| 00:00.07 | BarthezZ | there are different types |
| 00:00.28 | BarthezZ | but there really isn't much info about voip dect solutions |
| 00:00.40 | *** join/#asterisk felipe_ (~felipe@my.nada.kth.se) |
| 00:00.48 | BarthezZ | so when i have it, i'll publish my experience |
| 00:02.30 | BarthezZ | big advantage is GAP, in stead of paying 300or400 per handset, you can just buy cheap ones for 40 or something |
| 00:02.53 | BarthezZ | cause well, those things don't survive gravity that good, so you don't want really expensive handsets :p |
| 00:06.00 | Naikrovek | got a product number |
| 00:06.01 | Naikrovek | ? |
| 00:06.25 | Kobaz | is your google broken? |
| 00:06.36 | Kobaz | polycom wireless |
| 00:06.42 | BarthezZ | http://www.polycom.com/products/voice/wireless_solutions/dect_communications/infrastructure/dect_ws_300.html |
| 00:08.54 | Naikrovek | dude that is awesome |
| 00:08.58 | Naikrovek | holy schnikeys |
| 00:09.05 | Naikrovek | I R needing this |
| 00:09.13 | Naikrovek | for my roving band of phone people |
| 00:09.17 | BarthezZ | hehe |
| 00:09.23 | BarthezZ | yeah it's a really interesting product |
| 00:09.26 | BarthezZ | but quite pricy |
| 00:09.30 | Naikrovek | my india office could really use this |
| 00:09.42 | BarthezZ | the wireless server 600 is 1000 |
| 00:09.43 | Naikrovek | not really pricey when compared to voip phones |
| 00:09.52 | Naikrovek | the 600 can do quite a few handsets |
| 00:10.02 | Naikrovek | the 300 can do 12 |
| 00:10.08 | Naikrovek | 12 is probably enough |
| 00:10.17 | Naikrovek | will have to speak with mgmt about this tomorrow |
| 00:10.27 | BarthezZ | 12 with max 4 concurrent calls |
| 00:10.51 | *** join/#asterisk ickmund (~magnus@cli-5b7ee15c.bcn.adamo.es) |
| 00:11.47 | BarthezZ | Naikrovek sent me a mail if you want me to keep you up to date on my experience with it |
| 00:12.05 | Naikrovek | i'll watch for updates in here, if that's okay |
| 00:12.15 | BarthezZ | I'm not that active on irc :p |
| 00:12.17 | Naikrovek | oh |
| 00:12.19 | Naikrovek | okay then |
| 00:12.40 | BarthezZ | if your mail resides in my inbox somewhere i can give you an update :p |
| 00:14.28 | BarthezZ | check and marked as todo :) |
| 00:16.23 | Naikrovek | tyvm |
| 00:17.40 | BarthezZ | no problem, needing to inform me also helps me to really write something about it which could help more people on the subject :p |
| 00:17.57 | hipitihop | anyone here running zoiper sip client on ubuntu ? |
| 00:20.14 | Kobaz | hipitihop: use twinkle |
| 00:20.22 | Kobaz | the zoiper is a real pain |
| 00:20.28 | Kobaz | the new zoiper rather |
| 00:20.35 | Kobaz | apt-get install twinkle |
| 00:23.10 | hipitihop | Kobaz, I tried twinkle many moons ago before I put in asterisk and didn't find that too good abck then, I think codec support and choppy sound at the time |
| 00:23.23 | *** join/#asterisk dmz (~dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) |
| 00:24.03 | hipitihop | Kobaz, I think I found out about zoiper via teh asterisk book...hmm |
| 00:29.28 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 00:32.16 | hipitihop | Kobaz, and twinkle doesn't seem to have been updated since Feb 2009 |
| 00:35.01 | Naikrovek | BarthezZ: where do you see how many simultaneous calls can be done |
| 00:39.15 | BarthezZ | yeah somewhere on the polycom, that seems to be a well kept secret most of the time 8)7 |
| 00:40.06 | Naikrovek | yeah it does seem to be secret |
| 00:40.11 | Naikrovek | or at least well guarded |
| 00:40.44 | Naikrovek | would like to see more simultaneous calls out of the 300 but maybe it's a wireless protocol limitation |
| 00:41.23 | BarthezZ | iirc the 300 can do 4 |
| 00:41.28 | BarthezZ | so you need the 600 fast :P |
| 00:42.12 | BarthezZ | for the 600v3 it's 35 handsets and 11 channels |
| 00:42.16 | BarthezZ | i think |
| 00:42.39 | Naikrovek | seems like 1/3 |
| 00:45.35 | BarthezZ | yeah something like that, I don't know |
| 00:45.46 | BarthezZ | have no idea what the limitation would be internally |
| 00:45.59 | *** join/#asterisk ManxPower (~manxpower@193.sub-75-254-10.myvzw.com) |
| 00:46.14 | BarthezZ | wish i could buy two, one to mess around with at home and one for my customer :p |
| 00:46.15 | Naikrovek | probably depends on dect protocol or something, the number of calls that can be carried per access point |
| 00:47.56 | BarthezZ | could be |
| 00:48.07 | BarthezZ | don't know how many dect phones you can have in 1 room without interferance |
| 00:48.21 | BarthezZ | don't know that much about the technologie |
| 00:48.24 | Naikrovek | probably 4, but i'm guessing. |
| 00:48.29 | Naikrovek | yeah wikipedia needs a visit |
| 00:49.05 | BarthezZ | 120 available radiochannels |
| 00:51.12 | *** join/#asterisk chendy (~chatzilla@204.152.211.137) |
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| 00:53.58 | *** join/#asterisk sourcode (~code@ppp-58-8-238-110.revip2.asianet.co.th) |
| 00:55.20 | ManxPower | remember the dect frequencies are different in the USA than in Europe |
| 00:56.28 | BarthezZ | oh yes ofcourse, USA has 5 channels with 12 timeslots, EU 10 channels with 12 timeslots |
| 00:56.49 | ManxPower | in the USA it runs at 1800Mhz or somewhere close to that |
| 00:56.54 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 00:57.41 | BarthezZ | # Frequency: 1880 MHz1900 MHz in Europe, 1900 MHz-1920 MHz in China, 1910 MHz-1930 MHz in Latin America and 1920 MHz1930 MHz in the US and Canada |
| 00:57.44 | BarthezZ | # Carriers: 10 (1,728 kHz spacing) in Europe, 5 (1,728 kHz spacing) in the US |
| 00:57.47 | BarthezZ | # Time slots: 2 x 12 (up and down stream) |
| 00:58.05 | ManxPower | I must have had it backwards |
| 00:58.23 | BarthezZ | lol |
| 00:58.49 | ManxPower | I was so unhappy with the DECT phone I tried a few years ago I never tried them again. |
| 00:59.21 | BarthezZ | unhappy with dect? why |
| 01:00.37 | ManxPower | it started dropping out at a very short distance and even de-register (had to be placed back on the base to re-register) |
| 01:00.54 | ManxPower | I suspect it was just a poorly designed model. |
| 01:01.00 | ManxPower | Ended up just using cell phones instead |
| 01:01.01 | BarthezZ | think so |
| 01:01.26 | BarthezZ | i have a consumer grade dect thingy at home which easily reaches 100m oO |
| 01:05.18 | *** join/#asterisk coppice (~chatzilla@m121-202-82-46.smartone-vodafone.com) |
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| 01:13.27 | Naikrovek | 100m is awesome |
| 01:13.36 | Naikrovek | through cement blocks i wonder? |
| 01:15.43 | BarthezZ | 1 cement wall and after that open space |
| 01:16.49 | Naikrovek | same here |
| 01:34.19 | *** join/#asterisk pabelanger (~pabelange@CPE0013f7abc09a-CM0013f7abc096.cpe.net.cable.rogers.com) |
| 01:56.01 | Naikrovek | BarthezZ: 4 simultaneous calls for the 300, 12 for the 600v3 |
| 01:56.06 | Naikrovek | found it in the docs |
| 01:56.34 | Naikrovek | 11 calls per cell if a multi-cell environment |
| 01:56.38 | Naikrovek | for the 600v3 |
| 01:58.49 | *** join/#asterisk diegomad (~mad@190.158.77.2) |
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| 03:20.56 | TJNII | I need to cough up for a gigabit switch. |
| 03:21.13 | TJNII | 100mbit between the NAS box and the quad-core box just isn't cutting it. |
| 03:22.09 | TJNII | 217gigs copied this week. Can you tell I'm playing with my media collection? |
| 03:28.02 | *** join/#asterisk Akiraa (~Akiraaaa@79.112.44.66) |
| 03:44.51 | *** join/#asterisk voipnoob (shiva@detroit.slack.net) |
| 03:47.43 | voipnoob | For an Asterisk Server on the internet which will serve 1000 customers over the internet |
| 03:47.43 | voipnoob | with 150 max concurrent calls. The Asterisk Server will not be trunked directly. |
| 03:47.43 | voipnoob | It will be connected to another Asterisk Server in a different country which will be |
| 03:47.43 | voipnoob | doing all the termination & SIP trunking. |
| 03:47.43 | voipnoob | What kind of Bandwidth usage & Internet Connection would be required for the first |
| 03:47.44 | voipnoob | Asterisk Server? |
| 03:48.04 | voipnoob | Can anyone help with this or help me do the calculations? |
| 03:48.40 | p3nguin | What codecs will you use? What type of connection will you use? |
| 03:50.46 | voipnoob | All calls will be voice only. They will be calls originating from a SIP phone in an Asian country & will be terminating in a IP phone or a regular Phone outside the country (in the US/UK/Australia). |
| 03:51.00 | [TK]D-Fender | ~bandwidth |
| 03:51.01 | infobot | extra, extra, read all about it, bandwidth is This is a measure, in some amount of bits per second, of theamount of data that can be sent over a particular cable, interface, orbus. |
| 03:51.07 | [TK]D-Fender | ~codecs |
| 03:51.08 | infobot | i heard codecs is http://snipurl.com/wiki_codecs. If you have audio/codec problems, first try to 'disallow=all' and 'allow=ulaw' and see if that works. Anyone that tells you to use 'allow=all' is an idiot as it usually causes audio problems, or Number/Name: 1/g723, 2/gsm, 4/ulaw, 8/alaw, 16/g726, 32/adpcm, 64/slin, 128/lpc10, 256/g729, 512/speex, 1024/ilibc |
| 03:51.40 | [TK]D-Fender | voipnoob: take your codec + 20kbit overhead p/s /channel |
| 03:52.18 | voipnoob | Thanks [TK]D-Fender. How would I decide what is the right codec(s) to use for my need? |
| 03:52.34 | *** join/#asterisk Slugs_ (~Slugs@unaffiliated/slugs) |
| 03:52.36 | [TK]D-Fender | voipnoob: the biggest you can afford to use |
| 03:52.53 | *** join/#asterisk TehRabbitt (~rabbott56@c-71-59-82-2.hsd1.pa.comcast.net) |
| 03:53.05 | TehRabbitt | hey i'm back lol |
| 03:53.07 | voipnoob | What's the smallest I can use with decent Voice Quality :-)? |
| 03:53.20 | TehRabbitt | p3nguin: reflashed my router back to stock firmware 0_o so far so good |
| 03:53.44 | TehRabbitt | wanna try to connect to 300? lol |
| 03:56.24 | TJNII | voipnoob: If you're going country -> country latency & jitter may be the killer, not bandwidth. |
| 03:58.39 | p3nguin | tehrabbitt: Any audio from me? |
| 03:58.58 | voipnoob | D-Fender/TJNII - I am a C/C++ programmer, not a networking/VOIP/or hardware person. I will be bringing in a person with those kind of expertise. However, I am trying to obtain enough information - so I have an have an intelligent discussion for chosing such a person & validate his recommendations |
| 03:59.04 | ChannelZ | TehRabbitt: Save some trouble and make an extension that calls Echo() |
| 04:00.05 | p3nguin | I'm going to register my * to yours. Screw the softphone for now. |
| 04:01.02 | Slugs_ | My asterisk box fell off my desk into a toilet and now all calls go directly to voicmail. All calls sound like it's 'under water' PLZ HELP! |
| 04:01.42 | voipnoob | So what are the different factors for chosing Codec. I read the Asterisk book - looks like 2 codecs seem right -> GSM & G.729A for my needs. Not sure though |
| 04:01.55 | Slugs_ | all electronics are certified(tm) dry... |
| 04:01.55 | TehRabbitt | back |
| 04:02.00 | TehRabbitt | p3nguin: nothing :-\ |
| 04:02.24 | Slugs_ | ChannelZ ? |
| 04:02.25 | TJNII | eyes Slugs_ skeptically |
| 04:02.49 | voipnoob | Also I don't understand what the books says about G.729A licensing -> Yes (no for passthrough) |
| 04:02.52 | voipnoob | what does that mean/ |
| 04:02.57 | ChannelZ | Slugs_: You should replace computer with cats. |
| 04:03.13 | Slugs_ | ok ty |
| 04:03.40 | ChannelZ | They no like water so drive the water out. |
| 04:03.50 | ChannelZ | is win-win! |
| 04:03.53 | Slugs_ | thats perfect! |
| 04:04.12 | voipnoob | Considering that my Asterisk would only terminate in-country IP calls |
| 04:04.33 | ChannelZ | until cats shit in kitchen. But try not to think too far ahead. |
| 04:04.37 | TJNII | You need a licence to transcode to another codec, IIRC. |
| 04:05.00 | Slugs_ | ChannelZ, good point... |
| 04:05.05 | Slugs_ | one thing at a time |
| 04:05.30 | ChannelZ | Thank you, thank you, we're here all week |
| 04:05.52 | Slugs_ | ;) |
| 04:06.34 | ChannelZ | Anyone play with their own Upstart scripts? |
| 04:07.54 | ChannelZ | hmm guess his router upgrade didn't go as well as he thought |
| 04:08.09 | p3nguin | maybe |
| 04:08.13 | TJNII | haha |
| 04:08.25 | TJNII | "Another one bites the dust" |
| 04:09.35 | Naikrovek | i used to use IRC in politically charged channels back when modems were around to hang up on people |
| 04:09.46 | Naikrovek | there's a command you can type that will tell modems to hang up |
| 04:09.52 | Naikrovek | or, rather, it used to. older modems are affected |
| 04:10.06 | Naikrovek | it was great fun to silence people that way |
| 04:10.15 | TJNII | Pfft. Fascist. |
| 04:10.21 | Naikrovek | just bored |
| 04:10.27 | Naikrovek | and it's hilarious |
| 04:10.50 | Naikrovek | don't think it woudl work much anymore |
| 04:11.31 | ChannelZ | +++ATH |
| 04:11.35 | TJNII | There was a guy in ##networking who was all butthurt because he got banned from a quake server. He was hellbent on getting another DHCP address from his ISP to get retribution, but had no idea how DHCP worked. It was pathetically amusing. |
| 04:11.45 | Naikrovek | +++ATH0 i thought |
| 04:11.48 | Naikrovek | i used atho0 |
| 04:11.50 | Naikrovek | ath0 |
| 04:11.59 | p3nguin | +++ATH0 is right |
| 04:12.00 | Naikrovek | rooms would go quiet in a hurry |
| 04:12.05 | ChannelZ | Probably. My Hayes is rusty. |
| 04:12.33 | *** part/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 04:12.40 | TJNII | I remember working at a place that used RS232 modems to upload daily data. 3 years ago. |
| 04:13.16 | Naikrovek | http://www.securityspace.com/smysecure/catid.html?id=10020 |
| 04:13.25 | TJNII | That system must have been kick-ass and really expensive in its day. Every truck had a hand-held PC. |
| 04:14.29 | p3nguin | I bet there are still some routers out there that are vulnerable to the DCC SEND bug. |
| 04:14.29 | ChannelZ | I have a USRobotics modem around here somewhere that probably still works if not for the power supply |
| 04:15.06 | ChannelZ | A Courier! A sort of cadillac of modems. |
| 04:15.31 | *** part/#asterisk ChannelZ (~bobm@burner.com) |
| 04:15.35 | *** join/#asterisk ChannelZ (~bobm@burner.com) |
| 04:15.36 | ChannelZ | oops |
| 04:15.55 | p3nguin | DCC SEND LOLOLOLOLOLOL |
| 04:16.30 | p3nguin | Surprising no one exited. |
| 04:16.36 | ChannelZ | actually just "muscle spasm while mouse is over [X] button" |
| 04:16.44 | voipnoob | D-Fender - is codec the only consideration for bandwidth requirement |
| 04:18.03 | *** join/#asterisk Slugs_ (~Slugs@unaffiliated/slugs) |
| 04:18.08 | TJNII | runs a whois on ChannelZ and notes he is an hour away. |
| 04:18.14 | Slugs_ | wtf ;/ |
| 04:18.21 | TJNII | wonders where he can get a crate of sporks.... |
| 04:18.33 | Slugs_ | KFC |
| 04:18.35 | Slugs_ | ! |
| 04:19.46 | *** part/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 04:19.51 | Slugs_ | colonel sanders invented those |
| 04:19.57 | Slugs_ | it's a fact |
| 04:20.43 | *** join/#asterisk TehRabbitt (~rabbott56@c-71-59-82-2.hsd1.nj.comcast.net) |
| 04:20.48 | TehRabbitt | p3nguin: i'm back |
| 04:21.06 | TehRabbitt | i just learned.. dont turn "NAT" off on your router.. it pretty much turns ur router into a switch |
| 04:21.06 | Slugs_ | watches p3nguin waddle away |
| 04:21.07 | TehRabbitt | lol |
| 04:21.08 | TJNII | Hey, he didn't brick his router! |
| 04:21.19 | TehRabbitt | LOL |
| 04:21.26 | c0dyhi11 | is there decent supprot for asterisk16 and dahdi23 on freebsd? |
| 04:21.33 | TehRabbitt | TJNII: still having issues with SIP though :( |
| 04:21.39 | TJNII | Awwww |
| 04:22.13 | TehRabbitt | yup :( |
| 04:22.32 | TehRabbitt | cries, no more DD-WRT :( |
| 04:22.57 | TJNII | Buy a block of static IPs. It's fun! |
| 04:23.12 | TJNII | Put that server right on the damn internet! |
| 04:24.03 | ChannelZ | TJNII: An hour away from what? |
| 04:24.13 | TJNII | ChannelZ: You. :) |
| 04:24.27 | ChannelZ | Oh. Aren't you in Ft Collins or something? Or am I thinking of someone else |
| 04:24.32 | TJNII | Yea |
| 04:24.41 | TJNII | Well, Loveland. |
| 04:24.42 | ChannelZ | it's a different timezone there?? |
| 04:24.46 | TJNII | But close enough. |
| 04:25.01 | TehRabbitt | TJNII: static IP's from comcast.. yea no thanks 0_o can you talk 10 bucks each? |
| 04:25.05 | TJNII | No, I'm talking drive time. :P |
| 04:25.09 | ChannelZ | ahahahaha ok |
| 04:25.17 | TehRabbitt | ?? |
| 04:25.18 | ChannelZ | My brain works in mysterious ways |
| 04:25.35 | TJNII | TehRabbitt: $10 each? Ouch. |
| 04:25.40 | TJNII | Get a better ISP. |
| 04:26.06 | ChannelZ | I think mine is $5. |
| 04:26.31 | TJNII | I pay $8 for 6 or something close to that. |
| 04:26.32 | voipnoob | How does one chose a codec? Is one codec enough or more than one may be neccessary? What are the considerations here? |
| 04:26.36 | TehRabbitt | TJNII: ComCrap is the only thing in this area |
| 04:26.44 | TehRabbitt | 10Mbit connection for 69.95/mo |
| 04:26.57 | TehRabbitt | dynamic IP |
| 04:27.04 | TJNII | Qwest + local ISP. Twice the ISP goodness, only half the corporate crap. |
| 04:27.06 | TehRabbitt | if you want 1 static IP... 89.99 for the same service... |
| 04:27.15 | TehRabbitt | and every addnl IP is 10 bucks more |
| 04:27.24 | TehRabbitt | so 2 static IP's = 99.95 |
| 04:27.31 | TehRabbitt | 3 = 109.95 |
| 04:27.32 | TehRabbitt | etc |
| 04:27.37 | ChannelZ | Qwest never brought > 1.5mbit to my house so I dumped the bitches a couple months ago |
| 04:27.45 | ChannelZ | And they are getting bought by CenturyLink |
| 04:27.53 | TJNII | Yea, there's a reason I've never bothered to even price out Comcast. |
| 04:27.57 | TehRabbitt | lol |
| 04:28.07 | TehRabbitt | sadly it's the ONLY ISP in the area |
| 04:28.11 | voipnoob | D-Fender/THNII/anyone else? How does one chose a codec? Is one codec enough or more than one may be neccessary? What are the considerations here? |
| 04:28.13 | TehRabbitt | unless I wanna go T1 |
| 04:28.58 | TJNII | voipnoob: Honestly? I wouldn't worry to much about it. Your termination service and phones will play a bigger part than anything else. |
| 04:29.07 | TehRabbitt | so any idea why SIP still won't work? |
| 04:29.14 | drmessano | ~sipnat |
| 04:29.15 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 04:29.20 | drmessano | Follow the guide |
| 04:29.39 | *** join/#asterisk cmsouza (~chatzilla@189.121.152.138) |
| 04:29.41 | TehRabbitt | drmessano: followed it, several times.... it's beyond the guide |
| 04:29.44 | voipnoob | Tx, TJNII -> so now I need to calculate the bandwidth requirement for my Asterisk Server in my country & kind of connection I need - how can i do this calculation? |
| 04:30.54 | drmessano | TehRabbitt: Doubtful.. There's only a few settings that apply here.. Most follow some interpretation of the basic settings but never have it correct until they actually do what's required |
| 04:30.55 | voipnoob | let me repeat my setup -> |
| 04:30.56 | voipnoob | For an Asterisk Server on the internet which will serve 1000 customers over the internet |
| 04:30.56 | voipnoob | with 150 max concurrent calls. The Asterisk Server will not be trunked directly. |
| 04:30.56 | voipnoob | It will be connected to another Asterisk Server in a different country which will be |
| 04:30.56 | voipnoob | doing all the termination & SIP trunking. |
| 04:30.56 | voipnoob | What kind of Bandwidth usage & Internet Connection would be required for the first |
| 04:30.57 | voipnoob | Asterisk Server? |
| 04:31.41 | TJNII | voipnoob: Well, I don't know much about codecs but I wouldn't use anything with a bandwidth lower than ulaw. I would take that bandwidth, multiply it by the expected number of calls, multiply that by %20 (like [TK]D-Fender said) and go from there. |
| 04:31.49 | ChannelZ | TehRabbitt: Huh, I pay $65/mo for 10mbit/2mbit, Comcast Business, static IP |
| 04:31.52 | TJNII | gsm sounds kinda crappy. |
| 04:32.01 | p3nguin | tehrabbitt: Actually, I bet turning off NAT turns your router into a, uhm, router without NAT. |
| 04:32.04 | drmessano | Such things as using the server IP in the localnet setting, using externhost without DNS, etc |
| 04:32.30 | drmessano | SIP config for NAT is actually very simple |
| 04:32.48 | ChannelZ | Sorry, 12mbit/2mbit |
| 04:33.00 | drmessano | I pay $59.95 for 12/2 |
| 04:33.12 | ChannelZ | dynamic yes? |
| 04:33.29 | drmessano | I have a static as well |
| 04:33.46 | ChannelZ | hmm free perk for you I guess :) |
| 04:34.00 | voipnoob | TJNI - my Asterisk Server would be in Asia - All the calls would be either local to the country IP to IP calls or international calls forwarded to the Second Asterisk Server in Europe which would be run by the partner - he would take care of the 2nd one & it's trunking & termination |
| 04:34.12 | ChannelZ | Although inexplicably they bill mine as $14.95 with a $10 credit... because apparently that makes sense to them |
| 04:34.47 | cmsouza | voipnoob: if you are going to reinvite all the calls your bandwidth is unlikely to be a problem... |
| 04:34.51 | voipnoob | Consideing ulaw, should my calculation be 64Kbps * 150 |
| 04:34.53 | TJNII | voipnoob: I would try to get an idea of latency & jitter. I think that will kill you more than bandwidth. I don't have a good suggestion for doing so, there are some testers on the intarnets. |
| 04:34.56 | drmessano | I had so many problems with the home connection they've credited my Static IP for a couple years now |
| 04:35.15 | ChannelZ | nice |
| 04:35.25 | ChannelZ | problems not nice, but anyway.. |
| 04:35.27 | TJNII | goes off to bed. Busy day tomorrow. |
| 04:35.31 | voipnoob | Yes, TJNII - i will do that - as of now I am trying to write a basic requirements documents. |
| 04:35.32 | ChannelZ | nighty night |
| 04:36.05 | drmessano | I think they're not used to have Business Class boxes in home installations, and they're actually having to support a crappy, ancient neighborhood build-out |
| 04:36.14 | drmessano | having* |
| 04:36.22 | voipnoob | Is there any one else who can help me with my requirements? - now that TJNII Is off to bed |
| 04:36.40 | voipnoob | i am trying to figure out the bandwidth requirements |
| 04:37.25 | drmessano | voipnoob: calc.exe should be able to give you 64 x 150 |
| 04:37.30 | ChannelZ | I think he pretty much said what needed to be said |
| 04:37.57 | ChannelZ | and unless you're in the media stream of all those calls it's less than that |
| 04:38.02 | cmsouza | drmessano: windows user detected kkkk |
| 04:38.26 | drmessano | It's always worth a shot |
| 04:38.42 | voipnoob | drmessano -> So that would be around 10 Mbps |
| 04:39.10 | drmessano | voipnoob: If that is what the math tells you |
| 04:39.15 | cmsouza | voipnoob: as ChannelZ said, if you are going to reinvite the calls, your bandwidth will not be used for the voice part of the call, and so this is moot. |
| 04:40.20 | voipnoob | what do you mean -> reinvite the calls? |
| 04:40.29 | cmsouza | ~reinvite |
| 04:40.38 | drmessano | ~whatnow |
| 04:40.39 | infobot | extra, extra, read all about it, now is a good time to tell you that I have 6 gigabytes of data |
| 04:40.41 | *** join/#asterisk KingDavidNYC (~Chris1232@pool-71-190-78-39.nycmny.east.verizon.net) |
| 04:40.42 | cmsouza | ow bot... let me down this time |
| 04:40.48 | drmessano | haha |
| 04:40.48 | voipnoob | :-) |
| 04:40.52 | drmessano | ~book |
| 04:40.53 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 04:40.55 | drmessano | ^^^^^ |
| 04:41.00 | drmessano | ~drmessano |
| 04:41.01 | infobot | [drmessano] the leading cause of censorship in #asterisk, maybe a Doctor, not really a mess um a no..... The Doctor is accepting new patients, <drmessano> I am not a OB/GYN but I'll have a look anyway, and is earning his reputation daily |
| 04:41.01 | voipnoob | i read the asterisk book once |
| 04:41.43 | ChannelZ | reinvite == Asterisk gets the two end points in direct contact with each other for their media streams as opposed to them passing through Asterisk |
| 04:41.50 | drmessano | voipnoob: Maybe you should read it again before implementing this.. reinvite is a basic concept |
| 04:42.07 | drmessano | stabs autofs |
| 04:42.17 | voipnoob | yes, i saw the canreinvite parameter in the book |
| 04:42.22 | voipnoob | it's not explained in much details |
| 04:43.02 | ChannelZ | just realized Comcast prints his bills on a color inkjet printer.. mine looks better |
| 04:43.21 | voipnoob | For my IP->IP calls within the country - those would all go through the asterisk server, right? |
| 04:43.38 | voipnoob | the reinvite thing isn't relevant there, right? |
| 04:43.43 | drmessano | Comcast has been spamming me with paper bills too |
| 04:44.09 | drmessano | I am supposed to be set up for "not kill a tree mode" |
| 04:44.40 | ChannelZ | drmessano: well I somehow got signed up for 'EcoBill' even though I never asked (after getting 2 mailed bills) but they never actually sent me email that a new bill was available... |
| 04:45.10 | voipnoob | Assuming that out of my 150 concurrent calls 1/2 are IP to IP calls within the country -> then i would require 64Kps * 75 = approx 5Mbps bandwidth? |
| 04:45.17 | voipnoob | is that the right calculation? |
| 04:45.19 | *** join/#asterisk frk2 (~faraz@zivios/member/fkhan) |
| 04:45.33 | KingDavidNYC | hi everyone, quick question: I use system() in the dialplan to send a url out....I do system(wget ${url}).. does anyone here thinks of a better method?? |
| 04:45.41 | drmessano | Shoot, I needed to change the startup to /etc/init.d/EcoBill NoPaperBillKTHX |
| 04:45.50 | voipnoob | for the other calls, the would be passed over to the Asterisk server in Europe |
| 04:46.14 | voipnoob | so should I not allocate any bandwidth requirements for them at all? |
| 04:46.37 | *** join/#asterisk sourcode (~code@ppp-58-8-237-71.revip2.asianet.co.th) |
| 04:47.05 | drmessano | voipnoob: if 150 concurrent calls are passing through ONE of the servers, you calculate based on 150 calls |
| 04:47.11 | drmessano | Theres no "new math" here |
| 04:48.10 | cmsouza | voipnoob: i`ll type slowly, if you dont follow let me know |
| 04:48.27 | voipnoob | i am really confused cmsouza, drmessano |
| 04:48.35 | voipnoob | i am just starting to read up on these |
| 04:48.49 | drmessano | Its math |
| 04:48.50 | voipnoob | sorry if i my questions are very silly |
| 04:48.56 | drmessano | 150 concurrent calls = 150 times 1 call |
| 04:49.07 | drmessano | 1 call = 64k, then 150 x 64 |
| 04:49.17 | voipnoob | yes, but cmsouza said if a reinvite is there then i shouldn't worry about bandwidth for those calls |
| 04:49.33 | drmessano | Yeah, and you're not reinviting any of those calls |
| 04:49.33 | cmsouza | voipnoob: when a call starts, your * server has to know about it, guess what it should do with it and then send the remote host some message that says... somebody is trying to talk to you.. thats nice thats SIP |
| 04:50.57 | cmsouza | but a voip call is composed of more than just SIP, it has RTP also, RTP stands for REAL TIME PROTOCOL, this is where your bw matters |
| 04:50.59 | TehRabbitt | p3nguin: any way you could send me those ringtones? lol |
| 04:51.29 | p3nguin | Oh, I forgot. I'll work on it right now. |
| 04:52.20 | ChannelZ | here's a good one |
| 04:52.26 | cmsouza | if you reinvite the call, you send the audio portion of the call directly to the destination, it does not pass through your * server, does not waste your bw and therefore cannot be recorded and things alike (listening to audio DTMF etc) |
| 04:52.28 | ChannelZ | http://burner.com/snd_xbc.wav |
| 04:52.37 | TehRabbitt | p3nguin: aightt :) |
| 04:54.06 | cmsouza | handling reinvited 150 calls is trifle, but if you are going to process all those calls you`ll need what the math tells you. 10Mbit Up (sending the call to europe or wherever) and 10Mbit Down (receving the calls from your customers) |
| 04:55.55 | TehRabbitt | p3nguin: are they MP3 format or WAV btw? |
| 04:58.42 | ChannelZ | cmsouza: I think your math is flawed |
| 04:58.46 | voipnoob | cmsouza -> that was why I divided 150/2 & got 75. I was making an assumption that 50% of my calls would be IP->IP calls within my country |
| 04:59.03 | voipnoob | so it would be 75 * 64 |
| 04:59.21 | cmsouza | i didn`t do the math i just copied whathever someone wrote up there |
| 04:59.42 | ChannelZ | 64kbit = 8000 bytes |
| 05:00.26 | *** join/#asterisk githogori (~githogori@adsl-66-123-22-146.dsl.snfc21.pacbell.net) |
| 05:00.28 | voipnoob | ChannelZ what's the right calculation? |
| 05:01.20 | cmsouza | voipnoob: it does not matter if its within your country or else, if its IP->IP you can simply reinvite the call and free your bw. assume i am your customer and need to talk to ChannelZ who is also your customer |
| 05:01.22 | ChannelZ | For ulaw/alaw 8k/sec * calls = bytes/sec required |
| 05:01.56 | ChannelZ | so 150 calls is ~1.2mbit raw not including overhead |
| 05:02.53 | cmsouza | if i call ChannelZ i will use 64kbps or 8kBps and he will use another 8kBps but if the audio portion of the call is reinvited, it will go directly from me to channelZ and back, your bw will be untouched |
| 05:03.17 | coppice | ChannelZ: 1.2m BYTES/s |
| 05:03.58 | voipnoob | So channelZ, so if 50% of the calls are IP->IP, I would need .6 Mbps bandwidth |
| 05:03.59 | ChannelZ | yes sorry |
| 05:04.17 | ChannelZ | you did say mbit above, I misread |
| 05:04.34 | voipnoob | or is it .6 MBps? |
| 05:05.14 | ChannelZ | yes. So ~4.8mbit |
| 05:05.17 | cmsouza | capital B for bytes, but you probably buy bw by mbps nor mBps |
| 05:05.34 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-ubuoekjwwaamuwde) |
| 05:05.45 | ChannelZ | I'm having brain issues today |
| 05:07.08 | TehRabbitt | p3nguin: did you need my email again for the ringtones or did you save it lol |
| 05:07.19 | cmsouza | what i dont get is this 50% you are assuming... |
| 05:07.46 | *** join/#asterisk Milad (~milad@unaffiliated/slackark) |
| 05:10.42 | voipnoob | cmsouza -> I thought only IP-IP calls will be reinvited |
| 05:10.53 | voipnoob | i assumed 50% of my calls with IP->IP calls |
| 05:11.06 | cmsouza | and the other 50%? |
| 05:11.23 | voipnoob | the other 50% calls will be calls to US/UK/Europe (possibly to landlines/cellphones) |
| 05:11.38 | voipnoob | so can those also be re-invited? |
| 05:11.48 | voipnoob | will i not need to carry audio there also/ |
| 05:12.34 | cmsouza | now i understand it. if they are calls to landline phones no.. they should pass your * be forwarded to the other * in europe and then to POTS |
| 05:12.43 | ChannelZ | it depends are you terminating those calls yourself? Like on an E1? |
| 05:13.02 | voipnoob | channelZ - let me explain my scenario again |
| 05:13.05 | cmsouza | but, if they are calls to SIP extensions in europe, yes, they can and should be reinvited |
| 05:13.22 | voipnoob | My customers are in 1 country in Asia |
| 05:13.30 | voipnoob | I have 1 Asterisk server in the same country |
| 05:14.09 | voipnoob | IN can't Terminate to PSTN in my country for legal issues |
| 05:14.20 | voipnoob | so there will be no IP->landline calls inside the country |
| 05:14.34 | voipnoob | the IP -> landline calls will only be to foreign countries |
| 05:15.04 | voipnoob | My Countries Asterisk server would be connected to an Asterisk Server in Europe -> the European Asterisk Server will do the SIP Trunking/Terminations etc |
| 05:15.47 | voipnoob | My Country Asterisk Server would be on the internet. My 1000 customers with 150 concurrent calls will connect to my 1 Asterisk server through their internet connections |
| 05:15.55 | voipnoob | cmsouza/ChannelZ -> is this clear |
| 05:17.12 | voipnoob | Now assuming X% of my calls would be IP->IP in the country, Y% would be IP->landlines/cellphones outside the country & 100-X-Y would be IP->IPAbroad |
| 05:17.39 | voipnoob | how do I calculate the bandwidth required for my Asterisk Server & what sort of connection should I buy for hosting the Asterisk Server |
| 05:19.24 | TehRabbitt | p3nguin: ok, so apparenlty me flashing my router messed EVERYTHING up |
| 05:19.25 | cmsouza | calculate the bw required for the Y% of your calls, leave a safety margin, be sure you are capable of expanding this bandwidth if needed start your business |
| 05:19.29 | TehRabbitt | now i can't get incoming SIP @ all |
| 05:19.33 | TehRabbitt | hell, Gizmo can't even rout in |
| 05:19.45 | TehRabbitt | :( |
| 05:20.14 | voipnoob | cmsouza -> I don't need to provide at all for the X% & 100-X-Y? |
| 05:20.18 | voipnoob | only the Y |
| 05:20.57 | TehRabbitt | <--- SIP read from 24.171.65.27:5060 ---> |
| 05:20.58 | TehRabbitt | SIP/2.0 489 Bad event |
| 05:21.02 | TehRabbitt | :( |
| 05:21.38 | voipnoob | Assuming Y is 50% -> so that's 75 concurrent calls -> that's 75 * 64Kbps - so that's 4.7 Mbps |
| 05:21.51 | voipnoob | is that the right calculation? |
| 05:21.55 | cmsouza | just a few kbps for control messages, and yes, there is another issue, if both SIP phones are behind NATs the call will have to pass your * server, this is why you should leave 20% - 30% of safety margin |
| 05:22.44 | voipnoob | 20-30% because you are assuming that 20 to 30% of my 75 calls will be from behind NATs? |
| 05:22.50 | voipnoob | is that your assumption? |
| 05:23.24 | voipnoob | So overall I would need around 6 Mbps? |
| 05:23.34 | cmsouza | no.. just a reasonable margin if you want to be conservative buy 50% more bw |
| 05:23.44 | voipnoob | ok |
| 05:23.59 | voipnoob | Thanks a lot!!!!!! |
| 05:24.26 | cmsouza | i believe 6mbps should do the trick. anyway be aware of latency issues |
| 05:24.48 | cmsouza | satellite connections etc will probably bring you nightmares |
| 05:25.15 | voipnoob | what kind of connection should I buy for the Asterisk Server? |
| 05:25.38 | cmsouza | i would be more concerned about the quaslity and stability of your connection than about speed alone |
| 05:25.40 | voipnoob | should it be a T2 then? |
| 05:25.43 | TehRabbitt | anyone have any ideas why after flashing my firmware FROM DDWRT to Stock Linksys firmware now i can't receive calls FROM gizmo but I can make outgoing calls? |
| 05:26.13 | cmsouza | if you have access to this kind of stuff |
| 05:26.27 | ChannelZ | TehRabbitt: port forwarding |
| 05:26.28 | cmsouza | here in brazil they are REALLY expensive... |
| 05:26.45 | voipnoob | I am pretty sure T1 & E1 are avilable |
| 05:27.00 | TehRabbitt | ChannelZ: they are fowarded properly... |
| 05:27.04 | voipnoob | if it's very expensive - what's the alternative? |
| 05:27.08 | TehRabbitt | it was the first thing I did |
| 05:27.18 | cmsouza | but too damm slow |
| 05:27.28 | cmsouza | T1 = 2mpbs |
| 05:27.39 | voipnoob | cmsouza - what would be the alternative? |
| 05:27.44 | *** join/#asterisk blaines (~blaines@64-130-243-194.sparkplugbb.net) |
| 05:27.44 | ChannelZ | TehRabbitt: Maybe you set it up, but it might not be WORKING properly |
| 05:27.48 | voipnoob | if T1/T2 is very expensive/ |
| 05:27.49 | ChannelZ | I dunno |
| 05:28.19 | Milad | is there anyway to different hangup from 2 SIP channel ? I want to know which channel hangups the call |
| 05:29.21 | TehRabbitt | ChannelZ: so should I flash BACK to DDWRT against 90% of the general concensus here? |
| 05:30.00 | cmsouza | whenever i setup a * server that needs a reliable connection i buy 2 or more DSL or cable modem connections (about 100US$/4mpbs here in brazil) and set up a multi wan router that guarantees they are properly handled |
| 05:31.20 | voipnoob | DSL/Cable modems here have a speed of 1 to 2 Mbps |
| 05:31.35 | voipnoob | 2 of those should be enough then, right? |
| 05:31.45 | TehRabbitt | p3nguin: you still here? |
| 05:31.48 | *** join/#asterisk Tim_Toady (~moi@188.4.66.69.dsl.dyn.forthnet.gr) |
| 05:32.40 | cmsouza | yes.. be sure to buy from 2 different companies, so if one is having issues the other will hold your service up, slow, but up |
| 05:32.47 | voipnoob | ok |
| 05:32.57 | voipnoob | cmsouza - For Asterisk server hardware - i am looking at QuadCore 4GB box |
| 05:33.08 | voipnoob | is that enough/too much/too less |
| 05:33.15 | cmsouza | beautiful |
| 05:33.30 | voipnoob | do i need a RAID or something to go with it? |
| 05:33.43 | cmsouza | my first * started on ald celeron i found on the basement |
| 05:33.47 | voipnoob | :-) |
| 05:33.50 | ChannelZ | TehRabbitt: I don't know what that consensus was or was about... nor do I know why 'stock' it shouldn't work either all things being equal |
| 05:34.19 | TehRabbitt | 'stock' is very limited / has the Linksys firewall which is very restrictive |
| 05:34.20 | cmsouza | could handle 30 or more concurrent calls.. |
| 05:34.34 | TehRabbitt | also my signal went from 80% signal strength to 20% |
| 05:34.36 | TehRabbitt | same location |
| 05:35.11 | voipnoob | cmsouza -> i can also have a Flash Client which communicates through a propreitary protocol to an application sitting on top of the Asterisk Server -> the 20-30% margin of safety should take care of that bandwidth also, right? |
| 05:35.15 | cmsouza | if uptime is an issue raid 1 would help you |
| 05:35.26 | voipnoob | ok |
| 05:35.27 | ChannelZ | wait.. is your * connecting via wireless through your linksys to someplace else? |
| 05:35.48 | TehRabbitt | ChannelZ: Yes, Point to Point Network Bridge.. Very Reliable and Very Fast |
| 05:36.05 | TehRabbitt | Running cat5 is NOT an option, and neither is powerline ethernet |
| 05:36.14 | cmsouza | ChannelZ: TehRabbit has w wireless bridge between 2 routers at home |
| 05:36.33 | ChannelZ | sooo the router you're port forwarding on is different than the one your * is connected to |
| 05:36.41 | voipnoob | Another question -> if the communication needs to be encrypted (ssl or IPSec) between the client & Asterisk server - how much more bandwidth should be calculated for that |
| 05:36.44 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 05:36.48 | voipnoob | or is that already in the calculation? |
| 05:36.49 | ChannelZ | if the one * is connected to is truly acting as a bridge and not a router. |
| 05:36.56 | ManxPower | ~bandwidth |
| 05:36.57 | infobot | [~bandwidth] This is a measure in bits per second of the amount of data that can be sent over a particular cable, interface, or bus. |
| 05:37.05 | ManxPower | ~overhead |
| 05:37.06 | TehRabbitt | ChannelZ: it's truely acting as a bridge, NOT a router |
| 05:37.19 | ManxPower | there is a bandwidth calculator on voip-info.org |
| 05:37.24 | voipnoob | i.e. is the 64 Kbps - for unencrypted or encrypted? |
| 05:37.56 | cmsouza | voipnoob: IPSEC adds overhead 64 kbps is unencrypted 711-ulaw |
| 05:38.02 | *** part/#asterisk Milad (~milad@unaffiliated/slackark) |
| 05:38.05 | ManxPower | voipnoob, you do not need to use 64Kbps for calls |
| 05:38.18 | *** join/#asterisk Milad (~milad@unaffiliated/slackark) |
| 05:38.23 | ManxPower | and ulaw is about 80Kbps per call with the UDP and RTP overhead |
| 05:38.24 | ChannelZ | [internet] --> some router which should be doing port forwarding --> (wireless bridge) --> router * is connected to --> * server |
| 05:38.34 | cmsouza | ManxPower: he will be selling the service, i recommend against 729 or gsm |
| 05:38.43 | cmsouza | i believe quality is an issue here |
| 05:38.47 | ManxPower | cmsouza, then he had better get some accurate numbers |
| 05:38.48 | *** join/#asterisk kotp (~vgoff@96.2.187.66) |
| 05:39.24 | xargs | I got my asterisk setup working but my outbound audio isn't working |
| 05:39.38 | voipnoob | manzpower - for now i am just doing back of the napkin calculations - will hire a more knowledgable person for the final thing |
| 05:39.45 | cmsouza | ManxPower: you`re right |
| 05:39.59 | voipnoob | cmsouza - how much overhead more than 64Kbps for encrypted? |
| 05:40.18 | p3nguin | tehrabbitt: Yes, I'll be with you in a moment. |
| 05:40.38 | cmsouza | voipnoob: do all your calculations allowing for 10mbps you'll be on the safe side and your customers wont complain |
| 05:40.41 | voipnoob | MaxnPower-> what do you mean - you do not need to use 64Kbps for calls? |
| 05:40.45 | TehRabbitt | kk |
| 05:40.58 | ManxPower | there is a bandwidth calculator on voip-info.org |
| 05:41.08 | ManxPower | pick your codec, it will calculate the overhead |
| 05:41.47 | voipnoob | seems to be down -> http://blog.asteriskguide.com/bandcalc/bandcalc.php |
| 05:42.29 | ManxPower | Oh! Well then I'll be happy to take several hours of my time teaching you all about protocol overhead and codecs! |
| 05:42.48 | ManxPower | ~book |
| 05:42.49 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 05:42.50 | ManxPower | ~answers |
| 05:42.51 | infobot | [~answers] Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & Docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: /path/to/src/asterisk/UPGRADE*.txt |
| 05:42.52 | ManxPower | ~mailinglist |
| 05:42.53 | infobot | [~mailinglist] The Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 05:43.09 | ManxPower | 'night |
| 05:43.16 | voipnoob | what kind of % should I calculate for encryption overhead? |
| 05:43.44 | cmsouza | ipsec with aes some 5% |
| 05:44.23 | voipnoob | is that all? |
| 05:44.32 | cmsouza | depending on packet size it can go up to 15% |
| 05:44.41 | voipnoob | http://www.cisco.com/en/US/technologies/tk543/tk759/technologies_white_paper0900aecd80295ab0.pdf |
| 05:44.45 | voipnoob | this says it can double |
| 05:44.59 | voipnoob | for IPSec tunnening? |
| 05:45.07 | voipnoob | tunneling |
| 05:45.22 | voipnoob | i.e. 24Kbps can become 56Kbps |
| 05:46.13 | cmsouza | RTP packets are likely to be big, which reduces the overhead |
| 05:46.55 | cmsouza | but yes, in your worst case scenario it can double, although unlikely, i think |
| 05:47.29 | voipnoob | so I should assume for 25% |
| 05:47.58 | voipnoob | thanks again |
| 05:48.09 | TehRabbitt | p3nguin: I can't get inbound SIP working now at *all* esp gizmo |
| 05:48.42 | xargs | what does it mean when I see "Outbound Reregistration: Expiry for sip-provider.siphost.com is 60 sec (Scheduling reregistration in 45 s) |
| 05:49.51 | TehRabbitt | p3nguin: all my SIP connections are showing 404 errors |
| 05:50.13 | xargs | TehRabbitt: I got inbound SIP working... but my outbound audio is broken :( |
| 05:51.26 | TehRabbitt | xargs: how did you get inbound SIP working? |
| 05:51.33 | TehRabbitt | my outbound works great, i can't get inbound |
| 05:51.47 | xargs | TehRabbitt: you have to register with a sip provider |
| 05:52.01 | xargs | well |
| 05:52.03 | TehRabbitt | xargs: I am registered... it's returning "404 not found' errors |
| 05:52.15 | TehRabbitt | Gizmo5 is my incoming DID |
| 05:52.26 | TehRabbitt | worked *great* prior to reflashing my router |
| 05:52.49 | xargs | you might have the ports blocked? |
| 05:54.31 | p3nguin | tehrabbitt: Check your /notice window. |
| 05:56.08 | TehRabbitt | p3nguin: thanks.. |
| 05:56.15 | TehRabbitt | xargs: nope all ports are open |
| 05:56.57 | TehRabbitt | :-D |
| 05:57.51 | p3nguin | I included all the files even though the xml doesn't use them all. |
| 05:58.02 | p3nguin | Feel free to try to make them all work. I couldn't. |
| 05:58.51 | TehRabbitt | lol aightt |
| 05:59.00 | TehRabbitt | p3nguin: the only problem? my incoming line wont ring anymore :( |
| 05:59.10 | TehRabbitt | 404 error from Gizmo apparently |
| 05:59.23 | p3nguin | Maybe you broke something. |
| 05:59.31 | TehRabbitt | I havent touched * at all |
| 05:59.36 | TehRabbitt | just the router's firmware |
| 05:59.39 | TehRabbitt | soooo yea :( |
| 06:00.01 | ChannelZ | TehRabbitt: 404 seems more definitive than 'not responding'. You're getting nothing on your console? |
| 06:01.16 | TehRabbitt | ChannelZ: nothing :( |
| 06:01.46 | TehRabbitt | wait no it's coming in... |
| 06:02.24 | ChannelZ | I'm getting 'circuit-busy' again from you which implies my traffic is getting through |
| 06:03.05 | p3nguin | tehrabbitt: Did you get the file downloaded? |
| 06:03.56 | TehRabbitt | yep |
| 06:04.08 | TehRabbitt | trying mess with getting my server working again |
| 06:04.51 | p3nguin | I just didn't want to delete it if you hadn't had a chance to get it. |
| 06:05.11 | TehRabbitt | lol np and thanks again :) |
| 06:05.43 | TehRabbitt | <PROTECTED> |
| 06:05.46 | TehRabbitt | any ideas? |
| 06:06.35 | TehRabbitt | it's almost as if it's coming in but the dialplan doesn't know where to route the call anymore |
| 06:07.04 | p3nguin | That's all revealed in debug. |
| 06:08.28 | *** join/#asterisk gelo (~gelo@209.138.60.213.dynamic.mundo-r.com) |
| 06:08.47 | TehRabbitt | yep |
| 06:08.47 | TehRabbitt | :( |
| 06:10.17 | p3nguin | Your debug isn't complete, but I can see that calls do come in. |
| 06:11.34 | TehRabbitt | yea, idk why it's not going to SCCP anymore |
| 06:11.35 | xargs | YAYAYA |
| 06:11.40 | xargs | I got my outbound audio fixed |
| 06:11.44 | p3nguin | tehrabbitt: check /notice again. |
| 06:12.16 | p3nguin | Show me a complete sip debug of a failed call. |
| 06:12.52 | ChannelZ | then shake your money makers and show me some sugar |
| 06:14.06 | TehRabbitt | here it is... |
| 06:14.07 | TehRabbitt | http://pastebin.com/06rKLbCp |
| 06:15.04 | p3nguin | Why are you trying to call your Gizmo SIP number via VoIP.ms? |
| 06:15.40 | TehRabbitt | <PROTECTED> |
| 06:15.45 | TehRabbitt | 0_o |
| 06:15.48 | p3nguin | The debug says you are. |
| 06:15.49 | TehRabbitt | did i mess something up? heh |
| 06:16.08 | ChannelZ | He likes to make things as difficult as possible to debug. Why do straightforward tests and add features incrementally when you can create a Rube Goldberg config to begin with and try to make it work all at once? |
| 06:16.30 | TehRabbitt | LOL |
| 06:17.04 | ChannelZ | Wireless routers and firewalls and port forwarding and Skinny and remote service providers.... |
| 06:17.25 | ChannelZ | and 15 different contexts just to keep things interesting |
| 06:17.35 | TehRabbitt | heh |
| 06:18.15 | ChannelZ | anyways the last paste looks like an outgoing call? |
| 06:18.19 | ChannelZ | and again incomplete |
| 06:19.23 | p3nguin | Yeah outgoing to a phone number which is non-existent on the PSTN. |
| 06:19.43 | TehRabbitt | hmpp got it working |
| 06:19.56 | TehRabbitt | just added "9" to outgoing calls |
| 06:20.09 | TehRabbitt | so now to dial "out" you have to dial 9 first lol |
| 06:20.20 | ChannelZ | p3nguin: I just called it... |
| 06:20.34 | p3nguin | Wouldn't it be better to do things the right way from the beginning? |
| 06:20.46 | TehRabbitt | ok nvm 9 doesn't work... :( |
| 06:21.00 | TehRabbitt | exten => 9_1NXXNXXXXXX,1,Dial(SIP/voipms/${EXTEN}) |
| 06:21.09 | p3nguin | pattern fail |
| 06:21.19 | p3nguin | channelz: You called 17474945623? |
| 06:21.20 | ChannelZ | _91NXXNXXXXXX |
| 06:21.28 | TehRabbitt | ah |
| 06:21.48 | ChannelZ | p3nguin: Not with a 1 no, but does voip.ms require a 1 or only 10-digit? I have no idea, they're not my provider. |
| 06:21.54 | p3nguin | exten => _91NXXNXXXXXX,1,Dial(SIP/voipms/${EXTEN:1}) |
| 06:22.13 | p3nguin | channelz: That's a Gizmo5 SIP phone number. |
| 06:22.14 | ChannelZ | My guess if it's not working they don't want the 1 |
| 06:22.31 | ChannelZ | eh? |
| 06:22.42 | TehRabbitt | ok doesnt wanna work again :( |
| 06:23.01 | TehRabbitt | so it's something with my dialplan lol |
| 06:23.23 | p3nguin | I think VoIP.ms actually accepts both 10- and 11-digit dial. |
| 06:23.30 | TehRabbitt | yea they do |
| 06:23.38 | TehRabbitt | the +9 prefix wont work right |
| 06:23.41 | ChannelZ | i dunno what the fuck he's dialing and now I don't care anymore |
| 06:23.42 | p3nguin | but they do require the phone numbers to be valid in order to terminate. |
| 06:24.13 | ChannelZ | Incomplete debugs, no meaningful console output |
| 06:24.26 | p3nguin | screwing up configs |
| 06:24.46 | p3nguin | dialing Gizmo5 SIP numbers on the PSTN |
| 06:24.47 | TehRabbitt | ChannelZ: i'm posting everyting the CLI gives me |
| 06:24.51 | TehRabbitt | EVERYTHING |
| 06:24.54 | p3nguin | Start over. |
| 06:24.59 | TehRabbitt | start over? |
| 06:25.02 | p3nguin | right. |
| 06:25.03 | TehRabbitt | 0_o |
| 06:25.05 | ChannelZ | We're not seeing your initial INVITE, only the rejected reply. |
| 06:25.12 | ChannelZ | You appear to have no verbosity turned on either. |
| 06:25.18 | TehRabbitt | oh |
| 06:25.43 | TehRabbitt | the putty buffer only buffers so much as well... and it scrolls rather quickly so it's hard to catch it all |
| 06:25.48 | p3nguin | increase it |
| 06:26.09 | p3nguin | 100000 line should be enough. |
| 06:26.14 | TehRabbitt | k |
| 06:26.15 | p3nguin | lines |
| 06:27.18 | TehRabbitt | doesnt give me an option :-\ |
| 06:27.25 | p3nguin | Sure it does. |
| 06:27.48 | ChannelZ | Change Settings, Window |
| 06:28.07 | ChannelZ | "Lines of scrollback" |
| 06:29.06 | ChannelZ | SHOWER TIME! |
| 06:29.14 | ChannelZ | (it's like Hammer Time but with less pants) |
| 06:33.25 | TehRabbitt | http://pastebin.com/2a9A6Zmi |
| 06:33.30 | TehRabbitt | better? |
| 06:33.39 | p3nguin | http://imagebin.org/95809 |
| 06:35.19 | p3nguin | Looking for 17474945623 in users |
| 06:35.55 | TehRabbitt | better?? http://pastebin.com/gALw0Pp9 |
| 06:35.56 | TehRabbitt | lol |
| 06:36.28 | TehRabbitt | wait why would it be looking for that # in users? |
| 06:36.40 | p3nguin | The call is going to that SIP number. |
| 06:37.17 | p3nguin | Did you download that 25M file? |
| 06:37.34 | TehRabbitt | nvm i figured out what to do |
| 06:37.48 | TehRabbitt | and yea i donloaded it.. it's asking for a custom HTML URL to configure it though :-\ |
| 06:37.56 | p3nguin | I'm about to delete it. |
| 06:38.06 | p3nguin | What's asking for a custom URL? |
| 06:38.19 | *** part/#asterisk jmls (~jmls@host217-36-208-155.in-addr.btopenworld.com) |
| 06:38.46 | p3nguin | If you run the .exe file, it should extract and allow you to run setup. |
| 06:39.42 | p3nguin | Paste your sip.conf and extensions.conf again. I have to go soon, so I'll just configure them for you. :/ |
| 06:41.39 | TehRabbitt | figured out how to fix it :-D |
| 06:42.04 | TehRabbitt | It was a context issue that I overlooked when I setup voip.ms |
| 06:42.24 | TehRabbitt | the incoming and outgoing were using the same exact context... so it was matching it as an outgoing call |
| 06:42.36 | TehRabbitt | soooo it was coming in and imeditally rerouting it outward lol |
| 06:44.40 | TehRabbitt | p3nguin: the smartphone is asking for a config URL |
| 06:44.53 | TehRabbitt | during setup |
| 06:45.33 | p3nguin | I don't use it, so I don't know much about it. |
| 06:45.57 | TehRabbitt | hm |
| 06:46.03 | p3nguin | I just have a way with grabbing files that I think might be useful at some point, so that's why I had it. |
| 06:46.31 | TehRabbitt | btw... SetCallerID("number") do i place that at the *end* of a Dial()? |
| 06:46.37 | TehRabbitt | or before it |
| 06:46.41 | p3nguin | neither |
| 06:46.46 | TehRabbitt | where do I put it? |
| 06:47.10 | p3nguin | Set(CALLERID(num)=1234567890) in a priority before the Dial() command. |
| 06:47.59 | TehRabbitt | so how would that look? |
| 06:48.26 | TehRabbitt | exten => Set(CALLERID(num)=1234567890)? |
| 06:48.48 | p3nguin | exten _1NXXNXXXXXX,1,Set(CALLERID(num)=1234567890) |
| 06:48.50 | p3nguin | exten _1NXXNXXXXXX,n,Dial(SIP/voipms/${EXTEN}) |
| 06:49.06 | TehRabbitt | ah |
| 06:50.00 | p3nguin | If you wanted to do 7-digit dial on the 284 area code... |
| 06:50.02 | p3nguin | exten _NXXXXXX,1,Set(CALLERID(num)=1234567890) |
| 06:50.05 | p3nguin | exten _NXXXXXX,n,Dial(SIP/voipms/1284${EXTEN}) |
| 06:51.43 | p3nguin | fuxck. I left out the => in the extens. |
| 06:52.03 | p3nguin | This is what sleep deprivations causes. |
| 06:52.07 | p3nguin | -s |
| 06:52.21 | TehRabbitt | w00t lemme see if it works :) |
| 06:53.09 | p3nguin | Did you get them to activate the account? |
| 06:53.15 | TehRabbitt | sweet, I have 2-way calls now... now to only get external SIP working haha |
| 06:53.36 | TehRabbitt | and MoH |
| 06:53.44 | TehRabbitt | p3nguin: Yep, they activated it today.... |
| 06:53.49 | p3nguin | What was the problem? |
| 06:53.54 | p3nguin | or did they not say? |
| 06:53.56 | TehRabbitt | put 40 bucks on the account... |
| 06:54.03 | TehRabbitt | they said it's due to stricter signups |
| 06:54.07 | p3nguin | that should last you a year. |
| 06:54.09 | TehRabbitt | they've been getting alot of automated signups |
| 06:54.32 | TehRabbitt | yea, i figure 40-50 should last me till at least December from now |
| 06:54.33 | TehRabbitt | lol |
| 06:54.43 | TehRabbitt | esp since most of it's outgoing via Futurenine |
| 06:54.45 | voipnoob | I have a question about reinvite - is reinvite possible between 2 clients which are both on the internet - (i.e. not in the same intranet)? |
| 06:54.55 | voipnoob | if the Asterisk Server is also on the internet? |
| 06:55.09 | p3nguin | Do you even know what reinvites are? |
| 06:55.18 | xargs | p3nguin: I'm sleep deprived as well |
| 06:55.25 | xargs | p3nguin: my job is killing me |
| 06:55.59 | voipnoob | p3nguin - sort of -> it means that Asterisk gets the 2 clients directly in contact with each other |
| 06:56.05 | p3nguin | I'm usually more on my game, but this time I missed an important piece of syntax. |
| 06:56.05 | ChannelZ | voipnoob: if the two end points are capable of reaching each other through whatever means they are connected to the internet, yes |
| 06:56.07 | voipnoob | so the audio will not pass through the Asterisk Server |
| 06:56.19 | voipnoob | ok |
| 06:56.29 | p3nguin | So... |
| 06:56.51 | p3nguin | if you know what reinvites are and you know what it does, why would you ask what you asked? |
| 06:57.23 | voipnoob | I ask because I don't know that much about NAT/Firewalls et |
| 06:57.25 | voipnoob | etc |
| 06:57.54 | p3nguin | It has little to do with that. |
| 06:58.33 | p3nguin | If two nodes have the ability to communicate with each other directly, and if you want to enable reinvites, then let them talk to each other without asterisk being in the media stream. |
| 06:58.34 | kaldemar | it has very much to do with that if the clients are behind a NAT. |
| 06:59.35 | TehRabbitt | p3nguin: i'm exausted :( lol though now at least this is somewhat operational :-D |
| 06:59.43 | TehRabbitt | still need to get MoH working :( |
| 06:59.57 | xargs | TehRabbitt: did you get inbound fixed? |
| 07:00.01 | TehRabbitt | nope |
| 07:00.22 | TehRabbitt | well yes and no |
| 07:00.27 | TehRabbitt | inbound calls, yes |
| 07:00.32 | TehRabbitt | inbound SIP registration, No |
| 07:01.00 | p3nguin | I just registered to you. |
| 07:01.05 | p3nguin | thoth.tenehawk.com:5060 300 105 Registered Thu, 06 May 2010 02:00:51 |
| 07:01.12 | kaldemar | then again, there's the nonat option for canreinvite/directmedia. |
| 07:02.17 | *** join/#asterisk c0rnoTa (~c0rnoTa@94.25.173.118) |
| 07:03.26 | xargs | p3nguin: is it normal for reregistration to be ever 45-60 seconds? |
| 07:03.31 | xargs | *every |
| 07:03.39 | p3nguin | sure |
| 07:03.49 | *** join/#asterisk coppice (~chatzilla@m121-202-104-48.smartone-vodafone.com) |
| 07:03.51 | p3nguin | CallCentric always wanted me to reregister that often. |
| 07:04.00 | p3nguin | Then I quite using CallCentric. |
| 07:04.03 | xargs | was curious |
| 07:04.26 | xargs | Vitelity makes me rereg every 45sec |
| 07:04.30 | p3nguin | Turn off verbose and you'll stop seeing the message. |
| 07:05.41 | xargs | sip no debug |
| 07:05.43 | xargs | oops |
| 07:06.03 | p3nguin | sip set debug off? |
| 07:06.22 | xargs | no,I'm using ancient asterisk |
| 07:06.24 | xargs | 1.2 |
| 07:06.29 | xargs | sigh |
| 07:06.43 | p3nguin | oh |
| 07:06.58 | p3nguin | Any reason you don't come into the present times? |
| 07:07.42 | voipnoob | kaldemar/ChannelZ -> Assume 2 guys sitting in their homes - each getting an internet connection from their ISP -> in this case the 2 phones would be able to reach each other right -> and allow REINVITE? |
| 07:07.58 | xargs | p3nguin: this particular box is customer owned and is heavily patched with custom code |
| 07:08.14 | p3nguin | That could be a good reason to not upgrade. |
| 07:08.42 | p3nguin | If everything is working and you've backported any security patches, it's probably fine anyway. |
| 07:09.22 | kaldemar | voipnoob: only if their traffic can reach each other. |
| 07:10.56 | voipnoob | kaldemar - in the typical scenario I talked about -> 2 JoeSixpacks sitting in their basement with their internet connection there won't be anything to stop their traffic from reaching each other, right? |
| 07:11.07 | kaldemar | voipnoob: a NAT would. |
| 07:11.39 | ChannelZ | but 2 joe sixpacks sitting in their basement are looking for porn, not talking on the phone |
| 07:11.50 | p3nguin | :) |
| 07:11.51 | kaldemar | voipnoob: you can't make any assumptions on their connectivity unless you know their setup. but, if they are connected to your asterisk, you should know anyway. |
| 07:11.57 | *** join/#asterisk frk2 (~faraz@zivios/member/fkhan) |
| 07:12.04 | voipnoob | As i said before, I am asking this question because of my lack of knowledge about NAT - I know only the basics of NAT. for a typical internet user, does he have NAT |
| 07:12.22 | voipnoob | would there be a NAT at the ISP which would cause the problem |
| 07:12.44 | p3nguin | Some ISPs do NAT and give home users a private IP address. |
| 07:13.10 | voipnoob | p3nguin -> what about the majority of ISPs - do they do this or not? |
| 07:13.11 | kaldemar | voipnoob: typically, ISP's don't NAT client connections. but you never know. don't make assumptions without knowing. |
| 07:13.17 | voipnoob | ok, kaldemar |
| 07:13.20 | voipnoob | tx |
| 07:13.28 | ChannelZ | NAT is designed to allow multiple computers to share one real connection; as a byproduct they specifically prevent stray incoming traffic from reaching any of those computers. |
| 07:13.30 | kaldemar | voipnoob: but many users have NAT routers in their home. |
| 07:13.30 | p3nguin | The majority give public IP addresses to the end user. |
| 07:13.51 | voipnoob | ok, tx, everyone |
| 07:16.54 | xargs | p3nguin: the ISPs that give out a private IP address are lame |
| 07:17.00 | p3nguin | sometimes |
| 07:17.20 | xargs | screw private IPs |
| 07:17.21 | p3nguin | I see it a lot with WISPs. |
| 07:18.58 | ChannelZ | it's great for "joe sixpack" who otherwise has no idea how to maintain or secure his network |
| 07:19.38 | *** join/#asterisk b14ck (~rdegges@cpe-24-24-128-47.socal.res.rr.com) |
| 07:20.13 | b14ck | Sup all. I had a kinda weird question (maybe). But is there a way I can launch an AGI script from dialplan and run it in the background as a separate thread? Instead of having Asterisk wait for the AGI to finish executing before continuing with dialplan? |
| 07:20.24 | p3nguin | We have a local fiber ISP that normally gives you a private IP unless you ask for public. And if you ask for public, they give you a /29 and don't bother to tell you. |
| 07:20.24 | tuxx- | fork it. |
| 07:20.26 | tuxx- | :P |
| 07:20.35 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 07:20.38 | tuxx- | b14ck: use forking :) |
| 07:20.41 | b14ck | How do I *fork it* via the AGI() command? I can't just add & onto it |
| 07:20.46 | b14ck | It isn't a system() |
| 07:20.48 | b14ck | It's an AGI |
| 07:21.06 | tuxx- | i know, in your script fork it, not from the agi command |
| 07:21.06 | tuxx- | :P |
| 07:21.17 | b14ck | that's sloppy =/ |
| 07:21.20 | b14ck | there must be a better way! |
| 07:21.29 | tuxx- | if you find one, report back ;-) |
| 07:21.34 | p3nguin | I don't know why they won't bother telling the customer that they have a static /29. That seems like valuable information to me. |
| 07:21.36 | b14ck | anyone else? |
| 07:22.15 | fenrus | LOL |
| 07:22.22 | fenrus | an isp that hands out private ip-addresses |
| 07:22.29 | fenrus | retards |
| 07:22.30 | xargs | b14ck: you can also just use setsid |
| 07:22.49 | b14ck | xargs, what do you mean? |
| 07:22.55 | p3nguin | It's also quite common in a lot of middle eastern countries. |
| 07:23.09 | fenrus | morons |
| 07:23.18 | xargs | p3nguin: because they are a bunch of commies and want to control everythihg |
| 07:23.23 | ChannelZ | Well that settles it |
| 07:23.37 | kaldemar | fenrus: they may ask a little extra for a public address. |
| 07:23.44 | fenrus | xargs, you can do exactly the same controlling with non rfc1918 -addresses. |
| 07:24.08 | fenrus | kaldemar, they are allowed to charge for the extra work with the set up. |
| 07:24.34 | p3nguin | They could also charge for not having to do any work. |
| 07:24.37 | xargs | b14ck: yeah it's a unix command, it basically forks any process and makes the new process have a parent pid of 1 |
| 07:25.00 | p3nguin | There are no rules about charging for services; if the customer wants the service, he'll pay the asking price. |
| 07:25.01 | fenrus | they should be charged with "selling internet connections without giving full internet compatibility" |
| 07:25.09 | fenrus | crime against humanity |
| 07:25.23 | p3nguin | No one is forcing you to buy their services. |
| 07:25.26 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 07:25.29 | ChannelZ | Full internet compatbility? |
| 07:25.42 | b14ck | xargs, oh. But I mean--I think I read somewhere a long time about that there was something in Asterisk that allowed you to continue execution without waiting for the AGI script to finish. |
| 07:25.45 | xargs | any isp handing out private IPs just wants pain. Imagine the support calls... |
| 07:25.46 | b14ck | I'd like to do it that way instead. |
| 07:25.50 | kaldemar | fenrus: yes, like five bucks a month for a single click in an admin app. :) |
| 07:26.35 | fenrus | ChannelZ, port forwarding and NAT is fubar.. |
| 07:26.50 | *** join/#asterisk joelsolanki (joelsolank@124.125.151.87) |
| 07:26.57 | kaldemar | b14ck: http://www.voip-info.org/wiki/view/Asterisk+AGI#Forkandcontinuedialplan |
| 07:27.05 | fenrus | kaldemar, i'm not sure if that violates RIPEs rules about charging for ip-addresses.. |
| 07:27.56 | b14ck | =/ |
| 07:28.07 | ChannelZ | fenrus: yeah OK the millions of people using it every day are all delusional |
| 07:28.13 | xargs | kaldemar: that's basically the same thing I said to do |
| 07:28.19 | ChannelZ | b14ck: what version of asterisk? |
| 07:28.23 | tuxx- | forking \o/ |
| 07:28.46 | b14ck | ChannelZ, 1.6.2.7 (latest, stable) |
| 07:28.56 | fenrus | ChannelZ, if you're in control over NAT and port forwarding yourself it's your choice - but if your isp does it for you it's not that great. |
| 07:29.05 | ChannelZ | There supposedly is some sort of asyncronous AGI in 1.6 |
| 07:29.34 | fenrus | ChannelZ, and are you saying that the the world not would be a better place without NAT'ing firewalls ? |
| 07:30.04 | ChannelZ | No, it probably wouldn't for a lot of people. |
| 07:31.58 | ChannelZ | There's already enough wingnuts clicking anything that shows up in their email, installing shit because a website popped up a message telling them to do so.. Adding to that completely unfettered access? No thanks. Although it's only taking something bad and making it worse, |
| 07:33.12 | fenrus | imagine how IPv6 will be, where all machines can work fine without NAT - but the "home-firewall" instead of keeping NAT-states just drops all incoming non-established connections.. |
| 07:33.19 | fenrus | you just add rules to "open this port" |
| 07:33.28 | fenrus | instead of "open this port" and "forward it to XYZ" |
| 07:42.28 | *** join/#asterisk gelo (~gelo@209.138.60.213.dynamic.mundo-r.com) |
| 07:51.12 | *** join/#asterisk sulex (~sulex@host86-114-dynamic.21-79-r.retail.telecomitalia.it) |
| 08:04.23 | *** join/#asterisk gelo (~gelo@209.138.60.213.dynamic.mundo-r.com) |
| 08:07.13 | *** join/#asterisk MiserySoft (~LND@nat67.mia.three.co.uk) |
| 08:10.51 | xheliox | yawns |
| 08:16.22 | *** join/#asterisk TimeRider (steve@94-194-90-178.zone8.bethere.co.uk) |
| 08:17.47 | *** join/#asterisk sulex (~sulex@firewall.blindata.ch) |
| 08:18.51 | *** join/#asterisk voipnoob (shiva@69.31.82.90) |
| 08:23.13 | voipnoob | i have few security related questions |
| 08:23.50 | ChannelZ | Lock your doors at night, don't talk to strangers |
| 08:23.55 | voipnoob | :-) |
| 08:24.08 | *** join/#asterisk gelo (~gelo@209.138.60.213.dynamic.mundo-r.com) |
| 08:24.14 | voipnoob | ChannelZ - what part of the world are you from? |
| 08:24.36 | xheliox | He's from Mars. |
| 08:24.52 | xheliox | They get pretty good Internet access there, but the latency is a real bitch. |
| 08:24.56 | voipnoob | :-) |
| 08:25.45 | ChannelZ | Colorado USA |
| 08:25.56 | xheliox | Me too. |
| 08:26.02 | xheliox | Where at? |
| 08:26.10 | fenrus | this explains everything! |
| 08:26.14 | ChannelZ | West of Denver |
| 08:26.21 | xheliox | Golden, eh? |
| 08:26.21 | fenrus | americans are here to take over the world :) |
| 08:26.25 | ChannelZ | Yup |
| 08:26.42 | xheliox | Nice. Your part of the state has taken plenty of my money. |
| 08:26.49 | ChannelZ | fenrus: I don't want the world, I just want your half. |
| 08:26.56 | fenrus | ChannelZ, :D |
| 08:27.03 | fenrus | ChannelZ, if i can have yours? ;) |
| 08:27.15 | xheliox | ChannelZ: Ever wonder why Manifest Destiny stopped at the coast lines? ;) |
| 08:27.31 | ChannelZ | xheliox: lots of good strip clubs in Golden or something I don't know about? |
| 08:27.43 | xheliox | Well, Blackhawk really. |
| 08:27.56 | ChannelZ | oh. |
| 08:28.08 | ChannelZ | Well that was your choice then :) |
| 08:28.22 | xheliox | Wasn't really complaining. |
| 08:28.30 | xheliox | Just a fact. |
| 08:28.31 | ChannelZ | heh |
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| 08:28.48 | xheliox | voipnoob: What are your security concerns? |
| 08:29.34 | voipnoob | Let me explain my scenario first, xheliox |
| 08:30.35 | voipnoob | i am studying VOIP & trying trying to write up a basic req doc. I am more of a programmer, not a VOIP or networking guy. I will bring in other people later. I am not trying to get a basic understanding so that I can find a good VOIP/Networking guy & also work with him. |
| 08:30.55 | voipnoob | I meant "I am now trying to get a basic understanding" |
| 08:31.03 | voipnoob | "now" not "not" |
| 08:32.17 | voipnoob | My scenario is one Asterisk Server in my country in Asia - this doesn't SIP Trunk or terminate directly. It's connected to another Server in Europe for the SIP Trunk & Termination - a partner will take care of it so i am not concerned about it |
| 08:32.36 | voipnoob | I will have 1000 odd customers with 150 max concurrent calls |
| 08:32.55 | voipnoob | My customers are distinct - they have their own ISP connections. |
| 08:33.04 | voipnoob | My Asterisk Server will be on the internet |
| 08:33.11 | voipnoob | Till now I have figured out basic stuff. |
| 08:33.17 | voipnoob | Now next I am thinking about security |
| 08:33.46 | voipnoob | All the customers will connect to the Asterisk Server from the softphones/hardphones using their SIP id/password |
| 08:34.17 | voipnoob | is this enough for authentication purposes - do I have to think of something more to prevent unauthorised connections |
| 08:35.34 | kaldemar | voipnoob: a few pointers: http://svn.digium.com/svn/asterisk/tags/1.6.2.7/doc/tex/security.tex |
| 08:35.50 | voipnoob | tx, kaldemar - let me take a look |
| 08:36.10 | xheliox | kaldemar: Doesn't really cover risk analysis for passwords. |
| 08:36.19 | gelo | voipnoob: if i were you, i'd consider deploying an sbc |
| 08:36.44 | kaldemar | voipnoob: don't use numbers as device names (the client entries in sip.conf), some people use e.g. mac addresses. and use strong passwords of course. |
| 08:37.09 | voipnoob | ok |
| 08:37.22 | kaldemar | xheliox: yes, i know. but it spares us from writing that stuff here. :) |
| 08:38.01 | voipnoob | and a document i got at CISCO site - says that using IPSEC encrypted calls doubles the bandwidth requirement. Is there any better way for securing the calls itself? |
| 08:38.19 | *** join/#asterisk sourcode (~code@ppp-58-8-237-71.revip2.asianet.co.th) |
| 08:39.34 | xheliox | voipnoob: Are you concerned about the media being encrypted? because a lot of softphones aren't going to support that in the first place. |
| 08:40.07 | kaldemar | voipnoob: for SIP, there's really no STRP support in the release branches yet. IAX has encryption and key authentication. |
| 08:40.28 | xheliox | You need to be more concerned about password policies and non-standard usernames, I'd say. But even that.. in an unecrypted scenario.. is going to prevent the end user from being exploited if their network/computer is compermised. |
| 08:40.59 | xheliox | is going/isn't going |
| 08:42.49 | voipnoob | ok |
| 08:47.19 | voipnoob | but does encrypted streams actually double the bandwidth requirement? |
| 08:47.49 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 08:47.56 | xheliox | Sounds about right, give or take. |
| 08:49.18 | voipnoob | ok |
| 08:52.02 | voipnoob | would SRTP support as & when it gets added be cheaper than IPSEC or something like that in terms of bandwidth |
| 08:52.53 | *** join/#asterisk jkroon (~jkroon@dsl-244-3-100.telkomadsl.co.za) |
| 08:52.58 | jkroon | is store.digium.com down? |
| 08:53.13 | xheliox | jkroon: No. |
| 08:53.24 | *** part/#asterisk joelsolanki (joelsolank@124.125.151.87) |
| 08:53.38 | jkroon | crap. can't connect to it's https. thanks, then the problem is probably my side. |
| 08:53.44 | xheliox | Hmm. |
| 08:53.49 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 08:53.50 | xheliox | I didn't get to the secure order area. |
| 08:53.52 | xheliox | Hang on. |
| 08:54.13 | jkroon | missing proxy setting this side, i'm in, thanks. |
| 08:54.22 | xheliox | :) |
| 08:54.36 | xheliox | Where in South Africa are you? |
| 08:54.41 | jkroon | gauteng. |
| 08:55.01 | jkroon | and no, i'm not going via seacom. |
| 08:55.28 | kaldemar | voipnoob: should be cheaper in bandwidth than tunneling. |
| 08:55.42 | xheliox | Had to look that one up. I've been to Cape Town, but not that part of SA. |
| 08:55.57 | jkroon | hmm, i can't access the checkout page. ie, i can load up my cart but get a blank response on trying to checkout my cart. |
| 08:56.31 | *** join/#asterisk maxagaz (~maxagaz@soho2.i-xanadu.com) |
| 08:57.43 | xheliox | I just added something to my cart and logged in |
| 08:57.54 | xheliox | that's as far as I'm going as I don't need a $1900 analog card ;) |
| 08:58.50 | voipnoob | tx, kaldemar |
| 08:58.54 | xheliox | Short of adding my payment info, seems to be working. |
| 09:04.17 | aceio | hi |
| 09:04.33 | xheliox | Hello. |
| 09:05.05 | aceio | i am getting emmiting red orange led lights all 4 ports on b410p |
| 09:05.40 | aceio | i am using misdn drivers |
| 09:07.23 | aceio | but when i plug in the isdn line too the interface card i get i steady red light |
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| 09:25.16 | jkroon | xheliox, will try again, but I get stuck when I hit that "checkout" button. |
| 09:25.42 | jkroon | nm, works now. |
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| 09:44.10 | orioni | hi there , where can i find some implementation of asterisk for hotels ? general ideas what can be done and what are the trends of the used features of the hotel |
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| 10:18.46 | guyvdb_ | Hi, can someone point me in a direction to implement call waiting. Do I do it in the dial plan or is it configured else where? |
| 10:19.12 | guyvdb_ | call waiting on the SIP extentions |
| 10:28.37 | xheliox | If your SIP client can support another channel, it'll offer the call. |
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| 11:03.58 | carrar | bwhahah http://www.dailymotion.com/video/xa874c_tom-mabe-eavesdropping_news |
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| 11:56.58 | guyvdb_ | Hi, how do I setup recording of all calls on certain extentions? Is is thought automon? |
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| 12:03.51 | gelo | guyvdb_: take a look at Monitor() and/or MixMonitor() applications |
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| 12:07.14 | shadowhand | i have a general SIP question, but don't know where else to ask... we are using SNOM 320 phones and i'm trying to set up "action urls" |
| 12:07.28 | guyvdb_ | gelo: thx |
| 12:07.29 | shadowhand | but i can't figure out the parameter to pass for the incoming call phone number |
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| 12:17.41 | shadowhand | ah, found it... $remote seems to be the variable, but it a SIP address |
| 12:17.48 | shadowhand | guess there is no way to get the raw number |
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| 12:55.45 | grharry | Hi, I am using dahdi linux 2.3.0 with OpenVox B200P appart from the fact that some times the echo gets annoying all is ok ... I have to know if the dahdi linux version also supports the OpenVox G400P1 ( GSM ) ??? |
| 12:56.25 | shamelessn00b | Hi, we have a trunk group configured with a local cellular operator with CLI masking enabled, how can I set the CLI field and send the number to the operator |
| 12:57.22 | [TK]D-Fender | grharry: Other way around. More like does the CARD support it. Go to OpenVox's site and they should tell you. |
| 12:57.24 | shamelessn00b | inside my dialplan, which variable should I set, before I initiate the call |
| 12:58.52 | [TK]D-Fender | shamelessn00b: "core show aplication SetCallerPres" |
| 12:59.06 | shamelessn00b | [TK]D-Fender: thanks |
| 12:59.08 | shamelessn00b | :) |
| 12:59.10 | leifmadsen | is that done with CALLERID() now? |
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| 13:05.33 | shamelessn00b | how do I set to a particular number |
| 13:05.55 | shamelessn00b | that application only allows for restricting the cli info going on the network |
| 13:07.32 | [TK]D-Fender | shamelessn00b: CID is set the same way it always was |
| 13:07.50 | shamelessn00b | I donot know how its done |
| 13:08.44 | [TK]D-Fender | shamelessn00b: "core show function CALLERID" |
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| 13:28.10 | shamelessn00b | thanks TK I was able to initiate a call this time |
| 13:28.17 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 13:28.37 | shamelessn00b | but the operator is getting the ID of my SIp client instead of the value I'm setting |
| 13:28.59 | shamelessn00b | I registered on the asterisk box through sip id of 99000 |
| 13:29.23 | shamelessn00b | then dialed an extension where I'm calling the callerID() app |
| 13:29.29 | shamelessn00b | setting my CLI value to 1234567 |
| 13:29.49 | Baylink-work | shamelessn00b: recap: you're trying to set outgoing CNID on calls being sent over what kind of channel? GSM? |
| 13:30.07 | shamelessn00b | yes, GSM |
| 13:30.21 | *** join/#asterisk iCEBrkr (~icebrkr@72.251.206.106) |
| 13:30.29 | shamelessn00b | well, we have an SS7 link with the operator |
| 13:30.33 | shamelessn00b | so I'm dialing a trunk |
| 13:31.03 | Baylink-work | I shouldn't expect you'd be permitted to send CNID on ... Oh. You're talking to the back of the MTSO, not the wireless side. |
| 13:31.36 | *** join/#asterisk garymc (~chatzilla@host81-139-76-8.in-addr.btopenworld.com) |
| 13:31.44 | shamelessn00b | initiating a call to the switch, something like call(ss7/msc/<somenumber>) |
| 13:31.57 | shamelessn00b | and the call is landing on that subscriber number |
| 13:32.04 | Baylink-work | Ah. Then you're definitely out of my league; never worked with native SS7 |
| 13:32.14 | Baylink-work | Misunderstood the scroolback. |
| 13:32.40 | shamelessn00b | a possible workaround could be that I just set the SIP ID to a value that I want sent over the network |
| 13:32.54 | shamelessn00b | instead of using the callerID() application |
| 13:34.34 | Naikrovek | jeebus; lvm takes a long time |
| 13:34.49 | Naikrovek | is adding 800gb to a logvol |
| 13:35.28 | gelo | shamelessn00b: are you literally trying to send 1234567 as a caller ID, or are you using a DID assigned to you? |
| 13:36.11 | shamelessn00b | im trying to send 1234567 as caller id |
| 13:36.27 | shamelessn00b | but instead of sending that, asterisk is sending my sip id as the caller id |
| 13:36.49 | shamelessn00b | so Im getting 99000 (the ID I registered my sip client to the asterisk server) on my cell phone |
| 13:37.19 | shamelessn00b | the operator is sending whatever they get from us as the caller ID |
| 13:37.58 | shamelessn00b | so if I register with the asterisk box with the sip id of 99999 I'd get 99999 displayed on my cell phone after I initiate the call |
| 13:38.01 | gelo | is that legal in your country? really curious |
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| 13:42.00 | shamelessn00b | http://pastebin.com/R4WwCEAr |
| 13:42.06 | shamelessn00b | am I doing something wrong here? |
| 13:43.15 | [TK]D-Fender | shamelessn00b: exten => 33351100221,n,Set(CALLERID(name)=1234567) <--- NUMBER, not NAME |
| 13:44.22 | shamelessn00b | I did that on purpose because 'number is deprecated and doesn't work in 1.6. ' |
| 13:44.34 | tuxx- | CALLERID(num) |
| 13:44.34 | tuxx- | ;p |
| 13:44.46 | shamelessn00b | lemme try it with num only |
| 13:45.25 | shamelessn00b | ok works |
| 13:45.28 | shamelessn00b | thank you all |
| 13:45.30 | shamelessn00b | :) |
| 13:45.35 | tuxx- | :) |
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| 14:13.10 | devmod | Was FF* Converter ever available for dl? It's supposed to be opensource but I can't find the source anywhere... http://www.i6net.com/products/ff/ |
| 14:14.47 | aceio | hi all |
| 14:15.01 | aceio | where do i add this line |
| 14:15.03 | aceio | Dial(DAHDI/g1/${EXTEN}) |
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| 14:15.27 | tumaix | aceio: in the dialplan. |
| 14:15.30 | gelo | aceio: extensions.conf? |
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| 14:16.02 | aceio | tumaix: thx |
| 14:16.08 | aceio | gelo: thx |
| 14:16.13 | garymc | Lets be clear. What advantage using VPN server abroad to make voip calls have? |
| 14:17.16 | Gido-E | no nat? |
| 14:17.45 | *** join/#asterisk shamelessn00b (~chatzilla@58-65-172-114.nayatel.pk) |
| 14:18.19 | shamelessn00b | hey guys, one more thing, how can I achieve the same from asterisk CLI(if thats possible, using the originate command) |
| 14:18.38 | leifmadsen | huh? |
| 14:18.40 | [TK]D-Fender | shamelessn00b: the same what? |
| 14:18.49 | shamelessn00b | call a remote party |
| 14:18.50 | leifmadsen | use... 'originate' ? |
| 14:19.02 | shamelessn00b | and set my the CLI to what I want |
| 14:19.19 | shamelessn00b | without registering a SIP client and dialing into an extension |
| 14:19.24 | leifmadsen | same thing |
| 14:19.28 | leifmadsen | read the documentation |
| 14:19.35 | [TK]D-Fender | shamelessn00b: Use a Local channel as your calling channel |
| 14:19.39 | *** join/#asterisk Z_God (~julius@wlan226046.mobiel.utwente.nl) |
| 14:19.50 | shamelessn00b | oh ok, thanks again |
| 14:19.50 | [TK]D-Fender | shamelessn00b: do whatever you want. It's dialplan on both ends |
| 14:19.58 | Gido-E | shamelessn00b, for quick testing, i use callfiles. |
| 14:20.16 | Gido-E | i dont need also or oss support on the server |
| 14:20.26 | Gido-E | s/also/alsa |
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| 14:21.57 | jrz | anybody using localphone.com for incoming numbers / DID? |
| 14:23.11 | devmod | When using a queue, after an avail agent is found how can I make customer and agent meet in a ConfBridge instead of connecting the call directly? any pointers on how to get started on that ? |
| 14:25.08 | *** join/#asterisk Tim_Toady (~moi@188.4.66.69.dsl.dyn.forthnet.gr) |
| 14:25.46 | gelo | devmod: why would you want that? |
| 14:26.11 | devmod | gelo, I need to bring a third party to the conference eventually and cannot do the mixing on the endpoint |
| 14:27.59 | gelo | you could transfer the customer to a conference room later when needed... |
| 14:28.08 | *** join/#asterisk moy (~moy@bas1-unionville55-1177733627.dsl.bell.ca) |
| 14:28.39 | [TK]D-Fender | devmod: Depends what kind of member device the Queue is calling. |
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| 14:29.28 | devmod | [TK]D-Fender, SIP |
| 14:30.16 | [TK]D-Fender | devmod: Then its not happening. Use Local or Agent (effectively the same), and then you can use Dial() options for this |
| 14:30.18 | devmod | gelo, transferring would mean an interruption on the communication and since there is video involved that isnt an option |
| 14:30.49 | devmod | [TK]D-Fender, The agents could be logged in from SIP devices tho right? that would work? |
| 14:31.09 | [TK]D-Fender | devmod: Video is involved? |
| 14:31.15 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:31.20 | devmod | [TK]D-Fender, yes, video conf |
| 14:31.35 | [TK]D-Fender | devmod: * does not mix/conference video |
| 14:31.45 | devmod | yes, I dont need mixing |
| 14:31.46 | [TK]D-Fender | devmod: that idea is going down the tooilet, fast |
| 14:32.10 | [TK]D-Fender | devmod: Video = dead |
| 14:32.13 | leifmadsen | you can go point to point with video, but that's it |
| 14:32.14 | devmod | I just need audio mixing |
| 14:32.22 | devmod | I understand that |
| 14:32.24 | [TK]D-Fender | devmod: Use a Local channel to call the member witht he appropriate Dial() option |
| 14:32.26 | leifmadsen | video won't pass through the asterisk conference |
| 14:32.56 | devmod | leifmadsen, it will if between the first two users |
| 14:33.12 | leifmadsen | it will? never tried that, but that surprises me :) |
| 14:33.36 | devmod | yeah it does work, ofc assuming matching codec |
| 14:34.22 | gelo | and then you can join the conference with audio only? |
| 14:34.23 | devmod | [TK]D-Fender, not sure how would I modify the queue implementation to do that instead tho? |
| 14:34.26 | gelo | awesome |
| 14:34.27 | devmod | gelo, yes |
| 14:34.52 | [TK]D-Fender | devmod: member=Local/exten@context |
| 14:35.11 | [TK]D-Fender | devmod: then make your exten dial with the right options |
| 14:36.07 | devmod | [TK]D-Fender, I see. So how does the queue knows if the member picked up? |
| 14:36.20 | [TK]D-Fender | devmod: if the dial gets answered |
| 14:36.47 | [TK]D-Fender | devmod: Given you're about to toss both ends I'm not sure what the end impact will be like |
| 14:37.06 | devmod | [TK]D-Fender, ok got it, im gonna try that thanks |
| 14:37.17 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 14:37.45 | *** join/#asterisk mhaddog (~MhaddogM1@z65-50-118-232.ips.direcpath.com) |
| 14:38.53 | devmod | [TK]D-Fender, so with this approach I will need to keep track separately the availability of my agents, right? |
| 14:39.46 | [TK]D-Fender | devmod: Depends. AQM gives you a state device to work with. Look at the pieces and see if one combo fits |
| 14:40.04 | [TK]D-Fender | devmod: What you are looking to do is very irregular, so this may not work out too well |
| 14:40.29 | devmod | yeah I figured |
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| 14:47.22 | *** join/#asterisk wcselby (~wcselby@216.110.88.194) |
| 14:47.24 | wcselby | o/ |
| 14:47.34 | *** join/#asterisk gelpg (~chatzilla@94-21-234-59.pool.digikabel.hu) |
| 14:49.19 | *** join/#asterisk hugorebelo (~hugo@p5B277A47.dip.t-dialin.net) |
| 14:49.52 | *** join/#asterisk kerframil (~kerframil@gentoo/user/kerframil) |
| 14:51.37 | gelpg | hi, I have patterns in my extensions.conf for local calling |
| 14:52.10 | gelpg | but how do you set up a pattern if you don't know how long will be a pone number? |
| 14:53.26 | leifmadsen | gelpg: use the . pattern match |
| 14:53.40 | gelpg | exten => _0.,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})? |
| 14:53.50 | leifmadsen | _XXX.,1,NoOp(Variable length exten: ${EXTEN}) |
| 14:54.23 | leifmadsen | gelpg: yes, but read the best practices document because you need to filter characters out of the . in order to make your dialplan safe |
| 14:56.01 | gelpg | leifmadsen: this one? https://reviewboard.asterisk.org/r/507/ |
| 14:56.20 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:58.10 | Akiraa | Do you know if there are 3g/hdspa wireless modems which can be used to place calls from a computer? |
| 14:58.11 | *** join/#asterisk eliel (~eliels@201.234.94.226) |
| 14:58.23 | Akiraa | place/receive |
| 14:59.13 | leifmadsen | gelpg: http://svn.asterisk.org/svn/asterisk/trunk/README-SERIOUSLY.bestpractices.txt |
| 14:59.18 | leifmadsen | ~bestpractices |
| 14:59.28 | leifmadsen | infobot: bestpractices is http://svn.asterisk.org/svn/asterisk/trunk/README-SERIOUSLY.bestpractices.txt |
| 14:59.29 | infobot | leifmadsen: okay |
| 14:59.39 | Naikrovek | lol "readme-seriously" |
| 14:59.47 | Naikrovek | you should make it "whatever you do, don't read me" |
| 14:59.47 | gelpg | leifmadsen: thanks |
| 15:00.09 | leifmadsen | Naikrovek: "DONT-READ-THIS-BCUZ-YOURE-NOT-COOL.txt" |
| 15:00.19 | Naikrovek | hehe |
| 15:00.27 | leifmadsen | "fuck you I'm not cool! I'm reading the shit out of that document!" |
| 15:00.36 | Naikrovek | well if you tell them not to read it they're going to be super curious about why they can't read it, so they'll read it |
| 15:00.39 | leifmadsen | (at least that's what I'd say) |
| 15:00.45 | Naikrovek | lol |
| 15:00.51 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 15:01.01 | Naikrovek | you Canuks are alright |
| 15:01.05 | leifmadsen | ;) |
| 15:01.06 | Naikrovek | never met a Canadian I didn't like |
| 15:01.11 | Naikrovek | not once |
| 15:01.15 | leifmadsen | that's what she said |
| 15:01.20 | Naikrovek | hah |
| 15:01.21 | Naikrovek | haha |
| 15:02.32 | Mark22 | Naikrovek: I always copy readme.txt to do-not-read.txt |
| 15:02.42 | Mark22 | most peopel read at least one of these files ;) |
| 15:02.47 | *** join/#asterisk Akiraaa (~Akiraaaa@86.122.41.164) |
| 15:03.33 | jaytee | no nationality, Canadian or otherwise, is completely asshole free. I've never met an Aussie I didn't like but that doesn't mean Australia doesn't have assholes. |
| 15:03.57 | Naikrovek | i lived in australia |
| 15:04.02 | Naikrovek | there are lots of australians i don't like |
| 15:04.13 | Naikrovek | they frequent bars there and are threatened when you look them in the eye |
| 15:04.20 | Naikrovek | the same is true for anywhere |
| 15:04.23 | Nugget | zo umop apisdn |
| 15:04.30 | Naikrovek | Nugget: EXACTLY |
| 15:04.35 | leifmadsen | jaytee: there are definitely canadians I don't like :) |
| 15:04.55 | jaytee | I was just saying it's generalizing |
| 15:04.58 | *** join/#asterisk suprstar (~suprstar@216.56.88.4) |
| 15:05.04 | *** join/#asterisk sigius (~sigius@93-125-185-45.dsl.alice.nl) |
| 15:05.29 | jaytee | I just think you have to look a little harder in Canada or Australia than you do in the US for assholes. |
| 15:05.38 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 15:05.43 | Naikrovek | jaytee: yes |
| 15:05.54 | jaytee | heck, I run into one everytime I shave :-) |
| 15:06.04 | leifmadsen | gross ;) |
| 15:06.27 | jaytee | my face, dude! not any other regions |
| 15:06.46 | leifmadsen | gross |
| 15:06.47 | leifmadsen | :D |
| 15:07.54 | gelo | here in spain, i met 4 americans, 4 assholes; 2 australian, cool people |
| 15:07.58 | gelo | no canadians, sorry |
| 15:08.57 | devmod | [TK]D-Fender, when I call directly the Local extension , the transfer to the conf works. But when the Queue dials the extension the client side drops while the agent connects successfully :( |
| 15:09.19 | [TK]D-Fender | devmod: And I dont' see what you've done |
| 15:09.26 | suprstar | when I dial into voicemail,i hit VoiceMailMain(channel,ext) - it knows my # via rdnis and asks "Password.." - Can't I abort with * and have it do whatever? I want it to run VoiceMailMain() with no arg's and ask for a mailbox # ... |
| 15:10.04 | suprstar | err callerid not rdnis .. |
| 15:10.58 | [TK]D-Fender | suprstar: You can hit # or * during the message playback only |
| 15:11.54 | jrz | anybody using localphone.com for incoming numbers / DID? I have 2 DIDs, but it seems that localphone.com is not sending the DID per DID. But the same DID per DID |
| 15:11.58 | jrz | aka my account number |
| 15:12.13 | jaytee | be back later |
| 15:12.26 | suprstar | our existing production asterisk 0.9 system does this, altho it runs 'Voicemail2' instead if VoicemailMain, I can assume the guy who put this together hacked up the C code to make it do that then? |
| 15:12.26 | Gido-E | jrz investigate the sip headers |
| 15:12.29 | [TK]D-Fender | jrz: "is not sending the DID per DID" |
| 15:12.40 | [TK]D-Fender | jrz: "But the same DID per DID" |
| 15:12.42 | Gido-E | jzr some times you have to rewrite the s extension |
| 15:12.44 | [TK]D-Fender | jrz: Contradictory |
| 15:13.17 | jrz | Gido-E: I get emm Set("SIP/localphone-000000a5", "__FROM_DID=XXXXXXX") in new stac |
| 15:13.29 | jrz | and XXXXXXX is my accountnumber at localphone. |
| 15:13.45 | SuPrSluG | what is Content-Type: application/isup; version=ansi; ? Can asterisk handle this? I'm getting 1 way audio. |
| 15:13.55 | Gido-E | that is already your dialplan and when it works, it works :-) |
| 15:13.56 | jrz | [TK]D-Fender: mmm why's that :) it seemed logical when I typed it ;) |
| 15:14.01 | [TK]D-Fender | jrz: and you aren't looking at the complete call with SIP DEBUG to see the packets, nor do we see the full line showing the exten it landed on |
| 15:14.15 | [TK]D-Fender | jrz: Pastebin is your friend. |
| 15:14.17 | [TK]D-Fender | ~pb |
| 15:14.17 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 15:14.44 | devmod | [TK]D-Fender, http://pastebin.com/B54F2Mdg |
| 15:14.51 | SuPrSluG | this is supposed to be a straight sip trunk |
| 15:14.56 | [TK]D-Fender | suprstar: You seriously asking for support for an * 0.9 system in production? |
| 15:15.08 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:15.34 | suprstar | [tk]d-fender: no, i have a new 1.6 system in testing right now, and the users want the same functionality. |
| 15:16.13 | Gido-E | the same? |
| 15:17.32 | suprstar | [tk]d-fender: so im trying to figure out the best way of going about that. I could write an agi that reads in button presses, check mysql manually, and then calls VoiceMailMain without authenticating, I was hoping it was built in, setting in a config file somewhere tho.. |
| 15:18.05 | [TK]D-Fender | suprstar: huh? Rephrase exactly what you want here... |
| 15:21.06 | suprstar | [tk]d-fender: i need voicemail to be able to abort with * when VoiceMailMain asks "password", then run VoiceMailMain with no args, so the users can check a different voicemail box for a different extension. My existing system does it, i need to replicate it in 1.6. |
| 15:22.12 | suprstar | the only way i can think of to do that, is to write an agi that manually authenticates, and if they do, then call VoiceMailMain(ext). If I detect a *, then call VoiceMailMain() |
| 15:22.25 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 15:23.03 | [TK]D-Fender | suprstar: Using "*" to abort Voicemail = Already there. If you want them to be able to check any box, simply don't specify the box for VoiceMailMain |
| 15:24.15 | devmod | [TK]D-Fender, I added the asterisk log. When it gets to the transfer, I get the local extension instead of the caller... http://pastebin.com/M7emw51k |
| 15:24.56 | suprstar | [tk]d-fender: then the voicemail button on their phone would ask them for their mailbox # every time they press the 'messages' button on their phone. When they press 'messages', I dont know if they want to check their own or another mailbox |
| 15:26.20 | [TK]D-Fender | suprstar: so stop telling VoiceMailMain which box to look at first |
| 15:26.21 | *** join/#asterisk infernix (nix@unaffiliated/infernix) |
| 15:26.49 | [TK]D-Fender | suprstar: If you don't know which one they'll want, stop telling VoiceMailMain which one to use |
| 15:26.51 | infernix | hi. i'm trying to use the read command, but i can't figure out where asterisk expects the audio file to be. |
| 15:27.16 | infernix | Unable to open testaudio (no such file or dir) |
| 15:27.20 | [TK]D-Fender | infernix: in the sounds folder, unles you prefix with "/" which indicates an absolute path |
| 15:27.28 | infernix | does it expect a specific format? |
| 15:27.35 | infernix | it's not picking up .wav |
| 15:27.35 | [TK]D-Fender | infernix: No |
| 15:27.40 | infernix | weird |
| 15:27.43 | [TK]D-Fender | infernix: its not in the right place |
| 15:28.16 | infernix | its in /usr/share/asterisks/sounds/testaudio.wav |
| 15:28.20 | infernix | yet it's not being picked up |
| 15:28.30 | *** join/#asterisk Maliuta (~scooby@kiev.lusan.id.au) |
| 15:28.33 | infernix | oh wait |
| 15:28.34 | [TK]D-Fender | infAnd you aren't showing us the file and your attempt |
| 15:28.35 | *** join/#asterisk ThoMe (tm@tm.muc.de) |
| 15:28.38 | ThoMe | hiho |
| 15:28.40 | infernix | format_wav.c:156 check_header: Unexpected freqency 44100 |
| 15:28.40 | [TK]D-Fender | infernix: And you aren't showing us the file and your attempt |
| 15:28.42 | [TK]D-Fender | ~pb |
| 15:28.43 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 15:28.44 | [TK]D-Fender | ^^^^^^^^^ |
| 15:28.47 | ThoMe | when i try to play with "read()" a file: |
| 15:28.47 | ThoMe | [May 6 17:28:14] WARNING[5141]: file.c:664 ast_openstream_full: File 1 does not exist in any format |
| 15:28.50 | ThoMe | [May 6 17:28:14] WARNING[5141]: file.c:991 ast_streamfile: Unable to open 1 (format 0x4 (ulaw)): No such file or directory |
| 15:28.51 | *** join/#asterisk rttrey (~trey@andc-office-fw.atlantic.net) |
| 15:28.53 | ThoMe | its an alaw-file. |
| 15:28.55 | *** part/#asterisk rttrey (~trey@andc-office-fw.atlantic.net) |
| 15:29.00 | ThoMe | with placback() works fine |
| 15:29.03 | [TK]D-Fender | infWell it IS picking up the file according to that error... and it doesn't like the format |
| 15:29.07 | ThoMe | not so with Read() |
| 15:29.11 | ThoMe | any ideas why? |
| 15:29.14 | infernix | so it *does* expect a specific format :> |
| 15:29.17 | [TK]D-Fender | infernix: Well it IS picking up the file according to that error... and it doesn't like the format |
| 15:29.27 | *** join/#asterisk nitram (foo@superblob.com) |
| 15:29.31 | ThoMe | infernix: me? |
| 15:29.38 | [TK]D-Fender | infernix: 8 khz 16bit mono |
| 15:29.42 | [TK]D-Fender | infobot: For .wav |
| 15:29.53 | ThoMe | infobot: For .alaw |
| 15:30.06 | [TK]D-Fender | intthen again, GSM is a completely different format. A comletely DIFFERENT one, therefor not a "single" specific format |
| 15:30.19 | infernix | does some samplerate conversion |
| 15:30.23 | [TK]D-Fender | GAH |
| 15:30.36 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 15:30.36 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 15:36.27 | infernix | totally awesome, thanks |
| 15:38.35 | *** join/#asterisk Pazzo (~ugelt@reserved-225136.rol.raiffeisen.net) |
| 15:39.19 | [TK]D-Fender | infernix: Got it working? |
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| 15:50.57 | bent_screwdriver | anyone know of a good NPA/NXX site/listing for finding local numbers for an area? |
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| 15:58.28 | jmls | tries not to upset p3nguin today ... ;) |
| 15:59.25 | p3nguin | I don't remember being upset with you yesterday. |
| 15:59.26 | *** join/#asterisk orioni (~chatzilla@pc.tring.al) |
| 15:59.26 | orioni | how do i specify the authname on asterisk |
| 15:59.26 | orioni | my other server requires that |
| 15:59.26 | jmls | heh. I felt it over here :P |
| 15:59.41 | jmls | I asked a question badly, so I'm going to try again |
| 15:59.44 | orioni | the astrisk is to be connected to the othter system as an extension created on the other server |
| 16:00.14 | p3nguin | orioni: Servers are not extensions. |
| 16:00.35 | orioni | im getting on the asterisk this 120 No Authentication |
| 16:00.42 | orioni | on the sip show registry |
| 16:01.08 | p3nguin | orioni: Show me your register statement. |
| 16:01.10 | orioni | p3nguin : truing to create a sip trunk, ;( |
| 16:01.21 | p3nguin | Good luck with that. |
| 16:01.25 | p3nguin | ~trunk |
| 16:01.26 | infobot | well, trunk is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 16:01.26 | jmls | which is right ? a) Set(GROUP()=${AGENTEXTENSION}@inuse) or b) Set(GROUP()=${AGENTEXTENSION}@inuse) or c) doesn't matter |
| 16:01.43 | jmls | crap. |
| 16:01.47 | jmls | still got it wrong |
| 16:02.06 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 16:02.14 | jmls | which is right ? A) Set(GROUP(inuse)=${EXTEN}), B) Set(GROUP()=${AGENTEXTENSION}@inuse) or C) doesn't matter |
| 16:02.25 | orioni | but my other system does requires that i specify an authenticate username |
| 16:02.35 | orioni | of which variable on asterisk i cant find |
| 16:02.37 | jmls | ffs. AGENTEXTENSION == EXTEN |
| 16:03.06 | p3nguin | orioni: Show me the peer definition on serverA and the register statement from serverB. |
| 16:03.28 | orioni | the oter server is an freeswitch |
| 16:03.39 | p3nguin | orioni: Show me the peer definition on serverA and the register statement from serverB. |
| 16:03.41 | orioni | and the administrator told me that i have to send the authname |
| 16:03.49 | p3nguin | orioni: Show me the peer definition on serverA and the register statement from serverB. |
| 16:03.58 | orioni | i dont have access on that system man |
| 16:04.05 | p3nguin | Then we can't help you. |
| 16:04.17 | infernix | [TK]D-Fender: yep :) |
| 16:04.21 | orioni | i have to send the authenticate username that is = to my number |
| 16:04.28 | p3nguin | orioni: Show me the peer definition on serverA and the register statement from serverB. |
| 16:04.39 | infernix | i have one remaining question. is there a way to invoke voicemail for a mailbox with a message other than the default? |
| 16:05.03 | orioni | p3nguin is floding |
| 16:05.04 | orioni | :) |
| 16:05.12 | p3nguin | No I'm not, you fucking moron. |
| 16:05.16 | infernix | as in, i'd like to specify an audio file instead of the voicemail using the default/configured unavailable/busy message |
| 16:05.29 | p3nguin | lrn2irc |
| 16:05.49 | orioni | yeah , whatever |
| 16:06.10 | orioni | anyone know how to send an authenticated username from asterisk |
| 16:06.13 | jmls | orioni: not a good idea to "whatever" someone who helps as much as p3nguin does |
| 16:06.23 | p3nguin | Go read the glossary in the IRC manual... look up "flooding." |
| 16:06.40 | orioni | cmon dont get that personak |
| 16:07.03 | Tim_Toady | orioni in sip conf u can put a register statement like this: register => user[:secret[:authuser]]@host[:port][/extension] |
| 16:07.04 | wcselby | orioni - just use /block if you don't like what he's saying |
| 16:07.15 | wcselby | orioni - do you have a register statement in your sip.conf file? |
| 16:07.17 | orioni | is tghere an varialbe to be set ? i,.e authname user fromuser etc ? |
| 16:07.21 | wcselby | orioni - if so, we need to see it |
| 16:07.40 | wcselby | orioni - if not, you need to add it |
| 16:07.52 | leifmadsen | #asterisk release candidates for 1.4.32, 1.6.0.28, 1.6.1.20, and 1.6.2.8 now available: http://www.asterisk.org/node/49937 |
| 16:07.53 | infernix | or wait, i can just run Playback, then Voicemail(X|s) |
| 16:07.54 | orioni | i ahve it but no reigstratin possible |
| 16:08.00 | leifmadsen | #asterisk 1.6.0 and 1.6.1 branches going to security only maint. after this round of RCs. See http://bit.ly/9klSxt and http://bit.ly/aZ8o1P |
| 16:08.07 | orioni | no authentication is says |
| 16:08.11 | jmls | orioni: if you are connecting to a fw server, perhasps you should ask on the fw channel how to connect an asterisk server |
| 16:08.30 | orioni | 666800211:tronic12mat@telephony.dallavere.al/66644800211 |
| 16:08.37 | orioni | here is my reguistration string |
| 16:08.46 | Tim_Toady | thx for ur password |
| 16:08.57 | orioni | yep, it is changed :) |
| 16:09.00 | wcselby | orioni - did your freeswitch admin tell you what your authuser and password should be? |
| 16:09.05 | orioni | before copypaste :) |
| 16:09.22 | Tim_Toady | orioni see again the example i gave you ealrier with the authname |
| 16:09.42 | orioni | jmls : he told that the auth name is the same as the extension , 66644800211 |
| 16:09.50 | wcselby | orioni - you have a register => in front of that, right? |
| 16:10.00 | jmls | so, what is 666800211 ? |
| 16:10.00 | orioni | yep |
| 16:10.08 | p3nguin | (1101.07) <p3nguin> orioni: Show me your register statement. <--- I ask for it, and all I get is grief. |
| 16:10.13 | orioni | the devil number :) |
| 16:10.23 | Tim_Toady | orioni you need something like 666800211:tronic12mat:666800211@telephony.dallavere.al/66644800211 |
| 16:10.44 | orioni | let me try , and p3nguin cheer up man |
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| 17:08.12 | Naikrovek | holy crap LVM takes FOREVER to extend a partition |
| 17:08.21 | Naikrovek | it is a big partition, to be fair |
| 17:16.16 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:18.21 | *** join/#asterisk The-Bat (~The-Bat@59.162.86.164) |
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| 17:24.25 | *** join/#asterisk cmsouza (~chatzilla@c9067bea.static.spo.virtua.com.br) |
| 17:24.57 | *** join/#asterisk murdock_ut (~chatzilla@mail.kimballequipment.com) |
| 17:27.05 | leifmadsen | Naikrovek: you don't want to rush that kind of thing, for mistakes are costly :) |
| 17:27.17 | Naikrovek | yeah i know |
| 17:27.20 | murdock_ut | Ok. Here is a senario. A call B, A want to talk to C, B calls C, C answers and A, B, and C can all hear each other, then B drops out leaving A talking to C. An attended transfer almost accomplishes this except for the fact that at no time can A, B, and C talk to each other. |
| 17:27.30 | Naikrovek | my dumb ass forgot to snapshot the machine before i started, too. :( |
| 17:28.15 | [TK]D-Fender | murdock_ut: We call that a "boring 3-way call" Why are you overcomplicating this? |
| 17:28.18 | *** join/#asterisk diegomad (~mad@190.146.200.120) |
| 17:28.19 | Naikrovek | murdock_ut: transfers aren't supposed to do taht |
| 17:28.34 | [TK]D-Fender | murdock_ut: Indeed "transfer" has nothing to do with this |
| 17:28.34 | Naikrovek | transfers are supposed to transfer |
| 17:28.46 | Naikrovek | conference calling is another matter |
| 17:30.40 | murdock_ut | [TK]D-Fender: Wouldn't you need to transfer A and C into an * conference room? If you tried to use the phones conference or three way call feature wouldn't the conference end when B hung up? |
| 17:31.14 | [TK]D-Fender | murdock_ut: Depends on the phone. Most don't kill the other 2, but rather automatically hand them off like an attended transfer |
| 17:31.25 | [TK]D-Fender | murdock_ut: But you sure as hell don't press the "transfer" button to do this |
| 17:31.47 | [TK]D-Fender | murdock_ut: murdock_ut what are you using? |
| 17:31.57 | murdock_ut | [TK]D-Fender: For phones? |
| 17:31.59 | [TK]D-Fender | yes |
| 17:32.03 | murdock_ut | Snom. |
| 17:32.10 | [TK]D-Fender | murdock_ut: Have you TRIED? |
| 17:32.57 | murdock_ut | [TK]D-Fender: I was reading in the transfer section. I have not read up in the conference section yet. |
| 17:34.14 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 17:34.41 | [TK]D-Fender | SMRT |
| 17:35.00 | [TK]D-Fender | murdock_ut: Round hole, square peg |
| 17:35.19 | murdock_ut | [TK]D-Fender: The story of my life. |
| 17:36.41 | *** join/#asterisk SomethingISODD (~Dan@165puntacana96.codetel.net.do) |
| 17:37.16 | SomethingISODD | Hello all i am having an issue i have defined a context under a user in iax and i added the context into the extension config file but everytime i try a call i keep getting no context or extension |
| 17:37.53 | [TK]D-Fender | ~pb |
| 17:37.54 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 17:46.10 | murdock_ut | [TK]D-Fender: Well I guess when you hang up on a snom it just kills the conference and puts the participants back on hold. |
| 17:47.27 | [TK]D-Fender | murdock_ut: Perhaps you're doing it wrong. Or perhaps it considers your hanging up as a request to "split" them |
| 17:47.32 | [TK]D-Fender | Polycom > All |
| 17:49.49 | bent_screwdriver | pri showing "PRI span 1/0: Provisioned, Down, Active" right after i got this in the log: http://pastebin.com/UXYs4wtX |
| 17:50.13 | bent_screwdriver | restarted dahdi and get errors that module is in use while stopping |
| 17:57.01 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 18:07.01 | *** join/#asterisk rare1980_ (~rare1980@115.186.13.175) |
| 18:12.03 | *** join/#asterisk iscario (~div@laureades.davout02.net1.nerim.net) |
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| 18:21.50 | *** join/#asterisk SomethingISODD (SomethingI@206stb81.codetel.net.do) |
| 18:21.53 | *** join/#asterisk RobH_ (~robh@wikimedia/RobH) |
| 18:22.27 | SomethingISODD | Hello all can someone help me out please i am having an issues with getting a context to work using iax i have pasted both in to pastebin here is te url http://pastebin.com/xnRLcdZz any help would be great |
| 18:24.05 | pabelanger | SomethingISODD: looks good |
| 18:24.46 | SomethingISODD | second i will show the error i am getting1 |
| 18:25.44 | [TK]D-Fender | SomethingISODD: show BOTH sides |
| 18:25.50 | SomethingISODD | ok |
| 18:28.13 | iscario | hey , does anybody got a tutorial explaining how to configure asterisk to run with mysql only (without odbc support which doesn't seem to work on openbsd ) ? |
| 18:28.46 | [TK]D-Fender | iscario: To run what parts? |
| 18:28.59 | SomethingISODD | ok here is the one i am sending from http://pastebin.com/qX6XVw7Z i need a second to get the info from the receiving one |
| 18:29.51 | iscario | mysql configuration [TK]D-Fender (in order to interface later asterisk with a sip proxy) |
| 18:30.20 | [TK]D-Fender | iscario:What PART of Asterisk configuration? You can't shove EVERYTHING asterisk does in a DB |
| 18:30.27 | iscario | sorry |
| 18:31.08 | iscario | dynamic configuration [TK]D-Fender |
| 18:31.18 | [TK]D-Fender | iscario: ..... try again |
| 18:31.19 | iscario | for sip accounts |
| 18:31.25 | [TK]D-Fender | isqSIP accounts? |
| 18:31.38 | [TK]D-Fender | iscario: extconfig.conf |
| 18:32.10 | [TK]D-Fender | iscario: install asterisk-addons for MySQL support. Many docs in each tarball for this |
| 18:32.18 | iscario | yes i did it |
| 18:32.41 | iscario | but how to load modules and configure mysql module [TK]D-Fender |
| 18:32.58 | SomethingISODD | here is the reciving asterisk http://pastebin.com/MLGEBxtD |
| 18:33.41 | [TK]D-Fender | SomethingISODD: Your PBs don't mesh. Make 2 NWE ones. 1 with configs and CLI for siade A, and another one for B |
| 18:34.12 | iscario | cause i have some for odbc, but it can't work , so i juste would like to configure mysql (so that asterisk be able to work with mysql and how i could check it really works) |
| 18:34.21 | SomethingISODD | ok second let me do that |
| 18:35.15 | SomethingISODD | [TK]D-Fender do they both need to have cli? configured |
| 18:36.10 | [TK]D-Fender | iscario: Go read the realtime docs in your tarballs |
| 18:36.30 | [TK]D-Fender | SomethingISODD: dialplan, dirver, and CLI for BOTH sides |
| 18:36.49 | iscario | ok i didn't had a look there :( i'll read that, thx [TK]D-Fender |
| 18:37.06 | *** join/#asterisk Alagar (~Administr@122.164.41.80) |
| 18:37.25 | SomethingISODD | ok thanks second while i do that |
| 18:38.52 | *** join/#asterisk tankbusta (tankbusta@sslvpn.onlinetech.com) |
| 18:39.14 | tankbusta | I have CDR enabled. All internal calls are logged, however outgoing calls are not |
| 18:39.19 | tankbusta | anyone have any ideas? |
| 18:39.34 | tankbusta | (logged to cdr table in pgsql db) |
| 18:40.43 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 18:44.07 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 18:45.37 | *** join/#asterisk TehRabbitt (~rabbott56@c-71-59-82-2.hsd1.pa.comcast.net) |
| 18:45.48 | TehRabbitt | p3nguin: I got the service URLs working |
| 18:45.57 | p3nguin | yay? |
| 18:46.00 | TehRabbitt | yep |
| 18:46.03 | TehRabbitt | and background logo |
| 18:46.07 | p3nguin | Details! |
| 18:46.21 | TehRabbitt | you modify your SEPXXXXXXXXXX.XML file with the proper info |
| 18:46.31 | TehRabbitt | well SEPXXXXXXXX.cnf.xml |
| 18:46.48 | TehRabbitt | <authenticationURL></authenticationURL> |
| 18:46.48 | TehRabbitt | <directoryURL></directoryURL> |
| 18:46.48 | TehRabbitt | <idleURL></idleURL> |
| 18:46.48 | TehRabbitt | <informationURL>http://phone-xml.berbee.com/menu.xml</informationURL> |
| 18:46.48 | TehRabbitt | <messagesURL></messagesURL> |
| 18:46.48 | TehRabbitt | <proxyServerURL></proxyServerURL> |
| 18:46.49 | TehRabbitt | <servicesURL></servicesURL> |
| 18:46.57 | TehRabbitt | those are the options you can put in there |
| 18:46.58 | *** join/#asterisk chazzam (~chazz@173-24-238-25.client.mchsi.com) |
| 18:47.03 | p3nguin | oh |
| 18:47.14 | TehRabbitt | ServicesURL is the globe button on the left... |
| 18:47.22 | TehRabbitt | information is the center softkey |
| 18:47.26 | TehRabbitt | messages is what it says |
| 18:47.36 | TehRabbitt | idleURL is your background logo (in PNG format) |
| 18:48.33 | TehRabbitt | i just need to figure out how to get ringtones working :( |
| 18:48.48 | [TK]D-Fender | TehRabbitt: do not flood in here again... |
| 18:48.52 | [TK]D-Fender | TehRabbitt: APSTEBIN |
| 18:48.54 | [TK]D-Fender | PASTEBIN |
| 18:48.56 | TehRabbitt | [TK]D-Fender] sorry |
| 18:49.34 | p3nguin | Put the distinctivering xml files into the tftp root. Put the ringtones directory into the tftp root. Reboot the phone. |
| 18:49.48 | TehRabbitt | did that... nothing |
| 18:49.55 | jaytee | bad enough we've had flooding in the South and in Tennessee, we certainly don't need it in IRC |
| 18:52.02 | p3nguin | I have those options in my XMLDefault.cnf.xml, but I don't have anything in them. :/ |
| 18:52.15 | TehRabbitt | p3nguin: put something in them ;) |
| 18:52.20 | p3nguin | doing it now. |
| 18:53.02 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 18:53.34 | pabelanger | jaytee: Any major roadways out? I'm heading down that way in a few weeks. |
| 18:53.39 | SomethingISODD | [TK]D-Fender i redid both accounts |
| 18:53.44 | SomethingISODD | and yet i still get the same error |
| 18:53.59 | ruben23 | hi nayone can help, how do i setup the dial plan for an autralian toll free like this---> 1800 24 36 24 and 1800 011 846 |
| 18:54.30 | [TK]D-Fender | ruben23: exten => _1800XXXXXX,1,Whatever() |
| 18:54.36 | [TK]D-Fender | NEXT!@!!@!@ (c) BKW |
| 18:57.37 | *** join/#asterisk Alagar (~Administr@122.164.41.80) |
| 18:58.14 | pabelanger | ~next |
| 18:58.15 | infobot | it has been said that next is NEXT! |
| 18:58.55 | ruben23 | <PROTECTED> |
| 18:59.33 | [TK]D-Fender | ruben23: What you have to dial depends on where YOU are. Are we supposed to know how your provider requires you to format it? Ask THEM |
| 18:59.34 | pabelanger | ruben23: Ask your provider, they will tell you want you need to dial |
| 19:00.07 | SomethingISODD | Can someone take a look at this please |
| 19:00.08 | SomethingISODD | http://pastebin.com/J5CP6pR1 |
| 19:01.19 | [TK]D-Fender | SomethingISODD: I said **2 PASTEBINS** incliding IAX, dialplan, and CLI for EACH |
| 19:01.33 | SomethingISODD | ok |
| 19:01.47 | TehRabbitt | p3nguin: for the background logo: http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a00801c0764.shtml |
| 19:01.52 | *** join/#asterisk wcselby (~wcselby@208.180.112.123) |
| 19:01.57 | p3nguin | reading it now. |
| 19:02.12 | pabelanger | SomethingISODD: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/I_sect14_tt670.html |
| 19:02.34 | p3nguin | For the SIP image, you have to use a bmp... so now I have to either convert that or create a new image. |
| 19:02.44 | TehRabbitt | just convert to PNG using MS Paint |
| 19:02.57 | TehRabbitt | still can't get ringtones to work :( |
| 19:02.59 | TehRabbitt | it just wont load them |
| 19:03.09 | p3nguin | Maybe your phone doesn't support them. |
| 19:03.14 | TehRabbitt | no it does... |
| 19:03.33 | TehRabbitt | http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7921g/5_0_1/english/administration/guide/7921usr.html |
| 19:03.40 | *** join/#asterisk oDesk (~oDesk@188.48.93.48) |
| 19:04.21 | p3nguin | Did you watch the tftpd log when the phone boots? |
| 19:05.40 | p3nguin | https://supportforums.cisco.com/thread/289699;jsessionid=9327165FA68B637260D0454E76B34C86.node0?tstart=1 |
| 19:06.13 | *** part/#asterisk Doc (~scott@2001:470:1:8::2) |
| 19:06.43 | p3nguin | It looks like my phone needs an XML logo file for the screen. |
| 19:08.39 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 19:08.49 | TehRabbitt | p3nguin: where woudl that tftp log be? |
| 19:09.04 | SomethingISODD | [TK]D-Fender for some reason i am now not seeing it on the other server PBX b |
| 19:09.33 | Kobaz | [May 6 15:09:11] ERROR[11812]: chan_iax2.c:4545 handle_call_token: Call rejected, CallToken Support required. |
| 19:09.51 | Kobaz | but i have requirecalltoken=no |
| 19:10.09 | [TK]D-Fender | SomethingISODD: ..... |
| 19:10.12 | Kobaz | what's weird is only dialing with zoiper gives me that error |
| 19:10.24 | p3nguin | tehrabbitt: Mine is /var/log/daemon.log, but yours could be different. |
| 19:11.00 | SomethingISODD | but yet they are registrying fine |
| 19:12.45 | tankbusta | anyone familiar with CDR? |
| 19:12.59 | [TK]D-Fender | SomethingISODD: ..... |
| 19:16.28 | SomethingISODD | [TK]D-Fender register => username@peername should work fine right? |
| 19:16.39 | [TK]D-Fender | SomethingISODD: meaningless |
| 19:17.05 | SomethingISODD | hrm ok |
| 19:18.15 | TehRabbitt | p3nguin: http://pastebin.com/7VM4dE8x |
| 19:22.09 | TehRabbitt | p3nguin: Ah, it's case sensitve |
| 19:22.18 | TehRabbitt | lookign for Ringtone.xml not RINGTONE.xml |
| 19:22.40 | p3nguin | I gave it to you the way my phone looks for and loads it. |
| 19:22.58 | TehRabbitt | you gave it to me as "RINGTONE.xml i think 0_o or perhaps windows extracted it like that |
| 19:23.15 | TehRabbitt | only problem now is the ringtones show up in the menu but won't play lol |
| 19:23.27 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 19:23.29 | *** join/#asterisk prometheanfire (~mthode@70.88.176.253) |
| 19:23.53 | prometheanfire | do people in an itsp type of situation use dns srv records for load balencing? |
| 19:24.09 | Naikrovek | probably sometimes |
| 19:24.17 | Naikrovek | they also use SIP proxies sometimes |
| 19:24.21 | Naikrovek | like openSER i think |
| 19:25.16 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 19:26.09 | *** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net) |
| 19:26.10 | prometheanfire | ya, I'm trying to decide between kamailio and dns srv and didn't know if srv was viable |
| 19:28.04 | *** join/#asterisk afink (~chatzilla@204.26.87.226) |
| 19:29.04 | TehRabbitt | got it working... apparently debian is weird with case sensitivity |
| 19:29.07 | bmoraca_work | i'm sure they use both |
| 19:29.40 | prometheanfire | ya, gonna test both, it seems kamailio is REAL easy to set up |
| 19:29.48 | Qwell | TehRabbitt: *every* real OS is. |
| 19:29.56 | prometheanfire | indeed |
| 19:30.00 | p3nguin | tehrabbitt: Debian isn't weird with case sensitivity... You're using a Linux filesystem which, as almost all others, is case sensitive. That's why I didn't rename the files when I packaged them. |
| 19:30.06 | TehRabbitt | lol oh |
| 19:30.21 | prometheanfire | I love me some posix |
| 19:30.32 | TehRabbitt | you sent it as "RINGLIST.XML" it was looking for "Ringlist.xml" apparently lol |
| 19:30.36 | TehRabbitt | so perhaps it's my specific model phone |
| 19:30.42 | p3nguin | Your phone is hte weird one, then. |
| 19:30.50 | p3nguin | My phone wants it in caps. |
| 19:31.00 | p3nguin | as packaged |
| 19:31.13 | prometheanfire | it using tftp? |
| 19:31.19 | p3nguin | [root@asterisk ~]# tar czvvhf ringlist.tgz DISTINCTIVERINGLIST.XML RINGLIST.XML ringtones/ |
| 19:31.24 | prometheanfire | tftp isn't case sensitive I think |
| 19:31.27 | p3nguin | That's how I did it. |
| 19:32.03 | bmoraca_work | prometheanfire: certainly is |
| 19:32.18 | p3nguin | unless it's not on a case-sensitive file system. |
| 19:32.21 | prometheanfire | well, never mind then |
| 19:32.52 | bmoraca_work | upgrading my TouchPro 2 to WM 6.5...woo! |
| 19:33.24 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 19:38.27 | *** join/#asterisk defslap (~andy@defsdoor.gotadsl.co.uk) |
| 19:39.45 | *** join/#asterisk dhill (~dhill@fog.mindcry.org) |
| 19:40.32 | dhill | I have a polycom phone that hangs up (sends SIP/2.0 BYE) when asterisk enters VoicemailMain() and starts to play sound files... any idea why? |
| 19:41.08 | SomethingISODD | [TK]D-Fender this is what i am getting now but i do not see the attempt on pbx b chan_iax2.c:9189 socket_process: Call rejected by 201.229.221.53: No authority found |
| 19:41.20 | bmoraca_work | anyone go to Interop and see Barracuda Networks' new PBX based on Freeswitch? |
| 19:41.59 | Nugget | I just got rid of our barracuda load balancers. |
| 19:42.05 | Micc | isn't freeswitch an asterisk dirivitive? |
| 19:42.10 | bmoraca_work | yes |
| 19:42.18 | Qwell | no |
| 19:42.22 | bmoraca_work | Nugget: why's that? (and what'd you do with them?) |
| 19:42.34 | bmoraca_work | freeswitch is a fork of asterisk isn't it? |
| 19:42.41 | paulc | Thought it was more of a "fork, rip the guts out, rewrite a bunch of stuff" kinda deal |
| 19:43.14 | Micc | so asterisk is still superior, right? |
| 19:43.21 | Nugget | asterisk has more muffins. |
| 19:43.39 | Micc | I like muffins. |
| 19:43.49 | bmoraca_work | they seem to think it's not...i have no experience with freeswitch, though, so I couldn't say |
| 19:44.26 | Qwell | bmoraca_work: clearly they aren't biased at all *rolleyes* |
| 19:44.30 | SomethingISODD | Can anyone help me please and take a look at the pastebin http://pastebin.com/V80MczzX i am getting no errors or anything on PBX B |
| 19:44.36 | ChannelZ | I thought freeswitch was a totally separate thing by a disgruntled ex-asterisk dev |
| 19:44.41 | bmoraca_work | Qwell: indeed :P |
| 19:45.04 | Qwell | ChannelZ: pretty much |
| 19:45.31 | ChannelZ | Anyways, quick OT question - I am new to 64-bit linux; Is there some sort of compatibility layer to run 32-bit binaries, or is that a total mess? |
| 19:45.35 | *** join/#asterisk HyperJacker (~g123@24.145.223.188) |
| 19:45.58 | Qwell | no compat needed to run elf32 stuff. you do need to install 32-bit versions of whatever libs it wants though |
| 19:46.11 | Nugget | bmoraca_work: load balancing the webservers that host flightware.com -- I was unhappy with the support we got from barracuda. |
| 19:46.38 | ChannelZ | hmm ok. And do you remember what command it is I can see what libraries a binary links with? |
| 19:46.39 | bmoraca_work | Nugget: i meant "where'd they go when you got rid of them?" |
| 19:46.39 | Nugget | their bread and butter business is with their spam and virus filtering appliances, and it doesn't seem like they really care about the sideline businesses like load balancing |
| 19:46.46 | *** join/#asterisk tumaix (~tomaz@187.65.196.165) |
| 19:46.49 | p3nguin | You pretty much just install the 32-bit package and run the software. |
| 19:46.55 | Nugget | they're still in the rack, just not being used |
| 19:47.33 | bmoraca_work | ahh |
| 19:47.46 | Nugget | I'll yank them out the next time we're shuffling machines |
| 19:48.00 | ChannelZ | nevermind.. ldd |
| 19:48.21 | bmoraca_work | you can certainly give them to me...i'll uhm, dispose of them for you :P |
| 19:48.24 | Nugget | heh |
| 19:48.34 | Nugget | I'll probably get kim to ebay them or something |
| 19:48.37 | *** part/#asterisk prometheanfire (~mthode@70.88.176.253) |
| 19:49.14 | Nugget | or give them to my it guy to install some goofy linux shit onto for his home network |
| 19:49.53 | ChannelZ | looks like this doesn't use any 3rd party so's so I guess I'll just test it at home and see what happens. (I need to upgrade a work server and was going to go 64 while I was going to the trouble, but I have this one 3rd-party license server I have to use) |
| 19:49.55 | bmoraca_work | they already are goofy linux shit |
| 19:50.03 | ChannelZ | Thanks |
| 19:50.08 | Nugget | exactly |
| 19:50.42 | *** join/#asterisk Alagar (~Administr@122.164.41.80) |
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| 19:55.14 | SomethingISODD | can anyone help please |
| 19:56.08 | p3nguin | ~ask |
| 19:56.09 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 19:56.13 | SomethingISODD | i did above lol |
| 19:56.15 | SomethingISODD | Can anyone help me please and take a look at the pastebin http://pastebin.com/V80MczzX i am getting no errors or anything on PBX B |
| 19:57.03 | [TK]D-Fender | SomethingISODD: then you have no debug or verbose enabled |
| 19:57.18 | SomethingISODD | iax2 set debug on |
| 19:57.24 | SomethingISODD | and i load asterisk with asterisk -rcvvvvv |
| 19:57.35 | [TK]D-Fender | SomethingISODD: Then that box isn't the one being contacted |
| 19:58.08 | SomethingISODD | i have checked the ip and such its going to its the correct ip only asterisk box on this internet connection |
| 20:00.14 | [TK]D-Fender | SomethingISODD: IAX2 debug on server B will show it. |
| 20:00.14 | SomethingISODD | thats the thing now for some reason its not |
| 20:00.29 | garymc | [TK]D-Fender : Ive done something changed it back and now My sip phone take about 1 min to register through VPN |
| 20:01.32 | garymc | once it registers it works fine, but it takes soooooo long to register now. Also I havnt sorted the whole thing from not needing a Terminal restart of PPTPD each time i use it |
| 20:01.55 | garymc | any ideas on how I could diagnose this? |
| 20:02.47 | garymc | yeah ive even messed up the internet now |
| 20:02.52 | garymc | fuck sake |
| 20:03.45 | *** join/#asterisk MhaddogM1 (~MhaddogM1@adsl-64-223-140.mia.bellsouth.net) |
| 20:05.06 | TehRabbitt | p3nguin: will this work? |
| 20:05.07 | TehRabbitt | exten => 17474946623,2,Dial(SIP/voipms/2014860909, SIP/voipms/7324961477, SIP/voipms/2013178798,20) |
| 20:06.43 | TehRabbitt | apparently no :( |
| 20:07.21 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 20:09.52 | joako_ | The Asterisk GUI is not intended for use with Asterisk 1.6? |
| 20:11.43 | SomethingISODD | [TK]D-Fender i have tried a full reboot on the server and everything and i still dont see the call attempts on pbx b |
| 20:11.46 | *** join/#asterisk Alagar (~Administr@122.164.41.80) |
| 20:11.48 | SomethingISODD | but i show the error on pbx a |
| 20:12.33 | Micc | SomethingISODD, are you using iptables? |
| 20:12.44 | SomethingISODD | ya but i flushed and disabled it |
| 20:12.52 | [TK]D-Fender | SomethingISODD: I don't see trustworthy complete pastbins from both sides for identity, TCP dumps, or much else. You never gave the properly cleaned up PB that I asked for like 10 times. I'm very tired of wasting my time on this. |
| 20:12.54 | SomethingISODD | thinking that might have been the problem |
| 20:13.06 | TehRabbitt | can anyone give me an example ofan EXTEN => that will call 2 lines at ocne? |
| 20:13.08 | TehRabbitt | one* |
| 20:13.30 | [TK]D-Fender | TehRabbitt: s/,/&/ |
| 20:13.48 | TehRabbitt | ah.. |
| 20:15.52 | SomethingISODD | [TK]D-Fender i have provided the details you requested except the CLI print out from PBX B and the reason for that.. is that there is no cli print outs |
| 20:16.12 | SomethingISODD | wait now there is |
| 20:16.20 | [TK]D-Fender | SomethingISODD: I see no tcp dump attempts, prrof of IP, CLI output showing you're setting the debug levels required, etc |
| 20:16.39 | TehRabbitt | [TK]D-Fender: exten => 17474946623,2,Dial(SIP/voipms/2014860909&SIP/voipms/7324961477&SIP/voipms/2013178798,10) |
| 20:16.40 | TehRabbitt | like that? |
| 20:16.46 | *** join/#asterisk Alagar (~Administr@122.164.41.80) |
| 20:16.48 | [TK]D-Fender | TehRabbitt: Yes. |
| 20:17.03 | TehRabbitt | ah... aight :-\ i've gotta tweak it it's not working lol |
| 20:20.16 | TehRabbitt | forgot to use +1 |
| 20:20.55 | SomethingISODD | [TK]D-Fender http://pastebin.com/nvGYXXiN working on the tcp dump |
| 20:21.05 | SomethingISODD | but there is the nfo and the cli`s for both |
| 20:21.06 | SomethingISODD | brb sec |
| 20:24.05 | SomethingISODD | Do you still require the tcp dump [TK]D-Fender ? |
| 20:24.21 | [TK]D-Fender | SomethingISODD: not now that you seem to actually see CLI |
| 20:24.24 | [TK]D-Fender | <PROTECTED> |
| 20:24.32 | SomethingISODD | thats the phone number i am dialing |
| 20:24.45 | SomethingISODD | i am unsure why it would show as a username |
| 20:25.34 | *** join/#asterisk cesar_CR (~cesar@190.10.115.176) |
| 20:26.15 | SomethingISODD | [TK]D-Fender i fixed that username thing i will paste the updated cli |
| 20:26.41 | [TK]D-Fender | SomethingISODD: Now you're showing me more mismatched junk |
| 20:28.10 | SomethingISODD | what do you mean its exactly as i have pastebined.. except i forgot i changed the dial plan a bit to dial(IAX2/${EXTEN}@name |
| 20:28.16 | SomethingISODD | that was the only difference |
| 20:28.58 | SomethingISODD | http://pastebin.com/EFipEZgR |
| 20:29.02 | SomethingISODD | thats the new cli |
| 20:30.05 | p3nguin | tehrabbitt: +1? |
| 20:31.20 | SomethingISODD | p3nguin would you have a few minutes you could go over my issue and see if you can help me figure out a way to resolve it ? |
| 20:33.18 | SomethingISODD | if anyone has time to help and take alook at this maybe tell me what i am doing wrong here... here is the updated pastebin http://pastebin.com/1twKbn4X |
| 20:33.57 | p3nguin | somethingisodd: What is the problem I'm supposed to discover in this pastebin? |
| 20:34.20 | SomethingISODD | the issue i am having |
| 20:34.39 | SomethingISODD | i am getting a error from PBX b saying no context/extension but both are configured |
| 20:35.03 | SomethingISODD | [2010-05-06 13:07:35] WARNING[10261]: chan_iax2.c:9189 socket_process: Call rejected by 201.229.221.53: No such context/extension |
| 20:35.58 | p3nguin | Paste your entire extensions.conf and iax.conf. |
| 20:36.33 | SomethingISODD | from both servers or just pbx B? |
| 20:36.49 | p3nguin | both. |
| 20:37.00 | SomethingISODD | thats alot of stuff to paste. |
| 20:37.02 | p3nguin | Don't combine them. |
| 20:37.10 | SomethingISODD | both my extensions config`s are very large |
| 20:37.28 | *** join/#asterisk grEvenX (~even@85.19.87.179) |
| 20:38.02 | p3nguin | At least show me the peer definitions for both sides. |
| 20:38.11 | SomethingISODD | ok |
| 20:38.33 | SomethingISODD | when you say definitions you mean the actual account configuration correct? |
| 20:41.40 | p3nguin | yes |
| 20:41.54 | SomethingISODD | PBX A http://pastebin.com/03xxvw6G |
| 20:42.56 | p3nguin | That's on A, so it is the configuration for B. right? |
| 20:44.20 | SomethingISODD | this is the configurion for B http://pastebin.com/DKdwBUas |
| 20:44.34 | Marquis42 | gelo: ping |
| 20:44.46 | p3nguin | And it says that a call that comes from the other host should dial that host again. This is going to create a loop. Don't expect good results. |
| 20:45.34 | gelo | Marquis42: pong |
| 20:45.46 | SomethingISODD | other way around A is where my sip device is connected and i am calling into B from server A |
| 20:46.37 | Marquis42 | gelo: Am I to understand, based on this trace, that your SIP proxy is on the same system as Asterisk, with the proxy on 5060 and * on 5062? |
| 20:47.05 | gelo | Marquis42: yeah, you are right |
| 20:48.01 | p3nguin | somethingisodd: B's configuration says that a call from A should be routed into [local], and anything two or more digits will Dial() out of ZAP/g1. |
| 20:48.17 | Marquis42 | Ok, then are you using pedantic=yes in sip.conf? |
| 20:48.40 | SomethingISODD | exactly p3nguin |
| 20:48.41 | p3nguin | And the register statement tells A how to contact B. |
| 20:49.06 | p3nguin | somethingisodd: And the problem was that B rejects the call? |
| 20:49.13 | *** join/#asterisk DMeloUK (~Dominic_M@64.129.93.210) |
| 20:49.23 | SomethingISODD | exactly it says there is no such context/extenstion on every attempt |
| 20:49.29 | p3nguin | somethingisodd: B's IP address is 201.229.221.53? |
| 20:49.34 | SomethingISODD | and i am trying 18299665555 |
| 20:49.39 | SomethingISODD | yes |
| 20:50.03 | gelo | Marquis42: do you think it's necessary? |
| 20:50.19 | p3nguin | somethingisodd: Have you reloaded the configs after changing and saving them? |
| 20:50.35 | SomethingISODD | ya even shut down asterisk and restarted it |
| 20:50.42 | *** join/#asterisk creativx (~creadurex@197.82-134-19.bkkb.no) |
| 20:50.52 | Marquis42 | gelo: It might be. Things are little more wooly when Asterisk is colocated with a proxy server, because there are certain bits of the code that really aren't properly RFC3261. Pedantic helps in certain scenarios. |
| 20:51.42 | p3nguin | somethingisodd: On the CLI on B, run "dialplan show 18299665555@local" |
| 20:51.47 | SomethingISODD | k |
| 20:51.53 | Marquis42 | gelo: What I unfortunately suspect here is that even though the ports are different, that Asterisk is still considering these dialogs to be in the same "domain" and so is using that logic in the code. |
| 20:52.49 | Andrew_M_ | How do I make 2 DNs ring on 2 different line buttons on an Aastra set? |
| 20:53.17 | SomethingISODD | There is no existence of 18299665555@local extension |
| 20:53.20 | SomethingISODD | weird |
| 20:53.21 | gelo | Marquis42: this server has a virtual IP. could I use this IP as a domain for proxy, and physical IP for asterisk? do you think it would work? |
| 20:53.46 | p3nguin | somethingisodd: Run this "module reload pbx_config.so" |
| 20:54.03 | p3nguin | somethingisodd: Then run again dialplan show ... |
| 20:54.36 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 20:54.58 | SomethingISODD | l |
| 20:54.59 | SomethingISODD | ok |
| 20:55.12 | Marquis42 | gelo: Assuming I'm right (big assumption right at the moment), then absolutely. It's certainly worth trying if you can have Kamailio and Asterisk bind to different IPs. |
| 20:55.47 | SomethingISODD | p3nguin same thing |
| 20:55.56 | SomethingISODD | i have an idea moment |
| 20:55.56 | p3nguin | somethingisodd: OH! I figured it out. |
| 20:56.07 | p3nguin | somethingisodd: exten => _X.,1,dial(ZAP/g1/${EXTEN}) |
| 20:56.11 | p3nguin | somethingisodd: syntax error. |
| 20:56.22 | SomethingISODD | ? |
| 20:56.26 | p3nguin | somethingisodd: I did the same damn thing late last night. I left out the => |
| 20:56.41 | SomethingISODD | oh my god |
| 20:56.42 | SomethingISODD | dont tell me |
| 20:56.47 | *** join/#asterisk MiserySoft (~LND@89.193.33.29) |
| 20:56.49 | SomethingISODD | double checking |
| 20:57.19 | SomethingISODD | SOB |
| 20:57.40 | p3nguin | somethingisodd: After you see so many extensions.confs, you "see" those characters even when they do not exist. :/ |
| 20:58.02 | *** join/#asterisk Alagar (~Administr@122.164.41.80) |
| 20:58.04 | SomethingISODD | i know but this is messed up i have been working on it for three hours |
| 20:58.07 | SomethingISODD | ok let me try a call |
| 20:58.21 | p3nguin | don't forget dialplan reload. |
| 20:58.30 | SomethingISODD | already did |
| 20:58.45 | gelo | Marquis42: it's utterly worth trying. I even configured a workaround configuration making all calls go through asterisk, and wasn't getting a proper bye, but I didn't thought of pedantic... is a really good idea |
| 20:58.59 | garymc | isnt there another chat site other than freenode? |
| 20:59.03 | SomethingISODD | now to figure out what i did with my zaptel config |
| 20:59.08 | p3nguin | somethingisodd: I'm guessing that increasing core verbosity before running dialplan reload would have revealed the syntax error. |
| 20:59.24 | gelo | Marquis42: now i have homework enough to spend a funny friday :) |
| 20:59.27 | Marquis42 | gelo: Yeah, pedantic and your suggestion of binding to different IP addresses are probably your next steps. |
| 20:59.52 | Marquis42 | gelo: Haha, fair enough. Let me know how you get on with it. |
| 21:00.22 | SomethingISODD | i did it with four v1s |
| 21:00.25 | SomethingISODD | for v`s |
| 21:00.27 | gelo | Marquis42: i'll let you know. Thank you very much |
| 21:00.39 | gelo | good night, everybody! |
| 21:00.45 | p3nguin | somethingisodd: verbose 4 should be enough to see it. |
| 21:00.58 | p3nguin | somethingisodd: I'm going to break an exten and see what happens. |
| 21:01.03 | SomethingISODD | ok |
| 21:03.53 | p3nguin | somethingisodd: If you know what you're looking for, it stands out, but if I wasn't looking I would not see the error it is producing. |
| 21:04.32 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 21:05.24 | p3nguin | I can't fault you for not seeing it when you ran dialplan reload. |
| 21:06.01 | SomethingISODD | i should have actually been working on this in a room i could fully see my screen i guess |
| 21:06.05 | SomethingISODD | Be right back |
| 21:12.17 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
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| 21:51.21 | Kobaz | i believe you can |
| 21:51.23 | Kobaz | er |
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| 22:12.22 | *** join/#asterisk boodu (~boodu@175.158.129.128) |
| 22:13.16 | boodu | hi |
| 22:15.17 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 22:16.19 | *** join/#asterisk garymc (~chatzilla@host86-174-140-25.range86-174.btcentralplus.com) |
| 22:16.53 | ChannelZ | oh hai |
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| 22:50.07 | rare1980_ | hi all from where i can get asterisk cli logs? |
| 22:50.58 | rare1980_ | ? |
| 22:51.20 | Mark22 | look at /var/log/asterisk/full (if you have it) |
| 22:55.02 | Baylink-work | If that's not enough for you, you may have to do what we do, truss asterisk inside screen, and log that. |
| 22:59.40 | Mark22 | here what is displayed in the screen is also logged in that file |
| 22:59.51 | *** join/#asterisk Alagar (~Administr@122.164.41.80) |
| 23:01.25 | p3nguin | If you want the classic Google search back, sign this petition: http://www.petitiononline.com/classicG/petition.html |
| 23:01.34 | *** join/#asterisk drclue (~drclue@ip-65-49-163-54.wireless.dyn.beamspeed.net) |
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| 23:05.46 | drclue | Trying to get Festival to be happy with asterisk on ubuntu 10.04 After some fiddling I've got speech from the shell , but I've not been able to convince asterisk yet despite various tutorials |
| 23:06.05 | *** join/#asterisk jks (jks@193.189.93.254) |
| 23:07.37 | joako_ | Has anyone used the Digium Asterisk GUI with Asterisk 1.6? |
| 23:07.43 | Andrew_M | How do I make an extension ring on a specific button of a phone, like line 3, instead of the default line 1? |
| 23:08.16 | [TK]D-Fender | Andrew_M : "extensions" don't ring on phones. Phones get called. How the phone handles it is up to how you set up the phone |
| 23:09.31 | Andrew_M | D-Fender: So it must be handled in the phone set? |
| 23:10.43 | *** join/#asterisk boodu (~boodu@175.158.129.128) |
| 23:10.53 | Andrew_M | I have an Aastra 57i, and the only way I see is setting up a separate SIP account per line. That seems too complicated for this relatively simple task. |
| 23:11.57 | drclue | A separate sip per line would not seem the correct route either as multiple phones could want to answer those lines. |
| 23:13.14 | Andrew_M | Correct. Currently I only have 2 sets in the system, but once I have 1000+ sets, it will be impractical. |
| 23:13.29 | [TK]D-Fender | Andrew_M: Thats the way it works. |
| 23:13.31 | drclue | Anyways , that's not what I'm trying to figure out to day. vI'm trying to sort out the last little bit of making Festival happy with Asterisk |
| 23:14.19 | [TK]D-Fender | Andrew_M: * is not a Key System |
| 23:14.48 | Andrew_M | I'm coming from a CS1000, not key. |
| 23:15.29 | p3nguin | Have PBXs not always been designed this way? |
| 23:15.46 | p3nguin | So many people thinking Asterisk does things that I can't imagine why a PBX would do it. |
| 23:16.21 | Andrew_M | D-Fender: You are saying I should set up as many accounts per set as I want extensions in it, correct? |
| 23:16.39 | Corydon76-dig | Andrew_M: Check your phone documentation. There may be an additional SIP parameter that can be set to ring different lines |
| 23:17.01 | Corydon76-dig | but those SIP parameters are not standard. You need to check phone docs for which one to set |
| 23:17.40 | p3nguin | Why do you even need more than one line key to accept calls in the first place? |
| 23:18.24 | Andrew_M | Corydon76-dig: I've been looking, but it's 1,100+ pages! I am not in love with the Aastra 57i, as it seems stupid to me. I am waiting for some Polycoms to come in. |
| 23:19.12 | [TK]D-Fender | Andrew_M: You seem to be mixing a lot of terms up here |
| 23:19.29 | Andrew_M | p3nguin: Because a secretary may handle multiple extensions for people, and she needs to know how to answer each. |
| 23:19.35 | [TK]D-Fender | Andrew_M: An extension is just a number that you dial, which gets matched in your dialplan. |
| 23:19.56 | drclue | terminology can often be a pain :) |
| 23:20.12 | [TK]D-Fender | Andrew_M: And Each SIP device REGISTER's to *. How * can tell that device what to make a call look like on the phone is up to the phone |
| 23:20.29 | [TK]D-Fender | Andrew_M: In most cases the line keys are tied to registrations. |
| 23:20.44 | Andrew_M | D-Fender: Right, but I want ext 1000 on line 1, ext 2000 on line 3 of the set. |
| 23:20.59 | [TK]D-Fender | Andrew_M: Now you're talking about SPEED DIALS |
| 23:21.10 | [TK]D-Fender | Andrew_M: Which has absolutely nothing to do with * |
| 23:21.25 | [TK]D-Fender | Andrew_M: that is entirely within the scop of the phone. |
| 23:21.40 | p3nguin | I think he means exten 1000 would light up his line 1 key. |
| 23:21.44 | [TK]D-Fender | Andrew_M: if you want to set up speed dials that show the status of another device, that would be "presence" |
| 23:22.05 | [TK]D-Fender | p3nguin: I'll take a rephrase of that as "button" as opposed to "line key" |
| 23:22.06 | *** join/#asterisk Netgeeks (~chris@173.11.68.155) |
| 23:22.11 | joako_ | Andrew_M: Seems like you want RFC 3265 style shared line apperances which Asterisk does not support |
| 23:22.16 | Andrew_M | D-Fender: No speed dials. |
| 23:22.27 | Andrew_M | D-Fender: These are incoming calls. |
| 23:22.31 | p3nguin | I personally don't see why a phone needs more than one line to accept calls. |
| 23:23.20 | [TK]D-Fender | Andrew_M: What is "line 1" in your definition then? |
| 23:23.51 | Andrew_M | p3nguin: How do you answer a call Mr. Smith's if the caller called ext 1000 vs. Mr. Public if the caller called ext 2000? |
| 23:24.22 | drclue | p3nguin: I think it's often more a matter of semantics as relates to parking, and agent ques vs that traditional old school phone with a bunch of lighted cube buttons |
| 23:24.24 | p3nguin | andrew_m: "Company name, may I help you?" |
| 23:24.26 | [TK]D-Fender | Andrew_M: that is some really poor wording. please rephrase it. |
| 23:24.36 | Andrew_M | D-Fender: The Aastra has hard buttons line 1 through 4. |
| 23:25.04 | [TK]D-Fender | Andrew_M: If you want to indicate the content of the call... MANIPULATE THE CALLERID |
| 23:25.38 | [TK]D-Fender | Andrew_M: Incoming call from "MS:John Doe" |
| 23:25.42 | Andrew_M | D-Fender: Suppose I have soft buttons on the set. How do I configure the center left button to ring if ext 1000 was called, but lower right button if ext 2000 was called? |
| 23:25.45 | [TK]D-Fender | Andrew_M: Guess how you'd answer |
| 23:25.56 | [TK]D-Fender | Andrew_M: you can't |
| 23:26.20 | [TK]D-Fender | Andrew_M: this is the understanding gap. Asterisk does not control your phone <----- |
| 23:26.30 | p3nguin | Gasp! |
| 23:26.37 | Andrew_M | D-Fender: Got it. |
| 23:26.47 | joako_ | Andrew_M: You create a SIP account per phone-extension combo and assign 1 softkey per sip account registration. |
| 23:26.49 | [TK]D-Fender | Andrew_M: Asterisk passively sends it a call and the phone decides based on what the designer inteded. |
| 23:27.22 | p3nguin | joako_: Extensions have nothing to do with the SIP accounts. I have multiple extensions that all dial my phone. |
| 23:27.43 | Corydon76-dig | Andrew_M: Interestingly, the Aastra 57i manual includes documentation on how to configure THAT PHONE with Asterisk |
| 23:27.54 | Andrew_M | joako: I was just hoping this was not the answer, because it seems to complicated to implement on a large number of sets. |
| 23:28.14 | [TK]D-Fender | Andrew_M: Which in most cases means if your sole mental capacity for differentiating the nature of a call is by which magic button on the handset is flashing, then you are going to have to set up each button as a separate registration. 100% unique configuration entry |
| 23:28.18 | *** join/#asterisk boodu (~boodu@175.158.129.128) |
| 23:28.30 | p3nguin | People often use macros when they are doing the same operations on lots of end devices. |
| 23:28.53 | Andrew_M | Corydon76-dig: Yes, but I could not find this feature. |
| 23:29.13 | joako_ | Andrew_M: The other option is get RFC3265 style shared line appearances to work in asterisk and then you create one SIP account per extension: https://issues.asterisk.org/view.php?id=11688 |
| 23:29.18 | p3nguin | I used to have two SIP accounts for my phone. Then I realized that Asterisk can handle calls just fine with only one account on the phone. |
| 23:29.20 | Corydon76-dig | Andrew_M: why aren't you generating the configuration files and loading them onto a TFTP server? |
| 23:29.24 | [TK]D-Fender | Andrew_M: frankly Caller ID manipulation is so much simpler, and can convey more than the limited # of buttons a given handset may have |
| 23:29.32 | Corydon76-dig | Andrew_M: seems like an easy way to configure multiple phones |
| 23:29.36 | Corydon76-dig | See Appendix C |
| 23:29.55 | [TK]D-Fender | Corydon76-dig: probably has his appendix removed ;) |
| 23:30.35 | joako_ | That will replicate the behaviour of a typical legacy PBX you might be seeking... |
| 23:31.34 | [TK]D-Fender | joako_: Close... but not quite. |
| 23:32.03 | p3nguin | What is this legacy PBX behavior that people want so much from Asterisk? |
| 23:32.16 | [TK]D-Fender | p3nguin: Key System <- |
| 23:32.28 | p3nguin | Key systems were considered to be PBXs? |
| 23:32.49 | [TK]D-Fender | p3nguin: Larger PBX hmay have hunderds of lines. WTF is a phone with 5 buttons going to do to match that? |
| 23:33.04 | [TK]D-Fender | p3nguin: Many use the term interchangeably, and the lines are kinda grey |
| 23:33.23 | p3nguin | I didn't make the connection between PBX and key system. |
| 23:33.52 | p3nguin | I guess others have trouble making the separation. |
| 23:34.00 | Corydon76-dig | Andrew_M: See page 5-221 |
| 23:34.23 | Andrew_M | I come from CS1000, and the sets behave a certain way. Instead of retraining the organization I work for, I want to implement the same behaviour with Asterisk. I guess I may be too rigid... |
| 23:34.32 | Corydon76-dig | Set X-Line-Id to the line appearance number on which you want a call to appear |
| 23:35.10 | p3nguin | But, as [tk]d-fender said, that is going to limit your possibilities to the number of line buttons your phone has. |
| 23:35.22 | Corydon76-dig | Ah, it appears that's a parameter in the Alert-Info header. Still... |
| 23:36.14 | Andrew_M | p3nguin: Yes, it will limit it. |
| 23:36.18 | [TK]D-Fender | Corydon76-dig: hey, if there's a header for it, more the better... very oddball feature though |
| 23:36.24 | Corydon76-dig | Took me all of 20 minutes to find the manual, peruse it, and find the exact information you're looking for. I'd say you didn't search very well... |
| 23:36.56 | [TK]D-Fender | Corydon76-dig: Sure he did. he used the #asterisk search engine and the query only tood 20 minutes to run! |
| 23:37.55 | Andrew_M | Corydon76-dig: I spent hours trying to find an answer in the manuals and the Asterisk book. But I am new to this, only 6 weeks of experience. |
| 23:38.52 | Andrew_M | Thank you all for shedding light on this for me! |
| 23:38.56 | Corydon76-dig | You were looking for how to configure your Aastra phone in the Asterisk book? |
| 23:39.06 | Corydon76-dig | I think I see the problem |
| 23:39.11 | p3nguin | :) |
| 23:39.50 | Andrew_M | Corydon76-dig: It is funny now, but I did not know if it was a SIP, Dialplan, or phone feature. |
| 23:40.03 | *** join/#asterisk Alagar (~Administr@122.164.41.80) |
| 23:46.10 | *** join/#asterisk meshe (meshe@anapnea.net) |
| 23:46.19 | [TK]D-Fender | Also unfortunate is the fact that patch is for 1.4 No new features are being allowed for it, and its 3 officially branches behind. Has this been planned to port forward and actually merge? |
| 23:52.37 | meshe | I have a couple of did's from a voip provider that are sporadically working, on a working call it goes through the dial plan properly on a call that doesn't it stops at an Answer(); line and the sip debug is showing me a CANCEL response from the provider |
| 23:53.21 | meshe | the call end up at a "Comedian Mail" prompt and the console shows nothing after that |
| 23:54.31 | meshe | would i be correct in assuming the issue is coming from the DID provider? |