@c-76-26-221-76.hsd1.sc.comcast.net) |
| 00:57.59 | nnY_2 | i am debugging the sip for the user who had the open channel today |
| 00:58.04 | orionr | ~itsplist-us |
| 00:58.05 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 00:58.25 | nnY_2 | eww broadvoice? |
| 00:58.33 | nnY_2 | need to take them out |
| 00:58.42 | nnY_2 | google broadvoice terms of use :) |
| 00:59.06 | nnY_2 | i use vitelity. I had some dtmf issues with them, but they seem to be getting better |
| 01:05.01 | orionr | does anyone here have a teliax account? |
| 01:08.21 | *** join/#asterisk uski (n=uski@bre01-1-88-162-0-210.fbx.proxad.net) |
| 01:09.00 | uski | hi; i have a spa3000, and i'd like to control the distinctive ring feature from asterisk, i.e. i'd like to be able to do something like Dial(SIP/spa3000/<distinctive ring number>), is there a way to do so ? thx |
| 01:09.24 | uski | (the spa3000 is a 2-ports VoIP/PSTN bridge, with one FXO and one FXS port) |
| 01:14.18 | *** join/#asterisk mattwj2002 (n=matt@c-76-17-132-205.hsd1.mn.comcast.net) |
| 01:14.34 | mattwj2002 | hey everyone |
| 01:14.44 | orionr | Hey mattwj2002 |
| 01:15.04 | mattwj2002 | hey orionr |
| 01:16.30 | [TK]D-Fender | uski: http://www.voip-info.org/wiki/index.php?page_id=563 |
| 01:17.43 | uski | thanks |
| 01:17.58 | uski | it's precisely what i've been looking for :) sorry i couldn't find it myself |
| 01:18.04 | [TK]D-Fender | uski: You'll have to clean it up to 1.4 spec, but its a start |
| 01:18.34 | uski | it's definitely a start :) |
| 01:18.47 | mattwj2002 | TK I like that ATA |
| 01:19.16 | uski | (actually... im still using 1.2... it's in my linux distribution packages and i don't feel like upgrading everything to get 1.4) |
| 01:19.39 | mattwj2002 | does that have fxo and fxs or just fxs ports? |
| 01:19.57 | uski | mattwj2002, the SPA3000 ? it has one FXO and one FXS |
| 01:20.17 | mattwj2002 | I was actually wondering about the SPA2000 |
| 01:20.34 | uski | i think the SPA2000 only has "phone" ports |
| 01:20.49 | mattwj2002 | I don't have a real phone anyways |
| 01:20.57 | mattwj2002 | *phone line |
| 01:21.40 | [TK]D-Fender | mattwj2002: 2000 is an old model. Look for the 2102 / PAP2T-NA |
| 01:21.59 | mattwj2002 | yeah I actually have a PAP2 |
| 01:22.08 | mattwj2002 | it works all right |
| 01:22.16 | uski | i couldn't feel safe by only having voip lines - what if you loose power or internet access ? |
| 01:22.29 | mattwj2002 | cellphone |
| 01:22.30 | mattwj2002 | :) |
| 01:22.41 | uski | huh, true |
| 01:22.43 | uski | :) |
| 01:22.45 | angryuser | uski isdn or T1 |
| 01:23.21 | uski | angryuser, yea but it's a tad more expensive |
| 01:23.21 | [TK]D-Fender | hands angryuser an elephant gun and directs him towards an ant-hill. |
| 01:25.04 | uski | mattwj2002, does caller id works with your PAP2 ? |
| 01:25.24 | angryuser | [TK]D-Fender stop watching dirty movies |
| 01:25.29 | mattwj2002 | you know honestly I am not sure |
| 01:25.38 | mattwj2002 | hmmm |
| 01:25.46 | [TK]D-Fender | uski: Yes |
| 01:25.47 | mattwj2002 | let me see if I can check |
| 01:25.54 | uski | cause i can't get caller id to work with the FXS port of my SPA3000 and im not sure if it's my phone, the settings of the spa3000, or my extensions.conf |
| 01:26.18 | *** join/#asterisk chendy (n=chatzill@58.251.228.45) |
| 01:27.47 | uski | mattwj2002, don't bother - i'll eventually find out what's wrong with my setup |
| 01:27.54 | uski | (but thanks for looking :)) |
| 01:29.27 | mattwj2002 | no it is my pleasure to help |
| 01:30.11 | mchou | uski: what caller id are you referring to? Inbound or outbound? |
| 01:30.45 | uski | inbound, i.e. i want to display something on the callerid display of my phones when they get a call from the FXS port of the SIP3000 |
| 01:31.19 | uski | not sure if that's really inbound |
| 01:31.35 | mattwj2002 | yup uski my caller id works |
| 01:31.41 | mattwj2002 | just an idea... |
| 01:31.56 | uski | i successfully receive the callerid signals from my PSTN line for "incoming" calls from the PSTN, i.e. the ${CALLERID} variable has the callerid of the person calling me in the PSTN |
| 01:32.01 | mattwj2002 | is the call coming through as a guest ? |
| 01:32.30 | uski | well, no, but maybe asterisk somehow eats the callerid somewhere in my dialplan |
| 01:32.54 | *** join/#asterisk coppice (n=chatzill@27.202.17.210.dyn.pacific.net.hk) |
| 01:32.59 | mattwj2002 | yeah make sure it isn't set somewhere :) |
| 01:33.00 | uski | (i'm not suggesting that asterisk is at fault - i rather think that i'm not doing something correctly in my extensions.conf) |
| 01:34.09 | uski | <PROTECTED> |
| 01:34.09 | uski | <PROTECTED> |
| 01:34.09 | uski | <PROTECTED> |
| 01:34.09 | uski | <PROTECTED> |
| 01:34.21 | uski | when this runs, i see "Caller ID: 0......." in the console |
| 01:34.28 | uski | but the phone doesn't display any caller id |
| 01:34.50 | uski | (and yes, the phone rings, and i can take the call) |
| 01:35.00 | uski | so everything works apart from the callerid |
| 01:35.12 | mchou | what ver of asterisk is this? |
| 01:35.17 | uski | ... 1.2 |
| 01:35.18 | mattwj2002 | why are setting the caller id? |
| 01:35.41 | mattwj2002 | is this an incoming or outgoing extension |
| 01:36.23 | uski | it's called by the dialplan of the spa3000 (technically, the dialplan string is (S0<:66666>)) |
| 01:36.35 | mattwj2002 | ok |
| 01:36.48 | uski | which tells the spa3000 to route the incoming call to the extension 66666 |
| 01:37.46 | uski | maybe i should try with a recent version... |
| 01:37.51 | nebajoth | why is polycomm > linksys? |
| 01:37.56 | uski | i have no idea why debian keeps such an old version |
| 01:38.30 | [TK]D-Fender | uski: exten => 66666,4,SetCallerId(${CALLERIDNUM}) <-- useless line |
| 01:38.38 | uski | ok |
| 01:38.42 | [TK]D-Fender | uski: If you don't see CID, you probably misconfigured the ATA |
| 01:39.12 | uski | i thought of this, but i think the settings should be correct as i'm receiving the caller id from the PSTN |
| 01:39.28 | uski | this (should) mean that the settings are correct for my country |
| 01:39.35 | [TK]D-Fender | nebajoth: Quality. Better sound, more configurable, more solid construction. |
| 01:39.51 | nebajoth | how much better? |
| 01:39.55 | [TK]D-Fender | uski: wait.. are you trying to set CID for teh PSTN? |
| 01:40.04 | [TK]D-Fender | nebajoth: Where are you located? |
| 01:40.07 | mattwj2002 | yeah I am running the 1.6 beta so my configuration is a little different |
| 01:40.09 | mattwj2002 | :) |
| 01:40.12 | nebajoth | Ontario, Canada |
| 01:40.21 | [TK]D-Fender | nebajoth: Ok, forget Linksys probably. |
| 01:40.27 | uski | [TK]D-Fender, i'm trying to set the CID for a call which will make the phone connected to the FXS ring |
| 01:40.45 | uski | i.e. Dial(SIP/sip3000) makes the phone on the FXS ring |
| 01:40.46 | nebajoth | [TK]D-Fender: why? |
| 01:41.00 | [TK]D-Fender | uski: If the FXS isn't showing CID, then you have either misconfigured the FXS on the ATA, or your phone attached to it has issues |
| 01:41.13 | uski | hmm good call, i will try with another phone |
| 01:41.30 | [TK]D-Fender | nebajoth: Becaus in North America Polycom pricing is really close to Linksys and you get a lot of value out of polycom. |
| 01:41.39 | [TK]D-Fender | nebajoth: How many, and what kind of use? |
| 01:41.45 | mattwj2002 | hey uski |
| 01:41.56 | uski | mattwj2002, yea? |
| 01:41.59 | mattwj2002 | are you sure your receiving caller id from the PSTN? |
| 01:42.04 | uski | yes |
| 01:42.08 | uski | it's displaying in my console |
| 01:42.16 | uski | thanks to the NoOp line |
| 01:42.23 | nebajoth | [TK]D-Fender: 12, in a call center |
| 01:42.26 | uski | (which is there only for debugging purposes) |
| 01:42.32 | mattwj2002 | okay |
| 01:42.38 | [TK]D-Fender | nebajoth: What headsets? |
| 01:42.47 | mattwj2002 | just a thought anyways :) |
| 01:43.04 | uski | nebajoth, just a stupid question - i know that a lot of call centers use softphones with headsets connected to the soundcard, why don't you do that ? it's probably very cost effective |
| 01:43.15 | [TK]D-Fender | uski: EWWWWW!!!!!!!!!!!!! |
| 01:43.21 | uski | (oops) |
| 01:43.40 | mattwj2002 | I think most people prefer a real phone |
| 01:43.40 | *** join/#asterisk salzh (n=zhongxia@58.247.194.125) |
| 01:43.41 | mattwj2002 | :) |
| 01:43.45 | [TK]D-Fender | uski: For a person who spends their day on the phone, don't leave a piece of crap to work with.... |
| 01:43.50 | mattwj2002 | easier to dial for one |
| 01:44.19 | uski | yea |
| 01:44.33 | uski | why not getting a real voip phone then? |
| 01:44.35 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-0cc445adadeadcb5) |
| 01:44.40 | uski | instead of analog phones + ATA ? |
| 01:44.49 | uski | (price probably?) |
| 01:46.04 | mattwj2002 | one good thing about ata's |
| 01:46.21 | mattwj2002 | it sure helps with making cheap cordless phones |
| 01:47.04 | mattwj2002 | speaking of wireless phones..... |
| 01:47.05 | coppice | DECT IP phones aren't that expensive |
| 01:47.11 | uski | yea... i never really liked ATAs - not that i have a great experience in voip, but i think it's a big hack, especially when it comes to call status detection (tone detect, ...) |
| 01:47.26 | mattwj2002 | anyone know of a good wireless voip phone? |
| 01:47.28 | angryuser | mattwj2002 siemens has some cheap cordless phones like 650 ip one base 2 sip accounts |
| 01:47.40 | nebajoth | [TK]D-Fender: No headsets planned so far |
| 01:47.41 | mattwj2002 | thanks angryuser |
| 01:47.47 | nebajoth | uski: gah, good point |
| 01:47.55 | Katty | [TK]D-Fender: you geek around the clock. |
| 01:48.00 | Katty | [TK]D-Fender: when do you sleep?! |
| 01:48.12 | coppice | some of those siemens DECT phones support wideband voice, too |
| 01:48.20 | uski | (it's 3:48am here - who want to compete ? xD) |
| 01:48.26 | angryuser | i think fender is a robot |
| 01:48.33 | uski | wideband voice ? what's that |
| 01:48.36 | Katty | angryuser: lies. |
| 01:48.42 | angryuser | uski 3:48 hm gmt +2 ? |
| 01:48.43 | Katty | angryuser: i have seen human emotion. |
| 01:48.53 | uski | angryuser, yea |
| 01:48.55 | [TK]D-Fender | nebajoth: IP 320's should do for them then |
| 01:48.55 | nebajoth | I wouldn't mind going straight SIP rather than ATA |
| 01:49.02 | coppice | uski: voice that doesn't sound as crappy as an ordinary phone |
| 01:49.13 | Katty | [TK]D-Fender: go nap. |
| 01:49.14 | angryuser | Katty some people able to program huma emotion :) |
| 01:49.17 | nebajoth | [TK]D-Fender: those are headsets, or polycomm phones? |
| 01:49.19 | uski | coppice, ok |
| 01:49.21 | angryuser | uski country ? |
| 01:49.26 | Katty | jbot: emotion |
| 01:49.27 | jbot | i guess emotion is something apt doesn't have |
| 01:49.28 | [TK]D-Fender | Katty: I'm in EST. 8am-1am I may be on through. |
| 01:49.43 | Katty | jbot: emo |
| 01:49.43 | jbot | /wrists |
| 01:49.50 | [TK]D-Fender | ~cluemuffin |
| 01:49.50 | jbot | [~cluemuffin] A perfect blend of bran & ClueBat (tm). Not to be confused with the Chinese Fighting Muffin. |
| 01:49.53 | Katty | jbot: most excellent |
| 01:50.01 | Katty | [TK]D-Fender: most excellent. |
| 01:50.04 | [TK]D-Fender | :D |
| 01:50.07 | nebajoth | [TK]D-Fender: Where are you located? |
| 01:50.16 | uski | nebajoth, i've seen RF headsets which are meant to be connected to the standard "headset port" of phones; these RF headsets has a device which was able to simluate phone lifting/hang up |
| 01:50.21 | [TK]D-Fender | nebajoth: Montreal |
| 01:50.36 | nebajoth | [TK]D-Fender: beautiful city |
| 01:50.36 | Katty | nebajoth: canada explains everything. |
| 01:50.44 | [TK]D-Fender | uski: Yes, glorious PITA those are. |
| 01:50.48 | nebajoth | Katty: I'm Canadian too :P |
| 01:50.55 | Katty | well that explains it! |
| 01:50.57 | mattwj2002 | I can't seem to find the siemens 650 anywhere |
| 01:51.00 | nebajoth | :P |
| 01:51.05 | mattwj2002 | are you sure you have the right model number |
| 01:51.09 | angryuser | [TK]D-Fender canadian french accent is so fun |
| 01:51.20 | Katty | as in junky |
| 01:51.24 | nebajoth | [TK]D-Fender: I'm looking for a decent SIP provider in Ontario -- know any? |
| 01:51.24 | Katty | canadian french accent |
| 01:51.25 | angryuser | mattwj2002 sek |
| 01:51.27 | Katty | dear lord. shoot me. |
| 01:51.44 | [TK]D-Fender | Katty: Yup, he's "pure-laine" as they come... |
| 01:51.48 | mattwj2002 | sek? |
| 01:51.58 | Katty | i'd sooner get the post-it-notes than try to figure out what junky's saying |
| 01:52.00 | [TK]D-Fender | sec* |
| 01:52.01 | mattwj2002 | shoots katty |
| 01:52.09 | [TK]D-Fender | Katty: Same goes for me sometimes ;) |
| 01:52.21 | mattwj2002 | oh okay |
| 01:52.27 | Katty | [TK]D-Fender: you'd i'd just prefer to smack upside the head and say speak kat |
| 01:52.31 | Katty | mattwj2002: baroo? |
| 01:52.53 | [TK]D-Fender | blinks |
| 01:52.54 | mattwj2002 | :P |
| 01:52.58 | [TK]D-Fender | Katty: Say what? |
| 01:53.03 | Katty | mattwj2002: you, sir, do not parse. |
| 01:53.20 | Katty | [TK]D-Fender: what? |
| 01:53.35 | [TK]D-Fender | Katty: Who's on first? ;) |
| 01:53.48 | Katty | [TK]D-Fender: clearly not the guy on second or third base. |
| 01:53.55 | Katty | [TK]D-Fender: what are we talking about again? |
| 01:53.59 | mattwj2002 | lol |
| 01:54.11 | [TK]D-Fender | Katty: Something about you hitting me upside the head. |
| 01:54.13 | Katty | [TK]D-Fender: dagnabbit, speak kat! |
| 01:54.30 | Katty | [TK]D-Fender: oh yes. i do that on occasion when you do not parse properly. |
| 01:54.35 | [TK]D-Fender | Katty: "Bark". Its dog for "Mew" |
| 01:54.36 | *** join/#asterisk jeffspeff (n=jeff@c-98-240-112-228.hsd1.ky.comcast.net) |
| 01:54.45 | angryuser | mattwj2002 i dont remember the model exactly, try to search in their model line, i have one of this also http://www.estore.fr/go/siemens/s675ip.htm |
| 01:55.04 | Katty | [TK]D-Fender: well as long as no Yip!s are involved. |
| 01:55.12 | Katty | [TK]D-Fender: Punt(tm) |
| 01:55.19 | uski | "Standard features on the SPA921 include a high resolution graphical display, speakerphone, and a 2.5 mm head-set port." http://www.amazon.com/Voip-2-Line-Business-Phone/dp/B000F16HX8/?tag=srchprod-21 |
| 01:55.34 | mattwj2002 | I wonder if any of these phones work in the US |
| 01:55.44 | coppice | I thought "woof" was dog for mew, and "Bark" was tree for skin |
| 01:55.52 | Katty | well you could take them across the border and plug em in |
| 01:55.55 | uski | mattwj2002, is there any reason a SIP phone wouldn't work in the US ? |
| 01:56.05 | [TK]D-Fender | coppice: Don't get smarmy on us now! |
| 01:56.11 | mattwj2002 | well for one I don't want to buy a power adapter |
| 01:56.12 | mattwj2002 | :) |
| 01:56.13 | angryuser | mattwj2002 is ther any reason why they should not ? |
| 01:56.14 | uski | apart from the power adapter mayb |
| 01:56.29 | Katty | mattwj2002: power it with a couple lemons |
| 01:56.30 | [TK]D-Fender | mattwj2002: its friggin LINKSYS. Get a clue! |
| 01:56.36 | uski | angryuser, you live in france? |
| 01:56.55 | angryuser | uski sometimes .. |
| 01:56.58 | uski | woot, i like that SIP DECT phone |
| 01:57.01 | coppice | where's the clue in it being LINKSYS? |
| 01:57.05 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 01:57.11 | Katty | coppice: you're a clue. |
| 01:57.42 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 01:58.24 | [TK]D-Fender | coppice: that something so darn common here can be had cheap with local power supplies included. |
| 01:59.22 | mattwj2002 | man these are some pretty nice phones |
| 01:59.23 | mattwj2002 | :) |
| 01:59.53 | coppice | but they don't come with a universal supply, so you still end up buying supplies if you move them around |
| 02:00.11 | angryuser | i dont know about other but s675ip is very stable |
| 02:00.28 | uski | angryuser, do you have one ? |
| 02:00.37 | angryuser | uski yes |
| 02:00.43 | uski | i want one :D |
| 02:01.50 | uski | grrrrr why do they still use NiMh batteries |
| 02:02.28 | coppice | most things still use NiMh |
| 02:02.29 | [TK]D-Fender | coppice: I've moved them several kilometers and the power supply was still good! |
| 02:02.44 | coppice | how parochial |
| 02:02.46 | uski | angryuser, did you buy any additional phone ? if yes, which model ? |
| 02:02.51 | mattwj2002 | well I need to get to bed |
| 02:02.54 | mattwj2002 | good night all :) |
| 02:03.09 | uski | good night |
| 02:03.12 | angryuser | uski C450ip cheaper/basic no problems |
| 02:03.20 | *** part/#asterisk mattwj2002|sleep (n=matt@c-76-17-132-205.hsd1.mn.comcast.net) |
| 02:04.43 | angryuser | and i dont remember model, i will tell you tomorrow if you want, we have some random audio issies, no nat involved and ports set to good values, i wasnt able to trace it |
| 02:04.48 | *** join/#asterisk JonXP (n=me@c-24-99-164-175.hsd1.ga.comcast.net) |
| 02:05.07 | uski | angryuser, don't worry, i won't buy it tomorrow anyway |
| 02:05.10 | angryuser | but only one model |
| 02:05.23 | angryuser | s675ip works great |
| 02:05.28 | uski | so you're telling me the C450ip has issues and the s675ip is OK ? |
| 02:05.43 | angryuser | they are both ok |
| 02:05.55 | JonXP | Hey guys, I have incoming calls working fine, but I can't seem to get outgoing calls to work. It says the cause code is 99, anyone able to help me out on this? |
| 02:06.10 | angryuser | there is third one, and i dont remember which one |
| 02:07.29 | uski | ok |
| 02:08.11 | angryuser | uski aks me someday .. |
| 02:08.39 | JonXP | Or, more importantly, how do I change the caller ID info being transmitted to the PRI? I believe the fact that I'm sending a name is causing the issue. How do I turn taht off? |
| 02:08.44 | uski | angryuser, ok :) |
| 02:09.12 | uski | angryuser, do you know if it's possible to have a separate SIP extension for each phone on the same C460ip base ? |
| 02:09.28 | uski | that is, if i have one base and 6 phones, can i reach the 6 phones independently (and simultaneously) from the SIP ? |
| 02:09.58 | angryuser | on this one 6 sip account's 6 bases' but only 2 calls + one anolog |
| 02:10.12 | angryuser | analog* |
| 02:11.10 | angryuser | pff not 6 bass 6 phones ;) |
| 02:11.43 | angryuser | but 3 is a good number for mid used base |
| 02:16.08 | [TK]D-Fender | JonXP: enable PRI debug, verbsoe 10, and pastebin a failed call. |
| 02:16.10 | [TK]D-Fender | ~pb |
| 02:16.11 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 02:16.12 | [TK]D-Fender | ^^^^^^^^^^^ |
| 02:16.33 | JonXP | http://pastebin.com/m5045337d |
| 02:17.22 | JonXP | I know in that particular example it's choosing the first channel, not first group, I've already fixed that. |
| 02:17.34 | JonXP | The issue remains the same. :D |
| 02:19.52 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 02:21.53 | JonXP | To my newbie untrained eye, it appears that sending the "Display" element is causing it to choke (because the cause data is the same code for the Display element) |
| 02:22.14 | JonXP | But I don't know how to turn that off. :D |
| 02:22.25 | uski | angryuser, is it possible to make only a particular phone ring from the SIP ? or do all phones always ring simultaneously |
| 02:23.19 | angryuser | uski 6 separated sip accounts maximum 2 || calls max |
| 02:23.32 | uski | ok, thanks |
| 02:23.36 | JonXP | (We hired a consultant to set this up, but so far it has gone poorly, and I'm trying to fix it now) |
| 02:24.00 | [TK]D-Fender | JonXP: Presentation: Presentation permitted, user number passed network screening (1) '3923' ] <- try setting a valid 10-digit CID |
| 02:24.21 | JonXP | Can you point me in the correct direction? |
| 02:24.42 | JonXP | Perhaps a webpage or file to look in for that setting? |
| 02:24.53 | angryuser | also possible to define for each phone the callout account |
| 02:24.54 | [TK]D-Fender | JonXP: Set(CALLERID(num)=7701234567) |
| 02:25.14 | JonXP | Ahh, as part of the dial plan? |
| 02:25.23 | [TK]D-Fender | JonXP: Clearly |
| 02:25.46 | JonXP | Sorry, I am brand new to Asterisk, and learning through trial by fire. |
| 02:26.07 | [TK]D-Fender | JonXP: Asbestos suits are to to the right. |
| 02:26.14 | JonXP | I can't stand three weeks of our phone system being down. |
| 02:26.37 | [TK]D-Fender | JonXP: If its been 3 weeks, your consultant sucks |
| 02:27.06 | JonXP | This is why I am now doing it. :D |
| 02:28.33 | mchou | ok, stupid question. Sipbroker pattern are like _*X. Pap2 and other phones accept *XX (for supplementary services and the like). how to tell pap2 I wanna sipbroker? |
| 02:29.02 | mchou | not the supplementary stuff? |
| 02:30.31 | [TK]D-Fender | mchou: fix the dialplan on it |
| 02:30.51 | mchou | [TK]D-Fender: on pap2? |
| 02:30.56 | [TK]D-Fender | mchou: Yes. |
| 02:33.55 | JonXP | [TK]D-Fender: No error codes have changed, but a 10 digit CID is being sent |
| 02:34.28 | [TK]D-Fender | JonXP: Set(CALLERID(name)=JUSTATEST) |
| 02:34.58 | [TK]D-Fender | JonXP: And a though, try setting "pridialplan=national" in zapata.conf. This will require a reload of chan_sip or a restart of * |
| 02:35.42 | JT | reload of chan_sip? |
| 02:35.53 | [TK]D-Fender | chan_zap.so |
| 02:35.55 | [TK]D-Fender | :p |
| 02:35.59 | [TK]D-Fender | YOU KNOW WHAT I MEAN! |
| 02:36.04 | angryuser | im off, later |
| 02:36.05 | orionr | t |
| 02:36.19 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 02:37.17 | JonXP | [TK]D-Fender: I already have switchtype=national in my zapata.conf |
| 02:37.29 | JonXP | Will that be different? |
| 02:38.35 | [TK]D-Fender | JonXP: this isn't a PRI signalling, its teh dialplan indicator. |
| 02:38.45 | [TK]D-Fender | JonXP: Your signalling stas the same |
| 02:40.21 | JonXP | [TK]D-Fender: OK. |
| 02:40.39 | JonXP | [TK]D-Fender: Same problems. |
| 02:41.09 | JonXP | [TK]D-Fender: I have set teh CID name to "JUSTATEST" the number to a valid 10 digit number, and both show up as expected. |
| 02:42.14 | [TK]D-Fender | JonXP: Get the telco on the line and ask their opinion directly... |
| 02:42.24 | [TK]D-Fender | JonXP: Have them monitor the call. |
| 02:42.26 | JonXP | However, the error code remains 99, and the cause data remains 28. |
| 02:43.08 | JonXP | Well, out of curiosity...jsut for a lark...how do I prevent the CID info from being sent? |
| 02:43.18 | JonXP | Is it possible to have those info fields not sent at all? |
| 02:47.37 | [TK]D-Fender | hold on |
| 02:48.10 | [TK]D-Fender | JonXP: "core show application setcallerpres" |
| 02:49.06 | uski | is g722 (so-called wideband) supported by asterisk ? |
| 02:49.22 | uski | if no, i won't get a wideband-capable sip phone :) |
| 02:49.44 | JonXP | exten => _9NXXNXXXXXX,1,SetCallerPres(unavailable) |
| 02:49.49 | JonXP | That shoud be right? |
| 02:53.10 | JonXP | [TK]D-Fender: i tried both "prohib" and "unavailable" |
| 02:53.25 | [TK]D-Fender | JonXP: Looks ok. |
| 02:54.19 | JonXP | [TK]D-Fender: Both still had the "Display" and "Calling number" fields set. |
| 02:54.21 | JonXP | :-( |
| 02:55.49 | JonXP | [TK]D-Fender: Ah well, thanks for your help |
| 02:56.49 | *** part/#asterisk ricko73 (n=dhartman@wilug/newlug/ricko73) |
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| 03:00.48 | raasdnil | hi all. I am on CentOs 1.4. Installed Asterisk fine. Installed zaptel fine with ztdummy. load with modprobe, can see zaptel and ztdummy in the lsmod list, but iax2 reload complains that it needs zaptel timing.... any ideas? |
| 03:00.59 | *** join/#asterisk irisht (n=irisht@cpe-70-122-11-142.austin.res.rr.com) |
| 03:03.22 | [TK]D-Fender | JonXP: pastebin your zapata.conf |
| 03:03.43 | [TK]D-Fender | raasdnil: Perhaps * was built BEFORE zaptel and support was not built it. |
| 03:04.42 | JonXP | http://pastebin.com/d70493b51 |
| 03:05.05 | raasdnil | TK, so try re building asterisk? |
| 03:06.30 | [TK]D-Fender | raasdnil: Yes, modprobe ztdummy, run ztcfg -vvvv, and then trash your * source, re-extract and begin the process over |
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| 03:08.28 | [TK]D-Fender | JonXP: try "pridialplan=unknown" and "prilocaldialplan=unknown" |
| 03:08.32 | [TK]D-Fender | joan and restart * |
| 03:08.35 | [TK]D-Fender | jon |
| 03:08.47 | JonXP | OK |
| 03:10.49 | JonXP | Same error |
| 03:11.20 | mchou | Hey, why do most ITSPs charge for outbound toll-free calls while sipbroker does not? I mean how does sipbroker manage to do it? |
| 03:12.00 | [TK]D-Fender | mchou: Because the CALLEE gets charged |
| 03:12.30 | [TK]D-Fender | mchou: Many ITSP's let you terminate to 800's for no fee on your side. Plenty of fee ones too. Hybridize your setup |
| 03:12.40 | [TK]D-Fender | JonXP: Have your telco watch and advise. |
| 03:12.49 | [TK]D-Fender | JonXP: I've done what I can for now. |
| 03:12.58 | JonXP | Okie dokie, tahnks for your help. |
| 03:13.12 | JonXP | I just woner what I need to do if they say "Turn off CID" |
| 03:13.18 | *** join/#asterisk tobias (n=tobias@cpe-076-182-118-165.nc.res.rr.com) |
| 03:13.21 | JonXP | :D |
| 03:13.38 | JonXP | Oh well, figured I'd give it another shot before bed tonight. Once again, thanks. |
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| 03:21.45 | JonXP | [TK]D-Fender: For future reference...adding "Set(CALLERID(name)=)" to the dial plan made it work correctly |
| 03:22.09 | [TK]D-Fender | JonXP: lol.... |
| 03:22.16 | [TK]D-Fender | JonXP: Suckcess! |
| 03:22.20 | JonXP | Indeed! |
| 03:22.58 | JonXP | So my hunch was correct, I just had to figure out how to do it. :D |
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| 03:23.35 | *** mode/#asterisk [+o lmadsen] by ChanServ |
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| 03:26.03 | ReDNeQ | sup |
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| 03:40.43 | jeev | FENDER!!!!!!!!! |
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| 04:15.11 | nnY_2 | [TK]D-Fender btw I may need to contract some time from ya. Chasing down an issue with sip channels staying open after BYE. I have verbose logging enabled and working on gathering intel. More tomorrow. |
| 04:15.40 | nnY_2 | calling it quits for now. i'll let it burn in the meantime |
| 04:15.44 | [TK]D-Fender | k |
| 04:20.20 | heedly | is it possible to restrict what causes a retry for a .call file? |
| 04:24.13 | [TK]D-Fender | heedly: FAILURE |
| 04:27.17 | [TK]D-Fender | heedly: if you want restrict it, use a local channel, and a single retry and do all checks yourself. |
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| 06:46.17 | j0 | any recommendations for a sip or iax provider in the vancouver or seattle area? |
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| 06:47.13 | blackhole | Hi, if i made some changes in iax.conf do i need to restart my asterisk or would changes work and be accepted? |
| 06:50.46 | creativx | reload it, no restart needed |
| 06:51.00 | blackhole | creativx, what i need to do to reload it? |
| 06:51.58 | creativx | the CLI, iax2 reload i guess |
| 06:52.03 | creativx | in the CLI that is |
| 07:00.59 | kaldemar | if the bind address is changed, a restart is needed. |
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| 07:02.33 | creativx | indeed. |
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| 07:35.26 | roxlu | hi |
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| 07:49.11 | drmessano | Someone want to test something for me very quickly? |
| 07:50.32 | drmessano | Create an extension to dial sip/(some extension number you have on your system locally)@somedomain.com |
| 07:50.43 | drmessano | and see if it errors out or dials the local extension |
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| 08:10.57 | tapic | Do you have information about if G option of Dial command bridge the channels or not? |
| 08:11.55 | tapic | I want to redirect the callee to AGI script and at the end bridge the channels. |
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| 08:23.41 | mort_gib | Hi I have some problems with this error msg: chan_sip.c: Remote host can't match request BYE to call |
| 08:23.44 | mort_gib | Any insight |
| 08:24.10 | mort_gib | This drops calls, but makes it look a bit like a normal hangup |
| 08:25.07 | mort_gib | It's driving me nuts, not to mention my client |
| 08:25.56 | mort_gib | Google is not too helfull... |
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| 08:38.50 | kaldemar | mort_gib: looks like you're receiving a SIP 481 to a BYE. you'll have to give at least a trace of what's going on and tell what you're doing to get any help. |
| 08:41.20 | *** join/#asterisk techie (n=techie@m630e36d0.tmodns.net) |
| 08:47.42 | mort_gib | kaldemar: Well, thing is it works 95% of the time so I haven't got SIP trace turned on yet |
| 08:47.54 | mort_gib | -Could it e phone firmware?? |
| 08:48.01 | *** join/#asterisk abatkin (n=abatkin@81.130.203.231) |
| 08:48.30 | kaldemar | sure, based on that info it could be anything. |
| 08:49.00 | mort_gib | I just don't get this consistently... |
| 08:49.29 | mort_gib | So no good advice? -Other than turn on SIP trace and pray to my ancient angry goods :-) |
| 08:50.02 | *** join/#asterisk CybDev (i=cybdev@unaffiliated/cybdev) |
| 08:51.51 | kaldemar | find out what's happening when you get the 481 from the phone which causes that warning message. |
| 08:52.45 | mort_gib | So I have to up logging in * -in /etc/asterisk/asterisk.conf -Right, or do I have to set a global option in sip.conf |
| 08:54.05 | kaldemar | erm.. afaik you can't set up SIP debug in config files, but in the CLI. |
| 08:54.38 | mort_gib | Damn, that's what I feared! You see this happens 3-4 times a week... |
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| 09:39.58 | cjk | hi |
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| 09:40.14 | rukus | hello ppl. |
| 09:40.53 | cjk | i am actually having troubles to see the callerid of the call i pickup. now i am not sure if this should work in asterisk or if its still a feature that is not in asterisk? Is this related to RFC 3891? |
| 09:41.13 | rukus | any pointers to a setup I need, is there a way to get a client to register to a astrisk server, and then at the same time register the same client on an upstream sip_proxy ? (so ppl can roam ) |
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| 09:53.30 | mort_gib | rukus: more than one "identity" on the client |
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| 10:05.48 | _foxfire_ | hello, an1 got some experience with digiums B410P cards ? |
| 10:05.51 | *** join/#asterisk benneton (n=DELL@89.111.209.28) |
| 10:06.17 | benneton | Hi guys. Someone help me, please.. |
| 10:09.41 | *** join/#asterisk genin (n=phrame@ANice-252-1-10-118.w82-122.abo.wanadoo.fr) |
| 10:09.46 | genin | mornin |
| 10:10.25 | genin | does anyone know if there are some type of filters i can use to smooth out a foriegn accent so it sounds more british or american? |
| 10:10.58 | benneton | i can connect 2 asterisk pbx via iax. (both dyndns). but, when i add another pbx in iax.conf (type=friend) on first pbx, second cannot connect because i've added third one. (on first one i receive this message: Host xxx.xxx.xxx.xxx failed to authenticate as "third server") <<< it should use second server context in iax.conf, not third one?? |
| 10:11.08 | genin | something that was already created for asterisk or should i be looking at something that goes in between the phone and the users gateway |
| 10:11.31 | benneton | anyone have time to help me? thanx |
| 10:13.50 | _foxfire_ | benneton i am confused, you can configure as much servers as you want |
| 10:14.33 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 10:15.15 | _foxfire_ | i never has problems with this only be carefull that you use different lables in the [] |
| 10:15.59 | genin | this is difficult trying to figure out how to make indians in a call center sound like they are american or british |
| 10:16.45 | defswork | genin: don't make then friends |
| 10:17.14 | defswork | benneton even |
| 10:17.16 | defswork | :o |
| 10:17.40 | benneton | yes, it is logical to connect much servers i want |
| 10:17.42 | benneton | :) |
| 10:18.01 | benneton | but, it is something i'm trying to do without luck |
| 10:18.05 | genin | heh |
| 10:18.26 | benneton | :) |
| 10:18.44 | benneton | foxfire, what do you mean different labels? |
| 10:18.49 | benneton | in [] |
| 10:19.00 | benneton | isn't that a username? |
| 10:19.13 | _foxfire_ | each server entry starts with a lable like [server1] then the options |
| 10:19.22 | benneton | yes |
| 10:19.32 | benneton | i will send pastebin of my confs |
| 10:19.40 | benneton | on both servers |
| 10:19.57 | benneton | i repeat, i have dyndns on all 3 pbx |
| 10:20.29 | benneton | and dnsmgr enabled |
| 10:21.21 | _foxfire_ | not a problem , if you want you can use register comand on each server using host=dynamic |
| 10:22.14 | benneton | yes, but i have problems with register command... it fails to authenticate when server 3 is included? |
| 10:22.20 | benneton | weird thing, i say |
| 10:22.22 | benneton | :D |
| 10:22.50 | _foxfire_ | ok send me the pastebin link |
| 10:22.51 | benneton | everything is fine when 2 of them is connected |
| 10:22.53 | benneton | ok |
| 10:22.57 | benneton | wait, please |
| 10:25.14 | cjk | hi, i am having troubles to see the callerid of the call i pickup'ed. now i am not sure if this should work in asterisk or if its still a feature that is not in asterisk? Is this related to RFC 3891? |
| 10:38.09 | benneton | _foxfire_ |
| 10:38.15 | benneton | this is my pastebin |
| 10:38.22 | benneton | http://pastebin.com/m6e169d6d |
| 10:38.35 | benneton | thanks for your time |
| 10:52.20 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 10:52.27 | benneton | _foxfire_ |
| 10:52.50 | benneton | while I'm writing this to you, I've found a sollution |
| 10:53.13 | benneton | something came up fo me... :) |
| 10:54.17 | benneton | i will post my solution somewhere |
| 11:02.37 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 11:05.55 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 11:12.29 | benneton | why this is happening: 2nd iax.server trying to connect using 3rd user/peer (last one in iax.conf context) and then I get message on server1: Host xxx.xxx.xxx.xxx failed to authenticate as "third server") |
| 11:13.01 | benneton | damn the devil... |
| 11:13.05 | benneton | to hell :D |
| 11:13.50 | benneton | http://pastebin.com/m6e169d6d |
| 11:14.23 | benneton | (yes, i didn't find a solution) |
| 11:14.46 | *** join/#asterisk gnorbert (n=gnorbert@228-177.static.ew.hu) |
| 11:14.49 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 11:16.41 | gnorbert | Hi, I should play a sound in a meetme conference. I make it with call files, it calls the conference and plays the sound. Does somebody have an idea, how could I loop a sound file? |
| 11:16.50 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 11:18.45 | kaldemar | ~book |
| 11:18.46 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 11:18.51 | kaldemar | benneton: ^ |
| 11:20.09 | benneton | yes? |
| 11:20.17 | gnorbert | I wrote a script, that makes a call file in the directory after a given time interval, but that doensn't play it continously. |
| 11:20.17 | kaldemar | for examples. |
| 11:20.33 | benneton | ok |
| 11:20.41 | kaldemar | you're dialing a peer but your peer definitions don't have a username. |
| 11:20.54 | MaliutaLap | gnorbert: have you looked at the options to Play() |
| 11:21.21 | gnorbert | I use Application: Playback in the call file. |
| 11:21.35 | MaliutaLap | gnorbert: look at the options for it |
| 11:22.42 | gnorbert | MaliutaLap: Thank you, if I don't find anything (hope I find), I come back. :) But anyway thanks for the tip. :) |
| 11:22.48 | benneton | Book is just what i want |
| 11:22.59 | benneton | i didn't find solution in TFOT2 |
| 11:23.13 | benneton | hope there is everything i need |
| 11:23.15 | benneton | tnx |
| 11:24.08 | kaldemar | benneton: you need a correct username for outbound calls. |
| 11:25.30 | benneton | yes |
| 11:25.51 | *** join/#asterisk ez` (n=ez@c66.203.221-242.clta.globetrotter.net) |
| 11:25.58 | kaldemar | and your passwords don't match either. |
| 11:26.56 | kaldemar | when you dial out using IAX2/<peer>, asterisk uses the secret it finds in iax conf under [<peer>]. if the other end has something else as a secret in the corresponding context, you'll have an authentication problem. |
| 11:26.58 | benneton | do i need [server1]as peer [server2]as friend and vice versa on other BOX? |
| 11:27.21 | benneton | or i can do it as a friend? |
| 11:27.29 | kaldemar | that's why many people prefer to dial with IAX2/<username>:<secret>@<peer>/... |
| 11:28.13 | *** join/#asterisk shinao1 (n=shinao1@smtp.gtbank.com) |
| 11:28.17 | benneton | with [server1] type=peer username=server2 > on other side [server1] type=user |
| 11:28.29 | benneton | oops |
| 11:28.34 | benneton | with [server1] type=peer username=server2 > on other side [server2] type=user |
| 11:28.44 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 11:28.56 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 11:28.58 | _foxfire_ | benneton did youy get my message ? |
| 11:29.41 | kaldemar | you could use friend in both ends. |
| 11:30.06 | benneton | yes? |
| 11:31.12 | benneton | like this [server1] type=friend username=server2 -- OTHER SIDE [server2] type=friend username=server1 |
| 11:31.26 | benneton | (for calls in both directions) |
| 11:31.42 | *** part/#asterisk cplx (n=cplx@59.167.199.141) |
| 11:32.20 | kaldemar | yes. and in that case, if you dial with IAX/server1, you need same secrets in both ends. |
| 11:35.36 | benneton | THANKS A MILLION! |
| 11:35.41 | benneton | :D |
| 11:45.09 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-68-114.vif.net) |
| 11:45.25 | benneton | kaldermar - can i use RSA with this method you said: " dial with IAX2/<username>:<secret>@<peer>/..." |
| 11:46.52 | *** join/#asterisk mvanbaak_ (n=michiel@vanbaak.xs4all.nl) |
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| 11:52.20 | kaldemar | benneton: don't know. but with keys you can specify inkeys and outkeys for a peer, which makes it a bit more clear than with a secret. |
| 11:52.36 | *** join/#asterisk scampbell (n=scampbel@mail.scampbell.net) |
| 11:53.31 | benneton | ok |
| 11:53.38 | benneton | tnx again |
| 12:00.05 | gnorbert | MaliutaLap: Sorry, I found only options, but I have no idea, what can those be used for. |
| 12:01.05 | gnorbert | *only two options |
| 12:01.28 | gnorbert | Could you help a bit more? |
| 12:03.00 | gnorbert | Or anybody, who has an idea, how can I play a sound file continously with a call file? |
| 12:03.27 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 12:04.56 | *** join/#asterisk scampbell (n=scampbel@199.105.195.156) |
| 12:05.48 | kaldemar | gnorbert: point the call file to a looping extension. |
| 12:06.56 | gnorbert | kaldemar: That version, when I called an extension and played the sound file there, didn't work. |
| 12:08.11 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:08.13 | kaldemar | me do not understand. |
| 12:09.43 | kaldemar | gnorbert: aah, don't trigger the application with it, but an extension that loops the playback application. |
| 12:11.33 | kaldemar | if that didn't work, you could try telling how in form of pasting the callfile and a CLI trace. |
| 12:12.06 | gnorbert | kaldemar: Hmm, I think I don't fully understand it... |
| 12:12.10 | *** join/#asterisk LND (n=Lee@89.192.176.216) |
| 12:12.19 | kaldemar | i've done exactly the same thing you're going for, with success. |
| 12:12.34 | gnorbert | I have to play looped and not looped files in the same time. |
| 12:13.07 | *** join/#asterisk Gat0rvean (n=gredish@64.191.128.145) |
| 12:13.50 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 12:18.22 | *** part/#asterisk JenniferAkemi (n=akemi@conference/cluecon/x-ad776bedabbd831e) |
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| 12:20.51 | *** join/#asterisk mattwj2002 (n=matt@c-76-17-132-205.hsd1.mn.comcast.net) |
| 12:21.01 | mattwj2002 | good morning everyone :) |
| 12:23.18 | mattwj2002 | I just noticed I paid $1.40 for 70 minute call to Google 411 |
| 12:23.19 | mattwj2002 | :| |
| 12:23.47 | mattwj2002 | I know I didn't talk that long....is there any chance that the Asterisk box just didn't hang up? |
| 12:25.43 | [TK]D-Fender | mattwj2002: Look at your CDR |
| 12:27.09 | mattwj2002 | I can't |
| 12:27.21 | mattwj2002 | I blew away that Asterisk server a while back |
| 12:27.40 | mattwj2002 | it isn't that big of a deal |
| 12:27.48 | mattwj2002 | just wondering what could have happened |
| 12:29.12 | [TK]D-Fender | mattwj2002: Don't start the investigation after the cleaning staff have scrubbed everything down. Just asking looks immensely stupid. |
| 12:30.02 | mattwj2002 | good call |
| 12:31.03 | mattwj2002 | just for my own future reference.... |
| 12:31.21 | mattwj2002 | if I don't have the hangup command could this happen again? |
| 12:31.57 | [TK]D-Fender | mattwj2002: on an outbound call? Not likely. |
| 12:32.11 | [TK]D-Fender | mattwj2002: So just stop neurosing already... |
| 12:33.21 | mattwj2002 | okay |
| 12:39.09 | gnorbert | kaldemar: The other problem that I should be able to handle it from console with another program and be able to start play any kind of file. |
| 12:39.41 | gnorbert | So I can't say Playback, which file to play. |
| 12:40.47 | mattwj2002 | bye everyone |
| 12:40.48 | mattwj2002 | :) |
| 12:40.53 | *** part/#asterisk mattwj2002 (n=matt@c-76-17-132-205.hsd1.mn.comcast.net) |
| 12:43.28 | _foxfire_ | hello, an1 got some experience with digiums B410P cards ? |
| 12:46.44 | gnorbert | Does somebody have an idea, how can be solved from a call file, to loop a sound file? (I use Application:Playback) |
| 12:46.45 | kaldemar | gnorbert: use the Set option in the call file to set a filename to a variable. |
| 12:47.16 | kaldemar | and drop the Application:Playback already. |
| 12:48.15 | *** join/#asterisk LemensTS (i=LemensTS@adsl-70-238-187-205.dsl.stlsmo.sbcglobal.net) |
| 12:49.10 | kaldemar | loop Playback(${foo}) in the extension and put Set:foo=/path/to/bar in the call file. |
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| 12:55.50 | *** mode/#asterisk [+o lmadsen] by ChanServ |
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| 12:57.34 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 12:57.57 | gnorbert | kaldemar: I needed Application:Playback, because then I didn't need Extension in the call file (because I had application), But then if I leave Application:playback, what shall I write as extension to avoid it call itself twice? |
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| 13:06.24 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:08.14 | benneton | does dnsmgr refresh peer in "register => username:pass@someaddress" line? |
| 13:08.42 | genin | hey folks |
| 13:08.47 | genin | whats the best softphone |
| 13:08.50 | genin | you guys have used |
| 13:08.53 | benneton | or only option is to change iax2.h file: "#define IAX_DEFAULT_REG_EXPIRE 60 " to something else |
| 13:09.15 | benneton | genin: iax2 protocol > idefisk or zoiper |
| 13:09.28 | benneton | sip: i use X-lite |
| 13:09.33 | gnorbert | kaldemar: Sorry, but so what shall I write in the call file actually? I know the Channel line, the Priority line, and then the Set line, but what should be there, if not application? |
| 13:10.10 | benneton | gnorbert: context |
| 13:10.11 | genin | x lite? |
| 13:10.16 | benneton | genin: yes |
| 13:10.16 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 13:10.20 | genin | ah cool |
| 13:10.22 | genin | thanks |
| 13:10.25 | genin | ill try it out ;) |
| 13:10.28 | ManxPower | benneton: dnsmgr deals with DNS not registration |
| 13:10.39 | gnorbert | benneton: And then doesn't it call twice itself? |
| 13:11.01 | slugtwinturbo | guys, what's needed in order to create/have a number to which someone can dial a costumer of the voip network, how do I assign/give a number to them ? |
| 13:11.15 | benneton | gnorbert: use another context |
| 13:11.23 | benneton | with playback application |
| 13:11.37 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:11.56 | gnorbert | benneton: Ok, I make a try |
| 13:12.06 | *** join/#asterisk brodiem (n=brodiem@rrcs-24-227-171-186.sw.biz.rr.com) |
| 13:13.17 | benneton | ManxPower: i thought that dnsmgr is going through both, [usernames] and register line |
| 13:13.21 | benneton | tnx |
| 13:13.38 | ManxPower | benneton: it only looks up hostnames |
| 13:13.49 | benneton | hm |
| 13:14.13 | benneton | if this is in register line : register => username:pass@someaddress.com |
| 13:14.33 | benneton | there is host name |
| 13:14.44 | benneton | maybe it resolv it? |
| 13:15.09 | benneton | i have dynamic addresses on both server |
| 13:15.17 | ManxPower | benneton: That was not your question. I don't know the answer to your new question. |
| 13:15.35 | ManxPower | But I know that Asterisk does not work well with changing DNS |
| 13:15.42 | benneton | well, i thought we have a chat |
| 13:15.43 | benneton | np |
| 13:16.20 | *** join/#asterisk JenniferAkemi (n=akemi@conference/cluecon/x-e9fa02028d819a2c) |
| 13:18.08 | benneton | slugtwinturbo: do you need to assign PTSN number to a voip customer on your PBX? |
| 13:18.12 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 13:19.39 | gnorbert | benneton: It doesn't work, it looks like this now: http://pastebin.com/d5ddce4cb Sorry, I guess I missunderstood something. |
| 13:19.49 | benneton | no problem |
| 13:20.19 | *** join/#asterisk VaNNi (n=VaNNi___@38.98.61.142) |
| 13:20.30 | [TK]D-Fender | gnorbert: You didn't set an EXTENSION <- |
| 13:21.37 | gnorbert | [TK]D-Fender: But when I had EXTENSION, it called itself twice... |
| 13:22.01 | [TK]D-Fender | gnorbert: Because you were picking the SAME one. |
| 13:22.26 | benneton | gnorbert: default is mentioned two times |
| 13:22.30 | [TK]D-Fender | gnorbert: You call out to CHANNEL. that is one thing. It should NOT be related to the place in your dialplan that you will DUMP the call once its answered |
| 13:22.46 | [TK]D-Fender | benneton: That can be fine |
| 13:24.38 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 13:24.54 | gego | hello, |
| 13:25.14 | gego | how do i kill a sip channel? |
| 13:25.47 | [TK]D-Fender | gego: "soft hangup [channelwithoutbraces]" |
| 13:27.31 | gego | thanks - but the channel lives on - might have been too soft :? |
| 13:27.53 | [TK]D-Fender | gego: pastebin your channel dumps. |
| 13:29.15 | *** part/#asterisk JenniferAkemi (n=akemi@conference/cluecon/x-e9fa02028d819a2c) |
| 13:30.12 | brodiem | [TK]D-Fender: Do you know if the new polycom line (320/550/etc) added support to send NAT keep-alives? And without requiring you specifically tell it what your external IP is? |
| 13:30.53 | [TK]D-Fender | brodiem: thats is SIP 2.0 that has it, not model specific. As for the IP, not sure if you can do it without knowing |
| 13:31.01 | [TK]D-Fender | broGo download the admin guide. |
| 13:31.37 | brodiem | ty, yes I have one right here I was just too lazy to power it up and check heh |
| 13:33.16 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:34.45 | x86 | I had NO success with the IP330 |
| 13:34.50 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 13:35.06 | x86 | it kept locking up and sometimes randomly rebooting... |
| 13:35.10 | jaytee | no success with the IP330 for what? |
| 13:35.16 | x86 | would lock up right after registering with * |
| 13:35.30 | jaytee | I'm using several just fine here with * 1.4.x |
| 13:35.37 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 13:35.38 | jaytee | and an IP550 |
| 13:35.56 | x86 | I'm using 1.4.12.1 |
| 13:36.11 | jaytee | 12? or 21? |
| 13:36.13 | Kobaz | all of my polycom phones no longer keep track of missed calls after a firmware update I did a few weeks ago... anyone know offhand how to fix that? |
| 13:36.16 | x86 | 301, 501, and 601's work like a champ |
| 13:36.23 | x86 | jaytee: if I meant 21, I would have said 21 ;) |
| 13:36.38 | jaytee | ok. I'm on 1.4.15 |
| 13:36.48 | x86 | I've not tried any 550's or 650's |
| 13:36.53 | [TK]D-Fender | Kobaz: Go look in your provisioning. There is away to disable them. Make sure they haven't been. |
| 13:36.57 | jaytee | 550's are nice |
| 13:37.01 | x86 | but I've tried the 320 and 330, and both were complete fail |
| 13:37.12 | [TK]D-Fender | x86: And NO... your firmware version couldn't possibly be relevant here.... |
| 13:37.13 | *** join/#asterisk fnordus (n=dnall@70.71.225.48) |
| 13:37.19 | Kobaz | [TK]D-Fender: feature.8.name="calllist-missed" feature.8.enabled="1" |
| 13:37.32 | Kobaz | [TK]D-Fender: that's the only thing i can find |
| 13:37.50 | x86 | of course, the area in the building I was using them on was wired with Cat4, which can't do 100mbps, which is what the switch was negotiating with all the devices at |
| 13:38.10 | jaytee | ugh! that'll screw you up for sure. |
| 13:38.10 | *** join/#asterisk MrNaz (n=naz@ppp121-44-207-242.lns3.mel4.internode.on.net) |
| 13:38.17 | x86 | [TK]D-Fender: latest firmware and bootrom as of about 2 months ago... Polycom threw their arms up and had no idea what the problem was |
| 13:38.31 | x86 | jaytee: 301's and 501's can handle that just fine |
| 13:38.35 | jaytee | CAT4? I vaguely remember that being around for about a week before CAT5 came out in the late 80's. |
| 13:38.42 | *** join/#asterisk moy (n=moy@nat/ibm/x-1da505e958e9db00) |
| 13:38.54 | x86 | I should really force the switches down there to only run 10mbps |
| 13:39.02 | [TK]D-Fender | x86: thanks for providing LOGS, and an answer with a specific VERSION along with your configs |
| 13:39.04 | gego | [TK]D-Fender: ehhhm ... channel dumps? as in core show channel? |
| 13:39.20 | [TK]D-Fender | gego: "core show channels concise", "sip show channels" |
| 13:39.46 | x86 | jaytee: this building was built in 1993, and they hired an ELECTRICIAN to do the wiring... electrician found a damn good deal on a shit ton of cat4, and the powers that be wanted to save money, so of course that's what they went with |
| 13:40.25 | brodiem | x86: got to love how most electricians do their crimping, the jacket like 1/2" outside of the RJ45 |
| 13:40.36 | jaytee | and the powers that be are either gone or too high up to make it worth risking mentioning the wiring |
| 13:41.02 | x86 | they're still here |
| 13:41.06 | x86 | still pinching pennies |
| 13:41.43 | x86 | I put in cat6 for a whole department, on a brand new gigabit switch, but they wont let me do that for the rest of the building :( |
| 13:42.03 | jaytee | I've come across a few network runs that are terminated according to no color code I've ever seen, certainly not 568A or 568B. Who uses WhiteBrown / Brown for Send? |
| 13:42.34 | x86 | [TK]D-Fender: polycom couldn't figure out why it was locking up, so i figured no one could... returned it as defective and bought a 301 |
| 13:42.47 | x86 | jaytee: haha |
| 13:43.01 | [TK]D-Fender | x86: And your ability to pass us something useful is pretty much non-existant. |
| 13:43.02 | jaytee | and some that aren't even cat3 but seem to be some kind of plenum covered CAT2 or something. Like one twist per foot. |
| 13:43.25 | x86 | jaytee: I've seen blue, green, brown, orange, w/o, w/br, w/gr, w/bl before |
| 13:43.41 | gego | [TK]D-Fender: this is the output about the "stuck" channel: http://de.pastebin.ca/1093780 |
| 13:43.48 | x86 | [TK]D-Fender: i did two months ago before i RETURNED it ;) |
| 13:44.13 | x86 | jaytee: cat3 is about one twist per foot |
| 13:44.41 | [TK]D-Fender | x86: hindsight is 20/20 |
| 13:44.48 | jaytee | from what I understand they had this old guy named Floyd who was kinda handy with a hammer and they "edge-a-muhcated" him. |
| 13:45.09 | x86 | jaytee: *shudders* |
| 13:45.09 | [TK]D-Fender | gego: and "core show channels (without concise) |
| 13:47.27 | gego | [TK]D-Fender: there is (funny enough) less information http://de.pastebin.ca/1093785 without concise ?!? |
| 13:49.00 | [TK]D-Fender | gego: "concice" is "more", but less readable. Wanted to verify. Yup, that chan is WHACKED. I'd suggest a restart... thats about all thats left. |
| 13:51.03 | gego | [TK]D-Fender: tks a lot. i think i let it hang there until i have to restart anyhow |
| 13:51.41 | [TK]D-Fender | gego: "restart when convenient" |
| 13:51.41 | gego | [TK]D-Fender: all my f.. gxp phones forget their subscriptions after a restart :-( |
| 13:53.35 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:55.44 | x86 | ~grandstream |
| 13:55.45 | jbot | somebody said grandstream was the Yugo of VoIP hardware. Run. Run away now. |
| 13:55.54 | x86 | gego: read that |
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| 14:04.01 | *** join/#asterisk rabby (n=rabby@p4FCEA911.dip0.t-ipconnect.de) |
| 14:04.04 | rabby | hi |
| 14:04.29 | rabby | doesn't asterisk -r want to tell me about incomming calls if everythings works properly? |
| 14:04.53 | ManxPower | rabby: try "asterisk -rvvv" |
| 14:05.10 | *** join/#asterisk sgtpepper (n=ncorrare@200.61.187.185) |
| 14:05.12 | *** part/#asterisk sgtpepper (n=ncorrare@200.61.187.185) |
| 14:05.17 | rabby | i wonder why it does not :( i use a isdn fritz!card and the capi and fcpci module does not show any error; but i think it does not work in any way |
| 14:05.23 | *** join/#asterisk tobias (n=tobias@user-0c2hj2e.cable.mindspring.com) |
| 14:06.13 | ManxPower | First you say "everythings works properly" then you say "but i think it does not work in any way". Which is it? |
| 14:06.16 | rabby | verbose mode 3 does not show me anythink, too |
| 14:06.26 | ManxPower | then you are not receiving any calls |
| 14:06.52 | ManxPower | In any case, if you are using ISDN BRI I can't help you. Perhaps one of the europeople here can. |
| 14:06.55 | rabby | that's what i wonder about, too. the system does not tell me any error about the config or modules i use and installed, |
| 14:07.18 | rabby | where to find the europe people? |
| 14:07.23 | *** join/#asterisk c4t3l (n=rcallico@mail.questia.com) |
| 14:07.37 | rabby | is no one here who uses capi?!# |
| 14:07.37 | ManxPower | here during their business hours and early evening |
| 14:07.47 | *** part/#asterisk benneton (n=DELL@89.111.208.110) |
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| 14:08.21 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:09.16 | rabby | perhaps any one can have a look at http://nopaste.info/b2119a9222.html as i wonder about the lots of zeroes there. i guess, that may be the reason for asterisk not receiving my calls |
| 14:10.34 | *** join/#asterisk seanmh (i=seanmh@216.31.101.25) |
| 14:13.27 | [TK]D-Fender | rabby: do "set verbose 10", and "set debug 10". pastebin what displays for an incoming call attempt" |
| 14:13.40 | dominic1 | how can I call I mysql function with func_odbc? |
| 14:13.54 | dominic1 | a stored procedure... |
| 14:13.59 | dominic1 | sorry |
| 14:14.56 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 14:15.00 | rabby | dominic1: the same way like select queries. but talk about that in #mysql |
| 14:16.03 | rabby | [TK]D-Fender: nothing! that's the problem :( |
| 14:16.35 | Kobaz | is there any way to do a "background dial" like... so i have a thing dialing one phone for 10 seconds, then dialing that one phone plus another phone for 10 seconds... an incrementally increasing ring group sort of thing... the problem is now i'm placing two seperate calls to that original phone... ideally i want to ring phone one for 30 seconds, and then after 10, add phone two |
| 14:17.29 | [TK]D-Fender | rabby: Then its either a problem with your line, your card, its driver, or your channel driver setup. |
| 14:17.59 | [TK]D-Fender | Kobaz: nest them in Local channels. |
| 14:18.14 | Kobaz | [TK]D-Fender: mmm, k |
| 14:18.18 | [TK]D-Fender | Kobaz: dial multiple local channels each with their own offset delays. |
| 14:18.27 | Kobaz | ah |
| 14:19.36 | rabby | [TK]D-Fender: so i need to find out, but do not know how to do so. if i switch my telephone into the line, this receives the call. so i hope, that is not the source of problems. the driver does not show up errors. no error in dmesg and capiinfo tells me the strange output full of zeroes and i do not know if this is correct: http://nopaste.info/b2119a9222.html |
| 14:19.50 | [TK]D-Fender | Kobaz: Dial(Local/10@delay0&Local/20@delay10&Local/30@delay20) |
| 14:21.56 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.95.130) |
| 14:25.19 | *** join/#asterisk Ferdirand (n=max@2002:d5a2:16e4:0:0:0:0:1) |
| 14:25.40 | brodiem | haha look what I just scored: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=170246065816 |
| 14:26.27 | anonymouz666 | does zaptel-1.4 improves IRQ handling over zaptel-1.2? |
| 14:26.38 | ManxPower | anonymouz666: I don't think so |
| 14:26.48 | ManxPower | 1.2 had major IRQ updates at one point. |
| 14:26.49 | anonymouz666 | PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 - I need get rid of this |
| 14:26.54 | anonymouz666 | In a production server |
| 14:26.58 | anonymouz666 | it's almost impossible |
| 14:27.00 | ManxPower | anonymouz666: It sucks to be you. |
| 14:27.05 | anonymouz666 | really. |
| 14:27.11 | anonymouz666 | crap motherboards. |
| 14:27.17 | *** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 14:27.17 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:27.19 | [TK]D-Fender | anonymouz666: Yes |
| 14:27.23 | *** join/#asterisk Morrocco (n=morrocco@189.164.96.222) |
| 14:27.33 | ManxPower | anonymouz666: Contact Digium support. It won't help, but it might make you feel better. |
| 14:27.41 | [TK]D-Fender | anonymouz666: pastebin "cat /proc/interrupts" |
| 14:28.21 | anonymouz666 | ManxPower: I don't speak english to do that. So I won't feel better. My brain is trained just to read/write. Heh |
| 14:28.26 | anonymouz666 | [TK]D-Fender: one minute |
| 14:28.38 | Morrocco | Hi Guys, I would like to register a SIP service called Voiceline from net2phone with my Asterisk box, but I dont know how to do it, they dont give you any instructions so I got some sniffed information from the registration but Im not good with asterisk to test it, can some one help me? |
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| 14:29.20 | anonymouz666 | [TK]D-Fender: http://pastebin.com/m47dc277e |
| 14:29.32 | ManxPower | anonymouz666: Generally any solution to HDLC abort errors are things you don't want to do. i.e. disable onboard networking, disable onboard RAID, swap motherboards, switch to Sangoma, replace the onboard SATA with an addon SATA card, etc. |
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| 14:30.01 | [TK]D-Fender | anonymouz666: "dmesg" please |
| 14:30.24 | anonymouz666 | ManxPower: I just realized that. Since everything is disabled...there's nothing I can do. |
| 14:30.51 | ManxPower | anonymouz666: you can TRY 1.4, but you would have to upgrade Asterisk to 1.4 too. |
| 14:30.52 | anonymouz666 | it will be much faster to switch the motherboard. |
| 14:31.00 | ManxPower | exactly! |
| 14:31.00 | anonymouz666 | yeah I know |
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| 14:32.15 | minthome | would this work? GotoIfTime(23:00-9:00|mon-fri|*|*?placetogo,s,1) |
| 14:32.34 | [TK]D-Fender | minthome: Go try |
| 14:32.54 | anonymouz666 | [TK]D-Fender: http://pastebin.com/m490f5526 |
| 14:32.58 | minthome | heh, sadly, i can't really for another 7 hours |
| 14:33.08 | minthome | unless i change the time |
| 14:33.27 | [TK]D-Fender | minthome: Change your raqnge for the test |
| 14:33.53 | minthome | i know something like 7:00-22:00 would work |
| 14:33.55 | anonymouz666 | ManxPower: I just asked about zaptel-1.4 to try to make things faster. It's better to learn from others experience than waste your time to figure out that does not help to switch to zaptel-1.4. |
| 14:34.16 | ManxPower | anonymouz666: * understand, but when you have few other choices... |
| 14:34.19 | minthome | it's going around the 24:00/0:00 corner that I'm not sure about |
| 14:34.27 | [TK]D-Fender | minthome: Yes, should work |
| 14:34.37 | [TK]D-Fender | anonymouz666: YES <- |
| 14:35.17 | minthome | heh, yeah, it's that "should" that i'm worried about... guess i'll find out this afternoon |
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| 14:43.47 | *** join/#asterisk ejos (n=ejos@207.138.45.41) |
| 14:44.02 | ejos | Greetings! |
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| 14:48.33 | nikolaev | how can I register a SIP Phone which is behind NAT to the asterisk SIP Proxy ? |
| 14:49.06 | nikolaev | I see the asterisk's reply "Trying" but obvoiusly the packet does not reach the SIP Phone |
| 14:49.10 | JT | asterisk is NOT a sip proxy |
| 14:49.44 | nikolaev | okay |
| 14:50.30 | nikolaev | any workaround for the NAT issue ? |
| 14:52.28 | JT | follow the first guide at |
| 14:52.29 | JT | ~sipnat |
| 14:52.30 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 14:53.11 | dominic1 | anybody had trouble with odbc and a error like that PROCEDURE asterisk.test2 can't return a result set in the given context? I get this while calling a stored procedure |
| 14:54.13 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 14:55.13 | ejos | Somebody know how many nodes support asterisk in the best use case?, that is i heard that asterisk support only 300 nodes, that is true? |
| 14:55.21 | nikolaev | JT Thanks |
| 14:55.27 | *** join/#asterisk ta^3 (n=tacvbo@conference/cluecon/x-73cefa00c246e823) |
| 14:56.07 | ejos | and i have almost 8000 nodes on TDM |
| 14:57.50 | [TK]D-Fender | ejos: In large-scale setups like that people usually run SER in front and use * only as a back-end application server |
| 14:59.27 | ejos | [TK]D-Fender, SER? |
| 14:59.32 | [TK]D-Fender | ~ser |
| 14:59.33 | jbot | ser is, like, Sip Express Router - see http://www.iptel.org/ser/, or an old secret method of obtaining a havoc of NAT problems, or at #ser |
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| 15:00.01 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:01.00 | ejos | jbot, thanks both |
| 15:01.00 | jbot | ejos: pas de quoi |
| 15:01.18 | [TK]D-Fender | lol |
| 15:03.01 | uski | is there a way to set a variable (CALLERIDNAME actually) from an external program? when i receive a call, i currently do a reverse-phone-directory lookup using a script i wrote, and i store the results in a database. I'd like to alert the CALLERIDNAME so that it contains the actual name found by my program, so that it's displayed on the phones |
| 15:04.58 | [TK]D-Fender | uski: Go read the chapter on AGI in THE BOOK |
| 15:05.00 | [TK]D-Fender | ~book |
| 15:05.00 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 15:06.40 | uski | (FYI the link for the HTML is broken) |
| 15:07.20 | uski | [TK]D-Fender, may i ask... what's AGI ? i can't find it :/ |
| 15:07.25 | uski | oh found it |
| 15:07.27 | uski | thanks |
| 15:07.28 | cjk | hi, i am having troubles to see the callerid of the call i pickup'ed. now i am not sure if this should work in asterisk or if its still a feature that is not in asterisk? Is this related to RFC 3891? |
| 15:07.51 | [TK]D-Fender | uski: There is an entire CHAPTER ON IT in the book. GO READ |
| 15:07.54 | [TK]D-Fender | ~agi |
| 15:07.55 | jbot | hmm... agi is the Asterisk Gateway Interface... similar to CGI for web applications AGI lets you script call control and access databases using your favorite language. AGI wrappers are available for Python (pyst), Perl (astperl?) and other languages, or <reply> See also http://www.voip-info.org/wiki-Asterisk+AGI |
| 15:07.56 | [TK]D-Fender | ^^^^^^^^^^^ |
| 15:08.00 | uski | [TK]D-Fender, so am i |
| 15:08.04 | uski | thanks |
| 15:08.10 | uski | it's the Appendix C |
| 15:08.19 | uski | ...and Chapter 9 |
| 15:08.27 | [TK]D-Fender | uski: "Table Of Contents" is your friend. |
| 15:08.33 | *** join/#asterisk raasdnil (n=mikel@60-241-138-147.static.tpgi.com.au) |
| 15:08.39 | [TK]D-Fender | "This website is best viewed with ... YOUR EYES!" |
| 15:09.16 | uski | doesn't work for me |
| 15:09.25 | [TK]D-Fender | cjk: When you "pickup" YOU are calling out. You don't get CPID (called party ID" for outbound calls |
| 15:09.27 | uski | anyway, i got the PDF |
| 15:09.45 | [TK]D-Fender | uski: That wasn't in reference to Leif's site being down. |
| 15:09.49 | uski | ok |
| 15:09.55 | [TK]D-Fender | uski: Just a general statement. |
| 15:10.21 | *** part/#asterisk ejos (n=ejos@207.138.45.41) |
| 15:10.37 | cjk | [TK]D-Fender, i know that asterisk works like that. isn't that what RFC3891 (sip replaces) defines? how this could work? |
| 15:10.58 | *** join/#asterisk raasdnil (n=mikel@60-241-138-147.static.tpgi.com.au) |
| 15:11.03 | [TK]D-Fender | ~cipd |
| 15:11.10 | [TK]D-Fender | ~cpid |
| 15:11.10 | jbot | [~cpid] Called-Party ID is possible with * using patches on Mantis. See : http://bugs.digium.com/view.php?id=8824 |
| 15:11.18 | [TK]D-Fender | cjk: ^^^ |
| 15:11.50 | [TK]D-Fender | actually.. not sure that could work in this case. |
| 15:11.51 | cjk | [TK]D-Fender, lets check this out ;) thanks |
| 15:11.52 | [TK]D-Fender | nvm |
| 15:12.36 | *** part/#asterisk jivco (n=jivco@85.187.217.6) |
| 15:13.41 | raasdnil | [TK]D-Fender: I know it is like 10 hours later, but your suggestion of compiling zaptel, then trashing asterisk and re-compiling solved the problem of asterisk not spotting the zaptel drivers, thanks :) |
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| 15:15.02 | [TK]D-Fender | raasdnil: You're welcome |
| 15:15.07 | raasdnil | seeya |
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| 15:36.47 | uski | [TK]D-Fender, thanks, the AGI was what I needed - my script works like a charm |
| 15:36.52 | uski | :) |
| 15:41.01 | heedly | speaking of AGI, were do I get more detailed info about it failing? |
| 15:41.18 | heedly | All I see is AGISTATUS set to FAILURE, and it returning 0. |
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| 15:59.06 | lowtek | In asterisk 1.4.2x, FollowMe seems to ignore it's options, i.e, FollowMe(${EXTEN}|san), the options "san" work fine, but when I remove the options, it sill executes the standard functioanlity of asking for the caller to record their name. Any ideas? |
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| 16:05.00 | lowtek | Does the asterisk command "Wait" recognize fractions of a second? i.e., will Wait(.5) wait half a second? (500ms)? (1.4.current) |
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| 16:25.01 | [TK]D-Fender | lowtek: "core show application wait" |
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| 16:31.20 | saftsack | ~centos52bug |
| 16:31.20 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889 |
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| 16:39.29 | jjshoe | is it possible to get asterisk to make the xferbeep on completion of an attended xfer using the xfer button on the phone? (not the keycode) |
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| 16:41.04 | HardCor | is there a reason why I fail to download an iso image from http://dl2.digium.com/AsteriskNOW-1.0.2.1-x86-disc1.iso with my slow internet connection? I have tried to download via my Firefox, Download accilarator, lynx, but all fail in the middle of the process.... I have looked for a torrent of the same but failed to find it. Anyone have ideas? |
| 16:41.26 | zamba | how do i set up a sip trunk between two asterisks? |
| 16:42.01 | [TK]D-Fender | zamba: Same as you would any other ITSP. |
| 16:42.03 | _Bentley | does anyone know how I might find bkw? |
| 16:42.22 | zamba | [TK]D-Fender: ITSP? |
| 16:42.32 | discHead | HardCor-- if it were me, I would try curl next |
| 16:42.37 | zamba | internet telephony service provider.. got it :) |
| 16:42.44 | [TK]D-Fender | ~itsp |
| 16:42.44 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 16:42.51 | discHead | curl will pick up an interrupted download in the middle |
| 16:43.01 | zamba | [TK]D-Fender: well, i don't know how i do it for any other ITSP either, so :) |
| 16:43.02 | *** part/#asterisk _Bentley (i=bentley@secure.foodsled.com) |
| 16:43.13 | [TK]D-Fender | ~jerjerguide |
| 16:43.14 | jbot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 16:43.19 | HardCor | discHead: what is curl? |
| 16:43.49 | [TK]D-Fender | zamba: Inclues a sampe for NuFone. Its almost the same as setting up just a phone except that A registers to B, and the revers. A SIP call is just like any other. |
| 16:43.53 | discHead | curl is a command-line program, a Swiss-Army knife of a download utility |
| 16:44.21 | *** part/#asterisk hubguruJR (n=hubguruJ@mail.ntegratedsolutions.com) |
| 16:44.26 | discHead | http://curl.haxx.se/ |
| 16:45.00 | *** join/#asterisk rbd (n=rbd@adsl-074-229-183-112.sip.rmo.bellsouth.net) |
| 16:46.01 | HardCor | discHead: but if download-accilarator does not resume, why would any other app do it? My problem is that every download seems to through me out in the middle of the process... |
| 16:46.06 | rbd | hey guys... I'm trying to use the Record app to record out to a file ending with .temp (it will be bla.wav.temp)... it seems like this is not possible with the new format of the Record app... e.g. the format must be the same as the recorded file extension it looks like.... is it still possible to use the old Record(filename:format[|silence][|maxduration][|option]) syntax with asterisk 1.4? |
| 16:46.24 | discHead | I'm not sure, I don't have experience with Download Accelerator, I'm afraid |
| 16:46.41 | Qwell | DigitalIrony: wget can do that too... |
| 16:46.44 | Qwell | erm, discHead |
| 16:47.49 | discHead | It does? |
| 16:48.04 | Qwell | I didn't say does, I said can. |
| 16:48.05 | ManxPower | rbd: I doubt you can do what you want. I suggest recording the file with the .wav ending, then rename it if you want. |
| 16:48.21 | Qwell | wget -c |
| 16:49.02 | zamba | [TK]D-Fender: i |
| 16:49.03 | zamba | eh |
| 16:49.04 | HardCor | even wget would function like my attempt with lynx wouldn't it? |
| 16:49.12 | zamba | i'm in the other end now.. the other end of "register" |
| 16:49.27 | rbd | ManxPower: yeah, I record to .temp because I have a processing thread in my own app that looks for .wav files to process ....I'll probably have to record to .wav and then move the file to that dir once done... |
| 16:49.29 | Qwell | HardCor: You would just resume the download |
| 16:49.50 | HardCor | Qwell: with wget? |
| 16:49.57 | Qwell | wget -c |
| 16:50.12 | ManxPower | HardCor: since we don't know WHY your other programs are not resuming downloads, we can't tell you why or if using other apps would resume your download. we can only tell you apps that can resume a download. |
| 16:50.15 | HardCor | Qwell: let me try that, thanx! |
| 16:50.47 | ManxPower | Personally I've always had issues using wget to download files from Digium. It always seems to download the directory page, not the actual file. |
| 16:50.59 | Qwell | ManxPower: You're doing it wrong. |
| 16:50.59 | ManxPower | not directory, but "redirector" |
| 16:51.18 | HardCor | OKay guys, thanx, I'll try some more... |
| 16:51.28 | ManxPower | Qwell: Yeah, I know, right click on link, copy, paste in wget is not something that's supposed to work. |
| 16:52.00 | ManxPower | Well, at least on DIGIUM'S site. |
| 16:52.48 | ManxPower | I don't think that's HardCor's problem, but it could be. |
| 16:53.22 | ManxPower | Every time I download something from Digium I spend a few mins cursing whatever idiot created the new system. |
| 16:53.59 | *** join/#asterisk nny_1 (n=Scott_My@64.203.244.146) |
| 16:55.21 | ManxPower | http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/asterisk/releases/asterisk-1.4.21.2.tar.gz |
| 16:55.30 | ManxPower | like that crap. |
| 16:56.36 | *** join/#asterisk dwelsh (n=asterisk@ottawa-hs-69-20-226-218.s-ip.magma.ca) |
| 16:56.46 | nny_1 | hmm anyone wanna talk about echo.. I have been researching my ass off, and tuning various knobs buttons etc. We have an issue where the local telco is the CO for a T1 through embarq, and getting echo on local calls and not long distance. I am using the digium hardware echo canceler, and there isn't much to be adjusted there. However, in searching, I found that it is possible (? this may be a lie) to kick up the rx gain enough to assist the echo canceler |
| 16:57.06 | dwelsh | Hi. Is AsteriskGUI still available separately? I can only find it as part of AsteriskNow. |
| 16:57.09 | ManxPower | nny_1: which of the like 5 Digium EC are you using? |
| 16:57.16 | nny_1 | canceller* |
| 16:57.27 | ManxPower | nny_1: I can work with you on #asterisk-cli |
| 16:58.06 | ManxPower | nny_1: if the hardware EC is not working for you then you should contact Digium. |
| 16:58.14 | nny_1 | ManxPower: http://store.digium.com/productview.php?product_code=1TE122BF |
| 16:58.25 | Qwell | dwelsh: Yes. |
| 16:58.49 | dwelsh | Qwell: Do you know where to get it? |
| 16:59.01 | Qwell | via subversion |
| 16:59.06 | ManxPower | and no, increasing the rxgain has never helped with EC in my experience. But we switched to a carrier class EC hardware system a couple of years ago and we never needed to worry about it again. |
| 16:59.08 | Qwell | http://svn.digium.com/svn/asterisk-gui/trunk/ |
| 16:59.10 | dwelsh | Qwell: ok thanks |
| 17:00.15 | *** join/#asterisk ChicagoBud (n=Bud@38.104.180.122) |
| 17:01.12 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:06.14 | *** join/#asterisk XnOSX (n=XnOSX@142.156.15.62.static.jazztel.es) |
| 17:09.11 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 17:11.38 | dwelsh | Qwell: I'm having some trouble checking out the AsteriskGUI code. Is the command "svn co http://svn.digium.com/asterisk-gui/trunk"? |
| 17:11.48 | Qwell | should be |
| 17:11.50 | uski | any idea why a 3-digit extension would work and a 10-digit one wouldn't ? (from the context corresponding to a FXS port of a SPA3000) |
| 17:11.54 | Qwell | hold that thought |
| 17:12.14 | Qwell | mmm, yes |
| 17:12.27 | Qwell | oh, no |
| 17:12.31 | Qwell | svn/asterisk-gui/trunk/ |
| 17:12.35 | Qwell | you forgot the svn/ |
| 17:12.44 | uski | the dialplan of my SPA3000 is (7xxS0|0xxxxxxxxxS0|00x.|1[578]S0<:@gw0>|11[259]S0<:@gw0>) |
| 17:12.51 | dwelsh | Ah. It's working now. Thanks again |
| 17:19.31 | *** join/#asterisk JenniferAkemi (n=akemi@conference/cluecon/x-925c835e826954fc) |
| 17:21.58 | *** join/#asterisk petr7 (n=petr7@uidzr47156.sattnet.cz) |
| 17:22.40 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 17:23.58 | petr7 | Hello, I've got a problem with zaptel card AEX2400. Phone silent after an IRQ Miss. Is there any solution? |
| 17:26.22 | Qwell | petr7: I'd recommend calling support |
| 17:26.43 | Qwell | of course, make sure you've got the latest zaptel, etc |
| 17:27.51 | *** join/#asterisk teknoprep (n=cmr@unaffiliated/teknoprep) |
| 17:28.39 | petr7 | hmm, of course, I have lastest verions, but I'm trying to ask here, because of lack of informantion to this problem. |
| 17:28.50 | teknoprep | anyone else here using g729 with Bandwidth.com having one-way audio issues... its very random.. i am not using NAT at all on my box... it is multi-homed tho.. one port is behind the firewall on the DMZ with a WAN IP.. the other port is a LAN ip'd port |
| 17:29.03 | teknoprep | is multi-homing an asterisk box bad idea ? |
| 17:29.26 | Morrocco | Hi Guys, I would like to register a SIP service called Voiceline from net2phone with my Asterisk box, but I dont know how to do it, they dont give you any instructions so I got some sniffed information from the registration but Im not good with asterisk to test it, can some one help me? |
| 17:30.46 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 17:31.47 | *** join/#asterisk Deluzion (i=infamous@scarred.org) |
| 17:32.29 | Deluzion | I have a problem with our asterisk configuration. In particular the voicemail setup *comedian mail*....in /etc/asterisk/voicemail.conf we generally have a sub conf since the phone system is in place for 2 companies |
| 17:33.12 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:33.21 | *** join/#asterisk kand (n=kanderso@139.148.188.72.cfl.res.rr.com) |
| 17:34.43 | Deluzion | every time the phone system reloads, it merges the included sub conf into the main voicemail.conf and messes things up....have to delete the merged info and replace the include line....any idea on how to stop it from doing that? |
| 17:35.30 | Strom_M | you can't do #include in voicemail.conf |
| 17:35.40 | Strom_M | in fact, IIRC, the sample file tells you explicitly not to do that |
| 17:37.51 | Qwell | Strom_M: that it does |
| 17:38.27 | [TK]D-Fender | Doctor, it hurts when I riase my arm like this... |
| 17:39.20 | [TK]D-Fender | Morrocco: http://www.voip-info.org/wiki/index.php?page_id=2008 |
| 17:39.26 | *** join/#asterisk Linker3000L (n=chatzill@78.32.25.201) |
| 17:40.47 | Morrocco | Hi D-Fender, thats very old information it does not work, I would like to debug the new information that I have |
| 17:41.28 | *** join/#asterisk hsv-al (n=hsval@66.0.46.210) |
| 17:41.31 | hsv-al | hello all |
| 17:42.41 | [TK]D-Fender | Morrocco: Enable SIP DEBUG at * CLI and pastebin what it gives. Also include your sip.conf masking ONLY passwords |
| 17:43.48 | uski | i have a problem with my SPA3000; i have defined the following extension : 0xxxxxxxxx,1,Playback(transfer). When I take the phone on the FXS and I call 0123456789, the ATA sends "INVITE sip:0123456789@asteriskip SIP/2.0", asterisk responds with "SIP/2.0 407 Proxy Authentication Required", the ATA sends "ACK sip:0123456789@asteriskip" and "INVITE sip:0123456789@asteriskip SIP/2.0" again, and then asterisk answers "SIP/2.0 404 Not Found". |
| 17:43.48 | uski | .. wtf? |
| 17:43.52 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 17:44.11 | [TK]D-Fender | ~pb |
| 17:44.11 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 17:44.25 | uski | [TK]D-Fender, it's not long enough for a pastebin is it ? just one line |
| 17:44.34 | uski | it's not like i'm pasting 30 lines of code |
| 17:44.41 | [TK]D-Fender | uski: pastebin the ENTIRE call. |
| 17:44.44 | ManxPower | uski: the extension was not found in the specified context |
| 17:44.57 | styelz | _ |
| 17:45.01 | [TK]D-Fender | uski: and pastebin your dialplan |
| 17:45.07 | uski | 1s |
| 17:45.19 | *** join/#asterisk CanWood (n=chatzill@24.108.64.80) |
| 17:48.02 | *** join/#asterisk ta^3 (n=tacvbo@conference/cluecon/x-8665ade3104c22f9) |
| 17:48.28 | hsv-al | lulz @ comcast for using bind 8 still |
| 17:48.35 | Qwell | 4.2.2.2 |
| 17:48.36 | hsv-al | millions of customers vulnerable to dns poisoning/rogue redirection |
| 17:49.27 | ManxPower | you say it like you are suprized |
| 17:49.31 | hsv-al | qwell im extremely bored today, im sending mass emails to network departments at random universities stating I'm students there, asking if their system/providers are patched against the dns issue. |
| 17:49.45 | hsv-al | 3 replies so far, all saying they dindt know about it, 1 including harvard :) |
| 17:49.50 | Qwell | gj |
| 17:50.10 | ManxPower | hsv-al: So you are the disturbance in the Force I felt earlier. |
| 17:50.15 | hsv-al | This is peculiar how this bloomed in the last few days. |
| 17:50.44 | uski | [TK]D-Fender, here you are: http://rafb.net/p/KfXOxK12.html |
| 17:51.09 | uski | from the FXS of the SPA3000, i can call 770 and it works |
| 17:51.14 | uski | but 0xxxxxxxxx doesn't |
| 17:51.39 | [TK]D-Fender | uski: SIP DEBUG from * CLI |
| 17:51.45 | uski | ok |
| 17:53.53 | kand | I may be mistaken but in regards to uski's issue shouldnt patterns start with _ |
| 17:53.59 | styelz | yes |
| 17:54.21 | uski | ... that should be it |
| 17:54.23 | uski | thanks |
| 17:54.35 | [TK]D-Fender | yup |
| 17:54.46 | [TK]D-Fender | I jsut wanted the evidence first |
| 17:54.57 | uski | well at least i learnt how to provide useful debug data |
| 17:55.00 | styelz | perfectionist |
| 17:55.05 | styelz | ;) |
| 17:56.07 | *** join/#asterisk JenniferAkemi- (n=akemi@conference/cluecon/x-e79facb98c3c48b0) |
| 17:56.31 | [TK]D-Fender | uski: Yet still manage to have not actually followed through ;) |
| 17:59.09 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 17:59.56 | keith4 | using an itsp (SIP), what would be the way to add a PSTN line to a queue? |
| 18:02.40 | *** join/#asterisk fedya (n=fedya@rrcs-71-43-222-2.se.biz.rr.com) |
| 18:05.20 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
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| 18:07.06 | [TK]D-Fender | keith4 : ... pardon? |
| 18:08.04 | keith4 | yah... that made more sense in my head, sorry |
| 18:08.20 | keith4 | I have a queue with a bunch of sip user assigned to it |
| 18:08.57 | keith4 | i would also like to assign someone's POTS phone number to it, using our ITSP |
| 18:09.14 | keith4 | if possible |
| 18:09.41 | seanbright | member => Zap/g1/2222222222 |
| 18:09.45 | seanbright | ? |
| 18:09.47 | lesouvage | keith4: you want an external number to be an agent of a queue? |
| 18:09.53 | keith4 | lesouvage: yes |
| 18:10.00 | keith4 | seanbright: no Zap. SIP ITSP |
| 18:10.03 | seanbright | ohhh |
| 18:11.55 | *** join/#asterisk JenniferAkemi (n=akemi@conference/cluecon/x-2aebe14e77b940fc) |
| 18:14.37 | *** join/#asterisk dkwiebe (n=darren@h66-112-187-16.mcsnet.ca) |
| 18:14.50 | *** part/#asterisk korihor (n=korihor@190.78.32.60) |
| 18:15.29 | kand | keith4: You could use local as in member => Local/4079999999@oubound_context |
| 18:15.58 | keith4 | ooh, good idea |
| 18:16.47 | kand | np |
| 18:17.40 | *** join/#asterisk bbryant (n=brett@adsl-12-51-85.chs.bellsouth.net) |
| 18:19.59 | keith4 | kand: sweet, that works! |
| 18:20.07 | keith4 | hearts asterisk |
| 18:20.18 | kand | I have used that trick a few times. dido |
| 18:21.55 | dwelsh | I'm trying to set up database loggin for cdr using odbc. I get the error "ERROR[9347] cdr_odbc.c: cdr_odbc: Unable to connect to datasource: asterisk-connector" in /var/log/asterisk/messages. I know I can use the dsn because I can run the command "echo "select * from cdr;" | isql asterisk-connector" when logged in as "asterisk" |
| 18:23.41 | dwelsh | This is the ouput of "cdr status": [root@asterisk asterisk-1.4.21.2]# asterisk -rx "cdr status" |
| 18:23.41 | dwelsh | CDR logging: enabled |
| 18:23.41 | dwelsh | CDR mode: simple |
| 18:23.41 | dwelsh | CDR output unanswered calls: yes |
| 18:23.41 | dwelsh | CDR registered backend: ODBC |
| 18:23.42 | dwelsh | CDR registered backend: cdr_manager |
| 18:23.47 | dwelsh | CDR registered backend: cdr-custom |
| 18:23.50 | dwelsh | CDR registered backend: csv |
| 18:25.26 | *** join/#asterisk unlord (n=nathan@nextmeal.org) |
| 18:25.28 | unlord | hi |
| 18:25.34 | unlord | is there any good open source call accounting sfotware? |
| 18:26.10 | unlord | I have been tasked with building a call center, and I would like to find something that will let users keep track of who's called and if there are any open issues |
| 18:26.16 | unlord | sugarcrm looks like the best solution so far |
| 18:26.19 | heedly | What source file handles the call file behaviour? |
| 18:27.50 | keith4 | dwelsh: pastebin |
| 18:29.04 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 18:29.12 | dwelsh | keith4: what does that mean? |
| 18:29.27 | lesouvage | ~pastebin |
| 18:29.28 | jbot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 18:30.14 | *** join/#asterisk tacvbo (n=tacvbo@conference/cluecon/x-1a616ca470b364de) |
| 18:31.21 | *** join/#asterisk macros73 (n=cs@dsl093-063-232.pit1.dsl.speakeasy.net) |
| 18:31.33 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 18:34.28 | rbd | I'm trying to issue an AGI RECORD FILE command. I issue "record file /var/lib/ccd/recs_processing/lab1-int-011-ac/temp/677290_1218047497.607368 wav # 18000000 BEEP s=10" and the asterisk console shows (with agi debug on): "AGI Rx << temp buffer r - errno No such file or directory", followed by a very large amount of "AGI Rx << temp buffer r - errno Resource temporarily unavailable" messages. I've double checked and that directory exi |
| 18:34.30 | kand | unlord: for a call center take a look at vicidial, I now it has intergration with VTiger (a fork of sugarCRM). |
| 18:35.20 | kand | *now = know |
| 18:36.35 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
| 18:37.06 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 18:37.09 | uski | i have a weird problem with my spa3000; the dialplan has 06xxxxxxxxS0<:@gw0> in it, i.e. it will route all calls with 06 at the beginning and 8 numbers after that to the PSTN. When I call such a number, the call is routed through the PSTN but the DTMF tones the SPA3000 sends to the PSTN does not correspond to the number dialed (i listened and it's like it always sends the same tones) |
| 18:37.37 | *** join/#asterisk macros73 (n=cs@dsl093-063-232.pit1.dsl.speakeasy.net) |
| 18:37.41 | minthome | i'm looking to setup an informational extension only... |
| 18:37.54 | minthome | would i have to write my own recording menu for that? |
| 18:38.03 | uski | minthome, you could use Playback to play a sound you have recorded |
| 18:38.26 | minthome | yeah, i got that, but i'm not the one recording it... |
| 18:38.38 | uski | then have someone record it, then convert it to gsm format using sox on linux |
| 18:38.40 | minthome | i'd like to have a VoiceMail() app run, but not actually record anything |
| 18:38.46 | uski | oh ok |
| 18:38.59 | unlord | kand: does vtiger support call times with finer granualirty than 15 minutes? |
| 18:39.00 | uski | i know there is an asterisk function to record something |
| 18:39.08 | CanWood | MixMonitor |
| 18:39.10 | minthome | so the customer can call in and change the recording anytime |
| 18:39.21 | CanWood | oops, popped in half way through, sorry |
| 18:39.27 | unlord | I'm setting up a call center and I don't really mind if my users have to enter all this stuff manually, but it will be a nightmare if they can't keep track of who called who |
| 18:39.29 | codefreeze-lap | dwelsh: The message you see, "Unable to connect to datasource:" is in the cdr_odbc.c file, and is the result of odbc_init returning a val < 0; this can happen for more than one reason, the best way to find out is to run asterisk with verbose of 11 or more. |
| 18:40.10 | codefreeze-lap | dwelsh: core set verbose 40 |
| 18:40.40 | kand | unlord: I am not sure I understand your question... do you mean in tracking calls? |
| 18:41.39 | keith4 | unlord: asterisk can put call records in a database for you, and then you can do whatever you want with 'em |
| 18:41.41 | unlord | yes |
| 18:41.43 | *** join/#asterisk jamuse (n=josh@tamar.homelinux.com) |
| 18:42.06 | unlord | keith4: yah, that would help, but what I'm looking for is a way for the people on the call to manage the issues that come out of the call |
| 18:42.11 | unlord | so sugarcrm looks good for that |
| 18:42.17 | jamuse | Can someone explain the following error message: Rejected connect attempt from xxx.xxx.xxx.xxx, who was trying to reach 's@' |
| 18:42.21 | unlord | but I don't know what happens if your call is less than 15 minutes |
| 18:42.29 | jamuse | what is 's@' ? |
| 18:47.02 | kand | unlord: I dont know about the call times but why not fire up a VM machine and check it out: http://ftp.vicidialnow.org/pub/VicidialNOW/1.1/ISO/ |
| 18:47.18 | *** join/#asterisk academy (n=adam@unaffilated/academy) |
| 18:47.22 | *** join/#asterisk Peaceful (n=Peaceful@70.102.57.178) |
| 18:48.14 | academy | I've been given two servers to maintain. They're meant to be redundant - all that actually happens is someone manually changes a low ttl CNAME to point to one or the other. I've been asked to install Asterisk on each. Is there any way to put identical Asterisk instances with both inbound and outbound SIP on both servers so that it will actually work? The main issue would be inbound SIP I think - because there's no failover, both servers would need |
| 18:49.15 | keith4 | that's a sad excuse for a H-A setup |
| 18:50.10 | keith4 | read the H-A asterisk page in the wiki: http://www.voip-info.org/wiki/view/Asterisk+High+Availability+Solutions |
| 18:50.47 | academy | keith4: I agree completely. I'm sort of backed into a corner in that it has to be done this way. |
| 18:52.36 | heedly | minthome: the Record() page has an example of how to record several things at once. |
| 18:52.43 | dwelsh | Here is the error log when I started asterisk with more "-v"s: http://pastebin.com/m1ab79453 |
| 18:52.58 | academy | keith4: i.e. the servers are already in place and they do things like this. |
| 18:53.53 | keith4 | dwelsh: it tells you right there what the error is. FATAL: IDENT authentication failed for user "asterisk" |
| 18:54.04 | keith4 | sounds like MySQL |
| 18:54.46 | minthome | heedly, yeah, that's basically what I was going to write, but with confirmations on keeping the recording after playback etc... |
| 18:54.55 | minthome | just wondering if there was already something in place for that kinda deal |
| 18:54.56 | jamuse | I setup ipkall to talk directly to my asterisk box via sip, when I answer an incoming call I cant hear the caller. Any suggestions on how to debug that? |
| 18:55.22 | keith4 | jamuse: NAT? |
| 18:55.40 | jamuse | if it was a NAT problem wouldnt the phone not ring? |
| 18:55.47 | dwelsh | keith4: Yeah I saw that too, but I didn't know what to do. I figured it out. I had to remove the default line from /var/lib/pgsql/data/pg_hba.conf (a line with ident in it) |
| 18:55.56 | heedly | minthome: ya.. the wiki gives an example. |
| 18:56.14 | heedly | it moves them to sounds/local, you could change it for your own purposes. |
| 18:56.19 | keith4 | dwelsh: yes. you have to configure postgres *correctly* |
| 18:56.45 | minthome | how's festival doing these days anyway? |
| 18:56.48 | minthome | i never installed it |
| 18:56.55 | lesouvage | academy: a business called ranchenetworks has a solution that might fit your needs. |
| 18:57.31 | *** join/#asterisk Yourname` (i=Yourname@unaffiliated/yourname/x-837320) |
| 18:58.11 | lesouvage | academy: see http://blog.tmcnet.com/blog/tom-keating/asterisk/asterisk-and-ranch-networks-make-asterisk-secure-and-more-scalable.asp |
| 18:58.27 | heedly | it is OK. |
| 18:58.45 | heedly | I can't get the app to work through NAT'ed phones. |
| 18:58.45 | dwelsh | I thought I only had to add those two lines (host all 127.0.0.1/32 asterisk md5 and local all asterisk trust), but I had to remove the line "local all all ident sameuser" from the top |
| 18:58.51 | heedly | but the AGI perl script includes works fine. |
| 19:01.06 | outtolunc | hmm http://www.ranchnetworks.com/ |
| 19:01.22 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:02.00 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 19:03.10 | lesouvage | outolunc: oops, doens't look ok. |
| 19:03.26 | lesouvage | doens't=doesn't |
| 19:03.39 | *** join/#asterisk k-man_ (n=jason@unaffiliated/k-man) |
| 19:04.43 | outtolunc | more like ranchNOTworks.com <G> |
| 19:05.04 | outtolunc | falls off his chair |
| 19:05.42 | jamuse | if a call is routed correctly, i.e. the phone rings, is that an indication that NAT is not a problem? |
| 19:06.00 | ManxPower | jamuse: not really |
| 19:06.07 | *** join/#asterisk JenniferAkemi- (n=akemi@conference/cluecon/x-c628b5cda8538835) |
| 19:07.15 | ManxPower | jamuse: it might indicate that SIP signalling is working thru the NAT, it means nothing about RTP audio |
| 19:07.19 | seanbright | is there a magical way to maintain queue statistics across reloads and such? |
| 19:07.24 | *** join/#asterisk ta^3 (n=tacvbo@conference/cluecon/x-cc41af480f2b23e0) |
| 19:07.58 | putnopvut | seanbright: queues are flushed from memory on reloads, so no. |
| 19:08.02 | jamuse | hmm exacly my problem |
| 19:08.16 | putnopvut | You could maintain statistics yourself in the astdb, but there's no way to then apply them to the newly loaded queue after a reload. |
| 19:08.42 | seanbright | putnopvut: queues are dropped and recreated on reload? |
| 19:08.51 | putnopvut | seanbright: yes |
| 19:08.54 | seanbright | yikes |
| 19:08.55 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 19:09.02 | seanbright | ok, i'll try to code something up |
| 19:09.03 | ManxPower | jamuse: classic issue. Is Asterisk behind NAT? |
| 19:09.23 | seanbright | putnopvut: you converted to astobj2 in trunk, yes? |
| 19:09.28 | putnopvut | seanbright: yes |
| 19:09.40 | seanbright | and in 1.4? /me prays |
| 19:09.56 | ManxPower | trunk != 1.4 |
| 19:10.08 | seanbright | ManxPower: yes, thank you. |
| 19:10.16 | jamuse | ManxPower: yes, so I'll switch back to IAX. I'm trying to get ipkall to route my DID number to my iaxy |
| 19:10.17 | seanbright | thus the "and" |
| 19:10.32 | ManxPower | jamuse: you sure give up easy for something that is so simple to fix. |
| 19:10.35 | putnopvut | in 1.4, there's no astobj2 usage for queues, but reloads don't just immediately dump callers from queues though. |
| 19:10.43 | jamuse | ManxPower: so how do I fix it? |
| 19:11.04 | ManxPower | jamuse: Did you follow the SIPNAT instructions? |
| 19:11.06 | ManxPower | ~sipnat |
| 19:11.07 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 19:11.13 | seanbright | putnopvut: i'm just looking for a way to maintain the member stats across reload. i'll just monitor the queue log and be done with it. |
| 19:11.15 | jamuse | ManxPower: should there a difference in sound quality between SIP and IAX? |
| 19:11.21 | ManxPower | jamuse: no. |
| 19:11.41 | putnopvut | seanbright: yeah, you could do that or use a QueueStatus manager command right before the reload to get them at that time. |
| 19:11.52 | jamuse | thanks |
| 19:11.56 | seanbright | putnopvut: ah. good show. |
| 19:12.44 | putnopvut | heh, interesting. The QueueStatus manager action is unregistered twice when app_queue is unloaded. That seem wrong :) |
| 19:12.53 | putnopvut | s/seem/seems/ |
| 19:15.01 | *** join/#asterisk ZenBSDi (n=zenbsdi@unaffiliated/ZenBSDi) |
| 19:16.56 | dwelsh | Some one sent me a link to a page on voip-info.org that had links to third-pary cdr report programs. I can't find it now. Does anyone know where it is or know of a good program to use? |
| 19:17.20 | *** join/#asterisk stencil (n=stencil@206-248-163-143.dsl.teksavvy.com) |
| 19:17.42 | *** join/#asterisk shmaltz (n=chatzill@mail.dmaven.com) |
| 19:17.52 | shmaltz | anyone heard of nufone lately? |
| 19:18.46 | _ShrikE | lately? |
| 19:19.00 | jamuse | Manxpower: The solution at http://www.aocomputing.net/?p=3 did not seem to help. Would using IAX avoid the audio problem I'm having bc of NAT alltogether? |
| 19:19.07 | shmaltz | _ShrikeE ??? |
| 19:19.29 | lowtek | Will exten => _123XX,n+101,SomeFunction actually calculate the line + 101 or do I need to hardcode it? |
| 19:26.07 | *** join/#asterisk n3hxs (n=HAMming@151.196.87.132) |
| 19:27.43 | [TK]D-Fender | lowtek: No, it will not calculate, and you should not be using priority jumping. That was outmoded as of 1.2 |
| 19:27.47 | ManxPower | lowtek: that was removed in 1.4 |
| 19:28.06 | [TK]D-Fender | jamuse: PASTEBIN the SIP DEBUG from * CLI for your failed attempt |
| 19:28.07 | ManxPower | jamuse: no. |
| 19:28.29 | ManxPower | jamuse: it might be easier to set up the port forwarding with IAX2 (you only need one port). |
| 19:28.44 | lowtek | [TK]D-Fender: Labels? |
| 19:29.08 | [TK]D-Fender | lowtek: look at the applications you're using for the instructions. |
| 19:29.19 | ManxPower | lowtek: perhaps you should read all the upgrade files in your asterisk source code? |
| 19:29.56 | lowtek | Thanks for the help, and I am reading the instructions, in 1.4 "show application voicemail" says use option 'j' to 'jump to n+101' in case of an error ... |
| 19:30.30 | ManxPower | lowtek: that is for people that cannot or will not convert to the new 1.2+ way of doing things |
| 19:30.33 | [TK]D-Fender | lowtek: DON'T. Go look at what variables it sets on exit. THOSE you can choose to deal with after. |
| 19:30.35 | lowtek | Ahh. |
| 19:30.48 | shmaltz | anyone heard of nufone lately? |
| 19:30.50 | lowtek | Yea, that's what I'm doing now, checking VMSTATUS for FAILED. |
| 19:30.58 | lowtek | Thanks, guys. |
| 19:31.06 | [TK]D-Fender | lowtek: So clearly no need for "j" |
| 19:31.24 | jamuse | ManxPower: does IAX use significant less bandwidth than SIP? |
| 19:32.00 | ManxPower | jamuse: It might, depending on if you enable trunking and how many calls you have going between the same two servers at the same time. |
| 19:32.14 | outtolunc | still has $10 in my nufone accout.. woohoow |
| 19:32.43 | ManxPower | PERSONALLY, I have found IAX2 to be much less reliable than SIP. |
| 19:33.15 | ManxPower | However, my experience is not typical. |
| 19:36.13 | Yourname` | Hi. How can I change something like this for callwaiting? http://pastebin.ca/1094070 |
| 19:37.05 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 19:38.31 | *** join/#asterisk gaetronik (n=gaetan@eficaz2.manquehue.net) |
| 19:38.36 | gaetronik | hi |
| 19:38.38 | gaetronik | there |
| 19:38.45 | c4t3l | :D |
| 19:39.22 | gaetronik | <PROTECTED> |
| 19:39.23 | gaetronik | [Aug 6 12:25:21] VERBOSE[16388] logger.c: < Ext: 1 Cause: Requested channel not available (44), class = Network Congestion (resource unavailable) (2) |
| 19:39.57 | gaetronik | and when i want to make a call sometime i've congestion tone |
| 19:40.18 | gaetronik | it this an asterisk problem or a provider one |
| 19:40.57 | gaetronik | and how to trace the different steps of a call in the file |
| 19:41.02 | Yourname` | Basically, Agent101 is in a queue. The calls he gets from the queue are good, and I don't want callwaiting on the calls he gets from the queue. However, through an IVR when someone punches 101.. I want those calls to go to Agent101 with callwaiting. Is that possible? |
| 19:42.01 | jamuse | ManxPower: SIP debug info is at http://pastebin.com/d6ed7f2f6 |
| 19:42.29 | ManxPower | jamuse: don't mask the IPs. That is what I need to see. |
| 19:43.10 | ManxPower | Contact: <sip:123456@xxx.xxx.xxx.xxx> |
| 19:43.49 | ManxPower | I can't see that xxx.xxx.xxx.xxx is a local internal IP or external IP, that would tell me what nat setting needs to be changed. Just for future reference only mask PASSWORDS. |
| 19:44.15 | jamuse | ManxPower xxx.xxx.xxx.xxx is an external ip |
| 19:44.29 | *** join/#asterisk brian (n=brian@unaffiliated/brian) |
| 19:44.30 | jamuse | ManxPower: thanks will do |
| 19:44.35 | brian | What is Asterisk Realtime? |
| 19:44.44 | jamuse | ManxPower there were no internal IPs in the debug |
| 19:44.48 | ManxPower | brian: a way to replace the .conf files with a database. |
| 19:45.00 | brian | What does that have to do with virtualization? |
| 19:45.00 | ManxPower | jamuse: What audio ports did you portforward in your firewall? |
| 19:45.08 | ManxPower | brian: nothing whatsoever |
| 19:46.01 | jamuse | ManxPower: which audio ports need to be portforwarded? |
| 19:46.56 | ManxPower | jamuse: http://www.aocomputing.net/?p=3 tells you what the default is as well as how to change it. |
| 19:47.11 | jamuse | ManxPower: I used to have ipkall -> fwd -> my asterisk which worked fine. The only problem was that fwd's iax was not stable. I'd just like to bypass fwd now |
| 19:47.43 | *** join/#asterisk prg3 (n=prg3@playground.cein.ualberta.ca) |
| 19:48.27 | ManxPower | I normally require dinner, drinks before holding someone's hand, but I'll make an exception in this case. Port forward ports 10,000-20,000 UDP in your firewall to the internal IP of Asterisk. Also port forward 5060 UDP. |
| 19:48.32 | *** join/#asterisk CanWood (n=chatzill@24.108.64.80) |
| 19:48.59 | ManxPower | Or I could quote to document: "Our Asterisk server will also have to have ports 5060 (UDP), and the port range specified in ârtp.confâ (typically 10000-20000 UDP) forwarded to it. NEITHER of the remote phones connecting to our server need any forwarding." |
| 19:49.08 | jamuse | ManxPower: I got that thanks |
| 19:49.59 | ManxPower | jamuse: based on a quick reading of your pastebin, it looks like your asterisk side is set up correctly, now you just have to fix your router. |
| 19:50.24 | jamuse | ManxPower I dont have control over the firewall that does the port forwarding, so I guess its back to the IAX solution. thanks for taking the time to read my pastebin |
| 19:51.02 | ManxPower | jamuse: if you don't control the firewall, you cannot use Asterisk with SIP. |
| 19:51.16 | ManxPower | And if you have any IAX2 clients, you would need control of the firewall. |
| 19:51.16 | outtolunc | notes: without STUN/ICE |
| 19:51.43 | ManxPower | outtolunc: the firewall still needs to allow the packets thru |
| 19:52.06 | outtolunc | yeah, but you could tunnel them as there *has* to be some ports open <G> |
| 19:52.41 | ManxPower | outtolunc: and I could chop down trees and build a log cabin -- doesn't mean it's a good idea. |
| 19:52.46 | outtolunc | but i agreee wholeheartedly.. fix the router |
| 19:53.15 | outtolunc | actually thinks building a log cabin (in these times) as a good idea |
| 19:53.27 | outtolunc | as/is |
| 19:53.31 | ManxPower | outtolunc: Only if you are crazy. |
| 19:53.56 | outtolunc | hides his old fashion typewriter |
| 19:54.05 | ManxPower | outtolunc: I spent a weekend with eco hippie freaks. |
| 19:54.11 | jamuse | ManxPower: like I said I had the firewall setup to allow ipkall->fwd->my asterisk server, so now I just need to cut out the fwd step. When I configure ipkall to forward calls using IAX I get http://pastebin.com/dfdfbd7b, I'm not sure what that means though |
| 19:54.12 | n3hxs | hides his quills |
| 19:54.52 | ManxPower | jamuse: I cannot help you further. |
| 19:55.06 | jamuse | ManxPower: thanks for your time anyway |
| 19:55.24 | Deluzion | Strom_M, Qwell: sorry had to take off back to work. I understand it probably shouldn't be done that way ( #include statement in voicemail.conf ), though it does work doing so with 1.4.17 which is what the company is using..I'm inheriting the system and trying to get a feel for it....how would you suggest going about having 2 different areas for the config for voicemail? |
| 19:56.01 | ManxPower | Deluzion: #include is like taking your editor and inserting the first file into the 2nd file. |
| 19:56.04 | Deluzion | we're piggybacking off of the asterisk server for our daughter company at the moment and trying to keep the setup separated for easier migration in a month or two when we migrate to our own dedicated asterisk server |
| 19:56.42 | ManxPower | Deluzion: since you left you'll have to find someone to help you from scratch again. |
| 19:56.45 | Deluzion | ManxPower: right I understand it's like an injection e.g cat #include >> voicemail.conf but with a single line....which explains why when the system does a reload it does just that |
| 19:57.23 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 19:57.41 | Deluzion | but trying to figure out a way to keep /etc/asterisk/voicemail.conf with the daughter company's setup...whilst keeping ours in a separate file for easier migration |
| 19:57.51 | ManxPower | I really can't help you with that. |
| 19:57.56 | *** join/#asterisk chandoo (n=chandra@ool-4353bb46.dyn.optonline.net) |
| 19:58.57 | Deluzion | Does anyone happen to have a copy of the sample file for asterisk's voicemail.conf? I didn't see it on a quick search of voip-info and someone referenced it stating not to use #include's in it...curious to look it over since previous admin removed it from our installation |
| 19:59.31 | ManxPower | Deluzion: It's included in the Asterisk source. |
| 20:00.09 | ManxPower | Oddly, all the sample config files, as well as the official documentation is also in the Asterisk source. |
| 20:00.15 | Deluzion | yeah figured that, guess can just download it local and extract the archive, thanks |
| 20:00.21 | ManxPower | It's a pretty cool thing to have around. *hint* |
| 20:00.41 | Deluzion | yeah I know, twas just looking at this particular bug before delving deep, going to grab the source now |
| 20:01.57 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:02.01 | prg3 | I'm having some troubles with a Polycom SoundpointIP 600.. I'm not sure where the problem is, either asterisk/sip.conf, or the polycom config files.. does anyone have a working pair of both of those that I can take a look at? or a simple intro to how to writeup the polycom config files? (I couldn't get anything from voip-info that helped) |
| 20:02.24 | ManxPower | http://www.fnords.org/~eric/polycom-config-examples/ |
| 20:02.31 | *** join/#asterisk ta^3 (n=tacvbo@conference/cluecon/x-93ebd25ca6f374a4) |
| 20:02.42 | *** part/#asterisk roxlu (n=Roxlu@90-145-42-196.wxdsl.nl) |
| 20:02.55 | prg3 | ManxPower: Wow, that was quick :) Thanks! |
| 20:05.08 | seanbright | they call him the minute man... |
| 20:05.17 | seanbright | probably for a different reason though |
| 20:05.23 | seanbright | ba dum dum *ching* |
| 20:05.28 | prg3 | seanbright: thanks.. really.. that was helpful.. now I've got that image.. |
| 20:06.12 | seanbright | prg3: helpful? that brings my 'you are helpful' count up to... 1! |
| 20:07.44 | *** join/#asterisk grEvenX (n=even@193.71.255.75) |
| 20:14.17 | jameswf-home | weeeeeeeeeeee//// |
| 20:15.09 | *** join/#asterisk L84supper (n=L84suppe@c-24-13-254-249.hsd1.il.comcast.net) |
| 20:15.15 | n3hxs | weeeeeeeeeeee//// |
| 20:15.15 | n3hxs | [16:01] * atis_work (n=atis_wor@193.238.21 |
| 20:15.53 | n3hxs | oops |
| 20:16.21 | n3hxs | Don't know how that happend. |
| 20:18.54 | *** join/#asterisk murdock_ut (n=chatzill@70.102.148.44) |
| 20:20.58 | *** join/#asterisk JenniferAkemi (n=akemi@conference/cluecon/x-afb898159e26684c) |
| 20:23.33 | [TK]D-Fender | ~whee |
| 20:23.34 | jbot | [~whee] Weeeeeeeee! http://www.albinoblacksheep.com/flash/weeee |
| 20:24.02 | [TK]D-Fender | ok, I'm out for a while |
| 20:25.39 | *** join/#asterisk benneton (n=DELL@89.111.208.110) |
| 20:25.40 | *** part/#asterisk benneton (n=DELL@89.111.208.110) |
| 20:25.43 | *** join/#asterisk benneton (n=DELL@89.111.208.110) |
| 20:25.48 | *** part/#asterisk benneton (n=DELL@89.111.208.110) |
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| 20:40.30 | *** join/#asterisk ShaunWing (n=chatzill@dsl-244-213-50.telkomadsl.co.za) |
| 20:41.44 | ShaunWing | Please help. Trying to run asterisk as non root but when execute service asterisk start get error message cat: /var/run/asterisk.pid: No such file or directory;Automatically restarting Asterisk. |
| 20:43.07 | *** join/#asterisk JenniferAkemi (n=akemi@conference/cluecon/x-dc9f644c0d4ada53) |
| 20:43.16 | lowtek | ShaunWing: modify /etc/asterisk/asterisk.conf to specify /var/run/asterisk/ as your run dir and then create /var/run/asterisk with permissions of the user/group you're running from ... |
| 20:44.24 | ShaunWing | tx 4 the help |
| 20:45.16 | brodiem | ShaunWing: don't forget the other directories, like /var/log/asterisk, /var/spool/asterisk.. |
| 20:45.40 | *** join/#asterisk mvanbaak_ (n=michiel@vanbaak.xs4all.nl) |
| 20:49.03 | ShaunWing | I'm not sure what I'm doing with tehse commands as I've followed the instructions in "AsteriskFutureOf TelephonySecEdit.pdf" |
| 20:50.08 | tzafrir_laptop | ShaunWing, edit asterisk.conf and set varrundir to be /var/run/asterisk |
| 20:50.21 | ShaunWing | ok |
| 20:50.42 | tzafrir_laptop | And then: mkdir /var/run/asterisk; chown asterisk: /var/run/asterisk; chmod 755 /var/run/asterisk |
| 20:56.04 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 20:57.28 | *** join/#asterisk tacvbo (n=tacvbo@conference/cluecon/x-56c20d3436fdf321) |
| 20:58.28 | ShaunWing | at the moment my asterisk.conf has astrundir => /home/shaunw/asterisk-bin/run (with the shaunw being my non root user and installation directory) must I add varrundir=>/var/run/asterisk ? |
| 21:02.32 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:05.50 | *** join/#asterisk SebastianS_ (i=schu@adsl-dyn138.78-98-16.t-com.sk) |
| 21:10.21 | uski | can someone confirm me that i need 2 sections in sip.conf for each port of my ATA ? one for outgoing and one for incoming? |
| 21:10.35 | uski | i've been working on my setup for days and i'm all confused... |
| 21:15.17 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 21:20.09 | ShaunWing | I've tried all this and still the same error |
| 21:20.58 | *** part/#asterisk dkwiebe (n=darren@h66-112-187-16.mcsnet.ca) |
| 21:21.16 | kand | uski: I do not believe you need two sections for each port on your ATA. What type of ATA is it? |
| 21:22.16 | uski | kand: it's a SPA3000 from Linksys (Sipura) |
| 21:23.11 | kand | uski: I have a 2102 in use with only one registration per port. |
| 21:23.23 | uski | from my understanding, an entry in sip.conf is "unidirectionnal", i.e. asterisk will accept INVITEs if the host is marked as "friend" and will send INVITEs if the host is marked as "peer" |
| 21:23.43 | uski | kand: and you are able to send calls to the lines from asterisk, AND receive calls from the line to the asterisk ? |
| 21:26.15 | kand | uski: Basicly if type is friend then both ways, if peer then asterisk to client, if user then client to asterisk. They also change how asterisk authenticates them and other such goodies. |
| 21:26.51 | kand | uski: On my SPA I register as a friend in sip.conf |
| 21:26.52 | uski | kand: thanks. would you mind posting your sip.conf entry for the spa2102 on a pastbin (don't forget to remove the passwords) ? |
| 21:26.55 | ShaunWing | Any idea why when running service zaptel start I get ztcfg not executable |
| 21:26.58 | *** join/#asterisk Deeewayne (n=Deeewayn@conference/asterisk/x-c89979a2c6184e7e) |
| 21:26.58 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 21:27.03 | kand | np, 1s |
| 21:29.14 | kand | uski: http://pastebin.com/d5fe7657d also note that I pasted for both ports on my ATA |
| 21:29.18 | *** part/#asterisk brodiem (n=brodiem@rrcs-24-227-171-186.sw.biz.rr.com) |
| 21:29.24 | uski | yea ok, thx |
| 21:29.29 | kand | np |
| 21:32.09 | kand | ShaunWing: if you are starting it as non root I would first check its permissions ie. ls -la /sbin/ztcfg |
| 21:32.37 | ShaunWing | tx |
| 21:32.59 | ShaunWing | I only compiled ztdummy |
| 21:33.08 | ShaunWing | as no cards installed |
| 21:33.21 | ShaunWing | however its running but get error WARNING[8493]: chan_iax2.c:11111 load_module: Unable to open IAX timing interface: Permission denied |
| 21:33.35 | ShaunWing | [root@messaging run]# lsmod | grep ztdummy |
| 21:33.37 | ShaunWing | ztdummy 38856 0 |
| 21:33.38 | ShaunWing | zaptel 231496 1 ztdummy |
| 21:33.47 | ShaunWing | any ideas? |
| 21:33.58 | kand | ShaunWing: I am lazy so I am running my asteirsk/zaptel as root but I am sure it is permissions |
| 21:34.12 | kand | ShaunWing: have you looked at http://www.voip-info.org/wiki/view/Asterisk+non-root |
| 21:35.07 | uski | kand, when i have only one section in sip.conf for the FXS port of my spa3000, i get strange things happening when i call an internal extension from the phone; if i enter "770", i hear my phone sending 7 7 0 and i hear the tones back as if asterisk was sending me the tones by mistake |
| 21:35.13 | kand | ShaunWing: I think that has all you need to solve you problems. Based on what I am seeing there your /dev/zap probably doesnt have the right permissions |
| 21:35.34 | ShaunWing | tx |
| 21:36.35 | *** join/#asterisk wishes (n=wishes@60.234.20.178) |
| 21:36.47 | wishes | gmorning |
| 21:36.59 | uski | gevening |
| 21:37.04 | kand | uski: intersting... let me test something |
| 21:37.14 | wishes | hey does anyone here know much about queues ? |
| 21:37.38 | uski | kand, i don't have exactly the same sip.conf than you, i removed some parameters such as subscribecontext or callerid, but i don't think it's the problem |
| 21:38.01 | *** join/#asterisk nhuisman_work (n=nhuisman@aeko.IfA.Hawaii.Edu) |
| 21:39.10 | wishes | im trying to get the periodic announce to play when it first drops a user into the queue and then every 30 seconds after that until they get answered |
| 21:39.24 | wishes | but the problem is getting the announce to happen when they first hit the queue |
| 21:39.43 | putnopvut | wishes: you could always play the message to the caller before they actually enter the queue. |
| 21:40.17 | uski | kand, additionally the call does not get through, as if asterisk was "loopbacking" what the ATA sends back to the ATA |
| 21:41.10 | wishes | yeah but that means having a macro for each queue i suppose |
| 21:41.15 | kand | uski: but you can call other numbers correct (as in PSTN lines)? |
| 21:42.13 | uski | kand, well, it just worked now. given all the random problems and issues that i had (that were sometimes solved by restarting asterisk...), im pretty sure that my asterisk version is full of bugs. it's a very old version (1.2.13), patched by debian, and with their recent openssl "mistake" it seems that distribution patches are not always good |
| 21:42.27 | uski | so i think i'll give up using their old and ugly version and compile my own |
| 21:42.54 | uski | it didn't work, i tried again and it didn't work, and then it just worked... it's not something im used to see with linux |
| 21:44.54 | kand | uski: I would say move on over to 1.4.xx (18 is nice but I have 22 running and it seems very stable). But if you would like to debug this some more pb you cli with sip debug when you make a call. |
| 21:45.11 | kand | s/you/your/ |
| 21:45.17 | *** join/#asterisk macros73_ (n=cs@c-67-163-224-69.hsd1.pa.comcast.net) |
| 21:45.17 | ShaunWing | It solved it thanks. Hwo do I get zrdummy to load automatically? |
| 21:45.32 | ShaunWing | Also any idea whats causing this error: ERROR[8728]: res_config_pgsql.c:782 pgsql_reconnect: Postgresql RealTime: Failed to connect database server asterisk on 127.0.0.1. Check debug for more info. |
| 21:45.53 | *** join/#asterisk marv0997 (n=marv0997@200.107.127.10) |
| 21:45.59 | kand | ShaunWing ztdummy should load if zaptel finds no hardware, automaticly |
| 21:46.35 | uski | kand, my problem is that i am not familiar with all this, it's only my second setup and the first time i use an ATA. when i have a problem i never know if it's my fault (either from the way i configured the ATA or asterisk) or if it's a bug somewhere; so i'd just prefer to move to a more recent version which has months/years of bug fixed |
| 21:46.41 | marv0997 | Hello |
| 21:46.56 | uski | (and this is all rather frustrating - im done with ranting :)) |
| 21:47.20 | kand | ShaunWing: you would have to pb your debug |
| 21:47.36 | marv0997 | someone knows about grandstream products that ca give me a hand |
| 21:48.09 | ShaunWing | pb? |
| 21:48.10 | uski | marv0997, describe your problem and you'll see if someone can help :) |
| 21:48.41 | uski | (generally speaking it's not productive on IRC to ask if you can ask something - just go ahead and ask :)) |
| 21:48.47 | kand | uski: Ya I know how you feel however, I would have to say both 1.24 and 1.4.22 are fairly bug free..... |
| 21:49.08 | kand | ~pb |
| 21:49.09 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 21:49.19 | ShaunWing | tx |
| 21:49.23 | kand | np |
| 21:50.30 | ShaunWing | http://pastebin.com/m34ce791f |
| 21:50.52 | marv0997 | Thanks uski |
| 21:51.12 | uski | marv0997, so tell us what is your problem |
| 21:52.08 | marv0997 | HT-503 how to reset it to factory default, the deveci keep power cycling every 6 seconds |
| 21:52.38 | ShaunWing | Postgre error on loading pasted @ http://pastebin.com/m34ce791f Any help appreciated |
| 21:53.03 | kand | ShaunWing: I dont know to much about Postgresql but I am thinking mabey more permissions. Does Postgresql have a .sock file (like mysql)? if so check its permissions.... |
| 21:53.39 | uski | marv0997, have you tried the factory reset button ? |
| 21:54.01 | marv0997 | yes, no joy there |
| 21:54.16 | uski | i suggest trying with another power supply if you have one around |
| 21:54.26 | marv0997 | done that too |
| 21:54.56 | uski | well... then i don't know - did you recently flash its firmware or something? |
| 21:54.59 | uski | how did it start? |
| 21:55.06 | kand | marv0997: That sounds bad, where you using the FXO port and if so was it through a surge protector? |
| 21:55.14 | *** part/#asterisk Deeewayne (n=Deeewayn@conference/asterisk/x-c89979a2c6184e7e) |
| 21:56.20 | *** join/#asterisk Blackvel (n=blackvel@dslb-088-065-067-198.pools.arcor-ip.net) |
| 21:56.35 | marv0997 | yes using fxo, and yes had a surge protector, was changing something in the gui then reboot it, since then I have the problem |
| 21:56.40 | Blackvel | ~centos52bug |
| 21:56.40 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889 |
| 21:57.10 | *** join/#asterisk |dennis| (n=Dennis@200.32.231.18) |
| 21:58.01 | uski | marv0997, i have a stupid question.. how do you know it's power-cycling? sometimes there are LED error codes that may be confusing |
| 22:01.46 | marv0997 | well when you connect the power first time you power on, it turn on LEDs power, wan and lan then the internal relay sounds and after 2 sec's the lights on the wan and lan ports turn off, then it start is normal operation, after 6 seconds, it do all over again. |
| 22:02.35 | kand | marv0997: I dont know but it sounds like your box has a problem. Email support@grandstream.com they have always been very helpfull for me (even if it took a couple of days) |
| 22:03.30 | marv0997 | ok, I already did that, but thanks guys for your help |
| 22:04.04 | kand | marv0997: Have you already tried powering up with nothing else connected (no ethernet, FXO, or FXS)? |
| 22:05.38 | marv0997 | yes, I try that, and also the reset procedure without nothing connected too |
| 22:05.46 | *** join/#asterisk outtolunc (n=me@adsl-76-211-231-133.dsl.pltn13.sbcglobal.net) |
| 22:05.51 | *** join/#asterisk giesen (i=giesen@dirtypackets.net) |
| 22:06.03 | giesen | what's the best sccp channel to use for asterisk 1.4 |
| 22:06.09 | giesen | using it with a cisco 7936 |
| 22:06.26 | kand | marv0997: Dang, sorry that is it for my ideas. good luck. |
| 22:07.45 | Blackvel | still comparing and researching isdn bri cards (at least 2 s0, one TE-mode, one NT-mode) :) comparing zaphfc junghanns/beronet vs chan_capi vs misdn vs sirrix vs sangoma vs digium b410p (saw misdn bri support for this card in asterisk 1.4.21.1). anyone using it? is it working with mISDN at least as good as junghanns bristuff qozap driver? is MISDN working at least? read one some forums that it might crash? e.g some users use b |
| 22:09.17 | heedly | Blackvel: press Enter sooner |
| 22:09.30 | *** join/#asterisk Penggu (n=me@71.188.233.220.exetel.com.au) |
| 22:10.26 | Penggu | hi all. i wanted to simulate deman-dialling between vmware hosts using serial ports => guest os => serial/tcp converter => iaxmodem-running OS => asterisk |
| 22:10.38 | Penggu | i shouldn't have a problem with most of it |
| 22:10.46 | giesen | I'm currently using chan_sccp-b but it appears to be unworkable |
| 22:10.52 | Penggu | just the part between serial/tcp converter and iaxmodem on another host |
| 22:10.53 | giesen | resetting the phone after every call |
| 22:10.57 | *** join/#asterisk stencil (n=stencil@206-248-163-143.dsl.teksavvy.com) |
| 22:11.22 | Penggu | how would i configure the virtual com port for iaxmodem to accept tcp connections? |
| 22:11.23 | kand | Blackvel: I have sucessfully used two sangoma four port PRI cards reliably (in 1.2 a while ago) |
| 22:11.57 | Blackvel | heedly: you are right :) |
| 22:13.02 | Blackvel | kand: costs are comparable to digum b410p (because of 2 bri modules) and b410p would be 4 slots instead of 3 (and 3rd without bri module). so it mainly depends on quality and driver support |
| 22:13.35 | kand | giesen: For what it is worth, I have reliably implement ciscos by flashing them to sip... dont know if you want to go that way but it works. |
| 22:13.41 | Blackvel | i am struggling with all that asterisk / bristuff patching all the time (and now it should be a productive system) handling my company calls over Asterisk IVR |
| 22:14.18 | Blackvel | is dtmf hardware detection required somehow especially for IVR? not all cards seem to support that |
| 22:15.14 | giesen | kand: there's no sip firmware for this particular phone |
| 22:15.24 | giesen | I have about 40 other cisco's here running sip though =) |
| 22:15.54 | kand | Blackvel: I have no experience with a BRI line but I can recommend you go with a digum or sangoma card because (at least in my opion) they seem to be the most stable. With a PRI line I did not need a DTMF detection hardware. |
| 22:16.26 | Blackvel | ah i see |
| 22:16.29 | kand | giesen: *doh* |
| 22:16.53 | Blackvel | i would love going for misdn to support b410p in new asterisk 1.4 releases without having to patch anything |
| 22:17.54 | Blackvel | ... if misdn and chan_misdn is really stable enough for production. i am unsure. on wiki it says misdn is not production ready. is that still true? probably Digium wouldn't sell the card with asterisk built in driver support then? :) |
| 22:19.01 | uski | kand, i did what you told me (one entry in sip.conf for each port of my ATA) and it seems to be working - i'll see (over time) if the bugs were transient problems because of the numerous modifications i've been doing to the configuration or if they are more serious. thanks for your help |
| 22:19.45 | Blackvel | btw. what phones do you prefer in general for a little (1 person) company? now i forward my pstn number to some normal analog telephone. want to go more the business/professional way. maybe it's more clever to buy snom business voip with outlook dail and headset support than to buy some new analog or isdn phone? do you all use snom voip telephones for business or is there something better (like cisco) with price 200-300 euros? |
| 22:20.25 | *** join/#asterisk Dovid (n=chatzill@tony09-121-90.inter.net.il) |
| 22:20.35 | kand | uski: np, anytiem |
| 22:20.39 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:20.53 | uski | Blackvel, we had a discussion here 24h ago about siemens gigaset VoIP phones such as the c470 (entry level) or the s675 (more advanced) |
| 22:21.27 | kand | Blackvel: I am sure you are about to start a war.... But I love Polycom |
| 22:21.31 | uski | i'm not sure if these would be appropriate for you, but some relatives have a 3 persons company with 4 phones and the s675 would be just perfect for them |
| 22:21.38 | Blackvel | missed that :( |
| 22:21.39 | giesen | I'm partial to the cisco's, but then I'm made of money. |
| 22:21.55 | uski | someone looked to have some experience with the s675 and he said they are very stable |
| 22:21.59 | giesen | hugs his cisco 7971 |
| 22:22.03 | kand | giesen: Dont tell me you have the color touch screen ..... |
| 22:22.11 | giesen | aye. |
| 22:22.16 | uski | from what i've seen on the siemens faq, it looks like one s675 base can support 1 pstn call and 2 voip calls at the same time, i.e. 3 calls max |
| 22:22.24 | bijit | how can I configure two linksys switches to be in the same vlan? |
| 22:22.37 | giesen | kand: it's orgasmic |
| 22:22.42 | kand | giesen: I guess if you are going to go, go all out. The ciscos are nice looking phones tho..... |
| 22:22.46 | Blackvel | i mean it makes perfectly sense to buy business voip phone over isdn/analog phone? :) have to check siemens website |
| 22:23.14 | kand | giesen: lol, I saw/drooled on one once |
| 22:23.17 | Blackvel | i dont add siemens to my customer list. I don't really have to buy one (at least I did twice before) :) |
| 22:23.28 | giesen | I work for a cisco premier partner |
| 22:23.30 | uski | i do have some extensive experience with the analog siemens gigaset phones and i never had any issue with them (but i never bought the cheapest ones) |
| 22:23.33 | giesen | so we get all kinds of goodies cheap |
| 22:23.40 | kand | giesen: ah |
| 22:23.45 | Blackvel | uski: I run siemens analog too |
| 22:24.14 | Blackvel | but sharing private calls and business calls one telephone is going to make me crazy more and more (I am a contractor) |
| 22:24.35 | Blackvel | and since I am going for 100% asterisk ivr system which is up 24x7 |
| 22:25.05 | Blackvel | whats the price of a good business phone with cool display? 200-300 or more? cisco probably 500? |
| 22:25.34 | Blackvel | like headset, outlook integration. it would be cool to be able to send outlook contacts to phone for call number/name display |
| 22:26.08 | Blackvel | snom seems to support. but I have no experience about that. heard a cisco one time ago. it had nice sound |
| 22:26.24 | kand | Blackvel: I have to say the Polycom 650 (not the worlds coolest but very funcitonal), outlook integration is typicaly done with software but check out the free version of x-lite (softphone) |
| 22:26.30 | uski | i did something neat today that you may like; when asterisk receives a call, it reverse-lookups the phone number, and if the name of the caller is found it's added to the caller id display of the phone when it rings :) |
| 22:27.05 | *** join/#asterisk salaud (n=salaud@h-66-166-226-4.sttnwaho.covad.net) |
| 22:27.09 | Blackvel | very nice |
| 22:27.11 | uski | i couldn't find something for my country so i made that up from scratch, but i've seen on voip-info somewhere that there are packages to do that easily if you live in the us |
| 22:27.22 | uski | so, where are you living ? |
| 22:27.29 | Blackvel | does that only work for voip phones or analog too? clip no screening display? |
| 22:27.37 | uski | it works on my analog phones |
| 22:27.41 | Blackvel | wow |
| 22:27.47 | uski | siemens gigaset s1 color |
| 22:28.02 | uski | im using a SPA3000 to connect these to my asterisk server |
| 22:28.17 | uski | i had a hard time configuring the ATA to match my regional settings (tonality, ring settings, caller id, ...) |
| 22:28.21 | Blackvel | good old germany where I couldn't get voip contracting started 2004/2005 :) |
| 22:28.30 | Blackvel | i believe that |
| 22:29.25 | uski | honestly if you want to buy the equipment (i.e. if you don't have it already) then buy some voip phones (either dect phones or "fixed" phones, depending on your usage pattern and preferences) |
| 22:29.34 | uski | i always thought that ATAs were a big hack |
| 22:29.48 | salaud | hey there.... I have a SIP channel that won't hangup... the last message received was "BYE" ... but the channel doesn't hangup... Soft hangup won't work from the console... I need to clear it without restarting... if possible... |
| 22:29.54 | Blackvel | i had softphone problems (with soundcard, on laptop). probably I had had to go for a bluetooth headset but I didn't. real hardware is not too bad for business use (but of course I read that call centers use softphones) |
| 22:30.15 | salaud | anyone know why this issue of not hanging up might happen on SIP.... I've never seen it before. |
| 22:30.20 | salaud | thanks for the help |
| 22:30.40 | uski | Blackvel, yesterday i said "i know call centers use softphones, why don't you use that?" and i instantly got flamed :D |
| 22:30.58 | uski | ..and they were right: it's not very nice to give some crappy hardware/software to people who spend their day on the phone |
| 22:31.11 | uski | ooh, also, the siemens gigaset s675 supports "wideband" quality |
| 22:31.16 | uski | i.e. better sound quality, like skype |
| 22:31.42 | uski | your peers won't notice the difference if you reach them through a standard link, but if you use a voip link then they may |
| 22:31.49 | kand | salaud: I have had that issue before, I call them stuck channels. They seem preaty benign. Typicaly I just wait till an opertune time to restart |
| 22:32.35 | Blackvel | need to check that s675 |
| 22:32.36 | salaud | kand: Unfortunately... this is one client of many for me... I can't baby their system along.... I'd have to be able to detect the condition |
| 22:32.43 | Blackvel | very good |
| 22:33.09 | uski | Blackvel, it's a dect phone but it also has a headset port, so it may be a good compromise between a fixed phone and a dect phone |
| 22:33.34 | uski | im not affiliated with siemens but i thought that phone was good and if i was in the mood of getting a sip phone i'd probably go for this one |
| 22:33.58 | Blackvel | i wrote all down...polycom 650, cisco 7971, s675....probably I have to make the decision: b410p for 505 without phone or junghanns/beronet for ~300 + extra phone. HAHA! :) |
| 22:34.13 | kand | salaud: well....Um lets see as a hack you could script something to check for RTP streams weekly at midnight and restart asterisk if none.... |
| 22:34.17 | uski | but im not sure the s675 has the features you are looking for (outlook integration namely) |
| 22:34.35 | Blackvel | I think I need that badly |
| 22:34.44 | uski | why not periodically issuing "restart when convenient" commands to asterisk ? like daily ? |
| 22:34.47 | salaud | kand: Ah... problem is that they aren't benign in my case.... they make a phone look busy... |
| 22:34.52 | uski | oh no, that will not work |
| 22:35.06 | Blackvel | my database is just too big and it sucks to only have the number in display (without name and company and city) |
| 22:35.17 | kand | salaud: that is a problem... what version of asterisk? |
| 22:35.35 | salaud | perhaps the issue is that this phone doesn't have call-waiting enabled or something dumb like that... 1.4.17 |
| 22:35.51 | Blackvel | uski: headset/bluetooth support is cool too! |
| 22:35.53 | salaud | by "call-waiting" I mean in the FreePBX context... |
| 22:35.58 | uski | Blackvel, you mean... when you're making a call from the phone directory ? my "middle-class" gigaset s1 allows me to specify the name of the contacts alongside with their number |
| 22:36.13 | uski | and the s675 allows for vcard contacts i've read |
| 22:36.20 | uski | i didn't see the bluetooth support |
| 22:36.40 | uski | but it has a 2.5mm standard headset connector |
| 22:36.53 | Blackvel | uski: whats the main reason to go for voip phone if I have a bri card which connects asterisk to an isdn/analog pbx? I probably could go for a isdn/analog telephone too. is there any benefit of voip (of course asterisk runs as server too :) ) |
| 22:37.19 | uski | Blackvel, i honestly don't have the experience needed to answer that question |
| 22:37.34 | uski | im a noob in all this |
| 22:37.35 | kand | salaud: I vouch for 1.4.22 so mabey upgrade will catch this issue. Are you sure the sip-bye is reaching asteirsk? |
| 22:37.43 | salaud | hmmm... I'm thinking that even with FreePBX thinking it is ok to send more calls to the phone... this might not work.. because at some level enough channels might get stuck.... |
| 22:37.45 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 22:38.02 | Blackvel | uski: np. I am driving my guys on ippf crazy too :) |
| 22:38.09 | salaud | kand: yes... I looked at "sip show channel" and last received is BYE... |
| 22:38.21 | uski | salaud, then you have a way to detect the condition |
| 22:38.32 | Blackvel | upppppps |
| 22:38.40 | Blackvel | I have siemens s1 too :) |
| 22:38.44 | uski | ...but im not sure how to fix it |
| 22:38.56 | Blackvel | it has no interface to computer address book |
| 22:39.05 | uski | Blackvel, then it does have callerid and caller name support, and you can store contact names in addition to the numbers, and yea it has no computer interface |
| 22:39.09 | Blackvel | i'll check s675 |
| 22:39.32 | kand | salaud: what about rtptimeout? is it set? |
| 22:39.37 | Blackvel | uski: but where do you do the lookup? in a mysql/asterisk db or something? |
| 22:39.40 | salaud | uski: true... I could detect it that way.... and then force a "stop now"... but... seriously, in all the systems I have done since 2004.. I haven't seen this yet |
| 22:40.16 | salaud | kand: not aware of that setting... is it sip.conf or rtp.conf? |
| 22:40.18 | uski | salaud, i'm not saying it's normal nor clean to do it that way, but it's a temporary fix that should allow you to find a better solution over the long term (and that may just be... wait for a bug fix) |
| 22:40.24 | kand | sip.conf |
| 22:40.38 | salaud | uski: I appreciate that..... |
| 22:40.44 | wishes | ok i fixed my problem |
| 22:40.48 | salaud | kand: checking.... what is sane setting? |
| 22:41.02 | wishes | made some options into a macro and pressy 1 2 3 4 or 5 sent them into the macro where i had more control etc |
| 22:41.04 | Blackvel | wow, snom has cti integration |
| 22:41.06 | uski | Blackvel, im using something called AGI, it's an interface to control asterisk from an external program within a given context |
| 22:41.08 | kand | salaud: from mine- |
| 22:41.08 | kand | rtptimeout=300 |
| 22:41.08 | kand | rtpholdtimeout=300 |
| 22:41.10 | wishes | cheers ears, thanks for all the fish etc :) |
| 22:41.36 | Blackvel | uski: php/java? I used to write fastAGI Java control program for * in 2005 :) |
| 22:42.01 | salaud | kand: verified these are not set.... will set them now... |
| 22:42.03 | uski | Blackvel, in my extensions.conf, when i receive the call, i do exten => 55555,2,AGI(/path/to/cid.agi), and that cid.agi is actually a perl script that asks a free web-based reverse-loopkup service to do the lookup. the script then parses the result, and sends it back to asterisk using SET VARIABLE commands. |
| 22:42.24 | Blackvel | thats cool! |
| 22:42.44 | uski | Blackvel, if you do that don't forget to have timeouts for every blocking operation or it will get the call stuck |
| 22:42.46 | Blackvel | do you use Set(CALLERID(name) and Set(CALLERID(number) commands? |
| 22:42.51 | kand | salaud: They belong in the sip general section and are a seconds till hangup. Although I am not sure they will work since you can not softhangup |
| 22:43.14 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 22:43.18 | uski | Blackvel, sort of; the commands from the AGI are different; there is an example AGI script shipped with asterisk - use it |
| 22:43.38 | salaud | kand: worth a try..... wonder if it is a timer issue? |
| 22:43.47 | Blackvel | I forward the call to my mobile phone over an SIP provider (who supports clip no screening). so CALLERID(name) supports the forwarded number. but I am missing the text on my mobile: "call forward from..." |
| 22:43.53 | kand | salaud: I have long supected my ztdummy |
| 22:44.03 | Blackvel | in IPPF forum I got informed about sip protocol: indication |
| 22:44.08 | Blackvel | do you use that too? |
| 22:44.08 | *** join/#asterisk ReDNeQ (n=ReDNeQ@75.148.217.225) |
| 22:44.39 | uski | Blackvel, I have this in my code: print "SET VARIABLE \"CALLERID(name)\" \"$calleridname\" \"\"\n"; and $calleridname has the caller id name (sanitized) or something else such as "Private" (hidden number) or "Unknown" (if no result was found) |
| 22:45.06 | salaud | I'll I might compile a ztdummy again on this machine... I'm using module-assistant... and I created a package for this kernel on another machine and moved it here... and installed |
| 22:45.17 | salaud | debian... I should say |
| 22:45.53 | kand | salaud: Thats a good idea, I also found the newest version of zaptel help my ztdummy accuracy |
| 22:46.32 | Blackvel | uski: do you manually forward to your mobile too, perhaps? |
| 22:46.46 | salaud | I am TRYING to use debian/ubuntu packaging for stuff... I'm not so much lazy or uneducated about compilation... but, I have a lot of machines to maintain. |
| 22:46.58 | kand | salaud: promissingly since then I have not had any stuck sip channels just IAX (between a box with sangoma and one with ztdummy so...) |
| 22:47.48 | kand | salaud: Ya I hear ya. I only have 24 so it is managable |
| 22:47.59 | uski | Blackvel, i had some callback stuff setup from my previous "asterisk week" lol - what do you want to know? |
| 22:48.14 | salaud | kand: If I keep seeing this problem, I will try and upgrade that part... it sucks really bad since it is the Operator's phone that is getting stuck.. |
| 22:49.07 | salaud | thanks for the hints... I'll try back here in a day or so if the timeouts don't help... THANKS to all! |
| 22:49.12 | kand | salaud: It has been my experience the phone with the most issues (registration, jitter, ect) is always the most important, ie operator or CEO of our clients. Anyway good luck |
| 22:49.18 | Dovid | where is the init script for zaptel stored ? (the one that will modprobe zaptel for me ? |
| 22:49.47 | Blackvel | in etc or in source directory? |
| 22:49.50 | kand | Dovid: after make and make install run make config |
| 22:50.08 | Dovid | Kand: live box. i know i can find it there. dont wana do it now |
| 22:50.09 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 22:50.18 | Dovid | some one else set up the box. wana see what he did |
| 22:50.41 | Blackvel | if its there its in /etc/init.d/zaptel (was for 1.2) |
| 22:50.48 | kand | Dovid: most destros will put it in /etc/rc.d/init.d/zaptel |
| 22:50.49 | Blackvel | for 1.4 i am testing right now :) |
| 22:50.53 | *** join/#asterisk chandoo (n=chandra@ool-4353bb46.dyn.optonline.net) |
| 22:51.24 | Dovid | Blackdevel: not there. guess custom script. goto find it now :( |
| 22:52.59 | Dovid | if u find that it is else where let me know |
| 22:53.24 | Blackvel | hmmm |
| 22:53.38 | Blackvel | have to backup etc asterisk first |
| 22:53.47 | Blackvel | or it will overwrites/rename files |
| 22:53.54 | Dovid | nm |
| 22:53.58 | Dovid | found a test box :) |
| 22:54.36 | Dovid | http://pastebin.ca/1094253 |
| 22:54.52 | zamba | i'm trying to set up a sip trunk towards a non-standard port (5062).. how do i do this? |
| 22:55.05 | zamba | i tried specifying 5062 in the register command, but it still connects to 5060 |
| 22:55.06 | Blackvel | uh, too many manually edited files. I fear of overwrite even on /etc/modprobe and /etc/sysconfig/zaptel :) |
| 22:58.53 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 22:59.50 | drako | there is any way to limiting an amount of channels on a outgoing iax trunk? |
| 23:00.07 | Dovid | zamba: you can set ur default port to 5062 |
| 23:00.19 | zamba | Dovid: can't do that, the default port is still 5062 |
| 23:00.20 | zamba | eh |
| 23:00.22 | Dovid | or set up a peer on port 5062 and then have the register to that port |
| 23:00.24 | zamba | 5060 |
| 23:00.31 | zamba | but i'll try trunking with iax instead |
| 23:00.50 | Dovid | like regisster => foo:bar@peer_name instead of register => foo:bar@IP |
| 23:01.03 | *** join/#asterisk ReDNeQ (n=ReDNeQ@75.148.217.225) |
| 23:02.33 | Dovid | drako: I know there is a way in sip.conf. forgot the paramater. try it in iax.conf |
| 23:04.35 | *** join/#asterisk ReDNeQ (n=ReDNeQ@75.148.217.225) |
| 23:04.59 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 23:04.59 | kand | drako: There is no parameter for IAX (unfortunatly) but in the dial plan there is a handy func 'group' that you can use to track and count channels then conditionaly jump if over a limit |
| 23:06.27 | drako | kand, func group |
| 23:06.38 | Blackvel | wow, polycom 670. they are crazy. 396 euro for a telephone ;) |
| 23:06.39 | jameswf-home | can anyone think of a creative way to netcat a ztmonitor |
| 23:07.07 | Blackvel | thanks for tips. I am coming back. I promise :) g' night |
| 23:07.13 | Dovid | Blackdevel: where r u located ? I know in the US they arent that much. in the UK its a lot |
| 23:07.14 | kand | drako: http://www.voip-info.org/wiki/index.php?page=Asterisk+func+group |
| 23:07.26 | Blackvel | Dovid: germany |
| 23:07.27 | Dovid | Blackdevel: Threat or promies ? ;) |
| 23:07.36 | Dovid | Blackdevel: Get snom. i like em |
| 23:07.46 | Dovid | but their speakers suck |
| 23:08.04 | kand | Check out the Polycom 650 |
| 23:08.15 | Blackvel | right. wrote that down |
| 23:08.27 | kand | Ebay had them regularly :) |
| 23:08.29 | Blackvel | i hate not being able to go to specialist tsore |
| 23:08.34 | Blackvel | store |
| 23:08.50 | kand | Great phone, and really nice from an admin side (provisioning, support, features) |
| 23:09.02 | Blackvel | i had buying electronic stuff online. you can not hit anyone if something is not working as expected :) |
| 23:09.20 | kand | true... |
| 23:09.25 | Blackvel | kand: would you buy voip phone or isdn/analog (if asterisk is connected to a pbx) |
| 23:09.28 | Blackvel | I hate even... |
| 23:10.20 | tzafrir_laptop | jameswf-home, write to a pipe? |
| 23:10.23 | kand | VoIP the whole analog -> digital -> analog thing doesnt make much since and I would be willing to put money on having echo problems.... |
| 23:10.32 | Blackvel | snom sounds nice already. they have m3 too to get me mobile + headset + cti outlook integration. if i can copy my whole business phone directory to the snom directly that is going great |
| 23:11.04 | Blackvel | well germany is digital everywhere |
| 23:11.17 | Blackvel | and sip gateway to pstn doesn'T make it better (for handy forwarding) |
| 23:11.40 | *** part/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 23:11.45 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 23:11.52 | Blackvel | probably I should look for asterisk pbx integration card WITH echo hardware cancel |
| 23:11.56 | jameswf-home | tzafrir_laptop: didnt turn out so hot on pipe or std redirect |
| 23:12.15 | kand | I have not tried snom but I DONT recommend any linksys, grandstream (other than the ATA), or Astraa |
| 23:12.30 | heedly | Blackvel: are you big enough to actually hurt them? |
| 23:12.59 | JT | aastra and linksys are fine |
| 23:13.00 | Dovid | tzafrir_laptop: ztdummy was the issue. its working now. thanks |
| 23:13.00 | Blackvel | i don't have to be big. i can pay one to hurt. i need not to do all stuff myself :) |
| 23:13.06 | JT | grandstream is a pile of crap |
| 23:13.27 | Dovid | ~gradstream |
| 23:13.30 | *** join/#asterisk mvanbaak_ (n=michiel@vanbaak.xs4all.nl) |
| 23:13.32 | heedly | Blackvel: then do you have enough money to hire someone? |
| 23:13.34 | Dovid | ~grandstream |
| 23:13.34 | jbot | from memory, grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 23:13.34 | kand | ATA is not bad, phone is horrible (in my experience) |
| 23:13.35 | Blackvel | low budget endconsumer stuff. i also hate their ATA |
| 23:13.44 | heedly | maybe it would be cheaper to just buy several phones online |
| 23:13.50 | heedly | and return the ones you don't like. |
| 23:13.55 | Blackvel | heedly: depends if I am going to find a new freelancer project in September :) |
| 23:14.10 | tzafrir_laptop | ~gs |
| 23:14.10 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 23:14.13 | heedly | what about November? |
| 23:14.21 | heedly | would that make a difference.. |
| 23:15.02 | Blackvel | if not, probaby I will waste my money otherwise ... |
| 23:15.41 | Blackvel | that means 2 month no money. so I guess yes. |
| 23:16.08 | uski | can someone tell me what this "echo cancellation" is all about? from the name i know what it's supposed to do but so far (with my 2days old setup) i never heard any echo |
| 23:16.09 | Blackvel | oh nice . b410p has echo cancellation |
| 23:16.17 | JT | don't get a b410p |
| 23:16.24 | JT | uses misdn |
| 23:16.30 | uski | well i did, i can hear myself when i speak on the phone, but i never thought it was a problem |
| 23:16.49 | Blackvel | is it true that misdn isn't working as "someone" was looking for? |
| 23:17.11 | Blackvel | i read something about it the last couple of days (on wiki it says: not production ready) |
| 23:17.12 | JT | uski: echo is usually when you hear it coming from the far end, or the far end hears it coming from you |
| 23:17.26 | kand | uski: there is suppose to be a little 'talkback' but when you convert digital to analog there can sometimes be a disturbing echo |
| 23:17.28 | JT | Blackvel: misdn is basically ISDN4LINUX with a new name |
| 23:17.43 | JT | because misdn had such a bad reputation |
| 23:17.45 | Blackvel | jt: i4l was not good |
| 23:17.50 | drako | kand, i don't get it much so not sure if its going to work for what im looking for, i tell, i have 2 providers, first only accept 2 calls at same times, second few more, first is iax trunk. When i get the 2 calls going on the first provider i don't want asterisk to try to get there cuz it cost bw and time i just want it to jump to the other provider before go to the iax trunk |
| 23:17.56 | uski | maybe the SPA3000 has some echo cancellation enabled by default, i should disable it to try and see |
| 23:18.10 | Blackvel | jt: there are not really many options :) |
| 23:18.29 | JT | Blackvel: get a sangoma A500 or a junghanns card if you need a multiport card |
| 23:18.32 | Blackvel | either bristuff (zap), sirrix or sangoma |
| 23:18.36 | JT | generic hfc chipset for single port |
| 23:18.58 | JT | or better yet, get an external BRI to SIP gateway |
| 23:19.22 | Blackvel | 2 bri (2 ports, 4 lines, one NT, one TE) is the cheapest with junghanns. b410p even has 4 slots instead of 2 |
| 23:19.27 | Blackvel | astribank? |
| 23:19.35 | hardwire | anybody in oahu/kapolai? |
| 23:19.41 | kand | drako: I don't think it would cost that much BW or time to be conserned about but on the page I provided there is an example doing exactly what you requested. Let me know what in particular would you like me to explain and I will be more than happy to do so. |
| 23:19.42 | hardwire | I made need an "assist" |
| 23:19.44 | JT | no, not an astribank |
| 23:19.45 | JT | bri to sip |
| 23:20.03 | Blackvel | i had hope to find some solution without all that bristuff asterisk patching. |
| 23:20.10 | JT | Blackvel: there aren't many 2 port bri cards out there, plenty of 4 port though |
| 23:20.17 | Blackvel | so how is b410p working in asterisk 1.4 with misdn? :) |
| 23:20.25 | JT | Blackvel: you don't have to use bristuff anymore, you can just use zaptel |
| 23:20.31 | Blackvel | jt: most have no echo cancellation on them |
| 23:20.40 | JT | the A500 has an EC option |
| 23:20.50 | Blackvel | with 1.6 yes. it's not in 1.4 yet? |
| 23:20.56 | JT | what? |
| 23:20.59 | JT | Hardware EC. |
| 23:21.01 | Blackvel | bri |
| 23:21.03 | Blackvel | i mean |
| 23:21.05 | Blackvel | bri support |
| 23:21.20 | JT | nah it can be done in 1.4 i think, ask tzafrir_laptop |
| 23:21.21 | Blackvel | ahhhhhh |
| 23:21.29 | JT | bristuff is not that bad |
| 23:21.36 | JT | way better than misnd |
| 23:21.37 | Blackvel | you mean sangoma cards |
| 23:21.39 | JT | misdn |
| 23:21.42 | JT | yes |
| 23:21.58 | JT | but if you can afford it, get an external BRI to SIP gateway, will save you a world of pain |
| 23:22.08 | drako | kand, yes well at least it cost load on the server when trying to make 30 calls at same time and 28 of them failing every second :/ |
| 23:22.15 | Blackvel | i like their card design. but its ~200 euros more expensive than junghanns 2 bri (without EC) |
| 23:22.25 | Blackvel | its only for me :) hehe |
| 23:22.36 | Blackvel | just for my little business + pbx integration |
| 23:22.53 | Blackvel | if i had 5-10 employs that would be different |
| 23:22.58 | Blackvel | employees |
| 23:23.02 | JT | depends on local pricing really |
| 23:23.09 | JT | get some quotes |
| 23:23.16 | Blackvel | going for IVR way (accepting pstn calls) |
| 23:23.29 | Blackvel | but if card/asterisk/linux server crashes |
| 23:23.30 | kand | drako: Np, then why not use the group function and dialplan magic to skip the attempt? |
| 23:23.31 | Blackvel | its my money |
| 23:23.46 | JT | is PRI expensive there? |
| 23:23.47 | uski | is there a way to specify a timeout for a SIP call - not the timeout before the thrid party answers or anything, but the timeout before reaching the remote SIP peer; i'd like to setup a failover in case my internet connection is failed, so i want to detect that condition and route the call through the PSTN; i guess i could setup a timeout in the Call command and use the CONGESTION value of dialstatus, but with that there is not way to s |
| 23:23.47 | uski | eparate the timeouts |
| 23:23.47 | Blackvel | because companies will be pissed not being able to reach me at home/work |
| 23:23.57 | drako | kand, im trying |
| 23:24.04 | *** part/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 23:24.06 | JT | uski: in sip.conf, use qualify=yes |
| 23:24.10 | Blackvel | pri doesn't work here for telco |
| 23:24.12 | JT | for the relevant sip peer |
| 23:24.14 | kand | Blackvel: Sorry to interupt but why even use analog equipment, have you looked into a voip provider? |
| 23:24.18 | Blackvel | pri is 24-30 channels? |
| 23:24.24 | zamba | i'm trying to load the chan_iax2 module, and getting the following error: |
| 23:24.24 | uski | JT: and what happens in the dialplan when the SIP peer is unreachable ? |
| 23:24.25 | JT | Blackvel: you can't get PRI? o_O |
| 23:24.31 | zamba | Aug 7 01:24:12 WARNING[30652]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/chan_iax2: cannot open shared object file: No such file or directory |
| 23:24.31 | JT | kand: BRI is NOT analogue |
| 23:24.32 | kand | drako: if you need help pb what you got and I can assist |
| 23:24.48 | JT | uski: it goes to the next line in your dialplan |
| 23:24.51 | Blackvel | besides isdn there is 1 analog line |
| 23:24.58 | kand | JT: sorry, true but it is traditional service |
| 23:24.59 | uski | JT: does it set any variable ? |
| 23:25.02 | Blackvel | for telco. but that is no option |
| 23:25.14 | JT | kand: traditional in what sense? |
| 23:25.17 | Blackvel | kand: for my pstn number? no way. i won't port that |
| 23:25.20 | JT | Blackvel: what do you mean for telco? |
| 23:25.35 | Blackvel | kand: but i am going to use one for mobile phone forward |
| 23:25.58 | JT | yeah i would not recommend ever moving all services to a VoIPoI provider |
| 23:26.08 | Blackvel | jt: telekom germany |
| 23:26.31 | JT | Blackvel: i still don't know what you are saying, are you saying only telcos can buy PRI? |
| 23:26.34 | Blackvel | its o.k for me when asterisk picks up the call :) |
| 23:26.47 | kand | JT: I have very sucessfully but it isn't for everyone |
| 23:26.48 | Blackvel | jt: no |
| 23:27.06 | Blackvel | but help me, what is PRI exactly? T1? |
| 23:27.22 | JT | primary rate interface ISDN |
| 23:27.30 | JT | can run over T1 or E1, depends on country |
| 23:27.35 | Blackvel | do you mean 24-30 lines? or do you mean PRI phone/switch equipment? |
| 23:28.01 | JT | here it's delivered over E1 and available with between 10 and 30 active voice channels per circuit |
| 23:28.03 | Blackvel | ah in germany its s2m = 30 lines. probably is E1. that's not really my business I make money with :) |
| 23:28.27 | JT | in the US it's over T1 and avaiable with between 8 and 23 channels |
| 23:28.51 | Blackvel | i wanted to do that in 2004/2005 but java/j2ee/middleware/eai is going better for a contractor |
| 23:29.15 | JT | kand: if it's over dedicated DSL that doesn't use the Internet it's useable, but i wouldn't do it over Internet as the only means of voice connectivity |
| 23:29.16 | Blackvel | what are the costs for cisco call manager? |
| 23:29.22 | JT | lots :P |
| 23:29.25 | Blackvel | does it depend on size and voip stations? |
| 23:29.34 | Blackvel | i hate my middle size customer |
| 23:29.37 | Blackvel | its my own customer |
| 23:29.39 | JT | yeah it would |
| 23:29.48 | Blackvel | and now they put cisco voip phones on the desks |
| 23:29.52 | Blackvel | without asking me about asterisk |
| 23:30.02 | Blackvel | voip stuff was named on my bills |
| 23:30.09 | Blackvel | i gotta kick them in their ass :) |
| 23:30.19 | Blackvel | >100k? |
| 23:30.49 | JT | only for a big site |
| 23:31.00 | kand | JT: I have multiple locations that connect back to a data center using their existing internet connection (15x2 with diffserv) and the data center out to pure voip (multiple carriers). If the phone is not registered (for the most important people) I dial a cell phone number. |
| 23:31.16 | zamba | what do i need to get the chan_iax2 module loaded? |
| 23:31.32 | kand | JT: Works well for me and is cheap based on my usage. |
| 23:31.33 | Blackvel | so it cisco CM varies in price? not fixed? |
| 23:31.46 | JT | kand: is it dedicated dsl back aggregated back to your datacentre location? |
| 23:31.53 | JT | Blackvel: of course |
| 23:32.04 | Blackvel | jt: you forget one option. eicon/avm active BRI cards with capi 2.0 + chan_capi |
| 23:32.16 | Blackvel | but they are not really cheap too |
| 23:32.18 | JT | you've got handsets, phones, firmware licenses for phones, gateways, etc etc |
| 23:32.24 | JT | lol capi |
| 23:32.25 | kand | JT: no but with diffserv, g729, and only a handfull of channels from each location it hasn't been a problem |
| 23:32.33 | JT | good luck getting much support for capi |
| 23:32.46 | Blackvel | so whats that zaptel 1.6 thing with BRI in it? |
| 23:32.51 | JT | kand: oh ok, still sounds a little risky |
| 23:33.00 | Blackvel | BRI support? |
| 23:33.07 | JT | i think you can use BRI with 1.4 |
| 23:33.27 | Blackvel | but there needs to be a driver too? :) |
| 23:33.39 | JT | yeah you use the drivers from bristuff |
| 23:33.42 | Blackvel | do you know bristuff well? |
| 23:33.48 | JT | i've used it |
| 23:33.52 | JT | i didn't write it :P |
| 23:33.54 | Blackvel | hehe |
| 23:34.04 | kand | JT: I live on the bleading edge, but suprisingly very little issue in the year and half I have been running. Any time internet is down at a location cell phones kick in so no missed calls. Boss man happy, so I am happy! |
| 23:34.12 | Blackvel | is qozap.ko module better working than zaphfc.ko? |
| 23:34.21 | JT | kand: do the sites have any form of pstn backup? |
| 23:34.28 | Blackvel | i mean i dont say its not working |
| 23:34.31 | JT | Blackvel: they have different uses |
| 23:34.37 | Blackvel | i run with 0.2, now 0.3 and now i am testing 0.4 |
| 23:34.41 | JT | zaphfc is for single port hfc cards |
| 23:34.49 | JT | qozap is for quad and octal port cards |
| 23:34.55 | Blackvel | and dual? |
| 23:34.59 | JT | no idea |
| 23:35.02 | Blackvel | hehe |
| 23:35.03 | kand | JT: One or two have a fax line that in a REAL emergency I can use. |
| 23:35.28 | JT | kand: well you just want to make sure that the sites can call emergency services in a real emergency |
| 23:35.32 | Blackvel | its working most of the times. but i saw errors and asterisk not picking up the call anymore, too |
| 23:35.49 | JT | imho every site should have at least 1 analogue line for backup with a basic unpoweered handset connected |
| 23:35.51 | Blackvel | i can not afford that with my company in production mode ;) |
| 23:36.05 | JT | Blackvel: as i said, if you want reliable bri, get a gateway ;) |
| 23:36.05 | Blackvel | how much is bri -> sip gateway? |
| 23:36.18 | Blackvel | >500-1000 euros? |
| 23:36.19 | JT | varies |
| 23:36.28 | JT | i don't know how much it will cost in europe |
| 23:36.32 | JT | gets some quotes... |
| 23:36.43 | Blackvel | is there a us website? |
| 23:36.55 | JT | i dunno, at least patton and quintum make some |
| 23:37.03 | kand | JT: that is true and for that I am counting on the fact that everybody carries cell phones these days. So..... you never get something for nothing, and thats just par for the course. |
| 23:37.05 | Blackvel | jt: how does Digium sell B410P when it only supports mISDN? |
| 23:37.18 | Blackvel | ahh patton, heard that name yesterday |
| 23:37.23 | JT | kand: cell phones are not suitable as sole emergency contact devices |
| 23:37.38 | JT | cell networks are not as reliable as fixed line |
| 23:37.54 | JT | Blackvel: because they are US centric, the b410p was really an afterthought |
| 23:38.12 | JT | almost no-one in the US uses bri for voice |
| 23:38.22 | uski | JT: it's fun that you say that, i had a (small) argument this afternoon with one relative about this; he threw away an old very simple phone that i told him to keep in the garage in case of a power outage |
| 23:38.32 | Blackvel | its all about the drivers ;) |
| 23:38.39 | uski | all the other phones need the AC line voltage to work |
| 23:38.45 | JT | lol |
| 23:39.04 | uski | so if there is a power outage in the area... then he'll have no way to contact anyone |
| 23:39.19 | Blackvel | good point |
| 23:39.26 | Blackvel | have to check that with my bri setup |
| 23:39.27 | Blackvel | hehe |
| 23:39.29 | kand | JT: that is true but this is a business line and all our employees have signed paper work on file which states the phone may not be reliable for 911 service. |
| 23:39.30 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 23:39.34 | uski | afaik, fixed lines do have battery backups, but cellphones base stations do not |
| 23:40.08 | Blackvel | thanks so far! |
| 23:40.08 | uski | yea but if they have no way to contact 911 when they need it they may try to sue you for not providing them with a way to call for help |
| 23:40.09 | JT | kand: even if they've signed paperwork, it's not good if something bad happens and they aren't able to contact emergency services |
| 23:40.11 | Blackvel | cu |
| 23:40.33 | uski | i like the fact that my spa3000 just acts like a stupid bridge between its FXS and FXO port when it looses power |
| 23:40.42 | uski | it's a fail safe design |
| 23:40.51 | kand | JT: like I said before it may not be a good solution for everyone but as far as reliable good quality calls I would have to say it works well |
| 23:41.04 | JT | uski: yes and no, generally fixed lines run back to the exchange which has massive 2V lead acid cell banks with autostart generators, cell sites that aren't situated at an exchange have battery backup with no genset generalluy |
| 23:41.29 | JT | sometimes transportable generators can be deployed to a cell site, but that's not really for widespread power issues |
| 23:41.39 | JT | and the batteries don't last more than a coupld of hours |
| 23:41.45 | kand | JT: Just curious, what is your setup for emergency lines? |
| 23:41.49 | uski | JT: i guess it depends of the country; i live in france and i've seen quite a few cellphone base stations and they were far too small to have any battery bkup or diesel generator |
| 23:42.06 | uski | i should ask a former coworker who was working at nortel networks, designing cellphone base stations |
| 23:42.11 | uski | he should know about this :) |
| 23:42.20 | JT | uski: a micro or picocell might not have batteries |
| 23:42.28 | JT | but a full tower site would |
| 23:42.33 | uski | ok |
| 23:42.38 | JT | kand: i have a couple of analogue lines |
| 23:43.05 | kand | JT: Any voip termination? |
| 23:43.15 | JT | kand: only for overseas calls |
| 23:43.34 | JT | i use voip a lot more for my personal use though |
| 23:44.46 | kand | JT: when we started out we used the analog lines at each location, then consolidated to PRIs then voip. I understand your arguments but the desision here was $60 a location cut into the whole reason we went voip. So analog lines where dumped at half the locations, the rest kept them for fax |
| 23:45.05 | JT | wow only a $60 saving |
| 23:45.24 | JT | that seems pretty miserly |
| 23:45.26 | kand | Over many locations adds up per month |
| 23:45.31 | JT | even so |
| 23:45.35 | JT | cost of doing business |
| 23:45.38 | kand | hey, I am just the messenger |
| 23:45.40 | kand | lol |
| 23:45.42 | JT | heh |
| 23:46.03 | zamba | is there an easy way to block outgoing calls for a specific peer? |
| 23:46.07 | kand | luckly, we haven't had to test it but .... |
| 23:46.11 | JT | and for my own business venture, i have no problem dropping $60/mo at the drop of a hat |
| 23:46.25 | JT | dropping as in spending |
| 23:46.30 | JT | if i think it's necessary |
| 23:46.37 | [TK]D-Fender | zamba: Point them to a context that doesn't allow them to do things you don't want them to do. |
| 23:46.38 | uski | from my modest installation, most of my expenses come from faxes; i still have a dedicated landline for faxes because faxes are business-critical for me and i have yet to find a voip provider that supports T38 or something appropriate for faxing reliably |
| 23:47.06 | zamba | [TK]D-Fender: problem is that i have all sip users coming from openser |
| 23:47.24 | uski | welcome back [TK]D-Fender |
| 23:47.28 | kand | JT: Hey, they way I look at it your still saving and it is the cost of business. but my job is to make it work |
| 23:47.48 | [TK]D-Fender | zamba: ....and? |
| 23:48.11 | zamba | [TK]D-Fender: so the context is already defined by the openser peer defined in sip.conf |
| 23:48.22 | zamba | [TK]D-Fender: all users goes into the same context |
| 23:48.28 | JT | kand: i value availability pretty highly, heh, so i have about 3 different datacentre locations for some of my own stuff |
| 23:48.52 | kand | JT: ya, I have two each with a 6 server cluster |
| 23:49.08 | *** join/#asterisk Daejeo (n=chatzill@118.219.208.10) |
| 23:49.18 | kand | JT: and I only got the second one when the first data center had an outage |
| 23:49.22 | uski | kand, geez, a 6 servers cluster ? how many calls are you routing ? or is it just for redundancy ? |
| 23:49.24 | [TK]D-Fender | zamba: How do the calls look to? |
| 23:49.26 | Daejeo | is there any command to change alternative Tftp address using Telenet? |
| 23:49.29 | [TK]D-Fender | * |
| 23:49.38 | JT | i do co-lo for customers in one location |
| 23:49.44 | JT | where i have a full rack |
| 23:49.46 | kand | uski: 3 asterisk 2 mysql cluster 1 manager |
| 23:49.47 | Daejeo | is there any command to change alternative Tftp address using Telenet? cisco 7940 |
| 23:50.28 | zamba | [TK]D-Fender: what do you mean? |
| 23:50.30 | kand | JT: where are you anyway, US? |
| 23:50.48 | JT | australia |
| 23:51.05 | JT | E1s here not T1s ;) |
| 23:51.20 | [TK]D-Fender | zamba: What do the calls look lie to * when they come in? What peer do they hit? is CID trustworthy? How do you differentiate one caller from another? |
| 23:51.46 | kand | JT: neat, lol ya first time someone told me about E1 I thought he was an idiot, its obviously a T1. |
| 23:52.16 | zamba | [TK]D-Fender: openser sets callerid |
| 23:52.21 | JT | heh |
| 23:52.51 | JT | E1 is used in most parts of the world |
| 23:52.52 | [TK]D-Fender | zamba: then check for the CID before dialing out like normal |
| 23:53.10 | Daejeo | is there any command to change alternative Tftp address using Telenet? cisco 7940 |
| 23:53.23 | JT | what is telenet? |
| 23:53.33 | kand | JT: I have been jaded after working tech support through college. Users are automaticly wrong, but I have gotten over that. |
| 23:53.48 | jameswf-home | 32* channels psh.... who neds more than 24 |
| 23:54.02 | JT | ... |
| 23:54.03 | kand | Daejeo: you may be able to identify it using ? |
| 23:54.08 | JT | businesses? |
| 23:54.08 | drmessano | 8 FXOs should be enough for anyone |
| 23:54.11 | JT | lol |
| 23:54.22 | JT | it's 30 channels in pri btw |
| 23:54.24 | JT | not 32 |
| 23:54.27 | jameswf-home | if the line is busy well callers can just call back |
| 23:54.37 | jameswf-home | 2* D |
| 23:54.45 | kand | actualy I think it is 23 with d channel at 24 |
| 23:54.47 | uski | depends if you have one of these taxes numbers |
| 23:54.55 | uski | taxed* |
| 23:55.00 | JT | kand: yeah but T1 is available without pri signalling |
| 23:55.03 | JT | for voice |
| 23:55.09 | JT | you can get channelised robbed bit signalling |
| 23:55.14 | JT | that gives you 24 |
| 23:55.18 | JT | but RBS sucks :P |
| 23:55.19 | drmessano | If you need more than 8 FXOs, your people spend too much time on the phone and not enough time working <-- Premise under which I am building all my underpowered PBXs for "datsmypbx.com" |
| 23:55.21 | uski | if you get $0.5/min when someone is on hold, you'd prefer to have enough FXOs and keep people waiting |
| 23:55.31 | Daejeo | kand: telnet* |
| 23:55.32 | Nugget | telnet is eeeeeeevil! |
| 23:55.37 | kand | JT: That is what I have heard, I always ran a d-channel |
| 23:55.52 | jameswf-home | dchan=pri |
| 23:56.02 | Daejeo | is there any command to change alternative Tftp address using Telnet? cisco 7940 |
| 23:56.12 | kand | daejeo: Yes, in the telnet typing '?' showes you all avaliable commands where ever you are |
| 23:56.14 | JT | an E1 has 32 timeslots, so that's 1 for a D channel, 1 for multiframe sync, and 30 left for bearer channels |
| 23:56.24 | jameswf-home | Cisco is evil but their stocks are up |
| 23:56.24 | drmessano | thinks he heard someone say telnet |
| 23:56.27 | nhuisman_work | Daejeo, the config files it loads have a spot to change it |
| 23:56.30 | [TK]D-Fender | ~e1 |
| 23:56.30 | jbot | [~E1] E1 is the basic digital telephony circuit used everywhere except the US,Japan,Taiwan & Hong Kong (who use T1 and similar variants). E1 runs at 2.048Mbps. It can be an unstructured channel for data. It can be channelized, to provide 31 time slots of voice or data, each of 64kbps. Time slot 16 is used for D-Chan when used with PRI signalling. |
| 23:56.40 | jameswf-home | ~t1 |
| 23:56.41 | jbot | [~T1] T1 is the basic digital telephony circuit used in North America. T1 runs at 1.544 Mbps. It can be an unstructured channel for data. It can be channelized, to provide 24 time slots of voice or data, each of 64kbps. Time slot 24 is used for D-Chan when used with PRI signalling. |
| 23:56.43 | kand | daejeo: so telnet in, go to set the TFTP primary and type '?' |
| 23:56.47 | drmessano | ~j1 |
| 23:56.56 | jameswf-home | heh japan sux |
| 23:56.57 | JT | i wonder who wrote that e1 jbot entry |
| 23:57.02 | drmessano | Damnit, sank my battleship |
| 23:57.06 | kand | daejeo: I dont know of anything specific tho |
| 23:57.12 | JT | as i have never EVER seen an E1 for voice without PRI signalling |
| 23:57.28 | [TK]D-Fender | JT: I have |
| 23:57.37 | JT | where? |
| 23:58.00 | jameswf-home | MFC/R@ is rbs E1 |
| 23:58.05 | jameswf-home | *R2 |
| 23:58.08 | [TK]D-Fender | JT: Don't recall exactly where the guy was located, but it was east Europe somewhere IIRC |
| 23:58.24 | JT | ah mfc/r2 sucks to be those people |
| 23:58.32 | JT | but i think that uses audible tones? |
| 23:59.11 | Daejeo | ?,clear,debug,dns,erase,exit,ping,register,reset,show,test,timers,traceroute, tty,undebug |
| 23:59.15 | jameswf-home | JT i think for call data but control is rbs |
| 23:59.23 | Daejeo | kand: |
| 23:59.39 | jameswf-home | you have to set idle state in configs |