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| 09:24.25 | funky1 | hi all, still sturgling with sip registration, could anyone have a look and see what's going wrong, i really got no clue anymore, the whole problem is described here http://forums.digium.com/viewtopic.php?f=1&t=84744&p=196373&sid=fb669914fb5b1db8d288cffe6b7ab82b#p196373 |
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| 10:00.45 | boratynskikamil | Hello. I see my ISDN card with dahdi_hardware as plus and generated dahdi_conf with dahdi_genconf, after that I connected to Asterisk. dahdi show status presents my card but I don't see any spans and channels. Suggestions? |
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| 10:08.57 | kaldemar | boratynskikamil: have you included the file generated by dahdi_genconf in chan_dahdi.conf? have you restarted asterisk? |
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| 10:13.02 | boratynskikamil | kaldemar: I copied dahdi_genconf file to chan_dahdi.conf? is it better to include it in some way? |
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| 10:23.59 | kaldemar | boratynskikamil: no difference really. |
| 10:26.22 | boratynskikamil | kaldemar: So I generated typed: dahdi_genconf |
| 10:26.52 | boratynskikamil | Added #include dahdi-channels to chan_dahdi.conf [first line]. |
| 10:26.58 | boratynskikamil | And restarted asterisk at all. |
| 10:27.39 | kaldemar | first line is the wrong place. what dahdi_genconf generates belongs under [channels] |
| 10:28.09 | kaldemar | pastebin your whole chan_dahdi.conf |
| 10:29.25 | boratynskikamil | kaldemar: Ok, works. |
| 10:29.30 | boratynskikamil | I added #include after channels. |
| 10:29.45 | boratynskikamil | Thank you for help. :-) |
| 10:29.58 | kaldemar | np |
| 10:30.05 | boratynskikamil | kaldemar: Have you ever had any experience with OpenVox G400E card? GSM card. |
| 10:30.12 | kaldemar | no |
| 10:42.53 | boratynskikamil | kaldemar: And one more question? How did you define SIP passwords? plaintext? |
| 10:52.39 | kaldemar | boratynskikamil: depends. for peers you can use plaintext with the secret parameter or MD5 of user:realm:secret with the md5secret paramter. |
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| 11:08.26 | wdoekes | funky1: your example shows two different usernames. you did try the same username for the pbx as the twinkle one, right? |
| 11:09.15 | wizbit | on my dialplan i have created a simple menu, however, the user cannot press a number on the keyboard until the whole message has finished, is there a way to make key presses active even when the menu sound is being played? |
| 11:09.17 | wdoekes | other than that, I see a lack of Allow: headers in the asterisk register. that should not be a problem, but it might make a difference. |
| 11:09.39 | wdoekes | try a tool like sipp(1) with a register scenario, and play around with that |
| 11:10.30 | wdoekes | https://code.osso.nl/projects/sipp/browser/scenario/register.xml -- sipp -m 1 -sf register.xml -s username -ap password host |
| 11:10.47 | kaldemar | wizbit: app Background can be used like that. |
| 11:11.02 | mirela666 | wizbit: or app Read |
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| 11:12.06 | kaldemar | mirela666: how? |
| 11:13.31 | mirela666 | Read is playnig sound file untill reads given amount of DTMFs |
| 11:14.23 | mirela666 | Read(variable[,filename][,maxdigits][,option][,attempts][,timeout]) |
| 11:15.50 | wizbit | maybe this is the problem: exten => s,7,agi(googletts.agi,"Press 2, to leave a message.",en) |
| 11:17.02 | mirela666 | wizbit: take a look inside the .agi to see what is used to read DTMF |
| 11:17.08 | kaldemar | the documentation on app read is a bit misleading. |
| 11:17.18 | wizbit | ok |
| 11:17.23 | mirela666 | kaldemar: how? |
| 11:17.45 | kaldemar | mirela666: filename = "file(s) to play before reading digits or tone with option i" |
| 11:17.56 | kaldemar | it says "before". |
| 11:18.14 | kaldemar | but seems it does read during playback too nowadays. |
| 11:18.22 | mirela666 | hmmm |
| 11:18.32 | wizbit | http://forums.digium.com/viewtopic.php?p=182124 |
| 11:18.33 | wizbit | ace |
| 11:18.35 | mirela666 | I does, I used it for 3 years now |
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| 11:20.23 | wizbit | i could just convert what google plays into sound files and use Background() |
| 11:26.43 | funky1 | wdoekes: indeed i had tried same username as you thought as well, will look into and try the tool you mentioned |
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| 11:35.07 | mirela666 | wizbit: after a quick look at the script there is the explanation |
| 11:35.13 | mirela666 | wizbit: "If 'intkey' is set the |
| 11:35.14 | mirela666 | # script will wait for user input. Any given interrupt keys will cause the playback |
| 11:35.14 | mirela666 | " |
| 11:35.25 | mirela666 | damn sorry for multy rows |
| 11:35.28 | boratynskikamil | kaldemar: You mean, password, yes? Any reference to MD5 passwords? |
| 11:35.52 | boratynskikamil | The case is. I have to create 5 SIP and in general, would like to do it quite safe... |
| 11:38.06 | kaldemar | boratynskikamil: http://svn.asterisk.org/svn/asterisk/trunk/README-SERIOUSLY.bestpractices.txt |
| 11:38.10 | wizbit | mirela666: you clever sod |
| 11:38.19 | mirela666 | :P |
| 11:41.26 | funky1 | wdoekes: got sipp installed and using the command you gave me, but what am i actually looking at or do now? |
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| 11:44.39 | boratynskikamil | kaldemar: Thank you. |
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| 12:11.56 | wdoekes | funky1: did you run the command I gave you? |
| 12:12.00 | wdoekes | sipp -m 1 -sf register.xml -s username -ap password host |
| 12:12.59 | wdoekes | does it register? if not, you would alter the xml until it does (make it more similar to the twinkle register) |
| 12:14.05 | wdoekes | if sipp complains about bad input, you might need to add an extra linefeed after the <!DOCTYPE |
| 12:14.40 | wdoekes | (xml parser bug in older versions) |
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| 13:11.37 | funky1 | wdoekes: yes i tried it and i didn't register, will play with it and report back later, thanks ! |
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| 14:18.36 | sereal | Does anyone know why i'm getting "Reason: could not create SSL context: SSL error code" for cdr -> postgres issues? |
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| 14:19.59 | sereal | I can so psql from the asterisk box -> postgres fine so it's not silly connectivity issue |
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| 14:22.59 | sereal | I did at one point have things working fine, but I recently compiled a newer release of asterisk and *might* not of had the right library or set a config option. |
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| 14:37.15 | pii3 | anyone here using asterisk with softphones ? |
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| 14:40.30 | [TK]D-Fender | tons of people |
| 14:41.45 | TSM2 | no, none at all, not even sure people use asterisk any longer :0 |
| 14:41.46 | TSM2 | :p |
| 14:42.44 | sereal | Everyones upgrading to shouting these days. |
| 14:42.51 | TSM2 | WHAT |
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| 15:58.14 | asilva | Hello, I would like some help please, I have 2 asterisk server running asterisk 11.7.0 configured with DUNDi and IAX2 to dial between them, when I dial from one server to another I get a "NO AUTHORITY FOUND" and the call is rejected, using the same config under Asterisk version 1.8.25 works 100%, never had that problem before. more information from outputs and configuration on pastebin - http://pastebin.com/8AMsHw6T |
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| 16:09.12 | asilva | anyone ? |
| 16:09.54 | pabelanger | asilva, Did you read UPGRADE.txt and CHANGES? |
| 16:10.06 | pabelanger | there is likely a configuration change some place related to chan_iax2 |
| 16:10.19 | asilva | that's what I'm thinking but I cant seem to find it |
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| 16:11.11 | asilva | pabelanger: there is no information about upgrades on iax.conf or parameters on UPGRADE files |
| 16:12.07 | asilva | the only changes was from before 1.8 that the calltokens were added |
| 16:12.16 | asilva | and I sustain the confs from 1.8 to 11 |
| 16:13.27 | pabelanger | then you have a username / password mismatch |
| 16:13.48 | asilva | I use dbsecret from dundi keys, never used username and password |
| 16:14.07 | asilva | since the "account" is generated by DUNDi |
| 16:14.15 | asilva | as i mentioned 1.8 works perfectly! |
| 16:15.34 | asilva | based on sample configuration and the book(which has only SIP accounts for DUNDI as example) haven't done nothing wrong. |
| 16:19.13 | boom^time | Hello, I was wondering if there is a known reliability issue with AMI events. It seems like on rare occasions I'll miss a Hangup event leaving my application in a state believing a call is in progress. I've been trying to reproduce it while listening with tcpdump but of course it doesn't want to happen again. |
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| 17:18.46 | boom^time | danjenkins, what other events do you mean? I know that there are the channel creation events, bridging/etc, but I ignore those. |
| 17:18.46 | danjenkins | ah boom^time you tried in here first :D |
| 17:19.15 | danjenkins | yeah, so you're receiving a load of events that you just dont care about, all those events are tying up your ami connection |
| 17:19.20 | boom^time | haha yeah, I had a feeling the guys in dev would be more familiar with a bug in this |
| 17:20.03 | danjenkins | so boom^time - take a look at http://hungrygeek.holidayextras.co.uk/2012/05/14/elastix-apply-configuration-changes-problem/ |
| 17:20.17 | danjenkins | it'll give you more information on eventfilter in ami.conf |
| 17:20.26 | boom^time | Okay, currently I'm only using read=cc,call |
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| 17:21.06 | danjenkins | ah ok, that's good and that should be fine |
| 17:21.15 | danjenkins | but if you know exactly what events you care about |
| 17:21.27 | danjenkins | then I'd recommend adding eventfilters to your ami users |
| 17:21.29 | boom^time | I see, you can strip it right down |
| 17:21.35 | danjenkins | so you only ever get the ones you care about |
| 17:21.43 | danjenkins | then, if youre still having an issue |
| 17:21.57 | danjenkins | you can go back to digium and say, oi! |
| 17:22.10 | boom^time | That's my main problem, reproducing the issue. I wish I could reliably. |
| 17:22.20 | danjenkins | yeh :) i'd try out eventfilter |
| 17:22.31 | boom^time | I will, thank you for the advice |
| 17:22.54 | danjenkins | if that still doesnt help then maybe ask in asterisk-dev how you could try and record all the vents coming out from asterisk, and then record what your application receives |
| 17:23.10 | danjenkins | hope ive helped a little :) |
| 17:23.26 | boom^time | I've been trying to with tcpdump all morning, just can't get it to happen again. frustrating |
| 17:23.29 | boom^time | you have thank you. |
| 17:24.08 | danjenkins | no problem! |
| 17:38.40 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 17:39.20 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 17:43.00 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 17:46.48 | Kobaz | ooo yay |
| 17:46.52 | Kobaz | deadlock in 1.8.25 |
| 17:47.20 | *** join/#asterisk weinerk (~user@unaffiliated/weinerk) |
| 17:48.31 | *** part/#asterisk weinerk (~user@unaffiliated/weinerk) |
| 17:51.20 | drmessano | Impossible |
| 17:51.31 | ChannelZ-Wk | You're holding it wrong. |
| 17:51.53 | drmessano | All those were removed and replaced with quantum buffers |
| 17:51.58 | drmessano | Locking is so 2000s |
| 17:52.20 | coppice | do they teach locking at Yale? |
| 17:52.21 | *** join/#asterisk Zogot (~Leon@26.sub224.ddfr.nl) |
| 17:53.07 | Kobaz | heh |
| 17:53.15 | Kobaz | no one cares in -dev :( |
| 17:53.22 | asilva | pabelanger: So I fixed my problem. It was a permission issue on the astdb file(which is there the dundi account reads its secret for authentication) |
| 17:53.42 | Kobaz | code |
| 17:53.42 | Kobaz | http://pastebin.com/93bTrE38 not a true deadlock-deadlock in terms of locking |
| 17:53.56 | Kobaz | but something was locked up to the point that no new sip calls were starting |
| 18:00.27 | drmessano | Kobaz, I went through about 6 months of that when I had Asterisk on a VPS. I discovered that running Asterisk on a reduced footprint was fine for 20 concurrent calls, but getting slammed with a SIP attack from china over a 1G pipe would KILL the SIP stack in a second |
| 18:00.41 | drmessano | Everything else was fine.. But it was unrecoverable |
| 18:00.47 | drmessano | With a restart |
| 18:00.50 | drmessano | Err |
| 18:00.55 | drmessano | WITHOUT a restart |
| 18:01.00 | *** part/#asterisk LiuYan (~LiuYan@222.125.134.157) |
| 18:01.10 | Kobaz | hmm |
| 18:01.26 | TSM2 | restart of machine or restart of asterisk? |
| 18:01.36 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 18:01.43 | Kobaz | well yeah, sip calls from a 1g pipe would probably kill most boxes |
| 18:01.52 | *** join/#asterisk jpoz (~jpoz@158.sub-70-210-131.myvzw.com) |
| 18:01.54 | Kobaz | like, if it was maxed out |
| 18:01.55 | drmessano | Restarting Asterisk |
| 18:02.10 | drmessano | The machine was fine. Asterisk was even responsive. I could hit the CLI, check anything I wanted |
| 18:02.10 | TSM2 | put a rate limit on SIP packets per second |
| 18:02.12 | Kobaz | my personal build of 1.8.12 is rock solid |
| 18:02.16 | drmessano | Except SIP |
| 18:02.32 | drmessano | TSM2, did that in iptables, after I moved to a new provider that supported it |
| 18:02.36 | Kobaz | this is my first foray on a real server with 1.8.25 and the sip stack died within 24 hours |
| 18:03.13 | drmessano | fail2ban was useless (when is it not) because it wouldn't log anything before dying lol |
| 18:03.19 | *** join/#asterisk jpoz_ (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 18:03.20 | Kobaz | heh |
| 18:03.23 | drmessano | I generally hate the concept, but I at least tried it |
| 18:03.42 | drmessano | The rate limiting in iptables was brilliant and happy |
| 18:04.05 | TSM2 | drmessano: fail2ban does not work if the attack happens too quickly, it polls the log files using inotify |
| 18:04.26 | TSM2 | fail2ban is good at locking out FTP and SSH attempts though |
| 18:04.28 | drmessano | Every now and then I attack it, and have to wait 60 seconds for my phones to start working again after the IP is blocked temporarily |
| 18:04.38 | TSM2 | stop those pesky chineese bots |
| 18:04.40 | drmessano | I'm well aware |
| 18:04.41 | [TK]D-Fender | I left NTPD open on my server and got used as an NTPD DDoS relay and it ate up 50% of my bandwidth. Since adding a FW rule on my router level I can still see the dropped packet count rolling by, but it caps out at 10kbps..... |
| 18:04.54 | TSM2 | ahh the NTPD reflection attach |
| 18:05.04 | TSM2 | not as good though as a DNS reflection |
| 18:05.26 | [TK]D-Fender | yup, pissed me right off but my ISP clued me in as to what was happening and I caught it at my router level. |
| 18:05.40 | [TK]D-Fender | considering having them black-hole the IP and jsut a different one on my subnet |
| 18:06.05 | TSM2 | the only way really is to have an upstream router drop everything |
| 18:06.06 | [TK]D-Fender | fortunately not running named on it I guess ;) |
| 18:06.10 | [TK]D-Fender | yup |
| 18:06.16 | [TK]D-Fender | "black-hole" |
| 18:06.30 | TSM2 | our provider yesterday had a 100Gbps DDos attach, ouch |
| 18:06.51 | TSM2 | intermittant access to our DC, they sorted but took about 45mins |
| 18:07.06 | drmessano | I would love to have a 100Gbps DDoS |
| 18:07.14 | drmessano | Because that means I have 100Gbps |
| 18:07.16 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 18:07.19 | drmessano | and I currently do not |
| 18:08.08 | drmessano | I guess thats the same as wanting to have a blown transmission in my Lambo |
| 18:08.13 | TSM2 | if all ISP routers were setup correctly then reflection DDoS attacks would not happen much |
| 18:08.17 | drmessano | Because then, you know, Lambo |
| 18:09.24 | [TK]D-Fender | Which I'd pay to repair, immediately resell and buy something affordable and pocket the rest :) |
| 18:10.20 | coppice | affordable is relative. |
| 18:10.59 | coppice | some people would sell a corolla to use the money for something affordable |
| 18:11.13 | coppice | some people would repair a lambo with the loose change |
| 18:11.14 | drmessano | Top Gear makes me want a Mini |
| 18:11.38 | [TK]D-Fender | "reasonable to its rate of depreciation, and other operating expenses whil aloowing a significant return of current market value" |
| 18:15.47 | *** join/#asterisk nix8n82 (~AndChat27@24.143.10.36) |
| 18:28.11 | *** join/#asterisk funky1 (~funky@ip51cf100e.direct-adsl.nl) |
| 18:34.40 | paulc | now it works :) |
| 18:34.53 | paulc | I need screen to be set for audible bell, and putty to be set for visual |
| 18:34.53 | paulc | duh |
| 18:35.01 | paulc | uh - wrong window |
| 18:43.27 | ChannelZ-Wk | DING |
| 18:43.29 | ChannelZ-Wk | DING DING |
| 18:45.48 | drmessano | I think you have to remove it from the Blacklist |
| 18:49.41 | *** join/#asterisk c0ldg0ld (~c0ldg0ld@unaffiliated/c0ldg0ld) |
| 18:50.25 | *** part/#asterisk ddickenson (~c0ldg0ld@unaffiliated/c0ldg0ld) |
| 18:50.33 | *** join/#asterisk ddickenson (~c0ldg0ld@unaffiliated/c0ldg0ld) |
| 18:50.58 | *** part/#asterisk ddickenson (~c0ldg0ld@unaffiliated/c0ldg0ld) |
| 18:59.42 | *** join/#asterisk joeseemore (~joeseemor@bgareth.plus.com) |
| 19:00.27 | joeseemore | Hello, I need to use my pbx away from home - I have a static ip, what would be the best way of getting remote access securely - the easiest way |
| 19:01.27 | joeseemore | I also need to login from a softphone on my iphone |
| 19:02.09 | TSM2 | just open the ports |
| 19:02.36 | joeseemore | would that not lead to hacking attempts |
| 19:02.37 | TSM2 | you will have to do some NAT forwarding |
| 19:02.41 | joeseemore | dictionary attacks etc |
| 19:02.57 | TSM2 | make your passwords complex then |
| 19:03.46 | TSM2 | SIP does not use cleartext passwords, its a challenge response |
| 19:03.48 | joeseemore | is that enough to protect an outbound/inbound sip trunk? |
| 19:04.01 | *** join/#asterisk funky1 (~funky@ip51cf100e.direct-adsl.nl) |
| 19:04.20 | TSM2 | how do most other SIP services work |
| 19:04.32 | *** join/#asterisk Chotaire (chotaire@host-089-207-249-134.vipri.net) |
| 19:04.48 | TSM2 | other way is if your firewall supports VPN you could do it that wya |
| 19:04.51 | joeseemore | not sure |
| 19:05.05 | funky1 | wdoekes: tried but not sure how to understand what's going on, if u could take a look when u have time, would be great, posted my logs as grooverider at bottom here: http://forums.digium.com/viewtopic.php?f=1&t=84744&p=196450&sid=fbb79a68009af07c585902eb758fa1a6#p196450 |
| 19:06.08 | *** join/#asterisk theron (~theron@69.63.185.56) |
| 19:07.33 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 19:07.41 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 19:10.09 | drmessano | Most system exploits are not 0-day |
| 19:10.43 | drmessano | They're 2 or 10 year old nasty vulns that the sysadmin never patched |
| 19:11.08 | drmessano | Likely if it's a 2 year old exploit, it was fixed 1 year, 11 months, and 3 weeks prior |
| 19:11.21 | drmessano | Keep your system up to date, use strong passwords |
| 19:14.36 | *** join/#asterisk jhlavacek (~jirka@80.215.167.125) |
| 19:15.17 | *** part/#asterisk sicinthemind (~sicinthem@pool-71-100-232-20.tampfl.fios.verizon.net) |
| 19:16.06 | joeseemore | TSM2: I have tried sorting out a VPN but I couldn't get the help as i'm a newbie to Linux and Asterisk |
| 19:17.14 | joeseemore | TSM2: With the first option mentioned, you talk about a complex password - is that the password for the extension? |
| 19:22.38 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 19:24.45 | *** join/#asterisk wizbit (~wizbit@unaffiliated/wizbit) |
| 19:26.24 | Katty | FROSTED FLAKES |
| 19:26.26 | Katty | THEY"RE GREEEAATTTT! |
| 19:29.15 | wizbit | where can i get a uk british soundset for asterisk, all the sounds have american accents :-( |
| 19:31.17 | wizbit | Allison Smith sucks :( |
| 19:32.00 | *** join/#asterisk starks (~Eric@69.74.61.3) |
| 19:32.04 | starks | hi, anyone here familiar with getting cisco phones working on a switchvox? |
| 19:32.21 | _Corey_ | she's Canadian dude |
| 19:32.46 | starks | can't get the phones to register even though the sip commands are recognized |
| 19:43.02 | *** join/#asterisk julgr (~julgr@38.104.125.2) |
| 19:46.39 | *** join/#asterisk ghost75 (~quassel@ipservice-092-211-033-228.pools.arcor-ip.net) |
| 19:51.40 | *** join/#asterisk caveat- (hoax@2a01:4f8:191:9111:30::10) |
| 20:00.11 | *** join/#asterisk theron (~theron@69.63.185.56) |
| 20:04.54 | *** join/#asterisk starks (~Eric@69.74.61.3) |
| 20:05.11 | starks | hi, anyone familiar with putting cisco phones on asterisk/switchvox? |
| 20:05.35 | ghost75 | lots of work |
| 20:05.58 | starks | i've already turned the phone sip |
| 20:06.08 | starks | and the switchvox sees the sip commands |
| 20:06.12 | starks | just won't register the phone |
| 20:06.35 | ghost75 | lots of settings |
| 20:06.50 | *** join/#asterisk jpoz (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
| 20:06.55 | [TK]D-Fender | Then settings don't match |
| 20:07.11 | starks | if i provided a sipmac would you be able to help? |
| 20:08.17 | *** join/#asterisk jpoz_ (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 20:08.19 | [TK]D-Fender | that's half of the equation. |
| 20:08.56 | starks | try me |
| 20:09.26 | starks | is authpassword the phone password or voicemail password? |
| 20:11.56 | [TK]D-Fender | We don't know. |
| 20:12.01 | [TK]D-Fender | We've seen nothing. |
| 20:12.06 | [TK]D-Fender | maybe your username is off. |
| 20:12.09 | [TK]D-Fender | Maybe the password. |
| 20:12.18 | [TK]D-Fender | maybe you've restricted where they can register from |
| 20:12.22 | Katty | MAYBE. |
| 20:12.26 | Katty | it's a lack of blueberry muffins. |
| 20:12.30 | [TK]D-Fender | maybe you've restricted them from registering at all |
| 20:12.47 | Katty | no i'm fairly certain it's a lack of blueberry muffins. |
| 20:12.48 | ghost75 | to much maybe overall :) |
| 20:12.52 | starks | [TK]D-Fender, http://pastebin.com/zj8iZXyP |
| 20:13.01 | [TK]D-Fender | Katty: there's ALWAYS a lack of blueberry mussifns |
| 20:13.30 | [TK]D-Fender | muffins* |
| 20:13.56 | starks | i trust you voip sages can help me, digium is pretty bad at these cases |
| 20:14.26 | *** join/#asterisk Chotaire (chotaire@host-089-207-249-134.vipri.net) |
| 20:14.33 | starks | anyway, the phones are 7945/7965 |
| 20:15.52 | *** join/#asterisk bsdice (~bsdice@embinet.eu) |
| 20:16.16 | _Corey_ | I remember the SIP image on those 79xx phones being very picky about NAT settings |
| 20:16.25 | *** join/#asterisk g_r_eek (~g_r_eek@46-34-139.adsl.cyta.gr) |
| 20:16.49 | ghost75 | cisco needs tcp |
| 20:17.22 | [TK]D-Fender | No they don't |
| 20:17.38 | [TK]D-Fender | and this is still only half (or part of half) of the picture |
| 20:17.49 | [TK]D-Fender | <PROTECTED> |
| 20:18.08 | bsdice | <PROTECTED> |
| 20:18.10 | ghost75 | works better with tcp |
| 20:18.19 | ghost75 | and on proxy setting i have USECALLMANAGER |
| 20:18.51 | starks | do i set tcp in the xml or on the pbx? |
| 20:19.02 | starks | i'm using 8.5 sip btw |
| 20:19.34 | ghost75 | http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP |
| 20:19.45 | ghost75 | uh 8.5 is old |
| 20:20.31 | starks | i tried 9.3 but i hear it forces tcp |
| 20:21.14 | starks | and i don't feel like booting a linux distro off usb just to edit sip.conf on the pbx |
| 20:22.07 | [TK]D-Fender | Like that'd do you any good with a GUI walking all over it |
| 20:22.26 | [TK]D-Fender | Whichis where you should be able to go to set your device up in the first place. |
| 20:22.46 | starks | so where do i stand? i just want a friggin dial tone |
| 20:23.17 | ghost75 | this is nothing u configure in like 5minutes |
| 20:23.31 | [TK]D-Fender | starks: you stand nowhere because this is less than half of the picture |
| 20:23.33 | starks | my manager wants this done for 100+ phones |
| 20:23.44 | starks | we're cisco for god knows why |
| 20:23.47 | starks | *ditching |
| 20:24.06 | starks | **cisco call manager |
| 20:25.00 | starks | then let's do this step by step. i have a sip phone on this desk. it can run 8.5 or 9.3. my provisioning server "works" |
| 20:25.07 | sp00kz | ugh i used ccm back in 2004 it was the worst |
| 20:26.07 | starks | key questions: what should my switchvox extension config look like? disable nat? enable nat? phone password is set and i assume that's what i authorize with |
| 20:27.23 | wdoekes | funky1: you got the 200 |
| 20:27.37 | starks | i assume i don't need nat since the pbx is on our local net |
| 20:27.50 | wdoekes | but you forgot the -m 1 on the command line, so you were now load testing the server |
| 20:28.03 | ghost75 | nat depends on how u use it |
| 20:29.07 | wdoekes | funky1: so, if your twinkle registers, and the sipp scenario registers, but asterisk doesn't, then try to make the scenario more like the asterisk attempt |
| 20:31.46 | ghost75 | is * having any timeouts when sending sip invite to other peers? |
| 20:35.55 | funky1 | wdoekes: ok thanks will give it a go and see how far i get |
| 20:38.39 | *** join/#asterisk roentgen (~irc@openvpn/community/support/roentgen) |
| 20:47.56 | *** join/#asterisk whizzi (~whizzi@31-151-40-89.dynamic.upc.nl) |
| 20:50.56 | *** join/#asterisk XaV`S (~xavs@modemcable158.95-130-66.mc.videotron.ca) |
| 20:52.16 | *** join/#asterisk MauriceM_ (~MauriceM_@71-222-67-129.ptld.qwest.net) |
| 20:59.25 | *** join/#asterisk woleium (~woleium@209-82-119-242.dedicated.allstream.net) |
| 21:00.08 | woleium | lo peeps :). |
| 21:01.15 | woleium | in a trunk definition, if I want to disallow all codecs, but allow ulaw and T38 waht's the string to match T38 |
| 21:01.16 | woleium | ? |
| 21:01.28 | woleium | I have allow=ulaw&t38 |
| 21:01.50 | *** join/#asterisk DougsTech (~DougsTech@c-98-230-105-210.hsd1.al.comcast.net) |
| 21:01.52 | woleium | but I'm not sure, maybe it should be t38pt ?? |
| 21:03.24 | woleium | sorry, I should probably say in a sip trunk definition... :-$ |
| 21:04.42 | whizzi | t38 isn't a codec |
| 21:04.50 | whizzi | allow=ulaw |
| 21:04.51 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
| 21:05.11 | whizzi | and accept t38 on that particular trunk/peer |
| 21:05.13 | file | not... yet... MUAHAHAHAHAHA |
| 21:06.37 | whizzi | t38pt_udptl=yes would do the trick |
| 21:12.28 | mjordan | file: duh duh duhnnnn |
| 21:12.43 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.133) |
| 21:13.24 | woleium | Aah, I see thanks whizzi |
| 21:13.46 | whizzi | yw :) |
| 21:14.12 | *** join/#asterisk [sr] (~kvirc@pal-213-228-163-73.netvisao.pt) |
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| 21:18.58 | *** part/#asterisk Zogot (~Leon@26.sub224.ddfr.nl) |
| 21:19.30 | *** join/#asterisk MaliuatLap (~nobusines@eth637.qld.adsl.internode.on.net) |
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| 21:32.30 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
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| 21:39.43 | ghost75 | anyone seen sip 480 errors ? |
| 21:40.26 | whizzi | Temporary Unavailable ? |
| 21:40.41 | ghost75 | yes |
| 21:41.00 | whizzi | is your SIP trunk sending it or your own Asterisk instance ? |
| 21:41.16 | ghost75 | getting from sip peer |
| 21:41.52 | whizzi | directly after the INVITE I assume ? |
| 21:41.59 | ghost75 | yes |
| 21:42.08 | whizzi | then your SIP peer has issues |
| 21:42.19 | ghost75 | i am getting on about 90% outbound calls |
| 21:42.37 | whizzi | its either busy or blocking your account |
| 21:42.38 | ghost75 | if i connect a router with internal ata/voip, everything is ok though |
| 21:43.18 | ghost75 | isp is blocking * or what oO |
| 21:43.26 | whizzi | maybe |
| 21:43.37 | whizzi | or your provider is blocking the 200 OPTIONS for some reason |
| 21:44.06 | whizzi | you could try to set qualify=no on that particular peer to see if that helps |
| 21:44.15 | ghost75 | tried already |
| 21:44.23 | whizzi | didnt help? |
| 21:44.28 | ghost75 | nope |
| 21:45.11 | ghost75 | isp doesnt respond to qualify also |
| 21:45.29 | whizzi | hmm. Weird.. setting reregistering time to 2 hours for example? |
| 21:45.48 | ghost75 | so long? |
| 21:46.14 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 21:46.25 | whizzi | to try out, I dont know what the rereg-time on the trunk provider is ;) |
| 21:46.28 | ghost75 | on qualify the answer is: <--- SIP read from UDP:88.79.152.249:5060 ---> |
| 21:46.28 | ghost75 | SIP/2.0 403 Forbidden |
| 21:46.42 | whizzi | thats okay for Asterisk ;) |
| 21:46.54 | whizzi | as long as it says something SIP back, its considered as online |
| 21:46.54 | ghost75 | but wth they are blocking this |
| 21:47.23 | whizzi | they are not blocking it |
| 21:47.35 | whizzi | the software doesnt know how to respond correctly |
| 21:47.55 | whizzi | OR.. your peer isnt registered correctly |
| 21:48.08 | whizzi | that would explain why they send a 480 to you on INVITE |
| 21:48.27 | ghost75 | * doesnt know if its registered correctly? |
| 21:48.44 | ghost75 | at least inbound calls do work |
| 21:49.36 | ghost75 | registertimeout is set to 70 |
| 21:49.45 | whizzi | ghost75: sip show registry ... |
| 21:50.05 | ghost75 | all fine, i even monitor that |
| 21:50.16 | whizzi | State= registered? |
| 21:50.24 | ghost75 | yeah |
| 21:50.31 | whizzi | ok, thats good |
| 21:50.46 | whizzi | wonders where the single quote is on his Amiga :P |
| 21:50.54 | whizzi | never mind, uhm |
| 21:51.03 | ghost75 | amiga oO |
| 21:51.09 | whizzi | registertimeout to 70? Thats 70 seconds .. |
| 21:51.24 | ghost75 | default is 20 i think |
| 21:51.37 | whizzi | is now on an Amiga 1200 on IRC |
| 21:51.50 | whizzi | this doesnt do Asterisk though :P |
| 21:51.52 | ghost75 | that really IS old school |
| 21:52.10 | whizzi | it is.. but I need a hobby right? ;) |
| 21:52.44 | ghost75 | is * having any timeouts after the invite? |
| 21:52.56 | ghost75 | maybe the answer is not quick enough |
| 21:55.23 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.133) |
| 21:55.28 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:55.37 | whizzi | registertimeout is the number of seconds before Asterisk declares it as being "timeout" |
| 21:55.54 | ghost75 | yes |
| 21:56.07 | whizzi | so, it is not re-registering time ;) |
| 21:57.58 | whizzi | my best advice is to contact your trunk-provider to see if they have an explanation why you get 480s |
| 21:58.10 | ghost75 | <PROTECTED> |
| 21:58.30 | *** join/#asterisk CeBe (~CeBe@port-92-206-95-160.dynamic.qsc.de) |
| 21:58.36 | whizzi | or make a trace between the working ATA and compare it to the Asterisk one |
| 21:58.53 | ghost75 | they would just answer that i should use there own model router :< |
| 21:59.12 | whizzi | there is a difference somewhere, but it is hard to tell what or how |
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| 22:01.33 | *** join/#asterisk TheProf (~chatzilla@66.187.93.128) |
| 22:01.39 | TheProf | Hello. I hope you are doing very well. I am running Elastix and it is working amazingly. I had a question about connecting a regular fax machine -- can such a machine be connected to Elastix to allow faxes to come in via VOIP and emailed to people? |
| 22:02.07 | TheProf | I know the underpinning of Elastix is Asterisk so I thought to ask here. Thanks. |
| 22:02.30 | [TK]D-Fender | TheProf: Is the fax comes in... it won't be to a regular fax machine... |
| 22:02.47 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.141) |
| 22:03.03 | [TK]D-Fender | TheProf: that call would not go to your physiucal fax machine and then get miraculously turned into an e-mail |
| 22:03.19 | [TK]D-Fender | TheProf: You would instead have your PBX receieve it DIRECTLY and e-mail it off |
| 22:04.12 | TheProf | [TK]D-Fender: Hello. OK I understand. For incoming I'd need to give each user their own number and it would then go to appropriate email based on which number it came in from. |
| 22:04.32 | TheProf | [TK]D-Fender: What about for transmitting? Could I use the physical fax machine to do this via VOIP? |
| 22:04.50 | [TK]D-Fender | The phsical fax machine would ahve to be plugged into something. |
| 22:04.56 | [TK]D-Fender | Elastix is SOFTWARE |
| 22:05.08 | [TK]D-Fender | at which point you'd need a T.38 capable ATA |
| 22:05.25 | [TK]D-Fender | And be using a provider that supports it (both ways in fact) |
| 22:09.47 | TheProf | [TK]D-Fender: I understand. So I need an interface card (the ATA card?) in the voip server where I jack in the physical fax and then I check with my service provider for the T.38 codec. That sounds straightforward. |
| 22:10.04 | TheProf | Is there anything else I should consider (since I don't know what I don't know!) |
| 22:10.06 | [TK]D-Fender | T.38 protocol. It is not a codec |
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| 22:10.35 | [TK]D-Fender | You'll have to make sure your mail-server options are right on your server as well |
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| 22:11.43 | TheProf | Right - so it can send the appropriate emails once it receives the faxes. |
| 22:12.14 | [TK]D-Fender | correct |
| 22:12.54 | TheProf | [TK]D-Fender: Thanks. Is there a specific brand or specification for the ATA (in addition to the T.38 protocol) that is recommended? |
| 22:13.53 | WHiZZi | TheProf: I always used a dumb Fritz!Box to receive fax and send it as an email |
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| 22:15.33 | TheProf | WHiZZi: Hello. From my quick Google search it's basically a gateway with built-in phone port? |
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| 22:17.56 | WHiZZi | TheProf: Yes.. that is correct |
| 22:18.16 | TheProf | Neat thanks for that. |
| 22:18.17 | WHiZZi | TheProf: Asterisk cannot magically receive fax (T.38 or not) and make it into an email |
| 22:18.22 | WHiZZi | it needs software for that |
| 22:18.39 | WHiZZi | and there is software available for that, but it comes with a price |
| 22:19.12 | TheProf | WHiZZi: Right. I didn't mean it would magically do it per se -- I just was trying to figure out the actual wiring connection for it. Do you have the name of the software for that? |
| 22:19.15 | WHiZZi | the other option is to get an ATA which allows receiving T.38 and make it into an email, for example this Fritz!Box :) |
| 22:19.26 | TheProf | I'm going to check both out now. |
| 22:19.28 | WHiZZi | Fax2Email for Asterisk :P |
| 22:19.49 | [TK]D-Fender | [17:18]WHiZZiTheProf: Asterisk cannot magically receive fax (T.38 or not) and make it into an email <- first half, yes, it can |
| 22:20.09 | [TK]D-Fender | [17:19]WHiZZithe other option is to get an ATA which allows receiving T.38 and make it into an email, for example this Fritz!Box <- not needed |
| 22:21.39 | WHiZZi | [TK]D-Fender: I know, it can receive fax .. it cannot change it into e-mail |
| 22:21.49 | WHiZZi | [TK]D-Fender: but thanks for the clarification |
| 22:24.46 | TheProf | So many of the ATA devices I'm finding online are stand-alone devices. I did also see what looked like ATA adapters that can fit into a server. Could I use the adapter in order to manage less devices? |
| 22:25.57 | TheProf | Since our building has no POTS lines coming in. |
| 22:26.14 | [TK]D-Fender | There ar PCI(e) cards you could use, but you'd lose T.38 support for the fax machine. Thereefor not viable |
| 22:26.59 | TheProf | [TK]D-Fender: OK thank you. |
| 22:27.36 | [TK]D-Fender | they also cost substantially more and place a greater burdon on your server |
| 22:28.20 | TheProf | [TK]D-Fender: you mean in terms of processing cost? |
| 22:28.42 | [TK]D-Fender | processing, hardware driver support, etc |
| 22:29.54 | TheProf | OK. In terms of processing power I've got a box that was donated that is WAY overkill for VOIP and so it just sits doing next to nothing all day. I don't think I've ever see the load rise to any significant amount. So processing power I'd be OK but drivers, etc can be an issue as you said. |
| 22:30.29 | [TK]D-Fender | All pain, no gain |
| 22:32.20 | TheProf | I understand. Thank you for the help |
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