| 00:00.08 | *** part/#asterisk rcuza (n=rcuza@ool-18bd0fc5.dyn.optonline.net) |
| 00:03.55 | *** join/#asterisk remmo (n=junk@smack.isp.net.au) |
| 00:07.02 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 00:15.18 | *** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 00:16.11 | rue_mohr | is anyone here farmiliar with the state machine that tracks a channels status? |
| 00:16.22 | rue_mohr | for T1 |
| 00:20.07 | robin_sz | has anyone my camel? |
| 00:23.17 | justdave | nomadsoul: yeah, it works fine without any extra hardware if all you're doing is VoIP |
| 00:24.04 | justdave | that's how mine is at home, got a VoIP provider, SIP link inbound/outbound, and a few IP phones on the LAN |
| 00:24.35 | justdave | mind you, an ATA adapter plus a standard analog phone is probably cheaper than a good IP phone |
| 00:26.19 | robin_sz | cheaper, but 0.25 of the features |
| 00:26.29 | robin_sz | well 0.25 in an easily accessible way |
| 00:28.31 | robin_sz | ATAs are good for: connecting DECT phones to and connecting extension bells to |
| 00:30.50 | justdave | I have an FXS/FXO card now, haven't hooked it up yet |
| 00:31.01 | justdave | got it as a handout when I went to the Asterisk Bootcamp class |
| 00:31.22 | justdave | (work paid for that, never be able to afford it otherwise :) |
| 00:42.01 | nomadsoul | justdave: and do you know about any voip&sip provider that just let you call pc to pc for free (just to do some testing) |
| 00:42.35 | justdave | not offhand |
| 00:42.53 | nomadsoul | mmm |
| 00:43.06 | nomadsoul | i'm going bed now |
| 00:43.16 | nomadsoul | i think i'll come back again later to talk :P |
| 00:43.36 | nomadsoul | bye |
| 00:45.59 | boch | do you know if is it possible to decrease the gain of a ulaw file with sox ? |
| 00:47.20 | iCEBrkr | man sox |
| 00:47.22 | iCEBrkr | har har har |
| 00:51.26 | boch | Do not understand format type: ulaw |
| 00:51.41 | wunderkin | ul |
| 00:54.36 | robin_sz | ots amazing how many people seem to leave their ATA's connected to the world and accessible ... |
| 00:55.01 | robin_sz | is it illegal to change peoples callerid strigns to the names of cartoon characters? |
| 00:55.10 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.140.15) |
| 00:55.16 | jazzanova | can someone recommend a SIP provider in Canada ? |
| 00:55.59 | JT | robin_sz: is it illegal to tamper with the configuration of someone else's computer technology without permission? in a lot of countries, yes |
| 00:56.51 | robin_sz | oopsie :) |
| 00:59.55 | robin_sz | I thought I;d search out some settings hints for this Sipura ATA, so I pasted a few words fromt he config screen into google .. and instead of finding settings hints ,it seems to have found about 5000 fools with their ATA open to the world ... |
| 00:59.57 | *** join/#asterisk salviadud (n=dude@189.156.174.25) |
| 01:00.24 | salviadud | i'm stuck with a background |
| 01:00.46 | DocHolliday | iCEBrkr, do you 'broker' voip services? |
| 01:00.53 | DocHolliday | or is that 'brkr' for breaker? |
| 01:01.00 | salviadud | what would you guys do if you needed a welcome message, then inside that menu |
| 01:01.10 | iCEBrkr | I break things |
| 01:01.12 | salviadud | the user dials a really long number |
| 01:01.22 | salviadud | i can't get pattern matching on backbround :( |
| 01:01.42 | salviadud | what's a better way? |
| 01:01.52 | salviadud | oh, i mean, background |
| 01:02.08 | salviadud | well, if anybody could steer me right |
| 01:02.33 | salviadud | can 1.4 do pattern matching with the background application? |
| 01:02.42 | salviadud | i'm using 1.2... so |
| 01:02.55 | *** join/#asterisk znoG (n=gs@235-180-235-201.fibertel.com.ar) |
| 01:10.27 | *** join/#asterisk Mad|Cow (n=thirt@74.92.109.205) |
| 01:12.01 | Mad|Cow | Ok... I know this isnt a asterisk question.... but its related ;-). Atftpd timeouts out after transmitting 512 bytes. I cant transfer any files over 512 bytes without a timeout which is makeing it difficult to tftp boot my phones. Anyone have any ideas? |
| 01:16.10 | jazzanova | sounds like atftpd problem |
| 01:17.05 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 01:17.22 | Mad|Cow | jazzanova: Thanks for your insight.... that hadnt dawned on me ;-) |
| 01:18.34 | jazzanova | can someone recommend a SIP provider in Canada ? |
| 01:19.05 | jazzanova | madcow: have you tried it with a computer, and not a phone ? |
| 01:19.59 | jazzanova | pxe boot |
| 01:20.30 | Mad|Cow | jazzanova: yeah... it times out after 512 bytes... thats how I found it |
| 01:21.05 | jazzanova | try another ftpd server |
| 01:21.06 | cspot | jazzanova: i hear les.net mentioned alot |
| 01:22.17 | cspot | jazzanova: allo.com is another canadain provider |
| 01:22.21 | jazzanova | ok, thanks |
| 01:25.23 | *** join/#asterisk anthm][ (n=anthm@m810f36d0.tmodns.net) |
| 01:27.16 | LeddyHM | voicemeup is as well |
| 01:27.21 | jazzanova | thanks |
| 01:27.28 | jazzanova | they all work well with asterisk |
| 01:27.29 | jazzanova | ? |
| 01:27.33 | JT | yeah but who uses voip providers who advertise on irc |
| 01:27.48 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:27.53 | LeddyHM | free advertising :) |
| 01:28.06 | LeddyHM | for me it was a who could activate an account the fastest |
| 01:28.17 | LeddyHM | voicemeup to the rescue ;) |
| 01:28.49 | JT | LeddyHM: they advertise with an irc nick, that's unprofessional |
| 01:29.43 | LeddyHM | not as bad as a flyby spammer |
| 01:30.43 | JT | sure |
| 01:31.05 | LeddyHM | It doesn't bother me |
| 01:31.19 | LeddyHM | I needed a specific service, and they were here to help |
| 01:31.38 | JT | it wouldn't bother me so much if the advice given by the person with that nick wasn't so erroneous and outrageous |
| 01:31.50 | JT | it's also a lame name for an itsp |
| 01:32.07 | jazzanova | i need to make out-going in Vancouver, BC. |
| 01:32.21 | jazzanova | lots in parallel. |
| 01:32.40 | jazzanova | LeddyHM: how many channels dose voicemeup give ? |
| 01:32.42 | LeddyHM | technically, it's voip |
| 01:32.48 | LeddyHM | so who cares where it is ;) |
| 01:32.58 | Strom_M | um |
| 01:32.59 | Strom_M | latency |
| 01:33.00 | jazzanova | ehh.. well, no. |
| 01:33.02 | Strom_M | duh :) |
| 01:33.07 | jazzanova | it needs to go to a local phone in vancouver. |
| 01:33.08 | JT | latency, DIDs, etc |
| 01:33.13 | JT | rates |
| 01:33.14 | LeddyHM | get a faster connection |
| 01:33.19 | LeddyHM | he mentioned outbound |
| 01:33.20 | JT | location matters of course |
| 01:33.22 | JT | yes |
| 01:33.24 | *** join/#asterisk kn0x (n=atlantic@c-67-176-194-29.hsd1.il.comcast.net) |
| 01:33.30 | JT | so the location of the voip provider matters |
| 01:33.35 | LeddyHM | maybe I should just shutup |
| 01:33.36 | Strom_M | LeddyHM: a fast connection doesn't make another server physically closer |
| 01:33.38 | jazzanova | i want to play a message to 6000 numbers in vancouver. |
| 01:33.41 | Strom_M | speed != latency |
| 01:33.54 | Strom_M | jazzanova: you should just go burn in hell now. |
| 01:34.02 | Strom_M | phone spam == bad |
| 01:34.07 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 01:34.12 | jazzanova | strom: its not spam, its my customers. |
| 01:34.28 | JT | i'm sure they love automated recorded messages calling them up |
| 01:34.40 | LeddyHM | I hate that crap |
| 01:34.49 | LeddyHM | even if I was a customer of said company |
| 01:35.04 | JT | yeah |
| 01:35.12 | Strom_M | shit like that would make me take my business elsewhere |
| 01:35.21 | kn0x | <PROTECTED> |
| 01:35.22 | LeddyHM | yup |
| 01:35.42 | kn0x | ds3 = 672 calls |
| 01:35.45 | JT | kn0x: wrong channel? |
| 01:36.09 | kn0x | hmm, i dont get much a response in #openser generally JT |
| 01:36.18 | kn0x | i was hoping some SER people would be here |
| 01:36.23 | JT | you just need to go when people are active |
| 01:36.28 | kn0x | :X |
| 01:36.53 | kn0x | well i would imagine the dimensioning would be similar- what about an asterisk setup... |
| 01:37.03 | kn0x | no codec conversion just SIP passthrough |
| 01:37.12 | kn0x | SIP router implementation |
| 01:37.26 | JT | dimensioning would be similar to what? |
| 01:37.37 | JT | asterisk is a B2BUA not a proxy |
| 01:38.37 | kn0x | hmm |
| 01:39.46 | kn0x | JT- well can't OpenSER function as a B2BUA, how else do i manage my billing? |
| 01:40.36 | JT | it has billing modules but it is NOT a B2BUA |
| 01:40.43 | *** join/#asterisk ohadz (n=ohad@cpe-69-203-27-50.nyc.res.rr.com) |
| 01:41.29 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 01:42.31 | ohadz | anyone have experience installing on ubuntu 7.04? |
| 01:43.05 | kn0x | JT, I think i'm confused at the difference between a SIP proxy and a B2BUA- does a SIP proxy just hand out INVITES and A B2BUA hold the callstream through the duration? |
| 01:44.03 | kn0x | ohadz, use subversion.... |
| 01:44.05 | ohadz | is there any other good/cheap voip provider beside nufone? |
| 01:44.16 | kn0x | ohadz, for prepaid>? |
| 01:44.28 | kn0x | ohadz, i use vitelity.net for my prepaid- they are pretty good |
| 01:44.36 | ohadz | kn0x, most of the docs are talking about cvs.. |
| 01:45.22 | kn0x | i dont think digium maintains any cvs anymore |
| 01:45.51 | justdave | yeah, it's all svn now |
| 01:45.55 | kn0x | ohadz, svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk |
| 01:46.26 | kn0x | or asterisk-1.4 for the release version i believe |
| 01:46.31 | JT | kn0x: a B2BUA establishes different sip connections for different legs of the calls |
| 01:46.39 | JT | and remains in the signalling and often media patch |
| 01:47.11 | kn0x | JT- so what does a proxy do different? |
| 01:47.49 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 01:47.49 | JT | it simply passes through SIP messages to elsewhere based on predefined logic, usually doesn't touch media but there can be addons for media |
| 01:48.06 | JT | it doesn't play prompts or act as ivr |
| 01:48.38 | kn0x | JT- so how does it stay in the signalling ? what prevents the UA from not talking back to the end point? |
| 01:49.39 | kn0x | because I don't see how my SIP proxy could maintain usage records if it isn't informed by either side of the call... |
| 01:49.44 | kn0x | see what I'm saying? |
| 01:50.42 | JT | kn0x: a sip proxy can stay in the signalling path |
| 01:50.44 | JT | anyway |
| 01:50.54 | JT | sounds like this is outside your league |
| 01:51.03 | JT | you should consider hireing a consultant |
| 01:51.13 | kn0x | i have... im just trying to figure out |
| 01:51.25 | kn0x | their not good at communicating with me |
| 01:51.46 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:51.53 | kn0x | you say it 'can' stay in the signalling path, what is forcing it? |
| 01:52.19 | JT | it proxies |
| 01:52.20 | kn0x | why prevents the call from being 'reinvited' to the destination..... |
| 01:52.22 | JT | that's what it does |
| 01:52.26 | JT | eh |
| 01:52.32 | JT | reinvites are for media |
| 01:52.34 | JT | not signalling |
| 01:52.58 | kn0x | i see i see now |
| 01:53.11 | JT | media uses RTP, not SIP |
| 01:54.19 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 01:54.23 | kn0x | right right.... |
| 01:55.49 | ghento | Hi everyone. I have a question about the MP3Player() function - the docs say a user can stop playback by pressing any digit - does anyone know i there a way to catch this digit and store it? |
| 02:00.04 | salviadud | i got pattern matching to work on background :) |
| 02:00.16 | salviadud | i got this question, hear me out |
| 02:00.32 | salviadud | suppose i create a meeting, then by some chance, y got loads of wav files from al pacino |
| 02:00.39 | salviadud | and i create this little flash app |
| 02:00.42 | salviadud | that talks in xml |
| 02:00.53 | salviadud | and when i press a button, or say a wav file |
| 02:01.08 | salviadud | i inject al pacino to the conversation, with a simple dial and playback |
| 02:01.19 | salviadud | performance wise |
| 02:01.31 | salviadud | should i translate the wav files to an appropiate codec? |
| 02:01.42 | salviadud | would asterisk 1.4 take the beating? |
| 02:02.32 | salviadud | |
| 02:03.29 | salviadud | well, everybody's having dinner i guess |
| 02:03.42 | salviadud | peace out |
| 02:04.01 | ohadz | kn0x, now what?:) is the zaptel included in the trunk? |
| 02:07.04 | *** join/#asterisk kiwoneka (n=kiwoneka@KTNRON06-1168103823.sdsl.bell.ca) |
| 02:08.58 | *** join/#asterisk flashnet (n=flashnet@211.223.75.49) |
| 02:09.49 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 02:10.27 | justdave | zaptel is a separate module |
| 02:10.32 | justdave | in svn |
| 02:10.48 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 02:10.55 | justdave | I think you can swap zaptel for asterisk in the svn path |
| 02:11.12 | justdave | ohadz: ^^^ |
| 02:12.55 | ohadz | justdave, ^j^ |
| 02:14.06 | ohadz | justdave, like this -- svn checkout http://svn.digium.com/svn/zaptel/trunk zaptel ? |
| 02:14.29 | justdave | yeah, I think. going from memory here, which isn't that good :) |
| 02:16.06 | justdave | yeah, that looks correct, looking at my local svn checkout |
| 02:16.28 | justdave | (which is /svn/zaptel/branches/1.2) |
| 02:16.59 | justdave | [root@asterisk zaptel]# svn info |
| 02:16.59 | justdave | Path: . |
| 02:16.59 | justdave | URL: http://svn.digium.com/svn/zaptel/branches/1.2 |
| 02:17.00 | justdave | Repository Root: http://svn.digium.com/svn/zaptel |
| 02:19.28 | *** join/#asterisk rbd (n=rbd@adsl-074-229-183-112.sip.rmo.bellsouth.net) |
| 02:20.12 | rbd | hi guys, if I transfer a SIP call around to otherwise non-linked (e.g. non-trunked) asterisk boxes, is the call ID preserved or is a new call ID created for each leg....I guess this is more of a SIP question |
| 02:20.44 | rbd | leg's say box A transfers the call to box B, and box B sends it to box C, box C sends that same call to box A |
| 02:20.59 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 02:25.02 | kiwoneka | hello to all |
| 02:25.27 | kiwoneka | i need some quick callid help |
| 02:25.46 | kiwoneka | i just need call id to pass as is right to my polycoms |
| 02:26.10 | Strom_M | ~ask |
| 02:26.22 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 02:26.36 | justdave | what channel type is your incoming trunk? |
| 02:26.53 | ohadz | justdave, what's next? where do i find docs on how to actually set * up? |
| 02:27.03 | kiwoneka | i apologize |
| 02:27.12 | ohadz | i now i have the trunk of both * and zaptel on my machine.. |
| 02:27.17 | justdave | typically on most channels you can do something like "callerid=asreceived" on the inbound definition |
| 02:27.32 | justdave | ohadz: see the README files in each |
| 02:27.48 | kiwoneka | my provider is unlimitel, call id works perfectly on my zap channels |
| 02:28.20 | justdave | you zap is inbound or phones? |
| 02:28.31 | kiwoneka | inbound |
| 02:29.01 | justdave | and your unlimitel connection is sip? |
| 02:29.08 | kiwoneka | yes |
| 02:30.17 | justdave | do you have a callerid= line in your inbound definition in sip.conf? |
| 02:31.27 | rbd | say I have asterisk server A, and asterisk server B. B transfers a call to A...does a peer entry for B need to be in A's sip.conf, or is there a way to avoid this (possibly by allowing all from B's subnet or something)? |
| 02:31.33 | justdave | At this point I would suspect you do, and what's listed there is what calls coming in on that line show up as on the phone, right? |
| 02:32.37 | kiwoneka | i dont have registration on any of my inbound |
| 02:33.15 | justdave | rbd: usually if both servers are behind firewalls (which is how it's often done, since most of those situations are multiple offices), then you do canreinvite=no on the links, and asterisk just forwards it along (so there'll be three legs to the call, one from phone A to server A, one from server A to server B, one from server B to phone B) |
| 02:33.41 | justdave | if you want to allow reinvites, then the phones need to be able to see each other on the net |
| 02:33.54 | rbd | justdave: in this case, both servers only serve up IVR (AGI scripts and meetme conferences), there are no attached phones |
| 02:34.03 | kiwoneka | here is my inbound in my sip.conf http://pastebin.ca/476436 |
| 02:34.35 | rbd | justdave: so an incoming SIP call from a SIP trunk hits server A, A has an IVR frontend, and transfers to server B's meetme conference |
| 02:34.44 | *** join/#asterisk yxa (n=lonari@58.185.90.101) |
| 02:35.48 | justdave | yeah, then both servers need to know about the trunk, and the trunk (on the other end) also needs to support reinvites |
| 02:36.22 | justdave | if your trunk provider doesn't support reinvites then you're still going to have to forward the traffic via the IVR server until the call ends |
| 02:36.57 | justdave | kiwoneka: yeah, I don't see a callerid override in there. Does your SIP provider offer callerid? |
| 02:37.22 | justdave | or are you getting number only but not names? |
| 02:37.39 | justdave | there's a lot of sip providers out there that only provide numeric callerID, and don't pass the names |
| 02:38.26 | kiwoneka | justdave: yes |
| 02:38.41 | JT | all this talk of "sip" and "trunks" is making me feel unwell ;) |
| 02:42.08 | rbd | justdave: ok, thanks. so it sounds like the trunk needs to be in each box's sip.conf as a provider or a peer? |
| 02:43.25 | justdave | JT must be using the definition of a single connection carrying multiple channels, since sip can't do that :) |
| 02:43.57 | JT | yes and it's also a connection oriented protocol |
| 02:44.00 | justdave | on the closest thing sip has to a trunk, each call gets its own connection |
| 02:44.23 | justdave | PSTN gateway is probably what he really means |
| 02:44.45 | rbd | yeah by sip trunk I mean a pipe with multiple sip calls on it.... yeah that's a mix-in of some circuit-switched terminology |
| 02:45.06 | rbd | yeah it's from a PSTN GW up the line somewhere |
| 02:45.37 | ohadz | do i first install * and then zaptel? i am only using my network card.. i don't have a pri card or fxo cards.. |
| 02:45.40 | rbd | but yeah, to us it's just a packet-oriented connection |
| 02:46.18 | JT | ohadz: you probably don't need zaptel unless you need zap timing for something like meetme, moh or iax2 trunking |
| 02:46.26 | JT | ohadz: zaptel installs before asterisk |
| 02:46.34 | JT | otherwise asterisk is not aware of it |
| 02:46.56 | NormanAthol | just remember to modprobe ztdummy when you are finished to load it |
| 02:48.44 | ohadz | JT, so i won't need zaptel if i'm just using my machine as is (without any special digium cards) and with a sip phone? |
| 02:49.19 | JT | ohadz: yes and provided you're not doing one of those things i mentioned |
| 02:49.42 | ohadz | JT, ok. thanks. |
| 02:50.01 | *** join/#asterisk rbd (n=rbd@adsl-074-229-183-112.sip.rmo.bellsouth.net) |
| 02:50.46 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 02:54.17 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 02:57.52 | *** join/#asterisk _mike3_ (i=niter3@tweakin.com.ar) |
| 02:58.12 | _mike3_ | Is the X100P a test card or a card that can be used for everyday use? |
| 02:58.21 | JT | it's a toy |
| 02:59.19 | *** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
| 03:01.52 | _mike3_ | JT, which means what? |
| 03:02.04 | JT | don't use it if you can avoid it |
| 03:02.07 | JT | ~x100p |
| 03:02.09 | jbot | from memory, x100p is an obsolete card. You don't want to bother trying to make it (or any of the "digium compatible" clones) work. Get a TDM01B, and you will save your sanity, your hair, and countless other things. |
| 03:02.11 | _mike3_ | I don't need this for a company. Just for home use that I would like to use all the time. However, I need quaility.. |
| 03:02.50 | JT | well it's definitely not quality, they're no longer produced, so all the ones being sold now are quite crap |
| 03:03.39 | _mike3_ | got'cha. I need a low profile card though jbot |
| 03:03.47 | _mike3_ | I got a slim PC here. |
| 03:04.02 | JT | too hard basket |
| 03:04.07 | JT | get an ATA instead |
| 03:04.49 | _mike3_ | I have a Linksys ATA. |
| 03:04.49 | _mike3_ | will that work? |
| 03:04.50 | _mike3_ | unlocked linksys ata that is |
| 03:04.53 | ohadz | now that i've installed * i want to setup my sip phone... i have installed the samples.. do i need to move the extensions.conf.sample to extensions.conf? |
| 03:04.54 | _mike3_ | I was using it with Asterisk once before.. |
| 03:05.01 | JT | sure, as long as it has an FXO port |
| 03:05.13 | *** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 03:05.13 | JT | then why wouldn't it work now? :) |
| 03:05.26 | _mike3_ | hrm.. How do I configure it under Asterisk then? |
| 03:05.36 | _mike3_ | I don't need Zaptel for it do I |
| 03:05.38 | _mike3_ | ? |
| 03:05.48 | JT | set up an account in sip.conf for it |
| 03:05.49 | JT | nup |
| 03:06.06 | _mike3_ | oh ok |
| 03:06.33 | _mike3_ | so all inbound calls will go to this then I make my routes to where I want. Which section of the extensions.conf. So auto greeting and from there to where I want |
| 03:06.35 | _mike3_ | I take it? |
| 03:06.53 | JT | ~thebook |
| 03:06.55 | jbot | hmm... thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 03:07.14 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.138.229) |
| 03:07.37 | ohadz | anyone? |
| 03:09.17 | *** join/#asterisk morgala (n=blah@ppp134-56.lns3.mel6.internode.on.net) |
| 03:09.37 | morgala | hey everyone... |
| 03:10.15 | morgala | i am a little stuck... are there any AGI gurus out there? |
| 03:10.46 | JunK-Y | morgala: ask a specific question will be a good start. |
| 03:10.56 | JunK-Y | ~agi api |
| 03:10.58 | jbot | [agi api] at http://home.cogeco.ca/~camstuff/agi.html |
| 03:14.16 | morgala | ok... is there a way to delete files from within agi? |
| 03:15.05 | morgala | php is deleting files before they have finished being played with "control stream file" if i paus in the middle of playback... |
| 03:15.15 | morgala | sorry pause |
| 03:15.19 | *** join/#asterisk mattwj2005 (n=Matt@user-38q4155.cable.mindspring.com) |
| 03:15.27 | mattwj2005 | hi guys |
| 03:15.49 | mattwj2005 | I live in the US and I am thinking about buying an unlocked phone |
| 03:16.59 | mattwj2005 | do they work well for prepaid? |
| 03:18.15 | ohadz | i tried to define my sip phone both in sip.conf.samples and also in extensions.conf.samples -- i am still getting this error -- chan_sip.c:15479 handle_request_register: Registration from '"Hello Worldz" <sip:888@10.1.7.17>' failed for '10.1.7.167' - No matching peer found |
| 03:18.38 | JunK-Y | morgala: i dont understand, since control stream file is not deleting anything. |
| 03:19.51 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 03:20.39 | *** join/#asterisk Penggu (i=foobar@220-245-200-87.static.tpgi.com.au) |
| 03:20.40 | morgala | i do an unlink from php after doing control stream file |
| 03:21.16 | morgala | not really sure how to make the unlink happen only after stream file has finished |
| 03:22.26 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.169.195) |
| 03:23.01 | morgala | hey mattyj2005 can you pm me your config? |
| 03:23.13 | morgala | sorry... i mean ohadz |
| 03:25.43 | ohadz | morgala, which one? |
| 03:28.22 | *** join/#asterisk awannabe (n=brad@207-114-155-213.static.twtelecom.net) |
| 03:29.14 | awannabe | hey guys, anyway to place a call and not have to accept the invite? im having some problems with AgentCallBackLogin() and getting invite erros |
| 03:30.37 | ghento | Is it much less resource intensive to use Playback() with .wav files, over MP3Player() ? |
| 03:33.53 | kn0x | ghento, i dont know how much, but i'd imagine it is considering mp3 is highly compressed and .wav is usually uncompressed (as far as i know) |
| 03:34.41 | ghento | kn0x: thanks, good point |
| 03:34.57 | kn0x | no problem. |
| 03:35.52 | awannabe | maybe its these damn snoms |
| 03:35.54 | kn0x | optimally, you'd use a uniform codec.... |
| 03:36.10 | kn0x | awannabe, i was about to buy a snom, whats wrong with them? |
| 03:36.20 | awannabe | well, they do have their issues... |
| 03:36.35 | kn0x | hmmm.... interesting |
| 03:36.35 | JT | kn0x: where abouts are you located? |
| 03:36.48 | kn0x | JT, chicago, why do you ask? |
| 03:36.54 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:37.02 | JT | usa, right, no idea why you'd buy a snom then :) |
| 03:37.30 | kn0x | JT, i've heard great things about them... |
| 03:37.52 | kn0x | so what issues are people having? |
| 03:37.59 | ohadz | also.. is there a good voip provider that will allow me to setup an account/ DID and voip immediately ? |
| 03:38.07 | JT | they're just not that great for the price |
| 03:38.10 | JT | ~phones |
| 03:38.12 | jbot | i heard phones is http://bani.anime.net/phones/. SIP Hardphones in order of quality/suggestability : Polycom (any), Aastra 480i, Cisco 7940(+), Linksys SPA-9XX, Snom |
| 03:38.21 | JT | polycom is by far the most recommended phone here :) |
| 03:38.40 | kn0x | ohadz, what type of service: unlimited, prepaid, etc. ? |
| 03:39.00 | kn0x | jbot- oh thank you for spamming -_- |
| 03:39.11 | JT | spamming? |
| 03:39.41 | ohadz | kn0x, i guess prepaid.. |
| 03:39.42 | kn0x | well that looks like an attempt to get me to goto that website and buy a phone... |
| 03:40.04 | ohadz | what? |
| 03:40.08 | kn0x | ohadz, i use vitelity.net they're pretty good for 1.7cents/minute USA |
| 03:40.15 | *** join/#asterisk bbryant (i=Brett@12-214-191-64.client.mchsi.com) |
| 03:40.26 | kn0x | plus they have usa48 dids for unlimited 7.95/month |
| 03:41.03 | awannabe | anyway in * to let in place calls to any sip client without them registering, or making a peer entry? |
| 03:41.07 | kn0x | also voipjet.com has immediate service |
| 03:41.25 | ohadz | kn0x, i looked at their website-- they start at $35/m |
| 03:41.28 | kn0x | i've had reliability issues from them before... |
| 03:41.30 | kn0x | voipjet? |
| 03:41.45 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:42.13 | kn0x | neither have minimum usage reuirements, ohadz |
| 03:42.29 | JT | kn0x: what looks like an attempt you get you onto what website and buy what phone? |
| 03:42.45 | kn0x | JT, my appologies, i was wrong |
| 03:42.46 | JT | s/you get you/to get you/ |
| 03:43.11 | JT | there's no less than 5 brands listed, not sure what web site it'd be :P |
| 03:43.45 | kn0x | no that bot is useful actually |
| 03:44.14 | JT | kn0x: you should look into polycom phones, they're quite decent |
| 03:44.37 | ohadz | thanks morgala for your help. you got me working now. thanks again.. |
| 03:44.50 | _mike3_ | Hey guys.. I need a low profile FXO card. Anyone know of a good quaility card. |
| 03:44.51 | _mike3_ | ? |
| 03:45.06 | kn0x | <PROTECTED> |
| 03:45.14 | awannabe | _mike3_, sangoma or digium only... |
| 03:45.37 | ohadz | kn0x, would they vitallity get my up and running immediately ? |
| 03:45.41 | _mike3_ | awannabe, I know which manfacture to go with...... Just need a low profile card |
| 03:45.53 | awannabe | half height you mean? |
| 03:46.21 | kn0x | ohadz, vitelity.net (formely iax.cc) gave me my acc't immediately |
| 03:46.22 | _mike3_ | yes |
| 03:46.27 | _mike3_ | i have a slim pc here |
| 03:47.15 | awannabe | ahh, ok |
| 03:47.38 | _mike3_ | only thing I can find in a low profile format is the x100P. But I heard these cards are junk. I need something with quaility. |
| 03:47.47 | kn0x | JT, what do they mean by line appearances- registrations? |
| 03:48.07 | kn0x | do they mean how many simultaneous registrations? |
| 03:48.33 | ohadz | kn0x, and you could call immediately.. what about nufone.net? |
| 03:49.14 | JT | kn0x: yeah pretty much |
| 03:49.18 | kn0x | ohadz, yes, i think they are the same..... but im mad at nufone because they lost my number |
| 03:49.32 | JT | the 430 for example has 2 line appearances, but it can do 8 calls at once or something |
| 03:49.49 | kn0x | they like all of a sudden dropped their incomming service for 2 months |
| 03:49.52 | kn0x | last year |
| 03:50.04 | kn0x | JT, ohh ic |
| 03:52.11 | _mike3_ | guess I'll have to buy a FXO adapter. |
| 03:52.35 | _mike3_ | and make an extenion in sip.conf and have it register to it. Everything that comes in will pass to the auto greeting i guess. |
| 03:52.38 | _mike3_ | i'm sure that will work |
| 03:52.59 | JT | _mike3_: i thought you said you already has a sipura ata |
| 03:53.22 | _mike3_ | no I have a linksys. It's only has two FXS ports |
| 03:53.24 | _mike3_ | not FXO |
| 03:53.34 | JT | linksys bought sipura, same thing |
| 03:53.38 | JT | cisco owns linksys too |
| 03:54.05 | ohadz | kn0x, ah.. |
| 03:54.44 | kn0x | ohadz, idk what they've been like recently, but im still upset about that a year later. |
| 03:55.22 | kn0x | ohadz, i was reading that one of the co-owners had some mental stability issues |
| 03:55.32 | ohadz | hmm.. sounds like they are doing good biz |
| 03:55.34 | kn0x | ...thats was the source of the issues |
| 03:55.35 | ohadz | really? |
| 03:55.42 | ohadz | i heard good things though... |
| 03:55.44 | ohadz | who knows.. |
| 03:55.50 | kn0x | asterlink.com |
| 03:55.58 | kn0x | they are by far the most reliable |
| 03:56.08 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 03:57.07 | ohadz | i have * running.. and it registers my phone (thanks morgala) -- but then when i dial 8500 or ext 500 i get a bz signal and the following msgs -- == Using TOS bits 0 == Using CoS mark 0 |
| 04:00.06 | JT | those messages don't look relevant |
| 04:06.54 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 04:07.44 | *** join/#asterisk wolferine (n=profx@unaffiliated/wolferine) |
| 04:08.04 | wolferine | anyone ever use Ekiga with Asterisk ? |
| 04:08.30 | wolferine | im trying to talk between two devices (I have a Grandstream phone here as well) on my LAN |
| 04:09.12 | ohadz | what would explain that i get a bz signal when i try to dial the demo ext (500) ? |
| 04:10.09 | *** join/#asterisk RyanWi (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 04:10.20 | ohadz | my phone registered fine. i get a dialtone but i can't get to 8500, 1000, 500 or any other ext |
| 04:14.22 | JT | ohadz: what sort of phones? |
| 04:15.35 | ohadz | JT, i have only one phone (cisco spa921) but i thought that by having those ext in the ext.conf file i will be able to dial from my phone and here the demo.. |
| 04:15.58 | ohadz | tried to call ext 500 and 8500.. all i get is a busy signal JT |
| 04:16.55 | Dimik | <PROTECTED> |
| 04:17.41 | JT | s/cisco/linksys/ |
| 04:17.55 | JT | that model isn't a cisco one |
| 04:18.17 | ohadz | JT, i guess it's a linksys.. |
| 04:18.24 | ohadz | aren't they the same company ? |
| 04:19.02 | JT | no, cisco owns linksys |
| 04:19.06 | Dimik | linksys is owned by cisco |
| 04:19.08 | JT | linksys still make their own products |
| 04:19.17 | ohadz | right.. still same same |
| 04:19.27 | JT | ohadz: the phone is not a cisco though. |
| 04:19.33 | Dimik | linksys == shit |
| 04:19.33 | ohadz | now linksys has the cisco logos on their products:) |
| 04:19.47 | JT | Dimik: i wouldn't go that far |
| 04:19.52 | ohadz | JT, right. it'a linksys.. still doesn't resolve my issues :) |
| 04:19.54 | Dimik | good for SoHo |
| 04:19.57 | Dimik | and that's about it |
| 04:20.06 | jazzanova | <PROTECTED> |
| 04:20.07 | ohadz | Dimik, that is what i'm using it for:) |
| 04:20.32 | JT | Dimik: some people have good results with linksys phones in businesses |
| 04:21.08 | Dimik | JT, i was talking about the routers |
| 04:21.24 | Dimik | wasn't even aware linksys produced phones |
| 04:21.35 | Math` | whats wrong with their routers anyways |
| 04:22.55 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:23.07 | Dimik | the fact i guess that they're 802.11g |
| 04:23.17 | Dimik | 54mbps is good but the radius is pretty short |
| 04:23.35 | Dimik | also few models you have to activate with their special software which's only windows compatible |
| 04:23.45 | Dimik | and only then you can http to it to configure the actual thing |
| 04:24.08 | Dimik | it's all personal experience, i guess i just hadn't had luck with any that's all |
| 04:24.11 | ohadz | JT, how should i go about resolving this bz signal on all ext's? |
| 04:24.13 | Dimik | they're still good product |
| 04:25.14 | JT | busy, not "bz" :) |
| 04:25.20 | JT | ohadz: find out if the calls reach asterisk |
| 04:25.24 | JT | first of all... |
| 04:25.37 | JT | do they appear on the CLI with the verbosity set at 10 |
| 04:25.45 | Dimik | i need to install asterisk |
| 04:26.18 | JT | Dimik: certain linksys routers are good for using aftermarket firmware on too |
| 04:27.55 | Dimik | yeah i heard that |
| 04:30.58 | *** join/#asterisk c6vette (n=khagan@ip70-176-165-236.ph.ph.cox.net) |
| 04:31.23 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 04:31.28 | *** join/#asterisk ohadz (n=ohad@cpe-69-203-27-50.nyc.res.rr.com) |
| 04:32.00 | ohadz | JT, sorry got disconnected.. :/ so verbosity set to 10 is vvvvvvvvvvvvc? |
| 04:32.29 | JT | yes or set verbose 10 |
| 04:33.46 | dc3aes | hehehe -v^10x5 :P |
| 04:33.52 | *** join/#asterisk dhakatel (n=root@58.65.224.5) |
| 04:34.55 | JT | it actually has an unusual number for ther verbosity limit |
| 04:35.07 | JT | 2147483647 |
| 04:35.14 | JT | is the highest verbosity |
| 04:35.42 | dc3aes | interesting.. when i get my friends' boxes going ill sneak in and set that to their verbosity :) |
| 04:35.47 | ohadz | ok. i've done this -- core set verbose 10 Verbosity is at least 10*CLI> == Using TOS bits 0 == Using CoS mark 0 |
| 04:37.46 | ohadz | this is after i tried to dial 500 |
| 04:38.37 | JT | ohadz: the dial attempt should appear in there |
| 04:38.55 | ohadz | JT, i know but it doesnt.. :/ |
| 04:40.23 | JT | ohadz: check that the phone dialplan is correct |
| 04:40.27 | ohadz | i feel like i missed something during the install.. i svn'd the trunk of * into /usr/src and also zaptel.. then ran ./configure make, make install make samples.. set my sip phone on sip.conf and ext.conf.. that's it.. |
| 04:41.26 | JT | ohadz: no, i didn't say asterisk |
| 04:41.28 | JT | PHONE dialplan |
| 04:43.17 | *** join/#asterisk Cabal (n=Cabal@unaffiliated/cabal) |
| 04:43.29 | ohadz | JT what am i looking for in the my phone's dialplan? |
| 04:43.32 | *** join/#asterisk threat (i=threat@60-240-43-214.static.tpgi.com.au) |
| 04:43.46 | JT | what numbers it will send to the sip peer |
| 04:45.12 | ohadz | JT, you think the problem is with my phone's configuration? |
| 04:45.18 | *** join/#asterisk dudanogueira (n=duda@unaffiliated/bitdoidao/x-000001) |
| 04:45.33 | JT | yes, if nothing comes up in the asterisk cli |
| 04:45.42 | JT | you could always do a sip debug to check for sure |
| 04:47.01 | *** join/#asterisk dacter (n=dlittrel@host83.onr.com) |
| 04:48.39 | ohadz | JT, it registers -- my dialplan on the phones says -- (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) |
| 04:49.08 | JT | and when you dial a number, do any sip messages come in? |
| 04:49.19 | ohadz | nope |
| 04:49.29 | JT | then the dialplan needs fixing |
| 04:49.34 | ohadz | JT, nothing.. |
| 04:49.48 | JT | ? |
| 04:49.56 | ohadz | ok. |
| 04:55.26 | [TK]D-Fender | Why would he get SIP messages... he's only using VERBOSE. |
| 04:55.40 | [TK]D-Fender | And earlier got a blatant registration error. |
| 04:56.07 | [TK]D-Fender | we don't even know if its talking to * at all. |
| 04:56.22 | [TK]D-Fender | first step : SIP debug. |
| 04:56.35 | [TK]D-Fender | (aside from the obvious double-check of sip.conf) |
| 04:56.51 | [TK]D-Fender | ohadz, is your phone and * on the same subnet? |
| 04:58.44 | *** join/#asterisk Axelatino (n=Axel@200.12.40.89) |
| 04:59.19 | JT | [TK]D-Fender: already mentioned sip debug |
| 05:00.26 | [TK]D-Fender | jt : where? paste the line, because I missed it..... |
| 05:00.38 | Hmmhesays | anyone sitting next to a fax machine? |
| 05:00.45 | JT | < JT> you could always do a sip debug to check for sure |
| 05:01.25 | *** join/#asterisk ManxPower (n=manxpowe@133.sub-70-196-58.myvzw.com) |
| 05:01.41 | ohadz | [TK]D-Fender, yes. it also registers fine |
| 05:02.31 | [TK]D-Fender | JT : you never gave him the syntax to do so explicitly, not did I see any feedback to indicate taht he has. |
| 05:02.40 | *** part/#asterisk c6vette (n=khagan@ip70-176-165-236.ph.ph.cox.net) |
| 05:02.47 | [TK]D-Fender | ohadz, so IS you phone on the same subnet as *? |
| 05:03.07 | ohadz | [TK]D-Fender, yes |
| 05:03.21 | Hmmhesays | anyone? |
| 05:03.33 | *** join/#asterisk remmo (n=junk@smack.isp.net.au) |
| 05:03.37 | [TK]D-Fender | ohadz, so do "sip debug". then place a call. pastebin all of the CLI output./ |
| 05:04.09 | *** part/#asterisk ManxPower (n=manxpowe@133.sub-70-196-58.myvzw.com) |
| 05:05.18 | ohadz | [TK]D-Fender, http://pastebin.ca/476554 |
| 05:05.58 | [TK]D-Fender | ohadz, Doesn't look like everything... |
| 05:06.10 | JT | [TK]D-Fender: if he had any doubts what i meant, i'd expect him to ask what i meant |
| 05:06.14 | [TK]D-Fender | ohadz, that looks like the later half of a call. |
| 05:06.41 | [TK]D-Fender | JT : Any maybe the whole pile went over his head and simply said nothing. not a safe assumption :) |
| 05:06.53 | [TK]D-Fender | JT : around here, NONE are :) |
| 05:07.03 | JT | shrug, stay silent at your own peril |
| 05:07.12 | JT | if someone can't be bothered to ask, i can't be bothered to handhold |
| 05:07.21 | JT | depends how much they want their problem solved |
| 05:07.54 | *** part/#asterisk mattwj2005 (n=Matt@user-38q4155.cable.mindspring.com) |
| 05:08.46 | ohadz | [TK]D-Fender, sorry.. check this one -- http://pastebin.ca/476557 |
| 05:09.05 | [TK]D-Fender | SIP/2.0 401 Unauthorized |
| 05:09.07 | [TK]D-Fender | there you have it |
| 05:09.10 | [TK]D-Fender | bad pass |
| 05:09.23 | [TK]D-Fender | # |
| 05:09.23 | [TK]D-Fender | No user '888' in SIP users list |
| 05:09.23 | [TK]D-Fender | # |
| 05:09.23 | [TK]D-Fender | Found peer '888' for '888' from 10.1.7.167:5060 |
| 05:09.44 | Hmmhesays | ok record routing makes a hell of a lot more sense now |
| 05:09.48 | [TK]D-Fender | ohadz, pastebin your entry for that phone |
| 05:09.58 | JT | ohadz: if you didn't know what sip debug was, you should've asked |
| 05:10.24 | ohadz | JT, sorry i did know. i was doing 4 other things.. |
| 05:10.45 | [TK]D-Fender | JT : all he needs is to have THOUGHT he did for your assumption to fail silently :) |
| 05:10.48 | JT | oh ok, i thought you said the output of it showed no calls |
| 05:11.14 | [TK]D-Fender | JT : Got to trace the subtle bits... |
| 05:11.39 | JT | [TK]D-Fender: shrug, i have other stuff to do, like many others here :) |
| 05:11.39 | ohadz | [TK]D-Fender, I gotta hand it to ya. you're good;) |
| 05:11.57 | [TK]D-Fender | ohadz, I TRY.... |
| 05:12.07 | [TK]D-Fender | ohadz, And about that sip.conf entry..... |
| 05:12.27 | ohadz | http://pastebin.ca/476562 |
| 05:12.27 | JT | but if people wish to donate money, i promise i'll pay special attention to their problems :P |
| 05:12.42 | ohadz | :P |
| 05:13.11 | Hmmhesays | haha |
| 05:13.14 | Hmmhesays | sounds about right |
| 05:13.19 | [TK]D-Fender | ohadz, kill the "username=" bit from your phone entries, and make sure your phone's entry REALLY matches |
| 05:13.23 | Hmmhesays | ok what is the syntax for tcpdump |
| 05:13.32 | Hmmhesays | I can't freaking remember this late at night |
| 05:13.46 | ohadz | [TK]D-Fender, i just added that.. i used to not have it there.. |
| 05:14.04 | [TK]D-Fender | ohadz, go double check your phones. |
| 05:14.18 | [TK]D-Fender | ohadz, and pastebi "sip show peers" |
| 05:14.21 | JT | Hmmhesays: syntax to do what? |
| 05:14.31 | [TK]D-Fender | ohadz, I might also suggest you set them to "type=friend" |
| 05:14.33 | ohadz | ok. done. now what? |
| 05:14.51 | Hmmhesays | JT dump a sip trace into a file |
| 05:16.57 | *** join/#asterisk DarKnesS_WolF (n=sherif@unaffiliated/sherif) |
| 05:17.55 | [TK]D-Fender | ohadz, please pastebin "sip show peers".... |
| 05:18.41 | ohadz | ok. reloaded.. restarted... still the same... http://pastebin.ca/476564 |
| 05:19.17 | ohadz | http://pastebin.ca/476566 |
| 05:21.08 | [TK]D-Fender | ohadz, I'm still betting your password isn't right in your phone. |
| 05:21.16 | ohadz | did i forget to install something in the beginning..? am i missing something? |
| 05:21.22 | ohadz | hmm.. .passwd is 888 |
| 05:21.25 | ohadz | let me reset it.. |
| 05:22.56 | ohadz | [TK]D-Fender, just reset it.. still the same...dialplan on the phone maybe? (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) |
| 05:23.11 | [TK]D-Fender | ohadz, PASSWORD <-------------------- |
| 05:23.35 | [TK]D-Fender | ohadz, its not the dialplan. |
| 05:24.14 | [TK]D-Fender | ohadz, we can clearly see its dialing 500, calling *, matching the right user, and being refused. that leaves pass/domain issue on the phone |
| 05:24.18 | ohadz | [TK]D-Fender, i reentered the passwd into the admin webpage |
| 05:24.30 | ohadz | hmmm.. |
| 05:26.17 | *** join/#asterisk Exhar (n=Roy@213-73-139-87.cable.quicknet.nl) |
| 05:28.34 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 05:29.22 | *** part/#asterisk Axelatino (n=Axel@200.12.40.89) |
| 05:29.49 | ohadz | [TK]D-Fender, i tried to change it to 777 .. still.. |
| 05:29.51 | ohadz | http://pastebin.ca/476571 |
| 05:29.53 | ohadz | same thing.. |
| 05:30.10 | ohadz | the user and passwd are def. correct on the phone web interface... |
| 05:30.53 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:30.58 | [TK]D-Fender | ohadz, a blank in there is not filled out right |
| 05:32.53 | ohadz | [TK]D-Fender, in the passwd? |
| 05:33.16 | ohadz | on the phone? |
| 05:33.26 | [TK]D-Fender | ohadz, host a screenshot for someone to help you out, or check out a guide on its setup. |
| 05:33.56 | [TK]D-Fender | But I have to get some sleep now. I've pointed you to where you need to fix this. One step at a time now. |
| 05:33.59 | [TK]D-Fender | later all |
| 05:37.38 | ohadz | I found screen shots -- http://www.voipsyndicate.com/reviews/Linksys_SPA921/index2.html |
| 05:44.37 | *** join/#asterisk SophWise (n=SophWise@72.169.74.29) |
| 05:49.51 | *** join/#asterisk c6vette (n=khagan@ip70-176-165-236.ph.ph.cox.net) |
| 05:50.41 | *** join/#asterisk jacq (n=jal@203.187.143.130) |
| 05:52.47 | *** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au) |
| 05:56.51 | *** join/#asterisk friedrich| (n=friedric@e177243192.adsl.alicedsl.de) |
| 05:58.12 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 05:58.44 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 06:01.18 | *** join/#asterisk grEvenX (n=even@ti500720a080-4984.bb.online.no) |
| 06:11.06 | c6vette | / |
| 06:11.10 | *** part/#asterisk c6vette (n=khagan@ip70-176-165-236.ph.ph.cox.net) |
| 06:16.01 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 06:16.44 | *** join/#asterisk c6vette (n=khagan@ip70-176-165-236.ph.ph.cox.net) |
| 06:16.56 | *** part/#asterisk c6vette (n=khagan@ip70-176-165-236.ph.ph.cox.net) |
| 06:17.43 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 06:23.24 | *** join/#asterisk friedrich| (n=friedric@e177243192.adsl.alicedsl.de) |
| 06:24.04 | *** join/#asterisk Curus (n=Curus@10.8.185.213.dk-amb.res.sta.perspektivbredband.net) |
| 06:27.00 | Keltus | question - when the call is bridged using Dial(), how can I continue to record the call via Monitor()? |
| 06:27.37 | Corydon76-home | What do you mean? |
| 06:27.54 | Corydon76-home | Just because the call is bridged doesn't mean Monitor stops recording |
| 06:30.57 | Keltus | it does for me... |
| 06:32.11 | *** part/#asterisk Here_And_There (n=Here_And@pool-71-244-103-43.phlapa.fios.verizon.net) |
| 06:32.52 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 06:32.56 | *** join/#asterisk ghenry (n=ghenry@212.159.59.85) |
| 06:33.50 | Keltus | this is my script: |
| 06:33.53 | Keltus | [default] |
| 06:33.54 | Keltus | exten => 85,1,Zapateller(nocallerid) |
| 06:33.54 | Keltus | exten => 85,n,Monitor(wav,/home/admin/recording) |
| 06:33.54 | Keltus | exten => 85,n,Background(hello-world) |
| 06:33.54 | Keltus | exten => 85,n,Set(CALLERID(all)=bob <1234567890>) |
| 06:33.54 | Keltus | exten => 85,n,Dial(SIP/1234567890@proxy01.sipphone.com,20,r) |
| 06:33.55 | Keltus | exten => 85,n,Hangup() |
| 06:34.19 | ghenry | cool, my patch got into asterisk core! http://bugs.digium.com/view.php?id=9676 |
| 06:36.21 | Keltus | congrats |
| 06:36.45 | tzafrir | thanks for your contribution |
| 06:45.34 | *** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi) |
| 06:49.46 | *** join/#asterisk yidiyuehan (n=yidiyueh@58.185.253.70) |
| 06:50.11 | yidiyuehan | hi, guys, can any one explain to me how i can set up asterisk server outside of a NAT? |
| 06:50.48 | yidiyuehan | i am a bit confused with NAT issue as i think there is no way to put server outside NAT no matter dynamic or staic ip you are using |
| 06:50.51 | *** join/#asterisk Putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 06:52.42 | *** join/#asterisk ptblank (n=MURDER1@cpe-76-173-168-178.socal.res.rr.com) |
| 06:57.42 | dhakatel | can anybody help me how to port g729 in twinkle of linphone |
| 06:57.54 | dhakatel | *or |
| 06:58.12 | *** join/#asterisk FlatFoot (n=simon@80.88.192.83) |
| 06:58.37 | JT | yidiyuehan: what do you mean? |
| 06:59.19 | JT | kaldemar: is canreinvite=yes set in sip.conf? |
| 06:59.23 | JT | Keltus: i mean |
| 06:59.34 | JT | Keltus: is canreinvite=yes set? |
| 07:00.20 | yidiyuehan | JT, i mean, for my case, i have the ADSL router speedtouch st 585 and the local ip address is natted to public static ip address for server, in this case, is that a NAT in between? |
| 07:00.38 | yidiyuehan | as all the ports are forwarded by default |
| 07:01.01 | JT | can you restate that? i couldn't understand it apart from you have a speedtouch modem |
| 07:01.34 | yidiyuehan | well, like this, i have a ADSL speedtouch modem + router, |
| 07:03.22 | yidiyuehan | and my server has a local ip address within the network and it has been mapped to a public static ip address as well. |
| 07:03.23 | Keltus | JT: not sure, it's the default |
| 07:03.23 | Keltus | what does that option do? |
| 07:03.23 | kaldemar | JT: hello, hello. |
| 07:03.23 | JT | Keltus: set the media reinvites |
| 07:03.24 | Keltus | didn't think an option named that would have an effect on Monitor() |
| 07:03.27 | Keltus | but okay, I'll try |
| 07:03.38 | JT | Keltus: yes, it definitely have an effect |
| 07:03.42 | JT | make sure it is set to no |
| 07:03.44 | JT | not yes |
| 07:03.59 | yidiyuehan | as i have 16 public static ip addresses signed from my Telecom, and in this case is there a NAT in between? |
| 07:04.04 | JT | asterisk needs to see the media to record it |
| 07:04.32 | JT | yidiyuehan: if you've got a box acting as a router, then there's no NAT, if it's acting as NAT, then there is |
| 07:05.11 | yidiyuehan | JT, thanks but i still don't get it. |
| 07:05.33 | Keltus | JT: the option is not set. I will set it to no |
| 07:05.57 | yidiyuehan | typical setting will be, ADSL router with dynamic ip address, and some sip and rtp ports are forwarded, in this case there is a NAT right? |
| 07:06.24 | JT | yidiyuehan: port forwarding is different to NAT |
| 07:07.02 | JT | a device performing NAT or routing can be set to perform port forwarding too |
| 07:07.15 | JT | port forwarding is a very simple translation |
| 07:07.31 | JT | "if any packets come in for this port on this ip, send it to that port on that ip" |
| 07:09.10 | yidiyuehan | isn't it the meaning for NAT? I mean, packets are sent to dynamic ip over this port, and are forwarded to internal ip over another port |
| 07:09.20 | JT | no |
| 07:09.27 | JT | nat is dynamic automagic |
| 07:09.45 | JT | ports are chosen at random, and it must remember what connects to what |
| 07:09.59 | JT | you should read up on basic networking |
| 07:11.48 | Keltus | JT: that was it! thanks |
| 07:11.48 | yidiyuehan | okie ;-) i think so, i am reading some now. |
| 07:11.51 | yidiyuehan | thanks man ;-) |
| 07:12.31 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 07:15.39 | JT | Keltus: np |
| 07:15.40 | JT | Keltus: if the end points transmit media directly to each other, asterisk is not able to record it, that's why the media needs to go through asterisk there |
| 07:16.28 | Keltus | oh, so wait. doesn't the call quality suffer? |
| 07:16.55 | JT | yidiyuehan: the most important thing with NAT is that a computer on the Internet cannot ESTABLISH a connection to a computer behind NAT on a private LAN, in that case you must eiher port forward or the connection must be established from a computer inside the lan |
| 07:17.04 | JT | Keltus: not generally |
| 07:17.15 | JT | Keltus: unless asterisk is very far away from endpoints |
| 07:17.18 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 07:17.29 | JT | or you're trying to push hundreds or more of calls through the machine |
| 07:17.36 | *** join/#asterisk jacq (n=jal@203.187.143.130) |
| 07:17.37 | Keltus | gotcha |
| 07:17.57 | Keltus | actually |
| 07:18.02 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 07:18.04 | Keltus | I mean, it *is* one extra hop |
| 07:18.18 | JT | Keltus: are the points far apart? |
| 07:20.41 | Keltus | one is going to be anywhere in the US (customers), and the other will be erm, india or something |
| 07:21.07 | Keltus | the asterisk server is a dedicated server in illinois |
| 07:21.12 | JT | ok |
| 07:21.17 | *** join/#asterisk adorah (n=user@89-138-65-225.bb.netvision.net.il) |
| 07:21.21 | JT | in some cases it can improve call quality |
| 07:21.26 | JT | ifm asterisk is in a good location |
| 07:21.27 | Keltus | oh really? |
| 07:21.32 | Keltus | how would it do that |
| 07:21.44 | Keltus | ie. have a good connection to both ends? |
| 07:22.02 | *** join/#asterisk Ifaistos (n=stelios@ipa226.211.tellas.gr) |
| 07:22.59 | JT | if it's on good bandwidth |
| 07:23.02 | JT | also, you can't let 2 endpoints behind NAT reinvite media, it won't work |
| 07:23.14 | Keltus | oh and I guess it sort of hurts that right now the testcalls go from customer -> toll free resporg -> ipkall -> asterisk server |
| 07:23.41 | Keltus | so I'm going to change the toll free provider so that we can do customer -> toll free resporg -> asterisk server |
| 07:23.48 | Keltus | hoping it will improve call quality |
| 07:23.52 | Keltus | right now it's just barely acceptable |
| 07:25.18 | jacq | Keltus: your cusomter will be in india? |
| 07:26.01 | JT | ok |
| 07:26.47 | *** join/#asterisk Polis_ttt (n=your@194-237-172-225-no48.business.telia.com) |
| 07:27.33 | Keltus | jacq: no, the people that pick up calls |
| 07:27.40 | Keltus | half of them are there, half in canada |
| 07:28.08 | e-ddie | so, do any of them speak english? |
| 07:29.58 | Keltus | does it matter? |
| 07:31.56 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.169.195) |
| 07:33.37 | *** join/#asterisk lorinc (n=ang@pool-4344.adsl.interware.hu) |
| 07:34.13 | e-ddie | pretty much |
| 07:34.22 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 07:35.48 | *** join/#asterisk arcanine (n=arcanine@203.82.44.181) |
| 07:36.01 | MrWup | anyone know a method of keeping channel variables in the h extension when a channel goes zombie? cause atm they are all flushed |
| 07:36.08 | MrWup | before the h extension code executes |
| 07:45.49 | *** join/#asterisk simplexio (n=sim@ogw.oamk.fi) |
| 07:48.03 | *** join/#asterisk awk (n=bongo@dsl-242-80-16.telkomadsl.co.za) |
| 07:48.05 | arcanine | is there a method for call timer |
| 07:49.21 | awk | hi does anyone use Bankia voip billing? if so what is the cost of the product I can't see prices on their site. also if not does anyone know of a free multi user level billing system? |
| 07:49.27 | MrWup | is there any way i can edit and recompile only channel.c and install it? |
| 07:49.51 | arcanine | for billing purposes, once a call established the timer starts |
| 07:52.45 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 07:53.49 | awk | blah, does nobody use billing software here, everytime i ask i get no reply? |
| 07:53.59 | awk | is there a better channel I should be asking this in? |
| 07:54.16 | *** join/#asterisk zogulus (n=zogulus@58.98.adsl.brightview.com) |
| 07:54.18 | awk | please can i just get some head way instead of staring at this monitor all day awaiting some reply |
| 07:56.37 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:00.45 | jacq | MrWup: tried make all in channels folder? |
| 08:01.22 | MrWup | the bit of code i need to reconfigure is in channel.c in main |
| 08:01.40 | MrWup | im trying to stop the channel variables being flushed before the hangup event is executed when a channel goes zombie |
| 08:01.50 | MrWup | so i can at least access the variables of the channel which is about to die |
| 08:01.54 | MrWup | and do something useful |
| 08:03.53 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 08:05.47 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 08:06.26 | *** join/#asterisk abasic (n=alen@89-172-54-206.adsl.net.t-com.hr) |
| 08:06.38 | abasic | hello |
| 08:06.59 | abasic | i have problems with cdr and oracle db |
| 08:07.34 | abasic | msg: cdr_odbc: Error in Query -1 |
| 08:07.47 | abasic | asterisk realtime is working |
| 08:07.49 | *** join/#asterisk Dibbler_ (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com) |
| 08:08.46 | abasic | anybody |
| 08:09.07 | *** join/#asterisk Dibbler_ (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com) |
| 08:09.15 | awk | stfu biaaaaach, i've been waiting for my question to be answered around an hour earlier than yours |
| 08:10.07 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 08:10.46 | abasic | fya |
| 08:11.12 | awk | ;_; |
| 08:11.25 | *** join/#asterisk Mavvie (n=edwin@ppp161-18.lns4.syd6.internode.on.net) |
| 08:25.55 | *** join/#asterisk CVirus (n=GoD@196.205.192.117) |
| 08:26.18 | CVirus | A PRI/BRI line is connected to a WAN Interface in the device ... correct ? |
| 08:26.34 | JT | what? |
| 08:26.38 | awk | hello joe |
| 08:26.38 | DarKnesS_WolF | CVirus: ?? |
| 08:26.43 | DarKnesS_WolF | WAN !? |
| 08:26.49 | DarKnesS_WolF | CVirus: WAN ya 2fel ! |
| 08:27.13 | DarKnesS_WolF | CVirus: it go to zaptel card " Digital one " |
| 08:27.38 | CVirus | You'll need either Digium or Sangoma (or Cronyx) or OpenVox T1/E1 WAN interfaces. |
| 08:27.42 | CVirus | this is what voip-info says |
| 08:28.06 | JT | CVirus: do you have a question or not? |
| 08:28.34 | CVirus | JT: Where do you plug a PRI/BRI line ? |
| 08:28.37 | uwe | hello, how can i know what ulimit -c is set for an already running process (asterisk) !? i tried asking at #linuxhelp and #linux, but so far no one seem to have a clue how to figure that for an already running process |
| 08:28.37 | CVirus | do I* |
| 08:28.49 | JT | CVirus: into a card? |
| 08:28.52 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 08:29.08 | CVirus | JT: into an FXO port ? |
| 08:29.14 | JT | you will need the correct card of course |
| 08:29.15 | JT | no |
| 08:29.19 | JT | fxo is for an analogue line |
| 08:29.25 | JT | ~thebook |
| 08:29.40 | jbot | thebook is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 08:30.56 | *** part/#asterisk abasic (n=alen@89-172-54-206.adsl.net.t-com.hr) |
| 08:31.14 | CVirus | Thanks |
| 08:31.42 | *** join/#asterisk keulin (n=cray@AMontpellier-152-1-51-36.w83-197.abo.wanadoo.fr) |
| 08:45.47 | *** join/#asterisk hermuli (n=Eladamri@a88-112-255-26.elisa-laajakaista.fi) |
| 08:46.43 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 08:47.49 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.144.124) |
| 08:48.37 | uwe | DarKnesS_WolF, 2fel ? 2 like in 2a 2o 2e |
| 08:49.13 | DarKnesS_WolF | uwe: 2fel == a lock |
| 08:49.14 | DarKnesS_WolF | it's arabic |
| 08:49.34 | MrWup | anyone know how to suppress all warnings in PHP? |
| 08:49.55 | DarKnesS_WolF | uwe: we are using 2 for special arabic charctr.. |
| 08:49.56 | uwe | yeah, i thought so too ... |
| 08:50.06 | uwe | 3arabi ... |
| 08:50.09 | DarKnesS_WolF | uwe: we are using english letters to type arabic... much faster in english :-) |
| 08:50.13 | DarKnesS_WolF | uwe: yes 3arabi ;-) |
| 08:50.16 | DarKnesS_WolF | uwe: where ar u from ? |
| 08:50.21 | uwe | 3rifet |
| 08:50.35 | uwe | palestine |
| 08:51.01 | DarKnesS_WolF | uwe: hahah eshta :-) |
| 08:51.16 | uwe | u ? masr ? |
| 08:55.18 | uwe | DarKnesS_WolF, you know of any community working on arabic localized asterisk doc and/or support and/or development ? |
| 08:56.41 | DarKnesS_WolF | uwe: nop i wish i do ... |
| 08:56.47 | DarKnesS_WolF | uwe: yes i'm from egypt |
| 08:57.29 | HarryR | I think it'd be cool to have asterisk documentation in some of the other major languages, like spanish, arabic & mandarin |
| 08:57.35 | uwe | DarKnesS_WolF, i know here like 3-4 users scattered around, i thought a mailing list or something similar could be a good start, no ? |
| 08:57.45 | HarryR | who cares about french and italian :) |
| 08:58.55 | HarryR | oh hmm, arabic isn't even in the top 10 languages :\ Hindu would be better then |
| 09:00.00 | uwe | HarryR, :) a full arabic documentatin would be probably awful |
| 09:00.19 | *** join/#asterisk hannak (n=hanna@82.102.196.2) |
| 09:00.40 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 09:01.05 | HarryR | i'm sure it'd be very tedious to translate :\ |
| 09:01.33 | HarryR | but it'd be a very good task for somebody trying to learn MSA |
| 09:01.43 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 09:02.09 | HarryR | i'll have a go at translating it to Khmer when I'm good enough :\ |
| 09:02.21 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 09:03.32 | DarKnesS_WolF | uwe: yes sure |
| 09:05.52 | *** join/#asterisk michael-i (n=michael-@141.41.40.193) |
| 09:08.20 | MrWup | anyone know which bit of the * source in channel.c flushes the channel variables from a Zombie channel? |
| 09:08.48 | uwe | DarKnesS_WolF, any suggested places/domains , we are 2+ at #pslug |
| 09:11.53 | uwe | i can start one like astug@plug.ps, but that wouldnt be nice, i.e. .ps |
| 09:16.51 | ghenry | what pri card would you recommend? |
| 09:16.56 | ghenry | I'm in UK |
| 09:20.57 | *** join/#asterisk santibiotico (n=santi@ip23498.bcn.altecom.net) |
| 09:20.59 | santibiotico | hi |
| 09:21.25 | santibiotico | when using the page application..how do i define to call all users? |
| 09:21.41 | santibiotico | instead of specifying SIP/11&SIP/12$etc etc etc |
| 09:29.08 | *** join/#asterisk qdk_ (n=qdk@213.150.62.32) |
| 09:30.19 | simplexio | santibiotico: try SIP/__ .. no idea does that work. |
| 09:39.37 | *** join/#asterisk nasls_lsa (n=chatzill@87.203.68.253) |
| 09:51.42 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 09:51.44 | Chris-NB | hi |
| 09:51.55 | Chris-NB | anyone using a Thomson ST2030S phone? |
| 10:01.11 | *** join/#asterisk mkl1525 (n=qwertz@i59F77945.versanet.de) |
| 10:08.44 | mkl1525 | Hi, (* 1.2) trying to get access to my voip provider "sip show registry" shows the provider as "Registered" so I suppose connection is working. When calling from outside the only debug entry is "Unknown SIP media type in offer: video 62280 RTP/AVP 21 34" would this cause a problem or is it a fatal warning? |
| 10:10.10 | remmo | that should not cause any major problems |
| 10:13.00 | mkl1525 | remmo, thanks, so my guess that "Registered" stands for "is connected to the voip proivder" is correct? and if so - any reason why I don't see any call/info on verbose log when I call from outside? |
| 10:16.25 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 10:16.31 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 10:33.01 | jacq | registered only means your provider knows how to locate you |
| 10:33.46 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 10:34.27 | jacq | looks like you havent defined your sip peer properly |
| 10:34.33 | jacq | to support your codec |
| 10:36.12 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 10:38.37 | *** join/#asterisk Ast001 (n=uros@77-105-47-91.adsl-2.sezampro.yu) |
| 10:38.43 | Ast001 | hello |
| 10:43.24 | *** join/#asterisk znoG (n=gs@235-180-235-201.fibertel.com.ar) |
| 10:46.37 | MrWup | anyone know where ast_string_field_set is defined? |
| 10:49.01 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 10:51.49 | *** part/#asterisk Ast001 (n=uros@77-105-47-91.adsl-2.sezampro.yu) |
| 11:09.12 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:09.45 | defswork | can I test ring a handset from the asterisk cli ? |
| 11:13.51 | jacq | MrWup: include/asterisk/stringfields.h:275 ? |
| 11:14.02 | MrWup | thanks already found it though |
| 11:14.33 | MrWup | im trying to hack the * code to stop channel variables of zombie channel being flushed before the hangup extension code executes |
| 11:14.40 | MrWup | proving difficult to find where the variables are |
| 11:14.45 | MrWup | its all in channel.c somewhere |
| 11:14.55 | MrWup | round about line 3512 i think |
| 11:14.59 | MrWup | but not sure what to do |
| 11:15.00 | jacq | tried looking at doxygen on asterisk.org? |
| 11:15.10 | MrWup | yeah |
| 11:15.12 | MrWup | no luck |
| 11:18.33 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 11:19.33 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
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| 11:42.02 | nomadsoul | justdave: hi |
| 11:42.07 | defswork | odd - I've connected my Nokia N95 and added it to my ring group but asterisk says circuit-busy when I dial in. if I dial from AMI using originate the n95 rings ok |
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| 12:19.38 | sergee | i have a problem with Linksys 2102, it doesn't recognise key "3" dialed on attached phone, and pass numbers to asterisk without "3" :) does anybody know how to solve it? |
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| 12:30.39 | iCEBrkr | :/ |
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| 12:44.24 | ctooley | Anyone know of some good tuning parameters for the RHEL4 kernel when doing high volumes of SIP bridges? |
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| 12:53.55 | HarryR | ctooley, including RTP? |
| 12:55.10 | HarryR | ctooley, see: http://www.shell-tips.com/2006/11/25/fine-tuning-a-linux-apache-mysql-php-lamp-server/ (TCP section) and http://www-128.ibm.com/developerworks/linux/library/l-hisock.html |
| 12:56.26 | HarryR | other than that, the first thing I'd suggest is: not using Asterisk ;) |
| 12:57.29 | ctooley | HarryR, none of that stuff will help with RTP |
| 12:57.50 | ctooley | no matter how much you tune TCP, Apache or MySQL settings, it's not going to help RTP |
| 12:58.06 | HarryR | I was referring to the TCP stuff, not apache/mysql |
| 12:58.16 | HarryR | oh right, I see |
| 12:59.07 | Ryushin | So are there any pstn gateway that are free for non business use for local calls? I'm trying to use something else instead of my cell phone. |
| 12:59.26 | Ryushin | This will be for the US. |
| 12:59.56 | ctooley | Ryushin, Oddly enough, even local calls cost money to terminate |
| 13:00.52 | Putzz | heh |
| 13:00.57 | dacter | so... what's the "Authorization user name" in x-lite, and is that somehow different from the "user name"? |
| 13:01.33 | JT | HarryR: what tcp stuff? |
| 13:01.35 | matt_ | does anybody here use a service from the Finarea SA group |
| 13:02.04 | HarryR | ah I was pointing out general TCP tuning stuff for linux, not UDP |
| 13:02.13 | Ryushin | ctooley: Bummer. I'm just trying to cut back on some of my minutes on my cell phone. |
| 13:02.22 | JT | HarryR: i see |
| 13:02.30 | Ryushin | Guess I'll do some digging for the best price. |
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| 13:03.00 | matt_ | they dont seem to hang up the call properly, when i hangup the phone keeps ringing? does anybody know why, codec issue maybe |
| 13:03.09 | ctooley | Ryushin, I'd like to cut back on the minutes on my cell phone too. I stopped using it so much. |
| 13:03.29 | Putzz | can u say cheap? USA/Can calls are often terminated for 1c a min |
| 13:03.47 | ctooley | Putzz, depending on where you call. |
| 13:04.00 | Putzz | well us48 |
| 13:04.11 | Putzz | canada excluding NWT and yukon |
| 13:04.12 | ctooley | We've got rates that can go as low as .4 cents/minute but as high as 3 cents/minute. |
| 13:04.21 | ctooley | and that's just in the domestic 48 |
| 13:04.41 | Putzz | that still not bad coompared to cell rates cell carriers charge |
| 13:04.51 | Putzz | in canada .25-.35c |
| 13:04.54 | Putzz | a min |
| 13:04.56 | ctooley | It depends on where you call. It's expensive to terminate into areas where we don't have Access Relief, like Indian Reservations in Oklahoma |
| 13:05.09 | Putzz | true |
| 13:05.45 | ctooley | and almost always someone has to actually have a piece of copper running to the door of whoever you're calling. |
| 13:06.05 | ctooley | As more and more people move to VoIP it gets cheaper, but it's still expensive in some places... relatively speaking. |
| 13:08.03 | Ryushin | ctooley: Where are the rates as low as .4 cent a minute? |
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| 13:08.28 | ctooley | Ryushin, mostly to small pockets of mobile phones. I'm not sure |
| 13:09.04 | ctooley | Ryushin, plus you have to have carrier level volume to justify a NPA/NXX based rate deck. Otherwise we have a flat rate. |
| 13:09.43 | Ryushin | Okay, I guess I'll do some more digging. I guess I can live with cent a minute or so. You know of any carriers I should look at? |
| 13:10.01 | Putzz | google is a good place to start |
| 13:10.05 | *** part/#asterisk dhakatel (n=root@58.65.224.5) |
| 13:10.11 | Putzz | spoon feeding sucks |
| 13:10.18 | *** part/#asterisk trevarthan (n=trevarth@c-71-59-54-137.hsd1.ga.comcast.net) |
| 13:10.34 | Ryushin | I was looking at voip-info and there are a lot to choose from. Just didn't know if someone had already done their homework on this. |
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| 13:11.42 | ctooley | Ryushin, depends on what you're looking for. NuFone does do a lot of work with Asterisk though, they might be a good option. I work for one but we don't really have a retail resi package |
| 13:12.37 | ctooley | This isn't cool: src/add.c:1: error: CPU you selected does not support x86-64 instruction set |
| 13:12.54 | MrWup | im trying to hack the * code to stop channel variables of zombie channel being flushed before the hangup extension code executes |
| 13:13.01 | ctooley | the linux kernel seems to think that the CPU can do x86-64 instructions just fine. |
| 13:13.07 | MrWup | anyone know what code does that? |
| 13:13.11 | MrWup | somewhere in channel.c? |
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| 13:15.13 | dacter | ctooley: what does "ls -l /usr/src/" give you (and what flavor of linux?) |
| 13:15.29 | ctooley | dacter, CentOS 4.4 |
| 13:15.44 | sglinux | if I want to do a POTS only 'callback' setup with Asterisk, I need 2 POTS lines, right ? |
| 13:15.59 | ctooley | dacter, and the kernel source for the running kernel is in /usr/src/kernels |
| 13:16.10 | ctooley | and symlinked to /lib/modules/`uname -r`/build |
| 13:16.30 | ctooley | it's finding the kernel source. Zaptel 1.4.2.1 built fine. |
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| 13:17.35 | DrAk0 | hey, im trying to figure out why i get zero channels on misdn |
| 13:17.37 | DrAk0 | pbx*CLI> misdn show channels |
| 13:17.38 | DrAk0 | Chan List: (nil) |
| 13:20.08 | DrAk0 | Wed May 2 20:10:10 2007: P[ 0] Could not create channel on port:-1 with extensions:944413020 |
| 13:20.40 | sglinux | anyone running Asterisk in Singapore here ? |
| 13:21.25 | coppice | there seem to be quite a few users in .sg |
| 13:25.38 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 13:25.47 | tzafrir | ctooley, what is the output of: uname -r |
| 13:26.06 | mkl1525 | Hi, (* 1.2, snom 300|360, bellshare.com) I've got connection to my voip provider and the the calls get through. caller can hear and speak but the called party can just listen. no voice goes back from called to caller - any hints what could go wrong? |
| 13:26.22 | ctooley | tzafrir: problem resolved |
| 13:26.50 | rue_mohr | MrWup, I'm interested in channel.c, I need to add support for a newbridge 3624 |
| 13:26.59 | iCEBrkr | mkl1525: Typical of a NAT setup |
| 13:30.19 | *** part/#asterisk serotonin|work (i=ryan@mail.tankprofiler.com) |
| 13:30.36 | DrAk0 | any idea about ISDN problem_ |
| 13:30.39 | MrWup | asterisk sourcecode is a nightmare |
| 13:31.26 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 13:31.30 | mkl1525 | iCEBrkr, thanks for the hint, that's a good guess (having the phones in a test enviroment not the real net) |
| 13:33.30 | iCEBrkr | mkl1525: If it's all on the same network, it should be fine. |
| 13:34.12 | iCEBrkr | mkl1525: But typically, SIP doesn't play well over NAT. |
| 13:36.52 | nomadsoul | MrWup: have you tryed to use doxygen over it? you will have a nice doc to look at :D |
| 13:37.04 | MrWup | i think ive found the source to make it work |
| 13:37.10 | MrWup | trial and error eliminating, compiling =] |
| 13:39.42 | [TK]D-Fender | iCEBrkr: Only problem I ever had with SIP & NAT is Cisco PIX, and a few odd D-Link routers |
| 13:41.06 | iCEBrkr | [TK]D-Fender: Probably the ones that done have that magic UPNP feature |
| 13:41.23 | iCEBrkr | I've had good luck as well. |
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| 13:42.29 | DrAk0 | anyone? with a b410p working? im having problem getting the channels work. |
| 13:42.39 | rbd | hi guys, I have multiple asterisk servers that serve to frontend (IVR) incoming calls from a SIP provider. If I were to use DNS SRV to load balance across these servers, could I have it so that if a server got too full |
| 13:42.55 | *** join/#asterisk zogulus (n=zogulus@58.98.adsl.brightview.com) |
| 13:43.06 | rbd | it could refuse an incoming call, and the next server would be tried for it? |
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| 13:44.03 | blitzrage | rbd: depends if that's how DNS will work. I tend to use Transfer() to 302 redirect to the other server when I want to reject |
| 13:44.53 | iCEBrkr | blitzrage: You do something like, checking for the number of channels in use and then issue a Transfer() to do your 'load balancing'? Is that the concept? |
| 13:45.12 | rbd | blitzrage: makes sense. basically I just wanted to see if I could offer more inteligent load balancing than simple weighting/round robin |
| 13:45.26 | MrWup | bAAAHAHAHAHA |
| 13:45.27 | MrWup | ive done it |
| 13:45.32 | MrWup | ive finally hacked the * source |
| 13:45.33 | *** join/#asterisk joshaidan (n=brianj@thunderbay-voip-4.vianet.ca) |
| 13:45.36 | *** join/#asterisk Fieldy (i=3NJCOndh@gentoo/contributor/Fieldy) |
| 13:45.39 | iCEBrkr | rbd: You could go all out and build a 'traffic manager' :) |
| 13:45.48 | MrWup | and made it so that Zombie channels retain their variables in the h extension |
| 13:45.50 | MrWup | mooohahaha |
| 13:45.53 | iCEBrkr | MrWup: HAX0Rz |
| 13:45.54 | blitzrage | yes -- I do a load test to know how many calls I can do, then cut that down to 60%, use GROUP() and GROUP_COUNT() to track the number of calls on the system, and use Transfer() to force the call to the other server when it hits capacity |
| 13:45.58 | MrWup | and its so simple too |
| 13:46.04 | MrWup | took soooooo long to narrow it down though |
| 13:46.46 | rbd | blitzrage: sounds good, thanks |
| 13:47.34 | rbd | blitzrage: why 60%? is that a real-world safe figure? and what tool do you use to load test? |
| 13:48.47 | rbd | iCEBrkr: well there is SEP/OpenSEP whch does SIP load balancing, but the load balancing algos it uses are very simple....it doesn't seem to have an algo that can make use of external data (such as cached database query results or asterisk manager query data, etc) |
| 13:48.57 | rbd | it could always be enhanced though... |
| 13:48.59 | iCEBrkr | I used GROUP_COUNT() for outbound call throttling. |
| 13:49.01 | blitzrage | rbd: I use SIPp to load test. I use 60% because the load testing I did didn't involve call recordings, etc... etc... (simply setup), so I just picked 60% as a safe figure |
| 13:49.12 | blitzrage | SER/OpenSER* |
| 13:49.30 | rbd | oops, yeah |
| 13:49.43 | blitzrage | our topology uses OpenSER as the SIP registration end point, which distributes the calls among a cluster of Asterisk boxes |
| 13:50.03 | iCEBrkr | I should really look into SER |
| 13:50.49 | Peri | OpenSER > SER (but that's opinion) |
| 13:50.58 | jacq | blitzrage: using media proxy or rtp proxy? |
| 13:51.28 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 13:53.33 | anonymouz666 | blitzrage: and if you got let's say 6 T1's... what's used to distribute the inbound calls through the ast boxes? |
| 13:53.53 | blitzrage | anonymouz666: you don't -- they are coming from a T1 and are physically tied to a box |
| 13:53.56 | rue_mohr | can I get asterisk to only pick up a call aftera certian number of rings, or not at all? (as in I dont suspect a timer would do that) |
| 13:54.28 | blitzrage | jacq: using neither -- I use directrtpsetup=yes to send the data to the carriers directly |
| 13:54.47 | blitzrage | we don't do residential really, so most of our customers having a real IP and not behind NAT |
| 13:54.54 | blitzrage | if not, Asterisk proxies the media |
| 13:54.55 | *** join/#asterisk hfb (n=hfb@pool-72-67-156-130.lsanca.dsl-w.verizon.net) |
| 13:55.08 | blitzrage | rue_mohr: use Wait() before Answer() |
| 13:55.37 | rue_mohr | ok |
| 13:56.20 | rue_mohr | and then if the state changes it'll abort the answer |
| 13:56.48 | [TK]D-Fender | rue_mohr: Sort of. |
| 13:57.33 | [TK]D-Fender | rue_mohr: On analog you run the risk of picking up between rings where a caller could have hung up. Poor disconnect detection could lock up your channel potentially for a little bit |
| 13:58.13 | rue_mohr | :) it gets more fun than that, my channelbank signaling for a hangup isn't recognized by * |
| 13:58.48 | [TK]D-Fender | rue_mohr: namely? |
| 13:58.56 | rue_mohr | I'm just swimming through chan_zap.c to try to find the state machine for that |
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| 13:59.00 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:59.05 | rue_mohr | mainstreet 3624 |
| 14:02.07 | rue_mohr | I need to work out what sequence its sending |
| 14:02.27 | rue_mohr | I'm hoping there's already debug code around the state machine, for change logging |
| 14:07.23 | iCEBrkr | MrWup: um, question? Why would you want the variables to stick around? |
| 14:07.53 | MrWup | because in the h extension the zombie channel hangs up |
| 14:08.04 | MrWup | and you need to know which phone the zombie channel is associated with |
| 14:08.23 | MrWup | so you can play with the database where info about peoples numbers of calls are stored |
| 14:08.32 | MrWup | zombies happen when a call is transfered too |
| 14:08.35 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:08.38 | LeddyHM | how uber gay |
| 14:09.04 | MrWup | and obviously its difficult to track call transfers unless you can track what happens to the zombies |
| 14:09.08 | MrWup | (i.e. the caller who drops out) |
| 14:09.19 | iCEBrkr | I'm confused as to what you mean by zombies? |
| 14:09.20 | *** join/#asterisk wierdo (n=noname@digsys34-217.pip.digsys.bg) |
| 14:09.28 | LeddyHM | "you have to change your entire network and put your phones on canreinvit=yes" for dtmf to work |
| 14:09.40 | MrWup | iCEBrkr, read the source |
| 14:09.48 | MrWup | its quite clear about what a zomby is |
| 14:09.50 | MrWup | zombie even |
| 14:09.59 | iCEBrkr | MrWup: Did you just tell me to RTFS? :D |
| 14:10.04 | MrWup | =] |
| 14:10.06 | iCEBrkr | lol |
| 14:10.16 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:11.31 | iCEBrkr | MrWup: It's to early for spaghetti |
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| 14:12.37 | *** mode/#asterisk [+o mogorman] by ChanServ |
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| 14:15.08 | [TK]D-Fender | LeddyHM: Sounds like raging BS. |
| 14:15.21 | [TK]D-Fender | LeddyHM: RFC = STFU |
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| 14:20.32 | Mercestes | raging BS? |
| 14:22.29 | Polis_ttt | anyone that got a tip of a simple statistic-script that i can run, like php-script or so, that shows the load of my asterisk-mashin? like cpu and networkload? |
| 14:23.23 | *** join/#asterisk qdk_ (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
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| 14:23.48 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 14:24.28 | wwalker | what is the command from inside the asterisk cli to get a sip peer to reread its config? |
| 14:25.07 | Polis_ttt | wwalker: "sip reload" ? |
| 14:26.06 | anonymouz666 | after a read() can I use $ISNULL to check if the var contains something? Or Can I use ! using ${IF} with ${EXISTS}? |
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| 14:27.00 | wwalker | Polis_ttt: thx, but I was trying to reboot a polycom phone remotely (I'm 30 miles from the office). |
| 14:27.19 | iCEBrkr | Polis_ttt: dstat? |
| 14:28.53 | iCEBrkr | Polis_ttt: top |
| 14:28.53 | iCEBrkr | Polis_ttt: uptime |
| 14:28.54 | iCEBrkr | wwalker: Don't the polycoms have some sort of web interface? |
| 14:29.03 | MatBoy | Does someone know a webbased VoIP client that supports SIP ? |
| 14:29.05 | wwalker | fyi - sip notify polycom-check-cfg peer_name |
| 14:29.39 | Polis_ttt | iCEBrkr: dstat was good, thanks |
| 14:30.45 | wwalker | iCEBrkr yes |
| 14:30.45 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 14:33.36 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 14:33.37 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- Asterisk 1.4.4 (April 27, 2007) Asterisk 1.2.18 (April 24, 2007), Zaptel 1.2.17.1, 1.4.2.1 (April 25, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/#trixbox for trixbox support. |
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| 14:55.47 | docelmo | YAY! |
| 14:55.54 | iCEBrkr | docelmo: SHUTUP |
| 14:56.06 | docelmo | STFU |
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| 14:56.34 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177581822.dsl.bell.ca) |
| 14:56.35 | iCEBrkr | docelmo: I guess Damin is visiting Orlando come Memorial Day weekend |
| 14:56.41 | docelmo | eh |
| 14:56.53 | iCEBrkr | Not sure why |
| 14:56.56 | docelmo | I dunno.. havent talk to him in a few.. been wicked busy |
| 14:57.02 | docelmo | probably out to buy a clec down there |
| 14:57.13 | iCEBrkr | docelmo: Since I'm headed over to Tampa that weekend, we're gonna grab a beer |
| 14:57.28 | docelmo | have at it |
| 14:57.33 | iCEBrkr | docelmo: It seems important. Ed (his business partner) is coming with him |
| 14:57.39 | docelmo | haha |
| 14:57.43 | docelmo | they are going to buy someone |
| 14:57.48 | iCEBrkr | Probably |
| 14:57.59 | iCEBrkr | haha |
| 14:58.04 | docelmo | hehe |
| 14:58.24 | coppice | sounds better than ending up with a CLEC |
| 14:58.41 | coppice | the tee short stands a chance of coming debt free |
| 14:58.48 | iCEBrkr | coppice: haha |
| 14:59.05 | ManxPower | coppice: Other than the whole "All the regulations and the telcos are trying to put you out of business" thing, a CLEC would be fun. |
| 14:59.38 | iCEBrkr | coppice: Surprisingly, Damin's company has been successful. I think they're pushing close to 12yrs now? |
| 14:59.42 | iCEBrkr | yikes! 12yrs?!@# |
| 14:59.55 | iCEBrkr | Time flies |
| 15:00.46 | iCEBrkr | I remember having just under 3000 customers and about 200 hosted websites. and they were buying up all the mom and pop dial-up ISPs |
| 15:02.12 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 15:03.03 | docelmo | hehe |
| 15:03.14 | docelmo | he just bought a central office for his new data center |
| 15:04.05 | iCEBrkr | Buy buy buy |
| 15:06.48 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 15:09.06 | *** join/#asterisk rokjan (n=jj2@static-200-105-156-114.acelerate.net) |
| 15:10.16 | ManxPower | Mist be nice. I think it is a big deal when I spend $100 on a set of shelves |
| 15:10.49 | iCEBrkr | Ha! |
| 15:11.01 | Qwell[] | what? $100 for shelves? |
| 15:11.07 | Qwell[] | I can make them for like $8 |
| 15:11.41 | iCEBrkr | Qwell[]: I'm not to sure about your opensource shelves. :P |
| 15:11.53 | Qwell[] | they aren't open source |
| 15:11.59 | Qwell[] | I'll be damned if I'm gonna tell you how to make them |
| 15:12.04 | iCEBrkr | haha |
| 15:12.25 | ManxPower | They were actually $88 and a freestanding cabinet rather than shelves |
| 15:12.33 | ManxPower | stop ruining my examples with logic! |
| 15:13.25 | mogorman | Qwell, wood would cost more than 8 bucks |
| 15:13.35 | Qwell[] | not pressed wood |
| 15:13.38 | Qwell[] | ;p |
| 15:13.44 | coppice | not if its a very small piece |
| 15:13.56 | Qwell[] | coppice: that too |
| 15:14.03 | mogorman | any kind of wood Qwell |
| 15:14.14 | mogorman | unless your taking it from construction site |
| 15:14.17 | Qwell[] | mogorman: it's more "woodesque" |
| 15:14.19 | coppice | the sort that grows on trees? |
| 15:14.21 | mogorman | lol |
| 15:14.23 | Qwell[] | it's really just cardboard that I stole |
| 15:14.29 | Qwell[] | ...and painted |
| 15:14.33 | MrWup | why is it that setting __variables in a macro which completes upon dial doesnt set them properly |
| 15:14.33 | mogorman | uh hu.... |
| 15:14.39 | coppice | sounds very Ikea |
| 15:15.04 | Strom_M | the native american competitor to IKEA would probably be called "Chippaway" |
| 15:15.11 | *** part/#asterisk sglinux (n=chatzill@bb219-74-98-25.singnet.com.sg) |
| 15:15.37 | *** join/#asterisk Dimik_ (n=Dimik_@unaffiliated/dimik) |
| 15:15.39 | Mercestes | Strom_M, booo. That was bad.] |
| 15:15.56 | Strom_M | I never claim my jokes are actually any good |
| 15:16.12 | Mercestes | Very true. You only offer free delivery. |
| 15:16.18 | coppice | rather like Ikea and their furniture |
| 15:16.22 | Mercestes | precisely |
| 15:16.23 | Qwell[] | Mercestes: you never signed up for that trial ;) |
| 15:16.34 | *** join/#asterisk silentfury (i=anubis@CPE0001292d787f-CM000f9f5011d8.cpe.net.cable.rogers.com) |
| 15:16.35 | Mercestes | Yea, I know. I'm sorry...:( |
| 15:16.39 | Mercestes | I had my kids this weekend. |
| 15:16.42 | silentfury | hi guys |
| 15:16.52 | silentfury | i'm looking at a potential job where Asterisk is their main VoIP server |
| 15:16.58 | Mercestes | I'll do it, I promise. |
| 15:17.16 | Mercestes | silentfury, Then we'll be seeing alot more of you, I take it? |
| 15:17.17 | Qwell[] | Mercestes: excuses, excuses :p |
| 15:17.33 | silentfury | merc, maybe ;) |
| 15:17.35 | Mercestes | ~book |
| 15:17.41 | jbot | extra, extra, read all about it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:17.41 | Mercestes | ~docs |
| 15:17.53 | jbot | extra, extra, read all about it, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 15:17.53 | Mercestes | ~wglwat |
| 15:17.59 | jbot | methinks wglwat is well, good luck with all that |
| 15:18.00 | silentfury | is there a good crash course resourceE? |
| 15:18.02 | silentfury | I'm a quick learner - i'm hoping i can master this quickly. |
| 15:18.10 | Mercestes | jbot knows all |
| 15:18.12 | jbot | and don't you forget it |
| 15:18.27 | Mercestes | ~botsnack |
| 15:18.27 | jbot | Mercestes: :) |
| 15:18.37 | *** join/#asterisk d4rkst4r75 (n=d4rkst4r@85-18-66-28.ip.fastwebnet.it) |
| 15:18.42 | d4rkst4r75 | hi to all |
| 15:18.54 | Mercestes | hi |
| 15:19.03 | Mercestes | ASL? |
| 15:19.49 | d4rkst4r75 | i got an error with my E1 configuration: chan_zap.c: no D-channels avaiable! Using Primary channel 16 as D-channel anyway |
| 15:20.10 | d4rkst4r75 | my environment: asterisk 1.2 + libpri + wanpipe (sangoma board a102d) |
| 15:20.43 | d4rkst4r75 | my configuration in zaptel.conf is: |
| 15:20.45 | Mercestes | Is 16 really your Dchan? |
| 15:20.51 | d4rkst4r75 | yes, on E1 |
| 15:20.52 | Mercestes | ~pb |
| 15:20.54 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 15:21.04 | Strom_M | Mercestes: 16 is always the dchannel on E1 |
| 15:21.15 | Qwell[] | It's always channel 42 on a Q1 |
| 15:21.16 | d4rkst4r75 | yes |
| 15:21.22 | Mercestes | Strom_M, I'm sorry. *cries* I didn't know |
| 15:21.33 | Strom_M | Qwell[]: hot |
| 15:21.43 | d4rkst4r75 | the E1 comes up and is active for a while |
| 15:21.59 | d4rkst4r75 | after some time (randomly) the E1 goes down with that error |
| 15:22.23 | Mercestes | d4rkst4r75, Do you have just this one E1? |
| 15:22.33 | d4rkst4r75 | yes Mercestes |
| 15:22.40 | d4rkst4r75 | i can't try others |
| 15:22.44 | coppice | some people get really weird, and put the E1 D-channel on 31, but its rare |
| 15:23.04 | Mercestes | well, I was asking because sangoma has this weird thing with multiple telco timing sources and their latest firmware. |
| 15:23.30 | d4rkst4r75 | i've my sangoma configured as Master Clock source |
| 15:23.32 | Mercestes | pastebin your configs |
| 15:23.46 | d4rkst4r75 | ok |
| 15:24.55 | Strom_M | completely off topic but amusing nonetheless: http://hackedgadgets.com/2007/05/06/domino-pcs/ |
| 15:25.05 | Qwell[] | Strom_M: welcome to weeks ago |
| 15:25.05 | *** join/#asterisk slmnhq (n=salmanh@denali.asti-usa.com) |
| 15:27.20 | d4rkst4r75 | http://www.pastebin.ca/477269 |
| 15:28.34 | d4rkst4r75 | i've pasted all the configurations, the system, the logs before and after the span goes down |
| 15:29.06 | Mercestes | Nice |
| 15:29.24 | d4rkst4r75 | i've tried to downgrade to asterisk 1.2 and libpri 1.2 but the error is the same |
| 15:29.44 | d4rkst4r75 | the logs are for version 1.4, but i repeat: the problem is always the same |
| 15:30.27 | Mercestes | are you sure the span is up? Did you contact yoru telco? |
| 15:30.30 | d4rkst4r75 | yes |
| 15:30.31 | *** join/#asterisk _Sam-- (n=sam@fresco.kneedraggers.com) |
| 15:30.46 | d4rkst4r75 | the span goes up and is working fine when it is up |
| 15:31.03 | d4rkst4r75 | the problem is that it randomly goes down with the error i signaled |
| 15:31.14 | *** join/#asterisk salviadud (n=dude@189.156.174.25) |
| 15:31.31 | Mercestes | what does zttest say? |
| 15:31.34 | *** join/#asterisk DRoBeR (n=DRoBeR@212.145.188.221) |
| 15:31.38 | DRoBeR | Hello all. |
| 15:31.47 | d4rkst4r75 | i've not done the zttest |
| 15:31.57 | _Sam-- | sorry for off topic question -- does anyone know if Xlite works with Vista? (having a hard time getting a friends xlite to register) |
| 15:32.19 | *** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) |
| 15:32.21 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 15:32.28 | Mercestes | _Sam--, Try custom firewall settings, and run it as administrator. |
| 15:32.43 | _Sam-- | thanks will give it a shot.... |
| 15:32.43 | Mercestes | d4rkst4r75, Try that while I (we) read. |
| 15:32.57 | _Sam-- | (i had him disable the firewall already) |
| 15:33.24 | d4rkst4r75 | but my telco said that he sends a RR and get RR from my peer (asterisk) from a while. At a certain point, it sends RR but doesn't receive RR reply anymore. So he sends a SABME message and waits for the UA |
| 15:33.51 | d4rkst4r75 | that UA is not sended so the T200 Timer on the telco peer expiry and the link goes down |
| 15:35.17 | d4rkst4r75 | zttest give me an accuracy of 99.975586% |
| 15:35.21 | *** join/#asterisk ptblank (n=MURDER1@cpe-76-173-168-178.socal.res.rr.com) |
| 15:35.39 | d4rkst4r75 | Mercestes and the accuracy is always the same |
| 15:35.52 | Mercestes | d4rkst4r75, that's boarderline. 1: zapata.conf. are you pri_cpe or prI_net? |
| 15:36.00 | d4rkst4r75 | i'm pri_net |
| 15:36.14 | Mercestes | d4rkst4r75, is that correct for clock master? |
| 15:36.43 | _Sam-- | !seen zoa |
| 15:36.47 | *** join/#asterisk dracosilv (n=draco@CPE-65-29-47-173.wi.res.rr.com) |
| 15:36.56 | Mercestes | d4rkst4r75, try pri_cpe |
| 15:36.57 | *** join/#asterisk manopulus (n=manopulu@213.197.181.154) |
| 15:37.14 | Mercestes | wait. |
| 15:37.18 | manopulus | hello, is ooh323 latest are in 1.2.18 or only at 1.4.4? in asterisk-addons |
| 15:37.19 | Mercestes | nevermind. dont' try that |
| 15:37.29 | d4rkst4r75 | Mercestes, you say that when i'm pri_net i should set Clock Slave? |
| 15:37.42 | Mercestes | yoru the net in this case? |
| 15:37.50 | d4rkst4r75 | yes, i'm the net in this case |
| 15:38.10 | Mercestes | Ok, yea, leave it alone then, I'm confused. |
| 15:38.42 | tzafrir | manopulus, in asterisk-addons, both for 1.2 and for 1.4 |
| 15:39.03 | salviadud | i'm still using 1.2 |
| 15:39.03 | manopulus | tzafrir, thanks and it is same version? |
| 15:39.09 | Mercestes | d4rkst4r75, Hrm, I dunno. change the cable maybe? I'm worried about those 299 line errors and 2 crcs but other than that it looks ok. |
| 15:39.10 | *** join/#asterisk skyphyr (n=alanj@135.196.58.222) |
| 15:39.11 | salviadud | does 1.4 have a better mixmonitor? |
| 15:39.19 | tzafrir | there is a version for 1.2 and a version for 1.4 |
| 15:39.35 | Mercestes | pastebin zapata.conf too |
| 15:39.53 | d4rkst4r75 | ok, just one second Mercestes |
| 15:40.10 | *** join/#asterisk ^TheMask^ (n=mask@cm133.kappa157.maxonline.com.sg) |
| 15:41.00 | skyphyr | hi all - I've been running asterisk for my home phone for a couple of years now and my work is moving office. So looking for good voip phones and voip providers (we're in London though I imagine that's not hugely relevant) also any caveats/requirements running fax through asterisk. Thanks for any suggestions |
| 15:41.08 | d4rkst4r75 | www.pastebin.ca/477289 |
| 15:41.44 | *** join/#asterisk Peaceful (n=Peaceful@70.98.162.62) |
| 15:42.08 | ^TheMask^ | anyone around here that could spare a few precious minutes on a problem im facing? i bought a linksys spa-3102 and configured it to use fwdnet.. i was able to call in/call out 2 days back, but suddenly, now even with the echo tests or time test i get no audio in or out of my phone.. ive tested with my family in europe, no audio either.. but I am able to receive the calls/make the calls (it does connect) |
| 15:42.15 | ^TheMask^ | anyone any idea what could be wrong? |
| 15:42.50 | salviadud | are you connecting your spa directly to fwd? |
| 15:42.55 | Mercestes | d4rkst4r75, I dunno then, configs look ok to me. Did you try changing the cable? |
| 15:43.32 | *** join/#asterisk hijacked (i=mO4s@cerebus.clandestineresearch.com) |
| 15:43.42 | d4rkst4r75 | i can do another cable, but my question is: why randomly? |
| 15:43.43 | ^TheMask^ | salviadud: yes, configured fwd.pulver.com as the SIP proxy |
| 15:44.13 | ^TheMask^ | i made sure it goes through the correct line as well (Line 1) |
| 15:44.45 | Mercestes | d4rkst4r75, If it is a bad cable it would be random |
| 15:46.08 | d4rkst4r75 | yes, i'll do another cable |
| 15:46.22 | *** join/#asterisk wunderkin (i=wunderki@ip68-108-204-139.ph.ph.cox.net) |
| 15:46.59 | Peaceful | For the life of me, I cannot figure out why asterisk's (1.2.13) voicemail system keeps reporting a time 6 hours earlier than the system time. Anyone else have this problem? |
| 15:47.24 | Qwell[] | Peaceful: set a timezone |
| 15:47.27 | Qwell[] | in voicemail.conf |
| 15:48.25 | Peaceful | Qwell[]: I have: "tz=mountain" |
| 15:48.31 | iCEBrkr | Qwell[]: tz= is broken |
| 15:48.36 | tzanger | I'm not mountain |
| 15:48.42 | iCEBrkr | Peaceful: you gotta jam the tz= on the end of each mailbox definition |
| 15:48.47 | russellb | iCEBrkr: then fix it |
| 15:48.49 | tzanger | tz = eastern :-) |
| 15:48.54 | russellb | jk ... |
| 15:48.59 | iCEBrkr | russellb: I actually started looking into it. |
| 15:49.02 | Qwell[] | mountain time no longer exists |
| 15:49.03 | *** join/#asterisk imapfool (n=edhorton@216.23.111.98.nw.nuvox.net) |
| 15:49.07 | Qwell[] | sorry for the inconvenience |
| 15:49.10 | iCEBrkr | haha |
| 15:49.18 | *** join/#asterisk ploieel (n=ploieel@Fb2e6.f.ppp-pool.de) |
| 15:49.47 | iCEBrkr | tz is also broken in STRFTIME() |
| 15:49.48 | russellb | hehe |
| 15:49.48 | Peaceful | soo...mountain doesn't exist, or I need to dupe tz after each mailbox? |
| 15:50.06 | iCEBrkr | Peaceful: I fixed mine by putting tz at the end of each mailbox |
| 15:50.23 | codefreeze | <PROTECTED> |
| 15:50.35 | Qwell[] | codefreeze: You no longer exist either. Sorry. |
| 15:50.38 | iCEBrkr | codefreeze: tz=NULL |
| 15:50.42 | iCEBrkr | lol |
| 15:52.11 | salviadud | would it be illegal to put music from super nintendo games as MOH? |
| 15:52.37 | iCEBrkr | salviadud: It'd be illegal because it'd be so annoying. |
| 15:52.45 | Mercestes | salviadud, The fact that you have it in mp3 format is likely illegal so you may rest at ease having already violated the law. Doesn't get much worse from here. |
| 15:52.56 | salviadud | i'm from mexico |
| 15:53.07 | Mercestes | so? |
| 15:53.19 | Mercestes | did you steal music from mexican games? |
| 15:53.22 | iCEBrkr | Mercestes: Naa, he's going to hook the audio-out from his Nintendo up to his Asterisk box |
| 15:53.23 | imapfool | has anyone managed to get IMAP voicemail storage working in 1.4.4? |
| 15:53.24 | salviadud | i might be not breaking laws |
| 15:53.31 | Mercestes | hell, your likely in the US anways so the law still applies. |
| 15:53.35 | salviadud | i happen to own those games |
| 15:53.39 | salviadud | on cartridge |
| 15:53.41 | Corydon-w | iCEBrkr: your Asterisk machine wouldn't happen to have been a Windows at one time, would it? |
| 15:53.47 | iCEBrkr | Corydon-w: nope |
| 15:53.48 | Mercestes | you cannot OWN a game, only have a license to play a game |
| 15:53.52 | salviadud | so, i own them to some extent |
| 15:53.54 | Mercestes | unless you WRITE the game |
| 15:53.57 | Mercestes | troll |
| 15:54.03 | iCEBrkr | pwn |
| 15:54.12 | Corydon-w | iCEBrkr: did you set the hardware clock to be local time or UTC? |
| 15:54.18 | iCEBrkr | Corydon-w: local |
| 15:54.25 | Peaceful | yay! adding the tz thing to the individual voicemail lines worked! Thanks iCEBrkr!, |
| 15:54.26 | Corydon-w | iCEBrkr: that's probably the problem |
| 15:54.30 | salviadud | you guys are really on the offensive today |
| 15:54.37 | Mercestes | nah, just me. |
| 15:54.37 | salviadud | hehe |
| 15:54.40 | iCEBrkr | Corydon-w: I'm not familiar with with UTC as I tried that crap once and my time for everything was jacked up. |
| 15:54.43 | Mercestes | I'm alwasy offensive. |
| 15:54.57 | salviadud | well, if it helps |
| 15:55.03 | salviadud | i love breaking the law |
| 15:55.03 | DRoBeR | Mercestes: Yes, USA applies laws... but lot of them are stupid laws from stupid politics. :P |
| 15:55.19 | DRoBeR | Judas Priest? |
| 15:55.20 | Corydon-w | iCEBrkr: Linux was designed for the hardware clock to be UTC. The localtime crap is a hack meant to make it work with dual-boot Windows machines |
| 15:55.22 | Mercestes | DRoBeR, Point. but that was never in contention. ;) |
| 15:55.36 | iCEBrkr | Corydon-w: It's definitely something different between 1.2.x and 1.4.x |
| 15:55.49 | iCEBrkr | Corydon-w: ahhh good to know |
| 15:56.14 | iCEBrkr | Corydon-w: which is probably why my time was jacked up when I tried using UTC :) |
| 15:56.16 | salviadud | i haven't tried 1.4, i think i'll lose all my 1.2 config, that works |
| 15:56.28 | salviadud | is it worth it? |
| 15:56.34 | iCEBrkr | salviadud: It's fun! Do it! |
| 15:56.55 | Corydon-w | salviadud: you don't keep backups? |
| 15:57.00 | salviadud | ok, i'll just test it on another box |
| 15:57.02 | iCEBrkr | When I moved over to 1.4 I nuked all my conf's and ported my stuff over. Just incase there were any new options and such between the conf files. |
| 15:57.07 | Putzz | is everyone on 1.4 now? |
| 15:57.27 | Corydon-w | If you use nothing that was deprecated in 1.2, then your configs will still work in 1.4 |
| 15:57.40 | Corydon-w | although some things may be deprecated |
| 15:57.48 | Peaceful | Putzz: not me |
| 15:58.01 | iCEBrkr | I'm getting a deprecated message for voicemails. |
| 15:58.19 | salviadud | i want to try 1.4 to see how well it handles meetings |
| 15:58.20 | iCEBrkr | I forget what it's about, I think it's the 'ub' option or something of the like. |
| 15:58.29 | salviadud | say, i start a meeting |
| 15:58.45 | Corydon-w | iCEBrkr: yeah, we no longer prefix mailboxes with u or b, it's now in the second argument |
| 15:58.51 | salviadud | then i got all this wav files of al pacino connected to buttons on a flash |
| 15:59.08 | salviadud | via xml, i inject them with dial and play |
| 15:59.17 | salviadud | asterisk is the ultimate pranking software |
| 15:59.27 | iCEBrkr | Corydon-w: I'll fix it one day when I'm not being lazy :) |
| 15:59.30 | salviadud | pranksterisk |
| 15:59.52 | iCEBrkr | salviadud: you have too much time on your hands. |
| 16:00.09 | salviadud | my job's easy, that's all :) |
| 16:00.14 | salviadud | i work with windows |
| 16:00.21 | iCEBrkr | That sounds more like a headache |
| 16:00.36 | salviadud | well, it sucks like no other |
| 16:01.24 | salviadud | does anyone here work with linux ALL the time? |
| 16:01.28 | salviadud | i envy that |
| 16:01.33 | Corydon-w | I do |
| 16:01.42 | Putzz | I do |
| 16:01.50 | cpm | salviadud, pretty much. |
| 16:01.58 | iCEBrkr | salviadud: I'm on a Linux workstation doing PHP+MySQL web application development on Apache and we use svn... Does that count as all the time? |
| 16:02.36 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 16:02.49 | iCEBrkr | I can't remember the last time I fired up VMware for Xp |
| 16:03.00 | salviadud | that's all the time, yep |
| 16:03.22 | salviadud | have you checked out the OFBIZ project? |
| 16:03.35 | *** part/#asterisk imapfool (n=edhorton@216.23.111.98.nw.nuvox.net) |
| 16:04.01 | salviadud | i think it's very promising for lazy unix users to get a generic company started in no time |
| 16:04.11 | *** join/#asterisk imapfool (n=edhorton@216.23.111.98.nw.nuvox.net) |
| 16:04.23 | *** part/#asterisk imapfool (n=edhorton@216.23.111.98.nw.nuvox.net) |
| 16:04.29 | salviadud | and if you play with it, you can make it dial with festival or some funny thing like that |
| 16:05.18 | BSD_Tech | festival is text to speech |
| 16:05.23 | BSD_Tech | your thinking sphinx |
| 16:05.30 | BSD_Tech | or lumenvox |
| 16:05.34 | *** part/#asterisk Peaceful (n=Peaceful@70.98.162.62) |
| 16:05.45 | BSD_Tech | for voice control |
| 16:06.39 | joebob777as7 | can someone help me I have some simple questions... I am wanting to have 3 phone lines and about six phones in our new office. What hardware should I get? Should i get a voip router? etc... and what phones do you guys recommend? |
| 16:06.41 | salviadud | is sphinx open source? |
| 16:06.44 | BSD_Tech | yes |
| 16:07.11 | salviadud | well, i was thinking festival cause i was only thinking output |
| 16:07.41 | BSD_Tech | joeb depends on your needs and how much you can afford |
| 16:07.45 | salviadud | say, your estore made a sell, you make asterisk send you a phone call or something |
| 16:07.51 | BSD_Tech | but a tdm card |
| 16:08.29 | BSD_Tech | and if your in very basic need get 6 grandstream gxp2000 |
| 16:08.41 | BSD_Tech | else look at polycom |
| 16:09.03 | BSD_Tech | the server should be atleast a p3 1gz with 512 megs ram |
| 16:09.22 | BSD_Tech | and a 20 gig hd for voicemail storage |
| 16:09.42 | BSD_Tech | unless yur going to set limits on how many vm they can store |
| 16:10.07 | BSD_Tech | a voip router would be a + make sure it has qos in it |
| 16:10.37 | BSD_Tech | and if you can seperate you voip from your pc network your better off |
| 16:10.57 | *** join/#asterisk |Vulture| (n=|Vulture@136.246.189.72.cfl.res.rr.com) |
| 16:11.38 | BSD_Tech | but you need to do alot of research to meet your exact needs |
| 16:12.20 | BSD_Tech | bbl dr apt |
| 16:12.58 | [TK]D-Fender | ~gs |
| 16:13.01 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 16:13.01 | anonymouz666 | using cdr_mysql can I save into two different tables? |
| 16:13.38 | anonymouz666 | dbname=blah1,blah2 |
| 16:13.45 | anonymouz666 | it works? |
| 16:14.54 | *** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
| 16:14.54 | salviadud | i use spa 3000, is that junk? |
| 16:15.23 | [TK]D-Fender | salviadud: the FXO sometimes is flakey. Very "smart" device overall.... |
| 16:15.41 | salviadud | yea, it gets r done |
| 16:15.43 | [TK]D-Fender | salviadud: If you don't get hit with gain/echo problems, it works great |
| 16:17.22 | *** join/#asterisk FreezeS (n=bla@82.77.201.227) |
| 16:17.42 | FreezeS | hey guys |
| 16:17.51 | FreezeS | is there a problem with voip-info.org ? |
| 16:18.43 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 16:18.54 | DRoBeR | I "can't"/"don't know how to" turn on the asterisk-gui. I download it via SVN, compiled and "make install"ed ;). made the samples and check config. It prints the URI where I suppose to connect but there's no socket listening. I restarted asterisk service thinking that it would help me. :/ Can any one tell me how can I turn it on for trying it? Thanks. |
| 16:20.04 | Putzz | omg |
| 16:20.13 | Putzz | kick it it will work |
| 16:20.52 | DRoBeR | My apologies about my English. ^^ |
| 16:21.32 | Putzz | DRoBeR: read the topic pls |
| 16:21.44 | Qwell[] | Putzz: what about it? |
| 16:21.51 | DRoBeR | Wops, I didn't see the first channel. :S |
| 16:21.57 | DRoBeR | Thank you very much, Putzz. |
| 16:23.49 | Putzz | as per topic: Other fun channels: #asterisk-gui....Join #freepbx for freepbx/#trixbox for trixbox support. (GUI) |
| 16:24.16 | Qwell[] | well, it's neither freepbx or trixbox |
| 16:24.29 | Putzz | well gui |
| 16:24.43 | DRoBeR | I red it, Putzz. Thanks again. |
| 16:25.05 | Putzz | DRoBeR: dont mind me I guess I must be going coocoo |
| 16:25.49 | dacter | question: is it possible to define a channel without/before creating a dialplan? |
| 16:26.11 | [TK]D-Fender | dacter: Naturally, yes |
| 16:26.27 | [TK]D-Fender | dacter: Typically it won't GET you anywhere |
| 16:26.36 | [TK]D-Fender | dacter: but its all part of the job |
| 16:27.01 | *** part/#asterisk DRoBeR (n=DRoBeR@212.145.188.221) |
| 16:32.11 | MrWup | is there any easy way to check whether any of the 9 channels on a sip phone are being used? |
| 16:33.21 | penguinFunk | sip show channels |
| 16:34.08 | MrWup | i mean from a php app |
| 16:36.37 | kaldemar | use sip show channels from the php app. via manager interface for example. |
| 16:39.53 | *** join/#asterisk saftsack (n=saftsack@pD9E06549.dip.t-dialin.net) |
| 16:41.56 | *** join/#asterisk murdmath (n=vircuser@mail.kimballequipment.com) |
| 16:41.59 | murdmath | Howdy all. |
| 16:42.36 | murdmath | Is there a way to have a different ring tone when a parked call rings back? |
| 16:42.51 | Qwell[] | murdmath: ring tone is phone dependent |
| 16:43.41 | *** join/#asterisk NOT_guru (n=chatzill@24-241-103-142.static.stls.mo.charter.com) |
| 16:44.09 | *** join/#asterisk dlynes_laptop (n=dlynes@d207-216-161-56.bchsia.telus.net) |
| 16:45.21 | murdmath | Qwell: I can tell my phone which ringtone to use in a dial plan. Is there a way to call a specific dial plan when asterisk rings back to the phone... I'm not sure it that makes sense. |
| 16:47.34 | mvanbaak | murdmath: depends on what phone you use |
| 16:47.42 | mvanbaak | some use the _ALERT_INFO |
| 16:47.46 | mvanbaak | you can try that |
| 16:48.23 | murdmath | I would need to put that in a dial plan correct? |
| 16:48.29 | *** join/#asterisk dimas (n=ds@81.18.135.125) |
| 16:48.35 | murdmath | what part of the dial plan is called when a ringback happens? |
| 16:49.00 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.169.195) |
| 16:49.06 | mvanbaak | I have no idea |
| 16:49.14 | murdmath | That is my main question. |
| 16:49.18 | Strom_M | what do you mean "ringback"? |
| 16:49.35 | mvanbaak | Strom_M: my guess, timout in park |
| 16:49.44 | murdmath | When a call rings back after being parked. |
| 16:49.48 | mvanbaak | s/timout/timeout |
| 16:50.09 | Strom_M | and the award for "complete misuse of the word 'ringback'" goes to... |
| 16:50.21 | dlynes_laptop | Are the zaptel drivers capable of setting and retrieving CALLERID(num) and CALLERID(name) on DID multiline trunks for TDM400P, TDM2400P, A200, and A400 cards? |
| 16:50.30 | murdmath | Me. |
| 16:50.32 | murdmath | :) |
| 16:50.37 | Qwell[] | dlynes_laptop: I don't see why not |
| 16:50.43 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 16:50.59 | murdmath | The receptionist calles it ring back.. sorry. |
| 16:51.03 | dlynes_laptop | Qwell[]: ok, just wanted to make sure it wasn't just PRI's and BRI's that it was restricted to |
| 16:51.12 | Strom_M | setting for outbound calls is going to be tough, dlynes_laptop |
| 16:51.18 | dlynes_laptop | Strom_M: why? |
| 16:51.19 | Strom_M | because you CAN'T DO THAT ON ANALOG CIRCUITS |
| 16:51.27 | dlynes_laptop | Strom_M: Yes, you can |
| 16:51.38 | dlynes_laptop | Strom_M: It's called a DID trunk |
| 16:51.55 | Strom_M | well...coat me in pralines and call me jesus :) |
| 16:52.01 | mvanbaak | some countries dont support CALLERID(name) on landlines |
| 16:52.12 | dlynes_laptop | Strom_M: in Canada, they're tariffed, and in BC/Alberta it's a 4 line minimum |
| 16:52.33 | mvanbaak | like .nl |
| 16:52.33 | Strom_M | so how do you encode the caller ID information when setting up a call to the telco? |
| 16:52.35 | mvanbaak | brb |
| 16:52.50 | dlynes_laptop | mvanbaak: some LECs don't support CALLERID(name) on landlines or PRIs |
| 16:53.03 | dlynes_laptop | mvanbaak: Group Telecom/Bell doesn't support it, but Telus does |
| 16:53.31 | dlynes_laptop | Strom_M: Set(CALLERID(num)=6041234567) ; Set(CALLERID(name)=Joe Blow) ; |
| 16:53.44 | Strom_M | no, i meant on the physical circuit itself |
| 16:53.48 | [TK]D-Fender | dlynes_laptop: I'm guessing the only way they could do that is inband DTMF before bridging the call at which point you'd have to make a small IVR to process |
| 16:53.52 | Strom_M | what's the protocol? |
| 16:53.56 | dlynes_laptop | Strom_M: dtmf |
| 16:54.03 | Strom_M | how do you encode name in dtmf? |
| 16:54.13 | coppice | morse |
| 16:54.16 | *** part/#asterisk silentfury (i=anubis@CPE0001292d787f-CM000f9f5011d8.cpe.net.cable.rogers.com) |
| 16:54.20 | [TK]D-Fender | Strom_M: anything is encodable. |
| 16:54.37 | [TK]D-Fender | Strom_M: What nubar would attempt this and how is another matter ;) |
| 16:54.38 | Strom_M | i'm not doubting that it's possible; i'm just curious exactly how it's done |
| 16:55.31 | coppice | SMS works well enough |
| 16:56.23 | PioneerVM | anyone know how to pass the Incoming # called to a AGI script? |
| 16:56.23 | PioneerVM | It's apparently not available in the supplied variables to the script |
| 16:56.55 | dlynes_laptop | Strom_M: http://www.watsoncard.com/help/did.htm |
| 16:57.06 | justdave | is there a way for a sip registration to pull its password from an external source? (say an LDAP server?) |
| 16:58.38 | dlynes_laptop | Strom_M: Here's another article: http://resource.intel.com/telecom/support/gammalink/techtips/diddtmf.htm |
| 16:59.16 | Strom_M | dlynes_laptop: the first one doesnt even come close to answering my question |
| 16:59.23 | *** join/#asterisk pfn (n=pfnguyen@netblock-66-245-252-239.dslextreme.com) |
| 16:59.34 | MrChimpy | hmm. should I be able to tell the difference between a call to a busy line and a call to a non existant line on an E1? I'm just getting NO ANSWER reason code 0 for both. |
| 16:59.43 | justdave | (people registering to me I mean) |
| 16:59.43 | dlynes_laptop | Strom_M: no, but I thought it would explain what a did trunk was, for you |
| 16:59.51 | *** join/#asterisk ingenio (n=ingenio@12-216-99-16.client.mchsi.com) |
| 16:59.59 | Strom_M | I know what a DID trunk is |
| 17:00.28 | Strom_M | I just wasn't aware that there was a way to set your caller ID on calls /to/ the telephone company, and so therefore I'm asking you how it works |
| 17:01.06 | kippi | hey |
| 17:01.09 | ingenio | so I'd like to set up a small PBX for my business's two POTS lines. is this something asterisk might be used for? |
| 17:01.16 | Strom_M | ingenio: yes |
| 17:01.18 | kippi | how can I list registered extensions on the switch? |
| 17:01.26 | dlynes_laptop | Strom_M: I just finished talking to our man in charge at Telus for wholesale operations; he informed me that we would be able to set it, including the name |
| 17:01.41 | Strom_M | dlynes_laptop: on a per-call basis? |
| 17:01.55 | dlynes_laptop | Strom_M: yes |
| 17:01.58 | Strom_M | neat |
| 17:02.03 | Strom_M | but...how? |
| 17:02.18 | dlynes_laptop | Strom_M: I care more about whether I can...not how |
| 17:02.21 | murdmath | Strom_M: You need to make sure your PRI is set for NI2 also. |
| 17:02.34 | dlynes_laptop | murdmath: these are analog lines, not pri's |
| 17:02.45 | ingenio | Strom_M: is there anyway you can check out my (basic) needs? i posted on the asterisk forum but got no response.. i'd truly appreciate it |
| 17:02.47 | PioneerVM | I'm trying to have an AGI script answer "all" phone numbers but be able to do something different depending on which # was called -- anyone know how to pass the dialed # to the AGI script? |
| 17:03.03 | murdmath | PioneerVM Yes. |
| 17:03.06 | Strom_M | dlynes_laptop: well, finding out "how" will answer whether you can |
| 17:03.13 | Strom_M | ingenio: sure |
| 17:03.18 | PioneerVM | it doesn't seem to be passed to the script |
| 17:03.20 | ingenio | Strom_M: http://forums.digium.com/viewtopic.php?p=50562 |
| 17:03.23 | PioneerVM | how do i access it? |
| 17:03.31 | murdmath | PioneerVM: one sec |
| 17:03.48 | Strom_M | ingenio: dead easy |
| 17:03.57 | [TK]D-Fender | ingenio: Yes, all doable |
| 17:04.01 | Strom_M | how many telephone sets do you need? |
| 17:04.02 | ingenio | fantastic! |
| 17:04.04 | ingenio | just two |
| 17:04.12 | Strom_M | two options then: |
| 17:04.32 | Strom_M | either a TDM22B and two analog phones, or a TDM02B with two voip phones |
| 17:04.58 | ingenio | ok, i'll look into those right now! |
| 17:05.02 | ingenio | thank you very much |
| 17:05.14 | MrWup | gh |
| 17:05.39 | MrWup | hh |
| 17:06.19 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 17:06.38 | MrWup | kaldemar, sip show channels from the php script? |
| 17:06.43 | MrWup | how would i get the output? |
| 17:07.45 | murdmath | PioneerVM: Where can I paste stuff? |
| 17:07.53 | PioneerVM | dunno |
| 17:07.56 | kaldemar | MrWup: i've used it from a perl agi with a telnet connection. i don't know how to grab it with php. |
| 17:07.56 | Nugget | telnet is eeeeeeevil! |
| 17:07.56 | PioneerVM | pastbin.ca? |
| 17:08.03 | PioneerVM | never used it but someone told me to use that the other day |
| 17:08.17 | kaldemar | MrWup: uhh, cgi, not agi. |
| 17:08.19 | *** join/#asterisk neverblue (n=profx@unaffiliated/neverblue) |
| 17:08.39 | murdmath | PioneerVM: Thanks.. I tried pastebin.com... not good. Any way here you go. http://pastebin.ca/477406 |
| 17:08.42 | neverblue | can a .gsm file use an _ (underscore) in its filename? |
| 17:08.53 | PioneerVM | looking thx |
| 17:10.06 | murdmath | PioneerVM: That is for a speed dial that uses a mysql db. |
| 17:11.07 | PioneerVM | murd: this seems like it is for extension after the system is reached |
| 17:11.11 | PioneerVM | but not the incoming # |
| 17:11.21 | PioneerVM | I am trying to access the original # the user dialed |
| 17:11.24 | justdave | so I'm trying to compile Asterisk 1.4.4.... menuselect is telling me res_snmp has an unmet dependency of netsnmp. But netsnmp is really installed (as well as the related header files). What's it actually looking for to tell if it's installed or not? |
| 17:11.29 | PioneerVM | so if you called my system at 1-203-456-7890 |
| 17:11.36 | PioneerVM | i want the script to be able to see that # |
| 17:11.45 | Qwell[] | neverblue: yes, it can |
| 17:11.53 | murdmath | PioneerVM: So you want to see the number the user called to get to you. |
| 17:11.57 | PioneerVM | yes |
| 17:12.00 | neverblue | thanks Qwell |
| 17:12.09 | Strom_M | PioneerVM: do you have DNIS? |
| 17:12.10 | PioneerVM | but i think its being passed to me as "unknown" |
| 17:12.35 | PioneerVM | the extension.conf file can see the phone #, since i can use a pattern to do something based on it |
| 17:12.41 | PioneerVM | but the agi script cant see it for some reason |
| 17:13.01 | PioneerVM | Strom: not sure |
| 17:13.17 | PioneerVM | im using voicepulse and in extensions.conf I can use _XX. to match and send to script or _1xxxyyyzzzz to match the # |
| 17:13.19 | PioneerVM | both work |
| 17:13.37 | PioneerVM | but the script doesnt get the # passed, dnid and rdnis come up "unknown" in the passed info |
| 17:13.37 | MrWup | hmf |
| 17:13.46 | MrWup | theres an AGI function channel status: |
| 17:13.49 | MrWup | but thats pretty useless |
| 17:13.52 | PioneerVM | lol |
| 17:14.06 | MrWup | i need asterisk to report to me the status of all channels |
| 17:14.17 | MrWup | so i can show everyone who is doing what |
| 17:14.25 | PioneerVM | the only other option is if there is a varialbe in extensions.conf which has that # i can pass it on command line to script |
| 17:14.30 | PioneerVM | but dont know what that variable is |
| 17:15.02 | Strom_M | PioneerVM: well, when you pattern match, ${EXTEN} contains whatever matched that pattern |
| 17:15.18 | PioneerVM | i tried that but that shows "s" |
| 17:15.32 | PioneerVM | maybe i have to catch it earlier |
| 17:15.43 | [TK]D-Fender | PioneerVM: think where you are when you CALL the AGI. thats your EXTEN *now* |
| 17:15.44 | jsolares | are you calling a macro? |
| 17:16.16 | [TK]D-Fender | Clearly you should be storing original exten in a variable for retreival |
| 17:17.17 | PioneerVM | testing |
| 17:19.01 | PioneerVM | that did it thanks |
| 17:19.11 | PioneerVM | surprised there is no "original number" variable automatically |
| 17:19.30 | PioneerVM | i had to set it using the __VAR format to pass on to my subsection |
| 17:19.34 | PioneerVM | thanks for the help all |
| 17:19.56 | MrWup | hmmf |
| 17:20.05 | MrWup | theres gotta be a way to get that sip show channels from PHP |
| 17:20.10 | MrWup | that would be so useful |
| 17:20.15 | MrWup | solve all my problems in no time at all |
| 17:20.40 | PioneerVM | I'm using perl, can't help you on that one sorry |
| 17:21.02 | LeddyHM | plenty of ways to do it in php |
| 17:21.14 | MrWup | LeddyHM, how? |
| 17:21.22 | LeddyHM | google search |
| 17:21.24 | MrWup | i am! |
| 17:21.28 | MrWup | and turning up nothing |
| 17:21.29 | LeddyHM | you can connect to the manager interaface |
| 17:21.40 | LeddyHM | or even something as easy as a "system" call |
| 17:21.51 | *** join/#asterisk SwK (n=SwK@65.192.110.34) |
| 17:23.27 | neverblue | my background() isnt working properly, when i enter a key, the message still plays |
| 17:23.34 | neverblue | what could cause this? |
| 17:24.28 | [TK]D-Fender | neverblue: DTMF isn't being detected properly on your channel, or you have an ignorpat that matches, or something else. Pastebin your dialplan |
| 17:24.30 | justdave | the output from configure shows it finding net-snmp-config... |
| 17:25.24 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:26.29 | shido6 | Zzzz |
| 17:29.01 | neverblue | u mean the context I am working in? |
| 17:31.26 | [TK]D-Fender | neverblue: clearly... |
| 17:31.36 | MindTheGap | ~thebook |
| 17:31.50 | jbot | extra, extra, read all about it, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:31.50 | neverblue | nah, I think i might have it |
| 17:35.35 | *** join/#asterisk etfonhomey (n=etfonhom@74-140-209-191.dhcp.insightbb.com) |
| 17:35.59 | neverblue | hmm, that wasnt it |
| 17:36.00 | MrWup | ok ive managed to connect to the manager |
| 17:36.03 | MrWup | inteface |
| 17:36.05 | etfonhomey | [TK]D-Fender, are you around? |
| 17:36.11 | [TK]D-Fender | etfonhomey: Somewhat |
| 17:36.12 | MrWup | and ive logged in. and i can make it show dialplan |
| 17:36.18 | neverblue | my dtmfmode=info, which is correct for my Grandstream |
| 17:36.19 | MrWup | but when i do sip show channels nothing happens |
| 17:36.25 | MrWup | do only some commands work? |
| 17:36.42 | etfonhomey | [TK]D-Fender, I want to get the MWI light on a 501 to work. |
| 17:37.14 | etfonhomey | [TK]D-Fender, this is a small office and they only have one voicemail box. |
| 17:37.21 | [TK]D-Fender | etfonhomey: "mailbox=[boxnumber]@[context]" for that phone's entry is all you need |
| 17:37.41 | *** join/#asterisk Blackthorn (i=blacktho@72.236.88.10) |
| 17:37.53 | PioneerVM | anyone know how to AGI script errors during a call |
| 17:38.00 | etfonhomey | [TK]D-Fender, that's what I thought. Should I see a SIP subscription for the MWI? |
| 17:38.01 | PioneerVM | i looked at asterisk -r -vvvvv but dont see "errors" |
| 17:38.16 | [TK]D-Fender | etfonhomey: Believe so |
| 17:38.25 | Blackthorn | I am trying to send a call from one * to another * server. And when placing the call I get rejected with "no authorty found" any thoughts what this means? |
| 17:38.58 | [TK]D-Fender | Blackthorn: Means your user/peer account details don't match. |
| 17:39.37 | *** join/#asterisk keulin (n=cray@AMontpellier-152-1-38-159.w81-251.abo.wanadoo.fr) |
| 17:39.55 | etfonhomey | [TK]D-Fender, In phone.cfg, what are the correct settings for: msg.mwi.x.subscribe, msg.mwi.x.callBackMode, and msg.mwi.x.callBack (the mailbox is 104@default and the VM number is 3500) |
| 17:39.55 | neverblue | http://pastebin.ca/477466 <-- there is my context [TK]D-Fender |
| 17:39.58 | Blackthorn | thanks fender |
| 17:42.25 | *** part/#asterisk ingenio (n=ingenio@12-216-99-16.client.mchsi.com) |
| 17:42.43 | [TK]D-Fender | etfonhomey: in sip.conf, not sip.cfg |
| 17:43.03 | [TK]D-Fender | etfonhomey: leave the subscribe blank in sip.cfg |
| 17:43.46 | etfonhomey | [TK]D-Fender, OK. I'll try it. Thanks. |
| 17:43.52 | [TK]D-Fender | neverblue: You should not be running IVR's off of anything except "s". Time to hit the books again. |
| 17:44.18 | [TK]D-Fender | neverblue: You also never set any timeouts. |
| 17:44.47 | neverblue | ok, so can I still us s in an outgoing context? |
| 17:44.53 | neverblue | use* |
| 17:46.03 | *** join/#asterisk d4rkst4r75 (n=d4rkst4r@ip-41-112.sn1.eutelia.it) |
| 17:46.17 | Mercestes | d4rkst4r75, sup? fixed? |
| 17:46.26 | d4rkst4r75 | i Mercestes |
| 17:46.29 | coolbeans | Has anyone tried to connect Asterisk with Microsoft's solution? |
| 17:46.33 | d4rkst4r75 | i've rebuild the cable |
| 17:46.36 | d4rkst4r75 | but no hope |
| 17:46.44 | coolbeans | don't k+b me, it's a legit question ;) |
| 17:46.58 | Qwell[] | coolbeans: yes, but Vista kept sending SIP messages asking me if I wanted it to send me the admin password |
| 17:46.59 | danp | microsoft's solution? |
| 17:47.08 | d4rkst4r75 | <PROTECTED> |
| 17:47.13 | Qwell[] | </troll> |
| 17:47.31 | coolbeans | danp: They apparently have a new phone system coming out that is part of Exchange 2007. |
| 17:47.41 | Qwell[] | "part of", heh |
| 17:47.43 | danp | oh my |
| 17:47.45 | coolbeans | Qwell: lol |
| 17:47.46 | Qwell[] | ie; an extra $500/user |
| 17:47.57 | [TK]D-Fender | neverblue: "s" is not a CONTEXT, and hos nothing to do with "outgoing". |
| 17:48.03 | shido6 | + license |
| 17:48.24 | d4rkst4r75 | Mercestes: some other ideas? |
| 17:48.26 | neverblue | [TK]D-Fender, so i can never use an s in an outgoing context? |
| 17:48.54 | shido6 | you use "s" in a context when you dont need a number to dial inside of a context :) |
| 17:49.22 | [TK]D-Fender | neverblue: "S" IS NOT A CONTEXT. IT IS A STANDARD EXTENSIONS. |
| 17:49.40 | [TK]D-Fender | neverblue: You need to go re-read all the basics on extensions & IVRS |
| 17:49.49 | neverblue | you need to cut the caps |
| 17:49.51 | shido6 | "s" is an "exten" and takes the place of a number or pattern. For example: exten => s,1,Answer .. |
| 17:49.54 | neverblue | and tone it down a bit |
| 17:50.03 | neverblue | i never said s was a context |
| 17:50.23 | anonymouz666 | Exec("SIP/6000-086c5068", "NoOP()") in new stack |
| 17:50.28 | Mercestes | lol |
| 17:50.34 | Mercestes | get'em neverblue |
| 17:50.35 | [TK]D-Fender | neverblue: An what makes a context "outgoing"? |
| 17:50.39 | anonymouz666 | Up (None) |
| 17:50.54 | neverblue | [TK]D-Fender, i think we are done at this point |
| 17:50.55 | anonymouz666 | the call is 'active' with no apps associate |
| 17:50.55 | [TK]D-Fender | neverblue: Also, how would a phone DIAL "s"? |
| 17:50.56 | anonymouz666 | d |
| 17:51.08 | anonymouz666 | the call is totally stuck |
| 17:51.12 | anonymouz666 | lol |
| 17:51.31 | [TK]D-Fender | neverblue: Maybe if you had a soft-phone using that context it might be possible. Certainly nothing else I can think of. |
| 17:51.59 | d4rkst4r75 | Mercestes: there's a way to log in a file the output of pri intense debug span 1? |
| 17:52.06 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 17:52.30 | Blackthorn | fender, i seemed to have fixed the autentication problem. I'm not getting rejected on the recieving side with "request '@banned' does not exist" I have the context setup for default though.. thoughts? |
| 17:52.31 | Mercestes | d4rkst4r75, asterisk -r > output.txt && tail -f output.txt |
| 17:52.48 | d4rkst4r75 | that's right Mercestes :) |
| 17:53.13 | Mercestes | lol That's how I do it |
| 17:53.24 | Mercestes | there is some switch you can use that strips the color codes out but, I can never remember it. |
| 17:54.03 | [TK]D-Fender | Blackthorn: You'll have to really look at both sides carefully and make sure your understanding of WHOSE context is being referred to is implemented properly |
| 17:54.36 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 17:54.38 | d4rkst4r75 | i've to do asterisk -rcn > etc. etc. |
| 17:55.03 | Mercestes | I wouldn't cat asterisk onto /etc you need that |
| 17:56.12 | danp | anyone interested in a patch to have voicemail minimum message duration account for silence (if the message was ended because of silence)? |
| 17:56.21 | d4rkst4r75 | Mercestes, giving that command i've no console |
| 17:56.30 | Mercestes | d4rkst4r75, that is correct. |
| 17:56.37 | shido6 | tired of those 4 - 7 minute long vmails ? :) |
| 17:56.40 | Mercestes | that's what the tail -f etc. etc. is for |
| 17:56.43 | neverblue | You should not be running IVR's off of anything except "s" <-- is this referring to the background() ? |
| 17:57.05 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 17:57.38 | danp | shido6: it's annoying to get messages that are just silence but it's also annoying that the reported duration (in emails and such) is really duration + maxsilence |
| 17:58.07 | shido6 | should be a way to link voicemail with that voice detection app |
| 17:58.09 | Mercestes | danp: Looking forward to it in asterisk 1.4.3 |
| 17:58.26 | Mercestes | err..asterisk 1.4.5 |
| 17:58.38 | Mercestes | zomg. your up to 1.4.4? can't you ppl get it right the first time? |
| 17:58.47 | danp | i just checked the latest 1.2 svn. is a fix for that in 1.4? |
| 17:59.18 | *** join/#asterisk red9012 (n=marc3234@206-248-174-34.dsl.teksavvy.com) |
| 17:59.26 | justdave | there's no such thing as bug-free software :) |
| 17:59.29 | [TK]D-Fender | neverblue: Yes. Background is a tool used only for IVR's which is what you were attempting to do off of 101 |
| 17:59.30 | danp | it doesn't seem so |
| 17:59.56 | Mercestes | justdave, Sure there is, just look at Windows. |
| 18:00.02 | justdave | rotflmao |
| 18:01.00 | Mercestes | </troll for now> |
| 18:01.00 | Mercestes | :) |
| 18:01.14 | neverblue | in the ~thebook, the following example is given: exten=> 123,1,Background(helo-world) (Page 84) |
| 18:01.33 | Blackthorn | if you somone has a moment take a look at http://pastebin.ca/477500 perhaps i'm just missing something |
| 18:01.36 | neverblue | so I didnt realize you could NOT do that |
| 18:01.40 | d4rkst4r75 | Mercestes: i'm dumping D-Channel to a file |
| 18:01.47 | d4rkst4r75 | so i can send it to sangoma support |
| 18:01.47 | d4rkst4r75 | :) |
| 18:02.26 | [TK]D-Fender | Blackthorn: exten => 555,1,Dial(IAX2/remote_server/${exten}@default); |
| 18:02.53 | danp | http://pastie.caboo.se/59896 -- there's the diff for 1.2 if anyone's interested |
| 18:03.07 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 18:03.22 | [TK]D-Fender | neverblue: Regrettably I am rather disappointed with the book's (rev1) description of the "s" thatndard extension, the WaitExten app (I avoid), and IVR's in general. |
| 18:03.24 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 18:03.24 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:03.31 | shido6 | here blackthorn, http://pastebin.ca/477503 |
| 18:03.41 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-152-182-238.red.bezeqint.net) |
| 18:03.56 | neverblue | well, that line lead me to this confusing situation, im sorry for your frustration :) |
| 18:04.24 | [TK]D-Fender | neverblue: s'ok |
| 18:04.38 | neverblue | we just never seem to get a long :) |
| 18:04.38 | [TK]D-Fender | neverblue: rev2 is right around the corner, and maybe they did it RIGHT this time |
| 18:04.42 | MrWup | if i have an app which connects to the asterisk management interface every second to refresh the sip channel status... would that be going too far? |
| 18:05.06 | [TK]D-Fender | neverblue: I had a slight misinterpret on something you wrote, so I'm not going blameless on it :) |
| 18:05.15 | neverblue | ha |
| 18:05.16 | danp | should work with 1.4, too, but the whitespace is off just enough to make it fail |
| 18:05.16 | neverblue | lol |
| 18:05.20 | [TK]D-Fender | neverblue: I'm a bit better now |
| 18:05.51 | Mercestes | d4rkst4r75, Sangoma support is *very* good. |
| 18:07.34 | Mercestes | oh that reminds me. Corydon-w. relaxdtmf seemed to fix my resigna....err..my dtmf problem. |
| 18:09.15 | Mercestes | ... |
| 18:11.59 | Corydon-w | MrWup: why reconnect? |
| 18:12.51 | MrWup | php cant stay alive forever |
| 18:13.56 | *** join/#asterisk dbrummer (n=dan@64.221.232.247) |
| 18:14.06 | dbrummer | good afternoon |
| 18:14.36 | MrWup | maybe ill use a delphi app instead |
| 18:14.50 | dbrummer | i was wondering if someone could help me out with a quick configuration issue i'm having |
| 18:15.30 | *** join/#asterisk jlnt104 (n=JL@70.255.193.190) |
| 18:15.45 | jlnt104 | I need help with AGI |
| 18:15.50 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 18:16.21 | jlnt104 | is anyone awake |
| 18:16.29 | dbrummer | I have a TE210P card and I was wondering what the configuration would look like for zaptel.conf and asterisk/zapata.conf |
| 18:16.49 | Blackthorn | if i use exten => 555,1,Dial(IAX2/remote_server/${exten}@default); I get authority failed... if i use exten => 555,1,Dial(IAX2/remote_server/${exten}); I get "request '@banned' does not exist"... |
| 18:17.21 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 18:17.21 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:17.41 | anonymouz666 | channel.c:2274 __ast_read: Dropping deferred DTMF digits on SIP/6000-086d1800 |
| 18:17.43 | anonymouz666 | huhuhu |
| 18:17.46 | anonymouz666 | beautiful |
| 18:19.56 | jlnt104 | any ideas on the AGI |
| 18:19.56 | Blackthorn | oh btw, thanks for fixing that tyo shido6.. truck = trunk :P |
| 18:19.56 | anonymouz666 | how can that be possible? the read() app does not recognize my damn dtmfs |
| 18:19.56 | jlnt104 | how to execute it |
| 18:19.56 | etfonhomey | Qwell, is the appliance shipping yet? |
| 18:19.57 | shido6 | :) |
| 18:19.57 | Qwell[] | etfonhomey: ask sales |
| 18:19.57 | Mercestes | anonymouz666, inband? |
| 18:19.57 | Mercestes | jlnt104, exten s,1,AGI(path/to/your/app/nameofyourapp) |
| 18:19.57 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 18:19.58 | jlnt104 | hrm lol I feel retarded |
| 18:20.01 | anonymouz666 | no |
| 18:20.01 | Mercestes | then my work here is done. :) |
| 18:20.01 | anonymouz666 | it's rfc2833 |
| 18:20.09 | *** join/#asterisk KpoH (n=AID@host-86-106-252-180.moldtelecom.md) |
| 18:20.09 | anonymouz666 | inband does not work with g729 |
| 18:20.12 | Mercestes | anonymouz666, try info |
| 18:20.19 | KpoH | hi all |
| 18:20.21 | Mercestes | or better yet, dtmfmode=auto |
| 18:20.24 | coppice | rfc2833 is so last year |
| 18:20.25 | jlnt104 | You may think I am crazy but we use Fonality so are things different with all of the code that they have changed |
| 18:20.43 | Qwell[] | jlnt104: so call them for support |
| 18:20.47 | dbrummer | how do I configure the channels in zapata.conf for a dual-port PRI card? (TE210P) |
| 18:21.05 | Qwell[] | jlnt104: You realize that what fonality sells is over 2 years old? |
| 18:21.13 | jlnt104 | no |
| 18:21.16 | jlnt104 | didn't know that |
| 18:21.19 | KpoH | i have two servers with SRV record load balanced, i want to unregister sip user from one server, how can i do it? |
| 18:21.21 | Qwell[] | based on like 1.0.9 |
| 18:21.30 | NOT_guru | jlnt104 you can also try the #trixbox and or #freepbx channels |
| 18:21.55 | Qwell[] | jlnt104: personally, I'd cut your losses, and just move to open source asterisk, or BE |
| 18:22.26 | anonymouz666 | asterisk appliance is produced in USA? |
| 18:22.38 | jlnt104 | lol I was actually considering that |
| 18:22.39 | Qwell[] | anonymouz666: yes |
| 18:24.18 | [TK]D-Fender | NOT_guru: Fonality != FreePBX.... |
| 18:25.42 | NOT_guru | yes sir fender |
| 18:25.42 | NOT_guru | sorry |
| 18:25.59 | KpoH | peoples, how can i unregister sip peer? |
| 18:26.04 | dbrummer | i have two spans configured in zaptel.conf like so, span=1,1,0,esf,b8zs |
| 18:26.04 | dbrummer | span=2,1,0,esf,b8zs |
| 18:26.05 | dbrummer | bchan=1-23,25-47 |
| 18:26.05 | dbrummer | dchan=24,48 |
| 18:26.05 | dbrummer | , what would my zaptel configuration be like? |
| 18:26.09 | Mercestes | 1.0.9 is a reallly good release. |
| 18:29.20 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 18:29.43 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 18:29.49 | coppice | that will be on the greatest hits CD |
| 18:30.01 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 18:30.17 | dbrummer | anyone familiar with zapata.conf at all ? |
| 18:30.32 | bkruse_home | coppice: totally |
| 18:32.04 | Mercestes | dbrummer, I think you should have 1,1 and 2,2 if I understand it correctly but I could be completely off. |
| 18:32.24 | Blackthorn | i'm reall confused... Dial(IAX2/remote_server/${exten}@local); the reciving server rejects with thers no context local. but when i change local to default. I get failed to authenticate. Even though default is the correct context |
| 18:32.31 | Mercestes | dbrummer, the rest looks ok |
| 18:32.51 | *** join/#asterisk d4rkst4r75 (n=d4rkst4r@ip-41-112.sn1.eutelia.it) |
| 18:33.56 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 18:33.56 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:34.55 | dbrummer | Mercestes: I believe the second value in the span line is for timing |
| 18:35.06 | Mercestes | dbrummer, correct |
| 18:35.29 | dbrummer | Mercestes: would I need seperate timing for the different spans? |
| 18:35.49 | dbrummer | I believe my span configuration is good, I'm just having issues with the asterisk zapata.conf |
| 18:35.49 | Mercestes | dbrummer, that could be only for PRI's but as I understand it otherwise...yes. |
| 18:36.05 | dbrummer | I can only get 1-23 channels, I dont know how to configure the other channels on the second span |
| 18:36.34 | Mercestes | same as the first 23 channels. Just start with 25 and end with 47 |
| 18:37.00 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 18:37.16 | dbrummer | K, it gave me an error but Ill try again |
| 18:37.44 | Mercestes | do you wish to append span 2 onto span 1? |
| 18:37.49 | dbrummer | ERROR[25514]: chan_zap.c:10470 setup_zap: Unable to reconfigure channel '25-47' |
| 18:37.57 | dbrummer | i want them serperate |
| 18:37.58 | Mercestes | that error is very helpful |
| 18:38.46 | Mercestes | dbrummer, oh, then you just make a new group, a new context, and a new channel def and don't set anything for anything yo udon't want to change. |
| 18:39.25 | Mercestes | so if everything is the same as span 1 then you'd only really need like....3 lines. group=2 context=span2 channel -> 25-47 |
| 18:39.39 | dbrummer | ok |
| 18:39.48 | dbrummer | would it be ok to use the same context for both ? |
| 18:39.58 | Mercestes | not if you want to seperate them. but , sure it'd be ok |
| 18:40.10 | Mercestes | in that case you only need 2 lines |
| 18:40.25 | Mercestes | zapata.conf inherits previous values until reassigned |
| 18:41.11 | Mercestes | and i fyou want them to be "together" for dialing out, then you only need one line |
| 18:41.31 | Mercestes | or you could add 0 lines and just declare channel => 1-23,25-47 |
| 18:42.47 | *** join/#asterisk ZaVoid (n=zavoid@c-71-225-254-71.hsd1.pa.comcast.net) |
| 18:42.50 | ZaVoid | hello |
| 18:42.56 | Mercestes | 'ello |
| 18:43.00 | dbrummer | hmmm, it's still giving me errors |
| 18:43.14 | Mercestes | pastebin |
| 18:43.35 | dbrummer | same error as above |
| 18:43.36 | Mercestes | and check your ztcfg |
| 18:43.43 | dbrummer | it doesnt like the channel => 1-23,25-47 |
| 18:43.49 | Mercestes | s/pastebin/pastebin your configuration files/ |
| 18:44.09 | Mercestes | then your spans 25-47 are not up |
| 18:44.14 | dbrummer | zaptel.conf |
| 18:44.16 | dbrummer | span=1,1,0,esf,b8zs |
| 18:44.16 | dbrummer | span=2,1,0,esf,b8zs |
| 18:44.16 | dbrummer | bchan=1-23,25-47 |
| 18:44.16 | dbrummer | dchan=24,48 |
| 18:44.21 | Mercestes | ... |
| 18:44.26 | dbrummer | zapata.conf |
| 18:44.28 | Mercestes | PASTE BIN |
| 18:44.28 | dbrummer | [channels] |
| 18:44.28 | dbrummer | switchtype=national |
| 18:44.28 | dbrummer | context=default |
| 18:44.28 | dbrummer | signalling=pri_cpe |
| 18:44.28 | dbrummer | group=1 |
| 18:44.29 | Mercestes | ~pb |
| 18:44.54 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 18:44.54 | dbrummer | channel => 1-23 |
| 18:44.54 | ZaVoid | anyone know if i can disable g.723r63 and allow only g.723r53 |
| 18:44.54 | *** mode/#asterisk [+b %dbrummer!*@*] by Corydon-w |
| 18:48.31 | *** join/#asterisk freshfruit (n=j@202.162.43.1) |
| 18:49.03 | Mercestes | funny. his zaptel.conf is wrong |
| 18:49.10 | freshfruit | hi all |
| 18:49.16 | Mercestes | I think |
| 18:49.36 | Mercestes | mebbe not. Hi freshfruit |
| 18:49.56 | freshfruit | hi mercestes |
| 18:50.09 | errr | my asterisk crashed and dumped a core, http://fluxbox.pastebin.ca/477576 this is what was on the screen.. any ideas what could have caused this?? |
| 18:50.31 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 18:50.34 | anonymouz666 | what the variable who controls the t extension? |
| 18:50.40 | anonymouz666 | absoluttimeout |
| 18:50.41 | anonymouz666 | ? |
| 18:50.43 | anonymouz666 | timeout? |
| 18:50.48 | Qwell[] | either |
| 18:50.58 | Qwell[] | any timeout will go to t |
| 18:51.44 | anonymouz666 | I am typing the dtmfs.. read() recognize but before jump to next priority it timeouts and jumps to t extension |
| 18:52.44 | neverblue | if I wanted to call number (landline) 123-4567 from my ext 890, how would I do that with Originate in the AM? (the API on voip-info isnt enough :( ) |
| 18:52.58 | neverblue | using SIP, sorry, forgot to add that |
| 18:53.03 | jer | what am i doing wrong if i'm getting wrong password for invite for each time someone tries to call me? (setup is asterisk A to asterisk B (B is a client, no IAX set up; just SIP trunk)) |
| 18:53.10 | red9012 | chan_agent module has some bugs. |
| 18:53.22 | Mercestes | neverblue, you set up an account with a SIP Provider |
| 18:53.45 | red9012 | if you are in privacy mode, sometimes the dtmf are not registered (ie press 1 to accept, 2 to refuse) |
| 18:54.34 | neverblue | Mercestes, I beleive i have that part done, then I need to do Action: login |
| 18:54.35 | Mercestes | neverblue, Then you would use gotoiftime(*|*|00:01-11:59). |
| 18:55.05 | Mercestes | neverblue, Then you would have something like 890,1,Dial(SIP/1234567@provider) |
| 18:55.06 | neverblue | sorry, Mercestes I was referring to Action: Originate |
| 18:55.26 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 18:55.39 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 18:57.13 | Mercestes | neverblue, I dunno then. I always just use Dial. |
| 18:57.51 | neverblue | Mercestes, i am using php to communicate with the Asterisk Manager |
| 18:58.02 | Mercestes | neverblue, ok. |
| 18:58.52 | *** part/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 18:58.57 | Mercestes | neverblue, Have you looked at Flash Operator Panel? |
| 19:03.47 | neverblue | googling now :) |
| 19:05.10 | Mercestes | neverblue, It might have some useful source to look at. There is also some .net manager examples. |
| 19:05.22 | Mercestes | can't remember the name off the top of my head |
| 19:05.25 | neverblue | whats .net :) |
| 19:05.48 | Mercestes | google microsoft .net |
| 19:06.06 | neverblue | i wonder if PERL is very similar to PHP, when it comes to communicating with the manager? |
| 19:06.24 | Mercestes | seeing as it's pretty much text matching off of a telnet port I imagine they are all pretty similar. |
| 19:06.24 | neverblue | Mercestes, I think you missed the :) at the end of my question |
| 19:06.39 | Mercestes | neverblue, no, I didn't miss it. |
| 19:07.05 | Mercestes | just, decided I'd take the safe route and just answer. :) |
| 19:10.09 | freshfruit | what part of asterisk configuration that control user registration to server ? |
| 19:10.21 | Qwell[] | freshfruit: all of it |
| 19:10.28 | freshfruit | i mean |
| 19:10.36 | Mercestes | lol |
| 19:10.47 | Mercestes | sip.conf and iax.conf |
| 19:10.48 | freshfruit | my sip account still can make a call |
| 19:10.50 | [TK]D-Fender | freshfruit: You're probably thinking of sip.conf |
| 19:10.59 | freshfruit | allthough is not registered |
| 19:11.29 | freshfruit | is unregistered from server |
| 19:11.35 | freshfruit | but still can make a call |
| 19:11.58 | freshfruit | it is unregistered from server |
| 19:12.00 | freshfruit | but still can make a call |
| 19:12.58 | freshfruit | what wrong with my sip.conf ? |
| 19:18.56 | *** part/#asterisk rokjan (n=jj2@static-200-105-156-114.acelerate.net) |
| 19:20.21 | [TK]D-Fender | freshfruit: How would we know? You haven't SHOWN us anything |
| 19:20.26 | [TK]D-Fender | freshfruit: pastebin it. |
| 19:20.28 | [TK]D-Fender | ~pb |
| 19:20.41 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 19:20.41 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 19:20.53 | *** join/#asterisk Dimik_ (n=Dimik_@unaffiliated/dimik) |
| 19:21.54 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 19:22.23 | d4rkst4r75 | Mercestes |
| 19:22.28 | d4rkst4r75 | are you still there? |
| 19:25.25 | Mercestes | yes |
| 19:25.25 | *** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com) |
| 19:25.26 | d4rkst4r75 | Mercestes: http://www.pastebin.ca/477618 |
| 19:25.59 | d4rkst4r75 | Mercestes: that's the log of D-Chan when it goes down |
| 19:26.15 | Mercestes | d4rkst4r75, what did sangoma say? |
| 19:26.21 | d4rkst4r75 | nothing yet |
| 19:26.29 | d4rkst4r75 | they didn't answer to my email |
| 19:28.01 | Mercestes | looks like remote just stops responding |
| 19:28.16 | d4rkst4r75 | yes... |
| 19:28.36 | d4rkst4r75 | but i've no reason for that :( |
| 19:29.12 | Mercestes | maybe it's mad at you |
| 19:30.01 | *** join/#asterisk flambers (n=flambers@c9343fd2.virtua.com.br) |
| 19:31.17 | anonymouz666 | oh in some cases read() does not recognize the DTMFs.. :~ |
| 19:33.38 | Mercestes | anonymouz666, pulse or tone? |
| 19:34.40 | anonymouz666 | [May 8 15:34:20] WARNING[31413]: channel.c:2274 __ast_read: Dropping deferred DTMF digits on SIP/6000-089028d8 |
| 19:34.57 | Mercestes | try relaxdtmf |
| 19:36.43 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 19:37.00 | Mercestes | anonymouz666, are you calling "answer" first? |
| 19:37.10 | anonymouz666 | its read() |
| 19:38.16 | dacter | sip client to sip client call. woot. |
| 19:38.37 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 19:38.37 | *** mode/#asterisk [+o mog] by ChanServ |
| 19:40.00 | Mercestes | http://bugs.digium.com/view.php?id=14 |
| 19:40.36 | Hmmhesays | I wish there was a way to populate openser's usrloc database with asterisk registration ip addresses |
| 19:40.38 | Hmmhesays | damnit |
| 19:41.14 | Mercestes | ok, the only instances I'm seeing for "Deferred DTMF digits' is in source code, maybe it's your setup anonymouz666 |
| 19:41.45 | Mercestes | anonymouz666, pastebin |
| 19:44.19 | *** join/#asterisk SwK (n=SwK@65.192.110.34) |
| 19:47.01 | Hmmhesays | could someone fax me something please? |
| 19:50.38 | anonymouz666 | Mercestes: I am in trouble i have a read() inside a while loop() it works on the first... but 2 or 3 times the read always print user entered nothing |
| 19:52.17 | anonymouz666 | after I start to press lots of digits i got that warning |
| 19:53.02 | d4rkst4r75 | Mercestes: are you using sangoma boards? |
| 19:53.53 | Mercestes | d4rkst4r75, Just one. |
| 19:53.57 | Mercestes | anonymouz666, pastebin |
| 19:54.08 | d4rkst4r75 | Mercestes, a10x ? |
| 19:54.30 | xkev | hmmhesays: http://www.tpc.int/sendfax.html |
| 19:54.36 | Mercestes | a104d, yea |
| 19:54.41 | xkev | http://www.tpc.int/verify.html <- check you're in there |
| 19:55.44 | d4rkst4r75 | when install wanpipe, have you error when patching zaptel? |
| 19:55.48 | Mercestes | i followed the directions. |
| 19:56.01 | d4rkst4r75 | can you help me with a link? |
| 19:56.34 | d4rkst4r75 | Mercestes, also asterisk+zaptel+lipri versions? |
| 19:56.49 | Mercestes | d4rkst4r75, http://www.google.com/search?hl=en&q=sangoma+install+guide |
| 19:57.25 | Mercestes | d4rkst4r75, 1.2.13, 1.2.17, 1.2.3-r1 |
| 19:57.31 | xkev | hmmhesays, when checking, be sure to use spaces as shown in the format example |
| 19:57.39 | Mercestes | s/, /+/ |
| 19:58.03 | Mercestes | meh |
| 19:58.07 | Qwell[] | you lose |
| 19:58.12 | Mercestes | I lose. |
| 19:58.12 | d4rkst4r75 | Mercestses: asterisk 1.2.13, zaptel 1.2.17, libpri 1.2.3-r1 ? |
| 19:58.14 | Mercestes | s/, /+/g |
| 19:58.15 | xkev | erm, or maybe not.. anyway |
| 19:58.34 | Mercestes | d4rkst4r75, In that order |
| 19:58.41 | d4rkst4r75 | ko |
| 19:58.42 | d4rkst4r75 | ok |
| 19:58.49 | d4rkst4r75 | i would like to reinstall all the stuffs |
| 19:59.05 | d4rkst4r75 | i think it could be a wrong version of the software |
| 19:59.23 | Mercestes | then I would start with a little cat /dev/urandom > wahtever your harddrive is. |
| 19:59.38 | d4rkst4r75 | LOOL |
| 19:59.39 | Mercestes | good way to reinstall everything. :D |
| 19:59.56 | d4rkst4r75 | i don't think that "everything" is a good choice |
| 20:00.09 | Mercestes | oh. then I don't think urandom would be a good chioce. |
| 20:00.33 | Mercestes | Maybe you could try waiting for Sangoma? |
| 20:00.36 | Mercestes | di dyou call them? |
| 20:00.58 | d4rkst4r75 | i have time to do other tests while sangoma reply |
| 20:01.07 | d4rkst4r75 | i would like to test with other software versions to be sure |
| 20:01.22 | d4rkst4r75 | can you give me your wanpipe version? |
| 20:01.43 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 20:02.39 | Mercestes | 2.3.4-7 |
| 20:03.21 | slmnhq | I am trying to build a calling card service to provide consumers (in the US) the ability to dial international numbers |
| 20:04.25 | slmnhq | voipjet, voicepulse, offer PSTN termination rates of about 1-2 c/min in the US, and 5-10 c/min for numbers in Asia |
| 20:04.40 | slmnhq | (for example) |
| 20:05.04 | slmnhq | Alternatively, I could purchase bulk minutes from a Telco, like Verizon |
| 20:05.48 | slmnhq | Telco's require annual contracts |
| 20:05.54 | d4rkst4r75 | thk Mercestes |
| 20:05.55 | *** join/#asterisk jart (n=user@ool-43509aa5.dyn.optonline.net) |
| 20:06.03 | Mercestes | np |
| 20:06.05 | slmnhq | Any suggestions which is a more cost effective solution to use? |
| 20:06.16 | jart | does anyone know a voip service provider besides Level 3 that offers TCP SIP? |
| 20:07.24 | Mercestes | jart: CBeyond maybe? |
| 20:08.12 | Mercestes | slmnhq, depends on your usage. You could take advantage of those "1.2 cents a minute anywhere" telcos and run them out of business. |
| 20:08.26 | jart | any that don't require a contract? |
| 20:09.09 | d4rkst4r75 | Merces: libpri 1.2.17 |
| 20:09.11 | d4rkst4r75 | ? |
| 20:09.28 | slmnhq | Mercestes, so you're saying the Telcos will probably dole out rates based on geographical termination with in the US |
| 20:09.32 | slmnhq | ? |
| 20:09.33 | *** join/#asterisk vykarian (n=stefano@server.pennacchi.com.br) |
| 20:09.33 | Mercestes | no. 1.2.3-r1 |
| 20:09.45 | vykarian | hi all |
| 20:10.12 | vykarian | does someone knows a asterisk+skype addon/plugin that not the chanskype? |
| 20:10.14 | d4rkst4r75 | Mercestes: i asked: Mercestses: asterisk 1.2.13, zaptel 1.2.17, libpri 1.2.3-r1 ? ? |
| 20:11.02 | shido6 | is that the vmware fix? |
| 20:11.04 | Mercestes | slmnhq, where did I say that? But, basically, at a cost level, it's more expensive to call certain areas (Mexico has about 7 different cost "bands" alone) so yoru cost varies from region to region. Some Telcos try to create a flat rate that simply averages those costs, terminating below cost in some areas while "making up for it" in other areas. |
| 20:11.18 | shido6 | no.... i mean the vnc fix |
| 20:11.23 | Mercestes | slmnhq, if you only use the mfor expensive areas then you are always below cost, giving you temporary cheap terminatino until you drive them out of business. |
| 20:11.49 | *** join/#asterisk last1 (n=dood@86.34.213.191) |
| 20:12.02 | slmnhq | Mercestes, I see |
| 20:12.04 | Mercestes | d4rkst4r75, yes, yes you did. Then you: <d4rkst4r75> Merces: libpri 1.2.17 |
| 20:12.05 | Mercestes | ? and I corrected you. |
| 20:12.05 | shido6 | vykarian, is that the vnc fix ( skype ) ? |
| 20:13.31 | d4rkst4r75 | ok, thank you and sorry for my pedant :) |
| 20:14.13 | Mercestes | ah, nice word |
| 20:14.17 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 20:14.22 | *** join/#asterisk henryv2 (n=henry_vo@82-39-113-10.cable.ubr03.newy.blueyonder.co.uk) |
| 20:14.32 | slmnhq | I'm guessing that voipjet, freeworlddialup, etc that provide PSTN termination have negotiated some rates with their own telco for termination voip calls? |
| 20:14.42 | vykarian | dunno about that vnc fix.. I tried to search that now.. |
| 20:15.00 | Mercestes | slmnhq, a safe assumption, yes. |
| 20:15.10 | vykarian | I wonder about a patch like that for Unicall/R2 links for use Skype (www.skype.com) like a SIP account |
| 20:15.41 | vykarian | like that: http://www.chanskype.com/ |
| 20:16.37 | Hmmhesays | yeah |
| 20:16.39 | Hmmhesays | I use that |
| 20:16.41 | Hmmhesays | it works pretty well |
| 20:16.50 | Hmmhesays | can someone in here send me a fax |
| 20:16.52 | Hmmhesays | i'm in the US |
| 20:19.23 | JT | the shype channel driver is pretty hackish |
| 20:20.01 | JT | skype |
| 20:20.45 | *** part/#asterisk jart (n=user@ool-43509aa5.dyn.optonline.net) |
| 20:22.12 | Hmmhesays | it is but it works |
| 20:22.20 | Hmmhesays | and it works well if you set it up right |
| 20:22.48 | JT | not somthing you'd use for business though |
| 20:24.42 | Hmmhesays | why not? |
| 20:25.33 | JT | .. |
| 20:25.43 | JT | because it's a toyu |
| 20:25.43 | JT | toy |
| 20:25.55 | *** join/#asterisk ManxPower (n=manxpowe@stirprop-s4-0-0-21.ndcr2.datasync.net) |
| 20:26.21 | Hmmhesays | if you set it up right it works just fine |
| 20:26.34 | Hmmhesays | I've had multiple concurrent calls running through skype without issue |
| 20:27.20 | Hmmhesays | now someone send me a fax |
| 20:27.20 | JT | why would you want to? |
| 20:27.20 | drazak | Can asterisk handle tcp/ip, if so, what do clients need? |
| 20:27.20 | drazak | JT: :o |
| 20:27.21 | Hmmhesays | people like skype |
| 20:27.21 | Hmmhesays | drazak: openser |
| 20:27.21 | drazak | Hmmhesays: what now? |
| 20:27.23 | JT | drazak: your question doesn't make much sense |
| 20:27.36 | drazak | JT: If someone doesn't have a phone, but has internet, how can they get on an asterisk server? |
| 20:27.39 | Hmmhesays | asterisk cannot handle sip over tcp/ip |
| 20:27.44 | Hmmhesays | openser can convert it for you |
| 20:27.53 | JT | drazak: SIP or IAX2 |
| 20:27.55 | drazak | okay, openser seems not to be in portage |
| 20:27.57 | JT | uses UDP |
| 20:28.09 | JT | you do not need openser for what you want |
| 20:28.09 | drazak | JT: okay, udp or tcp/ip |
| 20:28.11 | ManxPower | Asterisk supports only UDP for protocols and audio |
| 20:28.17 | JT | drazak: UDP over IP |
| 20:28.28 | d4rkst4r75 | sorry Mercestes: what's your kernel version= |
| 20:28.29 | d4rkst4r75 | ? |
| 20:28.30 | Mercestes | drazak, portage? Gentoo? |
| 20:28.35 | drazak | Mercestes: yessir |
| 20:28.36 | JT | please don't say tcp/ip, choose one :P |
| 20:28.37 | henryv2 | Can anyone recommend which of the Digium TDM800P (presumably using HPEC if there is no hardware cancellation) or the Sangoma A20004d (with hardware echo cancellation) is better? |
| 20:28.43 | Hmmhesays | sorry TCP |
| 20:28.56 | Mercestes | drazak, good man |
| 20:28.59 | JT | henryv2: hardware ec of course |
| 20:29.02 | Mercestes | drazak, Did you try the voip overlay? |
| 20:29.06 | drazak | Mercestes: nah, no need |
| 20:29.27 | Mercestes | drazak, Ah, ok, you said openser was not in portage. Was going to suggest layman. :) |
| 20:29.29 | drazak | Mercestes: we're just doing asterisk for conference calls in a couple channels, don't need too much extra stuff, ya know |
| 20:29.37 | Hmmhesays | ser and openser can both do SIP over TCP |
| 20:30.04 | JT | drazak: do you actually need to use tcp for some silly reason? |
| 20:30.10 | drazak | JT: nah |
| 20:30.14 | JT | okay |
| 20:30.15 | drazak | JT: but that's how they asked me :P |
| 20:30.24 | drazak | anywho, what are some free clients that can do that, if someone has a mic? |
| 20:30.26 | *** join/#asterisk NoCarrier (n=NoCarrie@unaffiliated/badpacket) |
| 20:30.39 | Siya | bkruse: ? |
| 20:30.45 | JT | ekiga, idefisk |
| 20:30.53 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 20:31.08 | JT | NoCarrier: nice nick |
| 20:31.22 | NoCarrier | thank you |
| 20:31.22 | ManxPower | Remember, even if SIP is using TCP, the audio is still UDP |
| 20:31.34 | ManxPower | SIP is just SIGNALLING, not AUDIO |
| 20:31.39 | *** join/#asterisk AndrewGearhart (n=chatzill@h1.39.213.151.ip.alltel.net) |
| 20:31.57 | drazak | right right |
| 20:32.05 | AndrewGearhart | hey folks, can * be used to change the number reported on CallerID? |
| 20:32.37 | ManxPower | AndrewGearhart: Yes, with some limitations |
| 20:32.54 | Mercestes | AndrewGearhart, Yea, it doesn't work against the FBI |
| 20:33.01 | drazak | Got another question, I want to get the numbers of the people that call, is there an easy way to do that? |
| 20:33.04 | AndrewGearhart | Mercestes: darn! |
| 20:33.06 | AndrewGearhart | ;-) |
| 20:33.18 | Mercestes | drazak, yea, the CDRs |
| 20:33.35 | ManxPower | Mercestes: of he could look at the call lists on his phone. |
| 20:33.40 | drazak | Mercestes: the what nows? :P |
| 20:36.54 | drazak | Mercestes: how do I get those? |
| 20:36.58 | *** join/#asterisk saftsack (n=oliver@pD9E06549.dip.t-dialin.net) |
| 20:41.07 | JT | ~thebook |
| 20:41.20 | jbot | methinks thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 20:42.39 | salviadud | who is using bitchx? |
| 20:43.16 | JT | not i |
| 20:44.04 | LeddyHM | she's sitting next to me |
| 20:44.05 | LeddyHM | but I don' |
| 20:44.14 | LeddyHM | but I don't think she's who you are referring too |
| 20:47.50 | drazak | Is calling out from an asterisk box, that has an external #, free? |
| 20:50.23 | *** join/#asterisk pifiu (n=someone@216.5.79.1) |
| 20:50.47 | pifiu | i hate to ask this, but does anyone know of any dependable IAX providers? |
| 20:51.06 | JT | drazak: read the book |
| 20:52.11 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 20:52.34 | *** join/#asterisk Fieldy (i=w35uKc8k@gentoo/contributor/Fieldy) |
| 20:54.36 | drazak | JT: I'm reading it |
| 20:55.26 | ManxPower | All ITSPs suck. |
| 20:55.44 | *** join/#asterisk Strom_M (n=strom@adsl-66-127-181-92.dsl.lsan03.pacbell.net) |
| 20:56.51 | *** join/#asterisk kiscokid (n=ron@208.106.33.66) |
| 20:57.13 | henryv2 | LeddyHM: why are you sitting next to your x bitch? :-p |
| 20:57.41 | *** part/#asterisk Strom_M (n=strom@adsl-66-127-181-92.dsl.lsan03.pacbell.net) |
| 21:01.18 | Siya | ManxPower: why? |
| 21:01.44 | d4rkst4r75 | Mercestes: recompiled all the stuffs |
| 21:01.53 | d4rkst4r75 | i hope it'll work ;) |
| 21:02.23 | ManxPower | Siya: Many reasons. You can't make much of a profit charging 1/cent/min without massive volume. The internet is not very reliable for voice. Most of the companies are way under funded. |
| 21:03.04 | Siya | ManxPower: depends on your requirements |
| 21:03.35 | ManxPower | Siya: My requirements are for it to be reliable |
| 21:03.55 | Siya | I've not had any issues so far other than discovering that my current trunk providers screen certain calls (0870. for example) |
| 21:04.23 | JT | argh "trunk" word abuse :P |
| 21:04.25 | Siya | ManxPower: I've found it to be reliable, but I'm no business user so I might not notice all the glitches |
| 21:04.41 | Siya | JT: really, enlighten me |
| 21:04.42 | MindTheGap | does attribute md5secret have the same function as secret in sip.conf? if so, are there any other crypt methods allower in sip.conf? mu users passwords are not encrypted w md5... |
| 21:04.44 | *** join/#asterisk imapfool (n=edhorton@adsl-66-137-204-217.dsl.stlsmo.swbell.net) |
| 21:04.54 | JT | it's not a trunk |
| 21:04.56 | *** join/#asterisk Greek-Boy (n=Greek-Bo@196.45.144.42) |
| 21:05.17 | Siya | ManxPower: but maybe it helps that I work for an ISP and have full controll over my DSL line... ;) |
| 21:05.24 | Siya | JT why not? |
| 21:05.42 | henryv2 | I've been using one ITSP in the UK which provide a very reliable high quality service but you can still see the lack of funding if you call them for customer service. They're great if you don't need to contact anyone though! |
| 21:05.44 | JT | it's a connection oriented protocol |
| 21:06.33 | *** join/#asterisk Strom_M (n=strom@m1e0e36d0.tmodns.net) |
| 21:06.52 | henryv2 | but then if anyone has tried calling BT/Telewest/NTL/Virgin Media then you can also see a lack of high quality customer service in traditional telcos! |
| 21:07.01 | Siya | JT: afaik a SIP trunk does not refer to 802.1Q or (cisco-ISL) but to a logical connection between two voice switches |
| 21:07.24 | *** part/#asterisk BSD_Tech (n=bsdtech@adsl-69-230-166-20.dsl.irvnca.pacbell.net) |
| 21:07.25 | JT | Siya: it's an incorrect term, coined by trixbox/freepbx |
| 21:07.35 | JT | it refers not to a logical connection |
| 21:07.43 | JT | and if it did, that connection would have multiple channels |
| 21:07.50 | Hmmhesays | god I love t38's method of error checking |
| 21:08.05 | JT | pfft 802.1q is a fairly new user of the term, wasn't refering to that |
| 21:08.07 | Hmmhesays | "THROW MORE PACKETS AT IT" |
| 21:08.13 | Siya | henryv2: if anyone can point me to a SP where support is excelent at any time then please let me know :) |
| 21:08.29 | pifiu | so one can recommend any IAX providers that are reliable? |
| 21:08.47 | Hmmhesays | voipjet is ok |
| 21:08.50 | Hmmhesays | not great, but ok |
| 21:08.57 | Siya | JT: I've heard the term for many years and long before I heard about tirxbox/freepbx/asterisk |
| 21:08.58 | LeddyHM | not nufone |
| 21:09.04 | JT | Siya: it's wrong. |
| 21:09.20 | JT | there are lots of noobs in voice |
| 21:09.29 | JT | noobs repeat what other noobs say |
| 21:09.33 | JT | s/voice/voip/ |
| 21:10.10 | imapfool | I have had great service from voicepulse and IAX with their voicepulse connect service |
| 21:10.52 | Hmmhesays | steven tyler uses broadvoice |
| 21:10.52 | Hmmhesays | lol |
| 21:10.52 | *** join/#asterisk Fieldy (n=toon@gentoo/contributor/Fieldy) |
| 21:10.52 | Siya | JT: I was referring to well payed colleagues in the Voice industry, though I stand corrected if you say so. I'm more a Networking person than a voice specialist |
| 21:11.03 | JT | s/payed/paid/ |
| 21:11.07 | pifiu | whos steven tyler? |
| 21:11.07 | pifiu | lmao |
| 21:11.08 | JT | no such word as payed :P |
| 21:11.30 | Siya | JT :) common mistake of mine |
| 21:11.35 | JT | hehe |
| 21:11.43 | ManxPower | Voice lines can be called trunks. They frequently are. A trunk is a connection that handles 1 call. |
| 21:11.56 | ManxPower | if you can do more than 1 call then it is not a trunk |
| 21:12.05 | Greek-Boy | anyone know of a wake-up-call service run on asterisk? |
| 21:12.06 | Siya | ManxPower: I'd opt for stating at least more than one call |
| 21:12.06 | JT | a sip connection is not a trunk |
| 21:12.16 | Siya | JT: agreed |
| 21:12.36 | *** join/#asterisk axisys (n=axisys@c-69-143-190-152.hsd1.va.comcast.net) |
| 21:13.38 | Siya | henryv2: which ITSP is that? afaik the UK doesn't have a lot of voip providers... |
| 21:13.44 | MindTheGap | does attribute "md5secret" have the same function as "secret" in sip.conf? if so, are there any other crypt methods allower in sip.conf? my users passwords are not encrypted w md5... |
| 21:14.56 | ManxPower | Greek-Boy: there are a zillion of them. See the Wiki |
| 21:15.39 | imapfool | does anyone have any experience / success with imap voicemail storage in 1.4? |
| 21:17.25 | Sweeper | uh, recommendations for a 4 and 8 port fxo sip device? |
| 21:17.40 | Sweeper | I've tried the audiocodes and grandstreams, but D: |
| 21:18.53 | imapfool | I had good luck with an 8 port Astribank which is, in my case, is an 8 port FXO to USB device |
| 21:19.17 | Siya | bkruse: ? |
| 21:19.22 | shido6 | can you fax with that, imapfool? |
| 21:19.40 | JT | usb >:( |
| 21:19.55 | Sweeper | mmm |
| 21:19.57 | tzafrir | JT, do I really need to give my usual reply? |
| 21:20.42 | imapfool | I have not tried. I have had poor results faxing even with a T1 card to our LEC usinf IAXMODEM and Hylafax. We just use the old fashion method. |
| 21:20.44 | JT | get a better interface like ethernet ;) |
| 21:21.09 | tzafrir | Faxing from a different Astribank should work. Faxing from a non-astribank zaptel: with our latest sync code (provided that you apply the latest little sync patch we published) |
| 21:21.26 | Sweeper | mm |
| 21:21.36 | Sweeper | what kind of proc overhead does the astribank incur? |
| 21:23.37 | JT | tzafrir: ever going to make sip gateways? |
| 21:24.24 | tzafrir | not much. From our latest profiling, Zaptel's overhead was still quite higher than ours and the USB's |
| 21:25.12 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 21:25.26 | shmaltz | tzafrir ping |
| 21:25.34 | tzafrir | here |
| 21:25.51 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 21:25.51 | shmaltz | hi, tzafrir can you translate something from hebrew for me? |
| 21:26.04 | shmaltz | it's a post on wikipedia |
| 21:26.24 | tzafrir | shmaltz, what is it? |
| 21:26.31 | shmaltz | 讗讬 讗驻砖专 住转诐 诇拽讞转 驻住拽转 诪讬讚注 砖讗驻砖专 讘拽诇讜转 诇砖讻转讘讛 讘诇讬 拽讬砖讜专 诇诪拽讜专 讗讜 诪砖讛讜 讻讙讜谉 讝讛. 讗谞讬 诪讜讚注 讛讬讟讘 诇讞讜拽讬 讝讻讜讬讜转 讛讬讜爪专讬诐 (讜讙诐 诪驻注讬诇 讘讜讜讬拽讬砖讬转讜祝) 讗讱 讗讬谉 诇讛砖转诪砖 住转诐 讘砖讬诪讜砖 讛讜讙谉 讻砖诇讗 讞讬讬讘讬诐 讜讗讬谉 诇讜 注专讱 诪讜住祝 - 讜讘诪拽专讛 讝讛 讗讬谉 诇讜 注专讱 诪讜住祝, 讘讬讬讞讜讚 诇讗讜专 讛注讜讘讚讛 砖谞讬转谉 诇砖讻转 |
| 21:26.55 | JT | wrong window? |
| 21:26.56 | shmaltz | I understand he is saying he is knowledgable about the laws |
| 21:27.07 | shmaltz | now what does 注专讱 诪讜住祝 mean? |
| 21:27.11 | shmaltz | JT, nah |
| 21:27.19 | shmaltz | just hijacked :P |
| 21:27.27 | tzafrir | shmaltz, /j #israel ? |
| 21:27.54 | *** part/#asterisk imapfool (n=edhorton@adsl-66-137-204-217.dsl.stlsmo.swbell.net) |
| 21:30.25 | Strom_M | it translates to "Boy, I sure wish I'd paid attention in hebrew school" |
| 21:32.06 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 21:37.31 | *** join/#asterisk seele_ (n=seele@dns.tennis.com.co) |
| 21:37.52 | *** join/#asterisk Crescendo_ (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 21:38.39 | seele_ | please help I'm tryin to install a sangoma AFT102D, all the modules and compilations works fine ... but my sangoma still with red led on |
| 21:39.20 | Mercestes | <PROTECTED> |
| 21:39.23 | Mercestes | damnit |
| 21:39.46 | Mercestes | I wanna see what 注专讱 诪讜住祝 means. :D |
| 21:41.26 | *** join/#asterisk swyrus (n=ss@82-42-131-250.cable.ubr08.live.blueyonder.co.uk) |
| 21:41.27 | seele_ | when I look the channels in the asterisk CLI this shows http://www.pastebin.ca/477852 |
| 21:41.38 | *** join/#asterisk kp00 (n=kp00@85stb55.codetel.net.do) |
| 21:41.41 | kp00 | hi |
| 21:42.16 | seele_ | my zaptel.conf http://www.pastebin.ca/477857 |
| 21:43.13 | kp00 | how configure inbound ... sip account? |
| 21:43.52 | seele_ | and my zapata.conf http://www.pastebin.ca/477861 |
| 21:44.11 | *** join/#asterisk Strom_C (n=strom@adsl-66-127-181-92.dsl.lsan03.pacbell.net) |
| 21:45.03 | seele_ | please help I cant make outbound calls |
| 21:46.01 | tzafrir | seele_, what error do you get? |
| 21:46.48 | tzafrir | I understand that Sangoma have setup scripts of their own. I have no idea what they really do besides generating files. Any documentation on them? |
| 21:46.58 | seele_ | tzafrir, my full log for one call http://www.pastebin.ca/477867 |
| 21:47.16 | Mercestes | they also generate zaptel.conf |
| 21:47.56 | tzafrir | seele_, asterisk -rx 'zap show status' |
| 21:48.47 | seele_ | tzafrir, status http://www.pastebin.ca/477871 |
| 21:49.01 | seele_ | but the leds of the card still in red |
| 21:50.53 | *** join/#asterisk rogerz (i=fbpz@cpe-24-195-144-82.nycap.res.rr.com) |
| 21:51.19 | NOT_guru | tzafrir: I wanted to thank you again for the time you spent with me last week |
| 21:51.48 | NOT_guru | tzafrir: I have a much better understanding of things now if you would like to review the genzaptelconf again sometime |
| 21:52.15 | carrar | Nothing says thank you like a hooker at your door |
| 21:52.16 | rogerz | Old IT guy left, and didnt give the password to the asterisk box, any way to recover it? I'm root on the system |
| 21:52.33 | NOT_guru | tzafrir: but everything is working now, after I did some manual configuring |
| 21:52.40 | Siya | bkruse: ping |
| 21:52.45 | tzafrir | NOT_guru, any review is always appreciated |
| 21:52.50 | NOT_guru | tzafrir: but yah.... Thank You |
| 21:53.42 | NOT_guru | FYI: main problem I was having was lack of dev nodes due to kudzu screwing with things |
| 21:54.47 | carrar | rogerz, 1st time to log into a unix box? |
| 21:56.03 | rogerz | the main asterisk password for the web frontend is the asterisk password? |
| 21:56.32 | rogerz | new to asterisk obviously |
| 21:59.42 | seele_ | CLI log for one call http://www.pastebin.ca/477908 |
| 22:00.22 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 22:00.34 | seele_ | please help ... I need to configure the outbound calls urgent!! |
| 22:00.35 | *** join/#asterisk Blackthorn (n=support@w-l4.smyth.net) |
| 22:01.03 | Blackthorn | I'm having an issue with sending calls from one * to another through iax. Mind taking a look at http://www.pastebin.ca/477909 |
| 22:01.37 | NOT_guru | Rogers..the first screen on the "webfrontend" |
| 22:01.57 | NOT_guru | Rogers: does it say trixbox or freepbx? |
| 22:02.01 | NOT_guru | by chance |
| 22:02.21 | NOT_guru | Rogers: if so, you may want to change channels |
| 22:03.11 | seele_ | why if I disconnect the PRI cable from the interface 2 the alarm zap status still in ok ?? |
| 22:03.13 | NOT_guru | I will try to help.. I just need this info |
| 22:04.08 | *** join/#asterisk neverblue (n=profx@unaffiliated/neverblue) |
| 22:04.31 | neverblue | my background() is still not allowing me to enter my menu |
| 22:04.38 | Mercestes | pastebin |
| 22:04.42 | neverblue | what else could it be, my sip.conf is setup |
| 22:04.47 | *** join/#asterisk Strom_M (n=strom@adsl-66-127-181-92.dsl.lsan03.pacbell.net) |
| 22:04.52 | neverblue | me Mercestes ? |
| 22:05.02 | Mercestes | Yup |
| 22:05.51 | neverblue | http://www.pastebin.ca/477917 |
| 22:06.03 | neverblue | any questions, just ask |
| 22:06.40 | Mercestes | neverblue, more. |
| 22:06.44 | neverblue | :) |
| 22:06.52 | Mercestes | I need to see ${sounds} please |
| 22:06.54 | neverblue | the rest is just the default |
| 22:07.00 | neverblue | oh, sorry |
| 22:07.14 | neverblue | SOUNDS=/var/lib/asterisk/sounds/ |
| 22:07.44 | *** join/#asterisk _mm_ (n=mmclain@cpe-75-80-238-180.dc.res.rr.com) |
| 22:08.02 | neverblue | the waitexten was added to hopefully fix the issue |
| 22:08.37 | neverblue | my files are not wavs, does that matter? |
| 22:08.37 | seele_ | some channel for sangoma support ??? |
| 22:08.42 | *** part/#asterisk kp00 (n=kp00@85stb55.codetel.net.do) |
| 22:09.15 | Corydon-w | seele_: call the vendor on the telephone |
| 22:09.47 | seele_ | what telephone I cant configure yet !! LOL |
| 22:10.45 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-188-234.nycmny.east.verizon.net) |
| 22:12.12 | NOT_guru | ~thebook |
| 22:12.16 | jbot | from memory, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 22:13.17 | neverblue | any ideas Mercestes ? |
| 22:14.07 | *** join/#asterisk shadou (n=aj@unaffiliated/dj-fu) |
| 22:14.46 | zm23 | hello all, asterisk is not properly acting on REFER sip method. Instead of replacing the call it initiates another call. can anyone help ? |
| 22:21.37 | Mercestes | <PROTECTED> |
| 22:21.56 | neverblue | hey, np |
| 22:22.00 | Mercestes | remove all occurances of ${sounds} from extensions.conf as it appears in Background() commands and try again |
| 22:22.03 | *** join/#asterisk dj-fu (n=aj@unaffiliated/dj-fu) |
| 22:22.03 | neverblue | im in no rush zzzz |
| 22:22.19 | neverblue | ok, sure, thats something to try |
| 22:22.23 | Mercestes | <Polycom> Gee, lets make a great product and then find innovative ways to not sell or support the damned thing. |
| 22:22.44 | Mercestes | bascially asterisk assumes /var/lib/asterisk/sounds |
| 22:23.02 | neverblue | yes |
| 22:23.11 | neverblue | but that didnt work for me before |
| 22:24.01 | neverblue | <PROTECTED> |
| 22:24.07 | neverblue | yeah, see |
| 22:24.16 | neverblue | ill just add the dir into it |
| 22:24.53 | Mercestes | really, you shouldn't. |
| 22:25.08 | Mercestes | 1: do the files exist under /var/lib/asterisk/sounds and 2: does asterisk have read rights to those files? |
| 22:25.10 | neverblue | yes, I realize that |
| 22:25.24 | neverblue | yes, I can hear the menu |
| 22:25.30 | neverblue | just the keypresses arent working |
| 22:25.42 | neverblue | i cannot press 1 to go to my local phone |
| 22:25.44 | neverblue | sip/me |
| 22:26.21 | Mercestes | use dtmfmode=auto annnnnd.......Set(timeout(digits) instead of waitexten |
| 22:26.25 | neverblue | and I added dtmfmode=info |
| 22:26.35 | Mercestes | did I say info? sorry, I meant auto |
| 22:27.25 | neverblue | cause its a grandstream phone |
| 22:27.25 | Mercestes | lol |
| 22:27.25 | Mercestes | ~gs |
| 22:27.36 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 22:27.36 | Mercestes | jbot is asleep on the job. |
| 22:27.42 | neverblue | info is documented to be the choice for GS phones |
| 22:28.15 | neverblue | none the less, it should be =info |
| 22:28.20 | Mercestes | .... |
| 22:28.21 | neverblue | any other ideas? |
| 22:28.24 | Mercestes | then WTF are you asking questions? |
| 22:28.35 | neverblue | pardon? |
| 22:28.36 | *** join/#asterisk nomadsoul_ (n=nomadsou@unaffiliated/nomadsoul) |
| 22:28.44 | Mercestes | If you know everything why are you asking for help? |
| 22:28.53 | neverblue | hey hey hey |
| 22:29.05 | neverblue | i guess the touchiness is getting around |
| 22:29.17 | Mercestes | you haven't made many friends here as far as I can tell with ....whatever you call it. |
| 22:29.19 | neverblue | i explained why I added that line into my sip.conf |
| 22:29.21 | neverblue | thats it |
| 22:29.31 | Mercestes | you've been kinda demanding and more than a little backhanded. |
| 22:29.38 | neverblue | demanding? |
| 22:29.44 | Mercestes | and..I've already fixed yoru issue with my aforementioned suggestions so. |
| 22:29.46 | neverblue | i asked if you had anymore ideas |
| 22:29.49 | Mercestes | as you told me earlier, "I think we're done." |
| 22:29.51 | neverblue | how is that demanding? |
| 22:30.17 | Mercestes | you don't go to a mechanic with a broken car, ask for help and then go "no, your wrong, it's something else." |
| 22:30.28 | neverblue | you have obvious communication issues |
| 22:31.00 | neverblue | thanks for you help |
| 22:31.03 | Mercestes | lol |
| 22:32.15 | cspot | waiter, check please |
| 22:33.20 | cpm | are we leaving now? the conversation was just getting interesting |
| 22:33.45 | ManxPower | And I thought it was getting boring |
| 22:33.59 | cpm | well, yeah, that too |
| 22:34.46 | ManxPower | Is there anyone having problems that wants to actually listen to the advice? |
| 22:35.01 | neverblue | what advice? |
| 22:35.07 | neverblue | i didnt see any advice |
| 22:35.20 | neverblue | (i am assuming that was a taunt to keep the convo going) |
| 22:35.57 | ManxPower | neverblue: What is your specific problem that you need a fix for. |
| 22:36.31 | neverblue | my background() is still not allowing me to enter my menu |
| 22:36.38 | neverblue | on keypresses |
| 22:36.55 | neverblue | it plays fine, just wont let me goto an exten |
| 22:36.58 | ManxPower | neverblue: does DTMF work in other places like Voicemail? |
| 22:37.00 | neverblue | any* |
| 22:37.12 | ManxPower | no, dialing does not count as the DTMF is not sent to Asterisk |
| 22:37.13 | neverblue | no idea, this is the first place I tested it |
| 22:37.22 | *** join/#asterisk BSD_Tech (n=bsdtech@adsl-69-230-166-20.dsl.irvnca.pacbell.net) |
| 22:37.29 | ManxPower | neverblue: what phone are you using? |
| 22:37.41 | neverblue | Grandstream BT-100 |
| 22:38.58 | ManxPower | neverblue: Make sure the phone is set to RFC2833 aka AVT DTMF, then set dtmfmode=rfc2833 in sip.conf for that phone. Also, to make sure the call is actually matching the sip.conf entry, put context=INVALID in [general] and the correct context in the phone device section of sip.conf. |
| 22:39.26 | ManxPower | Don't ask me where to set the DTMF mode in the Grandstream. They are crap phones and should be banned from the universe. |
| 22:39.47 | neverblue | the context is setup correct, and why are you suggesting rfc2833? |
| 22:39.57 | _mm_ | manx: what phone(s) do you recommend? |
| 22:40.01 | ManxPower | neverblue: because rfc2833 is the "right way" to do DTMF. |
| 22:40.09 | ManxPower | _mm_: Polycom |
| 22:40.57 | ManxPower | ~phones |
| 22:41.14 | jbot | somebody said phones was http://bani.anime.net/phones/. SIP Hardphones in order of quality/suggestability : Polycom (any), Aastra 480i, Cisco 7940(+), Linksys SPA-9XX, Snom |
| 22:41.14 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 22:41.47 | ManxPower | neverblue: inband DTMF only works with the ulaw and alaw codecs and if you have any kind of network jitter issues, it won't be reliable there either. INFO type of DTMF is an old way of doing it. |
| 22:42.03 | neverblue | let me try each |
| 22:42.20 | ManxPower | neverblue: won't do you any good if you don't set it tot he same on the phone and asterisk |
| 22:42.36 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 22:42.39 | ManxPower | I assume you have done what I suggested with regards to context= ? |
| 22:50.07 | neverblue | ManxPower, u da man |
| 22:50.54 | ManxPower | does anyone happen to know what the sox format name for .WAV aka .wav49 aka GSM in a Microsoft WAV wrapper is called. |
| 22:51.01 | ManxPower | neverblue: I've been doing this for a very long time. |
| 22:51.20 | drazak | What do I have to do to make people connecting with sip clients work? |
| 22:51.47 | neverblue | the co-worker has setup in-audio/info all this time |
| 22:52.59 | *** join/#asterisk Rusty1 (n=Rusty1@cpe-72-226-96-74.nycap.res.rr.com) |
| 22:53.28 | drazak | It won't register the people that are trying |
| 22:53.48 | neverblue | login/pass incorrect? |
| 22:54.19 | ManxPower | Registration only tells the remote server what IP address is associated with a specific user/password. It does nothing else. |
| 22:54.34 | *** join/#asterisk Crescendo_ (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 22:55.14 | Blackthorn | i'm tryign to send calls from one * to another * thorugh iax and i'm getting rejected with "request '@default' does not exis" even though default does exist in extensions.cofn.. any ideas? |
| 22:55.48 | ManxPower | Blackthorn: the server is not sending a userid |
| 22:56.01 | ManxPower | hence the lack of anything before the @ |
| 22:56.25 | drazak | ManxPower: how do I setup a login/pass for them? |
| 22:57.28 | ManxPower | [userid] |
| 22:57.34 | ManxPower | secret=something |
| 22:57.39 | ManxPower | disallow=all |
| 22:57.42 | ManxPower | allow=ulaw |
| 22:57.45 | ManxPower | context=default |
| 22:58.03 | ManxPower | drazak: I recommend you stop using Asterisk and read The Book. |
| 22:58.07 | ManxPower | ~book |
| 22:58.27 | 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 |
| 22:58.27 | drazak | ManxPower: what chapter is it in? |
| 22:58.28 | drazak | As I don't see it :P |
| 22:58.28 | drazak | I've read chap 1-7 |
| 22:58.35 | drazak | I have it. |
| 22:58.46 | ManxPower | drazak: I have no idea, but I cannot imagine the book not showing you how to do one of the most basic things in the world with asterisk |
| 22:59.19 | ManxPower | It would like the owners manual for your car not showing you how to turn on the headlights. |
| 22:59.22 | tzafrir | I'm ring o ge chan_zap.c in runk o build |
| 23:00.27 | tzafrir | as I have no inenion of actually installing a newer zaptel on my precious system, I try to use --with-zaptel=/path/to/zaptel/source |
| 23:01.12 | tzafrir | I also put the symlinks include and zaptel to . in the zaptel source dir, so include/zaptel/zaptel.h would be easily spotted |
| 23:01.49 | Blackthorn | here is what i have http://www.pastebin.ca/477980 if you wouldn't mind taking a look when you have time |
| 23:02.31 | ManxPower | Blackthorn: What is the Dial line? |
| 23:02.33 | tzafrir | The configure script fails to build a zaptel program or something. Sadly, I can't see the actual program in the config.log |
| 23:02.51 | Blackthorn | exten => 555,1,Dial(IAX2/remote_server:mysecret@x.x.x.x/${exten}); |
| 23:03.24 | tzafrir | One thing I get: checking for ZT_TONE_DTMF_BASE in zaptel/zaptel.h... ./configure: line 32467: -I/home/tzafrir/Proj/Asterisk/DigiumRW/zaptel/branches/1.4/include: No such file or directory |
| 23:03.39 | ManxPower | Blackthorn: Dial(IAX2/iaxconfentry/${EXTEN}) |
| 23:03.54 | tzafrir | I did verify that /home/tzafrir/Proj/Asterisk/DigiumRW/zaptel/branches/1.4/include and /home/tzafrir/Proj/Asterisk/DigiumRW/zaptel/branches/1.4/include/zaptel/zaptel.h exist |
| 23:04.25 | tzafrir | anybody building chan_zap in trunk? |
| 23:05.02 | ManxPower | tzanger: If I ran trunk then I would be fired. It is as simple as that. |
| 23:05.44 | tzafrir | ManxPower, was that to me? |
| 23:05.56 | ManxPower | tzanger: more or less. |
| 23:05.58 | Mercestes | ManxPower, hell, I'm starting to think running Astersk is going to get me fired. |
| 23:06.19 | tzanger | heh |
| 23:06.19 | ManxPower | Mercestes: only if you take stupid chances like trying to run all your phones lines over the internet. |
| 23:06.36 | Mercestes | what's wrong with that? >.> |
| 23:06.42 | Blackthorn | the dial line is now exten => 7823333,1,Dial(IAX2/remote_server/${exten}); and i get no authority found messages on both sides. which as i was told ealier is user/pass error. but there identical |
| 23:07.06 | Blackthorn | auth = md5, password=mysecret |
| 23:07.10 | Blackthorn | for testing only of course |
| 23:08.01 | ManxPower | Blackthorn: remove the auth-md5 |
| 23:08.20 | ManxPower | Blackthorn: I just wanted to make sure the Dial line was correct before trying anything else. |
| 23:10.43 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583012.dsl.bell.ca) |
| 23:10.44 | Blackthorn | roger. ok md5 removed and i'm getting the no auth |
| 23:10.56 | Blackthorn | reloaded iax2 and exten.. |
| 23:11.15 | ManxPower | Blackthorn: iax2 debug on the receiving server |
| 23:11.28 | ManxPower | is it the EXACT same error message? |
| 23:12.19 | Blackthorn | <PROTECTED> |
| 23:12.28 | Blackthorn | thats the sending one |
| 23:13.13 | Blackthorn | chan_iax2.c:7159 socket_process: Host x,x,x,x failed to authenticate as remote_server |
| 23:13.47 | neverblue | thanks again ManxPower im outta here |
| 23:14.45 | *** part/#asterisk kiscokid (n=ron@208.106.33.66) |
| 23:15.05 | ManxPower | Blackthorn: honestly I'm just too tried to help you |
| 23:15.34 | Blackthorn | alrighty many thanks. the debug says the same. failed to authenticate the user name came thorugh just fine. i'll try back another time. |
| 23:23.14 | *** join/#asterisk Gpl_Source (n=The_natu@unaffiliated/gplsource) |
| 23:23.35 | *** join/#asterisk SwK (n=SwK@24.248.196.141) |
| 23:34.08 | shido6 | Zzz |
| 23:34.30 | *** join/#asterisk anthm (n=anthm@adsl-75-54-59-121.dsl.milwwi.sbcglobal.net) |
| 23:34.30 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:36.22 | *** join/#asterisk trcosta (n=hhlamar@201.15.216.158) |
| 23:43.38 | *** join/#asterisk Here_And_There (n=Here_And@pool-68-238-252-162.phlapa.fios.verizon.net) |
| 23:47.56 | *** join/#asterisk bmg505 (n=leon@196.209.180.71) |
| 23:51.11 | *** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 23:51.36 | *** join/#asterisk [[blah]asfd (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 23:52.05 | [[blah]asfd | is there such a thing as a good IAX2 ata? I have a linksys SIP ATA that works well, but I am looking now for IAX2. |
| 23:52.34 | shido6 | the iaxy |
| 23:53.07 | ManxPower | [[blah]asfd: no. |
| 23:53.50 | ManxPower | shido6: he said "good" |
| 23:53.51 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 23:53.51 | shido6 | unless you're in a hot environment |
| 23:53.51 | shido6 | it works well ... |
| 23:53.52 | ManxPower | or need a codec other than ulaw, alaw, or pcm |
| 23:53.55 | ManxPower | ..er..adpcm |
| 23:54.06 | ManxPower | or your network does not do bootp |
| 23:54.13 | ManxPower | or you need call pickup *8 |
| 23:54.19 | [[blah]asfd | well... good is relative. I have a low opinion of ATAs. But as far as ATAs go, does the iaxy suck? |
| 23:54.37 | shido6 | it does not suck... that much. |
| 23:55.19 | ManxPower | [[blah]asfd: They ran hot. newer versions may have fixed that. It has a very limited list of supported codecs. It does not support DNS. It requires bootp, not DHCP. chan_iax2 does not support call pickup via *8 |
| 23:55.42 | shido6 | so no domainname.boogie.down.productions.com |
| 23:55.50 | shido6 | but it knows 123.123.123.123 :) |
| 23:55.51 | [[blah]asfd | bootp is enough to turn me away. |
| 23:55.58 | [[blah]asfd | how about an IAX2 phone? |
| 23:56.04 | ManxPower | [[blah]asfd: you would want to confirm that about bootp |
| 23:56.57 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.169.195) |
| 23:58.12 | ManxPower | [[blah]asfd: there are reports of it randomly locking up, but I have not experienced that and suspect it might have been heat related. Oh, another thing that sucks about it is that it is $100 |