| 00:00.03 | dlynes | Yeah, so are we |
| 00:00.08 | [hC] | who are you working for? |
| 00:00.10 | dlynes | Having a hard time trying to find any in Vancouver |
| 00:00.12 | [hC] | northwest voip? :) |
| 00:00.14 | dlynes | 24/7 Communications |
| 00:00.45 | dlynes | We're a local interconnect company |
| 00:01.06 | dlynes | I think one fellow that used to be associated with us was trying to make some kind of pitch to Metrobridge at one point |
| 00:01.22 | [hC] | Ah... via PSD or something? |
| 00:01.36 | dlynes | Nah...there was a JV at that point called MXU Networks |
| 00:01.52 | dlynes | We dissolved that relationship, when we realized he was killing our business |
| 00:01.58 | [av]bani | Denmark: i have a snom 360 |
| 00:02.03 | [av]bani | so yes, you could say i've used it |
| 00:02.48 | [hC] | ah okay |
| 00:02.49 | Denmark | [av]bani : The snom is 168x? |
| 00:02.53 | [av]bani | no |
| 00:02.54 | [hC] | Maybe your company and my company should talk |
| 00:02.57 | dlynes | Well, actually...both of those clowns |
| 00:02.57 | [av]bani | i havent used pa168x |
| 00:03.00 | [hC] | we're focusing on SME clients |
| 00:03.08 | dlynes | Have you met Geoff Forrester or Michael Nugent? |
| 00:03.08 | [av]bani | i cant imagine the sound being much worse than gxp2000 |
| 00:03.09 | jeffgus | tdonahue-laptop, you mentioned you were using 8 channel banks earlier. what channel banks are you using? |
| 00:03.18 | [hC] | business termination only.. we sell hardware, configure it, phones, support, termination, provide internet, analog lines, the works |
| 00:03.26 | [hC] | all resold of course. |
| 00:03.34 | [hC] | geoff forrester sounds familiar, but nothing stands out in particular. |
| 00:03.44 | [hC] | tainted-: depends how many you want i guess. |
| 00:03.53 | dlynes | We're focussing on small and medium sized businesses, selling them PBXes, analog lines, voip lines, DSL, ... |
| 00:03.54 | tdonahue-laptop | jeffgus, technically it is only 2 channel banks, they are the 96 channel adtran channel banks |
| 00:04.01 | [hC] | tainted-: i normally dont *just* sell DID's, but i will, of course. probably $4-$5 CDN or so |
| 00:04.04 | Denmark | [av]bani : What makes you think its Open or Free? |
| 00:04.13 | [hC] | dlynes: so you're direct competition then. |
| 00:04.14 | [hC] | :) |
| 00:04.15 | tdonahue-laptop | it takes 4 T1's to feed that beast |
| 00:04.20 | dlynes | lol |
| 00:04.27 | tainted- | [hC] how much for origination / termination in 604 |
| 00:04.38 | jeffgus | tdonahue-laptop, ah, ok. i was looking at Rhino and the Adit 600 |
| 00:04.41 | Denmark | [av]bani : So far I have not found much on the website, though they seem to document the chip .. not on a technically deep level though.. |
| 00:04.45 | dlynes | Yeah, we're primarily focussing on our existing Nortel and Panasonic customers |
| 00:04.53 | [hC] | tainted-: im not up on prices, id have to look for you. |
| 00:04.58 | jeffgus | tdonahue-laptop, i heard the adrans had problems with AM radio stations sometimes |
| 00:05.00 | tainted- | i thought the GDP of Vancouver was 80% from the weed growers.. lol didn't know there were real businesses there |
| 00:05.01 | [hC] | dlynes: yeah, we're starting fresh. |
| 00:05.09 | [hC] | were you guys at techvibes? |
| 00:05.11 | [hC] | presenting? |
| 00:05.15 | jeffgus | tdonahue-laptop, and there is a AM radio station fairly close to this install |
| 00:05.15 | [hC] | the name rings a bell.. |
| 00:05.18 | dlynes | Geoff might have been |
| 00:05.22 | tdonahue-laptop | jeffgus, we haven't had any problems, these are the carrier grade ones though |
| 00:05.32 | dlynes | I wasn't, and Gurpreet I'm pretty sure wasn't |
| 00:05.47 | [hC] | nod.. |
| 00:06.06 | dlynes | Michael Nugent and Geoff Forrester are somehow associated with Talou Internet now |
| 00:06.07 | [hC] | i met a few people doig the same thing at techvibes there |
| 00:06.08 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 00:06.10 | tdonahue-laptop | jeffgus, i highly recommend the adtrans over rhinos for long distance runs, we have had all kinds of problems with the rhinos on long runs |
| 00:06.12 | jeffgus | tdonahue-laptop, so it's 96 lines and 4 cross over t1 lines into the pci boards? |
| 00:06.16 | *** join/#asterisk m_a_g_o (i=maxgluck@201.243.97.246) |
| 00:06.17 | exten123 | How to group sip extensions so that can dial by group like what ZAP got? |
| 00:06.17 | [hC] | but alot of them didnt have a fully developed platform, or a bad rep, or didnt handle everything |
| 00:06.28 | jeffgus | tdonahue-laptop, long runs being a 20-30 story building? |
| 00:06.45 | tdonahue-laptop | jeffgus, long runs being 900+ feet |
| 00:06.46 | dlynes | [hC]: Yeah, another fellow I know is starting out fresh, too |
| 00:06.50 | Denmark | [av]bani : (Not that I don't believe you .. I just havn't found it yet .. I would assume they would brag with it in order to sell more) |
| 00:06.52 | [hC] | whats his company called? |
| 00:06.55 | jeffgus | tdonahue-laptop, and what kind of problems where there? |
| 00:07.10 | dlynes | [hC]: He's going to be running a ITSP locally here for Level 3 customers |
| 00:07.19 | [hC] | Ahh |
| 00:07.20 | dlynes | [hC]: geoff forrester's? |
| 00:07.26 | tainted- | [hC] dlynes do u guys have web sites? |
| 00:07.28 | dlynes | [hC]: No idea...I don't talk to him anymore |
| 00:07.28 | [hC] | I wanted to start talks with level3' for termination. |
| 00:07.34 | [hC] | tainted-: www.voxter.ca |
| 00:07.44 | [hC] | its being developed still, and you wont find prices up there. |
| 00:07.44 | [hC] | :) |
| 00:07.44 | dlynes | tainted-: http://www.247communications.com/ |
| 00:07.51 | tainted- | [hC] wtf lol |
| 00:08.13 | dlynes | We don't have prices on ours either |
| 00:08.16 | tdonahue-laptop | jeffgus, phones not working, dtmf not being recognized, things that were never repeatable on demand |
| 00:08.16 | [hC] | tainted-: we are NOT marketing ourselves to resell just termination/origination |
| 00:08.20 | [hC] | we sell that as a package to our clients. |
| 00:08.24 | dlynes | But that's because we're an interconnect |
| 00:08.29 | dlynes | We do direct marketing |
| 00:09.15 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 00:09.20 | *** part/#asterisk franck (n=franck@tikiwiki/franck) |
| 00:09.24 | jeffgus | tdonahue-laptop, ah, okay. it seems the adit 600's are faily easy to purchase online. voipsupply sells them |
| 00:09.38 | jeffgus | tdonahue-laptop, i was having trouble finding pricing on the higher density solutions |
| 00:09.44 | dlynes | [hC]: the level 3 itsp thing is a joint venture with fairview wireless |
| 00:09.47 | jeffgus | the Axxium(SP?) for example |
| 00:10.14 | tdonahue-laptop | jeffgus, it was before my time, but i think we got them directly from adtran |
| 00:10.22 | dlynes | [hC]: part of the deal is that they become an official cisco channel partner for the new linksys pbx and associated hardware |
| 00:10.29 | tdonahue-laptop | jeffgus, but they do usually have a lead time on them |
| 00:10.31 | [hC] | dlynes: ahh. |
| 00:10.47 | [hC] | dlynes: we are reselling linksys and cisco phones, but not their pbx solutions |
| 00:10.48 | tainted- | "If you have an emergency, you may email 911 @ voxter.ca" |
| 00:10.55 | tainted- | sweet! lol |
| 00:11.14 | [hC] | "My toilet is clogged!" |
| 00:11.17 | [hC] | "Fire! Fire! Send help!" |
| 00:11.20 | tainted- | there is an alliga -- NO CARRIER |
| 00:11.24 | dlynes | [hC]: so how do those phones compare to the Aastra phones? |
| 00:11.33 | [hC] | We have a bunch of aastra phones |
| 00:11.35 | [hC] | and they are |
| 00:11.40 | [hC] | *drumroll* |
| 00:11.40 | [hC] | SHIT |
| 00:11.50 | [hC] | one of my partners seems attached to them for some reason |
| 00:11.54 | [hC] | but they are extremely buggy |
| 00:11.58 | dlynes | the linksys ones are better? |
| 00:12.00 | [hC] | maybe in a year when the firmware isnt so screwed up |
| 00:12.05 | [hC] | yeah, the 941/942 phones are much better |
| 00:12.06 | dlynes | i.e. spa841/spa941? |
| 00:12.08 | SwK | polycom > * |
| 00:12.13 | [hC] | 841 i dont bother with |
| 00:12.29 | SwK | 841's are about that same as a BT101 heh |
| 00:12.39 | [hC] | i mainly deploy polycom 501/601, cisco 7960/7970, linksys spa941/942, and linksys wip300 wifi phones if someone wants it |
| 00:12.39 | dlynes | ah....the sipura 3000, sipura 2000, and pap-2 all seem quite buggy, too |
| 00:12.47 | tainted- | aastra is shit? |
| 00:12.54 | dlynes | the bt100/bt101 sucks bad |
| 00:12.55 | [hC] | I use the pap2 sometimes.. home, etc. |
| 00:12.57 | dlynes | it looks like a toy, not a real phone |
| 00:13.01 | Denmark | dlynes : I have spa2k .. how is that buggy? |
| 00:13.07 | [hC] | and my spa2k is great. |
| 00:13.08 | tainted- | damn.. was gonna develop some XML browser apps for the 480i |
| 00:13.13 | SwK | aastra 9112i's seem buggy as hell when it comes to NAT |
| 00:13.27 | [hC] | tainted-: its buggy, the speakerphone sucks, audio quality is definitely sub par, |
| 00:13.31 | dlynes | Denmark: i often have calls where the sipura forgets its on the call, and just swaps it out for another call |
| 00:13.45 | [hC] | only in the last 3-4 revisions has the phone become able to retain a registration and function quasi normally' |
| 00:13.54 | SwK | i hate one that will register, but wont respond to a 407 when trying to send calls if the phone is natted (even tho the server is on a pub IP) |
| 00:14.01 | dlynes | Denmark: and other times where you end up getting a three way call for no reason |
| 00:14.02 | [hC] | dlynes: haha! really! ive never had that happen |
| 00:14.27 | tainted- | i'm primarily using polycom 301/501s and grandstream ATAs |
| 00:14.29 | [hC] | Im looking at putting a bounty on someone implementing polycom's non-limited BLF protocol |
| 00:14.31 | tainted- | what do u guys use for ATAs |
| 00:14.32 | dlynes | Denmark: and other times i get touch tones happening in the background |
| 00:14.38 | [hC] | this 7 line limit sucks. |
| 00:14.40 | Denmark | dlynes : Sounds like your phone is broken or something. |
| 00:14.45 | [hC] | dlynes: it sounds like you just have a bunk set up man. |
| 00:14.45 | SwK | dlynes: with a sipura SPA-2K series? |
| 00:14.58 | dlynes | Denmark: It's not a phone, it's an ata |
| 00:15.00 | [hC] | dlynes: ive never had any of htat, and ive been using sipura ata's (lots of them) for 2+ years |
| 00:15.14 | SwK | upgrade the firmware on that SPA |
| 00:15.19 | dlynes | [hC]: I've only had that crap happen at one customer |
| 00:15.26 | SwK | I use many man of them and never seen it |
| 00:15.31 | Denmark | Wallace78 : Try it ... you'll be amazed! |
| 00:15.44 | dlynes | [hC]: and another problem i've had is trying to do a *72/*73 on the spa3000's |
| 00:16.01 | dlynes | [hC]: it works on telus lines in vancouver, but not cloverdale |
| 00:16.09 | tainted- | do u guys offer directory services (411) |
| 00:16.14 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.170) |
| 00:16.22 | tainted- | i've been looking into that as a potential revenue stream |
| 00:16.49 | tainted- | it's hugely popular with SMBs here |
| 00:17.01 | Hmmhesays | almost beer time |
| 00:17.38 | tainted- | dlynes do u do SMATVs? |
| 00:17.45 | file[laptop] | Hmmhesays: almost? |
| 00:17.51 | Hmmhesays | yeap just about |
| 00:18.39 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:18.40 | dlynes | tainted-: no idea what smatv is |
| 00:18.47 | SwK | its past beer time |
| 00:18.47 | PakiPenguin_ | anyone played with astribank here? |
| 00:18.59 | SwK | astribank? |
| 00:18.59 | Hmmhesays | i'm going to dance with some hotties tonight and drink some beer |
| 00:19.02 | SwK | that the usb thing? |
| 00:19.35 | Ariel_ | Hmmhesays, have fun |
| 00:19.43 | Hmmhesays | I shall |
| 00:19.45 | [hC] | dlynes: ah, i do call forwarding on the pbx, not the ata. |
| 00:19.46 | PakiPenguin_ | yeah |
| 00:19.47 | Hmmhesays | it will be wonderful |
| 00:19.56 | [hC] | im 2 beers down |
| 00:19.56 | PakiPenguin_ | SwK, having issues with its install :( |
| 00:20.25 | tainted- | PakiPenguin_ i thought it was built specifically for asterisk |
| 00:20.26 | *** join/#asterisk Jon335_ (n=Jon335@unaffiliated/jon335) |
| 00:20.34 | *** part/#asterisk dasenjo (n=dasenjo@208.195.215.170) |
| 00:20.49 | PakiPenguin_ | tainted-, it needs some firmware to be loaded , which i cannot get it to load :( |
| 00:21.21 | Ariel_ | PakiPenguin_, do you have a link for this astribank |
| 00:21.22 | tainted- | did u try calling Xorcom |
| 00:21.29 | PakiPenguin_ | dont have their number |
| 00:21.31 | tainted- | Ariel_ http://www.xorcom.com/astribank/features.html |
| 00:21.40 | PakiPenguin_ | http://www.xorcom.com/drivers/astribank/Astribank_8.pdf |
| 00:21.59 | tainted- | PakiPenguin_ how much did u pay for it? |
| 00:22.14 | tainted- | the two relay ports look interesting |
| 00:22.23 | PakiPenguin_ | got it from www.digitnetworks.com |
| 00:22.34 | Denmark | dlynes : I know that spa2k is ata .. but you connect a phone. |
| 00:22.50 | Denmark | dlynes : I suspect your PSTN phone is broken.. |
| 00:23.06 | *** part/#asterisk Jon335_ (n=Jon335@unaffiliated/jon335) |
| 00:24.03 | tainted- | i just searched for 'xorcom' and digitnetworks.com coughed up a huge sql query |
| 00:24.05 | tainted- | select count(distinct p.products_id) as total from products p left join manufacturers m using(manufacturers_id), products_description pd left join specials s on p.products_id = s.products_id, categories c, products_to_categories p2c where p.products_status = '1' and p.products_id = pd.products_id and pd.language_id = '1' and p.products_id = p2c.products_id and p2c.categories_id = c.categories_id and ((pd.products_name like '%xor |
| 00:24.18 | tainted- | sql injection for free products anyone? |
| 00:24.37 | Hmmhesays | nice |
| 00:24.54 | Hmmhesays | its got a hairball, how cute |
| 00:25.06 | Denmark | eh? |
| 00:25.53 | Ariel_ | tainted-, thank you for the link |
| 00:26.07 | dlynes | Denmark: Not using the analog phone jack |
| 00:26.40 | dlynes | [hC]: We weren't doing call forwarding on the ata...we were doing it through the ata on the analog line |
| 00:26.44 | Denmark | dlynes : The spa2k has 3 connectors, 1 ethernet, and 2 fsx ports, right? |
| 00:27.02 | tainted- | PakiPenguin_ did u add the xorcom zaptel drivers to asterisk? |
| 00:27.13 | dlynes | Denmark: Oh yeah...i was thinking of the spa3k |
| 00:27.16 | PakiPenguin_ | i did tainted- |
| 00:27.24 | PakiPenguin_ | its included by default in 1.5 ( zaptel ) |
| 00:27.27 | tainted- | why are u upgrading firmware? |
| 00:27.28 | dlynes | Denmark: on the spa2k, we have those plugged directly into a KSU, not phones |
| 00:27.29 | Denmark | dlynes : Oh ok .. I don't know that one. |
| 00:27.42 | Denmark | ~ksu |
| 00:28.02 | dlynes | Denmark: ksu=key system unit (low end traditional pbx) |
| 00:28.07 | [hC] | dlynes: yeah.. |
| 00:28.15 | [hC] | dlynes: cloverdale is screwed, though.. hehe |
| 00:28.22 | PakiPenguin_ | tainted-, i am not |
| 00:28.30 | dlynes | [hC]: that whole customer was completely fubared |
| 00:28.38 | PakiPenguin_ | its supposed to take some firmware at the start ( when i plug it in ) |
| 00:28.38 | tainted- | PakiPenguin_ what is wrong wit hit? |
| 00:28.47 | PakiPenguin_ | i cant get zaptel to recognize it |
| 00:29.00 | dlynes | [hC]: they had super strong emf fields in the phone room, so it caused permanent damage to the computer, and major havok with the network cables |
| 00:29.15 | dlynes | [hC]: we ended up having to pull asterisk out of there, and throw in a Panasonic KSU |
| 00:29.35 | [hC] | doh :| |
| 00:29.40 | [hC] | that would be fun to debug |
| 00:29.45 | [hC] | why everything just 'breaks' randomly :) |
| 00:30.01 | *** join/#asterisk tdonahue-laptop (n=tdonahue@70.57.38.163) |
| 00:30.02 | tainted- | what do u guys do for alarms & gates |
| 00:30.09 | dlynes | [hC]: Yeah...like it would be working when we left, and then four hours later would mysteriously stop working |
| 00:30.31 | dlynes | tainted-: you could try using the X10 support in the Linux kernel for that |
| 00:30.32 | PakiPenguin_ | nothing :) |
| 00:30.39 | tainted- | PakiPenguin_ are u sure it's not USB issues? |
| 00:30.39 | dlynes | tainted-: and using an AGI script to control it |
| 00:30.40 | [hC] | tainted-: analog lines |
| 00:30.54 | [hC] | for alarms anyways |
| 00:31.09 | dlynes | tainted-: or use a entry pad system that includes supports for gates |
| 00:31.11 | tainted- | some gates require a lot of juice |
| 00:31.11 | PakiPenguin_ | i can see it connect and show up in messages |
| 00:31.54 | tainted- | dlynes [hC] what about intercoms etc |
| 00:32.09 | tainted- | PakiPenguin_ dunno.. did u try e-mailing xorcom? |
| 00:32.14 | dlynes | tainted-: asterisk has support for intercoms already |
| 00:32.19 | [hC] | yeah. |
| 00:32.19 | [hC] | heh |
| 00:32.22 | dlynes | tainted-: you use your soundcard to do it |
| 00:32.26 | tainted- | i mean in terms of hardware |
| 00:32.27 | *** join/#asterisk bkw_ (n=brian@c-68-32-112-142.hsd1.md.comcast.net) |
| 00:32.28 | [hC] | i implemented intercoms myself in asterisk |
| 00:32.33 | [hC] | on cisco or polycom |
| 00:32.36 | PakiPenguin_ | i am doing that at the moment |
| 00:32.43 | [hC] | or you can get ip enabled intercom buttons etc |
| 00:32.55 | *** join/#asterisk martianlobster (n=clarks@m815f36d0.tmodns.net) |
| 00:33.10 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.236) |
| 00:33.15 | tainted- | how about overhead speakers |
| 00:33.28 | dlynes | tainted-: like i said...soundcard |
| 00:33.31 | dlynes | tainted-: it's called paging |
| 00:33.43 | martianlobster | i want to set up an asterisk server to talk to viatalk. Can I do it from behind a router or will gnat mess up the protocall? |
| 00:34.08 | dlynes | tainted-: you typically set it up to use a horn |
| 00:34.14 | martianlobster | btw, what protocall is used to encode the packets for viatalk and astersirsk? is it SIP? |
| 00:34.24 | tainted- | soundcard on the asterisk box or in another box w/ a softclient |
| 00:34.35 | dlynes | tainted-: soundcard on the asterisk box |
| 00:34.40 | tainted- | oh wow |
| 00:35.16 | dlynes | tainted-: cheaper than dealing with pa amplifiers and ducking modules |
| 00:35.55 | dlynes | tainted-: you can also use your soundcard for streaming music on hold |
| 00:36.19 | tainted- | http://www.voip-info.org/wiki/view/Setting+up+paging+with+a+sound+card |
| 00:36.21 | tainted- | that? |
| 00:36.28 | dlynes | probably |
| 00:37.06 | Denmark | buying a phone with GPL firmware would rock. |
| 00:37.13 | *** join/#asterisk suge (i=gd@is.krazy.us) |
| 00:38.39 | Denmark | suge: Your nick is suck in danish. |
| 00:39.19 | [hC] | ok im gonna go home now i think. |
| 00:39.26 | [hC] | and sort out this drinking thing. |
| 00:39.29 | dlynes | yeah...good day for tennis :) |
| 00:39.38 | Denmark | [hC] : Ok, think about it! |
| 00:49.44 | *** join/#asterisk PBXtech (i=nik@156.sub-70-213-237.myvzw.com) |
| 00:52.17 | Darwin35 | what is asterisk where do I get it what does it run on adn how much is it ? |
| 00:53.24 | Ariel_ | Darwin35, you should know better then to ask that one... |
| 00:53.29 | *** part/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 00:54.03 | PBXtech | anyone know of a SIP provider who also sells TDM that can be resold? |
| 00:54.34 | Ariel_ | PBXtech, try race.com |
| 00:55.52 | *** join/#asterisk bkw_ (n=brian@c-68-32-112-142.hsd1.md.comcast.net) |
| 00:56.13 | *** join/#asterisk monthos (n=monthos@pewp.net) |
| 00:56.57 | PBXtech | they look small |
| 00:57.40 | monthos | Anyone wanna give me a hint on whats wrong? My asterisk is on my gateway, so its not technically behind my nat. im using sip and registering to another asterisk on the internet, my voip phone is a standard phone attached to a linksys voice router, behind my firewall |
| 00:57.55 | monthos | the phone can make and recieve calls. but it cannot hear |
| 00:58.11 | monthos | it does however, send its audio fine |
| 00:58.24 | Ariel_ | PBXtech, they are but they can do allot if you give them a call |
| 00:58.25 | Abydos313 | try changing codecs |
| 00:58.43 | *** join/#asterisk Laggy_McGee (n=jchadwic@pool-71-245-122-77.cmdnnj.fios.verizon.net) |
| 00:58.46 | Abydos313 | mine does exactly that with telasip and g729.. 711 works perfectly |
| 00:59.00 | monthos | alright, ill try it |
| 00:59.10 | Laggy_McGee | asterboy, X-Rob: you there? |
| 00:59.31 | Darwin35 | make sure you have 10000 -20000 udp open also |
| 00:59.50 | Darwin35 | and that your vibrator is attached to the serial port |
| 00:59.53 | monthos | lol |
| 00:59.57 | op3r | I alwats get G.729 lack of license errors |
| 01:00.10 | Ariel_ | PBXtech, there allot bigger then voipjet which allot of people use. |
| 01:00.22 | Darwin35 | you have to buy a license from digium and use the reg tool they provide |
| 01:01.11 | op3r | Darwin35: I got like 12 channel license |
| 01:01.12 | op3r | :( |
| 01:01.26 | Darwin35 | did you use the regtool |
| 01:01.49 | op3r | yep |
| 01:02.17 | op3r | but like 5 people on a vicidial and 8 people using softphones used it it crops up |
| 01:02.19 | Ariel_ | op3r, then call digium and get support that you paid for when you got there product... |
| 01:02.37 | Ariel_ | op3r, that is more then 12 |
| 01:02.56 | Laggy_McGee | lol |
| 01:03.11 | op3r | I know but the softphones I think have already built in G.729 |
| 01:03.27 | Ariel_ | op3r, yes but it's your channels that counts |
| 01:03.36 | Darwin35 | but the asterisk box has to have them also |
| 01:03.40 | PBXtech | [Ariel_]: do they sell TDM as well? resell global or something? |
| 01:04.04 | Ariel_ | PBXtech, the do have tdm setups as well as voip |
| 01:04.24 | PBXtech | hmm cool i will email them.. thx |
| 01:04.47 | *** join/#asterisk trbldwine (i=trbldwin@71.194.161.170) |
| 01:05.03 | Ariel_ | PBXtech, ask for carlos and tell them I sent you to them.... |
| 01:05.12 | Laggy_McGee | I have my ast server in a DMZ and when I try to connect to it, I get wicked "echo" - more like a CD skipping |
| 01:05.31 | Laggy_McGee | I brought it back behind the firewall and it works fine |
| 01:06.24 | PBXtech | they will know you as Ariel? |
| 01:06.28 | Laggy_McGee | what could call this? |
| 01:06.52 | Laggy_McGee | /call/cause |
| 01:07.07 | Ariel_ | PBXtech, yes |
| 01:07.15 | PBXtech | how do they compare to say sipstorm? |
| 01:07.20 | Ariel_ | Laggy_McGee, firewall not configured correctly |
| 01:07.30 | Ariel_ | PBXtech, don't know sipstorm |
| 01:07.38 | *** join/#asterisk ahattar (n=user@ool-43551487.dyn.optonline.net) |
| 01:07.44 | ahattar | hi all, |
| 01:07.53 | Laggy_McGee | Ariel_: plain old MASQ... every else works fine |
| 01:08.37 | Ariel_ | Laggy_McGee, even if it's behind a fw on the dmz does not mean ports are properly forwarded |
| 01:08.37 | Laggy_McGee | s/every/everyTHING/.... arg |
| 01:08.56 | ahattar | does anyone have experiance with vbuzzer to connect to asterisk? |
| 01:09.08 | Ariel_ | vbuzzer? |
| 01:09.08 | Darwin35 | that g729 /me fires all firewalls |
| 01:09.12 | tainted- | ahattar what's vbuzzer |
| 01:09.21 | Laggy_McGee | Ariel_: How so? it's a no-frills MASQing |
| 01:09.36 | monthos | Abydos313: yes it was a codec issue. thanks a million |
| 01:10.04 | Ariel_ | Laggy_McGee, what fw is it. |
| 01:10.12 | ahattar | tainted: www.vbuzzer.com (free did # area cocde 416) |
| 01:10.17 | Laggy_McGee | OpenWRT |
| 01:10.23 | Ariel_ | ahh |
| 01:10.59 | Laggy_McGee | Yeah... Literally no-frills MASQing. :) |
| 01:11.13 | Laggy_McGee | I know because I can see the chains! |
| 01:11.41 | Ariel_ | you are still forwarding ports correct? DMZ just means ports are open but that does not mean there forwarded to that port. |
| 01:11.48 | *** part/#asterisk ahattar (n=user@ool-43551487.dyn.optonline.net) |
| 01:12.41 | Laggy_McGee | I have two routers |
| 01:12.44 | Ariel_ | I use a linksys WRT54GS here and I don't have the asterisk box on the dmz. I just forward the registration ports I need. |
| 01:13.07 | Laggy_McGee | Internet > Router 1 > (DMZ) > Router 2 > LAN |
| 01:13.28 | Laggy_McGee | I have a literal DMZ, not a "fake" one |
| 01:13.39 | Ariel_ | double nat is not good for asterisk and sip |
| 01:14.06 | Laggy_McGee | I'll cross that bridge when I come to it... :) For now I'm only NATing once. |
| 01:14.20 | Laggy_McGee | LAN > Router 2 > DMZ |
| 01:14.28 | Laggy_McGee | Just running the ext 1000 test |
| 01:14.29 | Ariel_ | did you setup extenip and localnet settings on sip.conf |
| 01:14.41 | Laggy_McGee | No. All default settings |
| 01:15.02 | Laggy_McGee | just set up one extension in sip.conf |
| 01:15.04 | Ariel_ | well I guess you need to start there |
| 01:15.40 | Ariel_ | externip=realworld IP to your router, localnet=192.168.1.XXX/255.255.255.0 |
| 01:16.13 | Laggy_McGee | DMZ = 192.168.0.0/255.255.255.0; LAN = 192.168.1.0/255.255.0.0 |
| 01:16.18 | Ariel_ | after you set this up just need to reload sip |
| 01:16.23 | *** join/#asterisk P4C0 (n=pakw@200.124.22.34) |
| 01:16.40 | Ariel_ | Laggy_McGee, that is not going to work |
| 01:16.49 | Laggy_McGee | So, if the asterisk server is in the DMZ, localnet is 192.168.0.0/255.255.0.0? |
| 01:16.50 | Laggy_McGee | Why not? |
| 01:17.02 | Ariel_ | you need actuall external IP |
| 01:17.08 | Ariel_ | not one natted |
| 01:17.23 | Ariel_ | dmz makes no difference |
| 01:17.36 | Laggy_McGee | Router 1 is forwarding to the asterisk server (toast) |
| 01:17.45 | Laggy_McGee | forwarding 1194, that is |
| 01:18.17 | Ariel_ | when sip is sent from one box to a device it needs to know the proper routing via the internet |
| 01:18.27 | Laggy_McGee | er..... no.... now I'm confusing the OpenVPN ports. :) |
| 01:18.28 | Ariel_ | it's not a dmz issue but a nat address issue |
| 01:18.48 | P4C0 | nat help? I have asterisk server behind a firewall/nat, and my sip provider is outside and I don't use registry to register into it (it have ip based auth), it works fine but sometimes asterisk don't realize the person pick up the phone (when making calls to provider)... |
| 01:19.33 | P4C0 | Ariel_ the external ip is in global sip.conf file right? or inside the peer? |
| 01:19.35 | *** join/#asterisk mogorman (n=mogorman@68.62.237.103) |
| 01:19.49 | Ariel_ | global |
| 01:19.59 | Laggy_McGee | Ariel_: box? device? |
| 01:20.09 | Ariel_ | you set this up in the general section |
| 01:20.18 | P4C0 | strange think is that all of this happends when I only allow codec ulaw... |
| 01:20.47 | Ariel_ | hummm works fine with other codec's |
| 01:21.39 | P4C0 | Ariel_ yup |
| 01:21.50 | *** join/#asterisk Darwin35 (n=Darwin@c-24-9-75-234.hsd1.co.comcast.net) |
| 01:22.06 | P4C0 | no ulaw, alaw sorry |
| 01:22.38 | Ariel_ | does your provider support alaw? or are they in the states and support ulaw |
| 01:22.42 | Laggy_McGee | Ariel_: I don't have a static IP - can I use a hostname? |
| 01:23.10 | Ariel_ | Yes I use in mine externip=kasipbx.homedns.org |
| 01:23.13 | Laggy_McGee | Ariel_: Actually, no - we're not even up to that point yet. I'm not even talking to the outside world |
| 01:23.30 | Laggy_McGee | I'm just talking between the LAN and DMZ right now |
| 01:23.37 | Ariel_ | Laggy_McGee, yes but your sip devices is behind the other router |
| 01:24.45 | P4C0 | Ariel_: it says G711, when I tried with ulaw it fails, so I put it with alaw, works... so I change that to all my inside phones... (I think all go better if I use the same codec right?) |
| 01:25.07 | PakiPenguin_ | nothing :) |
| 01:25.07 | PakiPenguin_ | \ |
| 01:25.07 | Ariel_ | P4C0, normally yes |
| 01:25.08 | PakiPenguin_ | \".;l;klj ";' |
| 01:25.38 | P4C0 | Ariel_: I just found that that in the list of priority (for phones) I have alaw in the last... chaged it... but can this be the problem? |
| 01:25.46 | *** join/#asterisk gammacoder (n=chatzill@cpe-65-26-178-240.indy.res.rr.com) |
| 01:26.25 | Ariel_ | P4C0, it should not be. Alaw just take more bw 80k per call. It's less taxing since no compression needed so it should sound better |
| 01:26.48 | Laggy_McGee | Ariel_: My network is in extreme "setup mode" |
| 01:27.09 | P4C0 | Ariel_: in my firwall I am forwarding all udp packages (from provider) to the local ip of the asterisk server... |
| 01:27.16 | Laggy_McGee | Everything is open now... getting everything to work, and then slowing closing everything up |
| 01:28.04 | Laggy_McGee | Ariel_: And I'm taking baby steps with this asterisk install, hence I'm only trying to hit the 1000 test extension from my LAN to my DMZ |
| 01:28.14 | Laggy_McGee | once I get that working, I'll try hitting the outside world |
| 01:28.14 | Ariel_ | P4C0, sip you only really need 5060/61 and rtp ports 10,000 to 20,000 or like I do edit my rtp.conf and just put 10,001-11,000 I use webmin... |
| 01:29.19 | znoG | justinu|laptop: i managed to build the RJ21, the block 110 and got the TDM2400B up :) |
| 01:29.28 | Ariel_ | Laggy_McGee, I belive in the KISS setup. And two routers for my own view is a problem. I would start simple then go from there. |
| 01:29.30 | znoG | justinu|laptop: man it took a while to build the RJ21 and the block 110 |
| 01:29.38 | Ariel_ | znoG, great |
| 01:29.53 | P4C0 | Ariel_ didn't know the edit in rtp.conf, thanks |
| 01:29.55 | znoG | ah Ariel_ too of course :) |
| 01:30.28 | Laggy_McGee | Ariel_: Well, I did that... like I said - it works when everything's on the same subnet/router |
| 01:31.02 | Laggy_McGee | so now I want to set it up "correctly" |
| 01:31.50 | Ariel_ | Laggy_McGee, yes I understand. But in your setup you need to consider the asterisk box as being outside your network. Since your other devices are behind another router. |
| 01:32.11 | justinu|laptop | znoG: cool, congrats |
| 01:32.15 | Ariel_ | Nat issues are a real problem with asterisk. That is why people use a sip proxy in some of there setups |
| 01:32.19 | justinu|laptop | znoG: what country you in? |
| 01:32.23 | Laggy_McGee | Ariel_: shouldn't port forwarding (as above) rectify that? |
| 01:32.32 | Laggy_McGee | Is that what I really want - a SIP proxy? |
| 01:32.35 | Darwin35 | where can I purchase a kram |
| 01:32.43 | Darwin35 | I need a kram for my box |
| 01:32.44 | Laggy_McGee | The asterisk server in my LAN and the SIP proxy in my DMZ? |
| 01:32.51 | Ariel_ | Laggy_McGee, no |
| 01:33.43 | x86 | anyone get asterisk-oh323 module to compile correctly? |
| 01:33.55 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:33.56 | Darwin35 | ewww h323 |
| 01:34.02 | x86 | yeah i know |
| 01:34.03 | Darwin35 | banish him |
| 01:34.11 | x86 | one of my providers only supports h323 :( |
| 01:34.16 | Laggy_McGee | Ariel_: Is there a name for the issue I'm describing? "Horrible, stuttering echo"? |
| 01:34.20 | x86 | until next week, and then they will do SIP also |
| 01:34.40 | Ariel_ | yes lag and addressing miss match |
| 01:36.02 | Laggy_McGee | Ok. And, other than multiple NATing, what else can cause it? |
| 01:36.30 | Ariel_ | bw |
| 01:37.29 | *** join/#asterisk Derkommissar (n=Alberto@adsl-153-47-91.mia.bellsouth.net) |
| 01:37.59 | P4C0 | the cool think is that sometimes it works sometimes it don't :p hehe |
| 01:38.23 | P4C0 | s/think/thing |
| 01:38.24 | Laggy_McGee | Ariel_: what I'm having a problem with is the fact that I can connect just fine from my LAN, through both routers, to, say CallCentric, but I get this problem when I'm only going through one router to my DMZ |
| 01:38.30 | P4C0 | I'll go now, thanks Ariel_ |
| 01:39.31 | Ariel_ | Laggy_McGee, I still think it's a router issue then. |
| 01:39.40 | Laggy_McGee | Ariel_: how can lag not be a (noticable) issue when going through 2 routers, but become horrendous when going through only one? |
| 01:40.52 | Ariel_ | one is not configured correctly |
| 01:41.03 | Ariel_ | I have issues with linksys routers before. |
| 01:41.08 | Laggy_McGee | well that's pretty likely |
| 01:41.27 | Laggy_McGee | What types of things should I be looking for? |
| 01:43.06 | Laggy_McGee | Ariel_: this is the dump of my iptables for the router in question: http://pastebin.com/647213 |
| 01:46.00 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-92.apple.com) |
| 01:50.28 | x86 | MacDome: heya |
| 01:52.45 | Ariel_ | Laggy_McGee, humm I don't follow part of it. But seems like your closing allot |
| 01:53.27 | *** join/#asterisk mattwj2005 (n=Matt@user-12l3lqm.cable.mindspring.com) |
| 01:54.37 | Laggy_McGee | Ariel_: so if I try to remove the REJECTs and DROPs and see if it works...? |
| 01:54.49 | Ariel_ | maybe |
| 01:54.53 | mattwj2005 | what do you need to do to enable blind transfers? |
| 01:54.55 | Ariel_ | I am not that good with iptables |
| 01:55.00 | key2 | !seen kram |
| 01:55.06 | Laggy_McGee | Ariel_: no problem... I'll try |
| 01:55.25 | Ariel_ | mattwj2005, humm works fine on my polycom without any thing added |
| 01:55.56 | mattwj2005 | what I am trying to do is enable it on a stand phone by pressing #1 |
| 01:56.49 | Ariel_ | features.conf |
| 01:56.53 | cybertheq | ~~~ hello ~~~ are there any developers present? |
| 01:56.56 | jbot | cybertheq: okay |
| 01:57.15 | mattwj2005 | yeah I uncommented that line |
| 01:57.23 | mattwj2005 | do I need to do anything else? |
| 01:58.50 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 01:59.54 | *** join/#asterisk bkw_ (n=brian@c-68-32-112-142.hsd1.md.comcast.net) |
| 02:00.20 | Ariel_ | restart asterisk |
| 02:00.28 | mattwj2005 | is it possible I have my rfc setting wrong on my sip device? |
| 02:00.58 | cybertheq | hello, are there any developers present? |
| 02:01.18 | *** join/#asterisk vopi (n=kkk@202.139.197.105) |
| 02:01.20 | Ariel_ | mattwj2005, yes anything in asterisk is possible |
| 02:01.32 | Ariel_ | cybertheq, they might also be at asterisk-dev |
| 02:01.48 | vopi | hi alls |
| 02:01.53 | cybertheq | is that #asterisk-dev? |
| 02:02.12 | Ariel_ | cybertheq, well there was one. |
| 02:02.42 | cybertheq | I'll try |
| 02:07.29 | *** join/#asterisk froguz (n=froguz@83-136-222-201.adsl.terra.cl) |
| 02:16.47 | mattwj2005 | it doesn't appear to be it |
| 02:24.08 | *** join/#asterisk PBXtech (i=nik@236.sub-70-213-238.myvzw.com) |
| 02:24.30 | Darwin35 | gayasterisk the queer phoone system |
| 02:24.37 | Darwin35 | lol |
| 02:24.48 | PBXtech | is there a good/commerical call recording app for asterisk? |
| 02:24.59 | Darwin35 | homotelephonis |
| 02:27.39 | mogorman | whats wrong with app_monitor? |
| 02:27.45 | mogorman | and Darwin35 wtf? |
| 02:28.14 | Darwin35 | gaaydialtonis |
| 02:28.19 | Darwin35 | lol |
| 02:28.34 | mogorman | seriously your about to be kick banned |
| 02:28.43 | Darwin35 | just raising hell |
| 02:28.56 | Darwin35 | why I can joke I am family |
| 02:29.25 | PBXtech | he went to astricon |
| 02:29.29 | mogorman | and? |
| 02:30.26 | Darwin35 | <== is far from a homophobe I am family |
| 02:30.56 | mogorman | umm sure |
| 02:31.03 | Sedorox | lol |
| 02:31.39 | Darwin35 | mogorman ask kram he knows |
| 02:31.53 | Sedorox | Darwin35's been here as long as I can remember (about a year) :p |
| 02:32.02 | mogorman | i know Darwin35 has been here for a long time |
| 02:32.12 | mogorman | we fixed g729 for him and other bsd heads long ago |
| 02:32.26 | justinu|laptop | heh |
| 02:32.34 | Sedorox | I'm sure he doesn't have a problem with homosexuals... |
| 02:32.35 | Darwin35 | I was the one who orignaly ports asterisk to bsd |
| 02:32.36 | *** join/#asterisk {zombie} (i=zombie@soulasylum.penguincare.com.au) |
| 02:32.39 | Sedorox | maybe he is one? :p |
| 02:32.48 | mogorman | yeah thats what you have said Darwin35 |
| 02:33.08 | Darwin35 | ask Manx and bkw and anthm |
| 02:33.20 | mogorman | im not saying you didnt |
| 02:33.27 | Darwin35 | I use to get yelled at all the tiime when I was porting |
| 02:33.51 | xachen | ?? |
| 02:34.11 | Sedorox | ahahah |
| 02:34.32 | xachen | BSD only!! |
| 02:34.50 | Sedorox | bsd I use for anything involving network traffic... linux I use for my workstations |
| 02:36.00 | Darwin35 | bsd +sangoma = stable voip proxy |
| 02:36.30 | mogorman | groovvy Darwin35 |
| 02:37.01 | asterboy | brokeback asterisk? |
| 02:37.33 | Darwin35 | lol |
| 02:38.08 | Darwin35 | or bareback asterisk |
| 02:38.15 | *** join/#asterisk starlein (i=star@fo0bar.de) |
| 02:38.41 | asterboy | It can be a new fork |
| 02:39.00 | mogorman | yay yet another asaterisk fork |
| 02:39.02 | mogorman | yaaf |
| 02:39.30 | asterboy | linksys is making its own version |
| 02:39.36 | Abydos313 | did you see the movie |
| 02:39.40 | mogorman | que es? |
| 02:39.57 | asterboy | who is? |
| 02:40.10 | asterboy | what is... |
| 02:40.28 | mogorman | linksys is making is its own version??? |
| 02:40.33 | asterboy | ...the sound of one * zapping |
| 02:40.56 | asterboy | yes they are and expect it to be adopted as the NEW asterisk |
| 02:41.05 | *** join/#asterisk bkw_ (n=brian@c-68-32-112-142.hsd1.md.comcast.net) |
| 02:41.24 | Sedorox | asterboy: you sure you don't mean spoon and not fork :p |
| 02:41.28 | mogorman | you have any proof of this? |
| 02:41.34 | mogorman | or just conjecture? |
| 02:41.56 | Sedorox | well they do have that one package they sell... I figured it was asterisk based |
| 02:42.12 | asterboy | ewww...no spoons |
| 02:42.28 | mogorman | if it was we should figure it out |
| 02:42.29 | Sedorox | lol |
| 02:42.31 | mogorman | as i dont see binaries |
| 02:42.34 | mogorman | err source |
| 02:42.46 | mogorman | binaries with no source is a no no .... |
| 02:42.50 | *** part/#asterisk bkw_ (n=brian@c-68-32-112-142.hsd1.md.comcast.net) |
| 02:43.06 | Darwin35 | Cisco forks everything and that wich they cant for or copy the buy up |
| 02:43.20 | mogorman | show me proof |
| 02:43.24 | mogorman | or is it conjecture |
| 02:43.37 | Darwin35 | cisco bought linksys |
| 02:43.41 | asterboy | http://voxilla.com/name-News-article-sid-173.html |
| 02:43.49 | mogorman | show me linksys/cisco asterisk |
| 02:43.50 | asterboy | Read the 3rd paragraph from the bottom |
| 02:44.08 | Darwin35 | look at the new linksys device |
| 02:44.15 | Darwin35 | its asterisk based |
| 02:44.18 | asterboy | http://www.linksys.com/servlet/Satellite?c=L_Product_C2&childpagename=US%2FLayout&cid=1139414817110&pagename=Linksys%2FCommon%2FVisitorWrapper |
| 02:44.37 | asterboy | linksys wants * for their own |
| 02:45.06 | mogorman | where is the source or the changes mentioned? |
| 02:45.42 | tainted- | anyone use DLINK atas/voip routers? |
| 02:45.49 | mogorman | its not mentioned in their datasheet |
| 02:45.50 | mogorman | or the site |
| 02:47.05 | Qwell | sounds like FUD to me |
| 02:47.22 | mogorman | of course they are looking at it |
| 02:47.30 | mogorman | but i havent seen anything from them yet |
| 02:47.44 | Qwell | mogorman: I had a little chat with a cisco guy at VON |
| 02:47.58 | mogorman | i mean ive been told cisco went so far as to make an asterisk box |
| 02:48.01 | mogorman | but i never saw it |
| 02:48.07 | mogorman | or ever heard about it again |
| 02:48.18 | mogorman | others were gonna make boxes too |
| 02:48.22 | mogorman | but havent seen em |
| 02:48.39 | Darwin35 | snom did |
| 02:48.48 | mogorman | snom has an asterisk box? |
| 02:48.52 | mogorman | i thought they had a snom box |
| 02:48.56 | mogorman | that did snom sip stuff |
| 02:48.58 | Qwell | snomsterisk |
| 02:49.36 | mogorman | i mean im not trying to belittle anyone |
| 02:49.41 | mogorman | but if there is an asterisk box |
| 02:49.43 | mogorman | id love to buy it |
| 02:49.46 | mogorman | ^_^ |
| 02:49.54 | mogorman | for now my wrt and wgt will have to make me happy |
| 02:51.13 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 02:52.25 | tainted- | mogorman how is that working for you? (asterisk on wrt) |
| 02:53.17 | mogorman | well i did it to learn it |
| 02:53.23 | mogorman | but its good enough for a house |
| 02:53.31 | mogorman | i never pushed more than 6 channels |
| 02:53.36 | mogorman | wgt634u is better |
| 02:53.38 | mogorman | as it has usb port |
| 02:53.39 | tainted- | any transcoding? |
| 02:53.40 | mogorman | more ram |
| 02:53.45 | mogorman | ulaw to gsm |
| 02:53.48 | mogorman | 2 channels |
| 02:53.53 | tainted- | wow |
| 02:53.53 | mogorman | is all i have ever pushed |
| 02:54.02 | Sedorox | I need 4 powerbricks... 2 for WRT54G's.. and 2 for the 54GC's.... |
| 02:54.18 | Qwell | Sedorox: rig them up for POE |
| 02:54.24 | Qwell | PoE |
| 02:54.28 | tainted- | yea build a poe breakout box |
| 02:56.06 | mogorman | but go grab wgt634u its so easy |
| 02:56.26 | Qwell | Those are only like $40 too, aren't they? |
| 02:56.32 | mogorman | one place does |
| 02:56.35 | Sedorox | lol |
| 02:56.36 | mogorman | i heard its eol |
| 02:56.38 | Qwell | can't beat that |
| 02:56.41 | mogorman | most sell it for 60 |
| 02:56.43 | Sedorox | well I wanna sell the GC's |
| 02:56.44 | Qwell | oh |
| 02:57.06 | Sedorox | but the one GC is mine.. which I could do.. but I don't have anything to inject the power :p |
| 02:57.12 | mogorman | but its worth it just for the wifi card Qwell |
| 02:57.15 | mogorman | it goes for 90 |
| 02:57.21 | Qwell | mogorman: yeah...what's up with that? |
| 02:57.25 | mogorman | you could just buy wgt634us and rip them apart |
| 02:57.27 | mogorman | and sell the cards |
| 02:57.58 | Qwell | heh |
| 03:00.14 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:00.23 | *** join/#asterisk The_Isle_of_Mark (n=mark@c-68-85-63-96.hsd1.ga.comcast.net) |
| 03:05.56 | |omni| | coo..manager interface is working sweet |
| 03:06.05 | key2 | !seen mark |
| 03:06.11 | key2 | !seen kram |
| 03:10.45 | *** join/#asterisk S4w (n=sasa@adsl-3-166-86.mia.bellsouth.net) |
| 03:11.50 | S4w | hey guys I live in USA with bellsouth and I am having trouble with callprogress. Whenever I place a call the called partu answers but asterisk would not notice it. Are there any fixes for this? |
| 03:12.21 | mogorman | if you have an analog line you will not get reliable call progress |
| 03:12.44 | S4w | hmm |
| 03:12.48 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 03:12.58 | S4w | so that means that I am screwed up? |
| 03:13.06 | S4w | I just have a residential phone line |
| 03:13.08 | S4w | :-| |
| 03:13.09 | mogorman | well you can use callprogress |
| 03:13.18 | mogorman | but getting reliable call progress over analog is hard |
| 03:13.22 | *** join/#asterisk angom_h (n=angom@red-corp-201.130.165.94.telnor.net) |
| 03:13.24 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 03:13.30 | mogorman | you could ask em about kewlstart |
| 03:13.49 | S4w | I am using callprogress and koolstart |
| 03:13.59 | S4w | is there any way to fine\tune callprogress? |
| 03:14.18 | mogorman | well you can edit indications.conf |
| 03:14.23 | mogorman | but i wouldnt reccomend it |
| 03:14.25 | *** join/#asterisk coppice (n=chatzill@243.143.17.210.dyn.pacific.net.hk) |
| 03:14.30 | S4w | hmm |
| 03:14.31 | S4w | yes |
| 03:14.38 | S4w | it reads like chineese |
| 03:14.40 | S4w | :-S |
| 03:15.04 | Sedorox | ahh yes.. asterisk looked like that to me at first :p |
| 03:15.22 | mogorman | heh |
| 03:15.46 | SwK | heh |
| 03:16.00 | SwK | hah baseball sized hail |
| 03:16.32 | brookshire | there is not baseball sized hail |
| 03:16.38 | brookshire | it's only quarter sized! |
| 03:17.48 | SwK | brooks they are reporting baseball out side of decater |
| 03:20.15 | Abydos313 | ouch |
| 03:21.16 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 03:21.26 | CrashHD | abydos |
| 03:21.31 | CrashHD | isn't that a stargate world |
| 03:21.42 | Abydos313 | oh yeah! |
| 03:21.59 | Abydos313 | been a fan since 97 |
| 03:22.38 | terrapen | anyone here have an Aastra phone? |
| 03:22.55 | Sedorox | terrapen: no.. but let me know how it works out :p been eyeing them |
| 03:23.06 | terrapen | well, i can't get it to register |
| 03:23.13 | Sedorox | ah |
| 03:23.19 | terrapen | im wondering if im doing something wrong in sip.conf |
| 03:23.24 | terrapen | it looks correct |
| 03:23.30 | Yasheee | what's a cheapy SIP phone I can get off ebay for linux, *bsd or M$? |
| 03:23.50 | Sedorox | Yasheee: ummmm.... what do you mean linux, *bsd, or m$? |
| 03:24.16 | The_Isle_of_Mark | anyone here have any experience with a draytek 2900v? |
| 03:25.00 | *** join/#asterisk zimdog (n=zimdog@c-24-9-24-165.hsd1.co.comcast.net) |
| 03:25.29 | Yasheee | Sedorox: one that will work with any OS |
| 03:25.50 | Sedorox | you mean softphone? |
| 03:26.09 | Sedorox | hardphones don't matter on the operating system of a computer.. they don't even need a computer (besides whatever they terminate to...) |
| 03:32.09 | Yasheee | yes, just a set that is known to work well with the various software phones |
| 03:32.34 | Sedorox | ohhh.. headset? |
| 03:33.35 | Yasheee | yes |
| 03:33.54 | Sedorox | doesn't matter.. just pick one that sounds good.. on both ends.. so you may wanna read reviews on them |
| 03:34.10 | Sedorox | thats not dependant on operating system, softphone.. software.. or anything |
| 03:34.22 | Yasheee | I just thought asking here would be a good place, as I'm sure you guys have a lot of working experience with various ones |
| 03:34.30 | Sedorox | true... |
| 03:34.43 | Sedorox | I don't personally.. I have a shitty BT100 hardphone :p but someone else might |
| 03:36.07 | *** join/#asterisk fender21 (n=fender21@cpe-66-69-90-91.satx.res.rr.com) |
| 03:45.01 | *** join/#asterisk techman97_andy (n=me@70-98-20-60.dsl1.rsm.mn.frontiernet.net) |
| 03:45.06 | techman97_andy | hello all - wondering if I can get some sanity checks here...I config'd an * system about a year ago, but hadn't touched it since. Now, at a new company and looking for a refresher set of eyes...anyone out there tonight? |
| 03:45.37 | tainted- | techman97_andy what do u need? |
| 03:46.56 | techman97_andy | hello! Basically, here is my end-game config. I have two X100P cards that will take some legacy phone lines, and we have 6 SIP lines through VoiceEclipse. I have 7 stations that will be connecting to the * server. I'm in the WIKI (that's how I figured it out last time) and am having one bugger of a time getting the zaptel.conf / ztcfg to come back clean. |
| 03:47.47 | techman97_andy | the error I'm getting is "Line 221: Cannot get number of tones chanel 1" (and chanel 2) |
| 03:48.02 | techman97_andy | this has to be something completely easy that I'm completely overlooking. |
| 03:48.08 | techman97_andy | =S |
| 03:48.10 | techman97_andy | any hints? |
| 03:50.53 | techman97_andy | the only lines that I have enabled in the zaptel.conf file are: (without quotes) "fxsks=1-2", "loadzone=us", "defaultzone=us", and "channels=1-2" |
| 03:50.58 | tainted- | hmm that is an odd one |
| 03:51.08 | techman97_andy | it's nothing fancy....just two Wildcard X100Ps. |
| 03:51.17 | *** join/#asterisk bmg505 (n=leon@165.165.155.110) |
| 03:51.49 | techman97_andy | thinking I screwed up the fxs / fxo thing - I swapped out that line and tried the /sbin/ztcfg -vvvv again and got the same message no matter what |
| 03:52.15 | techman97_andy | the system found and registered up the two card as Wildcard X100Ps....so I know it sees them...=S |
| 03:52.47 | techman97_andy | (I'm kinda beating my head against the keyboard here...this isn't that hard as I remember...=P) |
| 03:52.48 | *** join/#asterisk lilo_ (i=levin@freenode/staff/pdpc.levin) |
| 03:58.19 | techman97_andy | any ideas anyone? |
| 03:58.21 | techman97_andy | =( |
| 03:59.50 | *** join/#asterisk somegeek (i=levin@unaffiliated/somegeek) |
| 03:59.53 | *** join/#asterisk bkw_ (n=brian@c-68-32-112-142.hsd1.md.comcast.net) |
| 04:00.58 | techman97_andy | ha ha! figured it out |
| 04:01.07 | techman97_andy | it was that last "channels=1-2" that was killing it |
| 04:01.24 | *** join/#asterisk CpuID2 (n=none@gentoo/contributor/cpuid) |
| 04:01.45 | techman97_andy | ya just whack at it enough and eventually you figure it out. |
| 04:01.47 | techman97_andy | rock on |
| 04:10.35 | *** join/#asterisk robl^ (n=robl@dsl093-025-118.hou1.dsl.speakeasy.net) |
| 04:12.51 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 04:25.52 | Sedorox | ok.. so voip gets hell when people can't call 911... but this operater doesn't lose her job after saying the boy was prank calling??? (http://www.msnbc.msn.com/id/12208992/) Sometimes... I hate the United States *watches Homeland come after him* |
| 04:28.51 | mogorman | to hear what really happened |
| 04:30.46 | Sedorox | yea... |
| 04:30.56 | mogorman | i wonder if i could do it |
| 04:31.01 | mogorman | its kinda a pain though |
| 04:31.04 | Sedorox | still pisses me off like that.. and I bet this will be the only press it gets.. unlike when there is a voip 911 issue... |
| 04:31.06 | mogorman | as you have to know the time call was made |
| 04:33.33 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 04:33.43 | *** join/#asterisk somegeek (i=levin@unaffiliated/somegeek) |
| 04:33.52 | techman97_andy | here's a weird question - in my previous * system, I didn't come across this, but in the newest release, I'm trying to get some ZAP channels (2 Wildcard x100p cards) running, but * doesn't have the "chan_zap.so" module loaded - how do I do that? |
| 05:03.02 | *** join/#asterisk tecnico (n=tecnico@user-24-236-120-2.knology.net) |
| 05:07.43 | drray | techman modprobe |
| 05:07.49 | techman97_andy | yeah, got it |
| 05:07.51 | techman97_andy | =) |
| 05:07.59 | drray | sorry.. just woke up |
| 05:08.09 | techman97_andy | np man - thanks for the response anyways...*smile* |
| 05:08.11 | techman97_andy | heheheheh |
| 05:08.11 | techman97_andy | O |
| 05:08.24 | techman97_andy | I'm sure I'll have a few more on here thoughout the weekend |
| 05:08.26 | techman97_andy | =) |
| 05:08.29 | drray | you are happy with 2 x100ps? |
| 05:09.02 | techman97_andy | they worked without much issue on my last * system - we're using them here for 911 and the main line |
| 05:09.06 | techman97_andy | they seem to work |
| 05:13.40 | *** join/#asterisk lokkju (n=lokkju@unaffiliated/lokkju) |
| 05:14.35 | lokkju | I am trying to just get voice conferancing set up on a PBX system, for now, and have the asterisk fully installed on a debian box, but am somewhat stuck on how to configure *just* inbout voice conference dialing |
| 05:15.17 | lokkju | do I need to set up a new sip phone, that I then put as the forward to number for a landline? |
| 05:15.30 | lokkju | or can I forward directly to a conf room? |
| 05:15.38 | drray | Meetme |
| 05:15.47 | lokkju | I know |
| 05:17.07 | lokkju | but how do I handle some sort of inbound VoIP connection? |
| 05:18.22 | drray | I would think that the s extension would feed to meetme |
| 05:19.24 | lokkju | hmm |
| 05:19.29 | drray | unless I don't understand what you want |
| 05:19.38 | lokkju | so <meetme extension>@<my server ip>? |
| 05:20.49 | drray | well, when you register a sip phone you then can pick it up and dial the meetme extension you created |
| 05:21.18 | drray | or you could use the s extension to link and inbound call to meetme (if your sip phone is on the outside) |
| 05:25.18 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.222) |
| 05:25.56 | *** join/#asterisk VxJasonxV (n=jason@unaffiliated/VxJasonxV) |
| 05:40.48 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 05:42.23 | *** join/#asterisk habakuk (n=apyles@64.71.190.176) |
| 05:46.08 | habakuk | Hey I'm having some weird 1 way audio issues with IAX. Anyone got some pointers as to why this is occuring? |
| 05:46.57 | habakuk | Iaxclient -> Asterisk = fine; iaxclient -> asterisk -> SIP provider = 1 way audio |
| 05:47.22 | habakuk | AND.. SIP -> asterisk -> sip provider =fine |
| 05:48.51 | habakuk | doing a tcpdump at asterisk server tells me I'm getting two way audio. But it's not getting delivered to the iax client |
| 05:51.04 | brookshire | habakuk: could be a codec problem |
| 05:52.57 | habakuk | brookshire: what did you have in mind? I'm using 711u ast - sip. gsm to iax2. Swithching IAX to gsm didn't change anything |
| 05:53.15 | lokkju | hmm |
| 05:53.31 | brookshire | did you force it to gsm in asterisk? |
| 05:53.47 | habakuk | IAX portion yes |
| 05:53.50 | brookshire | and how sure are you that the asterisk box is getting both sides of the sip |
| 05:53.56 | brookshire | both rtp and control? |
| 05:53.57 | lokkju | after installing asterisk, I should, at the very least, be able to call the server (sip:90@<myserver>), if 90 is set up as an echo test, right? |
| 05:54.14 | habakuk | brookshire: tcpdump -> ethereal -> .au |
| 05:54.34 | brookshire | habakuk: you might be sending it, but not receiving it |
| 05:54.41 | brookshire | is the asterisk box firewalled? |
| 05:54.50 | habakuk | no it's 2 way |
| 05:54.58 | habakuk | sip clients work fine |
| 05:55.08 | brookshire | so it's just iax |
| 05:55.21 | brookshire | does your iax client support gsm? |
| 05:55.25 | habakuk | just iax yeah.. thats the weird thing |
| 05:55.27 | habakuk | yeah |
| 05:55.48 | habakuk | yeah.. if I play an annoucement on the ASt server it works fine.. |
| 05:55.48 | brookshire | if it's iax, then it sounds like a codec issue |
| 05:55.53 | habakuk | echo works fine |
| 05:57.06 | habakuk | brookshire: hmm.. yeah it could be.. any other things to try? I'm fresh out of ideas |
| 05:57.53 | brookshire | try ulaw |
| 05:58.01 | brookshire | disallow=all |
| 05:58.04 | brookshire | allow=ulaw |
| 05:58.34 | habakuk | yeah I tried that.. and forced the client to only use ulaw |
| 05:58.35 | brookshire | if that works... then change ulaw back to gsm |
| 05:58.44 | brookshire | same problem? |
| 05:58.46 | habakuk | but no dice |
| 05:58.59 | brookshire | see.. usually with iax, if one side works, so does the other |
| 05:59.04 | brookshire | it's not adding up |
| 06:00.22 | habakuk | brookshire: yeah.. running ethereal on my laptop (with iax client) I'm seeing bidirectional iax packets |
| 06:00.38 | habakuk | brookshire: exactly.. it's driving me crazy.. |
| 06:00.42 | brookshire | f one side has trunk=yes and the other does not cannot validate the peer, you will get one-way audio. |
| 06:00.46 | X-Rob | habakuk, use a different iax client. |
| 06:00.59 | habakuk | X-Rob: I've tried 3 of them |
| 06:01.10 | X-Rob | wierd. |
| 06:01.31 | habakuk | brookshire: hmm.. I'll double check that.. Is that on by default? |
| 06:01.46 | brookshire | could be |
| 06:01.52 | X-Rob | no |
| 06:02.09 | habakuk | cause there is just one iax2 client -> 1 server -> sip provider |
| 06:02.49 | habakuk | X-Rob: ok I'll double check though |
| 06:03.17 | habakuk | yep no trunking.. |
| 06:04.08 | habakuk | running latest 1.2.6, updated time with ntp(slightly off).. fresh out of ideas |
| 06:05.05 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-175-120.lsanca.fios.verizon.net) |
| 06:05.35 | habakuk | anyone know if there is a tool to save IAX libpcap captures into audio files? |
| 06:05.44 | wasim | habakuk: ethereal |
| 06:05.55 | wasim | habakuk: oh, audio files, sorry, no |
| 06:06.03 | brookshire | can you post your iax.conf to pastebin? |
| 06:06.17 | brookshire | remove the passwords of course :) |
| 06:06.46 | *** join/#asterisk yaboo (n=jsirucka@220-245-131-131.static.tpgi.com.au) |
| 06:07.07 | Gamercjm | voipmasta: |
| 06:07.49 | habakuk | brookshire: sure its the default from 1.2.6 just using the guest account |
| 06:07.53 | habakuk | for testing |
| 06:09.45 | *** join/#asterisk forao (n=dfasdfs@ool-4354d60d.dyn.optonline.net) |
| 06:12.36 | *** join/#asterisk tainted- (n=identd@ppp-71-134-51-75.dsl.irvnca.pacbell.net) |
| 06:16.57 | yaboo | hello anyone using a cisco router for outbound pstn gateway |
| 06:23.19 | lokkju | so, I can see that my sip client is connecting (firewall logs) but asterisk isn't doing anything... and nothing shows in the asterisk logs, though it *IS* running |
| 06:23.47 | lilo | is there a channel staffer around? |
| 06:24.19 | Qwell | lilo: not too often |
| 06:24.44 | Qwell | lilo: anything we can help with? |
| 06:25.24 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 06:26.43 | brookshire | lilo: i can probably get someone :) |
| 06:26.50 | Qwell | yeah, was gonna say |
| 06:27.15 | lilo | I just recloaked a tor user, I think he's a regular |
| 06:27.26 | lilo | just let me know if you have any problems, he'll be labeled |
| 06:27.35 | lilo | (if that's okay) |
| 06:27.40 | brookshire | we use to have tor banned |
| 06:27.41 | *** join/#asterisk jimbe (n=jimbe@tor/contact-lilo-in-case-of-problems/x-1ab7739bd3e673e9) |
| 06:27.46 | lilo | he's escorted 8) |
| 06:27.48 | Qwell | heh |
| 06:27.50 | lilo | you still have tor banned |
| 06:28.09 | lilo | I know there are problems off and on |
| 06:28.11 | jimbe | thanks lilo! |
| 06:28.23 | lilo | we're also working on a reputation system to reduce the problems |
| 06:28.25 | brookshire | tor users were coming in here and dropping pedo porn links |
| 06:28.25 | lilo | jimbe: glad to help |
| 06:28.29 | brookshire | we had to do it |
| 06:28.34 | Qwell | brookshire: fun |
| 06:28.40 | lilo | brookshire: yeah, the abuse is a pain....the reputation system is simple, but I Think it'll help |
| 06:28.50 | lilo | erm think 8) |
| 06:31.30 | Qwell | What is the purpose of tor exactly? |
| 06:31.48 | jimbe | anonymizer |
| 06:31.59 | Qwell | you mean like the cloak I have? |
| 06:32.09 | jimbe | yep |
| 06:32.15 | Qwell | So, what's the point of it? |
| 06:32.23 | jimbe | http://tor.eff.org/ |
| 06:32.42 | jimbe | what's the point of cloaking? |
| 06:33.00 | Qwell | So I can troll on other channels and not have my true identity known |
| 06:33.01 | Qwell | wait...no |
| 06:33.03 | Qwell | :p |
| 06:33.19 | Qwell | because being packet flooded sucks |
| 06:33.50 | *** join/#asterisk dlynes (n=dlynes@S010600c09f9a0fc4.vc.shawcable.net) |
| 06:34.33 | jimbe | i'm special tho |
| 06:34.37 | jimbe | i've been 'tagged' :D |
| 06:35.46 | brookshire | i truthfully don't think tor is any more secure |
| 06:35.55 | jimbe | how so |
| 06:36.05 | brookshire | i hate the fact that random peers are receiving packets |
| 06:36.30 | brookshire | it's like.. it puts more info the hands of others |
| 06:36.51 | jimbe | it'd be pretty hard to piece all the info together into something usable |
| 06:37.01 | *** join/#asterisk froguz (n=froguz@83-142-222-201.adsl.terra.cl) |
| 06:37.06 | jimbe | my box thinks i just ssh'd in from miami lol |
| 06:37.06 | brookshire | yeah.. but sometimes a little bit is all you need |
| 06:37.39 | jimbe | well like Qwell said, just for trolling irc |
| 06:39.32 | jimbe | brookshire is there any way to get to the SIP Response after a DIAL()? |
| 06:39.46 | jimbe | ${DIALSTATUS} is encapsulating too much info |
| 06:42.22 | *** join/#asterisk Tili (i=Tili@218.19.66.54) |
| 06:42.29 | brookshire | i guess i don't understand your question |
| 06:42.41 | brookshire | i know how to make phones ring :) |
| 06:45.39 | *** join/#asterisk MGSsancho (n=user@ppp-67-126-243-88.dsl.irvnca.pacbell.net) |
| 06:56.08 | *** join/#asterisk tuxinator_linux (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 06:56.24 | drray | telwest (our new Telco provider is making a great 2nd impression ;/) |
| 06:57.59 | *** join/#asterisk KaBewM (n=DA-MAN@66-215-7-106.dhcp.psdn.ca.charter.com) |
| 07:04.32 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 07:21.32 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:24.28 | PakiPenguin | anyone here used astribanl? |
| 07:24.33 | PakiPenguin | astribank* |
| 07:28.14 | *** join/#asterisk lorinc (n=ang@caracas-3587.adsl.interware.hu) |
| 07:30.20 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:31.18 | PakiPenguin | tzafrir, around?? |
| 07:35.55 | *** join/#asterisk Shaun2222 (n=ndci@ip68-5-63-223.oc.oc.cox.net) |
| 07:38.35 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 07:43.51 | *** join/#asterisk Uzzi (n=Andrea@host238-239.pool875.interbusiness.it) |
| 07:44.18 | Uzzi | Hi guys! |
| 07:44.47 | Uzzi | Ther is someone who use asterisk with a zoltrix voice modem? |
| 08:04.42 | *** join/#asterisk SheriF_WorK (n=sherif@212.103.170.135) |
| 08:05.53 | bmg505 | I tried modems but 1.2.5/6 would not load the chan_modem*.so files |
| 08:06.12 | bmg505 | was bitching about some symbols |
| 08:07.05 | bmg505 | morning peeps |
| 08:08.08 | bmg505 | can I force * to re-initiate the fwd iax2 link at certain times in the day (in SA we must change ip addy every 24 hours so I force mine to 05:30) |
| 08:12.44 | *** join/#asterisk thx2000 (i=AgentFLY@adsl-66-51-192-221.dslextreme.com) |
| 08:13.38 | thx2000 | Does teliax not allow you to change the CIDName of outgoing calls? |
| 08:13.55 | Strom_M | thx2000, no one allows you to change the name |
| 08:14.06 | Strom_M | thats not how the north american network operates |
| 08:14.12 | thx2000 | doh |
| 08:14.48 | k31th | Anyone running asterisk on gentoo |
| 08:15.02 | Qwell | k31th: many people. ask your question |
| 08:15.36 | Uzzi | Ther is someone who use asterisk with a zoltrix voice modem? |
| 08:15.42 | FuriousGeorge | k31th: i do |
| 08:15.42 | k31th | Qwell: Well i wondered do you use the ebuilds |
| 08:15.46 | Qwell | no |
| 08:15.47 | FuriousGeorge | no |
| 08:16.00 | k31th | you dont? why is this out of interest ? |
| 08:16.12 | Qwell | because asterisk packages are pretty much junk |
| 08:16.28 | FuriousGeorge | old versions, for starters |
| 08:16.31 | k31th | I see, so how do you go about upgrading the source installs ? |
| 08:16.36 | mogorman | total junk Qwell |
| 08:16.38 | k31th | i noticed that |
| 08:16.43 | Qwell | by getting the latest source and compiling it |
| 08:16.43 | mogorman | and code is so much better up to date |
| 08:16.51 | Qwell | mogorman: indeed, trunk 4 lyfe |
| 08:16.54 | mogorman | but /me is heading to bed |
| 08:16.56 | mogorman | gnite people |
| 08:16.58 | FuriousGeorge | i wait for qwell to tell me too |
| 08:17.02 | FuriousGeorge | *to |
| 08:17.02 | Qwell | night |
| 08:17.02 | mogorman | Qwell, get back to work on skinny............ |
| 08:17.13 | k31th | Yeah sure i get that, so you install asterisk from source then 3 months time a new version is released |
| 08:17.15 | Qwell | mogorman: I realized tonight, that I now need a second monitor |
| 08:17.18 | k31th | wat do you do |
| 08:17.22 | FuriousGeorge | i make |
| 08:17.24 | mogorman | lol |
| 08:17.28 | k31th | just grab the latest source and compilie it again ? |
| 08:17.28 | Qwell | I want to get like a 7" lcd, for mythtv :p |
| 08:17.29 | FuriousGeorge | then i make install |
| 08:17.40 | mogorman | i have a 20 inch for my mythtv box |
| 08:17.42 | FuriousGeorge | k31th: yeah |
| 08:17.45 | Qwell | yeesh |
| 08:17.46 | FuriousGeorge | whats the big deal |
| 08:17.50 | mogorman | as i never ended up using it as a monitor |
| 08:17.52 | k31th | FuriousGeorge: didnt relise it was that simple |
| 08:17.54 | mogorman | i got this mac mini and lcd |
| 08:17.55 | mogorman | to use |
| 08:18.02 | mogorman | and i never did sit down at it |
| 08:18.10 | mogorman | im just addicted to laptops............... |
| 08:18.15 | k31th | FuriousGeorge: do you use emerge to install the source ? |
| 08:18.15 | FuriousGeorge | you download the file, extract it, make && make install |
| 08:18.17 | mogorman | except at work |
| 08:18.24 | mogorman | <PROTECTED> |
| 08:18.24 | k31th | that is possible iirc |
| 08:18.26 | CpuID2 | desktops rawk |
| 08:18.29 | Qwell | I can't stand typing on laptops |
| 08:18.34 | Qwell | well... |
| 08:18.39 | Qwell | I can't stand using laptop mice |
| 08:18.39 | mogorman | i agree |
| 08:18.49 | Qwell | keyboard I can deal with, I guess |
| 08:18.50 | k31th | yeah they both suck |
| 08:18.50 | mogorman | i love ibm m series keyboard i have |
| 08:19.02 | mogorman | but i dont have a desk / chair at home |
| 08:19.05 | Qwell | lame |
| 08:19.08 | mogorman | so its easier to work on the couch |
| 08:19.11 | k31th | mogorman: is it a clicky keyboard ? |
| 08:19.14 | mogorman | yeah |
| 08:19.19 | FuriousGeorge | k31th: on a clean stage 3 system i emerge asterisk only to get the dependencies, then i mask it so portage never tries to update it, then i grab the source |
| 08:19.20 | mogorman | its basically a typewritere |
| 08:19.21 | Qwell | I have a cheesy desk...an armoir(sp) |
| 08:19.52 | mogorman | you cant do an emerge deps for portage like you can with debian? FuriousGeorge |
| 08:20.01 | k31th | FuriousGeorge: I see maybe ill folow your aproache |
| 08:20.23 | k31th | waiting for xfce to emerge and a bunch of other stuff first 138 / 148 |
| 08:20.37 | FuriousGeorge | mogorman: i just let it install * too then i nuke /usr/lib/asterisk/modules |
| 08:20.50 | Qwell | FuriousGeorge: You can emerge -e it, and the deps will stay... |
| 08:21.11 | k31th | I thought it was possible to add the source to portage ? |
| 08:21.27 | k31th | im sure i installed cedega like that |
| 08:21.28 | FuriousGeorge | Qwell: it achieves the same thing, installs * and all the deps |
| 08:21.31 | FuriousGeorge | right? |
| 08:21.49 | mogorman | man i dont have any more gentoo boxes these days :( |
| 08:21.49 | FuriousGeorge | emptytree is just if you wanna recompile all the dependencies too, iirc |
| 08:21.58 | Qwell | -e is unmerge |
| 08:22.06 | mogorman | i just have debian, lfs box, and some random firmware builds i have made |
| 08:22.06 | FuriousGeorge | -C |
| 08:22.10 | FuriousGeorge | -e is emptytree |
| 08:22.12 | Qwell | ahh, I'm thinking rpm :P |
| 08:22.14 | mogorman | its dead to me |
| 08:22.17 | mogorman | death to rpm |
| 08:22.20 | mogorman | DEATH |
| 08:22.28 | Qwell | I haven't used rpm in a year or so |
| 08:22.38 | FuriousGeorge | not a fan either |
| 08:22.42 | k31th | Jesus I have to use RPMS at work |
| 08:22.52 | k31th | there al being nuked slowly with debian servers |
| 08:23.04 | k31th | I wouldnt mind getnooing them but i dont have the bollow |
| 08:23.07 | k31th | bollox |
| 08:23.13 | mogorman | i find it funny that redhat and co wont let it die |
| 08:23.15 | k31th | gentooing |
| 08:23.18 | Qwell | nothing wrong with gentoo |
| 08:23.18 | mogorman | its so obviously a bad idea |
| 08:23.33 | k31th | wat rpm ? |
| 08:23.59 | mogorman | okay time to sleep |
| 08:24.02 | mogorman | for real this time |
| 08:24.06 | k31th | nyt |
| 08:24.12 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-169.claranet.co.uk) |
| 08:26.54 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:31.33 | bmg505 | Uzzi: I could nopt get _ANY_ voice modem to work with * 1.2.6 and the commants was that it is depreciated and not supported anymore |
| 08:32.34 | *** join/#asterisk UlbabraB (n=caplaz@host241-43.pool8172.interbusiness.it) |
| 08:33.34 | FuriousGeorge | hey qwell i was thinking of relearning to program, but the only personal itch i have to scratch with asterisk is that server_a doesnt know about the presence status of server_b's peers. how would you go about solving a distributed roster like that? should they just poll eachother all the time? |
| 08:34.36 | Qwell | dunno |
| 08:34.54 | FuriousGeorge | :D |
| 08:35.13 | FuriousGeorge | way to go out on a limb :) |
| 08:35.20 | [av]bani | :o |
| 08:35.27 | Qwell | it's 1:30 am...what do you expect? :p |
| 08:35.35 | FuriousGeorge | fair enough |
| 08:35.44 | FuriousGeorge | ill ask you again in 20 hours :) |
| 08:37.10 | FuriousGeorge | [av]bani: ever setsipheader() with your snom |
| 08:38.48 | Uzzi | sigh |
| 08:44.00 | Uzzi | bmg505, I've installed 1.0.9 version |
| 08:44.02 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 08:44.58 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 08:46.14 | bmg505 | whats the problem? |
| 08:46.54 | k31th | FuriousGeorge: are you self employeed ? |
| 08:46.56 | FuriousGeorge | so go get a newer one |
| 08:47.02 | FuriousGeorge | k31th: yeah |
| 08:47.06 | FuriousGeorge | thank god for bartending |
| 08:47.12 | k31th | FuriousGeorge: where are you located ? |
| 08:47.22 | FuriousGeorge | nj/nyc area |
| 08:47.26 | k31th | ahh |
| 08:47.43 | k31th | yes iv become the director of my own ltd company lately |
| 08:47.51 | k31th | i know wat you mean about getting clients |
| 08:48.03 | Uzzi | I don't know how config asterisk!my modem have 1out and 1in!All my phone are connected in parallel mode with modem!I want to monitorize the out traffic! |
| 08:48.16 | k31th | tbh i got ahnded a bunch of clients for free... But we need more getting clients is the hardest thing |
| 08:48.29 | MacDome | hi x86 |
| 08:48.45 | FuriousGeorge | i just do general it for small businesses and individuals, but i think im ready to move on up (so to speak) |
| 08:48.56 | FuriousGeorge | now that qwell has trained me |
| 08:49.54 | k31th | ha ha |
| 08:50.11 | k31th | we do general IT + Linux bit a php etc |
| 08:50.18 | k31th | but im over in the UK |
| 08:50.27 | FuriousGeorge | thats my problem. i dont code |
| 08:50.27 | k31th | accross the pond so to speak |
| 08:50.37 | FuriousGeorge | i know a little pascal i learned 7 years ago and thats it |
| 08:50.37 | k31th | FuriousGeorge: i dont do much |
| 08:51.06 | FuriousGeorge | anyway, im gonna hit the sack |
| 08:51.13 | k31th | odd shell script to help my admining odd passeord change pages etc php is handy whensetting up linux for comapnys the like easy to use web interfaces |
| 08:51.13 | FuriousGeorge | good talking to you guys |
| 08:51.21 | k31th | Night dude |
| 08:51.33 | FuriousGeorge | yeah, i just got a book on php and sql |
| 08:51.39 | FuriousGeorge | and another on css and html |
| 08:51.52 | FuriousGeorge | i figure i could use a web page |
| 08:52.11 | FuriousGeorge | and im gonna borrow my buddies book on C |
| 08:52.13 | FuriousGeorge | anyway |
| 08:52.51 | FuriousGeorge | if you learn one non-OO language 7 years ago, how hard can learning others be, right? |
| 08:53.05 | bmg505 | Uzzi: monitoring outgoing traffic is difficult, u usually need special hardware unless the modem can do passive hook detection, I know of one minicom modem that used to do it |
| 08:53.18 | k31th | indeed |
| 08:53.21 | FuriousGeorge | later |
| 08:53.25 | k31th | 1999 |
| 08:53.28 | k31th | later |
| 08:53.31 | bmg505 | search the manual for passive hook detection, and u dont need * for that |
| 08:55.03 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 08:55.40 | Uzzi | bmg505, how I can know if my modem support this function? |
| 08:57.13 | *** join/#asterisk TripTrap (n=TripTrap@p54BA2235.dip0.t-ipconnect.de) |
| 08:59.14 | *** join/#asterisk astra^^ (n=muhajir_@59.145.104.74) |
| 08:59.49 | x86 | MacDome: talk about a delayed response ;) |
| 08:59.52 | x86 | haha |
| 09:00.20 | *** join/#asterisk Eggplant (i=No@dsl-469.cascadeaccess.com) |
| 09:00.46 | MacDome | x86: yeah, I was out and about |
| 09:01.21 | Dream_WEaver | Any way to set a timezone per user (besides in voicemail) so, let's say, I can set the user's tz in SayUnixTime. |
| 09:02.01 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:07.53 | JunK-Y | some1 familiar with app_ices? |
| 09:08.35 | Uzzi | bmg505, Who I hve to config asterisk to try to work with my modem? |
| 09:10.03 | bmg505 | there is a howto, on setting the voice modems I have used it before, u need a modem manual to see which voice commands you modem supports |
| 09:11.19 | *** join/#asterisk littleball (n=littleba@cm188.epsilon169.maxonline.com.sg) |
| 09:12.47 | littleball | hello, is there any utilities with GUI which can show the real time status of the asterisk (such as active channels....etc) |
| 09:12.49 | littleball | ? |
| 09:21.47 | x86 | MacDome: your scrollback buffer must be rather impressive ;) |
| 09:22.26 | MacDome | :) |
| 09:25.59 | Uzzi | bmg505, this?http://www.zoltrix.com/support_html/PUBLIC/MODEM/ATmanual/ATTCMAND.HTM |
| 09:34.12 | wasim | littleball: yes, check the wiki, flashop is one |
| 09:39.06 | bmg505 | Uzzi: those commands is the generic stuff, no voice etc. in it I have a doc somewhere, but will look and see I must search through 900+G of data so it could be a while |
| 09:40.11 | Uzzi | tnks |
| 09:42.35 | Uzzi | bmg505, http://www.zoltrix.com/support_html/modem/USEMODEM.HTM |
| 09:53.32 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 09:55.09 | *** join/#asterisk ToTo (n=ToTo@host129-167.pool872.interbusiness.it) |
| 10:06.42 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 10:10.49 | *** join/#asterisk madd (n=madd@panthera-systems.net) |
| 10:10.59 | madd | moin |
| 10:12.12 | madd | is it possible to fax over voip without isdn and modem only over voip? |
| 10:15.04 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 10:17.21 | x86 | madd: supposedly it is possible over SIP using the ulaw codec |
| 10:18.00 | x86 | madd: it works about 50% of the time for me (using app_rxfax.so), but IAXmodem+hylafax is supposedly better (have not tried it yet) |
| 10:18.07 | x86 | anyone good with AGI? |
| 10:18.24 | x86 | especially Asterisk::AGI for perl? |
| 10:19.09 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 10:19.38 | madd | x86: thanks |
| 10:22.23 | madd | x86: is the rxfax-module only for recieve? |
| 10:24.48 | *** join/#asterisk loick (n=loick@APlessis-Bouchard-152-1-76-210.w81-48.abo.wanadoo.fr) |
| 10:25.07 | x86 | madd: yeah, but there is an app_txfax too (i have not tested this at all) |
| 10:30.24 | *** join/#asterisk ToTo (n=ToTo@host129-167.pool872.interbusiness.it) |
| 10:31.45 | *** join/#asterisk LapTop006 (n=laptop00@sparc006.chriskaine.com.au) |
| 10:37.34 | redcap1 | I'm trying to dial out using my ISDN phone connected to my internal isdn, but get the following error: Channel 0/1, span 1 got hangup, cause 42 |
| 10:37.37 | *** join/#asterisk aslam (n=aslamr@dsl-146-24-20.telkomadsl.co.za) |
| 10:37.48 | redcap1 | does this sound familiar to anyone ? |
| 10:39.19 | *** join/#asterisk apardo (n=apardo@87.218.45.206) |
| 10:39.37 | *** join/#asterisk stuartcw (n=chatzill@softbank221025056004.bbtec.net) |
| 10:43.22 | *** join/#asterisk Kernel_Core (i=Kernel_C@83.217.236.50) |
| 10:48.01 | key2 | !seen kram |
| 11:09.44 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 11:10.40 | Kernel_Core | does chan_h323 driver supports rtptimeout ? |
| 11:11.37 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 11:12.06 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 11:12.37 | tzafrir_laptop | PakiPenguin_, here? |
| 11:24.06 | *** join/#asterisk hgaillac (n=Harry@196.18.119-80.rev.gaoland.net) |
| 11:24.19 | hgaillac | hello |
| 11:24.46 | tzafrir_laptop | hi |
| 11:25.25 | hgaillac | I really need help to disable 407 proxy authentication |
| 11:26.58 | X-Rob | tzafrir_laptop, you mean 'Re: ' isn't a good enough subject? |
| 11:28.23 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 11:29.01 | tzafrir_laptop | I was actually refering to some variation of "help" |
| 11:29.33 | *** join/#asterisk hertell (n=rene@jumbo52.adsl.netsonic.fi) |
| 11:29.51 | tzafrir_laptop | hgaillac, anything specific you need help with? a.k.a: the little details |
| 11:32.57 | *** join/#asterisk coppice (n=chatzill@99.155.17.210.dyn.pacific.net.hk) |
| 11:37.43 | hgaillac | tzafrir_laptop: look at asterisk-users mailing list subject "HELP!!!" I try to configure ser+asterisk in order to forward to asterisk sip:info@mydomain but asterisk ask for authentication i tried insecure and more |
| 11:41.01 | *** join/#asterisk pengyong (n=lala@222.185.16.241) |
| 11:51.45 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 11:55.52 | *** part/#asterisk hertell (n=rene@jumbo52.adsl.netsonic.fi) |
| 12:00.07 | *** join/#asterisk Kernel_Core (i=Kernel_C@52.230.dial-up.xter.net) |
| 12:00.31 | Kernel_Core | hi all , does chan_h323.so supports rtptimeout? |
| 12:01.22 | tzafrir_laptop | hgaillac, do you have an entry for "guest" in sip.conf? |
| 12:01.26 | Kernel_Core | or how do I set in h323.conf if there was no RTP activity for 60seconds , then asterisk should terminate the call ? |
| 12:02.57 | hgaillac | tzafrir_laptop: What do you mean guest in sip.conf ? |
| 12:04.58 | tzafrir_laptop | How exactly do you expect to authenticate those incoming calls? |
| 12:05.20 | *** join/#asterisk brif8 (n=chatzill@rrcs-71-41-50-162.se.biz.rr.com) |
| 12:05.46 | brif8 | How do I take a Cisco Phone (which I think is MGCP) and convert it to SIP ? |
| 12:08.06 | hgaillac | tzafrir_laptop: I want users to reach hunt group with uri sip:info@domain or sip:support@domain |
| 12:10.06 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:10.56 | *** join/#asterisk phez (n=phez@redcap.xs4all.nl) |
| 12:14.43 | *** join/#asterisk vopi (n=kkk@202.139.207.171) |
| 12:20.27 | *** join/#asterisk Tili (i=Tili@61.140.190.44) |
| 12:23.03 | drray | brif8 you need a sip image from cisco |
| 12:24.38 | brif8 | drray: I have them It seems I can't get the phone to see the tftp server |
| 12:25.56 | hgaillac | tzafrir_laptop: where do you set guest in sip.conf ? |
| 12:28.05 | *** join/#asterisk loick (n=loick@APlessis-Bouchard-152-1-45-144.w83-114.abo.wanadoo.fr) |
| 12:30.16 | SplasPood | Anyone have any clue as to why I can't load my zaptel.ko, I'm getting this message: |
| 12:30.17 | SplasPood | zaptel: disagrees about version of symbol copy_to_userzaptel: Unknown symbol copy_to_user |
| 12:30.39 | tzafrir_laptop | SplasPood, with what kernel? what distro? |
| 12:31.28 | tzafrir_laptop | Sounds like you built zaptel vs. the wrong kernel headers |
| 12:31.45 | SplasPood | 2.6.15, debian.. I'd guess that too, but I didn't.. |
| 12:33.34 | *** join/#asterisk Uzzi (n=Andrea@host68-238.pool874.interbusiness.it) |
| 12:33.36 | hgaillac | Ok no way to disable 407 proxy authrntication !!! :-( |
| 12:33.48 | *** join/#asterisk RoyK (n=roy@212.17.141.54) |
| 12:40.44 | SplasPood | hgaillac: You've been asking the same question, over and over, since yesterday? |
| 12:43.10 | tzafrir_laptop | hgaillac, exclemation marks will not get you answers... |
| 12:43.18 | hgaillac | Splaspood: I've been asking the same question over and over . can we disable proxy authentication in asterisk YES or NO ? |
| 12:43.23 | florz | tzafrir_laptop: why not! |
| 12:43.49 | SplasPood | hgaillac: You've been asking the same question, over and over, since yesterday? |
| 12:44.21 | SplasPood | hgaillac: I note tzafrir tried to help you not long ago.. |
| 12:45.36 | hgaillac | Splaspood: I wish to tank tzafrir for help |
| 12:49.39 | tzafrir_laptop | hgaillac, I am not aware of any way to disable it. But the common workaround I am aware it, is to refer it to the [guest] user in sip.conf |
| 12:53.16 | hgaillac | tzafrir_laptop: you mean i have to set [guest] in sip.conf with type=user ?! |
| 12:54.15 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 12:55.37 | tzafrir_laptop | yes. I can't think of anything smarter |
| 12:55.38 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 12:57.30 | *** join/#asterisk guyb_home (n=guy@115.251-7-195.ippool.ndo.com) |
| 12:58.33 | *** join/#asterisk backblue (n=moo@87-196-9-196.net.novis.pt) |
| 12:59.07 | *** join/#asterisk saftsack (n=saftsack@p54A7FDAA.dip.t-dialin.net) |
| 13:00.01 | guyb_home | stock sell off, any interest? e100p, te110p, rhino channel bank, cisco 7905g sip, etc uk only |
| 13:01.49 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:19.46 | tzafrir_laptop | hgaillac, asterisk-devel is not a simple extension of asterisk-users |
| 13:22.40 | hgaillac | tzafrir_laptop: I agree you i just want to know if proxy authentication can be disabled try to call sip:info@nxs.yi.org if possible |
| 13:23.44 | *** join/#asterisk austinnichols101 (n=austinni@dsl-10-169.cofs.net) |
| 13:23.49 | tzafrir_laptop | hgaillac, if the secret for the user is empty? |
| 13:25.17 | x86 | in the UK, what does NCFA mean? |
| 13:27.26 | hgaillac | tzafrir_laptop: secret is empty , insecure=very , type=peer realm=nxs.yi.org host=dynamic |
| 13:27.30 | *** join/#asterisk Uzzi (n=Andrea@host68-238.pool874.interbusiness.it) |
| 13:30.23 | *** join/#asterisk gammacoder (n=chatzill@cpe-65-26-178-240.indy.res.rr.com) |
| 13:33.35 | *** join/#asterisk pixolex (n=chatzill@87-196-155-139.net.novis.pt) |
| 13:34.10 | *** join/#asterisk vopi (n=kkk@202.139.208.94) |
| 13:34.15 | tzafrir_laptop | type=peer means that this is basically for outgoingcalls |
| 13:34.29 | tzafrir_laptop | You need something for incoming calls. This is type=user |
| 13:34.43 | tzafrir_laptop | or type=friend |
| 13:42.34 | *** join/#asterisk Lino` (i=Lino@i577BFDA8.versanet.de) |
| 13:43.34 | *** join/#asterisk valentines (i=valentin@modemcable101.208-202-24.mc.videotron.ca) |
| 13:44.39 | *** join/#asterisk pengyong (n=lala@222.185.16.241) |
| 13:50.04 | *** join/#asterisk mafkees (n=michiel@vanbaak.xs4all.nl) |
| 13:54.30 | hgaillac | OK disable proxy authentication should be impossible unlike sip express router |
| 14:00.04 | *** join/#asterisk bkw_ (n=brian@c-68-32-112-142.hsd1.md.comcast.net) |
| 14:01.20 | xachen | we still a weeeeeeebit drunk? :P |
| 14:01.53 | mafkees | :) |
| 14:02.14 | mafkees | is there something wrong with svn.digium.com? |
| 14:02.26 | mafkees | I cannot checkout a fresh copy of 1.2 |
| 14:02.26 | *** join/#asterisk coppice (n=chatzill@177.195.17.210.dyn.pacific.net.hk) |
| 14:03.04 | mafkees | root@sin { /usr/src/asterisk }$ svn checkout http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-1.2 |
| 14:03.07 | mafkees | svn: REPORT request failed on '/svn/asterisk/!svn/vcc/default' |
| 14:03.09 | mafkees | svn: REPORT of '/svn/asterisk/!svn/vcc/default': 400 Bad Request (http://svn.digium.com) |
| 14:03.26 | *** join/#asterisk The_Isle_of_Mark (n=mark@c-68-85-63-96.hsd1.ga.comcast.net) |
| 14:03.45 | bkw_ | lalala |
| 14:04.22 | xachen | mafkees: er :p |
| 14:05.06 | mafkees | I tried on 3 different internet connections |
| 14:05.16 | xachen | it would be their servres if its a 400 |
| 14:05.18 | mafkees | and also tried OpenBSD, macosx and debian |
| 14:05.27 | mafkees | so it's not my client |
| 14:05.51 | xachen | of course it woudln't be |
| 14:06.02 | *** join/#asterisk pixolex (n=chatzill@87-196-155-139.net.novis.pt) |
| 14:06.15 | The_Isle_of_Mark | anyone have any experience with draytek routers with fxs? |
| 14:08.28 | mafkees | hhmm, all the digium folks are sleeping ? |
| 14:13.47 | Darwin35 | its the weekend they are off work |
| 14:13.53 | *** join/#asterisk lokkju (n=lokkju@unaffiliated/lokkju) |
| 14:13.57 | Darwin35 | they will be back monday |
| 14:14.10 | The_Isle_of_Mark | wait wait wait...people get weekends off of work? |
| 14:14.31 | Darwin35 | yes |
| 14:14.36 | mafkees | wow |
| 14:14.47 | The_Isle_of_Mark | I gotta get a new job...sheesh |
| 14:14.47 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.237.93.Dial1.SanJose1.Level3.net) |
| 14:14.52 | The_Isle_of_Mark | weekends off |
| 14:15.33 | Darwin35 | heck I got a month off its nice |
| 14:15.38 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.237.93.Dial1.SanJose1.Level3.net) |
| 14:16.13 | The_Isle_of_Mark | rub it in...rub it in |
| 14:16.43 | Darwin35 | gives me tiime to pplay with my sangoma cards |
| 14:16.49 | key2 | !seen kran |
| 14:16.51 | key2 | !seen kram |
| 14:17.14 | Darwin35 | I think jbot is asleep |
| 14:18.01 | key2 | ~seen kram |
| 14:18.02 | jbot | kram <n=mark@pdpc/sponsor/digium/kram> was last seen on IRC in channel #asterisk, 5d 11h 11m 23s ago, saying: 'oh most certainly :)'. |
| 14:18.11 | key2 | Darkhalf :) |
| 14:18.21 | key2 | Darwin35:) |
| 14:18.28 | Darwin35 | Darkhalf who is that |
| 14:18.44 | Darwin35 | thnks |
| 14:18.53 | coppice | Lighthalf's evil twin |
| 14:18.55 | key2 | dunno but there is one otherwise my nick compl wouldnt have hanswered |
| 14:19.33 | key2 | am looking for a cheap middle east SIP provider |
| 14:20.26 | Darwin35 | they have sip service over there |
| 14:20.37 | key2 | dunno |
| 14:20.41 | Darwin35 | here I htoought is was the us and canada only |
| 14:20.44 | The_Isle_of_Mark | I think I am gonna get rid of this draytek..it doesn't seem to work for standard outbound calls... |
| 14:20.45 | key2 | that's why I ask for one that the prices are good |
| 14:21.06 | *** join/#asterisk A-Tuin (n=a-tuin@antz.me.uk) |
| 14:22.15 | mafkees | Darwin35: there are SIP providers in a lot of countries, not only us and canada |
| 14:23.05 | Darwin35 | I was yoking |
| 14:23.47 | *** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com) |
| 14:24.51 | *** join/#asterisk dokhench (n=dochench@adsl-065-080-180-134.sip.bna.bellsouth.net) |
| 14:27.15 | Darwin35 | its the sip things in life |
| 14:32.46 | *** join/#asterisk robl^ (n=robl@dsl093-025-118.hou1.dsl.speakeasy.net) |
| 14:33.27 | *** join/#asterisk brif8 (n=chatzill@rrcs-71-41-50-162.se.biz.rr.com) |
| 14:33.53 | brif8 | I have d/l zaptel 1.2.5 and it's patch how do install the patch and compile zaptel ? |
| 14:37.46 | brif8 | is the patch already applied to the zaptel-1.2.5.tar.gz file ? |
| 14:38.00 | mafkees | btw, the svn thing was my fault |
| 14:38.04 | mafkees | stupid squid :) |
| 14:39.22 | dokhench | question for you guys... have a call come in on a zap line, dials a polycom sip channel.. that polycom then transfers to another polycom(both are sip 1.6.2).. the second polycom is ringing but caller on the zap channel hears no ringing.. even tried specifying the generate ringing on the dial command.. problem is with polycoms or * dialplan? |
| 14:39.36 | Darwin35 | well when you let sealife do the job there are bound to be issues |
| 14:42.01 | brif8 | I have d/l zaptel 1.2.5 and it's patch how do install the patch and compile zaptel or is the patch already applied ? |
| 14:42.58 | Darwin35 | man patch |
| 14:43.17 | *** join/#asterisk techman97_andy (n=me@70-98-20-60.dsl1.rsm.mn.frontiernet.net) |
| 14:43.25 | Darwin35 | and if you pull svn head ver the patch shold be attached |
| 14:43.40 | *** join/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 14:43.44 | Darwin35 | not attached but already included |
| 14:44.08 | Darwin35 | what patch are you dealing with anyway ? |
| 14:45.51 | brif8 | I d/l from ftp.digium not svn. I got the zaptel.1.2.5 and zaptel.1.2.5-patch files |
| 14:47.10 | brodiem | brif8, you don't need both. |
| 14:47.29 | brodiem | brif8, if you're upgrading prev src, use the patch. Otherwise, just the 1.2.5 tarball |
| 14:47.56 | brif8 | brodiem: ok thanks |
| 14:48.34 | Darwin35 | the patch file is to patch 1.2.4 to mak eit 1.2.5 |
| 14:48.55 | brif8 | I see I'm new to patches obviously thanks |
| 14:49.58 | dokhench | question for you guys... asterisk is 1.2.5.. I have a call come in on a zap line, dials a polycom sip channel.. polycom answers and chats with zap line. that polycom then transfers to another polycom(both are sip 1.6.2).. the second polycom is ringing but caller on the zap channel hears no ringing.. even tried specifying the generate ringing on the dial command.. problem is with polycoms or * dialplan? |
| 14:51.07 | Darwin35 | dok yoou asked that if no one answer then no one knows |
| 14:51.18 | Darwin35 | dont repeat a question |
| 14:51.30 | dokhench | darwin35: thought maybe i didn't specify enough info.. that why i added the 1.2.5 |
| 14:51.40 | Darwin35 | if some one knows they will answer |
| 15:00.44 | *** join/#asterisk Igg-man (n=kc0itq@rrcs-67-53-20-210.west.biz.rr.com) |
| 15:01.28 | *** join/#asterisk hgaillac (n=Harry@196.18.119-80.rev.gaoland.net) |
| 15:02.24 | hgaillac | using http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+insecure |
| 15:03.17 | hgaillac | what does it mean "insecure=invite ; Do not require authentication of incoming INVITEs " is there a bug ? |
| 15:05.52 | *** join/#asterisk bkw_ (n=brian@c-68-32-112-142.hsd1.md.comcast.net) |
| 15:06.00 | *** join/#asterisk AvoidingDeadlock (n=brian@c-68-32-112-142.hsd1.md.comcast.net) |
| 15:06.04 | *** part/#asterisk AvoidingDeadlock (n=brian@c-68-32-112-142.hsd1.md.comcast.net) |
| 15:07.54 | *** join/#asterisk key2 (n=key2@gob75-2-81-56-64-17.fbx.proxad.net) |
| 15:08.37 | Rawplayer | RE |
| 15:09.51 | mafkees | wb |
| 15:11.17 | Darwin35 | if he is a raw player it must hurt |
| 15:11.59 | Darwin35 | bad yooke |
| 15:12.03 | Darwin35 | lol |
| 15:14.24 | Rawplayer | hehe |
| 15:15.30 | *** join/#asterisk coppice (n=chatzill@196.197.17.210.dyn.pacific.net.hk) |
| 15:17.01 | *** join/#asterisk docelmo (n=docelmo@55-65.126-70.tampabay.res.rr.com) |
| 15:17.17 | dokhench | question for you guys... asterisk is 1.2.5.. I have a call come in on a zap line, dials a polycom sip channel.. polycom answers and chats with zap line. that polycom then transfers to another polycom(both are sip 1.6.2).. the second polycom is ringing but caller on the zap channel hears no ringing.. even tried specifying the generate ringing on the dial command.. problem is with polycoms or * dialplan? |
| 15:30.23 | asterboy | watch cli with a lot of verbosity and see if it registers the ringing. |
| 15:30.48 | guyb_home | tzafrir: heard from Klaus Peter that the bristuff version before 0.3.0 does not respect rxgain + txgain on out going calls so no way to control echo - what bristuf ver in rapid experimental? |
| 15:30.53 | asterboy | could have something to do with the way you forward. |
| 15:31.39 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 15:35.43 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-175-120.lsanca.fios.verizon.net) |
| 15:38.20 | *** join/#asterisk X-Gen (n=x-gen@dsl-145-233-253.telkomadsl.co.za) |
| 15:39.17 | dokhench | asterboy: doing transfer, dial extension, then transfer again |
| 15:41.31 | asterboy | watch clie |
| 15:41.48 | asterboy | s/e// |
| 15:41.50 | *** join/#asterisk tdonahue-laptop (n=tdonahue@231.212.118.66.brainstorminternet.net) |
| 15:44.04 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
| 15:45.48 | tdonahue-laptop | morning all |
| 15:46.16 | tdonahue-laptop | anyone have any idea why Milliwatt works but Playback doesn't on my asterisk box? |
| 15:47.44 | *** join/#asterisk demigod2k (n=joey@cpe-24-210-97-162.twmi.res.rr.com) |
| 15:50.47 | *** join/#asterisk mogorman (n=mogorman@68.62.237.103) |
| 15:53.12 | drray | what format? |
| 15:53.22 | drray | does Record work? |
| 15:53.30 | tecnico | any idea on what this really means: "WARNING[1374]: chan_iax2.c:7552 socket_read: Received mini frame before first full voice frame" |
| 15:53.31 | *** part/#asterisk guyb_home (n=guy@115.251-7-195.ippool.ndo.com) |
| 15:59.08 | *** join/#asterisk salviadud (n=ralfalfa@201.135.13.124) |
| 15:59.37 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 16:00.40 | Hmmhesays | i'm alive |
| 16:00.57 | salviadud | no kiddin' dude |
| 16:00.59 | tdonahue-laptop | drray, the default files included with Asterisk, record should work, as I have verified the audio stream is reaching the server |
| 16:01.05 | tdonahue-laptop | congrats Hmmhesays |
| 16:01.48 | Hmmhesays | now its time to go through the phone numbers I got last night |
| 16:01.51 | Hmmhesays | and decide who to call |
| 16:02.07 | salviadud | you gonna do some prank calling? |
| 16:02.11 | Hmmhesays | hell no |
| 16:02.15 | salviadud | mixmonitor be your friend |
| 16:02.21 | Hmmhesays | wimmins dude, wimmins |
| 16:02.31 | salviadud | those kind of phone numbers |
| 16:02.35 | tdonahue-laptop | Hmmhesays, sounds like a real tough day you have planned for yourself |
| 16:02.36 | salviadud | i'd still prank them man |
| 16:02.54 | Hmmhesays | tdonahue-laptop saturday is my day off man |
| 16:03.01 | salviadud | that reminds me |
| 16:03.12 | salviadud | i got the phone number from this girl at school |
| 16:03.14 | tdonahue-laptop | not for me this weekend... |
| 16:03.26 | salviadud | i don't want to call her |
| 16:03.31 | Hmmhesays | i will |
| 16:03.54 | salviadud | you seem like a alpa-male desperado |
| 16:04.11 | Hmmhesays | um, is that good or bad? |
| 16:04.36 | salviadud | being an alpha-male is good, despeerado, not so good |
| 16:04.44 | *** part/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 16:04.47 | Hmmhesays | well i'm not desperate |
| 16:05.12 | salviadud | desperado and desperate are different |
| 16:05.29 | salviadud | desperate and desesperado are the same |
| 16:05.43 | Hmmhesays | desperado isn't bad, mean by yourself or loner type |
| 16:06.25 | salviadud | anyways man. it's just my opinion |
| 16:06.36 | salviadud | you do what you gotta do bro |
| 16:06.40 | Hmmhesays | lol |
| 16:09.07 | ManxPower | linksys.com seems to be down |
| 16:11.59 | salviadud | would anyone be interested in bullfrog venom? |
| 16:12.07 | Hmmhesays | hola |
| 16:12.10 | salviadud | i'd like to TALK about it |
| 16:12.14 | Hmmhesays | bat country by avenged sevenfold rocks |
| 16:13.40 | *** join/#asterisk hgaillac (n=Harry@196.18.119-80.rev.gaoland.net) |
| 16:13.43 | hgaillac | hello |
| 16:14.32 | *** join/#asterisk Kokey (n=jramirez@dsl-200-78-65-27.prod-infinitum.com.mx) |
| 16:14.47 | hgaillac | anybody could dial sip:info@nxs.yi.org or sip:music@nxs.yi.org for testing , thanks |
| 16:16.03 | tdonahue-laptop | hmm... anyone have any thoughts on why my TE110P was breaking my audio from Playback? |
| 16:17.28 | x86 | someone give me a non-US phone number, any valid phone number in the world... i want to test my call costing application |
| 16:18.55 | demigod2k | 011-44-190-282-4051 |
| 16:19.55 | Cybertoy | is that a toll number? |
| 16:20.46 | demigod2k | dont know what you mean by a toll number |
| 16:21.35 | Cybertoy | like a -1900 number |
| 16:22.09 | demigod2k | noclue just found it on the web to answer his question |
| 16:22.14 | demigod2k | tech support for some company |
| 16:22.33 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 16:22.40 | *** join/#asterisk pixolex (n=chatzill@87-196-215-20.net.novis.pt) |
| 16:23.31 | Cybertoy | x86, you can also try 011 41 32 511 2446 .... goes into my telemarketer feature.. swiss non-toll number.. :) |
| 16:24.34 | x86 | dont need the 011 part :) |
| 16:24.35 | x86 | thanks |
| 16:24.56 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 16:25.09 | x86 | $0.019 USD per minute to there |
| 16:25.31 | Cybertoy | sounds about right |
| 16:25.45 | Cybertoy | with voipdiscount.com it's free ;) |
| 16:25.53 | Cybertoy | at least for now ... |
| 16:28.09 | hgaillac | IS IT POSSIBLE TO DISABLE ASTERISK PROXY AUTHENTICATION YES OR NO ? |
| 16:28.20 | macTijn | DONT SCREAM |
| 16:28.29 | macTijn | WE CAN READ YOU LOUD AND CLEAR!\ |
| 16:28.35 | mogorman | what is asterisk proxy authentication? |
| 16:28.42 | file[laptop] | hgaillac: I think from the impression you've made, nobody is going to answer or take you seriously |
| 16:29.03 | mogorman | hey file how did you get to be an op in #asterisk |
| 16:29.07 | mogorman | but not me :( |
| 16:29.16 | file[laptop] | mogorman: I have magical powerz |
| 16:30.09 | PakiPenguin | there all powers gone! |
| 16:30.29 | tzafrir_laptop | A shreder is not good enough for it |
| 16:30.32 | *** join/#asterisk tecnico (n=tecnico@user-24-236-120-2.knology.net) |
| 16:31.07 | PakiPenguin | hehe tzafrir_laptop , magical shreder that is ;) |
| 16:31.12 | hgaillac | mogorman : 21.4.8 407 Proxy Authentication Required |
| 16:31.13 | hgaillac | <PROTECTED> |
| 16:31.13 | hgaillac | <PROTECTED> |
| 16:31.13 | hgaillac | <PROTECTED> |
| 16:31.13 | hgaillac | <PROTECTED> |
| 16:31.13 | hgaillac | <PROTECTED> |
| 16:31.15 | hgaillac | <PROTECTED> |
| 16:31.25 | x86 | ugh... |
| 16:31.31 | mogorman | you have the option to allow guest calls |
| 16:31.50 | mogorman | hgaillac, you are breaking several rules of conduct for this channel |
| 16:32.02 | mogorman | please stop it |
| 16:32.09 | file[laptop] | if the user portion of the From header matches a user entry though, it'll send back a 407 |
| 16:33.21 | tzafrir_laptop | So if I want to get calls through a trunk that may send an arbitrary user in the From header: I have a problem? |
| 16:33.42 | file[laptop] | users get priority over peers |
| 16:33.46 | hgaillac | file: thank i read it but asterisk is not a proxy |
| 16:33.48 | file[laptop] | for inbound matching, so yes |
| 16:33.54 | tzafrir_laptop | are peers considered at all? |
| 16:34.01 | file[laptop] | hgaillac: your point? |
| 16:34.08 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
| 16:34.12 | file[laptop] | tzafrir_laptop: yes |
| 16:34.47 | tzafrir_laptop | so if I have a peer with no secert line, is that a potential hole? |
| 16:35.06 | tzafrir_laptop | Or am I missing something? |
| 16:35.15 | file[laptop] | it takes more then that to get a peer entry to match |
| 16:35.30 | file[laptop] | but you can have no password entered... |
| 16:36.12 | hgaillac | try to call sip:info@nxs.yi.org |
| 16:36.59 | *** join/#asterisk luckyduck (i=lucky@gentoo/developer/luckyduck) |
| 16:37.07 | file[laptop] | hgaillac: if you want to accept any random person, then allow guest calls |
| 16:37.19 | file[laptop] | and send them to a context that only allows them to call a few places |
| 16:37.19 | tzafrir_laptop | In one place I pointed the calls from the guest user to a context where I checked the IP of the incoming call (assuming it's not as easy to forge as headers from the packet) and in case of a match to the specific trunk, sent it onwards. |
| 16:37.27 | tzafrir_laptop | But I consider this lame |
| 16:37.57 | hgaillac | file could you send me an example to gaillacharry@yahoo.fr please |
| 16:38.06 | file[laptop] | allowguest=yes |
| 16:38.10 | file[laptop] | context=inbound_public |
| 16:38.13 | file[laptop] | in [general] |
| 16:38.15 | file[laptop] | that's it... |
| 16:38.16 | Hmmhesays | file will you make me some coffee |
| 16:38.27 | file[laptop] | Hmmhesays: never! |
| 16:38.28 | mogorman | woohoo file[laptop] |
| 16:38.37 | Hmmhesays | fine |
| 16:39.05 | file[laptop] | I'm hungry though... so I think I'll go make food soon |
| 16:39.42 | tzafrir_laptop | Basically I assumed that the name of the channel must include the IP address, and parsed things from there. I don't have it with me he. I'll try to look it up |
| 16:40.03 | file[laptop] | tzafrir_laptop: interesting... |
| 16:44.14 | hgaillac | allowguest = yes no succes !!! |
| 16:44.27 | file[laptop] | then you do a sip debug and figure out what it's doing |
| 16:45.26 | techman97_andy | hey all - I'm trying to get a SIP phone to dial out through two ZAP channels (not at the same time...) I have the extensions.conf file set as far as I can tell, but when I dial, the CLI says it cannot create a ZAP channel. If I do a ZAP SHOW CHANNELS, I see both channels, but a ZAP SHOW STATUS shows both channels with a RED alarm. any ideas of what I can check on next? |
| 16:46.51 | techman97_andy | one more thing quick - the CLI says, "cause 0 - Unknown" in the error message |
| 16:47.35 | techman97_andy | -- Executing Dial("SIP/2000-df17", "Zap/1/6128452134") in new stack |
| 16:47.35 | techman97_andy | Apr 8 11:46:28 NOTICE[1735]: app_dial.c:1010 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) |
| 16:47.59 | asterboy | ~cli |
| 16:48.00 | jbot | extra, extra, read all about it, cli is a Command Line Interface, the best form of interface around, of course Call-Level Interface, originally developed by SQL Access Group, but now known as Microsoft's ODBC standard. an x86 assembly instruction Common Language Infrastructure (See mono or .net) |
| 16:48.13 | file[laptop] | techman97_andy: it would be helpful if you told us what type of zap channels |
| 16:49.00 | techman97_andy | the zap channels are a pair of x100P wildcards |
| 16:49.32 | hgaillac | sip debug does not provide help anyway i can't receice calls from other domains |
| 16:49.54 | file[laptop] | hgaillac: yes it will, it'll tell you what Asterisk is doing - whether it's matching to a user entry, or what |
| 16:50.56 | hgaillac | sip debug would help me if somebody call me ! |
| 16:51.13 | file[laptop] | it's not our obligation to call you |
| 16:51.25 | wasim | not unless you are a phone sex line |
| 16:51.28 | *** join/#asterisk guyb_home (n=guy@115.251-7-195.ippool.ndo.com) |
| 16:51.40 | wasim | and a hot blonde (or atleast sound like one) |
| 16:51.46 | hgaillac | you could listen music at sip:music@nxs.yi.org |
| 16:51.50 | techman97_andy | hahaha |
| 16:52.32 | techman97_andy | BLARG! 3 hours of trying things...the phone line was dead...WOW do I feel stupid. |
| 16:52.33 | *** join/#asterisk Evanrude (n=david@ip68-107-162-212.lu.dl.cox.net) |
| 16:52.53 | asterboy | stupid is what stupid does |
| 16:52.56 | techman97_andy | aye aye |
| 16:54.47 | *** join/#asterisk fender21 (n=fender21@cpe-66-69-90-91.satx.res.rr.com) |
| 16:55.17 | fender21 | Anyone around to answer a few questions on Asterisk? |
| 16:55.49 | Rawplayer | dunno |
| 16:56.18 | asterboy | nope nobody |
| 16:56.38 | asterboy | this is just a place for gay sex chat |
| 16:56.53 | fender21 | like a brokeback mountain pbx kind of thing? |
| 16:56.55 | asterboy | lol |
| 16:57.20 | fender21 | heh.. |
| 16:57.27 | techman97_andy | wow |
| 16:58.09 | tzafrir_laptop | techman97_andy, do those channels show up on 'zap show channels'? |
| 16:58.37 | techman97_andy | I figured it out tzafrir....amazing what works when you ensure you have PSTN dial tone coming through the line....*ANGRY SCREAM* |
| 16:58.45 | fender21 | I've spent most of this week reading up on Asterisk and I'm still a bit overwhelmed. I'm looking for a bit of brokeback guidance |
| 16:58.52 | dokhench | lol |
| 16:59.13 | dokhench | what question(s) fender? |
| 16:59.13 | *** join/#asterisk apardo (n=apardo@87.218.45.206) |
| 16:59.39 | fender21 | I'm looking to create an audioblog and want to use Asterisk to do it. |
| 16:59.54 | asterboy | phone diary |
| 16:59.55 | fender21 | answer the phone, key in a unique number, record an mp3, ftp it to a server |
| 17:00.03 | dokhench | ah |
| 17:00.04 | mafkees | fender21: ah cool, I'm thinking about that too |
| 17:00.07 | techman97_andy | here's one though - I remember having this issue last time I built a * box, but can't remember what I did. When I call an outside line from a SIP client through the x100p, I can hear the PSTN caller just fine, but my audio to them is really choppy for regular conversation, but if I just talk a long time - it gets better...but then drops off again when I stop talking. |
| 17:00.18 | dokhench | i was boggled for a moment at what exactly an "audio blog" would be. |
| 17:00.27 | fender21 | great mafkees.. |
| 17:00.32 | mafkees | fender21: I have this blogtool project |
| 17:00.36 | file[laptop] | techman97_andy: silence suppression enabled? |
| 17:00.44 | tzafrir_laptop | fender21, do consider the quality, though |
| 17:00.45 | techman97_andy | file: in the SIP client? |
| 17:00.49 | mafkees | fender21: and already have a ticket about creating audio entries |
| 17:00.51 | file[laptop] | techman97_andy: yes |
| 17:00.53 | techman97_andy | (which is xLite for testing here) |
| 17:00.58 | techman97_andy | k - lemme check |
| 17:01.04 | file[laptop] | set "transmit silence" to yes |
| 17:01.25 | tzafrir_laptop | Asterisk works with telephony-quality voice streams, mostly: 8khz, mono |
| 17:01.46 | tzafrir_laptop | Is that good enough for you? |
| 17:01.48 | mafkees | tzafrir_laptop: that format is always used ? |
| 17:01.51 | fender21 | Mafkees: what do you mean by ticket (forgive the ignorance) |
| 17:02.01 | *** join/#asterisk Dovid (n=Dovid@62.0.153.54) |
| 17:02.04 | mafkees | fender21: 'Request for new feature' |
| 17:02.19 | fender21 | oh..can't we do it on our own? |
| 17:02.31 | mafkees | tzafrir_laptop: if I use iax2 and set both ends to use g711A |
| 17:02.42 | mafkees | tzafrir_laptop: will it use 8khz mono too ? |
| 17:02.55 | tzafrir_laptop | mafkees, several. Internally 16-bit signed linear PCM is mostly used, I believe. If g711A is good enough for you, then fine |
| 17:03.02 | mafkees | fender21: yes, the ticket is on my blog development website. as feature for the blogtool |
| 17:03.31 | fender21 | cool! |
| 17:03.45 | mafkees | tzafrir_laptop: gheh, yeah, the quality will be medium, but the toy will be fun |
| 17:03.47 | Rawplayer | anyone in here using voipbuster? |
| 17:04.08 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 17:04.16 | mafkees | Rawplayer: yes me, but only for outgoing calls |
| 17:04.24 | fender21 | mafkees: what's your site? |
| 17:04.47 | mafkees | fender21: http://michiel.vanbaak.eu |
| 17:05.02 | mafkees | fender21: and the blog is on: http://www.mvblog.org |
| 17:05.07 | Rawplayer | mafkees: can you call without any echo problems? |
| 17:05.34 | mafkees | Rawplayer: it works fine here. My wife is not complaining, so I guess it's really ok |
| 17:05.57 | Rawplayer | haha |
| 17:06.02 | Rawplayer | yes it is then :) |
| 17:06.07 | techman97_andy | tzafrir: Thanks! that worked like a charm! |
| 17:06.20 | Rawplayer | then i think my soundcard is trashy |
| 17:06.24 | Rawplayer | its on a old laptop |
| 17:06.25 | techman97_andy | I'm on the phone right now...=P |
| 17:06.49 | fender21 | mafkees: You might want to check out this post on the Asterisk forum http://forums.digium.com/viewtopic.php?t=2821&highlight=record+mp3 |
| 17:06.56 | mafkees | Rawplayer: using a headset? or just the builtin speaker/mic ? |
| 17:07.28 | mafkees | fender21: I will be using .ogg |
| 17:07.41 | Rawplayer | using a headset |
| 17:08.34 | tzanger | hey does anyone here use polycoms and ftp provisioning? |
| 17:08.51 | tzanger | I've got my DHCP server returning a server that the polycom is seeing, but it's not parsing it correctly (which means I am not giving it the right format) |
| 17:08.51 | fender21 | can anyone recommend a cheap VOIP SIP service that will be easy for me to setup in Asterisk? I'm pretty clueless on SIP. |
| 17:09.08 | tzanger | I thought I could just return option 66 as "ftp://user:pass@ip.of.ftp.server/" |
| 17:09.16 | tzanger | the phone is definitely seeing that but it's obviously wrong :-) |
| 17:12.04 | *** join/#asterisk theorem_ (n=theorem@pool-71-251-210-104.nwrknj.fios.verizon.net) |
| 17:13.47 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 17:13.51 | tzanger | damn where's [tk]fender when you need him :-) |
| 17:15.41 | fender21 | Is that my brother? |
| 17:15.56 | *** join/#asterisk _Thor (i=CS@user-vc8fl7l.biz.mindspring.com) |
| 17:16.03 | *** join/#asterisk Kokey (n=jramirez@dsl-200-78-65-27.prod-infinitum.com.mx) |
| 17:16.08 | tzanger | fender21: heh |
| 17:16.14 | Ariel_ | fender21, cheap, easy and godd normally don't go together |
| 17:16.29 | Ariel_ | tzanger, let me check my ftp settings I deploy polycoms here |
| 17:16.37 | fender21 | I'm just trying to get a test rig up and running.. :-) nothing production yet |
| 17:16.51 | tzanger | Ariel_: you are a GODSEND |
| 17:17.17 | Ariel_ | tzanger, the only thing is we use a dhcp window$ box....(argh) |
| 17:17.53 | tzanger | Ariel_: that's fine, I've got the option66 being returned, what do you have for the value of option 66 |
| 17:19.15 | Ariel_ | damm pastebin is slow today |
| 17:20.23 | techman97_andy | fender21: I just signed up with VoiceEclipse - residential unlimited $20/mo. They allow you to log in with a SIPI username and password |
| 17:21.25 | fender21 | techman97_andy: Thanks for the information! If I go with this, I will be able to make incoming and outgoing calls to my Asterisk server (once I setup the config)? Is that right? |
| 17:22.25 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 17:23.19 | tzanger | Ariel_: pastebin.ca is generally much quicker |
| 17:23.33 | Ariel_ | tzanger, yes that is what I use. |
| 17:23.36 | Ariel_ | http://pastebin.ca/48643 |
| 17:23.45 | Ariel_ | it was my server not theres |
| 17:24.03 | tzanger | ahh okay you just give it a tftp-server-name |
| 17:24.07 | Ariel_ | but I just put the options for tftp not ftp which we actually have a ftp |
| 17:24.15 | Ariel_ | works |
| 17:24.34 | tzanger | hmm, interesting |
| 17:24.34 | ManxPower | few companies with unlimited plans allow you to use 3rd party hardware |
| 17:24.38 | *** join/#asterisk A-Tuin (n=a-tuin@antz.me.uk) |
| 17:24.39 | Dovid | yea |
| 17:24.49 | Ariel_ | for some reason we could not get it working if we put ftp instead of tftp |
| 17:24.55 | Dovid | most companies that have unlimited with ur hardware ususally equal crap |
| 17:25.01 | Dovid | exhibit A = Braodvoice |
| 17:25.24 | Ariel_ | voicepulse has been good to me |
| 17:25.31 | Ariel_ | also race.com |
| 17:25.54 | fender21 | Unlimited is not really a factor for the testing purpose.. I'm looking for good quality and price |
| 17:26.01 | Dovid | voicepulse is unlimited ? |
| 17:26.18 | Ariel_ | Dovid, they have both |
| 17:26.23 | Dovid | kk |
| 17:26.26 | Dovid | will look in to them |
| 17:26.26 | Ariel_ | unlimited is sip |
| 17:26.42 | Ariel_ | iax2 is what I use mainly since we dont really pass 10 dollars amonth on ld |
| 17:26.58 | Cybertoy | broadvoice has been good for me |
| 17:27.06 | Ariel_ | sometimes unlimited really does not make sence |
| 17:27.09 | Dovid | hehe |
| 17:27.21 | Dovid | braodvoice customer service is non existant |
| 17:27.27 | Dovid | as well as the voice quality |
| 17:27.28 | Ariel_ | Cybertoy, I have bv as well But they are not great for voice quaility and do tend to go down |
| 17:27.34 | Dovid | half the time dtmf dosent go thru |
| 17:27.51 | Cybertoy | customer service I agree that they suck. but I hardly need them |
| 17:28.06 | Cybertoy | never went down for me since May last year |
| 17:28.18 | Cybertoy | and I make many international calls to Switzerland and Brazil. |
| 17:28.25 | Cybertoy | esp Brazil ... |
| 17:28.40 | Cybertoy | so far we're happy. |
| 17:28.50 | Cybertoy | voipdiscount.com is a good backup at the moment. |
| 17:29.03 | Cybertoy | they're not unlimited... but free.. :) |
| 17:30.08 | fender21 | Cybertoy, how's the quality on voipdiscount.com? |
| 17:30.16 | Cybertoy | excellent. |
| 17:30.27 | Cybertoy | better than broadvoice for some destinations. |
| 17:30.30 | brodiem | i use voxee.com @1.1cpm with sixtel as backup and origination |
| 17:31.27 | salviadud | is voipdiscount sip? |
| 17:31.28 | PakiPenguin | voxee supports ANI brodiem ? |
| 17:31.38 | fender21 | Cybertoy: so are you charged per incoming call? |
| 17:32.17 | Cybertoy | fender, I have a voip-in number from voipbuster.com .. and there they don't charge me. |
| 17:32.28 | Cybertoy | fender, but I don't think they have US based did's... |
| 17:32.49 | *** join/#asterisk angom_h (n=angom@red-corp-201.130.165.94.telnor.net) |
| 17:32.52 | brodiem | PakiPenguin hmm I'm not sure |
| 17:33.26 | PakiPenguin | brodiem, can you send your own cli? |
| 17:33.29 | salviadud | i use voipbuster |
| 17:33.36 | salviadud | works ok |
| 17:33.37 | fender21 | Cybertoy: Do you know if voipdiscount.com is SIP? |
| 17:33.45 | Cybertoy | fender, yes... they are. |
| 17:33.48 | salviadud | is voipdiscount the same tyep of service Cibertoy? |
| 17:33.56 | fender21 | hmm..interesting! |
| 17:33.58 | Cybertoy | savia, yes.. they're the same company. |
| 17:34.03 | brodiem | PakiPenguin, oh.. I can't set a CID name with voxee but I can with sixtel |
| 17:34.09 | salviadud | same company... wow |
| 17:34.12 | Cybertoy | you only need to download their client to register... after that you can use SIP. |
| 17:34.18 | PakiPenguin | i know |
| 17:34.26 | salviadud | yeah, the same trick... |
| 17:34.36 | salviadud | so now i can prank call germany |
| 17:34.41 | salviadud | are the calls really free? |
| 17:34.48 | salviadud | or just super cheap? |
| 17:35.01 | Cybertoy | on voipdiscount.com I never charged a cent to my account. |
| 17:35.14 | Cybertoy | they don't even disconnect after 1 minute although that what it says on their page. |
| 17:35.23 | Cybertoy | so that's pretty free for me.. :) |
| 17:36.05 | salviadud | ^_^ prank calls here i come |
| 17:36.29 | Cybertoy | another option is fwdOUT.net ... |
| 17:36.50 | salviadud | mmm, let me see |
| 17:37.20 | fender21 | Cybertoy: did you order an inbound number from voipdiscount.com |
| 17:37.32 | Cybertoy | fender, no ... |
| 17:38.02 | Cybertoy | fender, but I did with voipbuster... in order to get an incoming number you need to put money on your account... |
| 17:38.20 | *** join/#asterisk dovid (n=Dovid@62.0.153.54) |
| 17:38.36 | Ariel_ | so how do you get voipdiscount to call via your asterisk box |
| 17:38.55 | fender21 | Ariel: That's what I was wondering :-) I'm so lost. |
| 17:39.11 | dovid | what do u need help with ? |
| 17:39.34 | fender21 | Dovid: I might be beyond help :-) I'm just soaking in all of the knowledge in this room |
| 17:39.38 | salviadud | should be easy |
| 17:39.40 | Ariel_ | voipdiscount says you need there software for windows ot work |
| 17:39.41 | salviadud | it's a sip channel |
| 17:39.44 | dovid | hehe |
| 17:39.47 | salviadud | thats BS |
| 17:39.50 | salviadud | you need it to register |
| 17:39.53 | dovid | hehe |
| 17:40.04 | salviadud | i wonder what's the syntax |
| 17:40.05 | dovid | do they give u the sip info to register with them ? |
| 17:40.11 | salviadud | no... |
| 17:40.13 | salviadud | you gotta hack it |
| 17:40.19 | salviadud | it should be on the wiki |
| 17:40.22 | salviadud | maybe... |
| 17:40.46 | dovid | salviadud: seems like they have thier own software and dont give out thier sip info |
| 17:40.57 | salviadud | of course they don't |
| 17:41.00 | salviadud | they're commercial |
| 17:41.38 | dovid | http://www.voipdiscount.com/en/index.html |
| 17:41.40 | salviadud | you could probably find out by placing a call with their client and run iptraf or ethereal |
| 17:41.51 | fender21 | Cybertoy: Thanks for all the info.. I would go with voipbuster.com but they don't have US numbers like you said. |
| 17:41.55 | dovid | yea, but against terms if u care about legalities |
| 17:42.05 | dovid | and i cant see em giving u a free did |
| 17:42.06 | salviadud | no i don't care about legalities |
| 17:42.08 | salviadud | i am from mexico |
| 17:42.16 | dovid | ;0 |
| 17:42.17 | dovid | :) |
| 17:42.20 | fender21 | Ariel_: If I go with voicepulse.com, I sign up and they give me the SIP info? Correct? |
| 17:42.28 | dovid | isnt voip ilegal in mexico ? |
| 17:42.51 | salviadud | it's not even regulated |
| 17:42.59 | Ariel_ | fender21, yes but read there web site. See which option you want and if your using asterisk you need the byod setup |
| 17:43.37 | Cybertoy | sorry.. had to step away... anyway... for voipdiscount you need their client to register... but after that you can use sip.voipdiscount.com ... |
| 17:43.38 | fender21 | Ariel_: Great, thank you! Once I have this SIP info, I can go into AMP and add them as a Trunk? |
| 17:44.57 | salviadud | Cybertoy, is the syntax the same as voipbuster.com? |
| 17:45.01 | brodiem | fender21, yes |
| 17:45.04 | Cybertoy | salvia, yes. |
| 17:45.12 | salviadud | were as you dial 00+countrycode.. blablabla |
| 17:45.20 | fender21 | brodiem, thanks! |
| 17:45.27 | Cybertoy | salviadud, exactly. |
| 17:45.42 | salviadud | alright |
| 17:48.11 | The_Isle_of_Mark | I keep asking occasionally in hopes someone does: Anyone here have any experience with Draytek voip routers? |
| 17:48.32 | Cybertoy | no |
| 17:53.05 | Ariel_ | here are the settings to get voipdiscount to work from asterisk and amp.. http://nerdvittles.com/index.php?p=127 |
| 17:53.58 | Ariel_ | The_Isle_of_Mark, I have never heard of draytek routers |
| 17:54.38 | *** join/#asterisk redcap1 (n=phez@redcap.xs4all.nl) |
| 17:55.09 | *** join/#asterisk MoutaPT (n=MoutaPT@85.139.183.36) |
| 17:56.29 | MoutaPT | Hi all, i'm trying to get mysql cdr working, php4-mysql already installed and lots of other packages but i still get mysql php Libraries not installed, i'm using debian |
| 17:56.33 | Katty | mew. |
| 17:56.39 | fender21 | Can anyone tell me what ZAP/g0 is reffering to in my trunk setup? |
| 17:56.39 | MoutaPT | any tip? |
| 17:56.43 | Qwell | MoutaPT: it shouldn't be using php at all... |
| 17:56.54 | Qwell | fender21: a group |
| 17:57.46 | fender21 | like a piece of hardware? It seems like it was there as a default |
| 17:58.05 | Ariel_ | fender21, it's a pstn trunk if your have it setup zap/1 actually on a amp or a@h setup |
| 17:58.22 | Ariel_ | fender21, yes it's a sample |
| 17:58.39 | fender21 | Ariel: perfect! THanks for the info, you guys are great |
| 17:59.28 | *** part/#asterisk guyb_home (n=guy@115.251-7-195.ippool.ndo.com) |
| 18:04.32 | *** join/#asterisk lo7k (n=lo7k@p5495168D.dip0.t-ipconnect.de) |
| 18:06.02 | lo7k | hi, does call parking work in current stable? i can't get it working. |
| 18:06.15 | ManxPower | lo7k, tes |
| 18:06.19 | ManxPower | yes |
| 18:06.27 | techman97_andy | hey all - if I'm looking to setup a traditional autoattendant with * for my office - can anyone give me a link or some advice on where to start? |
| 18:06.42 | Cybertoy | lo7k, works for me. |
| 18:06.55 | Ariel_ | techman97_andy, look at the sample files provided by asterisk |
| 18:07.09 | Ariel_ | ~docs |
| 18:07.10 | jbot | somebody said docs was probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf 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, or http://www.astmasters.net/howtos.html |
| 18:08.01 | techman97_andy | yeah, I'm pouring through the * WIKI right now and whatnot - just was wondering if anyone knew exactly a link or a good place to look. |
| 18:08.10 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 18:08.14 | *** part/#asterisk tecnico (n=tecnico@user-24-236-120-2.knology.net) |
| 18:08.38 | *** join/#asterisk tecnico (n=tecnico@user-24-236-120-2.knology.net) |
| 18:08.57 | Cybertoy | techman97, that's not easy as every autoattendant is different from the next one. |
| 18:09.12 | lo7k | I looked in the source and see only the info events for my keys but i can't see how the parking functions in res_features.c will be called |
| 18:09.34 | techman97_andy | all I'm looking to do with it is to have the system announce "hello, press 1 for sales, 2 for support, and 3 for the company directory. Press 0 for the receptionist"...fairly basic. |
| 18:09.51 | Cybertoy | lo7k, make sure you have "parkedcalls" included in the dial-plan. |
| 18:09.58 | techman97_andy | of course, that could be an IVR as well, but I think the dial-by-name that I saw in there was only in the autoattendant? |
| 18:10.40 | lo7k | Cybertoy, there should be a warning message if i forget it but it doesn't get this far |
| 18:11.27 | Cybertoy | techman, easiest if you create a context with s ext that picks up and announces your message and then you have extensions 1, 2, 3, ... in that context. |
| 18:11.48 | techman97_andy | gotcha - that would work. I'll start playing with that. |
| 18:11.58 | Cybertoy | techman97, I don't know what you're talking about. if you don't include the parkedcalls in the dialplan you don't have extension "700" to transfer your call to. |
| 18:12.13 | Cybertoy | techman, what happens if you call extension 700 ? |
| 18:12.41 | techman97_andy | Cyber - I think you have me confused with another question? |
| 18:12.44 | Ariel_ | techman97_andy, the extensions.conf.sample has an ivr predone there very simple for a demo which you can work from. |
| 18:12.56 | techman97_andy | cool! examples are good. thanks Ariel. |
| 18:13.13 | Cybertoy | uh damn... |
| 18:13.15 | Ariel_ | they should be at /usr/src/asterisk/configs/ |
| 18:13.17 | lo7k | Cybertoy, you mean me i guess. it does nothing. transfer works though |
| 18:13.20 | Cybertoy | I was answering the question about the parked calls |
| 18:13.36 | techman97_andy | Cyber: yeah - that's lo7k...=) |
| 18:13.39 | *** join/#asterisk trimi` (i=Whatt@62.162.242.13) |
| 18:13.52 | Cybertoy | lo7k, if you can't call ext 700 then you don't have parkedcalls included in your context. |
| 18:14.03 | fender21 | Cybertoy: I've got voipdiscount.com SIP setup as my trunk and i'm registered.. I registered a 2nd account to call my SIP but it shows it offline? Any suggestions on how to call? |
| 18:14.08 | ManxPower | lo7k, Look in features.conf and "show application dial" |
| 18:14.18 | ManxPower | this is for 1.2 of course |
| 18:15.13 | Cybertoy | fender21, uhm ... I'm not sure I fully understand your question. |
| 18:15.39 | Cybertoy | fender21, so you have an entry in sip.conf [voipdiscount] (or similar) ... |
| 18:15.55 | Cybertoy | fender21, and what's that 2nd account? |
| 18:15.57 | lo7k | ManxPower, i have parkext => 4, every telephone has include => parkedcalls in extensions.conf. if i call 4 nothing happens |
| 18:15.57 | fender21 | Cyber: well I believe I have my voicediscount.com SIP setup correctly in AMP.. but I don't think it's logged into the voicediscount.com server |
| 18:15.59 | ManxPower | Cybertoy, anytime anyone says "trunk" here, it frequently means they are running AMP/Asterisk@Home/FreePBX |
| 18:16.21 | Qwell | ManxPower: /svn trunk |
| 18:16.26 | fender21 | Cyber: I signed up with a 2nd voicediscount.com account to call the 1st one ? so I could test out my system. |
| 18:16.29 | Qwell | depends on context :p |
| 18:16.48 | ManxPower | Qwell, I still think he's running a "GUI" |
| 18:16.55 | Cybertoy | uhm... ok ... I don't know anything about AMP or a@h ... |
| 18:17.00 | Cybertoy | just plain old asterisk here. |
| 18:17.02 | Ariel_ | fender21, they might be letting only one registration from your IP address |
| 18:17.07 | ManxPower | fender21, we can't support AMP here, see the /topic |
| 18:17.23 | Ariel_ | fender21, is running a gui amp or freepbx |
| 18:17.28 | fender21 | oh okay...thanks Manxpower. I'm working my way up to Real Asterisk :-) |
| 18:17.38 | fender21 | gui amp via A@home |
| 18:17.40 | Ariel_ | amp is real asterisk |
| 18:17.43 | Cybertoy | fender21, if you don't have a voip-in number with voipdiscount.com you don't need to register there. |
| 18:18.15 | Cybertoy | nah ... real asterisk is asterisk with vi editor and lots of *.conf files to fool around with .. :) |
| 18:18.21 | fender21 | I don't have a voip-in number.. I'm just looking for a way to call into my asterisks server as I don't have hardware for phone lines |
| 18:18.26 | Ariel_ | Cybertoy, it has that too |
| 18:18.45 | Cybertoy | fender, right... so you have another sip provider that you register with? |
| 18:18.56 | fender21 | cyber: no, just voipdiscount.com |
| 18:19.06 | Cybertoy | fender, ok .. you need to get yourself a DID then... |
| 18:19.14 | Cybertoy | fender, like freedigits.com or stanaphone.com ... |
| 18:19.28 | Cybertoy | fender, they will give you a free phone number in the states. |
| 18:19.59 | Ariel_ | or http://www.kallfree.com/ |
| 18:20.04 | techman97_andy | another question: which conf file has the context name for voicemail? voicemail.conf? |
| 18:20.05 | Ariel_ | also gives you a free did |
| 18:20.14 | Cybertoy | or ipkall.com |
| 18:20.40 | Ariel_ | techman97_andy, it depends on your setup. but yes voicemail.conf would have a context like [default] |
| 18:21.28 | fender21 | cyber: thanks! I just received one from freedigits.com.. so now what? |
| 18:21.58 | Cybertoy | fender21, setup your asterisk so it registers you with that number... |
| 18:22.24 | thx2000 | is there a way to eliminate the delay when calling local extensions? |
| 18:22.24 | fender21 | cyber: Right.. Off for my 3rd mission of the day. Thanks again Cyber |
| 18:22.39 | techman97_andy | Ariel: thanks! got it! |
| 18:22.41 | ManxPower | fender21, perhaps you did not understand. Nobody here knows anything about the sick twisted and perverted way AMO has set up the Asterisk config files. |
| 18:22.42 | Cybertoy | no prob.. just keeping you busy.. :) |
| 18:22.44 | MoutaPT | php4-mysql already installed and lots of other packages but i still get mysql php Libraries not installed, i'm using debian, can any one help me? |
| 18:23.13 | Cybertoy | manxpower, hmm.. sick twisted and perverted? I think I have to look into that software then. |
| 18:23.19 | ManxPower | MoutaPT, you must have made a wrong turn. The #debian channel is down the hall, 4 doors down, on the left. |
| 18:23.30 | fender21 | ManxPower: Perhaps you missed the opening statements of Brokeback mountain pbx.. :-) |
| 18:23.47 | ManxPower | Cybertoy, have you ever seen the AMP config stuff. |
| 18:23.54 | *** join/#asterisk bkw_ (n=brian@c-68-32-112-142.hsd1.md.comcast.net) |
| 18:24.08 | Cybertoy | manx, no .. not really... |
| 18:24.38 | ManxPower | If you try to configure Asterisk the traditional way (/etc/asterisk/extensions.conf, for example) and then use AMP to configure something all your changes are overwritten. |
| 18:24.51 | ManxPower | you need to edit like extensions.conf.additional or some silliness like that |
| 18:26.15 | Cybertoy | manx, that's what happens when you look behind a gui of what's supposed to be easy. |
| 18:26.31 | ManxPower | exactly. |
| 18:27.10 | ManxPower | Which is why people using AMP should NOT ask for support here, they could easily get advice which is correct for people not running AMP, but will basically erase your configs if you follow the advice and are running AMP. |
| 18:27.30 | salviadud | manxpower, are you saying amp sucks? |
| 18:27.51 | salviadud | i think it sucks |
| 18:27.57 | ManxPower | salviadud, No. |
| 18:28.06 | salviadud | well, that's just me |
| 18:28.10 | salviadud | i don't like guis |
| 18:28.13 | ManxPower | I'm saying that if you run AMP and ask for advice here, you could end up erasing all your asterisk configs |
| 18:28.25 | salviadud | true dat holmes |
| 18:28.28 | ManxPower | salviadud, I don't like GUIs either. GUIs suck. |
| 18:28.36 | fender21 | Manx: This is the case anytime someone uses a frontend system. The questions I'm asking have yet to be related to A@H but more of how this whole thing works. |
| 18:28.36 | Ariel_ | Amp and freepbx works very well. But ManxPower is correct support for it should be at there location #freepbx |
| 18:28.47 | tzanger | hmm |
| 18:29.02 | tzanger | does anyone know what "misc file error 0x20000" means when an ip501 boots up? |
| 18:29.09 | ManxPower | fender21, "trunk" is an AMP term. It really means nothing here. |
| 18:29.42 | ManxPower | most people will assume "trunk" means "VoIP connection to an ITSP", but who knows what it means in AMP. |
| 18:29.56 | Ariel_ | fender21, asterisk@home uses amp and also should and is supported by the folks at freepbx |
| 18:30.24 | Ariel_ | ManxPower, it's the same a trunk is a setting to a voip provider either sip/iax2 or zap |
| 18:30.54 | Ariel_ | tzanger, I have not seen that error. Reboot the phone |
| 18:30.55 | fender21 | I've spent a week reading up on Asterisk trying to get hold of some of these concepts. It was time well spent by between Ariel & Cyber they have taught me more in 15 minutes than I read all of last week. |
| 18:30.56 | ManxPower | Ariel_, in traditional telecom terms a trunk can handle 1 call. |
| 18:31.15 | ManxPower | fender21, you didn't learn anything from The Book? |
| 18:31.18 | ManxPower | ~thebook |
| 18:31.19 | jbot | hmm... thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 18:31.19 | Ariel_ | ManxPower, yes I understand that. I have been around this industry for years now. |
| 18:31.34 | tzanger | Ariel_: it does reboot after that |
| 18:31.35 | tzanger | over and over and over |
| 18:31.43 | Ariel_ | ahh |
| 18:31.43 | tzanger | I was going ot factory reset it but that's not working |
| 18:31.51 | ManxPower | tzanger, could it be having problems updating it's firmware? |
| 18:31.54 | salviadud | ariel, back in the day, when the rotary phones were the bomb |
| 18:32.02 | fender21 | The book and wiki are both very helpful but skip over some of the basic concepts of connecting things for newbies. |
| 18:32.09 | ManxPower | tzanger, I seem to recall that I could NOT get my IP300 and IP500s to update to Bootrom 3.x |
| 18:32.21 | salviadud | or when they introduced horrible 80's music for MOH |
| 18:32.31 | fender21 | hehe |
| 18:32.43 | ManxPower | Maybe I just get pissed off when I spend 30 mins helping someone only to discover that my advice is totally invalid because they are running AMP |
| 18:32.46 | Ariel_ | salviadud, elevator music is what we called it |
| 18:32.52 | salviadud | man, i hate the oldschool system MOH |
| 18:32.53 | tzanger | ManxPower: nah, it's happy |
| 18:32.59 | salviadud | i use metallica! |
| 18:33.00 | tzanger | I didn't update the bootrom, just the sip.ld |
| 18:33.02 | tzanger | 1.6.5 |
| 18:33.19 | salviadud | studies have shown, that people want to hang up faster, when listening to metallica |
| 18:33.33 | tzanger | bootrom 2.6.1 |
| 18:33.34 | *** part/#asterisk |rt| (n=realthin@c-66-31-7-34.hsd1.nh.comcast.net) |
| 18:33.35 | salviadud | and metallica doesn't like their music being used that way, they want mo' money |
| 18:33.58 | Ariel_ | tzanger, the sip config for the new 1.5.plus uses one file for instead of icmp.cfg and sip.cfg.. So you could have them mixed up. |
| 18:33.58 | salviadud | and THAT'S why i still use metallica, just for my extension, at least |
| 18:34.09 | Cybertoy | talking about MoH ... reminds me that I wanted to setup an audio stream... |
| 18:34.23 | tzanger | Ariel_: I just have sip.cfg and phone1.cfg, and the very simple 000000000000.xml |
| 18:34.40 | *** join/#asterisk Evanrude (n=david@ip68-107-162-212.lu.dl.cox.net) |
| 18:34.57 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 18:35.02 | Ariel_ | how about your mac.cfg |
| 18:35.03 | *** join/#asterisk Dovid (n=Dovid@62.0.153.54) |
| 18:35.17 | Ariel_ | the mac.cfg is what calls up the other files |
| 18:35.29 | tzanger | don't have a mac.cfg |
| 18:35.39 | Ariel_ | macaddressofphone.cfg |
| 18:35.42 | tzanger | ahh |
| 18:35.45 | tzanger | I have those |
| 18:35.55 | Ariel_ | they point to the files you want to load on the phone |
| 18:36.09 | Cybertoy | you happen to have a 7970? |
| 18:36.16 | Ariel_ | polycom |
| 18:36.34 | tzanger | oh wait |
| 18:36.46 | tzanger | I had the directory directory called directory, but called contacts in that file |
| 18:37.16 | Ariel_ | and the phone one I use like the extension for the phone it self. like phone2000.cfg or in somecases the mac of the phone as well. |
| 18:37.23 | Cybertoy | someone gave me a 7970 to play with and it took me 3 days to get up and rinning with the new sip firmware and asterisk ... without cisco callmanager software... |
| 18:37.30 | tzanger | Ariel_: yes, that's what I did |
| 18:37.39 | Ariel_ | k |
| 18:37.46 | tzanger | [macaddress].cfg gives phone[exten].cfg and sip.cfg |
| 18:37.59 | Cybertoy | some things still not work the way I like them to. |
| 18:38.00 | thx2000 | anyone know why there would be a delay when calling extensions, even when i have no overlap in my dialplan? |
| 18:38.13 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 18:39.46 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 18:39.50 | tzanger | <APPLICATION APP_FILE_PATH="sip.ld" CONFIG_FILES="phone234.cfg, sip.cfg" MISC_FILES="misc" LOG_FILE_DIRECTORY="logs" OVERRIDES_DIRECTORY="overrides" CONTACTS_DIRECTORY="contacts"/> |
| 18:39.59 | tzanger | that's all I have in the [macaddress].cfg file |
| 18:39.59 | Ariel_ | thx2000, no but more info would be nice. sip/ zap via pstn ? |
| 18:40.31 | tzanger | but wait |
| 18:40.32 | Cybertoy | but I don't use it ... |
| 18:40.47 | Cybertoy | but the thought of running asterisk on an ipod is cool .. :) |
| 18:40.49 | thx2000 | sip, to sip, internal. Just for testing purposes, im trying to call line 2 of my sipura from line 1, and when i enter the extension it takes a good 5 seconds for it to start ringing.... |
| 18:40.50 | Ariel_ | <?xml version="1.0" standalone="yes"?> |
| 18:40.50 | Ariel_ | <!-- Default Master SIP Configuration File--> |
| 18:40.50 | Ariel_ | <!-- Edit and rename this file to <Ethernet-address>.cfg for each phone.--> |
| 18:40.50 | Ariel_ | <!-- $Revision: 1.12 $ $Date: 2003/06/17 15:26:10 $ --> |
| 18:40.50 | Ariel_ | <APPLICATION APP_FILE_PATH="sip.ld" CONFIG_FILES="phone215.cfg, sip.cfg, ipmid.cfg" MISC_FILES="" LOG_FILE_DIRECTORY=""/> |
| 18:40.53 | Ariel_ | is what I use |
| 18:41.01 | tzanger | Ariel_: doI need to say "phone1.cfg, phone234.cfg, sip.cfg" in there? I only have CHANGES from phone1.cfg in my phone234.cfg... |
| 18:41.14 | thx2000 | i eliminated all but one line in my dialplan so the only option i have is to call that extension, and it still waits |
| 18:41.51 | Strom_M | thx2000, you need to edit the dialplan on your sipura |
| 18:41.54 | Ariel_ | thx2000, is this normal asterisk or via amp/freepbx or a@h |
| 18:42.09 | tzanger | Ariel_: I think that was it |
| 18:42.09 | thx2000 | normal asterisk |
| 18:42.11 | Ariel_ | also press the # key will speed up the call |
| 18:42.16 | tzanger | I had misc files as a directory, not a list of misc files ot load |
| 18:42.24 | ManxPower | SIP devices usually have their OWN dialplan |
| 18:42.29 | Ariel_ | tzanger, I see |
| 18:42.33 | Strom_M | thx2000, look at your sipura's setup page and edit the dialplan there |
| 18:42.44 | thx2000 | yea, i've seen that in there...lemme see what i can do w/ that...thanx |
| 18:42.54 | *** join/#asterisk apardo (n=apardo@62-15-114-125.inversas.jazztel.es) |
| 18:42.58 | Ariel_ | Strom_C, lets just get him to dial the number and press# to see if it speed up first |
| 18:43.25 | thx2000 | Ariel_: that does speed it up |
| 18:43.45 | tzanger | Ariel_: what's your ipmid.cfg look like? I don't have that file at all |
| 18:44.00 | Ariel_ | thx2000, ok now what the sipura is waiting for is more numbers but it depends on what you want to dial. |
| 18:44.22 | Ariel_ | I don't recommend you edit the dial plan since it takes into account 911 and international numbers |
| 18:44.46 | Ariel_ | tzanger, I use the older version of sip.cfg and ipmid.cfg |
| 18:45.04 | Ariel_ | since I have been upgrading since over 2 years on the phones |
| 18:45.18 | Ariel_ | but it's basicly the new sip.cfg file just split up. |
| 18:45.26 | thx2000 | but this is all stuff that asterisk could handle anyway right? |
| 18:45.34 | Ariel_ | thx2000, no |
| 18:45.51 | Ariel_ | sip devices need to know your numbers due to sip sends them all at once |
| 18:46.02 | thx2000 | gotcha |
| 18:46.08 | Ariel_ | not like a phone which you pickup and dial each digit and the other end processit. |
| 18:46.59 | Ariel_ | tzanger, I can send you my few files I use. It's very basic and only configures to one line. I have very un-educated users here. |
| 18:47.46 | tzanger | no worries, I need all the help I can get :-) |
| 18:47.55 | Dovid | Ariel_: Post them on pastebin.com so we can all have a look |
| 18:47.57 | justinu|laptop | ~seen r_evolution |
| 18:47.59 | jbot | r_evolution <i=_evoluti@208.251.203.246> was last seen on IRC in channel #asterisk, 9d 21h 44m 26s ago, saying: 'or am I reading the question differently than you're asking it?'. |
| 18:48.00 | Ariel_ | pm your email and I will send them to you |
| 18:48.28 | Ariel_ | Dovid, there too big |
| 18:48.34 | Dovid | ? |
| 18:48.38 | Dovid | what r u having a problem with / |
| 18:48.40 | Dovid | ?* |
| 18:48.46 | *** part/#asterisk mogorman (n=mogorman@68.62.237.103) |
| 18:48.55 | Ariel_ | Dovid, not me. trying to help tzanger with polycom setups |
| 18:49.02 | Dovid | ah |
| 18:49.12 | Dovid | cant help with that. dont know em too well |
| 18:49.48 | Ariel_ | yeppie it's time to do the yum -y update on my server now.... almost finished with the basic setup.... |
| 18:50.29 | Dovid | Ariel_: What OS are you using ? |
| 18:51.28 | Ariel_ | Dovid, CentOS 4.3 final |
| 18:51.34 | Dovid | ok |
| 18:51.38 | Dovid | be carefull |
| 18:51.53 | Ariel_ | it's easy to setup the CentOS boxes |
| 18:52.05 | Dovid | cause there have been problems with a new kernel |
| 18:52.12 | Ariel_ | yes I know |
| 18:52.16 | Ariel_ | simple edit |
| 18:52.18 | Ariel_ | fixes |
| 18:52.25 | Dovid | :) |
| 18:52.36 | Dovid | u on ur first asterisk job ? |
| 18:52.59 | Ariel_ | Dovid, no I have been working on asterisk setups for over 3 years now |
| 18:53.07 | Dovid | ah |
| 18:53.08 | Ariel_ | my first one was on .5 beta |
| 18:53.23 | Ariel_ | I have many asterisk boxes up and running. |
| 18:53.32 | Strom_M | is there a way to show a list of all available dundi numbers on a network? |
| 18:54.16 | MoutaPT | Ariel_ did you try asterFAX? |
| 18:54.34 | |dennis| | Question: Dunno if this is the right place to ask but anyways..i have gxp2000 with fw 1.0.1.13 trying to upgrade to 1.1.0.1 but my tftp server says that the gxp 2000 islooking for lies that are not there in the zip..like ring1.bin,cfg000xxxxx etc..please help.. |
| 18:54.35 | Ariel_ | asterfax. hummm no. I have been using spandsp |
| 18:54.53 | MoutaPT | Does it handle Email to Fax? |
| 18:55.26 | Ariel_ | MoutaPT, hummm yes if you set it up. |
| 18:55.33 | Ariel_ | |dennis|, lies... |
| 18:55.43 | Ariel_ | rofl |
| 18:55.55 | MoutaPT | Ariel have you done it before? |
| 18:56.10 | |dennis| | oh sorry. Ariel_ i meant files.. |
| 18:56.22 | Cybertoy | anyone have a streaming audio musiconhold.conf they can share with me? |
| 18:56.22 | |dennis| | :) |
| 18:56.34 | Ariel_ | |dennis|, yes I knew that just was funny when I read it |
| 18:56.38 | Cybertoy | preferably something from shoutcast .. but I don't mind anything else. |
| 18:56.43 | |dennis| | :) |
| 18:56.45 | Ariel_ | but I don't have any gs phones |
| 18:57.02 | MoutaPT | How do I handle my users to only transfer calls to ext-local? |
| 18:57.20 | MoutaPT | currenlty if i allow Transfer calls they can transfer to landline... |
| 18:57.34 | Ariel_ | MoutaPT, if your in amp that is the only place the can transfer too local extensions |
| 18:58.03 | |dennis| | Ariel_ I am trying to convince our school/college to switch to asterisk from the nortel pbx they have here from the phone company..purchased a gxp-2000 to show them..but the voice quality of it is ok...not that good..which other phoens are better..? |
| 18:58.50 | Ariel_ | polycom |
| 18:58.50 | demigod2k | dennis, polycom. cisco. personally I wouldn't make the switch if they already have a nortel pbx |
| 18:59.03 | Abydos313 | why not? |
| 18:59.18 | |dennis| | why not? |
| 18:59.27 | demigod2k | depending on the box, it may be high-end equipment with a lot of infrastructure (he did say college) in place |
| 18:59.36 | Abydos313 | good point |
| 18:59.56 | demigod2k | what do you hope to get out of the deal. if you ask me it's a HUGE responsibility to undertake |
| 19:00.37 | |dennis| | well not too much actually...we are a small college in a third world country(belize) simple little pbx is sestup with 4 lines coming in...about 13 extensions setup presently...nothing more..no ivr, no voice mail, nothing is setup presently.. |
| 19:01.01 | demigod2k | ok then ya worth considering. just thought I'd throw that out there |
| 19:01.02 | Abydos313 | that is real small for college |
| 19:01.22 | demigod2k | polycom, cisco, anything for more than $100 a phone really |
| 19:01.34 | demigod2k | the gxp2k is cool "for the price" |
| 19:02.01 | |dennis| | yeah that why ipurchased it but should have expected less for what i paid... :) hmm..thanks..shall look into it... |
| 19:02.15 | |dennis| | Junior College actually... :) |
| 19:03.37 | MoutaPT | I'm using Asterisk 1.2.5 and some of my users refer Sjphone to take 30seconds to hangup calls and then report an error |
| 19:03.44 | MoutaPT | any one have seen this? |
| 19:09.01 | trimi` | !seen rene- |
| 19:11.40 | trimi` | which softphone its better for use with asterisk, SJLabs or X-Pro/eye-beam |
| 19:11.44 | trimi` | ??????? |
| 19:13.38 | Abydos313 | xlite and eyebeam are nice |
| 19:13.44 | Abydos313 | but eyebeam is not free |
| 19:13.57 | trimi` | im experianicng bad quality |
| 19:14.04 | Abydos313 | try diff codecs |
| 19:14.22 | trimi` | <Abydos313> i tried |
| 19:14.23 | Abydos313 | i'll test it when it comes out |
| 19:14.31 | trimi` | with g729 im experiancing bad quality |
| 19:14.40 | trimi` | and with g711 i cant use |
| 19:14.49 | trimi` | i got 128 KB for upload |
| 19:14.58 | NewSole | u need g723 |
| 19:14.59 | Abydos313 | i don't get incomming calls with g729..but 711 works perfectly |
| 19:15.46 | trimi` | <NewSole> i cant find softphone with g723 |
| 19:16.19 | NewSole | we have one we are testing now.... it will be released end of month |
| 19:16.38 | jimbe | NewSole is that Yuxin? |
| 19:16.45 | trimi` | <NewSole> Adore Softphone ? |
| 19:16.47 | NewSole | nope |
| 19:17.00 | jimbe | who is the manufacturer? |
| 19:17.01 | trimi` | working on WIN OS ? |
| 19:17.17 | trimi` | and give uf the website if you have one |
| 19:17.37 | _Thor | Anybody here knows mysql... I've tried the mysql board, but they are real jerks, you know, I post a question and none of them answer, you know |
| 19:17.41 | NewSole | yes.. wince.. winx86... and no it does not use that crappy iaxclient |
| 19:18.07 | Dovid | _Thor: i know a tiny lil bit |
| 19:18.10 | Dovid | what do u need ? |
| 19:18.13 | jimbe | NewSole will there be an ATA as well? |
| 19:18.21 | NewSole | it will be posted on http://www.virttel.com when done |
| 19:18.55 | NewSole | he have one that uses the pa168 chipset... but we have different flash |
| 19:18.58 | bkw_ | Jingle Bells |
| 19:19.19 | jimbe | hey bkw_ |
| 19:19.26 | bkw_ | wasabi |
| 19:19.54 | trimi` | <NewSole> how much its going to cost |
| 19:19.54 | trimi` | ? |
| 19:19.58 | jimbe | NewSole what's the price point? will there be IAX2 ATAs as well? |
| 19:20.14 | Dovid | _Thor: what do u need ? |
| 19:20.18 | _Thor | Dovid: my whole billing system for * is mysql based... but the freaking thing now keeps giving me "server gone away"messages |
| 19:20.24 | NewSole | it will be about 20$ |
| 19:20.46 | NewSole | it will be posted on website when done testing |
| 19:21.20 | jimbe | ohhh softphone |
| 19:21.20 | NewSole | jimbe.. we already have iax2 ATA's and Wifi's |
| 19:21.39 | jimbe | how much for the iax2 ata |
| 19:22.22 | NewSole | aya's support g729/g723/ilbc/ulaw/alaw/gsm |
| 19:22.32 | NewSole | same with wifi |
| 19:23.47 | *** part/#asterisk dokhench (n=dochench@adsl-065-080-180-134.sip.bna.bellsouth.net) |
| 19:24.12 | shido6 | iax2 ata's with g729 are about $100 with built in router |
| 19:24.25 | shido6 | it has g723 , too |
| 19:25.47 | shido6 | did you need a few? |
| 19:29.15 | jimbe | NewSole are they the iaxy? |
| 19:29.20 | jimbe | shido6 what model |
| 19:29.41 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 19:30.05 | shido6 | model? I havent given it a name yet |
| 19:30.56 | NewSole | Just like you have not DID My SER Server |
| 19:31.19 | NewSole | s/DID/DO |
| 19:31.45 | jimbe | shido6 you, JerJer and nufone are the biggest crooks in #asterisk |
| 19:31.58 | Qwell | NewSole: That sentence makes less sense now, if that's possible |
| 19:32.37 | Qwell | jimbe: personal attacks will NOT be accepted here. |
| 19:32.49 | Qwell | tolerated rather |
| 19:32.55 | jimbe | it's not personal, it's business |
| 19:33.06 | NewSole | shido6.. Charged me $$ upfront to do SER Server.... and he Could not even get server running... |
| 19:33.20 | NewSole | told me to speak to file to get it finished |
| 19:33.26 | jimbe | ask them about the pap2s, about the g729 licenses, about the downtime, the audio issues, the problems getting refunds |
| 19:33.32 | Qwell | lilo: Around? |
| 19:33.37 | Qwell | Please remove the tor troll |
| 19:33.51 | jimbe | Qwell do you work for nufone? |
| 19:33.55 | Qwell | No I do not |
| 19:34.16 | jimbe | oh, then why the defensiveness? |
| 19:34.40 | russellb | jimbe: take this to #nufone |
| 19:34.51 | Qwell | jimbe: Because, as I already said, personal attacks WILL NOT be tolerated |
| 19:35.20 | tzanger | Ariel_: do you have an ipmid.cfg you can put up somewhere for me? apparently it's required for distinctive ring and auto-answer |
| 19:35.38 | NewSole | Qwell... I am sorry I just did not want to see another Sucker get Pulled in By shido6 |
| 19:36.04 | NewSole | I did Twice... and no more.. I even gave him a second change and he blew it |
| 19:36.04 | *** join/#asterisk salaud (n=salaud@h-66-166-226-2.sttnwaho.covad.net) |
| 19:36.09 | Ariel_ | tzanger, normally I would but I am upgrading my servers today.. if you pm me your email I will send them to you. |
| 19:36.17 | jimbe | Qwell i'm sorry as well |
| 19:36.48 | lilo | jimbe: I've been asked to remove that escort cloak |
| 19:36.56 | *** join/#asterisk ComputerWarm (n=dan@HS196-230-97.nt.net) |
| 19:36.57 | tzanger | Ariel_: sounds good |
| 19:36.57 | NewSole | I told jimbe to look at voipsupply.... and not shido6... he asked why.... |
| 19:37.05 | salaud | For some reason my compiled, 1.2.6 asterisk, can't launch /usr/sbin/sendmail. Anyone seen this? It doesn't appear to a permissions issue... or at least one that I can find |
| 19:37.37 | lilo | should be resolved |
| 19:37.42 | salaud | I can log in as asterisk and run sendmail and successfully send an e-mail |
| 19:38.14 | *** join/#asterisk exonic_ (n=exonic@c-24-11-51-17.hsd1.mi.comcast.net) |
| 19:38.15 | salaud | any way to increase the debugging or something in app_voicemail? |
| 19:38.27 | NewSole | Qwell... does Asterisk have a fix yet for the h323 in large volume calls |
| 19:38.36 | exonic_ | Hey what's the name of the program that monitors packet loss of a traceroute? |
| 19:38.38 | Qwell | NewSole: dunno, I don't follow h323 |
| 19:38.54 | exonic_ | salaud, try set debug 50 ? |
| 19:39.06 | exonic_ | what are you trying to debug? |
| 19:39.09 | NewSole | tried the one in addons.. and it crashed server evey 2 min |
| 19:39.27 | bkw_ | Qwell, personal attacks won't be tolerated unless its JerJer right? |
| 19:39.36 | fourcheeze-away | sounds like something a high class tart would wear on a cold night |
| 19:39.44 | salaud | exonic_: I'm trying to find out why: WARNING[16963]: app_voicemail.c:1799 sendmail: Unable to launch '/usr/sbin/sendmail -t' |
| 19:39.45 | Qwell | bkw_: only one I've seen in a while... |
| 19:40.04 | bkw_ | salaud, you have no /usr/sbin/sendmail? |
| 19:40.07 | exonic_ | salaud, umm.. sendmail exists and is executabl? |
| 19:40.21 | salaud | exonic_: Yep.. exists and is executable by asterisk |
| 19:40.22 | shido6 | its called the IAD 100 S |
| 19:40.35 | shido6 | there is a 1 port FXS |
| 19:40.42 | shido6 | a 1 port FXO and a 1 port FXS |
| 19:40.47 | salaud | exonic_: bizarre.... I can log in as user asterisk and successfully send an e-mail |
| 19:40.52 | exonic_ | salaud, I doubt it is. You running asterisk as root or a non prived user? |
| 19:40.56 | shido6 | 1 wan and 1 LAN |
| 19:41.22 | salaud | exonic_: I'm running as asterisk... again.. I log in as asterisk and can send an e-mail find |
| 19:41.27 | salaud | exonic_: s/find/fine |
| 19:41.51 | ManxPower | salaud, can you log into asterisk and execute "/usr/sbin/sendmail -t" |
| 19:42.00 | salaud | ManxPower: Yep |
| 19:42.11 | ManxPower | rather than just issuing a mail command |
| 19:42.29 | salaud | ManxPower: Yep |
| 19:43.14 | salaud | Unfortunately the Warning message from app_voicemail doesn't specify permissions error, file not found, etc |
| 19:44.31 | salaud | This may have something to do with something compiled in? I was on the debian package 1.2.4 and it worked... compiled and installed.. now, it doesn't |
| 19:45.21 | exonic_ | salaud, never had that problem. I guess it depends on how asterisk executes the program.. is /usr/sbin/ in the PATH variable for user asterisk? |
| 19:45.27 | shido6 | the device can deal with SIP and IAX but only one or the other with a firmware upgrade |
| 19:46.04 | shido6 | you're better off with an IAXy using ulaw tho the g729 codec still needs work |
| 19:46.25 | salaud | exonic_: That, I'm sure about... where would I set that? |
| 19:46.26 | ComputerWarm | any gnugk users here? |
| 19:46.44 | salaud | exonic_: The user really has /bin/false as the shell.. so I'm not sure where to set a PATH |
| 19:47.17 | salaud | exonic_: I'm switching shell /bin/bash, just for testing |
| 19:47.48 | exonic_ | salaud, how do you 'login' as user asterisk .. because you're munging your env variables if you don't do it right. |
| 19:48.42 | salaud | exonic_: I'm just temporarily switching the shell to /bin/bash to login as asterisk. The asterisk daemon started when it was /bin/false |
| 19:49.19 | exonic_ | ahh |
| 19:49.24 | salaud | exonic_: Asterisk is just setting it's user and group as asterisk... I don't think the daemon actually logs in |
| 19:49.51 | tzafrir_laptop | salaud, why change the shell? isn't su -s /bin/bash - asterisk good enough? |
| 19:49.57 | salaud | exonic_: so... it's real hard to say where any PATH info is... is there an asterisk global var or something? |
| 19:50.22 | salaud | tzafrir_laptop: probably... but.. just force of habit... |
| 19:50.43 | ComputerWarm | does anyone know of a channel that supports gnugk? |
| 19:50.45 | PakiPenguin | okay , guys i have a problem , i have a sip provider , that i can register to , from a normal sip client , but when i try to register to it using asterisk , it doesnt work , it never registers , what could be wrong , kindly help me |
| 19:50.58 | *** join/#asterisk Strom_M (n=strom@66.159.243.59) |
| 19:51.07 | tzafrir_laptop | salaud, why are you trying to run a shell as the asterisk user? |
| 19:51.16 | shido6 | PakiPenguin: try "useragent" |
| 19:51.24 | tzafrir_laptop | just remember to run asterisk with -U asterisk |
| 19:51.33 | shido6 | what are you registering to the service provider with that works? |
| 19:51.45 | salaud | tzafrir_laptop: to test whether a permissions error is causing WARNING[16963]: app_voicemail.c:1799 sendmail: Unable to launch '/usr/sbin/sendmail -t' |
| 19:51.56 | PakiPenguin | X-lite / X-pro |
| 19:52.12 | shido6 | useragent, notifymimeype, fromdomain, realm, might be important |
| 19:52.32 | salaud | if I do ... su -s /bin/bash - asterisk ... I can launch /usr/sbin/sendmail -t |
| 19:52.34 | shido6 | useragent=X-PRO release (insert release here) |
| 19:52.45 | tzafrir_laptop | salaud, did you manually set the sendmail command? what doyou use for sendmail? |
| 19:53.20 | tzafrir_laptop | ls -l /usr/sbin/sendmail |
| 19:53.53 | salaud | tzafrir_laptop: I'm running postfix... so it's the sendmail command from the postfix package |
| 19:54.01 | PakiPenguin | hmms thanks shido6 , let me test it |
| 19:54.05 | fourcheeze-away | PakiPenguin: what makes you think it never registers? |
| 19:54.17 | shido6 | PakiPenguin: notifymimetype=text/plain |
| 19:54.29 | salaud | tzafrir_laptop: I can run '/usr/sbin/sendmail -t' as user asterisk after su -s /bin/bash - asterisk with no problems |
| 19:54.59 | fourcheeze-away | PakiPenguin: do you see your registration fail? |
| 19:55.40 | tzafrir_laptop | salaud, again, did you set the sendmail command yourself? |
| 19:55.46 | PakiPenguin | fourcheeze-away, no i dont :( , it just stucks up at Request Sent , when i do sip show registry |
| 19:56.26 | fourcheeze-away | PakiPenguin: you're sure you have the right IP number / domain name? |
| 19:56.30 | salaud | tzafrir_laptop: What do you mean by "set" do you mean in voicemail.conf? If so, I have tried it both ways. Both commenting and uncommenting the line in voicemail.conf |
| 19:56.32 | fourcheeze-away | no typos there? |
| 19:56.56 | PakiPenguin | yup |
| 19:56.58 | exonic_ | salaud, what version of asterisk? |
| 19:57.03 | tzafrir_laptop | Any chance you're trying to run the executable '/usr/sbin/sendmail -t' ? |
| 19:57.17 | salaud | exonic_: 1.2.6 |
| 19:57.18 | fourcheeze-away | fourcheeze-away: in that case debug the ip number |
| 19:57.24 | fourcheeze-away | PakiPenguin: ^^ |
| 19:57.38 | PakiPenguin | i am doing that |
| 19:57.40 | fourcheeze-away | and try register again |
| 19:57.43 | fourcheeze-away | ok |
| 19:57.51 | fourcheeze-away | got a sip trace to paste somewhere? |
| 19:57.52 | salaud | tzafrir_laptop: It's possible... but, even when I don't override the default I get the error... colud be a bug in the code though.. not sure.. |
| 19:58.01 | tzafrir_laptop | salaud, well, one possible way is to run everything under strace -f ... |
| 19:58.42 | salaud | tzafrir_laptop: yeah... but this is a production server... I can't run the daemon that way |
| 19:59.02 | PakiPenguin | ah strange |
| 19:59.04 | salaud | tzafrir_laptop: I suppose I could wait till late at night when things are low |
| 19:59.16 | exonic_ | salaud, hehe. how'd you get in this state on a production server? =) . just kiddin |
| 19:59.50 | salaud | exonic_: Because I took someone's advice on this channel to compile asterisk to solve my no DTMF recognized on IAX channels problem ;) |
| 19:59.59 | salaud | exonic_: and that was a higher priority fix |
| 20:00.08 | PakiPenguin | fourcheeze-away, i am trying to register it to a non standard port ( register => login:password@ip:8891 ) , but in sip debug , i see asterisk sending auth request on port 5060 |
| 20:00.29 | fourcheeze-away | ok |
| 20:00.41 | shido6 | err |
| 20:00.47 | shido6 | so in the user/peer you have port 8891? |
| 20:00.53 | shido6 | port=8891 |
| 20:00.55 | PakiPenguin | yeah |
| 20:00.56 | salaud | exonic_: People haven't been getting emailed their voicemails for two days... and I just found out |
| 20:01.05 | PakiPenguin | even my asterisk is listening on 8891 too |
| 20:01.12 | shido6 | you could just do an iptables hack if you're isp is blocking 5060 :) |
| 20:01.29 | exonic_ | salaud, ... does the user asterisk have privs to write to /tmp |
| 20:01.32 | salaud | exonic_: set debug 50 doesn't seem to do anything useful.. maybe that needs to be set as a compile flag |
| 20:01.34 | tzafrir_laptop | you can try strace -p |
| 20:01.47 | tzafrir_laptop | Though you'll have to find the right thread |
| 20:02.04 | salaud | tzafrir_laptop: That attaches to a currently running thread, I guess. |
| 20:02.19 | fourcheeze-away | PakiPenguin: which version of asterisk? |
| 20:02.28 | salaud | exonic_: no |
| 20:02.29 | tzafrir_laptop | salaud, each thread has its own pid |
| 20:02.29 | exonic_ | salaud, when all else fails, read the source. I am reading 1.2.6 and it's' the result of mkstemp() failed. |
| 20:02.50 | exonic_ | salaud, give privs to asterisk to write to /tmp .. |
| 20:03.04 | salaud | exonic_: There aren't any other possible function calls that will cause that message? |
| 20:03.23 | exonic_ | the debug message said app_voicemail.c:1799 .. that's where I went. |
| 20:03.27 | salaud | exonic_: If it were a file not found... would it give a different message |
| 20:03.34 | salaud | exonic_: oh... shit |
| 20:03.40 | salaud | exonic_: that's too easy ;) |
| 20:03.42 | PakiPenguin | thats the sip trace shido6 fourcheeze-away |
| 20:03.52 | fourcheeze-away | hmm |
| 20:04.16 | salaud | exonic_: testing |
| 20:05.05 | fourcheeze-away | PakiPenguin: what's at 192.168.1.7 ? |
| 20:05.12 | PakiPenguin | my asterisk |
| 20:05.21 | salaud | exonic_: ok... I owe you beer... next time you are in portland or at ETEL... let me know |
| 20:05.22 | shido6 | whatever it is it needs a nat=yes |
| 20:05.33 | exonic_ | salaud, heh |
| 20:06.04 | PakiPenguin | i have that in sip.conf |
| 20:06.15 | PakiPenguin | and the ip is in dmz , so no ports blocked |
| 20:07.30 | ManxPower | PakiPenguin, and you have localnet and externip set |
| 20:07.38 | fourcheeze-away | PakiPenguin: it looks to me like your sip supplier is going to hvae a problem finding it's way back to you |
| 20:07.42 | ManxPower | (assuming the asterisk server is behind nat) |
| 20:08.02 | file[laptop] | C'MON DO THE QT4 DANCE! |
| 20:08.16 | PakiPenguin | hmmm , a sec , let me set them up |
| 20:08.55 | |dennis| | my * box has a public ip address ....i want to use clients coming from the internet to connect to my * box. The clients may come from nat-ted nets. how can i get my * box to keep them connected? The clients manage to call other ppl on the * box but not vie versa. |
| 20:09.02 | Qwell | QT3 was so much cooler |
| 20:09.12 | demigod2k | qt quicktime or qt trolltech |
| 20:09.19 | |dennis| | i have set nat=yes and qualify=yes |
| 20:11.11 | file[laptop] | demigod2k: trolltech! |
| 20:11.26 | demigod2k | ya I'd have to agree. qt designer went downhill |
| 20:11.37 | demigod2k | I stopped using it at 4.x |
| 20:13.17 | PakiPenguin | http://pastebin.ca/48669 <-- after the externip and localnet setup |
| 20:13.35 | *** join/#asterisk opcode (i=adam@CPE006008148866-CM000f9fa8c50a.cpe.net.cable.rogers.com) |
| 20:13.48 | opcode | hey anyone here used chan_btp? |
| 20:14.25 | ComputerWarm | question maybe someone can help with this, how does one go about figuring out the math for 1/1 billing 6/6 billing and 30/6 billing? |
| 20:15.00 | *** part/#asterisk opcode (i=adam@CPE006008148866-CM000f9fa8c50a.cpe.net.cable.rogers.com) |
| 20:16.09 | PakiPenguin | ManxPower, take a look http://pastebin.ca/48669 |
| 20:17.01 | ManxPower | PakiPenguin, I'll give you my opinion after I get a large paypal from you. reading SIP debug is too hard to do for free. |
| 20:17.19 | ManxPower | ComputerWarm, what math? |
| 20:17.58 | ManxPower | 1/1 = billing in incriments of 1 second for the first min and 1 second for for any addtional mins. |
| 20:18.01 | ManxPower | That's pretty simple |
| 20:18.27 | ComputerWarm | ya 1/1 is but 30/6 for example isn`t |
| 20:18.37 | PakiPenguin | :) alright |
| 20:19.02 | *** part/#asterisk opc0de (i=adam@CPE006008148866-CM000f9fa8c50a.cpe.net.cable.rogers.com) |
| 20:19.14 | ManxPower | Hmm? take the total call seconds, subtract 30, bill for that 30 seconds, then divide the rest by 6, round up, bill that many seconds. |
| 20:19.31 | ManxPower | round up to the nearest 6 second boundry, that is |
| 20:20.13 | theorem_ | clever |
| 20:20.19 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 20:20.22 | ComputerWarm | ManxPower ya that would work |
| 20:20.31 | ManxPower | But it's saturday afternoon, I've had a few beers, so you should prolly confirm this with a sober person |
| 20:20.31 | ComputerWarm | thanks |
| 20:20.55 | ComputerWarm | lol is there sober people on Saturday :-) |
| 20:21.07 | ManxPower | or in other words "billing is in 6 second units, with a min of 30 seconds" |
| 20:22.13 | ComputerWarm | i wish i could just find a good billing system for gnugk and not have to worry about doing it myself |
| 20:22.24 | ComputerWarm | but i can`t seem to find anything that works very well |
| 20:22.39 | theorem_ | I'm sure there are people here who have their own billing systems |
| 20:23.02 | ComputerWarm | theorem_ ya same here but its a matter of finding them |
| 20:23.08 | theorem_ | if you query and are willing to offer $ , perhaps someone will come out of the woodwork ? |
| 20:23.14 | bkw_ | billing isn't a fun thing |
| 20:23.38 | theorem_ | but that's just a suggestion -- I do not do billing on my home system |
| 20:23.49 | ComputerWarm | bkw_ i agree i hate doing the figures for billing systems tried it before i could never get it to work correctly |
| 20:24.12 | bkw_ | you have prepaid, postpaid and live billing |
| 20:24.14 | bkw_ | all fun |
| 20:25.02 | ComputerWarm | all my customers a prepay |
| 20:25.07 | *** join/#asterisk rene- (n=rene-@201.137.74.112) |
| 20:27.56 | rene- | hey, i have been playing with chanspy, i can listen to any given call between an A and a B party, i would like to connect my self (e.g. C party ) to the A+B call and be able to talk to A but without B listening to what I (C) am saying, i think it is possible to do it with Meetme Conference Rooms, but i am not sure, and probably A would be the initiator of the conference, i would like also C to be the initiator, and have B to never listen to C, is it possib |
| 20:31.31 | *** join/#asterisk I-MOD (n=I-MOD@68.62.165.168) |
| 20:31.50 | exonic_ | rene-, if A falls in the forest does C make a noise? or does B? |
| 20:33.31 | *** join/#asterisk Qwell_ (n=north@unaffiliated/qwell) |
| 20:33.32 | rene- | if A falls over C or B's feet then a noise from them could be heard |
| 20:33.33 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 20:34.08 | exonic_ | rene-, sounds pretty custom to me :) |
| 20:34.49 | rene- | as in a feature available on a per pay basis? |
| 20:39.10 | MoutaPT | any one knows how to allow transfer call only to ext-local? |
| 20:39.15 | ComputerWarm | is the creator of AstBill around? |
| 20:39.33 | MoutaPT | currently transfer calls is allowing to outbound calls to pstn |
| 20:39.56 | *** join/#asterisk gammacoder (n=chatzill@cpe-65-26-178-240.indy.res.rr.com) |
| 20:41.03 | *** join/#asterisk Chotaire (i=chotaire@chotaire.net) |
| 20:47.38 | MoutaPT | I found one user with an wakeupcall of 200 minutes, is there any bug around this app or what could be wrong? |
| 20:48.59 | bkw_ | sounds like a bug to me |
| 20:52.31 | JunK-Y | some1 familiar with app_ices? |
| 20:54.13 | JunK-Y | im still getting: |
| 20:54.13 | JunK-Y | <PROTECTED> |
| 20:54.13 | JunK-Y | Apr 8 16:57:55 WARNING[11254]: app_ices.c:173 ices_exec: Write failed to pipe: Broken pipe |
| 21:09.39 | tzafrir_laptop | MoutaPT, the call needs to be in the context "ext-local" to begin with. Or something that contains only that context and not much more |
| 21:10.31 | Strom_M | hmm, this is an interesting problem |
| 21:11.40 | Strom_M | if I dial a local channel which then dials some other channel and connects, is there any way to have the local channel return to the dial plan and continue on with the next priority in the extension instead of unsupervising all the way back to the calling party? |
| 21:27.22 | *** join/#asterisk mogorman (n=mogorman@68.62.237.103) |
| 21:27.56 | *** join/#asterisk archvile (n=archvile@c-69-138-124-58.hsd1.fl.comcast.net) |
| 21:30.40 | archvile | hi, i just recently setup a asterisk server, and i was wondering how i can get the phone i have (SPA-841) to work with asterisk, asterisk answers my calls that i give to it, but im unable to give outgoing calls from asterisk unless i register the phone with my viatalk information, im sure its because i dont have an extention configured, and i tried to configure one but im not sure how to do so, anyone have any information? |
| 21:33.07 | Strom_M | archvile, read thebook |
| 21:33.09 | Strom_M | ~thebook |
| 21:33.10 | jbot | thebook is probably Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 21:36.41 | *** join/#asterisk znoG (n=gs@109-130-89-200.fibertel.com.ar) |
| 21:37.00 | The_Isle_of_Mark | hi all, anyone in here have any draytek voip experience? |
| 21:38.20 | zoa | yeah |
| 21:38.22 | zoa | they are slow |
| 21:38.23 | zoa | :) |
| 21:38.30 | zoa | at least that spaceship looking one is |
| 21:38.33 | zoa | extremely slow |
| 21:38.37 | zoa | (the gui) |
| 21:38.41 | archvile | Strom_M: this book isnt giving me any help really, everything is working fine i just need to be able to use my phone the (SPA-841) to call the asterisk server locally so i can setup the menu ect. i heard it was suppose to auto detect it but it doesnt |
| 21:38.47 | zoa | i also got a call from the isp |
| 21:38.50 | The_Isle_of_Mark | sure they are, but it works ok...I hav eone and I cant seem to figure out how to make it dial properly |
| 21:38.57 | zoa | to please disconnect whatever i connected to the adsl network |
| 21:38.59 | zoa | with one of them |
| 21:39.22 | Strom_M | archvile, are you using regular asterisk, or are you using asterisk@home / AMP / FreePBX? |
| 21:39.31 | archvile | asterisk@home |
| 21:39.33 | The_Isle_of_Mark | zoa: do you remember how to setup voip? |
| 21:39.43 | Strom_M | archvile, you should go to #freepbx and ask there |
| 21:40.19 | The_Isle_of_Mark | zoa: I have been through the docs and the only thing I can find is how to make it dial with a very limited dialplan |
| 21:40.36 | archvile | Strom_M: thanks |
| 21:41.29 | The_Isle_of_Mark | zoa, the only way I can get it to work is to create a dial number specifically for each number...I was wondering if there is a way to make it dial all numbers the same |
| 21:42.46 | zoa | The_Isle_of_Mark: iirc it was pretty easy through the gui ? |
| 21:43.06 | zoa | i didnt touch it in at least a year |
| 21:43.11 | zoa | so dont ask me how it was done |
| 21:43.37 | zoa | http://www.asteriskguru.com/tutorials/vigor_2900v_draytek.html |
| 21:43.41 | zoa | look here, maybe its similar |
| 21:44.12 | The_Isle_of_Mark | zoa, that is my problem, the sip works fine if I define each and every number I want to dial |
| 21:44.19 | zoa | what model do you have ? |
| 21:44.23 | The_Isle_of_Mark | 2900v |
| 21:44.27 | zoa | then check the website |
| 21:44.31 | zoa | we documented it for that oen |
| 21:44.33 | zoa | one |
| 21:44.38 | zoa | screenshots and everything |
| 21:44.40 | The_Isle_of_Mark | been all over for this |
| 21:45.16 | The_Isle_of_Mark | I can't seem to make it dia <number> as <number>@sipproxy.url |
| 21:45.21 | zoa | hmm thats really strange |
| 21:46.19 | The_Isle_of_Mark | zoa, and that tutorial, although good, does explain it |
| 21:46.28 | The_Isle_of_Mark | s/does/doesn't/ |
| 21:48.19 | The_Isle_of_Mark | say if I have a number 2125555555 I have to go into the router and put in a dial plan that is 2125555555@sipproxy.url and it works fine |
| 21:48.27 | The_Isle_of_Mark | ring any bells? |
| 21:49.44 | zoa | thats very strange |
| 21:49.47 | zoa | we didnt have that |
| 21:50.12 | The_Isle_of_Mark | maybe I am missing something. The docs for the thing are way less than perfect |
| 21:50.39 | zoa | if you find it, please do me a favor and post it in the comments section at asteriskguru |
| 21:51.00 | zoa | well, if you want to of course :) |
| 21:51.32 | zoa | im off now |
| 21:51.33 | zoa | cheers |
| 21:51.41 | The_Isle_of_Mark | ok thanks |
| 22:00.44 | *** join/#asterisk Mpls-Eric (n=ejo1@209.98.205.186) |
| 22:01.17 | Mpls-Eric | OK, feeling stupid at the moment, Shouldn't this work to get the 4 digit extension number from this header? "CALLREASON=<tel:4029> ;reason=no-answer" --- And here's what I have in the dial plan "Set(VMBOX=${CUT(${CALLREASON},:,2):4})"... I also tried with "Set(VMBOX=${CUT(${CALLREASON},:,1):4})"... |
| 22:01.59 | Mpls-Eric | I'm wondering if the ; in the var (I called it a header) is screwing with things? |
| 22:02.57 | *** join/#asterisk Winkie (n=urmom@cpc2-stre1-6-0-cust119.bagu.cable.ntl.com) |
| 22:03.01 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 22:03.16 | [TK]D-Fender | Yay, just set up my new IP 301 :) |
| 22:03.54 | brookshire | those are nice! |
| 22:03.55 | brookshire | :D |
| 22:04.21 | file | a moment to be by your side! |
| 22:04.25 | [TK]D-Fender | its "cute".... nothing glorious, just a solid little phone... |
| 22:05.27 | [TK]D-Fender | also a bonus in that I went from boxed to firmware upgraded and provisioned in jsut a few minutes from scratch (configs as well) |
| 22:07.13 | [av]bani | ick |
| 22:10.01 | Rawplayer | is anyone in here doing voip over openvpn? |
| 22:10.08 | Rawplayer | and wireless |
| 22:12.05 | Mpls-Eric | I've not tried openvpn |
| 22:12.19 | Mpls-Eric | does that use udp for transport? |
| 22:12.32 | Rawplayer | yes |
| 22:12.44 | Mpls-Eric | Have you tried it yet? |
| 22:13.04 | Rawplayer | yes but when i dial over openvpn with my softphone the network dies |
| 22:13.14 | Rawplayer | i cant reach the asterisk server anymore |
| 22:13.52 | Rawplayer | when i dont use openvpn its keeps on working fine over the same route to the asterisk server |
| 22:14.23 | dpryo | It only dies when you dial? |
| 22:14.31 | dpryo | or does it not work at all? |
| 22:15.53 | *** join/#asterisk saftsack (n=saftsack@p54A7FDAA.dip.t-dialin.net) |
| 22:16.37 | *** join/#asterisk CrummyGummy (n=wayne@dsl-145-122-123.telkomadsl.co.za) |
| 22:16.41 | *** join/#asterisk Darwin35 (n=Darwin@c-24-9-75-234.hsd1.co.comcast.net) |
| 22:19.57 | nextime | Rawplayer : i use iax trunk between many different * servers witouth problems, anyway, your problem is like an mtu problem, take a look of fragmentation, try to tracepath your route |
| 22:20.06 | nextime | *without |
| 22:21.26 | Rawplayer | dpryo: when i dial |
| 22:21.45 | Rawplayer | i can login sucsessfull with my softphone over the vpn connection |
| 22:21.53 | Rawplayer | but when i call someone then it stops working |
| 22:22.39 | nextime | Rawplayer : try to do some non-voip traffic over your openvpn tunnel, maybe with a file transfer via netcat, and watch if it stop the network |
| 22:22.49 | Rawplayer | that all works fine |
| 22:22.57 | Rawplayer | i also do samba over the vpn network |
| 22:23.00 | Rawplayer | without any problems |
| 22:23.50 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 22:24.00 | nextime | anyway, it sound to me like an mtu problem on the tunnel, not related on asterisk or on the voip channel directly |
| 22:24.54 | Rawplayer | hmm |
| 22:24.57 | Rawplayer | could be true |
| 22:25.58 | Rawplayer | when i ssh from my client where also the softphone is on to the asterisk server ssh dies to(also other traffic) but when i first ssh to the vpn endpoint and then ssh from the vpn endpoint to the asterisk server then the ssh session keeps alive |
| 22:26.09 | madd | r |
| 22:26.11 | madd | eree |
| 22:26.25 | Rawplayer | i'am gonne tcpdump some stuff |
| 22:26.32 | nextime | Rawplayer : try to do some traffic on ssh like catting some big log files |
| 22:27.15 | nextime | sometime when you have mtu problem over openvpn catting file can block your ssh session |
| 22:28.05 | *** join/#asterisk tainted- (n=identd@ppp-71-134-51-75.dsl.irvnca.pacbell.net) |
| 22:28.07 | tainted- | are there any atas or ipphones that support vpn? |
| 22:30.03 | Rawplayer | nextime: what should i change on the mtu? |
| 22:32.06 | nextime | Rawplayer : maybe you shuld use mtu 1436 on the tun/tap device, but you need to try different values and openvpn options to tune openvpn correctly when it as fragmentation problems, it strictly depend on your network config |
| 22:32.26 | Rawplayer | ok |
| 22:32.31 | nextime | start from openvpn --help | grep mtu is a good point :) |
| 22:32.34 | Rawplayer | i'am gonne play with that |
| 22:32.36 | Rawplayer | hehe |
| 22:32.43 | Rawplayer | i giggle first |
| 22:33.46 | tainted- | lol |
| 22:45.33 | Gamercjm | tainted |
| 22:45.36 | Gamercjm | from HTS? |
| 22:46.06 | *** join/#asterisk SparFux (n=player@e182020025.adsl.alicedsl.de) |
| 22:46.41 | SparFux | Hello! When I load zaphfc, my system hangs and I have to cold-start. What could cause it? I am using HFC-S bri card. |
| 22:51.37 | *** join/#asterisk Ad-Hoc (n=Nimbus@ppp71-adsl-118.ath.forthnet.gr) |
| 22:56.58 | *** join/#asterisk taz3r (n=tazer2@66-227-137-32.dhcp.bycy.mi.charter.com) |
| 22:57.00 | taz3r | hey |
| 22:57.10 | taz3r | I am having a problem with voicemail an A@H 2.6 |
| 22:57.44 | taz3r | when I call an extension that is offline (not registered) it says "<extension> is on the phone" |
| 22:58.00 | taz3r | it should say is unavailable or play the unavailable message |
| 22:58.09 | taz3r | any ideas? |
| 22:58.54 | *** join/#asterisk znoG (n=gs@109-130-89-200.fibertel.com.ar) |
| 22:58.56 | *** join/#asterisk flynux (i=r4dq3wl@cl-8.bru-01.be.sixxs.net) |
| 22:59.04 | *** join/#asterisk rene- (n=rene-@dsl-201-128-115-107.prod-infinitum.com.mx) |
| 22:59.16 | taz3r | bah, nobody is around I assume. |
| 22:59.33 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 23:00.48 | rene- | i have trouble understanding what the local channel does |
| 23:01.39 | *** join/#asterisk Evanrude (n=david@ip68-107-162-212.lu.dl.cox.net) |
| 23:02.00 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 23:02.54 | [TK]D-Fender | taz3r : Please read the channel topic.... |
| 23:04.53 | rene- | D-Fender: i read that someone was trying to implement a whisper feature for asterisk using local channels, do you think it is possible? |
| 23:04.56 | taz3r | err? |
| 23:06.02 | dlynes | SparFux: Are you using a recent version of the zaptel drivers in Europe? |
| 23:06.25 | znoG | if i have busydetect=no .. why would calls coming in/out of Zaptel be getting randomly dropped? |
| 23:06.34 | SparFux | It's the zaptel driver delivered with debian sarge and I just compiled it. |
| 23:06.38 | [TK]D-Fender | "whisper" can you clarify... |
| 23:06.46 | dlynes | SparFux: which version is it? |
| 23:06.53 | [TK]D-Fender | ~amp |
| 23:06.55 | jbot | hmm... amp is "amp is, like, NOT supported here! people using it should join #freepbx (the new name of amp)" |
| 23:07.00 | [TK]D-Fender | taz3r : See above |
| 23:07.41 | SparFux | dlynes: it's 1.2.5 |
| 23:07.44 | [TK]D-Fender | znoG : Got callpregress=yes by any chance? That'd do it |
| 23:07.56 | rene- | call center jargon for having A talk to B and then have C listen to both and make A able to listen to C but keep B from listening to B |
| 23:08.50 | dlynes | SparFux: have a look at the asterisk users mailing list |
| 23:08.54 | rene- | sorry, but keep C from listening to B |
| 23:09.04 | [TK]D-Fender | rene- : basically a "silent spy".... I suppose you can do that semi-transparently by conferencing them in and having them mute their mic. only diff would be that the call would be held for an instant while that happens... then again you could just use ChanSpy... |
| 23:09.06 | dlynes | SparFux: There's a recent thread on there of people having problems with zaphfc in europe |
| 23:09.18 | dlynes | SparFux: including solutions for the problem |
| 23:09.24 | SparFux | dlynes: I am already trying chan-capi and chan-misdn. chan-capi says, there is no capi, though misdn should support it and my card is detected by kernel. chan-misdn says, there is an unknown label "ast_load" |
| 23:10.03 | SparFux | dlynes: so perhaps it's my config euroisdn? |
| 23:10.19 | dlynes | SparFux: Yeah...it was somehting to do with that...can't remember exactly what |
| 23:10.24 | rene- | D-fender: i blew it, i meant whisper to be chan_spy plus the ability for A party to be able to listen to the C party without having B listening to C |
| 23:10.35 | SparFux | dlynes: I never connected the card to pstn though. |
| 23:10.37 | dlynes | SparFux: but it was something to do with zaphfc and HDLC or something |
| 23:10.39 | znoG | [TK]D-Fender: nope, i don't. |
| 23:11.24 | rene- | [TK]D-Fender: then some guy posted that he would try to mix two conferences using local channels to to acompplish that goal, does it sounds reasonable? |
| 23:11.55 | Hmmhesays | rene chanspy |
| 23:11.59 | znoG | [TK]D-Fender: it doesn't log anything, either. An error message such as "busy/hangup detected" or something would at least tell me why its dropping the connection. |
| 23:12.01 | SparFux | dlynes: hm.. don't know. I think it's the s0 hangup problem you mean, but what I have here is that the whole system freezes when I load zaphfc! |
| 23:12.49 | dlynes | SparFux: perhaps it's related to the same problem they're having |
| 23:13.00 | rene- | i think i have gotten it. is the purpose of local channels to make possible to be able to dial an extension the same way one dials a sip/zap/iax device? |
| 23:13.05 | [TK]D-Fender | rene- : ummm, still not sure how to make it "transparent" to the caller so they don't know they're being monitored.. its the "hold" factor that throws it |
| 23:13.18 | Hmmhesays | local channels serve many functions |
| 23:13.32 | SparFux | dlynes: what's your driver version of zaphfc? |
| 23:13.38 | rene- | Hmmhesays: would you care to elaborate on that? |
| 23:13.52 | SparFux | How to activate capi on misdn? |
| 23:13.52 | dlynes | SparFux: I'm not using zaphfc, nor am I in europe |
| 23:13.55 | Hmmhesays | you can use local channels to do cascading dial around |
| 23:13.59 | Hmmhesays | that is pretty kickass |
| 23:13.59 | SparFux | dlynes: ah, I see. |
| 23:14.35 | dlynes | SparFux: i'm only using x100p's and te410p |
| 23:14.54 | rene- | Hmmhesays: by Cascading Dial Around you mean that you have an extension that dials devices based on priority jumping? |
| 23:15.22 | Hmmhesays | rene- : example, you have a user with 3 different phones, one cell, one house phone and one ip phone |
| 23:16.12 | znoG | there we go, another Zaptel drop. |
| 23:16.15 | Hmmhesays | when you call that person, first you can make his ip phone ring.. then if they don't answer, call the house phone... while the ip phone still rings, in case they want to pick that one up... then have the cell phone ring, with the first two still ringing, so the user can pickup any one of the 3 ringing devices |
| 23:19.13 | rene- | but how does that works? dial statements have their timeouts so you basically dial the second device after the first dial statement has timed out, unless you do something like dial(CELLPHONE&IPPHONE&HOMEPHONE), which would dial them all at once |
| 23:19.40 | shido6 | how many rings do you want to go through before going to the next device? |
| 23:19.53 | rene- | say 5 |
| 23:20.39 | shido6 | what is that 20k ms ? |
| 23:20.43 | *** part/#asterisk SparFux (n=player@e182020025.adsl.alicedsl.de) |
| 23:20.56 | rene- | i believe so, 4 secs per ring |
| 23:24.59 | shido6 | http://pastebin.ca/48683 |
| 23:25.12 | shido6 | after 25k ms |
| 23:25.26 | shido6 | it goes to the next line and rings boink |
| 23:26.00 | znoG | i just found a post from somebody who had the same sort of random hangup problems with a X100P. Good to hear (i guess) that I'm not the only one with issues |
| 23:28.47 | rene- | thanks shido6 but i was wanting to look to an example using local channels, a concept i have yet to grasp |
| 23:28.54 | dlynes | znoG: is that on a digium x100p, or a clone? |
| 23:29.13 | znoG | digium x100p |
| 23:30.10 | dlynes | did you find a solution to it, then? |
| 23:30.50 | znoG | no, not at all |
| 23:30.57 | znoG | and it's driving me mad, happens so often |
| 23:31.08 | dlynes | oh |
| 23:31.19 | dlynes | does it happen with all x100p cards? or only certain ones? |
| 23:31.59 | znoG | i only have one |
| 23:32.05 | dlynes | ah |
| 23:32.22 | dlynes | just curious because i was going to start using them |
| 23:32.31 | *** join/#asterisk Saturn-- (i=Saturn@24.50.85.195) |
| 23:32.51 | dlynes | i've got one customer that's using two as well, without any reports of problems |
| 23:32.58 | dlynes | and they're both clones from ebay |
| 23:33.03 | Saturn-- | <PROTECTED> |
| 23:33.11 | Saturn-- | no hits when i searched for that before |
| 23:33.13 | Saturn-- | any ideas? |
| 23:34.03 | *** join/#asterisk Mourning (i=sa@w167.z208037076.nyc-ny.dsl.cnc.net) |
| 23:34.46 | tzafrir_laptop | Saturn--, that module is a left-over? Did you actually build codec_gsm.so on your latest asterisk build? |
| 23:34.54 | Mourning | does anyone know how to answer a callwaiting on an analog phone connected to a sipura 3000 running off of asterisk? |
| 23:35.07 | Saturn-- | I haven't built any previous version |
| 23:35.12 | dlynes | Saturn--: I'm not even seeing that symbol in any of the asterisk code |
| 23:35.22 | dlynes | Saturn--: Where did you get your codec_gsm.so from? |
| 23:35.46 | Saturn-- | just built latest tarball |
| 23:35.55 | dlynes | from where? |
| 23:36.02 | tzafrir_laptop | What is a PLT symbol? |
| 23:36.15 | Saturn-- | front page of the website |
| 23:36.20 | dlynes | tzafrir: something linker table |
| 23:36.39 | dlynes | Saturn--: which website? asterisk.org? |
| 23:36.49 | Saturn-- | yeah |
| 23:36.51 | *** join/#asterisk VoIPMasta (n=John@201.160.17.234.cableonline.com.mx) |
| 23:37.04 | dlynes | i'm running the same code you are |
| 23:37.09 | dlynes | but yet i don't have that symbol |
| 23:37.16 | tzafrir_laptop | Saturn--, do you build gsm with an external libgsm? |
| 23:37.33 | tzafrir_laptop | (or was that a Debian-specific patch?) |
| 23:37.55 | Saturn-- | no external libgsm |
| 23:37.57 | Saturn-- | this is netbsd |
| 23:38.21 | dlynes | Saturn--: it's probably looking for some external shared object that's not installed |
| 23:38.43 | rene- | Hmmhesays: i have found an example in the wiki that ilustrates the delayed dialing using Local channels. any other examples you could think of would help me a lot |
| 23:38.47 | dlynes | Saturn--: that shared object defines the symbol, 'Short_term_analysis_filteringx' |
| 23:39.49 | Saturn-- | building libgsm |
| 23:40.11 | Mourning | Anyone at all know anything about the SPA 3000? |
| 23:40.12 | dlynes | Saturn--: I guess netbsd doesn't have ports? |
| 23:40.18 | dlynes | Mourning: yes |
| 23:40.20 | Saturn-- | pkg_get |
| 23:40.45 | dlynes | Saturn--: ah...yeah...it's not an available package for freebsd..freebsd has it in ports |
| 23:40.53 | Mourning | dlynes: do you know an answer to my question about callwaiting on the SPA3k?? |
| 23:41.21 | Saturn-- | there is a codec_gsm.so in the directory |
| 23:41.36 | Saturn-- | which should have been whatever came with the tarball origionally |
| 23:42.07 | [TK]D-Fender | Mourning : what do you want to know? |
| 23:42.26 | dlynes | Mourning: not offhand, no |
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| 23:42.40 | dlynes | [TK]D-Fender: Mourning does anyone know how to answer a callwaiting on an analog phone connected to a sipura 3000 running off of asterisk? |
| 23:42.47 | Saturn-- | how can i rebuild that file without a reinstall? |
| 23:43.32 | Mourning | [TK]D-Fender: what dlynes said |
| 23:45.11 | rene- | Hmmhesays: Say i am autodialing, and i am in a call center type of environment, i am using either call files or manager to do the dialing out, if i set the originating channel to a local pseudo channel, i can make the connected call go to a context just as if it was an incoming call? |
| 23:45.50 | rene- | like passing the new Call tru answering-machine/fax-machine filters and then sending it to a Queue? |
| 23:46.04 | Mourning | [TK]D-Fender: any ideas? |
| 23:47.36 | tainted- | rene- yes - and i have done exactly that with some AGI scripting |
| 23:48.13 | tainted- | u can use the LOCAL channel and drop the call into a context after it's answered |
| 23:48.21 | tainted- | nasty hackage involved |
| 23:48.51 | rene- | tainted-: it is a cool feature, and something good to know, thanks |
| 23:50.30 | Mourning | Does anyone here have any experience with the SPA 3000? |
| 23:50.37 | tainted- | rene- but not sure if it will accomplish the call monitoring u want to do |
| 23:51.56 | Saturn-- | lol i only used a couple of cisco IP phones |
| 23:52.01 | Saturn-- | they are sorta confusing at first |
| 23:52.40 | Mourning | SPA 3000 isnt a phone, its an ATA with 1FXS and 1 FXO |
| 23:53.03 | dlynes | Mourning: They're talking about something altogether different |
| 23:53.05 | rene- | tainted-: it is ok, i have read in the asteriskguru forums that the feature is available in a commercial basis so if the people want that they will have to pay for it, but i have a better idea about how will auto dialing will be acompplished |
| 23:54.12 | tainted- | there are free autodialers |
| 23:54.19 | tainted- | just can't remember the names |
| 23:56.13 | Hmmhesays | rene, i use something similar to what is on the wiki for delayed dialing |
| 23:58.54 | *** part/#asterisk Mourning (i=sa@w167.z208037076.nyc-ny.dsl.cnc.net) |