| 00:02.23 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 00:09.08 | kaushal | p3nguin: Can i update you here if i have tested it successfully ? |
| 00:14.15 | WiretapWork | techknowlust: go to #freepbx |
| 00:15.02 | techknowlust | WiretapWork: yessri |
| 00:27.19 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 00:36.14 | *** join/#asterisk WiretapWork (~wiretap@unaffiliated/wiretap) |
| 00:39.15 | *** join/#asterisk tmrhmdv (~tmrhmdv@ool-4575afcd.dyn.optonline.net) |
| 00:43.19 | tmrhmdv | Umm, where can I get the latest of this package through apt? http://imgur.com/bIVII |
| 01:05.07 | tmrhmdv | are you guys alive or am I offline? :) |
| 01:06.51 | WIMPy | Does that mean we can only live if you are offline? |
| 01:12.06 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-67-253-245-174.rochester.res.rr.com) |
| 01:14.36 | tmrhmdv | Nope, I was just making sure :) It's okay, Google already helped |
| 01:15.55 | tmrhmdv | Entschuldigung |
| 01:16.13 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
| 01:18.10 | xpot-mobile | Question: having the following issues; can call from inside network through server to sip provider connecting to POTS or Wireless phone with two way voice traffic... can call from one internal ext to another int ext and phone rings and answers... however, int ext cannot hear each other, it appears to be RTP related, just can't narrow down the reason... any ideas? |
| 01:19.57 | ectospasm | xpot-mobile: sounds like standard SIP/NAT issue |
| 01:22.54 | xpot-mobile | ectospasm: I would agree... been banging my head against the server hoping it would fix it... server has an WAN nic and LAN nic... phones internal connect to internal LAN nic... so no firewall on that side ? (it is a Mikrotik though, so WAN is a bridge for some ports and LAN is a bridge for other ports) same device... thinking it might be in there somewhere |
| 01:23.47 | ectospasm | sounds like you're doing NAT between the LAN and the WAN, no? |
| 01:24.58 | xpot-mobile | yes, I have a similar setup at another location and it works very well... I have been trying to compare the two to see where the differences lie |
| 01:25.14 | *** join/#asterisk viaov (~viaov@64.253.187.219) |
| 01:26.54 | xpot-mobile | if I understand asterisk correctly, the server is not performing a translation if two internal ext comes in over LAN card, would they be NAT? |
| 01:26.54 | viaov | Hi after a recent upgrade to asterisk (through fonality) the system went down... the only thing I see is in the CLI there is this error every few seconds "No D-channels available! Using Primary channel 24 as D-channel anyway!" |
| 01:27.11 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 01:27.29 | ectospasm | viaov: your D-channel has gone down, meaning no PRI |
| 01:27.39 | ectospasm | check the cable, check the far end |
| 01:27.54 | ectospasm | ...if everything else didn't change. |
| 01:28.38 | viaov | ectospasm: well this started as a result of a software upgrade, just to be safe let me check the connections |
| 01:29.06 | ectospasm | viaov: what kind of PRI card is this? |
| 01:29.53 | xpot-mobile | I imagine that a call comes in from 192.168.1.x ext to 192.168.1.xx ext... they should hear each other since no firewall exists between them?? still cannot fathom why they cannot hear each other... I could understand why if I was dealing with a WAN ext calling a LAN ext. |
| 01:29.55 | ectospasm | xpot-mobile: if two endpoints on the same LAN are communicating, it's not NAT. |
| 01:30.26 | ectospasm | xpot-mobile: I dunno, could be a codec issue, or you're erroneously setting NAT |
| 01:30.40 | Maliuta | can we use the term "network segment" and not "LAN" ... it's more accurate, 'specially in these days of VLAN and VPN |
| 01:31.09 | xpot-mobile | exctospasm: bizzare.. I will look into your suggestions, thank you. |
| 01:31.28 | xpot-mobile | Maliuta: I will try to comply ;) |
| 01:31.50 | Maliuta | xpot-mobile: have you pastebin'd your sip.conf? |
| 01:32.06 | Maliuta | xpot-mobile: it should hold the key |
| 01:32.59 | viaov | ectospasm: do you know an easy way to tell from the shell? The box they have here was all prebuilt and setup by a contractor. |
| 01:33.17 | ectospasm | viaov: dahdi_hardware |
| 01:33.22 | viaov | I did check the cables though and everything seems good there |
| 01:33.39 | viaov | pci:0000:0b:04.0 wanpipe- 1923:0040 Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card |
| 01:34.04 | tmrhmdv | Fellas, I got stuck at installation: http://pastebin.com/MHR049S2 how can I fix that? :{ |
| 01:34.35 | tmrhmdv | Followed steps in Packages section of the new wiki |
| 01:35.01 | ectospasm | viaov: driver isn't loaded (it's what the '-' in wanpipe- means) |
| 01:35.10 | *** join/#asterisk WiretapWork (~wiretap@unaffiliated/wiretap) |
| 01:35.28 | viaov | ah...ha wouldnt that be done through /etc/init.d/dahdi start ? |
| 01:37.13 | ectospasm | viaov: maybe, I have ZERO familiarity with Sangoma hardware. |
| 01:37.37 | Maliuta | tmrhmdv: what debian version? (stable? unstable? testing?) |
| 01:37.57 | viaov | ectospasm: lol crap. Well I have more information than I did a few minutes ago at least. Thanks |
| 01:38.41 | tmrhmdv | Maliuta: Ubuntu Server 11.04 (natty) |
| 01:38.55 | Maliuta | riiiight |
| 01:39.33 | tmrhmdv | Is it no good for asterisk? |
| 01:39.34 | Maliuta | tmrhmdv: I know that the package you need (asterisk-core-sounds-en-gsm) is in the debian repos |
| 01:39.47 | Maliuta | tmrhmdv: I just don't like baby Debian :) |
| 01:39.53 | tmrhmdv | :) |
| 01:40.50 | Maliuta | tmrhmdv: just get the package direct from a debian mirror, it should install from the command line using dpkg with no problems |
| 01:41.35 | Maliuta | tmrhmdv: alternately http://packages.ubuntu.com/natty/asterisk-core-sounds-es-gsm |
| 01:41.44 | tmrhmdv | Maliuta: OK, thanks will do. I also did apt-get install asterisk-core-en-gsm and it installed but it was an old version |
| 01:42.01 | Maliuta | tmrhmdv: next time use _real_ Debian GNU/linux ;P |
| 01:42.17 | tmrhmdv | haha OK :) |
| 01:42.31 | Maliuta | ahh, the natty is 1.4.19, and that requires .22 |
| 01:42.46 | tmrhmdv | Exactly |
| 01:42.46 | Maliuta | takes a .22 to Ubuntu :P |
| 01:43.04 | tmrhmdv | :) |
| 01:43.20 | Maliuta | tmrhmdv: from memory that package has no dependencies in Deb |
| 01:44.11 | Maliuta | tmrhmdv: and "apt-cache show" holds me up on that one |
| 01:44.34 | xpot-mobile | Maliuta: http://pastebin.com/S4KV1vbQ .... took me a minute to maintain my security, let me know what you think |
| 01:46.47 | xpot-mobile | there are multiple sip includes that were all included appended into this pastebin |
| 01:46.50 | Maliuta | tmrhmdv: the package should be under debian/pool/main/a/astersisk on any debian mirror |
| 01:47.26 | Maliuta | xpot-mobile: have you tried pulling the nat=yes entries and putting canreinvite=no in there? |
| 01:48.04 | xpot-mobile | Maliuta: no, I will try that now |
| 01:48.47 | Maliuta | xpot-mobile: I put canreinvite=no in my [general] |
| 01:49.04 | tmrhmdv | Maliuta: Found it! Thank you :) |
| 01:49.14 | xpot-mobile | Maliuta: ok, and strip out the nat=yes from each ext? |
| 01:51.21 | Maliuta | yup |
| 01:52.09 | Maliuta | put Creedence on the turntable. |
| 01:53.15 | Maliuta | xpot-mobile: is the * box even behind a NAT? |
| 01:54.46 | xpot-mobile | Maliuta: negative, has a WAN static IP tables firewall allowing ports 5060, 10000-20000, and others. LAN static IP no firewall (other than the iptables with chain of 192.168.1.0/24 -j ACCEPT) |
| 01:54.48 | *** join/#asterisk capt_cassimir (~arch@178.73.219.93) |
| 01:55.03 | capt_cassimir | Greetings, nerds |
| 01:56.16 | Maliuta | xpot-mobile: so setting "canreinvite" means that everything passes through * ... because unless you're properly letting RTP through the NAT to the handsets you're screwed :) |
| 01:56.30 | Maliuta | xpot-mobile: using * as the man in the middle for that makes more sense |
| 01:58.09 | xpot-mobile | Maliuta: I do indeed want asterisk to be the man in the middle to maintain call stats and handling... I have lowered all iptables with the same results, this is what led me to believe that my chains are the problem... (although I could try it again with the conreinvite added to general) |
| 01:58.24 | xpot-mobile | *chains are NOT the problem |
| 01:59.35 | Maliuta | xpot-mobile: no, if the box running * is also the firewall then you can't do both (i.e. run * on it and pass RTP through to handsets behind it) |
| 02:00.49 | xpot-mobile | Maliuta: I do this same thing with 5 other servers... it works just fine ?? |
| 02:01.02 | xpot-mobile | just not on this one apparently |
| 02:01.22 | Maliuta | xpot-mobile: basically the firewall should let SIP and RTP packets in (through the INPUT chain) to * and then the handsets should be talking to * and every thing should have "canreinvite" and "canredirect" set to no (so doing it in [general] applies it to all] |
| 02:03.04 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 02:04.18 | xpot-mobile | Maliuta: correct my INPUT chains do allow the above mentioned ports (and other requirements) through... i even turned off iptables (no firewall) and issue still exists, I will add "canredirect=no" to the general as well |
| 02:06.29 | Maliuta | xpot-mobile: try doing a sip debug on a call between the two handsets with that (since everything should now be passing through the * box) |
| 02:07.25 | xpot-mobile | Maliuta: rgr, performing the operation now |
| 02:10.05 | Maliuta | capt_cassimir: I'm a geek, not a nerd ... you complete twat :P |
| 02:10.31 | capt_cassimir | I'm not a twat, i'm a neckbeard |
| 02:13.00 | Leddy | Executing [s@auto:1] Answer("SIP/sip.provider.com", "") in new stack <-- whats the second arg used for? Is that for passing the did? |
| 02:20.09 | *** part/#asterisk tmrhmdv (~tmrhmdv@ool-4575afcd.dyn.optonline.net) |
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| 02:24.11 | xpot-mobile | Maliuta: http://pastebin.com/KAWxcAgL <--- sip debug for call from ext xxx to ext nnn and vice versa |
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| 02:40.34 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 02:43.10 | Maliuta | xpot-mobile: about 1/2 way through that, looks like there might be some issues in your dialplan ... so far :) |
| 02:44.00 | capt_cassimir | so i'm reading The Book, and he has you create a new user for asterisk AND add the user to sudoers... |
| 02:44.06 | techknowlust | is there a simple way to know if a call is causing transcoding ?\ |
| 02:44.28 | capt_cassimir | you don't really get a security benefit of installing as a non-root user if the user has sudo... but i'll keep reading. |
| 02:45.09 | Maliuta | xpot-mobile: and you have an AGI script returning non-zero (i.e. bad) |
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| 02:47.01 | Maliuta | xpot-mobile: that was a call from one internal host to another? |
| 02:47.45 | Maliuta | xpot-mobile: I am also assuming you edited that to remove IP addresses. That makes it a little hard to follow. But I think you may need to look at the AGI and the dialplan |
| 02:49.07 | xpot-mobile | Maliuta: no, I am not on site now... that was two external phones behind the same NAT (ie: here at my house, which probably isn't helpful) |
| 02:49.32 | xpot-mobile | I will check out the AGI issue you mentioned |
| 02:52.22 | techknowlust | is it possible to see from the sip debug output whether a call has had to use transcoding ? |
| 02:52.51 | capt_cassimir | starts compiling |
| 02:53.40 | xpot-mobile | Maliuta: I do appreciate you checking that out for me... kudos to you ;) |
| 02:54.25 | Maliuta | techknowlust: there was no transcode, it was a native bridge |
| 02:54.47 | techknowlust | Maliuta: ? |
| 02:56.30 | Maliuta | techknowlust: it did look like there was an issue with the incoming call not being able to connect to anything (all lines busy from app Dial) |
| 02:57.21 | capt_cassimir | kicks off make for asterisk 1.8.5 |
| 02:57.22 | techknowlust | is there a way though of monitoring what calls are transcribed ? |
| 02:57.46 | Maliuta | transcribed? |
| 02:58.03 | techknowlust | transcoded* |
| 02:58.10 | Maliuta | that seems to be the wrong term for anything * related ... unless you're doing speech to text |
| 02:58.49 | techknowlust | sorry I meant transcoded. I'm trying to figure out if all the calls I've made have had to use transcoding between codecs |
| 02:59.18 | Maliuta | techknowlust: short of using the cli during calls I can't think of a way |
| 03:00.01 | p3nguin | techknowlust: Yes, there is a way to know if a call is being transcoded. Just look at the codecs in use for a call and see what apps are in use on said call. |
| 03:00.58 | Maliuta | p3nguin: that's basically what I said ... I think he's after a way to log them though |
| 03:01.03 | p3nguin | For past calls, I don't think you can see what codecs were used... not unless you thought of this in the past and set up something to document it so you can go back and read it. |
| 03:01.44 | p3nguin | In my opinion, it's not really important to know if transcoding occurred in the past. |
| 03:01.53 | Maliuta | shudders at the thought of an AGI that simply logs that stuff |
| 03:05.38 | p3nguin | DumpChan() contains that info. There may be some other app that you could wrap around it to filter out just the codec info. I'd imagine your idea of an AGI would be more reasonable, though. |
| 03:06.16 | techknowlust | p3nguin: I'm using two android phones at the moment. |
| 03:07.57 | techknowlust | p3nguin: the only line I think might be useful is this http://pastie.org/2215825 |
| 03:08.28 | techknowlust | seeing that there is an overlap of codecs, is that enough to say it wasn't transcoded |
| 03:08.31 | techknowlust | ? |
| 03:09.33 | p3nguin | That shows one leg of the call. |
| 03:09.52 | p3nguin | "us" means Asterisk. "peer" means a phone. |
| 03:10.06 | techknowlust | hmm ok. I can post the whole config if you like |
| 03:10.17 | p3nguin | There are two of those codec negotiations for a call between two phones. |
| 03:13.48 | capt_cassimir | What is an AGI? |
| 03:14.11 | techknowlust | http://pastie.org/2215844 p3nguin that's the full log |
| 03:17.39 | p3nguin | One side is using ulaw, the other side has gsm and ulaw available. It probably used gsm, and then transcoding would have occurred. |
| 03:18.13 | p3nguin | I don't see where is says what codec is used, just where it says what is available between "us" and the "peer." |
| 03:18.25 | p3nguin | Maybe maliuta can find it and tell us. |
| 03:18.43 | p3nguin | ~agi |
| 03:18.43 | infobot | well, agi is the Asterisk Gateway Interface... similar to CGI for web applications AGI lets you script call control and access databases using your favorite language. AGI wrappers are available for Python (pyst), Perl (astperl?) and other languages, or <reply> See also http://www.voip-info.org/wiki-Asterisk+AGI |
| 03:18.47 | p3nguin | capt_cassimir: ^^^ |
| 03:19.12 | capt_cassimir | That would have been my guss, thanks for clarifying |
| 03:20.06 | techknowlust | p3nguin: it would default to gsm even if ulaw were available on both ? |
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| 03:23.57 | p3nguin | It appears to me that gsm is the preferred codec on that one phone. |
| 03:26.13 | techknowlust | surely there should be a way of detecting this. weird |
| 03:30.57 | jmwpc | Question regarding Google talk & Google voice integration... I have everything working fine, except that when I try to log into Google talk with my IM client, incoming calls are no longer picked up by asterisk. When logged in to the asterisk console, the jabber messages also cease. I'm sure it's a problem with being logged in at multiple locations, does anyone know a way around this? Or am I stuck using 2 different accounts? |
| 03:31.57 | capt_cassimir | oh that's shitty, i hope there's a fix |
| 03:32.05 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 03:32.09 | p3nguin | It's because you're using two resources. |
| 03:32.14 | *** join/#asterisk ccesario (~ccesario@177.35.98.248) |
| 03:32.19 | p3nguin | I don't personally know of a workaround for it. |
| 03:33.12 | jmwpc | p3nguin: That's what I was afraid of. Maybe some kind of jabber proxy on the asterisk box, but that sounds overly complex. I think the second account is the best bet. |
| 03:35.12 | jmwpc | Aside from that problem, it's working great. I actually have it connected to 2 google accounts, my wife's and mine. With the Linksys ATA we now have 2 separate lines which is kind of handy at times (cell reception is bad). |
| 03:37.08 | p3nguin | You're having bad quality when using both accounts? |
| 03:38.18 | jmwpc | No, quality is fine. I was just saying that the cell phone reception at my house is spotty, so the 2 'land' lines come in handy now. |
| 03:38.26 | p3nguin | Ah, got it. |
| 03:38.32 | jmwpc | :) |
| 03:38.50 | capt_cassimir | jmwpc: that's awesome, i'm working on that type of setup now |
| 03:39.34 | capt_cassimir | Google voice routes to my cell phone, would be nice to route to a "landline" on occasion for better quality / reliability |
| 03:39.51 | jmwpc | capt_cassimir: Nice :) The quality is surprisingly good. I have only been working with asterisk for about 2 days now, so I'm sure my config isn't perfect, but it works. It's been a good learning experience. |
| 03:40.32 | capt_cassimir | Did you find a particular HOWTO? |
| 03:41.10 | jmwpc | capt_cassimir: This one -> http://pcprob.blogspot.com/2011/03/howto-use-google-and-asterisk-for-free.html |
| 03:41.17 | capt_cassimir | I'm working through the oreilly book now, trying to fortify my asterisk understanding |
| 03:41.32 | capt_cassimir | jmwpc: bookmarked, thanks |
| 03:41.47 | linuxgecko | jmwpc: i just recently got a google-voice/asterisk setup working right :) |
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| 03:42.02 | jmwpc | capt_cassimir: I even imported all of our google contacts into asterisk for caller id. That page links to the instructions. |
| 03:42.11 | capt_cassimir | whoa, fancy |
| 03:43.51 | jmwpc | linuxgecko: Looks like a lot of people are going that way. I'll drop my regular land line after a couple of weeks if everything remains stable :) |
| 03:50.34 | linuxgecko | jmwpc: it "Just Makes Sense" :) asterisk is free, g-voice is free, and finally, with 1.8, connecting the 2 is fairly simple. for most people, you have one cost. a one-time ATA purchase :) |
| 03:51.08 | linuxgecko | and i got mine second-hand, locked to vonage.. luckily there are great docs out there for unlocking them :) |
| 03:53.01 | linuxgecko | jmwpc: the big proble is up=time. john-q public is not well versed in setting up a server for phone service, with a 99.9% uptime situation |
| 03:53.29 | linuxgecko | as it is, my asterisk server is on a vobx :) |
| 03:54.00 | jmwpc | my asterisk server is also a vbox server ;) |
| 03:56.49 | *** join/#asterisk kaushal (~kaushal@49.248.16.122) |
| 03:56.58 | kaushal | Hi p3nguin |
| 03:57.04 | kaushal | Still it doesnot work |
| 03:57.14 | kaushal | Is there a way to enable debug ? |
| 04:03.00 | p3nguin | What are you trying to do? |
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| 04:47.39 | kaushal | p3nguin: please give me a moment |
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| 06:24.33 | joobie | hey guys.. polycom 321 phone - any idea why presence is not working on it? |
| 06:34.58 | WiretapSeven | ~ask |
| 06:34.58 | 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. |
| 06:44.18 | joobie | when a voicemail is left fo rthe user |
| 06:44.21 | joobie | the light doesnt light up |
| 06:44.34 | joobie | is there a setting that specifically controls this? |
| 06:45.30 | kaldemar | so you're not talking about presence but MWI. have you defined a mailbox for the device in sip.conf? |
| 06:45.43 | joobie | yes |
| 06:46.02 | joobie | mailbox=3019@default |
| 06:46.38 | joobie | and i have [default] 3019 => 3019,User,email@domain.com |
| 06:46.42 | joobie | .. within voicemail.conf |
| 06:47.19 | kaldemar | does the phone subscribe? do you see the notify message in sip debug when the box gets mail? |
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| 07:17.34 | ChannelZ | there might be a setting in the sphone its self to enable/disable MWI |
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| 07:29.12 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:29.14 | schmidts | good morning |
| 07:30.25 | irroot | morning shmidts |
| 07:30.31 | irroot | morning schmidts |
| 07:33.15 | schmidts | more coffee? |
| 07:42.46 | *** join/#asterisk gmaruzz (~gmaruzz@static-217-133-80-112.clienti.tiscali.it) |
| 07:43.28 | irroot | straight up in a IV tube please |
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| 07:51.36 | schmidts | :D |
| 07:51.53 | f2Knight | Q: using AGI is it possible to have your script return a set of variables to be used in the dialplan? e.g. I want to do a lookup in a database to see how many credits a caller has. Then assign that value to a channel Variable .. ${CALLERCREDITS} that I can use later to track when to disconnect the caller. |
| 07:53.17 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 07:55.54 | irroot | sure thing not hard do it here |
| 07:56.20 | irroot | just set the chan vars |
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| 08:01.37 | f2Knight | irroot, i am writing my AGI in python... but what would be the commands to send back from the AGI so I could read them in the dialplan? |
| 08:02.39 | irroot | not sure with the python bits still using the older php |
| 08:03.46 | f2Knight | irroot, okay well how about a php example? maybe I can see what its doing and modify it to my needs |
| 08:03.59 | f2Knight | irroot, at least it might get me looking in the right direction. |
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| 08:06.11 | irroot | $agi->set_variable("USERNAME",$username); |
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| 08:18.55 | f2Knight | irroot, i guess I need to tear open the phpagi app then to see just what -> set_variable is calling |
| 08:21.47 | f2Knight | irroot, awesome.. looks like pyst has a simiular function. Thanks! knowing what to look for was all I needed. |
| 08:29.31 | kaldemar | f2Knight: or just look at the documentation on the asterisk side. "agi show commands" in CLI. |
| 08:30.58 | f2Knight | kaldemar, that would have worked too , seems all I need is to do a SET VARIABLE "MY_VAR" "VALUE" |
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| 08:33.37 | kaldemar | and there are also python libraries if you want to use such. |
| 08:34.00 | f2Knight | Q: Does anyone know which is faster? the internal Asterisk DB or using MySQL / Sqlite ? And if AstDB, is there an easy way to backit up or use information in it with external services...? |
| 08:34.25 | f2Knight | kaldemar, i am leaning towards pyst/agilib.py |
| 08:35.11 | Chainsaw | f2Knight: If the MySQL instance is local, I would expect it to be as fast or faster. |
| 08:35.34 | Chainsaw | f2Knight: If it is several milliseconds of latency away on a different box, the astdb has an edge. |
| 08:36.09 | f2Knight | Chainsaw, the DB will be local to start, but may move to a SANS unit on a private network at a later point. |
| 08:37.28 | Chainsaw | f2Knight: Then it will win until you move it away, after which it will lose. |
| 08:38.03 | Chainsaw | f2Knight: But since you can't move the AstDB out to an external unit, you may want to consider other factors than just raw latency/performance. |
| 08:38.03 | f2Knight | Chainsaw, do you think it will be bad? performance wise? |
| 08:38.28 | Chainsaw | f2Knight: Bad? No. But that isn't what you asked. |
| 08:38.40 | f2Knight | Chainsaw, ahh okay . Thank you :) |
| 08:39.12 | Chainsaw | f2Knight: It is possible to have read-only access to the AstDB (it is a standard database format), but writes would be problematic. |
| 08:39.57 | f2Knight | Chainsaw, no I am content using MySQL, just didn't know if the over head of a AGI reading and writing to it would be that much of a performance hit... |
| 08:40.01 | f2Knight | heres what I am doing... |
| 08:40.36 | f2Knight | caller calls in , I verify there callerid and search a db for available credits. (credits are used for talking time) |
| 08:41.18 | f2Knight | I am thinking to dump them to a call Queue, where I have agents registered to handle the calls. |
| 08:41.49 | f2Knight | If there credit runs out they are disconnected from the queue... ( I think i can do that with queues) |
| 08:42.52 | f2Knight | I also want the caller to be able to press a digit to get out of the queue.. actually 2 keys 1 to skip to a random queue, and 1 to return to a menu. ( can you do that with queues?) |
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| 08:55.58 | kaldemar | f2Knight: see context option in queues.conf for exiting a queue with a digit. |
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| 08:58.26 | kaldemar | you really don't need agi for that, you can use func odbc from your dialplan directly. |
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| 09:29.48 | prash10x | how to execute commands in dialplan during a call |
| 09:30.45 | kaldemar | prash10x: see [applicationmap] in the sample features.conf. |
| 09:31.05 | prash10x | ok tx |
| 09:36.46 | jkroon | hi guys, looking to understand the impact of using srtp on load and bandwidth. are there any pointers I can look at? |
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| 09:45.46 | dym | Can someone tell me why my asterisk still plays english voice prompts? http://paste.debian.net/122963/ |
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| 09:50.48 | f2Knight | kaldemar, yes i know i could use func odbc.. but i need to keep the application logic away from prying eyes. I am being paid on commission from the system. and I do not want to have them just take the code and use it with out giving me my commission. |
| 09:51.10 | kaldemar | dym: that doesn't show asterisk playing anything. |
| 09:51.45 | dym | kaldemar: i know, cause thats the dialplan part. Asterisk just doesnt play the correct files, even though they are in the right place, etc. |
| 09:52.35 | f2Knight | kaldemar, I can run a python scrip through cxfreese and it makes an executable that runs on the system.this way I can keep the code and they can have the "app" I put a hook back to my server to look for a product key at random intervals. |
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| 09:59.49 | SunTsu | f2Knight: so, when your customer is security minded and restricts outgoing connections he loses your work? |
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| 10:01.25 | f2Knight | SunTsu, he already knows there is a call back for the license model. this really is not all that uncommon. its an outbound port 80 request to a simple service that checks the key and returns a code |
| 10:02.47 | SunTsu | f2Knight: he better know, else you could be running into trust issues |
| 10:03.10 | jkroon | anti-trust ... gotta love the world of proprietary work ... |
| 10:03.33 | f2Knight | SunTsu, he totally knows and signed and agreement. before i started work |
| 10:04.08 | SunTsu | f2Knight: still I find it somehow strange to do such things based on open source software |
| 10:05.00 | f2Knight | I mean personally I don't care its python or php or bash or whatever, I just want to be able to keep track of how many calls come in and for how long so i know how much he owes me :) so he knows I keep a connection back for logging systistics and reporting and licensing. |
| 10:05.45 | SunTsu | which brings me to the question of AGI's license |
| 10:06.33 | SunTsu | and about the legality of the kind of data tracking you say you do |
| 10:07.10 | SunTsu | but I'm sure you had some lawyers check all this |
| 10:08.51 | f2Knight | SunTsu, yes I did, we are very careful what I am keeping track of .. call logs only really ( and thats only a snapshot at night that is encrypted) but thats not the point. The point is this |
| 10:09.31 | f2Knight | how else do you keep a customer honest when your being paid by commission on call volume. when the system is deployed on all there hardware.? |
| 10:09.45 | f2Knight | you have to have some loop to verify the data |
| 10:10.39 | f2Knight | so I simply collect incoming callerID timestamp and durration, they keep all other billing information customer names cc info etc |
| 10:11.16 | f2Knight | that way I can tabulate what my 25% is :) |
| 10:12.36 | f2Knight | SunTsu, on AGI .. agi simply calls any external process. if I wrote it in C++ and compiled it it would be just the same as me writing it in anyother language. |
| 10:13.14 | f2Knight | only reason for compiling is to make sure that they just don't comment out the lic. check and billing code |
| 10:15.22 | SunTsu | f2Knight: I'm an open source advocate, so I'd prolly take a different licensing approach, and I don't believe in forcing customers to honesty, it's what music and film industry fail to do for years |
| 10:16.43 | f2Knight | SunTsu, how would you suggest it. ? I would be open to other models. but the customer is only paying a % on billable calls. |
| 10:18.29 | SunTsu | f2Knight: OK, if he wants it this way... |
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| 10:20.13 | SunTsu | I'd rather go with a fixed price |
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| 10:34.35 | puzzled | hi |
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| 10:55.56 | f2Knight | SunTsu, so would I but he didnt want to and wanted to pay a commission (this project has an easy potential of making a 50kmin of billiable talk time a month.) |
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| 11:10.17 | Psi|4ward | Hi, is there a Problem with IAX2 Trunk between 1.6 and 1.8? My Trunk are always UNREACHABLE and Asterisk sends only POKE and RETRY packets |
| 11:11.45 | Psi|4ward | POKE and REGREQ |
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| 11:18.11 | kaldemar | Psi|4ward: is there a problem with network connectivity? |
| 11:18.37 | Psi|4ward | nmap -sU -p 4569 says that the port is OPEN | Filtered |
| 11:19.02 | Psi|4ward | but inbetween theres a NAT but the connect to outside should work? |
| 11:21.41 | kaldemar | Psi|4ward: nmap says open|filtered also when packets are dropped, so that really does not prove anything. |
| 11:22.25 | Psi|4ward | is there another way to test the connection? |
| 11:25.15 | kaldemar | netcat is a quick way. shut down asterisk and do a "netcat -u -l 4569" in host1 and "netcat -u host1_address 4569" on host2. write text and hit enter, if it shows up on the other terminal then you have connectivity. |
| 11:25.33 | Psi|4ward | uh cool, dont know natcat supports udp |
| 11:25.42 | Psi|4ward | ill try it |
| 11:26.36 | kaldemar | the NAT is probably not configured to forward the port to your asterisk box. |
| 11:27.12 | Psi|4ward | but then the Asterisk-binhind-nat should be able to connect to asterisk-outside-nat |
| 11:27.55 | kaldemar | Psi|4ward: not necessarily. |
| 11:30.21 | *** join/#asterisk wasanzy (~emmanuel@196.201.43.55) |
| 11:30.37 | Psi|4ward | okay seems i have an inbound problem, thanks for the netcat hint! |
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| 11:41.09 | wasanzy | hi guys |
| 11:41.30 | wasanzy | by default, does asterisk comes with some sound codecs? |
| 11:41.50 | wasanzy | or do I hv to install them manually? |
| 11:44.05 | Psi|4ward | i installed the dev-packages and switched the support in make menuconfig on |
| 11:45.26 | Psi|4ward | kaldemar: do you how to change the port iax2 tries to connec to? i switched the standard-port to a open-one (6000) |
| 11:51.23 | kaldemar | Psi|4ward: depends on how you connect, but bindport defines what port asterisk listens to, registration statements take a port with @host:port, device definitions use "port" option and dialstrings take a port aswell with @host:port. |
| 11:51.58 | wasanzy | no one to answer me? |
| 11:52.48 | Psi|4ward | wasanzy: i installed the dev-packages and switched the support in make menuconfig on |
| 11:53.32 | kaldemar | wasanzy: yes it does come with plenty of codecs. |
| 11:55.07 | wasanzy | and how do I know they are active because am having problem with sound, I can hear it ring when some one call me using twinkle but I can't hear the person talk,neither can the person hear me talk |
| 11:55.16 | wasanzy | pls help me out |
| 11:55.25 | Psi|4ward | kaldemar: it works, many many much thanks ;) |
| 11:56.11 | Psi|4ward | oh it works only in one direction, but i think ill figure this out too |
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| 11:58.01 | kaldemar | wasanzy: did you configure codecs with allow and disallow lines in sip.conf? |
| 11:58.30 | kaldemar | wasanzy: is there a NAT involved in the network? |
| 11:58.59 | wasanzy | I did allow=all in the sip.conf when it was not working |
| 11:59.07 | wasanzy | no we are not using NAT |
| 11:59.37 | wasanzy | we have a wireless router we all use |
| 12:00.50 | kaldemar | change that to disallow=all and then allow the codecs you want one by one with allow lines. |
| 12:01.08 | kaldemar | then do a sip reload and dial away. |
| 12:01.16 | wasanzy | ok |
| 12:01.23 | wasanzy | will get back to u |
| 12:02.35 | wasanzy | gsm is one of the codecs right? |
| 12:03.06 | kaldemar | yes |
| 12:05.43 | wasanzy | kk |
| 12:08.43 | wasanzy | am trying to debug the sip and let u see the errors |
| 12:12.14 | wasanzy | kaldemar: have a look at the sip debug: http://pastebin.com/rn2NCtvE |
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| 12:16.33 | Psi|4ward | kaldemar: asterisk1 sends POKE on port 6000 but REGREQ on 4569? have u an idea why? |
| 12:17.16 | kaldemar | wasanzy: so you're running asterisk and two softphones on the same machine? |
| 12:17.42 | kaldemar | Psi|4ward: you haven't configured port 6000 to your registration statement in iax.conf. |
| 12:17.55 | Psi|4ward | i dont use register statements |
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| 12:18.37 | wasanzy | yes |
| 12:19.50 | wasanzy | <kaldemar>: will that not work> |
| 12:19.52 | wasanzy | ? |
| 12:20.38 | kaldemar | Psi|4ward: then it shouldn't even send REGREQ messages. check your iax.conf again for "register => ..." lines. |
| 12:20.44 | wasanzy | right now, it sound like, after I speak, it takes some time for the sound to come out, and is breaking too |
| 12:21.45 | kaldemar | wasanzy: the soft phones seem to be trying to use the same port for the RTP stream. try a scenario that is even remotely realistic. |
| 12:22.04 | Psi|4ward | kaldemar: youre right, there was a reg-string, killed it. now i get some POKE follwing an ACK and than it goes on with POKE |
| 12:22.49 | kaldemar | Psi|4ward: the POKE messages are because of "qualify" options in iax.conf. |
| 12:22.56 | wasanzy | so the two soft phone should be on a two different machines right? |
| 12:23.41 | kaldemar | wasanzy: sounds better. |
| 12:24.12 | wasanzy | ok let me try that and see |
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| 12:28.08 | wasanzy | hmm I can't get a different machine, so have to find a way out using it all on the same machine |
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| 12:30.52 | kaldemar | wasanzy: force all three to use different ports then. |
| 12:31.05 | Psi|4ward | kaldemar: you u take a look to my config? http://pastebin.com/ep2SkzhE |
| 12:31.30 | kaldemar | Psi|4ward: what should i look for? |
| 12:31.34 | wasanzy | hmmm, now the problem is how do I force all to use a different port? |
| 12:31.57 | Psi|4ward | perhaps ive some error in it |
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| 12:32.47 | kaldemar | wasanzy: asterisk in sip.conf and rtp.conf, the clients i don't know about. |
| 12:33.22 | wasanzy | ok |
| 12:33.38 | wasanzy | I can do it for the client so no problem |
| 12:34.08 | kaldemar | Psi|4ward: is it working? |
| 12:34.43 | Psi|4ward | Server2 connects to server1, but server1 does onle POKE POKE POKE ACK POKE .... |
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| 12:36.38 | wasanzy | in rtp.conf, do I put this: rtpbindport=n ? |
| 12:38.59 | kaldemar | wasanzy: where did you come up with that? |
| 12:39.50 | wasanzy | am just guessing as I don't know, I want to force the rtp audio to use a particular port |
| 12:41.08 | kaldemar | wasanzy: you should look at the sample configs for your installed version when poking options in the files. there you will find rtpstart and rtpend. |
| 12:41.34 | kaldemar | Psi|4ward: what do you mean by connect? |
| 12:41.58 | Psi|4ward | IAX2 peer status OK |
| 12:42.06 | wasanzy | I found that, but it didn't mention any thing concerning port that is why |
| 12:42.29 | wasanzy | should I change those values for the start and end? |
| 12:42.36 | kaldemar | Psi|4ward: are both boxes bound to port 6000? |
| 12:42.39 | Psi|4ward | perhaps its truly a problem with asterisk 1.6 and 1.8? |
| 12:42.45 | Psi|4ward | yes both on 6000 |
| 12:43.06 | kaldemar | which 1.6? |
| 12:43.25 | Psi|4ward | theone which says unreachable to 1.8 |
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| 12:43.45 | Psi|4ward | ive 1.6 with Gemeinschaft and 1.8 with FreePBX |
| 12:43.55 | kaldemar | which verson? there are three branches of 1.6.X, all of them different. |
| 12:44.41 | Psi|4ward | Asterisk 1.6.2.9-2+squeeze1 and Asterisk 1.8.4.4 |
| 12:44.48 | kaldemar | wasanzy: they define the range of ports that asterisk uses, change them as you wish. |
| 12:45.01 | wasanzy | oh ok |
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| 12:47.19 | wasanzy | what of running one soft-phone on the same machine as asterisk and another phone on a different machine, will that also conflict ports? |
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| 12:48.20 | fulcan | does the asterisk api except instruction from port 5060? |
| 12:48.32 | fulcan | like a peer? |
| 12:48.46 | wasanzy | me? |
| 12:48.49 | cusco | hu? |
| 12:48.57 | cusco | port 5060 is for sip |
| 12:49.02 | cusco | what do you mean by api? |
| 12:49.18 | fulcan | http://www.voip-info.org/wiki/view/Asterisk+manager+API |
| 12:49.41 | SuPrSluG | ami usese 5038 |
| 12:49.43 | cusco | manager |
| 12:49.50 | fulcan | cusco it's got to get its instructions somehow if not localhost.? |
| 12:49.51 | SuPrSluG | uses |
| 12:50.08 | cusco | have you enabled it in /etc/asterisk/manager.conf ? |
| 12:50.14 | cusco | also you can set a specific port there |
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| 12:52.22 | fulcan | cusco I was sure I was going to have to enable it, following enable, if I write a remote python script to say for instance 'forward calls to vm' and that script is sitting on a box that is 'not' asterisk, that chunk of code is going to have to get crammed down the asterisk gullet for digest somehow???? |
| 12:53.07 | fulcan | would that route be 5060 by chance? |
| 12:53.20 | fulcan | or could be used? |
| 12:53.20 | SuPrSluG | 5038 |
| 12:54.14 | SuPrSluG | default port of ami |
| 12:54.14 | fulcan | SuPrSluG ty |
| 12:55.04 | Psi|4ward | kaldemar: du have another tip for me? |
| 12:55.05 | SuPrSluG | you may want to create a different user and password |
| 12:55.23 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 12:55.47 | SuPrSluG | you can also permit/deny by ip/network address |
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| 12:58.57 | Psi|4ward | kaldemar: it now works in both directions, ive rebootet both machines |
| 12:59.06 | Psi|4ward | kaldemar: u probably have a wishlist? |
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| 13:00.59 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 13:19.17 | kaldemar | Psi|4ward: hmm.. did you reload or restart asterisk after making the port changes in configuration? |
| 13:19.33 | Psi|4ward | restarted asterisk per init.d script |
| 13:20.06 | kaldemar | ok. strange if a reboot made it work but a restart didn't. |
| 13:20.09 | kaldemar | wishlist? |
| 13:20.37 | Psi|4ward | amazon wishlist or smthg like that |
| 13:20.47 | Psi|4ward | i would thank u ;) |
| 13:21.18 | kaldemar | oh, i don't. it's all ok, help someone else if you can. |
| 13:21.31 | Psi|4ward | thanks ;) |
| 13:21.47 | Psi|4ward | perhaps uve a webdev-problem |
| 13:24.10 | kaldemar | thanks, i'm all good. :) |
| 13:25.49 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 13:28.09 | tzanger | greetz from .ro |
| 13:28.15 | tzanger | god bless the internet and asterisk |
| 13:28.16 | Kamineko | can someone help me with a trunk between asterisk and draytek 2820 ? |
| 13:29.10 | fulcan | where is the asterisk db? |
| 13:29.58 | kaldemar | fulcan: where astdbdir in asterisk.conf defines it to be. by default in /var/lib/asterisk. |
| 13:31.11 | fulcan | kaldemar is there any way to show the contents of it? |
| 13:31.51 | kaldemar | "database show" in CLI. |
| 13:31.52 | fulcan | kaldemar field parameters etc..? |
| 13:34.00 | *** join/#asterisk bchia (~Adium@nat/digium/x-ebxdzwcoqvwycxjk) |
| 13:34.04 | fulcan | does the asterisk agi update the asterisk db or the runtime memory? |
| 13:35.00 | *** join/#asterisk Ryushin (proxy@cl-412.phx-01.us.sixxs.net) |
| 13:36.10 | kaldemar | fulcan: the database commands modify both. |
| 13:37.16 | fulcan | if I wanted to write a script to forward a call to vm, I am trying to figure out exactly what is updated. since the configuration for the vm is in extensions.conf, if a line gets forwarded but this command is not updated in extensions.conf, this could create some confuzion. I am trying to figure out 'what is the sheet of music' so that all parties can be on the same sheet...? |
| 13:38.25 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:38.25 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 13:49.26 | *** part/#asterisk Ursinha (~ursinha@canonical/launchpad/ursinha) |
| 13:51.15 | *** join/#asterisk m_tadeu (~quassel@89.180.11.177) |
| 13:56.43 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 13:57.56 | *** join/#asterisk gavimobile (~user@87.68.161.167) |
| 13:58.07 | gavimobile | in the list of libraries needed for installing asterisk "Kernel headers (for building DAHDI drivers)" package cannot seem to be found. any ideas? |
| 13:58.16 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 14:03.23 | Chainsaw | gavimobile: What linux distribution are you using? |
| 14:04.12 | *** join/#asterisk pabelanger (~pabelange@nat/digium/x-xhuakbmbdwxnnitv) |
| 14:04.12 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:05.39 | kleszcz | gavimobile: apt-get install linux-headers-`uname -r` |
| 14:07.44 | m_tadeu | hi...I'mgetting a warning at asterisk startup saying "res_musiconhold.c: Unable to create timer: Success"...whatdoes this mean exaclty? |
| 14:10.02 | *** join/#asterisk fIorz (nobody@2001:1a50:503c::1) |
| 14:11.34 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:11.34 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:12.08 | *** join/#asterisk acidfoo (~nib@modemcable094.94-70-69.static.videotron.ca) |
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| 14:19.21 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:19.58 | p3nguin | m_tadeu: Sounds like you could be missing a timing mechanism, such as Dahdi. |
| 14:24.28 | m_tadeu | p3nguin: isn't dahdi just to contol fxo/fxs cards? |
| 14:25.21 | p3nguin | no |
| 14:25.26 | *** join/#asterisk garymc (~chatzilla@host81-139-157-61.in-addr.btopenworld.com) |
| 14:25.44 | p3nguin | It also has the ability to provide a timing source. |
| 14:26.47 | m_tadeu | I'm only using sip channels....should I install dahdi also? |
| 14:27.02 | dym | m_tadeu: always good to have it. |
| 14:27.10 | dym | as a timing device, as p3nguin already mentioned. |
| 14:27.14 | p3nguin | If you need a timing source, which you do, dahdi can provide it. |
| 14:27.41 | m_tadeu | I see....thanx for your help |
| 14:36.13 | *** join/#asterisk coppice (~chatzilla@116.92.29.9) |
| 14:37.22 | wasanzy | <p3nguin>: am still having problem with my sound, will you want to see "core show translation" output and see if that is good? |
| 14:38.12 | m_tadeu | still getting the same warning after installing dadhi |
| 14:39.02 | *** join/#asterisk zerohalo (~zerohalo@173-13-92-27-NewEngland.hfc.comcastbusiness.net) |
| 14:40.19 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:41.12 | wasanzy | <p3nguin>: my core show translation output: http://pastebin.com/YezjTK03 |
| 14:41.37 | kaldemar | m_tadeu: how did you install it? |
| 14:41.54 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 14:42.31 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 14:42.37 | m_tadeu | kaldemar: via package system...I'm using the packages.asterisk.org repository |
| 14:42.54 | wasanzy | <kaldemar>: I though u are gone |
| 14:43.07 | kaldemar | wasanzy: i was for some time. |
| 14:43.12 | wasanzy | am still having the sound problem |
| 14:43.49 | wasanzy | do want to see core show translation output? |
| 14:44.12 | wasanzy | I don't understand any thing in there |
| 14:44.39 | kaldemar | i saw it already. it won't offer anything. |
| 14:44.47 | kaldemar | still no sound at all? |
| 14:45.29 | wasanzy | yes,u can hear the phone ring alright but you can't hear each other talk |
| 14:46.44 | wasanzy | am confuse as what is actually wrong |
| 14:47.32 | *** join/#asterisk gavimobile (~user@87.68.161.167) |
| 14:49.56 | kaldemar | are they using different ports? |
| 14:50.26 | wasanzy | yes, but this time not running on the same machine |
| 14:50.48 | wasanzy | should I do the debug again? |
| 14:53.14 | kaldemar | by all means, do. |
| 14:54.03 | wasanzy | ok |
| 14:56.04 | wasanzy | http://pastebin.com/0Z696cfm |
| 14:57.22 | p3nguin | This time you have three computers: one with asterisk, one with a softphone, and another with another softphone? |
| 14:57.36 | *** join/#asterisk wonderworld (~ww@port-92-201-109-97.dynamic.qsc.de) |
| 14:57.45 | *** join/#asterisk joker2u (~root@li345-191.members.linode.com) |
| 14:58.02 | joker2u | is there an asterisk dev channel? |
| 14:58.14 | acidfoo | <PROTECTED> |
| 14:58.44 | wasanzy | no, is two computers, one with asterisk and soft-phone, the other with soft phone only |
| 15:00.58 | *** join/#asterisk engrxyz (~fgdfgfdg@212.23.51.7) |
| 15:01.19 | wasanzy | did you see any error? |
| 15:02.24 | *** part/#asterisk joker2u (~root@li345-191.members.linode.com) |
| 15:02.26 | p3nguin | When you're no longer trying to run a phone on the same computer as asterisk, and you're still having problems, then get back to me. |
| 15:03.22 | wasanzy | oh ok |
| 15:03.42 | wasanzy | I think that is what I hv to do now |
| 15:06.28 | *** join/#asterisk shine (~stroll@163.5.69.15) |
| 15:06.30 | shine | hi |
| 15:06.57 | kaldemar | wasanzy: asterisk is sending re-invites that have 127.0.0.1 as address for "emma". set directmedia=no for it in sip.conf. |
| 15:07.40 | p3nguin | I told him to do that a day or two ago. |
| 15:07.52 | wasanzy | under general? |
| 15:08.04 | p3nguin | Is that where we've both indicated to put it? |
| 15:08.10 | kaldemar | wasanzy: under [emma] |
| 15:08.21 | p3nguin | I specifically told you to put it in the phone's entry. |
| 15:08.56 | wasanzy | <p3nguin>: sorry if I didn't get you right. I might skipped that point am doing it now |
| 15:09.43 | p3nguin | When you come here asking for help, and people take their time to offer that help by making suggestions or recommendations, you should really follow those suggestions. |
| 15:09.57 | p3nguin | the first time. |
| 15:10.02 | p3nguin | without having to be told again. |
| 15:10.09 | wasanzy | ok thx |
| 15:10.59 | wasanzy | that is done, trying to call again n see |
| 15:16.26 | *** join/#asterisk sekil (~sekil@78.24.111.246) |
| 15:17.26 | *** join/#asterisk cerberus_za (~coert@196-215-58-254.dynamic.isadsl.co.za) |
| 15:18.43 | wasanzy | I did that, and now the other party can hear me a little and even with that, my voice take time to reach him and there is a break in it. |
| 15:19.51 | *** join/#asterisk sekil (~sekil@78.24.111.246) |
| 15:19.58 | *** join/#asterisk timeshell (~chatzilla@gw.lusi.on.ca) |
| 15:21.09 | wasanzy | my sip debug: http://pastebin.com/m7pxHiRD |
| 15:21.13 | *** join/#asterisk mboylan (~mboylan@66.206.176.224) |
| 15:22.05 | *** join/#asterisk luckman212 (~irc@2001:470:1f07:1225:8048:3812:9d88:a161) |
| 15:23.24 | mboylan | hi guys... we're in the middle of a project trying to upgrade the uni's phone system from 1.2 to 1.8. We have a distributed asterisk system with multiple routing boxes and multiple "endpoint" server where phones are registgered. We're starting with the backup routing box. We're noticing an issue where the calls will pass fine from 1.2 --> 1.8 --> 1.2 but only for the duration of the ring timeout. The "answer" is never passed back to the first 1.2 box. Th |
| 15:23.24 | mboylan | box releases the channel once the two 1.2 boxes are connected. But do you have any ideas why the answer wouldn't be passed back? |
| 15:23.45 | ChannelZ | wasanzy: sounds more like a sketchy network connection |
| 15:25.11 | wasanzy | what does that mean? |
| 15:25.33 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 15:25.52 | ChannelZ | your internet connection sucks |
| 15:26.20 | wasanzy | the two machines are in the same lan |
| 15:26.45 | *** join/#asterisk sekil (~sekil@78.24.111.246) |
| 15:26.47 | ChannelZ | Hmm. Bigger problem then. |
| 15:26.58 | wasanzy | how? |
| 15:27.15 | ChannelZ | though you're using GSM which always sounds like crap |
| 15:27.50 | wasanzy | I want to use a third machine but is not in the same network will that work? |
| 15:27.55 | ChannelZ | Did you compile Asterisk yourself? |
| 15:28.01 | wasanzy | yes |
| 15:28.07 | ChannelZ | What version of gcc? |
| 15:28.42 | wasanzy | 4.5.2 |
| 15:29.28 | ChannelZ | hmm. |
| 15:29.41 | *** join/#asterisk cyborg-one (1000@85-238-120-239.broadband.tenet.odessa.ua) |
| 15:29.49 | *** join/#asterisk jeffik (~chatzilla@TOROON63-1176243424.sdsl.bell.ca) |
| 15:30.35 | wasanzy | but one thing, why I want to use third machine and as advice by kaldemarn, one of the phone sit on the same machine with asterisk |
| 15:31.06 | *** join/#asterisk mateu (~mateu@missoula.org) |
| 15:31.43 | ChannelZ | worth a try |
| 15:31.46 | fulcan | mboylan sip or iax? |
| 15:32.18 | mboylan | fulcan: IAX between boxes. SIP to the phones. We tried doing this all SIP, but there's major reinvite differences between 1.2 and 1.8 that seemed to just not allow it to work |
| 15:32.18 | ChannelZ | nothing in your SIP debug looks 'wrong' so not sure what you're experiencing |
| 15:32.39 | fulcan | mboylan can you 'telnet 5060' (for sip) clean from box to box bidirectional? |
| 15:32.39 | Nugget | telnet is eeeeeeevil! |
| 15:32.40 | wasanzy | fulcan: are u asking me? |
| 15:32.42 | *** join/#asterisk gxdssoft (~gxdssoft@190.233.208.212) |
| 15:33.19 | mboylan | fulcan: The problem isn't sip related... it's iax |
| 15:33.24 | fulcan | mboylan why are you 'reinvite'ing? |
| 15:33.47 | fulcan | Nugget most people are on winblows so I default to suggestion test that way. |
| 15:33.50 | mboylan | fulcan: Why do we? To allow the asterisk boxes to drop out of the middle of the calls. |
| 15:34.30 | mboylan | regardless we're not using SIP trunks for these... they're iax |
| 15:34.33 | ChannelZ | SIP is UDP 99% of the time anyway, telnet isn't going to get you anywhere |
| 15:34.50 | fulcan | mboylan my understanding of reinvite = no is to drop the rtp stream. |
| 15:35.04 | fulcan | ChannelZ it will let you know if you got firewall issues. |
| 15:35.07 | ChannelZ | and it's "directmedia" now |
| 15:35.42 | mboylan | Right, it is. But it doesn't play well between versions... we've gone through this a bunch, even with Digium |
| 15:35.46 | mboylan | anywho |
| 15:36.33 | mboylan | The firewall was a nonissue as these were all on non-firwalled networks. We've since moved them to a firewalled network, yes. But the ports are allowed. |
| 15:37.06 | mboylan | the originating box just says " -- Format for call is ulaw |
| 15:37.06 | mboylan | <PROTECTED> |
| 15:37.07 | kaldemar | mboylan: search for calltoken and iax. the protocol had a change since 1.2. |
| 15:37.13 | ChannelZ | if you can call both ways it's not firewall, it's probably just something goofy with the versions |
| 15:37.25 | mboylan | kaldemar: Yeah, we had to turn off calltoken support too |
| 15:37.33 | mboylan | trying to mix these 1.2 w/ 1.8 is a pain in the ass, clearly |
| 15:37.37 | ChannelZ | I think my ITSP is running ancient chineese secret 1.2, and I am having DTMF issues |
| 15:37.39 | mboylan | but we can't just do the whole system overnight |
| 15:38.10 | mboylan | on the middle box you see |
| 15:38.12 | mboylan | -- IAX2/asterisk5-18420 is ringing |
| 15:38.12 | mboylan | <PROTECTED> |
| 15:38.12 | mboylan | <PROTECTED> |
| 15:38.27 | mboylan | but that answer status never makes it back to asterisk2, the originating box |
| 15:38.57 | wasanzy | kaldemar: are u there? |
| 15:39.27 | p3nguin | fulcan: Disallowing a reinvite (by setting canreinvite or directmedia to "no") keeps asterisk in the media stream. If you allow reinvites, asterisk will get out of the way and the phones can try to talk directly. |
| 15:39.54 | mboylan | right |
| 15:41.09 | mboylan | p3nguin: Any thoughts? This sets the whole project back again. We were going w/ SIP trunks first but then that failed miserably. Engineers from digium basically said it probably won't work. Trying IAX now and now we see this :( |
| 15:41.36 | mboylan | if I have to tell my boss it's all or nothing he's not going to be pleased I don't think |
| 15:41.55 | fulcan | p3nguin thank you for clearing that up. I was always a little unsure which direction it went. looks like I was backwards. :) |
| 15:42.04 | wasanzy | <p3nguin>: I am now using two machine which will connect to asterisk on other machine, but the other third machine is in a different network so am getting 503 service unavailable error in twinkle |
| 15:42.53 | wasanzy | how can I other machines in a different network but in the same office, to be able to connect to the asterisk server? |
| 15:43.32 | p3nguin | Do you have NAT between the two networks? If it's a WAN, you may just have a firewall problem. |
| 15:43.37 | fulcan | wasanzy with a router. |
| 15:43.41 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
| 15:44.37 | wasanzy | hmmm, I realized that machine is even connecting to a different ISP |
| 15:44.44 | wasanzy | hmmm |
| 15:45.08 | fulcan | wasanzy are you refering to the "routable www"? |
| 15:45.46 | wasanzy | no, I mean a different company providing internet for the third machine |
| 15:46.29 | fulcan | wasanzy different isp's not different networks. cause if you are on the www, all networks are connected by default. that's why it is called the www. |
| 15:46.44 | fulcan | wasanzy you may have a firewall issue. |
| 15:47.12 | p3nguin | If it's not on the same subnet, which it apparently isn't, then it's on a different network. |
| 15:47.36 | fulcan | p3nguin doesn't mean anything on the www. |
| 15:47.44 | p3nguin | Yes it does. |
| 15:48.26 | wasanzy | the thing is I can't even ping asterisk machine from the other machine |
| 15:48.27 | fulcan | p3nguin subnetmask means nothing on the www except to addressing. |
| 15:49.10 | p3nguin | My point was that you tried to tell him that if he has a second ISP it's not on a different network, but it actually is. |
| 15:49.29 | p3nguin | Because only something on the same subnet is on the same network. |
| 15:49.48 | fulcan | wasanzy can the host machine ping 4.2.2.2 and can the target machine do the same? |
| 15:50.11 | wasanzy | let me try |
| 15:50.52 | fulcan | www = the world routable network. all networks are routable to each other. |
| 15:51.34 | wasanzy | yes they can ping 4.2.2.2 |
| 15:51.56 | wasanzy | let me tell u about the network here |
| 15:51.57 | fulcan | now, can any machine on the web ping target or host |
| 15:51.59 | fulcan | ? |
| 15:52.06 | p3nguin | Just because they are routABLE does not mean his two devices within separate networks WILL communicate with each other correctly. |
| 15:52.09 | wasanzy | no |
| 15:52.42 | fulcan | p3nguin if you are ON the www, you are connect by default. |
| 15:52.44 | wasanzy | we have two routers here |
| 15:52.53 | p3nguin | fulcan: That would be false. |
| 15:53.21 | p3nguin | You're ON the www and I am ON the www, but you can't connect to my computer by default. |
| 15:53.32 | p3nguin | NAT prevents it. |
| 15:53.52 | fulcan | p3nguin I have configured WAY too many cisco and juniper routers for the www. |
| 15:53.58 | p3nguin | oooooooooooooooooooooooooooooohhhhhhhh |
| 15:54.04 | p3nguin | I'm impressed. |
| 15:54.21 | fulcan | p3nguin you are either on the routable www, or you are not. |
| 15:54.39 | wasanzy | fulcan: I can only be reach from the internet if am on the public network |
| 15:54.59 | fulcan | wasanzy yup, thats how it works. |
| 15:55.13 | wasanzy | u can do that bcos u do routing on your public IP in your router |
| 15:55.22 | wasanzy | but we don't have it that way |
| 15:55.35 | wasanzy | we are doing intranet sot of thing |
| 15:55.40 | fulcan | wasanzy what is the ip of your server? |
| 15:55.56 | wasanzy | 192.168.1.155 |
| 15:55.56 | fulcan | wasanzy so then you are using NAT |
| 15:56.07 | fulcan | forward the port. |
| 15:56.39 | p3nguin | If I ping 192.168.1.155, I get replies!!! I guess I really AM connected to your computer by the world routable network (www)! |
| 15:56.42 | p3nguin | lol |
| 15:57.11 | wasanzy | wow |
| 15:57.12 | fulcan | p3nguin 192.x is not routable |
| 15:57.21 | wasanzy | that is strange |
| 15:57.30 | p3nguin | chortles |
| 15:57.41 | wasanzy | hmmm |
| 15:58.18 | wasanzy | am lost now |
| 15:59.07 | fulcan | wasanzy if your server has a 192. address, it's in a LAN and your router is using NAT to translate the routable info from the www to your LAN segment. you have to forward your port to the server from your router. |
| 16:00.06 | wasanzy | ok |
| 16:00.36 | wasanzy | so what ports do I hv to forward? |
| 16:00.49 | fulcan | wasanzy are you using iax or sip? |
| 16:01.01 | wasanzy | sip |
| 16:01.30 | p3nguin | UDP 5060 and whatever UDP range is in rtp.conf |
| 16:01.38 | p3nguin | usually UDP 10000-20000 |
| 16:01.49 | wasanzy | ok |
| 16:02.12 | wasanzy | ok let me do that and get back to you guys |
| 16:02.20 | wasanzy | thank you so much for the help |
| 16:02.59 | wasanzy | but one other question, will thins affect the sound even if we are in the same network? |
| 16:03.03 | fulcan | forward port 5060 from your router ip (both udp and tcp) to your server. also forward udp 10,000 thru 20,000 (as a gross over kill) to the server. That huge range should be closed later after it works to only a few ports but open her up wide for testing an provisioning. |
| 16:03.55 | fulcan | wasanzy what do you mean by 'effect sound'? |
| 16:04.32 | p3nguin | The question was "will [this] affect the sound?" |
| 16:04.32 | *** join/#asterisk jsemar (~joel@office.appiction.com) |
| 16:04.43 | wasanzy | I mean when the two parties can not hear each other despite you can hear the phone ring in the same network |
| 16:04.55 | fulcan | because I have a recording server in London that accepts calls from the US and then trows it BACK to the US and sounds crystal clear after traveling the world. |
| 16:05.13 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-uvmigzkclkwazbxc) |
| 16:05.26 | fulcan | wasanzy that would be udp 10000-20000 issue usually. |
| 16:06.06 | wasanzy | I hv this 10001 - 20000 |
| 16:06.47 | p3nguin | If your rtp.conf has 10000-20000 listed, don't forward only 10001-20000 at the NAT. |
| 16:06.54 | p3nguin | You need to use the same port range. |
| 16:07.44 | fulcan | wasanzy you might be getting screwed by that 1 port number different AND is it udp forward or tcp? I have seen that mistake. |
| 16:07.47 | wasanzy | I have this 10001-20000 in the rtp.conf |
| 16:08.31 | fulcan | AND do you have both udp and tcp forwarded for port 5060? |
| 16:08.53 | p3nguin | I doubt his asterisk is listening on TCP, so that's not relevant. |
| 16:09.20 | wasanzy | I haven't done the port forwarding yet |
| 16:09.39 | wasanzy | am seeking permision from my boss first |
| 16:10.16 | wasanzy | is listening on udp not tcp |
| 16:10.38 | fulcan | p3nguin back in the early days, 5060 was tcp only. I still forward it out of habit and there are a number of phones that require it. |
| 16:11.06 | p3nguin | There are phones that use SIP over TCP, but that doesn't make his asterisk listen on TCP. |
| 16:11.22 | p3nguin | And if Asterisk isn't using TCP for SIP, it's worthless to forward it. |
| 16:12.49 | *** join/#asterisk radic (~radic@tmo-097-114.customers.d1-online.com) |
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| 16:25.58 | *** join/#asterisk sekil (~sekil@78.24.111.246) |
| 16:27.25 | *** join/#asterisk okei (bc81c390@gateway/web/freenode/ip.188.129.195.144) |
| 16:27.54 | okei | hello guys. how to see my running extensions.conf? i have many same config but i dont know which is true config |
| 16:28.02 | p3nguin | dialplan show |
| 16:28.03 | okei | can i debug this ? |
| 16:28.13 | okei | yes i try this but nothing |
| 16:28.27 | p3nguin | If it shows nothing, then you have no extensions configured. |
| 16:28.56 | okei | p3nguin: no nothing but no extensions, can i filter only extension string? |
| 16:29.07 | p3nguin | Check your asterisk.conf for the path. THen look in that path for extensions.conf. |
| 16:29.37 | p3nguin | "dialplan show" will show you ALL of the extensions that are configured and loaded. |
| 16:30.26 | p3nguin | If you want to select only one extension to show, use dialplan show <extension>@<context>. |
| 16:30.45 | p3nguin | If you want to see all the extensions in a context, use dialplan show <context>. |
| 16:31.08 | okei | ah okay 10x |
| 16:31.20 | p3nguin | 43z to you! |
| 16:31.47 | p3nguin | one zero ex to you! |
| 16:32.05 | p3nguin | ten x to you! |
| 16:32.11 | okei | : ) |
| 16:36.54 | *** join/#asterisk zorp75ck (~zorp75ck@146.186.115.44) |
| 16:38.03 | *** part/#asterisk fulcan (~root@li345-191.members.linode.com) |
| 16:42.36 | *** join/#asterisk sekil (~sekil@78.24.111.246) |
| 16:47.04 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 16:48.02 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 16:48.46 | *** join/#asterisk tyman (~tyler@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 16:49.08 | okei | p3nguin: can't find where is running extensions.conf file |
| 16:49.11 | okei | : / |
| 16:49.45 | p3nguin | What do you see when you run dialplan show in the asterisk cli? |
| 16:49.47 | *** join/#asterisk irroot (~irroot@197.171.135.196) |
| 16:51.09 | okei | p3nguin: http://pastebin.com/5HESid6y |
| 16:51.40 | p3nguin | There are your extensions. |
| 16:51.44 | p3nguin | Right there on the screen. |
| 16:51.46 | p3nguin | I see 'em. |
| 16:51.49 | p3nguin | You can't fool me! |
| 16:52.08 | okei | wtf? |
| 16:52.21 | okei | p3nguin: where is exts.conf? |
| 16:52.21 | p3nguin | 28 extensions in 22 contexts! |
| 16:52.29 | okei | yes but where is |
| 16:52.33 | okei | exts.conf |
| 16:52.33 | okei | mda! |
| 16:52.39 | p3nguin | typically, /etc/asterisk/extensions.conf |
| 16:52.45 | okei | this is not true conf |
| 16:52.53 | p3nguin | I'd say it is. |
| 16:52.53 | okei | i deleting this config but |
| 16:52.59 | okei | output is same |
| 16:53.01 | okei | : | |
| 16:53.10 | p3nguin | You changed it, saved, it and forgot to run "dialplan reload" afterward. |
| 16:53.28 | navaismo | amm I think you are using ael |
| 16:53.37 | okei | no |
| 16:53.41 | okei | i using default extension |
| 16:53.58 | p3nguin | There is no default. |
| 16:54.08 | okei | i know : / |
| 16:54.17 | p3nguin | If you have things "by default," you used the sample files. |
| 16:54.22 | p3nguin | And that was a mistake. |
| 16:54.57 | p3nguin | navaismo mentioned that you use ael. Check extensions.ael instead. |
| 16:55.37 | tyman | from the dialplan, how do I check the availability of a phone and, only if unavailable, redirect to another phone.? When the primary phone IS online, dont want the 2nd phone to ring. Is this a use queues scenario only? |
| 16:56.43 | p3nguin | You'll need to Dial the first phone, then check the DIALSTATUS of it. Then Goto the other phone if the first was UNAVAILABLE. |
| 16:57.54 | okei | p3nguin: fixd it 10x again : d |
| 16:57.58 | p3nguin | It would be something like GotoIf($[ "${DIALSTATUS}" = "UNAVAILABLE" ]?unavailablelabel) |
| 16:58.08 | p3nguin | okei: 19b to you! |
| 16:58.14 | tyman | ok perf |
| 16:58.41 | p3nguin | okei: 36d to you! |
| 16:58.55 | p3nguin | 36 DD to you! |
| 17:01.26 | ChannelZ | motorboats |
| 17:02.34 | *** join/#asterisk drynish (~drynish@modemcable039.7-200-24.mc.videotron.ca) |
| 17:02.37 | drynish | I need help :) |
| 17:02.55 | ChannelZ | Me too. I haven't pooped for days. |
| 17:03.04 | drynish | I have a pretty precarious setup right now... I didn't realized it |
| 17:03.08 | drynish | oh new version available |
| 17:03.35 | drynish | let me try the 1.8.5 |
| 17:03.36 | drynish | instead |
| 17:03.41 | p3nguin | lactulose |
| 17:04.03 | drynish | I doubt it will be good |
| 17:04.10 | drynish | I'm on svn and it's not working |
| 17:04.27 | drynish | I have no sound... my asterisk seems to do only invite |
| 17:04.46 | ChannelZ | was this a previously working system or..? |
| 17:04.51 | drynish | yes |
| 17:05.03 | drynish | but I had to upgrade since I added a fxo card |
| 17:05.14 | drynish | and I had to make dahdi working |
| 17:06.54 | drynish | so the only way to be able tocompile it with my kernel was to use the latest svn fonction |
| 17:06.56 | drynish | oups svn version |
| 17:07.09 | *** join/#asterisk fulcan (~root@li345-191.members.linode.com) |
| 17:07.12 | drynish | and when dahdi works (was compiled) the only way to be detected was using svn asterisk |
| 17:07.20 | drynish | but now my system is not working at all :( no sound |
| 17:08.35 | wasanzy | <p3nguin>: now, though I haven't done the port forwarding yet, but two machines can call each other on the same network |
| 17:09.33 | wasanzy | just that, one party could hear the other one, but one could not hear other other person |
| 17:10.19 | *** join/#asterisk TimeRider (~steve@host-92-27-131-175.static.as13285.net) |
| 17:11.06 | ChannelZ | drynish: no sound from your FXO or SIP channels? |
| 17:11.07 | fulcan | wasanzy sounds like the correct behavior for 'ports not being forwarded'. |
| 17:11.27 | drynish | sip |
| 17:11.31 | wasanzy | but they are on the same network |
| 17:11.46 | fulcan | wasanzy same LAN? |
| 17:11.46 | p3nguin | Right, because ports need forwared for clients on the same network. Pfft. |
| 17:11.50 | wasanzy | yes |
| 17:12.07 | wasanzy | same LAN |
| 17:12.11 | ChannelZ | wasanzy: so? An invidual machine might still be running a firewall.. Windows for instance |
| 17:12.11 | drynish | 1 day ago everything was working |
| 17:12.20 | drynish | same lan sure |
| 17:12.24 | fulcan | wasanzy like, pcs sitting next to each other with nothing inbetween them? |
| 17:12.47 | fulcan | ChannelZ very true. |
| 17:12.52 | ChannelZ | drynish: sounds like you broke something else, I'd rebuild a fresh DAHDI from sources, configure/build a fresh asterisk from sources |
| 17:12.54 | wasanzy | YES |
| 17:13.01 | drynish | oh sorry i was thinking you were talking to me |
| 17:13.53 | drynish | i think i will go with debian sid |
| 17:14.26 | fulcan | wasanzy one way audio is 99.997% of the time a port forwarding issue. .03% of the time (which is yet to happen to me personally) it is something else. Check your ports. |
| 17:14.29 | drynish | my setup for compiling is really bizarrre |
| 17:14.43 | ChannelZ | why |
| 17:14.49 | wasanzy | am using ubuntu |
| 17:14.51 | wasanzy | ok |
| 17:15.07 | wasanzy | let me just do the port forwarding and see |
| 17:15.13 | p3nguin | There is nothing to forward. |
| 17:15.20 | p3nguin | You have two computers on the same LAN. |
| 17:15.30 | p3nguin | Forwarding is not a part of that topology. |
| 17:16.05 | p3nguin | If you're running a firewall on either of the computers where the phones are, TURN THAT SHIT OFF. |
| 17:16.07 | McBoingbo | p3nguin: wassaaap, got my Asterisk 1.8 installed now, gonna try to couple it with the old server |
| 17:16.12 | fulcan | wasanzy keep in mind what ChannelZ said, if you are doing a test call from a PC and you are experiencing 1 way audio, then that machine has a firewall on it and the ports need to be forwarded there too! |
| 17:16.15 | ChannelZ | wasanzy: "your end" asks "the other end" what IP and port they should send their audio to you. "the other end" needs sufficient network access to get that traffic out to "your end". You must have sufficient network access to receive that traffic. |
| 17:16.31 | p3nguin | You don't forward ports on a machine. |
| 17:16.43 | p3nguin | You can stop blocking them. |
| 17:16.45 | ChannelZ | The reverse is also true. With that in mind you need to verify your network is allowing all this traffic... whether by firewalls, routers... |
| 17:16.47 | p3nguin | But forwarding sends them somewhere else. |
| 17:17.08 | p3nguin | And since it's a LAN that you are on, forwarding is not a part of the scenario. |
| 17:18.11 | fulcan | p3nguin a firewall is a firewall and most use NAT. NAT translates eathier one port at a time or all. but it's the exact same thing. |
| 17:18.38 | p3nguin | For a single machine? That's not accurate at all. |
| 17:18.41 | *** join/#asterisk vikapi (~quassel@124.125.34.134) |
| 17:18.42 | *** join/#asterisk batfastad (~benbradle@87-194-37-131.bethere.co.uk) |
| 17:18.44 | p3nguin | I would have expected someone that has configured WAY too many Cisco and Juniper routers on the world routable network (the www) to know that. |
| 17:18.58 | fulcan | p3nguin nat is nat. |
| 17:19.36 | p3nguin | NAT masquerades between on address space and another address space. A single machine does not fall into that category. |
| 17:19.44 | p3nguin | s/on/one/ |
| 17:19.49 | fulcan | p3nguin no such thing a NAT for pc vs. NAT for a router. it's still nat |
| 17:20.06 | p3nguin | ANd you don't have NAT on a single host machine that isn't a gateway. |
| 17:20.27 | atheos | unless your pc is a router (i.e. pfsense), it's not doing NAT |
| 17:20.30 | p3nguin | Why, you ask? There's nothing to masquerade. |
| 17:20.36 | p3nguin | atheos: exactly |
| 17:20.42 | *** part/#asterisk batfastad (~benbradle@87-194-37-131.bethere.co.uk) |
| 17:20.51 | McBoingbo | We have a wiener!! |
| 17:20.53 | McBoingbo | ding ding! |
| 17:20.58 | p3nguin | It's a single host. It can have a firewall, but it's not NAT. |
| 17:21.08 | p3nguin | No NAT, no ports to forward. |
| 17:21.17 | fulcan | p3nguin almost every firewall in the world is based on either iptable or ipforward BOTH use NATing. |
| 17:21.18 | p3nguin | Firewalls just block ports. |
| 17:21.22 | p3nguin | fulcan: You're wrong. |
| 17:21.31 | wasanzy | does ubuntu comes with a default firewall? |
| 17:21.49 | p3nguin | On a single host, you use the INPUT and OUTPUT chain. Neither involve NAT> |
| 17:21.54 | p3nguin | s/>/./ |
| 17:22.04 | p3nguin | wasanzy: Yes, it does. |
| 17:22.12 | wasanzy | which is? |
| 17:22.12 | p3nguin | wasanzy: but not using NAT, because it's not a router. |
| 17:22.13 | McBoingbo | infobot stay out of this! ;) |
| 17:22.37 | p3nguin | wasanzy: "iptables -L -nv" will show you if the firewall is doing anything. |
| 17:22.40 | wasanzy | I want to stop the firewall on if there is |
| 17:22.44 | fulcan | p3nguin thats what NATing is, a translation from INPUT to OUTPUT or FORWARD. |
| 17:22.52 | p3nguin | fulcan: Wrong again. |
| 17:23.17 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 17:23.18 | p3nguin | "iptables -L INPUT -nv" will show you if your firewall in blocking things going into that host. |
| 17:23.35 | ChannelZ | not even the FORWARD table is NAT |
| 17:23.47 | p3nguin | s/table/chain/ |
| 17:24.00 | wasanzy | ok |
| 17:24.00 | p3nguin | FUCK YOU infobot |
| 17:24.06 | McBoingbo | I know right? |
| 17:24.13 | ChannelZ | /ignore |
| 17:24.17 | p3nguin | "iptables -L OUTPUT -nv" will show you if your firewall in blocking things going out of that host which originated on that host. |
| 17:24.44 | p3nguin | I was trying to correct channelz's use of table there, and infobot thought I was trying to correct myself. |
| 17:24.56 | p3nguin | fulcan: If you aren't using -t nat, it's not NAT. Period. |
| 17:24.58 | McBoingbo | well then stop using that form lol |
| 17:25.04 | McBoingbo | you know you are just teasing the infobot |
| 17:25.17 | p3nguin | and neither INPUT, OUTPUT, nor FORWARD are in the nat table. They are not NAT. |
| 17:25.46 | p3nguin | If you want to NAT, you'll use the nat table and you'll do PREROUTING and POSTROUTING. That creates a NAT in iptables. |
| 17:26.09 | p3nguin | Again, someone who knows so God damned much about the world routable network (the www) should know this. |
| 17:26.19 | McBoingbo | calm down tiger |
| 17:26.30 | McBoingbo | quick gimme a Rib steak |
| 17:27.14 | wasanzy | Chain INPUT (policy ACCEPT 505 packets, 110K bytes) |
| 17:27.15 | wasanzy | <PROTECTED> |
| 17:27.16 | ChannelZ | and a bedpan |
| 17:27.31 | wasanzy | oh sorry 4 that, I should have pastebin |
| 17:27.44 | p3nguin | two lines, it's okay. |
| 17:27.53 | wasanzy | the port is not blocking any thing on the asterisk server |
| 17:27.53 | ChannelZ | no blockage there, and if OUTPUT looks similar. But what about the other machine? |
| 17:28.01 | fulcan | wasanzy drop your firewall, run a test call, listen to it work and then forward your ports and watch it work again. |
| 17:28.05 | p3nguin | wasanzy: If there is no other rule listed there, nothing is being blocked going into that host. |
| 17:28.25 | p3nguin | Since the default policy is ACCEPT, and there are no rules, everything is allowed in. |
| 17:28.40 | wasanzy | ok |
| 17:28.54 | *** join/#asterisk timahvo1 (~rogue@41.223.57.73) |
| 17:29.26 | fulcan | wasanzy 'iptables stop' is what you need. |
| 17:29.45 | p3nguin | On the other hand, if you ARE actually doing NAT, which you aren't by default, the nat table and PREROUTING chain can "take over" ports and forward them somewhere else. If you are concerned that something configured NAT erroneously, you can check with "iptables -t nat -L -nv" to see all the NAT chains. |
| 17:29.49 | wasanzy | ok |
| 17:29.50 | *** join/#asterisk diatonic (~diatonic_@mail.clearwater-research.com) |
| 17:30.29 | p3nguin | But again, there is no NAT for a single machine, so unless you configured it for a gateway, nothing is supposed to be in the nat table. |
| 17:30.56 | p3nguin | Did you check the OUTPUT chain like channelz said? |
| 17:32.24 | ChannelZ | and I keep not hearing about the _other_ machine. Or is that not involved again? I can't keep up. |
| 17:32.34 | p3nguin | The OUTPUT chain allows or blocks traffic from going out of the machine. |
| 17:32.51 | ChannelZ | I don't think it's been mentioned which way the one-way audio even is... |
| 17:32.58 | p3nguin | I think he still has only two hosts: one with asterisk and a soft phone, and one with a soft phone. |
| 17:33.15 | fulcan | p3nguin what if he has two firewalls up and doesn't know it OR another device in the chain that he has not accounted for? |
| 17:34.04 | p3nguin | Your suggestion of stopping iptables (flushing all chains) would solve it. That's what he SHOULD do most of the time. Firewalls just get in the way. |
| 17:34.08 | ChannelZ | right.. and the other with a softphone, what OS is it, what's the status if its firewall... |
| 17:34.57 | p3nguin | But if he verifies all the chains are set to ACCEPT and there are no rules, effectively it would be the same as shutting down the service. |
| 17:35.45 | p3nguin | service |
| 17:35.47 | p3nguin | since that's what stopping the services does, anyway. |
| 17:36.03 | fulcan | wasanzy unplug any and all firwalls, kill them dead. then use telnet to 'step' your way to and from the server. Telnet is a great tool because if it kills your attempt to access a port immediately, then you have a 'firewall' somewhere blocking it. if it just hangs, then that is a protocol conflict which is a good thing because comm CAN get through. But a quick connection kill means you were dropped. |
| 17:36.32 | p3nguin | You'll need some TCP daemon on the computer to be able to telnet to it. |
| 17:36.49 | p3nguin | Since SIP is UDP and he hasn't configured asterisk to use TCP, he can't telnet 5060. |
| 17:37.03 | fulcan | apt-get install telnet |
| 17:37.26 | p3nguin | If that particular telnet supports UDP, make sure you use the appropriate option to use UDP. |
| 17:37.34 | p3nguin | Otherwise, use netcat and its UDP option. |
| 17:37.49 | fulcan | p3nguin it doesn't have to suppot udp at all. |
| 17:38.11 | ChannelZ | it's a false test that proves little |
| 17:38.11 | p3nguin | telnet does not connect to UDP. |
| 17:38.13 | fulcan | all he is looking for is either a 'quick drop' or a hang. |
| 17:38.28 | p3nguin | And since SIP is UDP, you can't telnet it. |
| 17:38.54 | p3nguin | Until he configures his asterisk to use TCP for SIP, his SIP is just UDP. |
| 17:39.42 | fulcan | p3nguin it will 'hang' on a protocol conflict. He does care about whether asterisk speaks the lang at all, he is trying to diagnose whether something is inbetween it. |
| 17:40.06 | ChannelZ | What are you even talking about |
| 17:40.17 | p3nguin | If asterisk is listening on UDP 5060, and you telnet to 5060 (which is TCP), what do you THINK is going to happen? |
| 17:40.19 | fulcan | p3nguin quit putting the poor guy on another mission. |
| 17:40.38 | p3nguin | I'm not putting him on any mission. I don't want his asterisk to listen on TCP. |
| 17:41.12 | *** join/#asterisk nightrid3r (nightrid3r@91.176.218.162) |
| 17:41.19 | fulcan | p3nguin asterisk isn't even confirmed the problem at all @ this point. |
| 17:41.27 | p3nguin | exactly |
| 17:41.27 | ChannelZ | Telnet is a worthless tool in this scenario. |
| 17:41.33 | p3nguin | it sure is! |
| 17:41.56 | p3nguin | But you can't tell fulcan that. He knows way too much about configuring cisco and juniper routers on the world routable network (the www). |
| 17:42.10 | fulcan | ChannelZ damn, all those times I was wrong, I just guessed that telnet was telling me something. wow, my imagination. |
| 17:42.54 | p3nguin | Now telnet could be used for testing connectivity if the host has a service listening on TCP, or if your telnet supports UDP and you use the UDP telnet option. |
| 17:43.00 | ChannelZ | Telnet can tell you things. Here, no so much. We already know there is traffic moving around partially. |
| 17:43.21 | fulcan | you use telnet to test for drops. |
| 17:43.33 | ChannelZ | p3nguin: I think his assertion is that if a firewall is set to 'DROP' all traffic, telnet is going to sit there waiting for an ACK it will never get. |
| 17:43.59 | p3nguin | While that may happen, it wouldn't be definitive for testing Asterisk. |
| 17:44.02 | fulcan | those drops come from firewalls and closed ports. everything else is trivials when you are looking for WHAT is blocking your path. |
| 17:44.06 | ChannelZ | But it only tells 1% of the story |
| 17:44.21 | fulcan | ChannelZ that all he needs at this point |
| 17:44.25 | p3nguin | It assumes you have a rule that drops TCP and does something silly to UDP. |
| 17:44.27 | fulcan | ChannelZ not another mission |
| 17:44.30 | ChannelZ | No, we're past that. |
| 17:44.48 | ChannelZ | The call is being setup, how do you think that's happening? |
| 17:44.55 | p3nguin | I block all sorts of TCP and allow UDP. You telnet to me, you'll see dropped packets, but if you use an appropriate UDP test, you'll succeed. |
| 17:45.04 | fulcan | ChannelZ he's got a one way audio issue. something is blocking. |
| 17:45.10 | p3nguin | That's UDP. |
| 17:45.13 | p3nguin | RTP is UDP. |
| 17:45.14 | ChannelZ | Yes and telnet isn't going to help him figure out what! |
| 17:45.17 | p3nguin | Telnet won't test that. |
| 17:45.38 | fulcan | ChannelZ you DON't need a connection. I am trying to tell you this. |
| 17:45.45 | p3nguin | The SIP part works -- there is traffic flow between hosts. |
| 17:45.48 | ChannelZ | I give up |
| 17:45.58 | p3nguin | I'll try. |
| 17:46.20 | p3nguin | channelz: http://xkcd.com/386/ |
| 17:46.23 | ChannelZ | It's like saying I know definitiely my car is 100% functional because my garage door works. |
| 17:46.33 | ChannelZ | They're simply unrelated |
| 17:46.45 | p3nguin | or because the window rolls up and down. |
| 17:47.10 | ChannelZ | s/definitiely/definitively/ |
| 17:47.15 | fulcan | ChannelZ doing it right now to my asterisk box. telnet 10000 to asterisk give me a hung black screen. |
| 17:47.19 | ChannelZ | or whatever the heck it was I typed |
| 17:47.43 | ChannelZ | fulcan: Congradulations, you proved you're blocking one TCP port. |
| 17:47.49 | p3nguin | lol |
| 17:47.53 | fulcan | ChannelZ telnet is 'half assed' connected to udp port 10000 using tcp |
| 17:48.02 | p3nguin | I doubt that. |
| 17:48.27 | fulcan | ChannelZ no, I just proved I have access to port 10000 and nothing is in the way. |
| 17:48.43 | fulcan | p3nguin I'll send you a screen shot. |
| 17:48.45 | ChannelZ | just shrugs |
| 17:48.56 | *** join/#asterisk freakazoid0223 (~matt@pool-173-49-209-91.phlapa.fios.verizon.net) |
| 17:49.10 | p3nguin | Your screenshot will only show me what you think you've seen. |
| 17:49.19 | p3nguin | The fact is that if you telnet to 5060, it's TCP. |
| 17:49.23 | fulcan | p3nguin your a moron |
| 17:49.33 | p3nguin | points and laughs |
| 17:50.12 | p3nguin | Is now the appropriate time to point out that it's "you're a moron" or should I wait? |
| 17:50.34 | ChannelZ | I'd say misguided and leave it at that |
| 17:50.48 | fulcan | p3nguin try it yourself. i used winblows telnet 178.79.176.191 |
| 17:51.01 | fulcan | p3nguin telnet to 10000 |
| 17:51.06 | p3nguin | telnet asterisk.local 5060 |
| 17:51.07 | p3nguin | Trying 192.168.192.242... |
| 17:51.07 | p3nguin | telnet: Unable to connect to remote host: Connection refused |
| 17:51.18 | p3nguin | $ telnet asterisk.local 10000 |
| 17:51.18 | p3nguin | Trying 192.168.192.242... |
| 17:51.18 | p3nguin | telnet: Unable to connect to remote host: Connection refused |
| 17:51.23 | p3nguin | You know why it's refused? |
| 17:51.31 | p3nguin | 'CAUSE IT'S NOT TCP! |
| 17:51.33 | ChannelZ | because no means no! |
| 17:51.37 | fulcan | p3nguin cause your not going to the right port! |
| 17:51.51 | ChannelZ | Ohhh, all this time! Which one is the right one? |
| 17:51.58 | fulcan | p3nguin telnet pot IS closed, not 10000 |
| 17:51.58 | p3nguin | Right, because my asterisk.local doesn't listen on 5060 and 10000. Got it. |
| 17:52.17 | p3nguin | TCP 10000 is also closed. |
| 17:52.20 | p3nguin | I'll show you. |
| 17:52.41 | fulcan | p3nguin congrat, you can read. no shit, but udp is wide open |
| 17:52.47 | p3nguin | [root@cpe-e650 ~]# lsof -i tcp:10000 |
| 17:52.47 | p3nguin | [root@cpe-e650 ~]# lsof -i tcp:5060 |
| 17:52.47 | p3nguin | [root@cpe-e650 ~]# |
| 17:52.57 | p3nguin | Nothing listening; they are closed. |
| 17:53.16 | ChannelZ | please tell me the ice cream shop is still open though |
| 17:53.21 | p3nguin | telnet doesn't know about udp, and I just showed you. |
| 17:53.47 | p3nguin | telnet asterisk.local 80 |
| 17:53.49 | p3nguin | Trying 192.168.192.242... |
| 17:53.51 | p3nguin | Connected to asterisk.local. |
| 17:53.53 | p3nguin | Escape character is '^]'. |
| 17:53.57 | p3nguin | It sure does know about my TCP daemon, though. |
| 17:53.58 | fulcan | winblows does, and telnetd on linux does to |
| 17:53.59 | p3nguin | imagine that. |
| 17:54.06 | p3nguin | I'm not using telnetd. |
| 17:54.11 | p3nguin | We're talking about asterisk. |
| 17:54.22 | ChannelZ | WHo the heck even runs a telnet daemon anymore? |
| 17:54.27 | p3nguin | only fools |
| 17:54.38 | p3nguin | As nugget once said, telnet is evil. |
| 17:54.53 | p3nguin | Oh, wait, my cable modem runs telnet. |
| 17:55.15 | ChannelZ | Thus your cable modem is a fool. |
| 17:55.21 | p3nguin | ISP, rather |
| 17:55.35 | atheos | my cisco router runs telnet. damn cisco fool! |
| 17:56.02 | p3nguin | Most people using Cisco these days don't allow telnet to them and use ssh. |
| 17:56.21 | fulcan | telnet 178.79.176.191 10000 produces blank screen telnet 178.79.176.191 22 hangs, telnet 178.79.176.191 25 'connection refused" because the port is blocked. |
| 17:56.26 | p3nguin | Maybe on a LAN it's still common, but not from the public side. |
| 17:57.03 | ChannelZ | fulcan: and what do you have running on port 22 and 10000? |
| 17:57.04 | atheos | my cisco hardware is completely locked from the public, but it's still using telnet. |
| 17:57.06 | fulcan | p3nguin I am very much aware of this, its a testing tool that finds firewall screwups in a hurry because of it's bevavior. |
| 17:57.12 | p3nguin | 25 has a reject because telnetd is not running. |
| 17:57.19 | ChannelZ | 25 is smtpd |
| 17:57.23 | p3nguin | fuck |
| 17:57.32 | p3nguin | 25 has a reject because smtpd is not running. |
| 17:57.53 | fulcan | p3nguin I block all ports. especially 25 |
| 17:58.01 | p3nguin | hence the reject |
| 17:58.32 | fulcan | p3nguin yup telnet told me that one is blocked. did it's job. |
| 17:58.40 | p3nguin | its |
| 17:58.47 | p3nguin | its job |
| 17:58.48 | p3nguin | not it is job. |
| 17:59.09 | fulcan | p3nguin you enjoys C0Rrectig speliong? |
| 17:59.23 | fulcan | p3nguin Is dis beter? |
| 17:59.29 | ChannelZ | MOAR BETAR! |
| 17:59.36 | ssureshot | ok, finally back to my long distance issue, |
| 18:01.01 | ChannelZ | is that all? |
| 18:01.15 | fulcan | ChannelZ I hope so. |
| 18:02.16 | ChannelZ | I meant ssureshot. The comma gave me hope, but the following silence crushed it |
| 18:02.47 | fulcan | ChannelZ :) |
| 18:02.48 | p3nguin | hahaha |
| 18:02.51 | p3nguin | That's great! |
| 18:04.41 | ssureshot | ChannelZ: lol sorry,, I can't figure out why I can make local and long distance calls but I can't call any toll free numbers |
| 18:05.13 | Psi|4ward | i would like to route SIP-Trunk->Asterisk#1->IAX2-Trunk->Asterisk#2 but the called-incomming numbers is dropped so Asterisk#2 dosnt know what extension to ring - any ideas? |
| 18:05.15 | *** join/#asterisk cyborg-one (1000@85-238-109-111.broadband.tenet.odessa.ua) |
| 18:05.29 | *** join/#asterisk wesphillips (~wphill04@adsl-75-53-138-133.dsl.hstntx.sbcglobal.net) |
| 18:06.42 | ssureshot | I'd kill for any input on why It doen'st work lol |
| 18:07.28 | ssureshot | or even just a tip |
| 18:07.41 | ssureshot | concerning the issue not just any tip |
| 18:07.54 | _Corey_ | ssureshot: Could be a few things... make sure you're presenting caller-id |
| 18:08.04 | *** part/#asterisk fulcan (~root@li345-191.members.linode.com) |
| 18:08.16 | *** join/#asterisk wesphillips (~wphill04@adsl-75-53-138-133.dsl.hstntx.sbcglobal.net) |
| 18:08.25 | _Corey_ | make sure you're not using some "toll free" inbound trunk or something that's not capable of completing toll-free calls |
| 18:08.25 | *** part/#asterisk wesphillips (~wphill04@adsl-75-53-138-133.dsl.hstntx.sbcglobal.net) |
| 18:09.27 | wasanzy | am done with the port forwarding so going to try now |
| 18:10.03 | ssureshot | _Corey_: I have added num and name to the callerid,,, and the second item,,, Am I wrong in thinking since I can make long distance calls I should be able to make tollfree? |
| 18:10.17 | McBoingbo | p3nguin: any docs you can point me to that would be relevant to my Asterisk 1.2 talking to 1.8.5 setup? |
| 18:10.44 | *** join/#asterisk timahvo1 (~rogue@41.223.57.76) |
| 18:11.14 | wasanzy | p3nguin: should I remove the directmedia=no in the sip.conf, since the soft phones are no more on the same machine with asterisk? |
| 18:11.22 | ssureshot | This line is supposed to be setup just like my primary T1 that works ,, it's a secondary line incase of failure,, I'm just upgrading asterisk |
| 18:11.26 | _Corey_ | ssureshot: Maybe, I see it a lot when customers who have dedicated toll-free inbound circuits. Their carrier doesn't allow toll-free outbound |
| 18:11.29 | McBoingbo | checking http://www.voip-info.org/wiki/view/Asterisk+Connect+2+servers now |
| 18:13.03 | ssureshot | _Corey_: ah it's not dedicated toll free,, it should be fully functional,, this line doesn't have a tollfree incomming number just did's |
| 18:13.06 | _Corey_ | ssureshot: When you say "line" ... what kind of line? |
| 18:14.12 | ssureshot | _Corey_:Full T1 for voice,,, |
| 18:14.41 | _Corey_ | Call your carrier... they should know why they're blocking your calls :) |
| 18:15.42 | ssureshot | lol I just did but I didn't use that phrase,, maybe Ill play dumb this time... or well I guess it wouldn't be playing dumb haha |
| 18:17.02 | _Corey_ | I've had carriers reject calls because of some obscurity before, best to call them before wasting your time |
| 18:19.14 | wasanzy | the port forwarding is done but still, we can't hear each other. is only one partner that can hear |
| 18:27.33 | *** join/#asterisk douglas_carmicha (~dcarmich@209-242-50-10.rev.dls.net) |
| 18:27.44 | McBoingbo | there seems to be many ways to peer 2 Asterisk servers, but because 1 server is 1.8.5 and the main production server is 1.2.12, I need to be a little more careful, can anyone help get some communication going in between them? |
| 18:28.10 | douglas_carmicha | is there much of a functionality difference between chan_h323 and chan_ooh323? I'm trying to resolve a port conflict on FreeBSD between asterisk and ekiga, and the libraries used by ekiga (ptlib) conflict with what is used by chan_h323 (pwlib.) |
| 18:28.36 | wasanzy | pls some body help me out |
| 18:28.55 | *** join/#asterisk fulcan (~root@li345-191.members.linode.com) |
| 18:29.36 | wasanzy | fulcan: welcome back |
| 18:29.58 | fulcan | wasanzy hey my friend. how the old testing going? |
| 18:30.51 | wasanzy | I did the port forwarding in the router, but still only one person can hear his partner, and the other person can't hear him |
| 18:31.48 | fulcan | wasanzy are you on a winblows box right now or nix? |
| 18:31.48 | wasanzy | my latest sip debug is: http://pastebin.com/kQQFhWfh |
| 18:31.58 | wasanzy | ubuntu |
| 18:32.16 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 18:32.29 | wasanzy | both asterisk server and the two softphone machine are on ubuntu |
| 18:33.20 | fulcan | wasanzy I am actually asking about the box you are typing on right now. |
| 18:33.30 | wasanzy | ubuntu |
| 18:33.42 | wasanzy | is the same machine running asterisk |
| 18:34.16 | fulcan | wasanzy kk, is there a telnet client on it like utelnetd or just telnet? |
| 18:34.31 | wasanzy | telnet |
| 18:35.19 | fulcan | wasanzy do you have a second box you can connect too? |
| 18:35.53 | wasanzy | like connect from that machine to the asterisk box? |
| 18:36.17 | fulcan | wasanzy a laptop or or something on the side? hopefully on the same network? |
| 18:36.47 | wasanzy | the three machines are actually in the same network |
| 18:38.35 | wasanzy | I have this under the user who can not hear his partner: directmedia=no but one who can hear his partner doesn't have that |
| 18:38.41 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 18:39.17 | wasanzy | do I have to add it for all the two? |
| 18:39.32 | fulcan | telnet to all the machine on port 5060 and 10000 , you are interested in the behavior of 'how it doesn't work'. if it 'refuses' then you are block because of firewall or service is turned off. if it hangs, or 'delays' before it drops, then you are golden. can you check this from the different machines? |
| 18:41.01 | wasanzy | should I try it from the two machine having the phone to the asterisk server? |
| 18:41.19 | russellb | telnet is TCP though, and the traffic you're concerned with is UDP |
| 18:41.39 | fulcan | you want to do 'server -> client and then client -> server |
| 18:42.17 | fulcan | russellb it doesn't matter, we already had this discussion earlier with someone else. |
| 18:43.08 | russellb | unless you forwarded both, then it does matter, but ok, *goes back to work* |
| 18:43.25 | *** join/#asterisk dmz (~dmz@64.203.235.49.dyn-cm-pool-34.pool.hargray.net) |
| 18:43.55 | wasanzy | check pout nmap result first: http://pastebin.com/AKKYQSyh |
| 18:44.30 | fulcan | wasanzy not nmap, telnet! |
| 18:44.39 | wasanzy | ok |
| 18:45.20 | wasanzy | connection refuse on port 10000 from server to client 1 |
| 18:45.20 | Qwell | fulcan: umm, then you were wrong in the previous discussion too. |
| 18:46.03 | fulcan | Qwell no i am not. |
| 18:47.23 | wasanzy | connection refuse again to the other client as well |
| 18:47.34 | Qwell | Don't use telnet to test UDP. It will not work. Ever. |
| 18:48.22 | wasanzy | so what should I use? |
| 18:48.35 | fulcan | wasanzy telnet to port 10000 |
| 18:48.37 | Qwell | if you're trying to test Asterisk - test Asterisk. |
| 18:48.56 | Qwell | fulcan: Stop helping people with networking stuff. You have absolutely no idea what you are talking about. |
| 18:49.19 | fulcan | Qwell I am doing it right now to my own server. |
| 18:49.30 | Qwell | You are simply wrong. |
| 18:49.45 | fulcan | Qwell my commandline must be different from yours. |
| 18:49.59 | Qwell | Telnet does not use UDP. Period. |
| 18:50.07 | fulcan | Qwell try it! telnet 178.79.176.191 10000 |
| 18:50.24 | Qwell | netstat -plan | grep 10000 |
| 18:50.38 | Qwell | I will wager $500 that you are listening on TCP. |
| 18:50.42 | fulcan | Qwell your using the wrong damn command! |
| 18:50.57 | wasanzy | ok guys, pls will this affect some thing when place in the si: directmedia=no |
| 18:51.03 | fulcan | Qwell its REALLY easy to do! telnet 178.79.176.191 10000 |
| 18:51.12 | Qwell | go away |
| 18:51.23 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 18:52.00 | wasanzy | and if it is port issue, why will one hear other but the ohter can't hear him back? |
| 18:52.12 | *** join/#asterisk Nasga (~Nasga@AAmiens-157-1-106-45.w86-208.abo.wanadoo.fr) |
| 18:52.15 | Qwell | wasanzy: That would be a NAT issue. |
| 18:52.22 | Qwell | ~nat |
| 18:52.22 | infobot | it has been said that nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 18:52.57 | fulcan | wasanzy because your blocked somewhere and you don't even know whom/what is blocking you. telnet is an easy tool to use the will tell you whom, not what. |
| 18:53.19 | *** mode/#asterisk [+q fulcan!*@*] by Qwell |
| 18:53.23 | Qwell | go. away. |
| 18:54.43 | _Corey_ | uses telnet to make coffee |
| 18:54.44 | wasanzy | then the other person also shouldn't hear his partner if is a blocking issue |
| 18:55.04 | Qwell | wasanzy: It is a NAT issue, like I said. That is called one-way audio. It's incredibly common. |
| 18:55.05 | *** join/#asterisk joker2u (~root@li345-191.members.linode.com) |
| 18:55.15 | wasanzy | <Qwell> I did port forwarding already |
| 18:55.22 | *** mode/#asterisk [+q *!*@li345-191.members.linode.com] by Qwell |
| 18:55.27 | Qwell | wasanzy: It's more than port forwarding. |
| 18:55.34 | wasanzy | oh ok |
| 18:55.36 | Qwell | You need to configure Asterisk properly.. |
| 18:55.37 | Qwell | ~nat |
| 18:55.37 | infobot | somebody said nat was Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 18:56.20 | *** mode/#asterisk [-q fulcan!*@*] by Qwell |
| 18:57.24 | *** mode/#asterisk [-q *!*@li345-191.members.linode.com] by Qwell |
| 18:58.03 | jaytee | interesting, I use externip and localnet but I hadn't seen localmask nor do I have it set. |
| 18:58.30 | wasanzy | Qwell: so how do I solve the NAT issuse so that they can both hear each other? |
| 18:58.46 | Qwell | jaytee: yeah I don't know what localmask is |
| 18:58.50 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 18:59.47 | *** join/#asterisk fulcan (~root@li345-191.members.linode.com) |
| 19:00.28 | wasanzy | Qwell: are there? |
| 19:00.43 | Qwell | wasanzy: Look at the settings the bot gave you. Set those properly. |
| 19:01.19 | wasanzy | which settings? |
| 19:01.26 | wasanzy | externip? |
| 19:02.27 | *** join/#asterisk ChUbB (~IceChat7@cpc2-aztw12-0-0-cust229.aztw.cable.virginmedia.com) |
| 19:02.30 | SuPrSluG | yes. ie. your public ip. |
| 19:02.43 | wasanzy | hmm |
| 19:02.45 | SuPrSluG | localnet= your lan |
| 19:03.21 | wasanzy | in sip_nat.conf? |
| 19:05.30 | SuPrSluG | from the bot message -> "Usable in Asterisk sip.conf " |
| 19:06.17 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 19:06.54 | wasanzy | ok, under general? |
| 19:07.17 | SuPrSluG | yes |
| 19:07.38 | wasanzy | ok |
| 19:07.42 | wasanzy | am doing that |
| 19:08.17 | *** join/#asterisk radic (~radic@tmo-096-170.customers.d1-online.com) |
| 19:08.35 | wasanzy | localnet=192.168.1.155 right? |
| 19:10.11 | wasanzy | externip=mypublic ip right? |
| 19:10.57 | wasanzy | <SuPrSluG>: are you there? |
| 19:11.19 | SuPrSluG | yes |
| 19:11.34 | wasanzy | are my setting right now? |
| 19:12.47 | wasanzy | http://pastebin.com/4VLDgpUv my sip.conf |
| 19:15.31 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 19:15.39 | wasanzy | pls check and tell me if is ok |
| 19:16.26 | wasanzy | <SuPrSluG>: hmmm |
| 19:18.10 | SuPrSluG | localnet is the network address not the ip of the server. 192.168.1.0 localmask 255.255.255.0 in cidr 192.168.1.0/24 |
| 19:18.41 | wasanzy | ok |
| 19:18.58 | SuPrSluG | is is the NETWORK addresss |
| 19:19.14 | wasanzy | yes |
| 19:20.14 | wasanzy | and don't you think this directmedia=no will afftect some thing? |
| 19:20.17 | SuPrSluG | you can make network as large or small as you want using the subnet mask |
| 19:20.20 | *** join/#asterisk pabelanger (~pabelange@nat/digium/x-qolszrxirtnddczt) |
| 19:20.21 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 19:20.59 | wasanzy | I have this now: localnet=192.168.1.0/24 |
| 19:24.10 | fulcan | wasanzy did you turn the firewall off on your client yet? |
| 19:24.59 | wasanzy | they don't have firewall running |
| 19:26.12 | okei | guys, how to restart asterisk daemon from |
| 19:26.14 | okei | asterisk cli? |
| 19:26.22 | okei | stop now not working * 1.8 |
| 19:26.43 | Qwell | core restart now |
| 19:27.03 | wasanzy | ok |
| 19:27.24 | okei | 10x |
| 19:27.47 | fulcan | wasanzy what is the client software you are using? |
| 19:28.07 | wasanzy | twinkle |
| 19:28.31 | *** join/#asterisk sulex (~sulex@pdpc/supporter/professional/sulex) |
| 19:30.35 | *** join/#asterisk bn-7bc (~bjarne-im@pdpc/supporter/active/bn-7bc) |
| 19:31.13 | wasanzy | the same network |
| 19:33.20 | wasanzy | <SuPrSluG> are u there? |
| 19:34.33 | *** join/#asterisk moy (~moy@173.239.155.74) |
| 19:37.26 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 19:42.54 | Psi|4ward | can anyone help me with a dail-rule, i want forward all incomming calls to an iax2 trunk connected to a second asterisk |
| 19:44.28 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 19:45.30 | *** join/#asterisk ketema (~ketema@kjhmacpro.ketema.net) |
| 19:45.33 | SuPrSluG | yep |
| 19:45.53 | wasanzy | <SuPrSluG> are you back? |
| 19:47.40 | SuPrSluG | Psi|4ward, IAX2/<number>@ip:port |
| 19:48.07 | Psi|4ward | dont really know how to start right now |
| 19:48.37 | Psi|4ward | iave a FreePBX box and tried to set up incomming rule with destination "iax trunk" but this strips the called number |
| 19:52.12 | *** join/#asterisk tomaw_ (tom@freenode/staff/tomaw) |
| 19:54.41 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 19:56.41 | SuPrSluG | Psi|4ward, possibly use a misc destination and dial like the example i gave |
| 19:57.43 | *** join/#asterisk ffs (~garland@unaffiliated/ffs) |
| 20:00.02 | *** part/#asterisk irroot (~irroot@197.171.135.196) |
| 20:03.31 | Psi|4ward | hmm wont work, im to stupid right know |
| 20:06.50 | *** join/#asterisk vikapi (~quassel@124.125.34.134) |
| 20:13.41 | *** join/#asterisk coppice (~chatzilla@116.92.37.56) |
| 20:17.27 | Psi|4ward | created an custom extension and added a dial-rule to extensions_custom.conf but now i get "number not in service" |
| 20:18.11 | Psi|4ward | exten => ._,1,Dial(IAX2, .... |
| 20:22.48 | *** join/#asterisk godmachine-x6 (~godmachin@h214.179.90.75.dynamic.ip.windstream.net) |
| 20:24.07 | leifmadsen | psilikon: uhhhh.... ._ is wrong |
| 20:24.13 | leifmadsen | underscore first, not last |
| 20:24.30 | leifmadsen | Psi|4ward: ^^^^ |
| 20:24.37 | leifmadsen | psilikon: sorry, wrong tab completion :) |
| 20:24.40 | psilikon | heheh |
| 20:25.06 | Psi|4ward | dosent matter, hightlight is set to psi ;) |
| 20:26.57 | *** part/#asterisk fulcan (~root@li345-191.members.linode.com) |
| 20:27.14 | Psi|4ward | okay, the Dial command works now, but on asterisk#2 ill dont see the called number |
| 20:27.46 | Psi|4ward | is there any variable for the diald number? |
| 20:28.14 | Psi|4ward | Dial(..../${DID}) isnt it |
| 20:29.24 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 20:30.16 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
| 20:30.27 | *** join/#asterisk nightrid3r (~nightrid3@91.176.119.54) |
| 20:31.49 | SuPrSluG | it would be an inbound route |
| 20:32.26 | SuPrSluG | match what you're sending from server a |
| 20:32.43 | thegoat | is there a variable for the number that was dialed? |
| 20:33.41 | thegoat | psil|ward looks like we want the same thing ;-) |
| 20:33.49 | Psi|4ward | exactly |
| 20:36.06 | leifmadsen | ${EXTEN} ? |
| 20:37.08 | Psi|4ward | should there not be a debug-message like "Set("myvar","value") ? |
| 20:37.25 | serafie | also https://wiki.asterisk.org/wiki/display/AST/Asterisk+standard+channel+variables |
| 20:38.01 | serafie | Psi|4ward: https://wiki.asterisk.org/wiki/display/AST/Dialplan+Applications look at Verbose() and Noop() |
| 20:38.57 | Psi|4ward | hmm ok but this works only in these agi scripts dosnt it? |
| 20:39.09 | serafie | Dialplan |
| 20:39.14 | Psi|4ward | whoa asterisk is really hard stuff |
| 20:39.20 | serafie | It says Dialplan applications right there in the title. :) |
| 20:39.23 | leifmadsen | learning curve can be high |
| 20:39.27 | leifmadsen | that's why there is a good |
| 20:39.28 | leifmadsen | book* |
| 20:40.45 | Qwell | good ;) |
| 20:41.10 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 20:42.59 | Psi|4ward | thegoat: look at ${CALLERID(dnid)} |
| 20:44.04 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 20:46.49 | Psi|4ward | whoa HOST xxx failed to authenticate with user bb_user :\ |
| 20:46.53 | Psi|4ward | on the remote machine |
| 20:47.47 | Psi|4ward | my dial-rule looks like exten => _.,1,Dial(IAX2/benutzer:passwort@12.12.12.12:6000/${CALLERID(dnid)}) |
| 20:51.49 | Psi|4ward | but the trunk peer-name works ;) |
| 20:54.26 | *** join/#asterisk jonmasters (~jcm@edison.jonmasters.org) |
| 20:54.52 | thegoat | has anyone done any integration with google voice |
| 20:55.32 | *** join/#asterisk Bob_Pierce (~WESTMAN\p@216.36.132.162) |
| 20:56.32 | thegoat | i need asterisk to find my google voice number as the number dialed when a call comes in, not the number google voice is calling |
| 20:58.39 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 20:59.38 | *** join/#asterisk rgagnon (~rgagnon@rrcs-71-42-183-54.sw.biz.rr.com) |
| 21:01.16 | rgagnon | Question for anyone regarding cdr.conf and "endbeforehexten"..... The values for CDR(start), CDR(answer), CDR(end), CDR(duration), CDR(disposition), and CDR(billsec) don't seem to be available during the "h" extension, yet they appear correct in the Master.csv CDR File.... any thoughts? |
| 21:02.30 | rgagnon | "start" and "end" come up, but they are both equal to eachother, and equal to the end of the call only |
| 21:03.13 | rgagnon | and "disposition" shows "NO ANSWER" although the Master.csv shows "ANSWER", and the call completed with asterisk logs showing the channel answered |
| 21:09.13 | *** part/#asterisk bobb_WU (~bobb_WU@206.74.211.64) |
| 21:13.25 | *** join/#asterisk file (~file@neutrino-114-86.joshua-colp.com) |
| 21:19.53 | ChannelZ | thegoat: do you mean the extension? Doesn't it come in with no exten and thus goes to 's'? |
| 21:20.19 | *** join/#asterisk gravin (~gravin@217.71.50.60.brf01-home.tm.net.my) |
| 21:22.28 | thegoat | what happens is that when someone calls my gv number the dnid is the number that gv dials and not my google voice number), so say my gv number is 8005551212 and my did is 8005551213, the dnid variable shows 8005551213 and not 8005551212 |
| 21:26.32 | Qwell | thegoat: If it's going through Google Voice, and Google doesn't send that, I'm not sure how you'd change it. |
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| 21:32.09 | timeshell | Is OSLEC still the best echo canceller to use? |
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| 21:53.49 | Linuturk | I recently updated to 1.8.5.0 and now my pri stops receiving incoming calls after a bit |
| 21:54.00 | Linuturk | dahdi restart corrects the issue for a bit |
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| 21:54.15 | Linuturk | but, it isn't stable like it used to be with 1.8.2.4 |
| 21:54.39 | Micc | I need help with a lock problem on 1.6.2.19, I've got the core show locks when it happens, is anyone qualified to take a look at it? |
| 21:55.51 | Micc | http://pastebin.com/gU1LCH5n |
| 22:09.13 | *** part/#asterisk Bob_Pierce (~WESTMAN\p@216.36.132.162) |
| 22:10.10 | Psi|4ward | whoa if i hungup a forwarded call from asterisk#1 to #asterisk#2, asterisk#1 calls again :\ |
| 22:10.18 | Psi|4ward | can i hangup on asterisk#1 too? |
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| 22:18.29 | ShaunR | Anybody recommend a good SIP to phone adapter |
| 22:18.38 | ShaunR | just for home use, nothing special |
| 22:18.56 | ShaunR | was looking at the PAP2T-NA |
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| 22:25.08 | *** part/#asterisk ShaunR (~shaun@freenode/sponsor/NDChost.com) |
| 22:33.03 | timeshell | That should be suitable |
| 22:41.40 | Micc | Is 1.8 more stable than 1.6? |
| 22:42.04 | Micc | I know some versions might be less stable than others, but in general is 1.8 more stable because its a LTS release? |
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| 22:59.45 | timeshell | Is there anyway to use a flash stream for MOH? |
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