| 00:01.19 | *** join/#asterisk RoyK (n=roy@217-175-152.100710.adsl.tele2.no) |
| 00:02.36 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 00:03.40 | *** join/#asterisk Dane1 (n=DaneM@75.40.221.68) |
| 00:03.50 | *** part/#asterisk Dane1 (n=DaneM@75.40.221.68) |
| 00:04.14 | *** join/#asterisk toombaloomba (n=phph@do.you.like.my.frippers.com) |
| 00:11.37 | `p4r14h | had problems with a x100p in red alarm when hooked up to POTS in trixbox, recompiled zap modules, reinstalled EVERYTHING from scratch, tried a differnt PBX software (asterisk now) still in red alarm.......is this a bad card? |
| 00:11.42 | `p4r14h | im out of ideas =\ |
| 00:13.27 | iceyp | can someone tell me what is ment by this please: No user '6493375533' in SIP users list |
| 00:13.36 | iceyp | thats the number i was calling from ;/ |
| 00:16.19 | iceyp | From: <sip:6493375533@203.184.16.35>;tag=E53C7504-2637 |
| 00:16.19 | iceyp | To: <sip:6499742910@203.184.16.35:5060>;tag=as1b452cfd |
| 00:16.43 | iceyp | it's not picking up the 6499742910 in the sip.conf context |
| 00:17.23 | mihinomenest | can I do something like, "exten => s,1,dial(macro)" ? |
| 00:17.53 | tsurko | exten => s,1,Macro(macro_name) ->i think this is the correct syntax |
| 00:18.29 | mihinomenest | hmm. |
| 00:18.40 | mihinomenest | I'll google it in a minute. |
| 00:18.43 | iceyp | I have [general] with context=incomming , and then in [incomming] i have exten => 6499742910,1,Goto(break_incoming,s,1) |
| 00:18.55 | tsurko | i'm just curious - what softphones do you prefer to use on linux machones? |
| 00:19.24 | gambolputty | none |
| 00:19.50 | mihinomenest | iceyp: "6499742910" is the number you're calling from? |
| 00:20.30 | iceyp | no |
| 00:20.34 | iceyp | thats the numberi 'm calling |
| 00:21.15 | *** join/#asterisk tessier_ (n=treed@kernel-panic/sex-machines) |
| 00:27.26 | *** join/#asterisk rhombus (n=rhombus@S01060006257edf62.cg.shawcable.net) |
| 00:27.49 | rhombus | can I do conditional include statements in extensions.conf, or do I need to use a GotoIf statement? |
| 00:30.19 | *** join/#asterisk RA25 (n=RA25@c-66-31-1-124.hsd1.ma.comcast.net) |
| 00:30.19 | sbingner | rhombus, no conditional includes |
| 00:31.14 | rhombus | no? |
| 00:31.21 | rhombus | just by time, then? |
| 00:31.28 | rhombus | okay |
| 00:31.37 | rhombus | sbingner: thanks |
| 00:34.25 | iceyp | is exten => s,1,Goto(break_incoming,s,1) still a valid statement in 1.4? |
| 00:34.57 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 00:40.23 | rhombus | my Polycom phones are rejecting a leading * |
| 00:40.28 | rhombus | I get an immediate fast busy |
| 00:41.06 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 00:43.25 | JunK-Y | iceyp: sure, why? |
| 00:43.46 | iceyp | JunK-Y since moving to 1.4 my asterisk wont accept calls ne more |
| 00:44.01 | bkruse_home | who is a good provider that has a flat rate rather than per minute....... |
| 00:44.54 | JunK-Y | iceyp: CLI output? |
| 00:45.21 | iceyp | http://www.pastebin.ca/396782 |
| 00:45.45 | [TK]D-Fender | rhombus: You need to change your Polycom's dialplan |
| 00:46.51 | iceyp | JunK-Y that look normal? |
| 00:47.06 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 00:47.52 | JunK-Y | # |
| 00:47.53 | JunK-Y | Using INVITE request as basis request - 35630-3382992879-60889@mscak1 |
| 00:47.53 | JunK-Y | # |
| 00:48.05 | iceyp | yeh whats that about |
| 00:48.09 | JunK-Y | No user '6493375533' in SIP users list |
| 00:48.19 | iceyp | 6493375533 is what im calling from |
| 00:48.22 | iceyp | into the pabx |
| 00:48.36 | JunK-Y | it has nothing to do with goto. |
| 00:49.07 | iceyp | so i'm calling from PSTN number 6493375533 over PSTN into a SIP number of 6499742910 |
| 00:49.30 | iceyp | but its rather taking 6493375533 and saying im not a user so not authorized to make calls |
| 00:49.39 | iceyp | when im not trying to make calls, i'm trying to call the pabx |
| 00:52.13 | bkruse_home | anyone here have teliax? |
| 00:52.37 | [TK]D-Fender | iceyp: Your entry is a PEER that us used to PLACE calls. If you are tying to use that account to RECEIVE calls it should be "type=user" |
| 00:53.03 | iceyp | it's both a peer and a user, because it places calls and receives calls |
| 00:53.09 | iceyp | let me try it tho |
| 00:53.10 | rhombus | [TK]D-Fender: changed the dialplan |
| 00:53.17 | [TK]D-Fender | iceyp: then it should be "friend" |
| 00:53.36 | rhombus | it will take a *7, but not a *8 |
| 00:53.41 | rhombus | strange |
| 00:53.53 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 00:53.53 | JunK-Y | bkruse_home: nope |
| 00:53.54 | brian | Is there anyway to send a message as the ring like "This number should not be used, instead use ____" so i don't use up minutes? |
| 00:53.56 | rhombus | the dialplan has "*[2-9]|blablabla" |
| 00:53.59 | [TK]D-Fender | rhombus: paste its dialplan |
| 00:54.03 | rhombus | okay |
| 00:54.21 | iceyp | friend is doing the same thing |
| 00:54.31 | rhombus | dialplan.digitmap="*[2-9]|[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxxxxT|[2-9]xxxT|[2-9]xxT" |
| 00:54.31 | iceyp | let me try user |
| 00:54.51 | [TK]D-Fender | iceyp: Pastebin MORE of the call from the initial invite to the very end |
| 00:54.52 | rhombus | [TK]D-Fender: thar be it |
| 00:55.11 | iceyp | yaym using user now rings :) |
| 00:55.40 | iceyp | ok user appears to work, just need to get rid of digittimeout |
| 00:56.39 | [TK]D-Fender | rhombus: Personally I suugest : x.T|*.T|#.T |
| 00:57.18 | rhombus | yeah... I think we had this discussion before with ManxPower :) |
| 00:57.22 | iceyp | wat has replaced DigitTimeout |
| 00:57.42 | rhombus | but I will try the *.T, since this is the only thing we're using the * for |
| 00:58.07 | rhombus | is it safe to reload a dialplan while there are active calls? |
| 00:58.16 | rhombus | probably yes -- I'm just making sure |
| 00:59.52 | [TK]D-Fender | rhombus: yes |
| 01:00.15 | [TK]D-Fender | rhombus: Mind you you're going to need to reboot the phone for it to take |
| 01:00.54 | brian | how do I play a message like "Please dial XXX-XXX-XXXX instead of this number" instead of a ring? |
| 01:02.16 | JT | Playback() |
| 01:02.21 | *** join/#asterisk neuwald (n=felipe@200.96.162.16) |
| 01:02.58 | iceyp | can anyone tell me what variable replaced digittimeout |
| 01:03.22 | neuwald | I'm having here a problem with asterisknow: when I go to users tab, I got the message: Asterisk says it cannot find a required config file (contactinfo.conf) You will be now redirected to the main page |
| 01:03.24 | rhombus | yeah, I just did |
| 01:03.35 | rhombus | [TK]D-Fender: problem still there |
| 01:03.42 | neuwald | does anybody knows why this happening? of course, the file isn't there, but it's a known bug ? |
| 01:03.42 | rhombus | I'm starting to suspect a dialplan issue |
| 01:06.18 | *** join/#asterisk russellb (i=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 01:06.18 | *** mode/#asterisk [+o russellb] by ChanServ |
| 01:06.33 | *** join/#asterisk osiris (n=osiris@71.205.27.131) |
| 01:06.41 | rhombus | why would my polycom phone accept a *7, but not a *8? |
| 01:07.17 | JunK-Y | digitmaps, like [TK]D-Fender already said. |
| 01:07.57 | rhombus | well, my digitmap should allow a *8 |
| 01:08.07 | rhombus | *[2-9] |
| 01:08.10 | rhombus | is what I had |
| 01:08.11 | generalhan | can some one tell me how to determine what my NAT port number should be set to for a particular router ? or is it something that is typically configurable ? |
| 01:08.41 | JunK-Y | and it rebooted? |
| 01:09.04 | rhombus | yes |
| 01:09.16 | rhombus | it's puzzling |
| 01:09.17 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:09.31 | rhombus | i'm trying *.T now, and rebooting the phone |
| 01:09.43 | JunK-Y | in ur CLI: dialplan show *8@context? |
| 01:09.53 | *** join/#asterisk ShadowTech (n=jerespet@12.43.119.177) |
| 01:11.15 | rhombus | okay, get this |
| 01:11.23 | rhombus | when I dial *8, this is what appears in my asterisk CLI: |
| 01:11.32 | rhombus | Mar 15 19:10:41 NOTICE[4700]: chan_sip.c:10754 handle_request_invite: Nothing to pick up |
| 01:11.53 | JunK-Y | bingo, ya've ur answer. |
| 01:11.57 | JunK-Y | features.conf |
| 01:12.05 | rhombus | oh |
| 01:12.20 | rhombus | features.conf? |
| 01:12.58 | rhombus | but *8 is commented out in features.conf! |
| 01:13.01 | rhombus | what gives? |
| 01:14.56 | rhombus | JunK-Y? |
| 01:15.19 | russellb | it looks like your features.conf setting is fine, but your pickupgroup assignments are not |
| 01:15.56 | JunK-Y | i asked you the dialplan show, with that, ya will know whats going on exactly. |
| 01:16.00 | russellb | oh, you want it to not do that |
| 01:16.12 | russellb | *8 is the default probably |
| 01:16.15 | rhombus | okay, I get it |
| 01:16.19 | rhombus | I just put that together |
| 01:16.35 | rhombus | okay -- dumb -- I'll just move that feature to a different extension :\ |
| 01:16.55 | neuwald | I'm using asterisknow connected to a sip provider. When I call via PSTN asterisk, I heard who answer the call, but him doesn't heard me |
| 01:16.58 | neuwald | any help? |
| 01:17.11 | *** join/#asterisk topping (n=topping@adsl-71-146-152-95.dsl.pltn13.sbcglobal.net) |
| 01:17.33 | bkruse_home | neuwald: its a nat issue i bet |
| 01:17.34 | bkruse_home | nat=yes |
| 01:18.11 | Omer^ | yes thats a nat issue |
| 01:20.02 | generalhan | anyone here ever used an Aastra 9112i behind NAT ? |
| 01:22.13 | *** join/#asterisk dseeb_ (n=dcb@CPE-58-169-130-113.vic.bigpond.net.au) |
| 01:28.04 | *** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 01:30.14 | neuwald | bkruse_home : tks, i solved with canreinvite=no |
| 01:30.45 | *** part/#asterisk rhombus (n=rhombus@S01060006257edf62.cg.shawcable.net) |
| 01:33.08 | Strom_M | The HI/COCKS protocol (RFC 4373) |
| 01:34.56 | bkruse_home | neuwald: that has to d with nat. |
| 01:34.56 | bkruse_home | that makes sense, also |
| 01:36.11 | *** join/#asterisk sevard (n=sev@adsl-71-129-115-244.dsl.irvnca.pacbell.net) |
| 01:47.35 | neuwald | ok... does anybody here is using vono voip service (from brazil)? |
| 01:48.28 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 01:50.36 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.142.28) |
| 01:51.14 | *** join/#asterisk Hmmhesays (n=Neg@24-119-151-57.cpe.cableone.net) |
| 01:53.41 | *** join/#asterisk tb0301s (n=sowa@brln-4db1110d.pool.einsundeins.de) |
| 01:56.39 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 01:57.11 | *** join/#asterisk Hansin321 (n=Eric@71.237.90.115) |
| 02:00.11 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 02:01.31 | *** join/#asterisk [[blah]asfd (n=ckwall@71.195.199.149) |
| 02:02.14 | [[blah]asfd | i can call another extension and i can do playback where i hear audio just fine, but if i dial from an extension to asterisk which is connected to another server across the internet I do not get sound. |
| 02:02.37 | [[blah]asfd | would that be a symptom of traffic from rtp on ports 10000-20000 not making it to the right place? |
| 02:03.07 | JT | quite possibly |
| 02:04.21 | [[blah]asfd | could it be a symptom of anything else? |
| 02:04.47 | [[blah]asfd | i tried a tcpdump but am not seeing anything that would point a finger at this issue. |
| 02:04.51 | [[blah]asfd | at least that I can tell |
| 02:05.23 | [[blah]asfd | from a router i would just port forward udp ports 5060 and 10000-20000 to the other server, right? |
| 02:05.39 | *** join/#asterisk hyphen (n=hyphen@71.224.213.97) |
| 02:06.06 | JT | i'm sorry you'll have to explain or make a diagram for the whole end to end scenario |
| 02:06.33 | sevard | the man puts his parts in the woman's parts |
| 02:06.36 | sevard | and then babies are made. |
| 02:06.51 | [TK]D-Fender | [[blah]asfd: You need a whole whack of settings in sip.conf to work from behind NAT |
| 02:06.54 | [TK]D-Fender | ~sipnat |
| 02:06.55 | jbot | sipnat is, like, for for more information about configurtion of Asterisk with SIP behind NAT, see http://voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 02:07.01 | *** join/#asterisk michaelo (n=michaelo@adsl-068-159-111-129.sip.gsp.bellsouth.net) |
| 02:09.14 | [[blah]asfd | is iax stable enough for high volume servers? |
| 02:09.35 | [[blah]asfd | could it possibly support up to 100 concurrent connections? |
| 02:12.41 | sevard | <PROTECTED> |
| 02:13.00 | [[blah]asfd | which asterisk version? |
| 02:13.20 | sevard | 1.2 |
| 02:13.32 | [[blah]asfd | i had heard that if I want to do high iax volume i need to be on 1.4. at the time 1.4 was giving every one fits. |
| 02:14.58 | ltdwk | i found iax a bit flaky (i was using trunking) |
| 02:15.07 | ltdwk | i changed to sip |
| 02:18.30 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 02:18.40 | ltdwk | this was on 1.2.10 or there abouts |
| 02:22.01 | *** join/#asterisk dj-fu (n=ajc@202-74-195-152.ue.woosh.co.nz) |
| 02:22.32 | *** join/#asterisk ppyy (n=ppyy@222.188.129.138) |
| 02:23.36 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 02:34.55 | *** join/#asterisk kFuQ (n=somedude@67.185.123.34) |
| 02:38.51 | *** join/#asterisk sevard (n=sev@adsl-71-129-115-242.dsl.irvnca.pacbell.net) |
| 02:41.42 | *** join/#asterisk gr1ncheux (n=devine@unaffiliated/gr1ncheux) |
| 02:52.02 | carrar | Whats it mean in 1.4 when ZAP/pseudo channels are in Rsrvd State? |
| 02:52.16 | carrar | Assuming something is hung |
| 03:05.02 | [[blah]asfd | anyone here familiar with nufone.net? |
| 03:08.28 | [[blah]asfd | i was really excited to find them because they had decent quality, and the price was right... however, they do not respond to technical support. I requested (3 times now) to have my number ported to them. I have been doing this since december with no luck. I need help finding someone else. |
| 03:09.33 | [[blah]asfd | i like the prepay service... $10 got me 500 minutes. I can use them over multiple months until they run out... then i just pay for more. |
| 03:09.35 | carrar | heh |
| 03:09.44 | [[blah]asfd | anything out there that compares even a little bit? |
| 03:09.58 | carrar | Aer you getting what you paid for? |
| 03:10.19 | [[blah]asfd | like i said, the product is great... i just cannot ever talk to anyone. |
| 03:10.30 | JT | then the product is not that great |
| 03:11.12 | [[blah]asfd | if i could just get my number moved, i would never need tech support, I have you fine folks ;-) |
| 03:11.31 | [[blah]asfd | can anyone make a referral to another comapy? |
| 03:11.34 | [[blah]asfd | company |
| 03:13.07 | *** join/#asterisk gr1ncheux (n=devine@unaffiliated/gr1ncheux) |
| 03:17.47 | *** join/#asterisk Zand3r (n=Zand3r@host86-146-79-173.range86-146.btcentralplus.com) |
| 03:19.07 | [[blah]asfd | :-( sad |
| 03:20.25 | carrar | I do my own |
| 03:20.28 | carrar | thats the best way |
| 03:21.19 | [[blah]asfd | you have a provider though, right? |
| 03:21.29 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 03:21.30 | carrar | I am the provider |
| 03:21.46 | [[blah]asfd | so who do you connect to for non voip calls |
| 03:21.55 | carrar | several PSTN's |
| 03:22.08 | Zand3r | Hi... I am experimenting with a Digium TDM400P and a couple of Polycom phones (430 and 501 models). Volume seems a little low on the phones and there is some echo when the volume is cranked right up. There is an element of echo SIP to SIP and this becomes more pronounced for calls routed through the TDM400P. My understanding is that I should alter gain levels (either in the phone, for the TDM400P configuration, or both) |
| 03:22.08 | Zand3r | in order to alter the base volume and perhaps improve the echo. I wondered if anyone had any suggestions? |
| 03:22.16 | carrar | qwest, eli, gblx, mci |
| 03:22.25 | Zand3r | I am in the UK - I am unsure if that woudl affect gain configurations for the TDM400P ? |
| 03:22.43 | infinity1 | Zand3r: what zap drivers are you using? |
| 03:23.00 | flenders | Zand3r: use fxotune to tune the fxo modules |
| 03:23.12 | *** join/#asterisk ManxPower (n=manxpowe@66.sub-70-196-244.myvzw.com) |
| 03:23.29 | [[blah]asfd | carrar: thats not much different than what I am doing... so qwest is your provider. |
| 03:23.49 | flenders | Zand3r: I had over 38% of echo on the lines here, and they droped to 3%, then echocancel/echotraining did the rest |
| 03:24.11 | infinity1 | flenders: how do you measure echo as a percentage? |
| 03:24.35 | flenders | infinity1: you can run fxotune and it'll show you |
| 03:24.49 | Zand3r | infinity1: My base install was taken from the AsteriskNow BETA so I did not move away from drivers supplied with that. All configuration asside from adding the initial extensions was performed on the command line though. |
| 03:24.50 | flenders | fxotune -d -b <device> |
| 03:25.29 | infinity1 | Zand3r: i suggest you find out what version you're using. if you're using v1.2, your porblem will be easily sovled by downloading 1.4 trunk |
| 03:25.47 | flenders | asterisknow is based on 1.4, no? |
| 03:26.04 | Zand3r | Ahh - I see - Yes, Asterisk now is based on 1.4 |
| 03:28.17 | *** join/#asterisk Strom_M (i=strom@nat/digium/x-4c72ba6e12e5554f) |
| 03:28.50 | Zand3r | I get echo ratio = 0.3943 (1797.6 / 4559.4) and echo ratio = 0.3794 (1729.9 / 4559.4) for my lines 1 and 2 |
| 03:29.07 | flenders | that's a lot! |
| 03:29.13 | infinity1 | hmm ..after reading about asterisknow. isn't it a waste of a server running an "asterisK" distribution. |
| 03:29.17 | flenders | now run it with -i |
| 03:29.22 | infinity1 | wouldn't you want it to run on top for debian or ubuntu |
| 03:29.46 | flenders | infinity1: what's the difference? |
| 03:29.58 | JT | yeah i agree |
| 03:30.01 | JT | superflous gui |
| 03:30.10 | JT | being stuck on their distro |
| 03:30.11 | flenders | infinity1: I run on debian, but I tried asterisknow before, and it was alright |
| 03:30.26 | Zand3r | infinity1: In this instance we were dedicating a box to asterisk (the same as we'd have a box for an off the shelf PBX or equivalent) so it made sence to have AsteriskNow perform the heavy lifting of hardware setup, etc. |
| 03:31.02 | [TK]D-Fender | heavy Lifting for ahrdware setup? LOL |
| 03:31.13 | flenders | it's a 200KG server |
| 03:31.16 | [TK]D-Fender | * is a 10 minute install |
| 03:31.21 | Zand3r | Well it seemed like a good idea at the time :) |
| 03:31.31 | infinity1 | flenders: heh |
| 03:31.38 | [[blah]asfd | so who else in the us is using a sip provider... can anyone make a recommendation? |
| 03:31.55 | infinity1 | anyone if there are debian packages for 1.4.1? |
| 03:32.02 | [TK]D-Fender | Zand3r: Sounds like Genetic research is for you! Just find a REMOTE uncharted island to do it on, on Mr. Moreau? ;) |
| 03:32.03 | flenders | [[blah]asfd: voip-info.org is back up, have a look there |
| 03:32.19 | flenders | infinity1: I doubt it |
| 03:32.39 | infinity1 | flenders: why OH WHY!?? |
| 03:32.42 | [[blah]asfd | yeah.. but I am looking for a recommendation. nufone.net is on there, but I would tell people to stay away. |
| 03:33.10 | flenders | infinity1: I installed an old version once from the deb packages, but it was too old and too messy |
| 03:33.17 | infinity1 | [[blah]asfd: i use teliax and voipjet |
| 03:33.28 | flenders | infinity1: compiling was the best option for me |
| 03:33.39 | *** join/#asterisk e-milio (n=emilio@adsl-9-191-195.mia.bellsouth.net) |
| 03:33.40 | Zand3r | [TK]D-Fender: In hindsight I should have installed my favourite distribution, installed asterisk, given it a go and if all worked out then carry on and if I hit trouble (which it sounds like I most likely would not have done) then looked at AsteriskNow. However, at the time I thought running a distribution pre-configured for asterisk use might be easier and I'm fundamentally lazy. |
| 03:33.42 | infinity1 | i LOVE packages |
| 03:33.48 | flenders | [[blah]asfd: I use viatalk and it's pretty decent |
| 03:34.10 | [[blah]asfd | infinity1: any complaints |
| 03:34.40 | infinity1 | [[blah]asfd: hmmm .....once every few months i have a minor problem with one or the other. |
| 03:34.45 | Zand3r | After running with -i I now have echo ratio = 0.1602 (730.2 / 4559.4) and echo ratio = 0.1470 (670.4 / 4559.4) for lines 1 and 2 |
| 03:34.57 | infinity1 | there is usually an email explaining what the problem is an how to work around it (changing servr or whatever) |
| 03:34.59 | *** join/#asterisk xtr-II (i=01928375@S0106000c41ed11e1.vf.shawcable.net) |
| 03:35.09 | flenders | Zand3r: hmm, still a lot |
| 03:35.19 | flenders | Zand3r: but a lot better than before |
| 03:35.36 | Zand3r | flenders: I am just seeing if there's a noticeable improvement. |
| 03:35.55 | flenders | Zand3r: did you follow the instructions on voip-info? |
| 03:36.23 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 03:36.28 | [[blah]asfd | infinity1: if you could only have one, which provider is better? |
| 03:36.56 | Zand3r | flenders: I have the page open for fxotune - but perhaps I missed an explicit instruction, jsut checking. |
| 03:38.14 | Zand3r | flenders: Aha - I ran -i 1 but I see scrolling down that -i 5 should provide better results |
| 03:38.22 | e-milio | hello |
| 03:38.29 | creature1 | hi |
| 03:39.08 | e-milio | simple question: asterisk + digum card + sata = problem ? |
| 03:39.17 | flenders | Zand3r: I just followed those a few days ago, and the results were impressive |
| 03:39.35 | infinity1 | [[blah]asfd: i'd say about the same. though technically you can't have once. voipjet can onyl be used for outboudn calls |
| 03:40.33 | Zand3r | flenders: It didn't make much difference usign the 5 - I now have echo ratio = 0.1561 (711.8 / 4559.4) and echo ratio = 0.1537 (700.9 / 4559.4). I don;t know what these numbers mean but you seem to think they are high. Is there anythign else I can do? |
| 03:40.59 | infinity1 | Zand3r: half duplex |
| 03:41.29 | flenders | Zand3r: my lines here are all between .01 and .03 |
| 03:42.25 | creature1 | e-milio: you have a SATA PCI-card? |
| 03:42.29 | Zand3r | flenders: That's an order of magnitude different! - Is this line specific, i.e. something I could go shout at the telco about? |
| 03:42.54 | e-milio | creature1: sata hardrives |
| 03:43.03 | flenders | Zand3r: hmm, I don't know much about it... I was just sharing my experiences... :o) |
| 03:43.10 | flenders | Zand3r: maybe run HPEC? |
| 03:43.13 | creature1 | e-milio: that shouldn't be any problems |
| 03:43.53 | e-milio | creature1: with scsi it know to be not good... that why i ask. thanks |
| 03:44.00 | flenders | Zand3r: http://www.digium.com/en/products/software/hpec.php |
| 03:44.21 | creature1 | e-milio: that sounds crazy, that there's any problem with that combination (scsi) |
| 03:44.52 | ManxPower | e-milio: many SATA and RAID controllers lock interrupts for a long time -- causing lost audio data. GigEthernet also frequently does this |
| 03:45.43 | orlock | ManxPower: thats a gigabit interface, not just being on a gigabit switch, right? |
| 03:45.43 | creature1 | shouldn't be any problem with a card that doesn't share irq's |
| 03:46.28 | creature1 | (i suppose)= |
| 03:46.58 | e-milio | what can be done when lots of recordings need to be done ad speed of the hd is important ? |
| 03:47.24 | [[blah]asfd | e-milio: record to ramdisk instead |
| 03:47.32 | e-milio | mmhmmm |
| 03:47.45 | e-milio | [[blah]asfd: never heard of that one |
| 03:48.13 | Zand3r | flenders: I can report that whilst my numbers might not be as good as yours, the echo does seem to have vanished for calls going out over the analogue lines. So I owe you a big thanks! |
| 03:49.09 | andrew` | trying to update to 1.4 as my provider can't fix callerID issues with IAX in 1.2.13...i installed zaptel but asterisk's makemenuselect doens't think so |
| 03:49.17 | Zand3r | flenders: Don;t suppose you know how to increase the default volume of a polycom phone? :) |
| 03:49.20 | andrew` | the directions i've found online imply that happens automatically |
| 03:49.45 | flenders | Zand3r: nope, don't have polycoms |
| 03:49.58 | flenders | Zand3r: try tweaking the rx/tx gains now |
| 03:50.24 | Zand3r | flenders: For the card or on the phones ? |
| 03:50.32 | flenders | on zapata.conf |
| 03:50.48 | infinity1 | using 1.4 zaptel, gain doesn't seem to do much for me |
| 03:51.12 | infinity1 | except ruin the line quality. |
| 03:51.16 | flenders | infinity1: I use 1.4 too, and it made a big difference here |
| 03:51.21 | infinity1 | weird. |
| 03:52.08 | infinity1 | flenders: big difference for gain right? not ehc. |
| 03:52.10 | infinity1 | er ech |
| 03:52.11 | infinity1 | o |
| 03:52.21 | *** join/#asterisk bmg505 (n=leon@196.209.250.40) |
| 03:52.39 | flenders | yeah, gain |
| 03:53.11 | flenders | one of the lines here: rxgain=14.4 txgain=6.0 |
| 03:53.56 | *** join/#asterisk Strom_M (i=strom@nat/digium/x-9668a4fb1479da1e) |
| 03:54.11 | flenders | I tried the 0.0 as suggested in many places, and we could barely hear people, or be heard |
| 03:54.35 | sevard | http://store.telecomchinasourcing.com |
| 03:54.36 | sevard | wow |
| 03:54.37 | sevard | cheap prices |
| 03:54.52 | flenders | Zand3r: I spent a whole sunday trying to get it right |
| 03:55.01 | sevard | unless those boxes make omlets out of thin air there's no way in hell i'd buy one |
| 03:56.17 | andrew` | ah, rm -rf asterisk*, rerun ./configure and it detected it |
| 03:57.10 | mihinomenest | you have to wonder about a company that uses a visio stencil as their product image. |
| 03:59.41 | *** join/#asterisk intralanman (n=lanman@pool-71-253-253-149.nrflva.east.verizon.net) |
| 04:00.30 | tzafrir_laptop | rxgain 14.4? that's a lot |
| 04:00.47 | *** join/#asterisk intralanman (n=lanman@pool-71-253-253-149.nrflva.east.verizon.net) |
| 04:00.49 | *** part/#asterisk [[blah]asfd (n=ckwall@71.195.199.149) |
| 04:01.13 | flenders | tzafrir_laptop: I know, but it works |
| 04:01.30 | tzafrir_laptop | take a look at ztmonitor NUM -v , and see if you don't truncate the audio |
| 04:01.44 | flenders | tzafrir_laptop: I did... |
| 04:03.43 | *** join/#asterisk Cyber-Dogg (i=Cyber-Do@24-178-240-97.dhcp.stls.mo.charter.com) |
| 04:03.48 | *** join/#asterisk ezer (i=as@r190-64-41-94.dialup.adsl.anteldata.net.uy) |
| 04:04.12 | ezer | hello i am the most recent asterisk curious |
| 04:04.32 | ezer | anyone wanting to help me ? i knew about asterisk today |
| 04:04.44 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:06.01 | *** join/#asterisk topping (n=topping@204.152.96.238) |
| 04:09.55 | tzafrir_laptop | ezer, do you have eperince with Linux? |
| 04:10.18 | michaelo | Zand3r is the volume on your Polycoms high enough if you turn it up via the volume keys |
| 04:10.21 | michaelo | ? |
| 04:10.30 | ezer | a little |
| 04:10.55 | tzafrir_laptop | ezer, do you have linux already installed? |
| 04:10.57 | *** join/#asterisk Maveric (n=maveric@ip68-99-138-154.ph.ph.cox.net) |
| 04:10.59 | ezer | i know a bit about voip, but firstable i would like to know what is asterisk for |
| 04:11.07 | JT | firstable? |
| 04:11.11 | Zand3r | michaelo: No, my hearing is appalling but my colleagues believe that the volume could do with being higher even at the highest volume. |
| 04:11.23 | ezer | no i dont have yet.. wich linux would be right ? kubunutu for ex ? |
| 04:11.44 | JT | what is firstable? |
| 04:11.49 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 04:11.52 | flenders | JT: firstly |
| 04:12.01 | Zand3r | michaelo: When we turn the volume up to maximum (or approaching maximum) using the keys we do seem to get an aweful lot of echo (even on internal sip to sip calls) |
| 04:12.14 | ezer | haha yes sorry about my english |
| 04:12.15 | [TK]D-Fender | ezer: Here, go read the BOOK. It'll tell you all about * |
| 04:12.23 | [TK]D-Fender | ~book |
| 04:12.25 | jbot | from memory, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 04:12.25 | tzafrir_laptop | well, it comes with an Asterisk package (though not the latest, and in universe: not in the main repository) |
| 04:12.29 | [TK]D-Fender | Its free and everything too... |
| 04:12.50 | *** join/#asterisk ToyMan (n=Stuart@user-0cevdmv.cable.mindspring.com) |
| 04:13.07 | michaelo | used the handset or only when using speaker phone |
| 04:13.36 | mihinomenest | man. I can't believe how horrible some stuff sounds on an analog handset. |
| 04:14.39 | ezer | ok thanks.. but asterisk its an implementation of voip ? i need to programe something or it just to install and use ? |
| 04:14.49 | andrew` | decent did provider in the US? |
| 04:14.51 | Zand3r | michaelo: The problem seems to be using the handset - we have not used the speaker phone much. |
| 04:15.03 | Qwell | Just an FYI... I now own a copy of Windows Vista :( |
| 04:15.35 | JT | ezer: asterisk allows you to make a pbx, it does not have to use voip |
| 04:15.45 | michaelo | Zand3r: this is strange, should not be happening sip to sip. I can understand there being a problem with outside lines but not sip to sip |
| 04:16.12 | docelmo | Qwell you joined the dark side.. |
| 04:16.20 | Qwell | they sent me a copy |
| 04:16.25 | docelmo | Ive been running vista since November |
| 04:16.32 | docelmo | It takes some getting used to |
| 04:16.35 | ezer | i am a little lost i thing :(.. i read a little the book, but i couldn cathc the main idea from it |
| 04:16.41 | Qwell | oh, I'm not gonna actually use it |
| 04:17.20 | Zand3r | michaelo: That is what I thought - I originally thought it was restricted to the outside lines but it is not. It is only noticed at higher volumes. I wonder if some how there is a way to increase the base volume which may perhaps inadvertently resolve the echo at the same time? |
| 04:17.27 | ezer | whtas the diferrence of using asterisk or for example using skype, or another implementation ? |
| 04:17.54 | Qwell | ezer: Skype is slow, bloated, proprietary, and isn't a PBX |
| 04:18.13 | ezer | whats the diference of pbx and voip ? |
| 04:18.20 | Qwell | they aren't comparable |
| 04:18.38 | [TK]D-Fender | ezer: Go read THE BOOK. it will explain what * is all about |
| 04:18.45 | Qwell | ~book |
| 04:18.46 | jbot | rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 04:18.51 | michaelo | Zand3r: you can adjust the rx/tx gain on a Polycom but it requires a provisioning server and editing the phones config files. |
| 04:19.06 | ezer | ok |
| 04:19.54 | michaelo | ezer: A pbx is need when you want to have mulitple extensions internally along with functions like voicemail, etc. |
| 04:20.35 | Qwell | docelmo: somehow, twisted got you and I confused tonight :P |
| 04:20.46 | Qwell | figure that one out |
| 04:20.55 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 04:21.18 | Corydon76-home | Is docelmo short? |
| 04:21.24 | Qwell | Corydon76-home: ~6'3" :P |
| 04:21.56 | ezer | to asterisk to to be usefull i need to conect to a server so i cn call to external phone numbers ? |
| 04:22.00 | Juggie | i woudnt mess with him |
| 04:22.07 | Qwell | Juggie: I would - and it's FUN :P |
| 04:22.17 | Juggie | i've seen what he does to a plate of food |
| 04:22.22 | Qwell | I can get away with it though ;) |
| 04:22.55 | Zand3r | michaelo: This is interesting. I have set voice.aec.hs.enable="1" and voice.aes.hf.enable="1" and voice.aes.hs.enable="1" and voice.aes.hd.enable="1" and even though I am usign the handset they seem to have removed the echo and made the volume level louder. |
| 04:23.38 | michaelo | ezer: You can connect asterisk to the outside world via a VOIP provider or buy using a card or gateway to access analog or digital phone lines |
| 04:23.57 | Zand3r | michaelo: The polycom administration guide suggests that those settings only affect the speakerphone but evidently that is notthe case. Now, if I ould increase the default starting volume from 50% to somewhere around 65% I think things would be perfect. |
| 04:24.11 | Qwell | michaelo: "digital phone line" is a bit misleading |
| 04:24.21 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 04:24.53 | michaelo | Zand3r: great, hd, hs, hf are for handset, headset, handsfree(speaker phone) |
| 04:25.05 | ezer | for example a provider would be skype ? |
| 04:25.12 | Qwell | ezer: no |
| 04:25.55 | Qwell | ~itsp |
| 04:25.56 | jbot | i heard itsp is Internet Telephony Service Provider. An ITSP is a "VoIP Phone Company" |
| 04:26.06 | Qwell | ^ provider |
| 04:26.11 | michaelo | ezer: skype uses it's own protocol. You need a sip provider unless you want to buy another program to work with Asterisk to talk to skype |
| 04:26.18 | Zand3r | michaelo: That explains it - I found the manual a little misleading but that is perfect now ! |
| 04:26.41 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 04:26.43 | Mahmoud | hello |
| 04:26.54 | Mahmoud | any one here uses voicemail web interface |
| 04:27.06 | michaelo | Zand3r: you can set voice volume persist to 1 to make the phone remember volume between calls |
| 04:27.06 | Zand3r | michaelo: Going back to your original question, is there a configuration option for me to set the default volume level - the phone curently default to exactly half way between minimum and maximum and I oculd do with it being slightly higher as there's background noise here. |
| 04:28.09 | michaelo | Zand3r: I don't have a set of Polycom config files in front of me but its in the phone's sip.cfg file |
| 04:28.47 | [TK]D-Fender | Zand3r: Look for the 3 "persist" options in sip.cfg and set them all to "1" |
| 04:29.49 | Zand3r | michaelo: I see - I had seen the persist options but had thought it would be better to set the default volume higher rather than have them persist. I will turn persist on however then I think I have a solution everyone will behappy with. |
| 04:30.28 | Zand3r | [TK]D-Fender: Thanks. I had seen the persist options, wondered if there was an alternative method in terms of setting the default rather than having it persist but I think persist should suit me just fine. hanks for the help. |
| 04:31.08 | Qwell | Windows has detected startup. Cancel or allow? |
| 04:33.11 | [TK]D-Fender | Qwell: Ok. Fine. Sure. ? |
| 04:33.30 | [TK]D-Fender | Zand3r: That really is the best way. Let you tweak it as you go. |
| 04:33.55 | [TK]D-Fender | Zand3r: Oh and upgrade to SIP 2.1.0 , it'll double your volume setting resolution. |
| 04:34.14 | *** join/#asterisk simonr (n=simonr@xplr-ts-t11-208-114-158-94.barrettxplore.com) |
| 04:35.05 | Zand3r | [TK]D-Fender: Interesting. I updated to 2.1.0 yesterday - didn't notice the changes to volume settings |
| 04:36.01 | simonr | Has have people's experience with Asterisk 1.4 in production been? |
| 04:38.12 | *** join/#asterisk Teeli (n=tili@cm109.gamma248.maxonline.com.sg) |
| 04:38.18 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:39.23 | Qwell | way to go Microsoft! |
| 04:39.35 | Qwell | give me an OS, let me install it, let me setup a user account... |
| 04:39.40 | Qwell | BUT DON'T LET ME FREAKING LOGIN |
| 04:40.07 | Qwell | "The User Profile Service service failed the login" (real nice English there) |
| 04:40.33 | Juggie | what os? |
| 04:40.38 | Qwell | Vista ;p |
| 04:40.42 | Qwell | I'm not impressed, heh |
| 04:41.40 | Qwell | real nice - one single pitiful google hit |
| 04:41.47 | flenders | Qwell: I've been starring at the Vista DVD sitting on my desk for about 2 weeks... haven't had the courage to do it yet |
| 04:42.13 | Zand3r | Currently I currently dial multiple handsets by using a command such as Dial(SIP/201&SIP/202,20,r) - This works absolutely fine but the one down side is that when a call is answered by one phone, the second phone shows a missed call. If I put the extensions in a calling group, would this be resolved or is the best way of handling this to have the phone not display missed calls (if thats possible) ? |
| 04:44.25 | michaelo | I don't know of a way to avoid the missed calls unless you disable the missed call list entirely |
| 04:45.36 | clyrrad | is there a way to move the ASTDB from one server to another? |
| 04:47.19 | Juggie | .. /var/lib/asterisk |
| 04:47.23 | Juggie | i think its astdb |
| 04:47.56 | Zand3r | michaelo: Thanks - I'll do that |
| 04:47.56 | clyrrad | yep i see an astdb file there |
| 04:48.06 | clyrrad | is it as simple as just copying that file over? |
| 04:48.12 | Juggie | file or directory, i dont remember which but it should be there |
| 04:48.16 | Juggie | yes. |
| 04:48.17 | clyrrad | its a file there |
| 04:48.26 | Juggie | that should be all there is to it |
| 04:48.27 | clyrrad | -rw-r--r-- 1 root root 8192 Mar 16 00:30 astdb |
| 04:48.34 | clyrrad | oh cool :D |
| 04:48.35 | Juggie | you will need to shut down asterisk first |
| 04:48.42 | clyrrad | before copy? |
| 04:48.48 | Juggie | hmmmmm |
| 04:48.59 | *** join/#asterisk tg (i=tg@2001:618:1a23:0:0:0:0:1) |
| 04:49.01 | Juggie | no, but on the destination system |
| 04:49.02 | clyrrad | prolly just on destination server need to shut down asterisk |
| 04:49.06 | Juggie | yah. |
| 04:49.06 | clyrrad | yea makes sense |
| 04:49.16 | clyrrad | thanks Juggie |
| 04:49.29 | *** join/#asterisk tg (i=tg@x-net.hu) |
| 04:52.32 | clyrrad | Juggie: you are infact correct it worked - thanks bud |
| 04:52.38 | Juggie | np |
| 04:57.55 | *** join/#asterisk intralanman (n=lanman@pool-71-253-253-149.nrflva.east.verizon.net) |
| 05:02.15 | *** join/#asterisk intralanman (n=lanman@pool-71-253-253-149.nrflva.east.verizon.net) |
| 05:04.52 | *** join/#asterisk asterisky1 (n=gimmesom@ip70-190-159-144.ph.ph.cox.net) |
| 05:06.15 | asterisky1 | Hi to everyone |
| 05:07.10 | asterisky1 | please I need some major help before I go to a mental hospital, I'm having problems with TDM400P dialout |
| 05:07.51 | *** join/#asterisk TheMahmoud (n=fake@unaffiliated/mahmoud) |
| 05:07.51 | asterisky1 | I have my dialplan to dial local area codes without a 1 in the front |
| 05:07.55 | TheMahmoud | hello |
| 05:08.04 | TheMahmoud | _any_ one uses the web interface of voicemail? |
| 05:08.09 | TheMahmoud | vmail.cgi |
| 05:08.32 | asterisky1 | how ever the Phone company thinks im dialing a one in front, please help!!! |
| 05:08.36 | *** join/#asterisk orkid (n=orkid@bas1-barrie18-1242376711.dsl.bell.ca) |
| 05:09.21 | asterisky1 | in the cli window I can see that its only dialing the 10 digits |
| 05:09.44 | JT | asterisky1: are you using asterisk of freepbx/trixbox? |
| 05:09.55 | asterisky1 | JT I tried both |
| 05:10.24 | *** join/#asterisk intralanman (n=lanman@pool-71-253-253-149.nrflva.east.verizon.net) |
| 05:10.58 | asterisky1 | Im in Arizona and im using qwest as my provider |
| 05:12.33 | asterisky1 | worst of all, sometimes the calls go thru, and the problem I bealive is not from qwest cause I skip the pbx and calls go thru fine |
| 05:13.16 | asterisky1 | Has someone ever run into this??? |
| 05:13.42 | JT | asterisky1: might be freepbx putting in the 1 |
| 05:14.52 | asterisky1 | Jt thanks for bringing that up, how ever I first got the problem just on asterisk that I built on my self, and had made it work many times |
| 05:15.33 | asterisky1 | and then thinking something I did problably was wrong, installed free pbx |
| 05:15.47 | asterisky1 | ans still had the same problem |
| 05:16.22 | JT | pastebin.ca zapata.conf and zaptel.conf |
| 05:16.34 | JT | and show us the dial command you are using for this |
| 05:16.39 | ezer | wich linux do you recomend ? |
| 05:16.56 | asterisky1 | so I used my broadvoice acct and it worked, but not thru the tdm |
| 05:17.09 | flenders | ezer: any linux would do... pick the one you feel more confortable with |
| 05:17.25 | ezer | ok |
| 05:17.36 | *** join/#asterisk intralanman (n=lanman@pool-71-253-253-149.nrflva.east.verizon.net) |
| 05:17.40 | flenders | ezer: I would stay away of the desktop ones, though... too much stuff that you will never use |
| 05:18.16 | ezer | to windows users, wich would be a more friendly interface? i heard about kubuntu |
| 05:18.19 | asterisky1 | exten => _NXXXXXX,1,Dial(Zap/g1/${EXTEN}) |
| 05:18.36 | flenders | ezer: mate, I think you're on the wrong channel |
| 05:18.43 | ezer | :( |
| 05:18.45 | asterisky1 | and exten => _NXXNXXXXXX,1,Dial(Zap/g1/${EXTEN}) |
| 05:18.52 | ezer | why |
| 05:19.05 | flenders | ezer: try #linux first |
| 05:19.10 | JT | asterisky1: ok so they seem okay |
| 05:19.21 | flenders | we can't tell you of a more user friendly distro |
| 05:19.50 | ezer | ok oki wont ask any more question about linux.. i am intrested im asterisk.. lunux only as a tool to install it |
| 05:19.56 | asterisky1 | yeah im going crazy, seems no one has encounter thsi problem yet |
| 05:21.04 | JT | asterisky1: so done the pastebin yet? |
| 05:21.27 | *** join/#asterisk intralanman (n=lanman@pool-71-253-253-149.nrflva.east.verizon.net) |
| 05:21.38 | asterisky1 | no JT how do I run that? |
| 05:22.12 | JT | ~pb |
| 05:22.29 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 05:22.39 | JT | i am simply asking you to let me read the entire contents of zapata.conf and zaptel.conf by putting it into pastebin |
| 05:24.11 | asterisky1 | ok got it |
| 05:24.44 | *** join/#asterisk CrashHD (n=crashhd@c-67-166-155-233.hsd1.ca.comcast.net) |
| 05:24.54 | asterisky1 | [channels] |
| 05:24.54 | asterisky1 | ; |
| 05:24.54 | asterisky1 | ; Default language |
| 05:24.54 | asterisky1 | language=en |
| 05:24.54 | asterisky1 | context=incoming |
| 05:24.55 | asterisky1 | group=1 |
| 05:24.57 | asterisky1 | pulsedialing=yes |
| 05:24.59 | asterisky1 | relaxdtmf=yes |
| 05:25.01 | asterisky1 | context=incoming |
| 05:25.03 | asterisky1 | signalling=fxs_ks |
| 05:25.05 | asterisky1 | ;callerid=asreceived |
| 05:25.07 | asterisky1 | callwaiting=no |
| 05:25.08 | errr | geez |
| 05:25.09 | asterisky1 | channel => 3-4 |
| 05:25.14 | errr | use a paste bin |
| 05:25.16 | Qwell | ~pb |
| 05:25.29 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 05:25.29 | *** join/#asterisk ping2921 (n=marc3234@206-248-128-178.dsl.teksavvy.com) |
| 05:25.29 | asterisky1 | sorry , |
| 05:25.32 | JT | asterisky1: stop it |
| 05:25.40 | JT | asterisky1: i clearly said pastebin.ca |
| 05:25.43 | asterisky1 | please ignore the pulse dialing and the relaxdtmf |
| 05:25.43 | ping2921 | is asterisk compatible with mysql 5.0? |
| 05:25.44 | JT | then said pastebin |
| 05:25.53 | JT | then showed the pastebin entry in jbot |
| 05:26.00 | JT | what part said to paste in here |
| 05:26.13 | errr | ping2921: yes, I use mysql 5 with mine |
| 05:26.18 | JT | i want the ENTIRE CONTENTS of both files |
| 05:26.25 | ping2921 | errrr- are you using mysql-addons? |
| 05:26.28 | JT | do not skip a single line feed |
| 05:26.31 | errr | ping2921: yes |
| 05:26.49 | errr | ping2921: although all I have enabled at this point is the cdr |
| 05:26.56 | errr | but it works just fine |
| 05:27.13 | ping2921 | are you using innodb tables or myisam? |
| 05:27.24 | errr | I use default mysql tables |
| 05:28.26 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 05:33.48 | asterisky1 | JT http://pastebin.ca/397060 |
| 05:34.20 | asterisky1 | sent both files zapata and zaptel |
| 05:34.33 | *** join/#asterisk zeeesh (i=zeeesh@202.38.55.125) |
| 05:34.36 | zeeesh | hi |
| 05:35.13 | *** join/#asterisk kuku5 (n=kuku5@c-71-201-219-72.hsd1.il.comcast.net) |
| 05:35.20 | kuku5 | Anyone looking for a dedicated server ? |
| 05:35.26 | JT | asterisky1: where's channel 1-2? |
| 05:36.18 | asterisky1 | there is not fxo's or fxs in slots 1-2 |
| 05:36.42 | asterisky1 | just have 2 fxo's on 3&4 |
| 05:36.51 | JT | hmm ok |
| 05:36.54 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:37.02 | JT | and those files are both the full files on pastebin? |
| 05:37.10 | asterisky1 | yes |
| 05:37.32 | JT | okay |
| 05:37.43 | JT | have you tried using a handset on your exchange line |
| 05:37.49 | JT | and listening in while asterisk dials |
| 05:37.54 | JT | to see if you can hear the 1? |
| 05:38.03 | *** join/#asterisk brussel_ (n=brussel@cpe-24-165-7-252.san.res.rr.com) |
| 05:38.15 | asterisky1 | never even knew that I can do that |
| 05:38.31 | asterisky1 | how can I do it? |
| 05:38.33 | JT | you should be able to |
| 05:38.49 | JT | put a handset on the line, pickup handset, put to ear, make call with asterisk |
| 05:38.56 | *** join/#asterisk onglipo (n=onglipo@122.167.121.182) |
| 05:39.41 | asterisky1 | ok!!! I will do that |
| 05:41.25 | *** join/#asterisk MACscr (n=MACScr@adsl-75-23-89-176.dsl.peoril.sbcglobal.net) |
| 05:41.28 | asterisky1 | if it does not, would it be that I need to make changes in the zonedata?? |
| 05:41.59 | MACscr | can anyone recommend a good US based voip provider with per minute pricing? |
| 05:42.09 | MACscr | im thinking about going with Teliax |
| 05:42.44 | JT | asterisky1: i don't know if that would do anything |
| 05:43.11 | *** join/#asterisk onglipo (n=onglipo@122.167.121.182) |
| 05:43.50 | *** join/#asterisk kavit (n=kavit@178.132.233.220.exetel.com.au) |
| 05:43.52 | asterisky1 | im also going to connect the pbx with another provider ex: Cox and see if I get the same error |
| 05:44.08 | kavit | hey any aussie ISDN gurus around? |
| 05:44.23 | JT | kavit: not sure about guru, but ask |
| 05:44.28 | onglipo | Hello; ChanSpy doesnt seem to work when I am doing a Record() on a channel - any ideas what's up? |
| 05:45.03 | onglipo | When not Recording all is well with ChanSpy |
| 05:45.13 | kavit | JT: simple question.... does the cable need to be cross over from the Powertel box to TEXXXP card? |
| 05:45.18 | kavit | JT: what pins? |
| 05:46.28 | JT | not as far as i'm aware |
| 05:46.33 | JT | never used powertel myself |
| 05:46.44 | JT | use a standard ethernet non-crossover cable |
| 05:46.54 | JT | make sure the jumper is set to E1 on the card |
| 05:46.58 | asterisky1 | thanks alot for your help JT if I see you tomorrow in this room I'll let you know how it went |
| 05:47.12 | JT | it uses pin 1,2,4,5 for your info |
| 05:47.30 | kavit | ah yeah.... i was aware of that |
| 05:47.37 | kavit | but I get a red alarm |
| 05:47.48 | kavit | might reload the drivers |
| 05:47.58 | JT | kavit: have you got other working pris you setup on asterisk? |
| 05:48.01 | tzafrir_laptop | I need some feedback on my changes to the zaptel README: |
| 05:48.07 | tzafrir_laptop | http://svn.digium.com/view/zaptel/branches/1.2/README?r1=837&r2=2311 |
| 05:48.45 | tzafrir_laptop | latest version: http://svn.digium.com/view/zaptel/branches/1.2/README?view=markup |
| 05:49.08 | kavit | JT: none that I can access from here.... i just use the settings from voip-info australian settings page |
| 05:49.09 | lokkju_wrk | could someone help me getting IDEFisk to connect to asterisk? I keep on getting registration timeout errors, though asterisk sees the incoming connection |
| 05:49.18 | kavit | JT: anf it has always worked |
| 05:49.24 | tzafrir_laptop | More specifically: what do I need to do to have a kernel source on various distributions? |
| 05:49.26 | JT | kavit: feel free to send your setup to pastebin.ca |
| 05:49.28 | JT | kavit: hrm ok |
| 05:49.30 | lokkju_wrk | I think the issue may have something to do with asterisk seeing the port as 4569 |
| 05:49.36 | kavit | JT: gimme a sec |
| 05:49.43 | JT | kavit: ring powertel maybe then |
| 05:50.01 | *** join/#asterisk rrrobert (n=rrobert@mbl-82-51-38.dsl.net.pk) |
| 05:50.06 | tzafrir_laptop | The description I put there for RedHats is not god enough |
| 05:50.45 | kavit | JT: alright shall do.... can you look at my config just in case.... |
| 05:51.26 | JT | kavit: ok |
| 05:51.28 | kavit | JT: zapata.conf ---> http://pastebin.ca/397073 |
| 05:51.57 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 05:52.17 | JT | kavit: i wouldn't play with echo stuff unless there's a problem |
| 05:52.28 | JT | but zaptel.conf is more important here |
| 05:54.23 | kavit | JT: gimme a sec |
| 05:57.47 | kavit | JT: zaptel config http://pastebin.ca/397081 |
| 05:59.34 | JT | kavit: well i guess that look right too |
| 05:59.44 | JT | kavit: i assume you don't have a pri crossover cable handy? |
| 06:00.06 | TheMahmoud | ah, fixed the web interface voicemail |
| 06:00.23 | TheMahmoud | i just had to type <account>@<context> rather than just <Account> |
| 06:00.26 | TheMahmoud | as the username |
| 06:00.48 | JT | kavit: is there multiple ports? |
| 06:00.52 | TheMahmoud | however, to make it easier, i edited the vmail.cgi file to default the context to something when not mentioned |
| 06:01.30 | kavit | JT: just the one |
| 06:01.53 | JT | te110p? |
| 06:03.37 | kavit | JT: aye |
| 06:05.25 | kavit | JT: I just used a standard cross over cable, i used a network straight cable... to no avail |
| 06:05.31 | *** join/#asterisk intralanman (n=lanman@pool-71-253-253-149.nrflva.east.verizon.net) |
| 06:06.34 | *** join/#asterisk arooni (n=chatzill@c-24-19-10-29.hsd1.wa.comcast.net) |
| 06:06.37 | arooni | hey everyone |
| 06:06.47 | arooni | how can i get the file asterisk.ctl |
| 06:06.53 | arooni | i get the error: Unable to connect to remote asterisk (does / var/run/asterisk.ctl exist?) |
| 06:09.12 | arooni | and when i run it like... asterisk -cvvvv |
| 06:09.15 | arooni | i get to the CLI just fine |
| 06:11.17 | orlock | um |
| 06:11.18 | *** join/#asterisk intralanman (n=lanman@pool-71-253-253-149.nrflva.east.verizon.net) |
| 06:11.20 | *** join/#asterisk k-man (n=jason@unaffiliated/k-man) |
| 06:11.21 | orlock | (does / var/run/asterisk.ctl exist?)? |
| 06:11.31 | orlock | with the space? |
| 06:11.53 | JT | kavit: umm |
| 06:12.05 | JT | kavit: do you have a PRI crossover cable? |
| 06:12.14 | JT | kavit: they are totally different to network crossover cables |
| 06:12.25 | JT | ethernet crossover cables are useless for T1s and E1s |
| 06:12.33 | arooni | orlock: huh? |
| 06:14.19 | *** part/#asterisk MACscr (n=MACScr@adsl-75-23-89-176.dsl.peoril.sbcglobal.net) |
| 06:15.35 | *** join/#asterisk intralanman (n=lanman@pool-71-253-253-149.nrflva.east.verizon.net) |
| 06:15.37 | *** join/#asterisk ComputerGuru (n=Computer@81.10.82.188) |
| 06:16.24 | ComputerGuru | Hi guys |
| 06:17.13 | ComputerGuru | Quick question: Does anyone know if with VoIP it's possible to PLACE a phone call then RECEIVE key presses back? |
| 06:17.43 | JT | should be |
| 06:17.45 | ComputerGuru | I know I can set up a server that gets called and accepts key presses as menu selections, but I need the opposite: to place a call and have users do something with the menu. |
| 06:17.49 | JT | depends on your provider i guess |
| 06:17.58 | JT | some are crappy with inbound dtmf |
| 06:18.18 | ComputerGuru | JT_: I guess I'll have to create my own client for this though? |
| 06:18.35 | ComputerGuru | *JT |
| 06:18.36 | JT | no. |
| 06:19.16 | ComputerGuru | existing clients like iaxComm can receive incoming DTMF? |
| 06:19.45 | JT | sorry, you're not being clear enough on what you want to do? |
| 06:20.34 | *** join/#asterisk slinky (n=slinky@bent1.dsl.xmission.com) |
| 06:21.06 | ComputerGuru | I'd like to call someone, have the client play a wav/mp3 with a list of menu options. |
| 06:21.24 | ComputerGuru | then i need to be able to let the person i'm calling press a key in response |
| 06:21.43 | JT | sorry |
| 06:21.47 | JT | you want to call someone |
| 06:21.54 | JT | then you want THEM to play you a recording? |
| 06:22.03 | ComputerGuru | no, i want me to play them a recording |
| 06:22.06 | JT | then let them answer their own recording with dtmf? |
| 06:22.11 | ComputerGuru | no |
| 06:22.14 | ComputerGuru | answer my recording |
| 06:22.20 | JT | ComputerGuru: oh, you said have the client play a wav/mp3 |
| 06:22.26 | JT | hence confusion |
| 06:22.43 | ComputerGuru | sorry, i meant client as in the program on my pc. it's not server since it's making calls, not receiving them i guess... |
| 06:23.15 | JT | what you want is for your server to call someone, play the callED party a recording, then give them an option of responding with dtmf |
| 06:23.30 | ComputerGuru | yeah :) |
| 06:23.46 | orlock | arooni: the error you pasted had a space in the path. |
| 06:24.03 | arooni | hey JT >> any clue on these types of errors: 1) Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory 2) Unable to bind socket to /var/run/asterisk/asterisk.ctl: No such file or directory |
| 06:24.21 | JT | sounds like there's no such file or directory |
| 06:24.22 | orlock | arooni: ahh, no space then. |
| 06:24.23 | arooni | orlock yes i know that ... i pasted wrong |
| 06:24.35 | arooni | JT >> well i just created a directory |
| 06:24.37 | arooni | but do i need a file there |
| 06:24.50 | arooni | or is this like a temp file that asterisk will create |
| 06:25.51 | ComputerGuru | JT: Assuming of course that I send the call via a VoIP provider from my Asterisk server... Like I have Vonage and I place the call over their SIP network. |
| 06:27.19 | tzafrir_laptop | ls -l /var/run/asterisk |
| 06:27.30 | tzafrir_laptop | maybe it does not exist |
| 06:27.35 | JT | ComputerGuru: yes well it depends on provider network and making sure dtmf is setup up right |
| 06:27.48 | JT | ComputerGuru: vonage is pretty crap by the way |
| 06:28.23 | tzafrir_laptop | or maybe not owned by the asterisk user |
| 06:28.49 | ComputerGuru | JT: As far as quality and features, do you have a particular provider you would recommend? |
| 06:29.15 | JT | ComputerGuru: not really, a few others will have their recommendations |
| 06:29.24 | ComputerGuru | Just going over the checklist, so bear with me please :) |
| 06:29.28 | JT | i'm in australia so will only give actual recommendations here |
| 06:29.37 | ComputerGuru | oh ok |
| 06:30.05 | ComputerGuru | Asterisk can place calls via existing VoIP networks (other than it's own) ---- check |
| 06:30.54 | ComputerGuru | Asterisk can be configured to _place_ a call and accept menu choices via DTMF --- check |
| 06:32.27 | arooni | how can i dot this: Make sure the server running the process that uses callInitiate has file write access to the server running your Asterisk process. This means mapping a drive from your Asterisk server to your Railsserver and making sure the wakeup directory and outgoing directories are writeable to i |
| 06:32.43 | arooni | first of all... where is asterisk's 'wakup' directory |
| 06:32.53 | arooni | i know where the outgoing is |
| 06:34.26 | ComputerGuru | are there any commercial providers that use IAX? |
| 06:34.54 | arooni | ComputerGuru: i'm looking for an unlimited SIP provider |
| 06:35.01 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 06:35.03 | arooni | if you find one let me know.. i want unlimited phone calls for like $15/mo |
| 06:35.13 | *** join/#asterisk geoaxis (n=geoaxis@unaffiliated/geoaxis) |
| 06:35.43 | dlynes_laptop | whoot |
| 06:35.48 | dlynes_laptop | voip-info.org is back up :) |
| 06:35.48 | geoaxis | hello ... I am trying to configure SNMP for asterisk 1.4 ...nedd pointers (there is only one tutorial on the net and its not working for me ) |
| 06:35.53 | geoaxis | yup |
| 06:36.10 | dlynes_laptop | probably because the one tutorial on the net is for snmp on asterisk 1.2 |
| 06:36.15 | dlynes_laptop | not 1.4 |
| 06:36.40 | geoaxis | dlynes_laptop: there was no support for SNMP before 1.4 |
| 06:36.46 | ComputerGuru | arooni: logically, you can't use unlmited phone calls |
| 06:36.47 | geoaxis | 1.4 is the first to use SNMP |
| 06:36.49 | dlynes_laptop | sure there was |
| 06:36.52 | dlynes_laptop | Just not official |
| 06:36.58 | ComputerGuru | so just get one with either a big plan or really cheap rates |
| 06:36.58 | dlynes_laptop | res_snmp.so |
| 06:37.19 | geoaxis | well no use of hidden pieces of code ..(its as good as M$ then_ |
| 06:37.20 | orlock | bwhahah |
| 06:37.29 | orlock | somebody here left early to pick up his daughter |
| 06:37.33 | orlock | and is now trapped in the lift |
| 06:38.00 | clyrrad | SIP NOTIFY messages are sent on 5060 right? |
| 06:38.11 | clyrrad | by default i mean.... |
| 06:38.12 | orlock | clyrrad: generally yeah |
| 06:38.12 | dlynes_laptop | clyrrad: yes, unless you've specified an alternate port |
| 06:38.17 | arooni | ComputerGuru: right... but i guess i'm saying i dont want to pay per call... i want to pay for an 'unlimited' number of calls of which i wont be able to use htem all |
| 06:38.30 | clyrrad | alright - so if a phone is not responding to that message it means that port is blocked? |
| 06:38.37 | arooni | if i need to call a specific number 5 times in a row.... whats the best way of handling that? |
| 06:38.39 | clyrrad | actually I should say multiple phones |
| 06:39.34 | *** join/#asterisk lorinc (n=ang@pool-2896.adsl.interware.hu) |
| 06:39.35 | clyrrad | there are groups of phones in different locations, in one location no phones respond to SIP NOTIFY commands |
| 06:39.48 | orlock | clyrrad: tcpdump is your freind |
| 06:39.59 | clyrrad | wondering if it means that 5060 is blocked, but it if was, then the phones would not ring at all correct? |
| 06:40.25 | geoaxis | baaaahhh SNMP |
| 06:45.56 | *** join/#asterisk aaronr (n=arussell@87.127.234.100) |
| 06:49.25 | *** join/#asterisk sabakas1 (n=solapus@66.90.121.129) |
| 06:49.31 | dlynes_laptop | hrm...looks like there's been a few extremely useful patches in asterisk lately |
| 06:50.30 | drray | me no update asterisk |
| 06:50.54 | drray | me no update kernel |
| 06:51.24 | drray | me no nothing now that wiki broke |
| 06:52.08 | dlynes_laptop | drray: wiki's not broke, foo |
| 06:53.47 | drray | I'm going to send them some money |
| 06:54.04 | drray | it's a great resource that I did not know how much I needed until it was gone |
| 06:55.45 | arooni | has anyone messed with RAGI + asterisk here? |
| 06:56.03 | arooni | has anyone messed with RAGI + asterisk here? .. if so have you successfully set up a call handler? i'm having a bit of trouble |
| 06:58.28 | `mw | i havent used ragi but i've used ruby-agi |
| 06:58.49 | arooni | hm... i cant seem to make my call handlers work |
| 06:58.54 | arooni | any suggestions? |
| 07:04.37 | `mw | not especially, unless you are having some sort of ruby problem |
| 07:05.41 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 07:05.43 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:20.48 | *** join/#asterisk kev009 (n=kev009@ip70-162-43-70.ph.ph.cox.net) |
| 07:21.04 | kev009 | what's a good linux softphone for asterisk? |
| 07:21.31 | dlynes_laptop | How about kphone? |
| 07:21.40 | dlynes_laptop | Or Ekiga? |
| 07:21.51 | kev009 | will give them a try, thanks |
| 07:22.24 | arooni | if i want to redial a number 10 times |
| 07:22.43 | arooni | but cancel if user picks up.... can that just be number of retries |
| 07:23.55 | tengulre | the same problem: what's a good windows softphone for asterisk? (source code) |
| 07:26.35 | *** join/#asterisk freed0m (n=root@121.210.208.202) |
| 07:27.10 | *** part/#asterisk ComputerGuru (n=Computer@81.10.82.188) |
| 07:27.21 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 07:29.14 | dlynes_laptop | tengulre: try snom360 softphone |
| 07:29.18 | dlynes_laptop | tengulre: www.snom.de |
| 07:29.24 | freed0m | I have a question.. If i wanted people to be able to call my * box, enter an invoice number, and then have it tell them details about an order, like a status for example "waiting on parts" and how long we expect it to take etc, how would i go about doing that? |
| 07:29.30 | dlynes_laptop | tengulre: it's free for non-commercial use, and is not opensource |
| 07:31.05 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 07:32.27 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 07:34.09 | *** join/#asterisk vasquez (n=vasquez@85.183.64.6) |
| 07:35.36 | tengulre | dlynes_latop: thanks! |
| 07:35.38 | *** join/#asterisk aiksa[LV] (n=aiksa[LV@83.223.131.104) |
| 07:35.44 | aiksa[LV] | morning |
| 07:35.56 | aiksa[LV] | voip-info is back! |
| 07:36.12 | dlynes_laptop | very good :) |
| 07:36.13 | tengulre | I want to developt myself, but I don't known how to select a best SIP or IAX2 protocol stack? |
| 07:36.24 | dlynes_laptop | tengulre: sofa |
| 07:37.46 | dlynes_laptop | oops |
| 07:37.49 | dlynes_laptop | sofia i meant |
| 07:37.55 | dlynes_laptop | http://sofia-sip.sf.net/ |
| 07:38.56 | *** part/#asterisk infi (n=infi@about/linux/staff/infi) |
| 07:39.29 | aiksa[LV] | dlynes_laptop: dont pet the messenger |
| 07:39.47 | dlynes_laptop | aiksa[LV]: lol |
| 07:39.50 | aiksa[LV] | just wanted to let the crowd know, if you dont already |
| 07:40.00 | dlynes_laptop | tengulre: you can also try iaxclient.sf.net for an iax2 library |
| 07:41.09 | tengulre | dlynes_laptop: but the iaxclient can by use for visual studio c++/VB....delphi...? |
| 07:41.30 | dlynes_laptop | tengulre: i have no idea...I never do any windows development |
| 07:42.23 | dlynes_laptop | tengulre: i think if you're wanting to do windows development, and you're after a library, you're probably on your own |
| 07:43.00 | tengulre | dlynes_laptop: thank you give my ideas. |
| 07:43.51 | aiksa[LV] | http://iaxclient.sourceforge.net/ have libraries which can be used from visual studio |
| 07:44.20 | aiksa[LV] | at least I know a guy who does in one of our projectsa |
| 07:45.47 | tengulre | aiksa[LV]: did you successful use it for vs? |
| 07:47.22 | aiksa[LV] | as I said -- not me, but a guy on a project where there are two companies working for |
| 07:47.45 | aiksa[LV] | I pointed him to this library, and as far as i know he use it in his developments |
| 07:48.17 | aiksa[LV] | though I am not 100% sure whether he used VB, perhaps it was C++ from visual studio. |
| 07:48.51 | aiksa[LV] | nevertheless he told that the library was an easy to tuse. |
| 07:49.01 | aiksa[LV] | to use, sorry |
| 07:55.52 | `mw | anyone have problems with iax peers going unreachable in iax2 show peers but are still pingable? i have 3/5 so far that seem to be stuck in this state |
| 07:59.59 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:03.23 | arooni | anyone know how to tackle this: Unable to re-open DSP device /dev/dsp: Device or resource busy |
| 08:03.57 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 08:04.56 | kev009 | I'm trying TrixBox just for testing my new digi 400P card. I have a single FXO set up as a trunk, and a software SIP phone. The soft phone talks to asterisk fine, but I get "all circuits are busy now" when dialing out |
| 08:06.03 | fordfrog | hi, what is the way to dial both local extension and cell phone on incoming call? |
| 08:06.05 | arooni | god damn mycall files arent working |
| 08:11.47 | *** join/#asterisk shinux__ (n=shinux@196.201.159.106) |
| 08:15.13 | *** join/#asterisk Kapsel (i=kapsel@62.242.240.33) |
| 08:15.37 | drray | boy, the 7960's really dropped in price |
| 08:19.35 | *** join/#asterisk koma (n=koma@host147-80-static.17-80-b.business.telecomitalia.it) |
| 08:19.43 | koma | ji all |
| 08:21.41 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 08:22.33 | *** join/#asterisk arooni (n=chatzill@dsl081-163-148.sea1.dsl.speakeasy.net) |
| 08:22.41 | arooni | hey folks |
| 08:22.44 | *** join/#asterisk michael-i (n=michael-@Lc11f.l.pppool.de) |
| 08:23.00 | arooni | i'm having *loads of touble* with my call files... from asterisk command line it seems everyone is working : Mar 16 01:21:16 NOTICE[6034]: pbx_spool.c:279 attempt_thread: Call completed to SIP/proxy01.sipphone.com/14255331234 |
| 08:23.04 | arooni | but nothing happens |
| 08:24.29 | drray | are you moving them or writing them in the directory? |
| 08:24.46 | arooni | drray: moving |
| 08:26.11 | arooni | dday << it looks like i have some other issues http://pastie.caboo.se/47336 |
| 08:26.19 | arooni | drray: i mean... what do you think of those ? |
| 08:27.06 | *** join/#asterisk vgster (n=vgster@81.96.139.59) |
| 08:28.39 | drray | mismatched codecs? |
| 08:29.26 | arooni | drray: but would that prevent me from making an outbound call? |
| 08:29.37 | arooni | like i showed you by moving it to outgoing/ |
| 08:30.53 | drray | I'm not competent to answer your questions |
| 08:31.52 | *** join/#asterisk Ahrimanes (n=ma@81.7.159.2) |
| 08:33.04 | sbingner | arooni, you have no ilbc codec |
| 08:33.10 | sbingner | use gsm or ulaw and it'll probably work |
| 08:34.53 | *** join/#asterisk dseeb_ (n=dcb@58.169.152.56) |
| 08:35.58 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@c-67-188-233-116.hsd1.ca.comcast.net) |
| 08:39.25 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 08:43.43 | koma | <PROTECTED> |
| 08:43.43 | koma | <PROTECTED> |
| 08:43.43 | koma | <PROTECTED> |
| 08:43.43 | koma | <PROTECTED> |
| 08:43.44 | koma | <PROTECTED> |
| 08:44.46 | aiksa[LV] | what card do you have for analogue lines? |
| 08:45.05 | *** join/#asterisk TeleTommy (n=chatzill@p54a8bcc8.dip0.t-ipconnect.de) |
| 08:49.11 | *** join/#asterisk topping (n=topping@204.152.96.238) |
| 08:49.36 | *** join/#asterisk macTijn (i=martijn@linda.net.insecure.nl) |
| 08:50.11 | *** join/#asterisk skirmisha (n=viki@87-126-55-7.btc-net.bg) |
| 08:50.17 | skirmisha | hello guys |
| 08:50.33 | skirmisha | does anyone know if there is command for unregister a peer |
| 08:51.21 | sbingner | skirmisha, sip prune? |
| 08:51.33 | skirmisha | ? |
| 08:51.35 | sbingner | I think it makes it so they can't re-register till you sip relod tho not sure |
| 08:52.05 | sbingner | nah it looks like you can still re-register |
| 08:54.37 | *** join/#asterisk lokkju_wrk_ (n=lokkju@unaffiliated/lokkju) |
| 08:54.59 | skirmisha | that is ok |
| 08:55.12 | skirmisha | but sip prune says this peer is not real |
| 08:55.20 | skirmisha | or something like that |
| 08:56.19 | skirmisha | is not a Realtime peer, cannot be pruned. |
| 08:56.28 | skirmisha | do u get same error |
| 08:57.42 | skirmisha | damn |
| 08:57.56 | skirmisha | what asterisk is that if u don;t have control over peers |
| 09:00.17 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 09:01.47 | Mahmoud | lol |
| 09:02.09 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 09:02.21 | Mahmoud | democratic american asterisk |
| 09:02.33 | Mahmoud | where peers say "no" to asterisk |
| 09:03.02 | Mahmoud | otherwise, they would dial 911 |
| 09:03.44 | *** join/#asterisk lokkju_wrk (n=lokkju@unaffiliated/lokkju) |
| 09:04.09 | Mahmoud | disable his account? |
| 09:08.12 | koma | aiksa[LV] what card do you have for analogue lines? |
| 09:08.23 | dseeb_ | ~ seen voipy |
| 09:09.04 | jbot | voipy <n=pirch@a81-84-60-131.cpe.netcabo.pt> was last seen on IRC in channel #asterisk, 1d 11h 12m 42s ago, saying: 'Does anyone use Chan_cellphone and knows how to solve the bluetooth pairing prob on bluez-utils 3.7-1?'. |
| 09:09.04 | koma | i've an.. |
| 09:09.04 | koma | aiksa[LV] what card do you have for analogue lines? |
| 09:09.05 | koma | 01:0a.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
| 09:09.11 | koma | with wct6dm |
| 09:09.13 | koma | with wctdm |
| 09:09.17 | koma | module |
| 09:09.44 | koma | TDM400P |
| 09:10.32 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 09:10.35 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 09:14.35 | JT | koma: why repeat 3 times to someone not even here? |
| 09:15.04 | koma | to reply to a question :) |
| 09:19.29 | *** join/#asterisk topping (n=topping@204.152.96.238) |
| 09:19.44 | koma | JT can you help me please :| |
| 09:21.30 | koma | no eh? :° |
| 09:22.28 | JT | haven't had much experience with fax |
| 09:22.41 | JT | what do you have it doing to try and handle the fax? |
| 09:30.48 | *** join/#asterisk Exhar (n=Roy@84-105-192-215.cable.quicknet.nl) |
| 09:32.22 | koma | i need to handle a fax, |
| 09:33.09 | koma | but i can't see context-fax for the channel that i need |
| 09:33.29 | koma | ...so i have configured 1-31 for E1 |
| 09:33.42 | koma | and 32 33 34 35 for the analogic card |
| 09:34.04 | koma | but if i do zap show channels i see from 1-31 |
| 09:34.09 | koma | and not 32 33 34 35 |
| 09:48.48 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 09:51.34 | *** join/#asterisk sumasuma (n=kurukko@61.14.86.23) |
| 09:52.04 | sumasuma | i want to have video phone working with asterisk? it has problem in not passing few parameters in the SDP |
| 09:52.12 | sumasuma | anyother way to accomplish that ? |
| 09:52.58 | sumasuma | vp1 -> (all SDP parameters) asterisk -> (asterisk eaten few SDP parameters rest here) vp2 |
| 09:53.02 | sumasuma | so the video is poor |
| 09:53.26 | sumasuma | asterisk handles initially only audio call, once the extension is done, it handles the video |
| 09:53.29 | sumasuma | with a reinvite |
| 09:55.58 | *** join/#asterisk af_ (n=getsmart@ip-202-133.sn2.eutelia.it) |
| 09:57.25 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 09:58.21 | *** join/#asterisk [shodan] (n=shodan@ip073.99-113-216.pppoe4.joliette.intermonde.net) |
| 10:02.41 | *** join/#asterisk doctorzoidberg (n=doctorzo@85.20.86.178) |
| 10:04.19 | doctorzoidberg | hi everybody |
| 10:04.30 | sumasuma | hi |
| 10:05.08 | doctorzoidberg | is there a way to force asterisk to don't hangup even if nobody answer ? |
| 10:05.30 | *** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com) |
| 10:06.09 | sumasuma | you mean on Dial ? |
| 10:06.12 | doctorzoidberg | yep |
| 10:06.25 | sumasuma | asterisk will hangup if you don't have further instructions |
| 10:06.45 | sumasuma | show application dial on the cli will give you more options |
| 10:06.57 | doctorzoidberg | thanks |
| 10:07.09 | *** join/#asterisk rrrobert (n=rrobert@mbl-82-51-38.dsl.net.pk) |
| 10:08.20 | doctorzoidberg | my problem here is that asterisk doesn't detect the answer |
| 10:08.39 | doctorzoidberg | so even if the remote number answers the call, asterisk still hangup after 20 seconds |
| 10:08.50 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:09.48 | sumasuma | oh |
| 10:09.54 | sumasuma | is it an analog line ? |
| 10:10.03 | rrrobert | i am new to asterisk. i have an asterisk running. and some sip phones are also running on it.. what useful information can i get from the asterisk, regarding calls and other proceses.. |
| 10:10.14 | rrrobert | plz guide me.. i am newby |
| 10:10.25 | sumasuma | rrrobert: please http://www.voip-info.org |
| 10:11.20 | doctorzoidberg | sumasuma, yes, with zaptel module |
| 10:11.33 | sumasuma | doctorzoidberg: It is not problem with the zaptel module. |
| 10:12.14 | sumasuma | doctorzoidberg: when you call someone in the normal analog world, whoever calling should hangup first for the hangup completely |
| 10:12.35 | sumasuma | doctorzoidberg: it is not a bug, it came from legacy system for proper reasons |
| 10:12.45 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 10:16.00 | doctorzoidberg | sumasuma, the problem is that asterisk can't detect if the remote host answered (only on external -analog- calls) |
| 10:16.19 | doctorzoidberg | if I call an internal sip, iax or sccp phone it works |
| 10:17.38 | sumasuma | if the analog line is answered, asterisk is connecting the call to it ? |
| 10:17.51 | doctorzoidberg | yes |
| 10:18.05 | sumasuma | zaptel modules properly configured ? |
| 10:18.52 | doctorzoidberg | i hope |
| 10:18.57 | doctorzoidberg | i'll double-check it |
| 10:19.51 | sumasuma | how you checked it ? |
| 10:22.25 | doctorzoidberg | sumasuma, http://rafb.net/p/GffsGq42.html |
| 10:22.36 | doctorzoidberg | two answered calls |
| 10:22.49 | doctorzoidberg | that's _ODD_ |
| 10:23.48 | doctorzoidberg | the first call was answered by an automatic message from the mobile operator |
| 10:23.53 | doctorzoidberg | the second by myself |
| 10:24.55 | sumasuma | zaptel modules properly configured ? |
| 10:25.06 | sumasuma | check whether zaptel if configured for asterisk ? |
| 10:29.41 | *** join/#asterisk Ciber311 (n=Ciber311@user-12ld42j.cable.mindspring.com) |
| 10:33.26 | doctorzoidberg | sumasuma, yes, that's ok. I increased the dial() timeout just to check |
| 10:33.42 | doctorzoidberg | and discovered that the answer signal arrives late |
| 10:34.02 | doctorzoidberg | something like 10-15 seconds after the real answer |
| 10:34.29 | *** join/#asterisk inspired (n=mikael@85.221.7.59) [NETSPLIT VICTIM] |
| 10:35.00 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) [NETSPLIT VICTIM] |
| 10:35.37 | sumasuma | good |
| 10:35.43 | sumasuma | nice to see you got it worked |
| 10:37.18 | doctorzoidberg | it used to work before I had to replace the voip server (distro change, slackware to gentoo) |
| 10:40.33 | christo | I'm having problems getting status back from an Originate - can anybody see a solution to this: http://pastebin.ca/397228 ? |
| 10:42.29 | puzzled | hi |
| 10:43.21 | doctorzoidberg | btw, thanks sumasuma |
| 10:49.44 | *** join/#asterisk locodice (n=dinheiro@201-35-250-56.fnsce703.dsl.brasiltelecom.net.br) |
| 10:50.30 | *** join/#asterisk badcfe (n=cso@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) |
| 10:54.54 | *** part/#asterisk locodice (n=dinheiro@201-35-250-56.fnsce703.dsl.brasiltelecom.net.br) |
| 10:55.21 | *** join/#asterisk adde (n=adde@tn-84-218-52-159.dsl.bredbandsbolaget.se) |
| 10:56.26 | adde | Could someone please post me the example script supplied with asterisk to create submenu ivr? please... |
| 10:56.41 | christo | adde - use AGI |
| 10:58.22 | viperdude | has anyone ever had trouble with DTMF detection using PHPAGI? For instance if I try to detect 9 digits its ok but try 16 digits such as a credit card number it randomly drops some of the digits with no detectable pattern |
| 10:58.55 | *** join/#asterisk coppice (n=chatzill@90.203.17.210.dyn.pacific.net.hk) |
| 11:01.17 | rrrobert | i just wanna know which sip variables can i monitor, sip functionality guide, such that number of calls.. etc |
| 11:06.21 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:06.29 | *** join/#asterisk marexz (n=marexz@marexz.mil.lv) |
| 11:08.34 | viperdude | chriso: not sure about your orignate problem but have you managed to interface into Yahoo? it looks like that from your Dial string |
| 11:10.25 | adde | Could you recomend a Cheap UK SIp provider...anyone? |
| 11:11.06 | christo | viperdude - that's just a name for the iax trunk - it could equally well say 'potato' if that how I set it up at the other end |
| 11:11.07 | HarryR | adde, voiptalk.org |
| 11:11.19 | koma | what's the simplest way to configure a Card? |
| 11:13.28 | *** join/#asterisk avn (n=avn@daemon.hole.ru) |
| 11:16.38 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 11:17.34 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 11:19.50 | viperdude | chriso: yes i realise that, what I am asking is have you interfaced with Yahoo? |
| 11:22.10 | christo | lol no! |
| 11:22.12 | avn | Hello all, I have a small question... I have h323 terminal (hardware), gnugk and asterisk -- how I can redirect all calls from gatekeeper to asterisk? |
| 11:30.35 | *** part/#asterisk gelatinous (n=elanda@enron.xilogix.net) |
| 11:32.49 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:33.35 | *** join/#asterisk macTijn (i=martijn@linda.net.insecure.nl) |
| 11:34.22 | *** join/#asterisk sergee (i=opera@195.94.224.197) |
| 11:35.28 | *** join/#asterisk gr1ncheux (n=devine@unaffiliated/gr1ncheux) |
| 11:36.58 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 11:40.39 | sergee | i'm trying to use MeetMe (ztdummy, 2.6.16.21-0.13-smp), but when i have more then 2 people in the same conference room, i have quality issues: echo and cracks... zttest shows pretty good (imho) results: Best: 99.975586 -- Worst: 99.938965 -- Average: 99.960068, |
| 11:41.11 | sergee | are there any way to imrove quality/stability? in which direction should i dig? |
| 11:41.24 | mosty | sergee: in my experience, ztdummy sucks badly :( |
| 11:41.44 | *** join/#asterisk ComaVN (n=blaargh@unaffiliated/comavn) |
| 11:44.27 | sergee | mosty: are there any software replacement? |
| 11:44.57 | mosty | not in standard asterisk, that i know of |
| 11:45.14 | mosty | there is an asterisk fork that uses posix timers instead |
| 11:45.23 | sergee | OpenPBX? |
| 11:45.36 | mosty | yeah |
| 11:46.01 | sergee | mosty: did you try it in production? |
| 11:46.10 | mosty | no, not yet |
| 11:46.18 | sergee | mosty: is it better then meetme + ztdummy? |
| 11:46.32 | sergee | ahhh.. ok :) then i'll take a look |
| 11:46.49 | florz | mosty: OpenPBX really uses posix timers instead of zaptel timers, even if you do have zaptel hardware available? |
| 11:47.03 | mosty | let me know how it goes- because i really hate ztdummy |
| 11:47.13 | mosty | florz: not sure |
| 11:59.15 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 11:59.57 | FreezeS | I've got a problem with ChanIsAvail. It actually registers a user to the server, adding his id to regcontext |
| 12:00.12 | FreezeS | is this the correct way it should function ? |
| 12:01.54 | adde | Could somone just help me with basics so i have something to start with and go on from there... i want to start play the sound hello-world - wait for an 1-2-3 and play a different sond depending on what key is pressed... All examples i find are complex gotos which is overkill for my initial step of learning :) |
| 12:02.43 | *** join/#asterisk chrisknight (n=explodin@24.211.64.17) |
| 12:02.44 | *** join/#asterisk Ciber311 (n=Ciber311@user-12ld42j.cable.mindspring.com) |
| 12:03.06 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 12:03.09 | mosty | adde: three lines in your dialplan, one for each soundfile, and one for the pause. no gotos required |
| 12:03.39 | mosty | er, it will be more than three lines if you want a decision |
| 12:04.19 | mosty | use the read command, and create extensions for each option |
| 12:04.21 | adde | i have done this with a simple context but that only works for a phone in the context... i want this to my dialup context |
| 12:04.30 | adde | incoming calls context |
| 12:04.44 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 12:04.57 | adde | my incoming calls automaticly gets extension 1000 |
| 12:05.22 | mosty | well fix your incoming context then |
| 12:06.13 | adde | so if i put a sound with that conmtext i get that to play... but if i want it to play a sound saying press one for blahh 2 for blabla...wait for a button to pressed... thats what i havent succeeded in |
| 12:07.52 | giasai68 | hello, |
| 12:08.18 | giasai68 | i have some problem to authenticate a call in sip i got authentacated failed |
| 12:08.32 | giasai68 | can you hel me? |
| 12:09.22 | *** join/#asterisk dj-fu (i=ajc@202-74-195-152.ue.woosh.co.nz) |
| 12:18.16 | *** join/#asterisk HexDump (n=a@31.Red-217-126-215.staticIP.rima-tde.net) |
| 12:18.18 | HexDump | hi all |
| 12:20.41 | HexDump | I would like someone to tell me if I could use asterisk to log problems in my voip configuration, I mean, connection getting saturating (prudcing cuts in voice, etc...), statistics like maximun peek of kb consumed, etc.... It would be really nice. |
| 12:21.05 | adde | exten => 1000,3,WaitExten |
| 12:21.05 | adde | exten => 1,1,Playback(tt-monkeys) |
| 12:21.05 | adde | exten => 2,1,Playback(hello-world) |
| 12:21.12 | adde | what have i done wrong here |
| 12:29.10 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 12:29.25 | *** join/#asterisk tparcina (n=tparcina@cisco16.fesb.hr) |
| 12:29.27 | *** join/#asterisk step_quasar (n=step_qua@191-91-235-201.fibertel.com.ar) |
| 12:29.37 | tparcina | hi channel |
| 12:31.08 | tparcina | when asterisk calls someone, and that person pick's up the phone, I need to play message and lead called person thrue AA menu |
| 12:31.32 | tparcina | any sugestions, how to do that? how to play mesage when other party pick's up the phone? |
| 12:34.24 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:39.07 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 12:40.10 | mosty | adde: i find Read more reliable than WaitExten and Background |
| 12:42.04 | *** join/#asterisk friedrich| (n=friedric@e177252097.adsl.alicedsl.de) |
| 12:45.08 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 12:45.17 | *** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) |
| 12:46.00 | adde | most, could you give me a reall simpel example that i can go on with...? I really am stuck. |
| 12:48.12 | *** join/#asterisk MarkWD (n=MarkWD@rrcs-67-78-88-186.sw.biz.rr.com) |
| 12:48.37 | mosty | set debug 10 and set verbose 10 in asterisk |
| 12:48.52 | mosty | then test it and see what it's doing |
| 12:49.01 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-185-4.buckeyecom.net) |
| 12:49.57 | tparcina | how to play sound when called party picks up the phone? |
| 12:50.46 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:50.55 | mosty | tparcina, why would you want to do that? |
| 12:51.10 | adde | Telemarketing...hehe |
| 12:51.27 | adde | I hate those systems |
| 12:51.44 | mosty | i setup an asterisk box to break those systems |
| 12:52.03 | adde | Breakl in what way? |
| 12:52.10 | adde | Spamm back? |
| 12:52.32 | viperdude | tparcina: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial |
| 12:52.43 | viperdude | tparcina: check the A(x) option |
| 12:52.50 | mosty | no, it just answers and asks the callee to dial 1 if they are not a telemarketer |
| 12:53.14 | mosty | and plays hold music for the callee |
| 12:53.15 | adde | ahh... |
| 12:53.18 | viperdude | mosty: Coem to the UK and join the TPS |
| 12:53.29 | mosty | what's tps? |
| 12:53.32 | viperdude | Come |
| 12:53.51 | viperdude | Telephone Preference Service... register your number and no more telemarketers |
| 12:54.14 | adde | viperdude: we have that in sweden...but they still call |
| 12:54.30 | viperdude | adde: you need better regulation then hehe |
| 12:54.51 | florz | s/still/because of that/, no? That's at least what I heard to be the case in .de |
| 12:54.59 | adde | viper: no thanks... i like the swedish regulations... Pirate country #1 |
| 12:55.06 | viperdude | lol |
| 12:55.11 | florz | hehe :-) |
| 12:55.24 | tparcina | adde: yes, something like that. I have a bounh of people that need to hear same message. |
| 12:55.27 | adde | America/Hollywood are really starting to not like us here |
| 12:56.11 | adde | "Vote Tparcina for President" huh? :P |
| 12:56.18 | tparcina | mosty: yes, but people are willing to hear this message. it's for some party, where representative will thell them his opinion about certen thing... |
| 12:56.39 | tparcina | viperdude: thank you, i'll check that link |
| 12:57.47 | coppice | "Praise the Lord, and call Tparcina's toll free number now to make a donation."? :P |
| 12:58.05 | *** join/#asterisk redax (n=redax@r6.hu) |
| 12:58.31 | tparcina | adde: something like that ;)) |
| 12:58.34 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 12:58.47 | redax | hi, |
| 12:58.59 | adde | tparcina: dont forget that your biggest issue for the candidacy should be stop telespamming :P lol |
| 12:59.08 | tparcina | adde: but maybe I'll make some money from that :)) |
| 12:59.08 | giasai68 | i need to use cli command "ORIGINATE" FOR GENARATE A SIP CALL ON A tel numerb can you explain me the sintaxt please |
| 13:00.05 | redax | please, help... I have an analogue GSM adapter, connected to a X100P. I need to callforward the calls to the technical boy, at nighttime |
| 13:00.08 | tparcina | adde: yes, first I'll sell them 10 asterisks for telemarketing, then I'll sell even more to stop that first 10 telemarketers :)) |
| 13:00.50 | redax | how can I originate a call using the zap channel to execute a call forward in the GSM adapter like : '*21*<technicalboy_phonenum>#' |
| 13:00.53 | adde | Seriously, im really gettin ticked of here... I hate not understanding simple shit. Im trying to do the easiest menu in teh world but it will not respond on anything i press, it hangs up on me...grrr |
| 13:01.25 | mosty | adde: what does the asterisk comsole show? |
| 13:01.45 | tparcina | viperdude: I have check A(x), but I'm not sure will that do what I need. I eed to lead called person thrue auto atendant menu. |
| 13:01.48 | adde | incoming call... playing voicefile...then it hangs up |
| 13:02.16 | mosty | adde: paste the context at a paste site |
| 13:02.53 | adde | http://www.pastebin.ca/397360 |
| 13:03.17 | tparcina | mosty: how to put called person thrue auto atendant menu? |
| 13:03.40 | mosty | adde: for one thing, there's no priority 1 for extension 2 |
| 13:04.02 | adde | i tied that aswell |
| 13:04.03 | mosty | tparcina: i |
| 13:04.17 | mosty | i'm not really interested in helping you spam people sorry |
| 13:04.42 | mosty | adde: does it hangup immediately after playing the sound file? |
| 13:05.07 | adde | if i press a button |
| 13:05.33 | tparcina | mosty: If you don't want to help that's fine, but you can't call me a spammer, because I'm not. Nor will my product be used in that purpose. |
| 13:05.34 | mosty | can you paste the output thats on the console, with verbose and debug set to 10? |
| 13:06.30 | *** join/#asterisk Asteriskmonkey (n=pmullis@69.77.169.14) |
| 13:07.01 | Asteriskmonkey | has anyone had an issue where voicemail dosnt work on asterisk where there is an iax -> iax connection? |
| 13:08.15 | mosty | does asterisk close the pipe after sending agi variables to an agi script? |
| 13:08.59 | adde | http://www.pastebin.ca/397367 |
| 13:10.15 | mosty | adde: try Read instead of Background (it's nicer) |
| 13:10.55 | adde | same params |
| 13:11.36 | redax | is Originate what I need, if: I want to dial extension Y, if he picks up, dial outside number X, and bridge them? |
| 13:11.48 | mosty | adde, no. see the wiki |
| 13:11.51 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 13:11.56 | *** join/#asterisk hohum (n=dcorbe@c-71-62-76-68.hsd1.va.comcast.net) |
| 13:17.29 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:18.09 | tparcina | anybody, how to put called person thrue AA menu? |
| 13:18.11 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 13:22.03 | adde | in Gotoif, if last syntax is ?6:9 what does that mean.. |
| 13:22.46 | mosty | jump to 6 if the condition is true, otherwise jump to 9 |
| 13:22.48 | [TK]D-Fender | adde: "show application gotoif |
| 13:23.18 | adde | thanks... |
| 13:23.22 | *** join/#asterisk BigTrev (n=TrevPlus@host81-159-6-245.range81-159.btcentralplus.com) |
| 13:25.01 | *** join/#asterisk Strom_M (n=strom@216.207.245.1) |
| 13:26.13 | *** join/#asterisk ThoMe (n=tm@tm.muc.de) |
| 13:26.14 | ThoMe | hallo :-) |
| 13:26.16 | ThoMe | hello |
| 13:27.19 | ThoMe | i have a asterisk 1.2X with a digiuam 4-port isdncard. if i send a fax with a external ISDN-FAX-Device and try at the same time a call with a friend is the call broken, aborted. |
| 13:27.22 | ThoMe | why? |
| 13:28.02 | BigTrev | Hey, can somebody please tell me how to use the patchs at http://bugs.digium.com/view.php?id=7764 ? |
| 13:30.21 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 13:32.14 | *** join/#asterisk tic-tac (n=firemoth@office.supersoccer.co.uk) |
| 13:33.13 | *** join/#asterisk iCEBrkr (i=icebrkr@69.9.167.70) |
| 13:37.19 | *** join/#asterisk sabakas1 (n=solapus@66.90.121.129) |
| 13:37.19 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 13:37.23 | ThoMe | ah |
| 13:37.26 | ThoMe | echo canceling |
| 13:37.39 | *** join/#asterisk orn (n=orn@fw1.h12.ipf.is) |
| 13:37.57 | *** join/#asterisk iCEBrkr (i=icebrkr@cyberdyne.org) |
| 13:40.01 | adde | Where are sounds stored... |
| 13:40.33 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 13:40.33 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:43.42 | *** join/#asterisk jpmcallister (n=jpmcalli@kapla.escelsa.com.br) |
| 13:46.53 | tparcina | adde: are they in /var/lib/asterisk/srounds ? |
| 13:48.06 | *** join/#asterisk bkw_ (i=brian@adsl-70-143-50-36.dsl.tul2ok.sbcglobal.net) |
| 13:48.12 | orn | is it not possible to use many types of dtmf settings on the same trunk |
| 13:48.57 | orn | that is, make it recognize either inband and rfc2833 ? |
| 13:54.53 | *** join/#asterisk TaiSHi (n=jaquelin@zion.dattaweb.com) |
| 13:55.03 | TaiSHi | Hello everyone |
| 13:55.13 | TaiSHi | Would someone suggest me a good linux sip client? |
| 13:55.20 | TaiSHi | Using x-lite now... but noise is heavy |
| 13:55.38 | [TK]D-Fender | orn: No. Your trunk will use just 1 and * doesn't want to sit around tinking about 10 different signalling methods |
| 13:55.42 | masked | sjphone |
| 13:55.50 | [TK]D-Fender | Ekiga |
| 13:56.43 | TaiSHi | Ekiga requires quite some config to make it work |
| 13:56.49 | TaiSHi | Gonna try sjphone first |
| 13:58.43 | orn | [TK]D-Fender: Darn... Is there a way for * to force the device to use a method? |
| 13:59.04 | [TK]D-Fender | orn: dtmfmode= <- |
| 13:59.22 | orn | it doesn't seem to dictate to the device which method to use though |
| 13:59.37 | orn | because if i use inband, the PRI end devices work, but the sip devices don't |
| 13:59.38 | [TK]D-Fender | orn: And you can't FORCE the device to use a method, you can only tell * what to use and they had better agree... |
| 13:59.53 | [TK]D-Fender | orn: Most SIP devices use RFC2833 |
| 14:00.02 | [TK]D-Fender | orn: And you should actually KNOW that... |
| 14:00.10 | orn | i know, but PRI's don't |
| 14:00.10 | *** join/#asterisk TeleTommy (n=chatzill@p54a8bcc8.dip0.t-ipconnect.de) |
| 14:00.28 | orn | (ISDN PRI's) |
| 14:00.30 | [TK]D-Fender | orn: And configure your channels accordingly. Sipura/Linksys devices often use "SIP Info" for instance. |
| 14:00.53 | [TK]D-Fender | orn: Correct. Once you're on the PSTN its all just "audio". |
| 14:01.01 | BigTrev | anyone know how to fix the following: error while loading shared libraries: libiksemel.so.3: cannot open shared object file: No such file or directory |
| 14:01.03 | orn | and neither the SIP devices or the PRI in question are trunked directly to the asterisk |
| 14:01.26 | orn | so the possibility of having a seperate trunk for each is gone |
| 14:02.01 | [TK]D-Fender | ? |
| 14:02.08 | [TK]D-Fender | orn: Explain your setup |
| 14:02.31 | orn | trunking to a soft switch (with several thousand users, either PRI or ISDN) |
| 14:02.38 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 14:02.44 | orn | asterisk has a trunk with that switch |
| 14:02.57 | orn | * either PRI or SIP |
| 14:02.59 | orn | i meant to say |
| 14:03.02 | TaiSHi | masked: How do I set up user/pass on it ? |
| 14:03.40 | orn | and depending on the caller on that switch (whether he originates on SIP or PRI) a different DTMF method is employed |
| 14:05.53 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 14:09.05 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 14:10.53 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:11.56 | *** part/#asterisk tparcina (n=tparcina@cisco16.fesb.hr) |
| 14:12.30 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 14:14.45 | *** join/#asterisk phillipk (n=pkey@216.248.143.77) |
| 14:15.44 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 14:20.27 | *** join/#asterisk kink0 (n=k@161.pool62-37-205.static.orange.es) |
| 14:20.29 | kink0 | hello |
| 14:20.37 | redax | so, can Originate used to call an extension ; if he answered call the other party and bridge the call ? |
| 14:22.14 | kink0 | I still unable to get better zttest output, with results like Best: 100.000000 -- Worst: 99.963379 -- Average: 99.986630 |
| 14:22.30 | *** join/#asterisk Hmmhesays (n=Neg@24-119-151-57.cpe.cableone.net) |
| 14:22.30 | *** join/#asterisk Dovid (n=Dovid@85.159.160.207) |
| 14:22.50 | kink0 | even I have try two differents machines ( one a Dual Xeon FSB533, 3.2GH/2M L3 |
| 14:24.09 | kink0 | I have also dedicated one CPU to interrupts the digium card, and enable/dissable HT, recompiling kernel with a minimum, and so |
| 14:24.13 | *** join/#asterisk TaiSHi (n=TaiSHi@zion.dattaweb.com) |
| 14:24.14 | TaiSHi | Back |
| 14:24.26 | TaiSHi | Mmm, how can I cancel noise on softphones? |
| 14:24.43 | iCEBrkr | TaiSHi: close your office door :P |
| 14:24.48 | IPmonger | use a grounded microphone |
| 14:24.56 | TaiSHi | Meh |
| 14:25.07 | TaiSHi | Im using some ... shitty headsets |
| 14:25.11 | kink0 | TaiSHi, I have not noise ussing things like x-lite ( linux and windows versions ) |
| 14:25.19 | iCEBrkr | yeah, don't use the built-in mic on your laptop, it picks up the HD spining and fan noise :-D |
| 14:25.39 | TaiSHi | iCEBrkr, so, easily it could be the sound card ? |
| 14:25.51 | iCEBrkr | TaiSHi: probably not. I think it'd be the headset |
| 14:26.06 | *** join/#asterisk YonahW (n=chatzill@84.229.139.248) |
| 14:26.08 | iCEBrkr | TaiSHi: if you're using a $9.99 headset, the condenser mic in that thing sucks |
| 14:26.23 | TaiSHi | iCEBrkr, ah, you hitted the spot |
| 14:26.37 | TaiSHi | Actually, it's a 15$ :P |
| 14:26.37 | iCEBrkr | I use ExpressTalk with one of those gamer headsets. $24.00 |
| 14:26.40 | iCEBrkr | hehe |
| 14:26.55 | TaiSHi | I have a 5.1 headset @ home |
| 14:26.58 | iCEBrkr | I've talked to my parents on it for hours and I even asked them how it sounds. They can't tell the difference |
| 14:27.05 | TaiSHi | Mmm, how could I reduce the noise on those things? |
| 14:27.19 | iCEBrkr | TaiSHi: Does it have the foam around the mic? |
| 14:27.30 | iCEBrkr | What type of noise are you getting? |
| 14:27.36 | TaiSHi | Like a buzzz |
| 14:27.51 | TaiSHi | It doesn't has a fund |
| 14:27.52 | iCEBrkr | Like some sort of electronic buzz? |
| 14:27.53 | TaiSHi | Pure plastic |
| 14:27.58 | iCEBrkr | ahh |
| 14:28.00 | TaiSHi | Yeah, and background noise |
| 14:28.25 | TaiSHi | (heh, imagine my boss' mood right now... he's gonna cut my b....) |
| 14:28.39 | adde | ...Anyone who knows menus in extensions.conf please check this out: http://www.pastebin.ca/397443 Problem is described there together with conf and log... Thanks |
| 14:28.40 | iCEBrkr | Well you can dampen the background noise with the silly foam. Plus it'll help remove the 'popping' when you say Ps |
| 14:28.59 | iCEBrkr | Paul, Put, Peter, Pull, etc.. |
| 14:29.18 | TaiSHi | Ah, like spitting :P |
| 14:29.19 | iCEBrkr | The foam will also help eliminate the 'heavy breathing' into the phone :P |
| 14:29.35 | iCEBrkr | But the buzzing, sounds like it's just a cheap mic |
| 14:30.09 | iCEBrkr | You could also try to move the wire away from the computer a bit, it could be doing some sort of electrical induction |
| 14:30.32 | TaiSHi | adde, add this to start of menu (before answer on same extension as WaitExten) |
| 14:30.37 | TaiSHi | exten => s,2,Set(TIMEOUT(digit)=7) |
| 14:30.37 | TaiSHi | exten => s,3,Set(TIMEOUT(response)=20) |
| 14:30.54 | adde | testing |
| 14:31.00 | TaiSHi | iCEBrkr, it is a cheap one, very |
| 14:31.06 | iCEBrkr | hehe |
| 14:31.36 | adde | TaiShi: Before Answer?? |
| 14:31.52 | TaiSHi | Yes |
| 14:31.56 | TaiSHi | Like this... wait |
| 14:32.17 | iCEBrkr | OK, my question is, WHY the hell is my voicemail timestamps all jacked up from 1.2.x to 1.4.x |
| 14:32.23 | TaiSHi | http://www.pastebin.ca/397446 |
| 14:32.24 | iCEBrkr | I've set the tz option. |
| 14:32.58 | iCEBrkr | It's 4hrs ahead of time. |
| 14:33.10 | iCEBrkr | I suppose that's some sort of GMT shit |
| 14:35.20 | *** join/#asterisk Strom_M (i=strom@nat/digium/x-55a1ef787061925e) |
| 14:36.01 | *** join/#asterisk nfi|ermes (n=ermes@217.220.121.62) |
| 14:36.31 | *** join/#asterisk ellisdee (n=ellisdee@mail.globalgeophysical.com) |
| 14:37.31 | [TK]D-Fender | TaiSHi: Fix your DTMF mode |
| 14:37.36 | adde | TaiShi: http://www.pastebin.ca/397453 |
| 14:37.46 | adde | AFter your idea |
| 14:37.55 | adde | so, basicly no change |
| 14:38.26 | [TK]D-Fender | adde: That was meant for you actually |
| 14:38.42 | adde | where do i do that? |
| 14:39.06 | *** join/#asterisk badcfe (n=cso@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) |
| 14:40.15 | TaiSHi | [TK]D-Fender, ? |
| 14:40.25 | TaiSHi | Sorry, but no idea on that [TK]D-Fender |
| 14:40.49 | [TK]D-Fender | adde: Your call looks like its coming in from an un-authed source. |
| 14:41.02 | [TK]D-Fender | adde: What is the calls origin? |
| 14:41.22 | TaiSHi | dtmfmode=rfc2833 <-- this is what I have on external incoming calls (from ITSP) and it hears fine |
| 14:41.27 | [TK]D-Fender | adde: when in doubt try "dtmfmode=rfc2833" under [general] first |
| 14:42.03 | TaiSHi | Was that for me ?:P |
| 14:43.28 | adde | fender: it comes from my gizmo call-in number... |
| 14:43.43 | adde | Which ive set register |
| 14:43.43 | giasai68 | hello |
| 14:44.14 | [TK]D-Fender | adde: well try rfc2833 first, then inband next if the codec for the call is G711 |
| 14:44.25 | [TK]D-Fender | adde: Otherwise try "info" after |
| 14:44.33 | giasai68 | i need to send login and passwordo to sip proxi to authentivate i want insert this in info in extensions.conf |
| 14:44.38 | giasai68 | exten => _93X.,1,Dial(Sip/${EXTEN:1},A0:4c492a67@209.3.12.83:5070,,rt) |
| 14:46.51 | giasai68 | exten => _93X.,1,Dial(Sip/${EXTEN:1},A0:4c492a67@209.3.12.83:5070,,rt) |
| 14:46.59 | giasai68 | sorry wrong box |
| 14:47.50 | TaiSHi | [TK]D-Fender, rfc2833 screwed my login u_U |
| 14:47.57 | *** join/#asterisk Ciber311 (n=Ciber311@user-12ld42j.cable.mindspring.com) |
| 14:48.59 | iCEBrkr | Damnit.. Why didn't I run with this years ago?? www.grandcentral.com |
| 14:49.04 | adde | Fender: http://www.pastebin.ca/397469 My SIP.CONF |
| 14:49.39 | giasai68 | i need to insert in extex login and password to authentcate in a sip proxi is correct this syntax? : exten => _X.,1,Dial(sip/${EXTEN:1},login:password@host:port) |
| 14:49.52 | [TK]D-Fender | TaiSHi: I corrected myself, I was speaking to adde, not you. Was acciudent |
| 14:50.39 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 14:50.51 | TaiSHi | [TK]D-Fender, ok, any idea on my trouble ? |
| 14:51.35 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 14:54.39 | [TK]D-Fender | TaiSHi: Nope.... 1 at a time |
| 14:57.23 | adde | Fender. I instead of the menu at incoming i Dialed my softphone on pc... if i press buttons on the cellphone i could here them on pc.... shouldnt asterisk then be able to hear dialtones if its routing them through? |
| 14:58.35 | [TK]D-Fender | adde: for it to listen to AUDIO DTFM then you should use "dtmfmode=inband', but only if the codec is G711 |
| 14:59.13 | Sweeper | so anyone got work in new orleans? :/ |
| 14:59.30 | adde | Im really new at this... i havent really looked at codecs what is pre installed... I got my Gizmo dialin number and a softphone on pc. |
| 15:00.29 | adde | and basicly the only configs ive been messing with for now is sip and ext |
| 15:02.06 | *** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com) |
| 15:08.34 | *** join/#asterisk Mercestes (n=Merceste@cpe-24-175-82-3.houston.res.rr.com) |
| 15:10.18 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 15:10.38 | anonymouz666 | is it possible to "flash" a digital line? |
| 15:11.13 | Mercestes | anonymouz666, Like a sip line?? I don't thinks o. |
| 15:12.23 | anonymouz666 | like an analog line |
| 15:12.26 | adde | Fender: No ideas what i can try |
| 15:12.42 | anonymouz666 | to block collect calls |
| 15:12.43 | Mercestes | anonymouz666, : Oh, you mean a digiital analog line?? |
| 15:13.03 | anonymouz666 | yes |
| 15:13.30 | Mercestes | Call your telco and ask them if it's macromedia compliant. If has to be for flash to be enabled. |
| 15:14.46 | anonymouz666 | the legacy PBX do a flash for phones who can't get collect calls. but * is in front the legacy PBX, and I can't see a way to do the same flash. |
| 15:14.47 | Mercestes | If it's analog you can "flash" it. If it's digitial, you cannot "flash" it. If it's a digital PBX line like an old turnkey system then chances are it does internally emulate a flash for you. |
| 15:15.05 | Mercestes | how are you connected to the legacy pbx? |
| 15:15.15 | anonymouz666 | e1 port |
| 15:15.46 | Mercestes | There is a Flash() or SendFlash() command or something of that nature in which you can send a flash cmd across a pri line or an analog line. |
| 15:16.06 | *** join/#asterisk intralanman (n=lanman@pool-71-253-253-149.nrflva.east.verizon.net) |
| 15:16.27 | Mercestes | If your phone devices are SIP you can't just hang up and pick up the phone again. You'll have to use a *code or something. |
| 15:17.30 | anonymouz666 | no SIP phones |
| 15:17.40 | Mercestes | what kind of phones? |
| 15:17.57 | Mercestes | Oh...so it's Legacy PBX -> asterisk -> world??? |
| 15:17.58 | anonymouz666 | telco -> asterisk -> pbx -> tradional phones |
| 15:18.09 | Mercestes | Ahhh. |
| 15:18.19 | Mercestes | Just hook flash then. |
| 15:18.28 | anonymouz666 | flash() you mean |
| 15:18.46 | Mercestes | nah, hookflash PBX, pbx will deliver the flash to asterisk over the E1 |
| 15:19.10 | anonymouz666 | the legacy pbx send the flash do *, but * does not forward this flash |
| 15:19.12 | Mercestes | Unless you hooked asterisk up to a SIP provider or something silly like that. |
| 15:19.33 | anonymouz666 | i need to do the flash in * |
| 15:19.43 | Mercestes | How is * hooked to the Telco? |
| 15:20.03 | anonymouz666 | e1 |
| 15:20.32 | Mercestes | AFAIK it should just work then. |
| 15:20.56 | anonymouz666 | it does not |
| 15:21.34 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 15:21.34 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:21.44 | Mercestes | morning anthm. |
| 15:21.56 | anthm | hello |
| 15:22.18 | Mercestes | anonymouz666, Hrm. Might be able to do some exten => h,1,Voodoo() to emulate a hookflash. Or you can google asterisk PRI hookflash. |
| 15:23.16 | anonymouz666 | it is MFCr2 |
| 15:24.04 | *** join/#asterisk lorinc (n=ang@pool-8217.adsl.interware.hu) |
| 15:24.18 | Mercestes | did you google MFCr2? |
| 15:24.42 | Mercestes | like the third link down says "asterisk MFC r2" |
| 15:24.50 | Mercestes | of course...it's on voip-info.org =/ |
| 15:25.32 | mquin | v-i.o stil down? |
| 15:25.59 | *** join/#asterisk c4t3l (n=c4t3l@cpe-72-181-205-77.houston.res.rr.com) |
| 15:26.02 | Mercestes | No! no it's back up. |
| 15:26.31 | c4t3l | it up? |
| 15:26.49 | Mercestes | It is. |
| 15:26.52 | mquin | \o/ |
| 15:26.56 | Mercestes | :) |
| 15:26.59 | Mercestes | Oh happy day! |
| 15:27.05 | c4t3l | ellisdee hello |
| 15:27.06 | Mercestes | I can cancel my resignation now |
| 15:27.15 | c4t3l | mercestes hello as well |
| 15:27.36 | Mercestes | Well met. |
| 15:28.35 | Mercestes | So, anonymouz666: In beloved tribute to this happy day, I provide you my first voip-info link since after it's apocolypse. http://www.voip-info.org/wiki/view/Asterisk+MFC+R2 |
| 15:28.39 | Mercestes | bwahaha...that felt good. |
| 15:29.37 | c4t3l | god bless voip-info!! |
| 15:30.07 | c4t3l | i think some hacker kid brought it down to teach me a lesson |
| 15:30.13 | c4t3l | :) |
| 15:30.43 | c4t3l | ellisdee you better answer me boy! |
| 15:31.21 | coppice | what happened to voip-info? |
| 15:31.31 | c4t3l | hardware issue i think |
| 15:31.40 | coppice | and no backups? |
| 15:31.44 | c4t3l | at least thats what it said yesterday |
| 15:31.49 | HexDump | I would like someone to tell me if I could use asterisk to log problems in my voip configuration, I mean, connection getting saturating (prudcing cuts in voice, etc...), statistics like maximun peek of kb consumed, etc.... It would be really nice. |
| 15:31.49 | c4t3l | meh |
| 15:32.20 | c4t3l | have you tried mrtg HexDump |
| 15:32.39 | Mercestes | HexDump: Why would you use Asterisk to monitor your network? |
| 15:32.40 | coppice | I refuse any help on MFC/R2 to people who get their info from places like voip-info |
| 15:32.59 | c4t3l | nagios is better suited to mon net stuff |
| 15:33.01 | *** join/#asterisk intralanman (n=lanman@pool-71-253-253-149.nrflva.east.verizon.net) |
| 15:33.11 | c4t3l | hehe |
| 15:33.25 | Mercestes | coppice, Thank you, that was a happy thought. I feel better about myself now. |
| 15:33.31 | Mercestes | coppice, Where do you get your info? |
| 15:33.57 | c4t3l | coppice-info.org? |
| 15:34.21 | coppice | if voip-info shut down, most developers would have and easier life. its full of outdated and inaccurate crap |
| 15:34.22 | c4t3l | oh crap you're not an admin are you? |
| 15:34.34 | c4t3l | true that homie |
| 15:34.37 | Mercestes | c4t3l, lol. I've done that before. |
| 15:34.39 | Corydon-w | Backup doesn't help much if the problem isn't the hard drive. |
| 15:34.47 | HexDump | c4t3l: no mate, what's it? |
| 15:35.01 | Mercestes | coppice, Your always welcome to fix it. dictionary.com: wiki sometime. It's useful information. |
| 15:35.28 | c4t3l | HexDump if you are looking for network monitor software you should try something like nagios or cricket |
| 15:35.30 | HexDump | Mercestes: not my network, but i think it will be nice to have some statistics on band consuming (only for voip), etc... |
| 15:35.49 | redax | geez. |
| 15:35.54 | HexDump | I see. |
| 15:35.57 | coppice | Mercestes: it doesn't need fixing. it isn't needed at all. what you posted is just an old version of what you will find at the source of the R2 software |
| 15:35.59 | redax | how to dial #21# on a SIP trunk? |
| 15:36.07 | redax | like SIP/51/#21# |
| 15:36.09 | redax | not working |
| 15:36.25 | c4t3l | why you using #? |
| 15:36.26 | Mercestes | coppice, I find it assinine to state that you refuse to help people who do their own research, basically. |
| 15:36.35 | c4t3l | uh oh |
| 15:36.49 | *** join/#asterisk ParaNoir_ (n=daanb@84.53.99.162) |
| 15:36.51 | ParaNoir_ | Hey :) |
| 15:36.54 | Mercestes | coppice, besides, he asked a question about hook flashing, mentioned MFCr2, and I googled it and found that link. I'm not the one with the problem so *I'm* not hte one researching it. |
| 15:36.55 | c4t3l | how about that GPL version 3?? |
| 15:36.57 | redax | disabling callforward on a GSM adapter :) |
| 15:37.18 | c4t3l | hmm |
| 15:37.28 | TaiSHi | Mercestes, kill me. |
| 15:37.40 | Mercestes | coppice, Now if you wish to continue to not help because voip-info.org is beneath you then so be it, but..please do so silently so youd on't disrupt the rest of us trash who frequent slums like voip-info, please. |
| 15:37.44 | c4t3l | redax try senddtmf applicatin in dial plan |
| 15:37.52 | redax | oh. cool |
| 15:37.54 | anonymouz666 | Mercestes: thanks for the link but that does not help in any way. |
| 15:37.58 | redax | thanks c4t3l |
| 15:38.00 | c4t3l | senddtmf(#blah blah) |
| 15:38.06 | c4t3l | no prob |
| 15:38.07 | coppice | if they want to use any old crap they find somewhere on internet, instead of the information provided with the software, they are on their own. assinine is people posting bug fixes on voip-info, and not reporting them to the author |
| 15:38.20 | TaiSHi | science tbh |
| 15:38.22 | c4t3l | well i agree with that |
| 15:38.24 | TaiSHi | But I have a BIG problem |
| 15:38.27 | ParaNoir_ | i'm new to ISDN and new to asterisk, but have a quad BRI cards and when i do a cat /dev/zap/* i'll get Layer 1 DEACTIVATED for every port, can someone tell me how i can debug this? to clearify the problem.. |
| 15:38.27 | TaiSHi | On noise >.< |
| 15:38.48 | ParaNoir_ | it's /proc/zap ;) sorry |
| 15:38.51 | Mercestes | TaiShi: Sorry, stupidity osmosis kicking in. |
| 15:38.51 | c4t3l | how many channels can BRI support? |
| 15:39.29 | christo | Hi guys.. Can anybody see a solution to this: http://pastebin.ca/397349 ? |
| 15:39.47 | c4t3l | not trying to be a jerk , but did you compile the newest version of libpri ParaNoir_ |
| 15:39.50 | TaiSHi | Mercestes, thing is, I hear ALL the backbround noises |
| 15:40.02 | TaiSHi | I can kinda hear like supahman |
| 15:40.37 | *** join/#asterisk NewbePaul (n=paul@adsl-072-148-241-244.sip.asm.bellsouth.net) |
| 15:40.37 | *** join/#asterisk funxion (n=nunya@63.214.236.169) |
| 15:40.50 | Mercestes | TaiShi: Ok. So what kind of phones do you have and how is * connected to the world? |
| 15:41.03 | funxion | anyone know why chan_modem.so would be missing after compiling asterisk 1.2.14? |
| 15:41.08 | mihinomenest | so, I've been following the orderlyQ how-to. when I configure incoming calls to go to a queue, I get an avalanche of "Mar 16 16:37:17 NOTICE[356]: channel.c:1949 ast_read: Dropping incompatible voice frame on Local/100@default-f1b5,1 of format gsm since our native format has changed to (g729)" in sip.conf, I have "disallow=all," "allow=ulaw,g729" specified in [general] and the contexts for my provider. if I remove "g729" I get "no |
| 15:41.20 | *** join/#asterisk harleya (n=harleya@207.108.166.2) |
| 15:41.26 | *** join/#asterisk lokkju_wrk (n=lokkju@unaffiliated/lokkju) |
| 15:41.43 | ParaNoir_ | c4t3l: i don't know ;) it's not jerky i just did install-ZAPHFC, it gave an error and did it again, then it didn't give an error :) |
| 15:42.02 | ParaNoir_ | but there's an rpm named libpri-1.2.4-1.382 |
| 15:42.11 | ParaNoir_ | it's installed.... so looks like it is ;) |
| 15:42.20 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 15:42.24 | c4t3l | ohh |
| 15:42.33 | Mercestes | mihinomenest, redo your Moh to g729 transcoding |
| 15:42.36 | c4t3l | ParaNoir_ what distro |
| 15:42.38 | [TK]D-Fender | funxion: Support was dropped, just like its development during the 1.2 series |
| 15:42.38 | TaiSHi | x-lite + 9.99$ microphones |
| 15:42.46 | *** join/#asterisk redax (n=redax@r6.hu) |
| 15:42.46 | TaiSHi | headsets * |
| 15:42.47 | ParaNoir_ | CentOS it's the trixbox distro... |
| 15:42.49 | redax | c4t3l: aaaa. seems like Dial() has 'D' option |
| 15:42.56 | c4t3l | i tend to stay away from pre-comped bins |
| 15:43.02 | Mercestes | ParaNoir_, omg...ROFLMAO |
| 15:43.09 | Mercestes | ParaNoir_, That's great. I love it. :D |
| 15:43.13 | [TK]D-Fender | ~trixbox |
| 15:43.15 | jbot | it has been said that trixbox is unable to be supported here. Try joining #freepbx and asking there, or use the trixbox forums at http://www.trixbox.org/modules/newbb/ |
| 15:43.15 | ParaNoir_ | hey, i'm a beginner man :P |
| 15:43.22 | TaiSHi | ~wglwat |
| 15:43.24 | jbot | rumour has it, wglwat is well, good luck with all that |
| 15:43.24 | Mercestes | s'true tho. |
| 15:43.25 | ParaNoir_ | ahhh man ;) |
| 15:43.28 | redax | c4t3l: Dial(SIP/52,60,D(#21#) ) |
| 15:43.28 | [TK]D-Fender | ParaNoir_: Don't expect to learn anything except PAIN from it |
| 15:43.31 | redax | maybe.. let's see |
| 15:43.33 | ParaNoir_ | lol |
| 15:43.39 | ParaNoir_ | so what do you suggest? |
| 15:43.47 | ParaNoir_ | install went fine :P but it doesn't install ;) |
| 15:43.50 | Mercestes | ParaNoir_, #freepbx for starters.... |
| 15:44.00 | ParaNoir_ | didnt get zap started on AsteriskNOW |
| 15:44.02 | ParaNoir_ | ahhh ok ;) |
| 15:44.04 | ParaNoir_ | Thanks mate! |
| 15:44.07 | [TK]D-Fender | ParaNoir_: Trash it, install CentOS normally, download and compile * from source yourself and learn * |
| 15:44.07 | Mercestes | ParaNoir_, Or scrap that freebpx (and centos) crap and go with a pure asterisk install |
| 15:44.27 | Mercestes | but whatever you do, don't read that shit on the Internet, it's all lies and outdated crap. |
| 15:44.31 | ParaNoir_ | so what do i basically need? zaptel,asterisk and mISDN? |
| 15:44.31 | redax | grrr. |
| 15:44.32 | c4t3l | ok , screw all that . just use bsd |
| 15:44.37 | TaiSHi | Mercestes, ideas ? |
| 15:44.37 | mihinomenest | Mercestes: so, you're saying the problem is that I'm using RAW MoH and it needs to be G729. |
| 15:44.51 | Mercestes | mihinomenest, precisely! |
| 15:45.08 | redax | the problem is... the GSM adapter is connected via SPA3102 |
| 15:45.09 | mihinomenest | excellent. |
| 15:45.18 | redax | and #21# is blocked by the SPA3102 |
| 15:45.27 | Mercestes | mihinomenest, : NO!! Don't google! |
| 15:45.31 | Mercestes | It's all LIES! |
| 15:45.41 | Mercestes | The Internet is out to get you. It wille at your brain and format your harddrive |
| 15:45.45 | ParaNoir_ | :) |
| 15:45.45 | Mercestes | RTFS like a real man. |
| 15:45.57 | Mercestes | otherwise coppice won't help you. |
| 15:46.00 | mihinomenest | but all I want are small, convenient lies. |
| 15:46.02 | Qwell[] | redax: Most SIP devices can't dial # - that's usually interpreted as "send what I've dialed so far" |
| 15:46.07 | *** join/#asterisk ai-a[awol] (n=jake@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) |
| 15:46.12 | Mercestes | TaiSHi, So....your problem is..... |
| 15:46.22 | redax | Qwell[]: so that's it? no solution? |
| 15:46.28 | ParaNoir_ | will debian do the job too? |
| 15:46.35 | [TK]D-Fender | redax: Fix your dialplan |
| 15:46.36 | ai-a[awol] | I have SIP service "register => XXXXXX:XXXXX@XXXXX" in my sip.conf, how do i make a call using that register ? |
| 15:46.43 | Qwell[] | redax: pretty much |
| 15:46.46 | Mercestes | TaiSHi: When using $10 headsets on a 30-day trial of x-lite softphones hooked to your free copy of Asterisk on your network using mostly wallmart brand linksys switches....you get lots of background noise, aye? |
| 15:46.51 | Qwell[] | use something besides # in your dialplan |
| 15:47.02 | redax | [TK]D-Fender: I added the #xx to the spa3102 dialplan |
| 15:47.06 | [TK]D-Fender | ai-a[awol]: Gor read up on the basics of SIP peers/users in THE BOOK |
| 15:47.08 | Qwell[] | Mercestes: You mean you can't get perfect sound quality for $10? |
| 15:47.08 | [TK]D-Fender | ~book |
| 15:47.15 | jbot | well, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:47.15 | Mercestes | mihinomenest, hehe, please proceed them. I'm just trollin ga bit anyways. |
| 15:47.15 | redax | and if I use the FXS port, it allows |
| 15:47.24 | Mercestes | Qwell[]: *gasps* Of course you can! |
| 15:47.28 | redax | this case it should send the #21# on the FXO port |
| 15:47.29 | TaiSHi | Mercestes, actually, using cisco's |
| 15:47.29 | mihinomenest | Mercestes: seriously???? |
| 15:47.41 | Mercestes | mihinomenest, yea. I'm half-troll. |
| 15:47.50 | redax | Qwell[]: bad news |
| 15:47.58 | Mercestes | mihinomenest, Unless your asking about the headsets then......>.> yes. |
| 15:48.04 | mihinomenest | no. |
| 15:48.11 | mihinomenest | I have no interest in headsets. |
| 15:48.12 | Mercestes | TaiSHi: Cisco == expensive crap v/s linksys == cheap crap. |
| 15:48.14 | TaiSHi | Mercestes, also, Asterisk is free... |
| 15:48.33 | Mercestes | TaiSHi: What are you calling on yoru headsets? |
| 15:48.44 | redax | linksys accured sipura... and sipura wasn't that bad |
| 15:48.46 | *** join/#asterisk Penggu (i=foobar@220-245-200-87.static.tpgi.com.au) |
| 15:49.03 | Penggu | hi all. i want to somehow have a list of extensions in a list, eg: 200,201,205,207 |
| 15:49.08 | TaiSHi | 1 ear - microphone |
| 15:49.09 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 15:49.13 | TaiSHi | Full plastic |
| 15:49.24 | *** join/#asterisk bsd_tech (n=bsdtech@ppp-69-238-48-150.dsl.irvnca.pacbell.net) |
| 15:49.28 | bsd_tech | mornig |
| 15:49.31 | Penggu | and then match it in extensions.conf exten->2XX,1,IF(CALLER IN list fo extensions) |
| 15:49.31 | bsd_tech | lol |
| 15:49.33 | Mercestes | bsd_tech, morning. |
| 15:49.42 | redax | TaiSHi: if you're talking about a woman.. that's bad |
| 15:49.45 | Penggu | any suggestions? |
| 15:49.58 | redax | Penggu: sip show peers? |
| 15:49.58 | Mercestes | TaiSHi: .... what are you calling on your headset??? |
| 15:49.59 | bsd_tech | I just found my first original dial plan for asterisk 1.1 |
| 15:50.34 | Mercestes | bsd_tech, Tell us all about it! :) |
| 15:50.39 | bsd_tech | cleaning my hd and found all my old dialplans |
| 15:50.42 | Penggu | redax: i wanted to have variables that hold diff lists of extensions for diff purposes. eg voicemail_extensions = 200,201,205,208 |
| 15:50.49 | Penggu | and then |
| 15:50.52 | bsd_tech | just laughing at it |
| 15:50.59 | Penggu | extensions_that_can_dial_out = 300,255,etc |
| 15:51.05 | redax | Penggu: use AGI. ;-) |
| 15:51.06 | TaiSHi | Mercestes, I dont understand the question |
| 15:51.07 | Penggu | and use them in the dial plan |
| 15:51.13 | [TK]D-Fender | bsd_tech: 1.1? Wow... not sure how I missed that version ;) |
| 15:51.29 | Penggu | by saying (if exten is IN/NOT-IN [list]) blah |
| 15:51.37 | redax | yeah. the asterisk 1.1 was the best bugfree version ever |
| 15:51.51 | Mercestes | TaiSHi: Ok, youhave a headset in your ear, right? |
| 15:51.55 | Penggu | i notice the new functions in 1.2 .. CUT, EXISTS, etc.. but nothing quite useful in this regard |
| 15:51.58 | TaiSHi | Yes |
| 15:52.01 | Mercestes | TaiSHi: Are you calling something *before* you hear the noise?? |
| 15:52.25 | redax | why do you need a list of vm extensions? |
| 15:52.34 | Penggu | i supposed i could go agi... |
| 15:52.37 | TaiSHi | Not using any other audio soft, plus, just calling an extension (124) that gives me echo |
| 15:52.38 | redax | just make a context where vm extensions are routed :) |
| 15:52.47 | TaiSHi | Also tried calling my boss... uglyness audio u_U |
| 15:52.50 | Penggu | well, there's different contexts that i wanted to match |
| 15:52.51 | mihinomenest | fyi, googling for "gnu sox" isn't as productive as I thought it'd be. |
| 15:53.21 | Penggu | like, an ext might be want contexts a, c and f while another would be using only b, f, and i |
| 15:53.27 | [TK]D-Fender | Penggu: Go look up "asterisk expressions" on the wiki, and "show application gotoif" from * CLI |
| 15:53.36 | Penggu | expressions.. |
| 15:53.54 | redax | what!s the name of the "#" sign? |
| 15:53.57 | redax | what's |
| 15:54.09 | redax | in english, please ;-) |
| 15:54.19 | TaiSHi | pad ? |
| 15:54.23 | TaiSHi | almohadilla ? |
| 15:54.26 | TaiSHi | cuxinet ? |
| 15:54.29 | redax | pound? |
| 15:54.29 | Penggu | btw, we got an asterisk server made up from some company, they left the gui installed and running by default. |
| 15:54.43 | Penggu | are they bad, or is that an ok thing to do? |
| 15:55.05 | Penggu | my understanding was to have minimum stuff running to devote max resources to aster |
| 15:55.07 | Mercestes | TaiShi: So, now your making me assume things, like, your calling from one SIP softphone to another SIP softphone, aye?? |
| 15:55.09 | JoNate | redax: its pound |
| 15:55.22 | redax | thanks JoNate |
| 15:55.32 | TaiSHi | Mercestes, yes, or to an extension that does Echo() |
| 15:55.34 | *** join/#asterisk dasenjo (n=dasenjo@190.24.178.96) |
| 15:55.40 | redax | I heard 'bound' ;) |
| 15:55.45 | florz | Penggu: They _left_ it installed? Why did they install it in the first place? |
| 15:55.50 | TaiSHi | redax, sound ? |
| 15:55.51 | redax | out from the asterisk sounds |
| 15:55.56 | Mercestes | TaiSHi: Aren't you the one whose been going "please, my boss will fire me?" not too long ago?? |
| 15:56.05 | redax | yah :) |
| 15:56.13 | TaiSHi | Mercestes, yes ¬¬ |
| 15:56.20 | TaiSHi | Grandstream + softphone |
| 15:56.21 | Mercestes | yea, I remember you now. |
| 15:56.22 | Penggu | florz: beats me. when one of their techies came over i was giving them tips on how to config the dial-plan. i guess we chose the wrong bunch? |
| 15:56.29 | Mercestes | Here's a hint. |
| 15:56.34 | TaiSHi | Mercestes, glad you did, it isn't useful tho |
| 15:56.50 | JoNate | Mercestes...See if you can hook TaiSHi up with that Telepathy system we've been working on... |
| 15:56.50 | TaiSHi | Ok, hint me |
| 15:56.51 | Mercestes | TaiSHi: SIP does not cause echo. Sip does not cause static. Sip does not cause "noise." |
| 15:57.11 | TaiSHi | I supposse |
| 15:57.22 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 15:57.23 | Mercestes | It is not a supposition. It is a fact. |
| 15:57.41 | JoNate | isn't all rape brutal? |
| 15:57.41 | Mercestes | now with this newfound foundation to work upon, go forth, and troubleshoot that which you have wrought. |
| 15:57.47 | TaiSHi | Mercestes, so the problem radicates in my computer ? |
| 15:57.56 | TaiSHi | Well, on all of them u_U |
| 15:57.59 | JoNate | much better |
| 15:58.00 | TaiSHi | Meaning, hardware |
| 15:58.05 | redax | f**k... I need to send that '#21#' .. it's insane a box like this eats up other things like digits |
| 15:58.05 | Mercestes | the problem *could* radicate your computer......maybe. |
| 15:58.13 | Mercestes | I haven't ever seen it radicate a computer but it's possible. |
| 15:58.20 | *** join/#asterisk Zefk (n=Zefk@wsc-fo.b.astral.ro) |
| 15:58.26 | TaiSHi | o.O |
| 15:58.29 | Mercestes | TaiSHi: I'd focus more on discovering the problem. |
| 15:58.30 | TaiSHi | So, your ideas? |
| 15:58.39 | TaiSHi | Mercestes, problem = noise/background |
| 15:58.54 | Mercestes | TaiSHi: See, I already know how this movie ends so it'd be no fun if I just told you what's going to happen. |
| 15:58.54 | florz | Penggu: Well, I wouldn't judge by that alone, but it sounds a bit suspect ... |
| 15:59.04 | [TK]D-Fender | TaiSHi: Beg for political ayslum ;) |
| 15:59.09 | giasai68 | Dial("Zap/2-1", "Sip/393280472347@209.3.12.82:5070||r") in new stack |
| 15:59.10 | giasai68 | <PROTECTED> |
| 15:59.10 | giasai68 | [Mar 16 16:56:03] NOTICE[12115]: chan_sip.c:11719 handle_response_invite: Failed to authenticate on INVITE to '"xxxxx" <sip:xxxxxx@151.9.187.207:5070>;tag=as0bf97074' |
| 15:59.10 | christo | Hi guys.. Can anybody see a solution to this: http://pastebin.ca/397349? (sorry, irssi isn't logging and I stepped out for a sec) |
| 15:59.18 | giasai68 | got this warnig |
| 15:59.26 | giasai68 | anyone help me? |
| 15:59.35 | [TK]D-Fender | giasai68: You have no password and they need one. |
| 15:59.43 | [TK]D-Fender | giasai68: Read the BIG PRINT |
| 15:59.58 | Zefk | Hi. Is it possible to have Asterisk Dial into another sistem (without bridging any other call parties) and send some dtmf digits after that ? Thx |
| 16:00.04 | Mercestes | TaiSHi: And you now know that SIP causes neither. Good luck! |
| 16:00.15 | giasai68 | ok i have the password but i don't know where i must insert it |
| 16:00.18 | redax | there's really no serious documentation about linksys/sipura equipments? |
| 16:00.50 | christo | giasai68 - probably in your sip.conf |
| 16:01.08 | giasai68 | done it but don't' wirk |
| 16:01.11 | giasai68 | work |
| 16:01.12 | Mercestes | BRB, switchign computers. |
| 16:01.45 | bsd_tech | ok piecing together all the stuff I have done. lol |
| 16:01.47 | bsd_tech | man |
| 16:01.49 | [TK]D-Fender | giasai68: Dial("Zap/2-1", "Sip/393280472347:yourpasswordhere@209.3.12.82:5070||r") |
| 16:02.09 | [TK]D-Fender | giasai68: Although why are you dialing the IP directly? You should set up a SIP peer entry for that. |
| 16:02.33 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 16:02.58 | giasai68 | i'mgenerating a call from a analogic telephone connected trought a fxs port, i want transfer this call to a sip proxy (xxx.xxx.xxx.xxx) this proxy need to authentication with user and password |
| 16:03.14 | giasai68 | i'm setting in sip.conf all host login an password |
| 16:04.27 | *** join/#asterisk mkl1525 (n=qwertz@38.205.27.217.static.versanetonline.de) |
| 16:04.35 | TaiSHi | mercestes, bleh, Bluetooth and USB headsets arriving monday |
| 16:04.37 | TaiSHi | Screw analog u_U |
| 16:05.01 | TaiSHi | mercestes, thanks for the assistance |
| 16:05.29 | TaiSHi | I should try noise reduction soft, or try connecting @ home |
| 16:05.30 | mkl1525 | HI, does anybody know if it is possible to get the current firmware version of the snom phones attached to * (for all phones together) or if there is any other way to go? |
| 16:05.37 | mercestes | ..... |
| 16:05.54 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 16:05.58 | giasai68 | http://pastebin.ca/397566 my sip.conf |
| 16:06.01 | TaiSHi | I strongly think it's either headset or onboard audiocar |
| 16:06.03 | TaiSHi | card * |
| 16:06.03 | mercestes | TaiSHi: Very good! But....it's not the analog thats hurting you. |
| 16:06.15 | TaiSHi | Ok, not SIP, not Analog |
| 16:06.21 | mercestes | TaiShi: It's that $10 price tag that's hurting you. |
| 16:06.25 | TaiSHi | Maybe the 5u$d headset? |
| 16:06.33 | TaiSHi | Actually, it's 5 |
| 16:06.35 | mercestes | *That's* whats hurting you. |
| 16:06.43 | TaiSHi | In argentina it's around 15 pesos |
| 16:06.53 | TaiSHi | mercestes, ever knew a goddamn cheap boss? |
| 16:06.54 | mercestes | I have 15 pesos in my couch. |
| 16:06.56 | mercestes | and I live in the US. |
| 16:07.02 | mercestes | TaiSHi: yes.... |
| 16:07.15 | Penggu | is there something like If(condition) then Include(some context) ? |
| 16:07.20 | Penggu | i know there's gotoif |
| 16:07.27 | Penggu | but i wanted to include insted |
| 16:07.52 | mercestes | Penggu: I'm not an .ael expert...but..maybe .ael has if check ability |
| 16:08.51 | c4t3l | sorry lads. I just got out of the shower/ now whats this about a cheap boss? |
| 16:09.03 | TaiSHi | mercestes, thank you, I will copy/paste you |
| 16:09.06 | TaiSHi | Or better |
| 16:09.11 | TaiSHi | I will bring him here to |
| 16:09.15 | TaiSHi | Hear it from you |
| 16:09.22 | Penggu | so much to learn.. |
| 16:09.32 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 16:09.55 | Penggu | i thought i covered ground by reading that tfot book |
| 16:10.20 | giasai68 | http://pastebin.ca/397566 my sip.conf |
| 16:10.42 | lokkju_wrk | anyone any good at resolving IAX issues on asterisk? I'm trying to use IDEFisk, and keep getting registration timeout errors, even though asterisk sees the incoming connection |
| 16:11.27 | TaiSHi | lokkju_wrk, tried nat = yes ? |
| 16:11.45 | TaiSHi | Basic networking says, you may reach the target, but will he reach you? |
| 16:11.49 | lokkju_wrk | TaiSHi, for IAX? |
| 16:12.08 | c4t3l | okkju_wrk: have you tcpdumped on both sides to make sure that iax is good on bith? |
| 16:12.09 | TaiSHi | lokkju_wrk, I don't know IAX |
| 16:12.15 | c4t3l | both** |
| 16:12.22 | TaiSHi | But you should checkout |
| 16:12.29 | mercestes | brb |
| 16:12.42 | TaiSHi | ok |
| 16:13.33 | *** join/#asterisk TripleF_W (n=TripleFF@145-27.mc.cite.net) |
| 16:15.46 | lokkju_wrk | c4t3l, no, it looks like the the client is not recieving a response at all... it looks like * is sending back to the default IAX2 port instead of the nat out port |
| 16:15.55 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 16:16.27 | christo | is anybody able to reproduce this problem? http://pastebin.ca/397349 |
| 16:17.12 | TaiSHi | lokkju_wrk, there should be a port = variable.. |
| 16:17.32 | lokkju_wrk | TaiSHi, didn't you just say you don't know IAX? |
| 16:17.34 | TripleF_W | anyone have callerid probs ? with ths scenraio.. client phone -> trix (rpid/sendrpid enab)-> (rpid/sendrpid) itsp (*) |
| 16:17.52 | lokkju_wrk | TaiSHi, IAX is supposed to use the port that the communication was sent from |
| 16:19.27 | *** join/#asterisk qdk (n=qdk@193.164.155.44) |
| 16:19.50 | mercestes | TaiSHi: IAX2 always goes over port 4569. It makes no sense to have a port=variable if the port is always 4569 |
| 16:20.30 | lokkju_wrk | mercestes, IAX2 always goes over 4569 for INBOUND to the IAX2 server - but the client is supposed to be able to use *any* port, right? |
| 16:20.38 | *** join/#asterisk mkrufky (n=mk@unaffiliated/mkrufky) |
| 16:20.40 | TaiSHi | Ah |
| 16:20.43 | TaiSHi | Read my mind lokkju_wrk |
| 16:20.46 | lokkju_wrk | (client send from any port, that is) |
| 16:21.01 | TaiSHi | Client can use the port he wants, that's what you ment |
| 16:21.03 | mkrufky | does anybody know if digium has plans to sell PCIe digital cards? |
| 16:21.11 | lokkju_wrk | and that port is dynamically provisioned by either your system, and/or the NAT devices |
| 16:21.24 | mercestes | ...... |
| 16:22.26 | mercestes | I have only known IAX2 to go over 4569. |
| 16:22.48 | lokkju_wrk | mercestes, that would mean two IAX2 clients could not be behind the same NAT |
| 16:22.48 | mercestes | I guess if you *wanted* to ruin that perfect happiness by screwing with it.....you could. But..not by default. |
| 16:23.01 | mercestes | .... |
| 16:23.05 | mkrufky | dell's new servers.... at least the product line that we use here, are no longer shipping with PCI slots.... they're all PCI-X or PCIe .... would be nice to know that I can buy and new Dell 1U rack server and know that I can still get a 4-port pri card from digium to work with it |
| 16:23.08 | mercestes | NAT ports != IAx2 ports. |
| 16:23.15 | lokkju_wrk | the whole idea of IAX is that the server responds back along the channel opend by the client |
| 16:23.38 | mercestes | the whole idea of IAX2 is to use one friggin port so Firewalls don't twitch and die when they see voip coming at them. |
| 16:23.45 | mercestes | if you want random ports, use SIP. |
| 16:24.03 | lokkju_wrk | mercestes, that is on the server side, not client |
| 16:24.59 | mercestes | Ok. |
| 16:25.26 | lokkju_wrk | ah, I see what you are talking about |
| 16:25.57 | coppice | mkrufky: a PCI-X slot is the same as PCI slot when you don't enable the higher speed |
| 16:25.59 | mercestes | I'm glad. |
| 16:26.10 | lokkju_wrk | both the client and server use 4569, but the server is supposed to respond back to the port that it was recieved from (in case of a NAT device) |
| 16:26.43 | coppice | mrrufky: but they are all 64 bit slots, so you need a card that can run from 3.3V |
| 16:28.13 | mercestes | lokkju_wrk: I am not intimately familiar with the level 3 NAT protocol but what you end up with is a dynamic publicIP:assignedPort pointing to a InternalIP:4569 on one or both ends. |
| 16:28.18 | mkrufky | coppice: ah, so a 405P will work in a PCI-X |
| 16:28.19 | mkrufky | ? |
| 16:28.38 | lokkju_wrk | mercestes, right... now if I could figure out why * was not responding back to that port... |
| 16:28.49 | coppice | isn't the 405P the one that runs from 5V? You need a 410P |
| 16:29.11 | mkrufky | lol, then i got them mixed up |
| 16:29.25 | mkrufky | but yes, that helps very much, coppice ... thank you |
| 16:29.29 | mercestes | lokkju_wrk: to 4569? |
| 16:29.37 | Qwell[] | mkrufky: I would suggest calling Digium Sales if you aren't sure of something |
| 16:29.47 | mercestes | lokkju_wrk: Or the Nat port? I'm pretty sure responding to the proper NAT port is your router's job. |
| 16:30.07 | Qwell[] | mkrufky: They'd be more than happy to explain things to you if needed |
| 16:30.25 | mkrufky | Qwell[]: yes, of course... just shopping around for a new server now. calling digium was the next step |
| 16:30.42 | mkrufky | they're always very helpful on the telephone |
| 16:30.46 | lokkju_wrk | mercestes, the Asterisk is not behind a router - my setup is: iax2 client (idefisk) -> NAT (linksys) -> * |
| 16:30.51 | mercestes | lokkju_wrk: I guess if you wanted to submit some logs/packet traces and your configs that gave a good indication that IAX does not work over NAT, some developers here would be interested in reading it. |
| 16:31.18 | TripleF_W | so any reason * is not using the RPID.. its in the header but ${CALLERID} still shows the username part.. ( they used formuser) |
| 16:31.32 | lokkju_wrk | mercestes, but it *should* work |
| 16:31.38 | mercestes | lokkju_wrk: I agree. |
| 16:31.59 | mercestes | lokkju_wrk: Which linksys? |
| 16:32.05 | lokkju_wrk | is anyone else behind a nat device, and willing to try registering with my box? |
| 16:32.13 | lokkju_wrk | mercestes, openwrt (wrt54g) |
| 16:32.26 | lokkju_wrk | essentially, at this point, just an iptables nat config |
| 16:32.50 | mercestes | lokkju_wrk: Yea, I have mixed feelings on Openwrt. Difficult enough to give me that cool feeling when I figure something out, without the satisfaction of feeling productive. |
| 16:33.03 | mercestes | lokkju_wrk: And the wireless is about as flakey as Ana-Nicole.. |
| 16:33.20 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 16:33.20 | lokkju_wrk | mercestes, eh, it is better then the stock firmware, by a long shot |
| 16:33.51 | mercestes | lokkju_wrk: heh, Jury is out on that one. I love that it's linux but it's an hour long research project everytime I wanna fix something. |
| 16:33.59 | lokkju_wrk | but I just wish I could find some info on troubleshooting IAX |
| 16:34.09 | mercestes | lokkju_wrk: Anyways, see if you can see what the packets are doing going through the router. Try some port foward/opens and see if you can get it going. |
| 16:34.14 | lokkju_wrk | because it is driving me nutty right now |
| 16:34.16 | mercestes | lokkju_wrk: Or setup a static internal IP with static natting. |
| 16:34.47 | lokkju_wrk | mercestes, the whole point is to not have to do that - I want to use IAX so I don't have to worry about NAT devices |
| 16:34.59 | mercestes | TaiSHi: You can pay in blood if you wish. I also accept most global currencies and I offer an exchange rate for virgins. |
| 16:35.04 | [TK]D-Fender | TaiSHi: I don't *think* mercestes swings that way... perhaps you could ask bkw_ ;) |
| 16:35.24 | TaiSHi | bkw_, ;) |
| 16:35.26 | *** join/#asterisk svenna_ (n=svenna@p548d3cbb.dip0.t-ipconnect.de) |
| 16:35.34 | mercestes | [TK]D-Fender: I think he meant like raw meat.......he better have meant that atleast. |
| 16:35.46 | [TK]D-Fender | :D |
| 16:36.13 | [TK]D-Fender | <- Two tierd conversation? Nah... 20 layer cake :) |
| 16:36.15 | [TK]D-Fender | mmmmm CAKE |
| 16:36.22 | mercestes | lokkju_wrk: Well, there isn't much documented on troubleshooting IAX2 because there isn't much to troubleshoot. most things that go wrong are config file issues. |
| 16:37.06 | c4t3l | is there an isp issue of blocked ports?? eh? I was in the shower so i might have missed tht part |
| 16:37.19 | *** join/#asterisk ars247 (n=no@64-142-43-180.dsl.static.sonic.net) |
| 16:37.45 | mercestes | c4t3l: We don't know, he doesn't want to do static natting to troubleshoot it. |
| 16:37.54 | c4t3l | ahh |
| 16:38.07 | mercestes | lokkju_wrk: Install asterisk on the router and connect to the router. |
| 16:38.29 | mercestes | lokkju_wrk: Your using openwrt. Hell, for that matter, connect your client to the router too while your at it and just pass through. |
| 16:38.33 | c4t3l | well i know in our area that 4569 is blocked by TWC for their "digital phone" service |
| 16:38.49 | c4t3l | hehe |
| 16:39.18 | mquin | c4t3l: now that's petty |
| 16:39.18 | mercestes | c4t3l: But, c4t3l! Why shouldn't we use Roadrunner as a really cheap dedicated line for VoIP phone service???? |
| 16:39.30 | c4t3l | :) |
| 16:39.48 | *** join/#asterisk Ebola (n=Ebola@host86-143-156-147.range86-143.btcentralplus.com) |
| 16:41.02 | gambolputty | Port blocking of that nature is why we need Net Neutrality. |
| 16:41.16 | *** join/#asterisk JoelSolanki (i=Joel@220.224.43.241) |
| 16:41.34 | mercestes | gambolputty: no, we need net dictatorship. *one* person...in charge of it all...with assassination rights. |
| 16:41.51 | mercestes | gambolputty: may not be fair....but we'd have none of the BS and we'd all know what to expect. |
| 16:42.09 | coppice | how do US telcos get away with capacity caps they refuse to tell you? aren't there consumer protection laws about that kind of misrepresentation? |
| 16:42.14 | TaiSHi | Ok guys |
| 16:42.16 | TaiSHi | Im gonna logoff |
| 16:42.23 | TaiSHi | Take care |
| 16:42.27 | TaiSHi | Hope we see again... |
| 16:42.51 | mercestes | coppice: blame the PUC. |
| 16:42.54 | JoelSolanki | we are using g729 from digium but our clients have linksys pap2 and they want to use g723 for 2 channels. how can we have g723 support in asterisk ? |
| 16:43.16 | mercestes | JoelSolanki: tell them no. |
| 16:43.35 | mercestes | and why are you using G729? Use ulaw. It's warm and tasty |
| 16:43.48 | *** join/#asterisk Penggu (i=foobar@220-245-200-87.static.tpgi.com.au) |
| 16:43.53 | Penggu | is there anyw ay to go to a context without specifying the priority/extensions ? |
| 16:44.49 | fordfrog | anybody knows how to make an inbound call ring both on internal sip line and on a cell phone at the same time? |
| 16:45.07 | JoelSolanki | so there is no way of using g723 in asterisk ? |
| 16:45.11 | Strom_M | JoelSolanki: the TC400B is, AFAIK, the only way to transcode G.723 in asterisk is to use the TC400B card |
| 16:45.13 | gambolputty | the dial command can dial more than one number at the same time |
| 16:45.19 | Penggu | Dial(X&Y) |
| 16:45.55 | mercestes | fordfrog: dial(Sip/2134&Zap/7131234456) |
| 16:46.09 | coppice | JoelSolanki: it does g.723.1 passthrough. G.723.1 is kinda obsolete and expensive to licence. people have little incentive to implement it |
| 16:46.41 | *** join/#asterisk giasai68 (n=giasai@ip-240-130.sn2.eutelia.it) |
| 16:46.45 | *** join/#asterisk gr1ncheux (n=devine@unaffiliated/gr1ncheux) |
| 16:47.02 | giasai68 | hello |
| 16:47.23 | giasai68 | I'm tring to make a call trougth SIP protocol |
| 16:47.27 | fordfrog | mercestes: I need to look what Zap means :-) |
| 16:47.49 | giasai68 | when I make a call there is this error: http://pastebin.ca/397591 |
| 16:47.51 | JoelSolanki | ok |
| 16:47.53 | mercestes | fordfrog: "Phone line" basically. if your dialing a cell phone yo uhave to go out analog, pri, bri, isdn, or tin can with a string. |
| 16:47.56 | giasai68 | pls, help me |
| 16:47.59 | JoelSolanki | g723 to g729 works ?? |
| 16:48.06 | mercestes | fordfrog: more properly, anything using the zaptel driver. |
| 16:48.28 | Qwell[] | JoelSolanki: it can, but only indirectly. You'll have to go from g729 to ulaw/alaw to g723 - thereby using two channels |
| 16:48.31 | coppice | transcoding g723.1 to g729 sounds pretty horrible |
| 16:48.41 | Qwell[] | coppice: at best |
| 16:48.45 | mercestes | JoelSolanki: no, it doesn't. It's a federal offense and yoru clients could be jailed if you allowed them to implement such a thing. They should thank you for stopping them frmo attempting it. |
| 16:48.47 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 16:49.26 | JoelSolanki | oh k. |
| 16:49.32 | fordfrog | mercestes: the asterisk I'd like to configure this at has just sip providers, no other protocols, so I suppose I should route it somehow back to the cell phone via sip provider |
| 16:49.32 | mercestes | JoelSolanki: OK, not really, but..it'd save you some pain if you told them that. |
| 16:49.42 | coppice | pretty much any transcoding from one low bit rate voice codec to another sounds nasty |
| 16:49.43 | mercestes | fordfrog: That would be correct. |
| 16:50.15 | mercestes | fordfrog: Then it would be Dial(SIP/1234&Sip/7131234567@myserver.com:autheticate |
| 16:50.23 | JoelSolanki | my voip provider support g723 |
| 16:50.42 | JoelSolanki | can i pass g723 of linksys to my provider and asterisk in between ? |
| 16:50.54 | coppice | JoelSolanki: many do, but its really a hangover from the past |
| 16:51.11 | fordfrog | mercestes: ah, and will this propagate callers phone number to the cell phone? I cannot try it atm. |
| 16:51.16 | Qwell[] | You can do passthrough, but not any transcoding - ie; no applications like voicemail unless you convert all of your prompts to g723 first |
| 16:51.52 | JoelSolanki | i dont have voicemail service. i have only outbound. |
| 16:52.07 | JoelSolanki | Qwell: so u think g723 pass thru should work right ? |
| 16:52.15 | Qwell[] | it should, but ymmv |
| 16:53.20 | JoelSolanki | hmm |
| 16:54.22 | JoelSolanki | so for implementing g723 pass thru i need to modify sip.conf and enter following parameters.. disallow=all allow=G723.1 allow=g729 |
| 16:54.22 | JoelSolanki | right ? |
| 16:54.27 | *** join/#asterisk angryuser (n=Miranda@LPuteaux-151-43-4-37.w217-128.abo.wanadoo.fr) |
| 16:54.31 | angryuser | good day |
| 16:54.33 | Qwell[] | no, why would you allow g729? |
| 16:54.51 | JoelSolanki | i want both g729 and g723 |
| 16:54.58 | JoelSolanki | cant both work ? |
| 16:55.00 | angryuser | i saw module embedding options in menuselect of asteris 1.4.1 what is it for? |
| 16:55.02 | Qwell[] | Does your provider support g729? |
| 16:55.14 | JoelSolanki | yes my provider support both g729 and g723 |
| 16:55.50 | JoelSolanki | qwell: is it possible? |
| 16:56.11 | Qwell[] | should be, but, you really should do a bit of research into the issues involved |
| 16:56.33 | JoelSolanki | oh. have u noted any issues ? |
| 16:56.42 | mercestes | fordfrog: If that means "call them" then yes |
| 16:56.57 | giasai68 | when I make a call with SIP protocol there is this error: http://pastebin.ca/397591 |
| 16:57.00 | giasai68 | pls, help me |
| 16:57.18 | JoelSolanki | qwell: have u noted any error ? |
| 16:57.28 | fordfrog | mercestes: thank you, I'll try it next week |
| 16:57.34 | Qwell[] | JoelSolanki: yes, when people don't really understand what they're doing |
| 16:58.17 | *** join/#asterisk CunningPike_ (n=CunningP@dhcp-10-153.district.north-van.bc.ca) |
| 16:58.27 | JoelSolanki | :) |
| 16:58.38 | coppice | JoelSolanki: It sounds like everything involved supports G.729. You'd be crazy not to use it |
| 16:58.43 | *** part/#asterisk NewbePaul (n=paul@adsl-072-148-241-244.sip.asm.bellsouth.net) |
| 16:59.59 | JoelSolanki | coppice: problem is that linksys pap2 / cisco ata 186 and lot of other equipment can do concurrent 1 g729 only |
| 17:00.14 | JoelSolanki | therefore we are forced to do something to use g723 :( |
| 17:00.16 | coppice | those only do 1 G.723.1 too |
| 17:00.23 | Qwell[] | one of one or the other |
| 17:01.05 | coppice | basically their DSP isn't up to it, so they can only handle one channel of a high complexity codec |
| 17:01.14 | JoelSolanki | no g723 can be done 2 in those equpment |
| 17:01.27 | coppice | look again |
| 17:01.59 | JoelSolanki | i will do that. |
| 17:02.35 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:04.25 | *** join/#asterisk dahunter3 (n=dahunter@static-71-189-121-177.lsanca.dsl-w.verizon.net) |
| 17:05.01 | dahunter3 | Anyone know if I can configure a digium Te110p card to work with what verizon refers to as DIOD lines with, all they term as "wink"? |
| 17:07.01 | phearless | hello folks ! |
| 17:07.01 | phearless | How can I do a "call forwarding" if I receive a call to my direct extension, not from a "group call" ? |
| 17:11.08 | Strom_M | dahunter3: yes |
| 17:11.42 | Strom_M | dahunter3: though you would probably be better off getting a PRI |
| 17:11.42 | Penggu | anyone know of a command line util to do ini-style modifications? (sip.conf..) |
| 17:11.56 | dahunter3 | strom_M: Yeah, I hear that, but the quote they gave me for 4 lines was $200 one way and $650/month for pri |
| 17:13.00 | dahunter3 | strom_M: Any idea how to configure that in zaptel.conf and zapata.conf? Is it e&m=1-4 and then in zapata em_w=1-4 or something different (I'm assuming because that doesn't seem to work-- asterisk detects the lines as busy). |
| 17:15.36 | HexDump | bye all! |
| 17:16.11 | mercestes | dahunter3: get a different PRI quote then. |
| 17:16.39 | [TK]D-Fender | phearless: ...huh? |
| 17:17.18 | phearless | [TK]D-Fender: I can receive a call to my extension (404) |
| 17:17.35 | phearless | [TK]D-Fender: or I can receive a call that everybody receive, a group call |
| 17:17.49 | mercestes | phearless: YOu mean like a analog line, and you wan tto recieve a call, flash the line, and then send it back out over "line two" of hte hook flash? |
| 17:17.51 | phearless | [TK]D-Fender: I would like to forward ONLY direct calls |
| 17:18.09 | phearless | mercestes: let me think about this |
| 17:18.11 | mercestes | phearless: Oh. That's a dialplan thing |
| 17:18.22 | [TK]D-Fender | phearless: Its your dialplan, go shove an ASTDB lookup before your dial to see if you actually want to ring that phone. |
| 17:18.31 | mercestes | phearless: Just match direct calls v/s group calls in yoru dialplan |
| 17:18.38 | [TK]D-Fender | phearless: And make a little IVR so that you can set the forwarding value |
| 17:18.46 | phearless | okay guys |
| 17:18.58 | phearless | it a bit complex but I will try :) |
| 17:19.28 | phearless | I got Sipura 942 phones, so I can not use the call forwarding button of it ? |
| 17:19.38 | [TK]D-Fender | phearless: That isn't SELECTIVE |
| 17:19.42 | phearless | if I use this button ALL the calls are forwarded |
| 17:19.58 | [TK]D-Fender | phearless: Good... you are beginning to grasp the problem :) |
| 17:20.05 | phearless | okay |
| 17:20.10 | [TK]D-Fender | phearless: So time to learn how to work the dialplan |
| 17:20.31 | phearless | [TK]D-Fender / mercestes : thanks, I will work on the dialplan :) |
| 17:21.16 | mercestes | goodluck |
| 17:21.44 | b11d | . |
| 17:24.50 | phearless | oh |
| 17:25.00 | phearless | and by the way I got another question for the experts |
| 17:25.14 | [TK]D-Fender | phearless: "shoe function DB" , "show application gotoif" |
| 17:25.27 | mercestes | [TK]D-Fender: R OFLMAO |
| 17:26.12 | [TK]D-Fender | show* |
| 17:26.13 | phearless | when I receive a call, and I want to forward it to 411, and 411 is on "DnD mode", the call is cut |
| 17:26.32 | phearless | is there any smart ways to deal with it ? |
| 17:27.08 | [TK]D-Fender | phearless: well if you even CALL 411 the call is cut, no? |
| 17:27.21 | [TK]D-Fender | phearless: Should not only be because of forwarding |
| 17:27.41 | mercestes | phearless: Disable DND |
| 17:29.05 | phearless | when i call 411, I got the voicemail if 411 is on DnD |
| 17:29.05 | phearless | but when I redirect to it, it cuts the call |
| 17:29.05 | phearless | no voicemail |
| 17:29.08 | mercestes | phearless: does it work if DND is off? |
| 17:29.34 | phearless | yes it works |
| 17:29.39 | [TK]D-Fender | phearless: pastebin your dialplan for that extension. |
| 17:29.55 | mercestes | how are you group calling?? |
| 17:30.17 | phearless | I group call with the IVR |
| 17:30.30 | phearless | exten => 2,n,Queue(salesqueue|tTr|||10) |
| 17:30.33 | phearless | for ex like this |
| 17:30.37 | [TK]D-Fender | mercestes: I'm sure you're smelling what I'm smelling here... |
| 17:30.39 | phearless | [TK]D-Fender: ok 1s |
| 17:30.47 | mercestes | Queues + DND == :( |
| 17:30.52 | [TK]D-Fender | mercestes: Although I have been mistaken... |
| 17:31.04 | [TK]D-Fender | phearless: Indeed Queues do NOT like DND |
| 17:31.10 | phearless | ok [TK]D-Fender |
| 17:31.24 | [TK]D-Fender | phearless: get to that pastebin! |
| 17:34.35 | phearless | http://paste.lisp.org/display/38248/raw |
| 17:34.38 | phearless | it is the whole file |
| 17:34.42 | phearless | (sorry) |
| 17:35.31 | mercestes | you do realize that ";" is a comment, right? |
| 17:35.48 | [TK]D-Fender | phearless: And you do realize that you have exten => 411,1,Dial(SIP/411,30,tT) |
| 17:35.57 | [TK]D-Fender | and |
| 17:35.58 | phearless | noooo |
| 17:35.58 | [TK]D-Fender | exten => _4[0-2]X,1,Dial(SIP/${EXTEN},10,tT) |
| 17:36.03 | [TK]D-Fender | in the same damn context right? |
| 17:36.04 | phearless | 411 was an example for IRC |
| 17:36.27 | [TK]D-Fender | phearless: Give us SPECIFIC examples of problems ok. |
| 17:36.33 | phearless | 411 is a wireless phone |
| 17:36.37 | phearless | yes sorry |
| 17:36.44 | phearless | I will re-do a test |
| 17:37.04 | [TK]D-Fender | phearless: And STILL.... you have 2 patterns that match the same number in the same context! |
| 17:37.08 | [TK]D-Fender | phearless: that = BAD |
| 17:37.15 | mercestes | I coun t3 |
| 17:37.20 | mercestes | I count 3, rather |
| 17:37.46 | [TK]D-Fender | mercestes: ";include => errors" <- phew.. thank God its commented out! |
| 17:37.52 | mercestes | rofl |
| 17:37.54 | phearless | :D |
| 17:37.57 | mercestes | It's mostly comments |
| 17:38.08 | phearless | I can paste it wothout the comments |
| 17:38.12 | phearless | without |
| 17:38.15 | mercestes | It would be helpful. |
| 17:38.19 | phearless | no prob |
| 17:38.30 | mercestes | you want our comments on your file...we don't want yours. |
| 17:38.59 | [TK]D-Fender | mercestes: lol |
| 17:40.12 | phearless | http://rafb.net/p/SXygvt48.html |
| 17:40.19 | phearless | [TK]D-Fender mercestes here it is |
| 17:41.10 | [TK]D-Fender | phearless: Ok well what is the SPECIFIC example of a failed call? |
| 17:41.29 | phearless | I will do it now :) |
| 17:42.04 | [TK]D-Fender | phearless: And god God's sake get rid of that extension overlap in [local] |
| 17:42.33 | [TK]D-Fender | phearless: Which I am presently blaming for your problem |
| 17:43.42 | *** join/#asterisk thinwires (n=thinwire@ny-lancastercadent4g7-9d-77.buf.adelphia.net) |
| 17:44.20 | thinwires | hey guys, I need some personal opinions on phones, does anyone here use the Cisco CP-7941G phones? |
| 17:44.37 | phearless | [TK]D-Fender: http://rafb.net/p/DCZPz998.html |
| 17:44.43 | phearless | here is the log |
| 17:45.01 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 17:45.01 | phearless | 408 receive a call and FW it to 404 |
| 17:45.05 | *** join/#asterisk topping (n=topping@204.152.96.238) |
| 17:45.05 | phearless | 404 is on DnD |
| 17:45.23 | phearless | but when somebody calls 404 we got the voicemail |
| 17:47.17 | mercestes | Queues + DND = :( |
| 17:47.25 | mercestes | Here's a scenario. Let's use your girlfriend. |
| 17:47.28 | phearless | there's no queues |
| 17:47.36 | mercestes | ? |
| 17:47.54 | phearless | a mobile call the IVR and dial 408 |
| 17:48.30 | phearless | I got "include => local" in [incoming] |
| 17:49.09 | phearless | it is strange because the logs say : |
| 17:49.10 | phearless | <PROTECTED> |
| 17:49.20 | [TK]D-Fender | phearless: Ummm.. ok, you get voicemail... I see that you SHOULD. whats the problem? |
| 17:49.23 | phearless | but in fact in the mobile we never hear the voicemail message |
| 17:49.49 | mercestes | wait. |
| 17:49.56 | mercestes | are you trying to get voicemail on a mobile phone? |
| 17:50.10 | phearless | voicemail of 404 |
| 17:50.27 | phearless | mobile -call-> 408 -fw-> 404 |
| 17:51.00 | mercestes | <PROTECTED> |
| 17:51.07 | mercestes | you don't hear vm-intro? |
| 17:51.32 | [TK]D-Fender | mercestes: And in that pastebin you can see that 404 ANSWERS the call. It looks like a manual "transfer", not a "forward" |
| 17:51.39 | [TK]D-Fender | er... 408* |
| 17:51.54 | [TK]D-Fender | <PROTECTED> |
| 17:51.56 | [TK]D-Fender | <PROTECTED> |
| 17:52.15 | mercestes | What I want to know is how MOH is being played across Zap2-1 |
| 17:52.25 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 17:52.43 | [TK]D-Fender | mercestes: Like I said, it looks like they answer the phone and are manually starting a transfer. |
| 17:52.56 | [TK]D-Fender | mercestes: Which we have to proof is being properly finished |
| 17:53.00 | phearless | ok !! |
| 17:53.06 | mercestes | *nods* |
| 17:53.17 | mercestes | I see here that VM-intro is being played before MoH ends tho |
| 17:53.23 | mercestes | and then that error about the owned channel |
| 17:53.27 | phearless | in fact to transfer I do : [XFER] then I dial the number then I re-press [XFER] |
| 17:53.32 | [TK]D-Fender | mercestes: Corroborates everything doesn't it? ;) |
| 17:53.38 | mercestes | Indeed. |
| 17:53.57 | phearless | and when I dial the number I (408) can hear the voicemail |
| 17:53.59 | [TK]D-Fender | phearless: That sounds like an ATTENDED transfer, not a BLIND transfer |
| 17:54.02 | phearless | yes |
| 17:54.06 | [TK]D-Fender | phearless: Go read your phone's manual. |
| 17:54.06 | mercestes | ohhhh. |
| 17:54.07 | phearless | then I tried a blind transfer |
| 17:54.13 | phearless | and with a blind transfer it works |
| 17:54.20 | mercestes | He's hitting "transfer" again before it even plays back VM-Intro |
| 17:54.28 | phearless | yes mercestes |
| 17:54.36 | mercestes | so he's *dumping* the channel being fed MoH into the extension with DND. |
| 17:54.40 | thinwires | so, have you had any experiances with activating cisco phones to be used with SIP? |
| 17:54.51 | mercestes | phearless: yea, don't do that. |
| 17:55.53 | phearless | it is annoying because the blind Xfer button is a bit hidden on the screen of the phone |
| 17:56.02 | kink0 | I still unable to set worst over 99.98 : Best: 100.000000 -- Worst: 99.938965 -- Average: 99.981384 |
| 17:56.02 | phearless | and the attender xfer button is diplayed |
| 17:56.11 | kink0 | any suggestion ? |
| 17:56.18 | phearless | to do a blind one I have to press right, then blind xfer |
| 17:56.22 | phearless | quite boring |
| 17:56.26 | mercestes | .... |
| 17:56.32 | mercestes | What are you trying to do??? |
| 17:56.44 | kink0 | I did: try other computer, enable/dissable HT, recompile kernel, assign IRQ to other CPU , but the same. |
| 17:56.47 | [TK]D-Fender | phearless: like Don Henley says "Get over it" |
| 17:57.23 | JoNate | I hate snow |
| 17:57.47 | [TK]D-Fender | kink0: Whats the actual PROBLEM? |
| 17:58.02 | [TK]D-Fender | JoNate: I'm about to get buried under the white shit tonight... |
| 17:58.09 | kink0 | [TK]D-Fender, a bad worst |
| 17:58.14 | JoNate | TK: Where at? I'm in Jersey |
| 17:58.56 | [TK]D-Fender | JoNate: Montreal, QC |
| 17:59.11 | JoNate | TK:OI! Your getting it alot worse than I am! |
| 17:59.12 | kink0 | [TK]D-Fender, anyway I have not appreciate any audio quality loss at least with about 30 channels up, I am not sure if would affect while over 60 channels up |
| 17:59.29 | [TK]D-Fender | kink0: Thats jsut a NUMBER. What is the actual problem that it causes (excluding your morosing over it) :) |
| 18:00.00 | [TK]D-Fender | kink0: Sounds like you're worrying about problems that don't exist yet\ |
| 18:00.29 | kink0 | [TK]D-Fender, yes, I start be worried about that one day we lost D channel |
| 18:00.57 | kink0 | but I have not lost more D channels |
| 18:01.19 | [TK]D-Fender | kink0: What card, protocols, and setup are you running? |
| 18:01.21 | *** join/#asterisk nays85 (i=nays85@shell.thehostbusters.com) |
| 18:01.31 | *** join/#asterisk budmang (n=budman@12-210-54-193.client.mchsi.com) |
| 18:01.32 | Hmmhesays | bah something is messed up here /usr/bin/ld cannot find -lqt |
| 18:01.50 | budmang | any problems with running zaptel 1.4 and asterisk 1.2? |
| 18:01.51 | b11d | ldconfig -R |
| 18:01.54 | kuku5 | Anyone looking for a dedicated server ? |
| 18:01.58 | kink0 | [TK]D-Fender, TE405, euroisdn, span=1,1,1,ccs,hdb3,crc4 |
| 18:01.58 | budmang | kuku5 |
| 18:02.02 | budmang | why do you keep asking that |
| 18:02.27 | b11d | lets go fire up some of that bud mang.. |
| 18:02.33 | [TK]D-Fender | budmang: Nope, and you can use 2007 Mustang parts on your 1967 Shelby just fine too |
| 18:02.39 | Hmmhesays | cause he can't eat unless he sells servers |
| 18:02.44 | [TK]D-Fender | kink0: Sure you need that LBO? |
| 18:02.47 | kink0 | [TK]D-Fender, actually there 2 spans, conected to telco, ussing balum 75/120 ohms |
| 18:03.03 | kink0 | [TK]D-Fender, no, really not sure |
| 18:03.15 | budmang | [TK]D-Fender works fine for me I was just wondering if anyone had any issues |
| 18:03.15 | budmang | damn |
| 18:03.17 | [TK]D-Fender | kink0: How far is your server from your smartjack? |
| 18:03.39 | [TK]D-Fender | budmang: I seriously advise you to run the MATCHING version for your build of Asterisk |
| 18:03.41 | kink0 | about 3 meters |
| 18:03.43 | *** join/#asterisk lokkju_wrk_ (n=lokkju@unaffiliated/lokkju) |
| 18:03.49 | mercestes | kuku5: Oo, I am, I am! can I post you my credit card info here? |
| 18:03.57 | [TK]D-Fender | kink0: I'd advise 1,1,0 |
| 18:04.02 | kink0 | [TK]D-Fender, do you think would be wires length ? |
| 18:04.03 | mercestes | kuku5: IRC being the appropriate and effective marketing tool it is of course. |
| 18:04.08 | funxion | I upgraded my kernel and asterisk and now my tdm400p is seen as the first 4 channels where before it was at the end. Anyone know why that would happen and how I can get it to go back? |
| 18:04.08 | budmang | [TK]D-Fender: thanks |
| 18:04.29 | kink0 | yes, was normally 0, I just try 1 thinking about any problem with wiring |
| 18:04.45 | kink0 | no any noticiable changes if I set 0 or 1 |
| 18:05.34 | kuku5 | mercestes: :) Just starting out slowly - I have good bandwith thats all. |
| 18:05.58 | mercestes | kuku5: Someone promised me a monster server for asterisk work. That was *almost* on topic. |
| 18:06.09 | kuku5 | hehe |
| 18:06.10 | kink0 | other question, how is possible I still about 80% idle CPU ( Dual Xeon 2.8 ) while near 30 channels ussing g729 and soft ec ? |
| 18:06.10 | mercestes | ...he hasn't *paid* me yet, btw...... |
| 18:06.31 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 18:06.35 | funxion | I upgraded my kernel and asterisk and now my tdm400p is seen as the first 4 channels where before it was at the end. Anyone know why that would happen and how I can get it to go back? |
| 18:06.36 | kink0 | that would means one dual 2.8 would be able to carry over 100 simultaneous calls or so |
| 18:06.37 | mercestes | kink0: Your transcoding. Transcode less. |
| 18:06.45 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 18:06.46 | kuku5 | they promised you a server - so why did they have to pay |
| 18:06.49 | Dr-Linux | hi guys |
| 18:06.57 | *** join/#asterisk friedrich| (n=friedric@e177250137.adsl.alicedsl.de) |
| 18:07.00 | *** join/#asterisk Preytell (n=Preytell@seraph.contegix.com) |
| 18:07.04 | mercestes | kuku5: Getting the server is payment. |
| 18:07.16 | kink0 | mercestes, that is, I was expecting, as I read about bechmarks for digium cards, I would no able to pass over about 80 channels |
| 18:07.25 | mercestes | I accept all global currency, blood, souls, pets, and I offer an exchange rate in virgins. |
| 18:07.27 | kink0 | with this dual 2.8 |
| 18:07.45 | thinwires | I have a puppy, would that suffice? |
| 18:07.47 | mercestes | kink0: Psh. You could pass far more than that if you set it up right. |
| 18:07.57 | mercestes | thinwires: human pets. not dogs. ew. |
| 18:08.24 | thinwires | oh right right... well I'm sure I could rent a van and find some hoo....nice lady's |
| 18:08.26 | kink0 | mercestes, anywise I Choose DSP ec for the next cards, to help CPU load |
| 18:08.42 | mercestes | kink0: or you could transcode less. |
| 18:08.54 | kink0 | mercestes, how much calls do you estimate one dual 2.8 xeon would can do ? |
| 18:09.02 | mercestes | kink0: Try retranscoding your audio files into whatever codec your using globaly...you are using 1 codec everywhere, right? ulaw hopefully? |
| 18:09.06 | kink0 | no, no, I need g729 |
| 18:09.13 | mercestes | ... |
| 18:09.23 | mercestes | I need herpes in my face. Why do you need g729? |
| 18:09.24 | kink0 | g729 -> ulaw |
| 18:09.33 | mercestes | kink0: That would be transcoding. |
| 18:09.34 | funxion | why cant you send g729 |
| 18:09.50 | mercestes | kink0: And that's why you are using 20% cpu, and why you can't pass more than 80 calls. |
| 18:09.57 | kink0 | because I accept g729 from the IP world, and i need to send ulaw to the span |
| 18:10.08 | mercestes | kink0: And if you believe that 100% of your CPU is utilizable in a productive and stable manner.....*points and laughs* |
| 18:10.17 | thinwires | is ulaw the best for non-transcoding codecs? |
| 18:10.25 | mercestes | kink0: Your doomed to failure. |
| 18:10.38 | [TK]D-Fender | thinwires: Depends whats on the other side of the call. |
| 18:10.49 | funxion | are you using the IVR at all or just passing calls and an ip gateway |
| 18:10.51 | mercestes | thinwires: ulaw is the best codec. Not transcoding is better. you can not transcode in g729 for all I care..but..g729 is the devil. |
| 18:10.54 | kink0 | of course I assume I can not use 100% of CPU, I start this asterisk now with -M 80 |
| 18:11.13 | funxion | I like g729 |
| 18:11.17 | mercestes | funxion: Just remix the audio files in ulaw or g729 or whatever so the audio files are pretranscoded. Viola. |
| 18:11.30 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 18:11.32 | mercestes | funxion: Satanist! |
| 18:11.32 | funxion | yes Im aware |
| 18:11.43 | kink0 | yes, g729 is very bad if you see thinking on cpu load, but is generally accepted as good compression codec |
| 18:11.44 | thinwires | I have gsm running now :-o I have my own astrisk hooked to a provider and my phones to my astisk around different locations (home IP phones) |
| 18:12.08 | funxion | mercestes its for bw savnig purposes |
| 18:12.19 | funxion | saving |
| 18:12.54 | kink0 | anywise I am not ussing any audio files, just voIP termination, and when busy, congestion, etc.. I just return the cause |
| 18:12.55 | funxion | mercestes by any chance did you read my question earlier? |
| 18:13.18 | mercestes | funxion: probably. |
| 18:13.34 | funxion | kink0 what alse are you running and what card do you have |
| 18:13.38 | funxion | I upgraded my kernel and asterisk and now my tdm400p is seen as the first 4 channels where before it was at the end. Anyone know why that would happen and how I can get it to go back? |
| 18:13.38 | mercestes | funxion: Oh, I thought to myself "config file error" But I figured someone else would deliver hte message. |
| 18:13.59 | funxion | all the same configs |
| 18:14.06 | mercestes | funxion: Bu tI have to ask, what do you mean "first 4 channels" btw? |
| 18:14.10 | funxion | I went from asterisk 1.2.10 1.2.14 |
| 18:14.20 | thinwires | so does anyone here use Cisco IP phones? |
| 18:14.21 | funxion | channels 1-4 |
| 18:14.25 | mercestes | funxion: Are you using multiple interfaces as one massive zap channel? |
| 18:14.27 | kink0 | funxion, no any other softwares are running, the other are kernel, sshd and of course disk interrups ( SCSI ) |
| 18:14.31 | funxion | I also have a te205p |
| 18:14.32 | [TK]D-Fender | funxion: Fix your module load order |
| 18:14.41 | funxion | thnx |
| 18:14.48 | kink0 | funxion, the card is TE405, but I planned to migrate to TE412P to use DSP for ec |
| 18:14.52 | funxion | which modules should I look at |
| 18:15.25 | [TK]D-Fender | funxion: I'd say WCTDM & WCT4XXP |
| 18:15.31 | [TK]D-Fender | just a guess..... |
| 18:15.47 | [TK]D-Fender | er, maybe thats WCT2XXP for that card |
| 18:15.51 | funxion | kink0 I have that same card passing 96 simultaneous calls from g729 on a centrino dual core 1.8 and 1 gb RAM no problem |
| 18:16.15 | funxion | [TK]D-Fender I load those in the proper order |
| 18:16.41 | [TK]D-Fender | funxion: Maybe you should just cahge your config around and forget about it... |
| 18:16.50 | *** join/#asterisk intralanman (n=lanman@pool-71-253-253-149.nrflva.east.verizon.net) |
| 18:17.20 | kink0 | funxion, I am not sure how responds dual core vs dual xeon cpu's |
| 18:17.37 | funxion | my server is basically a laptop in a rackmount case with pci slots |
| 18:17.51 | funxion | its not taht powerful compared to what you have |
| 18:17.55 | funxion | and it werx fine |
| 18:17.56 | kink0 | even, I am not sure ( I saw not noticiable results ) to use 400Mz,533,800 or even 1066 bus |
| 18:18.45 | kink0 | I have also try CPU's with 512L3, and with 2M L3, also not noticiable difference. |
| 18:19.23 | kink0 | funxion, what is your zttest sumary output ? |
| 18:20.25 | funxion | for what Im working on now or the 4 port box? |
| 18:20.47 | kink0 | for your dual core with 4 spans |
| 18:21.01 | funxion | havent run zttest |
| 18:21.10 | *** part/#asterisk Preytell (n=Preytell@seraph.contegix.com) |
| 18:22.41 | *** join/#asterisk aydiosmio (i=aydiosmi@judecca.aculei.net) |
| 18:22.53 | aydiosmio | anyone familiar with a "PK" authenticaiton mechanism in SIP? |
| 18:23.36 | aydiosmio | SIP Asserted Identity |
| 18:23.37 | aydiosmio | P-Asserted-Identity |
| 18:24.52 | aydiosmio | http://www.ietf.org/rfc/rfc3325.txt |
| 18:24.53 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 18:28.03 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 18:28.07 | flujan | hi guys! |
| 18:28.21 | b11d | HI!!! |
| 18:28.35 | flujan | does mixmonitor works with wav format? |
| 18:28.46 | b11d | woah you got all serious.. i thought we'd become friends first. |
| 18:28.48 | flujan | It is just recording .raw files for me. :( |
| 18:29.02 | b11d | actually I dont know man |
| 18:29.38 | mercestes | flujan: Yes. Just use |wav |
| 18:29.50 | kink0 | anyone has placed two (or more) TE cards in one box ( and better if even used transcoding !! ) ? |
| 18:29.53 | mercestes | flujan: Search for 2wav2mp3 on voipinfo.org and you can even convert them into mp3 using that script. |
| 18:30.05 | mercestes | flujan: don't tell coppice you went there tho or he'll never help you again. |
| 18:30.31 | Corydon-w | kink0: you can, as long as you aren't using G.729 or iLBC on all those channels at once |
| 18:31.12 | kink0 | Corydon-w, I am evaluating how much hardware power would I require if I planned to plug 240 channels, DSP ec, g729 |
| 18:32.19 | Corydon-w | Well, the TC card is close, I hear |
| 18:32.24 | flujan | mercestes, why coppice will not help me anymore? ehehhe |
| 18:32.36 | kink0 | is for large system, I evaluate stability and costs, ussing last generation CPU servers ( even Quad Xeon ) or ussing the double number of boxes, every one with one TE |
| 18:32.37 | aydiosmio | flujan: yes it does, just specify soundfile.wav as the filename |
| 18:34.41 | mercestes | flujan: Voipinfo is beneath him. |
| 18:34.48 | aydiosmio | mercestes: mixmonitor doesn't use the extension option, it just uses the file extension format |
| 18:35.07 | aydiosmio | only monitor uses |mav |
| 18:35.11 | aydiosmio | only monitor uses |wav |
| 18:35.24 | *** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse) |
| 18:35.33 | mercestes | aydiosmio: Yea. It's hard for me to keep all tha tstraight in my head with no references. |
| 18:35.35 | *** join/#asterisk synthetiq (n=syntheti@vonmail.vonworldwide.com) |
| 18:37.07 | synthetiq | anyone here made http post scripts for polycoms in perl |
| 18:37.09 | synthetiq | ? |
| 18:38.03 | *** join/#asterisk kore (i=kore@mindwipe.org) |
| 18:38.32 | flujan | mercestes... lol... why? Is it serious? |
| 18:38.35 | flujan | eheheh |
| 18:39.15 | mercestes | flujan: ?? Coppice? oh, I never know with him. |
| 18:39.16 | [TK]D-Fender | Corydon-w: Deos the TC400 only transcode direct to a zaptel card, or can it be used for pure-voip transformation? (IE SIP-SIP calls, not just SIP-Zaptelcard) |
| 18:39.39 | Corydon-w | [TK]D-Fender: pure voip, I think. |
| 18:39.43 | [TK]D-Fender | synthetiq: I've heard of people who have, but not sure if they are lurking... why? |
| 18:39.54 | Corydon-w | [TK]D-Fender: it's kind of hard to tell since I don't actually have a card to test |
| 18:39.55 | synthetiq | i cant get past the authentication |
| 18:40.29 | [TK]D-Fender | Corydon-w: Just figuring you'd be more "in the know". Who amosgst staff that pops in here would know best? |
| 18:40.49 | Corydon-w | [TK]D-Fender: but given that that's what codec_zap is for, and given that I know the Asterisk architecture, I'd hazard a reasonable guess that it'll work with pure VOIP |
| 18:40.56 | mvanbaak | heya all |
| 18:41.00 | b11d | hey |
| 18:41.03 | Corydon-w | [TK]D-Fender: I am not employed by Digium |
| 18:41.13 | mvanbaak | me neither |
| 18:41.25 | [TK]D-Fender | Corydon-w: You know... I always just kinda assumed that you were :) |
| 18:41.26 | aydiosmio | I talked to a digium tech on the phone about the TC400 |
| 18:41.33 | aydiosmio | it does SIP-to-SIP |
| 18:41.47 | Corydon-w | [TK]D-Fender: I'm a core developer, yes |
| 18:42.04 | mvanbaak | and very active on the bugtracker :) |
| 18:42.16 | [TK]D-Fender | Corydon-w: I'm presuming you were then just a really active contributer at least... Do you work at a place that I might know of (in the telecom field)? |
| 18:42.19 | Corydon-w | Not in the past couple weeks, I haven't been |
| 18:42.37 | Corydon-w | [TK]D-Fender: not unless you know of VCCH |
| 18:42.41 | mvanbaak | Corydon-w: I know. I was talking overall |
| 18:42.48 | *** join/#asterisk queuetue (n=scott@70.54.254.134) |
| 18:42.55 | [TK]D-Fender | aydiosmio: That in mind I presume it should ahve little problems on inter-op with Sangoma cards? |
| 18:43.11 | queuetue | Is there any way to make voicemails mailed to users louder? |
| 18:43.13 | [TK]D-Fender | Corydon-w: Doesn't ring a bell... |
| 18:43.25 | mercestes | queuetue: sox. |
| 18:43.32 | Corydon-w | [TK]D-Fender: 888-VCCH-USA |
| 18:43.34 | kink0 | any way to group channels from differents boxes like if all TE cards would be in one ? ( of course TDMoE is not for this ) |
| 18:43.41 | aydiosmio | [TK]D-Fender: I can't confirm that |
| 18:43.59 | queuetue | mercestes: Is there some way to process them with sox before they get mailed? |
| 18:44.18 | aydiosmio | I only asked about SIP-SIP transcoding and double transcoding from g723 to g729 |
| 18:44.28 | mercestes | queuetue: rewrite/recompile voicemail.c I guess. Or do a dialplan voicemail system. |
| 18:44.28 | Corydon-w | kink0: you could use IAX2 trunking to bring in all trunks into the same context on one particular box |
| 18:44.32 | aydiosmio | which you can do by the way, which is great |
| 18:44.40 | Dr-Linux | hhm.. |
| 18:44.41 | *** join/#asterisk Zaw (i=zaw@unaffiliated/zaw) |
| 18:45.06 | queuetue | Why are they such low volume, anyway? |
| 18:45.16 | [TK]D-Fender | aydiosmio: And which as I've heard will chew up a transcoder from each set from the DSP. |
| 18:45.24 | Dr-Linux | i commented out the [default] moh and using [native] class, but it's still showing default class at CLI ? what could be the issue? :S |
| 18:45.29 | aydiosmio | yeah, will half the available channels |
| 18:45.41 | aydiosmio | but try findaing another card that will do that available now |
| 18:45.45 | Dr-Linux | -- Started music on hold, class 'default', on channel |
| 18:45.56 | [TK]D-Fender | Dr-Linux: And what do your DEVICES use as their MOH class? |
| 18:46.04 | Zaw | i have things working fine with asterisk *except* for sip-sip calls between office locations. i have sip-phone1 -- nat -- asterisk -- nat -- sip-phone2. when either dials the other via extension, it rings but when you answer neither party can hear the other... any ideas where i should start? |
| 18:46.09 | kink0 | Corydon... hmmm you means placing a head server (SIP) then pass IAX2 to the trunk, where every one span is located in differents boxes ? |
| 18:46.10 | [TK]D-Fender | aydiosmio: All none of them? ;) |
| 18:46.46 | aydiosmio | [TK]D-Fender: there's actually a really nice PCI device in the pipe |
| 18:46.49 | aydiosmio | lemme grab the link |
| 18:46.50 | Dr-Linux | [TK]D-Fender: hhm.. i got the issue, thanks |
| 18:46.52 | [TK]D-Fender | Zaw: Each phone's entry should have "nat=yes", "canreinvite=no", "qualify=yes" |
| 18:46.56 | [TK]D-Fender | Zaw: That should do it. |
| 18:46.58 | Dr-Linux | i mean i understand |
| 18:47.04 | Zaw | [TK]D-Fender: i'll check those, thank you |
| 18:47.22 | kink0 | is just to avoid If(congestion) then Dial(..../second_server) and so, a cascadable extension plan |
| 18:47.34 | aydiosmio | [TK]D-Fender: http://www.signalogic.com/index.pl?page=asterisk_ip_pbx |
| 18:48.35 | [TK]D-Fender | aydiosmio: Oh yeah.. I did see that one before, but I had trouble understanding it :) |
| 18:48.55 | aydiosmio | So no one has need to sipaddheader("P-Asserted-Identity") ? |
| 18:49.01 | [TK]D-Fender | aydiosmio: Thought he channel count does look SICK |
| 18:49.08 | aydiosmio | mhm |
| 18:49.10 | aydiosmio | $$$$ |
| 18:49.19 | aydiosmio | I called and got a quote |
| 18:49.33 | [TK]D-Fender | aydiosmio: Lack of a price leads to FEAR of the price ;) |
| 18:49.36 | aydiosmio | the base unit was about $200 more than the TC400 |
| 18:49.43 | aydiosmio | and has fewer channels |
| 18:50.13 | aydiosmio | the people there didn't speak english natively though |
| 18:50.17 | aydiosmio | hopefully I got the right price |
| 18:50.53 | Zaw | [TK]D-Fender: worked like a charm, thanks again |
| 18:51.12 | [TK]D-Fender | Zaw: np |
| 18:51.15 | *** part/#asterisk Zaw (i=zaw@unaffiliated/zaw) |
| 18:51.45 | [TK]D-Fender | aydiosmio: up to 2304 G.711 channels, 1152 G.729AB or iLBC channels, and/or 864 G.723 or GSM-AMR channels |
| 18:52.01 | [TK]D-Fender | aydiosmio: Looks like more channels than the TC400 to me... |
| 18:52.12 | *** join/#asterisk progcaribu (n=arturo@izones70.izones.net) |
| 18:53.23 | [TK]D-Fender | aydiosmio: Their stuff looks kinda bad-ass if you ask me.. the kind the big-boys use and cost accordingly... |
| 18:54.35 | *** join/#asterisk Omer^ (n=Omer@203.81.233.47) |
| 18:54.38 | kink0 | Corydon-w what I pretend is trunking serveral span that are not all in the same box |
| 18:54.57 | [TK]D-Fender | aydiosmio: Cool thing though is they seem to do EC in-line which means you could couple it with a lower model multi-port card to save a fair bit and help make up the cost. |
| 18:56.22 | b11d | i'm down with the polycom.. |
| 18:56.31 | b11d | but i've got enough to last me for a few more weeks |
| 18:56.32 | b11d | :) |
| 18:56.41 | synthetiq | you ever do any perl http post scripts? |
| 18:56.46 | mercestes | queuetue: most ppl don't complain |
| 18:56.54 | b11d | not specifically for polycoms no.. |
| 18:57.02 | synthetiq | you fail |
| 18:57.18 | b11d | hahaha |
| 18:57.27 | synthetiq | i cant get past the stupi authetnication |
| 18:57.31 | synthetiq | alwyas get a 501 error |
| 18:57.48 | b11d | sounds to me like YOU fail. |
| 18:57.49 | b11d | :) |
| 18:58.01 | synthetiq | you fail as polycom junkie i meant |
| 18:58.03 | b11d | :( |
| 18:58.23 | [TK]D-Fender | synthetiq: You trying to HTTP provision them dynamically? |
| 18:58.43 | synthetiq | definte dynamically |
| 18:58.49 | [TK]D-Fender | synthetiq: Make sure your header is right.... |
| 18:59.14 | [TK]D-Fender | synthetiq: Use Python with your web-server to create the config "on-the-fly"? |
| 18:59.21 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 18:59.21 | *** mode/#asterisk [+o denon] by ChanServ |
| 18:59.46 | synthetiq | <PROTECTED> |
| 19:01.14 | [TK]D-Fender | synthetiq: I'd suggest sniffing 2 requests, one yours, the other a traditional flat-file that works. |
| 19:01.41 | *** join/#asterisk ManxPower (n=manxpowe@stirprop-s4-0-0-21.ndcr2.datasync.net) |
| 19:02.17 | mcab | synthetiq: what exactly are you trying to do? |
| 19:02.30 | synthetiq | update dst crap |
| 19:02.54 | ManxPower | Any Nortel Modular ICS gurus around? If so, can you /msg me? I have a question about ANI and DID on E&M/Wink Tie lines |
| 19:03.12 | ManxPower | <-- trying to use up some of the good Asterisk karma he hs |
| 19:03.13 | mcab | synthetiq: ok, how is this involving HTTP post? :-) |
| 19:04.01 | synthetiq | as ooposed to manually going in and changing 600 phones.. |
| 19:05.24 | mcab | synthetiq: change the config on your provisioning server, reboot phones, voila... I'm still trying to see what you're doing with an HTTP post |
| 19:05.52 | synthetiq | violla its an old system not using tftp sever |
| 19:06.08 | mcab | ... |
| 19:06.09 | aydiosmio | [TK]D-Fender: I mean tthier base card has fewer channels |
| 19:06.15 | mcab | synthetiq: oh my |
| 19:06.16 | aydiosmio | 1-DSP option |
| 19:06.19 | synthetiq | yes |
| 19:06.26 | aydiosmio | as I recall |
| 19:06.29 | synthetiq | looks like ill have to manually do this |
| 19:06.33 | mcab | synthetiq: so, you're trying to POST *to* the phone's config webserver? |
| 19:06.37 | synthetiq | yes |
| 19:06.59 | synthetiq | just the variables for that coreConf.htm page |
| 19:08.13 | mcab | synthetiq: never tried that. I don't touch the web config stuff with a 10' pole. TBH, it might be easier installing a proper provisioning server :-7 |
| 19:13.42 | [TK]D-Fender | aydiosmio: How were they set up previously? |
| 19:14.56 | *** join/#asterisk Mercestes (n=Merceste@cpe-24-175-82-3.houston.res.rr.com) |
| 19:15.12 | *** join/#asterisk `p4r14h`work (n=josh@24-119-48-78.cpe.cableone.net) |
| 19:15.41 | aydiosmio | [TK]D-Fender: these cards cna have up to n DSP modules |
| 19:15.48 | aydiosmio | I was quoted for the 1 module option |
| 19:16.00 | `p4r14h`work | any way to set up SMTP authentication for send voicemail in email messages? |
| 19:16.50 | [TK]D-Fender | synthetiq: How were they set up previously? |
| 19:16.58 | [TK]D-Fender | aydiosmio: Sorry, poorly directed question :) |
| 19:17.20 | *** join/#asterisk topping (n=topping@204.152.96.238) |
| 19:17.25 | [TK]D-Fender | aydiosmio: Yeah I saw the DSP table, even the low end was basically about a tie for the TC400 |
| 19:17.37 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:18.05 | aydiosmio | `p4r14h`work: http://72.14.209.104/search?q=cache:hwph7rKrG0sJ:www.trixbox.org/modules/newbb/viewtopic.php%3Ftopic_id%3D884%26forum%3D2+asterisk+smtp+auth&hl=en&ct=clnk&cd=1&gl=us |
| 19:18.11 | aydiosmio | this might help you |
| 19:18.45 | aydiosmio | I don't know how * sends email, but it sounds like a local sendmail relay would work |
| 19:18.52 | aydiosmio | configured for auth |
| 19:21.54 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 19:22.57 | `p4r14h`work | aydiosmio, thanks for the great info :D |
| 19:22.58 | nays85 | anyone here have qwest ip termination? |
| 19:24.05 | b11d | i dont, but know some people that do. |
| 19:26.51 | *** join/#asterisk dasenjo (n=dasenjo@190.24.178.96) |
| 19:27.04 | nays85 | do you know pricing details? |
| 19:27.59 | nays85 | the quotes i'm getting (without negotiating) for tiered rates are ridiculously high |
| 19:28.03 | *** join/#asterisk ezway` (n=ez@c66.110.149-45.clta.globetrotter.net) |
| 19:29.12 | [TK]D-Fender | nays85: Then start shopping elsewhere now... |
| 19:32.19 | errr | Im trying to find some docs on making multiple asterisk servers able to communicate with each other, can anyone point me to some relevant documentation? |
| 19:32.30 | b11d | look into IAX |
| 19:32.49 | errr | b11d: me? |
| 19:33.02 | b11d | yes |
| 19:33.18 | errr | ok, do you know if I can do toll by pass with that? |
| 19:33.26 | b11d | nope, not sure. |
| 19:33.34 | b11d | all I know is you use IAX for linking multiple asterisk boxen.. |
| 19:33.38 | b11d | but I dont use it myself.. |
| 19:33.46 | errr | ah ok then, thanks |
| 19:33.51 | b11d | np.. |
| 19:34.54 | [TK]D-Fender | errr: lookup "asterisk dual servers" on the WIKI |
| 19:35.21 | [TK]D-Fender | errr: And forget "toll-bypass" as a tem. You're going direct server to server doing whatever you want in this case |
| 19:35.26 | [TK]D-Fender | term* |
| 19:35.52 | b11d | ahh.. b11d.. a constant source of misinformation. |
| 19:38.21 | *** join/#asterisk ToyMan (n=Stuart@user-0cevdmv.cable.mindspring.com) |
| 19:38.36 | errr | [TK]D-Fender: ah cool, so Ill be able to place a call from Server1 and use Server2's outbound truck then to avoid long distance charges? |
| 19:38.46 | errr | trunk* |
| 19:39.22 | [TK]D-Fender | ~b11d |
| 19:39.23 | jbot | b11d is a constant source of misinformation... |
| 19:39.32 | [TK]D-Fender | :O |
| 19:39.33 | *** join/#asterisk dahunter3 (n=dahunter@69-178-131-100.static-ip.telepacific.net) |
| 19:39.34 | errr | heh |
| 19:39.49 | [TK]D-Fender | errr: If you setup Server 2 to let them out, sure |
| 19:40.00 | errr | awesome |
| 19:40.57 | errr | its so cool what asterisk can do. intertel would like to charge you 100's of thousands of dollars to do this stuff |
| 19:41.35 | [TK]D-Fender | Aastra's 36-key attendant console & new 5i series phones look damn hot.... |
| 19:41.46 | [TK]D-Fender | errr: Yup.... |
| 19:42.01 | b11d | yeah the big telco's phear asterisk |
| 19:42.36 | errr | I went to an intertel demo a couple months ago and they were talking abou their new SIP system that me and my boss were betting big dollars on is based on asterisk |
| 19:43.10 | [TK]D-Fender | errr: My head office almost went with Intertel..... instead they are installing their Avaya IP Office TONIGHT.... |
| 19:43.22 | errr | awesmoe |
| 19:43.30 | errr | the intertel ticket system sucks ass |
| 19:43.33 | *** part/#asterisk mkrufky (n=mk@unaffiliated/mkrufky) |
| 19:43.38 | b11d | the State of Minnesota's Office of Enterprise Tech is rebranding and reselling Asterisk as some PoS called "VoiceRD" |
| 19:43.43 | gambolputty | I have heard of the Intertel SIP system, in limited use or something like that |
| 19:43.50 | b11d | throw some quasi-private company who partnered with them |
| 19:44.00 | errr | we are under contract for a few more months, I should have asterisk done by then to replace that crappy service they provide |
| 19:44.57 | [TK]D-Fender | OMG... 60 indications on 3 pages via LCD.. http://www.aastra.com/cps/rde/xchg/SID-3D8CCB73-17761597/04/hs.xsl/20684.htm |
| 19:45.39 | [TK]D-Fender | COMPLETELY sick... |
| 19:45.44 | kink0 | see u later guys |
| 19:48.07 | b11d | wow.. |
| 19:48.08 | *** join/#asterisk Waverly360 (n=irc@209.12.249.243) |
| 19:49.04 | *** join/#asterisk Zefk (n=Zefk@81.181.249.106) |
| 19:49.41 | [TK]D-Fender | b11d: Check the PDF at the bottom... this might displace the IP 601 for me... |
| 19:49.52 | [TK]D-Fender | b11d: As much as I love Polycom.... |
| 19:50.23 | b11d | know anyone who has actually used it yet? |
| 19:50.46 | Waverly360 | [TK]D-Fender: I have some questions for you :) |
| 19:51.18 | *** join/#asterisk MrChicken (n=MrChicke@200.71.58.39) |
| 19:51.20 | MrChicken | Hello |
| 19:51.29 | b11d | hi |
| 19:51.29 | *** join/#asterisk bsd_tech (n=bsdtech@ppp-69-238-48-150.dsl.irvnca.pacbell.net) |
| 19:52.02 | Waverly360 | [TK]D-Fender: Do you know what the maximum number of phones one instance of asterisk can support? |
| 19:52.10 | MrChicken | I wanted to ask you guys... can I connect a bluetooth headset for a cellphone to a computer running x-lite via bluetooth? |
| 19:52.37 | [TK]D-Fender | Waverly360: No idea on a limit.... |
| 19:52.52 | [TK]D-Fender | MrChicken: Yes |
| 19:53.13 | Waverly360 | [TK]D-Fender: In the past, both from a former co-worker of mine, and a various forums I've read, some feel that asterisk can't handle more than 100 phones reliably. |
| 19:53.42 | MrChicken | [TK]D-Fender ... is there any restriction on brands or anything? an |
| 19:53.49 | [TK]D-Fender | Waverly360: I've hear of installs around the 200 mark in here before... |
| 19:53.49 | mercestes | Waverly360: That's wrong. |
| 19:53.51 | Waverly360 | I need to know if anyone here is using asterisk with more phones than that, without any problem. |
| 19:53.58 | mercestes | I've had around 500. |
| 19:54.10 | [TK]D-Fender | MrChicken: Its just a mic & speaker to the computer, so I presume jsut about anything will do. |
| 19:54.41 | MrChicken | [TK]D-Fender ... and If I have like 10 of these devices, will they interfere with each other? |
| 19:54.59 | mercestes | Waverly360: And about 200 in a crisis call center. Yes, I do consulting work. :) |
| 19:55.05 | Waverly360 | MrChicken: You'll need some sort of driver on the PC I believe to support that device. |
| 19:55.11 | [TK]D-Fender | MrChicken: If you set them up right I suppose they should accomodate each other |
| 19:55.45 | *** join/#asterisk riddlebox (n=riddlebo@24-207-167-95.dhcp.stls.mo.charter.com) |
| 19:55.49 | Waverly360 | mercestes: Hmm. well, that eases my mind a bit. What about maximum amount of concurrent calls? |
| 19:56.19 | Waverly360 | mercestes: If I have a dual port PRI, that would give me access to about 46 lines at a time correct? |
| 19:56.30 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 19:56.47 | syzygyBSD | how stable is 1.4? |
| 19:57.06 | b11d | depends on what you're doing with Asterisk i guess |
| 19:57.13 | syzygyBSD | I am upgrading an old version and don't know which to choose |
| 19:57.18 | MrChicken | [TK]D-Fender ... I'm trying to get a cordless callcenter running... which would in your opinion have the least problems? Wireless analog phones connected to * via ATAs or Bluetooth headsets connected to X-lite? |
| 19:57.20 | mercestes | Waverly360: Not a problem. |
| 19:57.20 | b11d | what version are you upgrading from> |
| 19:57.21 | b11d | ? |
| 19:57.22 | syzygyBSD | voicemail... |
| 19:57.25 | syzygyBSD | 1.0.9 |
| 19:57.29 | b11d | wow.. |
| 19:57.37 | b11d | im not experienced enough to touch that one :) |
| 19:57.41 | mercestes | 1.0.9 is a *great* release. |
| 19:57.43 | syzygyBSD | if it ain't broke don't fix it... |
| 19:57.46 | syzygyBSD | its broke now... |
| 19:57.58 | [TK]D-Fender | MrChicken: Whats the point of it being "wireless?" |
| 19:58.03 | mercestes | syzygyBSD: Give 1.2.14 a try. I use 1.2.13 but I have an annoying voicemail forwarding bug. |
| 19:58.18 | b11d | 1.2.14 rocks.. 1.2.16 is the latest of the 1.2 I thought |
| 19:58.22 | b11d | but it doenst compile for me ;) |
| 19:58.24 | mercestes | I seem to recall having an issue with .14 that caused me to roll back. |
| 19:58.32 | [TK]D-Fender | syzygyBSD: If you must change I'd say the latest 1.2 series for now |
| 19:58.33 | syzygyBSD | it is, but I never trust the newest releases |
| 19:58.34 | mercestes | maybe i twas .16 I rolled back from. |
| 19:58.50 | syzygyBSD | ever since 1.2.9-1.2.12 broke agents |
| 19:59.13 | MrChicken | [TK]D-Fender ... my agents have to move around the building a lot ... |
| 19:59.13 | [TK]D-Fender | syzygyBSD: Keep in mind that chan_agent is like.. you know... toast |
| 19:59.14 | mercestes | syzygyBSD: .13 or .14 then. :) .15-.16 if your a little brave. |
| 19:59.20 | syzygyBSD | thanks for the insight |
| 19:59.33 | [TK]D-Fender | MrChicken: BEFORE getting the call, or AFTER? |
| 19:59.44 | mercestes | I miss 1.0.9 though |
| 19:59.47 | syzygyBSD | oh.. .I don't use agents on this server, but I would hate to find out the version I upgraded to didn't support voicemail |
| 20:00.21 | MrChicken | [TK]D-Fender ... once they recieve the call |
| 20:00.27 | syzygyBSD | lol, course they wouldn't, probably the most tested feature |
| 20:00.35 | *** join/#asterisk angryuser (n=Miranda@df01t2-212-194-222-248.d4.club-internet.fr) |
| 20:00.38 | Waverly360 | Has anyone had more than 500 phones connected to an asterisk box? |
| 20:00.56 | syzygyBSD | wait.. chan_agent is toast?... what replaced it? |
| 20:01.08 | mercestes | Waverly360: Why don't we make this EASY? HOw many phones are you planning on hooking up to one box??? |
| 20:01.16 | syzygyBSD | in case I need to upgrade another server... |
| 20:01.24 | angryuser | good evening, when i compiled asterisk i saw a module embedding in 'asterisk menuselect' what is it for? |
| 20:01.24 | b11d | and is thst 500 "registered" or 500 actually engaged in communications ? |
| 20:01.51 | Waverly360 | mercestes: It's not that easy. It's not just for us..it's for future customers. |
| 20:01.55 | [TK]D-Fender | MrChicken: Well I can't vouch for BT's range.. I think that might kill you |
| 20:02.11 | mercestes | Waverly360: Are you planning on using "Asterisk: The Open Source PBX" as a commercial soft switch? |
| 20:02.12 | Waverly360 | mercestes: I need to know what the upper limits are, so when installing this, we can know when to throw another server in the mix to balance the load |
| 20:03.47 | Waverly360 | mercestes: I'm not selling Asterisk, if that's what you're asking. I'm installing it as a service. |
| 20:03.52 | MrChicken | [TK]D-Fender ... so you'd suggest cordless phones? |
| 20:04.02 | MrChicken | (analog) |
| 20:04.05 | mercestes | Waverly360: That's not what I asked.. |
| 20:04.14 | *** join/#asterisk jaxxan (n=jaxxan@202.70.125.125) |
| 20:04.18 | jaxxan | hey guys |
| 20:04.35 | mercestes | Waverly360: Are you planning on hooking up *OFFICES* with a PBX...or are you planning on setting up a monster server with phones all over the planet connecting to your server from userland? |
| 20:05.02 | Waverly360 | mercestes: I'm planning on hooking up offices with a PBX. Sorry, I misunderstood the question. |
| 20:05.24 | mercestes | Waverly360: And how many offices do you expect to hookup that are +500 users? |
| 20:05.27 | [TK]D-Fender | MrChicken: I think THIS might be jsut up your alley : http://www.uniden.com/products/productdetail.cfm?product=ELX500&page=4 |
| 20:05.56 | Waverly360 | mercestes: Well, prior to this week, none. However, we've found a couple that were interested, but they have a lot of users. |
| 20:06.09 | Waverly360 | mercestes: I'm unsure of just how stable asterisk is as the user load increases. |
| 20:06.30 | *** join/#asterisk steffo (n=steffo@ip56505d7f.direct-adsl.nl) |
| 20:06.30 | [TK]D-Fender | Waverly360: Is chan_jello.so any indication? |
| 20:06.49 | mercestes | Don't listen to Fender. He uses OpenPBX. |
| 20:06.50 | Waverly360 | mercestes: I'm trying to get a feel for it from you guys, since I figured at least a few of you would have experience with that many phones. |
| 20:07.08 | Waverly360 | [TK]D-Fender: I don't know what chan_jello.so is. |
| 20:07.19 | techie | ha. |
| 20:07.36 | mercestes | Waverly360: Around 400-500 you start running into issues if you start doing any level of transcoding. My only experience with 500+ users involved the most retarded setup ever....but I managed to keep it together with minimal failures. |
| 20:07.58 | Waverly360 | mercestes: transcoding? What were you doing? |
| 20:07.59 | mercestes | Waverly360: true, I was more like "engineer scotty" than anything, "I'm giving it all shes got, captain!!!" But it held....barely. |
| 20:08.09 | [TK]D-Fender | Waverly360: Here's a primer on it : http://dictionary.reference.com/browse/sarcasm |
| 20:08.15 | mercestes | Waverly360: Any *sane* intelligent setup would be rock solid at tha tpoint..... |
| 20:08.21 | Waverly360 | [TK]D-Fender:...you're a punk :) |
| 20:08.28 | b11d | TK, you rock man.. |
| 20:08.37 | mercestes | Waverly360: *HOWEVER*.....there is a social context here too. Just how angry would 500 users be if their phones stopped working? |
| 20:08.58 | Waverly360 | mercestes: Probably pretty pissed. |
| 20:09.11 | mercestes | Waverly360: it would behoove you greatly to have a secondary (and maybe even a tertiary) asterisk PBX failover system available to any office over say..100 or so, just to cull the herds, wouldn't you say??? |
| 20:09.23 | b11d | mercestes, using the Scotty line, you need to remember that he never told Kirk exactly what he had up his sleeve, so "giving her everythign I've got" isnt really.. because he always kept that last card hidden to look like a miracle worker in the last minute. |
| 20:09.38 | b11d | thats Engineering basics :) |
| 20:09.39 | jaxxan | Is this the correct way to implement NFAS using Spans 2&3 on a TE410P with PRI's to a DMS100 ? |
| 20:09.45 | jaxxan | http://www.pastebin.ca/397811 |
| 20:09.46 | mercestes | Waverly360: So the question of "how many simultanious phones can * handle" is best for benchmarkin gand not for a business model. |
| 20:10.21 | mercestes | Waverly360: The answer is "it depends on your hardware" But realistically given a modest PC and a shitload of switches, probably around 2000 phones not doing anything, with around oh, 50 concurrent calls with those 2000 phones. |
| 20:10.33 | *** join/#asterisk ZefK (n=Zefk@wsc-fo.b.astral.ro) |
| 20:10.40 | riddlebox | you can create a sip extension on one asterisk server, then tell another asterisk server to use that as way to call out right? |
| 20:10.43 | mercestes | Waverly360: Nevermind the smoke and blue-green flames, they're normal. |
| 20:10.55 | jaxxan | i've just not ever used the trunkgroups before in zapata.conf so it's pretty new to me |
| 20:10.56 | b11d | riddlebox.. aye |
| 20:11.14 | jaxxan | the logical spans are still foreign to me, i'm not sure if i have them correctly |
| 20:11.25 | *** part/#asterisk Morph (i=gareth@mulder.wiked.org) |
| 20:12.12 | b11d | jaxxan.. whats up with those ,yellow in your span defs? |
| 20:12.20 | b11d | never seen that before |
| 20:12.29 | Waverly360 | mercestes: Oh definitely. For that scenario, I would have redundant servers standing by. Until recently, there was someone else here who made these decisions. He's gone now, and it's my responsibility now. |
| 20:12.40 | [TK]D-Fender | jaxxan: Clearly not NFAS given you specified 4 DCHAN 's... |
| 20:12.57 | [TK]D-Fender | jaxxan: And "no comment"on your lack of primary timing source... |
| 20:12.59 | Waverly360 | mercestes: I'm pretty familiar with asterisk, and it's workings, but he always led me to believe that asterisk couldn't support that many phones. I'm attempting to do my own research, as it's something we really need to know. |
| 20:13.17 | mercestes | he's wrong. |
| 20:13.18 | jaxxan | i get timing from the DMS100 |
| 20:13.23 | mercestes | or practical. |
| 20:13.27 | mercestes | one or the other |
| 20:14.05 | Waverly360 | mercestes: heh. It's hard to know what to trust. I mean, we don't really have the means to load test something that big...and I'd hate to sell it to a customer, and find out it doesn't work as expected. |
| 20:14.16 | b11d | Waverly360.. i hear that man. |
| 20:14.30 | *** join/#asterisk raidenz (i=raiden@205-200-66-136.static.mts.net) |
| 20:14.33 | jaxxan | [TK]D-Fender: yeah i need to remove that ... sec let me correct |
| 20:14.40 | raidenz | Hi guys |
| 20:14.49 | b11d | hi |
| 20:14.58 | Waverly360 | mercestes: There's always the possibility of splitting the load across multiple servers, but then I'd have to figure out how to make multiple asterisk servers talk to each other and route calls properly. |
| 20:15.58 | mercestes | Waverly360: Dialplan voodoo and IAX2 |
| 20:16.09 | raidenz | Has anyone successfully used ezstream or ices0 to stream MP3 data to an icecast server using app_ices? I have it setup for ogg but I want to stream MP3 data to it. |
| 20:16.39 | *** part/#asterisk MarkWD (n=MarkWD@rrcs-67-78-88-186.sw.biz.rr.com) |
| 20:17.07 | Waverly360 | mercestes: Yeah, but that's a LOT of dialplan voodoo |
| 20:17.35 | jaxxan | ok i've made the corrections: http://www.pastebin.ca/397820 |
| 20:17.49 | Waverly360 | mercestes: I mean, to be able to have queues ring people on other boxes, as well as having MeetMe and IVR's route properly.. |
| 20:18.03 | *** join/#asterisk bmg505 (n=leon@196.209.250.40) |
| 20:19.05 | Waverly360 | mercestes: That gets pretty heavy. I currently have it setup to where I can take multiple pbxes, set them up as a master and slaves through a web interface, and each of those pbxes can dial extensions on other boxes transparently. |
| 20:19.28 | jaxxan | [TK]D-Fender: i put the ,yellow in zaptel.conf a long time ago seems to work fine. it's just an added indicator yeah ? |
| 20:19.52 | [TK]D-Fender | jaxxan: I wasn't the one asing about that, though I don't think you need/want that really.. |
| 20:20.00 | b11d | yeah, that was me |
| 20:20.01 | b11d | :) |
| 20:20.14 | jaxxan | oh (= |
| 20:20.14 | Waverly360 | mercestes: Adding in all of the magic to build the config files properly through the web interface would be a nightmare at this point...but if asterisk can support that many phones, I can definitely put this off until I have more time to deal with it. |
| 20:20.15 | *** join/#asterisk qdk (n=qdk@80.243.125.204) |
| 20:20.24 | jaxxan | i can remove it. dont think it's a big deal though. |
| 20:20.28 | b11d | and I was just curious about it is all.. |
| 20:20.34 | b11d | didnt really have any point :) |
| 20:21.21 | mercestes | What? |
| 20:21.22 | jaxxan | damn, i messed up my dchans in that trunkgroup bah |
| 20:21.27 | mercestes | web interface? |
| 20:21.37 | mercestes | LOL...you are DOOOOOMED!!!!! |
| 20:21.45 | mercestes | you are not even worthy of this CHANNEL! |
| 20:21.51 | mercestes | ~trixbox |
| 20:21.52 | jbot | trixbox is probably unable to be supported here. Try joining #freepbx and asking there, or use the trixbox forums at http://www.trixbox.org/modules/newbb/ |
| 20:21.58 | mercestes | Take your 500 phones THERE baby! |
| 20:22.16 | mercestes | No consulting for you! Purge yourself of the sin of trixbox or begone! |
| 20:22.31 | Waverly360 | mercestes: I wrote my own. |
| 20:22.37 | riddlebox | mercestes, do you feel that way about the digium gui as well? |
| 20:22.42 | mercestes | Waverly360: even worse. |
| 20:22.45 | Waverly360 | mercestes: Why? |
| 20:22.46 | mercestes | riddlebox: Yes. |
| 20:22.55 | mercestes | riddlebox Guis are for girls. real men use flat files |
| 20:22.58 | riddlebox | lol |
| 20:23.10 | [TK]D-Fender | ~mercestes |
| 20:23.11 | jbot | hmm... mercestes is almost a total nub |
| 20:23.11 | mercestes | Next they'll want it to work in WINDOWS |
| 20:23.11 | tzafrir_laptop | the channel is for Asterisk questions. Even from Trixbox people (when they are phrased as Asterisk questions) |
| 20:23.18 | jaxxan | http://www.pastebin.ca/397854 has my correct dchans for the NFAS |
| 20:23.23 | mercestes | Shutup, Jbot. you know nothing. |
| 20:23.30 | [TK]D-Fender | mercestes: I don't like my girls "flat", ok? ;) |
| 20:23.39 | mercestes | lol @ D-Fender |
| 20:23.42 | Strom_M | ~strom |
| 20:23.43 | jbot | i heard strom is the coolest #asterisk lurker |
| 20:23.43 | mercestes | me neither. |
| 20:23.52 | Strom_M | wtf, i thought i was just some nub |
| 20:23.53 | Strom_M | ~strom_c |
| 20:23.54 | jbot | hmm... strom_c is just some nub |
| 20:23.57 | Strom_M | there we go |
| 20:24.00 | b11d | haha |
| 20:24.01 | mercestes | no, jbot. Mercestes is the dark overlord of #asterisk. Fear him. |
| 20:24.12 | mercestes | ... |
| 20:24.15 | mercestes | I hate you jbot. |
| 20:24.23 | [TK]D-Fender | ~jbot |
| 20:24.24 | jbot | jbot > mercestes |
| 20:24.27 | Waverly360 | Look, I'm not using trixbox. My manager and I wrote this interface from the ground up, to make asterisk easy to manage. |
| 20:24.29 | [TK]D-Fender | :D |
| 20:24.42 | b11d | haha |
| 20:24.54 | jaxxan | [TK]D-Fender: does that look better ? i'm mostly concerned with the [trunkgroups] |
| 20:24.59 | techie | 'easy to manage' |
| 20:25.03 | mercestes | Waverly360: Then join....#Waverly's super sweet web interface" I guess. |
| 20:25.14 | mercestes | If it's so easy why are you in here asking questions? |
| 20:25.29 | Waverly360 | mercestes: Are you really that hardcore against an interface? |
| 20:25.33 | mercestes | nice. |
| 20:25.36 | mercestes | Waverly360: Yes. |
| 20:26.25 | Waverly360 | mercestes: I'm asking questions about asterisk. |
| 20:26.25 | mercestes | Waverly360: My grandma needs an interface. That's why she has a MAC. |
| 20:26.25 | mercestes | graphics are the devil... |
| 20:26.25 | Waverly360 | mercestes: It would be different if I were the one doing these installs |
| 20:26.25 | mercestes | true bliss is pure and CLI. |
| 20:26.32 | Waverly360 | mercestes: the simple truth is, I need people who know nothing about asterisk to be able to install these systems |
| 20:26.35 | [TK]D-Fender | <PROTECTED> |
| 20:26.58 | jaxxan | [TK]D-Fender: 48 is the primary and 72 is the backup |
| 20:27.01 | mercestes | Waverly360: Then your in the right place. lol |
| 20:27.18 | Waverly360 | mercestes: You're telling me that for 50 customers, you'd maintain their dialplan using a text editor, rather than allowing them to do simple things themselves? |
| 20:27.23 | [TK]D-Fender | dchan=24,48,72,96 <- NO |
| 20:27.41 | Strom_M | NO NO NO (tesla girls, tesla girls) |
| 20:27.52 | mercestes | Waverly360: I'm telling you that for customer facing customizations I used a shitload of macros and a PHP page linked to a Mysql database with my Sip.conf and voicemail.conf using Res_mysql.conf. |
| 20:27.55 | jaxxan | [TK]D-Fender: how would i write that in zaptel.conf then? |
| 20:27.57 | *** join/#asterisk boch (n=fran@190.48.216.130) |
| 20:28.01 | mercestes | Waverly360: But I still wrote it by hand. |
| 20:28.03 | jaxxan | just 24, 48, 96 ? |
| 20:28.21 | mercestes | Waverly360: Customers (my grandma) got the GUI, and they basically just changed where their phone rang and for how long. |
| 20:28.24 | jaxxan | i'm only using spans 2 & 3 for the NFAS |
| 20:29.01 | MrChicken | [TK]D-Fender ... in a 20 - 30 sq meter area, which you think would be a better solution .... 10 cordless phones (3 900Mhz, 3 2.4 Ghz and 4 5.8 Ghz) connected to some ATAs, or 10 bluetooth headsets connected to x-lite? |
| 20:29.19 | [TK]D-Fender | jaxxan: Whats the point of NFAS w/ "backup" D-chan? |
| 20:29.35 | jaxxan | so i can have more than 23 channels |
| 20:29.38 | JacksLivr | Strom_M: howd the class go? |
| 20:29.49 | [TK]D-Fender | MrChicken: Well the X-lite solution = MUCH cheaper, but they can't do anything practical without being in front of the PC. |
| 20:29.52 | jaxxan | 46 channels is fine, and that's the way it's setup on the DMS100 |
| 20:30.23 | Strom_M | JacksLivr: i'm still proctoring the exam |
| 20:30.25 | [TK]D-Fender | MrChicken: And 20sq/m +/- 4x5 = who friggen casre about wireless? |
| 20:30.33 | jaxxan | is the [trunkgroup] section fine ? |
| 20:30.38 | [TK]D-Fender | MrChicken: let them RUN for it :) |
| 20:31.00 | mercestes | Waverly360: So the macros and initial setup got written by hand, yea. Pretty magic marcros too. I also had to do calls through php_agi's to read the database and construct dial plans based upon the mysql settings. |
| 20:31.02 | [TK]D-Fender | jaxxan: Not sure I understand the implementation... try it and see.... |
| 20:31.08 | Waverly360 | mercestes: *shrugs* I'm not dogging your way of doing things. To each his/her own. I just feel that what we have is more time and cost effective than doing even the slightest custom config. |
| 20:31.16 | jaxxan | is the spanmap part ok ? |
| 20:31.25 | jaxxan | that's my real question cause it's unfamiliar to me |
| 20:31.36 | mercestes | Waverly360: Ok.... |
| 20:31.48 | [TK]D-Fender | jaxxan: Not sure on the specific syntax... I'd just WIKI that part personally... no direct experience |
| 20:31.57 | jaxxan | kk |
| 20:32.01 | JacksLivr | Strom_M: pay attention, that dude to your left is CHEATING!!! |
| 20:32.10 | Waverly360 | mercestes: So how do your customers add new phones and extensions to the system? |
| 20:32.32 | mercestes | Waverly360: For customer facing interfaces, keep it simple. Dropdown box for their extension..give them three options. Ring their phone for x seconds, then go to A: Voicemail, B: hangup, C: Forward to another phone. Etc. |
| 20:32.37 | Strom_M | OH NOES |
| 20:33.01 | mercestes | Waverly360: They could add lines but not extensions (because they coudln't setup the phone.) Ihad my own scripts to setup new phones. |
| 20:34.12 | Waverly360 | mercestes: How do they add lines? |
| 20:35.14 | JacksLivr | i was just up a digium, picked up my iaxy |
| 20:37.14 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.91) |
| 20:37.43 | jaxxan | yeah didn't like that )= |
| 20:38.03 | Strom_M | JacksLivr: you were here? |
| 20:38.34 | JacksLivr | if here is the atrium building, then yes |
| 20:38.42 | Strom_M | did I meet you then? |
| 20:39.14 | JacksLivr | on sunday, i sat next to you at casablanca. |
| 20:39.34 | mercestes | Waverly360: Lines were a virtual construct because we were an all SIP operation. So we limited their concurrent calls under the guise of "lines" because that's what they were used to hearing. |
| 20:40.00 | Waverly360 | mercestes: How many asterisk customers do you manage? |
| 20:40.06 | *** join/#asterisk gr1ncheux (n=devine@unaffiliated/gr1ncheux) |
| 20:40.09 | Strom_M | ohhhh of course |
| 20:40.22 | Strom_M | sorry...parade of faces; i'm terrible with names :) |
| 20:40.35 | mercestes | Waverly360: At this setup, the 500 I was telling you about on one monster.....well.....on one kitty-dell really. |
| 20:40.38 | JacksLivr | it is too late, i have been shamed |
| 20:40.47 | mercestes | It was like a 3850 or some crap like that with no frills. |
| 20:40.50 | mercestes | I hated my life, btw. |
| 20:41.11 | mercestes | I remember you, John's Liver. |
| 20:41.20 | Waverly360 | mercestes: :) I'll bet. Originally, before I started here, we installed a custom pbx for a customer. |
| 20:41.29 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 20:41.34 | mercestes | Waverly360: Now the 80 I did at my next job was much better. |
| 20:41.37 | Waverly360 | mercestes: It's been a management headache. Anytime they wanted changes, we had to go make them. |
| 20:41.45 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 20:41.45 | *** mode/#asterisk [+o russellb] by ChanServ |
| 20:41.52 | *** join/#asterisk ToyMan (n=Stuart@user-0cevdmv.cable.mindspring.com) |
| 20:41.57 | ezway` | anyone using wakeup call ? |
| 20:42.04 | Waverly360 | mercestes: We've got 20 customers right now...I can't imagine managing all of them using flat text files. |
| 20:43.08 | Waverly360 | mercestes: So my manager and I wrote our own management interface from the ground up...around asterisk of course. You can setup phones, users, and the pstn devices from it. Plus, there's an IVR management system, and a way to upload custom audio files and ringtones. |
| 20:43.12 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 20:43.19 | mercestes | Waverly360: Been there ,don ethat. |
| 20:43.23 | Waverly360 | mercestes: There's no way we could keep up if we weren't doing that...but we wrote our initial configs. |
| 20:43.43 | mercestes | Waverly360: Are you using a database or text file mangling? |
| 20:43.57 | Waverly360 | mercestes: define text file mangling? |
| 20:44.06 | JacksLivr | ezway`: im not, but i just looked it up and will prolly be by tomorrow. |
| 20:44.11 | JacksLivr | looks cool |
| 20:44.33 | *** join/#asterisk Braxus (n=braxus@66.147.214.164) |
| 20:45.01 | JacksLivr | i was wanting to mess around with something like that that would read the weather to me when it woke me up |
| 20:45.05 | Waverly360 | mercestes: the web interface stores and manipulates all of the data within a database. Once all of the changes have been made, scripts are run using the data in the db to create all of the asterisk config files, then asterisk is reloaded or restarted, depending on the config files that were written. |
| 20:46.03 | [TK]D-Fender | ok, heading home, BBIAB |
| 20:46.44 | mercestes | Waverly360: Is you rwebpage editting a database...or a textfile... |
| 20:46.50 | carrar | Any recommendations for softphones for OSX users that do not crash? |
| 20:46.57 | mercestes | Waverly360: ew, nasty. |
| 20:47.14 | Waverly360 | mercestes: Why? |
| 20:49.06 | toombaloomba | carrar ive used X-lite on a PPC just fine |
| 20:49.23 | carrar | yeah xlite & SJ crash on 10.4.9 intel after a call |
| 20:49.40 | toombaloomba | if youre using intel then you can run any windows one you want with that parallels or whatever it is |
| 20:49.53 | carrar | thats just more overhead |
| 20:50.01 | mercestes | Waverly360: programmatic textfile mangling is error prone. |
| 20:50.24 | mercestes | I only programmatically mangle text files when clients don't pay me. |
| 20:50.31 | mercestes | that remind sme. |
| 20:50.55 | *** join/#asterisk CrashHD (n=crashhd@c-67-166-155-233.hsd1.ca.comcast.net) |
| 20:52.38 | Waverly360 | mercestes: error prone how? |
| 20:53.09 | Waverly360 | mercestes: I would say that using scripts to create the config files the same way every single time would be much less error prone than manually editting text files. Humans make mistakes, computers typically don't. |
| 20:57.24 | *** join/#asterisk l2cache (n=ghansen@64.128.254.98) |
| 20:58.01 | l2cache | I need to get mailbox notify working and the voicemail is stored on a different asterisk server than the one that the phones are registered to. Any ideas? |
| 21:01.16 | l2cache | anybody? |
| 21:01.26 | mercestes | Waverly360: I don't like textfile mangling |
| 21:01.39 | b11d | l2cache.. i'd love to hear how you accomplish that, when you do :) |
| 21:01.43 | Waverly360 | mercestes: To each his own :) |
| 21:01.53 | l2cache | lol..thank you |
| 21:01.59 | b11d | :) |
| 21:02.04 | *** join/#asterisk Shaun2222 (n=shaun@ip68-4-212-221.oc.oc.cox.net) |
| 21:02.10 | l2cache | Never should have routed all of the company's voicemail to one central server |
| 21:02.18 | b11d | well it makes some sense :) |
| 21:02.25 | l2cache | now i have no way to get the notify working since the voicemail server has no phones registered to it |
| 21:02.29 | b11d | you could probably write a script to do it, but the detaisl I dont know. |
| 21:02.36 | Waverly360 | mercestes: does that mean I'm not allowed to ask you anymore questions about asterisk? :) |
| 21:02.44 | *** join/#asterisk CrashHD (n=crashhd@67.166.155.233) |
| 21:03.34 | mercestes | Waverly360: You can ask. lol. |
| 21:04.02 | *** join/#asterisk NTJOCK (n=brian@txshirts.com) |
| 21:04.13 | mercestes | Waverly360: Worst I do with textfile mangling is cating polycom config files | though sed with s/_keyword_/_real value_/g > the file I want. |
| 21:05.30 | *** join/#asterisk gr1ncheux (n=devine@unaffiliated/gr1ncheux) |
| 21:05.39 | l2cache | b11d: well i AM going to figure this out. and ill let you know :) |
| 21:06.27 | *** join/#asterisk znoG (n=gs@OL132-95.fibertel.com.ar) |
| 21:06.33 | *** join/#asterisk CrashHD (n=crashhd@c-67-166-155-233.hsd1.ca.comcast.net) |
| 21:07.02 | *** join/#asterisk hfd (i=hfd@spc1-ayle2-0-0-cust228.asfd.broadband.ntl.com) |
| 21:07.35 | b11d | :) |
| 21:07.58 | Waverly360 | mercestes: I use HTML template to create templates of all of the config files that I need to modify, then I just use perl to write them out. |
| 21:08.49 | hfd | Hi all:) |
| 21:12.22 | syzygyBSD | hi |
| 21:12.37 | syzygyBSD | aww, wasn't there a bot that used to respond to that |
| 21:13.20 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 21:14.22 | b11d | yeah, its name is syzygyBSD |
| 21:14.39 | syzygyBSD | wow, what a coincidence |
| 21:14.55 | syzygyBSD | I just saw someone with that name, if only I could remember where |
| 21:15.24 | b11d | :) |
| 21:15.51 | NTJOCK | hi all. |
| 21:16.06 | NTJOCK | has anyone ever had a TDM400 that won't answer one particular port? |
| 21:16.31 | b11d | yepo |
| 21:16.38 | b11d | when the TDM400P had a bad FXO module |
| 21:16.44 | b11d | that was my experience anyway |
| 21:16.46 | NTJOCK | any hints on diagnosing which fxo module is bad? |
| 21:16.48 | Hmmhesays | finally I fixed my damn bootloader |
| 21:16.51 | b11d | they are in order.. |
| 21:16.54 | NTJOCK | aside from calling each POTS line individually |
| 21:16.58 | b11d | so if the fourth port doesnt work.. guess which module it is |
| 21:17.15 | NTJOCK | I was hoping that I could see on the console which module isn't getting answered |
| 21:17.38 | b11d | not sure anymore, i soon ditched my tdm400p and went with sangoma a104d's and a PRI. |
| 21:17.44 | NTJOCK | ah |
| 21:18.03 | NTJOCK | yea, we're probably going to be reducing our use of POTS lines in favor of VOIP channels. |
| 21:18.13 | b11d | cool |
| 21:18.15 | NTJOCK | I've been testing them for a while and the call quality is good to excellent. |
| 21:18.26 | b11d | yeah thats what everyone says :) |
| 21:18.27 | NTJOCK | I even tested it one time when I had our T-1 pegged out with a download |
| 21:18.39 | NTJOCK | I'm using Teliax |
| 21:18.49 | NTJOCK | someone here recommended them and they've been very very pleasant to work with |
| 21:18.52 | NTJOCK | and they support IAX |
| 21:18.55 | b11d | thats cool |
| 21:18.58 | NTJOCK | yea |
| 21:19.09 | NTJOCK | I put them on a pay as you go as a backup/rollover |
| 21:19.22 | NTJOCK | and I've been happy enough that I just moved my 800 service to them |
| 21:19.29 | NTJOCK | at 2.9c/min it's hard to beat |
| 21:19.38 | NTJOCK | we pay 6 to ATack&Terrorize |
| 21:19.42 | NTJOCK | and 10c in state |
| 21:20.08 | syzygyBSD | I never understood why instate was more then out of state |
| 21:20.13 | NTJOCK | because ti can be |
| 21:20.25 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.216.2) |
| 21:20.27 | NTJOCK | when your bent over a barrel with a pointy object against your a** you'll pay damn near anything! |
| 21:20.38 | mercestes | syzygyBSD: Taxes. If you leave the state there are no state taxes on the call, but if you call within the state they tax the shit out of you. |
| 21:20.47 | syzygyBSD | maybe you will, but I never put myself in that position |
| 21:20.50 | NTJOCK | right, again, see comment about barrel |
| 21:20.52 | cpm | does teliax just do termination? only thing I see on their page is 'features' |
| 21:21.07 | NTJOCK | I apologize for not knowing all the terms that well... |
| 21:21.12 | NTJOCK | what exactly do you mean by termination? |
| 21:21.17 | NTJOCK | just outbound? |
| 21:21.19 | NTJOCK | or just inbound? |
| 21:21.37 | cpm | no, just iax, handed off to my pbx, without any 'features' |
| 21:21.45 | NTJOCK | oh |
| 21:21.45 | NTJOCK | yea |
| 21:21.45 | *** join/#asterisk crlshn (i=kvirc@operaciones3.globalnet.hn) |
| 21:21.49 | NTJOCK | that's what we do presently |
| 21:21.53 | NTJOCK | I just disabled the "features" |
| 21:21.58 | cpm | I can't find that product, , , ah |
| 21:22.02 | NTJOCK | the pay as you go is a great "surge" service. |
| 21:22.07 | NTJOCK | $5/mo plus 2c minute |
| 21:22.39 | NTJOCK | for testing them I routed all my outbound long distance over their circuit. |
| 21:22.49 | NTJOCK | It really caused me to look and see that we weren't using our lines as much as I thought. |
| 21:23.15 | NTJOCK | I found that I could pay for 4 channels and 2500 minutes of use on their corporate plan with just what I'd save on 800 calls each month. |
| 21:23.22 | b11d | nice |
| 21:23.27 | NTJOCK | in otherwords, my 800 charges were more then their entire corporate plan. |
| 21:23.52 | NTJOCK | so I called AT&T and aksed what they do to me for canceling lines early in our contract. |
| 21:23.57 | NTJOCK | it's a hit, but still wroth it. |
| 21:24.15 | NTJOCK | AT $50/pots line (including LD) we'll save about $100 to $150/mo |
| 21:24.23 | NTJOCK | we currently have 1 fax, plus 4 pots lines |
| 21:24.29 | NTJOCK | and we're going to drop 3 pots lines. |
| 21:24.35 | NTJOCK | Keep one for "just in case" |
| 21:24.43 | NTJOCK | and move everything else over. |
| 21:25.18 | NTJOCK | The best indicator that the service is good is that I accidentally left in testing mode one morning and let my employees all call out on it. Nobody noticed anything except the calls sounding better. |
| 21:25.30 | NTJOCK | and one person's girlfriend asked why a 303 # showed up instead of our regular number. |
| 21:25.31 | b11d | that's satisfying eh |
| 21:25.31 | NTJOCK | lol |
| 21:25.42 | NTJOCK | yup. |
| 21:25.47 | cpm | hrmm .02 inbound/outbound, , , hrmm |
| 21:25.53 | NTJOCK | yea, it's cheap. |
| 21:27.09 | NTJOCK | I rewrote my extensions.conf to route local calls on POTS and LD on Teliax |
| 21:27.09 | NTJOCK | It basically let's us have the flexibility of a T-1 Voice setup with out the hassle or expense. |
| 21:27.09 | NTJOCK | in that we can have multiple channels. |
| 21:27.14 | NTJOCK | With pay as you go you get 10 channels by default.... |
| 21:27.20 | NTJOCK | but with the corp. plan you only get 4 |
| 21:27.25 | NTJOCK | and then it's $10 per channel extra. |
| 21:27.29 | NTJOCK | which is still cheap |
| 21:27.47 | NTJOCK | that's kinda silly but whatever..... it's still cheaper |
| 21:28.06 | NTJOCK | we had a heck of a time getting broadvoice to actually work in testing. |
| 21:28.08 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 21:28.20 | NTJOCK | and we had some very bad experiences early on with Vonage |
| 21:28.27 | NTJOCK | so I've been skeptical of call quality on VOIP. |
| 21:28.35 | NTJOCK | or external voip to be more specific. |
| 21:28.55 | *** part/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 21:29.15 | NTJOCK | I decided to give it another whirl because a buddy who has * and is on Damn Slow Link setup some backup/surge circuits on VOIP and said it worked well. |
| 21:29.16 | cpm | I get so-so service from voicepulse |
| 21:29.26 | NTJOCK | Teliax has been good so far. |
| 21:29.30 | NTJOCK | it was tricky to get setup |
| 21:29.33 | NTJOCK | but bulletproof after that |
| 21:29.59 | NTJOCK | The real charm will be when I try to get * to answer the phones for multiple company/identities. |
| 21:30.31 | NTJOCK | we have two websites that run as niche companies and they don't have phones yet... but a Pay as you go would be a simple way to give them their 10 minutes of calling a month. |
| 21:30.45 | NTJOCK | I know we could do it in *, just haven't tried yet. |
| 21:31.00 | NTJOCK | Just route those lines to a special spot in ext.cfg and give them their own announcements. |
| 21:31.03 | NTJOCK | :) |
| 21:31.05 | NTJOCK | or so I figure. |
| 21:34.48 | *** join/#asterisk Hmmhesays (n=Neg@24-119-151-57.cpe.cableone.net) |
| 21:35.02 | Hmmhesays | ok this cannot find -lqt thing is still driving me nuts |
| 21:35.25 | b11d | are the lib's installed? |
| 21:35.31 | b11d | ldconfig -R doenst pick them up? |
| 21:35.47 | Hmmhesays | the libs are install in /usr/lib/qt3.3/lib |
| 21:35.58 | b11d | what if you symlink them into /usr/lib ? |
| 21:36.04 | b11d | and then ldconfig -R and then -r |
| 21:36.11 | b11d | doh thats freebsd though.. |
| 21:36.34 | carrar | add the path to /etc/ld.so.conf |
| 21:36.40 | carrar | and run ldconfig |
| 21:39.21 | *** join/#asterisk guilherme-jorge (n=guilherm@200-170-201-134.core01.spo.ifx.net.br) |
| 21:39.32 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 21:40.38 | Hmmhesays | yeah I did |
| 21:41.11 | guilherme-jorge | hello all, I get the following message in asterisk console: " WARNING[20563]: chan_sip.c:2561 sip_write: Asked to transmit frame type 8, while native formats is 256 (read/write = 256/256)". When I say something the another part doesnt hear what I said... Any idea? |
| 21:41.15 | b11d | copy the libs to /usr/lib and run ldconfig -R |
| 21:41.17 | b11d | just to see if it works |
| 21:41.20 | Hmmhesays | i don't know what the hell lqt is though cause it ain't in my qt directory |
| 21:41.27 | Hmmhesays | -R is not an option in fedora core |
| 21:41.28 | *** join/#asterisk CaRb0n^ (n=Omer@203.81.233.47) |
| 21:41.30 | b11d | or whatever the "rescan" option is for linux |
| 21:41.57 | Hmmhesays | just ldconfig |
| 21:42.05 | b11d | ah |
| 21:42.14 | b11d | is there a verbose option? |
| 21:42.22 | b11d | fuck, strace the bastard |
| 21:42.30 | b11d | or whatever it is in linux |
| 21:42.41 | *** join/#asterisk aaronr (n=arussell@87.127.234.100) |
| 21:43.41 | *** join/#asterisk ZefK (n=Zefk@wsc-fo.b.astral.ro) |
| 21:43.47 | Asteriskmonkey | anyone had issues with asteirsk box -> iax2 -> asterisk box voicemail not working? |
| 21:44.12 | Hmmhesays | ok fedora is shitty |
| 21:44.15 | b11d | yeap. |
| 21:44.17 | Hmmhesays | i fixed that problem |
| 21:44.21 | b11d | what needs -qt? |
| 21:44.22 | Hmmhesays | now I get can't locate lGL |
| 21:44.24 | b11d | just compile without it |
| 21:44.27 | b11d | ah |
| 21:44.37 | Hmmhesays | i have not idea what gl is |
| 21:44.45 | Hmmhesays | open gl library maybe? |
| 21:44.54 | ZefK | ANyoane knows why Wait(1) waits for ~11sec ? |
| 21:44.58 | *** join/#asterisk TheSov (n=TheSov@dsl081-140-246.chi1.dsl.speakeasy.net) |
| 21:45.33 | b11d | I suspect you're right Hmmhesays |
| 21:45.51 | b11d | what are you compiling? |
| 21:45.52 | Hmmhesays | being i'm compiling the ati video card controller |
| 21:46.16 | b11d | aue |
| 21:46.17 | b11d | aye |
| 21:46.34 | denon | you're compiling the controller? |
| 21:46.38 | denon | new code for the asics? |
| 21:46.44 | ZefK | I'm running *1.4.0 and wait command waits with 10 sec more than the argument passed. Any hints ? |
| 21:47.18 | b11d | yeah Hmmhesays writes all the code for ATI |
| 21:47.20 | denon | ZefK: replace it with a noop() and make sure it's not something else taking the time? |
| 21:47.22 | b11d | you didnt know? |
| 21:47.32 | denon | b11d: hehe right |
| 21:47.35 | b11d | haha |
| 21:47.58 | denon | you could have told me he was an Apple coder, that I would have bought |
| 21:48.19 | b11d | :) |
| 21:48.29 | ZefK | denon: I can see the command in the console window. It is the only command in the context. |
| 21:49.00 | denon | ZefK: well, if you ask for suggestions .. |
| 21:49.07 | denon | you'll either need to follow em, or ignore em :) |
| 21:49.13 | TheSov | I'm trying to setup asterisk on an all VoIP setup and we have a sip provider and I would like to know how I would get asterisk to make calls out and recieve call in on that sip provider. I'm having trouble understanding the documentation in this respect. |
| 21:49.27 | b11d | how EXACTLY are you using Wait ? |
| 21:50.08 | b11d | lets see the exact line |
| 21:50.44 | ZefK | b11d: s => { Answer; Wait(1); } |
| 21:50.57 | b11d | oh, ael stuff |
| 21:51.14 | denon | yeah, my ael's a bit rusty too |
| 21:52.00 | *** join/#asterisk ToyMan (n=Stuart@user-0cevdmv.cable.mindspring.com) |
| 21:52.01 | TheSov | Can anyone help me with that? |
| 21:52.05 | *** join/#asterisk af_ (n=getsmart@ip-202-133.sn2.eutelia.it) |
| 21:52.48 | ZefK | I'll try in a sec to use the standard dialplan in extensions.conf ... |
| 21:52.54 | denon | ZefK: I'd still toss a noop(something) in there instead, and make sure it's not something screwy with answer |
| 21:53.22 | denon | nod, switching over to standard would help |
| 21:53.25 | denon | at least to test |
| 21:53.53 | eald | I have an asterisk trying te record every call with Monitor and there are soxmix process since last half hour |
| 21:55.04 | JunK-Y | eald: stop using the option to soxmix everythin? |
| 21:55.55 | ZefK | denon: the wait command works fine in extensions.conf |
| 21:56.01 | CrashHD | is there anyway to play voicemails from newest to oldest with asterisk? |
| 21:56.20 | denon | ZefK: I know ael can be a bit .. beta. . sometimes, though |
| 21:56.54 | lokkju_wrk | ok, so I am confused... I am trying to connect to asterisk via IDEFisk over IAX. IDEFisk works from this machine against FWD. However, it won't work against my * server. * sees the incoming communications, but the outgoing get dropped. The * has two interfaces. If I connect over the internal, 10.10.99.* address (over a VPN), everything works fine. If I connect via the external interface, then * sees the incoming traffic, but it's return |
| 21:56.54 | lokkju_wrk | traffic does not get through. Since IDEFisk works fine from this machine with FWD, it would see that it has to be either my iptables rules, or else something in front of the * box. I have totally cleared my * rules, and set them all to ACCEPT. Anything I am missing? |
| 21:56.54 | *** join/#asterisk bkunyiha (i=Billk@66-113-79-5.rev.ibsinc.com) |
| 21:57.22 | bkunyiha | I have a question on asterisk realtime |
| 21:57.32 | lokkju_wrk | (and there is no nat between the * box and the internet) |
| 21:58.39 | eald | JunK-Y: that could help |
| 21:59.02 | bkunyiha | When you make a call the Dial function cheaks the user in the cache before checking the database. How can you make asterisk check the database and not use the cache? |
| 22:00.39 | [TK]D-Fender | TheSov: Who's the provider, and what have you set up on * so far? |
| 22:00.47 | TheSov | speakeasy.net, actually they use onvoip |
| 22:01.02 | *** join/#asterisk juro (n=chatzill@dsl-241-66-90.telkomadsl.co.za) |
| 22:01.03 | TheSov | i have 2 extensions, the sip proxy info in the sip.conf |
| 22:01.29 | [TK]D-Fender | TheSov: And the 2 sip phones can call each other? |
| 22:01.47 | TheSov | yes by dialing the extension setup in the extensions.conf |
| 22:02.04 | juro | hi. does any1 here have information on how to interface a web-application (php) with asterisk? |
| 22:03.07 | [TK]D-Fender | TheSov: Ok, start by trying to make a SIP peer entry for the provider to dial out from. to use it you'd do something like "exten => _9x.,1,Dial(SIP/myproviderpeerentry/${EXTEN:1}) |
| 22:03.27 | bkruse | [TK]D-Fender: how many tiems do you say that a day? |
| 22:03.27 | bkruse | lol |
| 22:03.28 | *** join/#asterisk bmd (n=bmd@72.54.252.34) |
| 22:03.44 | [TK]D-Fender | bkruse: Actualy, the first time in MONTHS :) |
| 22:03.57 | bkruse | nice |
| 22:03.58 | bkruse | good |
| 22:04.13 | *** part/#asterisk l2cache (n=ghansen@64.128.254.98) |
| 22:04.20 | TheSov | fender: that will give them an outside line by dialing 9? |
| 22:06.07 | eald | JunK-Y: what I see now is that the sh soxmix don't die |
| 22:07.11 | *** join/#asterisk e-milio (n=emilio@pmr.pmrtechnologies.com) |
| 22:07.31 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 22:07.58 | e-milio | Hello All |
| 22:08.06 | juro | or could any1 point me to a ressource regarding interfacing asterisk with php? |
| 22:08.44 | e-milio | I know this might not be the right but I would like to ask opionions/recommendations for an IAX termination provider |
| 22:08.45 | [TK]D-Fender | TheSov: a number starting with 9, yes. |
| 22:09.15 | TheSov | [TK]D-Fender: thank you |
| 22:11.32 | JunK-Y | kill it? |
| 22:15.57 | *** join/#asterisk harleya (n=xyharley@c-67-166-122-212.hsd1.ut.comcast.net) |
| 22:19.09 | juro | hello?A |
| 22:19.42 | creature1 | hi |
| 22:20.37 | juro | sorry to ask this again but I am kind of desperate ;). does any1 here have information on how to interface a web-application (php) with asterisk? |
| 22:20.41 | *** join/#asterisk hohum (n=dcorbe@c-71-62-76-68.hsd1.va.comcast.net) |
| 22:20.55 | *** join/#asterisk jart (n=user@ool-43509aa5.dyn.optonline.net) |
| 22:21.05 | jart | hi peoples :) |
| 22:23.10 | [TK]D-Fender | juro: to do WHAT? |
| 22:23.24 | lokkju_wrk | whee, found my problem - is there any way to instruct asterisk to communicate back out on that same interface and ip address as the data came in on? specifically, for IAX? my issue is that my * box has two ips on one interface, and I am trying to connect via IAX to on of the ips, but * is then communicating back using the other ip... I do not want to make a global change though, routing wise |
| 22:24.32 | lokkju_wrk | juro, um, to control asterisk? |
| 22:24.35 | e-milio | juro: look at manager interface |
| 22:26.05 | mihinomenest | lokkju_wrk: I don't suppose you could just set the bindaddr... |
| 22:26.08 | juro | well, i am but a small developer building a crm with asterisk as the pbx. i have not used asterisk as of yet, but have to interface with it, i.e. make calls, hang up, get recording-ids, etc, from a php-driven web-application |
| 22:26.09 | creature1 | lokkju_wrk: might work for you to specify the bindaddr? but i suppose that's now what you want. |
| 22:27.16 | lokkju_wrk | nope, because I use it on the internal addy too |
| 22:27.39 | lokkju_wrk | (well, actually, internal, and both externals) |
| 22:28.04 | lokkju_wrk | different ip blocks, and different routes on the externals, for route failure reasons |
| 22:28.07 | creature1 | lokkju_wrk: i think you would have to set up the network route for that, i don't think it's something you can solve with asterisk. |
| 22:28.19 | lokkju_wrk | creature1, it isn't routing though |
| 22:28.21 | creature1 | but i'm far from sure |
| 22:28.40 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 22:29.12 | lokkju_wrk | it should be pretty simple - if you have a network card with 10.10.1.1 and 10.10.2.1, both on the same card, then asterisk *should* really respond back from the same ip that it was communicated to on. |
| 22:30.39 | creature1 | lokkju_wrk: that would be most logical, yes. but you've come to the conclusion looking at the debug that it's not that simple? |
| 22:30.43 | lokkju_wrk | creature1, the communication is coming in on 10.10.2.1, and going out on 10.10.1.1 (not real addys, obviously) |
| 22:30.47 | lokkju_wrk | 10.10.1.1 is the default for the card |
| 22:30.58 | lokkju_wrk | but any application can choose which to send from |
| 22:34.41 | creature1 | lokkju_wrk: i understand the problem, but i'm sorry to say i'm not sure how to solve that. |
| 22:35.05 | lokkju_wrk | creature1, yeah, I am thinking there has to be a setting somewhere... otherwise, it is a bug |
| 22:37.48 | TheSov | I'm still having trouble dialing out via our sip proxy |
| 22:38.28 | TheSov | I recieve an error starting with "app_dial.c:1081 dial_exec_full: Unable to create channel of type" |
| 22:38.52 | creature1 | TheSov: maybe you try to use a codec that' |
| 22:38.53 | TheSov | Ending with "no route to destination |
| 22:38.55 | creature1 | s not supported |
| 22:39.14 | creature1 | TheSov: type sip show registry to confirm that you are registered at first |
| 22:39.34 | TheSov | rcps.onvoip.net:5060 7737285106 104 Registered |
| 22:40.13 | TheSov | perhaps my extentions.conf is misconfigured |
| 22:40.19 | creature1 | TheSov: Paste your sip.conf and extensions.conf only masking out the passwords |
| 22:40.21 | creature1 | ~paste |
| 22:40.22 | jbot | rumour has it, paste is http://rafb.net/paste/ |
| 22:40.44 | lokkju_wrk | creature1, well, geez... that is just horrible design on asterisk's part |
| 22:40.47 | TheSov | Thank you |
| 22:41.12 | creature1 | lokkju_wrk: seems like, found that link i gave you interesting? |
| 22:41.49 | creature1 | lokkju_wrk: i'm far from expert on asterisk, just used it for ~2 weeks now, there may be a sollution that one of the more advanced users know of. |
| 22:42.48 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 22:43.39 | TheSov | creature1: I dont have the means to paste the whole thing can I simply paste the relavent sections? |
| 22:44.22 | [TK]D-Fender | juro: Start by reading THE BOOK to learn about the various interfaces you have at your disposal including AMI & AGI |
| 22:44.24 | [TK]D-Fender | ~book |
| 22:44.28 | jbot | hmm... book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 22:44.38 | creature1 | TheSov: sure |
| 22:45.10 | creature1 | TheSov: the user and the part of the users default context should be enough. |
| 22:45.36 | TheSov | http://rafb.net/p/YtF2P915.html |
| 22:46.38 | [TK]D-Fender | TheSov: where is your peer entry for [7001]? |
| 22:46.48 | TheSov | sip.conf |
| 22:46.49 | creature1 | TheSov: you need to.. ah d-fender already wrote :) |
| 22:47.06 | lokkju_wrk | mwahahah... I think I found a solution, though it will be a pain in the ass... use iptables to do rewriting based on the ip interface the communication came in on |
| 22:47.16 | juro | [TK]D-Fender: I read the chapter 9 but I couldn't see any information about placing calls etc. I do not want to become a Asterisk-guru, but am head developer for a crm to interface with Asterisk. |
| 22:47.22 | [TK]D-Fender | TheSov: Why don't we SEE this in your pastebin? And its probably be a good idea to inclu the full CLI output of your failed call as well |
| 22:47.30 | juro | (but that is only a small part of the project) |
| 22:48.01 | [TK]D-Fender | juro: Sorry, but you're going to HAVE to. There is no "Whatever juro wants" interface out there. this WILL take some learning on your part. |
| 22:48.16 | lokkju_wrk | [TK]D-Fender, you seem to be one of the experts - any idea on having two external interfaces, and making asterisk use the interface that the communication came in on? |
| 22:48.23 | TheSov | My mistake, I will try to be more thorough |
| 22:48.58 | [TK]D-Fender | lokkju_wrk: Typically it WILL respond on the interface it came in on. |
| 22:49.19 | lokkju_wrk | juro, crm to integrate? you mean, be able to place calls? depending on the CRM, there are already plugins for that |
| 22:49.23 | juro | [TK]D-Fender, NOT *damn* .... then I shall do it the hard way. Just a short question, can I set up Asterisk on Ubuntu-Server just to test it (so that it doesn't actually call out or something= |
| 22:49.30 | [TK]D-Fender | lokkju_wrk: Assuming you left bindaddr to 0.0.0.0 |
| 22:49.54 | juro | lokkju_wrk, custom crm - nothing of the shelve. |
| 22:50.25 | [TK]D-Fender | juro: Yes, once all the pacckage dependencies are in place of course you can install * on Ubuntu Server |
| 22:50.50 | lokkju_wrk | [TK]D-Fender, in my case, it is not - I have two *public* interfaces, xxx.xxx.242.70 and xxx.xxx.1.114, both on the same NIC. xxx.xxx.1.114 is the default (primary). I have my IAX client pointing at xxx.xxx.242.70, but when * responds, it comes from xxx.xxx.1.114 |
| 22:50.57 | *** join/#asterisk ToyMan (n=Stuart@user-0cevdmv.cable.mindspring.com) |
| 22:51.07 | juro | [TK]D-Fender, ok thanx. let's see if I can get that running on a vm-ware |
| 22:51.15 | creature1 | juro: should work just fine |
| 22:51.19 | [TK]D-Fender | juro: What is there to 'test"? *'s capabilities are relative well know, even by those who aren't actually using it it a given capacity... |
| 22:52.10 | [TK]D-Fender | lokkju_wrk: Well what interface was the incoming request targeting? perhaps a routing isse? |
| 22:52.27 | juro | [TK]D-Fender, it is easier seeing things work in a test environment as opposed to writing code and then hoping that it will interface properly on go-live day |
| 22:53.30 | lokkju_wrk | [TK]D-Fender, no, as I said, incoming request was targeted at xxx.xxx.242.70 |
| 22:54.03 | lokkju_wrk | [TK]D-Fender, default route sends everything over xxx.xxx.1.114 - but asterisk should still respond on the ip the request came in on, shouldn't it? |
| 22:54.36 | creature1 | juro: sounds better to me if you asked the person/company who ordered the application to set up a test server for you that is pretty simmilar to their own server. |
| 22:55.37 | lokkju_wrk | juro, creature1 has a good idea - even if they setup a test mirror of their existing config, and then you can vpn to it |
| 22:56.08 | juro | creature1, yes in the long run definetly. to start off and make a prliminary can-do list, it would really be easier just to see what I can do with PHP to start off with |
| 22:56.13 | mercestes | lokkju_wrk: Didn't work without the NAT, did it? |
| 22:56.25 | juro | (sorry for being slightly n00b in this matter ;) ) |
| 22:56.28 | lokkju_wrk | mercestes, what NAT? |
| 22:56.48 | mercestes | lokkju_wrk: For your IAX. |
| 22:56.51 | lokkju_wrk | mercestes, on the client side? it might - not sure |
| 22:56.56 | creature1 | juro: then i would suggest that you have a look at such a interface that's already built |
| 22:56.59 | mercestes | lokkju_wrk: Weren't you in here hours ago complainig that IAX didn't work over nat? |
| 22:57.22 | juro | creature1, you mean for an existing crm? |
| 22:57.25 | lokkju_wrk | mercestes, yeah, and this is why - my server is responding back from a different address then the original request was sent to |
| 22:57.31 | creature1 | juro: yeah, have a look at an existing crm |
| 22:57.58 | mercestes | lokkju_wrk: Did you do as I instructed and install asterisk on your router and setup IAX on the public side??? |
| 22:58.02 | creature1 | and read the asterisk docs, then you'll know what capabilities it have |
| 22:58.07 | mercestes | or are you bakc here trolling bug? |
| 22:58.22 | juro | creature1, unfortunately there is no existing crm that reproduces the workflow (not in a long run) |
| 22:58.27 | juro | no, no trolling bug here ;) |
| 22:58.43 | mercestes | juro: Not you, lokkju_wrk over there. |
| 22:59.05 | [TK]D-Fender | lokkju_wrk: I'm not sure based on your default route... |
| 22:59.10 | creature1 | juro: ok. have a look at this for starters: http://www.voip-info.org/wiki/view/Asterisk+AGI+php |
| 22:59.47 | TheSov | http://rafb.net/p/WXPwPf86.html |
| 23:00.46 | [TK]D-Fender | juro: Go see what has been done between SugarCRM , Request Tracker & Asterisk for an idea. |
| 23:00.52 | creature1 | TheSov: try with Dial(Dial(SIP/${EXTEN}@7001); |
| 23:00.54 | creature1 | oops |
| 23:01.16 | creature1 | Dial(SIP/${EXTEN:1}@7001) |
| 23:01.37 | juro | [TK]D-Fender, ok. thanx, will. do. should keep me from trolling for a while ... gone reading ;) |
| 23:01.49 | creature1 | TheSov: also, start asterisk in cli debug mode so that you are able to see what actually happens. |
| 23:02.21 | TheSov | ok |
| 23:02.22 | [TK]D-Fender | juro: No, you are definately SOMEWHERE above the level of "trolling". Exactly whree, or how far I'm unsure ;) |
| 23:02.52 | [TK]D-Fender | TheSov: Please pastebin the CLI output of a failed call. |
| 23:03.01 | juro | [TK]D-Fender, I'll get there ... learnt the hard way that there are no stupid questions .... so I keep asking them. |
| 23:03.20 | *** join/#asterisk doolph (n=doolph@200.105.35.219) |
| 23:03.25 | doolph | hi |
| 23:03.31 | [TK]D-Fender | juro: No, the saying is "there are dumb answers, only dumb QUESTIONS". |
| 23:03.35 | doolph | anyone can help me with SIP error messages? |
| 23:03.45 | [TK]D-Fender | doolph: like..... |
| 23:03.48 | lokkju_wrk | [TK]D-Fender, well, ever system is going to have a default public route - have you never run across a system running asterisk with two public ips, with asterisk running on the non-default one? |
| 23:03.57 | creature1 | doolph: just shoot, and the one/ones that can and wants to will help you |
| 23:04.08 | TheSov | ok i started asterisk with those switches and now I see a lot of messages about sending fake auth rejections |
| 23:04.16 | doolph | I am getting SIP response 484 "Address Incomplete" from the provider but they means 503 |
| 23:04.18 | [TK]D-Fender | lokkju_wrk: You lost me at HAVING 2 public IP's period ;) |
| 23:04.23 | creature1 | doolph: if you are going to show an error message that contains many rows then use pastebin |
| 23:04.26 | creature1 | ~pastebin |
| 23:04.27 | jbot | somebody said pastebin was a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/, or http://pastebin.com is usually painfully too slow and unresponsive to use, use one of the other pastebin sites, or dpaste.com is a very nice pastebin as well |
| 23:04.34 | juro | [TK]D-Fender, hmmm ... "there are dumb answers, only dumb QUESTIONS" - doesn't read very optimistic |
| 23:04.49 | doolph | there's ask asterisk understand 484 as 503 instead |
| 23:05.01 | doolph | there's anyway? |
| 23:05.34 | [TK]D-Fender | juro: Realism > optimism and != pessimism (though can often be confused for it) :D |
| 23:06.23 | juro | [TK]D-Fender, unfortunately. thanx girls/guys (damn political correctness) |
| 23:07.13 | TheSov | http://rafb.net/p/Kv8lRR44.html |
| 23:07.13 | [TK]D-Fender | juro: Good luck with your research. Shouldn't take you long to find out if it'll do what you need/hope |
| 23:07.14 | lokkju_wrk | [TK]D-Fender, funny... one our hosting boxes has 64, and that is nothing |
| 23:07.46 | lokkju_wrk | [TK]D-Fender, so, other then using the default ip as my target ip, you have no ideas? |
| 23:07.50 | juro | [TK]D-Fender, I am quite sure it does. But "quite" isn't a word you want to use when you are designing software |
| 23:07.58 | *** join/#asterisk stefmtl (n=stef@stef.istop.com) |
| 23:08.08 | [TK]D-Fender | lokkju_wrk: I referring more to the acknowledgement of my level of COMPETANCE with such matters ;) |
| 23:08.16 | lokkju_wrk | [TK]D-Fender, (of course, as I mentioned earlier, I think I *could* do something with iptables) |
| 23:08.42 | stefmtl | I have a lot of core dumps http://bugs.digium.com/file_download.php?file_id=13446&type=bug What can I do ? |
| 23:09.12 | [TK]D-Fender | TheSov: I'd be sure in your Linux CLI that you can even SEE that host... |
| 23:09.19 | stefmtl | I version 1.2.16 |
| 23:09.24 | [TK]D-Fender | TheSov: I might suspect it isn't resolving |
| 23:09.41 | TheSov | strange |
| 23:10.02 | lokkju_wrk | juro,seariously, look at some of the TAPI dialers for outlook and such... essentially, you are going to be telling asterisk "place this call from extension XXX to number XXX, and ih, tell the extension to pick up" |
| 23:10.14 | [TK]D-Fender | TheSov: Actually... are you sure that # you are dialing is in a legit format? |
| 23:10.53 | *** join/#asterisk ToyMan (n=Stuart@user-0cevdmv.cable.mindspring.com) |
| 23:11.03 | TheSov | what do u mean? is it an actual phone number? |
| 23:11.07 | *** part/#asterisk MrChicken (n=MrChicke@200.71.58.39) |
| 23:11.13 | juro | lokkjo_wrk, yes, and then I need * to tell me what the recording-id is for that specific call - no call reception in this system though |
| 23:11.29 | stefmtl | anyone experimenting the same issue than me ? |
| 23:11.31 | *** join/#asterisk dseeb_ (n=dcb@CPE-58-169-152-56.vic.bigpond.net.au) |
| 23:11.58 | TheSov | How else does someone dial a normal phone # via sip? |
| 23:12.55 | doolph | not me |
| 23:15.12 | Waverly360 | TheSov: exten => _X.,1,Dial(Zap/g2/5551212) ??? |
| 23:15.22 | Waverly360 | TheSov: Not exactly sure what you're asking |
| 23:15.34 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 23:17.23 | TheSov | I'm dialing from a voip phone to a pots phone via a sip provider I'm trying to find out why it doesnt work |
| 23:20.56 | Waverly360 | I'm not sure. |
| 23:21.21 | stefmtl | is the DDO_CRASH for THREADS compilation option is risky in production environment ? That is to say I could have not justified crashes ? |
| 23:22.07 | [TK]D-Fender | TheSov: Its very likely that your provider requires you to dial a 10 or 11 digit number.... |
| 23:22.22 | [TK]D-Fender | TheSov: You have to think outside of just YOUR BACKYARD :) |
| 23:23.00 | Waverly360 | TheSov: Don't take it personally. Fender's a punk everyday. :) |
| 23:23.58 | *** join/#asterisk X-Rob (n=Rob@ppp214-210.static.internode.on.net) |
| 23:24.36 | [TK]D-Fender | Waverly360: No, sometimes I'm blues, glam-rock, heavy metal, or classical :) |
| 23:24.39 | creature1 | [TK]D-Fender: he had missed the host parameter |
| 23:25.02 | Waverly360 | [TK]D-Fender: badump bump. |
| 23:25.06 | [TK]D-Fender | creature1: It was in his pastebin just fine. |
| 23:25.51 | creature1 | [TK]D-Fender: i meant fromdomain |
| 23:25.54 | Waverly360 | Anyone here have a preference to using internal PRI and Analog cards over remote PSTN gateway devices? |
| 23:26.10 | creature1 | [TK]D-Fender: and insecure |
| 23:26.24 | [TK]D-Fender | creature1: VERY few providers require "fromdomian" in my experience |
| 23:26.33 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 23:26.37 | [TK]D-Fender | creature1: And we're jsut dealing with OUTBOUND ATM |
| 23:26.47 | creature1 | [TK]D-Fender: may so be, but that solved his problems |
| 23:27.04 | doolph | omg this provider is stupid, is sending me 484 sip error code when they cannot connect |
| 23:27.08 | [TK]D-Fender | Waverly360: As in PCI VS SIP->PSTN gateways(a la audiocodes for example)? |
| 23:27.19 | lokkju_wrk | ok, extensions can call *43, and they show as peers... but now dailparties.agi is still not returning any extensions to call when calling between them |
| 23:27.26 | Waverly360 | [TK]D-Fender: yep |
| 23:27.44 | Waverly360 | [TK]D-Fender: I'd like to get away from using PCI devices at all if possible, and stick to external devices. |
| 23:28.08 | [TK]D-Fender | Waverly360: well for starters on PCI you can use SpanDSP with some reliability, the cost is way lower, typically a lot easier to set up, etc. |
| 23:29.11 | [TK]D-Fender | Waverly360: On the other hand gateways typicall encode to the codec of choice (if G.729 matters this can factor in), can allow for redundancy, allow physical speration of media & server, etc. |
| 23:29.21 | [TK]D-Fender | Waverly360: Each have very distinct merits. |
| 23:29.38 | [TK]D-Fender | lokkju_wrk: .... |
| 23:29.39 | [TK]D-Fender | ~freepbx |
| 23:29.41 | jbot | i heard freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 23:30.44 | lokkju_wrk | [TK]D-Fender, heh, right, forgot that that was put in their by freepbx |
| 23:30.53 | [TK]D-Fender | creature1: And I didn't see him say anywhere that he had succeeded with the use of "fromdomain". Where? |
| 23:31.59 | creature1 | [TK]D-Fender: i helped him in query |
| 23:32.59 | TheSov | yes I got it working |
| 23:33.10 | *** join/#asterisk Splat (n=splat@eth112.tas.adsl.internode.on.net) |
| 23:33.12 | [TK]D-Fender | creature1: Ah..... helps us help him doesn't it? ;) Ok, well glad it was discovered. |
| 23:33.17 | Waverly360 | [TK]D-Fender: Well, adding external devices like that would make adding and removing a PRI and or Analog port much simpler. |
| 23:33.23 | TheSov | Sorry I didn't inform you |
| 23:33.37 | creature1 | [TK]D-Fender: when we got it solved i informed you :) |
| 23:33.38 | [TK]D-Fender | TheSov: Don't mind me.. I'll just sit here confused :) |
| 23:34.17 | *** join/#asterisk X-Rob (n=Rob@ppp214-210.static.internode.on.net) |
| 23:34.21 | TheSov | Well its my fault because I do appreciate that you guys are all here helping others |
| 23:34.29 | [TK]D-Fender | Waverly360: Keep in mind the cost differnce IS big, but scale the solution to the task being performed. |
| 23:34.55 | [TK]D-Fender | Waverly360: Single co PBX? Nah... Co-location termination server? Hell yeah. |
| 23:34.56 | Waverly360 | [TK]D-Fender: How big are we talking? |
| 23:35.44 | Waverly360 | [TK]D-Fender: Yeah, we're talking about multi-location. I'm also curious about the possibility of multiple asterisk boxes sharing a single pri device. |
| 23:35.49 | [TK]D-Fender | Waverly360: In cases where you are terminating for a large number of sites you often see SER deployed with AudioCodes gateways and the like. |
| 23:36.26 | [TK]D-Fender | Waverly360: At a certain point * typically becomes more of an application server than "PBX" in those implementations |
| 23:36.54 | Waverly360 | [TK]D-Fender: I'm not sure I follow... |
| 23:38.43 | Waverly360 | [TK]D-Fender: Actually, it will have to wait..I didn't realize how late it was. |
| 23:38.50 | [TK]D-Fender | Waverly360: * hasn't traditionally scaled very large and was not built for redundancy, etc. SER is, and is oftn used as a front-end soft-switch for remote users and * only used for VM, IVR, etc. This allows you to do things like rotating DNS for proxies, and they can choose from amongst your gateways a path for any call to take |
| 23:39.19 | [TK]D-Fender | Waverly360: That should give you some food for thought :) |
| 23:39.22 | eald | I'll be glad if someone can help here, I have an asterisk 1.2.12.1 which have been running from around 2 months, now I had problem the soxmix and sh soxmix processes from Monitor command couldn't, then I stopped the asterisk server since the restart of that porcess it doesn't finnish it's load process with a constant 99% of cpu usage |
| 23:39.55 | eald | this problem begun 2 hours ago, and I can't start * since one hour ago |
| 23:41.21 | Waverly360 | [TK]D-Fender: Now that's something we've been talking about for awhile. I've not done any research (nor have I heard much about) SER. I'll do some research on that...At one point my manager had talked about using freeswitch in conjunction with asterisk. Freeswitch as the softswitch, and asterisk as the pstn connector so-to-speak. |
| 23:42.18 | denon | why use freeswitch at all? |
| 23:42.52 | Waverly360 | denon: well, that was his idea not mine. to my knowledge, freeswitch isn't ready to be used yet. |
| 23:42.57 | *** join/#asterisk TokyoJimu (n=jimmy@sunray2.nccom.com) |
| 23:43.16 | Waverly360 | denon: at any rate, we run 100mph here, so we never really have much time to follow through on even the simplest thought process. It was merely talk and speculation |
| 23:43.32 | Waverly360 | I have to go, I appreciate the help. You guys take it easy. Might be on later. |
| 23:43.43 | denon | heh, I guess |
| 23:43.55 | *** join/#asterisk ToyMan (n=Stuart@user-0cevdmv.cable.mindspring.com) |
| 23:44.56 | TheSov | ok now that I dial out working, im having issues with calling in, getting the message http://rafb.net/p/ibrsjU58.html |
| 23:46.24 | stefmtl | anyone using DDEBUG_THREADS DDO_CRASH compilation flags ? |
| 23:46.34 | *** join/#asterisk thansen|laptop (n=thansen@137.65.169.7) |
| 23:47.07 | [TK]D-Fender | TheSov: Set a context in [general] and set "insecure=very" as well. Then try again |
| 23:47.21 | TheSov | is that smart? |
| 23:47.30 | TheSov | cuz this is gonna be live |
| 23:47.57 | *** join/#asterisk xuser (n=boo@unaffiliated/xuser) |
| 23:47.58 | thansen|laptop | is it possible to enforce which codec is used? |
| 23:48.04 | *** join/#asterisk michaelo (n=michaelo@adsl-068-159-111-129.sip.gsp.bellsouth.net) |
| 23:48.07 | [TK]D-Fender | Theok, try to make a USER entry then with the host, context and "insecure=very" in it |
| 23:48.17 | [TK]D-Fender | TheSov: |
| 23:48.24 | TheSov | not to question your knowledge im just wondering if its ok in a buisness environment |
| 23:48.27 | xuser | Can asterik do e-fax? |
| 23:48.33 | [TK]D-Fender | TheSov: That will make getting a call from them rather easy |
| 23:48.49 | [TK]D-Fender | xuser: Explain "e-fax" |
| 23:49.28 | xuser | [TK]D-Fender: fax through email, using real numbers. |
| 23:52.11 | denon | xuser: asterisk fax stuff isnt terribly useful in the real world yet |
| 23:52.37 | *** join/#asterisk adde (n=adde@tn-84-218-52-59.dsl.bredbandsbolaget.se) |
| 23:53.41 | adde | Anyone who knows of a SIP Gateway in india? Kneed an Indian phonenumber and cheap calls within india... ? Ofcourse so it works with Asterisk....? |
| 23:54.10 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 23:54.28 | [TK]D-Fender | xuser: well if you have an analog or T1/E1 card you can use SpanDSP to send/receive faxes. You can then e-mail them out,e tc... |
| 23:55.22 | TheSov | [TK]D-Fender ok I set insecure=very in [general] of sip.conf, now when i call i get a generic subscriber is not in service message |
| 23:55.52 | [TK]D-Fender | TheSov: Try my second suggestion, and include the "fromdomain" in it too... |
| 23:57.59 | xuser | yeah like denon said, the fax stuff is work in pogress. |
| 23:58.30 | xuser | as with SpanDSP also. |