| 00:03.31 | *** join/#asterisk petris (~petris@2607:5300:60:5475:a9d7:583:99b:a901) |
| 00:13.58 | *** join/#asterisk chuckf (~chuckf@fedora/chuck) |
| 00:15.14 | *** join/#asterisk insha (~insha@mobile-166-137-185-188.mycingular.net) |
| 00:15.25 | *** join/#asterisk petris (~petris@2607:5300:60:5475:a9d7:583:99b:a901) |
| 00:18.31 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 00:26.27 | *** join/#asterisk bmurt (~brendan@64-121-18-195.c3-0.upd-ubr1.trpr-upd.pa.cable.rcn.com) |
| 00:27.56 | *** join/#asterisk petris (~petris@2607:5300:60:5475:a9d7:583:99b:a901) |
| 00:32.27 | *** join/#asterisk jpoz (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
| 00:34.13 | *** join/#asterisk jpoz_ (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
| 00:37.10 | *** join/#asterisk danitetus (~danitetus@90.Red-83-45-47.dynamicIP.rima-tde.net) |
| 00:42.38 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 00:42.44 | *** join/#asterisk chuckf_ (~chuckf@fedora/chuck) |
| 00:49.34 | *** join/#asterisk fidividi (fidividi@95.146.181.125) |
| 00:49.39 | fidividi | Hello |
| 00:49.56 | fidividi | I need some advice regarding my setup and issue with peers being unstable |
| 00:50.01 | fidividi | can anyone help please? |
| 00:52.11 | WIMPy | ~ask |
| 00:52.11 | 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. |
| 00:52.59 | fidividi | I moved from one datacenter to another, and all my sip peers/trunks/extensions keep going unreachable/lagged |
| 00:54.26 | fidividi | public ip address, same exact machine, i moved the VM from one XenServer to another |
| 00:55.37 | fidividi | <PROTECTED> |
| 00:56.15 | fidividi | on the xenserver, I am using network-routed method to use my additional subnet of IPs, and the VM is part of it |
| 00:57.16 | *** join/#asterisk chuckf (~chuckf@fedora/chuck) |
| 00:58.34 | *** join/#asterisk jsjc (~Adium@134.Red-83-60-134.dynamicIP.rima-tde.net) |
| 00:59.25 | fidividi | by network-routed xen, this is what i mean: http://kb.softescu.ro/server-administration/how-to-create-a-subnet-ipv4-for-the-vm-inside-xen-server/ |
| 01:09.37 | *** join/#asterisk edong23 (~quassel@mptc-dhcp-50-220.mptelco.com) |
| 01:11.48 | *** join/#asterisk chuckf (~chuckf@fedora/chuck) |
| 01:17.49 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 01:35.20 | woleium | lo peeps :) |
| 01:35.40 | woleium | does anyone happen to know if * supports far end camera control? |
| 01:36.25 | woleium | Specifically, I have a couple of polycom HDX units that support H.281/H.224 and H.323 Annex Q |
| 01:36.41 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 01:36.42 | woleium | all of which have been eaten by H.323 |
| 01:37.41 | woleium | oh, actually it seems just annex Q was. |
| 01:37.45 | WIMPy | If that's a RTP thing, it might just not care. It it's messages, I wouldn't have any hope. |
| 01:37.59 | woleium | that's what I thought too |
| 01:38.08 | woleium | but no dice. |
| 01:38.18 | woleium | I've not done a full debug capture ywt |
| 01:38.25 | woleium | is being lazy |
| 01:38.54 | woleium | thought I'd ask first - polycom hdx must be pretty popular |
| 01:40.01 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 01:40.10 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 01:41.35 | *** join/#asterisk chuckf (~chuckf@fedora/chuck) |
| 01:56.02 | *** join/#asterisk chuckf (~chuckf@fedora/chuck) |
| 02:04.14 | *** join/#asterisk litn (~blice@alrig.ht) |
| 02:08.33 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
| 02:09.38 | *** join/#asterisk D30 (~deo@203.177.9.66) |
| 02:24.47 | *** join/#asterisk chuckf (~chuckf@fedora/chuck) |
| 02:34.55 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 02:39.17 | *** join/#asterisk chuckf (~chuckf@fedora/chuck) |
| 02:43.12 | *** join/#asterisk wolrah (~wolrah@24.239.210.140) |
| 02:44.22 | *** join/#asterisk bkruse (~Adium@24.42.229.8) |
| 02:51.22 | *** join/#asterisk makubi (~makubi@xdsl-78-35-94-229.netcologne.de) |
| 03:06.14 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 03:07.33 | *** join/#asterisk Nemus (~Nemus@unaffiliated/nemus) |
| 03:08.03 | *** join/#asterisk chuckf (~chuckf@fedora/chuck) |
| 03:20.12 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 03:22.35 | *** join/#asterisk chuckf (~chuckf@fedora/chuck) |
| 03:24.55 | *** join/#asterisk raspberrypifan (~textual@71-22-220-224.gar.clearwire-wmx.net) |
| 03:28.33 | raspberrypifan | so is google voice completly dead now? |
| 03:35.16 | *** join/#asterisk tuxd00d (~tuxd00d@ip24-251-34-116.ph.ph.cox.net) |
| 03:40.22 | ghoti | raspberrypifan: I think as of the 15th. |
| 03:40.57 | raspberrypifan | hm |
| 03:41.15 | raspberrypifan | well i just got the incredible pbx on the pi and i wanna test try it |
| 03:41.26 | raspberrypifan | but u need a new account on there i think, but is it worth if there gonan takae it down in like 3 days |
| 03:42.14 | ghoti | can incredible pbx work with a normal SIP provider? |
| 03:42.19 | ghoti | there are plenty of those around... |
| 03:42.35 | raspberrypifan | i suppose but i want a free one to leanr how to use it |
| 03:43.31 | ghoti | And then you'd pay for service? Or do you just want stuff for free regardless? |
| 03:43.57 | raspberrypifan | im not sure, if i find asterisk useful then maybe |
| 03:44.20 | ghoti | There may be providers who give you a discount, perhaps even a free month, if you sign up for a period like a year. |
| 03:44.42 | ghoti | But I doubt you'll find a lot of companies who are keen on the idea of giving you something for nothing in return. |
| 03:44.48 | ghoti | That kind of thing puts a company out of business. |
| 03:45.11 | raspberrypifan | well what about this iptel thing |
| 03:45.14 | raspberrypifan | i had one of thsoe for a while |
| 03:45.17 | raspberrypifan | but i dont remember what it did anymore |
| 03:45.48 | *** join/#asterisk roentgen (~none@openvpn/community/support/roentgen) |
| 03:46.35 | ghoti | I don't know anything about iptel. |
| 03:47.00 | ghoti | But if you're serious about wanting to learn, why don't you just set up *two* asterisk boxes, and figure out how to have them call each other? |
| 03:47.34 | ghoti | Make one register with the other. Then get a couple of softphones, connect one to each VM, and see if you can have the phones call each other. |
| 03:47.44 | raspberrypifan | hmm id need anotehr phyiscal object to run asterisk i guess, and i cant seem to get it to run on a VM on mac, only the pi one is working |
| 03:47.50 | raspberrypifan | also i have no idea about sip trunks |
| 03:47.51 | raspberrypifan | or any of that |
| 03:47.55 | raspberrypifan | so id like to learn how to make that work |
| 03:48.00 | ghoti | Seems like a great learning experience, costs nothing, and doesn't cause some other company to bleed for your education. |
| 03:48.50 | ghoti | Setting up a sip trunk in Asterisk is seriously a no brainer. When you need it, buy the service, then spend 5 minutes googling example configuration. |
| 03:49.00 | raspberrypifan | hm |
| 03:49.05 | ghoti | In many cases, the sip trunk provider will tell you exactly what to add to your asterisk configuration anyway. |
| 03:49.12 | raspberrypifan | hm |
| 03:49.21 | raspberrypifan | well where cna i set up a second asterisk box at |
| 03:49.31 | ghoti | What VM software are you running? |
| 03:49.56 | ghoti | I set up multiple FreePBX Distro VMs in VirtualBox a while back, for testing. |
| 03:50.08 | raspberrypifan | ive tried two, virtualbox and vmware fusion and they both crap up no matter what version of asterisk or freepbx i try |
| 03:50.10 | raspberrypifan | they wont boot |
| 03:50.11 | raspberrypifan | at all |
| 03:50.30 | ghoti | Your experience appears to be different from mine. |
| 03:51.01 | ghoti | FreePBX is just centos. It runs on the most basic of hardware. |
| 03:52.00 | ghoti | If you're having trouble getting virtualbox running, there is a #vbox channel where folks can help. |
| 03:52.01 | raspberrypifan | what iso did u use on what vm |
| 03:52.11 | raspberrypifan | i know how to use VB for other things |
| 03:53.48 | ghoti | I tried both the PIAF-Green OVA to test an appliance, and FreePBX-5.211.65-11-x86_64-Full-1397332174.img to test the install process. Both worked flawlessly. |
| 03:54.08 | raspberrypifan | whats an appliance exactly? |
| 03:54.25 | ghoti | a pre-built VM, dedicated to one task. |
| 03:54.51 | raspberrypifan | hm would u recommen i download the green or the freepbx.img |
| 03:54.52 | ghoti | PIAF has problems. If you want a gui, FreePBX Distro is probably the way to go. |
| 03:55.38 | ghoti | If you don't need the GUI, just grab a CentOS install image and DIY from packages. |
| 03:55.53 | raspberrypifan | probably freepbx for now |
| 03:56.08 | raspberrypifan | how do you boot with a .img |
| 03:57.25 | raspberrypifan | can you use it like a .iso |
| 03:57.38 | *** join/#asterisk [[thufir]] (~thufir@S0106c8fb2677e8ca.vs.shawcable.net) |
| 04:00.14 | raspberrypifan | ghoti you still around |
| 04:01.34 | ghoti | Yup |
| 04:02.04 | ghoti | I'm trying to remember what I did. I either treated it like an ISO, or I used an alternate boot loader to get it in. |
| 04:02.39 | ghoti | Google "virtualbox boot usb" and check out some of the links. VBox doesn't support booting from USB directly, but people have developed workarounds (as they always do). |
| 04:03.12 | raspberrypifan | well aparently u can convert a .img to a .vdi |
| 04:03.14 | raspberrypifan | for direct use |
| 04:03.16 | raspberrypifan | in vb |
| 04:03.22 | ghoti | But you don't need to use the USB image. http://schmoozecom.com/distro-download.php includes ISOs. |
| 04:04.25 | ghoti | Come to think of it, that's what I used in the VM. But I also downloaded the USB image when I eventually installed it on hardware. |
| 04:04.36 | raspberrypifan | what did u use? |
| 04:04.41 | raspberrypifan | one of one from that page? |
| 04:04.44 | ghoti | yes |
| 04:05.06 | raspberrypifan | if im gonna virtualize it does it matter 32 or 64 |
| 04:05.07 | raspberrypifan | ? |
| 04:06.21 | ghoti | I suspect it doesn't matter much; I can't see that differences between 32 and 64 bit emulation would impact performance enough to affect your ability to learn how the tools work. |
| 04:06.34 | *** join/#asterisk chuckf (~chuckf@fedora/chuck) |
| 04:06.54 | raspberrypifan | right, so do u by any chance know how one would set up an ITSP to provide SIP trunking |
| 04:06.55 | ghoti | Note that if you go with FreePBX, if you stick with the GUI, you'll never really learn about Asterisk. |
| 04:08.44 | ghoti | raspberrypifan, a quick google search gave me this: http://www.freepbx.org/forum/freepbx/users/how-to-setup-a-sip-trunk-between-2-local-asterisk-servers-same-lan-no-nat-no-fir |
| 04:08.55 | ghoti | I can google more things for you if you like. :) |
| 04:09.19 | [TK]D-Fender | ghoti: Careful... that's my bag as #asterisk's resident Google-proxy :p |
| 04:09.46 | raspberrypifan | lol |
| 04:09.53 | ghoti | Heh! |
| 04:09.54 | raspberrypifan | well this is besides the first question |
| 04:09.59 | raspberrypifan | what hardware do u need to set up an ITSP |
| 04:10.24 | [TK]D-Fender | Depends. |
| 04:10.26 | ghoti | If you really want to provide a gateway to the PSTN, you need whatever hardware will connect you to the PSTN. |
| 04:10.47 | ghoti | But there are so many options... |
| 04:11.48 | raspberrypifan | hm |
| 04:12.02 | raspberrypifan | well this is a question for a country where SIP does not yet exist |
| 04:12.06 | ghoti | At the small scale, you could get an FXS device to talk to a POTS line. At the larger scale, you can get ISDN PRI cards from Digium, Sangoma and others. |
| 04:12.30 | raspberrypifan | how do you get the DIDs? |
| 04:12.51 | ghoti | That's up to the local regulators in that country. |
| 04:13.22 | ghoti | Is this hypothetical country open to the concept of having new competition in the phone industry, or is its phone company state-run? |
| 04:14.48 | raspberrypifan | well, im not sure where things stand now. Up until recently the only phone company was the state run one. Now the cable provider does give you phone service. |
| 04:15.53 | ghoti | And they're *not* state run? |
| 04:17.07 | raspberrypifan | the cable provider isnt |
| 04:21.05 | *** join/#asterisk chuckf (~chuckf@fedora/chuck) |
| 04:32.30 | ghoti | well, you probably approach the government regulator and ask them what the process is. Every country is different. Here in Canada, the CRTC makes new blocks of numbers available as new area codes come in to play. |
| 04:33.06 | raspberrypifan | hm so the government would have to say this block of numbers can be used by y'all, then how do u actually connect the numbers to the equipment? |
| 04:33.43 | [TK]D-Fender | Buy equipment. Hook into upstream telco equipement. |
| 04:33.56 | ghoti | ^. You maintain trunks with other providers. |
| 04:34.09 | ghoti | It's not cheap. And neither is the expertise that will set it up and maintain it for you. |
| 04:36.15 | raspberrypifan | hmm |
| 04:36.20 | raspberrypifan | im not sure i understand lol |
| 04:38.12 | ghoti | Okay, if you set yourself up as a telco, you might need to provide a "class 5 switch", which connects directly to subscribers. |
| 04:38.43 | ghoti | Your upstream telco uses "class 4" switches to route calls from telco to telco (or to other providers in other countries). |
| 04:38.53 | ghoti | I haven't thought about this stuff in years. |
| 04:38.57 | raspberrypifan | hm idk that they would allow a new telco |
| 04:39.02 | ghoti | A class 4 switch used to be called a "tandem", I'm not sure why. |
| 04:39.28 | ghoti | If they won't allow another telco, then you'd need to appear to the incumbent telco as a "subscriber". |
| 04:39.57 | raspberrypifan | so i would be buying minutes from them i suppose? |
| 04:40.07 | ghoti | You would need to get probably ISDN PRIs to connect to them. That's either 23 or 30 channels per trunk depending on where you are. |
| 04:40.21 | ghoti | How you get billed is up for negotiation. |
| 04:40.32 | ghoti | And out of scope for this channel. |
| 04:41.57 | ghoti | Anyway, once you get your PRI, you can connect it to an Asterisk box using Sangoma or Digium hardware. OR you can get a dedicated border controller. Something like a Sangoma SBC. |
| 04:42.11 | raspberrypifan | hm well i magine if the cable company is doing it then its possible to do it to provide ITSP service, because the phnoe company gives u a cable modem with a phone ata |
| 04:42.37 | ghoti | Hmm, actually, the Sangoma SBC doesn't talk PRI I think, so you'd need other hardware to act as that gateway. |
| 04:42.48 | *** join/#asterisk stasdizzi (~stasdizzi@159.224.69.125) |
| 04:44.04 | ghoti | If you want to claim to provide telephone service, you'll also need to learn how to comply with emergency requirements. In Canada, we use "911" to dial for fire/police, etc. A VoIP provider needs to provide battery backup on an ATA in order to be able to provide reliable 911 service. |
| 04:44.09 | ghoti | There's lots to consider. |
| 04:44.24 | *** join/#asterisk jhester (~jhester@ip-64-134-230-114.public.wayport.net) |
| 04:44.49 | raspberrypifan | well honestly those things are probably not really issues where im going. Its more of an issue of getting a phone company to agree to do it at all |
| 04:44.54 | raspberrypifan | so this PRI would come from the phone company? |
| 04:44.54 | ghoti | If you want to avoid dealing with that stuff, then I'd recommend selling this as an "Internet service" where people can set up an optional telephone number. |
| 04:46.03 | ghoti | Yes, a PRI comes from the phone company. |
| 04:46.10 | ghoti | You need hardware on your end to handle it. |
| 04:46.26 | ghoti | It will have either 23 channels or 30 channels depending on whether it's delivered on a T1 or an E1. |
| 04:46.51 | ghoti | I have to sleep. |
| 04:47.19 | raspberrypifan | oh well, come back tommorow. I really appreciate all the info your providing |
| 04:47.23 | *** join/#asterisk MrJoshGeddes (storm3y@unaffiliated/storm3y) |
| 04:50.45 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 04:52.02 | ghoti | np. nn. |
| 05:23.03 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 05:24.50 | *** join/#asterisk CeBe (~CeBe@port-92-206-57-178.dynamic.qsc.de) |
| 05:29.34 | *** join/#asterisk tuxd00d (~tuxd00d@ip24-251-34-116.ph.ph.cox.net) |
| 05:35.50 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
| 05:41.39 | *** join/#asterisk chuckf (~chuckf@fedora/chuck) |
| 06:02.25 | *** join/#asterisk raspberrypifan (~textual@71-22-220-224.gar.clearwire-wmx.net) |
| 06:05.04 | *** join/#asterisk raspberrypifan (~textual@71-22-220-224.gar.clearwire-wmx.net) |
| 06:12.29 | raspberrypifan | i instatled freepbx in a VM, and it is giving me an ip address, how do i make it so i can use it |
| 06:18.07 | *** join/#asterisk anon9002 (~anon@204.11.131.235) |
| 06:18.32 | *** join/#asterisk chuckf (~chuckf@fedora/chuck) |
| 06:24.55 | anon9002 | Anyone knows how to remotely force reboot an SPA-3000? PSTN is stuck in answering state. |
| 06:30.49 | *** join/#asterisk bulkorok (~Adium@85.183.61.47) |
| 06:37.45 | ChannelZ | try x.x.x.x/admin/reboot |
| 06:37.52 | ChannelZ | oh he left |
| 06:42.17 | *** join/#asterisk timahvo1 (~rogue@197.237.174.64) |
| 06:43.10 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 06:45.15 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:1c95:c691:88f9:5941) |
| 06:46.55 | *** join/#asterisk jhlavacek (~jirka@78.208.220.3) |
| 06:47.30 | *** join/#asterisk timahvo1 (~rogue@197.237.174.64) |
| 06:48.17 | *** join/#asterisk bulkorok (~Adium@85.183.61.47) |
| 06:49.35 | *** join/#asterisk jhester (~jhester@ip-64-134-225-218.public.wayport.net) |
| 07:02.32 | *** join/#asterisk chuckf (~chuckf@fedora/chuck) |
| 07:08.20 | *** join/#asterisk Zogot (~Adium@D4B2620B.static.ziggozakelijk.nl) |
| 07:15.22 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 07:19.09 | *** join/#asterisk jhester (~jhester@64.134.221.254) |
| 07:19.55 | *** join/#asterisk tparcina (~tomo@212.92.192.54) |
| 07:35.09 | *** join/#asterisk bstarek (~bstarek@41.230.24.208) |
| 07:43.58 | *** part/#asterisk Zogot (~Adium@D4B2620B.static.ziggozakelijk.nl) |
| 07:46.28 | *** join/#asterisk tzafrir (~tzafrir@81.218.177.19) |
| 07:46.46 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 07:50.40 | *** join/#asterisk D30 (~deo@203.177.9.66) |
| 07:52.25 | mushroomed | Hi, when I write "core show application ConfBridge" |
| 07:52.35 | mushroomed | I don't get it listed, I'm using Asterisk 1.8 |
| 07:55.38 | *** join/#asterisk calum_ (~calum_@cpc67428-harg5-2-0-cust142.7-1.cable.virginm.net) |
| 07:59.42 | *** join/#asterisk chuckf (~chuckf@fedora/chuck) |
| 08:06.18 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 08:09.47 | *** join/#asterisk jhester (~jhester@mobile-166-137-182-115.mycingular.net) |
| 08:21.44 | bstarek | Hello All |
| 08:22.23 | bstarek | I am still with the same problem as yesterday, when i can send SIP message, but cannot reply back :) |
| 08:22.45 | bstarek | I am willing to re-post my config files if needed :) |
| 08:22.55 | bstarek | thank you guys. |
| 08:32.36 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 08:36.55 | *** join/#asterisk hehol (~hehol@217.9.101.222) |
| 08:38.07 | bstarek | When i send an SIP message from a user, in the CLI mode it says it is send by "asterisk" user. |
| 08:41.48 | *** join/#asterisk Neoti (~Thunderbi@cpc5-nott16-2-0-cust33.12-2.cable.virginm.net) |
| 08:55.52 | *** join/#asterisk chuckf (~chuckf@fedora/chuck) |
| 08:58.04 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 08:59.46 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 09:14.10 | bstarek | Could I please show you my extensions for SIP messages, and you would give me your thoughts? |
| 09:23.31 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 09:39.04 | *** join/#asterisk yaaago (~kresp0@gateway/tor-sasl/kresp0) |
| 09:54.58 | *** join/#asterisk n3hxs (~Ed@pool-108-16-94-145.phlapa.fios.verizon.net) |
| 10:04.29 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 10:05.44 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 10:09.57 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 10:23.50 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 10:33.18 | *** join/#asterisk CeBe (~CeBe@port-92-206-57-178.dynamic.qsc.de) |
| 10:41.28 | *** join/#asterisk sgimeno (~sgimeno@163.117.206.10) |
| 10:53.05 | *** join/#asterisk chuckf (~chuckf@fedora/chuck) |
| 11:03.52 | *** join/#asterisk jhester (~jhester@mobile-166-137-182-115.mycingular.net) |
| 11:15.55 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 11:29.53 | *** join/#asterisk CeBe (~CeBe@port-92-206-57-178.dynamic.qsc.de) |
| 11:34.31 | *** join/#asterisk DougsTech (~DougsTech@c-98-230-105-210.hsd1.al.comcast.net) |
| 11:49.45 | *** join/#asterisk calum_ (~calum_@cpc67428-harg5-2-0-cust142.7-1.cable.virginm.net) |
| 11:59.35 | *** join/#asterisk Dovid (~Dovid@ool-2f113725.dyn.optonline.net) |
| 12:01.37 | *** join/#asterisk u0m3 (~u0m3@92.80.72.79) |
| 12:14.45 | *** join/#asterisk bmurt (~brendan@static-96-245-76-214.phlapa.fios.verizon.net) |
| 12:18.44 | *** join/#asterisk Xaviertoor (~Jr@177.43.144.157.dynamic.adsl.gvt.net.br) |
| 12:19.31 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:43.21 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 12:48.09 | *** join/#asterisk retentiveboy (~retentive@c-50-150-62-29.hsd1.ga.comcast.net) |
| 12:49.40 | *** join/#asterisk BCrookAtRA (~bcrook@2001:470:1f11:942::443) |
| 12:56.48 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 12:57.38 | *** join/#asterisk BCrookAtRA (~bcrook@2001:470:1f11:942::443) |
| 13:04.37 | *** join/#asterisk BCrookAtRA (~bcrook@2001:470:1f11:942::443) |
| 13:05.53 | *** join/#asterisk dkakoti (~dkakoti88@122.161.17.161) |
| 13:06.13 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
| 13:07.57 | dkakoti | hello everyone |
| 13:09.17 | Xaviertoor | Hello |
| 13:09.57 | *** join/#asterisk jwr_ (~quassel@205.196.167.54) |
| 13:10.19 | dkakoti | I need help regarding asterisk 11 with webrtc support |
| 13:10.59 | WIMPy | ~ask |
| 13:10.59 | 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. |
| 13:14.56 | dkakoti | which asterisk version is better support for webrtc? |
| 13:15.48 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:19.15 | Xaviertoor | dkakoti, https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support |
| 13:20.24 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 13:23.43 | *** join/#asterisk dkakoti (~dkakoti88@122.161.17.161) |
| 13:26.13 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 13:30.30 | *** join/#asterisk CeBe (~CeBe@port-92-206-57-178.dynamic.qsc.de) |
| 13:33.12 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 13:42.29 | *** join/#asterisk ttyS3 (~o@gateway/tor-sasl/omlib) |
| 13:58.28 | *** join/#asterisk slykens (~slykens@50-197-49-129-static.hfc.comcastbusiness.net) |
| 13:59.25 | slykens | HI all - anyone have experience with a T1 card not working properly after the kernel disables the IRQ (the "nobody cares" message)? I've enabled irqpoll on the kernel to no avail and spent time with Google, also to no avail |
| 14:05.21 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 14:11.15 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:11.15 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:11.35 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 14:15.08 | *** join/#asterisk netmonk (~netmonk@93-43-45-195.ip90.fastwebnet.it) |
| 14:27.07 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 14:32.10 | *** join/#asterisk bstarek (~bstarek@41.230.24.208) |
| 14:45.31 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 14:47.31 | Defraz | Hey all! I have a PRI and a few DIDs that come across that PRI and I use the exten=> _96XX,1,Dial(SIP/208236${EXTEN}@10.100.9.205) dial plan to forward it to my other Asterisk server but I am trying to find an example to pass the caller ID along with the call. I receive the Caller ID name and number from incoming calls but I know I need to forward that on as well. |
| 14:47.40 | Defraz | Could someone point me in the right direction? |
| 14:48.06 | Defraz | Number shows up of course but the name is lost. |
| 14:48.52 | WIMPy | No. |
| 14:49.23 | WIMPy | As long as you don't explicitely change the CallerID it is always transferred along. |
| 14:49.50 | Defraz | The CallerID seems to be that way but the Name does not move along. |
| 14:54.40 | *** join/#asterisk jwr_ (~quassel@205.196.167.54) |
| 14:55.03 | *** join/#asterisk jpoz (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
| 14:56.17 | *** join/#asterisk makubi (~makubi@xdsl-87-79-219-185.netcologne.de) |
| 15:05.55 | *** join/#asterisk navaismo (~navaismo@201.124.133.118) |
| 15:06.37 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-hnspmczwzwjdrgge) |
| 15:09.29 | *** join/#asterisk oatha (~oatha.inf@unaffiliated/athayde) |
| 15:13.42 | Defraz | hmmm my pri isn't delivering name is there a setting to process that in Asterisk? |
| 15:17.06 | *** join/#asterisk imox (~imox@p4FC5CA85.dip0.t-ipconnect.de) |
| 15:17.08 | slykens | defraz - you might need a Wait(1) in your diaplan before answering the call from the PRI. Sometimes Name comes "late". |
| 15:17.40 | sekil | name should come in setup |
| 15:17.51 | *** join/#asterisk bmurt (~brendan@static-96-245-76-214.phlapa.fios.verizon.net) |
| 15:20.29 | *** join/#asterisk BitEvil (~quassel@tor/regular/SpeedEvil) |
| 15:21.41 | *** join/#asterisk imox (~imox@p4FC5CA85.dip0.t-ipconnect.de) |
| 15:22.43 | *** join/#asterisk dipanjan (671b082b@gateway/web/freenode/ip.103.27.8.43) |
| 15:23.31 | dipanjan | I need to get the duration of a call in Perl AGI. The ANSWEREDTIME variable is returing blank. Can anyone please help? |
| 15:24.42 | *** join/#asterisk Vince-0 (~quassel@196-215-42-46.dynamic.isadsl.co.za) |
| 15:26.19 | slykens | sekil - I agree, but most times I have set up PRI I needed the Wait(1) to get cnam working properly. I think it has been this way for >10 years? |
| 15:27.40 | sekil | I don't know what you're on about...I'm in Europe though.. |
| 15:28.10 | sekil | SETUP message should have IEs with all the info... |
| 15:28.23 | sekil | if you're using some cnam server..that's another thing.. |
| 15:28.40 | file | some telcos will asynchronously do the CNAM DIP so it comes later |
| 15:28.51 | file | (thus why the Wait mentioned above) |
| 15:28.54 | slykens | It should come in the setup IEs, yes, but on this side of the pond the caller name doesn't always come in the initial setup signaling. |
| 15:29.30 | sekil | kewl..good to know then |
| 15:29.35 | *** join/#asterisk bmurt (~brendan@static-96-245-76-214.phlapa.fios.verizon.net) |
| 15:31.46 | file | knowledge is power! |
| 15:32.55 | sekil | bye |
| 15:33.18 | navaismo | money is power |
| 15:33.25 | slykens | speaking of knowledge - any ideas regarding a dahdi device that suffers a disabled IRQ with the "nobody cared" message? I think it's breaking a PRI circuit I would very much like to use. :) |
| 15:34.32 | WIMPy | Put the card in another slot and hope it helps. |
| 15:35.31 | slykens | Tried that, didn't do any good. It's the only device on the IRQ fwiw. The disabled IRQ message comes immediately on DAHDI loading. |
| 15:36.46 | WIMPy | The only device on the IRQ or the only device on that IRQ that has a driver loaded for it? |
| 15:37.47 | *** join/#asterisk toothe (~mongolian@unaffiliated/toothe) |
| 15:37.55 | toothe | when I try to run asterisk, I get this |
| 15:38.10 | toothe | "Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)" |
| 15:39.03 | *** part/#asterisk dipanjan (671b082b@gateway/web/freenode/ip.103.27.8.43) |
| 15:39.05 | *** join/#asterisk chuckf (~chuckf@fedora/chuck) |
| 15:39.21 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 15:39.43 | navaismo | asterins not running or the user cant use asterisk or selinux enabled |
| 15:40.00 | navaismo | s/asterins/asterisk/ |
| 15:40.17 | toothe | navaismo: Its not running... |
| 15:40.29 | navaismo | then start the service |
| 15:40.37 | navaismo | or start with asterisk -vvvvvvvvcg |
| 15:40.39 | toothe | there it goes |
| 15:40.42 | toothe | thanks |
| 15:41.10 | toothe | im actually brand new to asterisk |
| 15:41.22 | toothe | i just need to learn some basic stuff about it |
| 15:42.11 | slykens | only device on the IRQ according the lspci - irq 25. Also the only device listed in /proc/interrupts |
| 15:43.13 | WIMPy | Looks like somethign is broken then. |
| 15:44.17 | slykens | yes, discovering exactly what has been painful so far. |
| 15:49.05 | *** join/#asterisk gustavofranco (~gustavofr@2804:620:0:1:116a:fd1e:f551:8677) |
| 15:55.28 | *** join/#asterisk solmsted (~solmsted@pool-71-176-247-26.rcmdva.fios.verizon.net) |
| 15:56.00 | *** join/#asterisk ttyS3 (~o@gateway/tor-sasl/omlib) |
| 15:59.17 | toothe | can I setup a SIP server just does calls internally? |
| 15:59.21 | toothe | ie, doesn't touch the actual phone system |
| 16:03.19 | [TK]D-Fender | Yes |
| 16:03.29 | toothe | okay, that's what I'll do for this test. |
| 16:04.00 | *** join/#asterisk bmurt (~brendan@74-95-169-22-Philadelphia.hfc.comcastbusiness.net) |
| 16:04.18 | toothe | ty |
| 16:06.04 | toothe | this generic guide is talking about having a line that's like this: |
| 16:06.13 | toothe | [9993] ; And then it says to have this fit my dialing structure. |
| 16:06.20 | toothe | I don't know what it means by dialing structure. |
| 16:07.52 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-jifydxxutzkrjpvo) |
| 16:07.53 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 16:13.01 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 16:14.11 | [TK]D-Fender | depends where you got that [9993[ from |
| 16:14.20 | [TK]D-Fender | it means different things in different places |
| 16:15.08 | toothe | think I got it working... |
| 16:15.12 | toothe | i got it from this document: |
| 16:15.20 | toothe | http://www.valcom.com/techsupport/config-docs/Generic%20Asterisk%20SIP%20Configuration%20Guide.pdf |
| 16:15.30 | toothe | I'm not trying to become a SIP expert, just do some basic testing for security purposes. |
| 16:15.37 | [TK]D-Fender | ~book |
| 16:15.37 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:15.41 | [TK]D-Fender | ^^^^^ |
| 16:15.58 | [TK]D-Fender | Asterisk is not just a "SIP server", nor is it at-all complete in that respect |
| 16:16.50 | toothe | I would love to read the entire book, but I just don't have time. |
| 16:16.59 | toothe | I might just learned the tools that come with this project instead |
| 16:17.00 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 16:17.16 | [TK]D-Fender | that "guide" is also ANCIENT. To the tune of about a DECADE and that syntax wo't work as-is on anything remotely modern |
| 16:17.27 | toothe | ahh... |
| 16:17.31 | toothe | dadrn |
| 16:17.33 | toothe | darn* |
| 16:17.53 | toothe | can I give you a very small small list of what I want |
| 16:17.58 | toothe | and you can tell me how quicka nd simple this might take? |
| 16:18.11 | *** join/#asterisk Sprocks (~Sprocks@BMTNON3746W-LP130-03-1242451638.dsl.bell.ca) |
| 16:18.12 | toothe | i was tasked with learning a few tools that "exploit" flaws in SIP. |
| 16:18.28 | toothe | and i just wanna setup two SIP clients that talk to each other |
| 16:18.30 | toothe | nothing more. |
| 16:18.48 | [TK]D-Fender | Do you hae a alcom device? |
| 16:18.51 | [TK]D-Fender | Valcom* |
| 16:18.52 | toothe | no. |
| 16:19.20 | toothe | there,was able to register onto asteriks running on my Kali Linux VM |
| 16:19.31 | [TK]D-Fender | Then you should trash that PDF now |
| 16:19.37 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 16:19.44 | toothe | lol, okay |
| 16:20.05 | toothe | recommendations in the interrim? I really don't have time for a dense book. |
| 16:20.10 | [TK]D-Fender | Go read the book I linked |
| 16:20.32 | [TK]D-Fender | You need to have a minimal understanding of dialplan, and to set up SIP peers, etc. |
| 16:20.42 | [TK]D-Fender | You'll also need to know how to use CLI to watch things |
| 16:20.58 | toothe | I can't...no time. |
| 16:21.01 | toothe | thanks though. |
| 16:21.59 | toothe | looks like im in trouble. |
| 16:22.58 | [TK]D-Fender | Without understanding the bits you'll be using you won't have any basis for the evaluation you wanted it for. |
| 16:23.33 | toothe | i know, its an impossible task. I agree with you 100% |
| 16:24.04 | toothe | my task isnt to setup asterisks, its to learn some testing tools |
| 16:24.31 | SuperNull | Hey [TK] how have you been |
| 16:25.58 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 16:26.29 | *** join/#asterisk bstarek (~bstarek@41.230.24.208) |
| 16:26.53 | [TK]D-Fender | toothe: Guess they expected that first part to be handed to you then and that only 1 set of circumstances was valid. |
| 16:27.07 | [TK]D-Fender | toothe: Sounds like a terrible testing assumption. |
| 16:27.55 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 16:28.04 | toothe | by testing tools, these are security tools designed to exploit flaws in the protocol. |
| 16:28.11 | toothe | such as authentication downgrades |
| 16:30.48 | toothe | its okay, i'll get them a product that works hopefully |
| 16:40.17 | *** join/#asterisk Katty (~Katty@97-91-121-70.static.stls.mo.charter.com) |
| 16:40.34 | Katty | hi kiddos |
| 16:43.02 | *** join/#asterisk retentiveboy (~retentive@c-50-150-62-29.hsd1.ga.comcast.net) |
| 16:44.16 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 16:48.12 | *** join/#asterisk CeBe1 (~CeBe@port-92-206-106-8.dynamic.qsc.de) |
| 16:50.49 | *** join/#asterisk LemensTS (~t15@8.33.19.98) |
| 16:52.24 | LemensTS | Hello, I am having an odd issue. When I transfer or park a call, all the phones that are on that network go unreachable for ~20-30 seconds then come back as reachable. Phones at different locations are fine. Here is just a shot of the cli without debug on http://pastebin.com/UKEFE5sS |
| 16:52.33 | LemensTS | Polycom 321's btw |
| 16:53.34 | *** join/#asterisk cmendes0101 (~cmendes01@office.phone.com) |
| 17:00.55 | *** join/#asterisk jpoz (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
| 17:03.24 | *** join/#asterisk danjenkins_ (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 17:08.27 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 17:08.35 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 17:10.08 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 17:10.28 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:15.48 | *** join/#asterisk u0m3_ (~u0m3@92.80.72.79) |
| 17:17.21 | *** join/#asterisk u0m3__ (~u0m3@92.80.72.79) |
| 17:24.37 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 17:27.20 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 17:46.04 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 17:49.19 | *** join/#asterisk jpoz_ (~jpoz@207.173.72.195) |
| 17:51.12 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:54.58 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:55.14 | *** join/#asterisk theron (~theron@173.252.71.189) |
| 18:01.37 | *** join/#asterisk vlad_starkov (~vlad_star@212.34.52.66) |
| 18:05.10 | toothe | I'm trying to setup linphone to connect to my asterisk server. |
| 18:05.15 | toothe | But I can't seem to get the cofiguration done correctly. |
| 18:06.56 | toothe | when I enter my information in the wizard, the "Apply" button does not light up. |
| 18:10.35 | *** join/#asterisk jpoz (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
| 18:11.45 | *** join/#asterisk youjelly (~youjelly@39.32.43.20) |
| 18:12.52 | youjelly | hi guys, my asterisk is segfaulting, it starts successfully with no errors/or warnings, when I go to the console, and issue the exit command it segfaults, or if it processes a call it segfaults |
| 18:14.33 | pabelanger | ~backtrace |
| 18:14.34 | infobot | backtrace is a debugging tool that is useful when trying to track down asterisk crashes (use -g with asterisk to generate a core, then read doc/backtrace.txt). See also https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 18:14.36 | pabelanger | youjelly, ^ |
| 18:15.07 | youjelly | which directory does asterisk dump its core in? |
| 18:15.16 | pabelanger | that is OS specific |
| 18:15.19 | Qwell | youjelly: either the directory you were in when you started it, or /tmp/ |
| 18:15.21 | pabelanger | usually where you launched it from |
| 18:15.41 | Qwell | That's also assuming it was started with -g |
| 18:17.34 | newtonr | youjelly, be sure you compile with the BETTER_BACKTRACES compiler flag |
| 18:18.55 | newtonr | youjelly, if you have a crash and you are running on the latest version for your branch of Asterisk, then file an issue on the tracker. https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines |
| 18:21.13 | youjelly | Its not the latest version |
| 18:21.37 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 18:21.46 | youjelly | last time I fixed it by loading some module |
| 18:22.42 | youjelly | I've generated the backtrace file |
| 18:24.49 | youjelly | should I pastebin it? |
| 18:35.02 | youjelly | anyone there? |
| 18:36.22 | *** join/#asterisk oatha (~oatha.inf@unaffiliated/athayde) |
| 18:37.26 | newtonr | youjelly, you can pastebin it to see if any developers here will take a quick look. otherwise you'll need to file a JIRA issue. |
| 18:37.57 | newtonr | youjelly, you never mentioned what exact version of Asterisk you are using? |
| 18:39.04 | youjelly | http://pastebin.com/zQXFvh7x |
| 18:39.28 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 18:39.48 | *** join/#asterisk tuxd00d (~tuxd00d@ip24-251-34-116.ph.ph.cox.net) |
| 18:41.22 | youjelly | Asterisk 1.8.18.0 built by root @ yuave-dev on a x86_64 running Linux on 2012-11-12 08:01:30 UTC |
| 18:44.08 | file | this has a third party module |
| 18:44.24 | cj | hey folks |
| 18:44.37 | youjelly | which one |
| 18:44.39 | oatha | cj o/ |
| 18:44.41 | youjelly | I can just unload that |
| 18:44.43 | file | no idea, but it does |
| 18:45.00 | file | MRCP related methinks |
| 18:45.05 | cj | have any of you set up VIC-4FXS modules and had them routed back to an asterisk server? |
| 18:45.36 | cj | I'm new to SIP on cisco gear |
| 18:45.54 | cj | so I assume that I have to have the router register with the asterisk SIP server |
| 18:46.13 | cj | and that I need to route all outbound calls through the SIP server |
| 18:46.40 | cj | and need to tell the router which inbound numbers to route to which ports on the module |
| 18:47.19 | cj | but my cisco IOS-fu is weak |
| 18:47.37 | *** join/#asterisk dan_j (~IceChat77@unaffiliated/danfromuk) |
| 18:48.45 | dan_j | Hi, any idea why company_202!54321@dundibridge doesnt match the extension _company_202!X. ? |
| 18:49.13 | youjelly | just unloaded the module res_speech_unimrcp and app_unimrcp |
| 18:49.19 | youjelly | still segfault |
| 18:49.44 | youjelly | I'm not even using that app in the extension I'm calling |
| 18:49.47 | cj | youjelly: did you blacklist the module? |
| 18:50.01 | youjelly | its just answering, printing something on console using verbose and hanging up |
| 18:50.05 | cj | is there anything useful in /var/log/kern.log or equiv? |
| 18:50.10 | youjelly | as soon as it hangs up, it segfaults |
| 18:51.42 | dan_j | The CLI says Rejected connect attempt from 10.50.0.31, request 'company_202!54321@dundibridge' does not exist |
| 18:52.59 | dan_j | Is it an issue with the ! symbol? |
| 18:53.32 | youjelly | just blacklisted both mrcp modules at least it stopped segfaulting |
| 18:54.09 | *** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 18:54.11 | toothe | I just setup two simple asterisks accounts and was able to have my two clients register. Great. Can I have them call each other now? |
| 18:54.29 | toothe | what would be the process to having them calling each other? Surely that is a simple configuration. |
| 18:56.01 | [TK]D-Fender | [14:48]dan_jHi, any idea why company_202!54321@dundibridge doesnt match the extension _company_202!X. ? <-- you've forgotton ! is a PATTERN char |
| 18:56.03 | cj | toothe: what is there context? |
| 18:56.19 | cj | is there a rule in the dialplan to call the client when a number is entered? |
| 18:56.54 | cj | ogod. I just spelled their as there. I feel so ashamed. |
| 18:58.21 | youjelly | the weirdest part is |
| 18:58.30 | dan_j | [TK]D-Fender: ah. thanks. I'll try to delimit the exten with something else. |
| 18:58.37 | youjelly | I've blacklisted it, I start asterisk, load the module manually |
| 18:58.41 | youjelly | it starts working again |
| 18:58.42 | [TK]D-Fender | dan_j: That'd be the best start to it |
| 18:58.50 | toothe | cj: No specific context. Just setting up a VERY simple test environment. |
| 18:58.53 | toothe | cj: The simpler the better |
| 18:59.43 | dan_j | [TK]D-Fender: hmm. doesnt seem to work with a # either. |
| 19:00.03 | [TK]D-Fender | dan_j: show all the backup |
| 19:00.25 | *** join/#asterisk vlad_starkov (~vlad_star@212.34.52.66) |
| 19:00.32 | cj | toothe: well, you need a context for your calls to land in |
| 19:00.46 | toothe | cj: What do you mean by a context? |
| 19:01.27 | cj | toothe: in sip.conf, you have entries for your clients, no? |
| 19:01.32 | toothe | Yes. |
| 19:01.36 | toothe | just two so far. |
| 19:01.48 | toothe | actually, it'll probably only ever be two :-) |
| 19:01.52 | dan_j | [TK]D-Fender: http://pastebin.com/xRssmmkj |
| 19:02.55 | [TK]D-Fender | request 'keshersiptrunk_202!01618268771@dundibridge' <_ I still see ! |
| 19:04.17 | toothe | cj: What do you recommend I do? |
| 19:04.27 | dan_j | [TK]D-Fender: sorry. old cli. didnt realise. http://pastebin.com/DMy2C9mU |
| 19:05.47 | [TK]D-Fender | dan_j: You've got some serious cross-eye issues today.... |
| 19:06.03 | [TK]D-Fender | dan_j: exten => _kesher_201#X. |
| 19:06.12 | [TK]D-Fender | dan_j: 'keshersiptrunk_202#01618268771@dundibridge' |
| 19:06.21 | dan_j | Argh. Youre right. Let me get the right dialplan. |
| 19:06.25 | [TK]D-Fender | dan_j: kesher != keshersiptrunk |
| 19:06.35 | dan_j | The right plan is exactly the same but with keshersiptrunk. |
| 19:06.49 | cj | toothe: sorry for the delay. was over on #cisco |
| 19:07.18 | cj | toothe: in each one of those client configurations, add a line that says "context=default" |
| 19:07.22 | dan_j | [TK]D-Fender: third time lucky! http://pastebin.com/zRdTgsrV |
| 19:07.34 | cj | in extensions.conf, set up a [default] |
| 19:08.14 | cj | under that, put something like |
| 19:08.21 | [TK]D-Fender | exten => _keshersiptrunk_202#X. <-- the letter "n" is ALSO reserved. Grab a coffee. nix that. Grab the whole POT.... |
| 19:08.58 | dan_j | [TK]D-Fender: argh. any way to escape the n character so it matches an 'n'? |
| 19:09.07 | [TK]D-Fender | [] |
| 19:09.12 | [TK]D-Fender | read your basics... |
| 19:09.21 | [TK]D-Fender | you're mashing one violation after another here... |
| 19:10.09 | cj | exten => 1234,1,Dial(SIP/<first user>,5) |
| 19:10.10 | cj | <PROTECTED> |
| 19:10.32 | dan_j | Ok. back to school. |
| 19:10.35 | cj | exten => 4321,1,Dial(SIP/<second user>,5) |
| 19:10.36 | cj | <PROTECTED> |
| 19:11.14 | cj | then when you dial 1234 from your second user's phone it should dial your first user |
| 19:11.23 | cj | and when you dial 4321 from your first user's phone it should dial your second user |
| 19:11.30 | *** join/#asterisk bkruse (~Adium@74.51.115.113) |
| 19:11.37 | cj | of course you'll need to issue 'dialplan reload' from your console before that works |
| 19:20.04 | toothe | cj: Puting all that in right now.... |
| 19:20.50 | toothe | it appears to be dailing, but the other side doesn't seem to be ringing. |
| 19:27.52 | *** join/#asterisk dwayne (~dwayne@c-71-207-208-112.hsd1.al.comcast.net) |
| 19:30.28 | toothe | A pastebin of my setup so far: http://pastebin.com/Y0K4Mnxf |
| 19:30.34 | toothe | how do I have 9993 and 9991 calle ach other> |
| 19:30.36 | toothe | call each other* |
| 19:30.53 | toothe | they're both registered, but I Can't seem to dial th eother. |
| 19:31.45 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 19:33.59 | *** join/#asterisk tuxd00d (~tuxd00d@ip24-251-34-116.ph.ph.cox.net) |
| 19:34.53 | toothe | yeah |
| 19:35.59 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 19:37.46 | *** join/#asterisk elguero (~miguel323@2001:470:1f06:12c4::2) |
| 19:50.40 | *** join/#asterisk tuxd00d (~tuxd00d@ip24-251-34-116.ph.ph.cox.net) |
| 19:53.57 | *** join/#asterisk dorphalsig (b532ffa2@gateway/web/freenode/ip.181.50.255.162) |
| 19:55.00 | dorphalsig | Hello :). Does anybody know if you can create a dial plan using ale and traditional extensions.conf? |
| 19:58.55 | dorphalsig | Just found you actually can :) |
| 20:00.29 | Qwell | dorphalsig: ael just gets converted into normal dialplan |
| 20:04.31 | fidividi | whats the best way to shorten the keep-alive and refresh period for peers/trunks/extensions in asterisk, for those devices/peers that go unreachable/lagged? |
| 20:04.55 | fidividi | from the server side i mean, not peer side, as i have no access to that. |
| 20:05.42 | *** join/#asterisk danjenkins_ (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 20:06.23 | swiss__ | Anyone familiar with the GotoIf syntax? I've been having some trouble with mine no matter how many variations on it i try |
| 20:08.02 | dan_j | swiss__: don't ask. just post your code. otherwise no one will help you. use pastebin or similar. |
| 20:08.31 | *** join/#asterisk areisp (~quassel@unaffiliated/areisp) |
| 20:22.52 | navaismo | swiss__, GotoIF($["EXPRESION" ="value"]?ifresulttrue:ifresultfalse) |
| 20:23.08 | navaismo | change = for <,> != anything+} |
| 20:30.08 | swiss__ | thats what i've been lookign at, so here is my exten.conf lin |
| 20:30.12 | swiss__ | exten => s,n,GotoIF($[${DIGIT} = 1] ?30:40) |
| 20:33.13 | swiss__ | gah |
| 20:33.32 | swiss__ | i had tried quotes before with no luck but of course now that i asked and changed it it works |
| 20:38.47 | *** join/#asterisk RaMcHiP (~RaMcHiP@wsip-98-190-149-169.ph.ph.cox.net) |
| 20:40.34 | *** join/#asterisk jpoz (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
| 20:41.42 | swiss__ | my next question is even though this part works, when i leave the gotoif in the extensions.conf i dont see my dtmf data in the cli, if i remove the gotoif the dtmf data shows up, any ideas |
| 20:43.05 | *** part/#asterisk LemensTS (~t15@8.33.19.98) |
| 20:50.41 | navaismo | amm gotoif has nothing to do wih dtmf you may check your dialplan |
| 20:51.53 | swiss__ | i believe you, but thats whats happening. if i remove the gotoif then i get dtmf data back, with it there i dont. |
| 21:07.44 | *** join/#asterisk amessina_ (~amessina@50-196-241-78-static.hfc.comcastbusiness.net) |
| 21:11.21 | *** join/#asterisk ryan_turner|MTW (Ryan@2600:3c01::f03c:91ff:fe69:f4ab) |
| 21:19.21 | *** join/#asterisk bkruse (~Adium@74.51.115.113) |
| 21:22.06 | *** join/#asterisk BCrookAtRA (~bcrook@2001:470:1f11:942::443) |
| 21:34.42 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 21:37.24 | *** join/#asterisk tuxd00d (~tuxd00d@ip24-251-34-116.ph.ph.cox.net) |
| 21:38.28 | RaMcHiP | <PROTECTED> |
| 21:38.39 | *** join/#asterisk johndropper (~johndropp@cpe-065-190-162-078.nc.res.rr.com) |
| 21:39.56 | johndropper | hi |
| 21:40.02 | johndropper | here is my question |
| 21:40.04 | johndropper | http://paste.ubuntu.com/7459655/ |
| 21:40.23 | *** join/#asterisk TriJetScud (~TriJetScu@van-app-svr.ad.v10networks.ca) |
| 21:40.23 | *** join/#asterisk nanoha-sama (~nanoha-sa@van-app-svr.ad.v10networks.ca) |
| 21:50.35 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 21:52.24 | *** join/#asterisk nanoha-sama (~nanoha-sa@van-app-svr.ad.v10networks.ca) |
| 21:53.42 | johndropper | ok thanks |
| 21:54.10 | johndropper | Hello, |
| 21:54.11 | johndropper | My name is John and we have 165 phones that connect to asterisk via sip in a cloud (Hosted) environment. We are running Asterisk ver 11.6 raw on centos. We recently updated to the latest ver 11.9 asterisk and now the phones will not connect. It seems they are not connecting because rtp encryption is set to ON inside the phones themselves. I can set the phone to rtp OFF and they connect and work fine. They seem to have a valid sip |
| 21:54.13 | johndropper | connection via SIP SHOW PEER but when we try to dial out into a meetme conference we get the message: [2014-05-13 17:28:21] WARNING[24884][C-00000000]: chan_sip.c:10543 process_sdp: |
| 21:54.14 | johndropper | We are requesting SRTP for audio, but they responded without it! |
| 21:54.15 | johndropper | Any Ideas on how we can update to asterisk 11/9 and keep our phones working? |
| 21:54.16 | johndropper | Thank You so much, |
| 21:54.17 | johndropper | -John |
| 21:55.17 | *** join/#asterisk theron (~theron@173.252.71.189) |
| 22:12.18 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:13.02 | *** join/#asterisk dumby_PC (~dumby@50.8.119.230) |
| 22:50.08 | *** join/#asterisk joesmoe (~joesmoe@c-73-181-72-237.hsd1.co.comcast.net) |
| 22:52.02 | *** join/#asterisk Razva (Razva@unaffiliated/razva) |
| 22:52.27 | Razva | hey folks! I'm struggling here to setup CallerID but I'm stuck |
| 22:52.51 | Razva | the idea is simple: I want to setup a specific caller ID for any number with 07XXXXXXXX |
| 22:53.04 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 22:54.22 | [TK]D-Fender | "setup a specific caller ID" <- doesn't mean anything clear. Please rephrase what it is you are loking to do. |
| 22:55.08 | *** join/#asterisk petris (~petris@2607:5300:60:5475:a9d7:583:99b:a901) |
| 22:55.55 | Razva | example: if I call 0712345678 the SIP should use a specific caller id (example.: MyCompany). yes, the provider allows this. |
| 22:57.18 | [TK]D-Fender | that is yor dialplan. |
| 22:57.34 | [TK]D-Fender | ]make a match for that number. Set the callerid. Do whatever else you want |
| 22:57.42 | *** join/#asterisk igcewieling (~igcewieli@ip98-183-26-100.pn.at.cox.net) |
| 22:57.48 | Razva | I'm using this (got it from a tutorial): |
| 22:57.50 | Razva | [outgoing] |
| 22:57.51 | Razva | exten => _07XXXXXXXX,1,SetCallerID(2024561111) |
| 22:57.51 | Razva | exten => _07XXXXXXXX,n,Dial(IAX2/MyProvider/${EXTEN}) |
| 22:58.16 | Razva | buuut seems that SetCallerID returns No application 'SetCallerID' |
| 22:58.25 | igcewieling | Has anyone seen this before? If the nat translation was closed shouldn't it be getting connection refused or something like that? "[May 13 18:49:29] WARNING[13017]: chan_sip.c:3907 __sip_xmit: sip_xmit of 0x7ff5540060e0 (len 610) to 209.220.119.87:57855 returned -2: Interrupted system call" |
| 22:58.52 | igcewieling | I *think* it happened when I moved the endpoint to sip/tcp. Using Asterisk 11 |
| 22:59.17 | igcewieling | Razva: Stop! |
| 22:59.17 | [TK]D-Fender | Razva: that command has been dead for YEARS |
| 22:59.25 | [TK]D-Fender | Razva: "core show function CALLERID" |
| 22:59.34 | igcewieling | Razva: any documentation which refers to that application cannot be trusted |
| 22:59.46 | [TK]D-Fender | indeed |
| 23:00.36 | igcewieling | [TK]D-Fender: do you recall when the SetCallerID app was deprecated? Maybe around 2003 or so? |
| 23:00.53 | [TK]D-Fender | igcewieling: Little later than that. with 1.2 |
| 23:01.45 | Razva | hah. great. |
| 23:02.00 | Razva | sooo any idea where can I find an example, but without reading 100 pages? :) |
| 23:02.21 | Razva | yes, I know that I should, but I just want to see if this thing works :| |
| 23:03.12 | [TK]D-Fender | Razva: I just gave you the exact command to get the instructions for the function to use |
| 23:03.57 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-137-60.ks.ks.cox.net) |
| 23:06.00 | Razva | Command 'core show function CALLERID' failed. |
| 23:06.26 | navaismo | swiss__, and which asterisk version |
| 23:06.43 | *** part/#asterisk Nemus (~Nemus@unaffiliated/nemus) |
| 23:06.58 | [TK]D-Fender | Razva: Show us |
| 23:07.00 | [TK]D-Fender | ~pb |
| 23:07.01 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 23:07.02 | [TK]D-Fender | ^^^^ |
| 23:07.08 | [TK]D-Fender | and don't flood the channel with it |
| 23:08.11 | Razva | [TK]D-Fender sorry for bothering you, which file? the iax, or extensions? ore the core show? |
| 23:08.13 | Razva | *or |
| 23:11.48 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 23:11.50 | *** join/#asterisk petris (~petris@2607:5300:60:5475:a9d7:583:99b:a901) |
| 23:15.08 | Razva | igcewieling: http://pastebin.com/PA8g2hDn |
| 23:15.19 | Razva | I'm doing something very wrong there... |
| 23:16.07 | igcewieling | looks good at first glance, other than that screwed up dial line |
| 23:16.47 | *** join/#asterisk theron_ (~theron@173.252.71.189) |
| 23:17.23 | Razva | that's my best effort... :| baaah I hate when I don't understand where something is broken |
| 23:17.54 | navaismo | everyone... and is frustration not hate |
| 23:20.09 | igcewieling | It is lack of taking the time to learn enough to accomplish something |
| 23:20.12 | igcewieling | ~book |
| 23:20.13 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 23:20.43 | Razva | refresh please: http://pastebin.com/PA8g2hDn |
| 23:21.09 | Razva | smashes his head on the table |
| 23:21.17 | *** join/#asterisk petris (~petris@2607:5300:60:5475:a9d7:583:99b:a901) |
| 23:21.45 | igcewieling | Has anyone seen this before? If the nat translation was closed shouldn't it be getting connection refused or something like that? "[May 13 18:49:29] WARNING[13017]: chan_sip.c:3907 __sip_xmit: sip_xmit of 0x7ff5540060e0 (len 610) to 209.220.119.87:57855 returned -2: Interrupted system call" |
| 23:22.01 | navaismo | not me |
| 23:22.31 | navaismo | your fancy installation never met my asterisk installations :(...one day |
| 23:22.48 | Razva | ok I'm out of ideas folks, can you please point me to what I'm obviously doing wrong (like a 1000% newb, which I am) |
| 23:22.51 | igcewieling | navaismo: "automated" not "fancy" 8-| |
| 23:23.01 | Razva | yes, I'm reading https://wiki.asterisk.org/wiki/display/AST/The+Verbose+and+NoOp+Applications |
| 23:23.29 | igcewieling | navaismo: I've been wrestling with hanguphandlers. |
| 23:24.05 | igcewieling | Razva: Read up on the Dial applliction |
| 23:24.22 | navaismo | igcewieling, yeah i still use extension h and Agis |
| 23:24.55 | RaMcHiP | While people are chatting let me see if I can toss my question out there |
| 23:25.47 | navaismo | igcewieling, Razva , seems like some modules arent loaded, right? |
| 23:26.13 | Razva | navaismo I have no idea and I'm ready to drop this dead. |
| 23:26.30 | navaismo | ok first of all which asterisk version? |
| 23:26.30 | RaMcHiP | I am trying to get click to dial to work. I am using 7940 cisco hardphones. I have configured an asterisk extension and a vtiger user pointed to that extension. I have setup the PBXManager in vtiger to connect to asterisk. I have modified the $source variables to be "from-internal" and I cannot get click to dial to work. It acts like it is and tells me to pick up my extension but does |
| 23:26.31 | RaMcHiP | not dial out. |
| 23:26.35 | RaMcHiP | I am not getting any errors or anything |
| 23:26.58 | igcewieling | navaismo: looks to me like he installed from a package |
| 23:27.03 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 23:27.09 | navaismo | Razva, which version of asterisk? |
| 23:27.13 | Razva | yup. centos. |
| 23:27.17 | Razva | 11, from repo |
| 23:27.47 | [TK]D-Fender | Razva: Connection issue here. Show me where you got the error for the command I gave you earlier |
| 23:27.49 | igcewieling | Then you should contact the packager to find out why those applications are not available |
| 23:29.20 | igcewieling | Razva: never put a space after a comma in the dialplan. your verbose line would have logged that Verbose as starting with a space |
| 23:30.04 | Razva | http://pastebin.com/PA8g2hDn < just loaded some modules, the verbose is gone (wooohoo) but now I get Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) |
| 23:30.47 | *** part/#asterisk igcewieling (~igcewieli@ip98-183-26-100.pn.at.cox.net) |
| 23:30.54 | Razva | ok, space after comma removed |
| 23:32.01 | Razva | [TK]D-Fender I've rewrote everything (scraps from other tutorials + my own not-so-bright ideas), please take a look here http://pastebin.com/PA8g2hDn |
| 23:32.41 | [TK]D-Fender | {$CALLERID(all)} <- this is not how you reference a function |
| 23:32.47 | navaismo | sip peer is not registered |
| 23:32.55 | [TK]D-Fender | Hinst: It's just like variables |
| 23:33.39 | Razva | navaismo the module? |
| 23:34.15 | navaismo | no the device |
| 23:34.21 | navaismo | the phone |
| 23:34.29 | *** join/#asterisk petris (~petris@2607:5300:60:5475:a9d7:583:99b:a901) |
| 23:35.01 | Razva | [TK]D-Fender well the booboo is that I'm no programmer and no expert, I'm just a regular Joe who knows how to install a CentOS machine, so I suppose you understand my frustrantion. :) |
| 23:35.25 | [TK]D-Fender | You still need to read your variable & function basics. |
| 23:35.34 | [TK]D-Fender | We're all regular joes |
| 23:35.46 | Razva | yeah I suppose, or pay somebody and keep my sanity. :| |
| 23:35.56 | [TK]D-Fender | ${FUNCTION(arg1,arg2,etc)} |
| 23:36.09 | navaismo | RaMcHiP, sorry if i saw cisco 94XX in the same phrase I ignore that automagically, and you need at least provide logs of asterisk or how works that vtiger thingy, it use AMI call file or what? |
| 23:36.47 | Razva | [TK]D-Fender I suppose that we're talking about the Verbose line? |
| 23:36.59 | [TK]D-Fender | yes |
| 23:37.13 | Razva | same => n,Verbose(1,Outgoing Caller ID: ${CALLERID(all)}) |
| 23:37.14 | *** join/#asterisk sawgood1 (~sawgood@unaffiliated/sawgood) |
| 23:37.49 | [TK]D-Fender | Razva: Your Dial command also looks wrong |
| 23:37.57 | [TK]D-Fender | same => n,Verbose(1, Outgoing Caller ID: {$CALLERID(all)}) |
| 23:38.02 | [TK]D-Fender | was clearly wrong here |
| 23:38.26 | [TK]D-Fender | Your dial... is not referncig a peer entry in sip.conf, specifying a valid host, etc |
| 23:38.29 | *** join/#asterisk theron (~theron@173.252.71.189) |
| 23:38.50 | Razva | the $ was after the { => bad? |
| 23:39.04 | Razva | aha, ok, let me take a look |
| 23:39.14 | [TK]D-Fender | [19:35][TK]D-Fender${FUNCTION(arg1,arg2,etc)} |
| 23:40.08 | Razva | is thinking at how to rewrite that line |
| 23:41.01 | Razva | ok but the function is already in another () |
| 23:41.21 | Razva | so it should be like ( ${FUNCTION(arg1,arg2,etc)} ) |
| 23:41.25 | Razva | correct? |
| 23:41.39 | *** join/#asterisk theron_ (~theron@173.252.71.189) |
| 23:42.03 | *** join/#asterisk raspberrypifan (~textual@71-22-220-224.gar.clearwire-wmx.net) |
| 23:42.05 | [TK]D-Fender | that function reference needs to be whole regardless of where it's inside of |
| 23:43.14 | Razva | ${FUNCTION(arg1,arg2,etc)} == ${CALLERID(all)} |
| 23:43.15 | Razva | correct? |
| 23:43.43 | [TK]D-Fender | as far as similar syntax goes, yes |
| 23:43.58 | [TK]D-Fender | So onto phase 2.. your dial is not good. |
| 23:44.16 | Razva | horray, so same => n,Verbose(1, Outgoing Caller ID: ${CALLERID(all)}) == ok |
| 23:45.18 | Razva | http://pastebin.com/PA8g2hDn < added the sip.conf also |
| 23:45.28 | Razva | but I think it's missing the valid host |
| 23:46.30 | [TK]D-Fender | The thought process you skipped: Where is it supposed to SEND that call? |
| 23:46.52 | *** join/#asterisk petris (~petris@2607:5300:60:5475:a9d7:583:99b:a901) |
| 23:47.28 | Razva | mmm...to the number I'm dialing from the softphone? |
| 23:48.10 | Razva | oh, and another thing |
| 23:48.18 | [TK]D-Fender | You dial a number USING your softphone..... which matches [100]. Well... Asterisk accepts your request and starts processing it.. |
| 23:48.26 | [TK]D-Fender | one of those step is to DIAL something |
| 23:48.34 | [TK]D-Fender | Where is it supposed to SWEND that to? |
| 23:48.53 | Razva | just a sec |
| 23:49.17 | Razva | if I'm renaming my number 0731blahblah to 07XXXXXXXX I get Call from '100' (86.125.187.22:60190) to extension '0731059660' rejected because extension not found in context 'default'. |
| 23:49.40 | Razva | exten => 07XXXXXXXX,1,NoOp() |
| 23:50.28 | [TK]D-Fender | ? |
| 23:50.47 | Razva | aha, forgot the _ |
| 23:50.47 | [TK]D-Fender | I need to see accurate current configs & call output... |
| 23:51.00 | [TK]D-Fender | You are chopping bits up and I can't prove what actual condition anything is in... |
| 23:51.13 | Razva | ok, just a sec to make a decent pastebin |
| 23:52.48 | raspberrypifan | who was the nice guy i was taking to yuesterday |
| 23:54.59 | Razva | http://pastebin.com/PA8g2hDn |
| 23:55.19 | Razva | [TK]D-Fender please take a looksy here: http://pastebin.com/PA8g2hDn |
| 23:55.36 | raspberrypifan | ghoti |
| 23:55.55 | [TK]D-Fender | Razva: same => n,Dial(SIP/${EXTEN}) <- this has nowhere to go. |
| 23:56.00 | *** join/#asterisk johndropper (~johndropp@cpe-174-097-145-204.nc.res.rr.com) |
| 23:56.10 | [TK]D-Fender | Razva: You have to send a number.... to a place. You left out the place |
| 23:56.10 | johndropper | hey guys I need some help |
| 23:56.23 | johndropper | is anyone available to listen? |
| 23:56.24 | Razva | aaaha |
| 23:56.26 | [TK]D-Fender | ~ask |
| 23:56.26 | 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. |
| 23:56.40 | johndropper | ok infobot |
| 23:56.44 | johndropper | thanks |
| 23:56.56 | Razva | is pondering |
| 23:56.57 | [TK]D-Fender | johndropper: that's just a channel bot |
| 23:58.05 | Razva | well the idea is that after Dial it should be forwarded to...the SIP company, who should call the phone... |
| 23:58.26 | johndropper | We have 65 plus phone that are connected to our hosted server. The sip connections are remote. We upgraded from 11.6 to 11.9 and now the phones wont dial out. I go to the phone itnerface adn turn rtp off and then it dials out. Any ideas on how to get this working without logging into each phone and turning off rtp? |
| 23:59.17 | [TK]D-Fender | "turn rtpe off" doesn't make any sense.... |
| 23:59.27 | [TK]D-Fender | RTP = VOICE |
| 23:59.35 | [TK]D-Fender | How do you turn that "off" on a phone? |
| 23:59.58 | johndropper | RTP Encryption |