| 00:15.52 | *** join/#asterisk shapr (~shapr@nat/digium/x-qqcgueioioaqrxte) |
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| 00:33.06 | itsbroken | How can I check what version of dahdi asterisk is using |
| 00:41.04 | itsbroken | I just upgraded dahdi and when i restarted dahdi through /etc/init.d/dahdi it still says the old version in dmesg |
| 00:41.07 | itsbroken | any ideas? |
| 00:42.05 | ChannelZ | dahdi show version |
| 00:42.55 | ChannelZ | though that actually probably only tells you what version asterisk was built against, not necessarily what is running |
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| 01:49.34 | Godofmonkeys | anyone have any sample configs to work with simple signal? |
| 02:00.28 | shapr | Simple signal? |
| 02:00.46 | Godofmonkeys | yes, i'm having a bit of trouble with a 604 |
| 02:01.03 | shapr | itsbroken: If the old version of dahdi still shows at the end of dmesg, you probably haven't done "make install" or so |
| 02:01.06 | shapr | Godofmonkeys: What is that? |
| 02:01.16 | Godofmonkeys | my trunk provider |
| 02:01.30 | shapr | need more info |
| 02:01.36 | Godofmonkeys | pcap? |
| 02:01.41 | shapr | no, I mean... |
| 02:01.50 | shapr | What simple signal? digital, analog, or sip? |
| 02:01.54 | Godofmonkeys | sip |
| 02:02.15 | shapr | What's the actual problem? |
| 02:02.44 | Godofmonkeys | 604 user not found anywhere, i think it may be nat related, seems the contact header is sending out the private ip of the asterisk server |
| 02:02.53 | shapr | set externip |
| 02:03.06 | Godofmonkeys | I have. it reports it correctly once. |
| 02:03.36 | shapr | asterisk has a 192.168.* or 10.* or so ip? |
| 02:03.40 | Godofmonkeys | yes |
| 02:04.33 | shapr | Is the trouble that you can't call your SIP provider, or that an external extension cannot dial in? or what? |
| 02:05.05 | Godofmonkeys | cannot call out or in at the moment |
| 02:05.20 | Godofmonkeys | contact-uri is incorrect |
| 02:05.40 | shapr | Can you call from one internal extension to another? |
| 02:05.44 | Godofmonkeys | yes |
| 02:07.46 | Godofmonkeys | well; could. i just formatted the box, and am running through the extensions again. this is the 4th time today, same result every time |
| 02:23.08 | p3nguin | If Asterisk is behind a NAT, you have to configure it to work behind NAT. |
| 02:23.26 | ectospasm | Godofmonkeys: are you sure your simple signal credentials are correct? Has it ever worked? |
| 02:23.27 | p3nguin | There's more to it than just externip. |
| 02:24.05 | Godofmonkeys | yes, the credentials are correct, i get the 200/ok packets |
| 02:25.52 | ectospasm | Godofmonkeys: have you called Simple Signal about this? They may be able to help. |
| 02:26.15 | Godofmonkeys | ah no. they were little more help than giving me the pcap files |
| 02:27.38 | ectospasm | you didn't answer my previous question: has it ever worked? |
| 02:27.51 | Godofmonkeys | first time through |
| 02:27.56 | ectospasm | so what changed? |
| 02:29.06 | Godofmonkeys | no; this is the first time through all of it. i'm attempting to replace an older analog pbx being fed with an analog terminal adapter using sip users. now i have a trunk. |
| 02:29.19 | Godofmonkeys | and nice polycom phones |
| 02:29.43 | p3nguin | ~trunk |
| 02:29.43 | infobot | it has been said that trunk is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. the nose of an elephant |
| 02:29.44 | Godofmonkeys | p3nguin, i had also set the localnet variable |
| 02:29.59 | Godofmonkeys | indeed. ' |
| 02:30.31 | p3nguin | You also need to set nat and directmedia (canreinvite). |
| 02:31.58 | shapr | I have a trunk for all my sip papers, can I call that my sip trunk? |
| 02:32.02 | Godofmonkeys | both were set to yes |
| 02:33.12 | Godofmonkeys | it is also probably worth noting the asterisk server is in the dmz of my router |
| 02:33.28 | p3nguin | directmedia or canreinvite needs to be set to no when behind NAT. |
| 02:34.11 | p3nguin | Never, never, never use the DMZ setting for any residential network appliance. |
| 02:34.16 | p3nguin | s/for/on/ |
| 02:35.08 | p3nguin | A) No one I have ever met knows HTF to use it, and B) it usually isn't implemented well. |
| 02:35.26 | Godofmonkeys | only as a testing measure; i have to make it all work before i worry about what effing ports to use |
| 02:35.56 | p3nguin | It should never be used for "a testing measure" or any other time. Just don't use it. |
| 02:36.25 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 02:36.28 | p3nguin | It's very simple: forward UDP 5060 and whatever UDP range is defined in rtp.conf |
| 02:36.43 | Godofmonkeys | ok, one moment |
| 02:37.03 | ectospasm | but that doesn't explain the 604 |
| 02:37.20 | p3nguin | Not having a good sip.conf probably explains that. |
| 02:37.53 | p3nguin | Not having read The Book is often a cause of not having a good sip.conf. |
| 02:38.14 | ectospasm | ~thebook |
| 02:38.15 | infobot | extra, extra, read all about it, thebook is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org, or http://ofps.oreilly.com |
| 02:38.35 | shapr | Anyone know why only some programs would be able to DNS lookups with an empty /etc/resolv.conf? Or how they could get DNS results at all? |
| 02:38.44 | ectospasm | The Definitive Guide is in beta, linked (somewhere) on asteriskdocs.org |
| 02:39.08 | shapr | The one on ofps.oreilly.com is good |
| 02:40.38 | *** join/#asterisk sourcode (~code@ppp-58-8-196-188.revip2.asianet.co.th) |
| 02:41.23 | Godofmonkeys | wow. looks like my printer is going to be quite busy for a bit |
| 02:41.54 | shapr | Godofmonkeys: The ofps book is cc, so you'd do better to critique the parts you don't understand or feel aren't clear.. it's continually changing. |
| 02:47.30 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net) |
| 02:55.51 | ectospasm | waste of paper, IMO |
| 03:01.16 | *** join/#asterisk raden (~J@66-168-15-100.dhcp.stpt.wi.charter.com) |
| 03:01.29 | raden | whats average packet size of a G729 call ? |
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| 03:28.47 | coppice | they are about |<----->| that big |
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| 03:45.27 | dschuett | anyone want to take a look and see why this isn't working? http://pastebin.com/Ar6nLhv5 |
| 03:55.50 | jasonb | Can anyone suggest a fix or a way of debugging this?: http://fpaste.org/PRkO/ |
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| 04:06.38 | juliocesarlhg | hi |
| 04:06.47 | juliocesarlhg | is it posible to store cdr in oracle? |
| 04:08.35 | juliocesarlhg | ????? |
| 04:11.27 | p3nguin | mysql? |
| 04:12.29 | *** join/#asterisk benngard (~mabe@213.88.138.230) |
| 04:12.50 | p3nguin | Maybe you can use OBDC to get CDR to your Oracle DB. |
| 04:13.20 | juliocesarlhg | i would like to know if someone have try? |
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| 04:42.01 | tyrrexrrg | hi everyone |
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| 05:25.29 | tyrrexrrg | Hi everyone |
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| 05:37.13 | *** part/#asterisk jmls (~Julian@host217-36-208-155.in-addr.btopenworld.com) |
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| 06:04.39 | Godofmonkeys | hello, anyone avaliable for troubleshooting? |
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| 06:19.46 | p3nguin | godofmonkeys: What's the problem this time? |
| 06:21.12 | Godofmonkeys | p3nguin, jesus man, i get it, noobs suck. im still stuck on the 604. my provider sees that i am sending out my asterisk server public ip in the contact header |
| 06:21.25 | Godofmonkeys | err private ip |
| 06:22.00 | p3nguin | Do you have the capability to set NAT or no NAT on the provider? |
| 06:22.16 | Godofmonkeys | not at the provider, i dont think |
| 06:22.32 | p3nguin | Okay. I was going to have you set it to no nat if you had that capability. |
| 06:23.35 | p3nguin | With Asterisk, we typically configure all NAT settings on the Asterisk side of things and any device that supports NAT is left without NAT configuration. Asterisk has the ability to control it all, and if you set other devices to traverse NAT, it ruins Asterisk's ability to do it. |
| 06:23.44 | p3nguin | That's where I was going with that question. |
| 06:24.30 | p3nguin | Paste your entire sip.conf into a pastebin. Mask nothing but secret passwords. |
| 06:24.39 | Godofmonkeys | one moment please |
| 06:25.18 | Godofmonkeys | http://pastebin.com/awzCFgiN |
| 06:25.57 | Godofmonkeys | it's very basic, nothing but 1 user and the simpsig entries |
| 06:26.07 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 06:27.48 | Godofmonkeys | extensions.conf http://pastebin.com/hVfYGHa8 |
| 06:30.45 | juliocesarlhg | what is your problem? |
| 06:31.01 | Godofmonkeys | 604, user not found anywhere |
| 06:31.20 | juliocesarlhg | what r u trying to do? |
| 06:31.55 | Godofmonkeys | call out to a pots phone over voip with simple signal as a provider |
| 06:32.03 | *** join/#asterisk Tim_Toady (~moi@79.103.49.227) |
| 06:32.48 | juliocesarlhg | asterisk shows u what error |
| 06:32.51 | juliocesarlhg | paste it |
| 06:33.09 | Godofmonkeys | one moment please, i must generate the error |
| 06:34.56 | juliocesarlhg | ? |
| 06:35.01 | Godofmonkeys | http://pastebin.com/6Xh0QPMZ |
| 06:35.18 | Godofmonkeys | turned debug on to give you the most info i could |
| 06:36.43 | juliocesarlhg | first |
| 06:36.54 | juliocesarlhg | what kind of telephone u have? |
| 06:37.08 | Godofmonkeys | softphones. have polycoms at the office to use. |
| 06:37.14 | Godofmonkeys | x-lite |
| 06:37.27 | *** join/#asterisk joobie (~joobie@unaffiliated/moo0o0ooo00o0o0o) |
| 06:37.31 | juliocesarlhg | your provider what codec uses? |
| 06:38.35 | p3nguin | godofmonkeys: Okay, your sip.conf is a bit messed up. externip and localnet are not peer parameters, but you've erroneously put them in a peer rather than in the general section. |
| 06:38.51 | Godofmonkeys | ok, one moment |
| 06:38.52 | p3nguin | godofmonkeys: You've also revealed your password, so you might want to change that. |
| 06:41.51 | p3nguin | godofmonkeys: I would rather see it more like this: http://pastebin.com/xhEKt3bu |
| 06:42.33 | p3nguin | oops, bugfix: http://pastebin.com/Sse4hDqK |
| 06:43.30 | p3nguin | Are you using Asterisk 1.8.something? |
| 06:43.39 | Godofmonkeys | 1.6.x |
| 06:43.47 | wdoekes2 | (may I suggest that you do not use externip unless you're really sure you need it) |
| 06:44.08 | Work2Play_ | anyone around here any good at flashing a cisco 7975 to sip? I've gotta be close but I'm still stumbling around through the linux - not that experienced yet. Trying to get this thing on a new FreePBX box |
| 06:44.12 | wdoekes2 | (it usually breaks more than it fixes) |
| 06:44.14 | Godofmonkeys | i am sure, before it would dump my private ip |
| 06:44.40 | wdoekes2 | yes Godofmonkeys, sure it does, but that doesn't mean that your peer cannot cope with that |
| 06:45.02 | Godofmonkeys | they could not |
| 06:45.11 | Godofmonkeys | that was one of the first issues i had |
| 06:45.31 | wdoekes2 | ok |
| 06:46.36 | p3nguin | I know that username was changed to defaultuser in 1.8, but are you sure it is defaultuser in 1.6.somebranch.someversion? |
| 06:47.55 | p3nguin | If your Asterisk is behind a NAT, you *must* set externip or externhost and localnet. It's not optional if you want it to work. |
| 06:48.22 | Godofmonkeys | yes, when it was username it gave a warning |
| 06:48.28 | p3nguin | work2play_: What's the problem? Do you have the SIP firmware files? |
| 06:48.54 | Work2Play_ | yes I do - I just haven't used tftp a whole lot - |
| 06:49.02 | p3nguin | godofmonkeys: Okay, good. I personally use 1.4 branch and only admin 1.4 and 1.8 boxes. Never used 1.6.anything and have no reason to. |
| 06:49.20 | p3nguin | work2play_: Your tftpd is on a Linux host? |
| 06:49.26 | Godofmonkeys | the fix given results in: http://pastebin.com/ZTGRZzjX |
| 06:49.27 | Work2Play_ | I had added option 150 and option 66 to my router pointing to my freepbx server |
| 06:49.53 | p3nguin | Forget about FreePBX. We don't do FreePBX here; this is an Asterisk channel. |
| 06:50.11 | Work2Play_ | hrm - hold that thought - I just pointed it to my windows tftp host and now it's upgrading |
| 06:50.34 | p3nguin | perfect |
| 06:50.38 | *** join/#asterisk asterisk-learner (~chatzilla@77.42.241.114) |
| 06:50.48 | Work2Play_ | ok good to know - i'm taking small steps here so I can learn - trying to ease my way in here |
| 06:50.57 | p3nguin | You don't have to put the tftpd on the same system as Asterisk if you don't want to. If you do want to, I can help you do it. |
| 06:51.31 | Work2Play_ | well now the phone is going into the correct firmware upgrade screen but it's erroring out - so I must have the .cnf file formatted wrong |
| 06:53.07 | Work2Play_ | but hey - it's one baby step better |
| 06:53.09 | p3nguin | I run netkit-tftp and tftp-hpa in both stand-alone daemon as well as xinetd configurations, so if you need help with putting the tftpd on the same box as Asterisk, I bet I can tell you what you need to do. |
| 06:54.08 | Work2Play_ | thx - I appreciate it. I think I may have just found the problem on the windows side; about to find out |
| 06:54.37 | p3nguin | tftpd32 is pretty foolproof. |
| 06:54.51 | Work2Play_ | that's actually what i'm running (well tftpd64) |
| 06:55.15 | p3nguin | It's easy to manage, so that'll be a good one for you to use for now. |
| 06:55.17 | Work2Play_ | now it looks like it's gonna upgrade for half a second then says "auth fail" |
| 06:55.33 | Work2Play_ | talk about tiny increments of progress |
| 06:56.06 | p3nguin | I don't know why there would be any authentication involved at that level. |
| 06:56.57 | p3nguin | I don't personally have a 7975, but I have to assume the procedure is at least somewhat similar to that of a 7960. |
| 06:57.53 | Work2Play_ | does the 7960 do the xml files? I thought I had read that the 7940/7960 were a hair different than the 7945/7975 |
| 06:58.23 | p3nguin | For the 7960, you put in the files that you want it to use, set any cnf files that the phone reads to include the version name/numbers for the firmware you want to load, and then let 'er rip. |
| 06:58.40 | Work2Play_ | hrm- looks like cisco tac says I must upgrade specifically to 8.5(2) then I can proceed past |
| 06:59.16 | Work2Play_ | i am trying to load the absolute newest version of sip firmware - could be why |
| 06:59.20 | p3nguin | The 7940/7960 use a SIPDefault.cnf and a SIP<MAC>.cnf |
| 06:59.39 | p3nguin | If you have an old version on it, you may have to go in steps. |
| 06:59.58 | Work2Play_ | ok - ya the 7975 uses a cnf.xml which i guess has more elements to it |
| 07:00.04 | p3nguin | Like, for example, if you have SCCP 5.2, you can't go beyong SIP 6.1. |
| 07:00.15 | p3nguin | beyond |
| 07:00.31 | p3nguin | Those aren't the real values. |
| 07:00.50 | p3nguin | I'd have to look it up in the version/upgrade matrix. |
| 07:01.12 | Work2Play_ | right - i get what you're saying |
| 07:01.39 | Work2Play_ | i'm gonna try to upgrade sccp to 8.5.2 then try again |
| 07:01.53 | p3nguin | Then once you get SIP 6.1 on it, then you can ugprade to SIP 9.1 (again, example values). |
| 07:02.12 | p3nguin | What version do you have on it currently? |
| 07:03.00 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 07:03.04 | Godofmonkeys | p3nguin, http://pastebin.com/ZTGRZzjX |
| 07:03.21 | Work2Play_ | 75.8-3-2 |
| 07:03.24 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:03.49 | p3nguin | godofmonkeys: All I can tell from your debug is that the provider doesn't seem to have a peer configured for you. I don't quite understand why you would have all that info on a user but then not have a user on their system. |
| 07:04.25 | Godofmonkeys | if there is not a user, would i still be able to register? |
| 07:04.56 | p3nguin | It wouldn't make sense that you could. Things just don't add up. |
| 07:05.08 | p3nguin | Do they offer any type of Asterisk configuration support? |
| 07:05.46 | Godofmonkeys | Not asterisk directly, although they advertise as the perfect solution for asterisk viop services |
| 07:06.29 | Godofmonkeys | http://www.simplesignal.com/sip_asterisk.php |
| 07:07.39 | Godofmonkeys | then for configuration it's only "Asterisk Configuration |
| 07:07.39 | Godofmonkeys | Please check back for updates on Asterisk Configuration... " |
| 07:07.59 | p3nguin | Have you ever made any connections to an ITSP work? Would it make you feel better if you could get a different provider to work? |
| 07:08.11 | Godofmonkeys | contract :( |
| 07:08.53 | Godofmonkeys | we already use them with an older analog norstar system and a linksys ata voip adapter |
| 07:09.09 | Godofmonkeys | the norstar got hit by lightning and is crazy |
| 07:09.20 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 07:09.26 | Godofmonkeys | calls go off to nowhere, etc. that's when i found asterisk. |
| 07:09.28 | p3nguin | Do you have the ATA configured with the same credentials as what you're trying to use on Asterisk? |
| 07:09.31 | Work2Play_ | sweet p3nguin - it's taking the newer SCCP firmware now |
| 07:09.54 | p3nguin | work2play_: Once you get that done, then you should be able to switch over to a comparable version of SIP. |
| 07:09.56 | Godofmonkeys | no, it's still in production. called and requested another user |
| 07:11.28 | p3nguin | I was thinking I could set you up with a trial on another ITSP just to make sure your system works. |
| 07:11.48 | Godofmonkeys | sure, at this point i have to see something work |
| 07:12.19 | p3nguin | Give me a couple of minutes to set it up. |
| 07:12.25 | Godofmonkeys | thank you. |
| 07:14.18 | Work2Play_ | and there she blows - now she's takin' the newest SIP firmware. Awesome! |
| 07:16.57 | *** join/#asterisk oej (~olle@94.127.50.104) |
| 07:19.05 | wdoekes2 | p3nguin: you're very wrong about "you *must* set externip" |
| 07:20.01 | wdoekes2 | (1) if you set externip, the peer might think you're not natted and trust your port (which may be mangled by that nat router) |
| 07:20.43 | wdoekes2 | (2) if you set externip, some SIP-mangling ALG-routers will choke on the packets and break them |
| 07:24.38 | wdoekes2 | when the other end is an asterisk machine with nat options turned on, I've only seen externip have negative effects |
| 07:26.34 | ChannelZ | externip is to tell the *server* what its own IP is if it's behind a firewall |
| 07:27.17 | *** join/#asterisk jblack (~jblack@pool-71-181-219-109.sctnpa.east.verizon.net) |
| 07:28.16 | *** join/#asterisk moy (~moy@CPE002719f00364-CM0026f3a1701d.cpe.net.cable.rogers.com) |
| 07:28.56 | ChannelZ | If you're using an ALG that understands SIP then you've got to decide who wants to do the fudging. |
| 07:30.34 | Work2Play_ | ok so now this thing is fully up with the SIP firmware - that's awesome... now I just need to figure out how to assemble a good config file for it |
| 07:44.59 | *** part/#asterisk Balistic (~g@196.1.60.118) |
| 07:45.14 | asterisk-learner | hi, what is the latest asterisk pdf file version of "asterisk the future of telephony" ? |
| 07:45.46 | *** join/#asterisk kaldemar (~kaldemar@unaffiliated/kaldemar) |
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| 08:17.58 | jkroon | hi guys, i've got a client with a digium 4-port card, two FXS and two FXO (1:1 mapping). |
| 08:18.19 | jkroon | the provider provides the client with call waiting and basic hold functionality, but asterisk is being used for call recording. |
| 08:18.41 | jkroon | what the client wants is the ability to answer the alternative incoming call, but I have no idea how to even begin with doing that. |
| 08:19.51 | jkroon | basically asterisk will need to on the FXO port pick up that another call is incoming (waiting indications?), then if the FXS port gets flashed also flash the FXO port so as to allow the secondary call to come in - how will I tackle this problem? If at all possible? |
| 08:20.03 | shapr | You could register your analog card on digium.com and create a tech support case asking for install assistance. |
| 08:20.16 | shapr | At which point I would escalate your case to my supervisor :-/ |
| 08:20.35 | jkroon | hehe, will do that then - OK if I just copy&paste that description into the ticket? |
| 08:20.50 | shapr | Actually, I'd furiously search google/asterisk-sources/etc to see if I can figure it out myself. |
| 08:21.06 | shapr | jkroon: Yah, though I reserve the right to rewrite it a bit to clarify. |
| 08:21.15 | jkroon | lol - brownie points with supervisor ? (and trust me - i've googled a LOT) |
| 08:21.18 | jkroon | you're welcome. |
| 08:21.22 | shapr | jkroon: Oh, and please include country of use, provider, and signalling info if possible. |
| 08:21.48 | jkroon | crap - i don't think I took down the serial of that particular card before shipping it. |
| 08:22.10 | shapr | More that we get more than enough cases here in Digium tech support, and the more I can solve without passing the problem to someone else, the more chance we get everything done on time. |
| 08:22.11 | jkroon | chan_dahdi.conf + users.conf? |
| 08:22.30 | shapr | Would help, sure. |
| 08:24.12 | jkroon | feature request: the ability to get the serial number of a card from software! |
| 08:24.18 | shapr | not gonna happen |
| 08:24.30 | shapr | check the invoice, it's usually there |
| 08:25.07 | jkroon | i buy from miro (ZA), it's not. |
| 08:25.15 | shapr | Ah, I've dealt with them. |
| 08:25.19 | jkroon | i'll just log it against one of my many analog cards that is in the DB. |
| 08:25.38 | shapr | Would make our lives easier at Digium if you register the cards when you get 'em :-) |
| 08:26.42 | ChannelZ | I forgot to write down the serial before installing and putting the box into service. I'll get around to taking it down and opening it back up some day... |
| 08:26.57 | jkroon | impact/severity? |
| 08:27.05 | shapr | Once again, *really* makes things smoother if you register when you get 'em! |
| 08:27.36 | jkroon | chapr - makes my life easier too but it gets CRAZY here (three/four new servers going out in a week sometimes) |
| 08:27.48 | ChannelZ | thankfully it's worked flawlessly for 3 years so far (knocks on wood) |
| 08:27.57 | shapr | Yah, I understand, we all do the best we can. |
| 08:28.15 | jkroon | since dahdi 2.4.0 every single dahdi issue i've had was lightning related. |
| 08:28.30 | shapr | hahahaaa |
| 08:28.56 | jkroon | well, other than config/features, but no more random lockups and ports just stopping to function. |
| 08:29.34 | shapr | jkroon: My two favorite pix acquired here at Digium are 1. someone filed a PCI 3v card to fit into a PCI 5v slot and it fried 2. someone's card got thoroughly struck by lightning and they sent a picture of it for RMA |
| 08:30.06 | shapr | oh and the most awesome bonus is the loopback cable a customer had made for him, and he forced them to create in in AN ACTUAL LOOP before he would accept it! |
| 08:30.30 | Godofmonkeys | hahaha |
| 08:31.15 | shapr | jkroon: Yah, 2.4.0 is awesome... that and fxotune can solve most problems. |
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| 08:36.49 | jkroon | hehe, yea, i wrapped fxotune in a parallel variant, so it'll using dahdi_scan find all FXO ports, issue a fxotune for each one, writing the config for each one to /tmp/something and then recombine. |
| 08:37.02 | jkroon | shapr, 00223009 |
| 08:37.21 | jkroon | how do I add files to that? |
| 08:38.01 | shapr | You should get an initial email, reply to that with attached files. |
| 08:42.06 | shapr | looks |
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| 08:52.33 | shapr | jkroon: hey, a parallel fxotune is really cool, do you have that online somewhere? |
| 08:52.40 | *** join/#asterisk Cadey (~Cadey@62.84.178.106) |
| 08:53.16 | Cadey | Hi guys, how would I send the output of a sip debug into a text file so i can leave it running for a good period of time to debug some intermitant issues with media |
| 08:53.42 | Cadey | media setup |
| 08:53.43 | cjk_ | hi, my asterisk 1.8 receives faxes in T.38 mode using the ReceiveFAX application, but it doesnt use T.38 when forwarding to another SIP channel. any idea how i can tell my asterisk to "transcode" between T.38 and G711? |
| 08:54.13 | shapr | Cadey: turn on logging in /etc/asterisk/logger.conf turn on sip debug in the asterisk cli |
| 08:54.30 | Cadey | shapr, thanks :) |
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| 08:54.38 | Godofmonkeys | does that not force a register every 30 seconds or so? |
| 08:54.41 | schmidts | good morning |
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| 08:55.31 | shapr | cjk_: store and forward? |
| 08:55.36 | schmidts | does anyone know which ISDN signaling type is used in China? |
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| 08:56.11 | shapr | jkroon: case creation email bounced :-/ |
| 08:56.13 | cjk_ | shapr, hmm, that would kill those stupid fax reports :) |
| 08:56.33 | shapr | jkroon: uls.co.za bounced it as 550, sender blacklisted |
| 08:56.40 | shapr | I promise you, it's not spam :-) |
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| 08:59.56 | Cadey | shapr : sorry, how do i specify a location to log into, for example /mnt/blah/mylog.txt |
| 09:00.26 | shapr | not sure about that one |
| 09:00.48 | Cadey | i see it logs into syslot.local0 in the logger.conf examples |
| 09:01.30 | Cadey | shapr : seems its as simple as going... |
| 09:01.44 | Cadey | shapr : /mtn/blah/log.txt => debug |
| 09:02.03 | shapr | oh that is easy :-) |
| 09:02.25 | Cadey | ill test it :) |
| 09:02.31 | Cadey | see if it really is that simple |
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| 09:05.38 | jkroon | it is :) |
| 09:06.07 | Cadey | hum |
| 09:06.21 | festr_ | hi, I've upgraded from 1.8 asterisk to today asterisk-trunk and I've problem with H264 video now. Wireshark shows that All RTP packets to video phone from asterisk have MARK. Comparing to asterisk 1.8, MARK is only at the end of NAL frame. Any hints? |
| 09:06.26 | Cadey | its created the file but the sip debug its going into it |
| 09:06.49 | jkroon | you need to send verbose into it to get the SIP debug info. |
| 09:06.57 | Cadey | oh : |
| 09:06.58 | jkroon | i consider that a bug. |
| 09:06.59 | Cadey | :) |
| 09:07.01 | Cadey | my fail |
| 09:07.16 | jkroon | took me half a day to figure that one out. |
| 09:07.29 | jkroon | also prevents me from not sending the sip debug to the cli. |
| 09:07.32 | jkroon | which is annoying. |
| 09:07.49 | Cadey | yeah it is a tad |
| 09:08.05 | Cadey | oh well, i dont need to look at the cli today :) |
| 09:08.06 | Godofmonkeys | does sip debug force a register every 30 seconds or so? |
| 09:09.53 | atiti | does anyone have any good documentation/tutorial on how does channel masquerading work? |
| 09:11.31 | jkroon | Godofmonkeys, no. |
| 09:13.28 | shapr | Godofmonkeys: qualify |
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| 09:14.13 | Godofmonkeys | shapr, it seemed to force a re register every 30 seconds here |
| 09:14.21 | Godofmonkeys | sip debug on from the cli |
| 09:15.28 | shapr | that's weird |
| 09:15.43 | shapr | wait, you mean turning on qualify? what seemed to force a re-register? |
| 09:16.32 | Godofmonkeys | shapr, i say that, it does not seem to be behaving this way now |
| 09:17.02 | Godofmonkeys | looks like 120 seconds now |
| 09:18.37 | Godofmonkeys | shapr, the only command i used was 'sip set debug on' |
| 09:18.46 | Godofmonkeys | ramps up the expiry time |
| 09:18.52 | Godofmonkeys | from 3600 to 120 |
| 09:19.05 | Godofmonkeys | or 30 as i saw before |
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| 09:49.04 | cjk_ | hi, does asterisk 1.8 support T.38 gateway'ing ? |
| 09:49.51 | shapr | jkroon: hoi |
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| 10:03.06 | coppice | cjk_: not by default, but there are patches to add it |
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| 10:20.04 | BlackBishop | [2011-03-30 13:18:20] ERROR[28406]: chan_sip.c:13827 register_verify: 'TCP' is not a valid transport for 'dex'. we only use 'UDP'! ending call. |
| 10:20.09 | BlackBishop | what should I google for ? :/ |
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| 10:23.09 | novacall | hello, im trying to figure out what hardware/software to get to interface my desktop PC with my DECT 6.0 phone wirelessly.. |
| 10:23.55 | novacall | Im wondering if i can use osmocombb with some hardware to act as a client and connect to the dect 6.0 base.. |
| 10:24.25 | BlackBishop | why shouldn't I use tcp for sip ? |
| 10:25.20 | *** join/#asterisk cashback (~cashback@ip68-2-140-46.ph.ph.cox.net) |
| 10:25.35 | zkn | TCP needs to be supported (enabled) at the registrar end |
| 10:27.02 | BlackBishop | ok, and how do I enable that ? it's my asterisk .. |
| 10:27.33 | zkn | depending on the version of Asterisk, you may not be able to enable it |
| 10:27.52 | zkn | look into your sip.conf file |
| 10:32.32 | BlackBishop | 1.8.3.2 |
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| 10:38.24 | cjk_ | coppice, so what can t.38 be used for? just tiff2fax fax2tiff? |
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| 10:55.55 | coppice | just? that is what most people want to do |
| 10:57.09 | no1peanut | Trying to add the repositories for asterisk 1.8 : https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages ... import of gpg key fails - key not found on keyserver |
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| 11:37.08 | zkn | BlackBishop> 1.8.3.2 |
| 11:37.08 | zkn | <PROTECTED> |
| 11:37.08 | zkn | <PROTECTED> |
| 11:37.08 | zkn | <PROTECTED> |
| 11:37.08 | zkn | <PROTECTED> |
| 11:37.09 | zkn | <PROTECTED> |
| 11:37.11 | zkn | <PROTECTED> |
| 11:37.14 | zkn | <PROTECTED> |
| 11:37.16 | zkn | <PROTECTED> |
| 11:37.18 | zkn | <PROTECTED> |
| 11:37.20 | zkn | <PROTECTED> |
| 11:37.22 | zkn | <PROTECTED> |
| 11:37.24 | zkn | <PROTECTED> |
| 11:37.26 | zkn | <PROTECTED> |
| 11:37.30 | zkn | <PROTECTED> |
| 11:37.31 | zkn | umm |
| 11:37.34 | zkn | :D |
| 11:38.08 | zkn | BlackBishop: 1.8.3.2 supports TCP very well |
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| 12:29.10 | Poincare | zkn: you rang my lord? |
| 12:29.54 | zkn | heh.. sorry.. copy-paste accident :) |
| 12:37.03 | Cadey | my lord.... very SME.... |
| 12:37.33 | Cadey | sorry, S&M |
| 12:40.26 | WIMPy | The lord is (for?) S&M? |
| 12:41.44 | pabelanger | no1peanut: $ sudo apt-key adv --keyserver pgp.mit.edu --recv-keys 175E41DF |
| 12:41.46 | pabelanger | fixed wiki |
| 12:43.26 | no1peanut | pabelanger, Perfect :) |
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| 12:58.12 | BlackBishop | zkn: ok, then why do I get that error message ? I mean, I have it enabled in sip.conf tcpenable=yes with tcpbindaddr=0.0.0.0:5060 |
| 12:58.15 | BlackBishop | what else do I have to do ? |
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| 12:59.29 | Dovid | is. is there any way of seeing if a channel exisits ? |
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| 13:00.15 | Dovid | nm. i was lazzzy. google is my friend ;) |
| 13:03.18 | BlackBishop | google is a whore .. she's everybody's friend ... |
| 13:03.27 | BlackBishop | or at least .. should be .. |
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| 13:08.27 | neurosys | aside from static ports, is there any advantage to use IAX over SIP connecting 2 PBXs for intercorporate connectivity? |
| 13:09.24 | WIMPy | Bandwidth |
| 13:13.57 | neurosys | WIMPy: IAX uses less bandwidth? |
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| 13:16.13 | Dovid | ChanIsAvail is only for a resource and not for a channel ? |
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| 13:25.35 | WIMPy | neurosys: It only adds overhead per trunk, not per call. |
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| 13:37.39 | tuxx- | is the domain variable in asterisk 1.8.3.2 inside the sip.conf disabled or something? I tried to register to it when the domain variable was active, but it didnt work. i disabled it now, and it magically works. Can't find any documentation on it whatsoever |
| 13:37.54 | tuxx- | only documentation about < 1.6 |
| 13:45.39 | zkn | BlackBishop: what do you get when you do sip show peer <peer> ? I'm interested in the values of fileds: Prim.Transp. and Allowed.Trsp |
| 13:46.19 | wikki | is anyone here using asterisk for multi tenant conferenc calling? |
| 13:46.38 | wikki | if so are you just using the meetme feature or something else? |
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| 13:55.40 | donttrustem | Hi guy's I have a cisco ip phone 7960 but I don't seem to be able to get it to register to my sip provider? |
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| 13:57.22 | neurosys | WIMPy: sorry. got disconnected. Bandwidth using IAX2 is better? |
| 13:58.20 | BlackBishop | zkn: Allowed.Trsp : UDP |
| 13:58.30 | BlackBishop | <PROTECTED> |
| 13:58.33 | BlackBishop | hmmm |
| 13:58.39 | BlackBishop | so .. where do I change those !? |
| 13:59.19 | zkn | sip.conf |
| 13:59.44 | *** join/#asterisk ihor (~Miranda@194.44.15.90) |
| 13:59.46 | zkn | i'm sure you have transport=udp for peer(s) |
| 13:59.58 | zkn | change that to transport=tcp |
| 14:00.00 | zkn | or transport=tcp,udp |
| 14:02.52 | BlackBishop | ahuh ... |
| 14:02.59 | BlackBishop | tries |
| 14:04.45 | WIMPy | neurosys: It only adds overhead per trunk, not per call. |
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| 14:10.53 | BlackBishop | zkn: thanks :) |
| 14:11.09 | zkn | BlackBishop: wlcm |
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| 14:51.04 | wikki | wow, just found out a new customer of ours paid $180,000 for a phoen system for 40 users |
| 14:51.05 | wikki | :( |
| 14:51.22 | JerJer | i think my rates just went up |
| 14:51.44 | jkroon | ditto. |
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| 14:52.59 | wikki | for a nortel system too :o |
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| 15:34.32 | kuku | I need to check if the caller is a local extenion - how would I do that. |
| 15:35.25 | dschuett | agi script to check callerid |
| 15:36.08 | kuku | a dialplan had an option for that... |
| 15:37.14 | dschuett | ? |
| 15:37.28 | WIMPy | If you don't know that already, you're likely to have some serious security issues. |
| 15:37.38 | WIMPy | You should read the chapter about contexts. |
| 15:38.56 | kuku | I found it exten => callerid/s,1..... |
| 15:39.55 | WIMPy | other way round |
| 15:41.08 | kuku | yes- thank you |
| 15:41.51 | kuku | Whoever wrote this diaplan is a schmuck |
| 15:44.19 | kuku | Its allowing people from the outside, to call in, and then make an outbound call. |
| 15:44.28 | kuku | but they tied it into the default context. |
| 15:44.48 | kuku | They didnt seperate outside and inside dialplans |
| 15:46.43 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
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| 15:55.19 | kuku | Ok. So it seems they are not making inboud call.s |
| 15:55.35 | kuku | So how do I check how this hacker is making calls through thix pbx. |
| 15:55.47 | kuku | As in, where is the the entry point |
| 15:55.59 | kuku | sip show peers doesnt show anyone outside of the local lan |
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| 16:10.16 | kuku | Is there a reason why deny/permit would not be working ? |
| 16:20.16 | Work2Play_ | Hey All - If I wanted something similar to the Cisco 7975 (nice color display, amazing audio and speakerphone, and good feature set) but that played nice with Asterisk and was much easier to configure - what phone would be a good one to look at? |
| 16:21.08 | Work2Play_ | I know I'll want at least 4 line appearances too; the rest are just nice-to-have's for speeddial |
| 16:29.28 | Work2Play_ | It seems pretty pricey, but the Aastra 6739i looks like the best alternative... I just need to find out how hard they are to set up with asterisk |
| 16:29.44 | *** join/#asterisk mawhii (~mawhii@170.220.119.70.cfl.res.rr.com) |
| 16:30.24 | paulc | Work2Play_ I've played with the Aastras a bit, but not that model, and they're pretty nice. But top of my list would be Polycom - they're great phones. |
| 16:30.29 | *** part/#asterisk dschuett (~dschuett@wsip-68-15-229-108.om.om.cox.net) |
| 16:31.27 | Work2Play_ | paulc - good to know... I'm looking at voiplink and they didn't have the executive phones but I'll look elsewhere... I remember seeing them all over the Microsoft campus and they looked nice |
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| 16:42.41 | p3nguin | work2play_: Take a look at the new Cisco 500 series SIP phones. |
| 16:42.59 | Work2Play_ | looking |
| 16:43.09 | p3nguin | work2play_: I think the 525 is the "big" phone. |
| 16:43.53 | Work2Play_ | p3nguin - you're here a lot! Thx for the tip |
| 16:44.42 | p3nguin | I'm also not here a lot, as well. |
| 16:44.52 | Work2Play_ | lol |
| 16:45.34 | Work2Play_ | that's sweet - the 525G2 has vpn - could take that anywhere and hook back into the pbx - that's awesome. How funny would that look at starbucks you plug your own phone into the wall and open for business |
| 16:46.09 | p3nguin | I'm sure it happens often, actually. |
| 16:46.32 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 16:47.00 | nickfennell | Anyone know of any decent VoIP optimisation products |
| 16:47.02 | Work2Play_ | i don't travel otu of country often but I remember going to costa rica on business and relying heavily on my vpn and softphone |
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| 16:57.26 | Godofmonkeys | p3nguin, i must thank you again. you've earned me another 'telecommute' day to sort out the issues with simple signal :) |
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| 16:58.31 | p3nguin | godofmonkeys: I hope everything works out. If there's anything else I can do, let me know and I'll try to accommodate you. |
| 16:58.52 | Godofmonkeys | i have thier clueless tier 3 on the phone now... nobody else left :) |
| 16:59.30 | Godofmonkeys | seems though the from user is not correct, and they tell me that i'm tring to call my did no matter what number i dial |
| 16:59.31 | p3nguin | Isn't tier 3 supposed to be SysAdmin-level support? |
| 16:59.35 | Godofmonkeys | yeh |
| 17:00.59 | p3nguin | I would remove the fromuser setting. Or... take the config I gave you and use that as a template for their service. |
| 17:01.10 | Godofmonkeys | i did :) |
| 17:01.58 | p3nguin | As far as authentication to an ITSP, I use type=peer, username/defaultuser, and a secret. That should be plenty. |
| 17:02.09 | *** join/#asterisk sbruimen (~sbruimen@host10.ripc.redline.ru) |
| 17:03.13 | sbruimen | ÐдÑавÑÑвÑйÑе. ÐÐ¾Ð³Ñ Ð»Ð¸ Ñ Ð¿Ð¾Ð»ÑÑиÑÑ Ð·Ð´ÐµÑÑ Ð¿Ð¾Ð¼Ð¾ÑÑ Ð¿Ð¾ наÑÑÑойке PRI Ð´Ð»Ñ ÐºÐ°ÑÑоÑки DIGIUM ? |
| 17:03.20 | Godofmonkeys | will peer allow me incoming? |
| 17:04.10 | sbruimen | Hi! Can anyone help me to setup Digium card? |
| 17:04.36 | WIMPy | ~ask |
| 17:04.36 | 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. |
| 17:04.59 | WIMPy | sbruimen: The sample configs are a good starting point. Otherwise see above. |
| 17:05.59 | sbruimen | I try to connect Alcatel 4400 and Asterisk via PRI. PRI span is still down. dahdi_tool - OK. Digium TE410P |
| 17:06.45 | *** join/#asterisk timahvo1 (~rogue@41.223.57.73) |
| 17:07.07 | p3nguin | godofmonkeys: Peer allows calls in both directions. User is the only type that is one-way only. |
| 17:07.29 | WIMPy | sbruimen: Do you have the correct crossover cable? |
| 17:08.52 | sbruimen | WIMPy: yes. |
| 17:09.56 | sbruimen | WIMPy: led on 1 port in Asterisk is green. |
| 17:10.00 | WIMPy | sbruimen: What does 'dahdi show status' give you? |
| 17:10.38 | sbruimen | T4XXP (PCI) Card 0 Span 1 OK 0 0 0 CCS HDB3 CRC4 0 db (CSU)/0-133 feet |
| 17:10.53 | sbruimen | spans 2-4 are RED |
| 17:10.57 | WIMPy | Ok, that's a good start. |
| 17:11.31 | WIMPy | Don't see anything wrong so far. |
| 17:12.14 | sbruimen | i check manuals along 3 days (sorry about my english :) ) |
| 17:12.20 | *** join/#asterisk ks3 (~ksandy@74.203.195.1) |
| 17:14.32 | sbruimen | WIMPy: I am noob in E1 connections. I think that somethink is wrong in alcatel ISDN timings or my PRI library on * |
| 17:14.33 | Godofmonkeys | the 604 is gone; seems to be authentication related |
| 17:15.07 | WIMPy | sbruimen: What happens? Or doesn't happen? |
| 17:15.14 | Godofmonkeys | looks like it's on a reinvite |
| 17:15.35 | Godofmonkeys | if they turn off auth it will call out |
| 17:15.44 | sbruimen | WIMPy: can i show you my dmesg and pri debug? |
| 17:16.15 | p3nguin | godofmonkeys: Maybe they gave you the wrong use id or password. |
| 17:16.27 | Godofmonkeys | the register statments go through |
| 17:16.57 | WIMPy | sbruimen: That might hlep. |
| 17:17.07 | WIMPy | Use a pastebin for that. |
| 17:17.08 | p3nguin | godofmonkeys: Yeah, that's the weird part. |
| 17:17.37 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 17:18.47 | WIMPy | sbruimen: And please use pri set debug 2. |
| 17:18.59 | Godofmonkeys | seems i am not passing any credentials, they challenge per call. |
| 17:19.38 | *** join/#asterisk wonderworld (~ww@port-92-201-85-134.dynamic.qsc.de) |
| 17:19.40 | WIMPy | oh |
| 17:19.45 | WIMPy | sbruimen: That might hlep. |
| 17:19.50 | WIMPy | Use a pastebin for that. |
| 17:19.55 | WIMPy | sbruimen: And please use pri set debug 2. |
| 17:22.11 | *** join/#asterisk rkill (~james@nat/digium/x-yiqqfpyttrejhgqj) |
| 17:22.11 | Godofmonkeys | p3nguin, would a packet capture help? |
| 17:23.07 | p3nguin | godofmonkeys: Having the username/defaultuser and secret in the peer should provide the auth when requested. Are you using Dial(SIP/simpsig/${EXTEN}) to make calls through them? |
| 17:23.14 | *** join/#asterisk sbruimen (~sbruimen@host10.ripc.redline.ru) |
| 17:23.35 | Godofmonkeys | Dial(SIP/simpsig/${EXTEN},60) |
| 17:23.43 | p3nguin | same thing |
| 17:23.45 | sbruimen | WIMPy: sorry. |
| 17:23.50 | p3nguin | just with a dial timeout |
| 17:24.03 | Godofmonkeys | ah. didn't know if that would make a difference |
| 17:24.08 | p3nguin | nope |
| 17:24.35 | p3nguin | the ,60 just limits it to 60 seconds before it exits and goes to the next line in the extension. |
| 17:24.44 | Godofmonkeys | ok |
| 17:25.07 | Godofmonkeys | canreinvite goes in general or the simpsig section? |
| 17:25.13 | Godofmonkeys | seems to be listed twice |
| 17:25.15 | p3nguin | both |
| 17:25.18 | Godofmonkeys | ok |
| 17:25.33 | p3nguin | They aren't behind nat, so do you have nat=no in their peer entry? |
| 17:25.53 | Godofmonkeys | yes |
| 17:26.13 | sbruimen | WIMPy: http://217.144.98.133/~sbruimen/phone/log.txt |
| 17:26.23 | p3nguin | You'll have nat=yes in your general, but then you have to override it with nat=no in the peer to turn it back off, otherwise it would be inherited from general. |
| 17:26.49 | Godofmonkeys | as it is |
| 17:27.28 | sbruimen | WIMPy: PRI span 1/0: Provisioned, Down, Active |
| 17:28.38 | WIMPy | sbruimen: You don't receive anything. It the Port on the 44xx configured? |
| 17:30.54 | sbruimen | WIMPy: wat does it mean? how i can check it? board is "in service". All settings are as in manual from voip-info.org |
| 17:32.00 | WIMPy | sbruimen: Has that port been working before? |
| 17:32.00 | Godofmonkeys | p3nguin, failed to authenticate on invite |
| 17:32.51 | sbruimen | WIMPy: TE410p? or PRA2? |
| 17:33.07 | WIMPy | PRA2 |
| 17:33.57 | sbruimen | WIMPy: yes. this board has been working before |
| 17:34.53 | sbruimen | WIMPy: with our local E1 provider |
| 17:35.59 | sbruimen | WIMPy: i can try to use another PRA2 board. do you mean than my current PRA2 board is broken? |
| 17:36.35 | WIMPy | No, but I wonder if it's active. |
| 17:37.22 | sbruimen | WIMPY: | 1 | 9 | PRA2 | IN SERVICE| |
| 17:37.23 | *** join/#asterisk chasing`Sol (~cS@smtp.master-zone.net) |
| 17:37.59 | WIMPy | Anythning more you can get out of it? |
| 17:38.25 | sbruimen | <PROTECTED> |
| 17:38.25 | sbruimen | <PROTECTED> |
| 17:38.25 | sbruimen | <PROTECTED> |
| 17:38.25 | sbruimen | <PROTECTED> |
| 17:39.04 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 17:39.16 | WIMPy | How can it be busy? And what does (Master) mean? |
| 17:39.37 | sbruimen | WIMpy: i dont know :) |
| 17:40.35 | sbruimen | WIMpy: trkstat show me one of 30 channels as bysy, when i try call extension routed via this board trunk |
| 17:40.56 | sbruimen | busy |
| 17:42.17 | WIMPy | Are you sure the cable is ok? Have you tried it the other way round? |
| 17:42.41 | *** join/#asterisk atiti (~atiti@0103ds1-vir.0.fullrate.dk) |
| 17:42.55 | Godofmonkeys | said there is not an authentication srting in the invite |
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| 17:43.47 | sbruimen | WIMPy: im no 100% shure, but this cable work fine some time ago |
| 17:45.21 | WIMPy | Just thought that maybe it's only working one direction. |
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| 17:46.17 | cypher101 | any asterisk beginners in here , that want to group together ? :) |
| 17:46.40 | madduck | is it possible to initiate a meetme conference from the CLI? |
| 17:46.54 | madduck | i need to set one up to be used while I am offline |
| 17:47.50 | sbruimen | WIMPy: ohh! it is right, that connection would not established, and i must check my cable and PRA2 board? Why TE410p led is green? NO-signal led on PRA2 board is off |
| 17:48.50 | WIMPy | Well, if both ends have a signal, the cable must be ok. |
| 17:48.52 | Godofmonkeys | is there a way to pass the secret in the invite? |
| 17:55.18 | sbruimen | WIMPy: do you know, how i can debug this connection on alcatel side? |
| 17:56.16 | WIMPy | sbruimen: Sorry, no idea. |
| 18:01.31 | *** join/#asterisk dr__ (~duckz@78.96.101.150) |
| 18:05.41 | sbruimen | WIMPy: thanks. |
| 18:08.02 | *** join/#asterisk kaldemar (~kaldemar@unaffiliated/kaldemar) |
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| 18:09.11 | p3nguin | godofmonkeys: Asterisk is supposed to do that any time it is challenged. A typical invite will involve asking, being rejected and challenged, then providing auth credentials. |
| 18:09.59 | Godofmonkeys | so would there be a reason that fails? |
| 18:10.14 | Godofmonkeys | or atleast where i should look? |
| 18:10.42 | p3nguin | What version of Asterisk are you using? |
| 18:10.46 | Godofmonkeys | 1.6 |
| 18:10.52 | p3nguin | 1.6 is not a version |
| 18:10.56 | p3nguin | It's not even a branch. |
| 18:11.10 | Godofmonkeys | what command to ask it? |
| 18:11.17 | p3nguin | core show version |
| 18:11.46 | Godofmonkeys | Asterisk 1.6.2.11 built by root @ localhost.localdomain on a i686 running Linux on 2010-08-24 20:43:18 UTC |
| 18:12.58 | p3nguin | That's kind of old. Eight months... lots of things were fixed in eight months. |
| 18:14.43 | p3nguin | I would upgrade to 1.6.2.17.2 and see how that works out. |
| 18:15.38 | *** join/#asterisk Nombrandue (~Satan@ip174-71-68-157.om.om.cox.net) |
| 18:17.10 | Godofmonkeys | updating now |
| 18:18.11 | p3nguin | Back up your configs! |
| 18:19.25 | Godofmonkeys | of course :) |
| 18:19.59 | p3nguin | I never assume people back up important files. |
| 18:20.20 | Godofmonkeys | 15 years as a compiuer tech has taught me some hard lessons |
| 18:20.38 | Godofmonkeys | computer, as it is... |
| 18:20.53 | Godofmonkeys | guess it doesn't mean i can type |
| 18:21.48 | p3nguin | I still never assume people back up anything important, regardless of their proclaimed qualifications. |
| 18:22.43 | Nombrandue | I am having some strange RTP issues, but only from my VOIP provider and only for inbound. Asterisk runs noramally, sends out packets sip and RTP properly, and well formed SIP, but my provider (Broadvoice) kicks back that the peer rtp port it request, is unreachable. Outbound works perfectly though |
| 18:23.04 | p3nguin | As a matter of fact, I don't even assume *I* created a backup plan for things important -- I'm always going back to check that I'm backing up things just to find that I've already done it. |
| 18:23.17 | Godofmonkeys | i know the feeling |
| 18:23.53 | p3nguin | I also run into people who should know better who think RAID is acceptable as a backup. |
| 18:24.03 | Godofmonkeys | lol |
| 18:24.22 | p3nguin | That's when I show them that rm -f does, even on RAID. |
| 18:24.26 | Godofmonkeys | hahahah |
| 18:24.37 | p3nguin | s/m that/m what/ |
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| 18:25.54 | p3nguin | I admit that I put a lot of trust into RAID, but it's certainly no backup. |
| 18:26.19 | Godofmonkeys | indeed, gone is gone, if it's 1 drive or 200 |
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| 18:30.53 | Godofmonkeys | p3nguin, i have done the update, and it's still fail to auth |
| 18:32.21 | p3nguin | /: |
| 18:33.17 | Godofmonkeys | any way to force it in the dialplan? |
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| 19:12.35 | jaytee | anyone using Fail2Ban in here? |
| 19:14.12 | jaytee | I'd like to put an entire subnet range on the ignoreip line in /etc/fail2ban/jail.conf file but I can't find anything in the documentation or the FAQ to corroborate whether that is acceptable or if I have to list out all the IP addresses of the phones. |
| 19:15.46 | jaytee | ah, nevermind. found a good link about it. |
| 19:19.22 | p3nguin | ignoreip = 192.168.0.0/16 |
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| 19:24.37 | jaytee | p3nguin, thanks. I just found an example but it had the explicit mask syntax 192.168.1.0/255.255.255.0 and I'd rather do 192.168.1.0/24 instead |
| 19:30.02 | p3nguin | Slash notation certainly is neater and requires less typing, but functionally it is exactly the same. |
| 19:30.14 | fauxalliance | ;) |
| 19:31.02 | psilikon | Verizon fios westell 9100 is so lame and I think it is mangling my voip packets |
| 19:59.00 | *** part/#asterisk vrtigo1 (~fredw@vpn.lpga.com) |
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| 20:00.21 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 20:21.09 | Godofmonkeys | damn. now my incooming section is wrong. |
| 20:23.24 | Godofmonkeys | anyone feel like a quick look? http://pastebin.com/PtxcNM4V |
| 20:23.54 | Qwell | 1) What is it doing? |
| 20:24.01 | Qwell | 2) Why are you using priority 1 twice? |
| 20:24.11 | Qwell | 3) What is Log()? |
| 20:24.58 | Godofmonkeys | 1) tells me the did my provider gave me is not found in extensions.conf |
| 20:25.00 | Qwell | the same thing as Verbose(), apparently. Weird. That seems completely useless. |
| 20:25.08 | Qwell | So then why are you using s? |
| 20:25.15 | Godofmonkeys | 2) that was the example i found |
| 20:26.04 | Godofmonkeys | again, suggested to me by the 'real world example' i found |
| 20:26.13 | Qwell | ~book |
| 20:26.13 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 20:26.17 | Qwell | read about extensions |
| 20:26.43 | Godofmonkeys | have that bookmarked |
| 20:27.11 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 20:32.26 | eduzimrs | anyone here got experience with RedPhone Phone Bridge ? |
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| 20:35.05 | JerJer | eduzimrs: quite a awhile back i played with them |
| 20:35.19 | JerJer | may not even be the same hardware any more |
| 20:36.00 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 20:36.29 | eduzimrs | have you ever run an tcpdump at its port? That "explosions" of packages are normal ? they make no sense to me |
| 20:40.56 | JerJer | hmm... don't believe i tried that |
| 20:42.44 | eduzimrs | JerJer i dont know it that is normal |
| 20:42.52 | eduzimrs | JerJer i dont know if that is normal |
| 20:44.16 | Godofmonkeys | Qwell, updated the dialplan, and have included the debug with verbosity 4: http://pastebin.com/h3VbDNWg |
| 20:45.10 | Qwell | Godofmonkeys: and what is SIP/100? |
| 20:45.27 | Godofmonkeys | my extension |
| 20:45.33 | Qwell | show me |
| 20:45.36 | Godofmonkeys | for my softphone |
| 20:46.18 | Godofmonkeys | the entire extensions.conf : http://pastebin.com/wRz5uJcz |
| 20:46.44 | Qwell | and [100] in sip.conf? |
| 20:48.23 | Godofmonkeys | http://pastebin.com/JgwVRJWA |
| 20:48.56 | Qwell | doesn't look like a softphone to me |
| 20:49.14 | Godofmonkeys | it works |
| 20:49.28 | Godofmonkeys | can do echo test, and ring another extension i had |
| 20:49.50 | Qwell | but that extension can't be called by Asterisk |
| 20:49.55 | Qwell | rather, that phone |
| 20:50.06 | Godofmonkeys | so the error is in sip.conf |
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| 21:11.29 | *** join/#asterisk Schreiber1337 (cee4b465@gateway/web/freenode/ip.206.228.180.101) |
| 21:13.09 | Schreiber1337 | Trying to setup BLF on an extension, but when I put a "hint" into the extensions context I get "Auto fallthrough, channel 'SIP/4325-00000380' status is 'UNKNOWN'" any ideas? |
| 21:15.24 | *** join/#asterisk raden_work (~jon@66-191-96-74.static.eucl.wi.charter.com) |
| 21:16.08 | eduzimrs | Schreiber1337 how is your syntax at extensions ? |
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| 21:22.56 | Schreiber1337 | http://dl.dropbox.com/u/7097983/1.txt |
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| 21:27.44 | Jasnejac | Schreiber1337: there's no priority number in your jint line |
| 21:28.09 | Qwell | Jasnejac: That would be because it's a hint. |
| 21:28.37 | Qwell | I do question how the n priority would work, with it being after hint though. |
| 21:29.15 | Qwell | dialplan show 4326@internal will probably show some funkiness, like a priority of 0 |
| 21:30.26 | Jasnejac | why is there no prority to hints? anyone any idea? doesn't seem to make sense |
| 21:30.33 | Qwell | hint IS the priority |
| 21:30.53 | Jasnejac | and it fits into the schema where? |
| 21:31.56 | eduzimrs | Jasnejac probably he doesnt put the context at "subscribecontext" field. |
| 21:34.14 | Jasnejac | I'm struggling to understand the logic of hints. as far as I am concerned they are a singal, formed in one way or another depending on the endpoint. I don't understand why they should need to be in any way different to any other dialplan application |
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| 21:34.20 | Charrit | hi |
| 21:35.10 | Godofmonkeys | Qwell, i have made some changes, if you have the time. the call just gets hung up now, but there is a delay. http://pastebin.com/xuVpuJvB |
| 21:35.12 | Qwell | Jasnejac: The only time you would ever use a Hint() application is in priority 1. |
| 21:35.48 | Charrit | Is it possible to include channel variables into cdr report? |
| 21:36.23 | Jasnejac | Qwell: why so? you may not have decided where to send the call at that point |
| 21:36.34 | Qwell | Jasnejac: It has nothing to do with sending calls. |
| 21:36.59 | Jasnejac | ok, what has it to do with then? I am really confused here |
| 21:37.09 | Qwell | subscribing to device state |
| 21:38.24 | Jasnejac | so you are trying to see if a device is available? This is a lack of PSTN type knowledge on my part I feel |
| 21:39.07 | file | a hint is a mapping of a dialplan extension to a device, because if your phone asks to be notified of the state of an extension, for example 100, Asterisk has to know what device that extension represents |
| 21:39.14 | file | it could be a SIP phone named bob, or something else |
| 21:39.32 | *** join/#asterisk Schreiber1337 (cee4b465@gateway/web/freenode/ip.206.228.180.101) |
| 21:39.49 | Schreiber1337 | Hello all... |
| 21:39.51 | Jasnejac | ah, OK - legacy hardware device stuff. makes much more sense now but the implementation still doesn't! thank you |
| 21:40.07 | Qwell | Jasnejac: no, not legacy hardware |
| 21:40.07 | file | not legacy hardware device stuff |
| 21:40.51 | citywok | points ICBM @ bandwidth.com headquarters. |
| 21:40.52 | Jasnejac | soeey guys but yes it is. its a hangover from PSTN. it will change in a while when all these things are just software signals. |
| 21:40.53 | Schreiber1337 | Can someone assist with setting BLF... when I execute the "hint" statement I get an error... here is an example http://dl.dropbox.com/u/7097983/1.txt |
| 21:41.20 | file | blinks |
| 21:41.48 | Qwell | it has nothing to do with hardware or software |
| 21:42.37 | Qwell | Schreiber1337: dialplan show 4326@internal will probably show some funkiness, like a priority of 0, because of where your n priority is. |
| 21:44.55 | Schreiber1337 | @Qwell: So should I change 4326,n,Macro... to 4326,2,Macro... |
| 21:45.11 | Schreiber1337 | @Qwell: or am I confused |
| 21:45.21 | Qwell | Schreiber1337: That would be a test. Really, just moving the location of the n priority to above the hint would do it. |
| 21:45.40 | Qwell | dialplan show 4326@internal |
| 21:45.46 | Qwell | run that from the Asterisk CLI. |
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| 21:48.50 | Schreiber1337 | @Qwell: Shouldn't it show the hint somewhere? http://dl.dropbox.com/u/7097983/2.txt |
| 21:49.23 | Qwell | Schreiber1337: no, those get parsed elsewhere. core show hints, I believe |
| 21:50.50 | Schreiber1337 | @Qwell: Hmm... "There are no registered dialplan hints " and the Sidecars that I have setup to watch this extension are not seeing anything anymore... which is what I'm really trying to achieve. |
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| 21:52.13 | Schreiber1337 | @Qwell: I have the Sidecar keys set to "fnc=blf+sd+cp;sub=4326@$proxy;ext=4326@$proxy" and they were working, but calls were failing.. now call goes through, but no hint stored. |
| 21:52.20 | Charrit | if I make: exten => 600,n,Set(CDR(mycolumn)=myvalue) |
| 21:53.02 | Charrit | Will I be able to use CDR(mycolumn) in cdr_custom.conf for adding a custom field to Master.csv |
| 21:53.09 | Charrit | ? |
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| 22:01.30 | Schreiber1337 | @Qwell: Success! I had to move the "hint" line back before the Macro and make the Macro 2 instead of n... Thanks so much for pointing me in the right direction! |
| 22:01.51 | Qwell | How bizarre. That shouldn't be necessary. |
| 22:02.49 | Schreiber1337 | @Qwell: It works and I'm happy... frekin Cisco sidecar has been kicking my butt for a few hours... |
| 22:02.58 | Qwell | oh, that explains it then |
| 22:03.11 | Schreiber1337 | @Qwell: LOL |
| 22:04.36 | Jasnejac | Qwell: it isn't logical. Hints should be set by a dialplan application or moved to another config file that is processed separately IMHO. Then they make sense! |
| 22:05.04 | Jasnejac | however, I suspect I will be fighting a stream here |
| 22:05.12 | file | dialplan applications execute within the context of a call |
| 22:05.16 | file | hints exist outside the context of a call |
| 22:05.40 | Jasnejac | then they should not be included in a dialplan |
| 22:06.05 | Jasnejac | if they are different, keep them separate |
| 22:07.36 | Schreiber1337 | Jasnejac: OK... Please suggest what I should do differently... here is my sample config http://dl.dropbox.com/u/7097983/1.txt |
| 22:09.21 | Jasnejac | Schreiber1337: that's what you had before. sorry but I'm not looking at your problem directly, more the reason that it arises in the first place. since it now works and there was no logical way to know what to do, apparently, I think your problem supports my case |
| 22:10.34 | Schreiber1337 | Jasnejac: Should I have a separate context that just does all my "hint" statements and call it from "subscribecontext" in sip.conf? |
| 22:10.59 | Qwell | Schreiber1337: That is typically how people do it. |
| 22:12.05 | Schreiber1337 | @Qwell: do hint statements have to be called for each extension, or can I use a macro or just variables? |
| 22:12.29 | Qwell | I do not believe that you can use variables. Macros definitely not. |
| 22:12.48 | Qwell | Variables *might* work in 1.8, but it wouldn't anywhere else. |
| 22:12.59 | Schreiber1337 | @Qwell: OK... I'll give it a try... Thanks for everyone's help! |
| 22:15.20 | Jasnejac | is there anywhere anyone actually has a philosophical discussion about asterisk? |
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| 22:16.34 | Jasnejac | that sounds wrong - sorry - but is there anywhere where people actually talk about it rather than just solve problems, other than the email list? |
| 22:20.03 | Work2Play_ | anyone around today with experience getting a Cisco 7975 working via SIP? I got SIP loaded on it but it's not taking my config file - and all my google searches are turning up info from 2-3 years ago |
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| 22:26.56 | Schreiber1337 | Jasnejac: I think there is a developers channel for contributers |
| 22:29.15 | Jasnejac | Schreiber1337: I've moniterd it for ages, There's also a developers conference call that is open. I really feel there needs to be a philosphy shift but maybe that will only happen in freeswtich or opensips or kamailio, I hope not. They're pretty much alreadu there |
| 22:30.55 | Qwell | What kind of philosophy shift? |
| 22:32.28 | Jasnejac | away from pure telephont interfaces. the problem is that is unlikely to happen. the strength of asterisk among the other contenders for open-source defacto choice is the media capabilities. tropo addresses that simply and cheaply. |
| 22:32.51 | Qwell | who says Asterisk only supports "telephony interfaces"? |
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| 22:33.50 | Jasnejac | therefore the pstn interface becomes the king driver - viz Digium and Sangoma and others. The other choice is freeswitch but it has poor media capabilities, |
| 22:34.06 | Qwell | Because Asterisk doesn't support a myriad of other protocols? |
| 22:35.39 | Jasnejac | I know asterisk supports loads of stuff but SIP is where it is all at and, like it or not, is like the VHS/betamax stuff from many years ago. There will be an interface need for many years, sure, but as an example my local exchange is 100% digital. If they would let me I could have a cat 6 cable run into my house |
| 22:36.04 | Qwell | Are you suggesting Asterisk doesn't support SIP? |
| 22:36.57 | Jasnejac | lol, no of course not. I have run 250,000 SIP calls through asterisk boxes in the last two weeks. |
| 22:37.05 | Qwell | So then what is your point, exactly? |
| 22:37.09 | Jasnejac | whay I don't know is where SCF fits in all of this |
| 22:37.22 | Jasnejac | what* |
| 22:38.38 | Jasnejac | I'm looking for an argument that says stay with asterisk I guess. its what I'm happy with in most ways but I have to decide the way to go and I can't see asterisk as it at the moment. |
| 22:38.53 | Qwell | You still haven't presented an argument. |
| 22:41.31 | Jasnejac | an argument for what? moving away? that's easy - the software is illogical and does not do what it says on the tin, or at least all documentation. staying? equally easy - it works, it's supported and people care |
| 22:41.57 | Qwell | Point out a place where the documentation is incorrect. |
| 22:42.14 | *** join/#asterisk ketas-av (~ketas@kvlt.eu) |
| 22:42.28 | Qwell | The "documentation" he was using was "random samples I found somewhere on the Internet". That's hardly a problem we can fix. |
| 22:43.43 | Qwell | And saying something is "illogical" often means "I don't understand what it is doing". |
| 22:43.46 | Jasnejac | I'm looking for an argument that says stay with asterisk I guess. its what I'm happy with in most ways but I have to decide the way to go and I can't see asterisk as it at the moment. |
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| 22:44.38 | Jasnejac | I typed a long reply there and it disappeared! doh. probably best |
| 22:45.23 | Jasnejac | ok - illogical - not comfoming to nnormal or reasonable standards of linearity of thought or misunderstood. |
| 22:45.43 | Qwell | Give an example of the former definition. |
| 22:49.23 | Jasnejac | sory, in what terms? with asterisk? the dialplan load order can be circumvented and completely defeated by including every context in every other context. That is not logical or understandable and the performance in such circumstances is unknown. Therefore the state engine is wrong. |
| 23:12.14 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 23:12.14 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.3.2 (2011/03/17), 1.6.2.17.2 (2011/03/17), 1.4.40 (2011/02/28), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
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| 23:30.36 | aschneiderg | Hello guys. I have a XP100 clone hooked to my Centos Asterisk 1.6 test server. My analog line works fine until I start dahdi and asterisk services. Then it appears to be ocupied with no dial tone. I will appreciate if you point me to somewhere I can learn to solve this. |
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| 23:37.57 | ariel_ | hello folks |
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