| 00:00.46 | ChongLi | hmm, these are pretty expensive heh |
| 00:00.50 | VioByte | the cisco's 79XX series works good with asterisk |
| 00:00.58 | VioByte | well you get what you pay for :) |
| 00:01.22 | ChongLi | I have a really nice analog cordless setup with 3 handsets |
| 00:02.56 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:03.03 | VioByte | thats thats pretty easy with a ATA |
| 00:03.08 | ChongLi | how about a TDM400P? |
| 00:03.16 | *** join/#asterisk lokkju_wrk_ (n=lokkju@unaffiliated/lokkju) |
| 00:03.19 | VioByte | cost me 180$ for my Cisco 7940 + Power Adaptor and 2 year warranty |
| 00:03.25 | VioByte | never tried it |
| 00:03.26 | *** join/#asterisk elriah (i=elriah@adsl-074-164-217-189.sip.bhm.bellsouth.net) |
| 00:03.34 | ChongLi | hmmm |
| 00:03.46 | elriah | Hi all. We're deploying a bunch of asterisk realtime servers. How can I use both static realtime and flat file configs for my sip.conf? |
| 00:03.59 | elriah | It seems I can do either or but not both and I swear I read somewhere it was possible. |
| 00:03.59 | VioByte | the cisco i have supports 2 lines |
| 00:04.04 | ChongLi | but with an ATA I lose access to my PSTN? |
| 00:04.41 | VioByte | not sure never really tested out ata's much |
| 00:04.42 | ChongLi | I need an FXO |
| 00:05.07 | JT | the SPA-3102 has 1 FXS and 1 FXO port |
| 00:05.08 | *** join/#asterisk Caplain (n=shayne@pool-71-184-33-177.bstnma.east.verizon.net) |
| 00:05.27 | JT | different ATAs have different types and numbers of ports |
| 00:06.26 | ChongLi | ohhh |
| 00:06.56 | ChongLi | looks like what I need |
| 00:07.28 | JT | probably not as good as a TDM400P, but good enough for home use |
| 00:10.56 | ChongLi | is there an easy way to do encryption with asterisk? |
| 00:20.51 | JT | no |
| 00:20.51 | Keltus | hey JT, I was wondering if you had any recommendations for tollfree -> asterisk service providers |
| 00:20.51 | ChongLi | well, I don't exactly want to make calls to australia in the clear |
| 00:20.51 | ChongLi | with a sip softphone I can use zfone |
| 00:20.53 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 00:20.53 | blitzrage | ChongLi: you can use encryption with IAX2, or patch Asterisk with the SRTP stuff from he bug tracker |
| 00:20.53 | _DAW | Anyone using Audiocodes MP-XXX gateways with asterisk? |
| 00:20.55 | JT | Keltus: nup |
| 00:20.55 | _DAW | I am having an interesting problem. I get a 408 and disconnect from the MP-XXX after 10 minutes exactly on all outbound calls. |
| 00:20.56 | JT | ChongLi: why, exactly who is going to listen in? |
| 00:20.56 | justdave | you behind a NAT router? |
| 00:20.56 | ChongLi | these are for business calls |
| 00:20.56 | JT | ChongLi: you could always use a vpn |
| 00:21.03 | JT | ChongLi: so what |
| 00:21.03 | JT | ChongLi: so are hundreds of thousands of calls daily |
| 00:21.03 | ChongLi | ok, so just use a VPN then? |
| 00:21.03 | justdave | 10 minutes is a common idle timeout on a lot of routers, it might be misinterpreting what's "idle" |
| 00:21.04 | JT | ChongLi: WHO is going to listen in? |
| 00:21.04 | JT | if you are concerned about a threat, is it real or imagined? |
| 00:21.04 | ChongLi | JT: you never know |
| 00:21.05 | JT | imagined, ok. |
| 00:21.05 | VioByte | Keltus, i use junction networks. $2 per 800# and 2.8 or 2.9 cents inbound/outbound calls to any 48 states in the U.S. |
| 00:21.05 | VioByte | Keltus, there not that bad at all in voice quality. |
| 00:21.05 | justdave | re: someone listening in, I'd be more concerned about someone targetting your specific phone calls than someone randomly listening in on the net. |
| 00:21.06 | JT | the person has to have access to network backhaul on the patch between you and the other end |
| 00:21.06 | JT | justdave: exactly |
| 00:21.06 | JT | it's silly to imagine that people might randomly listen in |
| 00:21.06 | ChongLi | JT: I'm also concerned about ISP's throttling calls |
| 00:21.06 | Keltus | VioByte: do they support 1 888 numbers too? |
| 00:21.06 | Keltus | VioByte: and is it easy to set up with asterisk? |
| 00:21.07 | justdave | so if you've got stuff being talked about on your phone system that you really really want to be confidential then vpn is probably the way to do it. Otherwise there's not a lot of point in trying to hide it |
| 00:21.07 | JT | ChongLi: do you have any evidence that that is likely? |
| 00:21.07 | ChongLi | one ISP in my area has admitted to throttling all known voip protocals |
| 00:21.07 | VioByte | 800, 866, 877 most of the toll free number |
| 00:21.08 | VioByte | numbers* |
| 00:21.08 | JT | then don't use that isp |
| 00:21.08 | ChongLi | I don't, but there's no reason to believe my ISP won't eventually |
| 00:21.08 | Keltus | sounds good |
| 00:21.08 | ChongLi | I only have 2 ISPs in my area |
| 00:21.08 | VioByte | Keltus, http://www.junctionnetworks.com/pstngateway/ |
| 00:21.09 | VioByte | ask live support if you have any questions. |
| 00:21.10 | Keltus | wow, unlimited simulataneous calls? (channels?) |
| 00:21.11 | blitzrage | must be per-minute rates |
| 00:21.11 | JT | what's amazing about that if you pay per minute? |
| 00:21.11 | justdave | we use junctionnetworks for our inbound failover, they're great. |
| 00:21.11 | VioByte | it is per/min rate |
| 00:21.12 | blitzrage | if you're paying per-minte, you'd think they want as many channels as possible to be going on |
| 00:21.51 | justdave | we're only using inbound with them though |
| 00:21.51 | VioByte | http://www.junctionnetworks.com/pstngateway/pricing.php |
| 00:21.51 | VioByte | rates sheet |
| 00:22.20 | VioByte | i have a local did and a 800 directed to my asterisk box :) |
| 00:22.22 | justdave | service is great anyway, I hadn't actually looked at pricing before. :) Someone besides me set it all up, I just run the system connecting to it :) |
| 00:29.08 | elriah | Anyone here have asterisk realtime experience? |
| 00:29.48 | VioByte | did but then it got annoying seeing asterisk look up the same damn select 3 times in a row :( |
| 00:30.31 | last1 | anyone has a 7970 ? |
| 00:32.25 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 00:33.40 | ron_lautmann | I have a 7960 |
| 00:33.58 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 00:34.05 | last1 | unfortunately the 7960 isn't xml based like 7970 and that's where my problem lies |
| 00:34.13 | Keltus | 4c/min is a bit high though |
| 00:34.19 | Keltus | I've seen 2c/min |
| 00:34.28 | Keltus | do you guys know any that are not per minute? |
| 00:34.39 | Keltus | I don't see why per minute is a surprise... |
| 00:34.58 | JT | Keltus: it's not a surprise |
| 00:35.02 | last1 | I use voipstreet.com and vitelity.com |
| 00:35.04 | shmaltz | is pcix and pci express the same thing? |
| 00:35.06 | last1 | both are pretty good |
| 00:35.08 | JT | people have unlimited DIDs, but not 1800 |
| 00:35.47 | Keltus | right, all the 1 800s I've seen were per minute |
| 00:35.47 | JT | shmaltz: pci-x != pci-e/express, but sometimes people confuse terminology |
| 00:35.58 | shmaltz | JT, I do confuse them |
| 00:36.01 | JT | Keltus: yes, because it costs money per minute to terminate them |
| 00:36.20 | shmaltz | JT, is there a real hardware SATA RAID solution that uses PCI EXPRESS? |
| 00:37.17 | kavit | i sell 1800 numbers |
| 00:37.19 | kavit | and trust me |
| 00:37.23 | shmaltz | and what does RoHS comliant mean? |
| 00:37.29 | kavit | we pay suppliers by the minute |
| 00:37.45 | kavit | there is always a per min cost |
| 00:37.51 | kavit | PS: Hello |
| 00:38.21 | kavit | my test cluster has been running for a week with no dramas |
| 00:38.33 | kavit | time to move the test system into production |
| 00:38.38 | kavit | without telling people |
| 00:38.44 | VioByte | heheheh |
| 00:38.45 | kavit | hehe |
| 00:38.50 | VioByte | that will be fun. |
| 00:39.10 | JT | shmaltz: no idea what's real hardwar raid these days |
| 00:39.16 | *** join/#asterisk jazzanova (n=boris@S010600173f460b29.vc.shawcable.net) |
| 00:39.17 | jazzanova | hi |
| 00:39.24 | shmaltz | thank you JT |
| 00:39.30 | shmaltz | hi jazzanova |
| 00:39.40 | JT | shmaltz: rohs is something manufacturers must comply with in the EU if then want a chance to sell a product there |
| 00:39.42 | jazzanova | i would like to call a phone from a computer and play a wave file. how can i do it ? |
| 00:39.48 | jazzanova | is asterisk something that I can use ? |
| 00:39.53 | shmaltz | JT, thanks again |
| 00:40.00 | jazzanova | hey shmaltz |
| 00:40.02 | shmaltz | jazzanova, yes |
| 00:40.09 | JT | shmaltz: relates to low/no lead and other toxic materials |
| 00:40.21 | jazzanova | shmaltz: can you point me to a reading material ? |
| 00:40.22 | Keltus | kavit: what's the wholesale cost? |
| 00:40.30 | shmaltz | ~docs |
| 00:41.15 | jbot | [docs] 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 |
| 00:41.18 | shmaltz | ~asterisk wiki |
| 00:41.30 | jbot | i heard asterisk wiki is http://www.voip-info.org/wiki-Asterisk |
| 00:41.30 | jazzanova | is asterisk the correct tool to use ? |
| 00:41.30 | kavit | Keltus: we pay about 4cents per minute incoming |
| 00:41.30 | shmaltz | wake up jbot |
| 00:41.30 | kavit | but we dont do it bulk enough |
| 00:41.31 | shmaltz | jbot, you around?????????????// |
| 00:41.35 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 00:41.35 | Keltus | kavit: how do some small providers offer it for 1.9c/min or something |
| 00:41.37 | JT | Keltus: australian or american 1800s? |
| 00:41.45 | Keltus | american 1 800 |
| 00:41.45 | Keltus | just from the 48 states |
| 00:41.45 | JT | aud or usd? |
| 00:41.45 | Keltus | screw the other two |
| 00:41.46 | JT | Keltus: i meant kavit |
| 00:41.53 | JT | kavit: australain 1800s? |
| 00:42.01 | kavit | Keltus: 1.9c/min for inbound 1800 number in Australia? |
| 00:42.06 | kavit | who? |
| 00:42.06 | JT | stupid nicks starting with the same letter :P |
| 00:42.16 | kavit | yeah JT australian 1800 |
| 00:42.18 | shmaltz | can I PCI Express x16 slot take a PCI Express x4 card? |
| 00:42.30 | kavit | we sell of AUD 8 cents a minute |
| 00:42.34 | kavit | incoming |
| 00:42.36 | JT | nah Keltus is just an american who thinks everyone here is from america :P |
| 00:43.42 | kavit | oh |
| 00:43.42 | Keltus | kavit: oh, sorry for my americancentric view |
| 00:43.42 | kavit | i wanted to sign up |
| 00:43.42 | kavit | for 1.9c/min |
| 00:43.42 | kavit | heh fuck loyalty to suppliers |
| 00:43.43 | kavit | oops |
| 00:43.43 | kavit | i said the f word |
| 00:43.43 | kavit | :| |
| 00:43.43 | JT | hfs, no way |
| 00:44.40 | Keltus | reprimand? no cursing on this channel? |
| 00:44.47 | kavit | not sure |
| 00:45.17 | kavit | anyway I am trying to weigh up economic interest in making a clustering hot stand by solution |
| 00:45.20 | *** join/#asterisk BSD_Tech (n=bsdtech@ppp-69-239-113-126.dsl.irvnca.pacbell.net) |
| 00:45.29 | kavit | do you guys think people would buy something like that? |
| 00:45.57 | elriah | kavit: two scripts, one that rsyncs everything and another that pings the live IP and if dies, switches the local Ip and restarts networking. Done. |
| 00:46.36 | kavit | elriah: i am using shared disks and load balancing and heartbeat |
| 00:46.56 | kavit | i want to do something where you can add and remove boxes into a cluster |
| 00:47.03 | kavit | and it sets them up automatically |
| 00:47.08 | kavit | ease of administration |
| 00:47.11 | JT | kavit: people can already make their own failover solutions though |
| 00:47.11 | kavit | monitoring |
| 00:47.15 | kavit | and stuff |
| 00:48.10 | elriah | Anyone see this using realtime config (static) ?? 1.2.18 ---> WARNING[4543]: chan_sip.c:12984 reload_config: Section 'somepeer.123' lacks type |
| 00:48.29 | elriah | It appears for all my sip peers. There is a value for type for each one on metric 0 |
| 00:48.36 | kavit | JT: yeah but this makes it easier... the reason people use FreePBX and asterisk gui i guess |
| 00:49.01 | JT | mental illness ;) |
| 00:49.08 | kavit | hehe |
| 00:49.20 | kavit | JT: doesnt mean I can't make money off that |
| 00:49.32 | kavit | plus this way I can sell support as well |
| 00:50.21 | kavit | plus I am doing something where each machines config is stored in an LDAP directory that makes it easy to rebuild boxes and restore config |
| 00:50.26 | kavit | but it is a lot of work |
| 00:50.53 | kavit | if I wont make $$ of it, I probably wont do it, i will just write a howto for curious people and leave it at that |
| 00:51.31 | JT | heh |
| 00:51.42 | elriah | JT: You use RealTime? |
| 00:52.16 | JT | no |
| 00:53.23 | *** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net) |
| 00:58.56 | shmaltz | what is a 4-lane PCIe ? |
| 00:59.27 | Sweeper | I think they mean 4x |
| 01:00.50 | shmaltz | Sweeper, so a single PCI-e x16 wont do? |
| 01:02.59 | *** join/#asterisk jsolares (n=jsolares@216.106.164.193) |
| 01:03.39 | jsolares | why would after dialing via IAX2 with a resulting BUSY from ISDN and let it auto-fallthru be any different form setting PRI_CAUSE=17 and then do a hangup? |
| 01:04.37 | jsolares | i've looked at the pri debug info and it looks exactly the same, however i get the expected audio playback with the PRI_CAUSE but not the auto-fallthrough |
| 01:05.03 | Sweeper | shmaltz: I have the FEELING you can plug lower-x cards into higher-x slots, but I don't know for sure |
| 01:05.30 | shmaltz | Sweeper but does the 4 lane mean pci-e x4? |
| 01:05.40 | Sweeper | shmaltz: I'm pretty sure it does |
| 01:07.13 | jsolares | PCIe card will physically fit (and work correctly) in any slot that is at least as large as it is (e.g. an x1 sized card will work in any sized slot) |
| 01:07.48 | *** join/#asterisk grndslm (n=grndslm@host-69-59-127-198.nctv.com) |
| 01:09.08 | *** join/#asterisk MatN16 (n=akmdkmw@206-248-190-130.dsl.teksavvy.com) |
| 01:09.27 | MatN16 | Good day! |
| 01:10.11 | elriah | Anyone seen this error in static realtime? Apr 25 20:04:56 WARNING[4543]: chan_sip.c:12984 reload_config: Section 'test' lacks type |
| 01:10.21 | elriah | It does it for all my peers. |
| 01:10.23 | MatN16 | Would anyone know why quitting the voicemail with # return me to the originating context? |
| 01:10.29 | elriah | And type is in there, defined as friend. |
| 01:11.07 | [TK]D-Fender | MatN16, It continues the dialplan wherever you wre, because thats what it does. if you want it to actually hangup, then call it yourseif right after. |
| 01:11.22 | MatN16 | Ah, that makes a whole lot of sense |
| 01:11.28 | shmaltz | jsolares, after reading that I am under the impression that x4 and 4 lane is the same thing, am I correct? |
| 01:11.35 | jsolares | correct |
| 01:12.18 | shmaltz | jsolares, thank you |
| 01:12.44 | shmaltz | the only question left then is, is this real raid: |
| 01:12.46 | shmaltz | http://www.adaptec.com/en-US/products/raid_tech/sata_drives/SAS-3405/ |
| 01:13.59 | jsolares | what do you mean by real raid? |
| 01:14.07 | MatN16 | hardware raid, i presume |
| 01:14.43 | shmaltz | not fakeraid |
| 01:14.53 | jsolares | it doesnt really say :S |
| 01:14.58 | MatN16 | which is software raid |
| 01:14.59 | shmaltz | read this: |
| 01:15.01 | shmaltz | http://linuxmafia.com/faq/Hardware/sata.html |
| 01:15.30 | MatN16 | that card is hardware raid |
| 01:15.53 | *** join/#asterisk Avochelm (n=damo@gw-morphett.koalatelecom.com.au) |
| 01:15.57 | MatN16 | its got an onboard battery, which is always a sign that you have hardware raid |
| 01:16.08 | MatN16 | and its expensive, which is also a good sign |
| 01:16.18 | jsolares | but it doesnt really say :S |
| 01:16.19 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:17.17 | MatN16 | no, it doesn't, but then, you wouldn't pay 400$ for a software raid controller |
| 01:17.17 | MatN16 | this is the real deal, i'm sure |
| 01:17.17 | JT | err |
| 01:17.24 | JT | MatN16: a battery doesn't always mean real raid |
| 01:17.41 | MatN16 | explain |
| 01:17.57 | JT | some raid cards use the OS driver and offload to host cpu |
| 01:18.11 | MatN16 | possibly |
| 01:18.11 | shmaltz | MatN16, you talking about the 3405? |
| 01:18.31 | MatN16 | I don't know about adaptec, but anything LSI MegaRAID is a good bet |
| 01:18.34 | shmaltz | MatN15, what about the linux driver? is it included in 2.6.x? |
| 01:18.50 | MatN16 | I wouldn't know - but I know megaraid cards are a safe bet for Linux |
| 01:18.52 | shmaltz | MatN16, so which one are you talking about? |
| 01:19.07 | MatN16 | You want to run SATA? |
| 01:19.10 | MatN16 | or SAS? |
| 01:19.17 | JT | lsi make a bunch of cards |
| 01:19.24 | [TK]D-Fender | 3ware 8k-9k series |
| 01:19.25 | JT | that work under Linux |
| 01:19.27 | shmaltz | MatN16, SATA |
| 01:19.41 | MatN16 | I run vmware ESX on a megaraid PCI 150-4 |
| 01:19.45 | shmaltz | [TK]D-Fender, these are real hw raid? |
| 01:19.55 | MatN16 | which uses the same driver as its SCSI peers |
| 01:20.03 | JT | better be considering how much they cost |
| 01:20.07 | shmaltz | MatN16, you have a link to that? |
| 01:20.10 | JT | 3ware make awesome sata raid stuff |
| 01:20.12 | MatN16 | yeah hold on |
| 01:20.32 | [TK]D-Fender | shmaltz, yup, the best kind.... cost is far (500$ 4-port +/-) |
| 01:20.43 | [TK]D-Fender | fair* |
| 01:21.02 | shmaltz | I am realy looking for ~$300 |
| 01:21.22 | [TK]D-Fender | LSI & 3ware have the best kernel support for reasonably price raid cards. SATA raid = yummeh |
| 01:21.25 | MatN16 | J4K3: it does |
| 01:21.32 | JT | i wouldn't soft raid the OS |
| 01:21.36 | JT | and i'd never fake raid |
| 01:21.42 | ChongLi | ZFS with raid-z |
| 01:21.46 | shmaltz | J4k3, I use sw raid (Linux md devices) but they take too long to setup |
| 01:21.59 | groogs[h] | (as opposed to a hardware card where the drivers were just written because they had to be written so you'd buy the card..) |
| 01:22.04 | *** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au) |
| 01:22.04 | MatN16 | shmaltz: http://www.lsi.com/storage_home/products_home/internal_raid/megaraid_sata/megaraid_sata_1504/index.html |
| 01:22.04 | JT | shmaltz: i wouldn't use it for the OS partition |
| 01:22.14 | [TK]D-Fender | shmaltz, my pricing info was out of date : http://www.tigerdirect.com/applications/SearchTools/item-details.asp?EdpNo=1740122&CatId=1455 |
| 01:22.25 | shmaltz | JT, why not? lilo supports RAID 1 |
| 01:22.28 | MatN16 | shmaltz: the 150-4 has no battery protection, but then its only 400$ |
| 01:22.29 | shmaltz | and thats what I use |
| 01:22.30 | MatN16 | or 300$ |
| 01:22.45 | J4k3 | CF = Compact Flash (in an IDE adapter) |
| 01:22.48 | JT | shmaltz: too much trouble and futzing about especially if you want to split the array |
| 01:22.49 | MatN16 | I paid 250$ canadian, but then I work for a VAR with access to cost from the distributors |
| 01:22.54 | JT | and bring it back the way you want |
| 01:23.02 | groogs[h] | MatN16: whats the battery for? |
| 01:23.06 | shmaltz | MatN16, so can I get it for that price as well ;) |
| 01:23.15 | MatN16 | battery saves your data when theres an outage |
| 01:23.17 | shmaltz | groogs[h], cache |
| 01:23.20 | JT | groogs[h]: umm all real raid cards have batteries and ram cache |
| 01:23.22 | ChongLi | ZFS requires no battery |
| 01:23.39 | MatN16 | it allows the card to write whats in ram before dying |
| 01:23.39 | ChongLi | with raid-z you do not need a card either |
| 01:23.43 | groogs[h] | haven't heard of UPSes ? |
| 01:23.45 | ChongLi | ZFS protects against the write-hole |
| 01:23.56 | groogs[h] | they keep the machine running, too, when there's an outage... |
| 01:24.01 | JT | groogs[h]: UPSes don't protext against everything |
| 01:24.09 | ChongLi | http://opensolaris.org/os/community/zfs/ |
| 01:24.19 | groogs[h] | no, you're right. |
| 01:24.20 | *** join/#asterisk flewid (n=flewid@216.145.22.66) |
| 01:24.21 | JT | groogs[h]: ram cache is extremely important in raid to ensure data continuity |
| 01:24.24 | flewid | 'afternoon |
| 01:24.32 | groogs[h] | but neither does a battery. :) |
| 01:24.33 | shmaltz | flewid, you live in a box? |
| 01:24.39 | JT | groogs[h]: ? |
| 01:24.42 | JT | groogs[h]: huh? |
| 01:25.00 | flewid | shmaltz: a house shaped like a box? |
| 01:25.06 | ChongLi | zfs ensures data consistency |
| 01:25.20 | shmaltz | no, I meant a social box, it's not after noon in most of the world |
| 01:25.21 | MatN16 | shmaltz: the link I sent you is for a card that has 64MB of ram if I recall. Its one of the only card that tricks vmware ESX in installing on regular SATA drives :) (the damn thing wont normally settle for anything less than SCSI, FCAL or NFS-mounted storage) |
| 01:25.24 | groogs[h] | i would tend to think that a raid setup where if you lose power, you can trash the array has serious problems.. (if that's what you mean anyway) |
| 01:25.31 | ChongLi | why buy hardware that doesn't work as well as software? |
| 01:26.03 | groogs[h] | if you lose data, ok.. the battery doesn't protect that anyways, if you lose power to the machine (but the raid card has power to write its cache), you still lose everything in memory that the OS hasn't told to write yet |
| 01:26.20 | *** join/#asterisk hexa (n=hexa@modemcable189.156-200-24.mc.videotron.ca) |
| 01:26.22 | flewid | anyone here using pfsense as a router perchance? |
| 01:26.25 | groogs[h] | like i said though, i just use software raid. cheaper cards, and mdadm is well well tested |
| 01:26.35 | flewid | i'm having a bitch of a time getting my phone to register through it to a remote * box |
| 01:26.41 | ChongLi | groogs[h]: look into zfs |
| 01:26.44 | J4k3 | my big problem with hard raid was |
| 01:26.45 | ChongLi | it's the best |
| 01:26.54 | shmaltz | flewid, what does asterisk say? |
| 01:26.55 | J4k3 | my first hard SATA raid card died before I even got the damned thing deployed. |
| 01:26.59 | JT | MatN16: also, battery on raid cards usually restores the cache AFTER the power is reapplied |
| 01:27.03 | J4k3 | so I was like... if I do this, I have to buy two |
| 01:27.03 | shmaltz | flewid, whats your network topology? |
| 01:27.03 | flewid | shamltz: sip 401 unauth |
| 01:27.08 | J4k3 | one to use, one for spare. |
| 01:27.18 | J4k3 | the costs start adding up |
| 01:27.20 | shmaltz | flewid, it's a one way comm problem |
| 01:27.24 | MatN16 | JT: you are probably right - I'm not very knowledgeful of storage |
| 01:27.53 | shmaltz | flewid, make sure that asteirsk knows it's nat, by adding nat=yes |
| 01:27.53 | hexa | Hi all... when I run asterisk I see about 16 asterisk processes... is there any way to set the number of proceses it starts ? ( not sure if it's threads or processes)... |
| 01:27.53 | flewid | shmaltz: asterisk server is remote, behind no firewall, i can register fine with a softphone |
| 01:27.53 | MatN16 | flewid: sip doesn't play nice with firewalls |
| 01:27.53 | shmaltz | fromt the same network? |
| 01:27.56 | shmaltz | flewid, waht phoen you using? |
| 01:27.58 | flewid | shamltz: my internal network is cable modem -> pfsense -> (vlan1/2/3) -> cisco 2950 -> cisco 7970 phone |
| 01:28.02 | shmaltz | *phone |
| 01:28.24 | shmaltz | flwid, make sure you tell the cisco phone its behind nat |
| 01:28.26 | JT | groogs[h]: a lot of handwaving with your justifications there. fact is, inconsistant RAID array == very bad news. inconsistant user data == bad news, but not so bad |
| 01:28.41 | JT | groogs[h]: hence why the battery backup is so important |
| 01:28.41 | flewid | shamltz: yup, nat=yes, qualify=2000, externip=MYLOCALIP |
| 01:28.43 | flewid | tried all that |
| 01:29.01 | ChongLi | JT: zfs guarantees consistent data |
| 01:29.16 | JT | ChongLi: can you please stop repeating the same thing over and over, i don't care |
| 01:29.28 | JT | 100% data consistency is almost impossible to guarantee |
| 01:29.29 | groogs[h] | JT: i just don't see how the battery guarantees you anything |
| 01:29.34 | J4k3 | well, shouldn't the whole system have a battery backup, dual psus, etc? |
| 01:29.41 | groogs[h] | really, it guarantees if your UPS dies, or someone kicks out the power |
| 01:29.41 | JT | J4k3: yes |
| 01:29.43 | J4k3 | it should know long before the mains power actually fails |
| 01:30.07 | shmaltz | flewid, there is a setting that should go in the SIPmac.cfg file |
| 01:30.21 | groogs[h] | or rather, it guarantees that the stuff in cache to be written isn't lost .. but not the stuff in system memory |
| 01:30.21 | flewid | oh yah? |
| 01:30.29 | shmaltz | flewid, http://www.cisco.com/en/US/products/sw/voicesw/ps2156/products_administration_guide_chapter09186a00801d1977.html |
| 01:30.32 | JT | groogs[h]: the battery backup on raid cards is mainly there to mitigate the REDUCED reliability that that aspect of raid brings to data integrity |
| 01:30.37 | shmaltz | scroll down to nat enabled |
| 01:30.37 | flewid | i tried nat>true</true> |
| 01:30.40 | flewid | k sec |
| 01:30.45 | JT | groogs[h]: yes, that is very important |
| 01:30.52 | JT | losing user data is one thing |
| 01:31.02 | JT | having the whole array unreadable is another |
| 01:31.22 | groogs[h] | yeah, i'm just saying, i think the whole raid design is broken if that is possible |
| 01:31.33 | flewid | shmaltz: i put this in the <line> declaration for the extension i'm trying to register? |
| 01:31.42 | JT | groogs[h]: none of the alternatives are very rosy |
| 01:31.46 | groogs[h] | i don't have any experience with setting up/administrating hardware raid though, |
| 01:31.58 | flewid | shmaltz: because i do have an internal * server, so i can't get too nuts with the nat stuff on the phone |
| 01:32.01 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 01:32.05 | shmaltz | flewid, no, you put it in the SIPmac.cfg or SIPDefault.cfg file |
| 01:32.30 | flewid | and SEPxxx.conf.xml = same difference ? |
| 01:32.45 | groogs[h] | seems like it should be an ACID type transaction.. if it gets partially written, it might as well not be there.. eg, it has to write all the data, then turn on a bit to make it 'show up' |
| 01:33.06 | J4k3 | groogs[h]: yeah but, poop happens. |
| 01:33.13 | J4k3 | except I've found poop happens a lot less these days |
| 01:33.15 | groogs[h] | if something happens (eg, power loss) before it turns on that bit, it means the data is not done, so it should be dropped |
| 01:33.25 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 01:33.35 | groogs[h] | when it powers back up, it should ignore that data |
| 01:33.37 | J4k3 | most of the poop I had occur here involved bad SCSI cables (I...hate....scsi...) or bad drive firmware. |
| 01:33.47 | JT | this is why power should never be lost ;) |
| 01:33.51 | groogs[h] | of course, i've never designed a file system or anything... just saying :) |
| 01:34.03 | groogs[h] | J4k3: yeah, raid should catch that though, hopefully |
| 01:34.20 | J4k3 | groogs[h]: yeah, when it didn't tend up taking the whole bus/controller down. |
| 01:34.21 | groogs[h] | and of course, you have backups, so you never REALLY lose the 'whole' array. right, backups.? :) |
| 01:34.24 | flewid | shmaltz: yah, mine isn't SIP<MAC> it's SEP<MAC>.conf.xml (what came w/ the phone) |
| 01:34.33 | flewid | let me try these options out |
| 01:34.33 | flewid | sec |
| 01:34.43 | J4k3 | groogs[h]: realtime backups are best... |
| 01:34.49 | J4k3 | preferably offsite... |
| 01:34.51 | J4k3 | ;) |
| 01:35.32 | groogs[h] | do you do realtime? |
| 01:35.52 | J4k3 | haha no |
| 01:36.09 | J4k3 | well, I do for the database pretty much |
| 01:36.13 | groogs[h] | there is an argument though, for doing differential backups,, if some corruption happens (eg, someone maliciously deletes a bunch of files) you don't want that "copied" to your backups |
| 01:36.28 | kavit | you could take a look at drbd |
| 01:36.33 | kavit | raid 1 over tcp |
| 01:36.34 | J4k3 | yeah, unless you have tiered backups |
| 01:36.50 | J4k3 | kavit: yep... I'm seriously considering doing that across town using an 802.11a link. |
| 01:36.51 | groogs[h] | thats what i do with mine, hardlinks with rsync |
| 01:37.00 | shmaltz | is this real raid: |
| 01:37.02 | shmaltz | http://www.lsi.com/storage_home/products_home/internal_raid/megaraid_sata/megaraid_sata_1504/index.html |
| 01:37.08 | *** part/#asterisk litage (n=nick@203.220.55.70) |
| 01:37.11 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 01:37.16 | shmaltz | it's PCI based, so I will not take it, but just trying to find out |
| 01:37.44 | JT | if you have the resources to do a full backup very regularly, that's even better than regular differential backups |
| 01:37.59 | *** join/#asterisk rikstah (n=rick@rhamnett.plus.com) |
| 01:38.11 | J4k3 | shmaltz: I had a card that appeared identical in appearance, except 6 SATA ports... its processor topped out at 40MB/sec or so on RAID1. |
| 01:38.21 | J4k3 | it actually turned into one hell of a bottleneck under load |
| 01:38.29 | groogs[h] | JT: thats the nice thing about rsync+hardlinks (eg, rdiff-backup does it this way, also easy to script) ... each "differential" backup IS a full backup |
| 01:38.31 | jsolares | meh fuck me sideways, damn R2 and sending indication 14 when it shouldnt beacuse the other end is busy |
| 01:38.32 | shmaltz | hmmmmmm, thats no good |
| 01:38.34 | rikstah | Hi there, does anyone here have a Mac that would kindly like to test my Asterisk Desktop Manager application (i just need to know if the installer works and the app launches). It should take 1 minutes. I'd really appreciate it |
| 01:38.39 | groogs[h] | the files that haven't changed are just hardlinked to the previous backup |
| 01:38.41 | kavit | i have a spare hot swap hdd for every machine |
| 01:38.48 | kavit | that I use once a week |
| 01:38.51 | shmaltz | J4k3, I guess you had this: |
| 01:38.52 | shmaltz | http://www.lsi.com/storage_home/products_home/internal_raid/megaraid_sata/megaraid_sata_1506/index.html |
| 01:38.57 | kavit | and cycle it |
| 01:39.27 | JT | groogs[h]: not really, it's just one full backup with lots of diffs |
| 01:39.28 | groogs[h] | JT: i agree though, i remember trying to restore from tapes years ago, its a huge PITA. |
| 01:39.32 | kavit | but it is an expensive way of doing it |
| 01:39.33 | J4k3 | shmaltz: Adaptec AAR-2610 or something like that... its actually a Dell-branded product |
| 01:39.36 | JT | which is fine if you are only using 1 piece of media |
| 01:39.45 | J4k3 | Adaptec never directly marketed the 6 port card themselves, just the 4 and 8. |
| 01:39.46 | groogs[h] | JT: yeah, it does require one piece of media |
| 01:39.58 | kavit | if something breaks I just chuck in the backed up hdd |
| 01:40.09 | J4k3 | but there was no attempt to cover up Adaptec's name on the device. |
| 01:40.10 | JT | jsolares: my condolences.. maybe one day you'll get real PRI there |
| 01:40.13 | rikstah | anyone? |
| 01:40.14 | groogs[h] | but its nice being able to use "cp" as your restore tool |
| 01:40.14 | kavit | and boot from it and have to restore only one week worth of backups |
| 01:40.50 | jsolares | well it's ISDN -> asterisk -> IP -> asterisk -> R2 -> wireless endpoints |
| 01:41.00 | J4k3 | well, its not the exact same card looking closer |
| 01:41.12 | JT | jsolares: why? |
| 01:41.22 | J4k3 | but... the CPU looks identical, the specs otherwise look identical.... I'd be sure to put a BIG FAN on that little heatsink :P |
| 01:41.37 | jsolares | well ISDN and the first asterisk box are in the central office, the IP is a fiber link hundreds of miles into the jungle |
| 01:41.49 | jsolares | the mighty jungle, alas no lions down here in guatemala |
| 01:42.05 | jsolares | they had those R2/wireless endpoints equipment already |
| 01:42.12 | JT | hmm |
| 01:42.12 | J4k3 | I saw a cougar last night I suspect... In Texas. |
| 01:42.13 | J4k3 | ;) |
| 01:42.21 | J4k3 | and I don't mean a Mercury |
| 01:43.04 | jsolares | they had R2 end to end with some 4/1 E1 compressors, but they were wasting E1 ports on their central office equipment |
| 01:43.12 | Brandon_W | JT - Wouldn't happen to know the default password for newly upgraded polycom SIP phones would ya? PlcmSpIp doesn't seem to work |
| 01:44.16 | JT | 456 |
| 01:44.46 | JT | 123 is the user menu password |
| 01:44.49 | Brandon_W | d'oh |
| 01:44.55 | Brandon_W | ha, excellent |
| 01:44.57 | Brandon_W | thanks sir |
| 01:45.03 | Brandon_W | been forever since I've had to do that |
| 01:45.10 | JT | all in the polycom sip administrator's guide :P |
| 01:45.35 | shmaltz | is this real: |
| 01:45.36 | shmaltz | http://store.3ware.com/?category=10&subcategory=8&productid=9650SE-2LP |
| 01:45.48 | J4k3 | 3ware's stuff has always been fly |
| 01:46.05 | Brandon_W | JT - So I see now. THanks :) |
| 01:46.13 | Brandon_W | I just now got the thing to update |
| 01:46.51 | JT | nice |
| 01:46.54 | JT | what model? |
| 01:46.59 | Brandon_W | IP 500 |
| 01:47.11 | Brandon_W | it didn't like several of the bootroms I threw at it |
| 01:47.29 | Brandon_W | and before that the ftp server didn't want to play nice |
| 01:47.59 | J4k3 | shmaltz: that 2 port doesn't do onboard battery if that matters to you |
| 01:48.11 | J4k3 | or well... it doesn't have "raid battery" support. |
| 01:48.12 | JT | shmaltz: bloody better be real for that price :) |
| 01:48.16 | shmaltz | J4k3, the batt doesn't bother me, I just want to know that it's real raid |
| 01:48.22 | J4k3 | its real |
| 01:48.25 | J4k3 | its coprocessed |
| 01:48.51 | J4k3 | you'll get a full assortment of firmware based bugs, no worry ;) |
| 01:48.59 | shmaltz | J4k3, so I take your word it will work :P |
| 01:49.00 | J4k3 | (nah, 3ware's stuff is generally awesome) |
| 01:49.05 | jsolares | meh i really hate R2 now, i added 2 "events" to pri isdn, codes 1 and 38, also added them to dialstatus and cdr and all for naught it was just a matter of suppressing that damn progress message |
| 01:49.19 | flewid | hum |
| 01:49.33 | JT | jsolares: R2 is inband signalling isn't it? |
| 01:49.35 | flewid | shmaltz: yah, same difference |
| 01:49.40 | jsolares | yep |
| 01:49.42 | J4k3 | shmaltz: 2 port also somewhat sucks because you can't do hot spares. |
| 01:49.44 | flewid | i can register the exact same extension through idefisk using sip though |
| 01:49.47 | flewid | just not this damn 7970 |
| 01:49.48 | J4k3 | if you're coloing, spares are nice. |
| 01:49.57 | shmaltz | J4k3, I wnat just RAID1 |
| 01:50.02 | notoriousrab1982 | does anybody know how to get asterisk to start automatically when you boot up your machine, to save typing asterisk -cvvv to get it to load? |
| 01:50.03 | shmaltz | flewid, you sure it applied it? |
| 01:50.09 | J4k3 | shmaltz: that'll definetly do the job then |
| 01:50.13 | jsolares | it works rather well with unicall |
| 01:50.19 | rikstah | Hi there, does anyone here have a Mac that would kindly like to test my Asterisk Desktop Manager application (i just need to know if the installer works and the app launches). It should take 1 minutes. I'd really appreciate it |
| 01:50.32 | jsolares | rikstah: send it over |
| 01:50.46 | jsolares | if it fucks up my macbook i'll go hunting for you tho |
| 01:50.54 | shmaltz | J4k3, will it work with 2.6.x? |
| 01:50.54 | rikstah | jsolares, thanks, that'd be great...1 second i'll give you a URL |
| 01:50.57 | jsolares | and i'll put you thru what R2 put me thru |
| 01:51.08 | *** join/#asterisk Caplain (n=shayne@pool-71-184-33-177.bstnma.east.verizon.net) |
| 01:51.12 | jsolares | :) |
| 01:52.58 | J4k3 | shmaltz: hrm... not sure, I've got a LOT more experience with FreeBSD than I do linux. |
| 01:53.43 | shmaltz | J4k3, according to 3ware linux 2.6 kernel (not disro based) should work, which means it's hardware and will work :) |
| 01:53.45 | shmaltz | thank you |
| 01:54.13 | jsolares | Mercestes: using V21 didn't fix the noises, not sure if it's less, we're using g729 and they are not really audible anymore but with gsm it's noise hell |
| 01:56.23 | *** join/#asterisk kiscokid (n=xxx@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 01:56.58 | shmaltz | http://www.pricegrabber.com/p__3ware_3WARE_9650SE_2LP_INT_SATA_II_RAID,__29265187 |
| 01:57.38 | Snake-Eyes | When a sip code 200 (success) is sent, does that mean the call would be considered "connected" by User Agent ? |
| 01:57.54 | shmaltz | quick vote: |
| 01:57.55 | J4k3 | painfully expensive, but you gotta pay to play... |
| 01:57.55 | shmaltz | this one: |
| 01:57.57 | shmaltz | http://www.pricegrabber.com/search_techspecs_full.php/masterid=34516012 |
| 01:58.03 | shmaltz | or this one: |
| 01:58.05 | shmaltz | http://www.pricegrabber.com/search_techspecs_full.php/masterid=24582167 |
| 01:58.27 | shmaltz | J4k3, this is for a ambulance dispatching system, it has to be expensive :P |
| 01:58.31 | J4k3 | I would not buy a via chipset for an intel cpu |
| 01:58.42 | J4k3 | I mean... Intel makes the best chipsets on earth |
| 01:59.05 | J4k3 | well, except for serverworks, which is basically like saying "that warehouse behind Intel" |
| 01:59.21 | shmaltz | J4k3, but I am going to use it with Digium cards, and in my experience Asus mobo are better for that |
| 01:59.23 | J4k3 | so the 965 wins |
| 01:59.33 | J4k3 | well |
| 01:59.37 | shmaltz | J4k3, what is serverworks? |
| 01:59.39 | J4k3 | I wouldn't buy an Intel branded 965 board |
| 01:59.59 | J4k3 | Serverworks is another manufacturer of chipsets, but they're generally priced pretty high. |
| 02:00.10 | flewid | does anyone know if there's a service to tell me what type of nat i'm behind? |
| 02:00.24 | JT | shmaltz: what sort of digium card? |
| 02:00.32 | shmaltz | JT, single span T1 |
| 02:00.40 | JT | i see |
| 02:00.45 | shmaltz | J4k3, why not 965? |
| 02:00.50 | shmaltz | JT, why? |
| 02:01.09 | JT | as if you'd go asus for something possibly supporting life-preservation operations |
| 02:01.19 | JT | like J4k3 said, intel chipsets... |
| 02:01.26 | J4k3 | shmaltz: I'd get an 965 board, but not an actual intel-branded motherboard. |
| 02:01.27 | JT | decent server brands |
| 02:01.31 | shmaltz | JT, don't worry there is a backup system |
| 02:01.46 | shmaltz | JT, like? |
| 02:01.49 | J4k3 | I'd likely get an Asus P2B (the new one) ;) |
| 02:01.55 | J4k3 | http://www.newegg.com/Product/Product.aspx?Item=N82E16813131030 |
| 02:02.12 | JT | shmaltz: supermicro |
| 02:02.21 | kiscokid | what's wrong with intel branded mboards? |
| 02:02.40 | drfreeze | I'm looking for VoIP providers with a good reputation for quality and service. Any suggestions? |
| 02:02.53 | shmaltz | drfreeze, nufone.net |
| 02:02.53 | JT | tyan are also not that bad |
| 02:03.02 | drfreeze | shmaltz: you use them? |
| 02:03.09 | J4k3 | kiscokid: nothing really, just generally lack features and they're designed for mass produced desktop systems (unless you get an actual $erver grade motherboard |
| 02:03.09 | shmaltz | yes |
| 02:03.12 | J4k3 | ) |
| 02:03.16 | shmaltz | drfreeze, yes I do |
| 02:04.04 | *** join/#asterisk Q-At-Home (n=Queue@S01060016b69e6ff5.ed.shawcable.net) |
| 02:04.16 | shmaltz | I am assuming T 775 is the same as LGA775, am I correct? |
| 02:04.35 | J4k3 | yes |
| 02:04.55 | J4k3 | put one together before? the new socket is neat |
| 02:05.13 | J4k3 | just make sure you know you've got it in there right before you go applying a bunch of assembly pressure. |
| 02:05.34 | kiscokid | j4k3: those "ball grid arrays" are cool |
| 02:07.09 | shmaltz | doesn't look like supermicro maeks a mobo with support for Intel Core CPUs |
| 02:07.11 | *** join/#asterisk stridernzl (n=neville@125-237-117-207.jetstream.xtra.co.nz) |
| 02:07.52 | JT | shmaltz: i know they make plenty that support dual and quad core xeon |
| 02:08.15 | shmaltz | JT, that's overkill for my app |
| 02:08.24 | J4k3 | well, buying an equal performance xeon generally costs a good chunk more money |
| 02:08.32 | J4k3 | so... thats a pretty significant additional expense |
| 02:08.58 | J4k3 | you could probably build a second somewhat weaker pcserver for the same amount as the additional costs to build a really nice xeon box. |
| 02:09.11 | shmaltz | so againg vote, intel or asus? |
| 02:09.38 | shmaltz | *again |
| 02:09.50 | J4k3 | the only two choices are those two mobos? (the intel 965, or asus via thingie?) |
| 02:09.56 | J4k3 | if so, I vote for the intel 965 for sure. |
| 02:10.05 | shmaltz | J4k3, unless you tell me another one :P |
| 02:10.05 | J4k3 | via chipsets are a hassle |
| 02:10.11 | J4k3 | the Asus P5B |
| 02:10.17 | J4k3 | $120ish at newegg |
| 02:10.30 | J4k3 | http://www.newegg.com/Product/Product.aspx?Item=N82E16813131030 |
| 02:10.32 | J4k3 | thats my suggestion |
| 02:10.42 | J4k3 | I've heard *Really* good things about it |
| 02:10.46 | J4k3 | and for $120 I damn well should. |
| 02:12.39 | shmaltz | J4k3, thanks I am taking that one, makes tons of sense, has 2 PCI 1 x16 and 2 x4 slots |
| 02:12.43 | shmaltz | perfect for me |
| 02:13.07 | shmaltz | sorry 1 x4 slot |
| 02:14.55 | J4k3 | yeah it seems to be set up for more 'professional' applications. |
| 02:15.16 | shmaltz | J4k3, the price3 and the looks is exactly what I want |
| 02:17.13 | *** join/#asterisk ksharpe (i=nid@port0136-adx-adsl.cwjamaica.com) |
| 02:18.09 | shmaltz | J4k3, this memory works with that Mobo? |
| 02:18.11 | shmaltz | http://www.pricegrabber.com/p__Antec_SOLO_Silver_Black_Mini_Tower,__20384690 |
| 02:18.16 | shmaltz | wrong link |
| 02:18.18 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 02:18.23 | shmaltz | http://www.pricegrabber.com/p__Kingston_1GB_PC2_5300_DDR2_DIMM_Memory,__8442609 |
| 02:18.32 | ksharpe | Hi all |
| 02:18.51 | ksharpe | I am a newbie to asterisk |
| 02:19.15 | shmaltz | ksharpe, welcome aboard |
| 02:19.16 | ksharpe | wanted to know how many SIP clients can asterisk handle without breakdown |
| 02:19.19 | ksharpe | thanks |
| 02:19.24 | illsci | 2 or 3 |
| 02:19.36 | shmaltz | kshrarpe, that depends on how many ppl fit in my office |
| 02:19.40 | shmaltz | or your office |
| 02:19.52 | ksharpe | I wanted to purchase a channel bank |
| 02:19.54 | shmaltz | which really depends on how big my or your office is |
| 02:19.55 | ksharpe | 16E1 |
| 02:19.56 | JT | illsci: what? |
| 02:20.01 | illsci | im joking... |
| 02:20.02 | ksharpe | 480 lines |
| 02:20.17 | shmaltz | kshrpe, then make sure you know what you are doing |
| 02:20.18 | JT | ksharpe: that's probably not a good idea anyway |
| 02:20.18 | illsci | JT.... what's up with the asterisk vuln.... have you guys discussed that in here at all |
| 02:20.19 | ksharpe | wanted to know if asterisks can handle that |
| 02:20.30 | JT | ksharpe: what if the server goes down |
| 02:20.35 | JT | ksharpe: better to do a cluster |
| 02:20.42 | shmaltz | ksharpe, in theory yes |
| 02:20.53 | ksharpe | ok |
| 02:20.57 | shmaltz | but like JT says, don't |
| 02:21.12 | ksharpe | JT: will do eventually but need to get this up and running |
| 02:21.20 | illsci | how do you cluster asterisk? would you just have two different boxes register to your voip provider? |
| 02:21.21 | ksharpe | another question |
| 02:21.50 | JT | ksharpe: well, 4 or 8 port PRI cards are not cheap, having another box isn't that much investment looking at the bigger picture |
| 02:21.57 | ksharpe | how many Quad E1 cards can a single server handle? |
| 02:22.15 | JT | ksharpe: digium do not recommend the use of more than 2 digium cards in a box |
| 02:22.39 | pfn | SEQNO cycled: 9043968 59227 |
| 02:22.41 | pfn | wtf is that |
| 02:22.42 | ksharpe | JT: asterisk can handle 240 simultaneous calls like that |
| 02:22.58 | illsci | I dont understand why you would want to do that... why would you get channel banks and the analoge cards rather than just buying new ip phones |
| 02:22.59 | ksharpe | JT: or do I need to tweak? |
| 02:23.08 | JT | pfn: something about sequency numbers |
| 02:23.11 | illsci | and have everything be digital |
| 02:23.12 | ksharpe | it's not phone lines |
| 02:23.19 | ksharpe | I am trying to do PRI to SIP |
| 02:23.19 | JT | ksharpe: probably with the right hardware specs |
| 02:23.30 | ksharpe | was told that it was the best way to go |
| 02:23.37 | ksharpe | calls coming in from PSTN |
| 02:23.40 | ksharpe | not from extensions |
| 02:23.48 | JT | ksharpe: so no channel banks? |
| 02:23.50 | ksharpe | infact there are no extensions |
| 02:24.05 | illsci | why would you ever want to get channel banks? |
| 02:24.08 | ksharpe | meaning physical people |
| 02:24.10 | JT | pfn: sequence numbers, rather |
| 02:24.21 | JT | illsci: because you have a lot of analogue lines... |
| 02:24.24 | illsci | i can see if it you spent like thousands of dollars on phones already... but if you havent why would you do that |
| 02:24.29 | pfn | hmm, looks like some recent rtp change in asterisk... |
| 02:24.31 | ksharpe | I was told asterisks can't handle more than 100 simultaneous calls |
| 02:24.48 | JT | ksharpe: not so, depends on hardware |
| 02:24.50 | ksharpe | I need 600 simultaneous calls |
| 02:24.53 | JT | and it's asterisk, not asterisks |
| 02:25.01 | ksharpe | my bad |
| 02:25.01 | JT | ksharpe: you should use multiple machines anyway |
| 02:25.32 | JT | illsci: sometimes the infrastructure precludes the use of ethernet |
| 02:25.42 | illsci | JT: like when |
| 02:25.45 | JT | illsci: or the cost of ip phones is too great |
| 02:25.46 | ksharpe | JT: So multiple machines max 2 Digium cards? |
| 02:25.46 | *** part/#asterisk hexa (n=hexa@modemcable189.156-200-24.mc.videotron.ca) |
| 02:25.56 | JT | illsci: cat 2 or 3 cable |
| 02:26.02 | illsci | werd |
| 02:26.03 | JT | illsci: long long runs |
| 02:26.23 | kiscokid | Did anyone here go to the Asterisk Bootcamp class offered by Digium? Is it worth $3000? |
| 02:26.33 | JT | lol |
| 02:26.43 | JT | stay in here long enough, it'll be a boot camp ;) |
| 02:26.46 | kavit | i'd say no kiscokid |
| 02:26.52 | illsci | kiscokid: it is if someone pays for you |
| 02:26.58 | kavit | all information is available online |
| 02:27.02 | kavit | just persevere |
| 02:27.09 | kavit | and teach yourself |
| 02:27.14 | pfn | well, it's worth $3k if you aren't going to spend a weekend doing it yourself |
| 02:27.14 | illsci | if you work for some company screwing you on a salary but offers training... why not |
| 02:27.18 | pfn | or if you're on the company budget |
| 02:27.23 | kavit | unless you have $3000 lying around |
| 02:27.35 | kavit | and you want to help digium with the asterisk effort |
| 02:27.45 | JT | and a cupboard full of CCNAs, MCSEs and A+s, etc :P |
| 02:27.48 | Brandon_W | JT - For as much as I hate polycom config files the phone works now... Time to go home several hours late :P |
| 02:27.59 | JT | Brandon_W: heh ok |
| 02:28.14 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 02:28.25 | kiscokid | ok, thanks for the advice |
| 02:29.12 | ksharpe | another newbie question - I am having trouble setting up an AGI app |
| 02:29.33 | ksharpe | when I call (using 7777) from a SIP ext. it answers but I hear nothing |
| 02:30.24 | ksharpe | any one know why that could be? |
| 02:31.08 | Q-At-Home | are you answering the channel first? (is that still required?) |
| 02:31.22 | ksharpe | I tried both |
| 02:31.36 | ksharpe | answer and also let the agi app answer |
| 02:31.41 | ksharpe | fyi its phpagi |
| 02:32.07 | shmaltz | I like this type of latency: |
| 02:32.08 | shmaltz | 3779/3779 67.80.x.x D N 5060 OK (28 ms) |
| 02:32.28 | shmaltz | they are on the same providers network, about 70 miles apart, this is nice |
| 02:32.41 | Q-At-Home | old faces and new faces, hi everyone... its been almost a year since I've been in here :) |
| 02:32.59 | shmaltz | Q-At-Home, I don't recall your name |
| 02:33.21 | JT | shmaltz: i prefer D N 5060 OK (30 ms) |
| 02:33.22 | JT | well |
| 02:33.23 | Q-At-Home | used to be Q-At |
| 02:33.25 | Q-At-Home | -work |
| 02:33.27 | JT | 30ms is not great atm |
| 02:33.33 | JT | but it goes down to 10ms :D |
| 02:33.34 | jsolares | lol |
| 02:33.47 | shmaltz | JT, what's wrong with 28ms? |
| 02:34.05 | flenders | has anyone used chanskype? |
| 02:34.05 | Q-At-Home | bkw still poke his head in here? |
| 02:34.06 | shmaltz | anyhow, conversations still sound good up to ~225ms |
| 02:34.13 | flewid | [-----------] |
| 02:34.18 | JT | shmaltz: for some reason i thought 30 was better than 28 :P |
| 02:34.20 | flewid | er :) sorry |
| 02:34.23 | JT | usually it's lower |
| 02:34.26 | shmaltz | Q-At-Home, last I have seen of him was on dev channel |
| 02:35.07 | ksharpe | any one can help with my AGI problem? |
| 02:35.28 | Q-At-Home | you sure you have the sound files needed to play to the channel? |
| 02:35.32 | flewid | hmm, i just noticed that the sip headers from idefisk show a rport=XXXXX option, whereas the sip headers from the cisco do not |
| 02:35.37 | ksharpe | its using text to speech |
| 02:35.48 | Q-At-Home | festival? |
| 02:36.00 | ksharpe | I believe flite |
| 02:36.11 | Q-At-Home | make sure its running as the right user |
| 02:36.17 | Q-At-Home | never used flite |
| 02:36.31 | ksharpe | trixbox 2.0 comes with it by default |
| 02:36.46 | ksharpe | I tried vanilla asterisk got same result |
| 02:36.48 | boch | how can i interrupt a playback with a dtmf tone ? |
| 02:36.52 | jsolares | i've never liked trixbox |
| 02:36.56 | ksharpe | :) |
| 02:37.11 | ksharpe | the weird thing is the vmware version works fine |
| 02:37.14 | Q-At-Home | dont use playback, use background I believe |
| 02:37.23 | JT | ~trixbox |
| 02:37.35 | jbot | Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org |
| 02:37.50 | ksharpe | lol |
| 02:38.05 | Q-At-Home | hahahaha |
| 02:39.38 | ksharpe | this might sound incorrect but I am in transition from windoze ...anyhow does festival need to be running as a service? |
| 02:39.52 | *** join/#asterisk jnfuller (n=joshfull@S01060016cb851f2c.vs.shawcable.net) |
| 02:39.54 | ksharpe | or is it standalone |
| 02:40.10 | ksharpe | just run outputs file and that's it |
| 02:40.13 | ksharpe | ? |
| 02:40.19 | jsolares | boch: in an AGI with STREAM FILE, in your dialplan background would be your best bet |
| 02:41.02 | boch | jsolares, but Background() waits for an extension number, that in my case is very probably that it doesnt exist |
| 02:42.04 | jsolares | maybe controlplayback |
| 02:42.12 | jsolares | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ControlPlayback |
| 02:43.07 | jnfuller | I'm looking at /main/frame.c in 1.4.3 and wondering comparing to /doc/rtp-packetization if there might be an inversion in the alaw and ulaw definitions. I can't get the allow=ulaw:10 to force codec to 10ms ptime no matter what I do. Has anyone here gotten this to work? We're currently hardcoding the ptime to 10ms by changing frame.c. |
| 02:44.03 | *** join/#asterisk asteriskguy (n=learnast@cpe-75-80-111-113.socal.res.rr.com) |
| 02:44.25 | asteriskguy | does anyone have any experience with HylaFAX+ and IAXMODEM? |
| 02:44.57 | Q-At-Home | I've been meaning to play with that... working ok? |
| 02:45.41 | jsolares | laters all |
| 02:45.49 | asteriskguy | Kinda |
| 02:46.06 | asteriskguy | Got it to work but not sure how it got my email address to send me the tiff file |
| 02:46.27 | Q-At-Home | thats a hylafax option, shouldnt have anything to do with iaxmodem |
| 02:46.55 | asteriskguy | but where is that located at? Is there like a config file that I need to edit? |
| 02:47.15 | Q-At-Home | sec, gotta go back in my brain |
| 02:47.27 | asteriskguy | lol...thanks Q-At-Home |
| 02:47.30 | JT | festival runs daemonised |
| 02:48.27 | ksharpe | thx |
| 02:48.31 | ksharpe | JT: thanks |
| 02:49.07 | *** join/#asterisk flashnet (n=flashnet@211.223.75.49) |
| 02:49.13 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 02:49.13 | ksharpe | what the difference between Asterisk 1.2.x and 1.4.x? |
| 02:49.25 | asteriskguy | bug fixes |
| 02:49.25 | [TK]D-Fender | ksharpe, 0.2.x |
| 02:49.31 | asteriskguy | lol TK |
| 02:49.53 | Q-At-Home | http://www.hylafax.org/man/current/faxrcvd.1m.html |
| 02:50.22 | ksharpe | [TK]D-Fender: just for testing :) |
| 02:50.34 | asteriskguy | kharp, I know it has SLA while 1.2 doesn't |
| 02:50.48 | [TK]D-Fender | ksharpe, There have been plenty of articles, changelogs, etc. go READ :) |
| 02:51.07 | ksharpe | asteriskguy: thx |
| 02:51.23 | *** join/#asterisk fx0 (n=fx0@cypher.punk.net) |
| 02:52.21 | shmaltz | which asteisk TAPI program would you recommend? |
| 02:52.21 | asteriskguy | Q-At-Home, this weird because in my FaxDispatch I only have FILETYPE=Tif and SENDTO FaxMaster |
| 02:52.34 | Q-At-Home | does faxmaster exist as a user |
| 02:52.38 | [TK]D-Fender | Major core rewrite. major SIP changes + SLA (looks like a bastardized hack right now), AEL2, users.conf (one big bastard family now), I believe they changed SCCP/skinny significanlty. Redisndged build process that does away with excess makefile manipulation for apps. |
| 02:53.06 | asteriskguy | no |
| 02:53.14 | asteriskguy | or at least not that I know of |
| 02:53.16 | Q-At-Home | an alias at least? |
| 02:53.39 | Q-At-Home | setup an alias or change faxmaster to a real user/working alias |
| 02:54.02 | asteriskguy | ok, I'll check that tomorrow |
| 02:54.09 | ksharpe | [TK]D-Fender: thx ... I have been reading everything else works except what I want it to do...which run a simple AGI script |
| 02:54.29 | Q-At-Home | also check that you have a working MTA on the box |
| 02:54.34 | asteriskguy | MTA? |
| 02:54.37 | Q-At-Home | i.e sendmail or such |
| 02:54.42 | illsci | mail transfer agent |
| 02:54.42 | asteriskguy | oh...I do |
| 02:54.44 | Q-At-Home | mail transfer agent |
| 02:55.07 | Q-At-Home | check the logs, see if its even trying to send the message, should see some undeliverables |
| 02:55.13 | asteriskguy | it was able to send the test fax to my email in tif formant |
| 02:55.23 | Q-At-Home | cool |
| 02:55.24 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 02:55.24 | asteriskguy | I just wasn't sure how it got my email |
| 02:55.49 | Q-At-Home | usually through a pipe, or a sendmail local delivery thing |
| 02:55.56 | asteriskguy | I was trying so many different things, I got kinda lost and wasn't sure what works |
| 02:56.02 | Q-At-Home | :) |
| 02:56.14 | asteriskguy | I guess I'll scrap everything and rebuild |
| 02:56.32 | Q-At-Home | nah, thats a bit extreme |
| 02:56.51 | Q-At-Home | do this, echo test |mail faxmaster |
| 02:56.54 | Q-At-Home | see if you get the message |
| 02:57.12 | Q-At-Home | if not, edit your aliases and add one |
| 02:57.20 | Q-At-Home | rebuild your aliases with "newaliases" |
| 02:57.26 | Q-At-Home | then try again |
| 02:57.42 | Q-At-Home | or, change FaxMaster in your FaxDispatch to root |
| 02:57.52 | Q-At-Home | sombody gets roots mail I hope :) |
| 02:58.01 | Q-At-Home | err somebody |
| 02:58.30 | Q-At-Home | anyone from link2voip here? |
| 02:59.27 | asteriskguy | Ok I'll try that tomorrow when I get to work |
| 02:59.29 | asteriskguy | Thanks |
| 02:59.32 | Q-At-Home | have fun |
| 03:00.08 | asteriskguy | I will. Thanks |
| 03:00.24 | asteriskguy | gtg, be back in 30 |
| 03:03.42 | illsci | hey what module listens on port tcp 2000 |
| 03:03.50 | illsci | and do you need it if you use iax2 |
| 03:04.38 | illsci | and what one listens on udp 4520... |
| 03:07.27 | ksharpe | what's the best newbie tutorial to setup asterisk, zapatel from scratch? |
| 03:07.50 | blitzrage | www.asteriskdocs.org |
| 03:08.40 | ksharpe | which is the best distro to use? |
| 03:08.50 | JT | more specifically |
| 03:08.51 | illsci | debian ofcourse |
| 03:08.52 | JT | ~thebook |
| 03:08.54 | jbot | thebook is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 03:08.55 | *** join/#asterisk d3wayne (n=deeewayn@c-68-62-209-143.hsd1.al.comcast.net) |
| 03:09.38 | ksharpe | would ubuntu server qualify as debian? |
| 03:09.43 | JT | ~thewiki |
| 03:09.45 | jbot | i guess thewiki is at http://www.voip-info.org/wiki-Asterisk |
| 03:09.45 | illsci | sure |
| 03:10.09 | kiscokid | anyone have anything good to say about AsteriskNOW? |
| 03:10.22 | ksharpe | thanks for the links |
| 03:11.08 | d3wayne | try it: www.asterisknow.org |
| 03:11.36 | *** join/#asterisk sevard (i=chuck-th@adsl-71-129-115-242.dsl.irvnca.pacbell.net) |
| 03:11.53 | JT | it might be an ok toy ;) |
| 03:12.54 | illsci | JT: what listen's on those ports... |
| 03:13.02 | illsci | tcp 2000 and udp 4520 |
| 03:13.03 | JT | what ports? |
| 03:13.23 | *** join/#asterisk aydiosmio (i=aydiosmi@judecca.aculei.net) |
| 03:13.42 | aydiosmio | how can I force * to return a 403? |
| 03:13.45 | jnfuller | the AN distro is really limiting and the gui isn't better than ssh |
| 03:13.51 | JT | whatever you put there i guess, illsci |
| 03:13.55 | aydiosmio | or under what circumstance would * return a 403? |
| 03:14.00 | JT | ssh > * ;) |
| 03:14.13 | illsci | well... there are some default modules that obviously are being loaded and listening on those ports |
| 03:14.20 | illsci | i doubt im using them so i want to disable them |
| 03:14.21 | [TK]D-Fender | ksharpe, best distro is whichever you feel most comfortable administering and installing the pre-requisites to compile * (you should avoid anay attempts with packaged binary *) |
| 03:14.27 | *** join/#asterisk visba (n=dca[lapt@c-24-8-53-17.hsd1.co.comcast.net) |
| 03:14.34 | JT | illsci: default modules that i don't have... |
| 03:14.37 | *** join/#asterisk Fieldy (i=FJbAxkSi@gentoo/contributor/Fieldy) |
| 03:14.44 | jnfuller | I like ubuntu 6.10 myself. We use it in our labs. |
| 03:14.46 | ksharpe | what about http://www.asteriskguru.org is that any good? |
| 03:14.50 | JT | illsci: is it definitely asterisk using them? |
| 03:14.55 | illsci | yes |
| 03:15.06 | [TK]D-Fender | ksharpe, I usually suggest CentOS / Slackware. Debian is very respectable as well so long as you actually compile *. |
| 03:15.07 | ksharpe | excellent |
| 03:15.16 | JT | must be some strange channel drivers or something |
| 03:15.18 | [TK]D-Fender | ksharpe, go read... THE BOOK |
| 03:15.21 | [TK]D-Fender | ~book |
| 03:15.23 | jbot | i guess book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 03:15.24 | illsci | stkn_: why do you say to compile everythign |
| 03:15.31 | illsci | stkn_: i meant... |
| 03:15.42 | illsci | TK weird... |
| 03:15.51 | JT | illsci: try the letter [ |
| 03:15.56 | JT | or character i should say |
| 03:16.10 | jnfuller | apt-get asterisk doesn't really get you anywhere. You need to go through the build and actually learn the dependencies. |
| 03:16.26 | JT | illsci: standard practice to compile asterisk |
| 03:16.26 | illsci | why |
| 03:16.32 | shido6 | :) |
| 03:16.33 | JT | illsci: prepackaged versions suck |
| 03:16.42 | JT | and might not include what you want |
| 03:16.57 | illsci | honestly I think it includes more than you might want |
| 03:17.04 | illsci | but then you can disable all the modules you dont use |
| 03:17.07 | illsci | with modules.conf right.. |
| 03:17.14 | illsci | and then it gets updated with the os... |
| 03:17.17 | JT | or include more |
| 03:17.23 | illsci | and its apt-get install asterisk |
| 03:17.24 | JT | they come with stupid defaults too |
| 03:17.29 | JT | yeah stuff that |
| 03:17.36 | JT | I choose when i upgrade asterisk |
| 03:17.38 | JT | not the OS |
| 03:17.45 | illsci | you do chose |
| 03:17.49 | blitzrage | w00t book |
| 03:17.59 | JT | blitzrage: so where is number 2? :P |
| 03:18.08 | blitzrage | in progress |
| 03:18.15 | *** part/#asterisk visba (n=dca[lapt@c-24-8-53-17.hsd1.co.comcast.net) |
| 03:18.19 | JT | heh |
| 03:18.37 | blitzrage | writing a book takes for-cocking-ever |
| 03:18.44 | [TK]D-Fender | illsci, because packaged bit only come with precompiled options and aren't necessarily geared for your kernel (like zaptel). |
| 03:19.00 | jnfuller | I've tried both methods and I find downloading the code and compiling is the better way to go. We need to lock ptime to 10ms in our labs to interface with nortel SST and you can't rebuild a binary to fix the issue. |
| 03:19.03 | blitzrage | and are usually out of date |
| 03:19.15 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) |
| 03:19.34 | blitzrage | it usually takes about 5 minutes longer to build as opposed to using pkg'd binary |
| 03:19.49 | jnfuller | Plus, I've got corporate security nattering in my ear over every * disclosure to make sure I'm running the latest/greatest. |
| 03:19.52 | ksharpe | in little experience the apt-get install asterisk doesn't work out of the box |
| 03:20.03 | ksharpe | it chokes on the mp3 module |
| 03:20.07 | [TK]D-Fender | illsci, and not having to wait forever for the pacakage maintainer to wake and realize that 15 releases have passed them by (that NEVER happens in the Debian world, EVER) |
| 03:20.12 | ksharpe | which you have to edit and comment out |
| 03:20.18 | ksharpe | compile works best |
| 03:20.20 | illsci | oh it does too |
| 03:20.32 | ksharpe | I used the ubuntu |
| 03:20.36 | illsci | you dont have to run stable infact hardly anyone does |
| 03:20.38 | ksharpe | and that's what happened |
| 03:21.19 | [TK]D-Fender | illsci, Still they are often (usually) behind with options you might want not available, etc. Do it from source, do it right, and KNOW what you're getting. |
| 03:21.34 | [TK]D-Fender | Use the Source Luke! |
| 03:21.44 | illsci | to each their own... |
| 03:22.00 | JT | illsci: good luck on your own :P |
| 03:22.58 | illsci | well, the majority of systems engineers and admins would probably agree using a packaging system is much better than trying to maintain a box where you compile everythign from source |
| 03:23.15 | illsci | so im far from being on my own in that respect |
| 03:23.24 | illsci | and apt in general is a very good package manager |
| 03:23.38 | JT | illsci: yes but we're not talking about everything. |
| 03:23.43 | JT | we're talking about asterisk |
| 03:23.46 | JT | i know apt is good |
| 03:23.47 | illsci | it works in a deterministic maner... |
| 03:24.01 | JT | i just find the packaged versions of it to be shit |
| 03:24.03 | jnfuller | Some things are good, like being able to use uname in a command line to download kernel source and headers, but * is such a moving target that it's really better to keep to major releases or svn |
| 03:24.52 | *** join/#asterisk jmacz (n=jmacz@201.244.170.241) |
| 03:25.01 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 03:26.14 | [TK]D-Fender | illsci, lets just say that you don't want to find yourself jumping * versions and having your configs obsoleted behind your back (since these sort of things DO happen), and take an otherwise working install and putting a newer untested version in its place in PRODUCTION just because of some abstract call of apt to update everything. |
| 03:26.19 | jnfuller | Does debian even have zaptel packages? |
| 03:26.35 | illsci | TK well learn to use apt |
| 03:26.38 | illsci | and that wont happen |
| 03:27.38 | illsci | you can configure it to not update certain packages should you choose to do so and then apt-get upgrade to your hearts content with no fear of that or its dependancies being updated |
| 03:27.55 | jnfuller | Are you asking a serious question or just distro trolling? |
| 03:28.10 | illsci | we were just discussing |
| 03:28.17 | JT | trolling |
| 03:28.18 | [TK]D-Fender | illsci, well you are welcome to run it as you like. We've just stated some of the reasoning behind ours. so on to bigger and better things no.... |
| 03:28.21 | illsci | oh god |
| 03:28.53 | JT | illsci: "learn to use apt" oh seriously |
| 03:28.53 | illsci | then i wont continue discussing it |
| 03:28.54 | JT | illsci: i've found the apt packaged version of asterisk to be a major disappointment |
| 03:29.04 | JT | anyway, people want different things out of their asterisk setup |
| 03:29.06 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 03:29.06 | *** mode/#asterisk [+o mog] by ChanServ |
| 03:29.06 | illsci | i said to each his own... and then "good luck on your own" oh seriously |
| 03:29.11 | [TK]D-Fender | jnfuller, Not even a distro fight. packaged * in any distro = bad in the eyes of most. Amongst the most experienced here, even more so. |
| 03:29.30 | JT | illsci: anyway, you can package your compiled version up if you want |
| 03:29.45 | J4k3 | hehe, using a distro is frowned upon by some here, I'm sure ;) |
| 03:29.47 | illsci | http://www.infrastructures.org/papers/turing/turing.html |
| 03:29.58 | J4k3 | then theres the mac users. |
| 03:30.01 | J4k3 | :P |
| 03:30.22 | JT | illsci: well you made it sound like the view that compiling asterisk was a minority view amongst asterisk professionals, the majority disagrees with that |
| 03:30.33 | [TK]D-Fender | J4k3, Oh God... "Why won't Zaptel compile?!?!", "Why can't I use my Mac Mini's modem as an FXO?!" |
| 03:31.01 | illsci | well... i wouldnt equate asterisk professionals with professional sysadmins then... or engineers |
| 03:31.11 | mog | packages in a distro are fine for things that are not your most important app that you want to hand deal with |
| 03:31.24 | J4k3 | [TK]D-Fender: "because apple is sux at teh unix?" |
| 03:31.30 | jnfuller | As long as you understand the limitations, a mac actually does quite well as an asterisk SIP-only pbx. |
| 03:31.31 | J4k3 | thats my answer at least |
| 03:31.34 | illsci | compile it... no one cares... let it go |
| 03:31.46 | jnfuller | But I still wouldn't fink install asterisk |
| 03:31.48 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 03:32.03 | J4k3 | jnfuller: well, PPCs are great little workhorses. |
| 03:32.27 | JT | illsci: your are being a troll |
| 03:32.30 | [TK]D-Fender | J4k3, Yeah... thats why Apple went Intel ;) |
| 03:32.32 | JT | most of us are sysadmins |
| 03:32.34 | illsci | i am not |
| 03:32.55 | [TK]D-Fender | mog, Surface mount I'm sure.... probably still in the anti-static bag ;) |
| 03:33.04 | mog | lol |
| 03:33.10 | illsci | you said it was a bad idea to use packages... i never said compiling was stupid or a bad idea... i mearly pointed out the benefits of using packages.. |
| 03:33.15 | mog | we have a g4 sawtooth at digium with some hw |
| 03:33.26 | J4k3 | [TK]D-Fender: well, x86 got fast ;) |
| 03:33.52 | illsci | and the fact that apt happens to work in a determinist manner unlike ./configure make make install |
| 03:33.58 | J4k3 | unluckily its still a fast hack, but its a hack that works so its hard to argue with |
| 03:34.00 | JT | illsci: yeah there are some |
| 03:34.07 | jnfuller | I'm not a sysadmin and I've tried both routes. I find it easier and faster to use a blend of apt and source when compiling asterisk. I use apt for dependencies and development environments, kernel updates, etc. and stick to the asterisk tarballs. It's a nice blend that is actually really fast once you get used to it all. |
| 03:34.14 | [TK]D-Fender | J4k3, Benchmarks had proven that for AGES the promice of PPC was a complete waste. Adobe released some very real numbers and Apple went to cry in the corner... |
| 03:34.15 | JT | illsci: umm, deterministic manner, sure you're not just buzzwording? |
| 03:34.33 | illsci | if you're a sysadm |
| 03:34.36 | illsci | you should read this paper |
| 03:34.39 | illsci | http://www.infrastructures.org/papers/turing/turing.html |
| 03:34.46 | JT | why? |
| 03:34.55 | [TK]D-Fender | JT, jnfuller , illsci, (me) : Ok, this topic has run its course and can only head down-hill. All valid point have been made. NEXT! |
| 03:34.56 | [TK]D-Fender | :) |
| 03:35.07 | illsci | moving on... |
| 03:35.12 | [TK]D-Fender | thank you :) |
| 03:35.28 | [TK]D-Fender | (hopefully all) |
| 03:35.46 | jnfuller | So did anyone have any comments on my sip packetization problem? |
| 03:35.55 | jnfuller | err rtp |
| 03:36.06 | J4k3 | [TK]D-Fender: well... the PPC wasn't ever a fast thinker, just a good data mover. |
| 03:36.13 | J4k3 | err ever = never |
| 03:36.18 | JT | illsci: they can't even spell infrastructure :) |
| 03:36.34 | J4k3 | those old macs will hustle I/O like a real server, by comparison to a PC from the same timeframe. |
| 03:36.48 | *** join/#asterisk wunderkin (n=kev@dslstat-ppp-95.fastq.com) |
| 03:36.48 | [TK]D-Fender | JT : OK, pull yer line in cap'n! |
| 03:37.01 | JT | eh? |
| 03:37.22 | [TK]D-Fender | JT : STOP TROLLING OR WE'LL USE YOU FOR SHARK-BAIT! |
| 03:37.48 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:37.51 | JT | [TK]D-Fender: not trolling, i just don't take advice from sites that cannot spell |
| 03:38.19 | [TK]D-Fender | jnfuller, So did I see that right that you needed 10ms SIP packets to interface with an SST? |
| 03:38.33 | jnfuller | Yeppers |
| 03:38.57 | *** join/#asterisk Utah_Dave (n=boucha@d103.digis.net) |
| 03:39.01 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:39.31 | jnfuller | What we're hoping for is that we can use allow:ulaw:10 to force this but it doesn't seem to work. I'm wondering if the packet assertation is just stub code at this point or if there might be an error in frame.c |
| 03:40.20 | [TK]D-Fender | jnfuller, Nasty.. esp if you had to route that over the internet. Packet overhead = stupid. * is locked to a single frame size per codec, isn't it? |
| 03:41.16 | jnfuller | Supposedly 1.4 allows specification of ptime in sip.conf, but seems to slave to 20 unless we edit frame.c by hand and change ulaw to 10 from 20. |
| 03:41.48 | JT | sounds like you have to interface with a piece of junk :/ |
| 03:41.49 | jnfuller | It's like the setting :10 gets ignored and I can't seem to figure out how the detection is parsed by chan_sip |
| 03:42.05 | jnfuller | Nortel SST, you could say that. |
| 03:42.20 | *** join/#asterisk chiardon (n=chiardon@hostip-79-244.axesat.com) |
| 03:43.00 | jnfuller | rfc-noncompliant POS |
| 03:43.09 | JT | what a pain |
| 03:43.10 | jnfuller | rolling out drafts into production, yay |
| 03:43.33 | jnfuller | We workaround by forking code, but I hate doing that |
| 03:43.57 | [TK]D-Fender | jnfuller, Nortel has a habit of operating under the terms "No Comment" rather that RFC. |
| 03:44.05 | *** join/#asterisk chiardon (n=chiardon@hostip-79-244.axesat.com) |
| 03:44.15 | JT | heh |
| 03:44.44 | JT | a guy from nortel came to an open voip meeting we had here "we need to better integrate our products with things like asterisk" |
| 03:45.01 | *** join/#asterisk chiardon (n=chiardon@hostip-79-244.axesat.com) |
| 03:45.34 | jnfuller | Funny, the static we get from them here in canada is that they won't even recognize * exists |
| 03:45.50 | *** join/#asterisk chiardon (n=chiardon@hostip-79-244.axesat.com) |
| 03:46.00 | [TK]D-Fender | JT : While their BCM is the biggest POS ever, and you have to pay like $300 just to be ALLOWED to talk SIP to it for 1 channel. |
| 03:46.18 | JT | [TK]D-Fender: nice |
| 03:47.08 | [TK]D-Fender | JT : I fought bids from Norstar & Avaya IP Office to get my * setup at work. head office got the Avaya (sellouts). |
| 03:47.18 | *** join/#asterisk chiardon (n=chiardon@hostip-79-244.axesat.com) |
| 03:47.20 | [TK]D-Fender | JT : 1.5 years later than us that is. |
| 03:47.24 | jnfuller | We have sn07 and sn09 loads in the sst and they can't even negotiate sdp with each other, the 07 will tank in a pukefest saying the packets are too big |
| 03:47.41 | JT | [TK]D-Fender: it's took 1.5yr longer to implement? |
| 03:47.59 | JT | jnfuller: thought about talking H.323 to them? |
| 03:48.16 | [TK]D-Fender | jnfuller, Personal suggestion : if you can run an intermediary server to do transcoding between your primary & the SST. |
| 03:48.18 | *** join/#asterisk chiardon (n=chiardon@hostip-79-244.axesat.com) |
| 03:48.28 | jnfuller | we're 100% sip/sip-t |
| 03:48.35 | [TK]D-Fender | JT : they are SLOW to move and go the most fixed boring ways. |
| 03:48.49 | JT | jnfuller: sure, but telco products usually work better on H.323 |
| 03:48.57 | JT | use the best tool for the job, and all |
| 03:48.58 | [TK]D-Fender | JT : They still think its cool to ALWAYS have their poor receptionist answer EVERY call, and IVR is a sad BACKUP. |
| 03:49.14 | [TK]D-Fender | JT : they are ass-backwards and catching up to YESTERDAY |
| 03:49.23 | JT | actually that's often good customer service |
| 03:49.34 | JT | nothing wrong with receptionist answering calls |
| 03:49.43 | JT | increases customer satisfication usually |
| 03:49.48 | JT | sure it depends on the market |
| 03:49.50 | JT | but sitll |
| 03:49.51 | JT | still |
| 03:50.11 | [TK]D-Fender | JT : its SLOW, it annoys me, and makes me feel guilty for always asking her to route my calls instead of having DID's, or an IVR I can puch an ext into within 2 sec flat. |
| 03:50.24 | JT | [TK]D-Fender: oh, no DIDs? well. |
| 03:50.47 | *** part/#asterisk Fieldy (i=FJbAxkSi@gentoo/contributor/Fieldy) |
| 03:50.52 | [TK]D-Fender | JT : Receptionist is a guilt-driven IVR that can never offer more for 99% of calls |
| 03:51.23 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 03:51.35 | [TK]D-Fender | JT : I want 1. Sales 2. Acc Rec . 3. Acc Pay. 4. Purchasing. # of the bloody person I want and not even wait for the instructions! |
| 03:51.51 | JT | heh :) |
| 03:51.56 | JT | some IVRs speed you up |
| 03:51.59 | JT | some slow you down |
| 03:52.03 | [TK]D-Fender | JT : A good IVR will nail down what you want 95% in the first menu on a single press. |
| 03:52.07 | JT | some make it near impossible to speak to an operator |
| 03:52.19 | [TK]D-Fender | JT : usually only TELCO IVR's seriously punish you :) |
| 03:52.23 | JT | yeah |
| 03:52.29 | JT | voice recognition |
| 03:52.34 | JT | what the fsck is wrong with dtmf |
| 03:52.40 | jnfuller | Ours is vr, it's painful |
| 03:53.09 | [TK]D-Fender | JT : I do entirely advocate the use of "0" at ALL TIMES to be able to reach a human. But like all things, that itself should be a choice the the caller. Empower them, not the other way around. |
| 03:53.10 | *** join/#asterisk bmg505 (n=leon@196.209.182.248) |
| 03:53.29 | jnfuller | I just shout fuck a few times into the handset and get connected to a human instantly |
| 03:53.37 | illsci | hahaha |
| 03:53.51 | [TK]D-Fender | JT : VR IVR's are the most annoying POS in the world and should only be used by the severy physically disabled. |
| 03:54.03 | [TK]D-Fender | jnfuller, Sweet |
| 03:54.13 | blitzrage | wow, I'm surprised Asterisk doesn't play a tone when you start one-touch recording |
| 03:54.23 | jnfuller | Yeah, it's especially sweet at work |
| 03:54.24 | [TK]D-Fender | jnfuller, That works on a number of street-corners as well ;) |
| 03:55.10 | JT | rofl |
| 03:55.12 | illsci | when i was at symantec i used to wish they used asterisk to automate calling customers to notify them of down devices... they paid a ton of money for this phone system that everyone hated |
| 03:55.45 | [TK]D-Fender | blitzrage, I don't want to meet your mom! |
| 03:56.00 | blitzrage | I just want... |
| 03:56.01 | [TK]D-Fender | blitzrage, ! ! ! |
| 03:56.02 | *** join/#asterisk logicwrath (n=some@c-68-60-121-112.hsd1.mi.comcast.net) |
| 03:56.23 | JT | illsci: down devices? |
| 03:56.29 | [TK]D-Fender | blitzrage, There's something seriously wrong with the fact that still isn't getting old with us :) |
| 03:56.34 | illsci | yeah.. like firewalls or ids sensors... |
| 03:56.43 | illsci | this was at their mssp, their soc |
| 03:56.44 | blitzrage | [TK]D-Fender: no, it just means we're awesome |
| 03:56.50 | *** join/#asterisk techie (n=gus@voip.routedsystems.com) |
| 03:56.55 | [TK]D-Fender | blitzrage, truely and utterly |
| 03:57.19 | blitzrage | positively |
| 03:57.23 | file | I don't want to know your name |
| 03:57.27 | [TK]D-Fender | blitzrage, so any chance you could get me a roster for that * training camp thingy here next month? ;) |
| 03:57.32 | illsci | they had sla's on notifying customers of down devices... so the engineer would get an email and then have to find a possibly custom call list and then call it until someone answered and tell them their device was down |
| 03:57.33 | blitzrage | I just want ! ! ! |
| 03:57.45 | blitzrage | [TK]D-Fender: not really my thing... |
| 03:57.54 | blitzrage | [TK]D-Fender: contact Lisa, as she'll have the info |
| 03:58.02 | [TK]D-Fender | file, I don't have time to talk, make sex, go shopping... |
| 03:58.10 | blitzrage | lol |
| 04:00.16 | blitzrage | yah, I should probably go to bed soon |
| 04:00.26 | jnfuller | Hey speaking of training, has anyone here done the dCap? |
| 04:00.59 | aptura | a roaster? |
| 04:01.18 | [TK]D-Fender | blitzrage, file : on the topic of dCap..... new blade came in :D |
| 04:01.31 | file | ut roh |
| 04:01.50 | [TK]D-Fender | http://www.mantisswords.com/kin_tori_katana_and_wakizashi.htm |
| 04:02.13 | jnfuller | is the bootcamp worthwhile? |
| 04:02.24 | file | [TK]D-Fender: through the streets you got chopping people up? |
| 04:02.53 | russellb | i prefer the highheelcamp |
| 04:02.53 | [TK]D-Fender | jnfuller, Nobody I can think of cares about it, and its a question of you... |
| 04:03.10 | [TK]D-Fender | russellb, one day... in band-camp....... |
| 04:03.19 | russellb | :) |
| 04:03.23 | blitzrage | I prefer the naked boobs camp |
| 04:03.24 | jnfuller | I'm actually trying to get out of goinf |
| 04:03.26 | jnfuller | going |
| 04:03.43 | blitzrage | almost everyone learns something they didn't know at the bootcamp |
| 04:03.54 | russellb | i'd go. |
| 04:03.55 | blitzrage | I help teach it, and always pick something up new, or solidify something |
| 04:04.10 | russellb | i want to get my dCap too, hehe |
| 04:04.16 | [TK]D-Fender | jnfuller, it cause tooth decay! Every time someone gets a dCap, a faerie gets eaten by a croc! |
| 04:04.24 | russellb | :( |
| 04:04.31 | russellb | i probably would fail |
| 04:04.35 | blitzrage | nah... I jest, I jest |
| 04:04.37 | russellb | i never actually *use* this stuff |
| 04:04.37 | jnfuller | So since it's been forced upon me I should just cave |
| 04:04.47 | blitzrage | russellb: heh, I hear ya :) |
| 04:04.49 | file | russellb: rewrite the code so you pass! |
| 04:04.52 | blitzrage | russellb: that's why I keep sending you ideas |
| 04:04.58 | russellb | blitzrage: yep :) |
| 04:05.07 | blitzrage | I can't code, but I can implement like a banshee! |
| 04:05.15 | russellb | heck yeah |
| 04:05.31 | russellb | d3wayne: ^^^^ |
| 04:05.37 | [TK]D-Fender | blitzrage, leave your "implement" out of this, kthxplzbibi |
| 04:05.38 | file | so sleepy.... |
| 04:05.38 | russellb | blitzrage: i gave that one to d3wayne :) |
| 04:05.41 | blitzrage | d3wayne: THAT'D BE HAWT |
| 04:05.54 | russellb | unless he doesn't want it |
| 04:05.58 | russellb | or i just end up coding it tonight. |
| 04:06.03 | blitzrage | d3wayne: russellb just told me that I have to bug you everyday until you do something about it :D |
| 04:06.14 | blitzrage | russellb: do it while you have the motivation :D |
| 04:06.18 | russellb | heh |
| 04:06.30 | file | erm direct |
| 04:06.33 | russellb | it is ... 911 |
| 04:06.36 | blitzrage | file: you need teh zleep :D |
| 04:07.01 | jnfuller | I think you spelled erection wrong |
| 04:07.10 | file | blitzrage: so do you! |
| 04:08.09 | blitzrage | file: its true |
| 04:08.52 | d3wayne | blitzrage & russellb: I'm almost on it...not really, but it could be fun |
| 04:09.33 | d3wayne | file: don't fall asleep, freddy krueger will get you |
| 04:09.46 | jnfuller | babbling nutbars on team 1040am... I love it when announcers fill between periods and run out of things to say |
| 04:10.06 | blitzrage | d3wayne: totally cool! |
| 04:10.16 | blitzrage | crazy handy for clustering |
| 04:10.35 | blitzrage | thanks for Corydon for fixing the use of CUT() in a DUNDi mapping, I'm halfway there! |
| 04:10.43 | *** join/#asterisk Pinga-Fogo (n=zorra@201.86.90.135.adsl.gvt.net.br) |
| 04:10.43 | russellb | ha |
| 04:11.04 | russellb | i fought with that patch for a while |
| 04:11.07 | russellb | it felt dirty, heh |
| 04:11.14 | Pinga-Fogo | hello people, i'm a asterisk user from brazil ,, |
| 04:11.16 | blitzrage | russellb: oh yah? well thx! :) |
| 04:11.18 | Pinga-Fogo | hello for all, |
| 04:11.21 | russellb | but, i wasn't willing to offer a better solution, so in it went |
| 04:11.42 | blitzrage | now I can check for the number of channels in use on a remote system |
| 04:12.03 | russellb | dynamic weights? :) |
| 04:12.10 | blitzrage | nope, not using that |
| 04:12.13 | blitzrage | that's only in trunk right? |
| 04:12.15 | Pinga-Fogo | sorry for my intromission but i need some help, someone can help-me about disable echo cancelation in asterisk dialplan ? |
| 04:12.33 | russellb | you can't do it from the dialplan as far as i know |
| 04:12.53 | blitzrage | trunk_usage => trunk_usage,0,SIP,${GROUP_COUNT(${CUT(NUMBER|#|2)}@pbx)}-${GROUP_COUNT(${CUT(NUMBER|#|1)}@${CUT(NUMBER|#|2)})} |
| 04:12.59 | jnfuller | no, that's not done in the plan at all |
| 04:12.59 | blitzrage | ; dundi.conf |
| 04:13.04 | russellb | omg |
| 04:13.12 | Pinga-Fogo | ohh |
| 04:13.13 | russellb | blitzrage.... |
| 04:13.19 | russellb | good lord |
| 04:13.21 | blitzrage | russellb: its hawt |
| 04:13.39 | jnfuller | What's the EC problem you're trying to fix? |
| 04:13.51 | russellb | blitzrage: over my head :) |
| 04:13.54 | blitzrage | there is two lookups there: the first one is for the trunks for the PBX, and the 2nd is the number of trunks in use for a user |
| 04:13.54 | Pinga-Fogo | i have this situattion |
| 04:14.03 | Pinga-Fogo | i answer a voice call |
| 04:14.09 | blitzrage | russellb: what?! I wish I was going to AstriDevCon so I could show you my dialplan then |
| 04:14.13 | blitzrage | I'd boggle your mind |
| 04:14.19 | Pinga-Fogo | and after i transfer this call to extension *25 for example |
| 04:14.23 | russellb | i'd like to see it sometime |
| 04:14.25 | Pinga-Fogo | in extension *25 i need to |
| 04:14.30 | blitzrage | russellb: we'll have to have a conference call sometime |
| 04:14.33 | Pinga-Fogo | disable echo cancellation and call a sipfax |
| 04:14.35 | russellb | sweet |
| 04:14.56 | blitzrage | this kind of stuff is useful too: exten => s,n,Set(PBX_REMOTE_TRUNK_USAGE=${IF($[${ISNULL(${REMOTE_TRUNK_USAGE})}]?0:${CUT(REMOTE_TRUNK_USAGE|-|1)})}) |
| 04:15.06 | Pinga-Fogo | i need to disable echo canceller in a running call |
| 04:15.19 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:15.21 | blitzrage | it would make more sense if I showed you REMOTE_TRUNK_USAGE: |
| 04:15.22 | blitzrage | exten => s,n,Set(REMOTE_TRUNK_USAGE=${CUT(DUNDILOOKUP(${USERNAME}#${PBX}|trunk_usage|b)|/|2)}) |
| 04:15.46 | blitzrage | so I pretty much showed you the logic backwards :) that line above calls the DUNDi mapping I showed first |
| 04:16.22 | JT | Pinga-Fogo: sip fax, what the hell? |
| 04:16.31 | blitzrage | puts the return value (with the SIP/ stripped off) into the REMOTE_TRUNK_USAGE, then the 2nd thing I pasted splits based on the hyphen, and assigns 0 if DUNDILOOKUP returned NULL |
| 04:16.39 | jnfuller | You're using the voice codec to pass the fax inband? |
| 04:16.56 | Pinga-Fogo | yes |
| 04:16.59 | blitzrage | you people are seeing gold here, and not doing a thing :D |
| 04:17.04 | Pinga-Fogo | i'm using alaw codec |
| 04:17.18 | Pinga-Fogo | to pass faz in me local network |
| 04:17.20 | russellb | blitzrage: lol .. |
| 04:17.27 | Pinga-Fogo | ops |
| 04:17.34 | Pinga-Fogo | to send faz in me local network |
| 04:17.41 | Pinga-Fogo | in sip to sip faxes all is ok |
| 04:17.46 | jnfuller | to an ata termination? |
| 04:17.48 | blitzrage | I guess I'm one of the few who can read multiple embedded functions without glazed eyes :) |
| 04:17.51 | jnfuller | or to zaptel? |
| 04:17.58 | Pinga-Fogo | in ata is ok |
| 04:18.04 | Pinga-Fogo | the problem is in a zaptel termination |
| 04:18.15 | *** join/#asterisk DaveCanoe (n=Dave@adsl-065-007-135-002.sip.asm.bellsouth.net) |
| 04:18.17 | jnfuller | some ata will turn off the EC on the fy |
| 04:18.19 | jnfuller | fly |
| 04:18.19 | Pinga-Fogo | in the zaptel i need to disable echo cancelation before send or receive faz |
| 04:18.27 | Pinga-Fogo | yes |
| 04:18.42 | Pinga-Fogo | i nedd to turn off the zaptel ec on the fly (or on the dialplan) |
| 04:18.48 | jnfuller | is faxdetect set in zapata.conf? |
| 04:18.50 | JT | i don't understand why you'd be transfering a fax call |
| 04:18.56 | aptura | blitzrage was that one of your easier embeded functions? :) |
| 04:19.04 | blitzrage | yes |
| 04:19.08 | aptura | gee |
| 04:19.08 | blitzrage | funny enough it was ;) |
| 04:19.15 | blitzrage | let me see if I can find a nutzo one |
| 04:19.34 | aptura | That is the area I really need to concentrate on. |
| 04:20.45 | aptura | I am just trying to figure out my bracing as I think it is screwing with the dialplan and forcing the call to use trunkld instead of trunk. |
| 04:21.01 | jnfuller | do you often bridge sip to zap? |
| 04:21.15 | Pinga-Fogo | the faz detect not work on my case |
| 04:21.26 | jnfuller | You might want to mess with the taps and training values |
| 04:21.30 | JT | Pinga-Fogo: why on earth are you transfering a fax call? |
| 04:22.15 | jnfuller | toll bypass perhaps, fax over ip to termination, drops back into tdm |
| 04:22.40 | Pinga-Fogo | [JT]: is common in brazil, the people makes a voice call and request the faz signal |
| 04:22.41 | JT | there is no reason to transfer a fax call that i can think of |
| 04:22.47 | JT | you can detect it |
| 04:22.57 | JT | Pinga-Fogo: that sounds silly |
| 04:23.01 | blitzrage | this one is a bit odd..... |
| 04:23.03 | blitzrage | Set(HUNT_MEMBERS(${Q_ID})=${IF($[${EXISTS(${POS})} & ${POS} > 1 & ${POS} != ${COUNT}]?$[${POS} - 1]:0)}) |
| 04:23.31 | Pinga-Fogo | [JT]: silly ? sorry, i have a bad english hehehehe |
| 04:23.38 | blitzrage | although it doesn't really do anything fancy... it just sets the position of a loop and goes back to the beginning if at the end of the loop |
| 04:23.40 | jnfuller | It's not common but I've done it in a call a few times |
| 04:23.40 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 04:24.10 | Pinga-Fogo | jt immagine this case |
| 04:24.15 | Pinga-Fogo | a person callme |
| 04:24.24 | Pinga-Fogo | i answer the call on a sip phone |
| 04:24.36 | Pinga-Fogo | and the person request to me the faz signal |
| 04:24.41 | blitzrage | that is used in huntgroups, which is basically me creating a simple Queue() in dialplan logic. That as part of the looping routine which can start at any position in a list of available phones, and keeps track of where it started and when someone answered (think roundrobin-memory) |
| 04:24.45 | Pinga-Fogo | and i transfer the call to the faz extension |
| 04:24.55 | Pinga-Fogo | faz (faz) |
| 04:24.58 | JT | Pinga-Fogo: yes i've never seen any business that does that |
| 04:25.00 | JT | fax |
| 04:25.48 | Pinga-Fogo | [JT]: this is common in brazil ,, |
| 04:25.51 | aptura | yea no kidding. |
| 04:25.55 | Pinga-Fogo | and to solve my problem |
| 04:25.56 | jnfuller | It's done in the military from time to time, drop to fax and then back to voice, but I've never forwarded to a fax |
| 04:25.58 | Pinga-Fogo | i need to |
| 04:26.14 | blitzrage | on this one is just stupid... :) |
| 04:26.16 | blitzrage | exten => _X.,n,Exec(${IF($[${HASH(feature_status|vm)}]?${IF($[${ISNULL(${DIAL_FLAGS})}]?Set(DIAL_FLAGS=d):Set(DIAL_FLAGS=${DIAL_FLAGS}d))}:Verbose(1|Voicemail not active for ${USERNAME}@${PBX}(${USER_ID}) so not setting 'd' option in DIAL_FLAG))}) |
| 04:26.16 | Pinga-Fogo | disable the echo cancellation |
| 04:26.17 | kavit | hey I want to listen to analog signals from a fax machine... the crackly notes do it for me! |
| 04:26.25 | JT | Pinga-Fogo: why not just have a number just for faxing? |
| 04:26.33 | Pinga-Fogo | no |
| 04:26.35 | kavit | JT: you think I can start a fax 1900 sepcial number service |
| 04:26.38 | aptura | good grief |
| 04:27.13 | jnfuller | Can you hairpin the call back over zap to a landline? |
| 04:27.27 | JT | Pinga-Fogo: why not? |
| 04:27.48 | russellb | blitzrage: that's just insane |
| 04:27.52 | jnfuller | Or... set up a hylafax server |
| 04:27.54 | aptura | blitzrage pretty impressive |
| 04:27.57 | blitzrage | russellb: yes... yes it is :) |
| 04:28.14 | *** join/#asterisk JunK-Y (n=junky@74.56.205.105) |
| 04:28.37 | aptura | now can you figure out the rubics cube with your eyes closed? |
| 04:29.16 | Pinga-Fogo | [JT]: this is not common in brazil |
| 04:29.21 | aptura | Actually that is rubiks cube :) |
| 04:29.42 | blitzrage | If the STATUS of VM is active, then we check if the DIAL_FLAGS is NULL, if so, then we Set the DIAL_FLAGS to 'd', otherwise we set DIAL_FLAGS to what it was + d. If the STATUS of VM is inactive, then we Verbose() some text out saying we're not setting the 'd' option because VM isn't active (nests IF() functions) |
| 04:29.54 | Pinga-Fogo | the people calls the main number of the business |
| 04:29.59 | JT | Pinga-Fogo: it'd be smarter |
| 04:30.08 | jnfuller | http://www.voip-info.org/wiki-Asterisk+fax discusses the transfer to zap scenario |
| 04:30.11 | Pinga-Fogo | its occurs in 80% of brazil enterprises |
| 04:30.19 | blitzrage | what the IF() functions is really doing is passing back the Application to be executed by Exec() |
| 04:30.31 | JT | Pinga-Fogo: doesn't mean it's a good idea |
| 04:30.46 | jnfuller | looks like they use a separate context to force cancelling values |
| 04:30.56 | blitzrage | IF($[...]?IF($[...]?true:false):false) |
| 04:31.17 | blitzrage | the 2nd IF is executed if the 1st IF() is true, else, its not :) |
| 04:31.38 | aptura | What would be usefull is ast to dial its owner when somone steps in front of the motion camera. |
| 04:31.55 | JT | aptura: yeah it can do that |
| 04:32.02 | aptura | and which camera |
| 04:32.24 | JT | aptura: any |
| 04:32.28 | aptura | :) |
| 04:32.37 | JT | aptura: motion detection software for linux is freely available |
| 04:32.44 | JT | just get a CCTV DVR card |
| 04:32.49 | aptura | I need a spare server but have all the cctv gear here to set it up some time. |
| 04:32.51 | JT | and a camera |
| 04:33.46 | Pinga-Fogo | [JT]: i find a app to do all i nedd (zapEC) but i cant install this |
| 04:34.08 | JT | ed-209? |
| 04:34.25 | jnfuller | You could always steal a channel to use as outbound fax with no cancellation |
| 04:34.25 | [TK]D-Fender | http://en.wikipedia.org/wiki/ED-209 |
| 04:34.28 | joe-f | has anyone extended CDR to include meetme conference PIN's entered? (I know `lastdata` stores the CID.. but not PIN) |
| 04:34.35 | [TK]D-Fender | JT, yay motion detection! |
| 04:34.43 | JT | heh |
| 04:34.57 | Pinga-Fogo | Bristuff & zaphfc applications |
| 04:35.00 | *** join/#asterisk dhakatel (n=root@58.65.224.5) |
| 04:35.05 | jnfuller | there would be occasional collisions on inbound calls if the volume ramped up inbound. |
| 04:35.27 | JT | [TK]D-Fender: ah |
| 04:35.33 | jnfuller | but you could always forward inbound on the one channel to the other channels |
| 04:36.18 | JT | Pinga-Fogo: still sounds like a case where the business practice should be improved, not asterisk |
| 04:36.35 | JT | transfering fax calls, over sip, no less, sounds like a really stupid idea |
| 04:36.50 | JT | but if you have any luck, well great :) |
| 04:37.31 | jnfuller | Hmmm... can you reuse zap channels in zapata? |
| 04:39.12 | blitzrage | ok, bedtime! |
| 04:39.41 | Pinga-Fogo | [JT]: hmmm... the problem is,, how i change a business prectice from a a country ?? and i have the same problem on the credit card POS |
| 04:39.57 | Pinga-Fogo | when i need to disable ec on data calls |
| 04:41.47 | JT | Pinga-Fogo: it's amazing fax even works at all over sip |
| 04:41.48 | jnfuller | fractionalizing your pri across multiple contexts is probably the only way to pull that off. |
| 04:42.19 | JT | Pinga-Fogo: i know you have fax to email services in brazil, for example |
| 04:42.29 | JT | it's not that hard to tell someone to call a fax number |
| 04:44.20 | jnfuller | If you only have one real did it would be |
| 04:44.20 | Pinga-Fogo | [JT]: yes, its not soo hard |
| 04:44.20 | aptura | what would be the case where ast would use the _1NXX vs the _1800 dial context when dialing a 800 number? Mine is doign this and have no idea what the cause is. trunk=zap and of coure trunkld=sip/sipprovider info. |
| 04:44.20 | [TK]D-Fender | Pinga-Fogo, You would do what any sane person would do. you would get a completely seperate analog line for that fax/POS keep * the HELL AWAY FROM IT. |
| 04:44.32 | aptura | TK I thought ast has resolved the fax issues |
| 04:44.52 | jnfuller | This ignores the customer with an ancient fax machine that requires post-call handshaking |
| 04:45.31 | jnfuller | Which is pretty much every tactical fax on the market |
| 04:45.35 | aptura | ohh k so it is a protocol issue |
| 04:46.05 | JT | aptura: ast can work in some capacities with fax, it still doesn't have T.38 endpoint support |
| 04:46.53 | Pinga-Fogo | if asterisk have t38 endpoing all my problems go out ... |
| 04:46.53 | Pinga-Fogo | heheheh |
| 04:46.54 | jnfuller | Not all fax machines have dedicated phones built into them. Some are passthrough devices. Boat anchors, yes, but you would be amazed at some of the trouble tickets I get. |
| 04:47.24 | JT | jnfuller: what do you mean by dedicated phone? |
| 04:47.34 | *** join/#asterisk ber123 (i=brad@neu.cow.org) |
| 04:47.55 | ber123 | anyone have popping/clicking issues from ztdummy in dual CPU conferencing application? |
| 04:47.57 | jnfuller | Some fax machines do not have the capability to originate calls and are passively sitting on the line awaiting user intervention |
| 04:47.59 | aptura | JT shaw digital has fax support. I wonder what shaw did to resolve it. Wife wanted to try it and the box is about as big as a polycom phone and contains a battery. I have not tryed fax across it but its reported to work. |
| 04:48.19 | JT | aptura: they probably don't run asterisk |
| 04:48.40 | aptura | Jt probebly not but some thing simular. |
| 04:48.42 | jnfuller | Hi, I call you, you answer... hey, can we fax? okay, let's go... bee doo bee daah (drop into fax mode) done, back into voice call |
| 04:48.53 | JT | aptura: similar, as in costing heaps more :) |
| 04:49.14 | aptura | what ever the case Shaw is winning the minds of its customers. |
| 04:49.14 | JT | jnfuller: i'm glad no-one is retarded enough to do fax calls like that in Australia :) |
| 04:50.05 | jnfuller | What ata's do shaw use? They might have echo-cancel detection which would generate options, methods or reinvited |
| 04:50.08 | jnfuller | reinvites |
| 04:50.17 | aptura | the box is made by motorola. |
| 04:50.32 | aptura | IT has 6 lites on its front |
| 04:50.38 | JT | even sipura ATAs support T.38, supposedly |
| 04:50.47 | aptura | half the size accomidated the 4 hour battery. |
| 04:51.04 | JT | aptura: does it run off a cable tv network? |
| 04:51.06 | aptura | I should get a pic of it and send it your way. |
| 04:51.09 | aptura | yes |
| 04:51.09 | jnfuller | I've only ever gotten inband to work with a sipura box |
| 04:51.15 | jnfuller | using ulaw |
| 04:51.26 | JT | aptura: does the cable tv network even have power backup for 4 hours? |
| 04:51.34 | Pinga-Fogo | jnfuller this is all i need but after go to faz mode i dont need to back to voice |
| 04:51.49 | Pinga-Fogo | only hangup the call |
| 04:52.05 | aptura | yes. Alpha UPS boxes on the power poles. JT what city do you live in thay may be on your power poles. |
| 04:52.17 | JT | aptura: sydney |
| 04:52.25 | aptura | well disregard then :) |
| 04:52.29 | JT | how big are these UPSes? |
| 04:52.45 | aptura | Alpha is more of a local company in the states and have there boxes here in Canada. |
| 04:52.51 | jnfuller | http://broadband.motorola.com/catalog/productdetail.asp?ProductID=513 <= is that the shaw ata? |
| 04:52.58 | aptura | Look like 2.5 feet square. |
| 04:53.22 | aptura | jnfuller no |
| 04:53.27 | aptura | that is flat. |
| 04:53.27 | aptura | :) |
| 04:53.47 | JT | anyway, if you actually need a reliable service, you wouldn't go with cable |
| 04:54.10 | aptura | sbv5220 |
| 04:54.33 | *** join/#asterisk r_open (n=Administ@58.65.224.5) |
| 04:54.35 | aptura | http://broadband.motorola.com/catalog/productdetail.asp?ProductID=463 |
| 04:54.40 | jnfuller | dear god it's hideous |
| 04:55.11 | *** part/#asterisk r_open (n=Administ@58.65.224.5) |
| 04:55.18 | aptura | its a little big |
| 04:55.19 | aptura | ) |
| 04:55.22 | aptura | :) |
| 04:55.22 | jnfuller | Ya, built in ata fax negotiation |
| 04:55.43 | JT | aww, no operator transfer? ;) |
| 04:56.09 | asteriskguy | JT do you know anything about HylaFAX+? |
| 04:56.19 | aptura | its a great fax server |
| 04:56.37 | jnfuller | Here's an idea, Pinga-Fongo... What about dumping the sip call to an ATA hooked up to a fax machine, say a sipura 3000 or something and letting it take care of the echo-cancelling? |
| 04:56.38 | JT | i know it does some fax termination |
| 04:56.54 | asteriskguy | yeah, I need to build about 50 or so IAXMODEM to work with it |
| 04:56.58 | ber123 | anyone have experience with ss7 |
| 04:56.59 | *** join/#asterisk paavum (n=Dorphals@200.71.58.39) |
| 04:57.02 | jnfuller | Dump the AD/DAC out of asterisk |
| 04:57.04 | paavum | Hello |
| 04:57.21 | paavum | Im running * 1.2.18 with latest zaptel 1.2 |
| 04:57.28 | asteriskguy | aptura, where do I configure the email on HylaFAX+ so it send to certain people? |
| 04:57.44 | paavum | yet when I finish a call I allways get a message that reads BAD EIP value |
| 04:57.51 | paavum | and then just hangs up the comp |
| 04:57.55 | Pinga-Fogo | [jnfuller]: i have a pap2 ,, in the pap 2 the ec is turned off and all is ok |
| 04:58.13 | Pinga-Fogo | the problem is in the analog PTSN lines |
| 04:58.18 | paavum | kin anybody help me out? |
| 04:58.24 | Pinga-Fogo | all ptsn lines is conected to my asterisk ins a tdm04 |
| 04:58.57 | ber123 | bad EIP sounds like buffer overflow in some part of the code |
| 04:59.09 | ber123 | i would try recompiling with a diff version of asterisk |
| 04:59.12 | ber123 | just to test |
| 04:59.21 | paavum | oki... |
| 04:59.32 | asteriskguy | JT do you know? |
| 04:59.45 | ber123 | what is generating the bad EIP error |
| 04:59.51 | ber123 | asterisk or the kernel module zaptel |
| 04:59.54 | JT | asteriskguy: no |
| 05:00.02 | asteriskguy | I got it to work today but don't know how it got my email address |
| 05:00.18 | paavum | it says something about zaptel just before it hangs |
| 05:00.21 | *** part/#asterisk Utah_Dave (n=boucha@d103.digis.net) |
| 05:00.22 | asteriskguy | oh ok...cool...I guess I'll retry again tomorrow then |
| 05:00.24 | asteriskguy | :) |
| 05:00.48 | ber123 | you compiled the zaptel from source? |
| 05:00.53 | paavum | yes |
| 05:01.01 | ber123 | is it a brand new version? |
| 05:01.01 | paavum | on an SMP kernel machine |
| 05:01.09 | paavum | yes |
| 05:01.21 | ber123 | i would revert back to a known stable zaptel |
| 05:01.27 | ber123 | to see if that is the issue |
| 05:01.29 | paavum | which is...? |
| 05:01.34 | asteriskguy | paavum: is it ABE or OpenSource? |
| 05:01.46 | paavum | OpenSource |
| 05:02.20 | ber123 | i've been having issues with ztdummy on dual cpu box today |
| 05:02.27 | ber123 | i had to downgrade to single CPU until i can figure it out |
| 05:02.46 | aptura | JT your a dial issue mistro? |
| 05:02.49 | asteriskguy | oh, nevermind then, because I know the SMP for rPath/Poundkey Linux that comes with ABE has issues |
| 05:02.58 | aptura | mathching mistro that is? |
| 05:02.59 | aptura | :) |
| 05:03.12 | paavum | really? |
| 05:03.16 | *** join/#asterisk anderiv (n=anderiv@207-67-87-34.static.twtelecom.net) |
| 05:03.24 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 05:03.24 | ber123 | if its suspect SMP issue, then you can always build a parallel single cpu kernel to test |
| 05:03.27 | JT | ztdummy is crap |
| 05:03.28 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 05:03.33 | asteriskguy | according to Digium's tech, yes |
| 05:03.39 | JT | aptura: ? |
| 05:03.40 | ber123 | whats better than ztdummy for running a conf bridge |
| 05:03.41 | paavum | i'll use zaptel 1.2.16 |
| 05:03.52 | ber123 | with no digium cards |
| 05:03.54 | JT | ber123: real hardware zaptel timing, or app_conference |
| 05:03.59 | asteriskguy | well, that was the problem, we had a uniproc kernel before |
| 05:04.03 | ber123 | ah ok |
| 05:04.16 | ber123 | i guess i could slap a fxo card in there |
| 05:04.29 | paavum | If I downloaded an old version of zaptel |
| 05:04.34 | ber123 | ill check out app_conference too |
| 05:04.36 | asteriskguy | when we changed it to an SMP kernel there were some issues with it. But I think there's a fix coming out or even out already |
| 05:04.37 | paavum | should I download an old version of asterisk as well? |
| 05:04.39 | JT | you could always try compiling your kernel with Hz=1000, and maybe even PREEMPT |
| 05:05.04 | aptura | JT, its odd. TRUNK=zap/g1 and TRUNKLS=SIP/mywholesaleprovider:5060 yet when I dial 1800 the number is read by _1NXXN context and not _1800 context. does not make sence. |
| 05:05.46 | asteriskguy | aptura, did you have _1NXXN before _1800? |
| 05:05.49 | JT | aptura: i guess you'll need to pastebin extensions.conf |
| 05:06.27 | asteriskguy | asterisk uses the first pattern match that it finds |
| 05:06.38 | ber123 | really? i woudl think it would match on most specific route |
| 05:06.48 | JT | it depends |
| 05:06.53 | JT | i can't remember on what exactly |
| 05:06.58 | ber123 | hehe |
| 05:07.02 | JT | seems to be a bit of voodoo |
| 05:07.03 | asteriskguy | let me rephrase that, if I remember it right, the first specific match |
| 05:07.13 | aptura | In fact it was. So I switched the dial context to where the dial plan is reads _1800 first does not make sence. |
| 05:07.19 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 05:07.22 | JT | but try including the more specific match from another context |
| 05:07.24 | aptura | JT sure. give a sec. |
| 05:07.33 | paavum | If I downloaded an old version of zaptel should I download an old version of asterisk as well? |
| 05:07.40 | paavum | asteriskguy ... may I priv u? |
| 05:07.49 | asteriskguy | sure |
| 05:09.41 | asteriskguy | aptura, does it work? |
| 05:11.05 | aptura | asteriskguy It did but removed the 9 dial out issue no need for it. |
| 05:11.08 | aptura | http://www.pastebin.ca/458905 |
| 05:11.28 | aptura | I thought mabey it was the bracing but no. |
| 05:12.52 | aptura | local dialing is no issue. |
| 05:14.03 | asteriskguy | are you using TRUNKLD for your local also? |
| 05:14.04 | jnfuller | Oh man, I'm listening to the canucks game over the internet and it's been over forever. We already lost like an hour ago. Damn. |
| 05:14.20 | jnfuller | What crappy lag. |
| 05:14.28 | aptura | asteriskguy no just trunk. |
| 05:15.07 | aptura | jnfuller last nights Canucks win was nuts. Streets were swarming with fans |
| 05:15.14 | asteriskguy | have you check your SIP format? |
| 05:15.52 | ber123 | anyone have suggestions besides sipp for putting test calls over the network? |
| 05:15.54 | aptura | the dial plan worked perfectly. I just removed TRUNKMSD and removed 9. |
| 05:16.10 | jnfuller | Aptura: ha ha |
| 05:16.20 | ber123 | looking to stress test some systems with 200+ calls |
| 05:16.35 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 05:17.07 | asteriskguy | aptura, but ur not even using those in your dialplan |
| 05:17.28 | aptura | no thay were removed. |
| 05:18.27 | aptura | what was exten => _91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) |
| 05:18.38 | *** part/#asterisk dhakatel (n=root@58.65.224.5) |
| 05:18.40 | aptura | is now _1866NXXXXXX,1,Dial(${TRUNK}/${EXTEN}) |
| 05:19.07 | asteriskguy | oh |
| 05:19.12 | asteriskguy | ok |
| 05:19.18 | asteriskguy | makes sense now |
| 05:19.55 | asteriskguy | but technically that should've worked too |
| 05:20.32 | aptura | thats what i have been saying. I think its a bug and this is a older version. |
| 05:20.41 | asteriskguy | 1.2? |
| 05:21.00 | aptura | 1.2.7.1 |
| 05:21.20 | JT | aptura: that pastebin looks imcomplete |
| 05:21.30 | JT | aptura: are these contexts included elsewhere? |
| 05:22.22 | aptura | no. In fact its native context included with the install. I just modified trunk and trunkld to refelct what I was using. It worked with 9 but in this case, its crossing context. |
| 05:22.31 | jnfuller | g'nite all |
| 05:22.50 | JT | aptura: well your phone, what context does it send calls into? |
| 05:23.47 | Snake-Eyes | Any one have any idea's why asterisk would send a SIP 200 instead of SIP 183 for call thats still ringing and hasnt been pickuped (using the normal Dial() comand) ? |
| 05:23.50 | aptura | the ip500? dial plan is never altered on the phone. What is it you want to see because I dont think that is the issue. |
| 05:24.20 | JT | aptura: the ip500 must have its calls sent to a context in asterisk, which context is it? |
| 05:24.31 | JT | not interested in the ip500 dialplan |
| 05:24.59 | aptura | k hold |
| 05:25.55 | aptura | sipura ata doing it also. |
| 05:26.02 | JT | sure |
| 05:26.07 | aptura | I am going to return it back to what it was. |
| 05:26.13 | asteriskguy | Alright fellas, good night |
| 05:26.17 | JT | you did not paste enough of your dialplan for anyone to diagnose the problem. |
| 05:26.19 | asteriskguy | later JT |
| 05:26.30 | JT | you are leaving out critical bits of information |
| 05:26.35 | JT | asteriskguy: cya |
| 05:26.46 | *** join/#asterisk threat (n=phix@60-240-43-214.static.tpgi.com.au) |
| 05:28.19 | aptura | JT, take a look at that. |
| 05:28.22 | aptura | http://www.pastebin.ca/458920 |
| 05:29.13 | JT | why aren't you pasting your whole dialplan? are you trying to hide something? |
| 05:29.26 | JT | because you still have not included the context that the sip phone uses |
| 05:29.31 | JT | which is vitally important |
| 05:30.17 | JT | also looks like the remnants of freepbx |
| 05:30.25 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 05:30.27 | Snake-Eyes | This is wierd UA->SER->*->PSTN Term , PSTN sends SIP 183 while the end phone rings. Thus you're not "connected" until they pick up, but go the opposite direction: PSTN -> * -> SER-> UA, Asterisk sends out a SIP 200 back to PSTN when the call gets to it so the PSTN phone "connects" while * and SER are still ringing the UA. |
| 05:34.59 | aptura | okay 1800 goes though zap but it still for some off reason want to hit sixtel |
| 05:35.08 | aptura | anyway im outa here. |
| 05:35.13 | JT | aptura: yeah uhuh |
| 05:35.17 | aptura | I will deal with this tomarro. |
| 05:35.23 | JT | aptura: next time try not to make our job so hard :P |
| 05:36.02 | aptura | JT, version 1.2.1 is online. I think if you google extentions.conf 1.2.1 it might come up. Same one as mine with some minor changes. |
| 05:36.24 | aptura | version 1.2.7.1 that is |
| 05:36.31 | aptura | night. |
| 05:36.35 | JT | aptura: not sure what you're talking about, next time just pastebin the whole thing minus passwords |
| 05:36.46 | aptura | k will do tomarro. |
| 05:37.37 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 05:38.48 | paavum | thx bye |
| 05:45.15 | *** part/#asterisk Pinga-Fogo (n=zorra@201.86.90.135.adsl.gvt.net.br) |
| 05:47.37 | *** join/#asterisk flashnet (n=flashnet@211.223.75.49) |
| 05:58.13 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 06:05.56 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 06:10.31 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 06:11.49 | *** join/#asterisk grEvenX (n=even@ti500720a080-4710.bb.online.no) |
| 06:13.39 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.138.189) |
| 06:14.32 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.169.195) |
| 06:15.38 | *** join/#asterisk saftsack (n=oliver@p54A7088D.dip0.t-ipconnect.de) |
| 06:16.25 | *** join/#asterisk chris_1 (n=chris@ng1.kurtkrenn.com) |
| 06:16.42 | chris_1 | hello |
| 06:19.00 | chris_1 | * behind nat, and the ring status goes on, when I hangup the call |
| 06:19.10 | chris_1 | why this? |
| 06:24.03 | *** join/#asterisk af_ (n=getsmart@81-174-46-10.f5.ngi.it) |
| 06:37.14 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-267548.home.otenet.gr) |
| 06:37.35 | *** join/#asterisk JT_ (n=jon@unaffiliated/jt) |
| 06:37.50 | *** part/#asterisk remmo (n=junk@smack.isp.net.au) |
| 06:38.10 | *** part/#asterisk ksharpe (i=nid@port0136-adx-adsl.cwjamaica.com) |
| 06:39.43 | *** join/#asterisk Ast001 (n=uros@212.200.221.184) |
| 06:39.49 | Ast001 | Hello |
| 06:39.52 | *** join/#asterisk pagec (n=pagec@static-64-115-80-189.isp.broadviewnet.net) |
| 06:39.56 | Ast001 | I have one question for asterisk gurus |
| 06:40.57 | Ast001 | I have sound problem with moh |
| 06:41.02 | Ast001 | sound cuting |
| 06:41.12 | Ast001 | I did ping |
| 06:41.27 | Ast001 | and I wonder if this result may be reason for problems |
| 06:41.28 | Ast001 | rtt min/avg/max/mdev = 7.081/12.722/30.458/4.750 ms |
| 06:41.40 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 06:41.48 | Ast001 | min is 7 max is 30.45 |
| 06:41.51 | hi365 | anyone using the polycom 650? |
| 06:43.26 | Ast001 | also zttest gives me |
| 06:43.31 | Ast001 | Best: 100.000000 -- Worst: 85.339355 -- Average: 98.135376 |
| 06:44.16 | *** join/#asterisk santoshr (i=1063@203.199.110.93) |
| 06:44.29 | JT | Ast001: that's awful |
| 06:44.42 | Ast001 | what zttest or ping |
| 06:44.44 | Ast001 | or both |
| 06:44.47 | JT | zttest |
| 06:44.51 | JT | ping seems fine |
| 06:45.10 | Ast001 | i see |
| 06:45.13 | Ast001 | what can I do |
| 06:45.26 | JT | zttest should never go below 99.97% |
| 06:45.39 | Ast001 | ah oh |
| 06:47.03 | Ast001 | can I do something about it |
| 06:47.21 | santoshr | i am trying to run a Dial with the L option. how to set the the filenames "LIMIT_TIMEOUT_FILE" |
| 06:47.26 | JT | make sure you aren't sharing irqs with zaptel stuff |
| 06:47.56 | Ast001 | thereis no irq sharing according to /proc/interrupts |
| 06:48.21 | Ast001 | except IHC... and eth1 |
| 06:48.33 | Ast001 | but zaptel card does not share |
| 06:49.03 | JT | you might need to try moving the card around and removing unnecessary cards |
| 06:49.11 | *** join/#asterisk sashion (n=djbdsf@dsl-241-199-240.telkomadsl.co.za) |
| 06:49.21 | sashion | is this bad? |
| 06:49.22 | sashion | pbx.c:5738 pbx_builtin_serialize_variables: Data Buffer Size Exceeded! |
| 06:50.31 | Ast001 | At the moment I have 2 cards |
| 06:50.40 | Ast001 | 1 eth0 for internet and 2 eth1 for LAN |
| 06:50.43 | Ast001 | and isdn pri card |
| 06:51.21 | hi365 | anyone using the polycom 650? |
| 06:51.27 | *** join/#asterisk Benny_132 (n=benmarti@144.131.132.107) |
| 06:52.10 | Benny_132 | hi guys jsut a quick question when dialing out over isdn im getting a double ringing sound |
| 06:53.53 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 06:54.36 | Ast001 | one more question I used wireshark program and discover that every packet (when using gsm codec) has checksum error if wireshark |
| 06:55.17 | Ast001 | all packet which server send to peers are black and have warning checksum error |
| 06:55.32 | Ast001 | there is no error if I use ulaw alaw or ilbc |
| 06:57.32 | boch | when there are more than one sip active call my asterisk stops reading dtmf, do you know why? |
| 06:58.48 | *** join/#asterisk Ahrimanes (n=ma@81.7.159.2) |
| 07:06.16 | *** join/#asterisk mkl1525 (n=mkl1525@i59F76998.versanet.de) |
| 07:09.16 | liversmudge | morning all ... HELP!! ... setting callerid .. adhoc to withhold the number ... any ideas? |
| 07:09.49 | mkl1525 | Hi, (* 1.2 Setup: snom360 - * - E1 - ISDN) when calling out our phones nearly always miss the first 1-2 seconds of a call, using G711.a atm - did anyone experience such an behavior and has a hint how to turn off this problem? |
| 07:10.55 | liversmudge | Hi mkl we have the same * setup with E1 lines and have a few snom360's on the net in the office as test phones .. never had this problems you are getting |
| 07:11.25 | JT | are the phones behind nat or any type of firewalling? |
| 07:12.33 | liversmudge | there shouldnt be a problem with the nat side of things as I presume the phone has rung out b4 its answered .. and its on the answering side that mkl looses the data |
| 07:12.44 | mkl1525 | JT, no |
| 07:12.46 | liversmudge | sounds like a buffer is filling |
| 07:12.58 | JT | liversmudge: umm it makes a difference with sip calls |
| 07:13.08 | JT | especially if they don't negotiate nat properly |
| 07:13.53 | liversmudge | Im not fully up on this but I thought that the sip negotiation and data was sent over the same negotiated port |
| 07:13.58 | *** join/#asterisk fadey (n=fadey@239.Red-80-36-91.staticIP.rima-tde.net) |
| 07:14.14 | fadey | join #asterisk-dev |
| 07:14.20 | JT | err |
| 07:14.28 | JT | liversmudge: signalling is over udp 5060 |
| 07:14.47 | liversmudge | we have around 700 sip and iax connections on a few * boxes .. most (around 90%) are NAT |
| 07:14.55 | liversmudge | yes and no |
| 07:14.55 | JT | liversmudge: media uses the RTP protocol, over a pair of ports between 10000 and 20000 for default asterisk |
| 07:15.43 | liversmudge | exactly |
| 07:15.43 | JT | so what's the yes and no about it? |
| 07:15.43 | liversmudge | but the port on the nat side is negotiated at registration |
| 07:16.08 | JT | sure |
| 07:16.53 | boch | JT, i lose dtmf in 2 or more sip phones behind the same nat when using simultaeusly.. could this be related with a nat problem? |
| 07:17.41 | JT | not sure, it's possible i guess |
| 07:17.42 | *** join/#asterisk JoanaDoe (n=JoanaDoe@193.174.25.23) |
| 07:17.46 | JT | what dtmf mode do you use? |
| 07:18.08 | boch | tried with all supported, info rfc and inband |
| 07:18.25 | JT | what codec? |
| 07:19.10 | boch | ulaw |
| 07:19.26 | JT | hmm |
| 07:19.49 | JT | weird |
| 07:20.10 | boch | very |
| 07:20.31 | boch | when a phone hangs the other one recovers dtmf |
| 07:20.40 | boch | lol |
| 07:20.49 | JT | sounds like an outright bug |
| 07:20.57 | JT | dtmf, in what direction? |
| 07:21.05 | JT | and are the phones called or calling |
| 07:21.16 | boch | both phones calling the pbx |
| 07:21.58 | JT | what direction is the dtmf, and what asterisk version? |
| 07:22.14 | boch | asterisk 1.4.2, what do you mean by direction ? |
| 07:22.42 | JT | dtmf can be sent or received |
| 07:23.13 | *** join/#asterisk Curus (n=Curus@10.8.185.213.dk-amb.res.sta.perspektivbredband.net) |
| 07:23.14 | boch | the pbx cant read the dtmf |
| 07:23.27 | JT | sending dtmf from the phone then |
| 07:23.31 | Curus | Stupid question: Where do I report zaptel bugs? |
| 07:23.39 | JT | bugs.digium.com |
| 07:24.09 | Curus | I can't pick zaptel/wct... as a category |
| 07:24.39 | Curus | I can search for them, but not report bugs in those categories |
| 07:24.52 | JT | did you register? |
| 07:25.10 | Curus | Yes |
| 07:25.22 | JT | no idea then |
| 07:25.23 | Curus | I can report bugs in other categories, typically general |
| 07:25.55 | Curus | Maybe I'll just have to write to support@digium.com |
| 07:26.51 | liversmudge | Curus : did you say the hand sets are behind a NAT router? or on the same net as the pbx? |
| 07:27.19 | Curus | liversmudge: I think you got the wrong guy there |
| 07:28.01 | liversmudge | oops boch .. same question |
| 07:28.01 | liversmudge | sorry |
| 07:28.19 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 07:28.50 | fadey | Hi, everyone. Was wondering what is normaly a codebase for asterisk releases. Always thought that at some point a revision from the trunk was being announced a release |
| 07:29.39 | fadey | however now I' looking through the differences between today's svn trunk and asterisk-1.4.3 and there are quite a lot of |
| 07:29.59 | oej | For 1.4, the base is /svn/asterisk/branches/1.4 |
| 07:30.00 | JT | trunk is probably closer to 1.6, i think |
| 07:30.17 | oej | Trunk is the base for a new branch, which we use to produce a release |
| 07:30.23 | fadey | thanks |
| 07:30.28 | oej | But don't use SVN in production, since it may be buggy |
| 07:30.34 | boch | liversmudge, yes, a linux box |
| 07:30.50 | oej | We implement fixes in the 1.4 branch, test them and when we are done, produce a release on ftp.digium.com as a .tar.gz package |
| 07:31.52 | liversmudge | whats does your [sip show peers] look like? |
| 07:32.18 | liversmudge | morning all ... HELP!! ... setting callerid .. adhoc to withhold the number ... any ideas? |
| 07:32.25 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.169.195) |
| 07:32.28 | boch | liversmudge, both registered |
| 07:33.01 | liversmudge | can you paste the output to pastebin.com and show me the URL |
| 07:33.08 | JT | pastebin.ca |
| 07:33.12 | JT | pastebin.com is awful |
| 07:33.13 | liversmudge | either |
| 07:33.16 | liversmudge | realy |
| 07:33.21 | liversmudge | Ill stop using it |
| 07:33.31 | JT | pastebin is so incredibly slow, and it has horrible formatting |
| 07:33.34 | JT | .com |
| 07:33.36 | JT | that is |
| 07:33.57 | liversmudge | ah we use the formatting for some code |
| 07:34.00 | boch | liversmudge, there is no errors, the output is normal like there were no problem |
| 07:34.20 | JT | liversmudge: paste.rafb.net is better than pastebin.com if you need that stuff |
| 07:34.21 | liversmudge | ok then |
| 07:34.48 | liversmudge | Firefox can't find the server at paste.rafb.net. |
| 07:35.17 | JT | ~pb |
| 07:35.30 | 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 |
| 07:35.30 | JT | must be a differen url |
| 07:35.30 | JT | different |
| 07:35.31 | liversmudge | Ill google it |
| 07:36.11 | liversmudge | http://rafb.net/paste/ |
| 07:36.18 | liversmudge | and very nice too |
| 07:36.33 | JT | pastebin.ca is still king :) |
| 07:36.35 | Keltus | is it just me, or does the asterisk web UI not work in IE and Opera (onyl firefox) |
| 07:37.03 | JT | Keltus: it might be only you using the gui here ;) |
| 07:37.16 | Keltus | oh |
| 07:38.11 | liversmudge | funny that Keltus when I tried the gui ... in a time of total loss of personal control .. only IE worked for me!!! |
| 07:38.30 | Keltus | it doesn't save the service providers when I edit them |
| 07:38.35 | Keltus | but in firefox it does |
| 07:38.38 | liversmudge | it was an older version of IE .. not that ie7 crap |
| 07:38.39 | Keltus | in opera, it doesn't even let me log in |
| 07:38.43 | Keltus | I have IE6 |
| 07:39.20 | JT | sounds pretty compatible ;) |
| 07:42.09 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-40-188.socal.res.rr.com) |
| 07:48.09 | Keltus | argh |
| 07:48.34 | Keltus | I just want to set up asterisk with ipkall (free WA number) to get a test call |
| 07:48.43 | Keltus | so I need something like freeworlddialup too? |
| 07:49.40 | chris_1 | hello |
| 07:50.44 | Keltus | yeah, washington |
| 07:50.48 | JT | not sure what ipkall is, but don't see why it'd need FWD |
| 07:52.38 | Keltus | http://www.ipkall.com/ |
| 07:52.52 | Keltus | I think it takes a phone number, and forwards it to your SIP number |
| 07:53.03 | Keltus | it needs a "SIP proxy" |
| 07:53.10 | Keltus | can I need my own SIP proxy? aka my asterisk server |
| 07:54.00 | chris_1 | when i dial from my sip-phone out and hangup immediately - * doesn't hangup the call. |
| 07:54.12 | chris_1 | what to do? |
| 07:54.54 | chris_1 | sip & * is behind nat; dial out via sip-trunk. |
| 07:55.07 | JT | chris_1: where is the call going out to? |
| 07:55.14 | JT | connecting to a SIP ITSP, right |
| 07:55.22 | JT | no such thing as sip "trunk" btw :) |
| 07:56.19 | chris_1 | yes to a SIP ITSP |
| 07:56.48 | chris_1 | in sip.conf externip, localnet, nat=yes is set |
| 07:56.49 | JT | Keltus: you could probably just put your asterisk server in |
| 07:56.57 | JT | might need to accept unauthenticated calls though |
| 07:57.09 | *** join/#asterisk penguinFunk (n=penguin@87.224.86.46) |
| 07:57.17 | JT | chris_1: what sort of number are you calling via the itsp? |
| 07:57.53 | Keltus | JT: so my asterisk server is a "SIP Proxy"? |
| 07:58.10 | Keltus | what port does it run on, so I can turn on that port on the firewall |
| 07:58.16 | JT | Keltus: it isn't technically, but a B2BUA should be good enough :) |
| 07:58.23 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:58.35 | JT | udp 5060, udp 10000-20000 |
| 07:58.52 | Keltus | jesus, that's a lot of ports |
| 07:59.24 | JT | you can narrow it down in rtp.conf, but there's no need to unless you have a good reason |
| 08:00.50 | Keltus | how do I know my SIP phone number for asterisk? |
| 08:01.28 | JT | it just needs to be the number of a valid extension in the context that the sip call would come in though |
| 08:02.45 | Keltus | hmm... then do I need any service provider at all? |
| 08:02.58 | Keltus | so people can call that WA number, and it will ring asterisk... |
| 08:03.02 | JT | apart from ipkall, no |
| 08:03.03 | Keltus | I guess I should rtfm |
| 08:03.10 | JT | yes, asterisk is your service provider |
| 08:03.29 | Keltus | cause in the GUI, it asks you to put in service providers so it was a bit confusing |
| 08:03.39 | Keltus | but I think it's clear now... anyone wanna test calling me? |
| 08:03.53 | Keltus | I just have the default setup |
| 08:09.02 | oej | The GUI is very confusing |
| 08:10.51 | Keltus | seems too easy... I just install AsteriskNow and it's all done..? |
| 08:14.28 | *** join/#asterisk sashion (i=synergy@wbs-196-2-106-110.wbs.co.za) |
| 08:14.38 | sashion | ERROR[13859]: pbx.c:5738 pbx_builtin_serialize_variables: Data Buffer Size Exceeded! |
| 08:14.59 | sashion | can that error cause issues? I get it everytime app_queue makes an agent call |
| 08:24.15 | *** join/#asterisk saftsack (n=saftsack@pD9E065E8.dip.t-dialin.net) |
| 08:26.24 | *** part/#asterisk Ast001 (n=uros@212.200.221.184) |
| 08:28.14 | simplexio | Does anyone how method to play backround music and then stop it from AGIConnection |
| 08:30.55 | simplexio | every method i have tried play whole track before exiting or acceptin new dialplan commands |
| 08:31.25 | simplexio | another solution would be realtime moh configuraton but that dosen't exist. right? |
| 08:36.47 | *** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi) |
| 08:43.07 | *** join/#asterisk Ast001 (n=uros@212-200-217-62.smin.sezampro.yu) |
| 08:58.47 | *** join/#asterisk Polis_ttt (n=your@194-237-172-225-no48.business.telia.com) |
| 08:59.12 | *** join/#asterisk jazzanova (n=boris@S010600179a9d8a9d.vc.shawcable.net) |
| 08:59.22 | jazzanova | hi |
| 08:59.34 | jazzanova | how can I install asterisk on fedora core 3 ? |
| 09:00.32 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 09:04.31 | jazzanova | Anvil: can I upgrade to another fedora version ? |
| 09:07.39 | *** join/#asterisk abasic (n=alen@83-131-98-126.adsl.net.t-com.hr) |
| 09:07.48 | abasic | hello there |
| 09:08.12 | abasic | I need a help with real-time and postgresql |
| 09:09.02 | abasic | hello |
| 09:10.16 | abasic | anybody |
| 09:10.32 | jazzanova | i am here abasic |
| 09:10.35 | jazzanova | but i can't help you |
| 09:10.37 | jazzanova | :) |
| 09:11.08 | *** join/#asterisk mquin (n=mike@pdpc/supporter/active/mquin) |
| 09:11.23 | sashion | abasic: check http://www.asteriskguru.com/tutorials/realtime_pgsql.html |
| 09:11.25 | abasic | no probs man |
| 09:12.31 | abasic | done that |
| 09:12.39 | jazzanova | chmac: i don't have physical access to machine right now |
| 09:14.09 | abasic | i used that site ... but phones registration failed with msg: No matching peer found |
| 09:15.18 | abasic | user is in the table, a can read all datas with realtime load sipusers name |
| 09:15.46 | simplexio | abasic: google asteris postgresql realtime, and you get very good howto |
| 09:16.20 | simplexio | should read log before answering |
| 09:16.23 | abasic | ok thx man |
| 09:17.23 | simplexio | does it make connection to postgresql and do you hava extconfig.conf configured |
| 09:18.22 | simplexio | if it makes conenction to postgre and it still dosent find sip_peers nad you dont get any notice about db conenction failing, you didnt configure asterisk to actually look those sip users from db |
| 09:18.23 | abasic | exctonfig.conf is configured, and astersik is using odbc connection to postgres db |
| 09:18.39 | *** join/#asterisk hermuli (n=Eladamri@a88-112-255-26.elisa-laajakaista.fi) |
| 09:18.43 | simplexio | i have similiar configuration running |
| 09:20.29 | simplexio | are you sure that you have user right in db |
| 09:20.33 | abasic | i have db up and running, odbc working, asterisk running , exconfig have line sipusers => odbc,asterisk,sip_conf (asterisk from res_odbc) |
| 09:21.47 | simplexio | im not sure but dont you need sippeers => odbc,asterisk,sip_conf too in there |
| 09:22.06 | abasic | i have users in db, and i can read user datas from asterisk |
| 09:22.11 | abasic | i have that line too |
| 09:22.21 | abasic | i'm using asterisk 1.4.3 |
| 09:22.34 | simplexio | i think i have 1.4.2 |
| 09:22.51 | abasic | same problems is with 1.4.2 |
| 09:23.09 | abasic | what about db premissions |
| 09:23.19 | simplexio | that could be one too |
| 09:24.01 | abasic | db server and asterisk are on the same host |
| 09:24.21 | abasic | i can make insert in tables from that pc |
| 09:24.36 | simplexio | and select * and update |
| 09:25.00 | lenne_dk | Can I make a timeout between I dial out, and I get the response "...is making progress passing it to..." |
| 09:25.22 | abasic | insert and update yes, i made select user from asterisk |
| 09:25.27 | lenne_dk | My uplink is having problems, and I want to terminate the call and retry automatically |
| 09:43.34 | *** part/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 09:46.52 | *** join/#asterisk Greek-Boy (n=grb@196.46.105.176) |
| 09:49.19 | *** join/#asterisk nasls_lsa (n=chatzill@85.75.128.115) |
| 09:51.40 | *** join/#asterisk mquin (n=mike@pdpc/supporter/active/mquin) |
| 10:01.30 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 10:10.56 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 10:16.20 | nasls_lsa | hello , |
| 10:17.11 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 10:28.19 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 10:31.03 | Mahmoud | why can't I call "*33*4646#" this number? |
| 10:31.12 | Mahmoud | it's 100% an asterisk issue |
| 10:31.22 | Mahmoud | exten *33*4646#,1,answer() is not working |
| 10:31.27 | Mahmoud | exten *33*4646#,n,echo() |
| 10:31.33 | Mahmoud | exten *33*4646#, hangup() |
| 10:31.53 | Mahmoud | err exten *33*4646#,n,hancup() |
| 10:32.03 | Mahmoud | s/hancup/hangup/ |
| 10:33.04 | Mahmoud | asterisk is bitchy about * digit |
| 10:35.24 | sashion | can this cause issues? |
| 10:35.24 | sashion | ERROR[17902]: pbx.c:5738 pbx_builtin_serialize_variables: Data Buffer Size Exceeded! |
| 10:36.25 | Mahmoud | gaaah |
| 10:39.37 | *** join/#asterisk NormanAthol (n=filenotf@203.208.76.227) |
| 10:40.26 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:41.21 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 10:47.58 | menfin | is jbot a bot ? |
| 10:48.41 | *** join/#asterisk X-Rob_ (n=rob-x@58.167.128.40) |
| 10:48.52 | *** join/#asterisk Ifaistos (n=stelios@ipa226.211.tellas.gr) |
| 10:49.24 | nasls_lsa | I believe it is |
| 10:51.24 | *** join/#asterisk Dibbler_ (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com) |
| 11:06.20 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 11:07.19 | inspired | I have *21*XXXXXXXX# - how can I Goto *21*XXXXXXXX ? StripLSD is gone and I can't find the right way to manipulate the string into a substring |
| 11:13.45 | *** join/#asterisk Cabal (n=Cabal@unaffiliated/cabal) |
| 11:16.31 | *** join/#asterisk Ast001 (n=uros@212-200-217-62.smin.sezampro.yu) |
| 11:18.39 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 11:18.40 | Ast001 | can someone tell me ps can * write recorded calls to another linux machine |
| 11:19.34 | Ast001 | does savecallsin=192.168.1.4:/var/www/ |
| 11:19.34 | Ast001 | <PROTECTED> |
| 11:20.01 | mosty | Ast001: use samba or nfs |
| 11:20.14 | Ast001 | yes nfs must work on other machine of course |
| 11:20.38 | Ast001 | but I wonder what I need to write in agents.conf |
| 11:20.41 | e-ddie | i use ntfs |
| 11:21.06 | *** join/#asterisk mquin (n=mike@pdpc/supporter/active/mquin) |
| 11:21.13 | Ast001 | what did you write in agents.conf |
| 11:21.25 | Ast001 | savecallsin=x.x.x.x:/path ? |
| 11:21.42 | mosty | Ast001: nothing. just let it write to the default location, but mount an nfs or samba share there |
| 11:24.04 | Ast001 | i see |
| 11:24.08 | Ast001 | thak you mosty |
| 11:24.19 | Ast001 | one more question |
| 11:24.39 | Ast001 | at the moment zttest gives me unacceptable resulrs |
| 11:24.42 | Ast001 | results |
| 11:24.46 | abasic | need help with realtime asterisk and postgres |
| 11:24.59 | Ast001 | I have isdn pri card and 2 nerwork cards for LAN and internet |
| 11:25.15 | Ast001 | can i buy Cisco router and move out one network card |
| 11:25.31 | Ast001 | and have better results with interrupts |
| 11:26.29 | Ast001 | so cisco router to go to optic cable and other side or router to switch |
| 11:26.36 | abasic | anybody |
| 11:26.43 | Ast001 | is it possible ? is it big pain to configure ? |
| 11:26.57 | mosty | abasic: ask the question, maybe someone can help |
| 11:27.40 | mosty | Ast001: cisco routers can route, yes. will removing a NIC fix your interrupt problem? i don't know |
| 11:28.17 | abasic | unabe to match sip users in table and asterisk |
| 11:28.38 | abasic | got the msg: No matching peer found |
| 11:29.10 | mosty | abasic: what happens if you do "sip show peer <peername>" in the console? |
| 11:29.16 | abasic | nothing |
| 11:29.26 | mosty | set verbose 10 |
| 11:29.31 | mosty | then try again |
| 11:30.37 | abasic | 0 sip peers |
| 11:30.50 | Ast001 | I dont know what to disable in bios anymore |
| 11:31.00 | Ast001 | i wonder can i disable sound card |
| 11:31.10 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.voop.net) |
| 11:31.11 | abasic | realtime load sipusers name xxxx show user |
| 11:31.46 | mosty | abasic: did you setup extconfig.conf? |
| 11:32.12 | abasic | yeah; sipusers => odbc,asterisk,sip_conf |
| 11:32.25 | abasic | yeah; sippeers => odbc,asterisk,sip_conf |
| 11:32.49 | abasic | asterisk v. 1.4.2 |
| 11:32.55 | mosty | can you login to the odbc connection with isql ? |
| 11:33.31 | abasic | yes |
| 11:33.59 | abasic | postgres is on the same host as asterisk |
| 11:34.42 | mosty | turn on postgres logging so that you can see login attempts. do you see login attempts when you do a sip show peer foo? |
| 11:36.02 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:36.03 | abasic | how to turn on logging, i'm new with postgres |
| 11:36.57 | mosty | i think there are logging options with comments in postgresql.conf |
| 11:41.07 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 11:41.19 | JT | reading the postgresql documentation would probably be most fruitful |
| 11:45.05 | *** join/#asterisk saftsack (n=oliver@p54a7088d.dip0.t-ipconnect.de) |
| 11:47.40 | *** join/#asterisk MaximusXM (n=Miranda@81-208-83-242.fastres.net) |
| 11:47.47 | MaximusXM | Hiù |
| 11:48.47 | MaximusXM | anyone can help me? |
| 11:49.43 | *** part/#asterisk MaximusXM (n=Miranda@81-208-83-242.fastres.net) |
| 11:50.27 | *** join/#asterisk darkmug (n=dennis@143.106.7.170) |
| 11:56.09 | *** join/#asterisk _omer (i=_omer@9-237-154-202.wol.net.pk) |
| 11:56.10 | _omer | hi |
| 11:56.26 | _omer | any help or documents about SIP EXPRESS ROUTER |
| 11:57.54 | *** join/#asterisk frigidzephyr (i=frigidze@nat/digium/x-b4dca39305f6c4bf) |
| 12:01.30 | *** join/#asterisk champster (n=asterisk@AH.tescogroup.com) |
| 12:02.42 | *** join/#asterisk Fieldy (i=FYQMSb0f@gentoo/contributor/Fieldy) |
| 12:02.43 | synthetiq | www.iptel.org for set |
| 12:02.44 | synthetiq | ser |
| 12:03.02 | champster | I am using a macro to dial a "1" or "2" to our overhead paging system, and it takes to long for the voice traffic to connect. |
| 12:03.44 | champster | Is there either a way to speed it up, a way to lay a beep after the macro, or a way to keep the call in the marco? |
| 12:07.16 | _omer | my first day in SER ..... I am good in asterisk but SER is like french to me ;) |
| 12:07.28 | *** part/#asterisk joe_bag_odonuts (n=erne@dsl027-164-165.atl1.dsl.speakeasy.net) |
| 12:11.46 | abasic | mosty r u there |
| 12:13.05 | abasic | i manage somehow to read db log file |
| 12:13.37 | DrukenHME | _omer: ser is confusing if you dun know it... kinda like asterisk :) do your research..... |
| 12:14.47 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 12:15.20 | abasic | need help with real-time asterisk w/ postgres |
| 12:15.21 | *** join/#asterisk coppice (n=chatzill@94.143.17.210.dyn.pacific.net.hk) |
| 12:16.57 | abasic | anybody |
| 12:17.41 | *** join/#asterisk jcims (n=chatzill@cpe-24-210-60-100.columbus.res.rr.com) |
| 12:18.28 | abasic | hello |
| 12:19.23 | synthetiq | ask the question dont assume we know your question |
| 12:20.08 | jcims | hey folks, i'd like to build a small call broadcasting app for my church and the local ball teams. does anyone have something started already for asterisk? |
| 12:20.28 | jcims | everything i've seen so far is commercial. |
| 12:20.41 | abasic | voip phohe is unable to log to asterisk w real-time configuration No matching peer found |
| 12:21.22 | synthetiq | jcims as in multicasting |
| 12:21.23 | synthetiq | ? |
| 12:21.57 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 12:22.03 | jcims | sure, basically call a bunch of numbers on a predefined list to say practice is cancelled or whatever |
| 12:22.33 | abasic | <PROTECTED> |
| 12:22.42 | synthetiq | why make it an app when you can do it with a bash script |
| 12:22.57 | *** join/#asterisk Ifaistos (n=stelios@ipa226.211.tellas.gr) |
| 12:23.01 | jcims | i dont' want to show the coaches how to run a bash script. :) |
| 12:23.18 | jcims | it needs to be either entirely phone driven or done with some help with a web app |
| 12:23.28 | nasls_lsa | can anyone help me with mISDN ? |
| 12:23.29 | synthetiq | they call into the number and you do a system call to run the bash script |
| 12:24.10 | _omer | DrukenHME : asterisk was confusing too when I was newbie....:) |
| 12:24.19 | _omer | take care guys..see you later |
| 12:24.20 | _omer | Bye |
| 12:24.22 | jcims | so they record the message, asterisk saves it then kicks off a script to roll through the outcall numbers? |
| 12:24.36 | synthetiq | yup |
| 12:24.41 | jcims | sweet |
| 12:24.46 | champster | I am using a macro to dial a "1" or "2" to our overhead paging system, and it takes to long for the voice traffic to connect. |
| 12:24.48 | champster | Is there either a way to speed it up, a way to lay a beep after the macro, or a way to keep the call in the marco? |
| 12:25.02 | jcims | it's been a while since i've done anything with asterisk, i'm a little rusty :) should be a good project |
| 12:25.20 | jcims | thanks! |
| 12:25.53 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 12:26.35 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 12:26.50 | abasic | <PROTECTED> |
| 12:26.56 | abasic | need help |
| 12:27.11 | synthetiq | with english? |
| 12:27.34 | abasic | that too |
| 12:28.51 | champster | how do you play a sound file durring a call? I mean while two people a speaking. |
| 12:29.32 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 12:29.37 | lirakis | hello |
| 12:30.17 | lirakis | is there any way i can initiate a sip call from the command line? .. or possibly though some kind of script? .. like i want to call a certain # every 10 minutes or something from cron |
| 12:30.48 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
| 12:30.53 | frigidzephyr | champster: look at the application dial , u know like " show application dial" on the CLI, look at option L |
| 12:31.07 | frigidzephyr | champster: thats the only way I've heard of |
| 12:35.09 | *** join/#asterisk Zefk (n=Zefk@wsc-fo.b.astral.ro) |
| 12:39.21 | abasic | is anybody using asterisk 1.4 real-time w/ postgres? I can't register user from db on asterisk |
| 12:44.47 | tzanger | abasic: yes, but I'd suggest using ODBC |
| 12:45.34 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 12:46.31 | flujan | hi guys... I am using call-limit=1 on my sip.conf... Some users that are answering a queue sometimes appear in the Busy state. The queue is not clearing the state of the user to allow he/she to receive additional calls... |
| 12:46.41 | abasic | i do, i use res_odbc |
| 12:46.46 | flujan | how can I reset a specific channel status on asterisk? |
| 12:48.54 | tzanger | abasic: so why are you looking for a pgsql-specific variant? I'm curious? Is odbc not doing what you require? |
| 12:51.04 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 12:51.53 | abasic | i really dont know. I'm using odbc w posgres db. DB is up and running, asterisk is running ... |
| 12:52.26 | FreezeS | what't the dialed peer number is 1.4.2 ? (${DNID} isn't working anymore) |
| 12:53.17 | abasic | When i try to connect my phone i've got: No matching peer found |
| 12:58.16 | nasls_lsa | [Apr 26 15:59:53] NOTICE[19606]: chan_sip.c:5159 process_sdp: No compatible codecs, not accepting this offer! how can I see that codecs does my device supports ? I use allow=gsm allow=g723 allow=g729 at sip.conf and doesn't work :/ |
| 13:01.25 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 13:02.40 | mosty | nasls_lsa: login to your device |
| 13:03.38 | *** join/#asterisk tuxick (n=userMurf@tuxick.xs4all.nl) |
| 13:03.40 | tuxick | lo |
| 13:03.57 | nasls_lsa | I did , says that supports PCMU ( ulaw ) , PCMA ( alaw ) G.723.1 , ( g723 ) G.729(A/B) - g729 , GSM |
| 13:03.59 | tuxick | how to keep that "Cirpack Keepalive" noise out of log? |
| 13:04.13 | tuxick | and why doesn't set debug 0 do this? |
| 13:04.18 | nasls_lsa | I put at sip.conf disallow=all allow=gsm |
| 13:04.37 | *** join/#asterisk Dovid (n=Dovid@bzq-82-81-246-67.cablep.bezeqint.net) |
| 13:04.41 | nasls_lsa | and I get the error .. even if I say allow= to the other codecs too .. |
| 13:04.44 | mosty | nasls_lsa: did you reload? |
| 13:04.49 | nasls_lsa | yes , |
| 13:04.53 | nasls_lsa | many times :/ |
| 13:05.15 | Dovid | Tzafrir: Do u know by any chance know the number for Bezeq that I can dial that will tell me the number of the line that I am on ? |
| 13:08.45 | tzafrir | I don't know of one |
| 13:09.44 | tzafrir | But if you have an extra Asterisk connected to Bezeq, just call to that other Asterisk (or even a plain phone with callerid on) and see what is the caller number |
| 13:10.38 | tzafrir | nasls_lsa, in 'sip debug' you can see exactly what codecs both parties claim to support |
| 13:10.55 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 13:11.06 | tzafrir | though this is encoded in a binary bit field. 'show codecs' will show you the value of each bit |
| 13:11.51 | nasls_lsa | so , I restart the phone and see ... |
| 13:13.20 | *** part/#asterisk Ahrimanes (n=ma@81.7.159.2) |
| 13:13.34 | FreezeS | does anybody know what's the dialed peer number in 1.4.2 ? (${DNID} isn't working anymore) |
| 13:14.08 | FreezeS | I have a PRA card and I want to route the incomming calls to different contexts based on the dialed number |
| 13:14.39 | nasls_lsa | exten => _.51515151,1,Dial(SIP/102) |
| 13:14.45 | nasls_lsa | exten => _.51515152,1,Dial(SIP/103) |
| 13:14.52 | nasls_lsa | something like that FreezeS ? |
| 13:15.20 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 13:15.26 | *** join/#asterisk naitram (n=ttech@216.77.58.40) |
| 13:15.40 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:15.44 | jm|work | hello |
| 13:15.46 | jm|work | hello [TK]D-Fender |
| 13:16.02 | [TK]D-Fender | *yawn* |
| 13:16.10 | jm|work | how could I make something distinctive to know which trunk a particular call was going out on? In the ring or something? |
| 13:16.33 | jm|work | rather than announce it at the start with Festival or something |
| 13:16.50 | mosty | jm|work: different ringing sounds? |
| 13:17.05 | jm|work | mosty: for outgoing calls? Cool. Where should I look? |
| 13:18.07 | mosty | jm|work: well the dial command can play music on hold instead of ringing. use "music" files that contain your ringing sounds |
| 13:18.15 | mosty | so see the dial command in the wiki |
| 13:18.17 | candyman50 | Anyone know if it's possible to use multiple audio files in a Read() or GET DATA function? |
| 13:18.20 | naitram | I have a need to dial additional digits during a call. Imagine, im talking to bob and bob says open the gate. I would like to remain talking to bob and hit a series of keystrokes that are then acted upon by asterisk to run some scripts to open the gate. Im sure this can be done but I don't know enough about the dial plan. Help please. |
| 13:18.22 | jm|work | oh right, yeah. |
| 13:18.27 | FreezeS | nasls_lsa: I had something like exten => s,1,GoTo(${DNID},s,1) |
| 13:18.32 | jm|work | that's quite straightforward. Thanks :) |
| 13:18.56 | nasls_lsa | no ideas |
| 13:19.30 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
| 13:20.10 | [TK]D-Fender | naitram: lookup "dynamic features" and features.conf in the WIKI |
| 13:20.17 | champster | Does anyone know why I am getting "app_meetme.c: Cmdline: 5488|K"? It is kicking users from the meetme that I use for overhead paging. |
| 13:22.32 | naitram | [TK]D-Fender: ok will do, thanks |
| 13:23.08 | [TK]D-Fender | jm|laptop: Forget messing with MoH while you dial and a ton of other junk. Just play a fix recording saying "Using provider X" and be done with it. |
| 13:23.25 | candyman50 | candyman50: Answer to my own question --> no |
| 13:23.26 | [TK]D-Fender | candyman50: have you tried? |
| 13:25.17 | *** join/#asterisk Uatec (n=uatecuk@adsl.ntsols.com) |
| 13:25.20 | Uatec | Grettings |
| 13:25.23 | Uatec | Greetings |
| 13:26.40 | LeddyHM | I guess Uatec doesn't come in peace, he should be banned |
| 13:27.29 | champster | lol |
| 13:28.53 | [TK]D-Fender | Shoot to kill, Jim! |
| 13:29.28 | Uatec | I come in peace. |
| 13:29.38 | Uatec | LeddyHM, why would you assume i do not? |
| 13:29.52 | LeddyHM | because you didn't say so |
| 13:29.59 | Uatec | nor did you |
| 13:30.19 | LeddyHM | I did not say "Greetings" either |
| 13:30.30 | Uatec | that's just rude |
| 13:30.40 | Uatec | I actually come in peaceful inquisition. |
| 13:30.44 | LeddyHM | I'm not a sci fi person, but even *I* know the proper saying :) |
| 13:31.01 | Uatec | we're not in scifi |
| 13:31.05 | Uatec | we're in freenode |
| 13:31.22 | Uatec | in to how i can stop my asterisk from picking up the phone, whenever someone tries to send a fax? |
| 13:31.28 | [TK]D-Fender | Go in PIECES :D |
| 13:31.30 | LeddyHM | We also don't say "Greetings" |
| 13:31.30 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 13:31.51 | LeddyHM | regardless, don't get your panties in a bunch |
| 13:31.57 | [TK]D-Fender | Uatec: It either auto-answers all calls, or none at all. |
| 13:32.09 | Uatec | I have two numbers coming in on my ISDN line, one of which i want to answer, the other one i want to leave for the fax machine |
| 13:32.10 | champster | nanu-nanu |
| 13:32.16 | Uatec | even on different numbers [TK]D-Fender? |
| 13:32.17 | tuxick | Uatec: it can't know it's an incoming fax |
| 13:32.24 | champster | and shazbot |
| 13:32.34 | tuxick | ahah |
| 13:32.36 | Uatec | tuxick, yes it can in this case. cos the fax is coming in on a different number |
| 13:32.47 | tuxick | Uatec: that changes the question :) |
| 13:32.50 | Uatec | :) |
| 13:32.53 | Uatec | and the answer, i hope |
| 13:33.21 | LeddyHM | you shouldn't care, you've been shot :) |
| 13:33.32 | LeddyHM | however, I digress |
| 13:33.38 | tuxick | Uatec: don't make it listen on that channel |
| 13:33.48 | tuxick | but i just got started with asterisk, so can't help you there |
| 13:33.56 | Uatec | oh |
| 13:34.12 | Uatec | the problem is, both numbers come in on the same physical wire in to the back of my asterisk box |
| 13:34.41 | LeddyHM | ah hah |
| 13:34.58 | frigidzephyr | Uatec: is the fax just being passed through to an extension, or do you have some kind of fax answering\processing module\application |
| 13:35.10 | tuxick | Uatec: i'm sure they have port/channel numbers assigned to them |
| 13:35.27 | Uatec | frigidzephyr, we i have a separate fax hardware |
| 13:35.39 | frigidzephyr | Uatec: can't you just route the msns to a different extension? |
| 13:35.52 | Uatec | frigidzephyr, the fax hardware is not connected to asterisk |
| 13:36.09 | frigidzephyr | Uatec: is the fax hardware in front of asterisk, on the line before it? |
| 13:36.11 | tuxick | it should just be told to ignore that channel :) |
| 13:36.20 | Uatec | frigidzephyr, nope, it's parallel to it |
| 13:36.40 | frigidzephyr | Uatec: hmm, so both get rang at the same time? |
| 13:36.44 | Uatec | tuxick, thing is, it's on the same port... |
| 13:36.58 | Uatec | frigidzephyr, i can't. asterisk keeps picking up. |
| 13:37.33 | *** join/#asterisk squall (n=squall@ns2.squallnetwork.net) |
| 13:37.53 | *** join/#asterisk ctaloi (n=ctaloi@rrcs-208-125-53-58.nys.biz.rr.com) |
| 13:37.53 | frigidzephyr | Uatec: Asterisk will pick up , even if that msns is routed nowhere? |
| 13:37.57 | frigidzephyr | Uatec: not sure if you can do anything about that =D |
| 13:38.25 | tuxick | isn't it a matter of exten => somemsn,1,Ignore() or whatever the term is? |
| 13:38.43 | frigidzephyr | Uatec: See if theres a different way you can set all that up besides parallel? |
| 13:38.47 | Uatec | tuxick, i get a "no extension" message, and it hangs up |
| 13:39.04 | Uatec | frigidzephyr, no, becuase both numbers come in down the same line |
| 13:39.05 | tuxick | at least you're close then :) |
| 13:42.40 | *** join/#asterisk darylvoip (n=darylvoi@c-71-224-42-97.hsd1.pa.comcast.net) |
| 13:43.23 | frigidzephyr | Uatec: could you have asterisk answer that msns and pass it through to the fax machine?, instead of having them parallel? |
| 13:43.41 | frigidzephyr | Uatec: like call out another port? |
| 13:44.11 | [TK]D-Fender | Uatec: * can't be psychic. It has no way of knowing its a fax call until it answers, at which point its TOO LATE. |
| 13:44.19 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 13:44.26 | tuxick | Uatec: Busy or something? |
| 13:44.26 | frigidzephyr | TK: his fax calls come in on a seperate msns i think he is saying |
| 13:44.26 | [TK]D-Fender | Uatec: You're using a dumb analog line, right? |
| 13:44.26 | tuxick | frigidzephyr: have you located the context to use? |
| 13:45.36 | tuxick | uhm |
| 13:45.36 | tuxick | frigidzephyr: i meant Uatec :) |
| 13:45.36 | frigidzephyr | hehe |
| 13:45.37 | tuxick | Uatec: if you have that bit, i get exten => yourmsn,1,Busy(15) or so would do? |
| 13:45.38 | frigidzephyr | TK: i thought it said this was ISDN |
| 13:45.38 | [TK]D-Fender | Uatec: nvm, I see where you wtoe the answer to my question. |
| 13:45.38 | [TK]D-Fender | Uatec: What are you matching on for incoming calls? |
| 13:45.39 | *** join/#asterisk SnrWup (i=Neil@i-83-67-202-134.freedom2surf.net) |
| 13:45.39 | SnrWup | herro |
| 13:45.48 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 13:52.28 | naitram | [TK]D-Fender: could you look at this and maybe tell me why it doesn't work, features.conf line dooropen => #91101,self,Goto,MAST200 91101 |
| 13:52.38 | naitram | extension exten => 1101,1,Set(DYNAMIC_FEATURES=dooropen) |
| 13:52.50 | naitram | doesn't seem to do anything |
| 13:53.12 | naitram | MAST200 91101 does work if dialed |
| 13:53.16 | [TK]D-Fender | naitram: you also need to use one of the following in your Dial statement to have access to it : "tTwW" |
| 13:53.54 | naitram | [TK]D-Fender: ok will try, :) |
| 13:54.07 | [TK]D-Fender | naitram: And I also don't know EXACTLY what you can do in there regardless... I have never impleneted anything suing it. |
| 13:54.22 | [TK]D-Fender | implemented* |
| 13:54.24 | [TK]D-Fender | dskjhdfshkjdsfkhlja |
| 13:55.10 | champster | Does anyone know why I am getting "app_meetme.c: Cmdline: 5488|K"? It is kicking users from the meetme that I use for overhead paging. |
| 13:56.16 | [TK]D-Fender | champster: Maybe you should be showing us CLI output of this happening and the code you use to pgae in the first place.... We aren't psychic you know. |
| 13:56.28 | [TK]D-Fender | champster: and make sure to PASTEBIN IT. |
| 13:56.29 | [TK]D-Fender | ~pb |
| 13:56.39 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 13:57.17 | Brandon_W | oooo... overhead paging |
| 13:57.31 | Brandon_W | My bosses may get a kick outta that. :) |
| 13:58.11 | Brandon_W | Morning all |
| 13:58.17 | Brandon_W | even if it is 10am |
| 13:58.41 | SnrWup | [TK]D-Fender - is there any way to log all calls into a database and record calls so that even if a PSTN caller calls asterisk, speaks to SIP1, gets transfered to SIP2 and finally to SIP3, the entire route is logged in a database and also recorded to a sound file? so i can see later that Mr. Pstn caller called, got put through to sip1 for x minutes then to sip2 for y minutes and finally to sip3 for z minutes, and then i can play a wav of a |
| 13:58.55 | champster | I will do that. the odd thing is that it just started on Monday and there have been no changes for weeks. |
| 13:59.10 | [TK]D-Fender | SnrWup: Nothing so perfect out there, so NO. |
| 13:59.22 | SnrWup | (the transfers are effected by the aastra 9133i sip user hitting their transfer hardkey and transfering it) |
| 13:59.34 | SnrWup | [TK]D-Fender could such a system be cobbled together? |
| 14:00.00 | SnrWup | i dont expect it all on a plate, just if i know it can be done, i can spend the time trying to achieve it |
| 14:00.01 | [TK]D-Fender | SnrWup: It'd require massive rewrites to * code...... falls into the "Dream On" category. |
| 14:00.09 | SnrWup | oh |
| 14:00.36 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
| 14:00.36 | SnrWup | what do all the big corps do for call logging then? |
| 14:00.46 | SnrWup | they just lose data on callers when transfers happen? |
| 14:01.24 | ManxPower | There was some duscussion of changing the CDR stuff to track call paths, but i'm too groogy to remember much more |
| 14:02.32 | SnrWup | does GiantCrop Inc. use some other sort of PBX system then? |
| 14:02.41 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:02.49 | SnrWup | i can't see how u can run a proper customer database of calls without being able to track who the customer spoke to |
| 14:03.28 | SnrWup | especially... when u have say a receptionist who takes all calls, then transfers the call to x person if x person wants to speak to the caller |
| 14:04.27 | codefreeze | SnrWup: I'm in the middle of coding a new event logging system. Trouble with sip-ish stuff is that, if * isn't informed of a change, it can't be logged. If the sip phones xfer calls between each other without * having to get involved, then no dice. |
| 14:05.26 | SnrWup | codefreeze thing is... asterisk does actually know about it |
| 14:05.26 | SnrWup | for example, this is what came up in the asterisk console |
| 14:05.36 | SnrWup | SIP1 calls SIP2. SIP2 calls SIP3 and transfers SIP1 to SIP3 |
| 14:05.48 | SnrWup | there was a call from SIP1 to SIP2 to begin with, obviously |
| 14:06.01 | SnrWup | then the call from SIP2 to SIP3, again obviously (to do attended transfer) |
| 14:06.08 | SnrWup | then after the transfer was finally made |
| 14:06.13 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
| 14:06.17 | SnrWup | asterisk set up a call from SIP1 to SIP3 |
| 14:06.26 | SnrWup | (without SIP1 ever having made the call in reality) |
| 14:06.34 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 14:06.37 | *** join/#asterisk SECGOD (n=traderz@65.114.86.29) |
| 14:06.38 | codefreeze | OK, then you should be able to track them, SnrWup |
| 14:06.39 | SnrWup | and when asterisk created the SIP1 to SIP3 call, SIP2 was cut off |
| 14:06.51 | *** join/#asterisk joshaidan (n=brianj@thunderbay-voip-4.vianet.ca) |
| 14:07.00 | *** part/#asterisk SECGOD (n=traderz@65.114.86.29) |
| 14:07.24 | SnrWup | well it seems so, cause asterisk certainly generates shit for when this happens |
| 14:07.31 | SnrWup | but ive just got no clue how to harness whats going on |
| 14:07.41 | SnrWup | the dialplan seems pretty useless for finding out |
| 14:08.15 | [TK]D-Fender | SnrWup: Giant Corp doesn't use Asterisk. Deal with it and move on. |
| 14:08.34 | SnrWup | [TK]D-Fender what would they use? |
| 14:08.49 | codefreeze | [TK]D-Fender: the eternal optimist! ;) |
| 14:08.49 | [TK]D-Fender | SnrWup: Go do some market research and find out. |
| 14:09.04 | SnrWup | i don't mind paying tbh as long as i get decent functionality |
| 14:09.05 | ManxPower | can't you Set(__CALLCOOKIE=${UNIQUEID}) and then at the end of the call get that info into the CDR? |
| 14:09.41 | [TK]D-Fender | codefreeze: I'm a REALIST.... that means that my beliefs are based on facts, not hopes ;) |
| 14:09.49 | SnrWup | ManxPower problem is, there doesnt seem to be a way to link the resulting SIP1->SIP3 call to the original calls |
| 14:09.58 | SnrWup | the unique ID is totally different |
| 14:09.59 | ManxPower | SnrWup: CDRs are used for BILLING. Nobody cares that SIP1 was still part of the call since from a billing standpoint we only care about the call |
| 14:09.59 | ManxPower | SIP1 -> SIP2 and SIP2 -> SIP3 |
| 14:10.21 | SnrWup | so CDRs arnt really much help then |
| 14:10.29 | ManxPower | SnrWup: That is why we set CALLCOOKIE at the start of the first call. and never again |
| 14:10.56 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 14:11.07 | SnrWup | ManxPower and that variable remains the same even when its like SIP1 talking to SIP3? |
| 14:11.34 | ManxPower | for all calls processing you would check to see if CALLCOOFIE is not empty. the __ prefix says "set this in the current channel and all children channels |
| 14:11.34 | ManxPower | " |
| 14:12.03 | ManxPower | SnrWup: I've not done this, but the basic idea should be sound |
| 14:12.13 | codefreeze | SnrWup: There's accountcodes you can assign from the dialplan, which can help track, and you can set userfield... you have to play around and see if anything will help. |
| 14:12.14 | ManxPower | SnrWup: you should be reading README.variables |
| 14:12.31 | SnrWup | ManxPower did u get that from the wiki? cant find much on callcookie on google |
| 14:12.47 | [TK]D-Fender | SnrWup: ITS JUST THE NAME HE GAVE THAT VARIABLE. |
| 14:12.48 | ManxPower | SnrWup: CALLCOOKIE is just a random variable name I picked. |
| 14:12.52 | [TK]D-Fender | stupid capslock... |
| 14:12.59 | SnrWup | ohhhh i see |
| 14:12.59 | ManxPower | SnrWup: the Wiki should be the LAST place you look for actual documentation |
| 14:12.59 | *** part/#asterisk Aji-Dahaka (n=kjelderg@fluffy.dname.net) |
| 14:13.06 | SnrWup | ohhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhh |
| 14:13.16 | SnrWup | a transfered call is a child channel? |
| 14:13.16 | ManxPower | the first place should be the CLI and the docs directory of the asterisk source code. |
| 14:13.24 | SnrWup | well hell |
| 14:13.27 | SnrWup | that solves everything then |
| 14:14.10 | ManxPower | SnrWup: THAT is something I'm not totally sure of. I guess it might depend on the way the transfer happens. try it and see |
| 14:14.38 | SnrWup | cause i can just slot in a check into every sip to sip call to check whether the callcookie is set, if so, see if that cookie is in the database, if it is, log it as a transfer from the original |
| 14:14.43 | SnrWup | ManxPower yeah i'll give it a go thanks |
| 14:15.24 | *** join/#asterisk unik-rados (n=rados@c-68-62-71-239.hsd1.mi.comcast.net) |
| 14:15.25 | ManxPower | codefreeze: only girls use DTMF transfers! |
| 14:15.50 | Uatec | tuxick,frigidzephyr, i figured it out |
| 14:15.56 | simplexio | does anyone have idea howto define musiconhold class with agi conenction or howto play file on backround |
| 14:15.57 | Uatec | some numpty had disconnected the fax machine |
| 14:15.58 | Uatec | ACH! |
| 14:16.20 | frigidzephyr | Uatec: roflz isnt that the way it is sometimes |
| 14:16.29 | [TK]D-Fender | codefreeze: More like "Incest : A game te whole FAMILY can play!" |
| 14:16.37 | frigidzephyr | Uatec: 1. Check to see if its plugged in |
| 14:16.40 | frigidzephyr | Uatec: hehe |
| 14:16.50 | errr | with the CDR for mysql is it possible to set the name of the table that will be used, or is that something thats hard coded? |
| 14:17.41 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
| 14:17.42 | simplexio | or anyidea where i find moh realtime patch ? |
| 14:17.46 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:18.18 | Uatec | frigidzephyr, i checked with the guy. he said it was |
| 14:18.20 | Uatec | what a liar |
| 14:18.48 | *** part/#asterisk inspired (n=mikael@85.221.7.59) |
| 14:18.52 | *** join/#asterisk Gpl_Source (n=The_natu@unaffiliated/gplsource) |
| 14:18.53 | Uatec | is it normal for a digium s101i iaxy to spam: -- Accepted AUTHENTICATED TBD call from 192.168.232.173 to the CLI continually? |
| 14:19.07 | champster | sorry for the delay, here is the paste for the meetme kick problem: http://paste.debian.net/26552 |
| 14:19.08 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 14:21.52 | ManxPower | Uatec: no. It is normal for asterisk to do that when getting a call from an IAXy |
| 14:22.17 | naitram | anyone know how to get the features.conf to work with asterisk. I have a simple example dooropen => #9,self,Playback,1101open16 in my features.conf, I have exten => 1101,1,Set(DYNAMIC_FEATURES=dooropen) in the exten I want to be able to dial #9 from and nothing works, the script for dooropen is never run |
| 14:23.52 | [TK]D-Fender | naitram: Did you mod your dial command like I told you to? |
| 14:24.08 | ManxPower | naitram: you are on a Zap channel or MGCP or SCCP channel, right? |
| 14:24.27 | naitram | the dial string is exten => 1101,n,Dial(SIP/SYS1AMP,20,gT), no I am on a SIP channel |
| 14:25.17 | ManxPower | naitram: 1) for some reason I thought w/W was the activation char. I assume you are testing this when you are ON an ACTIVE calls, right? |
| 14:26.07 | naitram | ManxPower: yes, active call, W is supposed to be for recording? right |
| 14:26.31 | ManxPower | naitram: I think it is for "recording and any other girly DTMF feature" |
| 14:26.45 | naitram | ManxPower: ok will try |
| 14:26.48 | ManxPower | naitram: see the examples on the Wiki or extensions.conf.sample or the mailing list |
| 14:27.02 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
| 14:27.36 | [TK]D-Fender | ManxPower: No, he WANTS an in-line DTMF feature. He wast to dynamically call the dooropen script he has setup in-line with his call. |
| 14:27.43 | *** join/#asterisk krp (n=krp@mar92-10-82-239-65-214.fbx.proxad.net) |
| 14:27.53 | krp | hi |
| 14:27.57 | naitram | ManxPower: thought thats what I had. They show a playback example |
| 14:28.31 | krp | just wanted to know if there was a solid voicexml solution for asterisk |
| 14:28.33 | ManxPower | naitram: half the info in the wiki is wrong, so try to confirm in other places on the web |
| 14:28.51 | jm|work | has "if" been deprecated?! |
| 14:28.52 | krp | and opensource |
| 14:29.01 | jm|work | <PROTECTED> |
| 14:29.02 | ManxPower | naitram: I assume you have no problems calling into and logging into voicemail from that phone? |
| 14:29.22 | ManxPower | jm|work: perhaps it was replaced with $IF but UPGRADE.txt would be the place to check |
| 14:29.28 | jm|work | :) |
| 14:30.00 | champster | meetme-kick - it appears that web-meetme's cbEnd.php was kicking the users. |
| 14:30.04 | *** join/#asterisk Polis_ttt (n=your@194-237-172-225-no48.business.telia.com) |
| 14:31.07 | naitram | ManxPower: no problem. Although not using voicemail; just doen't seem to be sending the tones? Is that a phone thing. Or maybe a SIP setup thing. I never see anything going out on wireshark after I hit #9. Is that set up with the sip session? |
| 14:31.37 | krp | don't tell me no one knows about voicexml here ;) |
| 14:32.40 | ManxPower | naitram: try voicemail or any other app that has to read DTMF to confirm your DTMF works |
| 14:33.00 | ManxPower | krp: I'll bet nobody here uses voicexml. search the mailinglist archives |
| 14:33.02 | ManxPower | ~mailinglist |
| 14:33.14 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 14:33.55 | coppice | has anyone here used http://www-x.antd.nist.gov/nistnet/ to simulate a VoIP path? |
| 14:34.06 | boch | i lose dtmf in 2 or more sip phones behind the same nat when using simultaeusly.. could this be related with a nat problem? |
| 14:34.44 | ManxPower | naitram: using inband dtmf on a compressed codec would break dtmf |
| 14:34.55 | ManxPower | inband dtmf would just show up as a voice packet |
| 14:35.34 | naitram | ManxPower: ok will check that out |
| 14:36.50 | ManxPower | The weather in N Alabama sucks todat |
| 14:38.18 | naitram | ManxPower: middle al, aint no better. Im in the Gump |
| 14:39.54 | *** join/#asterisk shinao1 (n=shinao1@196.1.179.225) |
| 14:42.27 | ManxPower | naitram: where are you again? |
| 14:42.37 | *** join/#asterisk plasmid (n=noway@c-68-46-97-136.hsd1.pa.comcast.net) |
| 14:42.44 | ManxPower | Gads the people on Slashdot know nothing about T-1s |
| 14:42.48 | naitram | ManxPower: OK, definitely getting . |
| 14:43.04 | naitram | sip dtmf signals, now |
| 14:43.16 | naitram | still no feature activation |
| 14:44.15 | naitram | ManxPower: where are you...., im in the gump, you know ... Montgomery |
| 14:46.16 | *** join/#asterisk chefrs (n=joe@c-24-8-226-145.hsd1.co.comcast.net) |
| 14:46.40 | chefrs | Anyone know how to make it so that when you direct dial a voicemail, instead of the default "Please leave your message, after the tone" you get that person's actual voicemail message? |
| 14:49.43 | ManxPower | I am 1 hr NE of Birmingham AL |
| 14:49.48 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 14:50.07 | krp | ManxPower: ok |
| 14:50.18 | tuxick | Uatec: you could still add a rule to at least delay picking up :) |
| 14:50.31 | krp | ManxPower: what's your opinion on voicexml ? |
| 14:50.34 | Uatec | <PROTECTED> |
| 14:51.25 | *** part/#asterisk chefrs (n=joe@c-24-8-226-145.hsd1.co.comcast.net) |
| 14:51.30 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 14:51.44 | tuxick | Uatec: to the context now picking up |
| 14:51.50 | ManxPower | krp: I am not qualified to have an opinion of VoiceXML. But that does not stop me from having one. I don't use and have never used VoiceXML. I don't know of anyone that has used VoiceXML. |
| 14:51.58 | tzanger | ManxPower: hahahah |
| 14:52.18 | illsci | hey do cell phone companies allow you to forward voicemails to different carriers? |
| 14:52.24 | *** join/#asterisk `pariah (n=josh@unaffiliated/pariah) |
| 14:52.27 | ManxPower | illsci: no |
| 14:52.31 | illsci | lame |
| 14:53.07 | ManxPower | illsci: frequently you can't even forward voicemails between voicemail boxes on the same carrier in different parts of the country |
| 14:53.31 | jm|work | speex or festival? |
| 14:53.34 | jm|work | ~speex |
| 14:53.48 | illsci | does like cingular have voip gateways... |
| 14:54.19 | illsci | or the major cell phone carriers... |
| 14:55.16 | ManxPower | jm|work: could be both since they do TOTALLY different things |
| 14:56.05 | ManxPower | speex is a codec (perhaps you meant Sphinx?). Sphinx is voice recognition. Festival is voice synth. |
| 14:56.39 | [TK]D-Fender | jm|laptop: Which again... are 2 totally differnt things :) |
| 14:57.00 | jm|work | :( |
| 14:57.14 | jm|work | I thought there was an alternative to festival |
| 14:57.56 | [TK]D-Fender | jm|laptop: Plenty of alternatives to Festival. Most common is Cepstral. |
| 14:58.05 | illsci | att makes a good one too |
| 14:58.10 | illsci | its not free though |
| 14:59.47 | ManxPower | Cepstral is voice synth that is cheap |
| 15:01.14 | *** join/#asterisk pks (n=pks@81.210.81.3) |
| 15:01.24 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 15:03.01 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 15:03.23 | BSD_Tech | Cepstral is like flite |
| 15:03.29 | BSD_Tech | FLite = free |
| 15:03.59 | BSD_Tech | I have the source for app_flite |
| 15:04.00 | *** join/#asterisk IOscanner (n=IOscanne@cpe-76-187-194-128.tx.res.rr.com) |
| 15:05.12 | IOscanner | Anyone have or know of a good document that explains how to track calls via the CDR records. We enabled the uniqueid field, but it changes. |
| 15:05.28 | ManxPower | *grumle* I was supposed to do outside plant work today |
| 15:05.50 | ManxPower | IOscanner: ask SnrWup |
| 15:05.50 | *** join/#asterisk wasim_ (n=wasim@pdpc/supporter/active/wasim) |
| 15:06.02 | IOscanner | Is there a better wayt to track calls as they go through IVR, Queues to answer or hangup |
| 15:06.29 | mosty | IOscanner: what do you want to track exactly? |
| 15:06.49 | IOscanner | I am trying to get call times |
| 15:07.01 | IOscanner | Call abandoned |
| 15:07.14 | IOscanner | answered |
| 15:07.26 | Uatec | who here knows how much of a piece of crap a digium s101i iaxy is? |
| 15:07.27 | mosty | IOscanner: there is a field in the CDR for that, why don't you just read that? |
| 15:07.30 | IOscanner | and minutes on the call |
| 15:07.39 | Qwell[] | Uatec: iaxy works pretty well in my experiences |
| 15:08.05 | Qwell[] | disclaimer: I now work for Digium. My experiences with it are pre-Digium though |
| 15:08.07 | IOscanner | Yes but asterisk puts lines and lines you can't trace it when you have lots of calls |
| 15:09.26 | IOscanner | per call with no way to determine which records are related |
| 15:09.32 | mosty | IOscanner: what do you mean? it's a simple table, you can look up whatever you like in it |
| 15:09.43 | mosty | IOscanner: one line per call... |
| 15:10.01 | IOscanner | I am getting more than one line per call |
| 15:10.30 | IOscanner | I get records for Queues trying each extension |
| 15:10.43 | unik-rados | speaking of tts engines... which one is the best then? I'm using Cepstral right now but people say that it sounds too "robotic". is there something better out ther ethat works with Asterisk? |
| 15:10.47 | [TK]D-Fender | Uatec: It has no features, no friendly interface, and is only good with * direct. |
| 15:11.28 | [TK]D-Fender | IOscanner: You don't use CDR's to trak queue calls, you use the queue log itself. |
| 15:11.42 | [TK]D-Fender | Uatec: I'd never touch it. |
| 15:12.18 | *** join/#asterisk merlinn (n=merlin@fartjam.com) |
| 15:13.13 | *** part/#asterisk BSD_Tech (n=bsdtech@ppp-69-239-113-126.dsl.irvnca.pacbell.net) |
| 15:13.16 | IOscanner | So we can't use the uniqueid to track calls that are related? |
| 15:14.18 | merlinn | hi |
| 15:14.34 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 15:14.34 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:14.40 | merlinn | I'm in a fix, I need to know if I can scale an asterisk platform past any example I've seen |
| 15:15.05 | blitzrage | yes -- its called clustering |
| 15:15.12 | blitzrage | more boxes == more calls |
| 15:15.13 | merlinn | I need to route 4M minutes a day |
| 15:15.26 | blitzrage | how many simultaneous calls does that equal out to? |
| 15:15.30 | merlinn | urm |
| 15:15.43 | merlinn | about 50 pri's maxed out |
| 15:15.47 | merlinn | so... |
| 15:16.32 | merlinn | so |
| 15:16.32 | merlinn | assuming it's stady |
| 15:16.32 | blitzrage | 1200 calls |
| 15:16.32 | merlinn | steady |
| 15:16.32 | blitzrage | wait |
| 15:16.32 | blitzrage | PRI |
| 15:16.32 | merlinn | yeah |
| 15:16.32 | blitzrage | 1150 |
| 15:16.32 | merlinn | well |
| 15:16.34 | blitzrage | 23 channels per link |
| 15:16.37 | merlinn | yeah |
| 15:16.41 | blitzrage | 1150 |
| 15:16.42 | plasmid | is it possible for a user to send a text message to my VOIP cell phone through the IVR?? IVR extension 5 connects to VOIP cell phone. |
| 15:16.47 | merlinn | but that's assuming it's steady |
| 15:16.51 | merlinn | I'm guessing a peak of 3000 |
| 15:16.55 | blitzrage | you could 2 that with a pair of dual-quad-core 2.8GHz Xeons |
| 15:17.07 | merlinn | I can ? |
| 15:17.23 | merlinn | have you seen any examples of this kind of load threshold? |
| 15:17.26 | blitzrage | I can get at least 600 simultaneous calls, with media, through one of those boxes at < 20% cpu |
| 15:18.21 | blitzrage | I've done my own load testing |
| 15:18.21 | merlinn | sweet |
| 15:18.21 | *** join/#asterisk wunderkin (n=kev@dslstat-ppp-95.fastq.com) |
| 15:18.23 | plasmid | blitzrage, what kind of hardware does your asterisk box have? |
| 15:18.23 | blitzrage | funny enough, I don't just ask other people, I actually test this stuff |
| 15:18.23 | merlinn | well |
| 15:18.24 | blitzrage | plasmid: try looking at what I just said |
| 15:18.24 | merlinn | congratulations |
| 15:18.24 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 15:18.30 | merlinn | I don't have the resource to waste time when 90% of people are telling me I'm wasting time using asterisk in the first place |
| 15:18.30 | plasmid | pfff.. i am not about to scroll 100 lines up. |
| 15:18.30 | merlinn | but I totally agree with your point |
| 15:18.58 | blitzrage | yah, because figuring out what your system can handle is a waste of time |
| 15:18.58 | blitzrage | plasmid: try like 8 |
| 15:19.05 | blitzrage | <blitzrage> you could 2 that with a pair of dual-quad-core 2.8GHz Xeons |
| 15:19.22 | *** join/#asterisk syalcin (n=syalcin@165.236.171.10) |
| 15:19.25 | blitzrage | fuck people are lazy in this channel |
| 15:19.30 | codefreeze | blitzrage: but 50 t1's? In a single box or two? |
| 15:19.36 | justdave | hmm, so after upgrading to asterisk 1.2.18, using coming in on iax2 links get dumped when they enter MeetMe, and the log shows this: |
| 15:19.38 | plasmid | blitzrage, that was a line referring to an answer to someone else you arrogant prick. |
| 15:19.39 | justdave | Apr 26 08:13:18 NOTICE[26059]: chan_iax2.c:3167 iax2_read: I should never be called! Hanging up. |
| 15:19.55 | Qwell[] | codefreeze: 42 port DS3 card from Telecomjoshliefvoxmart Enterprises |
| 15:19.56 | blitzrage | codefreeze: ummm... I'm not doing PRI to be honest -- but I was doing 2.5MB of ulaw traffic, which worked out to like 600 simultaneous calls |
| 15:19.59 | plasmid | sheesh.. the ppl here. |
| 15:20.11 | blitzrage | plasmid: its true -- I *am* an arrogant prick |
| 15:20.21 | Qwell[] | (A division of Qwell Communications) |
| 15:20.23 | plasmid | that explains a lot. |
| 15:20.23 | Mercestes | It is true |
| 15:20.31 | Mercestes | He is an arrogant prick |
| 15:20.34 | codefreeze | blitzrage is so humble, he's proud of it! |
| 15:20.39 | Qwell[] | Mercestes: totally |
| 15:20.45 | Qwell[] | but, he's allowed to |
| 15:20.47 | blitzrage | I just have a short attention span with some people |
| 15:20.50 | SnrWup | is it possible to detect how many and which channels are currently in use by a particular SIP phone? so for example, I have an application running on every workstation which has a list of every user in the office. asterisk can link to PHP to send user status to each app. so at the moment everyone can see whether a user has set himself as DND (by pressing a PHP softkey). but what if i want to give information like User X is busy on 2 channe |
| 15:20.50 | Mercestes | true |
| 15:21.08 | *** join/#asterisk nettie (n=nettie@ns.coolgadgets.it) |
| 15:21.19 | merlinn | does your winning personality extend beyond irc? |
| 15:21.25 | blitzrage | it sure does! |
| 15:21.31 | blitzrage | I'm also a porn star |
| 15:21.34 | merlinn | are you a consultant? |
| 15:21.46 | merlinn | or gainfully employed |
| 15:21.53 | Qwell[] | blitzrage: Do you *really* want Tilghman to know that? |
| 15:21.53 | blitzrage | yes, but I'm not taking on any more clients since I have too many already |
| 15:22.03 | merlinn | fair enough. |
| 15:22.04 | blitzrage | Qwell[]: I only do girls :) |
| 15:22.14 | merlinn | what's your daily rate? |
| 15:22.19 | blitzrage | $125/hr |
| 15:22.21 | merlinn | out of idle curiosity |
| 15:22.24 | merlinn | lol |
| 15:22.29 | blitzrage | and people tell me I don't charge enough |
| 15:22.33 | merlinn | yeah |
| 15:22.45 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 15:22.46 | codefreeze | his wife, mainly... |
| 15:22.48 | *** join/#asterisk elriah (i=elriah@adsl-074-164-217-189.sip.bhm.bellsouth.net) |
| 15:22.49 | merlinn | you're under charging |
| 15:22.55 | blitzrage | merlinn: don't I know it |
| 15:22.57 | Mercestes | merlinn, Well, since I'm a lazy fuck that just wandered into this channel, what exactly are you trying to do again? |
| 15:23.06 | nettie | Hi guys, I keep getting Primary D-Channel on span 2 down messages in my asterisk console .. I think it's one of my BRI port because I cant make calls using them. Anyone know what could be wrong please? |
| 15:23.13 | merlinn | terminate a lot of calls |
| 15:23.20 | blitzrage | but people who run Asterisk are notoriously cheap! |
| 15:23.24 | merlinn | no |
| 15:23.27 | merlinn | well yeah |
| 15:23.30 | syalcin | can anyone help me with faxing over pri? |
| 15:23.34 | elriah | Hi all. Anyone using realtime static in 1.2.16-18? I'm using it for SIP peers, or trying to at least, and when I reload I get errors saying that asterisk can't find the 'type' field. Clearly, the type field exists and with a value of 'friend' for each. Any suggestions? |
| 15:23.39 | merlinn | I need to use asterisk because I need to use the agi functionality |
| 15:23.43 | Mercestes | nettie, What does span 2 go to ? |
| 15:23.46 | blitzrage | yah, its pretty sweet |
| 15:23.47 | merlinn | otherwise i'd buy a soft switch and be done |
| 15:23.48 | *** join/#asterisk littleball (n=littleba@bb220-255-154-109.singnet.com.sg) |
| 15:23.56 | Mercestes | merlinn, How many calls? |
| 15:24.05 | merlinn | probably about 3000 simultaneously at peak |
| 15:24.13 | mosty | elriah: try turning on logging for your db server, and see if the queries arent working |
| 15:24.15 | Mercestes | merlinn, ...on one box? |
| 15:24.16 | merlinn | but potentially growing to four times that within 12 months |
| 15:24.19 | merlinn | no |
| 15:24.20 | Qwell[] | merlinn: That's gonna be 2-3 servers |
| 15:24.21 | nettie | Mercestes |
| 15:24.21 | littleball | hello, my asterisk cli print : Apr 26 23:31:16 NOTICE[27611]: rtp.c:576 ast_rtp_read: Unknown RTP codec 127 received |
| 15:24.22 | nettie | veronica*CLI> zap show status |
| 15:24.23 | nettie | Description Alarms IRQ bpviol CRC4 |
| 15:24.23 | nettie | duoBRI PCI ISDN Card 1 Span 1 [TE] (car OK 0 0 0 |
| 15:24.23 | nettie | duoBRI PCI ISDN Card 1 Span 2 [TE] (car OK 0 0 0 |
| 15:24.25 | littleball | what does it mean? |
| 15:24.25 | merlinn | 'll use as many boxes as necessary |
| 15:24.26 | blitzrage | but yah -- dual-quad-core xeon (Dell 2950) will do at least 600 calls, with media (non-transcoding) at < 20% CPU |
| 15:24.29 | *** join/#asterisk DrukenLPY (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 15:24.30 | Qwell[] | probably more |
| 15:24.33 | Mercestes | nettie, *slaps* Pastebin! |
| 15:24.36 | blitzrage | I tested with SIPp |
| 15:24.40 | nettie | Mercestes it was short :p |
| 15:24.43 | merlinn | all the people I know that use asterisk terminate 10 clals simultaneously though |
| 15:24.44 | Qwell[] | or one sunfire T2000 ;) |
| 15:24.52 | mosty | blitzrage: i'm playing with one of those boxes right now :) |
| 15:24.55 | blitzrage | SIPp --> Asterisk (load) --> Asterisk (media generator) |
| 15:24.58 | Mercestes | merlinn, I've done a bit more than 10 but never 3000. |
| 15:24.58 | merlinn | and my voice knowledge is purely appliance |
| 15:24.59 | Qwell[] | (Sun is having an AWESOME sale right now, FYI) |
| 15:25.04 | blitzrage | mosty: they are great boxes |
| 15:25.13 | Qwell[] | (very very VERY good deals on T1000 and T2000s) |
| 15:25.16 | blitzrage | Qwell[]: I don't know anything about Sun though :) |
| 15:25.22 | Qwell[] | blitzrage: put Linux on it |
| 15:25.26 | blitzrage | Qwell[]: hrmmmm |
| 15:25.28 | elriah | mostly: It loads the peer, just not all the values. The peers show up in 'sip show peers' but whey I show a specific peer, my values are all the defaults. |
| 15:25.33 | blitzrage | I think we use Sun V20 or something for our softswitches |
| 15:25.43 | Mercestes | merlinn, You might wanna server farm from OpenSer to Asterisk boxes if you wanna do that without problems. |
| 15:25.44 | Qwell[] | zaptel won't work (yet), but it'll be fairly easy to make it work |
| 15:25.54 | Qwell[] | the drivers should, in theory, already be endianized |
| 15:26.10 | Qwell[] | (I know for sure that the TC400b is..) |
| 15:26.14 | merlinn | I want to shoot myself for using something opensource for this |
| 15:26.20 | merlinn | but I don't really have an option I don't htink |
| 15:26.23 | [TK]D-Fender | Qwell[]: Who'd use zaptel for that many T1's? Audiocodes gateways...... |
| 15:26.34 | Qwell[] | [TK]D-Fender: nah, you wouldn't really |
| 15:26.40 | mosty | blitzrage: the pci slots could be easier to access, but that's my only gripe so far |
| 15:26.40 | elriah | Qwell[]: Any clue why when using RealTime static SIP peers in 1.2.16-18 I get an error indicating asterisk can't find the 'type' value? It's in the db. Thanks for any insight. |
| 15:26.43 | Qwell[] | SIP or IAX2 probably |
| 15:26.46 | [TK]D-Fender | merlinn: Sure you do... Broadsoft :) |
| 15:26.49 | merlinn | no |
| 15:26.55 | merlinn | I use broadsoft |
| 15:26.59 | merlinn | this isn't a similar thing really |
| 15:27.03 | blitzrage | I use OpenSER on the front end as the registration point, then use to dispatcher module to drop calls to a cluster of asterisk boxes which use a replicated database backend to pull information from the DB using func_odbc in the dialplan (so no dialplan reloads are needed for changes to the DB from the GUI), and store VM using ODBC for access to VM from any Asterisk box |
| 15:27.09 | mosty | elriah: yes but did you check the queries? |
| 15:27.23 | elriah | mostly: No, doing that now... |
| 15:27.24 | blitzrage | mosty: yah, the boxes are hosted in a colo in Florida (I'm in Toronto), so I don't care how difficult it is :)) |
| 15:27.27 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 15:27.28 | elriah | mostly: (thanks) |
| 15:27.37 | merlinn | all I'm doing is premium rate call routing |
| 15:28.27 | [[blah]asfd | what does this mean when dialing out of a T1: !! Got reject for frame 96, retransmitting frame 96 now, updating n_r! |
| 15:28.28 | mosty | blitzrage: it's a tool-less setup, but i needed a pocket knife to open up the tab locking the pci slots in place |
| 15:28.28 | [[blah]asfd | what does that mean? |
| 15:28.37 | blitzrage | codefreeze: I think I did it in 1 box :) |
| 15:29.36 | blitzrage | I'm using the very high density card from telcomjoshleifvoxmartenterprisesinc |
| 15:29.36 | Dovid | Tzafrir: are you in here ? |
| 15:29.39 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:29.40 | littleball | hi, what is AST_FORMAT? |
| 15:29.46 | nettie | Mercestes any idea? |
| 15:30.00 | illsci | what do you mean "so many" exploited asterisk boxes |
| 15:30.08 | nettie | Mercestes well span2 goes to a BRI |
| 15:30.28 | illsci | they released a vuln yesterday that dealt with an option in their sip module |
| 15:30.30 | nettie | maybe I try reset it |
| 15:30.30 | Mercestes | blitzrage, "People who use asterisk are notoriously cheap." You know an awful lot about a product you have a negative opinion of. :P |
| 15:30.41 | illsci | woops wrong channel :) |
| 15:30.57 | merlinn | I don't think that means he has a negative opinion of it |
| 15:31.03 | Mercestes | nettie, What does span 2 go to? please don't paste anything |
| 15:31.11 | plasmid | is it possible for a user to send a text message to my VOIP cell phone through the IVR?? IVR extension 5 connects to VOIP cell phone. |
| 15:31.34 | *** part/#asterisk illsci (n=illsci@evil.hack3rs.org) |
| 15:31.42 | blitzrage | Mercestes: I have a negative opinion about it? |
| 15:31.46 | nettie | Mercestes what you mean with "go to"? |
| 15:32.00 | Mercestes | blitzrage, That would be my perception, yes. |
| 15:32.01 | blitzrage | I have a negative opinion about people -- I have an infinitely positive opinion about Asterisk |
| 15:32.14 | Mercestes | blitzrage, Ahh. Thanks for the clarification. |
| 15:32.35 | blitzrage | I wouldn't have been using Asterisk for 5 years if I hated it :) |
| 15:32.35 | Mercestes | nettie, What is it plugged into? |
| 15:32.35 | codefreeze | Just curious, how is the formation of a mega-corp like telcomjoshleifvoxmartenterprisesinc accomplished? |
| 15:32.35 | *** join/#asterisk fourcheeze (n=rich@85-189-96-153.rcg-global.managedbroadband.co.uk) |
| 15:32.37 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 15:32.44 | *** join/#asterisk hfb (n=hfb@pool-72-67-156-130.lsanca.dsl-w.verizon.net) |
| 15:32.46 | blitzrage | codefreeze: very carefully |
| 15:33.00 | simplexio | plasmid: there is sms application but bo idea howto use it |
| 15:33.13 | fourcheeze | anyone using asterisk with avaya ip office 4.0? |
| 15:33.27 | fourcheeze | our avaya seems to be insisting on using Tel: urls instead of SIP |
| 15:33.38 | nettie | Mercestes TELCO---ISDN----duoBRI---asterisk |
| 15:34.08 | simplexio | so anyone idea how i get realtime musiconhold or patch for musiconhold so i can use files instead predefined classes |
| 15:34.09 | Mercestes | nettie, Make sure your zaptel.conf is setup right. |
| 15:34.39 | nettie | Mercestes it looks like, worked before, when I Start using it stopped :( |
| 15:34.40 | Mercestes | nettie, What type of card are you using? |
| 15:35.36 | nettie | I'll pastebin it -- I'm using a duoBRI pci card. http://www.junghanns.net/en/duoBRI2_produkt.html |
| 15:36.26 | plasmid | simplexio, sms application. where did u see it? i google for it but can't find something remotely familiar. |
| 15:36.43 | Mercestes | nettie, Alright. hrm. |
| 15:37.01 | tzafrir | Dovid, yes |
| 15:37.01 | nettie | Mercestes http://paste.uni.cc/14857 |
| 15:37.10 | *** part/#asterisk frigidzephyr (i=frigidze@nat/digium/x-b4dca39305f6c4bf) |
| 15:37.11 | simplexio | plasmid: ./doc/sms.txt |
| 15:37.25 | plasmid | simplexio, gotcha. Thankx :-) |
| 15:37.36 | simplexio | plasmid: in asterisk-1.4.2 source code dir.. |
| 15:37.47 | plasmid | kk |
| 15:38.43 | Mercestes | nettie, I dunno. |
| 15:39.40 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:40.06 | tzanger | there, new asterisk box mostly setup and waiting for tonight :-) |
| 15:40.20 | blitzrage | sweet deals! |
| 15:40.42 | Polis_ttt | anyone used astguiclient/vicidial ? |
| 15:40.59 | tzanger | nope |
| 15:41.13 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:44.32 | drako | there is a way to change permission of zap devices on /dev at load time? |
| 15:44.44 | jm|work | stupid Festival |
| 15:44.50 | tzafrir | Dovid, well, actually not |
| 15:46.10 | Mercestes | BRB. I have to uninstall .net framework 1.1 |
| 15:46.33 | *** part/#asterisk merlinn (n=merlin@fartjam.com) |
| 15:53.31 | Dovid | tzafrir: i was told they have one :( oh well. fun fun tonight |
| 15:53.43 | *** join/#asterisk swyrus (n=woder@demon-gw.denbridgemarine.com) |
| 15:53.51 | *** part/#asterisk swyrus (n=woder@demon-gw.denbridgemarine.com) |
| 15:54.22 | *** join/#asterisk jsolares (n=jsolares@216.106.164.193) |
| 15:55.11 | Uatec | awesome |
| 15:56.49 | jm|work | ~celliax |
| 16:03.22 | *** join/#asterisk djflux (n=djflux@mm.shermfin.com) |
| 16:04.13 | drako | there is a way to change permission of zap devices on /dev at load time? |
| 16:04.35 | drako | i have some rules on udev.d/zaptel.conf but my devices keep gettig root as owner |
| 16:04.41 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 16:07.51 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 16:11.26 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 16:14.08 | naitram | ok, I give up! Despite being sure that sip dtmf messages are being sent on the wire, having include => featuremap, include => applicationmap in my context, having Dial(...,gTW) in my dial string, dooropen => #9,self/caller,Playback,tt-monkeys in my features.conf. Nothing happens when I am connected when I dial #9. HELP! Looked all over the web. This can not be that unusual a thing is it? |
| 16:14.23 | *** join/#asterisk jtoy (n=jtoy@mail.backchannelmedia.com) |
| 16:14.39 | [TK]D-Fender | naitram: Go verify that your DTMF is working at all. |
| 16:14.51 | [TK]D-Fender | naitram: Try a DTMF transfer, etc. |
| 16:14.52 | jtoy | can anyone recommend a server model to use for a asterisk server? is ther esome server that a lot of pople use? |
| 16:15.10 | jtoy | I am looking to get a 2u server so I can put a couple of pci cards in for pri/t1 |
| 16:15.15 | [TK]D-Fender | jtoy: Nope, soo many different ones, not a lot in common. |
| 16:15.26 | *** join/#asterisk ToyMan (n=Stuart@74-32-9-93.dsl1.mdl.ny.frontiernet.net) |
| 16:15.38 | [TK]D-Fender | jtoy: Many respect IBM's i3XX series |
| 16:15.41 | naitram | [TK]D-Fender: isn't that what this is? What has to be changed to make it a transfer? |
| 16:16.03 | [TK]D-Fender | naitram: just try transfering that call via features.conf. |
| 16:18.24 | *** join/#asterisk ringhals (i=fwuser@firewall.drgutah.com) |
| 16:19.44 | ringhals | I am getting a couple warining I was hoping someone could shed some light on. |
| 16:19.49 | ringhals | Apr 24 17:28:04 WARNING[3730] chan_zap.c: Ring/Off-hook in strange state 6 on channel 23 |
| 16:20.11 | ringhals | Apr 26 10:12:03 NOTICE[6262] channel.c: Dropping incompatible voice frame on Local/2114@asterisks01-fa37,2 of format ulaw since our native format has changed to slin |
| 16:20.52 | naitram | ok, I know this is all newbie stuff but bear with me please. So uncomment out blindxfer => #1 and then add Set(DYNAMIC_FEATURES=blindxfer) and when I dial #1 then lets say 1101, I should transfer to 1101? |
| 16:21.16 | [TK]D-Fender | naitram: Sounds about right, though I haven't used it myself. |
| 16:21.24 | naitram | ok will try |
| 16:21.42 | *** join/#asterisk dlynes_laptop (n=dlynes@d207-216-161-56.bchsia.telus.net) |
| 16:23.53 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 16:25.28 | naitram | [TK]D-Fender: nope doesn't work, I get signal 11, then signal 1 etc... from my client back to the asterisk machine but asterisk doesn't seem to do a thing with it. I have verbose mode and debug mode on in cli and don't even see it recognize the sip dtmf messages |
| 16:25.56 | [TK]D-Fender | naitram: You may simply not have DTMF functional for your phone at all. Do some more tests. |
| 16:26.42 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:26.50 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 16:27.15 | naitram | [TK]D-Fender: but the phone is sending the dtmf signals via sip requests to the server. I can see them with wireshark. |
| 16:27.45 | [TK]D-Fender | naitram: Yeah... and maybe * is IGNORING them because the modes don't match. GO TEST |
| 16:28.03 | *** part/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 16:28.06 | zuez | Hmm, Version: 3.1(MF.G2) on a C7960, is that SCCP? or Media Gateway? |
| 16:29.28 | naitram | [TK]D-Fender: test what, I guess thats the main point here. Don't know what to test. What modes does asterisk accept for dtmf? |
| 16:30.35 | naitram | anyone successfully used blindxfer with sip clients. or more specifically with AAstra 480i phone |
| 16:32.22 | xkev | blind is fine with my polycom and ciscos |
| 16:33.03 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 16:33.19 | ManxPower | Anyone have any ideas about this? "Unable to do INITIAL ProSLIC powerup on module 2" |
| 16:33.25 | naitram | xkev: anyway you can show me your extensions.conf and features.conf for these phones? |
| 16:33.38 | [TK]D-Fender | naitram: make an IVR and prove that your phone is set up right to match * at all. |
| 16:34.14 | naitram | [TK]D-Fender: an IVR ? |
| 16:36.53 | naitram | [TK]D-Fender: if you mean can I dial the phone and dial out from it, yeah of course. Not sure what you mean exactly. Are you questioning whether the phone is even working with asterisk |
| 16:37.16 | [TK]D-Fender | naitram: Make an IVR menu in your dialplan to test your DTMF against. |
| 16:37.29 | [TK]D-Fender | naitram: If I have to explain this to you tyy... |
| 16:37.31 | [TK]D-Fender | ~osmosis |
| 16:37.36 | jbot | [osmosis] the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
| 16:37.36 | [TK]D-Fender | try* |
| 16:38.51 | ringhals | So I have not been able to find much of an explanation on the following warning could someone shed some light on this for me? |
| 16:38.54 | ringhals | Dropping incompatible voice frame on Local/2114@asterisks01-c87d,2 of format ulaw since our native format has changed to slin |
| 16:39.01 | naitram | [TK]D-Fender: well obviously, this is all voluntary so... you don't really have to explain anything. But thanks for the patience |
| 16:39.34 | xkev | my extensions.conf(s) are huge. features.conf is irrelevant for sip transfer |
| 16:39.47 | xkev | I'm not doing like * or # codes or whatever for transfres |
| 16:39.49 | jsolares | naitram: what do you have as dtmfmode in your sip.conf? |
| 16:40.01 | xkev | I hit 'transfer' then 'blind' on my phone, and dial |
| 16:40.06 | [TK]D-Fender | xkev: it was a means to an end, which was an attempt to see if his OTHER real purpose had any hopes of success. |
| 16:40.17 | xkev | ..which is a problem if you use chan_local or chan_agent |
| 16:41.14 | xkev | [featuremap] |
| 16:41.14 | xkev | blindxfer => ## ; Blind transfer |
| 16:41.18 | xkev | does that help? :) |
| 16:41.35 | xkev | also I run cvs from 20050311 |
| 16:41.40 | xkev | it's 1.2..kinda |
| 16:42.02 | naitram | jsolares: rfc2833 |
| 16:42.18 | ManxPower | naitram: 90% of DTMF transfer problems are really DTMF problems |
| 16:42.23 | jsolares | and are you sure your phones are also in that mode and it works ok? |
| 16:43.57 | jsolares | by IVR he probably meant doing a Background(some-audio) and then pressing an existing extension on that context to see if that would ring |
| 16:43.58 | naitram | jsolares: no, im not sure. I will check. That may be the whole problem. Thanks, Didn't realize where that association was. thanks |
| 16:44.05 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 16:44.59 | naitram | ManxPower: yeah, i see . I just have a lot of vocabulary issues right now. New to pbx's and telephony |
| 16:45.32 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 16:45.32 | *** mode/#asterisk [+o anthm] by ChanServ |
| 16:47.09 | *** join/#asterisk horvath (i=horvath@i.bring.lots.of.bad-ne.ws) |
| 16:47.50 | jsolares | heh |
| 16:49.10 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 16:49.33 | horvath | I can't get any sound playback from any of default gsm asterisk sounds or any of the extra ulaw ones I added. Swift() works fine for playback so its not a one way audio issue... |
| 16:50.05 | horvath | asterisk 1.4.2 |
| 16:50.41 | horvath | I see it play the file on the CLI but again no sound. |
| 16:54.01 | *** part/#asterisk jtoy (n=jtoy@mail.backchannelmedia.com) |
| 16:58.25 | ManxPower | horvath: What Digium Zap card are you using? |
| 16:58.48 | horvath | ManxPower: none. I do have ztdummy loaded |
| 16:59.03 | horvath | ManxPower: it says ZTDUMMY/1 1 UNCONFIGUR 0 0 0 |
| 17:00.27 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 17:02.20 | horvath | ManxPower: actually when I use the SayDigits() cmd it freezes |
| 17:02.33 | horvath | ManxPower: the CLI doesent move after the 4 |
| 17:02.40 | horvath | ManxPower: until I hang up |
| 17:03.00 | horvath | ManxPower: it seems to sit there trying to play the file but no audio output |
| 17:04.07 | *** join/#asterisk ucfmethod (n=ucfmetho@office.eyestreet.com) |
| 17:04.16 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:08.55 | *** join/#asterisk `pariah (n=josh@unaffiliated/pariah) |
| 17:09.52 | *** join/#asterisk Peri (n=redanti@xtreme-12-48.dyn.aci.on.ca) |
| 17:10.17 | Peri | Anyone having issues with wrapuptime not working? Also having a problem with agent penalties in queues |
| 17:14.03 | *** join/#asterisk vicentealencar (i=MyIdent@200.234.208.152) |
| 17:16.54 | *** join/#asterisk Qwell[] (i=qwell@pdpc/sponsor/digium/Qwell) |
| 17:16.54 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 17:18.02 | horvath | ManxPower: thanks, seems to be an issue with ztdummy because without the module loaded it works fine. |
| 17:19.45 | vicentealencar | does anyone here know where I can find a AGI tutorial that shows all the possible communications between the script and asterisk? |
| 17:19.59 | JunK-Y | ~agi api |
| 17:20.18 | jbot | it has been said that agi api is at http://home.cogeco.ca/~camstuff/agi.html |
| 17:21.20 | vicentealencar | cool |
| 17:23.26 | JunK-Y | not the latest version, but its actually a good start. |
| 17:23.49 | *** join/#asterisk Gpl_Source (n=The_natu@unaffiliated/gplsource) |
| 17:26.00 | vicentealencar | thanks! |
| 17:32.28 | *** join/#asterisk Zefk (n=Zefk@wsc-fo.b.astral.ro) |
| 17:33.17 | Zefk | Hi, How can I convert a wav to wav6? (format_wav.c:139 check_header: Not a wav file 6). Thanks |
| 17:34.17 | [TK]D-Fender | Zefk: "man sox" |
| 17:35.03 | Peri | so no one has had any issues with wrapuptime not working? |
| 17:35.26 | Peri | i'm running asterisk-1.2.16 |
| 17:38.29 | Zefk | [TK]D-Fender: If I convert to wav with sox that mean I convert to wav file 6? |
| 17:38.33 | ManxPower | Peri: your extensive search of the mailing lists was not helpful |
| 17:38.52 | [TK]D-Fender | Zefk: Quite likely. |
| 17:38.52 | ManxPower | Zefk: Mono, 8khz, 16 bit |
| 17:38.54 | ManxPower | I think |
| 17:39.11 | Zefk | Thanks |
| 17:39.43 | Peri | nope, my search of the mailing list only confirmed that i've got my variables straight |
| 17:40.01 | ManxPower | This can't be good: |
| 17:40.02 | ManxPower | Opened pseudo zap interface, measuring accuracy... |
| 17:40.02 | ManxPower | 99.938965% 99.938965% 99.938965% 99.938965% 99.938965% 99.938965% |
| 17:44.14 | *** join/#asterisk DJ-Pyro_ (n=abrezins@gateway.brevient.net) |
| 17:44.28 | *** join/#asterisk TokyoJimu (n=TokyoJim@cpe-24-165-1-28.san.res.rr.com) |
| 17:44.28 | [TK]D-Fender | ManxPower: You're right.... those are CONSISTENT results. they MUST be fake ! ;) |
| 17:44.30 | vicentealencar | I've read in the AGI API tutorial that EXEC runs a specific asterisk application with arguments |
| 17:44.43 | vicentealencar | what are exactly the asterisk applications? |
| 17:45.09 | [TK]D-Fender | vicentealencar: There is a whole chapter on this in THE BOOK. Go read... |
| 17:45.11 | [TK]D-Fender | ~book |
| 17:45.25 | jbot | from memory, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:45.33 | vicentealencar | is it Asterisk and the future of telephony? |
| 17:45.54 | vicentealencar | yeah it is |
| 17:45.57 | vicentealencar | I'll take a look |
| 17:48.23 | ManxPower | [TK]D-Fender: 99.93 seems rather low to me |
| 17:48.54 | ManxPower | [TK]D-Fender: I need to make up another cable, I keep getting GREEN/YELLOW/RED/YELLOW/GREEN on my Asterisk/Adtran CB link |
| 17:49.15 | *** join/#asterisk mtaht4 (n=m@c-71-198-215-76.hsd1.ca.comcast.net) |
| 17:49.33 | [TK]D-Fender | ManxPower: Catch the wave! |
| 17:50.56 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 17:57.55 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 18:03.08 | irule | how can I choose wich zap channels to choose? |
| 18:03.22 | irule | <PROTECTED> |
| 18:03.35 | irule | that was not bad english, that is pressure lol |
| 18:03.44 | *** join/#asterisk Qwell[] (i=qwell@pdpc/sponsor/digium/Qwell) |
| 18:03.44 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 18:04.42 | irule | hello |
| 18:14.21 | *** join/#asterisk cstamas (n=cstamas@dsl51B63B22.pool.t-online.hu) |
| 18:15.52 | cstamas | hi all |
| 18:17.41 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
| 18:18.33 | cstamas | I have a TE205P card. One port is connected to an Ericsson PBX |
| 18:19.21 | cstamas | Ericsson is set up to put the call to this E1 trunk, but I always get the following error message: |
| 18:19.48 | cstamas | -- Extension '' in context 'eric' from '' does not exist. Rejecting call on channel 0/1, span 1 |
| 18:20.28 | *** join/#asterisk SoftIce (n=bongo@vc-196-207-45-253.3g.vodacom.co.za) |
| 18:20.37 | SoftIce | hi please can somebody sugest as to why zttool isn't compiling |
| 18:20.38 | SoftIce | http://www.pastebin.ca/459809 |
| 18:20.44 | *** join/#asterisk red9012 (n=marc3234@206-248-153-243.dsl.teksavvy.com) |
| 18:21.24 | brimstone | SoftIce, appears to be a problem with your libnewt |
| 18:21.40 | SoftIce | strange, I can try recompile libnewt-dev |
| 18:22.06 | SoftIce | maybe an outdated version as to what zaptel 1.4 requirs? |
| 18:23.40 | brimstone | perhaps, i'm not aware of any specifics to libnewt and zttool, just that libnewt has to exist |
| 18:24.00 | CrazyTux | How do I setup perm. agents i.e. they are always logged in without logging in? |
| 18:25.37 | russellb | put them in queues.conf |
| 18:26.22 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 18:31.49 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 18:33.06 | unik-rados | does anyone here tried using asterisk as a voicemail system with Avaya? |
| 18:33.52 | CrazyTux | russellb, as Agents? |
| 18:34.06 | russellb | no |
| 18:34.16 | CrazyTux | russellb, as? |
| 18:34.23 | russellb | the device itself |
| 18:34.29 | russellb | check out the sample queues.conf ... |
| 18:34.52 | anonymouz666 | where can I find more about SLA documentation? |
| 18:35.09 | russellb | documentation about documentation? |
| 18:35.42 | russellb | the stuff that exists is ... 1) blog post on asterisk.org 2) configs/sla.conf.sample 3) doc/sla.pdf 4) mailing list posts in the archives |
| 18:36.26 | Corydon-w | unik-rados: No, but I've done that with a Nitsucko and a Tadiran |
| 18:37.03 | Qwell[] | Nitsucko? That's a real brand? |
| 18:38.14 | Corydon-w | That's what the customer called it |
| 18:38.15 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 18:38.27 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 18:45.32 | CrazyTux | russellb, how would I route calls directly through a peer? |
| 18:45.35 | CrazyTux | russellb, i.e. an IP |
| 18:49.50 | *** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) |
| 18:50.45 | [TK]D-Fender | CrazyTux: What are you calling through this peer? |
| 18:50.51 | CrazyTux | [TK]D-Fender, PSTN |
| 18:51.28 | [TK]D-Fender | CrazyTux: not recommended, but you'd do something like "member => SIP/provider/1234567" |
| 18:51.54 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:51.58 | [TK]D-Fender | CrazyTux: it could lead to a phone getting a massive number of calls and kill batteries on a cell, etc. |
| 18:52.13 | CrazyTux | [TK]D-Fender, so something like.. |
| 18:52.20 | *** join/#asterisk FarrisG (n=lckirk@gateway.wiquest.com) |
| 18:52.23 | CrazyTux | [TK]D-Fender, member => SIP/hostname/CALLER_ID |
| 18:52.45 | [TK]D-Fender | CrazyTux: No, not callerid, the # you want to dial. jsut shove it the way you would were you to DIAL it. |
| 18:53.05 | CrazyTux | [TK]D-Fender, so what is "provider" |
| 18:53.11 | CrazyTux | [TK]D-Fender, constant? |
| 18:53.17 | [TK]D-Fender | CrazyTux: no, its a SAMPLE. |
| 18:53.38 | [TK]D-Fender | CrazyTux: you put your PEER in there. |
| 18:53.52 | CrazyTux | [TK]D-Fender, Ohhh ok gotcha |
| 18:54.27 | anonymouz666 | russellb: the blog post is great. |
| 18:54.45 | russellb | thanks |
| 18:55.31 | *** join/#asterisk marcan (n=marcanso@160.10.7.121) |
| 18:59.36 | *** join/#asterisk kikoafonso (n=rafonso@cronopio.rits.org.br) |
| 19:06.30 | SoftIce | please can somebody here that uses freepbx and has used asterisk gui tell me what they prefure? |
| 19:07.11 | DrukenLPY | pico :) |
| 19:07.23 | russellb | pfft ... vim |
| 19:07.31 | SoftIce | goodness |
| 19:07.31 | russellb | but seriously, asterisk-gui ! |
| 19:07.37 | SoftIce | russellb: really? |
| 19:07.42 | SoftIce | what makes it better than freepbx? |
| 19:07.43 | russellb | yes, but i'm biased |
| 19:07.55 | SoftIce | heh :) |
| 19:08.04 | SoftIce | russellb: if you where not biost? |
| 19:08.08 | SoftIce | biased :) |
| 19:08.16 | brimstone | SoftIce, look at your conf files after you realize that AMP can't do something you need |
| 19:08.35 | SoftIce | brimstone: I do realise that it messes things up |
| 19:08.41 | russellb | asterisk-gui is *much* nicer at letting you manually edit the config when you need to |
| 19:09.04 | IOscanner | It is not hard to modify AMP to do what you want |
| 19:09.29 | SoftIce | russellb: well what about with regards to things like, umm call reports, does ast-gui have that too built in? |
| 19:09.45 | russellb | don't think it has any of that yet |
| 19:09.59 | SoftIce | I see, not hard to implement though |
| 19:10.06 | russellb | nope .. |
| 19:10.09 | SoftIce | well i'm meaning if you using a2billing, etc |
| 19:10.13 | SoftIce | its all in there allready |
| 19:10.34 | SoftIce | what type of damage do you get running a combination of freepbx and asterisk gui? |
| 19:10.49 | russellb | the machine will catch on fire |
| 19:10.56 | russellb | really, it's a bad idea |
| 19:11.04 | SoftIce | heh :) |
| 19:11.19 | SoftIce | specially seeing that fact anything not writen by freepbx will get overwriten when applying changes, etc |
| 19:11.30 | DrukenLPY | catastrophie?? |
| 19:14.37 | SoftIce | 1 last question I think, does ast-gui have implemeting and writing directly to fop configs? |
| 19:14.42 | brimstone | SoftIce, both AMP and asterisk-gui take different mentalities on managing the configs, so it's highly likely that one will break the other one to the ponit of not working |
| 19:15.01 | brimstone | s/ponit/point/ |
| 19:15.18 | Mercestes | IOscanner, it's not hard to edit a conf file to do what you want either. |
| 19:15.19 | Corydon-w | brimstone: I think he's talking about Flash Operator Panel |
| 19:15.42 | Corydon-w | (a manager interface) |
| 19:16.02 | brimstone | Corydon-w, that was in reference to the question about running both on the same box, if you notice the 5 sec time delay, i can't type that fast |
| 19:16.15 | Peri | FOP isn't to be trusted entirely either |
| 19:16.17 | Corydon-w | brimstone: sure you can... ;-) |
| 19:17.16 | brimstone | 360 words a minute? |
| 19:17.53 | SoftIce | ok last question |
| 19:17.58 | SoftIce | if you make modifications in asterisk-gui |
| 19:18.06 | SoftIce | what happens if you modify the configs directly |
| 19:18.12 | SoftIce | does it £*"%*"£% like freepbx ? |
| 19:18.16 | SoftIce | and overwrite changes? |
| 19:18.21 | brimstone | for the most part, they're reflected back in asterisk-gui |
| 19:18.33 | brimstone | but either way, the gui's going to just ignore anything it can't parse |
| 19:18.57 | Corydon-w | brimstone: like the contents of func_odbc ? ;-) |
| 19:19.11 | brimstone | like the contents of res_corydon |
| 19:21.28 | Mercestes | Hrm. |
| 19:21.44 | Mercestes | res_corydon. What methods does *that* object provide? :D |
| 19:22.21 | SoftIce | who uses odbc anyway |
| 19:22.30 | SoftIce | mysql all the way |
| 19:22.30 | brimstone | methods you're not old enough for Mercestes |
| 19:22.49 | Mercestes | I shall never be old enough for those methods. |
| 19:22.56 | brimstone | no one is... |
| 19:23.03 | Mercestes | well, except Corydon |
| 19:23.08 | dlynes_laptop | and file |
| 19:23.10 | Corydon-w | brimstone: coil was plenty old |
| 19:23.13 | Mercestes | and file. |
| 19:23.16 | Mercestes | and may be Manxpower |
| 19:23.16 | dlynes_laptop | file loves to be defiled |
| 19:23.23 | *** part/#asterisk horvath (i=horvath@i.bring.lots.of.bad-ne.ws) |
| 19:23.28 | dlynes_laptop | by corydon of course |
| 19:23.35 | J4k3 | ODBC = Over Dose of Backward Computing |
| 19:23.38 | brimstone | yikes, that explains some problems with that child |
| 19:23.44 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 19:24.00 | Mercestes | So if someone breaks up with File, are they defiling themselves? |
| 19:24.28 | dlynes_laptop | No, they broke up with him because they got syfileus |
| 19:26.07 | file | hrm? |
| 19:28.01 | *** join/#asterisk _DAW (n=chatzill@adsl-222-56-130.msy.bellsouth.net) |
| 19:29.39 | *** join/#asterisk Siya (n=djerk@217-195-248-251.dsl.easynet.nl) |
| 19:30.02 | Siya | ello |
| 19:30.31 | *** join/#asterisk Strom_M (n=strom@135.196.213.180) |
| 19:30.39 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-ce6e94c256e3a810) |
| 19:30.41 | Mercestes | Did you say hello? |
| 19:30.52 | bkruse | Mercestes: hello :"D |
| 19:31.03 | Mercestes | Heya, bk. |
| 19:31.08 | J4k3 | lo. |
| 19:32.33 | _DAW | I noticed that there was a bugfix for the 256 byte temp storage buffer limit in app_voicemail in 1.4.3. Is there a similar problem with the page application? Or are all asterisk applications limited to 256 bytes? |
| 19:35.22 | Siya | Mercestes: I did :) |
| 19:35.47 | Siya | bkruse: hello! |
| 19:36.20 | irule | hi guys, how can I send a busy signal in case the only ZAP channel is busy? it currently dials another extension that is not busy |
| 19:36.39 | SnrWup | you cant |
| 19:36.43 | bkruse | Siya: hey |
| 19:36.48 | SnrWup | it would require a recodig of the * source |
| 19:36.59 | SnrWup | =] |
| 19:37.05 | SnrWup | in truth i have no idea |
| 19:37.13 | Siya | bkruse: Any progress on multi-extension dialing in *now? |
| 19:37.26 | bkruse | Siya: no, i have some other priorities for the week, my apologies :[ |
| 19:37.40 | Siya | bkruse: I know what that's like... |
| 19:37.43 | bkruse | ;[ |
| 19:39.15 | Siya | bkruse: I managed to config it in extensions.conf for now, not very generic but it works for me (for now) |
| 19:39.17 | *** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@145-27.mc.cite.net) |
| 19:39.30 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 19:39.45 | _VoiceMeUp_COM | when dialing from peer 1 -> peer 2 to outbound proxy , the rtp is sent back to peer1 even if peer2 says canreinvite=no |
| 19:39.47 | _VoiceMeUp_COM | any idea ? |
| 19:40.05 | bkruse | Siya: make a macro for it |
| 19:40.07 | _VoiceMeUp_COM | i also added fromdomain=peer2 but the VIA still says peer1 |
| 19:40.12 | _VoiceMeUp_COM | how can i cagne via ? |
| 19:40.15 | _VoiceMeUp_COM | change |
| 19:40.16 | bkruse | or, better yet, hmm, just /msg me and we will talk about it ;] |
| 19:40.20 | bkruse | or chat in the asterisk-gui room! |
| 19:41.46 | Siya | then again... |
| 19:42.13 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:42.21 | _VoiceMeUp_COM | hmm |
| 19:42.26 | _VoiceMeUp_COM | so any via rewriteavail ? |
| 19:42.42 | Mercestes | irule, If you try to dial out your only zap channel and it's busy you should get Congestion() what exactly is broken? |
| 19:42.50 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:43.58 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 19:43.58 | *** mode/#asterisk [+o anthm] by ChanServ |
| 19:45.41 | *** join/#asterisk jmacz (n=jmacz@190.25.33.122) |
| 19:46.25 | *** join/#asterisk ibismalta (n=chris@62.173.5.186) |
| 19:46.28 | irule | Mercestes I have no idea, I just noticed this behavior, and I also would like to understando how to make sure how to handle different status like busy or what if the telco is not giving me a dial tone propperly |
| 19:47.15 | ibismalta | does anybody know if there is any other option to raise the volume of the TDM400P apart from the attenuation levels? |
| 19:47.26 | *** join/#asterisk NLok (i=Lok@gunlok.rh.rit.edu) |
| 19:47.28 | Mercestes | The variable ${DIALSTATUS} should help. |
| 19:48.47 | irule | Mercestes http://pastebin.ca/459955 this is my current newbie dialplan if you dont mind taking a look, thanks |
| 19:48.51 | jm|laptop | anybody know why I am being udp flooded from darkvoip.net ? |
| 19:49.02 | jm|laptop | even when * isn't running ... |
| 19:49.20 | irule | all other criticism is always taken constructively :) |
| 19:49.26 | vicentealencar | ~agi api |
| 19:49.35 | jbot | i heard agi api is at http://home.cogeco.ca/~camstuff/agi.html |
| 19:49.36 | Siya | jm|laptop: sounds like a nice DOS |
| 19:49.52 | jm|laptop | yes; I thought so |
| 19:49.55 | jm|laptop | it's not working very well |
| 19:49.58 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:50.03 | jm|laptop | but it is taking quite some bandwidth |
| 19:50.21 | Siya | UDP can do that, TCP will back off. UDP won't |
| 19:50.29 | SoftIce | dont touch me? |
| 19:50.47 | ibismalta | I have been working with asterisk for 2 years but I installed it on a dell optiflex using FC6 and I seem I cannot control the volime levels |
| 19:50.59 | Siya | jm|laptop: have you sniffed any of the packets? |
| 19:51.01 | ibismalta | I have made txgain upto 46 but it does not seem to amke a difference |
| 19:51.09 | jm|laptop | yeah |
| 19:51.17 | jm|laptop | looks like random crap: all 172 in length |
| 19:51.18 | Siya | and? |
| 19:51.21 | jm|laptop | random or encrypted |
| 19:51.39 | jm|laptop | 20:51:32.496308 IP 64.246.18.157.13772 > 192.168.1.102.18796: UDP, length 172 |
| 19:51.39 | jm|laptop | E.....@./.5t@......f5.Il...o..?.&...).hEz...................................wqs{|tnosrmknx}xrsuojinz|smmppnmqy||~~zvsty}~...~zzzz{.....tt}.......{...yu....wt}...}rmmorqlijnswvplil~..|uz...us.......}{. |
| 19:51.54 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
| 19:52.17 | Siya | if they're very evenly spaced then I suspect someone is using iperf or something to target you |
| 19:52.38 | Siya | 192.168 is private though |
| 19:52.50 | Siya | must be using an open port in ur firewall |
| 19:52.50 | jm|laptop | Siya: I know but I have binat going on |
| 19:52.58 | jm|laptop | that tii |
| 19:53.00 | jm|laptop | too* |
| 19:53.01 | Siya | jm|laptop: not stateful then |
| 19:53.25 | jm|laptop | I guess I'll firewall them :S |
| 19:53.32 | jm|laptop | I did visit their website the other day |
| 19:55.35 | Mercestes | irule, what exactly does ignorepat => do again? |
| 19:58.08 | [TK]D-Fender | ~mercestes |
| 19:58.14 | jbot | well, mercestes is almost a total nub |
| 19:58.14 | *** join/#asterisk JunK-Y (n=junky@modemcable105.205-56-74.mc.videotron.ca) |
| 19:58.21 | Mercestes | Shush, Fender. |
| 19:58.31 | [TK]D-Fender | I said nothing ;) |
| 19:58.44 | Mercestes | irule, What type of phones are you using and why do they require an "ignorepat"? And why are you using Ignorepat twice in a context? |
| 20:00.02 | *** join/#asterisk yenno (i=becks@80-219-246-216.dclient.hispeed.ch) |
| 20:00.04 | jm|laptop | 8115 packets captured |
| 20:00.04 | jm|laptop | 8115 packets received by filter |
| 20:00.04 | jm|laptop | 0 packets dropped by kernel |
| 20:00.04 | jm|laptop | real 2m39.408s |
| 20:00.06 | Mercestes | irule, and does this phone line have call waiting on it? |
| 20:00.27 | jm|laptop | firewalling isn't going to help when it has already gone down the line :| |
| 20:00.27 | vicentealencar | hey guys, is it possible to make the agi(script) function return a value? |
| 20:00.28 | yenno | hi, i want asterisk to simulate ringing when clients call each other. what do i need to change? my current extensions.conf: http://pastebin.ca/459986 |
| 20:00.39 | vicentealencar | I mean, I want it to return true or false |
| 20:00.43 | vicentealencar | is it possible/ |
| 20:02.05 | irule | Mercestes I think that ignorepat = 9 is to get a dialtone after 9 is dialed to make the user think they have an external line waiting to dial, thats what they expect |
| 20:02.45 | Mercestes | irule, Yea, I googled that because it was faster. |
| 20:02.55 | irule | Mercestes ignorepat twice? Ill have to check that, I was not aware of that thanks for noticing, I am unnewbieing as we speak lol |
| 20:03.40 | irule | Mercestes the type of phones? I am using xten sip phone, a sipura ata and a couple plain old telephones from the stone age |
| 20:04.02 | Mercestes | Does ignorepat really work with a SIP softphone? |
| 20:04.10 | *** join/#asterisk ohadz (n=ohad@19-231-13-72.static.cosmoweb.net) |
| 20:04.11 | wunderkin | no |
| 20:04.20 | Mercestes | I didn't think so |
| 20:04.47 | irule | Mercestes I am not aware of the fact that it has call waiting or not, where is the code related to it so that I investigate? ....ignorepat with sip?b beats me, I have not tested it yet lol |
| 20:05.02 | *** join/#asterisk n0n0x (n=n0n0x@200.127.112.200) |
| 20:05.05 | ohadz | hi all, does anyone know why the linksys ip phone spa921 on * sounds like a robot when in one of the conf 5000-5005 rooms? |
| 20:05.06 | Mercestes | irule, You hook a phone to it and call yoru # twice. |
| 20:05.25 | vicentealencar | is it possible to make the agi(script) application return a specific value, like true or false? |
| 20:05.38 | n0n0x | hi!, quesiton: someone running asterisk on FC5 with free codec_g729 module?? |
| 20:06.07 | *** join/#asterisk phearless (n=phear@host217-34-75-65.in-addr.btopenworld.com) |
| 20:06.20 | n0n0x | i'm getting this error when trying to load the module: "wrong ELF class: ELFCLASS32" |
| 20:07.57 | irule | Mercestes my number twice? |
| 20:08.04 | irule | what do you mean? |
| 20:08.31 | Mercestes | n0n0x, it means that you can't be that class as an elf. I suggest choosing human. they can be any class, level to 20, and dual class instead of multi class without penalties. |
| 20:08.43 | [TK]D-Fender | irule: You can't use ignorepat with SIP, only analog Zaptel channels |
| 20:08.55 | Mercestes | irule, get a phone. Plug it into your phone line. Dial a number. Answre the phone. Dial your number again. See if it beeps. |
| 20:09.07 | n0n0x | Mercestes, fixed, i forgot that my box is running FC5 x86_64 :P |
| 20:09.11 | n0n0x | Mercestes, thank you! |
| 20:09.14 | Mercestes | Note, there is no "hangup" directive anywhere in that so you have to have *two* calls going at the same time to test it |
| 20:09.21 | Mercestes | n0n0x, Glad I could help! |
| 20:09.38 | ohadz | anyone? |
| 20:09.58 | [TK]D-Fender | the "3d20" modifier makes ALL the difference ;) |
| 20:10.00 | irule | oh, ok thanks |
| 20:10.06 | Mercestes | Of course. :D |
| 20:10.56 | *** join/#asterisk `p4r14h (n=j0sh@206-192.69-92-cpe.cableone.net) |
| 20:11.05 | n0n0x | Mercestes, actually i didn't read your first message, hahahaha :P |
| 20:11.13 | *** join/#asterisk ringhals (i=fwuser@firewall.drgutah.com) |
| 20:11.31 | n0n0x | pretty funny, should it be at bash.org :P |
| 20:11.31 | Mercestes | n0n0x, I noticed. :D |
| 20:12.40 | *** join/#asterisk pariah (n=j0sh@unaffiliated/pariah) |
| 20:13.56 | *** part/#asterisk naitram (n=ttech@216.77.58.40) |
| 20:15.38 | yenno | i want asterisk to simulate ringing when clients call each other. what do i need to change? my current extensions.conf: http://pastebin.ca/459986 |
| 20:15.55 | *** join/#asterisk Siya (n=djerk@217-195-248-251.dsl.easynet.nl) |
| 20:15.57 | Siya | re |
| 20:18.13 | dlynes_laptop | yenno: assuming you donm't have any problems, ringing should be generated automatically |
| 20:18.13 | ohadz | hi all, does anyone know why the linksys ip phone spa921 on * sounds like a robot when in one of the conf 5000-5005 rooms? |
| 20:18.24 | yenno | ok dlynes_laptop... well i don't hear ringing on my device when i'm calling somebody |
| 20:19.11 | dlynes_laptop | I'm just curious if anyone has wanpipe working with zaptel 1.2.13 and higher, or any of the 1.4 series zaptels? |
| 20:19.47 | dlynes_laptop | yenno: what technology are their phones using? |
| 20:20.09 | dlynes_laptop | yenno: zaptel? sip? h323? iax2? isdn? |
| 20:20.12 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 20:20.39 | yenno | sip |
| 20:20.54 | dlynes_laptop | yeah...ringing shojuld be automatically generated then |
| 20:21.07 | syzygyBSD | asterisk seems to be dropping calls recently, more recently after I updated to 1.2.17 but it was droping them before |
| 20:21.19 | yenno | so asterisk should send back a 200 OK to my phone :/ but he doesn't |
| 20:21.39 | syzygyBSD | I can't find a cause, or a way to duplicate it, are there some suggestions for debugging it? |
| 20:22.00 | syzygyBSD | I *Think* it is only sip -> zap calls |
| 20:22.38 | bkruse | syzygyBSD: ANY error msgs'? |
| 20:22.44 | syzygyBSD | no |
| 20:22.55 | syzygyBSD | makes it harder |
| 20:22.55 | *** join/#asterisk jazzanova (n=boris@S01060011954e59bc.vc.shawcable.net) |
| 20:22.56 | jazzanova | hi |
| 20:23.22 | ringhals | syzyg I am having the same issue |
| 20:23.34 | syzygyBSD | what version are you on? |
| 20:23.40 | ringhals | 1.2.17 |
| 20:23.44 | jazzanova | i want to call a phone using a modem and play a recording. i just installed asterisk. instructions talk about zaptel. do i need zaptel for this ? |
| 20:23.45 | ringhals | and I ahve issues both ways |
| 20:24.26 | ringhals | but I have issues with people calling in to sip phone / into meetme rooms / and sip out |
| 20:25.09 | syzygyBSD | ringhals: yes, issues both ways, but I haven't noticed them zap-zap or sip-sip |
| 20:25.18 | ringhals | me either |
| 20:25.21 | syzygyBSD | zap-sip and sip-zap... |
| 20:25.26 | ringhals | I do not have any zap - zap devices |
| 20:25.53 | syzygyBSD | hmm, I am wondering if a reboot would help |
| 20:26.01 | ringhals | no |
| 20:26.02 | syzygyBSD | <PROTECTED> |
| 20:26.05 | ringhals | not for me anyway |
| 20:26.08 | syzygyBSD | couldn't hurt... |
| 20:26.23 | ringhals | I rebooted 3 times today LOL |
| 20:26.30 | syzygyBSD | ok then... probably not |
| 20:27.46 | DJ-Pyro_ | what's the easiest way to rewrite a sip header, asterisk can't do it, can openser? |
| 20:27.51 | syzygyBSD | ringhals: have you tried http://www.voip-info.org/wiki/view/Asterisk+PSTN+interface+debugging yet? |
| 20:28.21 | ringhals | syzygyBSD: not yet |
| 20:28.24 | ringhals | but I will |
| 20:30.03 | syzygyBSD | ya, I am going to try callprogress=no |
| 20:30.08 | syzygyBSD | hopefully it will fix it |
| 20:30.12 | ringhals | for sure |
| 20:30.18 | ringhals | I will need to look at my config |
| 20:30.25 | ringhals | btw I do not seem to have zmonitor |
| 20:31.38 | syzygyBSD | ztmonitor? |
| 20:31.42 | jazzanova | can someone point to docs on using a simple modem with asterisk ? |
| 20:31.51 | jazzanova | no VOIP |
| 20:32.04 | syzygyBSD | ~docs |
| 20:32.15 | jbot | methinks 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 |
| 20:32.17 | syzygyBSD | hmm... |
| 20:32.17 | syzygyBSD | !book |
| 20:32.50 | *** join/#asterisk Rahail (n=Rahail@209-19-88-240.detroit.mi.D-Conn.net) |
| 20:32.51 | aydiosmio | jazzanova: you mean dialing a modem with AT? |
| 20:33.03 | aydiosmio | through a serial port? |
| 20:33.04 | Rahail | Any one care to help me with asterisk |
| 20:33.11 | Rahail | ? uninstall |
| 20:33.21 | Rahail | ! uninstall |
| 20:33.23 | jazzanova | aydiosmio: yes |
| 20:34.04 | ringhals | hum.. |
| 20:35.00 | syzygyBSD | Rahail: ask a real question |
| 20:35.19 | aydiosmio | jazzanova: doesn't look like there's support for it directly, but sending commands to a modem is exceedingly easy with a perl script (AGI) |
| 20:35.35 | aydiosmio | Device::Modem is a perl extension written in pure perl that allows to communicate with modems, or generically with AT command set compliant devices |
| 20:35.36 | jazzanova | aydiosmio: how can I play a message into it? |
| 20:35.40 | syzygyBSD | I know at one point there was a chan_modem |
| 20:35.48 | ringhals | I have callprogress=no and busydetect=no |
| 20:35.54 | Rahail | Stopping retransmission on '7cd54d331e1d1eed71659ab5123d4b1a@72.4.160.99' of Request 102: Match Found |
| 20:35.58 | syzygyBSD | have or had? |
| 20:35.59 | Rahail | what is this |
| 20:36.03 | aydiosmio | that requires voice modem support, which is far more comlicated |
| 20:36.12 | ringhals | had |
| 20:36.17 | ringhals | and have :-) |
| 20:36.30 | ibismalta | does anybody know if there is any other option to raise the volume of the TDM400P apart from the attenuation levels in zapata.conf? |
| 20:36.44 | jazzanova | aydiosmio: voice modem support in the kernel ? |
| 20:37.09 | aydiosmio | here we go |
| 20:37.16 | aydiosmio | use Modem::Vgetty; |
| 20:37.19 | aydiosmio | http://search.cpan.org/~yenya/Modem-Vgetty-0.03/Vgetty.pm |
| 20:37.42 | aydiosmio | that should do what you want |
| 20:38.16 | jazzanova | ok, great. |
| 20:38.23 | jazzanova | thank you :) |
| 20:38.24 | syzygyBSD | ringhals: hmm, I had callprogress=yes, hopefully =no will fix my problem |
| 20:38.44 | ringhals | I hope so also |
| 20:39.09 | ringhals | I am going to redo my entire zapata.conf as it is a copy over from a very very old version |
| 20:39.30 | *** join/#asterisk emiquelito (n=evandro@200-170-201-155.core01.spo.ifx.net.br) |
| 20:39.31 | syzygyBSD | ringhals: ztmonitor is in your zaptel source folder |
| 20:39.35 | *** part/#asterisk emiquelito (n=evandro@200-170-201-155.core01.spo.ifx.net.br) |
| 20:39.39 | ringhals | thanks |
| 20:44.19 | *** join/#asterisk altaergo (n=alta@mail.americanfamilyfunds.com) |
| 20:48.27 | altaergo | I have an asterisknow vmware image I'm having a hard time getting connect to broadvoice. I'm looking for error messages, but 'asterisk -r' is giving me info overload... Anyone know of a file I can grep that would give me just the login stuff? |
| 20:48.27 | altaergo | well, I see there's an asterisk now channel, maybe I'm just in the wrong place |
| 20:55.58 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 20:56.06 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:58.24 | *** part/#asterisk jazzanova (n=boris@S01060011954e59bc.vc.shawcable.net) |
| 20:59.01 | dlynes_laptop | altaergo: /var/log/asterisk/full, usually |
| 20:59.39 | dlynes_laptop | I'm just curious if anyone has wanpipe working with zaptel 1.2.13 and higher, or any of the 1.4 series zaptels? |
| 21:01.27 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 21:01.40 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 21:10.09 | [TK]D-Fender | dlynes_laptop, I do - Wanpipe 2.3.4-7 |
| 21:10.58 | jsolares | lovely R2 :(, now i'm getting wrong callerid |
| 21:12.29 | *** join/#asterisk alrs_ (i=nobody@cpe-76-81-15-170.socal.res.rr.com) |
| 21:13.35 | [TK]D-Fender | dlynes_laptop, 1.2.16 |
| 21:14.53 | syzygyBSD | hmm, if I turn off callprogress on my pri how will it affect my dialplan? |
| 21:16.29 | Rahail | ? upgrade |
| 21:16.53 | _VoiceMeUp_COM | : when dialing from peer 1 -> peer 2 to outbound proxy , the rtp is sent back to peer1 even if peer2 says canreinvite=no |
| 21:16.53 | _VoiceMeUp_COM | <PROTECTED> |
| 21:16.56 | _VoiceMeUp_COM | any idea ? |
| 21:17.02 | bkruse | fromuser |
| 21:17.04 | _VoiceMeUp_COM | any call i place to that switch hairpins back to peer1 |
| 21:17.15 | _VoiceMeUp_COM | bkruse me ? |
| 21:17.21 | _VoiceMeUp_COM | ip auth |
| 21:17.27 | bkruse | _VoiceMeUp_COM: ya, give it a shot, still |
| 21:17.30 | _VoiceMeUp_COM | hmm |
| 21:17.35 | bkruse | the username still has to come across, and thus gets routed somewhere |
| 21:18.12 | _VoiceMeUp_COM | well path is .. cisco phone -> peer1 --> peer2 --> outboudn switch |
| 21:18.12 | _VoiceMeUp_COM | flow is doign this.. cisco --> peer 1 --> peer2 --> outbound switch --> peer 1 on answer |
| 21:18.19 | bkruse | alot of times, i do IP auth but it gets assigned credentials of another user (if you dont pay close attention) and go to the wrong context, OR, (i think your case) gets matched to a user which ip is the same of the one coming in, then getting your loopback, in a sense |
| 21:18.27 | _VoiceMeUp_COM | ah |
| 21:18.28 | bkruse | oh, hmm |
| 21:18.29 | _VoiceMeUp_COM | yeah |
| 21:18.33 | _VoiceMeUp_COM | that whats happening |
| 21:18.52 | _VoiceMeUp_COM | ok so from user should be waht.. since its ip based. |
| 21:19.12 | _VoiceMeUp_COM | canreinvite = no |
| 21:19.23 | _VoiceMeUp_COM | so why would it even loop lol ill try |
| 21:19.34 | bkruse | k |
| 21:20.01 | _VoiceMeUp_COM | fromdomain=peer2ip |
| 21:20.03 | _VoiceMeUp_COM | i added |
| 21:20.12 | _VoiceMeUp_COM | also.. it mods the contact in the FROM.. but not the VIA |
| 21:20.14 | bkruse | k |
| 21:20.16 | _VoiceMeUp_COM | formuser should fix ? |
| 21:20.25 | _VoiceMeUp_COM | or cough SEr i guess |
| 21:20.25 | bkruse | fromuser will populate that field, i hope |
| 21:20.26 | bkruse | try it |
| 21:20.29 | bkruse | ha |
| 21:20.34 | _VoiceMeUp_COM | but its not the ip |
| 21:20.40 | _VoiceMeUp_COM | i mean not the user but IP the prob |
| 21:20.49 | _VoiceMeUp_COM | i would need a fromvia = lol |
| 21:20.55 | *** join/#asterisk kippi (n=cowen@89.240.231.255) |
| 21:21.22 | kippi | hey |
| 21:21.32 | kippi | what is the best way to link two asterisk systems together? |
| 21:22.24 | blitzrage | SIP or IAX2 |
| 21:22.33 | _VoiceMeUp_COM | same |
| 21:22.33 | _VoiceMeUp_COM | sip |
| 21:22.43 | blitzrage | I use SIP with no problems |
| 21:22.48 | kippi | any reason to go with IAX2? |
| 21:23.22 | blitzrage | well, if your devices are SIP and you link the two Asterisk boxes via IAX2, then you can't reinvite the media directly to the endpoints |
| 21:23.54 | blitzrage | honestly, just keep it all SIP if that's what the endpoints are |
| 21:23.59 | _VoiceMeUp_COM | lol |
| 21:24.01 | _VoiceMeUp_COM | even wort |
| 21:24.08 | _VoiceMeUp_COM | i called that # via another trunk |
| 21:24.11 | _VoiceMeUp_COM | and doign same |
| 21:24.24 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:25.00 | kippi | ah ok, sip it is |
| 21:25.13 | kippi | guess there are loads of docs on how to do it? |
| 21:25.16 | justdave | if your different boxes are in different private LANs you're not going to want to reinvite the media anyway |
| 21:27.23 | *** join/#asterisk BSD_Tech (n=bsdtech@ppp-69-239-113-126.dsl.irvnca.pacbell.net) |
| 21:27.58 | kippi | going on a hunt now to read up on sip links |
| 21:28.13 | *** join/#asterisk mesfet (n=iw3grx@89-96-227-146.ip14.fastwebnet.it) |
| 21:28.33 | *** join/#asterisk zeeesh (n=aadilism@202.125.143.67) |
| 21:28.35 | zeeesh | hi |
| 21:29.01 | mesfet | Hi all. I have a question: which is the best driver for 2 HFC-S cards? bristuff, vISDN, .... |
| 21:29.49 | Rahail | any one here have free time to help me with asterisk |
| 21:29.56 | *** join/#asterisk RaYmAn-Bx (i=rayman@skumler.dk) |
| 21:33.16 | Corydon-w | ~ask |
| 21:33.20 | 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. |
| 21:33.22 | [TK]D-Fender | Rahail, Ask a specific question and you may get a specific answer |
| 21:34.55 | [TK]D-Fender | There! Snug as a bug in a hermetically sealed straight-jacket! ;) |
| 21:35.03 | Corydon-w | [TK]D-Fender: I'm harmless. :-P |
| 21:35.12 | *** join/#asterisk killfill (n=killfill@201.215.221.113) |
| 21:35.14 | killfill | hi |
| 21:35.16 | [TK]D-Fender | Corydon-w, "Mostly" ;) |
| 21:35.17 | kippi | is there a how to for connecting asterisk boxes together? |
| 21:35.23 | _VoiceMeUp_COM | duck tape |
| 21:35.24 | Corydon-w | I don't bite hard |
| 21:35.24 | _VoiceMeUp_COM | ;) |
| 21:35.33 | kippi | hehe |
| 21:35.34 | [TK]D-Fender | kippi, CrazyGlue (tm) |
| 21:35.48 | killfill | asterisk-addons 1.4.0 doesnt compile with asterisk 1.4.3: src/chan_h323.c:250: warning: passing arg 7 of `ast_channel_alloc' makes pointer from integer without a cast |
| 21:35.52 | killfill | any tips?.. :) |
| 21:36.15 | killfill | oh.. there is a new version |
| 21:36.32 | [TK]D-Fender | killfill, Slap yourself now... we're too tired :) |
| 21:36.39 | killfill | jeje |
| 21:36.42 | russellb | i got it. |
| 21:36.48 | [TK]D-Fender | killfill, Ok.. except for russellb... he's bad-ass :D |
| 21:36.53 | russellb | or! |
| 21:36.57 | russellb | ~lart killfill |
| 21:36.57 | jbot | pushes the wall down onto killfill whilst whistling innocently |
| 21:36.58 | *** join/#asterisk Supaplex (i=supaplex@166-70-62-199.ip.xmission.com) |
| 21:37.47 | russellb | [TK]D-Fender: busy right now? I'm not doing anything if you want to chat about SLA ... |
| 21:38.07 | Supaplex | as public_html (mod_userdir) is to apache, as ____________ is to asterisk? |
| 21:38.07 | [TK]D-Fender | russellb, Sure, I've got a few |
| 21:38.28 | russellb | [TK]D-Fender: ok, well, where would you like me to start? |
| 21:39.48 | *** join/#asterisk VioByte (n=afreeman@unallocated-63-210-246-146.cvg1.bytehosting.com) |
| 21:39.50 | dlynes_laptop | [TK]D-Fender: so the problem with it working with 1.2.13 has been eliminated then? |
| 21:40.06 | dlynes_laptop | [TK]D-Fender: I just remember it stopped working with 1.2.13 because of all the new changes added to support the new cards |
| 21:41.43 | [TK]D-Fender | dlynes_laptop, I've been interruption free... |
| 21:43.53 | *** join/#asterisk Acidcrawl (n=Miranda@12.168.96.254) |
| 21:44.42 | Acidcrawl | can you analyze the inbound number in the extensions.conf and route based on that? |
| 21:45.05 | *** join/#asterisk thoughtpolice (n=austin@c75-111-146-82.plaicmtc01.tx.dh.suddenlink.net) |
| 21:45.35 | *** join/#asterisk Exhar (n=Roy@213-73-139-87.cable.quicknet.nl) |
| 21:45.37 | zeeesh | i hv peer 100 at asterisk server "A" and peer 200 at asterisk server "B" how can both communicate with each other by using SIP through xlite??? |
| 21:46.38 | *** join/#asterisk uppal (n=uppal@2001:618:400:888c:218:deff:fe9f:a77f) |
| 21:47.11 | *** join/#asterisk ToyMan (n=Stuart@72.168.167.241) |
| 21:49.17 | zeeesh | how asterisk server "A" users can call asterisk server "B" users .... will anybody guide ... pls ?///// |
| 21:51.23 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-163-90.ny325.east.verizon.net) |
| 21:53.57 | _VoiceMeUp_COM | xzeeesh |
| 21:53.58 | _VoiceMeUp_COM | cant |
| 21:54.03 | _VoiceMeUp_COM | need to share reg's on db |
| 21:54.07 | _VoiceMeUp_COM | use SER |
| 21:54.33 | Qwell[] | _VoiceMeUp_COM: You're kidding, right? |
| 21:54.55 | _VoiceMeUp_COM | well.. he means have USER A reged to server A call a user B on Server B ? |
| 21:55.26 | _VoiceMeUp_COM | coz if they not shared.. then Box B wont know anything on authentification of user A , right ? |
| 21:55.39 | _VoiceMeUp_COM | so both box needs both boxes reg's |
| 21:55.51 | _VoiceMeUp_COM | maybe im just not reading question right |
| 21:56.08 | jm|laptop | for those that remember my udp 'DoS' - I emailled the guy and it has stopped now |
| 21:56.16 | zeeesh | guys look complete question is |
| 21:56.16 | jm|laptop | ... which is nice. |
| 21:56.16 | lude | wow |
| 21:56.19 | lude | got SLA working |
| 21:56.45 | _VoiceMeUp_COM | jm|laptop ? which guy ? |
| 21:56.45 | zeeesh | i hv peer 100 and 101 at asterisk server "A" both can communicate with each other by using this extension ( "exten => 100,1,Dail(SIP/100)" and "exten => 101,1,Dial(SIP/101)" same i hv 2 peers 200 and 201 at asterisk server "B" both also 200 and 201 can comunicate with each other ... how can asterisk server "A" peers(100,101) and asterisk server "B" peers (200,201) can comunicate with each other by using SIP not IAX ... through xlite ..??? |
| 21:56.45 | _VoiceMeUp_COM | ;) |
| 21:56.45 | lude | i *really* wish i could configure the speed dial on these 7960's from tftp though |
| 21:56.55 | bkruse | lude: you cannot? |
| 21:57.01 | bkruse | lude: how much provisioning experience do you have? |
| 21:57.08 | jm|laptop | _VoiceMeUp_COM: owner of the domain that was "DoS"ing me |
| 21:57.10 | _VoiceMeUp_COM | zeesh ,: wahts the CLI say when you try ? |
| 21:57.11 | lude | "enough" |
| 21:57.12 | bkruse | anyone in here have alot of provisioning experience, or maybe just in SNOM for now. |
| 21:57.19 | _VoiceMeUp_COM | jm|laptop ;) #define domain |
| 21:57.28 | _VoiceMeUp_COM | #define DOS style |
| 21:57.30 | lude | it's one of the variables, like auto answer, that you just can't set via tftp |
| 21:57.37 | _VoiceMeUp_COM | jus got rid of a DNS DOs here |
| 21:57.41 | jm|laptop | er darkvoip.net ....?! |
| 21:57.42 | *** join/#asterisk Exhar (n=Roy@213-73-139-87.cable.quicknet.nl) |
| 21:57.47 | lude | and setting up SLA for 4 bridged lines on like 25 phones |
| 21:57.48 | jm|laptop | it wasn't a DoS |
| 21:57.48 | lude | not cool |
| 21:57.51 | jm|laptop | it was a mistake |
| 21:57.54 | *** part/#asterisk RaYmAn-Bx (i=rayman@skumler.dk) |
| 21:57.57 | Qwell[] | _VoiceMeUp_COM: uhh... |
| 21:58.06 | jm|laptop | and the admin resolved it quite quickly so I have no hard feelings |
| 21:58.09 | Qwell[] | _VoiceMeUp_COM: There are at least 2 methods to do that, VERY easily |
| 21:58.09 | _VoiceMeUp_COM | iptables -add 64.246.18.157 >> /ev/null |
| 21:58.16 | _VoiceMeUp_COM | Qwell k |
| 21:58.25 | zeeesh | <_VoiceMeUp_COM>:some time user authtication failed msg or channel unavailable |
| 21:58.26 | _VoiceMeUp_COM | not involving DB' ? |
| 21:58.30 | _VoiceMeUp_COM | realtime ? |
| 21:58.37 | _VoiceMeUp_COM | yeah |
| 21:58.48 | _VoiceMeUp_COM | since right now your BOX doesnt hold users from box b |
| 21:58.51 | Qwell[] | no |
| 21:58.51 | _VoiceMeUp_COM | you could use REALTIME |
| 21:58.57 | zeeesh | <_VoiceMeUp_COM> mostly user authntication problem |
| 21:58.59 | Qwell[] | iax2 switch or dundi |
| 21:59.03 | _VoiceMeUp_COM | or .. lets see Qwell[] answer |
| 21:59.03 | Qwell[] | trivial |
| 21:59.06 | _VoiceMeUp_COM | ah |
| 21:59.11 | _VoiceMeUp_COM | he #defined Not using IAX |
| 21:59.19 | _VoiceMeUp_COM | so your stuck with dundi |
| 21:59.20 | Qwell[] | Who says he has to? |
| 21:59.23 | zeeesh | yes not using IAX |
| 21:59.44 | _VoiceMeUp_COM | dundi is your soliution |
| 21:59.52 | _VoiceMeUp_COM | dundi could lookup where to send call |
| 22:00.03 | zeeesh | ok let me try .... |
| 22:00.05 | _VoiceMeUp_COM | since right now sip show peers wont show users of box b |
| 22:00.06 | zeeesh | thanx |
| 22:00.13 | _VoiceMeUp_COM | thank Qwell[] ;) |
| 22:00.31 | _VoiceMeUp_COM | im just doign a verbose on Qwell 's answers |
| 22:00.56 | bkruse | Qwell[] > all |
| 22:03.29 | _DAW | Does the page application have a 256 byte limitation like the voicemail app did? |
| 22:03.46 | Rahail | can some one help with asterisk |
| 22:03.56 | bkruse | haha |
| 22:03.58 | bkruse | i needed that |
| 22:04.00 | Qwell[] | Rahail: not unless you ask a question about it |
| 22:05.26 | _DAW | Looks like voicemail was fixed in 1.4.3 but page app still only appears to use the first 256 bytes. |
| 22:05.41 | Rahail | well I install asterisk register the extension but i dont know what this sip debug saying |
| 22:07.47 | Rahail | ? pastebing |
| 22:07.51 | Rahail | ? pastebin |
| 22:07.59 | Corydon-w | ~pb |
| 22:08.02 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 22:08.24 | Rahail | thank you |
| 22:08.27 | Rahail | Qwell http://pastebin.ca/460183 |
| 22:09.00 | _VoiceMeUp_COM | sip prune realtime Qwell[] |
| 22:09.28 | *** join/#asterisk sandorp (n=sandor@firewall2.wsi.net) |
| 22:10.12 | Corydon-w | Rahail: seems clear enough. Extension 5555 doesn't exist. |
| 22:10.59 | Rahail | i am trying to send this call to trunk |
| 22:11.44 | _VoiceMeUp_COM | so no way to change a vIA line right ? |
| 22:11.56 | _VoiceMeUp_COM | the via just a line saying passed torugh this box.. |
| 22:13.51 | *** join/#asterisk cspot (i=cspot@ip68-109-8-207.pn.at.cox.net) |
| 22:14.07 | Rahail | Corydon-w |
| 22:14.13 | Rahail | so what am I doign wrong then |
| 22:14.41 | *** join/#asterisk uppal (n=uppal@2001:618:400:888c:218:deff:fe9f:a77f) |
| 22:15.46 | *** join/#asterisk keith80403 (n=keith804@64-17-89-46.us.warpdriveonline.com) |
| 22:16.39 | *** join/#asterisk Exhar (n=Roy@213-73-139-87.cable.quicknet.nl) |
| 22:17.41 | Rahail | http://pastebin.ca/460194 |
| 22:20.41 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 22:20.47 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
| 22:22.13 | *** part/#asterisk ohadz (n=ohad@19-231-13-72.static.cosmoweb.net) |
| 22:22.24 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 22:22.25 | diclophis-work | hello all |
| 22:22.33 | diclophis-work | whats this all about: "failed to read gains: Invalid argument" |
| 22:22.37 | diclophis-work | i get that when i restart asterisk |
| 22:24.58 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 22:25.40 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
| 22:32.00 | *** join/#asterisk hlribeiro (i=hugo@201.14.62.25) |
| 22:34.27 | Siya | Anyone who can point me in the right direction with a public *(now) server and NAT-ed Cisco 7960 phones |
| 22:34.53 | Siya | Everything works but I think the nat rules timeout, registry doesn't break |
| 22:35.05 | Siya | when I dial from the phone everything works again |
| 22:35.30 | Siya | after a while the phones don;t ring when called |
| 22:37.02 | Siya | I have 2 7960's both in the same LAN so having them talk to eachother directly rather than via their public (nat-ed) address would be great too... |
| 22:37.12 | Siya | Anyone? |
| 22:37.26 | *** join/#asterisk mwebb (n=chatzill@cpe-76-183-188-152.tx.res.rr.com) |
| 22:37.48 | jsolares | Siya: try using qualify=yes in the sip.conf area for each phone, that should keep the connection going |
| 22:37.54 | *** join/#asterisk Bobocop (n=Bobocop@ip-89-174-122-119.multimo.gtsenergis.pl) |
| 22:39.16 | Siya | jsolares: thank you |
| 22:39.29 | Siya | do I need to restart my phones for that or only *? |
| 22:39.39 | *** part/#asterisk mwebb (n=chatzill@cpe-76-183-188-152.tx.res.rr.com) |
| 22:39.54 | jsolares | only * |
| 22:42.00 | Siya | :) |
| 22:42.00 | *** join/#asterisk kavit (n=kavit@ppp167-236-231.static.internode.on.net) |
| 22:42.42 | Rahail | can some one help me to add trunk |
| 22:43.03 | Rahail | ~trunk |
| 22:43.12 | jbot | i heard trunk is my trunk my trunk; my lovely asterisk trunk (check it out) |
| 22:44.25 | Bobocop | Hi! Does anyone know why I'm getting lots of repeated messages like "Stopping retransmission on '66c461eb3783df3833bf8f12753dea0e@192.168.20.1' of Request 102: Match Found"? |
| 22:44.36 | Rahail | lol |
| 22:44.42 | Bobocop | with 0 calls volume |
| 22:45.01 | *** join/#asterisk nays85 (i=nays85@shell.thehostbusters.com) |
| 22:45.42 | nays85 | any ex-voxee customers see the thread on dslreports? |
| 22:45.45 | Bobocop | Rahail what help do u need with trunk? |
| 22:45.57 | Rahail | I cant find way to setup trunk |
| 22:46.03 | Rahail | how dO I set it up |
| 22:46.18 | Rahail | everything I found was about freepbx trunk |
| 22:46.20 | Rahail | now asterisk |
| 22:46.21 | Rahail | :( |
| 22:47.01 | Rahail | 209.19.88.6 this the ip i want send call to |
| 22:47.13 | Bobocop | well it's just easier to do with freepbx installed - few clicks and u're up&running ;) |
| 22:47.31 | Rahail | do you know how to install it on fedora |
| 22:47.33 | Rahail | freepbx |
| 22:47.37 | ringhals | wow I have having crazy ammounts of dropped calls since my upgrade to 1.2.17 |
| 22:47.49 | Bobocop | Rahail, sure. w8 |
| 22:48.24 | Rahail | cool |
| 22:48.30 | Bobocop | Rahail: www.freepbx.org -> download |
| 22:48.41 | Rahail | i did it |
| 22:48.47 | Rahail | but didnt work an i delet |
| 22:49.16 | ringhals | I have callprogress=no and busydetect=no any suggestions? |
| 22:50.04 | Bobocop | Rahail: have you met requirements? |
| 22:50.17 | Bobocop | libxml installed etc? |
| 22:50.54 | Rahail | i have no idea |
| 22:51.51 | Bobocop | Rahail: here is the list: http://www.aussievoip.com/wiki/freePBX-2.1beta1Install |
| 22:52.17 | Bobocop | anyway, let's configure your trunk by hand |
| 22:52.28 | *** part/#asterisk FarrisG (n=lckirk@gateway.wiquest.com) |
| 22:52.33 | bkruse | ~russell |
| 22:52.43 | jbot | from memory, russell is a n00b |
| 22:52.43 | bkruse | ~russellb |
| 22:52.47 | jbot | russellb is Russell Bryant <russell@digium.com>, aka drumkilla, or 1337. He like... just totally f'n rocks you know! |
| 22:52.48 | Bobocop | so - u need to have trunk connected to some ip. is it SIP peer? or IAX? |
| 22:53.04 | bkruse | russellb: thats totally you |
| 22:53.25 | Rahail | sip |
| 22:53.49 | bkruse | ~russellb |
| 22:53.51 | jbot | i heard russellb is Russell Bryant <russell@digium.com>. He is also kind of a big deal, people know him. |
| 22:54.33 | Rahail | Bobocop its sip |
| 22:56.10 | file | moo? |
| 22:58.13 | Corydon-w | ~file |
| 22:58.42 | Bobocop | Rahail:open sip.conf then, add sip peer |
| 22:58.44 | Rahail | Bobocop |
| 22:59.06 | diclophis-work | is it possible to kill a SIP call from the CLI? |
| 22:59.06 | Bobocop | [name_of_ur_trunk] |
| 22:59.30 | Rahail | in sip.conf |
| 22:59.30 | file | jbot: file? |
| 22:59.33 | jbot | well, file is Joshua Colp, jcolp@digium.com, and is the best looking software developer at Digium |
| 22:59.33 | Rahail | ? |
| 22:59.36 | Bobocop | diclophis: use soft hangup |
| 22:59.47 | Corydon-w | ~file /etc/passwd |
| 23:00.08 | Corydon-w | ~file vim |
| 23:00.08 | Bobocop | diclophis-work: soft hangup <tab> |
| 23:00.10 | sandorp | I have a remote user using x-lite as a SIP client; his connections work on/off; the 2 extensions he's tried to use from 2 PC's both show UNREACHABLE; how do I figure out WHY he is unreachable? |
| 23:00.12 | diclophis-work | awesome |
| 23:00.12 | diclophis-work | thanks |
| 23:00.27 | Bobocop | Rahail: yes, in sip.conf |
| 23:00.36 | russellb | jbot: file is also a muffin eater |
| 23:00.39 | Bobocop | diclophis-work: welcome :) |
| 23:00.51 | Rahail | then |
| 23:01.16 | russellb | stupid bot ... |
| 23:01.24 | Bobocop | LOL |
| 23:01.32 | Corydon-w | russellb: type ~file whatever |
| 23:01.44 | russellb | yeah, he msg'd me |
| 23:01.57 | russellb | ~file asterisk |
| 23:02.10 | Bobocop | sandorp: use sip debug.... maybe thereis NAT involved? |
| 23:02.39 | Corydon-w | ~corydon |
| 23:02.43 | sandorp | NAT on the remote end? yes, I'm sure he's using NAT behind a single cable modem |
| 23:02.45 | Rahail | sorry |
| 23:02.53 | *** join/#asterisk Fieldy (i=tN8PAmm7@gentoo/contributor/Fieldy) |
| 23:02.57 | Bobocop | Rahail: add new section, start with [some_trunk_name] |
| 23:03.11 | Corydon-w | jbot: corydon? |
| 23:03.22 | russellb | jbot: bkruse |
| 23:03.24 | jbot | it has been said that bkruse is a bowling maniac |
| 23:03.31 | bkruse | hahaha |
| 23:03.45 | Bobocop | sandorp: have u added nat=yes in peer definition? |
| 23:03.57 | bkruse | so true! |
| 23:04.27 | russellb | jbot: corydon |
| 23:04.28 | jbot | corydon is probably one of the most l33t Asterisk developers around. He has been around longer than you have. |
| 23:04.29 | sandorp | hmm, I used freepbx to set up the system; where would that option be? |
| 23:04.38 | Corydon-w | Heh, nice |
| 23:04.46 | russellb | :) |
| 23:04.47 | Rahail | Bobocop id id that |
| 23:04.50 | Rahail | what else I have to do now |
| 23:07.09 | Bobocop | sandorp: in extensions |
| 23:07.18 | sandorp | sip_additional.conf has nat=yes |
| 23:07.26 | sandorp | for each extension |
| 23:07.52 | Bobocop | sandorp: it's an option in an extension definition, visible after creating the extension |
| 23:08.06 | Bobocop | sandorp: ok |
| 23:10.27 | sandorp | Bobocop: the extension works every now and then; once the call terminates the extension becomes unreachable; restarting asterisk sometimes fixes it, sometimes it doesn't |
| 23:11.27 | Bobocop | sandorp: does asterisk have public ip? or is it behind NAT too? |
| 23:12.06 | sandorp | public |
| 23:12.24 | sandorp | public and static, to be exact |
| 23:14.17 | Bobocop | well, maybe NAT gateway at xlite's site drops translations too fast. try adding qualify=yes to friend definition |
| 23:15.48 | sandorp | Bobocopy: qualify already = yes |
| 23:16.15 | Bobocop | ok, start sip debug then... |
| 23:16.41 | Bobocop | sip debug peername |
| 23:17.41 | sandorp | sorry, how do I put it into debug mode exactly? |
| 23:20.29 | *** join/#asterisk BBHoss (n=hoss@216.186.235.254) |
| 23:21.14 | BBHoss | anybody that has used voipstreet with AstNOW |
| 23:22.22 | BBHoss | can i still use iax.conf like the setup guide says |
| 23:24.05 | Bobocop | sandorp: open asterisk console, write: sip debug peer <peername> |
| 23:24.34 | *** join/#asterisk notoriousrab1982 (n=notoriou@207.47.34.74.static.nextweb.net) |
| 23:24.37 | Bobocop | sandorp: and try make a call |
| 23:24.59 | notoriousrab1982 | has anyone tried using centos5 as a distribution for asterisk? |
| 23:26.15 | Bobocop | I'm still looking for answer what are those "chan_sip.c: Stopping retransmission on '4bc5ac8646344d533f75a2c36937ac17@192.168.20.1'" messages are |
| 23:26.19 | Bobocop | anyone? |
| 23:27.06 | file | they are messages that a normal user would never need to see... they are debug messages |
| 23:29.57 | pfn | hmm, digium doesn't have any cards that would support a ds3, do they.... |
| 23:30.15 | _DAW | doesnt sangoma? |
| 23:31.40 | ringhals | notoriousr I have been told centos5 is a great choice |
| 23:31.56 | pfn | http://www.sangoma.com/datasheets/p_aft-et3-specs |
| 23:31.57 | Bobocop | file: thx, I see that those are visible only in debug mode. I'm troubleshouting voice quality issues, so I'm interested in any strange msgs ;) why so many of them, even there are no active calls? |
| 23:32.01 | pfn | that doesn't appear to work for voice |
| 23:32.22 | pfn | take tdm over ds3 and spit it out as sip |
| 23:33.19 | sandorp | Bobocop: of course both of the remote extensions work now that debug is turned on :\ |
| 23:34.12 | xkev | a lucent max tnt |
| 23:34.23 | xkev | cheap on baye |
| 23:34.33 | BBHoss | anybody that has used voipstreet with AstNOW here? |
| 23:34.34 | notoriousrab1982 | thanks ringhals - i am just wondering if anyone has successfully installed asterisk on centos5, having some problems - think it is because of centos5 |
| 23:35.35 | xkev | but the code won't likely handle pri signalling which delivers CNAM in a delayed facility ie |
| 23:35.35 | xkev | ..though I know someone who has code for it that does, apparently |
| 23:37.08 | sandorp | Bobocop: ok, 1 of the extensions is unreachable now, what am I looking for in the debug output? |
| 23:37.11 | Bobocop | sandorp: LOL as always ;) |
| 23:37.29 | sandorp | and is there a way to save the output to a file? |
| 23:37.55 | Bobocop | sandorp: look at headers - ip address of your friend |
| 23:38.32 | Bobocop | sandorp: u have to define logging to file in /etc/asterisk/logger.conf |
| 23:39.19 | Bobocop | can you ping from asterisk the pc with xlite installed? |
| 23:39.35 | sandorp | no, it is behind a firewall |
| 23:41.10 | sandorp | firewall is pingable |
| 23:42.26 | *** join/#asterisk Fieldy (i=ccYOgxQN@gentoo/contributor/Fieldy) |
| 23:44.36 | sandorp | Bobocop: looks like logging is set up OK; the headers right before I see where I got the error all start with SIP/2.0 OK ... no other error indicators |
| 23:45.22 | sandorp | however, I see a number of lines where it appears to try to read from the remote site but gets 0 lines |
| 23:46.10 | sandorp | <-- SIP read from 66.74.166.17:23070: |
| 23:46.10 | sandorp | --- (0 headers 1 lines) --- |
| 23:46.44 | Bobocop | sandorp: remind me plz, what kind of problem u have with calls from/to that xlite? |
| 23:47.01 | sandorp | they work sometimes, fail most of the time |
| 23:47.15 | Bobocop | sandorp: only with one direction? |
| 23:47.28 | sandorp | both directions |
| 23:47.35 | sandorp | he can't call me, I can't call him |
| 23:48.20 | Bobocop | sandorp: what kind of message you see on the console when u can't call xlite? |
| 23:48.22 | sandorp | I see in the debug output that the 2nd PC that is showing "OK" is able to send a series of headers after the "SIP read" message |
| 23:48.42 | sandorp | the broken one sends nothing |
| 23:49.45 | sandorp | all I see for the broken extension are the two lines I pasted above |
| 23:51.44 | *** join/#asterisk quidpro (n=quidpro@bas3-toronto12-1088945292.dsl.bell.ca) |
| 23:51.52 | Bobocop | sandorp: hmm... let me think it over :) |
| 23:52.22 | quidpro | Hmm.. anybody know of the "proper" way to upgrade Zaptel when using Sangoma/WanRouter drivers? I upgraded, now I get a Kernel Panic on boot. |
| 23:53.39 | tzanger | quidpro: which version of the wanpipe drivers are you running? |
| 23:54.26 | quidpro | One sec... |
| 23:54.38 | quidpro | I think 2.3.4.7 |
| 23:54.49 | sandorp | I see the following in the log file (but not on console): Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 23:55.01 | tzanger | that should work |
| 23:55.09 | tzanger | I'm running that with a TE407 as we speak |
| 23:55.14 | tzanger | (talking on it, in fact) |
| 23:55.15 | Bobocop | sandorp: I'd suspect that NAT device |
| 23:55.22 | quidpro | Tzanger: With Zaptel 1.2.17.1? |
| 23:55.23 | sandorp | and: db.c: Unable to find key '6004' in family 'CFU' |
| 23:55.33 | tzanger | no 1.4 |
| 23:55.43 | sandorp | 6004 = extention that does not work |
| 23:55.45 | quidpro | Different ballgame. :) |
| 23:56.02 | JT | tzanger: you run wanpipe on a digium card? |
| 23:56.08 | tzanger | nope |
| 23:56.21 | JT | hmm |
| 23:57.02 | Bobocop | sandorp: I'm not sure what that message is, but it's more like internal inventory handling, or stg like that :) |
| 23:57.19 | _VoiceMeUp_COM | quidproi would use a fresh zaptel download |
| 23:57.39 | _VoiceMeUp_COM | becasue wanpipe install has a new way.. to do stuff.. that if its sangoma and not dig |
| 23:57.45 | Bobocop | sandorp: the strangest thing is that the connection is established, then breaks, then establishes again |
| 23:58.08 | sandorp | Bobocop: yes, very strange |
| 23:58.16 | Bobocop | sandorp: maybe u could try to use any iax soft phone to see if it works flawlesly |
| 23:58.17 | quidpro | VoiceMeup: What do you mean by "fresh download"? This is from Digium. |
| 23:58.54 | sandorp | I was begninning to wonder if IAX might be more reliable ... I don't really understand the difference between the two protocols |