| 00:00.04 | *** join/#asterisk netman (~netman@88.148.119.130) |
| 00:00.08 | WIMPy | Nope there are three different bitrates for G.722. |
| 00:00.19 | lvlinux | WIMPy: breaking out the UTF characters huh? |
| 00:00.27 | lvlinux | WIMPy: 3 bitrates? really? |
| 00:00.34 | WIMPy | I'm not on UTF. |
| 00:00.36 | lvlinux | I thought it was straight 64kb? |
| 00:00.49 | WIMPy | Yes, it can be used with 48, 56 or (default) 64 kbps. |
| 00:00.49 | lvlinux | WIMPy: oh yes that's right, in DE land |
| 00:00.54 | lvlinux | ah, interesting |
| 00:01.07 | lvlinux | can you use the other bitrates in * ? |
| 00:01.18 | WIMPy | I don't think so. |
| 00:15.14 | *** join/#asterisk netman (~netman@88.148.119.130) |
| 00:22.22 | Lope | okay I've got asterisk receiving a call from my DID, and forwarding the call to my VoIP phones... Now I would like to record all calls in OGG or MP3 format. Can someone help me with that please? |
| 00:28.32 | WIMPy | You have to do it outside Asterisk. |
| 00:29.24 | Lope | WIMPy: so I guess I need to tell asterisk to record the calls to wav, and then encode with an external program? |
| 00:30.31 | *** join/#asterisk netman (~netman@88.148.119.130) |
| 00:34.40 | [TK]D-Fender | Correct |
| 00:45.41 | *** join/#asterisk netman (~netman@88.148.119.130) |
| 00:53.20 | Lope | context myphones { testsip0 => {Dial(SIP/testsip0); }; works fine |
| 00:53.55 | Lope | context myphones { testsip0 => { MixMonitor(${UNIQUEID}.wav,ab); Dial(SIP/testsip0); StopMixMonitor(); Hangup();}; says declined ? |
| 00:54.09 | Lope | it loads the ael without errors |
| 00:54.40 | [TK]D-Fender | AEL = garbage you're better off without |
| 00:54.46 | Lope | but when I try call testsip0 with the MixMonitor it says declined. |
| 00:54.47 | [TK]D-Fender | makes debugging a PITA and only limits you |
| 00:54.56 | [TK]D-Fender | Show us the CALL. |
| 00:55.08 | [TK]D-Fender | What your dialplan says doesn't matter if we can't prove what's arriving |
| 00:55.13 | Lope | oh, a guy who doesn't use it yet told me it's the present and the future |
| 00:55.15 | Lope | :) |
| 00:55.31 | Lope | okay, how can I show the call? |
| 00:55.35 | [TK]D-Fender | * CLI <- |
| 00:56.00 | Lope | asterisk command line interface? |
| 00:56.04 | [TK]D-Fender | yes |
| 00:56.11 | [TK]D-Fender | where you go to see things actually happen |
| 00:59.20 | Lope | okay here's my terminal https://128.199.77.95:14573/ |
| 00:59.20 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 00:59.45 | Lope | what debug options do you suggest? |
| 01:00.50 | [TK]D-Fender | What is that link supposed to be? |
| 01:00.52 | *** join/#asterisk netman (~netman@88.148.119.130) |
| 01:00.59 | Lope | open it and you'll see my terminal |
| 01:01.06 | [TK]D-Fender | PASTBIN it yourself |
| 01:01.13 | [TK]D-Fender | Don't expect us to run it for you |
| 01:01.27 | Lope | haha, I didn't give open write access to my terminal |
| 01:01.30 | [TK]D-Fender | "What channel driver are you using for your devices? |
| 01:01.39 | Lope | just making the process more streamlined |
| 01:01.48 | Lope | so don't need to pastebin anything |
| 01:02.07 | [TK]D-Fender | Crap will scroll by and people can't go back when they start looking later |
| 01:02.22 | [TK]D-Fender | <[TK]D-Fender> "What channel driver are you using for your devices? <- |
| 01:02.24 | Lope | I've got a DID number that forwards to my asterisk server, that transfers the calls to a VOIP softphone |
| 01:02.57 | lvlinux | Lope he means are you using SIP or IAX or MGCP or ?? |
| 01:02.58 | Lope | What is a channel driver? |
| 01:03.04 | Lope | SIP |
| 01:03.08 | [TK]D-Fender | WHICH SIP? |
| 01:03.11 | [TK]D-Fender | there are TWO now |
| 01:03.18 | lvlinux | so your channel driver is either chan_sip or chan_pjsip |
| 01:03.23 | [TK]D-Fender | CHAN_SIP or CHAN_PJSIP |
| 01:03.31 | Lope | chan sip |
| 01:03.39 | [TK]D-Fender | "core set verbose 10" |
| 01:03.42 | [TK]D-Fender | "sip set debug on" |
| 01:03.46 | [TK]D-Fender | PASTEBIN the failed attempt |
| 01:04.38 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 01:05.02 | Lope | okay, I don't have the mixmonitor module loaded :) |
| 01:06.05 | [TK]D-Fender | Last time I told you to NOLOAD the ones you specifically didn't want listening for conenctions |
| 01:06.41 | Lope | D-Fender: what about the security implications of running about 100 modules I don't need? |
| 01:06.53 | [TK]D-Fender | This is likely to become a very stupid game chasing down every one you think you didn't need as they fail on you one by one as you start actually configuring your system |
| 01:07.09 | [TK]D-Fender | You don't know what you're leaving OUT. |
| 01:07.45 | Lope | Well I've come this far, I might as well continue a bit further before killing the modules.conf and starting again |
| 01:08.31 | lvlinux | Lope do what TK says. He's only trying to help you. You can fiddle fart around with it after you get it working :-) |
| 01:08.41 | [TK]D-Fender | "Finds self in ditch. Looks for bigger shovel" |
| 01:08.53 | lvlinux | LOL ^^^^^ hehe |
| 01:09.17 | Lope | challenge accepted! haha |
| 01:13.06 | MaliutaLap | [TK]D-Fender: remember The Simpsons ... "dig up stupid" :) |
| 01:13.42 | [TK]D-Fender | Duct tape can't fix stupid... but it can muffle the sounds... |
| 01:16.04 | Lope | okay so I loaded the mixmonitor and wav modules. Now the call completes without any obvious errors, but voice only goes one way in the call when using mixmonitor, vs not using mixmonitor. |
| 01:16.14 | *** join/#asterisk netman (~netman@88.148.119.130) |
| 01:17.26 | Lope | I am getting this error though, which sounds related: WARNING[3220]: chan_sip.c:4175 retrans_pkt: Retransmission timeout reached on transmission 90042b526727b56d32329c7a335e6516 for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions |
| 01:19.39 | Lope | okay the problem is not a lack of a module. |
| 01:19.59 | Lope | I've tried autoload=yes to rule out it being a module issue. |
| 01:21.57 | Lope | Any ideas why I only get audio in one direction when using MixMonitor? https://128.199.77.95:14573/ |
| 01:24.50 | Lope | Here's a paste of the log when using MixMonitor and all modules loaded: http://codepad.org/USElYQMm |
| 01:26.21 | Lope | the notice at the 2nd last line is just SPAM, I've not yet filtered it out. |
| 01:27.04 | *** join/#asterisk Oatmeal (~Suzeanne@75-103-145-152.ccrtc.com) |
| 01:30.05 | lvlinux | Lope you didn't have "sip set debug on" before doing the call. That's the info needed. Do that and then PB again. |
| 01:30.18 | Lope | lvlinux: ok |
| 01:32.16 | [TK]D-Fender | Because now you have * sitting in the MIDDLE and that is screwing things up because of your improper networking setup |
| 01:32.20 | Lope | when i enable SIP debugging there's mountains of noise in my logs though |
| 01:32.36 | [TK]D-Fender | Not "noise" , that's PROOF |
| 01:32.56 | Lope | hehe you mean the SPAM from my lack of network filtering? |
| 01:33.20 | Lope | problem is my softphones have dynamic IP's? |
| 01:33.37 | [TK]D-Fender | no |
| 01:34.10 | [TK]D-Fender | it means you didn't set things up right and the wrong addresses are being trusted for where to send audio to. |
| 01:34.33 | Lope | ah, I see |
| 01:35.11 | Lope | How can I secure it? |
| 01:35.21 | [TK]D-Fender | Secure what? |
| 01:35.27 | [TK]D-Fender | you are flipping topics.... |
| 01:35.41 | Lope | How can I stop all the garbage being received by my asterisk server? |
| 01:35.57 | [TK]D-Fender | firewalls obviously |
| 01:36.11 | Lope | so would I need to use fail2ban? |
| 01:36.19 | [TK]D-Fender | that's one option |
| 01:36.38 | [TK]D-Fender | And we're still off-topic |
| 01:36.49 | Lope | well, I could to ip whitelisting, but that doesn't help me with my dynamic IP softphones? |
| 01:39.14 | [TK]D-Fender | </rhetorical> |
| 01:39.39 | youtmon | from the perimeter firewall allow only IP subnets of your carrier network inbound access only |
| 01:40.01 | youtmon | your fw noramlly has an implict deny rulel |
| 01:40.32 | Lope | When you say carrier network, you mean the DID provider? |
| 01:41.09 | Lope | youtmon: okay, how can I differentiate between my DID provider and my softphones in iptables? |
| 01:42.20 | youtmon | do you have a tele carrier you are paying for SIP service? |
| 01:43.48 | Lope | no, I'm renting a DID to receive incoming calls at my asterisk server, from there I transfer the call to my softphone |
| 01:44.47 | Lope | the calls are received via SIP |
| 01:44.52 | youtmon | softphone service by who? |
| 01:45.25 | Lope | I've created 2 SIP accounts on my asterisk server, that I connect to using my softphones. |
| 01:46.19 | youtmon | so you want to connect remotely to asterisk from any location where there's internet service? |
| 01:46.44 | Lope | yes |
| 01:47.07 | Lope | makes firewalling a little challenging :) |
| 01:47.19 | youtmon | well you are exposed at that point |
| 01:48.01 | Lope | I imagine just using a non-standard SIP port number would help a lot? |
| 01:48.02 | youtmon | there is no challenge in your case its all or nothing .... look at your setup like an SMTP server |
| 01:48.46 | Lope | But when I tried using a non-standard SIP port number, my softphones connected but could not carry audio in both directions |
| 01:48.47 | youtmon | you could change your port to another but its like security with obscurity at that point |
| 01:49.20 | Lope | yes, i know it's not a proper solution, but what's the chances of them guessing my phone number? |
| 01:49.31 | youtmon | you may need to use DNS with a service record for sip to get around the port change |
| 01:50.01 | Lope | well, if I put the port in my DNS there's almost no point in using a non-standard port. |
| 01:51.00 | Lope | looks like it's a job for fail2ban? |
| 01:51.23 | youtmon | it helps the client to use the correct port for SIP instaed of default 5060 |
| 01:51.54 | Lope | well, my clients accepted host addresses in the format ip:port and they connected. |
| 01:52.03 | Lope | It's only bi-directional audio that was a problem. |
| 01:52.31 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 01:52.47 | youtmon | could be asymmetric routing too but I don't know ur network |
| 01:53.32 | youtmon | could be RTP that could be the issue as well |
| 01:54.12 | youtmon | try from the LAN side (behind the firewall) if that works then its your NW topology / firewall |
| 01:54.23 | Lope | Okay I'll try the non-standard port again. |
| 01:54.35 | [TK]D-Fender | Won't change the audio issue |
| 01:57.42 | Lope | okay, actually it works with a non-standard port now. Maybe I had a codec problem earlier, or one of my softphones needed a proper restart. |
| 01:57.56 | [TK]D-Fender | no |
| 01:58.13 | youtmon | is there a doc for outbound routes setups? have two GV profile, if a specific area code is dialed to use one and not the other |
| 01:58.36 | [TK]D-Fender | look at the number before you choose to dial |
| 01:58.42 | [TK]D-Fender | or make more specific patterns |
| 02:05.31 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 02:07.23 | Lope | Without MixMonitor, I get audio, with MixMonitor, no audio. Here's the MixMonitor debug: http://codepad.org/V0hFPFcq |
| 02:11.28 | Lope | I'll try extensions.conf instead of extensions.ael |
| 02:12.25 | youtmon | had to change dial patterns to old style . now its working gr8 |
| 02:14.04 | youtmon | Sending to 192.228.141.117:5060 (no NAT) ??? |
| 02:14.32 | Lope | the call gets cut after 32 seconds, which is the destruction time of the SIP dialog |
| 02:14.48 | youtmon | where is the NAT taking place, on asterisk or perimeter router? |
| 02:15.10 | Lope | the NAT is at the softphone side |
| 02:15.36 | youtmon | the logs posted is from client side or asterisk? |
| 02:15.47 | Lope | from asterisk |
| 02:16.09 | youtmon | right, so is asterisk doing a NAT? |
| 02:18.25 | Lope | "doing a NAT" ? you mean asterisk is performning NAT? no way. |
| 02:18.38 | Lope | asterisk server is on a static IP |
| 02:18.59 | youtmon | RFC1918 or ARIN regsitered? |
| 02:19.28 | youtmon | or RIPE? |
| 02:22.23 | Lope | geez I have no idea about the things you're asking, will google |
| 02:24.28 | youtmon | 192.228.141.117 ip of asterisk? |
| 02:24.58 | youtmon | or 192.168.0.170 |
| 02:25.39 | youtmon | or 192.168.0.77 |
| 02:28.28 | Lope | 192.228.141.117 my router's IP and the other two are soft phones. |
| 02:28.54 | Lope | Okay I turned mixmonitor off again to get a comparative log, and low and behold there was no sound again. |
| 02:29.15 | Lope | I changed the bind port back to standard and suddenly audio works again (mixmonitor still off) |
| 02:29.36 | Lope | So I guess one or both of my softphones can't handle a non-standard SIP port. |
| 02:29.54 | Lope | For now I've just temporarily blocked everything on 5060 that's not my IP. |
| 02:30.04 | Lope | To keep the SPAM out. |
| 02:32.14 | Lope | okay, now using the standard SIP port and mixmonitor on, audio works from testsip1 -> testsip0 but not testsip0 -> testsip1 |
| 02:32.53 | youtmon | from the local lan side? |
| 02:33.33 | Lope | both softphones are on my local LAN |
| 02:33.54 | youtmon | ensure fw is turned off on both sip phones |
| 02:34.16 | youtmon | diable the fw service |
| 02:34.38 | Lope | it's not a firewall issue, because audio works in both directions with MixMonitor off |
| 02:35.47 | Lope | could there be something wrong with this command? MixMonitor(${UNIQUEID}.wav,ab); |
| 02:38.19 | youtmon | MixMonitor(filename.extension,[options,[command]]) |
| 02:41.50 | Lope | any ideas why MixMonitor is breaking my audio? |
| 02:44.31 | [TK]D-Fender | it isn't |
| 02:44.39 | [TK]D-Fender | I've already explained it |
| 02:45.05 | Lope | DFender: when I don't use MixMonitor, audio works in both directions, when I do use it, audio only flows one way. |
| 02:45.51 | [TK]D-Fender | <[TK]D-Fender> Because now you have * sitting in the MIDDLE and that is screwing things up because of your improper networking setup |
| 02:45.55 | [TK]D-Fender | Mixmonitor is not to blame |
| 02:46.03 | [TK]D-Fender | you have SCREWED UP SETTINGS somewhere |
| 02:46.09 | [TK]D-Fender | SIP port is NOT IT |
| 02:46.12 | [TK]D-Fender | that is NOT AUDIO |
| 02:46.58 | [TK]D-Fender | The reason they might work without mixmonitor in the middle is they are allowed to REINVITE to one another and successfully work it out |
| 02:47.13 | Lope | TKDFender how can I debug my network setup with regards to it flowing through *? |
| 02:47.16 | [TK]D-Fender | when you have sit int he middle that fuckup causes audio to be told to go the wrong way |
| 02:47.57 | Lope | TKDfender: could the problem be that I have 2 softphones behind the same NAT? |
| 02:48.09 | Lope | trying to call each other |
| 02:48.33 | youtmon | RTP stream should occur between softphones |
| 02:48.56 | [TK]D-Fender | No, it sure as hell shouldn't |
| 02:48.57 | youtmon | per you they are on the same logical segment. |
| 02:49.48 | *** join/#asterisk protem (~protem@unaffiliated/protem) |
| 03:00.53 | Lope | okay sound works when I call from a real phone to my DID provider, which trunks to my asterisk server which transfers to my softphone. |
| 03:01.00 | Lope | (both directions) |
| 03:06.03 | Lope | ok cool, I just inspected the WAV file, it works. |
| 03:07.50 | *** join/#asterisk Oatmeal (~Suzeanne@75-103-145-152.ccrtc.com) |
| 03:42.55 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 03:44.54 | *** join/#asterisk Lope (~Lope@192.228.141.117) |
| 03:45.16 | Lope | Maybe I should run a STUN/TURN server to overcome my NAT problems. Anyone done it? |
| 04:07.12 | *** join/#asterisk youtmon (~yout@c-73-46-212-142.hsd1.fl.comcast.net) |
| 04:29.10 | *** join/#asterisk linocisco (~linocisco@45.112.176.139) |
| 04:31.12 | linocisco | digium E1 card for one E1 line and cheap server to service inbound call center +CRM with 50 max agents, what server I should buy? |
| 04:59.24 | *** join/#asterisk almostworking (~almostwor@unaffiliated/almostworking) |
| 05:10.26 | *** join/#asterisk juned (~juned@202.131.119.122) |
| 05:15.03 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 05:23.11 | *** join/#asterisk goddva (~goddva@77.40.154.242) |
| 06:10.12 | Lope | Okay, I definitely don't have any networking problems now. I have now connected both softphones to the asterisk server each via their own separate VPN connection on separate subnets 10.0.0.x and 10.0.1.x. When testsip1 calls testsip0 there is no audio in any direction. When testsip0 calls testsip1 then audio works in both directions. http://codepad.org/TZvEhZku since each softphone has it's own private VPN connection to the asterisk server, that rules out t |
| 06:10.12 | Lope | he possibility that the 2 softphones are negotiating in a way that they can speak directly to each other. The audio is definitely flowing through the asterisk server in both cases. But when using MixMonitor something breaks. What could it be? |
| 06:13.58 | *** join/#asterisk TheIgorGuy (~TheIgorGu@173-29-238-144.client.mchsi.com) |
| 06:14.24 | Lope | i even set `iptables -P FORWARD DROP` to be extra sure that they can't communicate directly and re-tested. Same result as above. |
| 06:17.00 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 06:17.42 | *** join/#asterisk youtmon (~yout@c-73-46-212-142.hsd1.fl.comcast.net) |
| 06:20.39 | *** join/#asterisk TheIgorGuy (~TheIgorGu@173-29-238-144.client.mchsi.com) |
| 06:23.16 | *** join/#asterisk elitas (~elitas@213.226.135.203) |
| 06:25.31 | ChannelZ | Are you saying it works if you don't use MixMonitor? |
| 06:34.41 | *** join/#asterisk MrFancyPants (~TheIgorGu@173-29-238-144.client.mchsi.com) |
| 06:36.44 | craigify | If you use MixMonitor, you must have audio flowing through Asterisk, or else there is no way asterisk could record the audio stream. Look into canreinvite, directmedia, directrtpsetup config options for chan_sip to force audio through asterisk. Then try mixmonitor as a proof of concept that it works. |
| 06:38.10 | craigify | going back to idle, then sleep. good luck. |
| 06:43.26 | *** join/#asterisk linocisco (~linocisco@45.112.176.149) |
| 06:48.17 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 06:54.07 | linocisco | digium E1 card for one E1 line and cheap server to service inbound call center +CRM with 50 max agents, what server I should buy? |
| 06:55.36 | [TK]D-Fender | linocisco, How many more timeas are you going to ask this questions? |
| 06:55.43 | [TK]D-Fender | Same thing every time |
| 06:56.01 | [TK]D-Fender | Digium was selling 4 port cards 15 YEARS ago. |
| 06:56.10 | [TK]D-Fender | What kind of computers were around then? |
| 06:56.15 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 06:56.39 | [TK]D-Fender | WHATEVER THE HELL YOU WANT |
| 06:56.54 | [TK]D-Fender | CRM says nothing meaningful |
| 07:00.48 | *** join/#asterisk bof22 (~Thunderbi@185.13.183.107) |
| 07:25.08 | *** join/#asterisk netman (~netman@88.148.119.130) |
| 07:38.53 | *** join/#asterisk Aboba (~Bob@S010614cc209fc3d3.gv.shawcable.net) |
| 07:45.14 | *** join/#asterisk Oatmeal (~Suzeanne@75-103-145-152.ccrtc.com) |
| 07:46.43 | *** join/#asterisk pchero_work (~pchero@109.70.54.56) |
| 07:55.29 | Lope | ChannelZ: Yes, I'm saying my softphone to softphone calls work if I don't use MixMonitor. |
| 07:56.08 | Lope | craigify: thanks for the tips, I'll see if I can do that :) |
| 08:02.30 | ChannelZ | Codecs of both phones? |
| 08:03.09 | Lope | g722 |
| 08:03.51 | Lope | Please help, I don't know how to debug this/ |
| 08:04.55 | *** join/#asterisk linocisco (~linocisco@45.112.176.29) |
| 08:05.28 | linocisco | which RAID is best used in call center environment? |
| 08:07.49 | ChannelZ | I can't imagine disk usage being through the roof in that application, whether space or throughput.. |
| 08:08.21 | ChannelZ | It' |
| 08:08.39 | Lope | Okay weird, now I just found that it's got nothing to do with mixmonitor. |
| 08:08.43 | ChannelZ | It'd probably depend more on what kind of server it is and how many drives you could/wanted to cram into the thing |
| 08:10.55 | ChannelZ | I wouldn't figure it would, Lope, which is why I asked about codecs.. as the only time I could think that things would stop working by introducting MixMonitor was if you were doing something like g.729 and didn't actually have any licenses (or ran out) |
| 08:11.11 | Lope | I can make calls from testsip0 -> testsip1 and there's audio working in both directions. But calls from testsip1 -> testsip0, no audio at all. As far as I can tell I've got identical context rules. |
| 08:11.45 | Lope | dialpad show http://codepad.org/gWv5rQ4v |
| 08:11.53 | Lope | dialplan i mean. |
| 08:12.26 | Lope | oh maybe my one softphone is selecting a dodgy codec. |
| 08:13.19 | ChannelZ | It would be far more instructive to see a sip debug than your dialplan. |
| 08:13.22 | Lope | Well my linux based softphone (testsip0) shows G722 is selected nomatter who calls who. |
| 08:13.32 | Lope | okay |
| 08:17.18 | Lope | SIP debug when testsip1 calls testsip0 and there is NO AUDIO at all. When 0 calls 1 audio works perfectly. No MixMonitor or Monitor involved in either case. http://codepad.org/pBwLEicx |
| 08:18.33 | *** join/#asterisk pchero_work (~pchero@109.70.54.56) |
| 08:19.44 | ChannelZ | ok so you have a weird NAT setup going |
| 08:20.08 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:20.14 | Lope | both of my softphones are connected to my asterisk server via VPN |
| 08:20.30 | Lope | I was hoping to eliminate all NAT and network issues by doing that |
| 08:21.04 | ChannelZ | They think their IPs are 192.168.x; But they are coming from 10.x |
| 08:21.23 | Lope | yeah it's weird. Maybe it's because of DNS? |
| 08:21.31 | ChannelZ | mmno |
| 08:22.17 | Lope | my asterisk server's VPN IPs are 10.0.0.1 and 10.0.1.1. And the two softphones are 10.0.0.2 and 10.0.1.2 |
| 08:22.17 | ChannelZ | It's not necessarily a problem, but needs to be configured right. The IP of your asterisk box is 10.0.1.1 ? |
| 08:22.45 | Lope | each softphone VPN connection is on it's own subnet. |
| 08:22.53 | ChannelZ | sorry 10.0.0.1 |
| 08:23.26 | Lope | my asterisk server's IPs are 10.0.0.1 and 10.0.1.1 |
| 08:23.53 | Lope | For comparison, this is when testsip0 calls testsip1 and the audio works perfectly in both directions: http://codepad.org/MvlYUJ2x |
| 08:24.33 | Lope | It's very weird indeed, why are we seeing the 192.168.0 stuff at all? |
| 08:25.18 | ChannelZ | Because that's what your softphones are telling asterisk. They are presumably unaware of the VPN. |
| 08:25.33 | Lope | dumbass softphones! |
| 08:25.34 | Lope | hehe |
| 08:25.45 | ChannelZ | Which isn't necessarily a problem, a lot of phones behind NAT have no idea what's going on. |
| 08:26.05 | Lope | Well I connected the VPNs to eliminate all possibilities of NAT problems. |
| 08:26.13 | ChannelZ | I see this is CSipSimple, these are Android or something? |
| 08:26.26 | Lope | CSipSimple is on android. SFLphone is on linux. |
| 08:28.32 | Lope | It makes no sense, because the default gateway on 192.168.0.x is unable to route to 10.0.x.x ... |
| 08:29.04 | ChannelZ | Well let me try to explain this. Asterisk can be made to know that a peer is behind NAT and that it's likely going to lie about its IP address. |
| 08:32.00 | ChannelZ | But it gets confusing when asterisk's own LAN happens to look the same as the LAN of the remote end. And if I have it straight in my head, you've actually made it worse because of the VPN |
| 08:32.28 | Lope | It's very very strange... because even though these 3 programs are getting fancy detecting IP's they shouldn't be messing with, at no point do they figure out the internet IP of my asterisk server. They keep talking to it via the VPN. |
| 08:32.51 | Lope | But the asterisk server manages to SOMEHOW figure out the internet IP of my router... it's bizarre. |
| 08:34.20 | ChannelZ | That's based more on your routing/network setup than anything. chan_sip can only bind to one interface, or all of them. |
| 08:34.42 | ChannelZ | anyway it doesn't matter |
| 08:34.59 | ChannelZ | let me go back to looking at these sip dumps |
| 08:35.16 | Lope | thanks :) |
| 08:35.50 | Lope | I'll BRB, I'm going to kill my default route and see how these softphones enjoy that. |
| 08:39.23 | ChannelZ | that went well |
| 08:39.38 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 08:40.14 | *** join/#asterisk netman (~netman@88.148.119.130) |
| 08:41.12 | *** join/#asterisk Lope (~Lope@192.228.141.117) |
| 08:42.24 | Lope | the bizarreness continues. Even when I deleted the default route on my phone and on my laptop. So that they could only possibly dream of using the VPN to connect to asterisk or each other... The exact same problem persisted (audio works both ways when 0 calls 1, but no audio at all when 1 calls 0) And even with no default routes available to either device, they still send their 192.168.0.x IP's to the Asterisk server. |
| 08:43.31 | Lope | If I setup a TURN server, will it eliminate all of this crap where they try to connect to each other instead of route through asterisk? |
| 08:45.20 | ChannelZ | They're not trying to connect to each other. You just have created this IP hell where you have a bunch of devices on the same LAN, but have made another LAN with a VPN |
| 08:46.30 | Lope | I see. Okay I'll connect one machine to a VPN and use it for my routing etc. |
| 08:46.35 | Lope | BRB |
| 08:46.47 | ChannelZ | The phones think their IPs are 192.168.x but the traffic is going over 10.x networks, of which asterisk is also on the same network. |
| 08:46.53 | ChannelZ | shrugs |
| 08:48.46 | linocisco | ChannelZ, how much RAM memory is enough for serving 50 agents? 4GB or 8GB or 16 GB ? |
| 08:52.18 | ChannelZ | ??? if all it's doing basically is asterisk, 4GB is enough I'm sure. Asterisk is nowhere near that big. It'd be more about CPU if it's having to transcode 50 g.729 calls simultaneously or something |
| 08:52.29 | MaliutaLap | linocisco: how much transcoding? |
| 08:55.24 | *** join/#asterisk netman (~netman@88.148.119.130) |
| 08:58.16 | *** join/#asterisk Lope (~Lope@192.228.141.117) |
| 08:58.50 | Lope | SIP is such a pain in the nuts. |
| 08:59.58 | ChannelZ | Need to see your sip.conf |
| 09:00.39 | Lope | I connected my laptop to a VPN so I had an IP/gatway in another country. Yet it persisted in communicating on my 192.168.0.x LAN. So I added iptables rules to DROP everything to or from my phone's LAN IP. And after that there was no audio nomatter who called whom. hahaha |
| 09:00.45 | Lope | okay |
| 09:00.59 | ChannelZ | basically your nat settings are jacked |
| 09:02.00 | *** join/#asterisk Tim_Toady (~fuzzy@snf-33276.vm.okeanos.grnet.gr) |
| 09:02.16 | Lope | http://codepad.org/6uqjJYGk |
| 09:03.26 | *** join/#asterisk Tim_Toady (~fuzzy@snf-33276.vm.okeanos.grnet.gr) |
| 09:05.32 | ChannelZ | try adding nat=auto_force_rport,comedia under [general] which may or may not be right because it looks like you're running asterisk 11 and these things changed names and I don't really remember when. |
| 09:10.35 | Lope | it didn't help |
| 09:10.44 | Lope | there were no resulting errors from your command |
| 09:10.52 | ChannelZ | although when you went away last time you seemed to imply uou were changing something about the whole network again so god only knows what you have going now |
| 09:11.13 | ChannelZ | Did you reload chan_sip? |
| 09:11.27 | Lope | i did a core reload |
| 09:12.12 | Lope | now I've got both devices just hooked up to my lan, but I've got an iptables rule on my laptop to reject any INPUT/OUTPUT from/to my phone's LAN IP |
| 09:12.39 | Lope | what about setting up a STUN server? |
| 09:12.40 | ChannelZ | ...!? |
| 09:13.24 | Lope | ChannelZ: the REJECT rule is to try force my phones to communicate via the asterisk server...? |
| 09:13.27 | ChannelZ | Everything's on the same LAN.. which I'm guessing is now all 192.168.x? |
| 09:13.32 | Lope | yeah |
| 09:14.04 | *** join/#asterisk youtmon (~yout@c-73-46-212-142.hsd1.fl.comcast.net) |
| 09:14.28 | Lope | I've tried separating them using different default gateways and so on, but they just kept trying to go via the lan, they both realized "hey, that other machine is also on 192.168.0.x "so it *MUST* be the same physical network... right?" |
| 09:14.45 | ChannelZ | Show me another sip debug now that you've jacked everything |
| 09:14.55 | Lope | okay |
| 09:14.57 | ChannelZ | and then stop "trying things" |
| 09:16.29 | Lope | http://codepad.org/jtCGIIjv |
| 09:18.22 | Lope | hmm, Well I can see why it would be a problem when I've not connected either phone to a VPN. Because the asterisk server would see that both phones are on the same internet IP then it might tell them to try talk to each other. |
| 09:18.37 | ChannelZ | Stop it with that |
| 09:18.55 | ChannelZ | Now you have a new problem. |
| 09:19.17 | ChannelZ | All your devices, including Asterisk, are on the same LAN. But you've told the phones to contact asterisk via its external IP |
| 09:19.28 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 09:19.35 | Lope | But I DID connect my laptop to a VPN so that it's IP/gateway was in another country, and that didn't help. |
| 09:19.48 | ChannelZ | puts his face in his hands |
| 09:19.53 | Lope | okay. Well there's no audio in either direction with your option. |
| 09:20.44 | ChannelZ | which related to your previous setup with VPN. You can just change the entire way your network is routed and seen by your devices and expect the same config to work |
| 09:20.51 | Lope | I've got a 3G connection now, I can connect one device with that, but if your option works, that should not be necessary |
| 09:21.37 | Lope | Should I give the 3G a try? |
| 09:22.12 | ChannelZ | I meant to say you CAN'T just change.... not can |
| 09:22.15 | ChannelZ | but anyway |
| 09:23.44 | ChannelZ | I have to go to bed and can't keep up with this moving target that is your setup. Is this just playing around and testing or what is the end goal? It would be far more useful to know how you intend to use this and get it working for that case. |
| 09:24.21 | ChannelZ | You will find (and already have) that it can be difficult to get a peer like a cell phone to work in a single configuration for both when it's on your LAN and when it's not. |
| 09:26.55 | ChannelZ | Just for the sake of 'wow, it works', with both of your phones on LAN, configure both of them to contact your asterisk server on its LAN IP, not the WAN IP. And it aught to all work just fine assuming your iptables isn't so upside-down the traffic is being blocked |
| 09:27.26 | Lope | I've tried connecting my phone to mobile data now, and nomatter who calls who, I can now get audio from my phone to my laptop, but not the other way. |
| 09:27.52 | Lope | I'll see what's going on now with the IP's in the SIP debug. |
| 09:29.09 | Lope | ChannelZ thanks for your help my friend. I've stated my goal many times, but perhaps you've not seen it. I want to rent DID numbers, and forward those DID numbers to my VOIP softphones. That actually works without any problems, so my main goal has been achieved. |
| 09:29.23 | ChannelZ | The last thing you pastebin'd was probably a routing problem because while your phones had 192.168.x IPs, and knew it, they were contacting asterisk at its external IP.. but the packets were coming from their LAN IP. Asterisk was telling them to send their RTP to its external address, not its LAN address. |
| 09:29.31 | Lope | However it would be nice if the softphones could call each other as well, I didn't think it would be such a big deal to get it working. |
| 09:30.05 | ChannelZ | There are two halves to NAT with Asterisk; 1. when clients are behind NAT but aren't really aware of it; 2. when asterisk its self is behind NAT |
| 09:30.33 | Lope | well it's correct that asterisk was telling them to send data to it's external address. The asterisk server is a VPS separate from my softphones. |
| 09:30.59 | ChannelZ | but it's physically in the same place |
| 09:31.02 | Lope | My asterisk has a dedicated internet IP, no NAT there. |
| 09:31.02 | ChannelZ | on the name wire |
| 09:31.17 | Lope | my soft phones are on dynamic IPs |
| 09:31.30 | Lope | right now on physically separate networks. 3g and fiber. |
| 09:32.12 | ChannelZ | wait a min I think I'm looking at the wrong window at this point |
| 09:32.14 | ChannelZ | AARGH |
| 09:35.54 | Lope | Okay this is a very interesting SIP DEBUG log |
| 09:36.41 | Lope | Here the mobile phone is on 3g, the laptop is on fiber, and the laptop is still sending it's LAN IP which has no relevance at this point. http://codepad.org/2SlXD095 |
| 09:37.06 | Lope | Yet the phone which is also connected to a LAN, is not sending it's LAN IP, it's sending it's internet IP, which is much more relevant. |
| 09:39.13 | Lope | Okay I lie. The phone is sending it's freakin LAN IP as well. 172.20.x.x. Buggers. |
| 09:44.13 | Lope | is there some guide to installing TURN? |
| 09:48.41 | ChannelZ | set in sip.conf your nat to nat=force_rport,comedia (IE remove auto_ because it's not seeing your phones as behind NAT (and they sort of aren't but are lying like they are) |
| 09:48.46 | *** join/#asterisk defsdoor (~andy@cpc73037-sutt4-2-0-cust62.19-1.cable.virginm.net) |
| 09:50.08 | Lope | no audio laptop > phone |
| 09:50.19 | ChannelZ | and with that last suggestion I'm going to bed as I can't keep track of everything you keep changing and testing and what state anything is actually in |
| 09:51.27 | Lope | no audio from phone > laptop, but I'm on a common lan again, I'll borrow my friend's phone again and try the 3g. |
| 09:51.55 | *** join/#asterisk netman (~netman@88.148.119.130) |
| 09:52.00 | Lope | Where can I find a guide for setting up a TURN relay? |
| 09:54.36 | Lope | using 3g it's the same as when I used 3g the last time with nat=auto_force_rport,comedia. one way audio. |
| 09:58.04 | *** join/#asterisk sommarnatt (~sommarnat@193.180.18.201) |
| 10:01.33 | Lope | I also tried nat=no nat=never nat=yes. Even when setting nat=never the softphones still send their LAN IP's |
| 10:11.58 | Lope | I'm giving iax a try. |
| 10:12.23 | Lope | I'm having trouble getting my IAX user registered on SFLphone (linux smartphone) |
| 10:13.25 | Lope | my iax.conf is simply (where spaces are newlines) [testiax0] host=dynamic type=peer secret=a context=myphones |
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| 10:29.41 | gavimobile | can asterisk prevent a forwarded phone call from going to a cellphones voicemail? |
| 10:30.34 | gavimobile | i think it used to be on my pbx that it would forward the call to my cellphone if i wasnt there and if i didnt answer on my cellphone than the caller would recieve the voice mailbox for that extention rather than my celphones voicemail |
| 10:32.35 | Chainsaw | gavimobile: Generally you can deactivate the voicemail forwarding via the cellphone. |
| 10:32.55 | Chainsaw | gavimobile: In which case it would simply ring longer, and you can then cut in your Asterisk voicemail when you see fit. |
| 10:38.08 | gavimobile | Chainsaw: thanks |
| 10:38.15 | gavimobile | ill contact my provider to disbale voicemail |
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| 10:38.50 | Chainsaw | gavimobile: It's better that way. It is very difficult to reliably detect a voicemail forward. |
| 10:40.51 | gavimobile | i dont need a voicemail, thats fine |
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| 15:55.18 | pa | hello |
| 15:56.19 | pa | i was wondering: is there any plan to extend asterisk to be a message router between various physical and virtual lines? i mean routing sms/mms to and from the various virtual networks such as XMPP/telegram/whatsapp/gtalk etcetera |
| 15:56.35 | pa | i suppose it could already work with gtalk (since it's supported, at least the audio) |
| 15:56.48 | [TK]D-Fender | No. |
| 15:57.05 | [TK]D-Fender | Asterisk is a Telephony Platform, not a "messaging" platform |
| 15:57.48 | [TK]D-Fender | Now if you have an interface on different media types and each supports messaging that * can use then it's on you to create this as some sort of "relay" |
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| 16:08.51 | pa | [TK]D-Fender, so supporting routing calls between POTS or GSM and virtual networks such as gtalk/skype/whatever can be considered, but messages not? |
| 16:09.37 | [TK]D-Fender | Don't call it "routing" either. |
| 16:09.41 | [TK]D-Fender | ~b2bua |
| 16:09.41 | infobot | i guess b2bua is a Back 2 Back User Agent. Additional information is available on wikipedia: http://en.wikipedia.org/wiki/Back-to-back_user_agent |
| 16:09.57 | [TK]D-Fender | DIALPLAn is what makes a call go out and bridge. |
| 16:10.03 | lvlinux | pa: Asterisk supports some sms, but I think it's pretty basic |
| 16:10.07 | [TK]D-Fender | It doesn't "just happen". Neither will any kind of messaging |
| 16:14.11 | lvlinux | pa: this type of thing is what XMPP support in * is usually used for: https://www.mundoopensource.com.br/en_page_xmpp_asterisk_pratical_example/ |
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| 17:08.23 | sgduser | hi.. trying to install asterisk on ubuntu 15.04 64-bits, all other requirements -dahdi, libpri, and pjsip are installed- after make menuselect is done when i run make.. i get this error ==> http://bit.ly/1PG455w |
| 17:08.45 | sgduser | hi.. trying to install asterisk on ubuntu 15.04 64-bits, all other requirements dahdi, libpri, and pjsip are installed after make menuselect is done when i run make.. i get this error ==> http://bit.ly/1PG455w |
| 17:11.15 | [TK]D-Fender | <PROTECTED> |
| 17:11.27 | [TK]D-Fender | it is telling you precisely what to do... |
| 17:17.19 | file | you also don't have shared libraries for PJSIP installed |
| 17:17.29 | file | so pjsip wasn't built right, or is using the wrong ones |
| 17:18.45 | sgduser | strange cuz it never showed any error let me try to re-install pjsip once again |
| 17:22.56 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 17:27.22 | marceloamorim | guys, I had a problem on my HDD, and I had an asterisk with E1 - openr2 , should I install openr2 to work properly or can I make work without this? |
| 17:27.54 | WIMPy | Didn't you just say, you need it? |
| 17:28.15 | marceloamorim | but it was in old versions |
| 17:29.21 | WIMPy | So you are about to try a new version now? |
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| 17:33.43 | sgduser | i have tried to remove pjproject, i followed the same steps on asterisk wiki and this is what i get http://bit.ly/1Svm8yc |
| 17:34.07 | marceloamorim | I think there is new DAHDI, I installed the system 2 years ago |
| 17:34.50 | file | sgduser, because that rm only receives .a files and a .pc file, not .so |
| 17:36.41 | file | and based on your original error you also have a PJSIP install in /usr/local/lib |
| 17:37.23 | sgduser | because "sudo make uninstall" also fails!! |
| 17:37.59 | craigify | If you installed pjproject using ubuntu, you'll need to tell ubuntu to remove the package for you |
| 17:38.31 | craigify | also, if this project is for something stable, you should potentially consider using an LTS release, which has long term support (LTS) |
| 17:38.40 | craigify | 15 is not a LTS release....just fyi |
| 17:39.00 | craigify | if you remove pjproject, you're going to use chan_sip instead? |
| 17:39.08 | sgduser | no im installing from source!! |
| 17:39.39 | sgduser | no.. i want to use pjsip but trying to rebuild pjsip |
| 17:39.44 | C1pher2 | Could someone suggest a software to stress test a asterisk server? |
| 17:40.02 | [TK]D-Fender | sipp <- |
| 17:43.24 | craigify | sgduser, ok. do you have multiple versions of the shared library installed? |
| 17:44.24 | sgduser | i installed asterisk before on the same machine, due to some problems im trying to reinstall once again |
| 17:46.13 | craigify | ok so you downloaded the pjsip source first, compiled it, and installed? |
| 17:46.18 | craigify | pjproject |
| 17:46.20 | craigify | I mean |
| 17:46.30 | sgduser | yes |
| 17:46.55 | craigify | so you did not install two separate versions of pjproject, correct? |
| 17:47.03 | craigify | or compile |
| 17:47.06 | craigify | I should say |
| 17:47.57 | sgduser | no.. its the same version!!! |
| 17:48.09 | sgduser | 2.4.5 |
| 17:49.20 | craigify | I don't have much experience in pjproject |
| 17:49.29 | craigify | /usr/bin/ld: /usr/local/lib/libpjsip-ua-x86_64-unknown-linux-gnu.a |
| 17:49.43 | craigify | do you know why it's using that file? |
| 17:49.43 | *** join/#asterisk italorossi (~Adium@187.60.66.11) |
| 17:49.46 | craigify | and not a shared library? |
| 17:50.06 | sgduser | no idea.. newbie to linux too!! |
| 17:50.39 | craigify | do you know how compilers and linkers work? |
| 17:50.57 | sgduser | no |
| 17:51.32 | craigify | well, that sure makes compiling from source that much harder :) |
| 17:52.02 | craigify | as mentioned before, something does not look right with pjproject |
| 17:52.19 | sgduser | any advice? |
| 17:53.07 | craigify | https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject |
| 17:53.09 | craigify | did you read that? |
| 17:53.12 | sgduser | yes |
| 17:53.18 | craigify | did you use --enable-shared? |
| 17:54.09 | sgduser | you mean ldconfig? yes i did! |
| 17:54.21 | craigify | no in your configure line |
| 17:54.26 | craigify | it's in that wiki |
| 17:54.43 | craigify | I think the one time I tried messing with pjproject, I didn't install from source |
| 17:57.46 | craigify | oh I see your second pastebin, and there is definitely shared objects |
| 17:58.06 | sgduser | how to remove pjsip from my system? |
| 17:58.07 | craigify | /usr/bin/ld: /usr/local/lib/libpjsip-ua-x86_64-unknown-linux-gnu.a(sip_inv.o): relocation R_X86_64_32S against `.rodata' can not be used when making a shared object; recompile with -fPIC |
| 17:58.22 | craigify | I don't know why when you ran configure on pjproject, it didn't properly set that |
| 18:00.40 | craigify | what is the configure line you used to compile pjproject |
| 18:00.44 | craigify | before make |
| 18:02.24 | sgduser | " |
| 18:02.24 | sgduser | # ./configure --prefix=/usr --enable-shared --disable-sound --disable-resample --disable-video --disable-opencore-amr CFLAGS='-O2 -DNDEBUG'" |
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| 18:05.13 | craigify | that's exactly what the wiki says |
| 18:05.33 | craigify | at this point, I'd start removing everything and starting over...which looks like where you are now |
| 18:05.52 | sgduser | yes.. i followed the wiki.. infact, i've been following the wiki only!! |
| 18:05.52 | craigify | and trying more limited configuration options in asterisk |
| 18:06.02 | craigify | and probably getting annoyed |
| 18:06.22 | craigify | if pjproject compiled without errors |
| 18:06.32 | craigify | it might be something you did in installing asterisk |
| 18:06.53 | sgduser | so.. how do i start over? |
| 18:07.08 | craigify | make clean, perhaps make distclean |
| 18:07.22 | sgduser | thats in pjsip, or asterisk? |
| 18:07.22 | craigify | in the source tree |
| 18:07.28 | craigify | both |
| 18:07.31 | sgduser | ok |
| 18:09.10 | craigify | there is something going on with the shared object |
| 18:09.12 | sgduser | done without error |
| 18:09.23 | craigify | the error states requiring -fPIC |
| 18:09.28 | sgduser | shall install pjsip now? |
| 18:09.38 | craigify | your guess is as good as mine |
| 18:09.57 | craigify | I'd try installing pjproject from a package |
| 18:10.02 | craigify | then compiling asterisk |
| 18:10.05 | craigify | see what that yields |
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| 18:14.42 | craigify | you might need to delete the *.so files for pjproject from your source install |
| 18:19.22 | sgduser | i will give a try on a fresh machine.. see what happens..thanks.. i learned alot |
| 18:22.23 | *** join/#asterisk linocisco (~linocisco@103.25.14.254) |
| 18:22.32 | file | duplicate pjproject installs will be the death of me |
| 18:22.37 | file | or, multiple |
| 18:23.39 | linocisco | what server hardware should I choose to service 50 agents call center + CRM to save cost as much as possible |
| 18:25.32 | saint_ | linocisco if you go for a free phone system as Asterisk, don t be cheap on the hardware. No matter what you will save on Cisco / Shoretel / Avaya / Alcatel . |
| 18:26.33 | [TK]D-Fender | linocisco: You've been answered on this a dozen times already. |
| 18:26.54 | [TK]D-Fender | linocisco: in multiple channels |
| 18:28.08 | [TK]D-Fender | linocisco: Stop asking the same useless question for months on end. |
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| 18:31.52 | linocisco | I would like to know minimum hardware requirement to service 50 agents. for example, how much RAM, How much HDD or RAID (0,1,5 or 10), |
| 18:31.58 | linocisco | or CPU speed |
| 18:32.14 | saint_ | do you guys know of any SIP deskphone which are intrinsically safe ? |
| 18:32.16 | linocisco | based on that , I would like to seek affordable one in the market. |
| 18:32.34 | saint_ | linocisco 2.5Ghz, 8GB Ram, 500GB HD. |
| 18:32.46 | saint_ | here. |
| 18:32.47 | saint_ | lol |
| 18:32.47 | linocisco | saint_, need RAID or not? |
| 18:32.52 | saint_ | linocisco your call. |
| 18:32.57 | WIMPy | saint_: LOL. (to the phone) |
| 18:33.03 | saint_ | linocisco how badly do you want HD safety ? |
| 18:33.11 | [TK]D-Fender | I ran that spec on a 3 ghz single core xeon with 2gb of ram |
| 18:33.28 | saint_ | [TK]D-Fender i was trying to shut down this conversation he he |
| 18:33.39 | [TK]D-Fender | I already tried yesterday |
| 18:33.43 | linocisco | saint_, that first thing is call performance, the second thing is to store recording for 1 month at least |
| 18:33.43 | saint_ | linocisco take [TK]D-Fender conf. here. you have it now. |
| 18:33.44 | [TK]D-Fender | SAME stupid quesiton every damn day |
| 18:34.17 | saint_ | linocisco everything depend on how much verbose you put in your logs. 500GB will cover a long time before you need to clean up |
| 18:34.59 | saint_ | I was looking at TalkAPhone for Intrinsically safe , but they don t have any deskphone like that. I heard about FITRE in Italy.. Do you guys know anyone in north america ? Ascom maybe ? |
| 18:35.00 | [TK]D-Fender | linocisco: My * install only takes < 10GB, system has 2 GB. You can use WHATEVER THE HELL YOU WANT. |
| 18:35.26 | [TK]D-Fender | linocisco: There is NO SPEC for this. You metrics are wortless for picking hardware. |
| 18:35.43 | saint_ | linocisco i've been running on 50GB for years now with no issue |
| 18:35.51 | linocisco | [TK]D-Fender, u dont use RAID for 50 agents call center? |
| 18:36.10 | [TK]D-Fender | linocisco: Wouldn't you? |
| 18:37.37 | [TK]D-Fender | linocisco: You're talking about the central business tool of a COMPANY. Why are you even ASKING if you should be doing RAID? Do you WANT to get screwed by a single HD dying on you? Do you WANT to get fired when your attempt to be "cheap" causes their system to go down? |
| 18:38.42 | [TK]D-Fender | linocisco: You shouldn't be asking any of these questions. If you are responsible for deploying a soultion you should already KNOW the answer to all of this, not be coming here week after week asking the same stupid question. |
| 18:41.13 | saint_ | linocisco are you located in Myanmar ? |
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| 18:42.23 | [TK]D-Fender | linociscosaint_, that first thing is call performance, the second thing is to store recording for 1 month at least <- look at space required by the format you are going to store your recordings in. This is ALSO documented. Do not waste people's time on this. |
| 18:43.14 | [TK]D-Fender | Documented, AND easily proven by acutally just recording a single call. |
| 18:45.58 | saint_ | this fire department is fucked up. They went from $300/month in phone bills to barely $8/month , but they are pissed off because they can t understand how asterisk works. nobody wanted to learn when i was ready to teach. then this one guy came and say "hey, i'm taking over, show me basically how i can add an extension and change prompts" - but the guy has NO training IT. He only does word processing and know how to print. Just ssh into |
| 18:45.59 | saint_ | <PROTECTED> |
| 18:46.01 | saint_ | fucked up people. |
| 18:46.05 | saint_ | i can t stand it. |
| 18:46.47 | robmal | Start a fire. |
| 18:46.56 | robmal | Erm, let them burn. |
| 18:46.56 | saint_ | haha |
| 18:47.45 | saint_ | i got them this awesome deal for free with a TTS company so the voice would be nice when people called. i did not renew it. I actually left this fire dept. I have other things to take care of in my life. |
| 18:48.09 | robmal | You should recommend linocisco as our authorised asterisk troll, he'll help them. |
| 18:49.00 | saint_ | oh yeah , good one. stand by , let me send his contact info . lol |
| 18:50.43 | saint_ | i never thank all the people who helped me in this chan. I remember malcolmd / mjordan when i found a couple of bugs, [TK]D-Fender for sure, and probably some other folks. it's been running at the fire dep for 2 years , and so in my house with no issue. i got Digium phones , and use Flowroute as a sip provider. So here it is.. thank you :) |
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| 18:52.39 | thansen | I've traditionally used sip phones/trunk and was just sent some shoretell 230 phones that I'd like to integrate into my dial plan..is that even remotely feasible? |
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| 18:53.26 | lvlinux | thansen: as long as the phones support SIP and aren't locked to a provider they should work fine. |
| 18:54.10 | thansen | lvlinux, pretty sure they don't work with sip but I could be mistaken |
| 18:54.38 | thansen | I just don't have any experience with mgcp |
| 18:54.49 | lvlinux | Asterisk does support MGCP. |
| 18:55.01 | [TK]D-Fender | thansen: You should know what your own product supports. I'm sure they have VERY FINE MANUALS you should be reading. |
| 18:56.04 | thansen | [TK]D-Fender, pretty sure I just clarified what it supports..just after some general knowledge about integrating sip/mgcp phones in the same setup |
| 18:56.50 | lvlinux | thansen: As long as you can get Asterisk to talk to them then it should work fine. |
| 18:57.11 | thansen | as I mentioned, I have 0 experience with mgcp so I'm wondering if it's a total waste of time to try and make them work together or just order different phones |
| 18:57.14 | lvlinux | You just do Dial(MGCP/blahblah) instead of Dial(SIP/blah) |
| 18:58.00 | [TK]D-Fender | thansen: You have the phones. You have Asterisk. You know the protocol to configure. * comes with sample configs. Is there even an excuse to not "Just Do It"? |
| 18:58.08 | lvlinux | When you have them setup, they get dropped into a dialplan context just like any other SIP phone |
| 18:59.01 | lvlinux | thansen: look at mgcp.conf and start messing with it. There are tutorials and info about mgcp and * out there too. |
| 18:59.20 | lvlinux | It's not a big deal. |
| 18:59.33 | thansen | lvlinux, yeah, been looking at it..I'll proceed based on the responses I've got here |
| 19:00.11 | malcolmd | saint_: yay :) |
| 19:01.17 | thansen | lvlinux, I'm not seeing any kind of auth/identifying information for going over the internet for connection (ie: not VPN'd) |
| 19:01.43 | thansen | similar to the sip secret/username etc |
| 19:01.52 | saint_ | @malcolmd I did some pretty simple installations for a couple of customers, going from 2 phones with an online asterisk to a couple of hundred in NYC, saving them so much $$$$$$$$ on their bill. Nothing fancy, but it works. I'll try Call Center features down the road. but I'm focusing on iPhone apps right now. I thought about making an iPad app that would let you configure your Asterisk, but I'm not sure people would use it .. |
| 19:02.21 | thansen | how does mgcp do that, or is it not meant to have multiple connections from the same ip |
| 19:02.33 | thansen | in my case, the office ip |
| 19:02.50 | mjordan | saint_: glad to hear, np |
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| 19:28.56 | stkoch | WIMPy: hi, I have found a commentary from you because of AST_FORMAT_INC http://infobot.rikers.org/%23asterisk-dev/20110208.html.gz |
| 19:29.31 | stkoch | I have ported the chan_lantiq from Asterisk 1.8 to Asterisk11 |
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| 19:31.19 | stkoch | The AST_FORMAT_ULAW have changed from 4 to 100003 |
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| 19:32.23 | stkoch | But the Lantiq TAPI Driver does only allow uint8 values |
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| 19:39.09 | WIMPy | stkoch: You shouldn't need to know the values. |
| 19:39.50 | stkoch | WIMPy: asterisk gives this |
| 19:39.52 | stkoch | channel.c:4301 __ast_read: Dropping incompatible voice frame on TAPI/1 of format unknown since our native format has changed to (ulaw) |
| 19:41.33 | WIMPy | You should use the constants fron the Asterisk header files. |
| 19:41.44 | stkoch | This was set in chan_lantiq.c::lantiq_setup_rtp(): |
| 19:41.50 | stkoch | rtpPTConf.nPTup[IFX_TAPI_COD_TYPE_MLAW] = AST_FORMAT_ULAW; |
| 19:42.11 | stkoch | but nPTup is uint_8 |
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| 19:44.18 | stkoch | AST_FORMAT_ULAW is 100003 (to big) |
| 19:48.19 | WIMPy | You probably shouldn;t define anyting yourself. I don;t think I changed any definitions. |
| 19:49.25 | rmudgett | Even in v1.8 the format type was changed to format_t which is a typedef of int64_t |
| 19:52.32 | stkoch | rmudgett: but the AST_FORMAT_ULAW was 4 (passes uint8) |
| 19:52.58 | stkoch | I have changed AST_FORMAT_INC for test purposes from 100000 to 100 |
| 19:53.11 | stkoch | Then sound works with the chan_lantiq |
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| 19:59.46 | AsteriskNoob | * thinks my phone is dead while my is always up also my phone is behind NAT, however I have 30 extensions and only 2 of them have this issue |
| 19:59.54 | stkoch | Should it be done with wrapping chan_lantiq <-> Lantiq TAPI Driver ? |
| 20:00.18 | [TK]D-Fender | AsteriskNoob: The answer isn't any different in here.... |
| 20:02.07 | AsteriskNoob | I checked with my network guy, he is against disabling sip inspect |
| 20:03.07 | AsteriskNoob | he informed this due to using udp and tls at same time on * |
| 20:03.32 | AsteriskNoob | however I m not convinced that idea |
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| 20:04.54 | AsteriskNoob | [Tk]D-Fender any other work around like increasing or decreasing Keep-Alives? |
| 20:06.01 | Penguin | Asterisk extensions are not affected by keepalive in the network layer. |
| 20:07.18 | [TK]D-Fender | AsteriskNoob: if that's what it is failing then yes |
| 20:07.32 | [TK]D-Fender | AsteriskNoob: Qualify timeout is what it is |
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| 20:26.38 | intheworld | hi everyone |
| 20:27.39 | intheworld | what is host in iax ? is it my host address or asterisk server ip ? |
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| 20:28.36 | lvlinux | I think hostname and IP both work there. Somebody correct me if wrong. |
| 20:29.35 | [TK]D-Fender | Host is where that device is supposed to be contacted |
| 20:29.40 | [TK]D-Fender | The same as for SIP |
| 20:29.44 | intheworld | can work asterisk without elastic ??? |
| 20:30.02 | [TK]D-Fender | intheworld: Please fix the phrasing of your question |
| 20:32.30 | [TK]D-Fender | has never needed an elastic for his Asterisk servers. Occasionally duct tape, but never an elastic... |
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| 20:35.45 | Penguin | I'd rather use a draw string. |
| 20:38.26 | MaliutaLap | I prefer gaffer tape to duct tape |
| 20:38.49 | MaliutaLap | [TK]D-Fender: I don't think we're doing phrasing anymore |
| 20:39.22 | [TK]D-Fender | I'm not seeing a clarification, so it would appear not.... |
| 20:39.53 | MaliutaLap | [TK]D-Fender: https://www.youtube.com/watch?v=d2CE0DjDdVA |
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